From 1b52a76815aa068a3477bfdfaf58718d41d7f9b4 Mon Sep 17 00:00:00 2001 From: The Android Open Source Project Date: Tue, 3 Mar 2009 18:28:36 -0800 Subject: auto import from //depot/cupcake/@135843 --- arm-hybrid-22k/Makefile | 66 - arm-hybrid-22k/bin/arm-hybrid-22k | Bin 485468 -> 0 bytes arm-hybrid-22k/host_src/.DS_Store | Bin 6148 -> 0 bytes arm-hybrid-22k/host_src/._.DS_Store | Bin 82 -> 0 bytes arm-hybrid-22k/host_src/arm-hybrid-22k.mak | 25 - arm-hybrid-22k/host_src/eas.h | 1062 ---- arm-hybrid-22k/host_src/eas_build.h | 36 - arm-hybrid-22k/host_src/eas_chorus.h | 53 - arm-hybrid-22k/host_src/eas_config.c | 619 --- arm-hybrid-22k/host_src/eas_config.h | 191 - arm-hybrid-22k/host_src/eas_debugmsgs.h | 44 - arm-hybrid-22k/host_src/eas_host.h | 83 - arm-hybrid-22k/host_src/eas_hostmm.c | 660 --- arm-hybrid-22k/host_src/eas_main.c | 461 -- arm-hybrid-22k/host_src/eas_report.c | 264 - arm-hybrid-22k/host_src/eas_report.h | 77 - arm-hybrid-22k/host_src/eas_reverb.h | 54 - arm-hybrid-22k/host_src/eas_types.h | 268 - arm-hybrid-22k/host_src/eas_wave.c | 423 -- arm-hybrid-22k/host_src/eas_wave.h | 74 - arm-hybrid-22k/lib/libarm-hybrid-22k.a | Bin 189624 -> 0 bytes arm-hybrid-22k/lib_src/ARM-E_filter_gnu.s | 134 - .../lib_src/ARM-E_interpolate_loop_gnu.s | 131 - .../lib_src/ARM-E_interpolate_noloop_gnu.s | 130 - arm-hybrid-22k/lib_src/ARM-E_mastergain_gnu.s | 109 - arm-hybrid-22k/lib_src/ARM-E_voice_gain_gnu.s | 166 - arm-hybrid-22k/lib_src/ARM_synth_constants_gnu.inc | 153 - arm-hybrid-22k/lib_src/arm-hybrid-22k_lib.mak | 25 - arm-hybrid-22k/lib_src/eas_audioconst.h | 97 - arm-hybrid-22k/lib_src/eas_chorus.c | 604 --- arm-hybrid-22k/lib_src/eas_chorusdata.c | 34 - arm-hybrid-22k/lib_src/eas_chorusdata.h | 160 - arm-hybrid-22k/lib_src/eas_ctype.h | 41 - arm-hybrid-22k/lib_src/eas_data.c | 37 - arm-hybrid-22k/lib_src/eas_data.h | 131 - arm-hybrid-22k/lib_src/eas_effects.h | 61 - arm-hybrid-22k/lib_src/eas_fmengine.c | 785 --- 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arm-hybrid-22k/lib_src/eas_pan.c | 98 - arm-hybrid-22k/lib_src/eas_pan.h | 66 - arm-hybrid-22k/lib_src/eas_parser.h | 98 - arm-hybrid-22k/lib_src/eas_pcm.c | 1482 ------ arm-hybrid-22k/lib_src/eas_pcm.h | 359 -- arm-hybrid-22k/lib_src/eas_pcmdata.c | 35 - arm-hybrid-22k/lib_src/eas_pcmdata.h | 157 - arm-hybrid-22k/lib_src/eas_public.c | 2597 ---------- arm-hybrid-22k/lib_src/eas_reverb.c | 1154 ----- arm-hybrid-22k/lib_src/eas_reverbdata.c | 34 - arm-hybrid-22k/lib_src/eas_reverbdata.h | 486 -- arm-hybrid-22k/lib_src/eas_rtttl.c | 1197 ----- arm-hybrid-22k/lib_src/eas_rtttldata.c | 41 - arm-hybrid-22k/lib_src/eas_rtttldata.h | 70 - arm-hybrid-22k/lib_src/eas_smf.c | 1203 ----- arm-hybrid-22k/lib_src/eas_smf.h | 49 - arm-hybrid-22k/lib_src/eas_smfdata.c | 66 - arm-hybrid-22k/lib_src/eas_smfdata.h | 66 - arm-hybrid-22k/lib_src/eas_sndlib.h | 406 -- arm-hybrid-22k/lib_src/eas_synth.h | 395 -- arm-hybrid-22k/lib_src/eas_synth_protos.h | 60 - arm-hybrid-22k/lib_src/eas_synthcfg.h | 70 - arm-hybrid-22k/lib_src/eas_vm_protos.h | 1086 ----- arm-hybrid-22k/lib_src/eas_voicemgt.c | 3971 --------------- arm-hybrid-22k/lib_src/eas_wavefile.c | 867 ---- arm-hybrid-22k/lib_src/eas_wavefile.h | 63 - arm-hybrid-22k/lib_src/eas_wavefiledata.c | 33 - arm-hybrid-22k/lib_src/eas_wt_IPC_frame.h | 82 - arm-hybrid-22k/lib_src/eas_wtengine.c | 661 --- arm-hybrid-22k/lib_src/eas_wtengine.h | 171 - arm-hybrid-22k/lib_src/eas_wtsynth.c | 1251 ----- arm-hybrid-22k/lib_src/eas_wtsynth.h | 66 - arm-hybrid-22k/lib_src/hybrid_22khz_mcu.c | 5149 -------------------- 92 files changed, 37895 deletions(-) delete mode 100644 arm-hybrid-22k/Makefile delete mode 100755 arm-hybrid-22k/bin/arm-hybrid-22k delete mode 100644 arm-hybrid-22k/host_src/.DS_Store delete mode 100644 arm-hybrid-22k/host_src/._.DS_Store delete mode 100644 arm-hybrid-22k/host_src/arm-hybrid-22k.mak delete mode 100644 arm-hybrid-22k/host_src/eas.h delete mode 100644 arm-hybrid-22k/host_src/eas_build.h delete mode 100644 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a/arm-hybrid-22k/Makefile +++ /dev/null @@ -1,66 +0,0 @@ -LOCAL_PATH := $(call my-dir) -include $(CLEAR_VARS) - -LOCAL_SRC_FILES = \ - lib_src/eas_chorus.c \ - lib_src/eas_chorusdata.c \ - lib_src/eas_data.c \ - lib_src/eas_fmengine.c \ - lib_src/eas_fmsynth.c \ - lib_src/eas_fmtables.c \ - lib_src/eas_ima_tables.c \ - lib_src/eas_imaadpcm.c \ - lib_src/eas_imelody.c \ - lib_src/eas_imelodydata.c \ - lib_src/eas_math.c \ - lib_src/eas_midi.c \ - lib_src/eas_mididata.c \ - lib_src/eas_mixbuf.c \ - lib_src/eas_mixer.c \ - lib_src/eas_ota.c \ - lib_src/eas_otadata.c \ - lib_src/eas_pan.c \ - lib_src/eas_pcm.c \ - lib_src/eas_pcmdata.c \ - lib_src/eas_public.c \ - lib_src/eas_reverb.c \ - lib_src/eas_reverbdata.c \ - lib_src/eas_rtttl.c \ - lib_src/eas_rtttldata.c \ - lib_src/eas_smf.c \ - lib_src/eas_smfdata.c \ - lib_src/eas_voicemgt.c \ - lib_src/eas_wavefile.c \ - lib_src/eas_wavefiledata.c \ - lib_src/eas_wtengine.c \ - lib_src/eas_wtsynth.c \ - lib_src/hybrid_22khz_mcu.c \ - host_src/eas_config.c \ - host_src/eas_hostmm.c \ - host_src/eas_main.c \ - host_src/eas_report.c \ - host_src/eas_wave.c - -LOCAL_CFLAGS+= -O2 -D UNIFIED_DEBUG_MESSAGES -D EAS_HYBRID_SYNTH \ - -D _IMELODY_PARSER -D _RTTTL_PARSER -D _OTA_PARSER \ - -D _WAVE_PARSER -D _REVERB_ENABLED \ - -D _CHORUS_ENABLED -D NUM_OUTPUT_CHANNELS=2 \ - -D _SAMPLE_RATE_22050 -D MAX_SYNTH_VOICES=32 \ - -D NUM_PRIMARY_VOICES=8 \ - -D _8_BIT_SAMPLES -D _FILTER_ENABLED -D _IMELODY_PARSER \ - -D _IMA_DECODER - -LOCAL_C_INCLUDES:= \ - $(LOCAL_PATH)/host_src/ \ - $(LOCAL_PATH)/lib_src/ - -LOCAL_ARM_MODE := arm - -LOCAL_MODULE := libsonivox - -LOCAL_COPY_HEADERS_TO := libsonivox -LOCAL_COPY_HEADERS := \ - host_src/eas.h \ - host_src/eas_types.h - -include $(BUILD_SHARED_LIBRARY) diff --git a/arm-hybrid-22k/bin/arm-hybrid-22k b/arm-hybrid-22k/bin/arm-hybrid-22k deleted file mode 100755 index ea80a08..0000000 Binary files a/arm-hybrid-22k/bin/arm-hybrid-22k and /dev/null differ diff --git a/arm-hybrid-22k/host_src/.DS_Store b/arm-hybrid-22k/host_src/.DS_Store deleted file mode 100644 index 5008ddf..0000000 Binary files a/arm-hybrid-22k/host_src/.DS_Store and /dev/null differ diff --git a/arm-hybrid-22k/host_src/._.DS_Store b/arm-hybrid-22k/host_src/._.DS_Store deleted file mode 100644 index c9474ea..0000000 Binary files a/arm-hybrid-22k/host_src/._.DS_Store and /dev/null differ diff --git a/arm-hybrid-22k/host_src/arm-hybrid-22k.mak b/arm-hybrid-22k/host_src/arm-hybrid-22k.mak deleted file mode 100644 index 8874f31..0000000 --- a/arm-hybrid-22k/host_src/arm-hybrid-22k.mak +++ /dev/null @@ -1,25 +0,0 @@ -# -# Auto-generated sample makefile -# -# This makefile is intended for use with GNU make. -# Set the paths to the tools (CC, AR, LD, etc.) -# - -vpath %.c host_src - -CC = C:\Program Files\GNUARM\bin\arm-elf-gcc.exe -AS = C:\Program Files\GNUARM\bin\arm-elf-as.exe -LD = C:\Program Files\GNUARM\bin\arm-elf-gcc.exe -AR = C:\Program Files\GNUARM\bin\arm-elf-ar.exe - -%.o: %.c - $(CC) -c -O2 -o $@ -I host_src -D UNIFIED_DEBUG_MESSAGES -D EAS_HYBRID_SYNTH -D _IMELODY_PARSER -D _RTTTL_PARSER -D _OTA_PARSER -D _WAVE_PARSER -D _REVERB_ENABLED -D _CHORUS_ENABLED $< - -%.o: %.s - $(AS) -o $@ -EL -mcpu=arm946e-s -mfpu=softfpa -I lib_src --defsym CHECK_STACK=0 --defsym REVERB=0 --defsym CHORUS=0 --defsym STEREO_OUTPUT=1 --defsym SAMPLE_RATE_22050=1 --defsym SAMPLES_8_BIT=1 --defsym FILTER_ENABLED=1 $< - -OBJS = eas_main.o eas_report.o eas_wave.o eas_hostmm.o eas_config.o - -arm-hybrid-22k: $(OBJS) - $(LD) -o $@ $(OBJS) libarm-hybrid-22k.a -lm - diff --git a/arm-hybrid-22k/host_src/eas.h b/arm-hybrid-22k/host_src/eas.h deleted file mode 100644 index 0bb04fe..0000000 --- a/arm-hybrid-22k/host_src/eas.h +++ /dev/null @@ -1,1062 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas.h - * - * Contents and purpose: - * The public interface header for the EAS synthesizer. - * - * This header only contains declarations that are specific - * to this implementation. - * - * DO NOT MODIFY THIS FILE! - * - * Copyright Sonic Network Inc. 2005, 2006 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 852 $ - * $Date: 2007-09-04 11:43:49 -0700 (Tue, 04 Sep 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_H -#define _EAS_H - -#include "eas_types.h" - -/* for C++ linkage */ -#ifdef __cplusplus -extern "C" { -#endif - -/* library version macro */ -#define MAKE_LIB_VERSION(a,b,c,d) (((((((EAS_U32) a <<8) | (EAS_U32) b) << 8) | (EAS_U32) c) << 8) | (EAS_U32) d) -#define LIB_VERSION MAKE_LIB_VERSION(3, 6, 10, 14) - -typedef struct -{ - EAS_U32 libVersion; - EAS_BOOL checkedVersion; - EAS_I32 maxVoices; - EAS_I32 numChannels; - EAS_I32 sampleRate; - EAS_I32 mixBufferSize; - EAS_BOOL filterEnabled; - EAS_U32 buildTimeStamp; - EAS_CHAR *buildGUID; -} S_EAS_LIB_CONFIG; - -/* enumerated effects module numbers for configuration */ -typedef enum -{ - EAS_MODULE_ENHANCER = 0, - EAS_MODULE_COMPRESSOR, - EAS_MODULE_REVERB, - EAS_MODULE_CHORUS, - EAS_MODULE_WIDENER, - EAS_MODULE_GRAPHIC_EQ, - EAS_MODULE_WOW, - EAS_MODULE_MAXIMIZER, - EAS_MODULE_TONECONTROLEQ, - NUM_EFFECTS_MODULES -} E_FX_MODULES; - -/* enumerated optional module numbers for configuration */ -typedef enum -{ - EAS_MODULE_MMAPI_TONE_CONTROL = 0, - EAS_MODULE_METRICS -} E_OPT_MODULES; -#define NUM_OPTIONAL_MODULES 2 - -/* enumerated audio decoders for configuration */ -typedef enum -{ - EAS_DECODER_PCM = 0, - EAS_DECODER_SMAF_ADPCM, - EAS_DECODER_IMA_ADPCM, - EAS_DECODER_7BIT_SMAF_ADPCM, - EAS_DECODER_NOT_SUPPORTED -} E_DECODER_MODULES; -#define NUM_DECODER_MODULES 4 - -/* defines for EAS_PEOpenStream flags parameter */ -#define PCM_FLAGS_STEREO 0x00000100 /* stream is stereo */ -#define PCM_FLAGS_8_BIT 0x00000001 /* 8-bit format */ -#define PCM_FLAGS_UNSIGNED 0x00000010 /* unsigned format */ -#define PCM_FLAGS_STREAMING 0x80000000 /* streaming mode */ - -/* maximum volume setting */ -#define EAS_MAX_VOLUME 100 - -/*---------------------------------------------------------------------------- - * EAS_Init() - *---------------------------------------------------------------------------- - * Purpose: - * Initialize the synthesizer library - * - * Inputs: - * polyphony - number of voices to play (dynamic memory model only) - * ppLibData - pointer to data handle variable for this instance - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_Init (EAS_DATA_HANDLE *ppEASData); - -/*---------------------------------------------------------------------------- - * EAS_Config() - *---------------------------------------------------------------------------- - * Purpose: - * Returns a pointer to a structure containing the configuration options - * in this library build. - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC const S_EAS_LIB_CONFIG *EAS_Config (void); - -/*---------------------------------------------------------------------------- - * EAS_Shutdown() - *---------------------------------------------------------------------------- - * Purpose: - * Shuts down the library. Deallocates any memory associated with the - * synthesizer (dynamic memory model only) - * - * Inputs: - * pEASData - handle to data for this instance - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_Shutdown (EAS_DATA_HANDLE pEASData); - -/*---------------------------------------------------------------------------- - * EAS_Render() - *---------------------------------------------------------------------------- - * Purpose: - * Parse the Midi data and render PCM audio data. - * - * Inputs: - * pEASData - buffer for internal EAS data - * pOut - output buffer pointer - * nNumRequested - requested num samples to generate - * pnNumGenerated - actual number of samples generated - * - * Outputs: - * EAS_SUCCESS if PCM data was successfully rendered - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_Render (EAS_DATA_HANDLE pEASData, EAS_PCM *pOut, EAS_I32 numRequested, EAS_I32 *pNumGenerated); - -/*---------------------------------------------------------------------------- - * EAS_SetRepeat() - *---------------------------------------------------------------------------- - * Purpose: - * Set the selected stream to repeat. - * - * Inputs: - * pEASData - handle to data for this instance - * streamHandle - handle to stream - * repeatCount - repeat count (0 = no repeat, -1 = repeat forever) - * - * Outputs: - * - * Side Effects: - * - * Notes: - * 0 = no repeat - * 1 = repeat once, i.e. play through twice - * -1 = repeat forever - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetRepeat (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 repeatCount); - -/*---------------------------------------------------------------------------- - * EAS_GetRepeat() - *---------------------------------------------------------------------------- - * Purpose: - * Gets the current repeat count for the selected stream. - * - * Inputs: - * pEASData - handle to data for this instance - * streamHandle - handle to stream - * pRrepeatCount - pointer to variable to hold repeat count - * - * Outputs: - * - * Side Effects: - * - * Notes: - * 0 = no repeat - * 1 = repeat once, i.e. play through twice - * -1 = repeat forever - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetRepeat (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 *pRepeatCount); - -/*---------------------------------------------------------------------------- - * EAS_SetPlaybackRate() - *---------------------------------------------------------------------------- - * Purpose: - * Set the playback rate. - * - * Inputs: - * pEASData - handle to data for this instance - * streamHandle - handle to stream - * rate - rate (28-bit fractional amount) - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetPlaybackRate (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_U32 rate); -#define MAX_PLAYBACK_RATE (EAS_U32)(1L << 29) -#define MIN_PLAYBACK_RATE (EAS_U32)(1L << 27) - -/*---------------------------------------------------------------------------- - * EAS_SetTransposition) - *---------------------------------------------------------------------------- - * Purpose: - * Sets the key tranposition for the synthesizer. Transposes all - * melodic instruments by the specified amount. Range is limited - * to +/-12 semitones. - * - * Inputs: - * pEASData - handle to data for this instance - * streamHandle - handle to stream - * transposition - +/-12 semitones - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetTransposition (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 transposition); -#define MAX_TRANSPOSE 12 - -/*---------------------------------------------------------------------------- - * EAS_SetSynthPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Set the polyphony of the synthesizer. Value must be >= 1 and <= the - * maximum number of voices. This function will pin the polyphony - * at those limits - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * synthNum - synthesizer number (0 = onboard, 1 = DSP) - * polyphonyCount - the desired polyphony count - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetSynthPolyphony (EAS_DATA_HANDLE pEASData, EAS_I32 synthNum, EAS_I32 polyphonyCount); - -/*---------------------------------------------------------------------------- - * EAS_GetSynthPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current polyphony setting of the synthesizer - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * synthNum - synthesizer number (0 = onboard, 1 = DSP) - * pPolyphonyCount - pointer to variable to receive polyphony count - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetSynthPolyphony (EAS_DATA_HANDLE pEASData, EAS_I32 synthNum, EAS_I32 *pPolyphonyCount); - -/*---------------------------------------------------------------------------- - * EAS_SetPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Set the polyphony of the stream. Value must be >= 1 and <= the - * maximum number of voices. This function will pin the polyphony - * at those limits - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - handle returned by EAS_OpenFile - * polyphonyCount - the desired polyphony count - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetPolyphony (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 polyphonyCount); - -/*---------------------------------------------------------------------------- - * EAS_GetPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current polyphony setting of the stream - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - handle returned by EAS_OpenFile - * pPolyphonyCount - pointer to variable to receive polyphony count - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetPolyphony (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 *pPolyphonyCount); - -/*---------------------------------------------------------------------------- - * EAS_SetPriority() - *---------------------------------------------------------------------------- - * Purpose: - * Set the priority of the stream. Determines which stream's voices - * are stolen when there are insufficient voices for all notes. - * Value must be in the range of 1-255, lower values are higher - * priority. The default priority is 50. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - handle returned by EAS_OpenFile - * polyphonyCount - the desired polyphony count - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetPriority (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 priority); - -/*---------------------------------------------------------------------------- - * EAS_GetPriority() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current priority setting of the stream - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - handle returned by EAS_OpenFile - * pPriority - pointer to variable to receive priority - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetPriority (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 *pPriority); - -/*---------------------------------------------------------------------------- - * EAS_SetVolume() - *---------------------------------------------------------------------------- - * Purpose: - * Set the master volume for the mixer. The default volume setting is - * 90 (-10 dB). The volume range is 0 to 100 in 1dB increments. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * volume - the desired master volume - * - * Outputs: - * - * - * Side Effects: - * overrides any previously set master volume from sysex - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetVolume (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 volume); - -/*---------------------------------------------------------------------------- - * EAS_GetVolume() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the master volume for the mixer in 1dB increments. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * volume - the desired master volume - * - * Outputs: - * - * - * Side Effects: - * overrides any previously set master volume from sysex - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_I32 EAS_GetVolume (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle); - -/*---------------------------------------------------------------------------- - * EAS_SetMaxLoad() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the maximum workload the parsers will do in a single call to - * EAS_Render. The units are currently arbitrary, but should correlate - * well to the actual CPU cycles consumed. The primary effect is to - * reduce the occasional peaks in CPU cycles consumed when parsing - * dense parts of a MIDI score. Setting maxWorkLoad to zero disables - * the workload limiting function. - * - * Inputs: - * pEASData - handle to data for this instance - * maxLoad - the desired maximum workload - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetMaxLoad (EAS_DATA_HANDLE pEASData, EAS_I32 maxLoad); - -/*---------------------------------------------------------------------------- - * EAS_SetMaxPCMStreams() - *---------------------------------------------------------------------------- - * Sets the maximum number of PCM streams allowed in parsers that - * use PCM streaming. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - handle returned by EAS_OpenFile - * maxNumStreams - maximum number of PCM streams - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetMaxPCMStreams (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 maxNumStreams); - -/*---------------------------------------------------------------------------- - * EAS_OpenFile() - *---------------------------------------------------------------------------- - * Purpose: - * Opens a file for audio playback. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * locator - pointer to filename or other locating information - * pStreamHandle - pointer to stream handle variable - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_OpenFile (EAS_DATA_HANDLE pEASData, EAS_FILE_LOCATOR locator, EAS_HANDLE *pStreamHandle); - -#ifdef MMAPI_SUPPORT -/*---------------------------------------------------------------------------- - * EAS_MMAPIToneControl() - *---------------------------------------------------------------------------- - * Purpose: - * Opens a ToneControl file for audio playback. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * locator - pointer to filename or other locating information - * pStreamHandle - pointer to stream handle variable - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_MMAPIToneControl (EAS_DATA_HANDLE pEASData, EAS_FILE_LOCATOR locator, EAS_HANDLE *pStreamHandle); - -/*---------------------------------------------------------------------------- - * EAS_GetWaveFmtChunk - *---------------------------------------------------------------------------- - * Helper function to retrieve WAVE file fmt chunk for MMAPI - *---------------------------------------------------------------------------- - * pEASData - pointer to EAS persistent data object - * streamHandle - stream handle - * pFmtChunk - pointer to pointer to FMT chunk data - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetWaveFmtChunk (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_VOID_PTR *ppFmtChunk); -#endif - -/*---------------------------------------------------------------------------- - * EAS_GetFileType - *---------------------------------------------------------------------------- - * Returns the file type (see eas_types.h for enumerations) - *---------------------------------------------------------------------------- - * pEASData - pointer to EAS persistent data object - * streamHandle - stream handle - * pFileType - pointer to variable to receive file type - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetFileType (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 *pFileType); - -/*---------------------------------------------------------------------------- - * EAS_ParseMetaData() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - file or stream handle - * playLength - pointer to variable to store the play length (in msecs) - * - * Outputs: - * - * - * Side Effects: - * - resets the parser to the start of the file - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_ParseMetaData (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 *pPlayLength); - -/*---------------------------------------------------------------------------- - * EAS_Prepare() - *---------------------------------------------------------------------------- - * Purpose: - * Prepares the synthesizer to play the file or stream. Parses the first - * frame of data from the file and arms the synthesizer. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - file or stream handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_Prepare (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle); - -/*---------------------------------------------------------------------------- - * EAS_State() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the state of an audio file or stream. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - file or stream handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_State (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_STATE *pState); - -/*---------------------------------------------------------------------------- - * EAS_RegisterMetaDataCallback() - *---------------------------------------------------------------------------- - * Purpose: - * Registers a metadata callback function for parsed metadata. To - * de-register the callback, call this function again with parameter - * cbFunc set to NULL. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - file or stream handle - * cbFunc - pointer to host callback function - * metaDataBuffer - pointer to metadata buffer - * metaDataBufSize - maximum size of the metadata buffer - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_RegisterMetaDataCallback ( - EAS_DATA_HANDLE pEASData, - EAS_HANDLE streamHandle, - EAS_METADATA_CBFUNC cbFunc, - char *metaDataBuffer, - EAS_I32 metaDataBufSize, - EAS_VOID_PTR pUserData); - -/*---------------------------------------------------------------------------- - * EAS_GetNoteCount () - *---------------------------------------------------------------------------- - * Returns the total number of notes played in this stream - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - file or stream handle - * pNoteCount - pointer to variable to receive note count - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetNoteCount (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 *pNoteCount); - -/*---------------------------------------------------------------------------- - * EAS_CloseFile() - *---------------------------------------------------------------------------- - * Purpose: - * Closes an audio file or stream. Playback should have either paused or - * completed (EAS_State returns EAS_PAUSED or EAS_STOPPED). - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - file or stream handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_CloseFile (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle); - -/*---------------------------------------------------------------------------- - * EAS_OpenMIDIStream() - *---------------------------------------------------------------------------- - * Purpose: - * Opens a raw MIDI stream allowing the host to route MIDI cable data directly to the synthesizer - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * pStreamHandle - pointer to variable to hold file or stream handle - * streamHandle - open stream or NULL for new synthesizer instance - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_OpenMIDIStream (EAS_DATA_HANDLE pEASData, EAS_HANDLE *pStreamHandle, EAS_HANDLE streamHandle); - -/*---------------------------------------------------------------------------- - * EAS_WriteMIDIStream() - *---------------------------------------------------------------------------- - * Purpose: - * Send data to the MIDI stream device - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - stream handle - * pBuffer - pointer to buffer - * count - number of bytes to write - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_WriteMIDIStream(EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_U8 *pBuffer, EAS_I32 count); - -/*---------------------------------------------------------------------------- - * EAS_CloseMIDIStream() - *---------------------------------------------------------------------------- - * Purpose: - * Closes a raw MIDI stream - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - stream handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_CloseMIDIStream (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle); - -/*---------------------------------------------------------------------------- - * EAS_Locate() - *---------------------------------------------------------------------------- - * Purpose: - * Locate into the file associated with the handle. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - file handle - * milliseconds - playback offset from start of file in milliseconds - * - * Outputs: - * - * - * Side Effects: - * the actual offset will be quantized to the closest update period, typically - * a resolution of 5.9ms. Notes that are started prior to this time will not - * sound. Any notes currently playing will be shut off. - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_Locate (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 milliseconds, EAS_BOOL offset); - -/*---------------------------------------------------------------------------- - * EAS_GetRenderTime() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current playback offset - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * - * Outputs: - * Gets the render time clock in msecs. - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetRenderTime (EAS_DATA_HANDLE pEASData, EAS_I32 *pTime); - -/*---------------------------------------------------------------------------- - * EAS_GetLocation() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current playback offset - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - file handle - * - * Outputs: - * The offset in milliseconds from the start of the current sequence, quantized - * to the nearest update period. Actual resolution is typically 5.9 ms. - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetLocation (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 *pTime); - -/*---------------------------------------------------------------------------- - * EAS_Pause() - *---------------------------------------------------------------------------- - * Purpose: - * Pauses the playback of the data associated with this handle. The audio - * is gracefully ramped down to prevent clicks and pops. It may take several - * buffers of audio before the audio is muted. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * streamHandle - file or stream handle - * - * Outputs: - * - * - * Side Effects: - * - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_Pause (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle); - -/*---------------------------------------------------------------------------- - * EAS_Resume() - *---------------------------------------------------------------------------- - * Purpose: - * Resumes the playback of the data associated with this handle. The audio - * is gracefully ramped up to prevent clicks and pops. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * streamHandle - file or stream handle - * - * Outputs: - * - * - * Side Effects: - * - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_Resume (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle); - -/*---------------------------------------------------------------------------- - * EAS_GetParameter() - *---------------------------------------------------------------------------- - * Purpose: - * Set the parameter of a module. See E_MODULES for a list of modules - * and the header files of the modules for a list of parameters. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * module - enumerated module number - * param - enumerated parameter number - * pValue - pointer to variable to receive parameter value - * - * Outputs: - * - * - * Side Effects: - * - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetParameter (EAS_DATA_HANDLE pEASData, EAS_I32 module, EAS_I32 param, EAS_I32 *pValue); - -/*---------------------------------------------------------------------------- - * EAS_SetParameter() - *---------------------------------------------------------------------------- - * Purpose: - * Set the parameter of a module. See E_MODULES for a list of modules - * and the header files of the modules for a list of parameters. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * handle - file or stream handle - * module - enumerated module number - * param - enumerated parameter number - * value - new parameter value - * - * Outputs: - * - * - * Side Effects: - * - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetParameter (EAS_DATA_HANDLE pEASData, EAS_I32 module, EAS_I32 param, EAS_I32 value); - -#ifdef _METRICS_ENABLED -/*---------------------------------------------------------------------------- - * EAS_MetricsReport() - *---------------------------------------------------------------------------- - * Purpose: - * Displays the current metrics through the EAS_Report interface. - * - * Inputs: - * pEASData - instance data handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_MetricsReport (EAS_DATA_HANDLE pEASData); - -/*---------------------------------------------------------------------------- - * EAS_MetricsReset() - *---------------------------------------------------------------------------- - * Purpose: - * Displays the current metrics through the EAS_Report interface. - * - * Inputs: - * pEASData - instance data handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_MetricsReset (EAS_DATA_HANDLE pEASData); -#endif - -/*---------------------------------------------------------------------------- - * EAS_SetSoundLibrary() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the location of the sound library. - * - * Inputs: - * pEASData - instance data handle - * streamHandle - file or stream handle - * pSoundLib - pointer to sound library - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetSoundLibrary (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_SNDLIB_HANDLE pSndLib); - -/*---------------------------------------------------------------------------- - * EAS_SetHeaderSearchFlag() - *---------------------------------------------------------------------------- - * By default, when EAS_OpenFile is called, the parsers check the - * first few bytes of the file looking for a specific header. Some - * mobile devices may add a header to the start of a file, which - * will prevent the parser from recognizing the file. If the - * searchFlag is set to EAS_TRUE, the parser will search the entire - * file looking for the header. This may enable EAS to recognize - * some files that it would ordinarily reject. The negative is that - * it make take slightly longer to process the EAS_OpenFile request. - * - * Inputs: - * pEASData - instance data handle - * searchFlag - search flag (EAS_TRUE or EAS_FALSE) - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetHeaderSearchFlag (EAS_DATA_HANDLE pEASData, EAS_BOOL searchFlag); - -/*---------------------------------------------------------------------------- - * EAS_SetPlayMode() - *---------------------------------------------------------------------------- - * Some file formats support special play modes, such as iMode partial - * play mode. This call can be used to change the play mode. The - * default play mode (usually straight playback) is always zero. - * - * Inputs: - * pEASData - instance data handle - * handle - file or stream handle - * playMode - play mode (see eas_types.h for enumerations) - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetPlayMode (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 playMode); - -#ifdef DLS_SYNTHESIZER -/*---------------------------------------------------------------------------- - * EAS_LoadDLSCollection() - *---------------------------------------------------------------------------- - * Purpose: - * Downloads a DLS collection - * - * Inputs: - * pEASData - instance data handle - * streamHandle - file or stream handle - * locator - file locator - * - * Outputs: - * - * - * Side Effects: - * May overlay instruments in the GM sound set - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_LoadDLSCollection (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_FILE_LOCATOR locator); -#endif - -/*---------------------------------------------------------------------------- - * EAS_SetFrameBuffer() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the frame buffer pointer passed to the IPC communications functions - * - * Inputs: - * pEASData - instance data handle - * locator - file locator - * - * Outputs: - * - * - * Side Effects: - * May overlay instruments in the GM sound set - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetFrameBuffer (EAS_DATA_HANDLE pEASData, EAS_FRAME_BUFFER_HANDLE pFrameBuffer); - -#ifdef EXTERNAL_AUDIO -/*---------------------------------------------------------------------------- - * EAS_RegExtAudioCallback() - *---------------------------------------------------------------------------- - * Purpose: - * Registers callback functions for audio events. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - file or stream handle - * cbProgChgFunc - pointer to host callback function for program change - * cbEventFunc - pointer to host callback functio for note events - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_RegExtAudioCallback (EAS_DATA_HANDLE pEASData, - EAS_HANDLE streamHandle, - EAS_VOID_PTR pInstData, - EAS_EXT_PRG_CHG_FUNC cbProgChgFunc, - EAS_EXT_EVENT_FUNC cbEventFunc); - -/*---------------------------------------------------------------------------- - * EAS_GetMIDIControllers() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current state of MIDI controllers on the requested channel. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - file or stream handle - * pControl - pointer to structure to receive data - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetMIDIControllers (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_U8 channel, S_MIDI_CONTROLLERS *pControl); -#endif - -/*---------------------------------------------------------------------------- - * EAS_SearchFile - *---------------------------------------------------------------------------- - * Search file for specific sequence starting at current file - * position. Returns offset to start of sequence. - * - * Inputs: - * pEASData - pointer to EAS persistent data object - * fileHandle - file handle - * searchString - pointer to search sequence - * len - length of search sequence - * pOffset - pointer to variable to store offset to sequence - * - * Returns EAS_EOF if end-of-file is reached - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_SearchFile (EAS_DATA_HANDLE pEASData, EAS_FILE_HANDLE fileHandle, const EAS_U8 *searchString, EAS_I32 len, EAS_I32 *pOffset); - -#ifdef __cplusplus -} /* end extern "C" */ -#endif - -#endif /* #ifndef _EAS_H */ diff --git a/arm-hybrid-22k/host_src/eas_build.h b/arm-hybrid-22k/host_src/eas_build.h deleted file mode 100644 index 3663322..0000000 --- a/arm-hybrid-22k/host_src/eas_build.h +++ /dev/null @@ -1,36 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * host_src\eas_build.h - * - * Contents and purpose: - * This file contains the build configuration for this - * build. The buildGUIDStr is a GUID created during - * the build process and is guaranteed to be unique - * for each build. - * - * Copyright Sonic Network Inc. 2006 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - * This file was autogenerated by buildid.exe - *---------------------------------------------------------------------------- -*/ - -#ifndef _GUID_0cbd200ab054487f9ece7d3c8852e426_ -#define _GUID_0cbd200ab054487f9ece7d3c8852e426_ - -#define _BUILD_VERSION_ "0cbd200a-b054-487f-9ece-7d3c8852e426" -#define _BUILD_TIME_ 0x4743b9dc - -#endif /* _GUID_0cbd200ab054487f9ece7d3c8852e426_ */ diff --git a/arm-hybrid-22k/host_src/eas_chorus.h b/arm-hybrid-22k/host_src/eas_chorus.h deleted file mode 100644 index 0e9057f..0000000 --- a/arm-hybrid-22k/host_src/eas_chorus.h +++ /dev/null @@ -1,53 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_chorus.h - * - * Contents and purpose: - * Contains parameter enumerations for the Chorus effect - * - * - * Copyright Sonic Network Inc. 2006 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 309 $ - * $Date: 2006-09-12 18:52:45 -0700 (Tue, 12 Sep 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef EAS_CHORUS_H -#define EAS_CHORUS_H - -/* enumerated parameter settings for Chorus effect */ -typedef enum -{ - EAS_PARAM_CHORUS_BYPASS, - EAS_PARAM_CHORUS_PRESET, - EAS_PARAM_CHORUS_RATE, - EAS_PARAM_CHORUS_DEPTH, - EAS_PARAM_CHORUS_LEVEL -} E_CHORUS_PARAMS; - -typedef enum -{ - EAS_PARAM_CHORUS_PRESET1, - EAS_PARAM_CHORUS_PRESET2, - EAS_PARAM_CHORUS_PRESET3, - EAS_PARAM_CHORUS_PRESET4 -} E_CHORUS_PRESETS; - - -#endif \ No newline at end of file diff --git a/arm-hybrid-22k/host_src/eas_config.c b/arm-hybrid-22k/host_src/eas_config.c deleted file mode 100644 index c45fbb7..0000000 --- a/arm-hybrid-22k/host_src/eas_config.c +++ /dev/null @@ -1,619 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_config.c - * - * Contents and purpose: - * This file contains the Configuration Module interface (CM). The CM - * is a module compiled external to the library that sets the configuration - * for this build. It allows the library to find optional components and - * links to static memory allocations (when used in a static configuration). - * - * DO NOT MODIFY THIS FILE! - * - * NOTE: This module is not intended to be modified by the customer. It - * needs to be included in the build process with the correct configuration - * defines (see the library documentation for information on how to configure - * the library). - * - * Copyright Sonic Network Inc. 2004-2006 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 796 $ - * $Date: 2007-08-01 00:15:25 -0700 (Wed, 01 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas.h" -#include "eas_config.h" - - -#ifdef _MFI_PARSER -/*---------------------------------------------------------------------------- - * Vendor/Device ID for MFi Extensions - * - * Define the preprocessor symbols to establish the vendor ID and - * device ID for the MFi PCM/ADPCM extensions. - *---------------------------------------------------------------------------- -*/ -const EAS_U8 eas_MFIVendorIDMSB = (MFI_VENDOR_ID >> 8) & 0xff; -const EAS_U8 eas_MFIVendorIDLSB = MFI_VENDOR_ID & 0xff; -const EAS_U8 eas_MFIDeviceID = MFI_DEVICE_ID; -#endif - -/*---------------------------------------------------------------------------- - * - * parserModules - * - * This structure is used by the EAS library to locate file parsing - * modules. - *---------------------------------------------------------------------------- -*/ - -/* define the external file parsers */ -extern EAS_VOID_PTR EAS_SMF_Parser; - -#ifdef _XMF_PARSER -extern EAS_VOID_PTR EAS_XMF_Parser; -#endif - -#ifdef _SMAF_PARSER -extern EAS_VOID_PTR EAS_SMAF_Parser; -#endif - -#ifdef _WAVE_PARSER -extern EAS_VOID_PTR EAS_Wave_Parser; -#endif - -#ifdef _OTA_PARSER -extern EAS_VOID_PTR EAS_OTA_Parser; -#endif - -#ifdef _IMELODY_PARSER -extern EAS_VOID_PTR EAS_iMelody_Parser; -#endif - -#ifdef _RTTTL_PARSER -extern EAS_VOID_PTR EAS_RTTTL_Parser; -#endif - -#if defined (_CMX_PARSER) || defined(_MFI_PARSER) -extern EAS_VOID_PTR EAS_CMF_Parser; -#endif - -/* initalize pointers to parser interfaces */ -/*lint -e{605} not pretty, but it works */ -EAS_VOID_PTR const parserModules[] = -{ - &EAS_SMF_Parser, - -#ifdef _XMF_PARSER - &EAS_XMF_Parser, -#endif - -#ifdef _WAVE_PARSER - &EAS_Wave_Parser, -#endif - -#ifdef _SMAF_PARSER - &EAS_SMAF_Parser, -#endif - -#ifdef _OTA_PARSER - &EAS_OTA_Parser, -#endif - -#ifdef _IMELODY_PARSER - &EAS_iMelody_Parser, -#endif - -#ifdef _RTTTL_PARSER - &EAS_RTTTL_Parser, -#endif - -#if defined (_CMX_PARSER) || defined(_MFI_PARSER) - &EAS_CMF_Parser -#endif -}; -#define NUM_PARSER_MODULES (sizeof(parserModules) / sizeof(EAS_VOID_PTR)) - -/*---------------------------------------------------------------------------- - * Data Modules - *---------------------------------------------------------------------------- -*/ - -#ifdef _STATIC_MEMORY -extern EAS_VOID_PTR eas_SMFData; -extern EAS_VOID_PTR eas_Data; -extern EAS_VOID_PTR eas_MixBuffer; -extern EAS_VOID_PTR eas_Synth; -extern EAS_VOID_PTR eas_MIDI; -extern EAS_VOID_PTR eas_PCMData; -extern EAS_VOID_PTR eas_MIDIData; - -#ifdef _XMF_PARSER -extern EAS_VOID_PTR eas_XMFData; -#endif - -#ifdef _SMAF_PARSER -extern EAS_VOID_PTR eas_SMAFData; -#endif - -#ifdef _OTA_PARSER -extern EAS_VOID_PTR eas_OTAData; -#endif - -#ifdef _IMELODY_PARSER -extern EAS_VOID_PTR eas_iMelodyData; -#endif - -#ifdef _RTTTL_PARSER -extern EAS_VOID_PTR eas_RTTTLData; -#endif - -#ifdef _WAVE_PARSER -extern EAS_VOID_PTR eas_WaveData; -#endif - -#if defined (_CMX_PARSER) || defined(_MFI_PARSER) -extern EAS_VOID_PTR eas_CMFData; -#endif -#endif - -/*---------------------------------------------------------------------------- - * - * Effects Modules - * - * These declarations are used by the EAS library to locate - * effects modules. - *---------------------------------------------------------------------------- -*/ - -#ifdef _ENHANCER_ENABLED -extern EAS_VOID_PTR EAS_Enhancer; -#define EAS_ENHANCER_INTERFACE &EAS_Enhancer -#ifdef _STATIC_MEMORY -extern EAS_VOID_PTR eas_EnhancerData; -#define EAS_ENHANCER_DATA &eas_EnhancerData -#else -#define EAS_ENHANCER_DATA NULL -#endif -#else -#define EAS_ENHANCER_INTERFACE NULL -#define EAS_ENHANCER_DATA NULL -#endif - -#ifdef _COMPRESSOR_ENABLED -extern EAS_VOID_PTR EAS_Compressor; -#define EAS_COMPRESSOR_INTERFACE &EAS_Compressor -#ifdef _STATIC_MEMORY -extern EAS_VOID_PTR eas_CompressorData; -#define EAS_COMPRESSOR_DATA &eas_CompressorData -#else -#define EAS_COMPRESSOR_DATA NULL -#endif -#else -#define EAS_COMPRESSOR_INTERFACE NULL -#define EAS_COMPRESSOR_DATA NULL -#endif - -#ifdef _MAXIMIZER_ENABLED -extern EAS_VOID_PTR EAS_Maximizer; -#define EAS_MAXIMIZER_INTERFACE &EAS_Maximizer -#ifdef _STATIC_MEMORY -extern EAS_VOID_PTR eas_MaximizerData; -#define EAS_MAXIMIZER_DATA &eas_MaximizerData -#else -#define EAS_MAXIMIZER_DATA NULL -#endif -#else -#define EAS_MAXIMIZER_INTERFACE NULL -#define EAS_MAXIMIZER_DATA NULL -#endif - - -#ifdef _REVERB_ENABLED -extern EAS_VOID_PTR EAS_Reverb; -#define EAS_REVERB_INTERFACE &EAS_Reverb -#ifdef _STATIC_MEMORY -extern EAS_VOID_PTR eas_ReverbData; -#define EAS_REVERB_DATA &eas_ReverbData -#else -#define EAS_REVERB_DATA NULL -#endif -#else -#define EAS_REVERB_INTERFACE NULL -#define EAS_REVERB_DATA NULL -#endif - -#ifdef _CHORUS_ENABLED -extern EAS_VOID_PTR EAS_Chorus; -#define EAS_CHORUS_INTERFACE &EAS_Chorus -#ifdef _STATIC_MEMORY -extern EAS_VOID_PTR eas_ChorusData; -#define EAS_CHORUS_DATA &eas_ChorusData -#else -#define EAS_CHORUS_DATA NULL -#endif -#else -#define EAS_CHORUS_INTERFACE NULL -#define EAS_CHORUS_DATA NULL -#endif - -#ifdef _WIDENER_ENABLED -extern EAS_VOID_PTR EAS_Widener; -#define EAS_WIDENER_INTERFACE &EAS_Widener -#ifdef _STATIC_MEMORY -extern EAS_VOID_PTR eas_WidenerData; -#define EAS_WIDENER_DATA &eas_WidenerData -#else -#define EAS_WIDENER_DATA NULL -#endif -#else -#define EAS_WIDENER_INTERFACE NULL -#define EAS_WIDENER_DATA NULL -#endif - -#ifdef _GRAPHIC_EQ_ENABLED -extern EAS_VOID_PTR EAS_GraphicEQ; -#define EAS_GRAPHIC_EQ_INTERFACE &EAS_GraphicEQ -#ifdef _STATIC_MEMORY -extern EAS_VOID_PTR eas_GraphicEQData; -#define EAS_GRAPHIC_EQ_DATA &eas_GraphicEQData -#else -#define EAS_GRAPHIC_EQ_DATA NULL -#endif -#else -#define EAS_GRAPHIC_EQ_INTERFACE NULL -#define EAS_GRAPHIC_EQ_DATA NULL -#endif - -#ifdef _WOW_ENABLED -extern EAS_VOID_PTR EAS_Wow; -#define EAS_WOW_INTERFACE &EAS_Wow -#ifdef _STATIC_MEMORY -#error "WOW module requires dynamic memory model" -#else -#define EAS_WOW_DATA NULL -#endif -#else -#define EAS_WOW_INTERFACE NULL -#define EAS_WOW_DATA NULL -#endif - -#ifdef _TONECONTROLEQ_ENABLED -extern EAS_VOID_PTR EAS_ToneControlEQ; -#define EAS_TONECONTROLEQ_INTERFACE &EAS_ToneControlEQ -#ifdef _STATIC_MEMORY -extern EAS_VOID_PTR eas_ToneControlEQData; -#define EAS_TONECONTROLEQ_DATA &eas_ToneControlEQData -#else -#define EAS_TONECONTROLEQ_DATA NULL -#endif -#else -#define EAS_TONECONTROLEQ_INTERFACE NULL -#define EAS_TONECONTROLEQ_DATA NULL -#endif - -/*lint -e{605} not pretty, but it works */ -EAS_VOID_PTR const effectsModules[] = -{ - EAS_ENHANCER_INTERFACE, - EAS_COMPRESSOR_INTERFACE, - EAS_REVERB_INTERFACE, - EAS_CHORUS_INTERFACE, - EAS_WIDENER_INTERFACE, - EAS_GRAPHIC_EQ_INTERFACE, - EAS_WOW_INTERFACE, - EAS_MAXIMIZER_INTERFACE, - EAS_TONECONTROLEQ_INTERFACE -}; - -EAS_VOID_PTR const effectsData[] = -{ - EAS_ENHANCER_DATA, - EAS_COMPRESSOR_DATA, - EAS_REVERB_DATA, - EAS_CHORUS_DATA, - EAS_WIDENER_DATA, - EAS_GRAPHIC_EQ_DATA, - EAS_WOW_DATA, - EAS_MAXIMIZER_DATA, - EAS_TONECONTROLEQ_DATA -}; - -/*---------------------------------------------------------------------------- - * - * Optional Modules - * - * These declarations are used by the EAS library to locate - * effects modules. - *---------------------------------------------------------------------------- -*/ - -#ifdef _METRICS_ENABLED -extern EAS_VOID_PTR EAS_Metrics; -#define EAS_METRICS_INTERFACE &EAS_Metrics -#ifdef _STATIC_MEMORY -extern EAS_VOID_PTR eas_MetricsData; -#define EAS_METRICS_DATA &eas_MetricsData -#else -#define EAS_METRICS_DATA NULL -#endif -#else -#define EAS_METRICS_INTERFACE NULL -#define EAS_METRICS_DATA NULL -#endif - -#ifdef MMAPI_SUPPORT -extern EAS_VOID_PTR EAS_TC_Parser; -#define EAS_TONE_CONTROL_PARSER &EAS_TC_Parser -#ifdef _STATIC_MEMORY -extern EAS_VOID_PTR eas_TCData; -#define EAS_TONE_CONTROL_DATA &eas_TCData -#else -#define EAS_TONE_CONTROL_DATA NULL -#endif -#else -#define EAS_TONE_CONTROL_PARSER NULL -#define EAS_TONE_CONTROL_DATA NULL -#endif - -/*lint -e{605} not pretty, but it works */ -EAS_VOID_PTR const optionalModules[] = -{ - EAS_TONE_CONTROL_PARSER, - EAS_METRICS_INTERFACE -}; - -EAS_VOID_PTR const optionalData[] = -{ - EAS_TONE_CONTROL_DATA, - EAS_METRICS_DATA -}; - -/*---------------------------------------------------------------------------- - * EAS_CMStaticMemoryModel() - *---------------------------------------------------------------------------- - * Purpose: - * This function returns true if EAS has been configured for - * a static memory model. There are some limitations in the - * static memory model, see the documentation for more - * information. - * - * Outputs: - * returns EAS_TRUE if a module is found - *---------------------------------------------------------------------------- -*/ -EAS_BOOL EAS_CMStaticMemoryModel (void) -{ -#ifdef _STATIC_MEMORY - return EAS_TRUE; -#else - return EAS_FALSE; -#endif -} - -/*---------------------------------------------------------------------------- - * EAS_CMEnumModules() - *---------------------------------------------------------------------------- - * Purpose: - * This function is used to find pointers to optional modules. - * - * Inputs: - * module - module number - * - * Outputs: - * returns a pointer to the module function table or NULL if no module - *---------------------------------------------------------------------------- -*/ -EAS_VOID_PTR EAS_CMEnumModules (EAS_INT module) -{ - - if (module >= (EAS_INT) NUM_PARSER_MODULES) - return NULL; - return parserModules[module]; -} - -/*---------------------------------------------------------------------------- - * EAS_CMEnumData() - *---------------------------------------------------------------------------- - * Purpose: - * This function is used to find pointers to static memory allocations. - * - * Inputs: - * dataModule - enumerated module number - * - * Outputs: - * Returns handle to data or NULL if not found - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, dataModule) used only when _STATIC_MEMORY is defined */ -EAS_VOID_PTR EAS_CMEnumData (EAS_INT dataModule) -{ - -#ifdef _STATIC_MEMORY - switch (dataModule) - { - - /* main instance data for synthesizer */ - case EAS_CM_EAS_DATA: - return &eas_Data; - - /* mix buffer for mix engine */ - case EAS_CM_MIX_BUFFER: - /*lint -e{545} lint doesn't like this because it sees the underlying type */ - return &eas_MixBuffer; - - /* instance data for synth */ - case EAS_CM_SYNTH_DATA: - return &eas_Synth; - - /* instance data for MIDI parser */ - case EAS_CM_MIDI_DATA: - return &eas_MIDI; - - /* instance data for SMF parser */ - case EAS_CM_SMF_DATA: - return &eas_SMFData; - -#ifdef _XMF_PARSER - /* instance data for XMF parser */ - case EAS_CM_XMF_DATA: - return &eas_XMFData; -#endif - -#ifdef _SMAF_PARSER - /* instance data for SMAF parser */ - case EAS_CM_SMAF_DATA: - return &eas_SMAFData; -#endif - - /* instance data for the PCM engine */ - case EAS_CM_PCM_DATA: - /*lint -e{545} lint doesn't like this because it sees the underlying type */ - return &eas_PCMData; - - case EAS_CM_MIDI_STREAM_DATA: - return &eas_MIDIData; - -#ifdef _OTA_PARSER - /* instance data for OTA parser */ - case EAS_CM_OTA_DATA: - return &eas_OTAData; -#endif - -#ifdef _IMELODY_PARSER - /* instance data for iMelody parser */ - case EAS_CM_IMELODY_DATA: - return &eas_iMelodyData; -#endif - -#ifdef _RTTTL_PARSER - /* instance data for RTTTL parser */ - case EAS_CM_RTTTL_DATA: - return &eas_RTTTLData; -#endif - -#ifdef _WAVE_PARSER - /* instance data for WAVE parser */ - case EAS_CM_WAVE_DATA: - return &eas_WaveData; -#endif - -#if defined (_CMX_PARSER) || defined(_MFI_PARSER) - /* instance data for CMF parser */ - case EAS_CM_CMF_DATA: - return &eas_CMFData; -#endif - - default: - return NULL; - } - -#else - return NULL; -#endif -} - -/*---------------------------------------------------------------------------- - * EAS_CMEnumFXModules() - *---------------------------------------------------------------------------- - * Purpose: - * This function is used to find pointers to optional effects modules. - * - * Inputs: - * module - enumerated module number - * pModule - pointer to module interface - * - * Outputs: - * Returns pointer to function table or NULL if not found - *---------------------------------------------------------------------------- -*/ -EAS_VOID_PTR EAS_CMEnumFXModules (EAS_INT module) -{ - - if (module >= NUM_EFFECTS_MODULES) - return NULL; - return effectsModules[module]; -} - -/*---------------------------------------------------------------------------- - * EAS_CMEnumFXData() - *---------------------------------------------------------------------------- - * Purpose: - * This function is used to find pointers to static memory allocations. - * - * Inputs: - * dataModule - enumerated module number - * pData - pointer to handle variable - * - * Outputs: - * Returns handle to data or NULL if not found - *---------------------------------------------------------------------------- -*/ -EAS_VOID_PTR EAS_CMEnumFXData (EAS_INT dataModule) -{ - - if (dataModule >= NUM_EFFECTS_MODULES) - return NULL; - return effectsData[dataModule]; -} - -/*---------------------------------------------------------------------------- - * EAS_CMEnumOptModules() - *---------------------------------------------------------------------------- - * Purpose: - * This function is used to find pointers to optional modules. - * - * Inputs: - * module - enumerated module number - * - * Outputs: - * returns pointer to function table or NULL if no module - *---------------------------------------------------------------------------- -*/ -EAS_VOID_PTR EAS_CMEnumOptModules (EAS_INT module) -{ - - /* sanity check */ - if (module >= NUM_OPTIONAL_MODULES) - return EAS_FALSE; - return optionalModules[module]; -} - -/*---------------------------------------------------------------------------- - * EAS_CMEnumOptData() - *---------------------------------------------------------------------------- - * Purpose: - * This function is used to find pointers to static memory allocations. - * - * Inputs: - * dataModule - enumerated module number - * - * Outputs: - * Returns handle to data or NULL if not found - *---------------------------------------------------------------------------- -*/ -EAS_VOID_PTR EAS_CMEnumOptData (EAS_INT dataModule) -{ - - if (dataModule >= NUM_OPTIONAL_MODULES) - return NULL; - return optionalData[dataModule]; -} - - diff --git a/arm-hybrid-22k/host_src/eas_config.h b/arm-hybrid-22k/host_src/eas_config.h deleted file mode 100644 index d16be4a..0000000 --- a/arm-hybrid-22k/host_src/eas_config.h +++ /dev/null @@ -1,191 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_config.h - * - * Contents and purpose: - * This header declares the Configuration Module interface (CM). The CM - * is a module compiled external to the library that sets the configuration - * for this build. It allows the library to find optional components and - * links to static memory allocations (when used in a static configuration). - * - * NOTE: This module is not intended to be modified by the customer. It - * needs to be included in the build process with the correct configuration - * defines (see the library documentation for information on how to configure - * the library). - * - * DO NOT MODIFY THIS FILE! - * - * Copyright 2005 Sonic Network Inc. - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -// sentinel -#ifndef _EAS_CONFIG_H -#define _EAS_CONFIG_H - -#include "eas_types.h" - -/* list of enumerators for optional modules */ -typedef enum { - EAS_CM_FILE_PARSERS = 1 -} E_CM_ENUM_MODULES; - -/* list of enumerators for module and memory pointers */ -typedef enum { - EAS_CM_EAS_DATA = 1, - EAS_CM_MIX_BUFFER, - EAS_CM_SYNTH_DATA, - EAS_CM_MIDI_DATA, - EAS_CM_SMF_DATA, - EAS_CM_XMF_DATA, - EAS_CM_SMAF_DATA, - EAS_CM_PCM_DATA, - EAS_CM_MIDI_STREAM_DATA, - EAS_CM_METRICS_DATA, - EAS_CM_OTA_DATA, - EAS_CM_IMELODY_DATA, - EAS_CM_RTTTL_DATA, - EAS_CM_WAVE_DATA, - EAS_CM_CMF_DATA -} E_CM_DATA_MODULES; - -typedef struct -{ - int maxSMFStreams; - void *pSMFData; - void *pSMFStream; -} S_EAS_SMF_PTRS; - -typedef struct -{ - int maxSMAFStreams; - void *pSMAFData; - void *pSMAFStream; -} S_EAS_SMAF_PTRS; - -/*---------------------------------------------------------------------------- - * EAS_CMStaticMemoryModel() - *---------------------------------------------------------------------------- - * Purpose: - * This function returns true if EAS has been configured for - * a static memory model. There are some limitations in the - * static memory model, see the documentation for more - * information. - * - * Outputs: - * returns EAS_TRUE if a module is found - *---------------------------------------------------------------------------- -*/ -EAS_BOOL EAS_CMStaticMemoryModel (void); - -/*---------------------------------------------------------------------------- - * EAS_CMEnumModules() - *---------------------------------------------------------------------------- - * Purpose: - * This function is used to find pointers to optional modules. - * - * Inputs: - * module - module number - * - * Outputs: - * returns a pointer to the module function table or NULL if no module - *---------------------------------------------------------------------------- -*/ -EAS_VOID_PTR EAS_CMEnumModules (EAS_INT module); - -/*---------------------------------------------------------------------------- - * EAS_CMEnumData() - *---------------------------------------------------------------------------- - * Purpose: - * This function is used to find pointers to static memory allocations. - * - * Inputs: - * dataModule - enumerated module number - * - * Outputs: - * Returns handle to data or NULL if not found - *---------------------------------------------------------------------------- -*/ -EAS_VOID_PTR EAS_CMEnumData (EAS_INT dataModule); - -/*---------------------------------------------------------------------------- - * EAS_CMEnumFXModules() - *---------------------------------------------------------------------------- - * Purpose: - * This function is used to find pointers to optional effects modules. - * - * Inputs: - * module - enumerated module number - * pModule - pointer to module interface - * - * Outputs: - * Returns pointer to function table or NULL if not found - *---------------------------------------------------------------------------- -*/ -EAS_VOID_PTR EAS_CMEnumFXModules (EAS_INT module); - -/*---------------------------------------------------------------------------- - * EAS_CMEnumFXData() - *---------------------------------------------------------------------------- - * Purpose: - * This function is used to find pointers to static memory allocations. - * - * Inputs: - * dataModule - enumerated module number - * pData - pointer to handle variable - * - * Outputs: - * Returns handle to data or NULL if not found - *---------------------------------------------------------------------------- -*/ -EAS_VOID_PTR EAS_CMEnumFXData (EAS_INT dataModule); - -/*---------------------------------------------------------------------------- - * EAS_CMEnumOptModules() - *---------------------------------------------------------------------------- - * Purpose: - * This function is used to find pointers to optional modules. - * - * Inputs: - * module - enumerated module number - * - * Outputs: - * returns pointer to function table or NULL if no module - *---------------------------------------------------------------------------- -*/ -EAS_VOID_PTR EAS_CMEnumOptModules (EAS_INT module); - -/*---------------------------------------------------------------------------- - * EAS_CMEnumOptData() - *---------------------------------------------------------------------------- - * Purpose: - * This function is used to find pointers to static memory allocations. - * - * Inputs: - * dataModule - enumerated module number - * - * Outputs: - * Returns handle to data or NULL if not found - *---------------------------------------------------------------------------- -*/ -EAS_VOID_PTR EAS_CMEnumOptData (EAS_INT dataModule); - -#endif /* end _EAS_CONFIG_H */ diff --git a/arm-hybrid-22k/host_src/eas_debugmsgs.h b/arm-hybrid-22k/host_src/eas_debugmsgs.h deleted file mode 100644 index 3fff38f..0000000 --- a/arm-hybrid-22k/host_src/eas_debugmsgs.h +++ /dev/null @@ -1,44 +0,0 @@ -/* Auto-generated from source file: eas_chorusdata.c */ -/* Auto-generated from source file: eas_imelodydata.c */ -/* Auto-generated from source file: eas_mididata.c */ -/* Auto-generated from source file: eas_pan.c */ -/* Auto-generated from source file: eas_wavefiledata.c */ -/* Auto-generated from source file: eas_voicemgt.c */ -/* Auto-generated from source file: eas_ota.c */ -/* Auto-generated from source file: eas_mixbuf.c */ -/* Auto-generated from source file: eas_rtttl.c */ -/* Auto-generated from source file: eas_reverb.c */ -/* Auto-generated from source file: eas_fmsynth.c */ -/* Auto-generated from source file: eas_pcmdata.c */ -/* Auto-generated from source file: eas_chorus.c */ -/* Auto-generated from source file: eas_math.c */ -/* Auto-generated from source file: eas_fmengine.c */ -/* Auto-generated from source file: eas_smfdata.c */ -/* Auto-generated from source file: eas_fmtables.c */ -/* Auto-generated from source file: eas_imelody.c */ -/* Auto-generated from source file: eas_public.c */ -/* Auto-generated from source file: eas_rtttldata.c */ -/* Auto-generated from source file: eas_reverbdata.c */ -/* Auto-generated from source file: eas_wtengine.c */ -/* Auto-generated from source file: eas_imaadpcm.c */ -{ 0x2380b977, 0x00000006, "eas_imaadpcm.c[305]: IMADecoderLocate: Time=%d, samples=%d\n" }, -{ 0x2380b977, 0x00000007, "eas_imaadpcm.c[328]: IMADecoderLocate: Looped sample, numBlocks=%d, samplesPerLoop=%d, samplesInLastBlock=%d, samples=%d\n" }, -{ 0x2380b977, 0x00000008, "eas_imaadpcm.c[335]: IMADecoderLocate: Byte location in audio = %d\n" }, -{ 0x2380b977, 0x00000009, "eas_imaadpcm.c[345]: IMADecoderLocate: bytesLeft = %d\n" }, -/* Auto-generated from source file: eas_midi.c */ -/* Auto-generated from source file: eas_otadata.c */ -/* Auto-generated from source file: eas_ima_tables.c */ -/* Auto-generated from source file: eas_data.c */ -/* Auto-generated from source file: eas_pcm.c */ -/* Auto-generated from source file: eas_mixer.c */ -/* Auto-generated from source file: eas_wavefile.c */ -/* Auto-generated from source file: eas_wtsynth.c */ -/* Auto-generated from source file: eas_smf.c */ -/* Auto-generated from source file: eas_wave.c */ -/* Auto-generated from source file: eas_hostmm.c */ -{ 0x1a54b6e8, 0x00000001, "eas_hostmm.c[586]: Vibrate state: %d\n" }, -{ 0x1a54b6e8, 0x00000002, "eas_hostmm.c[601]: LED state: %d\n" }, -{ 0x1a54b6e8, 0x00000003, "eas_hostmm.c[616]: Backlight state: %d\n" }, -/* Auto-generated from source file: eas_config.c */ -/* Auto-generated from source file: eas_main.c */ -{ 0xe624f4d9, 0x00000005, "eas_main.c[106]: Play length: %d.%03d (secs)\n" }, diff --git a/arm-hybrid-22k/host_src/eas_host.h b/arm-hybrid-22k/host_src/eas_host.h deleted file mode 100644 index 0db0e30..0000000 --- a/arm-hybrid-22k/host_src/eas_host.h +++ /dev/null @@ -1,83 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_host.h - * - * Contents and purpose: - * This header defines the host wrapper functions for stdio, stdlib, etc. - * The host application must provide an abstraction layer for these functions - * to support certain features, such as SMAF and SMF-1 conversion. - * - * DO NOT MODIFY THIS FILE! - * - * Copyright 2005 Sonic Network Inc. - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -// sentinel -#ifndef _EAS_HOST_H -#define _EAS_HOST_H - -#include "eas_types.h" - -/* for C++ linkage */ -#ifdef __cplusplus -extern "C" { -#endif - -/* initialization and shutdown routines */ -extern EAS_RESULT EAS_HWInit(EAS_HW_DATA_HANDLE *hwInstData); -extern EAS_RESULT EAS_HWShutdown(EAS_HW_DATA_HANDLE hwInstData); - -/* memory functions */ -extern void *EAS_HWMemSet(void *s, int c, EAS_I32 n); -extern void *EAS_HWMemCpy(void *s1, const void *s2, EAS_I32 n); -extern EAS_I32 EAS_HWMemCmp(const void *s1, const void *s2, EAS_I32 n); - -/* memory allocation */ -extern void *EAS_HWMalloc(EAS_HW_DATA_HANDLE hwInstData, EAS_I32 size); -extern void EAS_HWFree(EAS_HW_DATA_HANDLE hwInstData, void *p); - -/* file I/O */ -extern EAS_RESULT EAS_HWOpenFile(EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_LOCATOR locator, EAS_FILE_HANDLE *pFile, EAS_FILE_MODE mode); -extern EAS_RESULT EAS_HWReadFile(EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *pBuffer, EAS_I32 n, EAS_I32 *pBytesRead); -extern EAS_RESULT EAS_HWGetByte(EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p); -extern EAS_RESULT EAS_HWGetWord (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p, EAS_BOOL msbFirst); -extern EAS_RESULT EAS_HWGetDWord (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p, EAS_BOOL msbFirst); -extern EAS_RESULT EAS_HWFilePos (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 *pPosition); -extern EAS_RESULT EAS_HWFileSeek (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 position); -extern EAS_RESULT EAS_HWFileSeekOfs (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 position); -extern EAS_RESULT EAS_HWFileLength (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 *pLength); -extern EAS_RESULT EAS_HWDupHandle (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_FILE_HANDLE* pFile); -extern EAS_RESULT EAS_HWCloseFile (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file); - -/* vibrate, LED, and backlight functions */ -extern EAS_RESULT EAS_HWVibrate(EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state); -extern EAS_RESULT EAS_HWLED(EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state); -extern EAS_RESULT EAS_HWBackLight(EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state); - -#ifdef __cplusplus -} /* end extern "C" */ -#endif - - -/* host yield function */ -extern EAS_BOOL EAS_HWYield(EAS_HW_DATA_HANDLE hwInstData); -#endif /* end _EAS_HOST_H */ diff --git a/arm-hybrid-22k/host_src/eas_hostmm.c b/arm-hybrid-22k/host_src/eas_hostmm.c deleted file mode 100644 index 7e58838..0000000 --- a/arm-hybrid-22k/host_src/eas_hostmm.c +++ /dev/null @@ -1,660 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_hostmm.c - * - * Contents and purpose: - * This file contains the host wrapper functions for stdio, stdlib, etc. - * This is a sample version that maps the requested files to an - * allocated memory block and uses in-memory pointers to replace - * file system calls. The file locator (EAS_FILE_LOCATOR) handle passed - * HWOpenFile is the same one that is passed to EAS_OpenFile. If your - * system stores data in fixed locations (such as flash) instead of - * using a file system, you can use the locator handle to point to - * your memory. You will need a way of knowing the length of the - * data stored at that location in order to respond correctly in the - * HW_FileLength function. - * - * Modify this file to suit the needs of your particular system. - * - * EAS_MAX_FILE_HANDLES sets the maximum number of MIDI streams within - * a MIDI type 1 file that can be played. - * - * EAS_HW_FILE is a structure to support the file I/O functions. It - * comprises the base memory pointer, the file read pointer, and - * the dup flag, which when sets, indicates that the file handle has - * been duplicated. If your system uses in-memory resources, you - * can eliminate the duplicate handle logic, and simply copy the - * base memory pointer and file read pointer to the duplicate handle. - * - * Copyright 2005 Sonic Network Inc. - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 795 $ - * $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifdef _lint -#include "lint_stdlib.h" -#else -#include -#include -#include -#endif - -#include "eas_host.h" - -/* Only for debugging LED, vibrate, and backlight functions */ -#include "eas_report.h" - -/* this module requires dynamic memory support */ -#ifdef _STATIC_MEMORY -#error "eas_hostmm.c requires the dynamic memory model!\n" -#endif - -#ifndef EAS_MAX_FILE_HANDLES -#define EAS_MAX_FILE_HANDLES 32 -#endif - -/* - * this structure and the related function are here - * to support the ability to create duplicate handles - * and buffering it in memory. If your system uses - * in-memory resources, you can eliminate the calls - * to malloc and free, the dup flag, and simply track - * the file size and read position. - */ -typedef struct eas_hw_file_tag -{ - EAS_I32 fileSize; - EAS_I32 filePos; - EAS_BOOL dup; - EAS_U8 *buffer; -} EAS_HW_FILE; - -typedef struct eas_hw_inst_data_tag -{ - EAS_HW_FILE files[EAS_MAX_FILE_HANDLES]; -} EAS_HW_INST_DATA; - -/*---------------------------------------------------------------------------- - * EAS_HWInit - * - * Initialize host wrapper interface - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_HWInit (EAS_HW_DATA_HANDLE *pHWInstData) -{ - - /* need to track file opens for duplicate handles */ - *pHWInstData = malloc(sizeof(EAS_HW_INST_DATA)); - if (!(*pHWInstData)) - return EAS_ERROR_MALLOC_FAILED; - - EAS_HWMemSet(*pHWInstData, 0, sizeof(EAS_HW_INST_DATA)); - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_HWShutdown - * - * Shut down host wrapper interface - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_HWShutdown (EAS_HW_DATA_HANDLE hwInstData) -{ - - free(hwInstData); - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWMalloc - * - * Allocates dynamic memory - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, hwInstData) hwInstData available for customer use */ -void *EAS_HWMalloc (EAS_HW_DATA_HANDLE hwInstData, EAS_I32 size) -{ - return malloc((size_t) size); -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWFree - * - * Frees dynamic memory - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, hwInstData) hwInstData available for customer use */ -void EAS_HWFree (EAS_HW_DATA_HANDLE hwInstData, void *p) -{ - free(p); -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWMemCpy - * - * Copy memory wrapper - * - *---------------------------------------------------------------------------- -*/ -void *EAS_HWMemCpy (void *dest, const void *src, EAS_I32 amount) -{ - return memcpy(dest, src, (size_t) amount); -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWMemSet - * - * Set memory wrapper - * - *---------------------------------------------------------------------------- -*/ -void *EAS_HWMemSet (void *dest, int val, EAS_I32 amount) -{ - return memset(dest, val, (size_t) amount); -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWMemCmp - * - * Compare memory wrapper - * - *---------------------------------------------------------------------------- -*/ -EAS_I32 EAS_HWMemCmp (const void *s1, const void *s2, EAS_I32 amount) -{ - return (EAS_I32) memcmp(s1, s2, (size_t) amount); -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWOpenFile - * - * Open a file for read or write - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_HWOpenFile (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_LOCATOR locator, EAS_FILE_HANDLE *pFile, EAS_FILE_MODE mode) -{ - EAS_HW_FILE *file; - FILE *ioFile; - int i, temp; - - /* set return value to NULL */ - *pFile = NULL; - - /* only support read mode at this time */ - if (mode != EAS_FILE_READ) - return EAS_ERROR_INVALID_FILE_MODE; - - /* find an empty entry in the file table */ - file = hwInstData->files; - for (i = 0; i < EAS_MAX_FILE_HANDLES; i++) - { - /* is this slot being used? */ - if (file->buffer == NULL) - { - /* open the file */ - if ((ioFile = fopen(locator,"rb")) == NULL) - return EAS_ERROR_FILE_OPEN_FAILED; - - /* determine the file size */ - if (fseek(ioFile, 0L, SEEK_END) != 0) - return EAS_ERROR_FILE_LENGTH; - if ((file->fileSize = ftell(ioFile)) == -1L) - return EAS_ERROR_FILE_LENGTH; - if (fseek(ioFile, 0L, SEEK_SET) != 0) - return EAS_ERROR_FILE_LENGTH; - - /* allocate a buffer */ - file->buffer = EAS_HWMalloc(hwInstData, file->fileSize); - if (file->buffer == NULL) - { - fclose(ioFile); - return EAS_ERROR_MALLOC_FAILED; - } - - /* read the file into memory */ - temp = (int) fread(file->buffer, (size_t) file->fileSize, 1, ioFile); - - /* close the file - don't need it any more */ - fclose(ioFile); - - /* check for error reading file */ - if (temp != 1) - return EAS_ERROR_FILE_READ_FAILED; - - /* initialize some values */ - file->filePos = 0; - file->dup = EAS_FALSE; - - *pFile = file; - return EAS_SUCCESS; - } - file++; - } - - /* too many open files */ - return EAS_ERROR_MAX_FILES_OPEN; -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWReadFile - * - * Read data from a file - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, hwInstData) hwInstData available for customer use */ -EAS_RESULT EAS_HWReadFile (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *pBuffer, EAS_I32 n, EAS_I32 *pBytesRead) -{ - EAS_I32 count; - - /* make sure we have a valid handle */ - if (file->buffer == NULL) - return EAS_ERROR_INVALID_HANDLE; - - /* calculate the bytes to read */ - count = file->fileSize - file->filePos; - if (n < count) - count = n; - - /* copy the data to the requested location, and advance the pointer */ - if (count) - EAS_HWMemCpy(pBuffer, &file->buffer[file->filePos], count); - file->filePos += count; - *pBytesRead = count; - - /* were n bytes read? */ - if (count!= n) - return EAS_EOF; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWGetByte - * - * Read a byte from a file - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, hwInstData) hwInstData available for customer use */ -EAS_RESULT EAS_HWGetByte (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p) -{ - - /* make sure we have a valid handle */ - if (file->buffer == NULL) - return EAS_ERROR_INVALID_HANDLE; - - /* check for end of file */ - if (file->filePos >= file->fileSize) - { - *((EAS_U8*) p) = 0; - return EAS_EOF; - } - - /* get a character from the buffer */ - *((EAS_U8*) p) = file->buffer[file->filePos++]; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWGetWord - * - * Returns the current location in the file - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, hwInstData) hwInstData available for customer use */ -EAS_RESULT EAS_HWGetWord (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p, EAS_BOOL msbFirst) -{ - EAS_RESULT result; - EAS_U8 c1, c2; - - /* read 2 bytes from the file */ - if ((result = EAS_HWGetByte(hwInstData, file, &c1)) != EAS_SUCCESS) - return result; - if ((result = EAS_HWGetByte(hwInstData, file, &c2)) != EAS_SUCCESS) - return result; - - /* order them as requested */ - if (msbFirst) - *((EAS_U16*) p) = ((EAS_U16) c1 << 8) | c2; - else - *((EAS_U16*) p) = ((EAS_U16) c2 << 8) | c1; - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWGetDWord - * - * Returns the current location in the file - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, hwInstData) hwInstData available for customer use */ -EAS_RESULT EAS_HWGetDWord (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p, EAS_BOOL msbFirst) -{ - EAS_RESULT result; - EAS_U8 c1, c2,c3,c4; - - /* read 4 bytes from the file */ - if ((result = EAS_HWGetByte(hwInstData, file, &c1)) != EAS_SUCCESS) - return result; - if ((result = EAS_HWGetByte(hwInstData, file, &c2)) != EAS_SUCCESS) - return result; - if ((result = EAS_HWGetByte(hwInstData, file, &c3)) != EAS_SUCCESS) - return result; - if ((result = EAS_HWGetByte(hwInstData, file, &c4)) != EAS_SUCCESS) - return result; - - /* order them as requested */ - if (msbFirst) - *((EAS_U32*) p) = ((EAS_U32) c1 << 24) | ((EAS_U32) c2 << 16) | ((EAS_U32) c3 << 8) | c4; - else - *((EAS_U32*) p)= ((EAS_U32) c4 << 24) | ((EAS_U32) c3 << 16) | ((EAS_U32) c2 << 8) | c1; - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWFilePos - * - * Returns the current location in the file - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, hwInstData) hwInstData available for customer use */ -EAS_RESULT EAS_HWFilePos (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 *pPosition) -{ - - /* make sure we have a valid handle */ - if (file->buffer == NULL) - return EAS_ERROR_INVALID_HANDLE; - - *pPosition = file->filePos; - return EAS_SUCCESS; -} /* end EAS_HWFilePos */ - -/*---------------------------------------------------------------------------- - * - * EAS_HWFileSeek - * - * Seek to a specific location in the file - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, hwInstData) hwInstData available for customer use */ -EAS_RESULT EAS_HWFileSeek (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 position) -{ - - /* make sure we have a valid handle */ - if (file->buffer == NULL) - return EAS_ERROR_INVALID_HANDLE; - - /* validate new position */ - if ((position < 0) || (position > file->fileSize)) - return EAS_ERROR_FILE_SEEK; - - /* save new position */ - file->filePos = position; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWFileSeekOfs - * - * Seek forward or back relative to the current position - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, hwInstData) hwInstData available for customer use */ -EAS_RESULT EAS_HWFileSeekOfs (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 position) -{ - - /* make sure we have a valid handle */ - if (file->buffer == NULL) - return EAS_ERROR_INVALID_HANDLE; - - /* determine the file position */ - position += file->filePos; - if ((position < 0) || (position > file->fileSize)) - return EAS_ERROR_FILE_SEEK; - - /* save new position */ - file->filePos = position; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWFileLength - * - * Return the file length - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, hwInstData) hwInstData available for customer use */ -EAS_RESULT EAS_HWFileLength (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 *pLength) -{ - - /* make sure we have a valid handle */ - if (file->buffer == NULL) - return EAS_ERROR_INVALID_HANDLE; - - *pLength = file->fileSize; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWDupHandle - * - * Duplicate a file handle - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_HWDupHandle (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_FILE_HANDLE *pDupFile) -{ - EAS_HW_FILE *dupFile; - int i; - - /* make sure we have a valid handle */ - if (file->buffer == NULL) - return EAS_ERROR_INVALID_HANDLE; - - /* find an empty entry in the file table */ - dupFile = hwInstData->files; - for (i = 0; i < EAS_MAX_FILE_HANDLES; i++) - { - /* is this slot being used? */ - if (dupFile->buffer == NULL) - { - - /* copy info from the handle to be duplicated */ - dupFile->filePos = file->filePos; - dupFile->fileSize = file->fileSize; - dupFile->buffer = file->buffer; - - /* set the duplicate handle flag */ - dupFile->dup = file->dup = EAS_TRUE; - - *pDupFile = dupFile; - return EAS_SUCCESS; - } - dupFile++; - } - - /* too many open files */ - return EAS_ERROR_MAX_FILES_OPEN; -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWClose - * - * Wrapper for fclose function - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_HWCloseFile (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file1) -{ - EAS_HW_FILE *file2,*dupFile; - int i; - - - /* make sure we have a valid handle */ - if (file1->buffer == NULL) - return EAS_ERROR_INVALID_HANDLE; - - /* check for duplicate handle */ - if (file1->dup) - { - dupFile = NULL; - file2 = hwInstData->files; - for (i = 0; i < EAS_MAX_FILE_HANDLES; i++) - { - /* check for duplicate */ - if ((file1 != file2) && (file2->buffer == file1->buffer)) - { - /* is there more than one duplicate? */ - if (dupFile != NULL) - { - /* clear this entry and return */ - file1->buffer = NULL; - return EAS_SUCCESS; - } - - /* this is the first duplicate found */ - else - dupFile = file2; - } - file2++; - } - - /* there is only one duplicate, clear the dup flag */ - if (dupFile) - dupFile->dup = EAS_FALSE; - else - /* if we get here, there's a serious problem */ - return EAS_ERROR_HANDLE_INTEGRITY; - - /* clear this entry and return */ - file1->buffer = NULL; - return EAS_SUCCESS; - } - - /* no duplicates -free the buffer */ - EAS_HWFree(hwInstData, file1->buffer); - - /* clear this entry and return */ - file1->buffer = NULL; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWVibrate - * - * Turn on/off vibrate function - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, hwInstData) hwInstData available for customer use */ -EAS_RESULT EAS_HWVibrate (EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state) -{ - EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x1a54b6e8, 0x00000001 , state); - return EAS_SUCCESS; -} /* end EAS_HWVibrate */ - -/*---------------------------------------------------------------------------- - * - * EAS_HWLED - * - * Turn on/off LED - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, hwInstData) hwInstData available for customer use */ -EAS_RESULT EAS_HWLED (EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state) -{ - EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x1a54b6e8, 0x00000002 , state); - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWBackLight - * - * Turn on/off backlight - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, hwInstData) hwInstData available for customer use */ -EAS_RESULT EAS_HWBackLight (EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state) -{ - EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x1a54b6e8, 0x00000003 , state); - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * - * EAS_HWYield - * - * This function is called periodically by the EAS library to give the - * host an opportunity to allow other tasks to run. There are two ways to - * use this call: - * - * If you have a multi-tasking OS, you can call the yield function in the - * OS to allow other tasks to run. In this case, return EAS_FALSE to tell - * the EAS library to continue processing when control returns from this - * function. - * - * If tasks run in a single thread by sequential function calls (sometimes - * call a "commutator loop"), return EAS_TRUE to cause the EAS Library to - * return to the caller. Be sure to check the number of bytes rendered - * before passing the audio buffer to the codec - it may not be filled. - * The next call to EAS_Render will continue processing until the buffer - * has been filled. - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, hwInstData) hwInstData available for customer use */ -EAS_BOOL EAS_HWYield (EAS_HW_DATA_HANDLE hwInstData) -{ - /* put your code here */ - return EAS_FALSE; -} - diff --git a/arm-hybrid-22k/host_src/eas_main.c b/arm-hybrid-22k/host_src/eas_main.c deleted file mode 100644 index 809a132..0000000 --- a/arm-hybrid-22k/host_src/eas_main.c +++ /dev/null @@ -1,461 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_main.c - * - * Contents and purpose: - * The entry point and high-level functions for the EAS Synthesizer test - * harness. - * - * Copyright Sonic Network Inc. 2004 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 775 $ - * $Date: 2007-07-20 10:11:11 -0700 (Fri, 20 Jul 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifdef _lint -#include "lint_stdlib.h" -#else -#include -#include -#include -#include -#endif - -#include "eas.h" -#include "eas_wave.h" -#include "eas_report.h" - -/* determines how many EAS buffers to fill a host buffer */ -#define NUM_BUFFERS 8 - -/* default file to play if no filename is specified on the command line */ -static const char defaultTestFile[] = "test.mid"; - -EAS_I32 polyphony; - -/* prototypes for helper functions */ -static void StrCopy(char *dest, const char *src, EAS_I32 size); -static EAS_BOOL ChangeFileExt(char *str, const char *ext, EAS_I32 size); -static EAS_RESULT PlayFile (EAS_DATA_HANDLE easData, const char* filename, const char* outputFile, const S_EAS_LIB_CONFIG *pLibConfig, void *buffer, EAS_I32 bufferSize); -static EAS_BOOL EASLibraryCheck (const S_EAS_LIB_CONFIG *pLibConfig); - -/* main is defined after playfile to avoid the need for two passes through lint */ - -/*---------------------------------------------------------------------------- - * PlayFile() - *---------------------------------------------------------------------------- - * Purpose: - * This function plays the file requested by filename - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ - -static EAS_RESULT PlayFile (EAS_DATA_HANDLE easData, const char* filename, const char* outputFile, const S_EAS_LIB_CONFIG *pLibConfig, void *buffer, EAS_I32 bufferSize) -{ - EAS_HANDLE handle; - EAS_RESULT result, reportResult; - EAS_I32 count; - EAS_STATE state; - EAS_I32 playTime; - char waveFilename[256]; - WAVE_FILE *wFile; - EAS_INT i; - EAS_PCM *p; - - /* determine the name of the output file */ - wFile = NULL; - if (outputFile == NULL) - { - StrCopy(waveFilename, filename, sizeof(waveFilename)); - if (!ChangeFileExt(waveFilename, "wav", sizeof(waveFilename))) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error in output filename %s\n", waveFilename); */ } - return EAS_FAILURE; - } - outputFile = waveFilename; - } - - /* call EAS library to open file */ - if ((reportResult = EAS_OpenFile(easData, filename, &handle)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_OpenFile returned %ld\n", reportResult); */ } - return reportResult; - } - - /* prepare to play the file */ - if ((result = EAS_Prepare(easData, handle)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_Prepare returned %ld\n", result); */ } - reportResult = result; - } - - /* get play length */ - if ((result = EAS_ParseMetaData(easData, handle, &playTime)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_ParseMetaData returned %ld\n", result); */ } - return result; - } - EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0xe624f4d9, 0x00000005 , playTime / 1000, playTime % 1000); - - if (reportResult == EAS_SUCCESS) - { - /* create the output file */ - wFile = WaveFileCreate(outputFile, pLibConfig->numChannels, pLibConfig->sampleRate, sizeof(EAS_PCM) * 8); - if (!wFile) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Unable to create output file %s\n", waveFilename); */ } - reportResult = EAS_FAILURE; - } - } - - /* rendering loop */ - while (reportResult == EAS_SUCCESS) - { - - /* we may render several buffers here to fill one host buffer */ - for (i = 0, p = buffer; i < NUM_BUFFERS; i++, p+= pLibConfig->mixBufferSize * pLibConfig->numChannels) - { - - /* get the current time */ - if ((result = EAS_GetLocation(easData, handle, &playTime)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_GetLocation returned %d\n",result); */ } - if (reportResult == EAS_SUCCESS) - reportResult = result; - break; - } - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Parser time: %d.%03d\n", playTime / 1000, playTime % 1000); */ } - - /* render a buffer of audio */ - if ((result = EAS_Render(easData, p, pLibConfig->mixBufferSize, &count)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_Render returned %d\n",result); */ } - if (reportResult == EAS_SUCCESS) - reportResult = result; - } - } - - if (result == EAS_SUCCESS) - { - /* write it to the wave file */ - if (WaveFileWrite(wFile, buffer, bufferSize) != bufferSize) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "WaveFileWrite failed\n"); */ } - reportResult = EAS_FAILURE; - } - } - - if (reportResult == EAS_SUCCESS) - { - /* check stream state */ - if ((result = EAS_State(easData, handle, &state)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_State returned %d\n", result); */ } - reportResult = result; - } - - /* is playback complete */ - if ((state == EAS_STATE_STOPPED) || (state == EAS_STATE_ERROR)) - break; - } - } - - /* close the output file */ - if (wFile) - { - if (!WaveFileClose(wFile)) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error closing wave file %s\n", waveFilename); */ } - if (reportResult == EAS_SUCCESS) - result = EAS_FAILURE; - } - } - - /* close the input file */ - if ((result = EAS_CloseFile(easData,handle)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_Close returned %ld\n", result); */ } - if (reportResult == EAS_SUCCESS) - result = EAS_FAILURE; - } - - return reportResult; -} /* end PlayFile */ - -/*---------------------------------------------------------------------------- - * main() - *---------------------------------------------------------------------------- - * Purpose: The entry point for the EAS sample application - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -int main( int argc, char **argv ) -{ - EAS_DATA_HANDLE easData; - const S_EAS_LIB_CONFIG *pLibConfig; - void *buffer; - EAS_RESULT result, playResult; - EAS_I32 bufferSize; - int i; - int temp; - FILE *debugFile; - char *outputFile = NULL; - - /* set the error reporting level */ - EAS_SetDebugLevel(_EAS_SEVERITY_INFO); - debugFile = NULL; - - /* process command-line arguments */ - for (i = 1; i < argc; i++) - { - /* check for switch */ - if (argv[i][0] == '-') - { - switch (argv[i][1]) - { - case 'd': - temp = argv[i][2]; - if ((temp >= '0') || (temp <= '9')) - EAS_SetDebugLevel(temp); - else - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Invalid debug level %d\n", temp); */ } - break; - case 'f': - if ((debugFile = fopen(&argv[i][2],"w")) == NULL) - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Unable to create debug file %s\n", &argv[i][2]); */ } - else - EAS_SetDebugFile(debugFile, EAS_TRUE); - break; - case 'o': - outputFile = &argv[i][2]; - break; - case 'p': - polyphony = atoi(&argv[i][2]); - if (polyphony < 1) - polyphony = 1; - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "Polyphony set to %d\n", polyphony); */ } - break; - default: - break; - } - continue; - } - } - - /* assume success */ - playResult = EAS_SUCCESS; - - /* get the library configuration */ - pLibConfig = EAS_Config(); - if (!EASLibraryCheck(pLibConfig)) - return -1; - if (polyphony > pLibConfig->maxVoices) - polyphony = pLibConfig->maxVoices; - - /* calculate buffer size */ - bufferSize = pLibConfig->mixBufferSize * pLibConfig->numChannels * (EAS_I32)sizeof(EAS_PCM) * NUM_BUFFERS; - - /* allocate output buffer memory */ - buffer = malloc((EAS_U32)bufferSize); - if (!buffer) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Error allocating memory for audio buffer\n"); */ } - return EAS_FAILURE; - } - - /* initialize the EAS library */ - polyphony = pLibConfig->maxVoices; - if ((result = EAS_Init(&easData)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "EAS_Init returned %ld - aborting!\n", result); */ } - free(buffer); - return result; - } - - /* - * Some debugging environments don't allow for passed parameters. - * In this case, just play the default MIDI file "test.mid" - */ - if (argc < 2) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "Playing '%s'\n", defaultTestFile); */ } - if ((playResult = PlayFile(easData, defaultTestFile, NULL, pLibConfig, buffer, bufferSize)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %d playing file %s\n", playResult, defaultTestFile); */ } - } - } - /* iterate through the list of files to be played */ - else - { - for (i = 1; i < argc; i++) - { - /* check for switch */ - if (argv[i][0] != '-') - { - - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "Playing '%s'\n", argv[i]); */ } - if ((playResult = PlayFile(easData, argv[i], outputFile, pLibConfig, buffer, bufferSize)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %d playing file %s\n", playResult, argv[i]); */ } - break; - } - } - } - } - - /* shutdown the EAS library */ - if ((result = EAS_Shutdown(easData)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "EAS_Shutdown returned %ld\n", result); */ } - } - - /* free the output buffer */ - free(buffer); - - /* close the debug file */ - if (debugFile) - fclose(debugFile); - - /* play errors take precedence over shutdown errors */ - if (playResult != EAS_SUCCESS) - return playResult; - return result; -} /* end main */ - -/*---------------------------------------------------------------------------- - * StrCopy() - *---------------------------------------------------------------------------- - * Purpose: - * Safe string copy - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -static void StrCopy(char *dest, const char *src, EAS_I32 size) -{ - int len; - - strncpy(dest, src, (size_t) size-1); - len = (int) strlen(src); - if (len < size) - dest[len] = 0; -} /* end StrCopy */ - -/*---------------------------------------------------------------------------- - * ChangeFileExt() - *---------------------------------------------------------------------------- - * Purpose: - * Changes the file extension of a filename - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL ChangeFileExt(char *str, const char *ext, EAS_I32 size) -{ - char *p; - - /* find the extension, if any */ - p = strrchr(str,'.'); - if (!p) - { - if ((EAS_I32)(strlen(str) + 5) > size) - return EAS_FALSE; - strcat(str,"."); - strcat(str,ext); - return EAS_TRUE; - } - - /* make sure there's room for the extension */ - p++; - *p = 0; - if ((EAS_I32)(strlen(str) + 4) > size) - return EAS_FALSE; - strcat(str,ext); - return EAS_TRUE; -} /* end ChangeFileExt */ - -/*---------------------------------------------------------------------------- - * EASLibraryCheck() - *---------------------------------------------------------------------------- - * Purpose: - * Displays the library version and checks it against the header - * file used to build this code. - * - * Inputs: - * pLibConfig - library configuration retrieved from the library - * - * Outputs: - * returns EAS_TRUE if matched - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL EASLibraryCheck (const S_EAS_LIB_CONFIG *pLibConfig) -{ - - /* display the library version */ - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "EAS Library Version %d.%d.%d.%d\n", - pLibConfig->libVersion >> 24, - (pLibConfig->libVersion >> 16) & 0x0f, - (pLibConfig->libVersion >> 8) & 0x0f, - pLibConfig->libVersion & 0x0f); */ } - - /* display some info about the library build */ - if (pLibConfig->checkedVersion) - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tChecked library\n"); */ } - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tMaximum polyphony: %d\n", pLibConfig->maxVoices); */ } - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tNumber of channels: %d\n", pLibConfig->numChannels); */ } - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tSample rate: %d\n", pLibConfig->sampleRate); */ } - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tMix buffer size: %d\n", pLibConfig->mixBufferSize); */ } - if (pLibConfig->filterEnabled) - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tFilter enabled\n"); */ } -#ifndef _WIN32_WCE - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tLibrary Build Timestamp: %s", ctime((time_t*)&pLibConfig->buildTimeStamp)); */ } -#endif - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tLibrary Build ID: %s\n", pLibConfig->buildGUID); */ } - - /* check it against the header file used to build this code */ - /*lint -e{778} constant expression used for display purposes may evaluate to zero */ - if (LIB_VERSION != pLibConfig->libVersion) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Library version does not match header files. EAS Header Version %d.%d.%d.%d\n", - LIB_VERSION >> 24, - (LIB_VERSION >> 16) & 0x0f, - (LIB_VERSION >> 8) & 0x0f, - LIB_VERSION & 0x0f); */ } - return EAS_FALSE; - } - return EAS_TRUE; -} /* end EASLibraryCheck */ - diff --git a/arm-hybrid-22k/host_src/eas_report.c b/arm-hybrid-22k/host_src/eas_report.c deleted file mode 100644 index d4dd22c..0000000 --- a/arm-hybrid-22k/host_src/eas_report.c +++ /dev/null @@ -1,264 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_report.c - * - * Contents and purpose: - * This file contains the debug message handling routines for the EAS library. - * These routines should be modified as needed for your system. - * - * Copyright 2005 Sonic Network Inc. - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 659 $ - * $Date: 2007-04-24 13:36:35 -0700 (Tue, 24 Apr 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifdef _lint -#include "lint_stdlib.h" -#else -#include -#include -#include -#endif - -#include "eas_report.h" - -static int severityLevel = 9999; - -/* debug file */ -static FILE *debugFile = NULL; -int flush = 0; - -#ifndef _NO_DEBUG_PREPROCESSOR - -/* structure should have an #include for each error message header file */ -S_DEBUG_MESSAGES debugMessages[] = -{ -#ifndef UNIFIED_DEBUG_MESSAGES -#include "eas_config_msgs.h" - - -#include "eas_host_msgs.h" -#include "eas_hostmm_msgs.h" -#include "eas_math_msgs.h" -#include "eas_midi_msgs.h" -#include "eas_mixer_msgs.h" -#include "eas_pcm_msgs.h" -#include "eas_public_msgs.h" -#include "eas_smf_msgs.h" -#include "eas_wave_msgs.h" -#include "eas_voicemgt_msgs.h" - -#ifdef _FM_SYNTH -#include "eas_fmsynth_msgs.h" -#include "eas_fmengine_msgs.h" -#endif - -#ifdef _WT_SYNTH -#include "eas_wtsynth_msgs.h" -#include "eas_wtengine_msgs.h" -#endif - -#ifdef _ARM_TEST_MAIN -#include "arm_main_msgs.h" -#endif - -#ifdef _EAS_MAIN -#include "eas_main_msgs.h" -#endif - -#ifdef _EAS_MAIN_IPC -#include "eas_main_ipc_msgs.h" -#endif - -#ifdef _METRICS_ENABLED -#include "eas_perf_msgs.h" -#endif - -#ifdef _COMPRESSOR_ENABLED -#include "eas_compressor_msgs.h" -#endif - -#ifdef _ENHANCER_ENABLED -#include "eas_enhancer_msgs.h" -#endif - -#ifdef _WOW_ENABLED -#include "eas_wow_msgs.h" -#endif - -#ifdef _SMAF_PARSER -#include "eas_smaf_msgs.h" -#endif - -#ifdef _OTA_PARSER -#include "eas_ota_msgs.h" -#endif - -#ifdef _IMELODY_PARSER -#include "eas_imelody_msgs.h" -#endif - -#ifdef _WAVE_PARSER -#include "eas_wavefile_msgs.h" -#endif - -#if defined(_CMX_PARSER) || defined(_MFI_PARSER) -#include "eas_cmf_msgs.h" -#endif - -#if defined(_CMX_PARSER) || defined(_MFI_PARSER) || defined(_WAVE_PARSER) -#include "eas_imaadpcm_msgs.h" -#endif - -#else -#include "eas_debugmsgs.h" -#endif - -/* denotes end of error messages */ -{ 0,0,0 } -}; - -/*---------------------------------------------------------------------------- - * EAS_ReportEx() - * - * This is the error message handler. The default handler outputs error - * messages to stdout. Modify this as needed for your system. - *---------------------------------------------------------------------------- -*/ -void EAS_ReportEx (int severity, unsigned long hashCode, int serialNum, ...) -{ - va_list vargs; - int i; - - /* check severity level */ - if (severity > severityLevel) - return; - - /* find the error message and output to stdout */ - /*lint -e{661} we check for NULL pointer - no fence post error here */ - for (i = 0; debugMessages[i].m_pDebugMsg; i++) - { - if ((debugMessages[i].m_nHashCode == hashCode) && - (debugMessages[i].m_nSerialNum == serialNum)) - { - /*lint -e{826} */ - va_start(vargs, serialNum); - if (debugFile) - { - vfprintf(debugFile, debugMessages[i].m_pDebugMsg, vargs); - if (flush) - fflush(debugFile); - } - else - { - vprintf(debugMessages[i].m_pDebugMsg, vargs); - } - va_end(vargs); - return; - } - } - printf("Unrecognized error: Severity=%d; HashCode=%lu; SerialNum=%d\n", severity, hashCode, serialNum); -} /* end EAS_ReportEx */ - -#else -/*---------------------------------------------------------------------------- - * EAS_Report() - * - * This is the error message handler. The default handler outputs error - * messages to stdout. Modify this as needed for your system. - *---------------------------------------------------------------------------- -*/ -void EAS_Report (int severity, const char *fmt, ...) -{ - va_list vargs; - - /* check severity level */ - if (severity > severityLevel) - return; - - /*lint -e{826} */ - va_start(vargs, fmt); - if (debugFile) - { - vfprintf(debugFile, fmt, vargs); - if (flush) - fflush(debugFile); - } - else - { - vprintf(fmt, vargs); - } - va_end(vargs); -} /* end EAS_Report */ - -/*---------------------------------------------------------------------------- - * EAS_ReportX() - * - * This is the error message handler. The default handler outputs error - * messages to stdout. Modify this as needed for your system. - *---------------------------------------------------------------------------- -*/ -void EAS_ReportX (int severity, const char *fmt, ...) -{ - va_list vargs; - - /* check severity level */ - if (severity > severityLevel) - return; - - /*lint -e{826} */ - va_start(vargs, fmt); - if (debugFile) - { - vfprintf(debugFile, fmt, vargs); - if (flush) - fflush(debugFile); - } - else - { - vprintf(fmt, vargs); - } - va_end(vargs); -} /* end EAS_ReportX */ -#endif - -/*---------------------------------------------------------------------------- - * EAS_SetDebugLevel() - * - * Sets the level for debug message output - *---------------------------------------------------------------------------- -*/ - -void EAS_SetDebugLevel (int severity) -{ - severityLevel = severity; -} /* end EAS_SetDebugLevel */ - -/*---------------------------------------------------------------------------- - * EAS_SetDebugFile() - * - * Redirect debugger output to the specified file. - *---------------------------------------------------------------------------- -*/ -void EAS_SetDebugFile (void *file, int flushAfterWrite) -{ - debugFile = (FILE*) file; - flush = flushAfterWrite; -} /* end EAS_SetDebugFile */ - diff --git a/arm-hybrid-22k/host_src/eas_report.h b/arm-hybrid-22k/host_src/eas_report.h deleted file mode 100644 index 9d7c8e8..0000000 --- a/arm-hybrid-22k/host_src/eas_report.h +++ /dev/null @@ -1,77 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_report.h - * - * Contents and purpose: - * This file contains the debug message handling routines for the EAS library. - * These routines should be modified as needed for your system. - * - * DO NOT MODIFY THIS FILE! - * - * Copyright 2005 Sonic Network Inc. - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -/* sentinel */ -#ifndef _EAS_REPORT_H -#define _EAS_REPORT_H - -#define _EAS_SEVERITY_NOFILTER 0 -#define _EAS_SEVERITY_FATAL 1 -#define _EAS_SEVERITY_ERROR 2 -#define _EAS_SEVERITY_WARNING 3 -#define _EAS_SEVERITY_INFO 4 -#define _EAS_SEVERITY_DETAIL 5 - -/* for C++ linkage */ -#ifdef __cplusplus -extern "C" { -#endif - -#ifndef _NO_DEBUG_PREPROCESSOR - -/* structure for included debug message header files */ -typedef struct -{ - unsigned long m_nHashCode; - int m_nSerialNum; - char *m_pDebugMsg; -} S_DEBUG_MESSAGES; - -/* debug message handling prototypes */ -extern void EAS_ReportEx (int severity, unsigned long hashCode, int serialNum, ...); - -#else - -/* these prototypes are used if the debug preprocessor is not used */ -extern void EAS_Report (int severity, const char* fmt, ...); -extern void EAS_ReportX (int severity, const char* fmt, ...); - -#endif - -extern void EAS_SetDebugLevel (int severity); -extern void EAS_SetDebugFile (void *file, int flushAfterWrite); - -#ifdef __cplusplus -} /* end extern "C" */ -#endif - -#endif diff --git a/arm-hybrid-22k/host_src/eas_reverb.h b/arm-hybrid-22k/host_src/eas_reverb.h deleted file mode 100644 index a2535fb..0000000 --- a/arm-hybrid-22k/host_src/eas_reverb.h +++ /dev/null @@ -1,54 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_reverb.h - * - * Contents and purpose: - * Contains parameter enumerations for the Reverb effect - * - * - * Copyright Sonic Network Inc. 2006 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 300 $ - * $Date: 2006-09-11 17:37:20 -0700 (Mon, 11 Sep 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_REVERB_H -#define _EAS_REVERB_H - - -/* enumerated parameter settings for Reverb effect */ -typedef enum -{ - EAS_PARAM_REVERB_BYPASS, - EAS_PARAM_REVERB_PRESET, - EAS_PARAM_REVERB_WET, - EAS_PARAM_REVERB_DRY -} E_REVERB_PARAMS; - - -typedef enum -{ - EAS_PARAM_REVERB_LARGE_HALL, - EAS_PARAM_REVERB_HALL, - EAS_PARAM_REVERB_CHAMBER, - EAS_PARAM_REVERB_ROOM, -} E_REVERB_PRESETS; - - -#endif /* _REVERB_H */ diff --git a/arm-hybrid-22k/host_src/eas_types.h b/arm-hybrid-22k/host_src/eas_types.h deleted file mode 100644 index f0293ef..0000000 --- a/arm-hybrid-22k/host_src/eas_types.h +++ /dev/null @@ -1,268 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_types.h - * - * Contents and purpose: - * The public interface header for the EAS synthesizer. - * - * This header only contains declarations that are specific - * to this implementation. - * - * DO NOT MODIFY THIS FILE! - * - * Copyright Sonic Network Inc. 2004 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 726 $ - * $Date: 2007-06-14 23:10:46 -0700 (Thu, 14 Jun 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_TYPES_H -#define _EAS_TYPES_H - -/* EAS_RESULT return codes */ -typedef long EAS_RESULT; -#define EAS_SUCCESS 0 -#define EAS_FAILURE -1 -#define EAS_ERROR_INVALID_MODULE -2 -#define EAS_ERROR_MALLOC_FAILED -3 -#define EAS_ERROR_FILE_POS -4 -#define EAS_ERROR_INVALID_FILE_MODE -5 -#define EAS_ERROR_FILE_SEEK -6 -#define EAS_ERROR_FILE_LENGTH -7 -#define EAS_ERROR_NOT_IMPLEMENTED -8 -#define EAS_ERROR_CLOSE_FAILED -9 -#define EAS_ERROR_FILE_OPEN_FAILED -10 -#define EAS_ERROR_INVALID_HANDLE -11 -#define EAS_ERROR_NO_MIX_BUFFER -12 -#define EAS_ERROR_PARAMETER_RANGE -13 -#define EAS_ERROR_MAX_FILES_OPEN -14 -#define EAS_ERROR_UNRECOGNIZED_FORMAT -15 -#define EAS_BUFFER_SIZE_MISMATCH -16 -#define EAS_ERROR_FILE_FORMAT -17 -#define EAS_ERROR_SMF_NOT_INITIALIZED -18 -#define EAS_ERROR_LOCATE_BEYOND_END -19 -#define EAS_ERROR_INVALID_PCM_TYPE -20 -#define EAS_ERROR_MAX_PCM_STREAMS -21 -#define EAS_ERROR_NO_VOICE_ALLOCATED -22 -#define EAS_ERROR_INVALID_CHANNEL -23 -#define EAS_ERROR_ALREADY_STOPPED -24 -#define EAS_ERROR_FILE_READ_FAILED -25 -#define EAS_ERROR_HANDLE_INTEGRITY -26 -#define EAS_ERROR_MAX_STREAMS_OPEN -27 -#define EAS_ERROR_INVALID_PARAMETER -28 -#define EAS_ERROR_FEATURE_NOT_AVAILABLE -29 -#define EAS_ERROR_SOUND_LIBRARY -30 -#define EAS_ERROR_NOT_VALID_IN_THIS_STATE -31 -#define EAS_ERROR_NO_VIRTUAL_SYNTHESIZER -32 -#define EAS_ERROR_FILE_ALREADY_OPEN -33 -#define EAS_ERROR_FILE_ALREADY_CLOSED -34 -#define EAS_ERROR_INCOMPATIBLE_VERSION -35 -#define EAS_ERROR_QUEUE_IS_FULL -36 -#define EAS_ERROR_QUEUE_IS_EMPTY -37 -#define EAS_ERROR_FEATURE_ALREADY_ACTIVE -38 - -/* special return codes */ -#define EAS_EOF 3 -#define EAS_STREAM_BUFFERING 4 -#define EAS_BUFFER_FULL 5 - -/* EAS_STATE return codes */ -typedef long EAS_STATE; -typedef enum -{ - EAS_STATE_READY = 0, - EAS_STATE_PLAY, - EAS_STATE_STOPPING, - EAS_STATE_PAUSING, - EAS_STATE_STOPPED, - EAS_STATE_PAUSED, - EAS_STATE_OPEN, - EAS_STATE_ERROR, - EAS_STATE_EMPTY -} E_EAS_STATE; - -/* constants */ -#ifndef EAS_CONST -#define EAS_CONST const -#endif - -/* definition for public interface functions */ -#ifndef EAS_PUBLIC -#define EAS_PUBLIC -#endif - -/* boolean values */ -typedef unsigned EAS_BOOL; -typedef unsigned char EAS_BOOL8; - -#define EAS_FALSE 0 -#define EAS_TRUE 1 - -/* scalar variable definitions */ -typedef unsigned char EAS_U8; -typedef signed char EAS_I8; -typedef char EAS_CHAR; - -typedef unsigned short EAS_U16; -typedef short EAS_I16; - -typedef unsigned long EAS_U32; -typedef long EAS_I32; - -typedef unsigned EAS_UINT; -typedef int EAS_INT; -typedef long EAS_LONG; - -/* audio output type */ -typedef short EAS_PCM; - -/* file open modes */ -typedef EAS_I32 EAS_FILE_MODE; -#define EAS_FILE_READ 1 -#define EAS_FILE_WRITE 2 - -/* file locator e.g. filename or memory pointer */ -typedef const void *EAS_FILE_LOCATOR; - -/* handle to stream */ -typedef struct s_eas_stream_tag *EAS_HANDLE; - -/* handle to file */ -typedef struct eas_hw_file_tag *EAS_FILE_HANDLE; - -/* handle for synthesizer data */ -typedef struct s_eas_data_tag *EAS_DATA_HANDLE; - -/* handle to persistent data for host wrapper interface */ -typedef struct eas_hw_inst_data_tag *EAS_HW_DATA_HANDLE; - -/* handle to sound library */ -typedef struct s_eas_sndlib_tag *EAS_SNDLIB_HANDLE; -typedef struct s_eas_dls_tag *EAS_DLSLIB_HANDLE; - -/* pointer to frame buffer - used in split architecture only */ -typedef struct s_eas_frame_buffer_tag *EAS_FRAME_BUFFER_HANDLE; - -/* untyped pointer for instance data */ -typedef void *EAS_VOID_PTR; - -/* inline functions */ -#ifndef EAS_INLINE -#if defined (__XCC__) -#define EAS_INLINE __inline__ -#elif defined (__GNUC__) -#define EAS_INLINE inline static -#else -#define EAS_INLINE __inline -#endif -#endif - -/* define NULL value */ -#ifndef NULL -#define NULL 0 -#endif - -/* metadata types for metadata return codes */ -typedef enum -{ - EAS_METADATA_UNKNOWN = 0, - EAS_METADATA_TITLE, - EAS_METADATA_AUTHOR, - EAS_METADATA_COPYRIGHT, - EAS_METADATA_LYRIC, - EAS_METADATA_TEXT -} E_EAS_METADATA_TYPE; - -/* metadata callback function */ -typedef void (*EAS_METADATA_CBFUNC) (E_EAS_METADATA_TYPE metaDataType, char *metaDataBuf, EAS_VOID_PTR pUserData); - -/* file types for metadata return codes */ -typedef enum -{ - EAS_FILE_UNKNOWN = 0, - EAS_FILE_SMF0, - EAS_FILE_SMF1, - EAS_FILE_SMAF_UNKNOWN, - EAS_FILE_SMAF_MA2, - EAS_FILE_SMAF_MA3, - EAS_FILE_SMAF_MA5, - EAS_FILE_CMX, - EAS_FILE_MFI, - EAS_FILE_OTA, - EAS_FILE_IMELODY, - EAS_FILE_RTTTL, - EAS_FILE_XMF0, - EAS_FILE_XMF1, - EAS_FILE_WAVE_PCM, - EAS_FILE_WAVE_IMA_ADPCM, - EAS_FILE_MMAPI_TONE_CONTROL -} E_EAS_FILE_TYPE; - -/* enumeration for synthesizers */ -typedef enum -{ - EAS_MCU_SYNTH = 0, - EAS_DSP_SYNTH -} E_SYNTHESIZER; - -/* external audio callback program change */ -typedef struct s_ext_audio_prg_chg_tag -{ - EAS_U16 bank; - EAS_U8 program; - EAS_U8 channel; -} S_EXT_AUDIO_PRG_CHG; - -/* external audio callback event */ -typedef struct s_ext_audio_event_tag -{ - EAS_U8 channel; - EAS_U8 note; - EAS_U8 velocity; - EAS_BOOL8 noteOn; -} S_EXT_AUDIO_EVENT; - -typedef struct s_midi_controllers_tag -{ - EAS_U8 modWheel; /* CC1 */ - EAS_U8 volume; /* CC7 */ - EAS_U8 pan; /* CC10 */ - EAS_U8 expression; /* CC11 */ - EAS_U8 channelPressure; /* MIDI channel pressure */ - -#ifdef _REVERB - EAS_U8 reverbSend; /* CC91 */ -#endif - -#ifdef _CHORUS - EAS_U8 chorusSend; /* CC93 */ -#endif -} S_MIDI_CONTROLLERS; - -/* iMode play modes enumeration for EAS_SetPlayMode */ -typedef enum -{ - IMODE_PLAY_ALL = 0, - IMODE_PLAY_PARTIAL -} E_I_MODE_PLAY_MODE; - -typedef EAS_BOOL (*EAS_EXT_PRG_CHG_FUNC) (EAS_VOID_PTR pInstData, S_EXT_AUDIO_PRG_CHG *pPrgChg); -typedef EAS_BOOL (*EAS_EXT_EVENT_FUNC) (EAS_VOID_PTR pInstData, S_EXT_AUDIO_EVENT *pEvent); - -#endif /* #ifndef _EAS_TYPES_H */ diff --git a/arm-hybrid-22k/host_src/eas_wave.c b/arm-hybrid-22k/host_src/eas_wave.c deleted file mode 100644 index 02fed6e..0000000 --- a/arm-hybrid-22k/host_src/eas_wave.c +++ /dev/null @@ -1,423 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_wave.c - * - * Contents and purpose: - * This module contains .WAV file functions for the EAS synthesizer - * test harness. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 658 $ - * $Date: 2007-04-24 13:35:49 -0700 (Tue, 24 Apr 2007) $ - *---------------------------------------------------------------------------- -*/ - -/* lint complaints about most C library headers, so we use our own during lint step */ -#ifdef _lint -#include "lint_stdlib.h" -#else -#include -#include -#endif - -#include "eas_wave.h" - -/* .WAV file format tags */ -const EAS_U32 riffTag = 0x46464952; -const EAS_U32 waveTag = 0x45564157; -const EAS_U32 fmtTag = 0x20746d66; -const EAS_U32 dataTag = 0x61746164; - -#ifdef _BIG_ENDIAN -/*---------------------------------------------------------------------------- - * FlipDWord() - *---------------------------------------------------------------------------- - * Purpose: Endian flip a DWORD for big-endian processors - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -static void FlipDWord (EAS_U32 *pValue) -{ - EAS_U8 *p; - EAS_U32 temp; - - p = (EAS_U8*) pValue; - temp = (((((p[3] << 8) | p[2]) << 8) | p[1]) << 8) | p[0]; - *pValue = temp; -} - -/*---------------------------------------------------------------------------- - * FlipWord() - *---------------------------------------------------------------------------- - * Purpose: Endian flip a WORD for big-endian processors - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -static void FlipWord (EAS_U16 *pValue) -{ - EAS_U8 *p; - EAS_U16 temp; - - p = (EAS_U8*) pValue; - temp = (p[1] << 8) | p[0]; - *pValue = temp; -} - -/*---------------------------------------------------------------------------- - * FlipWaveHeader() - *---------------------------------------------------------------------------- - * Purpose: Endian flip the wave header for big-endian processors - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -static void FlipWaveHeader (WAVE_HEADER *p) -{ - - FlipDWord(&p->nRiffTag); - FlipDWord(&p->nRiffSize); - FlipDWord(&p->nWaveTag); - FlipDWord(&p->nFmtTag); - FlipDWord(&p->nFmtSize); - FlipDWord(&p->nDataTag); - FlipDWord(&p->nDataSize); - FlipWord(&p->fc.wFormatTag); - FlipWord(&p->fc.nChannels); - FlipDWord(&p->fc.nSamplesPerSec); - FlipDWord(&p->fc.nAvgBytesPerSec); - FlipWord(&p->fc.nBlockAlign); - FlipWord(&p->fc.wBitsPerSample); - -} -#endif - -/*---------------------------------------------------------------------------- - * WaveFileCreate() - *---------------------------------------------------------------------------- - * Purpose: Opens a wave file for writing and writes the header - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ - -WAVE_FILE *WaveFileCreate (const char *filename, EAS_I32 nChannels, EAS_I32 nSamplesPerSec, EAS_I32 wBitsPerSample) -{ - WAVE_FILE *wFile; - - /* allocate memory */ - wFile = malloc(sizeof(WAVE_FILE)); - if (!wFile) - return NULL; - wFile->write = EAS_TRUE; - - /* create the file */ - wFile->file = fopen(filename,"wb"); - if (!wFile->file) - { - free(wFile); - return NULL; - } - - /* initialize PCM format .WAV file header */ - wFile->wh.nRiffTag = riffTag; - wFile->wh.nRiffSize = sizeof(WAVE_HEADER) - 8; - wFile->wh.nWaveTag = waveTag; - wFile->wh.nFmtTag = fmtTag; - wFile->wh.nFmtSize = sizeof(FMT_CHUNK); - - /* initalize 'fmt' chunk */ - wFile->wh.fc.wFormatTag = 1; - wFile->wh.fc.nChannels = (EAS_U16) nChannels; - wFile->wh.fc.nSamplesPerSec = (EAS_U32) nSamplesPerSec; - wFile->wh.fc.wBitsPerSample = (EAS_U16) wBitsPerSample; - wFile->wh.fc.nBlockAlign = (EAS_U16) (nChannels * (EAS_U16) (wBitsPerSample / 8)); - wFile->wh.fc.nAvgBytesPerSec = wFile->wh.fc.nBlockAlign * (EAS_U32) nSamplesPerSec; - - /* initialize 'data' chunk */ - wFile->wh.nDataTag = dataTag; - wFile->wh.nDataSize = 0; - -#ifdef _BIG_ENDIAN - FlipWaveHeader(&wFile->wh); -#endif - - /* write the header */ - if (fwrite(&wFile->wh, sizeof(WAVE_HEADER), 1, wFile->file) != 1) - { - fclose(wFile->file); - free(wFile); - return NULL; - } - -#ifdef _BIG_ENDIAN - FlipWaveHeader(&wFile->wh); -#endif - - /* return the file handle */ - return wFile; -} /* end WaveFileCreate */ - -/*---------------------------------------------------------------------------- - * WaveFileWrite() - *---------------------------------------------------------------------------- - * Purpose: Writes data to the wave file - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -EAS_I32 WaveFileWrite (WAVE_FILE *wFile, void *buffer, EAS_I32 n) -{ - EAS_I32 count; - - /* make sure we have an open file */ - if (wFile == NULL) - { - return 0; - } - -#ifdef _BIG_ENDIAN - { - EAS_I32 i; - EAS_U16 *p; - p = buffer; - i = n >> 1; - while (i--) - FlipWord(p++); - } -#endif - - /* write the data */ - count = (EAS_I32) fwrite(buffer, 1, (size_t) n, wFile->file); - - /* add the number of bytes written */ - wFile->wh.nRiffSize += (EAS_U32) count; - wFile->wh.nDataSize += (EAS_U32) count; - - /* return the count of bytes written */ - return count; -} /* end WriteWaveHeader */ - -/*---------------------------------------------------------------------------- - * WaveFileClose() - *---------------------------------------------------------------------------- - * Purpose: Opens a wave file for writing and writes the header - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ - -EAS_BOOL WaveFileClose (WAVE_FILE *wFile) -{ - EAS_I32 count = 1; - - /* return to beginning of file and write the header */ - if (wFile->write) - { - if (fseek(wFile->file, 0L, SEEK_SET) == 0) - { - -#ifdef _BIG_ENDIAN - FlipWaveHeader(&wFile->wh); -#endif - count = (EAS_I32) fwrite(&wFile->wh, sizeof(WAVE_HEADER), 1, wFile->file); -#ifdef _BIG_ENDIAN - FlipWaveHeader(&wFile->wh); -#endif - } - } - - /* close the file */ - if (fclose(wFile->file) != 0) - count = 0; - - /* free the memory */ - free(wFile); - - /* return the file handle */ - return (count == 1 ? EAS_TRUE : EAS_FALSE); -} /* end WaveFileClose */ - -#ifdef _WAVE_FILE_READ -#ifdef _BIG_ENDIAN -#error "WaveFileOpen not currently supported on big-endian processors" -#endif -/*---------------------------------------------------------------------------- - * WaveFileOpen() - *---------------------------------------------------------------------------- - * Purpose: Opens a wave file for reading and reads the header - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ - -WAVE_FILE *WaveFileOpen (const char *filename) -{ - WAVE_FILE *wFile; - struct - { - EAS_U32 tag; - EAS_U32 size; - } chunk; - EAS_U32 tag; - EAS_I32 startChunkPos; - EAS_INT state; - EAS_BOOL done; - - /* allocate memory */ - wFile = malloc(sizeof(WAVE_FILE)); - if (!wFile) - return NULL; - - /* open the file */ - wFile->write = EAS_FALSE; - wFile->file = fopen(filename,"rb"); - if (!wFile->file) - { - free(wFile); - return NULL; - } - - /* make lint happy */ - chunk.tag = chunk.size = 0; - startChunkPos = 0; - - /* read the RIFF tag and file size */ - state = 0; - done = EAS_FALSE; - while (!done) - { - - switch(state) - { - /* read the RIFF tag */ - case 0: - if (fread(&chunk, sizeof(chunk), 1, wFile->file) != 1) - done = EAS_TRUE; - else - { - if (chunk.tag != riffTag) - done = EAS_TRUE; - else - state++; - } - break; - - /* read the WAVE tag */ - case 1: - if (fread(&tag, sizeof(tag), 1, wFile->file) != 1) - done = EAS_TRUE; - else - { - if (tag != waveTag) - done = EAS_TRUE; - else - state++; - } - break; - - /* looking for fmt chunk */ - case 2: - if (fread(&chunk, sizeof(chunk), 1, wFile->file) != 1) - done = EAS_TRUE; - else - { - startChunkPos = ftell(wFile->file); - - /* not fmt tag, skip it */ - if (chunk.tag != fmtTag) - fseek(wFile->file, startChunkPos + (EAS_I32) chunk.size, SEEK_SET); - else - state++; - } - break; - - /* read fmt chunk */ - case 3: - if (fread(&wFile->wh.fc, sizeof(FMT_CHUNK), 1, wFile->file) != 1) - done = EAS_TRUE; - else - { - fseek(wFile->file, startChunkPos + (EAS_I32) chunk.size, SEEK_SET); - state++; - } - break; - - /* looking for data chunk */ - case 4: - if (fread(&chunk, sizeof(chunk), 1, wFile->file) != 1) - done = EAS_TRUE; - else - { - startChunkPos = ftell(wFile->file); - - /* not data tag, skip it */ - if (chunk.tag != dataTag) - fseek(wFile->file, startChunkPos + (EAS_I32) chunk.size, SEEK_SET); - else - { - wFile->dataSize = chunk.size; - state++; - done = EAS_TRUE; - } - } - break; - - default: - done = EAS_TRUE; - break; - } - } - - /* if not final state, an error occurred */ - if (state != 5) - { - fclose(wFile->file); - free(wFile); - return NULL; - } - - /* return the file handle */ - return wFile; -} /* end WaveFileOpen */ -#endif - - - diff --git a/arm-hybrid-22k/host_src/eas_wave.h b/arm-hybrid-22k/host_src/eas_wave.h deleted file mode 100644 index ca388f5..0000000 --- a/arm-hybrid-22k/host_src/eas_wave.h +++ /dev/null @@ -1,74 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_wave.h - * - * Contents and purpose: - * Writes output to a .WAV file - * - * DO NOT MODIFY THIS FILE! - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_types.h" - -/* sentinel */ -#ifndef _EAS_WAVE_H -#define _EAS_WAVE_H - -/* .WAV file format chunk */ -typedef struct { - EAS_U16 wFormatTag; - EAS_U16 nChannels; - EAS_U32 nSamplesPerSec; - EAS_U32 nAvgBytesPerSec; - EAS_U16 nBlockAlign; - EAS_U16 wBitsPerSample; -} FMT_CHUNK; - -/* .WAV file header */ -typedef struct { - EAS_U32 nRiffTag; - EAS_U32 nRiffSize; - EAS_U32 nWaveTag; - EAS_U32 nFmtTag; - EAS_U32 nFmtSize; - FMT_CHUNK fc; - EAS_U32 nDataTag; - EAS_U32 nDataSize; -} WAVE_HEADER; - -typedef struct { - WAVE_HEADER wh; - FILE *file; - EAS_BOOL write; - EAS_U32 dataSize; -} WAVE_FILE; - -WAVE_FILE *WaveFileCreate (const char *filename, EAS_I32 nChannels, EAS_I32 nSamplesPerSec, EAS_I32 wBitsPerSample); -EAS_I32 WaveFileWrite (WAVE_FILE *wFile, void *buffer, EAS_I32 n); -EAS_BOOL WaveFileClose (WAVE_FILE *wFile); -WAVE_FILE *WaveFileOpen (const char *filename); - -#endif /* end #ifndef _EAS_WAVE_H */ - - - diff --git a/arm-hybrid-22k/lib/libarm-hybrid-22k.a b/arm-hybrid-22k/lib/libarm-hybrid-22k.a deleted file mode 100644 index f3999f0..0000000 Binary files a/arm-hybrid-22k/lib/libarm-hybrid-22k.a and /dev/null differ diff --git a/arm-hybrid-22k/lib_src/ARM-E_filter_gnu.s b/arm-hybrid-22k/lib_src/ARM-E_filter_gnu.s deleted file mode 100644 index 871cd7d..0000000 --- a/arm-hybrid-22k/lib_src/ARM-E_filter_gnu.s +++ /dev/null @@ -1,134 +0,0 @@ -@*********************************************************** -@ Function: WT_VoiceFilter -@ Processor: ARM -@ Description: -@ Implements a 2-pole low-pass filter with resonanance -@ -@ Usage: -@ void WT_VoiceFilter( -@ S_FILTER CONTROL *pFilter, -@ S_WT_FRAME *pWTFrame); -@ -@ Copyright 2005 Sonic Network, Inc. -@**************************************************************** -@ Revision Control: -@ $Revision: 496 $ -@ $Date: 2006-12-11 14:33:26 -0800 (Mon, 11 Dec 2006) $ -@**************************************************************** -@ -@ where: -@ S_FILTER_CONTROL *pFilter -@ PASSED IN: r0 -@ -@ S_WT_FRAME *pWTFrame -@ PASSED IN: r1 -@**************************************************************** - - .include "ARM_synth_constants_gnu.inc" - - .arm - .text - - - .global WT_VoiceFilter - - -@ Register usage -@ -------------- -pFilter .req r0 -pWTFrame .req r1 -pBuffer .req r2 -numSamples .req r3 - -z1 .req r4 -z2 .req r5 -b1 .req r6 -b2 .req r7 -K .req r8 - -tmp0 .req r1 @ reuse register -tmp1 .req r9 -tmp2 .req r10 - - -@SaveRegs RLIST {r4-r10, lr} -@RestoreRegs RLIST {r4-r10, pc} - - - .func WT_VoiceFilter -WT_VoiceFilter: - - STMFD sp!, {r4-r10, lr} - -@ -@ Setup passed parameters in their destination registers -@---------------------------------------------------------------- - - LDR pBuffer, [pWTFrame, #m_pAudioBuffer] - LDR numSamples, [pWTFrame, #m_numSamples] - - @load state variables from pFilter structure - LDRSH z1, [pFilter, #m_z1] - LDRSH z2, [pFilter, #m_z2] - - @load coefficients from pWTFrame structure - LDR K, [pWTFrame, #m_k] - LDR b1, [pWTFrame, #m_b1] - LDR b2, [pWTFrame, #m_b2] - - RSB b1, b1, #0 @ b1 = -b1 - RSB b2, b2, #0 @ b2 = -b2 - MOV b2, b2, ASR #1 @ b2 = b2 >> 1 - MOV K, K, ASR #1 @ K = K >> 1 - -@ -@ Start processing -@---------------------------------------------------------------- - - LDRSH tmp0, [pBuffer] @ fetch sample - -FilterLoop: - SMULBB tmp2, z1, b1 @ tmp2 = z1 * -b1 - SMLABB tmp2, z2, b2, tmp2 @ tmp2 = (-b1 * z1) + (-b2 * z2) - - MOV z2, z1 @ delay line - - SMLABB tmp0, tmp0, K, tmp2 @ tmp1 = (K * x[n]) + (-b1 * z1) + (-b2 * z2) - - LDRSH tmp1, [pBuffer, #NEXT_OUTPUT_PCM] @ fetch next sample - - MOV z1, tmp0, ASR #14 @ shift result to low word - STRH z1, [pBuffer], #NEXT_OUTPUT_PCM @ write back to buffer - - SMULBB tmp2, z1, b1 @ tmp2 = z1 * -b1 - - SUBS numSamples, numSamples, #2 @ unroll loop once - - SMLABB tmp2, z2, b2, tmp2 @ tmp2 = (-b1 * z1) + (-b2 * z2) - - SMLABB tmp1, tmp1, K, tmp2 @ tmp1 = (K * x[n]) + (-b1 * z1) + (-b2 * z2) - - MOV z2, z1 @ delay line - - MOV z1, tmp1, ASR #14 @ shift result to low word - - LDRGTSH tmp0, [pBuffer, #NEXT_OUTPUT_PCM] @ fetch next sample - - STRH z1, [pBuffer], #NEXT_OUTPUT_PCM @ write back to buffer - - BGT FilterLoop -@ save z terms -@---------------------------------------------------------------- - - STRH z1, [pFilter, #m_z1] - STRH z2, [pFilter, #m_z2] - -@ Return to calling function -@---------------------------------------------------------------- - - LDMFD sp!,{r4-r10, lr} - BX lr - - .endfunc - .end - diff --git a/arm-hybrid-22k/lib_src/ARM-E_interpolate_loop_gnu.s b/arm-hybrid-22k/lib_src/ARM-E_interpolate_loop_gnu.s deleted file mode 100644 index 847a7f0..0000000 --- a/arm-hybrid-22k/lib_src/ARM-E_interpolate_loop_gnu.s +++ /dev/null @@ -1,131 +0,0 @@ -@*********************************************************** -@ Function: WT_Interpolate -@ Processor: ARM-E -@ Description: the main synthesis function when fetching -@ wavetable samples. -@ C-callable. -@ -@ Usage: -@ void WT_Interpolate( -@ S_WT_VOICE *pWTVoice, -@ S_WT_FRAME *pWTFrame); -@ -@ Copyright Sonic Network Inc. 2004 -@**************************************************************** -@ Revision Control: -@ $Revision: 496 $ -@ $Date: 2006-12-11 14:33:26 -0800 (Mon, 11 Dec 2006) $ -@**************************************************************** -@ -@ where: -@ S_WT_VOICE *pWTVoice -@ PASSED IN: r0 -@ -@ S_WT_FRAME *pWTFrame; -@ PASSED IN: r1 -@**************************************************************** - - .include "ARM_synth_constants_gnu.inc" - - .arm - .text - - .global WT_Interpolate - - -@ Register usage -@ -------------- -pWTVoice .req r0 -pWTFrame .req r1 - -numSamples .req r2 -phaseIncrement .req r3 -pOutputBuffer .req r4 - -tmp0 .req r1 @reuse register -tmp1 .req r5 -tmp2 .req r6 - -pLoopEnd .req r7 -pLoopStart .req r8 - -pPhaseAccum .req r9 -phaseFrac .req r10 -phaseFracMask .req r11 - -@SaveRegs RLIST {r4-r11,lr} -@RestoreRegs RLIST {r4-r11,pc} - - .func WT_Interpolate -WT_Interpolate: - - STMFD sp!,{r4-r11,lr} - -@ -@ Fetch parameters from structures -@---------------------------------------------------------------- - - LDR pOutputBuffer, [pWTFrame, #m_pAudioBuffer] - LDR numSamples, [pWTFrame, #m_numSamples] - - LDR phaseIncrement, [pWTFrame, #m_phaseIncrement] - LDR pPhaseAccum, [pWTVoice, #m_pPhaseAccum] - LDR phaseFrac, [pWTVoice, #m_phaseFrac] - LDR phaseFracMask,=PHASE_FRAC_MASK - - LDR pLoopStart, [pWTVoice, #m_pLoopStart] - LDR pLoopEnd, [pWTVoice, #m_pLoopEnd] - ADD pLoopEnd, pLoopEnd, #1 @ need loop end to equal last sample + 1 - -InterpolationLoop: - SUBS tmp0, pPhaseAccum, pLoopEnd @ check for loop end - ADDGE pPhaseAccum, pLoopStart, tmp0 @ loop back to start - - .ifdef SAMPLES_8_BIT - LDRSB tmp0, [pPhaseAccum] @ tmp0 = x0 - LDRSB tmp1, [pPhaseAccum, #1] @ tmp1 = x1 - .else - LDRSH tmp0, [pPhaseAccum] @ tmp0 = x0 - LDRSH tmp1, [pPhaseAccum, #2] @ tmp1 = x1 - .endif - - ADD tmp2, phaseIncrement, phaseFrac @ increment pointer here to avoid pipeline stall - - SUB tmp1, tmp1, tmp0 @ tmp1 = x1 - x0 - SMULBB tmp1, phaseFrac, tmp1 @ tmp1 = phaseFrac * tmp2 - -@ This section performs a gain adjustment of -12dB for 16-bit samples -@ or +36dB for 8-bit samples. For a high quality synthesizer, the output -@ can be set to full scale, however if the filter is used, it can overflow -@ with certain coefficients and signal sources. In this case, either a -@ saturation operation should take in the filter before scaling back to -@ 16 bits or the signal path should be increased to 18 bits or more. - - .ifdef SAMPLES_8_BIT - MOV tmp0, tmp0, LSL #6 @ boost 8-bit signal by 36dB - .else - MOV tmp0, tmp0, ASR #2 @ reduce 16-bit signal by 12dB - .endif - - ADD tmp1, tmp0, tmp1, ASR #(NUM_EG1_FRAC_BITS-6) @ tmp1 = tmp0 + (tmp1 >> (15-6)) - @ = x0 + f * (x1 - x0) == interpolated result - - STRH tmp1, [pOutputBuffer], #NEXT_OUTPUT_PCM @ *pOutputBuffer++ = interpolated result - -@ carry overflow from fraction to integer portion - ADD pPhaseAccum, pPhaseAccum, tmp2, LSR #(NUM_PHASE_FRAC_BITS - NEXT_INPUT_PCM_SHIFT) - AND phaseFrac, tmp2, phaseFracMask @ nphaseFrac = frac part - - SUBS numSamples, numSamples, #1 - BGT InterpolationLoop - -@ update and store phase - STR pPhaseAccum, [pWTVoice, #m_pPhaseAccum] - STR phaseFrac, [pWTVoice, #m_phaseFrac] - - LDMFD sp!,{r4-r11,lr} - BX lr - - .endfunc - .end - diff --git a/arm-hybrid-22k/lib_src/ARM-E_interpolate_noloop_gnu.s b/arm-hybrid-22k/lib_src/ARM-E_interpolate_noloop_gnu.s deleted file mode 100644 index 6343762..0000000 --- a/arm-hybrid-22k/lib_src/ARM-E_interpolate_noloop_gnu.s +++ /dev/null @@ -1,130 +0,0 @@ -@*********************************************************** -@ Function: WT_InterpolateNoLoop -@ Processor: ARM-E -@ Description: the main synthesis function when fetching -@ wavetable samples. -@ C-callable. -@ -@ Usage: -@ void WT_InterpolateNoLoop( -@ S_WT_VOICE *pWTVoice, -@ S_WT_FRAME *pWTFrame); -@ -@ Copyright Sonic Network Inc. 2004 -@**************************************************************** -@ Revision Control: -@ $Revision: 496 $ -@ $Date: 2006-12-11 14:33:26 -0800 (Mon, 11 Dec 2006) $ -@**************************************************************** -@ -@ where: -@ S_WT_VOICE *pWTVoice -@ PASSED IN: r0 -@ -@ S_WT_FRAME *pWTFrame; -@ PASSED IN: r1 -@**************************************************************** - - .include "ARM_synth_constants_gnu.inc" - - .arm - .text - - - .global WT_InterpolateNoLoop - - -@ Register usage -@ -------------- -pWTVoice .req r0 -pWTFrame .req r1 -pOutputBuffer .req r2 -numSamples .req r3 - -phaseIncrement .req r4 -pPhaseAccum .req r5 -phaseFrac .req r6 -phaseFracMask .req r7 - -tmp0 .req r1 @ reuse register -tmp1 .req r8 -tmp2 .req r9 - - -@SaveRegs RLIST {r4-r9,lr} -@RestoreRegs RLIST {r4-r9,pc} - - .func WT_InterpolateNoLoop -WT_InterpolateNoLoop: - - STMFD sp!, {r4-r9,lr} - -@ -@ Fetch parameters from structures -@---------------------------------------------------------------- - - LDR pOutputBuffer, [pWTFrame, #m_pAudioBuffer] - LDR numSamples, [pWTFrame, #m_numSamples] - - LDR phaseIncrement, [pWTFrame, #m_phaseIncrement] - LDR pPhaseAccum, [pWTVoice, #m_pPhaseAccum] - LDR phaseFrac, [pWTVoice, #m_phaseFrac] - LDR phaseFracMask,=PHASE_FRAC_MASK - -InterpolationLoop: - - .ifdef SAMPLES_8_BIT - LDRSB tmp0, [pPhaseAccum] @ tmp0 = x0 - LDRSB tmp1, [pPhaseAccum, #1] @ tmp1 = x1 - .else - LDRSH tmp0, [pPhaseAccum] @ tmp0 = x0 - LDRSH tmp1, [pPhaseAccum, #2] @ tmp1 = x1 - .endif - - ADD tmp2, phaseIncrement, phaseFrac @ increment pointer here to avoid pipeline stall - - SUB tmp1, tmp1, tmp0 @ tmp1 = x1 - x0 - SMULBB tmp1, phaseFrac, tmp1 @ tmp1 = phaseFrac * tmp2 - -@ This section performs a gain adjustment of -12dB for 16-bit samples -@ or +36dB for 8-bit samples. For a high quality synthesizer, the output -@ can be set to full scale, however if the filter is used, it can overflow -@ with certain coefficients and signal sources. In this case, either a -@ saturation operation should take in the filter before scaling back to -@ 16 bits or the signal path should be increased to 18 bits or more. - - .ifdef SAMPLES_8_BIT - MOV tmp0, tmp0, LSL #6 @ boost 8-bit signal by 36dB - .else - MOV tmp0, tmp0, ASR #2 @ reduce 16-bit signal by 12dB - .endif - - ADD tmp1, tmp0, tmp1, ASR #(NUM_EG1_FRAC_BITS-6) @ tmp1 = tmp0 + (tmp1 >> (15-6)) - @ = x0 + f * (x1 - x0) == interpolated result - - STRH tmp1, [pOutputBuffer], #NEXT_OUTPUT_PCM @ *pOutputBuffer++ = interpolated result - -@ carry overflow from fraction to integer portion - ADD pPhaseAccum, pPhaseAccum, tmp2, LSR #(NUM_PHASE_FRAC_BITS - NEXT_INPUT_PCM_SHIFT) - AND phaseFrac, tmp2, phaseFracMask @ nphaseFrac = frac part - - SUBS numSamples, numSamples, #1 - BGT InterpolationLoop - -@ Clean up and store any changes that were caused during the loop -@---------------------------------------------------------------- - - @ update and store phase - STR pPhaseAccum, [pWTVoice, #m_pPhaseAccum] - STR phaseFrac, [pWTVoice, #m_phaseFrac] - -@ -@ Return to calling function -@---------------------------------------------------------------- - - LDMFD sp!,{r4-r9,lr} - BX lr - - .endfunc - .end - diff --git a/arm-hybrid-22k/lib_src/ARM-E_mastergain_gnu.s b/arm-hybrid-22k/lib_src/ARM-E_mastergain_gnu.s deleted file mode 100644 index b4e905b..0000000 --- a/arm-hybrid-22k/lib_src/ARM-E_mastergain_gnu.s +++ /dev/null @@ -1,109 +0,0 @@ -@*********************************************************** -@ Function: SynthMasterGain -@ Processor: ARM-E -@ Description: Copies 32-bit synth output to 16-bit buffer -@ with saturated gain control -@ C-callable. -@ -@ Usage: -@ SynthMasterGain( -@ pInputBuffer -@ pOutputBuffer, -@ nGain, -@ nNumLoopSamples -@ ); -@ -@ Copyright Sonic Network Inc. 2004 -@**************************************************************** -@ Revision Control: -@ $Revision: 496 $ -@ $Date: 2006-12-11 14:33:26 -0800 (Mon, 11 Dec 2006) $ -@**************************************************************** -@ -@ where: -@ long *pInputBuffer -@ PASSED IN: r0 -@ -@ EAS_PCM *pOutputBuffer -@ PASSED IN: r1 -@ -@ short nGain -@ PASSED IN: r2 -@ -@ EAS_U16 nNumLoopSamples -@ PASSED IN: r3 -@ -@**************************************************************** - - .include "ARM_synth_constants_gnu.inc" - - .arm - .text - - .func SynthMasterGain -SynthMasterGain: - - .global SynthMasterGain @ allow other files to use this function - - - - - -@ Stack frame -@ ----------- - .equ RET_ADDR_SZ, 0 @return address - .equ REG_SAVE_SZ, 0 @save-on-entry registers saved - .equ FRAME_SZ, (8) @local variables - .equ ARG_BLK_SZ, 0 @argument block - - .equ PARAM_OFFSET, (ARG_BLK_SZ + FRAME_SZ + REG_SAVE_SZ + RET_ADDR_SZ) - -@ Register usage -@ -------------- -pnInputBuffer .req r0 -pnOutputBuffer .req r1 -nGain .req r2 -nNumLoopSamples .req r3 - - STMFD sp!,{r4-r6,r14} @Save any save-on-entry registers that are used - - LDR r6, =0x7fff @constant for saturation tests - -loop: - LDR r4, [pnInputBuffer], #4 @fetch 1st output sample - - LDR r5, [pnInputBuffer], #4 @fetch 2nd output sample - - SMULWB r4, r4, nGain @output = gain * input - - CMP r4, r6 @check for positive saturation - MOVGT r4, r6 @saturate - CMN r4, r6 @check for negative saturation - MVNLT r4, r6 @saturate - - SMULWB r5, r5, nGain @output = gain * input - - STRH r4, [pnOutputBuffer], #NEXT_OUTPUT_PCM @save 1st output sample - - CMP r5, r6 @check for positive saturation - MOVGT r5, r6 @saturate - CMN r5, r6 @check for negative saturation - MVNLT r5, r6 @saturate - STRH r5, [pnOutputBuffer], #NEXT_OUTPUT_PCM @save 2nd output sample - - SUBS nNumLoopSamples, nNumLoopSamples, #2 - BGT loop - -@ -@ Return to calling function -@---------------------------------------------------------------- - - LDMFD sp!,{r4-r6, lr} @ return to calling function - BX lr - -@***************************************************************************** - - .endfunc @ end of function/procedure - - .end @ end of assembly code - diff --git a/arm-hybrid-22k/lib_src/ARM-E_voice_gain_gnu.s b/arm-hybrid-22k/lib_src/ARM-E_voice_gain_gnu.s deleted file mode 100644 index 4517a3d..0000000 --- a/arm-hybrid-22k/lib_src/ARM-E_voice_gain_gnu.s +++ /dev/null @@ -1,166 +0,0 @@ -@*********************************************************** -@ Function: WT_VoiceGain -@ Processor: ARM-E -@ Description: the main synthesis function when fetching -@ wavetable samples. -@ C-callable. -@ -@ Usage: -@ Usage: -@ WT_VoiceGain( -@ S_WT_VOICE *pWTVoice, -@ S_WT_FRAME *pWTFrame); -@ -@ Copyright 2004, 2005 Sonic Network, Inc. -@**************************************************************** -@ Revision Control: -@ $Revision: 814 $ -@ $Date: 2007-08-02 10:34:53 -0700 (Thu, 02 Aug 2007) $ -@**************************************************************** -@ -@ where: -@ S_WT_VOICE *psVoice -@ PASSED IN: r0 -@ -@ S_WT_FRAME *pWTFrame -@ PASSED IN: r1 -@**************************************************************** - - - - .include "ARM_synth_constants_gnu.inc" - - .arm - .text - - .global WT_VoiceGain - -@ Register usage -@ -------------- -pWTVoice .req r0 -pWTFrame .req r1 -pInputBuffer .req r2 -pMixBuffer .req r3 - -tmp0 .req r4 -tmp1 .req r5 -tmp2 .req r1 @ reuse register -tmp3 .req r6 - -numSamples .req r9 - - .if STEREO_OUTPUT -gainIncLeft .req r7 -gainIncRight .req r8 -gainLeft .req r10 -gainRight .req r11 - .else -gainIncrement .req r7 -gain .req r8 - .endif - - -@ register context for local variables -@SaveRegs RLIST {r4-r11,lr} -@RestoreRegs RLIST {r4-r11,pc} - - .func WT_VoiceGain -WT_VoiceGain: - - STMFD sp!, {r4-r11,lr} - - LDR pInputBuffer, [pWTFrame, #m_pAudioBuffer] - LDR pMixBuffer, [pWTFrame, #m_pMixBuffer] - LDR numSamples, [pWTFrame, #m_numSamples] - -@---------------------------------------------------------------- -@ Stereo version -@---------------------------------------------------------------- -@ NOTE: instructions are reordered to reduce the effect of latency -@ due to storage and computational dependencies. -@---------------------------------------------------------------- - - .if STEREO_OUTPUT - - LDR tmp0, [pWTFrame, #m_prevGain] - LDR tmp1, [pWTFrame, #m_gainTarget] - - LDRSH gainLeft, [pWTVoice, #m_gainLeft] - LDRSH gainRight, [pWTVoice, #m_gainRight] - - MOV gainIncLeft, gainLeft - SMULBB gainLeft, tmp0, gainLeft - - SMULBB gainIncLeft, tmp1, gainIncLeft - SUB gainIncLeft, gainIncLeft, gainLeft - MOV gainLeft, gainLeft, ASR #(NUM_MIXER_GUARD_BITS - 2) - MOV gainIncLeft, gainIncLeft, ASR #(SYNTH_UPDATE_PERIOD_IN_BITS + NUM_MIXER_GUARD_BITS - 2) - - MOV gainIncRight, gainRight - SMULBB gainRight, tmp0, gainRight - - SMULBB gainIncRight, tmp1, gainIncRight - SUB gainIncRight, gainIncRight, gainRight - MOV gainRight, gainRight, ASR #(NUM_MIXER_GUARD_BITS - 2) - MOV gainIncRight, gainIncRight, ASR #(SYNTH_UPDATE_PERIOD_IN_BITS + NUM_MIXER_GUARD_BITS - 2) - - LDRSH tmp0, [pInputBuffer], #2 - -StereoGainLoop: - LDR tmp1, [pMixBuffer] - - ADD gainLeft, gainLeft, gainIncLeft - - SMLAWB tmp1, gainLeft, tmp0, tmp1 - - LDR tmp2, [pMixBuffer, #4] - - ADD gainRight, gainRight, gainIncRight - - STR tmp1, [pMixBuffer], #4 - - SMLAWB tmp2, gainRight, tmp0, tmp2 - - SUBS numSamples, numSamples, #1 - - LDRGTSH tmp0, [pInputBuffer], #2 - - STR tmp2, [pMixBuffer], #4 - - BGT StereoGainLoop - -@---------------------------------------------------------------- -@ Mono version -@---------------------------------------------------------------- - .else - - LDR gain, [pWTFrame, #m_prevGain] - MOV gain, gain, LSL #(NUM_MIXER_GUARD_BITS + 4) - LDR gainIncrement, [pWTFrame, #m_gainTarget] - MOV gainIncrement, gainIncrement, LSL #(NUM_MIXER_GUARD_BITS + 4) - SUB gainIncrement, gainIncrement, gain - MOV gainIncrement, gainIncrement, ASR #SYNTH_UPDATE_PERIOD_IN_BITS - -MonoGainLoop: - - LDRSH tmp0, [pInputBuffer], #NEXT_OUTPUT_PCM @ fetch voice output - - LDR tmp1, [pMixBuffer] @ get left channel output sample - ADD gain, gain, gainIncrement @ gain step to eliminate zipper noise - SMULWB tmp0, gain, tmp0 @ sample * local gain - - MOV tmp0, tmp0, ASR #1 @ add 6dB headroom - ADD tmp1, tmp0, tmp1 - STR tmp1, [pMixBuffer], #4 @ save and bump pointer - - SUBS numSamples, numSamples, #1 - BGT MonoGainLoop - - .endif @end Mono version - - LDMFD sp!,{r4-r11,lr} - BX lr - - .endfunc - .end - diff --git a/arm-hybrid-22k/lib_src/ARM_synth_constants_gnu.inc b/arm-hybrid-22k/lib_src/ARM_synth_constants_gnu.inc deleted file mode 100644 index b4cd29a..0000000 --- a/arm-hybrid-22k/lib_src/ARM_synth_constants_gnu.inc +++ /dev/null @@ -1,153 +0,0 @@ -@*********************************************************** -@ File: ARM_synth_constants.inc -@ Processor: ARM -@ Description: Contains constants and defines, most of which -@ are mirrored in synth.h -@ -@ Copyright Sonic Network Inc. 2004 -@**************************************************************** -@ Revision Control: -@ $Revision: 741 $ -@ $Date: 2007-06-22 16:39:21 -0700 (Fri, 22 Jun 2007) $ -@**************************************************************** - - - .ifdef SAMPLE_RATE_8000 - .equ SYNTH_UPDATE_PERIOD_IN_BITS, 5 - .equ BUFFER_SIZE_IN_MONO_SAMPLES, 32 - .endif - - .ifdef SAMPLE_RATE_16000 - .equ SYNTH_UPDATE_PERIOD_IN_BITS, 6 - .equ BUFFER_SIZE_IN_MONO_SAMPLES, 64 - .endif - - .ifdef SAMPLE_RATE_20000 - .equ SYNTH_UPDATE_PERIOD_IN_BITS, 7 - .equ BUFFER_SIZE_IN_MONO_SAMPLES, 128 - .endif - - .ifdef SAMPLE_RATE_22050 - .equ SYNTH_UPDATE_PERIOD_IN_BITS, 7 - .equ BUFFER_SIZE_IN_MONO_SAMPLES, 128 - .endif - - .ifdef SAMPLE_RATE_24000 - .equ SYNTH_UPDATE_PERIOD_IN_BITS, 7 - .equ BUFFER_SIZE_IN_MONO_SAMPLES, 128 - .endif - - .ifdef SAMPLE_RATE_32000 - .equ SYNTH_UPDATE_PERIOD_IN_BITS, 7 - .equ BUFFER_SIZE_IN_MONO_SAMPLES, 128 - .endif - - .ifdef SAMPLE_RATE_44100 - .equ SYNTH_UPDATE_PERIOD_IN_BITS, 8 - .equ BUFFER_SIZE_IN_MONO_SAMPLES, 256 - .endif - - .ifdef SAMPLE_RATE_48000 - .equ SYNTH_UPDATE_PERIOD_IN_BITS, 8 - .equ BUFFER_SIZE_IN_MONO_SAMPLES, 256 - .endif - - -@ if the OUTPUT PCM sample is 16-bits, then when using indexed addressing, -@ the next sample is this many bytes away - .equ NEXT_OUTPUT_PCM, 2 - -@**************************************************************************** -@/* macros for fractional phase accumulator */ - .equ NUM_PHASE_FRAC_BITS, 15 - - .equ PHASE_FRAC_MASK, 0x7FFF - -@ shift for phase accumulator when fraction carries over - .ifdef SAMPLES_8_BIT - .equ NEXT_INPUT_PCM_SHIFT, 0 - .endif - - .ifdef SAMPLES_16_BIT - .equ NEXT_INPUT_PCM_SHIFT, 1 - .endif - -@**************************************************************************** - .equ NUM_MIXER_GUARD_BITS, 4 - -@**************************************************************************** -@/* Envelope 1 (EG1) calculation macros */ - .equ NUM_EG1_FRAC_BITS, 15 - -@**************************************************************************** - - .equ NUM_ENHANCER_FILTER_COEF_FRAC_BITS, 5 - -@**************************************************************************** - -@ -@ I've temporarily given up on the idea of getting ADS/RV and gcc to -@ handle a struct in a compatible fashion. Switching to old fashion EQU -@ - - .if FILTER_ENABLED -@************************************** -@ typedef struct s_filter_tag - .equ m_z1, 0 - .equ m_z2, 2 - .endif - -@************************************** -@ typedef struct s_wt_frame_tag - .equ m_gainTarget, 0 - .equ m_phaseIncrement, 4 - - .if FILTER_ENABLED - .equ m_k, 8 - .equ m_b1, 12 - .equ m_b2, 16 - .equ m_pAudioBuffer, 20 - .equ m_pMixBuffer, 24 - .equ m_numSamples, 28 - .equ m_prevGain, 32 - .else - .equ m_pAudioBuffer, 8 - .equ m_pMixBuffer, 12 - .equ m_numSamples, 16 - .equ m_prevGain, 20 - .endif - - -@************************************** -@ typedef struct s_wt_voice_tag - .equ m_pLoopEnd, 0 @ /* points to last PCM sample (not 1 beyond last) */ - .equ m_pLoopStart, 4 @ /* points to first sample at start of loop */ - .equ m_pPhaseAccum, 8 @ /* points to first sample at start of loop */ - .equ m_phaseFrac, 12 @ /* points to first sample at start of loop */ - - .if STEREO_OUTPUT - .equ m_gainLeft, 16 @ /* current gain, left ch */ - .equ m_gainRight, 18 @ /* current gain, right ch */ - .endif - - -@**************************************************************************** -@ enhancer - .equ m_nEnhancerFeedForward1, 0 - .equ m_nEnhancerFeedback1, 1 - .equ m_nDriveCoef, 2 - .equ m_nEnhancerFeedback2, 3 - .equ m_nWet, 4 - .equ m_nDry, 5 - - .equ m_zF0L, 6 @ filter 1 zero state var, left - .equ m_zF1L, 8 @ filter 1 pole state var, left - .equ m_zF2L, 10 @ filter 2 zero state var, left - .equ m_zF0R, 12 @ filter 1 zero state var, right - .equ m_zF1R, 14 @ filter 1 pole state var, right - .equ m_zF2R, 16 @ filter 2 zero state var, right - -@**************************************************************************** - - - diff --git a/arm-hybrid-22k/lib_src/arm-hybrid-22k_lib.mak b/arm-hybrid-22k/lib_src/arm-hybrid-22k_lib.mak deleted file mode 100644 index 43edd6d..0000000 --- a/arm-hybrid-22k/lib_src/arm-hybrid-22k_lib.mak +++ /dev/null @@ -1,25 +0,0 @@ -# -# Auto-generated sample makefile -# -# This makefile is intended for use with GNU make. -# Set the paths to the tools (CC, AR, LD, etc.) -# - -vpath %.c lib_src - -CC = C:\Program Files\GNUARM\bin\arm-elf-gcc.exe -AS = C:\Program Files\GNUARM\bin\arm-elf-as.exe -LD = C:\Program Files\GNUARM\bin\arm-elf-gcc.exe -AR = C:\Program Files\GNUARM\bin\arm-elf-ar.exe - -%.o: %.c - $(CC) -c -O2 -o $@ -I lib_src -I host_src -D NUM_OUTPUT_CHANNELS=2 -D _SAMPLE_RATE_22050 -D MAX_SYNTH_VOICES=32 -D NUM_PRIMARY_VOICES=8 -D EAS_HYBRID_SYNTH -D _8_BIT_SAMPLES -D _FILTER_ENABLED -D _IMELODY_PARSER -D _RTTTL_PARSER -D _OTA_PARSER -D _WAVE_PARSER -D _REVERB_ENABLED -D _CHORUS_ENABLED -D _IMA_DECODER -D NATIVE_EAS_KERNEL $< - -%.o: %.s - $(AS) -o $@ -EL -mcpu=arm946e-s -mfpu=softfpa -I lib_src --defsym CHECK_STACK=0 --defsym REVERB=0 --defsym CHORUS=0 --defsym STEREO_OUTPUT=1 --defsym SAMPLE_RATE_22050=1 --defsym SAMPLES_8_BIT=1 --defsym FILTER_ENABLED=1 $< - -OBJS = eas_mididata.o eas_pan.o eas_wavefiledata.o eas_wavefile.o eas_smfdata.o eas_imelody.o eas_math.o eas_fmengine.o ARM-E_filter_gnu.o eas_chorusdata.o eas_wtengine.o eas_ota.o eas_rtttldata.o eas_reverbdata.o eas_public.o ARM-E_interpolate_loop_gnu.o eas_rtttl.o eas_reverb.o ARM-E_voice_gain_gnu.o eas_fmsynth.o eas_midi.o eas_otadata.o eas_wtsynth.o eas_mixbuf.o eas_imaadpcm.o hybrid_22khz_mcu.o eas_smf.o eas_chorus.o eas_pcm.o eas_mixer.o eas_data.o eas_imelodydata.o eas_pcmdata.o eas_ima_tables.o eas_fmtables.o ARM-E_interpolate_noloop_gnu.o ARM-E_mastergain_gnu.o eas_voicemgt.o - -arm-hybrid-22k.a: $(OBJS) - $(AR) rc lib$@ $(OBJS) - diff --git a/arm-hybrid-22k/lib_src/eas_audioconst.h b/arm-hybrid-22k/lib_src/eas_audioconst.h deleted file mode 100644 index 1cfa404..0000000 --- a/arm-hybrid-22k/lib_src/eas_audioconst.h +++ /dev/null @@ -1,97 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_audioconst.h - * - * Contents and purpose: - * Defines audio constants related to the sample rate, bit size, etc. - * - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_AUDIOCONST_H -#define _EAS_AUDIOCONST_H - -/*---------------------------------------------------------------------------- - * These macros define the various characteristics of the defined sample rates - *---------------------------------------------------------------------------- - * BUFFER_SIZE_IN_MONO_SAMPLES size of buffer in samples - * _OUTPUT_SAMPLE_RATE compiled output sample rate - * AUDIO_FRAME_LENGTH length of an audio frame in 256ths of a millisecond - * SYNTH_UPDATE_PERIOD_IN_BITS length of an audio frame (2^x samples) - *---------------------------------------------------------------------------- -*/ - -#if defined (_SAMPLE_RATE_8000) -#define BUFFER_SIZE_IN_MONO_SAMPLES 32 -#define _OUTPUT_SAMPLE_RATE 8000 -#define AUDIO_FRAME_LENGTH 1024 -#define SYNTH_UPDATE_PERIOD_IN_BITS 5 - -#elif defined (_SAMPLE_RATE_16000) -#define BUFFER_SIZE_IN_MONO_SAMPLES 64 -#define _OUTPUT_SAMPLE_RATE 16000 -#define AUDIO_FRAME_LENGTH 1024 -#define SYNTH_UPDATE_PERIOD_IN_BITS 6 - -#elif defined (_SAMPLE_RATE_20000) -#define BUFFER_SIZE_IN_MONO_SAMPLES 128 -#define _OUTPUT_SAMPLE_RATE 20000 -#define AUDIO_FRAME_LENGTH 1638 -#define SYNTH_UPDATE_PERIOD_IN_BITS 7 - -#elif defined (_SAMPLE_RATE_22050) -#define BUFFER_SIZE_IN_MONO_SAMPLES 128 -#define _OUTPUT_SAMPLE_RATE 22050 -#define AUDIO_FRAME_LENGTH 1486 -#define SYNTH_UPDATE_PERIOD_IN_BITS 7 - -#elif defined (_SAMPLE_RATE_24000) -#define BUFFER_SIZE_IN_MONO_SAMPLES 128 -#define _OUTPUT_SAMPLE_RATE 24000 -#define AUDIO_FRAME_LENGTH 1365 -#define SYNTH_UPDATE_PERIOD_IN_BITS 7 - -#elif defined (_SAMPLE_RATE_32000) -#define BUFFER_SIZE_IN_MONO_SAMPLES 128 -#define _OUTPUT_SAMPLE_RATE 32000 -#define AUDIO_FRAME_LENGTH 1024 -#define SYNTH_UPDATE_PERIOD_IN_BITS 7 - -#elif defined (_SAMPLE_RATE_44100) -#define BUFFER_SIZE_IN_MONO_SAMPLES 256 -#define _OUTPUT_SAMPLE_RATE 44100 -#define AUDIO_FRAME_LENGTH 1486 -#define SYNTH_UPDATE_PERIOD_IN_BITS 8 - -#elif defined (_SAMPLE_RATE_48000) -#define BUFFER_SIZE_IN_MONO_SAMPLES 256 -#define _OUTPUT_SAMPLE_RATE 48000 -#define AUDIO_FRAME_LENGTH 1365 -#define SYNTH_UPDATE_PERIOD_IN_BITS 8 - -#else -#error "_SAMPLE_RATE_XXXXX must be defined to valid rate" -#endif - -#endif /* #ifndef _EAS_AUDIOCONST_H */ - diff --git a/arm-hybrid-22k/lib_src/eas_chorus.c b/arm-hybrid-22k/lib_src/eas_chorus.c deleted file mode 100644 index bc42237..0000000 --- a/arm-hybrid-22k/lib_src/eas_chorus.c +++ /dev/null @@ -1,604 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_chorus.c - * - * Contents and purpose: - * Contains the implementation of the Chorus effect. - * - * - * Copyright Sonic Network Inc. 2006 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 499 $ - * $Date: 2006-12-11 16:07:20 -0800 (Mon, 11 Dec 2006) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_data.h" -#include "eas_effects.h" -#include "eas_math.h" -#include "eas_chorusdata.h" -#include "eas_chorus.h" -#include "eas_config.h" -#include "eas_host.h" -#include "eas_report.h" - -/* prototypes for effects interface */ -static EAS_RESULT ChorusInit (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData); -static void ChorusProcess (EAS_VOID_PTR pInstData, EAS_PCM *pSrc, EAS_PCM *pDst, EAS_I32 numSamples); -static EAS_RESULT ChorusShutdown (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT ChorusGetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue); -static EAS_RESULT ChorusSetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value); - -/* common effects interface for configuration module */ -const S_EFFECTS_INTERFACE EAS_Chorus = -{ - ChorusInit, - ChorusProcess, - ChorusShutdown, - ChorusGetParam, - ChorusSetParam -}; - - - -//LFO shape table used by the chorus, larger table would sound better -//this is a sine wave, where 32767 = 1.0 -static const EAS_I16 EAS_chorusShape[CHORUS_SHAPE_SIZE] = { - 0, 1608, 3212, 4808, 6393, 7962, 9512, 11309, 12539, 14010, 15446, 16846, 18204, 19519, 20787, 22005, 23170, - 24279, 25329, 26319, 27245, 28105, 28898, 29621, 30273, 30852, 31356, 31785, 32137, 32412, 32609, 32728, - 32767, 32728, 32609, 32412, 32137, 31785, 31356, 30852, 30273, 29621, 28898, 28105, 27245, 26319, 25329, - 24279, 23170, 22005, 20787, 19519, 18204, 16846, 15446, 14010, 12539, 11039, 9512, 7962, 6393, 4808, 3212, - 1608, 0, -1608, -3212, -4808, -6393, -7962, -9512, -11309, -12539, -14010, -15446, -16846, -18204, -19519, - -20787, -22005, -23170, -24279, -25329, -26319, -27245, -28105, -28898, -29621, -30273, -30852, -31356, -31785, - -32137, -32412, -32609, -32728, -32767, -32728, -32609, -32412, -32137, -31785, -31356, -30852, -30273, -29621, - -28898, -28105, -27245, -26319, -25329, -24279, -23170, -22005, -20787, -19519, -18204, -16846, -15446, -14010, - -12539, -11039, -9512, -7962, -6393, -4808, -3212, -1608 -}; - -/*---------------------------------------------------------------------------- - * InitializeChorus() - *---------------------------------------------------------------------------- - * Purpose: Initializes chorus parameters - * - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ChorusInit (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData) -{ - S_CHORUS_OBJECT *pChorusData; - S_CHORUS_PRESET *pPreset; - EAS_I32 index; - - /* check Configuration Module for data allocation */ - if (pEASData->staticMemoryModel) - pChorusData = EAS_CMEnumFXData(EAS_MODULE_CHORUS); - - /* allocate dynamic memory */ - else - pChorusData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_CHORUS_OBJECT)); - - if (pChorusData == NULL) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Failed to allocate Chorus memory\n"); */ } - return EAS_ERROR_MALLOC_FAILED; - } - - /* clear the structure */ - EAS_HWMemSet(pChorusData, 0, sizeof(S_CHORUS_OBJECT)); - - ChorusReadInPresets(pChorusData); - - /* set some default values */ - pChorusData->bypass = EAS_CHORUS_BYPASS_DEFAULT; - pChorusData->preset = EAS_CHORUS_PRESET_DEFAULT; - pChorusData->m_nLevel = EAS_CHORUS_LEVEL_DEFAULT; - pChorusData->m_nRate = EAS_CHORUS_RATE_DEFAULT; - pChorusData->m_nDepth = EAS_CHORUS_DEPTH_DEFAULT; - - //chorus rate and depth need some massaging from preset value (which is sample rate independent) - - //convert rate from steps of .05 Hz to value which can be used as phase increment, - //with current CHORUS_SHAPE_SIZE and rate limits, this fits into 16 bits - //want to compute ((shapeSize * 65536) * (storedRate/20))/sampleRate; - //computing it as below allows rate steps to be evenly spaced - //uses 32 bit divide, but only once when new value is selected - pChorusData->m_nRate = (EAS_I16) - ((((EAS_I32)CHORUS_SHAPE_SIZE<<16)/(20*(EAS_I32)_OUTPUT_SAMPLE_RATE)) * pChorusData->m_nRate); - - //convert depth from steps of .05 ms, to samples, with 16 bit whole part, discard fraction - //want to compute ((depth * sampleRate)/20000) - //use the following approximation since 105/32 is roughly 65536/20000 - /*lint -e{704} use shift for performance */ - pChorusData->m_nDepth = (EAS_I16) - (((((EAS_I32)pChorusData->m_nDepth * _OUTPUT_SAMPLE_RATE)>>5) * 105) >> 16); - - pChorusData->m_nLevel = pChorusData->m_nLevel; - - //zero delay memory for chorus - for (index = CHORUS_L_SIZE - 1; index >= 0; index--) - { - pChorusData->chorusDelayL[index] = 0; - } - for (index = CHORUS_R_SIZE - 1; index >= 0; index--) - { - pChorusData->chorusDelayR[index] = 0; - } - - //init delay line index, these are used to implement circular delay buffer - pChorusData->chorusIndexL = 0; - pChorusData->chorusIndexR = 0; - - //init LFO phase - //16 bit whole part, 16 bit fraction - pChorusData->lfoLPhase = 0; - pChorusData->lfoRPhase = (CHORUS_SHAPE_SIZE << 16) >> 2; // 1/4 of total, i.e. 90 degrees out of phase; - - //init chorus delay position - //right now chorus delay is a compile-time value, as is sample rate - pChorusData->chorusTapPosition = (EAS_I16)((CHORUS_DELAY_MS * _OUTPUT_SAMPLE_RATE)/1000); - - //now copy from the new preset into Chorus - pPreset = &pChorusData->m_sPreset.m_sPreset[pChorusData->m_nNextChorus]; - - pChorusData->m_nLevel = pPreset->m_nLevel; - pChorusData->m_nRate = pPreset->m_nRate; - pChorusData->m_nDepth = pPreset->m_nDepth; - - pChorusData->m_nRate = (EAS_I16) - ((((EAS_I32)CHORUS_SHAPE_SIZE<<16)/(20*(EAS_I32)_OUTPUT_SAMPLE_RATE)) * pChorusData->m_nRate); - - /*lint -e{704} use shift for performance */ - pChorusData->m_nDepth = (EAS_I16) - (((((EAS_I32)pChorusData->m_nDepth * _OUTPUT_SAMPLE_RATE)>>5) * 105) >> 16); - - *pInstData = pChorusData; - - return EAS_SUCCESS; -} /* end ChorusInit */ - -/*---------------------------------------------------------------------------- - * WeightedTap() - *---------------------------------------------------------------------------- - * Purpose: Does fractional array look-up using linear interpolation - * - * first convert indexDesired to actual desired index by taking into account indexReference - * then do linear interpolation between two actual samples using fractional part - * - * Inputs: - * array: pointer to array of signed 16 bit values, typically either PCM data or control data - * indexReference: the circular buffer relative offset - * indexDesired: the fractional index we are looking up (16 bits index + 16 bits fraction) - * indexLimit: the total size of the array, used to compute buffer wrap - * - * Outputs: - * Value from the input array, linearly interpolated between two actual data values - * - *---------------------------------------------------------------------------- -*/ -static EAS_I16 WeightedTap(const EAS_I16 *array, EAS_I16 indexReference, EAS_I32 indexDesired, EAS_I16 indexLimit) -{ - EAS_I16 index; - EAS_I16 fraction; - EAS_I16 val1; - EAS_I16 val2; - - //separate indexDesired into whole and fractional parts - /*lint -e{704} use shift for performance */ - index = (EAS_I16)(indexDesired >> 16); - /*lint -e{704} use shift for performance */ - fraction = (EAS_I16)((indexDesired>>1) & 0x07FFF); //just use 15 bits of fractional part - - //adjust whole part by indexReference - index = indexReference - index; - //make sure we stay within array bounds, this implements circular buffer - while (index < 0) - { - index += indexLimit; - } - - //get two adjacent values from the array - val1 = array[index]; - - //handle special case when index == 0, else typical case - if (index == 0) - { - val2 = array[indexLimit-1]; //get last value from array - } - else - { - val2 = array[index-1]; //get previous value from array - } - - //compute linear interpolation as (val1 + ((val2-val1)*fraction)) - return(val1 + (EAS_I16)MULT_EG1_EG1(val2-val1,fraction)); -} - -/*---------------------------------------------------------------------------- - * ChorusProcess() - *---------------------------------------------------------------------------- - * Purpose: compute the chorus on the input buffer, and mix into output buffer - * - * - * Inputs: - * src: pointer to input buffer of PCM values to be processed - * dst: pointer to output buffer of PCM values we are to sume the result with - * bufSize: the number of sample frames (i.e. stereo samples) in the buffer - * - * Outputs: - * None - * - *---------------------------------------------------------------------------- -*/ -//compute the chorus, and mix into output buffer -static void ChorusProcess (EAS_VOID_PTR pInstData, EAS_PCM *pSrc, EAS_PCM *pDst, EAS_I32 numSamples) -{ - EAS_I32 ix; - EAS_I32 nChannelNumber; - EAS_I16 lfoValueLeft; - EAS_I16 lfoValueRight; - EAS_I32 positionOffsetL; - EAS_I32 positionOffsetR; - EAS_PCM tapL; - EAS_PCM tapR; - EAS_I32 tempValue; - EAS_PCM nInputSample; - EAS_I32 nOutputSample; - EAS_PCM *pIn; - EAS_PCM *pOut; - - S_CHORUS_OBJECT *pChorusData; - - pChorusData = (S_CHORUS_OBJECT*) pInstData; - - //if the chorus is disabled or turned all the way down - if (pChorusData->bypass == EAS_TRUE || pChorusData->m_nLevel == 0) - { - if (pSrc != pDst) - EAS_HWMemCpy(pSrc, pDst, numSamples * NUM_OUTPUT_CHANNELS * (EAS_I32) sizeof(EAS_PCM)); - return; - } - - if (pChorusData->m_nNextChorus != pChorusData->m_nCurrentChorus) - { - ChorusUpdate(pChorusData); - } - - for (nChannelNumber = 0; nChannelNumber < NUM_OUTPUT_CHANNELS; nChannelNumber++) - { - - pIn = pSrc + nChannelNumber; - pOut = pDst + nChannelNumber; - - if(nChannelNumber==0) - { - for (ix = 0; ix < numSamples; ix++) - { - nInputSample = *pIn; - pIn += NUM_OUTPUT_CHANNELS; - - //feed input into chorus delay line - pChorusData->chorusDelayL[pChorusData->chorusIndexL] = nInputSample; - - //compute chorus lfo value using phase as fractional index into chorus shape table - //resulting value is between -1.0 and 1.0, expressed as signed 16 bit number - lfoValueLeft = WeightedTap(EAS_chorusShape, 0, pChorusData->lfoLPhase, CHORUS_SHAPE_SIZE); - - //scale chorus depth by lfo value to get relative fractional sample index - //index is expressed as 32 bit number with 16 bit fractional part - /*lint -e{703} use shift for performance */ - positionOffsetL = pChorusData->m_nDepth * (((EAS_I32)lfoValueLeft) << 1); - - //add fixed chorus delay to get actual fractional sample index - positionOffsetL += ((EAS_I32)pChorusData->chorusTapPosition) << 16; - - //get tap value from chorus delay using fractional sample index - tapL = WeightedTap(pChorusData->chorusDelayL, pChorusData->chorusIndexL, positionOffsetL, CHORUS_L_SIZE); - - //scale by chorus level, then sum with input buffer contents and saturate - tempValue = MULT_EG1_EG1(tapL, pChorusData->m_nLevel); - nOutputSample = SATURATE(tempValue + nInputSample); - - *pOut = (EAS_I16)SATURATE(nOutputSample); - pOut += NUM_OUTPUT_CHANNELS; - - - //increment chorus delay index and make it wrap as needed - //this implements circular buffer - if ((pChorusData->chorusIndexL+=1) >= CHORUS_L_SIZE) - pChorusData->chorusIndexL = 0; - - //increment fractional lfo phase, and make it wrap as needed - pChorusData->lfoLPhase += pChorusData->m_nRate; - while (pChorusData->lfoLPhase >= (CHORUS_SHAPE_SIZE<<16)) - { - pChorusData->lfoLPhase -= (CHORUS_SHAPE_SIZE<<16); - } - } - } - else - { - for (ix = 0; ix < numSamples; ix++) - { - nInputSample = *pIn; - pIn += NUM_OUTPUT_CHANNELS; - - //feed input into chorus delay line - pChorusData->chorusDelayR[pChorusData->chorusIndexR] = nInputSample; - - //compute chorus lfo value using phase as fractional index into chorus shape table - //resulting value is between -1.0 and 1.0, expressed as signed 16 bit number - lfoValueRight = WeightedTap(EAS_chorusShape, 0, pChorusData->lfoRPhase, CHORUS_SHAPE_SIZE); - - //scale chorus depth by lfo value to get relative fractional sample index - //index is expressed as 32 bit number with 16 bit fractional part - /*lint -e{703} use shift for performance */ - positionOffsetR = pChorusData->m_nDepth * (((EAS_I32)lfoValueRight) << 1); - - //add fixed chorus delay to get actual fractional sample index - positionOffsetR += ((EAS_I32)pChorusData->chorusTapPosition) << 16; - - //get tap value from chorus delay using fractional sample index - tapR = WeightedTap(pChorusData->chorusDelayR, pChorusData->chorusIndexR, positionOffsetR, CHORUS_R_SIZE); - - //scale by chorus level, then sum with output buffer contents and saturate - tempValue = MULT_EG1_EG1(tapR, pChorusData->m_nLevel); - nOutputSample = SATURATE(tempValue + nInputSample); - - *pOut = (EAS_I16)SATURATE(nOutputSample); - pOut += NUM_OUTPUT_CHANNELS; - - //increment chorus delay index and make it wrap as needed - //this implements circular buffer - if ((pChorusData->chorusIndexR+=1) >= CHORUS_R_SIZE) - pChorusData->chorusIndexR = 0; - - //increment fractional lfo phase, and make it wrap as needed - pChorusData->lfoRPhase += pChorusData->m_nRate; - while (pChorusData->lfoRPhase >= (CHORUS_SHAPE_SIZE<<16)) - { - pChorusData->lfoRPhase -= (CHORUS_SHAPE_SIZE<<16); - } - } - } - - } -} /* end ChorusProcess */ - - - -/*---------------------------------------------------------------------------- - * ChorusShutdown() - *---------------------------------------------------------------------------- - * Purpose: - * Initializes the Chorus effect. - * - * Inputs: - * pInstData - handle to instance data - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ChorusShutdown (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData) -{ - /* check Configuration Module for static memory allocation */ - if (!pEASData->staticMemoryModel) - EAS_HWFree(pEASData->hwInstData, pInstData); - return EAS_SUCCESS; -} /* end ChorusShutdown */ - -/*---------------------------------------------------------------------------- - * ChorusGetParam() - *---------------------------------------------------------------------------- - * Purpose: - * Get a Chorus parameter - * - * Inputs: - * pInstData - handle to instance data - * param - parameter index - * *pValue - pointer to variable to hold retrieved value - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ChorusGetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue) -{ - S_CHORUS_OBJECT *p; - - p = (S_CHORUS_OBJECT*) pInstData; - - switch (param) - { - case EAS_PARAM_CHORUS_BYPASS: - *pValue = (EAS_I32) p->bypass; - break; - case EAS_PARAM_CHORUS_PRESET: - *pValue = (EAS_I8) p->m_nCurrentChorus; - break; - case EAS_PARAM_CHORUS_RATE: - *pValue = (EAS_I32) p->m_nRate; - break; - case EAS_PARAM_CHORUS_DEPTH: - *pValue = (EAS_I32) p->m_nDepth; - break; - case EAS_PARAM_CHORUS_LEVEL: - *pValue = (EAS_I32) p->m_nLevel; - break; - default: - return EAS_ERROR_INVALID_PARAMETER; - } - return EAS_SUCCESS; -} /* end ChorusGetParam */ - - -/*---------------------------------------------------------------------------- - * ChorusSetParam() - *---------------------------------------------------------------------------- - * Purpose: - * Set a Chorus parameter - * - * Inputs: - * pInstData - handle to instance data - * param - parameter index - * *pValue - new paramter value - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ChorusSetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value) -{ - S_CHORUS_OBJECT *p; - - p = (S_CHORUS_OBJECT*) pInstData; - - switch (param) - { - case EAS_PARAM_CHORUS_BYPASS: - p->bypass = (EAS_BOOL) value; - break; - case EAS_PARAM_CHORUS_PRESET: - if(value!=EAS_PARAM_CHORUS_PRESET1 && value!=EAS_PARAM_CHORUS_PRESET2 && - value!=EAS_PARAM_CHORUS_PRESET3 && value!=EAS_PARAM_CHORUS_PRESET4) - return EAS_ERROR_INVALID_PARAMETER; - p->m_nNextChorus = (EAS_I8)value; - break; - case EAS_PARAM_CHORUS_RATE: - if(valueEAS_CHORUS_RATE_MAX) - return EAS_ERROR_INVALID_PARAMETER; - p->m_nRate = (EAS_I16) value; - break; - case EAS_PARAM_CHORUS_DEPTH: - if(valueEAS_CHORUS_DEPTH_MAX) - return EAS_ERROR_INVALID_PARAMETER; - p->m_nDepth = (EAS_I16) value; - break; - case EAS_PARAM_CHORUS_LEVEL: - if(valueEAS_CHORUS_LEVEL_MAX) - return EAS_ERROR_INVALID_PARAMETER; - p->m_nLevel = (EAS_I16) value; - break; - - default: - return EAS_ERROR_INVALID_PARAMETER; - } - return EAS_SUCCESS; -} /* end ChorusSetParam */ - - -/*---------------------------------------------------------------------------- - * ChorusReadInPresets() - *---------------------------------------------------------------------------- - * Purpose: sets global Chorus preset bank to defaults - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ChorusReadInPresets(S_CHORUS_OBJECT *pChorusData) -{ - - int preset = 0; - int defaultPreset = 0; - - //now init any remaining presets to defaults - for (defaultPreset = preset; defaultPreset < CHORUS_MAX_TYPE; defaultPreset++) - { - S_CHORUS_PRESET *pPreset = &pChorusData->m_sPreset.m_sPreset[defaultPreset]; - if (defaultPreset == 0 || defaultPreset > CHORUS_MAX_TYPE-1) - { - pPreset->m_nDepth = 39; - pPreset->m_nRate = 30; - pPreset->m_nLevel = 32767; - } - else if (defaultPreset == 1) - { - pPreset->m_nDepth = 21; - pPreset->m_nRate = 45; - pPreset->m_nLevel = 25000; - } - else if (defaultPreset == 2) - { - pPreset->m_nDepth = 53; - pPreset->m_nRate = 25; - pPreset->m_nLevel = 32000; - } - else if (defaultPreset == 3) - { - pPreset->m_nDepth = 32; - pPreset->m_nRate = 37; - pPreset->m_nLevel = 29000; - } - } - - return EAS_SUCCESS; -} - - -/*---------------------------------------------------------------------------- - * ChorusUpdate - *---------------------------------------------------------------------------- - * Purpose: - * Update the Chorus preset parameters as required - * - * Inputs: - * - * Outputs: - * - * - * Side Effects: - * - chorus paramters will be changed - * - m_nCurrentRoom := m_nNextRoom - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ChorusUpdate(S_CHORUS_OBJECT *pChorusData) -{ - S_CHORUS_PRESET *pPreset = &pChorusData->m_sPreset.m_sPreset[pChorusData->m_nNextChorus]; - - pChorusData->m_nLevel = pPreset->m_nLevel; - pChorusData->m_nRate = pPreset->m_nRate; - pChorusData->m_nDepth = pPreset->m_nDepth; - - pChorusData->m_nRate = (EAS_I16) - ((((EAS_I32)CHORUS_SHAPE_SIZE<<16)/(20*(EAS_I32)_OUTPUT_SAMPLE_RATE)) * pChorusData->m_nRate); - - /*lint -e{704} use shift for performance */ - pChorusData->m_nDepth = (EAS_I16) - (((((EAS_I32)pChorusData->m_nDepth * _OUTPUT_SAMPLE_RATE)>>5) * 105) >> 16); - - pChorusData->m_nCurrentChorus = pChorusData->m_nNextChorus; - - return EAS_SUCCESS; - -} /* end ChorusUpdate */ diff --git a/arm-hybrid-22k/lib_src/eas_chorusdata.c b/arm-hybrid-22k/lib_src/eas_chorusdata.c deleted file mode 100644 index caee1ed..0000000 --- a/arm-hybrid-22k/lib_src/eas_chorusdata.c +++ /dev/null @@ -1,34 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_chorusdata.c - * - * Contents and purpose: - * Contains the static data allocation for the Chorus effect - * - * - * Copyright Sonic Network Inc. 2006 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 550 $ - * $Date: 2007-02-02 09:37:03 -0800 (Fri, 02 Feb 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_chorusdata.h" - -S_CHORUS_OBJECT eas_ChorusData; - diff --git a/arm-hybrid-22k/lib_src/eas_chorusdata.h b/arm-hybrid-22k/lib_src/eas_chorusdata.h deleted file mode 100644 index 4420ddd..0000000 --- a/arm-hybrid-22k/lib_src/eas_chorusdata.h +++ /dev/null @@ -1,160 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_chorusdata.h - * - * Contents and purpose: - * Contains the prototypes for the Chorus effect. - * - * - * Copyright Sonic Network Inc. 2006 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 309 $ - * $Date: 2006-09-12 18:52:45 -0700 (Tue, 12 Sep 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_CHORUS_H -#define _EAS_CHORUS_H - -#include "eas_types.h" -#include "eas_audioconst.h" - -//defines for chorus - -#define EAS_CHORUS_BYPASS_DEFAULT 1 -#define EAS_CHORUS_PRESET_DEFAULT 0 -#define EAS_CHORUS_RATE_DEFAULT 30 -#define EAS_CHORUS_DEPTH_DEFAULT 39 -#define EAS_CHORUS_LEVEL_DEFAULT 32767 - -#define EAS_CHORUS_LEVEL_MIN 0 -#define EAS_CHORUS_LEVEL_MAX 32767 - -#define EAS_CHORUS_RATE_MIN 10 -#define EAS_CHORUS_RATE_MAX 50 - -#define EAS_CHORUS_DEPTH_MIN 15 -#define EAS_CHORUS_DEPTH_MAX 60 - -#define CHORUS_SIZE_MS 20 -#define CHORUS_L_SIZE ((CHORUS_SIZE_MS*_OUTPUT_SAMPLE_RATE)/1000) -#define CHORUS_R_SIZE CHORUS_L_SIZE -#define CHORUS_SHAPE_SIZE 128 -#define CHORUS_DELAY_MS 10 - -#define CHORUS_MAX_TYPE 4 // any Chorus numbers larger than this are invalid - -typedef struct -{ - EAS_I16 m_nRate; - EAS_I16 m_nDepth; - EAS_I16 m_nLevel; - -} S_CHORUS_PRESET; - -typedef struct -{ - S_CHORUS_PRESET m_sPreset[CHORUS_MAX_TYPE]; //array of presets - -} S_CHORUS_PRESET_BANK; - -/* parameters for each Chorus */ -typedef struct -{ - EAS_I32 lfoLPhase; - EAS_I32 lfoRPhase; - EAS_I16 chorusIndexL; - EAS_I16 chorusIndexR; - EAS_U16 chorusTapPosition; - - EAS_I16 m_nRate; - EAS_I16 m_nDepth; - EAS_I16 m_nLevel; - - //delay lines used by the chorus, longer would sound better - EAS_PCM chorusDelayL[CHORUS_L_SIZE]; - EAS_PCM chorusDelayR[CHORUS_R_SIZE]; - - EAS_BOOL bypass; - EAS_I8 preset; - - EAS_I16 m_nCurrentChorus; // preset number for current Chorus - EAS_I16 m_nNextChorus; // preset number for next Chorus - - S_CHORUS_PRESET pPreset; - - S_CHORUS_PRESET_BANK m_sPreset; - -} S_CHORUS_OBJECT; - - -/*---------------------------------------------------------------------------- - * WeightedTap() - *---------------------------------------------------------------------------- - * Purpose: Does fractional array look-up using linear interpolation - * - * first convert indexDesired to actual desired index by taking into account indexReference - * then do linear interpolation between two actual samples using fractional part - * - * Inputs: - * array: pointer to array of signed 16 bit values, typically either PCM data or control data - * indexReference: the circular buffer relative offset - * indexDesired: the fractional index we are looking up (16 bits index + 16 bits fraction) - * indexLimit: the total size of the array, used to compute buffer wrap - * - * Outputs: - * Value from the input array, linearly interpolated between two actual data values - * - *---------------------------------------------------------------------------- -*/ -static EAS_I16 WeightedTap(const EAS_I16 *array, EAS_I16 indexReference, EAS_I32 indexDesired, EAS_I16 indexLimit); - -/*---------------------------------------------------------------------------- - * ChorusReadInPresets() - *---------------------------------------------------------------------------- - * Purpose: sets global Chorus preset bank to defaults - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ChorusReadInPresets(S_CHORUS_OBJECT *pChorusData); - -/*---------------------------------------------------------------------------- - * ChorusUpdate - *---------------------------------------------------------------------------- - * Purpose: - * Update the Chorus preset parameters as required - * - * Inputs: - * - * Outputs: - * - * - * Side Effects: - * - chorus paramters will be changed - * - m_nCurrentChorus := m_nNextChorus - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ChorusUpdate(S_CHORUS_OBJECT* pChorusData); - -#endif /* #ifndef _EAS_CHORUSDATA_H */ - - diff --git a/arm-hybrid-22k/lib_src/eas_ctype.h b/arm-hybrid-22k/lib_src/eas_ctype.h deleted file mode 100644 index 8503870..0000000 --- a/arm-hybrid-22k/lib_src/eas_ctype.h +++ /dev/null @@ -1,41 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_ctype.h - * - * Contents and purpose: - * This is a replacement for the CRT ctype.h functions. These - * functions are currently ASCII only, but eventually, we will want - * to support wide-characters for localization. - * - * Copyright (c) 2005 Sonic Network Inc. - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 429 $ - * $Date: 2006-10-19 23:50:15 -0700 (Thu, 19 Oct 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_CTYPE_H -#define _EAS_CTYPE_H - -EAS_INLINE EAS_I8 IsDigit (EAS_I8 c) { return ((c >= '0') && (c <= '9')); } -EAS_INLINE EAS_I8 IsSpace (EAS_I8 c) { return (((c >= 9) && (c <= 13)) || (c == ' ')); } -EAS_INLINE EAS_I8 ToUpper (EAS_I8 c) { if ((c >= 'a') && (c <= 'z')) return c & ~0x20; else return c; } -EAS_INLINE EAS_I8 ToLower (EAS_I8 c) { if ((c >= 'A') && (c <= 'Z')) return c |= 0x20; else return c; } - -#endif - diff --git a/arm-hybrid-22k/lib_src/eas_data.c b/arm-hybrid-22k/lib_src/eas_data.c deleted file mode 100644 index bb60ef2..0000000 --- a/arm-hybrid-22k/lib_src/eas_data.c +++ /dev/null @@ -1,37 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_data.c - * - * Contents and purpose: - * Contains a data allocation for synthesizer - * - * Copyright Sonic Network Inc. 2004 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 547 $ - * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $ - *---------------------------------------------------------------------------- -*/ - -// includes -#include "eas_data.h" - -// globals -S_EAS_DATA eas_Data; -S_VOICE_MGR eas_Synth; -S_SYNTH eas_MIDI; - diff --git a/arm-hybrid-22k/lib_src/eas_data.h b/arm-hybrid-22k/lib_src/eas_data.h deleted file mode 100644 index 0a47d04..0000000 --- a/arm-hybrid-22k/lib_src/eas_data.h +++ /dev/null @@ -1,131 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_data.h - * - * Contents and purpose: - * This header defines all types, to support dynamic allocation of the - * memory resources needed for persistent EAS data. - * - * Copyright 2004 Sonic Network Inc. - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 842 $ - * $Date: 2007-08-23 14:32:31 -0700 (Thu, 23 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_DATA_H -#define _EAS_DATA_H - -#include "eas_types.h" -#include "eas_synthcfg.h" -#include "eas.h" -#include "eas_audioconst.h" -#include "eas_sndlib.h" -#include "eas_pcm.h" -#include "eas_pcmdata.h" -#include "eas_synth.h" -#include "eas_miditypes.h" -#include "eas_effects.h" - -#ifdef AUX_MIXER -#include "eas_auxmixdata.h" -#endif - -#ifdef JET_INTERFACE -#include "jet.h" -#endif - -#ifdef _METRICS_ENABLED -#include "eas_perf.h" -#endif - -#ifndef MAX_NUMBER_STREAMS -#define MAX_NUMBER_STREAMS 4 -#endif - -/* flags for S_EAS_STREAM */ -#define STREAM_FLAGS_PARSED 1 -#define STREAM_FLAGS_PAUSE 2 -#define STREAM_FLAGS_LOCATE 4 -#define STREAM_FLAGS_RESUME 8 - -/* structure for parsing a stream */ -typedef struct s_eas_stream_tag -{ - void *pParserModule; - EAS_U32 time; - EAS_U32 frameLength; - EAS_I32 repeatCount; - EAS_VOID_PTR handle; - EAS_U8 volume; - EAS_BOOL8 streamFlags; -} S_EAS_STREAM; - -/* default master volume is -10dB */ -#define DEFAULT_VOLUME 90 -#define DEFAULT_STREAM_VOLUME 100 -#define DEFAULT_STREAM_GAIN 14622 - -/* 10 dB of boost available for individual parsers */ -#define STREAM_VOLUME_HEADROOM 10 - -/* amalgamated persistent data type */ -typedef struct s_eas_data_tag -{ -#ifdef _CHECKED_BUILD - EAS_U32 handleCheck; -#endif - EAS_HW_DATA_HANDLE hwInstData; - - S_EFFECTS_MODULE effectsModules[NUM_EFFECTS_MODULES]; - -#ifdef _METRICS_ENABLED - S_METRICS_INTERFACE *pMetricsModule; - EAS_VOID_PTR pMetricsData; -#endif - - EAS_I32 *pMixBuffer; - EAS_PCM *pOutputAudioBuffer; - -#ifdef AUX_MIXER - S_EAS_AUX_MIXER auxMixer; -#endif - -#ifdef _MAXIMIZER_ENABLED - EAS_VOID_PTR pMaximizerData; -#endif - - S_EAS_STREAM streams[MAX_NUMBER_STREAMS]; - - S_PCM_STATE *pPCMStreams; - - S_VOICE_MGR *pVoiceMgr; - -#ifdef JET_INTERFACE - JET_DATA_HANDLE jetHandle; -#endif - - EAS_U32 renderTime; - EAS_I16 masterGain; - EAS_U8 masterVolume; - EAS_BOOL8 staticMemoryModel; - EAS_BOOL8 searchHeaderFlag; -} S_EAS_DATA; - -#endif - diff --git a/arm-hybrid-22k/lib_src/eas_effects.h b/arm-hybrid-22k/lib_src/eas_effects.h deleted file mode 100644 index 01e64c0..0000000 --- a/arm-hybrid-22k/lib_src/eas_effects.h +++ /dev/null @@ -1,61 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_effects.h - * - * Contents and purpose: - * Defines a generic effects interface. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_EFFECTS_H -#define _EAS_EFFECTS_H - -#include "eas_types.h" - -typedef struct -{ - EAS_RESULT (*pfInit)(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData); - void (*pfProcess)(EAS_VOID_PTR pInstData, EAS_PCM *in, EAS_PCM *out, EAS_I32 numSamples); - EAS_RESULT (*pfShutdown)(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData); - EAS_RESULT (*pFGetParam)(EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue); - EAS_RESULT (*pFSetParam)(EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value); -} S_EFFECTS_INTERFACE; - -typedef struct -{ - EAS_RESULT (*pfInit)(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData); - void (*pfProcess)(EAS_VOID_PTR pInstData, EAS_I32 *in, EAS_I32 *out, EAS_I32 numSamples); - EAS_RESULT (*pfShutdown)(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData); - EAS_RESULT (*pFGetParam)(EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue); - EAS_RESULT (*pFSetParam)(EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value); -} S_EFFECTS32_INTERFACE; - -/* mixer instance data */ -typedef struct -{ - S_EFFECTS_INTERFACE *effect; - EAS_VOID_PTR effectData; -} S_EFFECTS_MODULE; - -#endif /* end _EAS_EFFECTS_H */ - diff --git a/arm-hybrid-22k/lib_src/eas_fmengine.c b/arm-hybrid-22k/lib_src/eas_fmengine.c deleted file mode 100644 index 9c3da66..0000000 --- a/arm-hybrid-22k/lib_src/eas_fmengine.c +++ /dev/null @@ -1,785 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_fmengine.c - * - * Contents and purpose: - * Implements the low-level FM synthesizer functions. - * - * Copyright Sonic Network Inc. 2004, 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 795 $ - * $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -/* includes */ -#include "eas_types.h" -#include "eas_math.h" -#include "eas_audioconst.h" -#include "eas_fmengine.h" - -#if defined(EAS_FM_SYNTH) || defined(EAS_HYBRID_SYNTH) || defined(EAS_SPLIT_HYBRID_SYNTH) || defined(EAS_SPLIT_FM_SYNTH) -#include "eas_data.h" -#endif - -/* externals */ -extern const EAS_I16 sineTable[]; -extern const EAS_U8 fmScaleTable[16]; - -// saturation constants for 32-bit to 16-bit conversion -#define _EAS_MAX_OUTPUT 32767 -#define _EAS_MIN_OUTPUT -32767 - -static S_FM_ENG_VOICE voices[NUM_FM_VOICES]; - -/* local prototypes */ -void FM_SynthMixVoice (S_FM_ENG_VOICE *p, EAS_U16 gainTarget, EAS_I32 numSamplesToAdd, EAS_PCM *pInputBuffer, EAS_I32 *pBuffer); - -/* used in development environment */ -#if defined(_SATURATION_MONITOR) -static EAS_BOOL bSaturated = EAS_FALSE; - -/*---------------------------------------------------------------------------- - * FM_CheckSaturation() - *---------------------------------------------------------------------------- - * Purpose: - * Allows the sound development tool to check for saturation at the voice - * level. Useful for tuning the level controls. - * - * Inputs: - * - * Outputs: - * Returns true if saturation has occurred since the last time the function - * was called. - * - * Side Effects: - * Resets the saturation flag - *---------------------------------------------------------------------------- -*/ -EAS_BOOL FM_CheckSaturation () -{ - EAS_BOOL bTemp; - bTemp = bSaturated; - bSaturated = EAS_FALSE; - return bTemp; -} -#endif - -/*---------------------------------------------------------------------------- - * FM_Saturate() - *---------------------------------------------------------------------------- - * Purpose: - * This inline function saturates a 32-bit number to 16-bits - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * Returns a 16-bit integer - *---------------------------------------------------------------------------- -*/ -EAS_INLINE EAS_I16 FM_Saturate (EAS_I32 nValue) -{ - if (nValue > _EAS_MAX_OUTPUT) - { -#if defined(_SATURATION_MONITOR) - bSaturated = EAS_TRUE; -#endif - return _EAS_MAX_OUTPUT; - } - if (nValue < _EAS_MIN_OUTPUT) - { -#if defined(_SATURATION_MONITOR) - bSaturated = EAS_TRUE; -#endif - return _EAS_MIN_OUTPUT; - } - return (EAS_I16) nValue; -} - -/*---------------------------------------------------------------------------- - * FM_Noise() - *---------------------------------------------------------------------------- - * Purpose: - * A 31-bit low-cost linear congruential PRNG algorithm used to - * generate noise. - * - * Inputs: - * pnSeed - pointer to 32-bit PRNG seed - * - * Outputs: - * Returns a 16-bit integer - *---------------------------------------------------------------------------- -*/ -EAS_INLINE EAS_I16 FM_Noise (EAS_U32 *pnSeed) -{ - *pnSeed = *pnSeed * 214013L + 2531011L; - return (EAS_I16) ((*pnSeed >> 15) & 0xffff); -} - -/*---------------------------------------------------------------------------- - * FM_PhaseInc() - *---------------------------------------------------------------------------- - * Purpose: - * Transform pitch cents to linear phase increment - * - * Inputs: - * nCents - measured in cents - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * nResult - int.frac result (where frac has NUM_DENTS_FRAC_BITS) - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_I32 FM_PhaseInc (EAS_I32 nCents) -{ - EAS_I32 nDents; - EAS_I32 nExponentInt, nExponentFrac; - EAS_I32 nTemp1, nTemp2; - EAS_I32 nResult; - - /* convert cents to dents */ - nDents = FMUL_15x15(nCents, CENTS_TO_DENTS); - nExponentInt = GET_DENTS_INT_PART(nDents) + (32 - SINE_TABLE_SIZE_IN_BITS - NUM_EG1_FRAC_BITS); - nExponentFrac = GET_DENTS_FRAC_PART(nDents); - - /* implement 2^(fracPart) as a power series */ - nTemp1 = GN2_TO_X2 + MULT_DENTS_COEF(nExponentFrac, GN2_TO_X3); - nTemp2 = GN2_TO_X1 + MULT_DENTS_COEF(nExponentFrac, nTemp1); - nTemp1 = GN2_TO_X0 + MULT_DENTS_COEF(nExponentFrac, nTemp2); - - /* - implement 2^(intPart) as - a left shift for intPart >= 0 or - a left shift for intPart < 0 - */ - if (nExponentInt >= 0) - { - /* left shift for positive exponents */ - /*lint -e{703} */ - nResult = nTemp1 << nExponentInt; - } - else - { - /* right shift for negative exponents */ - nExponentInt = -nExponentInt; - nResult = nTemp1 >> nExponentInt; - } - - return nResult; -} - -#if (NUM_OUTPUT_CHANNELS == 2) -/*---------------------------------------------------------------------------- - * FM_CalculatePan() - *---------------------------------------------------------------------------- - * Purpose: - * Assign the left and right gain values corresponding to the given pan value. - * - * Inputs: - * psVoice - ptr to the voice we have assigned for this channel - * psArticulation - ptr to this voice's articulation - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * the given voice's m_nGainLeft and m_nGainRight are assigned - *---------------------------------------------------------------------------- -*/ -static void FM_CalculatePan (EAS_I16 pan, EAS_U16 *pGainLeft, EAS_U16 *pGainRight) -{ - EAS_I32 nTemp; - EAS_INT nNetAngle; - - /* - Implement the following - sin(x) = (2-4*c)*x^2 + c + x - cos(x) = (2-4*c)*x^2 + c - x - - where c = 1/sqrt(2) - using the a0 + x*(a1 + x*a2) approach - */ - - /* - Get the Midi CC10 pan value for this voice's channel - convert the pan value to an "angle" representation suitable for - our sin, cos calculator. This representation is NOT necessarily the same - as the transform in the GM manuals because of our sin, cos calculator. - "angle" = (CC10 - 64)/128 - */ - /*lint -e{703} */ - nNetAngle = ((EAS_I32) pan) << (NUM_EG1_FRAC_BITS -7); - - /* calculate sin */ - nTemp = EG1_ONE + FMUL_15x15(COEFF_PAN_G2, nNetAngle); - nTemp = COEFF_PAN_G0 + FMUL_15x15(nTemp, nNetAngle); - - if (nTemp > SYNTH_FULL_SCALE_EG1_GAIN) - nTemp = SYNTH_FULL_SCALE_EG1_GAIN; - else if (nTemp < 0) - nTemp = 0; - - *pGainRight = (EAS_U16) nTemp; - - /* calculate cos */ - nTemp = -EG1_ONE + FMUL_15x15(COEFF_PAN_G2, nNetAngle); - nTemp = COEFF_PAN_G0 + FMUL_15x15(nTemp, nNetAngle); - - if (nTemp > SYNTH_FULL_SCALE_EG1_GAIN) - nTemp = SYNTH_FULL_SCALE_EG1_GAIN; - else if (nTemp < 0) - nTemp = 0; - - *pGainLeft = (EAS_U16) nTemp; -} -#endif /* #if (NUM_OUTPUT_CHANNELS == 2) */ - -/*---------------------------------------------------------------------------- - * FM_Operator() - *---------------------------------------------------------------------------- - * Purpose: - * Synthesizes a buffer of samples based on passed parameters. - * - * Inputs: - * nNumSamplesToAdd - number of samples to synthesize - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void FM_Operator ( - S_FM_ENG_OPER *p, - EAS_I32 numSamplesToAdd, - EAS_PCM *pBuffer, - EAS_PCM *pModBuffer, - EAS_BOOL mix, - EAS_U16 gainTarget, - EAS_I16 pitch, - EAS_U8 feedback, - EAS_I16 *pLastOutput) -{ - EAS_I32 gain; - EAS_I32 gainInc; - EAS_U32 phase; - EAS_U32 phaseInc; - EAS_U32 phaseTemp; - EAS_I32 temp; - EAS_I32 temp2; - - /* establish local gain variable */ - gain = (EAS_I32) p->gain << 16; - - /* calculate gain increment */ - /*lint -e{703} use shift for performance */ - gainInc = ((EAS_I32) gainTarget - (EAS_I32) p->gain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS); - - /* establish local phase variables */ - phase = p->phase; - - /* calculate the new phase increment */ - phaseInc = (EAS_U32) FM_PhaseInc(pitch); - - /* restore final output from previous frame for feedback loop */ - if (pLastOutput) - temp = *pLastOutput; - else - temp = 0; - - /* generate a buffer of samples */ - while (numSamplesToAdd--) - { - - /* incorporate modulation */ - if (pModBuffer) - { - /*lint -e{701} use shift for performance */ - temp = *pModBuffer++ << FM_MODULATOR_INPUT_SHIFT; - } - - /* incorporate feedback */ - else - { - /*lint -e{703} use shift for performance */ - temp = (temp * (EAS_I32) feedback) << FM_FEEDBACK_SHIFT; - } - - /*lint -e{737} */ - phaseTemp = phase + temp; - - /* fetch sample from wavetable */ - temp = sineTable[phaseTemp >> (32 - SINE_TABLE_SIZE_IN_BITS)]; - - /* increment operator phase */ - phase += phaseInc; - - /* internal gain for modulation effects */ - temp = FMUL_15x15(temp, (gain >> 16)); - - /* output gain calculation */ - temp2 = FMUL_15x15(temp, p->outputGain); - - /* saturating add to buffer */ - if (mix) - { - temp2 += *pBuffer; - *pBuffer++ = FM_Saturate(temp2); - } - - /* output to buffer */ - else - *pBuffer++ = (EAS_I16) temp2; - - /* increment gain */ - gain += gainInc; - - } - - /* save phase and gain */ - p->phase = phase; - p->gain = gainTarget; - - /* save last output for feedback in next frame */ - if (pLastOutput) - *pLastOutput = (EAS_I16) temp; -} - -/*---------------------------------------------------------------------------- - * FM_NoiseOperator() - *---------------------------------------------------------------------------- - * Purpose: - * Synthesizes a buffer of samples based on passed parameters. - * - * Inputs: - * nNumSamplesToAdd - number of samples to synthesize - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void FM_NoiseOperator ( - S_FM_ENG_OPER *p, - EAS_I32 numSamplesToAdd, - EAS_PCM *pBuffer, - EAS_BOOL mix, - EAS_U16 gainTarget, - EAS_U8 feedback, - EAS_I16 *pLastOutput) -{ - EAS_I32 gain; - EAS_I32 gainInc; - EAS_U32 phase; - EAS_I32 temp; - EAS_I32 temp2; - - /* establish local gain variable */ - gain = (EAS_I32) p->gain << 16; - - /* calculate gain increment */ - /*lint -e{703} use shift for performance */ - gainInc = ((EAS_I32) gainTarget - (EAS_I32) p->gain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS); - - /* establish local phase variables */ - phase = p->phase; - - /* establish local phase variables */ - phase = p->phase; - - /* recall last sample for filter Z-1 term */ - temp = 0; - if (pLastOutput) - temp = *pLastOutput; - - /* generate a buffer of samples */ - while (numSamplesToAdd--) - { - - /* if using filter */ - if (pLastOutput) - { - /* use PRNG for noise */ - temp2 = FM_Noise(&phase); - - /*lint -e{704} use shift for performance */ - temp += ((temp2 -temp) * feedback) >> 8; - } - else - { - temp = FM_Noise(&phase); - } - - /* internal gain for modulation effects */ - temp2 = FMUL_15x15(temp, (gain >> 16)); - - /* output gain calculation */ - temp2 = FMUL_15x15(temp2, p->outputGain); - - /* saturating add to buffer */ - if (mix) - { - temp2 += *pBuffer; - *pBuffer++ = FM_Saturate(temp2); - } - - /* output to buffer */ - else - *pBuffer++ = (EAS_I16) temp2; - - /* increment gain */ - gain += gainInc; - - } - - /* save phase and gain */ - p->phase = phase; - p->gain = gainTarget; - - /* save last output for feedback in next frame */ - if (pLastOutput) - *pLastOutput = (EAS_I16) temp; -} - -/*---------------------------------------------------------------------------- - * FM_ConfigVoice() - *---------------------------------------------------------------------------- - * Purpose: - * Receives parameters to start a new voice. - * - * Inputs: - * voiceNum - voice number to start - * vCfg - configuration data - * pMixBuffer - pointer to host supplied buffer - * - * Outputs: - * - * Side Effects: - * - * Notes: - * pFrameBuffer is not used in the test version, but is passed as a - * courtesy to split architecture implementations. It can be used as - * as pointer to the interprocessor communications buffer when the - * synthesis parameters are passed off to a DSP for synthesis. - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pFrameBuffer) pFrameBuffer not used in test version - see above */ -void FM_ConfigVoice (EAS_I32 voiceNum, S_FM_VOICE_CONFIG *vCfg, EAS_FRAME_BUFFER_HANDLE pFrameBuffer) -{ - S_FM_ENG_VOICE *pVoice; - EAS_INT i; - - /* establish pointer to voice data */ - pVoice = &voices[voiceNum]; - - /* save data */ - pVoice->feedback = vCfg->feedback; - pVoice->flags = vCfg->flags; - pVoice->voiceGain = vCfg->voiceGain; - - /* initialize Z-1 terms */ - pVoice->op1Out = 0; - pVoice->op3Out = 0; - - /* initialize operators */ - for (i = 0; i < 4; i++) - { - /* save operator data */ - pVoice->oper[i].gain = vCfg->gain[i]; - pVoice->oper[i].outputGain = vCfg->outputGain[i]; - pVoice->oper[i].outputGain = vCfg->outputGain[i]; - - /* initalize operator */ - pVoice->oper[i].phase = 0; - } - - /* calculate pan */ -#if NUM_OUTPUT_CHANNELS == 2 - FM_CalculatePan(vCfg->pan, &pVoice->gainLeft, &pVoice->gainRight); -#endif -} - -/*---------------------------------------------------------------------------- - * FM_ProcessVoice() - *---------------------------------------------------------------------------- - * Purpose: - * Synthesizes a buffer of samples based on calculated parameters. - * - * Inputs: - * nNumSamplesToAdd - number of samples to synthesize - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - * Notes: - * pOut is not used in the test version, but is passed as a - * courtesy to split architecture implementations. It can be used as - * as pointer to the interprocessor communications buffer when the - * synthesis parameters are passed off to a DSP for synthesis. - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pOut) pOut not used in test version - see above */ -void FM_ProcessVoice ( - EAS_I32 voiceNum, - S_FM_VOICE_FRAME *pFrame, - EAS_I32 numSamplesToAdd, - EAS_PCM *pTempBuffer, - EAS_PCM *pBuffer, - EAS_I32 *pMixBuffer, - EAS_FRAME_BUFFER_HANDLE pFrameBuffer) -{ - S_FM_ENG_VOICE *p; - EAS_PCM *pOutBuf; - EAS_PCM *pMod; - EAS_BOOL mix; - EAS_U8 feedback1; - EAS_U8 feedback3; - EAS_U8 mode; - - /* establish pointer to voice data */ - p = &voices[voiceNum]; - mode = p->flags & 0x07; - - /* lookup feedback values */ - feedback1 = fmScaleTable[p->feedback >> 4]; - feedback3 = fmScaleTable[p->feedback & 0x0f]; - - /* operator 3 is on output bus in modes 0, 1, and 3 */ - if ((mode == 0) || (mode == 1) || (mode == 3)) - pOutBuf = pBuffer; - else - pOutBuf = pTempBuffer; - - if (p->flags & FLAG_FM_ENG_VOICE_OP3_NOISE) - { - FM_NoiseOperator( - p->oper + 2, - numSamplesToAdd, - pOutBuf, - EAS_FALSE, - pFrame->gain[2], - feedback3, - &p->op3Out); - } - else - { - FM_Operator( - p->oper + 2, - numSamplesToAdd, - pOutBuf, - 0, - EAS_FALSE, - pFrame->gain[2], - pFrame->pitch[2], - feedback3, - &p->op3Out); - } - - /* operator 4 is on output bus in modes 0, 1, and 2 */ - if (mode < 3) - pOutBuf = pBuffer; - else - pOutBuf = pTempBuffer; - - /* operator 4 is modulated in modes 2, 4, and 5 */ - if ((mode == 2) || (mode == 4) || (mode == 5)) - pMod = pTempBuffer; - else - pMod = 0; - - /* operator 4 is in mix mode in modes 0 and 1 */ - mix = (mode < 2); - - if (p->flags & FLAG_FM_ENG_VOICE_OP4_NOISE) - { - FM_NoiseOperator( - p->oper + 3, - numSamplesToAdd, - pOutBuf, - mix, - pFrame->gain[3], - 0, - 0); - } - else - { - FM_Operator( - p->oper + 3, - numSamplesToAdd, - pOutBuf, - pMod, - mix, - pFrame->gain[3], - pFrame->pitch[3], - 0, - 0); - } - - /* operator 1 is on output bus in mode 0 */ - if (mode == 0) - pOutBuf = pBuffer; - else - pOutBuf = pTempBuffer; - - /* operator 1 is modulated in modes 3 and 4 */ - if ((mode == 3) || (mode == 4)) - pMod = pTempBuffer; - else - pMod = 0; - - /* operator 1 is in mix mode in modes 0 and 5 */ - mix = ((mode == 0) || (mode == 5)); - - if (p->flags & FLAG_FM_ENG_VOICE_OP1_NOISE) - { - FM_NoiseOperator( - p->oper, - numSamplesToAdd, - pOutBuf, - mix, - pFrame->gain[0], - feedback1, - &p->op1Out); - } - else - { - FM_Operator( - p->oper, - numSamplesToAdd, - pOutBuf, - pMod, - mix, - pFrame->gain[0], - pFrame->pitch[0], - feedback1, - &p->op1Out); - } - - /* operator 2 is modulated in all modes except 0 */ - if (mode != 0) - pMod = pTempBuffer; - else - pMod = 0; - - /* operator 1 is in mix mode in modes 0 -3 */ - mix = (mode < 4); - - if (p->flags & FLAG_FM_ENG_VOICE_OP2_NOISE) - { - FM_NoiseOperator( - p->oper + 1, - numSamplesToAdd, - pBuffer, - mix, - pFrame->gain[1], - 0, - 0); - } - else - { - FM_Operator( - p->oper + 1, - numSamplesToAdd, - pBuffer, - pMod, - mix, - pFrame->gain[1], - pFrame->pitch[1], - 0, - 0); - } - - /* mix voice output to synthesizer output buffer */ - FM_SynthMixVoice(p, pFrame->voiceGain, numSamplesToAdd, pBuffer, pMixBuffer); -} - -/*---------------------------------------------------------------------------- - * FM_SynthMixVoice() - *---------------------------------------------------------------------------- - * Purpose: - * Mixes the voice output buffer into the final mix using an anti-zipper - * filter. - * - * Inputs: - * nNumSamplesToAdd - number of samples to synthesize - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void FM_SynthMixVoice(S_FM_ENG_VOICE *p, EAS_U16 nGainTarget, EAS_I32 numSamplesToAdd, EAS_PCM *pInputBuffer, EAS_I32 *pBuffer) -{ - EAS_I32 nGain; - EAS_I32 nGainInc; - EAS_I32 nTemp; - - /* restore previous gain */ - /*lint -e{703} */ - nGain = (EAS_I32) p->voiceGain << 16; - - /* calculate gain increment */ - /*lint -e{703} */ - nGainInc = ((EAS_I32) nGainTarget - (EAS_I32) p->voiceGain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS); - - /* mix the output buffer */ - while (numSamplesToAdd--) - { - /* output gain calculation */ - nTemp = *pInputBuffer++; - - /* sum to output buffer */ -#if (NUM_OUTPUT_CHANNELS == 2) - - /*lint -e{704} */ - nTemp = ((EAS_I32) nTemp * (nGain >> 16)) >> FM_GAIN_SHIFT; - - /*lint -e{704} */ - { - EAS_I32 nTemp2; - nTemp = nTemp >> FM_STEREO_PRE_GAIN_SHIFT; - nTemp2 = (nTemp * p->gainLeft) >> FM_STEREO_POST_GAIN_SHIFT; - *pBuffer++ += nTemp2; - nTemp2 = (nTemp * p->gainRight) >> FM_STEREO_POST_GAIN_SHIFT; - *pBuffer++ += nTemp2; - } -#else - /*lint -e{704} */ - nTemp = ((EAS_I32) nTemp * (nGain >> 16)) >> FM_MONO_GAIN_SHIFT; - *pBuffer++ += nTemp; -#endif - - /* increment gain for anti-zipper filter */ - nGain += nGainInc; - } - - /* save gain */ - p->voiceGain = nGainTarget; -} - diff --git a/arm-hybrid-22k/lib_src/eas_fmengine.h b/arm-hybrid-22k/lib_src/eas_fmengine.h deleted file mode 100644 index 4ddc12b..0000000 --- a/arm-hybrid-22k/lib_src/eas_fmengine.h +++ /dev/null @@ -1,121 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_fmengine.h - * - * Contents and purpose: - * Declarations, interfaces, and prototypes for FM synthesize low-level. - * - * Copyright Sonic Network Inc. 2004 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 664 $ - * $Date: 2007-04-25 13:11:22 -0700 (Wed, 25 Apr 2007) $ - *---------------------------------------------------------------------------- -*/ - -/* sentinel */ -#ifndef _FMENGINE_H -#define _FMENGINE_H - -/* check for split architecture */ -#if defined (EAS_SPLIT_HYBRID_SYNTH) || defined(EAS_SPLIT_FM_SYNTH) -#define FM_OFFBOARD -#endif - -/* output level to mix buffer (3 = -24dB) */ -#define FM_GAIN_SHIFT 3 -#define FM_MONO_GAIN_SHIFT 9 - -/* voice output level for stereo 15 = +6dB */ -#define FM_STEREO_PRE_GAIN_SHIFT 11 -#define FM_STEREO_POST_GAIN_SHIFT 10 - -/* modulator input level shift (21 = -30dB) */ -#define FM_MODULATOR_INPUT_SHIFT 21 - -/* feedback control level shift (7 = 0dB) */ -#define FM_FEEDBACK_SHIFT 7 - -/* synth final output level */ -#define SYNTH_POST_GAIN_SHIFT 14 - -/* LFO modulation to gain control */ -#define FM_LFO_GAIN_SHIFT 12 - -/* sine table is always a power of 2 - saves cycles in inner loop */ -#define SINE_TABLE_SIZE_IN_BITS 11 -#define SINE_TABLE_SIZE 2048 - -/* operator structure for FM engine */ -typedef struct -{ - EAS_U32 phase; /* current waveform phase */ - EAS_U16 gain; /* current internal gain */ - EAS_U16 outputGain; /* current output gain */ -} S_FM_ENG_OPER; - -typedef struct -{ - S_FM_ENG_OPER oper[4]; /* operator data */ - EAS_I16 op1Out; /* op1 output for feedback loop */ - EAS_I16 op3Out; /* op3 output for feedback loop */ - EAS_U16 voiceGain; /* LFO + channel parameters */ -#if (NUM_OUTPUT_CHANNELS == 2) - EAS_U16 gainLeft; /* left gain multiplier */ - EAS_U16 gainRight; /* right gain multiplier */ -#endif - EAS_U8 flags; /* mode bits and noise waveform flags */ - EAS_U8 feedback; /* feedback for Op1 and Op3 */ -} S_FM_ENG_VOICE; - -typedef struct -{ - EAS_U16 gain[4]; /* initial operator gain value */ - EAS_U16 outputGain[4]; /* initial operator output gain value */ - EAS_U16 voiceGain; /* initial voice gain */ - EAS_U8 flags; /* mode bits and noise waveform flags */ - EAS_U8 feedback; /* feedback for Op1 and Op3 */ -#if (NUM_OUTPUT_CHANNELS == 2) - EAS_I8 pan; /* pan value +/-64 */ -#endif -} S_FM_VOICE_CONFIG; - -typedef struct -{ - EAS_U16 gain[4]; /* new operator gain value */ - EAS_I16 pitch[4]; /* new pitch value */ - EAS_U16 voiceGain; /* new voice gain */ -} S_FM_VOICE_FRAME; - -/* bit definitions for S_FM_ENG_VOICE.flags */ -#define FLAG_FM_ENG_VOICE_OP1_NOISE 0x10 /* operator 1 source is PRNG */ -#define FLAG_FM_ENG_VOICE_OP2_NOISE 0x20 /* operator 2 source is PRNG */ -#define FLAG_FM_ENG_VOICE_OP3_NOISE 0x40 /* operator 3 source is PRNG */ -#define FLAG_FM_ENG_VOICE_OP4_NOISE 0x80 /* operator 4 source is PRNG */ - -#ifdef FM_OFFBOARD -extern EAS_BOOL FM_StartFrame (EAS_FRAME_BUFFER_HANDLE pFrameBuffer); -extern EAS_BOOL FM_EndFrame (EAS_FRAME_BUFFER_HANDLE pFrameBuffe, EAS_I32 *pMixBuffer, EAS_I16 masterGain); -#endif - -/* FM engine prototypes */ -extern void FM_ConfigVoice (EAS_I32 voiceNum, S_FM_VOICE_CONFIG *vCfg, EAS_FRAME_BUFFER_HANDLE pFrameBuffer); -extern void FM_ProcessVoice (EAS_I32 voiceNum, S_FM_VOICE_FRAME *pFrame, EAS_I32 numSamplesToAdd, EAS_PCM *pTempBuffer, EAS_PCM *pBuffer, EAS_I32 *pMixBuffer, EAS_FRAME_BUFFER_HANDLE pFrameBuffer); - -#endif -/* #ifndef _FMENGINE_H */ - diff --git a/arm-hybrid-22k/lib_src/eas_fmsynth.c b/arm-hybrid-22k/lib_src/eas_fmsynth.c deleted file mode 100644 index 83f0087..0000000 --- a/arm-hybrid-22k/lib_src/eas_fmsynth.c +++ /dev/null @@ -1,910 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * fmsynth.c - * - * Contents and purpose: - * Implements the high-level FM synthesizer functions. - * - * Copyright Sonic Network Inc. 2004 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 795 $ - * $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -// includes -#include "eas_host.h" -#include "eas_report.h" - -#include "eas_data.h" -#include "eas_synth_protos.h" -#include "eas_audioconst.h" -#include "eas_fmengine.h" -#include "eas_math.h" - -/* option sanity check */ -#ifdef _REVERB -#error "No reverb for FM synthesizer" -#endif -#ifdef _CHORUS -#error "No chorus for FM synthesizer" -#endif - -/* - * WARNING: These macros can cause unwanted side effects. Use them - * with care. For example, min(x++,y++) will cause either x or y to be - * incremented twice. - */ -#define min(a,b) ((a) < (b) ? (a) : (b)) -#define max(a,b) ((a) > (b) ? (a) : (b)) - -/* pivot point for keyboard scalars */ -#define EG_SCALE_PIVOT_POINT 64 -#define KEY_SCALE_PIVOT_POINT 36 - -/* This number is the negative of the frequency of the note (in cents) of - * the sine wave played at unity. The number can be calculated as follows: - * - * MAGIC_NUMBER = 1200 * log(base2) (SINE_TABLE_SIZE * 8.175798916 / SAMPLE_RATE) - * - * 8.17578 is a reference to the frequency of MIDI note 0 - */ -#if defined (_SAMPLE_RATE_8000) -#define MAGIC_NUMBER 1279 -#elif defined (_SAMPLE_RATE_16000) -#define MAGIC_NUMBER 79 -#elif defined (_SAMPLE_RATE_20000) -#define MAGIC_NUMBER -308 -#elif defined (_SAMPLE_RATE_22050) -#define MAGIC_NUMBER -477 -#elif defined (_SAMPLE_RATE_24000) -#define MAGIC_NUMBER -623 -#elif defined (_SAMPLE_RATE_32000) -#define MAGIC_NUMBER -1121 -#elif defined (_SAMPLE_RATE_44100) -#define MAGIC_NUMBER -1677 -#elif defined (_SAMPLE_RATE_48000) -#define MAGIC_NUMBER -1823 -#endif - -/* externs */ -extern const EAS_I16 fmControlTable[128]; -extern const EAS_U16 fmRateTable[256]; -extern const EAS_U16 fmAttackTable[16]; -extern const EAS_U8 fmDecayTable[16]; -extern const EAS_U8 fmReleaseTable[16]; -extern const EAS_U8 fmScaleTable[16]; - -/* local prototypes */ -/*lint -esym(715, pVoiceMgr) standard synthesizer interface - pVoiceMgr not used */ -static EAS_RESULT FM_Initialize (S_VOICE_MGR *pVoiceMgr) { return EAS_SUCCESS; } -static EAS_RESULT FM_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex); -static EAS_BOOL FM_UpdateVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32 numSamples); -static void FM_ReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum); -static void FM_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum); -static void FM_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum); -static void FM_UpdateChannel (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel); - - -/*---------------------------------------------------------------------------- - * Synthesizer interface - *---------------------------------------------------------------------------- -*/ -const S_SYNTH_INTERFACE fmSynth = -{ - FM_Initialize, - FM_StartVoice, - FM_UpdateVoice, - FM_ReleaseVoice, - FM_MuteVoice, - FM_SustainPedal, - FM_UpdateChannel -}; - -#ifdef FM_OFFBOARD -const S_FRAME_INTERFACE fmFrameInterface = -{ - FM_StartFrame, - FM_EndFrame -}; -#endif - -/*---------------------------------------------------------------------------- - * inline functions - *---------------------------------------------------------------------------- - */ -EAS_INLINE S_FM_VOICE *GetFMVoicePtr (S_VOICE_MGR *pVoiceMgr, EAS_INT voiceNum) -{ - return &pVoiceMgr->fmVoices[voiceNum]; -} -EAS_INLINE S_SYNTH_CHANNEL *GetChannelPtr (S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice) -{ - return &pSynth->channels[pVoice->channel & 15]; -} -EAS_INLINE const S_FM_REGION *GetFMRegionPtr (S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice) -{ -#ifdef _SECONDARY_SYNTH - return &pSynth->pEAS->pFMRegions[pVoice->regionIndex & REGION_INDEX_MASK]; -#else - return &pSynth->pEAS->pFMRegions[pVoice->regionIndex]; -#endif -} - -/*---------------------------------------------------------------------------- - * FM_SynthIsOutputOperator - *---------------------------------------------------------------------------- - * Purpose: - * Returns true if the operator is a direct output and not muted - * - * Inputs: - * - * Outputs: - * Returns boolean - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL FM_SynthIsOutputOperator (const S_FM_REGION *pRegion, EAS_INT operIndex) -{ - - /* see if voice is muted */ - if ((pRegion->oper[operIndex].gain & 0xfc) == 0) - return 0; - - /* check based on mode */ - switch (pRegion->region.keyGroupAndFlags & 7) - { - - /* mode 0 - all operators are external */ - case 0: - return EAS_TRUE; - - /* mode 1 - operators 1-3 are external */ - case 1: - if (operIndex != 0) - return EAS_TRUE; - break; - - /* mode 2 - operators 1 & 3 are external */ - case 2: - if ((operIndex == 1) || (operIndex == 3)) - return EAS_TRUE; - break; - - /* mode 2 - operators 1 & 2 are external */ - case 3: - if ((operIndex == 1) || (operIndex == 2)) - return EAS_TRUE; - break; - - /* modes 4 & 5 - operator 1 is external */ - case 4: - case 5: - if (operIndex == 1) - return EAS_TRUE; - break; - - default: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL,"Invalid voice mode: %d", - pRegion->region.keyGroupAndFlags & 7); */ } - } - - return EAS_FALSE; -} - -/*---------------------------------------------------------------------------- - * FM_CalcEGRate() - *---------------------------------------------------------------------------- - * Purpose: - * - * Inputs: - * nKeyNumber - MIDI note - * nLogRate - logarithmic scale rate from patch data - * nKeyScale - key scaling factor for this EG - * - * Outputs: - * 16-bit linear multiplier - *---------------------------------------------------------------------------- -*/ - -static EAS_U16 FM_CalcEGRate (EAS_U8 nKeyNumber, EAS_U8 nLogRate, EAS_U8 nEGScale) -{ - EAS_I32 temp; - - /* incorporate key scaling on release rate */ - temp = (EAS_I32) nLogRate << 7; - temp += ((EAS_I32) nKeyNumber - EG_SCALE_PIVOT_POINT) * (EAS_I32) nEGScale; - - /* saturate */ - temp = max(temp, 0); - temp = min(temp, 32767); - - /* look up in rate table */ - /*lint -e{704} use shift for performance */ - return fmRateTable[temp >> 8]; -} - -/*---------------------------------------------------------------------------- - * FM_ReleaseVoice() - *---------------------------------------------------------------------------- - * Purpose: - * The selected voice is being released. - * - * Inputs: - * psEASData - pointer to S_EAS_DATA - * pVoice - pointer to voice to release - * - * Outputs: - * None - *---------------------------------------------------------------------------- -*/ -static void FM_ReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum) -{ - EAS_INT operIndex; - const S_FM_REGION *pRegion; - S_FM_VOICE *pFMVoice; - - /* check to see if voice responds to NOTE-OFF's */ - pRegion = GetFMRegionPtr(pSynth, pVoice); - if (pRegion->region.keyGroupAndFlags & REGION_FLAG_ONE_SHOT) - return; - - /* set all envelopes to release state */ - pFMVoice = GetFMVoicePtr(pVoiceMgr, voiceNum); - for (operIndex = 0; operIndex < 4; operIndex++) - { - pFMVoice->oper[operIndex].envState = eFMEnvelopeStateRelease; - - /* incorporate key scaling on release rate */ - pFMVoice->oper[operIndex].envRate = FM_CalcEGRate( - pVoice->note, - fmReleaseTable[pRegion->oper[operIndex].velocityRelease & 0x0f], - fmScaleTable[pRegion->oper[operIndex].egKeyScale >> 4]); - } /* end for (operIndex = 0; operIndex < 4; operIndex++) */ -} - -/*---------------------------------------------------------------------------- - * FM_MuteVoice() - *---------------------------------------------------------------------------- - * Purpose: - * The selected voice is being muted. - * - * Inputs: - * pVoice - pointer to voice to release - * - * Outputs: - * None - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pSynth) standard interface, pVoiceMgr not used */ -static void FM_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum) -{ - S_FM_VOICE *pFMVoice; - - /* clear deferred action flags */ - pVoice->voiceFlags &= - ~(VOICE_FLAG_DEFER_MIDI_NOTE_OFF | - VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF | - VOICE_FLAG_DEFER_MUTE); - - /* set all envelopes to muted state */ - pFMVoice = GetFMVoicePtr(pVoiceMgr, voiceNum); - pFMVoice->oper[0].envState = eFMEnvelopeStateMuted; - pFMVoice->oper[1].envState = eFMEnvelopeStateMuted; - pFMVoice->oper[2].envState = eFMEnvelopeStateMuted; - pFMVoice->oper[3].envState = eFMEnvelopeStateMuted; -} - -/*---------------------------------------------------------------------------- - * FM_SustainPedal() - *---------------------------------------------------------------------------- - * Purpose: - * The selected voice is held due to sustain pedal - * - * Inputs: - * pVoice - pointer to voice to sustain - * - * Outputs: - * None - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pChannel) standard interface, pVoiceMgr not used */ -static void FM_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum) -{ - const S_FM_REGION *pRegion; - S_FM_VOICE *pFMVoice; - EAS_INT operIndex; - - pRegion = GetFMRegionPtr(pSynth, pVoice); - pFMVoice = GetFMVoicePtr(pVoiceMgr, voiceNum); - - /* check to see if any envelopes are above the sustain level */ - for (operIndex = 0; operIndex < 4; operIndex++) - { - - /* if level control or envelope gain is zero, skip this envelope */ - if (((pRegion->oper[operIndex].gain & 0xfc) == 0) || - (pFMVoice->oper[operIndex].envGain == 0)) - { - continue; - } - - /* if the envelope gain is above the sustain level, we need to catch this voice */ - if (pFMVoice->oper[operIndex].envGain >= ((EAS_U16) (pRegion->oper[operIndex].sustain & 0xfc) << 7)) - { - - /* reset envelope to decay state */ - pFMVoice->oper[operIndex].envState = eFMEnvelopeStateDecay; - - pFMVoice->oper[operIndex].envRate = FM_CalcEGRate( - pVoice->note, - fmDecayTable[pRegion->oper[operIndex].attackDecay & 0x0f], - fmScaleTable[pRegion->oper[operIndex].egKeyScale >> 4]); - - /* set voice to decay state */ - pVoice->voiceState = eVoiceStatePlay; - - /* set sustain flag */ - pVoice->voiceFlags |= VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF; - } - } /* end for (operIndex = 0; operIndex < 4; operIndex++) */ -} - -/*---------------------------------------------------------------------------- - * FM_StartVoice() - *---------------------------------------------------------------------------- - * Purpose: - * Assign the region for the given instrument using the midi key number - * and the RPN2 (coarse tuning) value. By using RPN2 as part of the - * region selection process, we reduce the amount a given sample has - * to be transposed by selecting the closest recorded root instead. - * - * This routine is the second half of SynthAssignRegion(). - * If the region was successfully found by SynthFindRegionIndex(), - * then assign the region's parameters to the voice. - * - * Setup and initialize the following voice parameters: - * m_nRegionIndex - * - * Inputs: - * pVoice - ptr to the voice we have assigned for this channel - * nRegionIndex - index of the region - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * success - could find and assign the region for this voice's note otherwise - * failure - could not find nor assign the region for this voice's note - * - * Side Effects: - * psSynthObject->m_sVoice[].m_nRegionIndex is assigned - * psSynthObject->m_sVoice[] parameters are assigned - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT FM_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex) -{ - S_FM_VOICE *pFMVoice; - S_SYNTH_CHANNEL *pChannel; - const S_FM_REGION *pRegion; - EAS_I32 temp; - EAS_INT operIndex; - - /* establish pointers to data */ - pVoice->regionIndex = regionIndex; - pFMVoice = GetFMVoicePtr(pVoiceMgr, voiceNum); - pChannel = GetChannelPtr(pSynth, pVoice); - pRegion = GetFMRegionPtr(pSynth, pVoice); - - /* update static channel parameters */ - if (pChannel->channelFlags & CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS) - FM_UpdateChannel(pVoiceMgr, pSynth, pVoice->channel & 15); - - /* init the LFO */ - pFMVoice->lfoValue = 0; - pFMVoice->lfoPhase = 0; - pFMVoice->lfoDelay = (EAS_U16) (fmScaleTable[pRegion->lfoFreqDelay & 0x0f] >> 1); - -#if (NUM_OUTPUT_CHANNELS == 2) - /* calculate pan gain values only if stereo output */ - /* set up panning only at note start */ - temp = (EAS_I32) pChannel->pan - 64; - temp += (EAS_I32) pRegion->pan; - if (temp < -64) - temp = -64; - if (temp > 64) - temp = 64; - pFMVoice->pan = (EAS_I8) temp; -#endif /* #if (NUM_OUTPUT_CHANNELS == 2) */ - - /* no samples have been synthesized for this note yet */ - pVoice->voiceFlags = VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET; - - /* initialize gain value for anti-zipper filter */ - pFMVoice->voiceGain = (EAS_I16) EAS_LogToLinear16(pChannel->staticGain); - pFMVoice->voiceGain = (EAS_I16) FMUL_15x15(pFMVoice->voiceGain, pSynth->masterVolume); - - /* initialize the operators */ - for (operIndex = 0; operIndex < 4; operIndex++) - { - - /* establish operator output gain level */ - /*lint -e{701} */ - pFMVoice->oper[operIndex].outputGain = EAS_LogToLinear16(((EAS_I16) (pRegion->oper[operIndex].gain & 0xfc) - 0xfc) << 7); - - /* check for linear velocity flag */ - /*lint -e{703} */ - if (pRegion->oper[operIndex].flags & FM_OPER_FLAG_LINEAR_VELOCITY) - temp = (EAS_I32) (pVoice->velocity - 127) << 5; - else - temp = (EAS_I32) fmControlTable[pVoice->velocity]; - - /* scale velocity */ - /*lint -e{704} use shift for performance */ - temp = (temp * (EAS_I32)(pRegion->oper[operIndex].velocityRelease & 0xf0)) >> 7; - - /* include key scalar */ - temp -= ((EAS_I32) pVoice->note - KEY_SCALE_PIVOT_POINT) * (EAS_I32) fmScaleTable[pRegion->oper[operIndex].egKeyScale & 0x0f]; - - /* saturate */ - temp = min(temp, 0); - temp = max(temp, -32768); - - /* save static gain parameters */ - pFMVoice->oper[operIndex].baseGain = (EAS_I16) EAS_LogToLinear16(temp); - - /* incorporate key scaling on decay rate */ - pFMVoice->oper[operIndex].envRate = FM_CalcEGRate( - pVoice->note, - fmDecayTable[pRegion->oper[operIndex].attackDecay & 0x0f], - fmScaleTable[pRegion->oper[operIndex].egKeyScale >> 4]); - - /* if zero attack time, max out envelope and jump to decay state */ - if ((pRegion->oper[operIndex].attackDecay & 0xf0) == 0xf0) - { - - /* start out envelope at max */ - pFMVoice->oper[operIndex].envGain = 0x7fff; - - /* set envelope to decay state */ - pFMVoice->oper[operIndex].envState = eFMEnvelopeStateDecay; - } - - /* start envelope at zero and start in attack state */ - else - { - pFMVoice->oper[operIndex].envGain = 0; - pFMVoice->oper[operIndex].envState = eFMEnvelopeStateAttack; - } - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * FM_UpdateChannel() - *---------------------------------------------------------------------------- - * Purpose: - * Calculate and assign static channel parameters - * These values only need to be updated if one of the controller values - * for this channel changes. - * Called when CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS flag is set. - * - * Inputs: - * nChannel - channel to update - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - the given channel's static gain and static pitch are updated - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pVoiceMgr) standard interface, pVoiceMgr not used */ -static void FM_UpdateChannel (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel) -{ - S_SYNTH_CHANNEL *pChannel; - EAS_I32 temp; - - pChannel = &pSynth->channels[channel]; - - /* convert CC7 volume controller to log scale */ - temp = fmControlTable[pChannel->volume]; - - /* incorporate CC11 expression controller */ - temp += fmControlTable[pChannel->expression]; - - /* saturate */ - pChannel->staticGain = (EAS_I16) max(temp,-32768); - - /* calculate pitch bend */ - /*lint -e{703} */ - temp = (((EAS_I32)(pChannel->pitchBend) << 2) - 32768); - - temp = FMUL_15x15(temp, pChannel->pitchBendSensitivity); - - /* include "magic number" compensation for sample rate and lookup table size */ - temp += MAGIC_NUMBER; - - /* if this is not a drum channel, then add in the per-channel tuning */ - if (!(pChannel->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL)) - temp += (pChannel->finePitch + (pChannel->coarsePitch * 100)); - - /* save static pitch */ - pChannel->staticPitch = temp; - - /* Calculate LFO modulation depth */ - /* mod wheel to LFO depth */ - temp = FMUL_15x15(DEFAULT_LFO_MOD_WHEEL_TO_PITCH_CENTS, - pChannel->modWheel << (NUM_EG1_FRAC_BITS -7)); - - /* channel pressure to LFO depth */ - pChannel->lfoAmt = (EAS_I16) (temp + - FMUL_15x15(DEFAULT_LFO_CHANNEL_PRESSURE_TO_PITCH_CENTS, - pChannel->channelPressure << (NUM_EG1_FRAC_BITS -7))); - - /* clear update flag */ - pChannel->channelFlags &= ~CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS; - return; -} - -/*---------------------------------------------------------------------------- - * FM_UpdateLFO() - *---------------------------------------------------------------------------- - * Purpose: - * Calculate the LFO for the given voice - * - * Inputs: - * pVoice - ptr to the voice whose LFO we want to update - * psEASData - pointer to overall EAS data structure - used for debug only - * - * Outputs: - * - * Side Effects: - * - updates LFO values for the given voice - *---------------------------------------------------------------------------- -*/ -static void FM_UpdateLFO (S_FM_VOICE *pFMVoice, const S_FM_REGION *pRegion) -{ - - /* increment the LFO phase if the delay time has elapsed */ - if (!pFMVoice->lfoDelay) - { - /*lint -e{701} */ - pFMVoice->lfoPhase = pFMVoice->lfoPhase + (EAS_U16) (-fmControlTable[((15 - (pRegion->lfoFreqDelay >> 4)) << 3) + 4]); - - /* square wave LFO? */ - if (pRegion->region.keyGroupAndFlags & REGION_FLAG_SQUARE_WAVE) - pFMVoice->lfoValue = (EAS_I16)(pFMVoice->lfoPhase & 0x8000 ? -32767 : 32767); - - /* trick to get a triangle wave out of a sawtooth */ - else - { - pFMVoice->lfoValue = (EAS_I16) (pFMVoice->lfoPhase << 1); - /*lint -e{502} */ - if ((pFMVoice->lfoPhase > 0x3fff) && (pFMVoice->lfoPhase < 0xC000)) - pFMVoice->lfoValue = ~pFMVoice->lfoValue; - } - } - - /* still in delay */ - else - pFMVoice->lfoDelay--; - - return; -} - -/*---------------------------------------------------------------------------- - * FM_UpdateEG() - *---------------------------------------------------------------------------- - * Purpose: - * Calculate the synthesis parameters for an operator to be used during - * the next buffer - * - * Inputs: - * pVoice - pointer to the voice being updated - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL FM_UpdateEG (S_SYNTH_VOICE *pVoice, S_OPERATOR *pOper, const S_FM_OPER *pOperData, EAS_BOOL oneShot) -{ - EAS_U32 temp; - EAS_BOOL done; - - /* set flag assuming the envelope is not done */ - done = EAS_FALSE; - - /* take appropriate action based on state */ - switch (pOper->envState) - { - - case eFMEnvelopeStateAttack: - - /* the envelope is linear during the attack, so add the value */ - temp = pOper->envGain + fmAttackTable[pOperData->attackDecay >> 4]; - - /* check for end of attack */ - if (temp >= 0x7fff) - { - pOper->envGain = 0x7fff; - pOper->envState = eFMEnvelopeStateDecay; - } - else - pOper->envGain = (EAS_U16) temp; - break; - - case eFMEnvelopeStateDecay: - - /* decay is exponential, multiply by decay rate */ - pOper->envGain = (EAS_U16) FMUL_15x15(pOper->envGain, pOper->envRate); - - /* check for sustain level reached */ - temp = (EAS_U32) (pOperData->sustain & 0xfc) << 7; - if (pOper->envGain <= (EAS_U16) temp) - { - /* if this is a one-shot patch, go directly to release phase */ - if (oneShot) - { - pOper->envRate = FM_CalcEGRate( - pVoice->note, - fmReleaseTable[pOperData->velocityRelease & 0x0f], - fmScaleTable[pOperData->egKeyScale >> 4]); - pOper->envState = eFMEnvelopeStateRelease; - } - - /* normal sustaining type */ - else - { - pOper->envGain = (EAS_U16) temp; - pOper->envState = eFMEnvelopeStateSustain; - } - } - break; - - case eFMEnvelopeStateSustain: - pOper->envGain = (EAS_U16)((EAS_U16)(pOperData->sustain & 0xfc) << 7); - break; - - case eFMEnvelopeStateRelease: - - /* release is exponential, multiply by release rate */ - pOper->envGain = (EAS_U16) FMUL_15x15(pOper->envGain, pOper->envRate); - - /* fully released */ - if (pOper->envGain == 0) - { - pOper->envGain = 0; - pOper->envState = eFMEnvelopeStateMuted; - done = EAS_TRUE; - } - break; - - case eFMEnvelopeStateMuted: - pOper->envGain = 0; - done = EAS_TRUE; - break; - default: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL,"Invalid operator state: %d", pOper->envState); */ } - } /* end switch (pOper->m_eState) */ - - return done; -} - -/*---------------------------------------------------------------------------- - * FM_UpdateDynamic() - *---------------------------------------------------------------------------- - * Purpose: - * Calculate the synthesis parameters for this voice that will be used to - * synthesize the next buffer - * - * Inputs: - * pVoice - pointer to the voice being updated - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * Returns EAS_TRUE if voice will be fully ramped to zero at the end of - * the next synthesized buffer. - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL FM_UpdateDynamic (S_SYNTH_VOICE *pVoice, S_FM_VOICE *pFMVoice, const S_FM_REGION *pRegion, S_SYNTH_CHANNEL *pChannel) -{ - EAS_I32 temp; - EAS_I32 pitch; - EAS_I32 lfoPitch; - EAS_INT operIndex; - EAS_BOOL done; - - /* increment LFO phase */ - FM_UpdateLFO(pFMVoice, pRegion); - - /* base pitch in cents */ - pitch = pVoice->note * 100; - - /* LFO amount includes LFO depth from programming + channel dynamics */ - temp = (fmScaleTable[pRegion->vibTrem >> 4] >> 1) + pChannel->lfoAmt; - - /* multiply by LFO output to get final pitch modulation */ - lfoPitch = FMUL_15x15(pFMVoice->lfoValue, temp); - - /* flag to indicate this voice is done */ - done = EAS_TRUE; - - /* iterate through operators to establish parameters */ - for (operIndex = 0; operIndex < 4; operIndex++) - { - EAS_BOOL bTemp; - - /* set base phase increment for each operator */ - temp = pRegion->oper[operIndex].tuning + - pChannel->staticPitch; - - /* add vibrato effect unless it is disabled for this operator */ - if ((pRegion->oper[operIndex].flags & FM_OPER_FLAG_NO_VIBRATO) == 0) - temp += lfoPitch; - - /* if note is monotonic, bias to MIDI note 60 */ - if (pRegion->oper[operIndex].flags & FM_OPER_FLAG_MONOTONE) - temp += 6000; - else - temp += pitch; - pFMVoice->oper[operIndex].pitch = (EAS_I16) temp; - - /* calculate envelope, returns true if done */ - bTemp = FM_UpdateEG(pVoice, &pFMVoice->oper[operIndex], &pRegion->oper[operIndex], pRegion->region.keyGroupAndFlags & REGION_FLAG_ONE_SHOT); - - /* check if all output envelopes have completed */ - if (FM_SynthIsOutputOperator(pRegion, operIndex)) - done = done && bTemp; - } - - return done; -} - -/*---------------------------------------------------------------------------- - * FM_UpdateVoice() - *---------------------------------------------------------------------------- - * Purpose: - * Synthesize a block of samples for the given voice. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * number of samples actually written to buffer - * - * Side Effects: - * - samples are added to the presently free buffer - * - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL FM_UpdateVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32 numSamples) -{ - S_SYNTH_CHANNEL *pChannel; - const S_FM_REGION *pRegion; - S_FM_VOICE *pFMVoice; - S_FM_VOICE_CONFIG vCfg; - S_FM_VOICE_FRAME vFrame; - EAS_I32 temp; - EAS_INT oper; - EAS_U16 voiceGainTarget; - EAS_BOOL done; - - /* setup some pointers */ - pChannel = GetChannelPtr(pSynth, pVoice); - pRegion = GetFMRegionPtr(pSynth, pVoice); - pFMVoice = GetFMVoicePtr(pVoiceMgr, voiceNum); - - /* if the voice is just starting, get the voice configuration data */ - if (pVoice->voiceFlags & VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET) - { - - /* split architecture may limit the number of voice starts */ -#ifdef MAX_VOICE_STARTS - if (pVoiceMgr->numVoiceStarts == MAX_VOICE_STARTS) - return EAS_FALSE; - pVoiceMgr->numVoiceStarts++; -#endif - - /* get initial parameters */ - vCfg.feedback = pRegion->feedback; - vCfg.voiceGain = (EAS_U16) pFMVoice->voiceGain; - -#if (NUM_OUTPUT_CHANNELS == 2) - vCfg.pan = pFMVoice->pan; -#endif - - /* get voice mode */ - vCfg.flags = pRegion->region.keyGroupAndFlags & 7; - - /* get operator parameters */ - for (oper = 0; oper < 4; oper++) - { - /* calculate initial gain */ - vCfg.gain[oper] = (EAS_U16) FMUL_15x15(pFMVoice->oper[oper].baseGain, pFMVoice->oper[oper].envGain); - vCfg.outputGain[oper] = pFMVoice->oper[oper].outputGain; - - /* copy noise waveform flag */ - if (pRegion->oper[oper].flags & FM_OPER_FLAG_NOISE) - vCfg.flags |= (EAS_U8) (FLAG_FM_ENG_VOICE_OP1_NOISE << oper); - } - -#ifdef FM_OFFBOARD - FM_ConfigVoice(voiceNum, &vCfg, pVoiceMgr->pFrameBuffer); -#else - FM_ConfigVoice(voiceNum, &vCfg, NULL); -#endif - - /* clear startup flag */ - pVoice->voiceFlags &= ~VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET; - } - - /* calculate new synthesis parameters */ - done = FM_UpdateDynamic(pVoice, pFMVoice, pRegion, pChannel); - - /* calculate LFO gain modulation */ - /*lint -e{702} */ - temp = ((fmScaleTable[pRegion->vibTrem & 0x0f] >> 1) * pFMVoice->lfoValue) >> FM_LFO_GAIN_SHIFT; - - /* include channel gain */ - temp += pChannel->staticGain; - - /* -32768 or lower is infinite attenuation */ - if (temp < -32767) - voiceGainTarget = 0; - - /* translate to linear gain multiplier */ - else - voiceGainTarget = EAS_LogToLinear16(temp); - - /* include synth master volume */ - voiceGainTarget = (EAS_U16) FMUL_15x15(voiceGainTarget, pSynth->masterVolume); - - /* save target values for this frame */ - vFrame.voiceGain = voiceGainTarget; - - /* assume voice output is zero */ - pVoice->gain = 0; - - /* save operator targets for this frame */ - for (oper = 0; oper < 4; oper++) - { - vFrame.gain[oper] = (EAS_U16) FMUL_15x15(pFMVoice->oper[oper].baseGain, pFMVoice->oper[oper].envGain); - vFrame.pitch[oper] = pFMVoice->oper[oper].pitch; - - /* use the highest output envelope level as the gain for the voice stealing algorithm */ - if (FM_SynthIsOutputOperator(pRegion, oper)) - pVoice->gain = max(pVoice->gain, (EAS_I16) vFrame.gain[oper]); - } - - /* consider voice gain multiplier in calculating gain for stealing algorithm */ - pVoice->gain = (EAS_I16) FMUL_15x15(voiceGainTarget, pVoice->gain); - - /* synthesize samples */ -#ifdef FM_OFFBOARD - FM_ProcessVoice(voiceNum, &vFrame, numSamples, pVoiceMgr->operMixBuffer, pVoiceMgr->voiceBuffer, pMixBuffer, pVoiceMgr->pFrameBuffer); -#else - FM_ProcessVoice(voiceNum, &vFrame, numSamples, pVoiceMgr->operMixBuffer, pVoiceMgr->voiceBuffer, pMixBuffer, NULL); -#endif - - return done; -} - diff --git a/arm-hybrid-22k/lib_src/eas_fmsynth.h b/arm-hybrid-22k/lib_src/eas_fmsynth.h deleted file mode 100644 index 76f8adc..0000000 --- a/arm-hybrid-22k/lib_src/eas_fmsynth.h +++ /dev/null @@ -1,81 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_fmsynth.h - * - * Contents and purpose: - * Implements the FM synthesizer functions. - * - * Copyright Sonic Network Inc. 2004 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 90 $ - * $Date: 2006-07-11 20:18:13 -0700 (Tue, 11 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef fmsynthH -#define fmsynthH - -#include "eas_data.h" - -#if defined (_FM_SYNTH) - -/* FM envelope state */ -typedef enum { - eFMEnvelopeStateAttack = 0, - eFMEnvelopeStateDecay, - eFMEnvelopeStateSustain, - eFMEnvelopeStateRelease, - eFMEnvelopeStateMuted, - eFMEnvelopeStateInvalid /* should never be in this state! */ -} E_FM_ENVELOPE_STATE; - -/*------------------------------------ - * S_OPERATOR data structure - *------------------------------------ -*/ -typedef struct s_operator_tag -{ - EAS_I16 pitch; /* operator pitch in cents */ - EAS_U16 envGain; /* envelope target */ - EAS_I16 baseGain; /* patch gain (inc. vel & key scale) */ - EAS_U16 outputGain; /* current output gain */ - EAS_U16 envRate; /* calculated envelope rate */ - EAS_U8 envState; /* envelope state */ - EAS_U8 pad; /* pad to 16-bits */ -} S_OPERATOR; -#endif - -typedef struct s_fm_voice_tag -{ - S_OPERATOR oper[4]; /* operator data */ - EAS_I16 voiceGain; /* LFO + channel parameters */ - EAS_U16 lfoPhase; /* LFO current phase */ - EAS_I16 lfoValue; /* LFO current value */ - EAS_U16 lfoDelay; /* keeps track of elapsed delay time */ - EAS_I8 pan; /* stereo pan value (-64 to +64) */ - EAS_I8 pad; /* reserved to maintain alignment */ -} S_FM_VOICE; - -#ifdef _FM_EDITOR -extern S_FM_REGION newPatch; -extern S_FM_REGION OriginalPatch; -#endif - -extern EAS_U32 freqTable[]; - -#endif diff --git a/arm-hybrid-22k/lib_src/eas_fmtables.c b/arm-hybrid-22k/lib_src/eas_fmtables.c deleted file mode 100644 index 25c6961..0000000 --- a/arm-hybrid-22k/lib_src/eas_fmtables.c +++ /dev/null @@ -1,368 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_fmtables.c - * - * Contents and purpose: - * Contains lookup tables for the FM synthesizer - * - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - * - *---------------------------------------------------------------------------- -*/ - - -#include "eas_types.h" - -/* this table is needed by the DSP and the main processor */ -const EAS_U8 fmScaleTable[16] = -{ - 0,8,16,24,32,40,48,56,64,72,80,96,128,160,192,255 -}; - -/* these tables are needed on the main processor */ -#ifndef _DSP_CODE -const EAS_I16 fmControlTable[128] = -{ - -32768,-14313,-12265,-11067,-10217,-9558,-9019,-8563, - -8169,-7821,-7510,-7228,-6971,-6734,-6515,-6312, - -6121,-5942,-5773,-5613,-5462,-5317,-5180,-5049, - -4923,-4802,-4686,-4575,-4467,-4364,-4264,-4167, - -4073,-3982,-3894,-3808,-3725,-3644,-3565,-3488, - -3414,-3341,-3269,-3200,-3132,-3066,-3001,-2937, - -2875,-2814,-2754,-2696,-2638,-2582,-2527,-2473, - -2419,-2367,-2316,-2265,-2216,-2167,-2119,-2071, - -2025,-1979,-1934,-1889,-1846,-1803,-1760,-1718, - -1677,-1636,-1596,-1556,-1517,-1478,-1440,-1403, - -1366,-1329,-1293,-1257,-1221,-1186,-1152,-1118, - -1084,-1051,-1018,-985,-953,-921,-889,-858, - -827,-796,-766,-736,-706,-677,-648,-619, - -590,-562,-534,-506,-479,-452,-425,-398, - -371,-345,-319,-293,-268,-242,-217,-192, - -168,-143,-119,-95,-71,-47,-23,0 -}; - -const EAS_U16 fmRateTable[128] = -{ - 32767,32764,32758,32747,32731,32712,32688,32659, - 32627,32590,32548,32503,32453,32398,32340,32277, - 32211,32140,32065,31985,31902,31815,31724,31628, - 31529,31426,31319,31208,31094,30976,30854,30728, - 30599,30466,30330,30191,30048,29902,29752,29599, - 29443,29285,29123,28958,28790,28619,28445,28269, - 28090,27909,27725,27538,27349,27158,26964,26769, - 26571,26371,26169,25965,25760,25552,25343,25132, - 24920,24706,24490,24274,24056,23836,23616,23394, - 23172,22948,22724,22499,22273,22046,21819,21591, - 21363,21135,20906,20676,20447,20217,19987,19758, - 19528,19298,19069,18840,18610,18382,18153,17926, - 17698,17471,17245,17020,16795,16571,16347,16125, - 15903,15683,15463,15245,15027,14811,14596,14382, - 14169,13957,13747,13538,13331,13125,12920,12717, - 12516,12316,12117,11921,11725,11532,11340,0 -}; - -const EAS_U16 fmAttackTable[15] = -{ - 27,54,109,327,655,1310,2730,4095, - 4681,5461,6553,8191,10922,16383,32767 -}; - -const EAS_U8 fmDecayTable[16] = -{ - 4,7,10,15,20,25,30,35,40,50,60,70,80,90,100,127 -}; - -const EAS_U8 fmReleaseTable[16] = -{ - 10,15,20,25,30,35,40,45,50,60,70,80,90,100,113,127 -}; -#endif - -/* this table is needed only on the DSP */ -#if defined(_DSP_CODE) || !defined(_SPLIT_ARCHITECTURE) -//--------------------------------------------------------------------- -// sineTable -// -// Contains sine lookup table -//--------------------------------------------------------------------- - -const EAS_I16 sineTable[2048] = -{ - 0,101,201,302,402,503,603,704, - 804,905,1005,1106,1206,1307,1407,1507, - 1608,1708,1809,1909,2009,2110,2210,2310, - 2410,2511,2611,2711,2811,2911,3012,3112, - 3212,3312,3412,3512,3612,3712,3811,3911, - 4011,4111,4210,4310,4410,4509,4609,4708, - 4808,4907,5007,5106,5205,5305,5404,5503, - 5602,5701,5800,5899,5998,6096,6195,6294, - 6393,6491,6590,6688,6786,6885,6983,7081, - 7179,7277,7375,7473,7571,7669,7767,7864, - 7962,8059,8157,8254,8351,8448,8545,8642, - 8739,8836,8933,9030,9126,9223,9319,9416, - 9512,9608,9704,9800,9896,9992,10087,10183, - 10278,10374,10469,10564,10659,10754,10849,10944, - 11039,11133,11228,11322,11417,11511,11605,11699, - 11793,11886,11980,12074,12167,12260,12353,12446, - 12539,12632,12725,12817,12910,13002,13094,13187, - 13279,13370,13462,13554,13645,13736,13828,13919, - 14010,14101,14191,14282,14372,14462,14553,14643, - 14732,14822,14912,15001,15090,15180,15269,15358, - 15446,15535,15623,15712,15800,15888,15976,16063, - 16151,16238,16325,16413,16499,16586,16673,16759, - 16846,16932,17018,17104,17189,17275,17360,17445, - 17530,17615,17700,17784,17869,17953,18037,18121, - 18204,18288,18371,18454,18537,18620,18703,18785, - 18868,18950,19032,19113,19195,19276,19357,19438, - 19519,19600,19680,19761,19841,19921,20000,20080, - 20159,20238,20317,20396,20475,20553,20631,20709, - 20787,20865,20942,21019,21096,21173,21250,21326, - 21403,21479,21554,21630,21705,21781,21856,21930, - 22005,22079,22154,22227,22301,22375,22448,22521, - 22594,22667,22739,22812,22884,22956,23027,23099, - 23170,23241,23311,23382,23452,23522,23592,23662, - 23731,23801,23870,23938,24007,24075,24143,24211, - 24279,24346,24413,24480,24547,24613,24680,24746, - 24811,24877,24942,25007,25072,25137,25201,25265, - 25329,25393,25456,25519,25582,25645,25708,25770, - 25832,25893,25955,26016,26077,26138,26198,26259, - 26319,26378,26438,26497,26556,26615,26674,26732, - 26790,26848,26905,26962,27019,27076,27133,27189, - 27245,27300,27356,27411,27466,27521,27575,27629, - 27683,27737,27790,27843,27896,27949,28001,28053, - 28105,28157,28208,28259,28310,28360,28411,28460, - 28510,28560,28609,28658,28706,28755,28803,28850, - 28898,28945,28992,29039,29085,29131,29177,29223, - 29268,29313,29358,29403,29447,29491,29534,29578, - 29621,29664,29706,29749,29791,29832,29874,29915, - 29956,29997,30037,30077,30117,30156,30195,30234, - 30273,30311,30349,30387,30424,30462,30498,30535, - 30571,30607,30643,30679,30714,30749,30783,30818, - 30852,30885,30919,30952,30985,31017,31050,31082, - 31113,31145,31176,31206,31237,31267,31297,31327, - 31356,31385,31414,31442,31470,31498,31526,31553, - 31580,31607,31633,31659,31685,31710,31736,31760, - 31785,31809,31833,31857,31880,31903,31926,31949, - 31971,31993,32014,32036,32057,32077,32098,32118, - 32137,32157,32176,32195,32213,32232,32250,32267, - 32285,32302,32318,32335,32351,32367,32382,32397, - 32412,32427,32441,32455,32469,32482,32495,32508, - 32521,32533,32545,32556,32567,32578,32589,32599, - 32609,32619,32628,32637,32646,32655,32663,32671, - 32678,32685,32692,32699,32705,32711,32717,32722, - 32728,32732,32737,32741,32745,32748,32752,32755, - 32757,32759,32761,32763,32765,32766,32766,32767, - 32767,32767,32766,32766,32765,32763,32761,32759, - 32757,32755,32752,32748,32745,32741,32737,32732, - 32728,32722,32717,32711,32705,32699,32692,32685, - 32678,32671,32663,32655,32646,32637,32628,32619, - 32609,32599,32589,32578,32567,32556,32545,32533, - 32521,32508,32495,32482,32469,32455,32441,32427, - 32412,32397,32382,32367,32351,32335,32318,32302, - 32285,32267,32250,32232,32213,32195,32176,32157, - 32137,32118,32098,32077,32057,32036,32014,31993, - 31971,31949,31926,31903,31880,31857,31833,31809, - 31785,31760,31736,31710,31685,31659,31633,31607, - 31580,31553,31526,31498,31470,31442,31414,31385, - 31356,31327,31297,31267,31237,31206,31176,31145, - 31113,31082,31050,31017,30985,30952,30919,30885, - 30852,30818,30783,30749,30714,30679,30643,30607, - 30571,30535,30498,30462,30424,30387,30349,30311, - 30273,30234,30195,30156,30117,30077,30037,29997, - 29956,29915,29874,29832,29791,29749,29706,29664, - 29621,29578,29534,29491,29447,29403,29358,29313, - 29268,29223,29177,29131,29085,29039,28992,28945, - 28898,28850,28803,28755,28706,28658,28609,28560, - 28510,28460,28411,28360,28310,28259,28208,28157, - 28105,28053,28001,27949,27896,27843,27790,27737, - 27683,27629,27575,27521,27466,27411,27356,27300, - 27245,27189,27133,27076,27019,26962,26905,26848, - 26790,26732,26674,26615,26556,26497,26438,26378, - 26319,26259,26198,26138,26077,26016,25955,25893, - 25832,25770,25708,25645,25582,25519,25456,25393, - 25329,25265,25201,25137,25072,25007,24942,24877, - 24811,24746,24680,24613,24547,24480,24413,24346, - 24279,24211,24143,24075,24007,23938,23870,23801, - 23731,23662,23592,23522,23452,23382,23311,23241, - 23170,23099,23027,22956,22884,22812,22739,22667, - 22594,22521,22448,22375,22301,22227,22154,22079, - 22005,21930,21856,21781,21705,21630,21554,21479, - 21403,21326,21250,21173,21096,21019,20942,20865, - 20787,20709,20631,20553,20475,20396,20317,20238, - 20159,20080,20000,19921,19841,19761,19680,19600, - 19519,19438,19357,19276,19195,19113,19032,18950, - 18868,18785,18703,18620,18537,18454,18371,18288, - 18204,18121,18037,17953,17869,17784,17700,17615, - 17530,17445,17360,17275,17189,17104,17018,16932, - 16846,16759,16673,16586,16499,16413,16325,16238, - 16151,16063,15976,15888,15800,15712,15623,15535, - 15446,15358,15269,15180,15090,15001,14912,14822, - 14732,14643,14553,14462,14372,14282,14191,14101, - 14010,13919,13828,13736,13645,13554,13462,13370, - 13279,13187,13094,13002,12910,12817,12725,12632, - 12539,12446,12353,12260,12167,12074,11980,11886, - 11793,11699,11605,11511,11417,11322,11228,11133, - 11039,10944,10849,10754,10659,10564,10469,10374, - 10278,10183,10087,9992,9896,9800,9704,9608, - 9512,9416,9319,9223,9126,9030,8933,8836, - 8739,8642,8545,8448,8351,8254,8157,8059, - 7962,7864,7767,7669,7571,7473,7375,7277, - 7179,7081,6983,6885,6786,6688,6590,6491, - 6393,6294,6195,6096,5998,5899,5800,5701, - 5602,5503,5404,5305,5205,5106,5007,4907, - 4808,4708,4609,4509,4410,4310,4210,4111, - 4011,3911,3811,3712,3612,3512,3412,3312, - 3212,3112,3012,2911,2811,2711,2611,2511, - 2410,2310,2210,2110,2009,1909,1809,1708, - 1608,1507,1407,1307,1206,1106,1005,905, - 804,704,603,503,402,302,201,101, - 0,-101,-201,-302,-402,-503,-603,-704, - -804,-905,-1005,-1106,-1206,-1307,-1407,-1507, - -1608,-1708,-1809,-1909,-2009,-2110,-2210,-2310, - -2410,-2511,-2611,-2711,-2811,-2911,-3012,-3112, - -3212,-3312,-3412,-3512,-3612,-3712,-3811,-3911, - -4011,-4111,-4210,-4310,-4410,-4509,-4609,-4708, - -4808,-4907,-5007,-5106,-5205,-5305,-5404,-5503, - -5602,-5701,-5800,-5899,-5998,-6096,-6195,-6294, - -6393,-6491,-6590,-6688,-6786,-6885,-6983,-7081, - -7179,-7277,-7375,-7473,-7571,-7669,-7767,-7864, - -7962,-8059,-8157,-8254,-8351,-8448,-8545,-8642, - -8739,-8836,-8933,-9030,-9126,-9223,-9319,-9416, - -9512,-9608,-9704,-9800,-9896,-9992,-10087,-10183, - -10278,-10374,-10469,-10564,-10659,-10754,-10849,-10944, - -11039,-11133,-11228,-11322,-11417,-11511,-11605,-11699, - -11793,-11886,-11980,-12074,-12167,-12260,-12353,-12446, - -12539,-12632,-12725,-12817,-12910,-13002,-13094,-13187, - -13279,-13370,-13462,-13554,-13645,-13736,-13828,-13919, - -14010,-14101,-14191,-14282,-14372,-14462,-14553,-14643, - -14732,-14822,-14912,-15001,-15090,-15180,-15269,-15358, - -15446,-15535,-15623,-15712,-15800,-15888,-15976,-16063, - -16151,-16238,-16325,-16413,-16499,-16586,-16673,-16759, - -16846,-16932,-17018,-17104,-17189,-17275,-17360,-17445, - -17530,-17615,-17700,-17784,-17869,-17953,-18037,-18121, - -18204,-18288,-18371,-18454,-18537,-18620,-18703,-18785, - 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* - * File: - * eas_ima_tables.c - * - * Contents and purpose: - * Contains the constant tables for IMA encode/decode - * - * Copyright (c) 2005 Sonic Network Inc. - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 760 $ - * $Date: 2007-07-17 23:09:36 -0700 (Tue, 17 Jul 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_types.h" - -/*---------------------------------------------------------------------------- - * ADPCM decode tables - *---------------------------------------------------------------------------- -*/ -const EAS_I16 imaIndexTable[16] = -{ - -1, -1, -1, -1, 2, 4, 6, 8, - -1, -1, -1, -1, 2, 4, 6, 8 -}; - -const EAS_I16 imaStepSizeTable[89] = -{ - 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, - 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, - 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, - 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, - 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, - 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, - 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, - 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, - 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 -}; - diff --git a/arm-hybrid-22k/lib_src/eas_imaadpcm.c b/arm-hybrid-22k/lib_src/eas_imaadpcm.c deleted file mode 100644 index 68bf257..0000000 --- a/arm-hybrid-22k/lib_src/eas_imaadpcm.c +++ /dev/null @@ -1,368 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_imaadpcm.c - * - * Contents and purpose: - * Implements the IMA ADPCM decoder - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 847 $ - * $Date: 2007-08-27 21:30:08 -0700 (Mon, 27 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_data.h" -#include "eas_host.h" -#include "eas_pcm.h" -#include "eas_math.h" -#include "eas_report.h" - -// #define _DEBUG_IMA_ADPCM_LOCATE - -/*---------------------------------------------------------------------------- - * externs - *---------------------------------------------------------------------------- -*/ -extern const EAS_I16 imaIndexTable[]; -extern const EAS_I16 imaStepSizeTable[]; - -/*---------------------------------------------------------------------------- - * prototypes - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT IMADecoderInit (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState); -static EAS_RESULT IMADecoderSample (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState); -static void IMADecoderADPCM (S_DECODER_STATE *pState, EAS_U8 nibble); -static EAS_RESULT IMADecoderLocate (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 time); - -/*---------------------------------------------------------------------------- - * IMA ADPCM Decoder interface - *---------------------------------------------------------------------------- -*/ -const S_DECODER_INTERFACE IMADecoder = -{ - IMADecoderInit, - IMADecoderSample, - IMADecoderLocate -}; - -/*---------------------------------------------------------------------------- - * IMADecoderInit() - *---------------------------------------------------------------------------- - * Purpose: - * Initializes the IMA ADPCM decoder - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) common decoder interface - pEASData not used */ -static EAS_RESULT IMADecoderInit (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState) -{ - pState->decoderL.step = 0; - pState->decoderR.step = 0; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * IMADecoderSample() - *---------------------------------------------------------------------------- - * Purpose: - * Decodes an IMA ADPCM sample - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT IMADecoderSample (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState) -{ - EAS_RESULT result; - EAS_I16 sTemp; - - /* if high nibble, decode */ - if (pState->hiNibble) - { - IMADecoderADPCM(&pState->decoderL, (EAS_U8)(pState->srcByte >> 4)); - pState->hiNibble = EAS_FALSE; - } - - /* low nibble, need to fetch another byte */ - else - { - /* check for loop */ - if ((pState->bytesLeft == 0) && (pState->loopSamples != 0)) - { - /* seek to start of loop */ - if ((result = EAS_HWFileSeek(pEASData->hwInstData, pState->fileHandle, (EAS_I32) (pState->startPos + pState->loopLocation))) != EAS_SUCCESS) - return result; - pState->bytesLeft = pState->byteCount = (EAS_I32) pState->bytesLeftLoop; - pState->blockCount = 0; - pState->flags &= ~PCM_FLAGS_EMPTY; - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMADecoderSample: Rewind file to %d, bytesLeft = %d\n", pState->startPos, pState->bytesLeft); */ } - } - - /* if start of block, fetch new predictor and step index */ - if ((pState->blockSize != 0) && (pState->blockCount == 0) && (pState->bytesLeft != 0)) - { - - /* get predicted sample for left channel */ - if ((result = EAS_HWGetWord(pEASData->hwInstData, pState->fileHandle, &sTemp, EAS_FALSE)) != EAS_SUCCESS) - return result; -#ifdef _DEBUG_IMA_ADPCM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Predictor: Was %d, now %d\n", pState->decoderL.acc, sTemp); */ } -#endif - pState->decoderL.acc = pState->decoderL.x1 = sTemp; - - /* get step index for left channel - upper 8 bits are reserved */ - if ((result = EAS_HWGetWord(pEASData->hwInstData, pState->fileHandle, &sTemp, EAS_FALSE)) != EAS_SUCCESS) - return result; -#ifdef _DEBUG_IMA_ADPCM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Step: Was %d, now %d\n", pState->decoderL.step, sTemp); */ } -#endif - pState->decoderL.step = sTemp & 0xff; - - if (pState->flags & PCM_FLAGS_STEREO) - { - /* get predicted sample for right channel */ - if ((result = EAS_HWGetWord(pEASData->hwInstData, pState->fileHandle, &sTemp, EAS_FALSE)) != EAS_SUCCESS) - return result; - pState->decoderR.acc = pState->decoderR.x1 = sTemp; - - /* get step index for right channel - upper 8 bits are reserved */ - if ((result = EAS_HWGetWord(pEASData->hwInstData, pState->fileHandle, &sTemp, EAS_FALSE)) != EAS_SUCCESS) - return result; -#ifdef _DEBUG_IMA_ADPCM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Step: Was %d, now %d\n", pState->decoderR.step, sTemp); */ } -#endif - pState->decoderR.step = sTemp & 0xff; - - pState->blockCount = pState->blockSize - 8; - pState->bytesLeft -= 8; - } - else - { - pState->blockCount = pState->blockSize - 4; - pState->bytesLeft -= 4; - } - } - else - { - - /* get another ADPCM data pair */ - if (pState->bytesLeft) - { - - if ((result = EAS_HWGetByte(pEASData->hwInstData, pState->fileHandle, &pState->srcByte)) != EAS_SUCCESS) - return result; - - /* decode the low nibble */ - pState->bytesLeft--; - pState->blockCount--; - IMADecoderADPCM(&pState->decoderL, (EAS_U8)(pState->srcByte & 0x0f)); - - if (pState->flags & PCM_FLAGS_STEREO) - IMADecoderADPCM(&pState->decoderR, (EAS_U8)(pState->srcByte >> 4)); - else - pState->hiNibble = EAS_TRUE; - } - - /* out of ADPCM data, generate enough samples to fill buffer */ - else - { - pState->decoderL.x1 = pState->decoderL.x0; - pState->decoderR.x1 = pState->decoderR.x0; - } - } - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * IMADecoderADPCM() - *---------------------------------------------------------------------------- - * Purpose: - * Decodes an IMA ADPCM sample - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static void IMADecoderADPCM (S_DECODER_STATE *pState, EAS_U8 nibble) -{ - EAS_INT delta; - EAS_INT stepSize; - - /* get stepsize from table */ - stepSize = imaStepSizeTable[pState->step]; - - /* delta = (abs(delta) + 0.5) * step / 4 */ - delta = 0; - if (nibble & 4) - delta += stepSize; - - if (nibble & 2) - /*lint -e{702} use shift for performance */ - delta += stepSize >> 1; - - if (nibble & 1) - /*lint -e{702} use shift for performance */ - delta += stepSize >> 2; - - /*lint -e{702} use shift for performance */ - delta += stepSize >> 3; - - /* integrate the delta */ - if (nibble & 8) - pState->acc -= delta; - else - pState->acc += delta; - - /* saturate */ - if (pState->acc > 32767) - pState->acc = 32767; - if (pState->acc < -32768) - pState->acc = -32768; - pState->x1 = (EAS_PCM) pState->acc; - - /* compute new step size */ - pState->step += imaIndexTable[nibble]; - if (pState->step < 0) - pState->step = 0; - if (pState->step > 88) - pState->step = 88; - -#ifdef _DEBUG_IMA_ADPCM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "In=%u, Pred=%d, Step=%d\n", nibble, pState->acc, imaStepSizeTable[pState->step]); */ } -#endif -} - -/*---------------------------------------------------------------------------- - * IMADecoderLocate() - *---------------------------------------------------------------------------- - * Locate in an IMA ADPCM stream - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT IMADecoderLocate (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 time) -{ - EAS_RESULT result; - EAS_I32 temp; - EAS_I32 samplesPerBlock; - EAS_I32 secs, msecs; - - /* no need to calculate if time is zero */ - if (time == 0) - temp = 0; - - /* not zero */ - else - { - - /* can't seek if not a blocked file */ - if (pState->blockSize == 0) - return EAS_ERROR_FEATURE_NOT_AVAILABLE; - - /* calculate number of samples per block */ - if (pState->flags & PCM_FLAGS_STEREO) - samplesPerBlock = pState->blockSize - 7; - else - samplesPerBlock = (pState->blockSize << 1) - 7; - - /* break down into secs and msecs */ - secs = time / 1000; - msecs = time - (secs * 1000); - - /* calculate sample number fraction from msecs */ - temp = (msecs * pState->sampleRate); - temp = (temp >> 10) + ((temp * 49) >> 21); - - /* add integer sample count */ - temp += secs * pState->sampleRate; - -#ifdef _DEBUG_IMA_ADPCM_LOCATE - EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x2380b977, 0x00000006 , time, temp); -#endif - - /* for looped samples, calculate position in the loop */ - if ((temp > pState->byteCount) && (pState->loopSamples != 0)) - { - EAS_I32 numBlocks; - EAS_I32 samplesPerLoop; - EAS_I32 samplesInLastBlock; - - numBlocks = (EAS_I32) (pState->loopStart / pState->blockSize); - samplesInLastBlock = (EAS_I32) pState->loopStart - (numBlocks * pState->blockSize); - if (samplesInLastBlock) - { - if (pState->flags & PCM_FLAGS_STEREO) - samplesInLastBlock = samplesInLastBlock - 7; - else - /*lint -e{703} use shift for performance */ - samplesInLastBlock = (samplesInLastBlock << 1) - 7; - } - samplesPerLoop = numBlocks * samplesPerBlock + samplesInLastBlock; - temp = temp % samplesPerLoop; -#ifdef _DEBUG_IMA_ADPCM_LOCATE - EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x2380b977, 0x00000007 , numBlocks, samplesPerLoop, samplesInLastBlock, temp); -#endif - } - - /* find start of block for requested sample */ - temp = (temp / samplesPerBlock) * pState->blockSize; -#ifdef _DEBUG_IMA_ADPCM_LOCATE - EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x2380b977, 0x00000008 , temp); -#endif - - } - - /* seek to new location */ - if ((result = EAS_PESeek(pEASData, pState, &temp)) != EAS_SUCCESS) - return result; - -#ifdef _DEBUG_IMA_ADPCM_LOCATE - EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x2380b977, 0x00000009 , pState->bytesLeft); -#endif - - /* reset state */ - pState->blockCount = 0; - pState->hiNibble = EAS_FALSE; - if ((pState->state != EAS_STATE_PAUSING) && (pState->state != EAS_STATE_PAUSED)) - pState->state = EAS_STATE_READY; - - return EAS_SUCCESS; -} - diff --git a/arm-hybrid-22k/lib_src/eas_imelody.c b/arm-hybrid-22k/lib_src/eas_imelody.c deleted file mode 100644 index 9f4d541..0000000 --- a/arm-hybrid-22k/lib_src/eas_imelody.c +++ /dev/null @@ -1,1738 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_imelody.c - * - * Contents and purpose: - * iMelody parser - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 797 $ - * $Date: 2007-08-01 00:15:56 -0700 (Wed, 01 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -/* lint doesn't like the way some string.h files look */ -#ifdef _lint -#include "lint_stdlib.h" -#else -#include -#endif - -#include "eas_data.h" -#include "eas_miditypes.h" -#include "eas_parser.h" -#include "eas_report.h" -#include "eas_host.h" -#include "eas_midi.h" -#include "eas_config.h" -#include "eas_vm_protos.h" -#include "eas_imelodydata.h" -#include "eas_ctype.h" - -// #define _DEBUG_IMELODY - -/* increase gain for mono ringtones */ -#define IMELODY_GAIN_OFFSET 8 - -/* length of 32nd note in 1/256ths of a msec for 120 BPM tempo */ -#define DEFAULT_TICK_CONV 16000 -#define TICK_CONVERT 1920000 - -/* default channel and program for iMelody playback */ -#define IMELODY_CHANNEL 0 -#define IMELODY_PROGRAM 80 -#define IMELODY_VEL_MUL 4 -#define IMELODY_VEL_OFS 67 - -/* multiplier for fixed point triplet conversion */ -#define TRIPLET_MULTIPLIER 683 -#define TRIPLET_SHIFT 10 - -static const char* const tokens[] = -{ - "BEGIN:IMELODY", - "VERSION:", - "FORMAT:CLASS", - "NAME:", - "COMPOSER:", - "BEAT:", - "STYLE:", - "VOLUME:", - "MELODY:", - "END:IMELODY" -}; - -/* ledon or ledoff */ -static const char ledStr[] = "edo"; - -/* vibeon or vibeoff */ -static const char vibeStr[] = "ibeo"; - -/* backon or backoff */ -static const char backStr[] = "cko"; - -typedef enum -{ - TOKEN_BEGIN, - TOKEN_VERSION, - TOKEN_FORMAT, - TOKEN_NAME, - TOKEN_COMPOSER, - TOKEN_BEAT, - TOKEN_STYLE, - TOKEN_VOLUME, - TOKEN_MELODY, - TOKEN_END, - TOKEN_INVALID -} ENUM_IMELODY_TOKENS; - -/* lookup table for note values */ -static const EAS_I8 noteTable[] = { 9, 11, 0, 2, 4, 5, 7 }; - -/* inline functions */ -#ifdef _DEBUG_IMELODY -static void PutBackChar (S_IMELODY_DATA *pData) -{ - if (pData->index) - pData->index--; - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "PutBackChar '%c'\n", pData->buffer[pData->index]); */ } -} -#else -EAS_INLINE void PutBackChar (S_IMELODY_DATA *pData) { if (pData->index) pData->index--; } -#endif - - -/* local prototypes */ -static EAS_RESULT IMY_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset); -static EAS_RESULT IMY_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT IMY_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime); -static EAS_RESULT IMY_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode); -static EAS_RESULT IMY_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState); -static EAS_RESULT IMY_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT IMY_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT IMY_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT IMY_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT IMY_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value); -static EAS_RESULT IMY_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue); -static EAS_BOOL IMY_PlayNote (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData, EAS_I8 note, EAS_INT parserMode); -static EAS_BOOL IMY_PlayRest (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData); -static EAS_BOOL IMY_GetDuration (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_I32 *pDuration); -static EAS_BOOL IMY_GetLEDState (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData); -static EAS_BOOL IMY_GetVibeState (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData); -static EAS_BOOL IMY_GetBackState (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData); -static EAS_BOOL IMY_GetVolume (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_BOOL inHeader); -static EAS_BOOL IMY_GetNumber (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_INT *temp, EAS_BOOL inHeader); -static EAS_RESULT IMY_ParseHeader (S_EAS_DATA *pEASData, S_IMELODY_DATA* pData); -static EAS_I8 IMY_GetNextChar (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_BOOL inHeader); -static EAS_RESULT IMY_ReadLine (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE fileHandle, EAS_I8 *buffer, EAS_I32 *pStartLine); -static EAS_INT IMY_ParseLine (EAS_I8 *buffer, EAS_U8 *pIndex); - - -/*---------------------------------------------------------------------------- - * - * EAS_iMelody_Parser - * - * This structure contains the functional interface for the iMelody parser - *---------------------------------------------------------------------------- -*/ -const S_FILE_PARSER_INTERFACE EAS_iMelody_Parser = -{ - IMY_CheckFileType, - IMY_Prepare, - IMY_Time, - IMY_Event, - IMY_State, - IMY_Close, - IMY_Reset, - IMY_Pause, - IMY_Resume, - NULL, - IMY_SetData, - IMY_GetData, - NULL -}; - -/*---------------------------------------------------------------------------- - * IMY_CheckFileType() - *---------------------------------------------------------------------------- - * Purpose: - * Check the file type to see if we can parse it - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT IMY_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset) -{ - S_IMELODY_DATA* pData; - EAS_I8 buffer[MAX_LINE_SIZE+1]; - EAS_U8 index; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_CheckFileType\n"); */ } -#endif - - /* read the first line of the file */ - *ppHandle = NULL; - if (IMY_ReadLine(pEASData->hwInstData, fileHandle, buffer, NULL) != EAS_SUCCESS) - return EAS_SUCCESS; - - /* check for header string */ - if (IMY_ParseLine(buffer, &index) == TOKEN_BEGIN) - { - - /* check for static memory allocation */ - if (pEASData->staticMemoryModel) - pData = EAS_CMEnumData(EAS_CM_IMELODY_DATA); - else - pData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_IMELODY_DATA)); - if (!pData) - return EAS_ERROR_MALLOC_FAILED; - EAS_HWMemSet(pData, 0, sizeof(S_IMELODY_DATA)); - - /* initialize */ - pData->fileHandle = fileHandle; - pData->fileOffset = offset; - pData->state = EAS_STATE_ERROR; - pData->state = EAS_STATE_OPEN; - - /* return a pointer to the instance data */ - *ppHandle = pData; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * IMY_Prepare() - *---------------------------------------------------------------------------- - * Purpose: - * Prepare to parse the file. Allocates instance data (or uses static allocation for - * static memory model). - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT IMY_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_IMELODY_DATA* pData; - EAS_RESULT result; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_Prepare\n"); */ } -#endif - - /* check for valid state */ - pData = (S_IMELODY_DATA*) pInstData; - if (pData->state != EAS_STATE_OPEN) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - - /* instantiate a synthesizer */ - if ((result = VMInitMIDI(pEASData, &pData->pSynth)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitMIDI returned %d\n", result); */ } - return result; - } - - /* parse the header */ - if ((result = IMY_ParseHeader(pEASData, pData)) != EAS_SUCCESS) - return result; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Prepare: state set to EAS_STATE_READY\n"); */ } -#endif - - pData ->state = EAS_STATE_READY; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * IMY_Time() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the time of the next event in msecs - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * pTime - pointer to variable to hold time of next event (in msecs) - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) common decoder interface - pEASData not used */ -static EAS_RESULT IMY_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime) -{ - S_IMELODY_DATA *pData; - - pData = (S_IMELODY_DATA*) pInstData; - - /* return time in milliseconds */ - /*lint -e{704} use shift instead of division */ - *pTime = pData->time >> 8; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * IMY_Event() - *---------------------------------------------------------------------------- - * Purpose: - * Parse the next event in the file - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT IMY_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode) -{ - S_IMELODY_DATA* pData; - EAS_RESULT result; - EAS_I8 c; - EAS_BOOL eof; - EAS_INT temp; - - pData = (S_IMELODY_DATA*) pInstData; - if (pData->state >= EAS_STATE_OPEN) - return EAS_SUCCESS; - - /* initialize MIDI channel when the track starts playing */ - if (pData->time == 0) - { - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Event: Reset\n"); */ } -#endif - /* set program to square lead */ - VMProgramChange(pEASData->pVoiceMgr, pData->pSynth, IMELODY_CHANNEL, IMELODY_PROGRAM); - - /* set channel volume to max */ - VMControlChange(pEASData->pVoiceMgr, pData->pSynth, IMELODY_CHANNEL, 7, 127); - } - - /* check for end of note */ - if (pData->note) - { - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Stopping note %d\n", pData->note); */ } -#endif - /* stop the note */ - VMStopNote(pEASData->pVoiceMgr, pData->pSynth, IMELODY_CHANNEL, pData->note, 0); - pData->note = 0; - - /* check for rest between notes */ - if (pData->restTicks) - { - pData->time += pData->restTicks; - pData->restTicks = 0; - return EAS_SUCCESS; - } - } - - /* parse the next event */ - eof = EAS_FALSE; - while (!eof) - { - - /* get next character */ - c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_FALSE); - - switch (c) - { - /* start repeat */ - case '(': - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter repeat section\n", c); */ } -#endif - - if (pData->repeatOffset < 0) - { - pData->repeatOffset = pData->startLine + (EAS_I32) pData->index; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Repeat offset = %d\n", pData->repeatOffset); */ } -#endif - } - else - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Ignoring nested repeat section\n"); */ } - break; - - /* end repeat */ - case ')': - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "End repeat section, repeat offset = %d\n", pData->repeatOffset); */ } -#endif - /* ignore invalid repeats */ - if (pData->repeatCount >= 0) - { - - /* decrement repeat count (repeatCount == 0 means infinite loop) */ - if (pData->repeatCount > 0) - { - if (--pData->repeatCount == 0) - { - pData->repeatCount = -1; -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Repeat loop complete\n"); */ } -#endif - } - } - -//2 TEMPORARY FIX: If locating, don't do infinite loops. -//3 We need a different mode for metadata parsing where we don't loop at all - if ((parserMode == eParserModePlay) || (pData->repeatCount != 0)) - { - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Rewinding file for repeat\n"); */ } -#endif - /* rewind to start of loop */ - if ((result = EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->repeatOffset)) != EAS_SUCCESS) - return result; - IMY_ReadLine(pEASData->hwInstData, pData->fileHandle, pData->buffer, &pData->startLine); - pData->index = 0; - - /* if last loop, prevent future loops */ - if (pData->repeatCount == -1) - pData->repeatOffset = -1; - } - } - break; - - /* repeat count */ - case '@': - if (!IMY_GetNumber(pEASData->hwInstData, pData, &temp, EAS_FALSE)) - eof = EAS_TRUE; - else if (pData->repeatOffset > 0) - { - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Repeat count = %dt", pData->repeatCount); */ } -#endif - if (pData->repeatCount < 0) - pData->repeatCount = (EAS_I16) temp; - } - break; - - /* volume */ - case 'V': - if (!IMY_GetVolume(pEASData->hwInstData, pData, EAS_FALSE)) - eof = EAS_TRUE; - break; - - /* flat */ - case '&': - pData->noteModifier = -1; - break; - - /* sharp */ - case '#': - pData->noteModifier = +1; - break; - - /* octave */ - case '*': - c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_FALSE); - if (IsDigit(c)) - pData->octave = (EAS_U8) ((c - '0' + 1) * 12); - else if (!c) - eof = EAS_TRUE; - break; - - /* ledon or ledoff */ - case 'l': - if (!IMY_GetLEDState(pEASData, pData)) - eof = EAS_TRUE; - break; - - /* vibeon or vibeoff */ - case 'v': - if (!IMY_GetVibeState(pEASData, pData)) - eof = EAS_TRUE; - break; - - /* either a B note or backon or backoff */ - case 'b': - if (IMY_GetNextChar(pEASData->hwInstData, pData, EAS_FALSE) == 'a') - { - if (!IMY_GetBackState(pEASData, pData)) - eof = EAS_TRUE; - } - else - { - PutBackChar(pData); - if (IMY_PlayNote(pEASData, pData, c, parserMode)) - return EAS_SUCCESS; - eof = EAS_TRUE; - } - break; - - /* rest */ - case 'r': - case 'R': - if (IMY_PlayRest(pEASData, pData)) - return EAS_SUCCESS; - eof = EAS_TRUE; - break; - - /* EOF */ - case 0: -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Event: end of iMelody file detected\n"); */ } -#endif - eof = EAS_TRUE; - break; - - /* must be a note */ - default: - c = ToLower(c); - if ((c >= 'a') && (c <= 'g')) - { - if (IMY_PlayNote(pEASData, pData, c, parserMode)) - return EAS_SUCCESS; - eof = EAS_TRUE; - } - else - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Ignoring unexpected character '%c' [0x%02x]\n", c, c); */ } - break; - } - } - - /* handle EOF */ -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Event: state set to EAS_STATE_STOPPING\n"); */ } -#endif - pData->state = EAS_STATE_STOPPING; - VMReleaseAllVoices(pEASData->pVoiceMgr, pData->pSynth); - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * IMY_State() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current state of the stream - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * pState - pointer to variable to store state - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) common decoder interface - pEASData not used */ -static EAS_RESULT IMY_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pState) -{ - S_IMELODY_DATA* pData; - - /* establish pointer to instance data */ - pData = (S_IMELODY_DATA*) pInstData; - - /* if stopping, check to see if synth voices are active */ - if (pData->state == EAS_STATE_STOPPING) - { - if (VMActiveVoices(pData->pSynth) == 0) - { - pData->state = EAS_STATE_STOPPED; -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_State: state set to EAS_STATE_STOPPED\n"); */ } -#endif - } - } - - if (pData->state == EAS_STATE_PAUSING) - { - if (VMActiveVoices(pData->pSynth) == 0) - { -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_State: state set to EAS_STATE_PAUSED\n"); */ } -#endif - pData->state = EAS_STATE_PAUSED; - } - } - - /* return current state */ - *pState = pData->state; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * IMY_Close() - *---------------------------------------------------------------------------- - * Purpose: - * Close the file and clean up - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT IMY_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_IMELODY_DATA* pData; - EAS_RESULT result; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Close: close file\n"); */ } -#endif - - pData = (S_IMELODY_DATA*) pInstData; - - /* close the file */ - if ((result = EAS_HWCloseFile(pEASData->hwInstData, pData->fileHandle)) != EAS_SUCCESS) - return result; - - /* free the synth */ - if (pData->pSynth != NULL) - VMMIDIShutdown(pEASData, pData->pSynth); - - /* if using dynamic memory, free it */ - if (!pEASData->staticMemoryModel) - EAS_HWFree(pEASData->hwInstData, pData); - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * IMY_Reset() - *---------------------------------------------------------------------------- - * Purpose: - * Reset the sequencer. Used for locating backwards in the file. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT IMY_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_IMELODY_DATA* pData; - EAS_RESULT result; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Reset: reset file\n"); */ } -#endif - pData = (S_IMELODY_DATA*) pInstData; - - /* reset the synth */ - VMReset(pEASData->pVoiceMgr, pData->pSynth, EAS_TRUE); - - /* reset time to zero */ - pData->time = 0; - pData->note = 0; - - /* reset file position and re-parse header */ - pData->state = EAS_STATE_ERROR; - if ((result = EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->fileOffset)) != EAS_SUCCESS) - return result; - if ((result = IMY_ParseHeader (pEASData, pData)) != EAS_SUCCESS) - return result; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Reset: state set to EAS_STATE_ERROR\n"); */ } -#endif - - pData->state = EAS_STATE_READY; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * IMY_Pause() - *---------------------------------------------------------------------------- - * Purpose: - * Pauses the sequencer. Mutes all voices and sets state to pause. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT IMY_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_IMELODY_DATA *pData; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Pause: pause file\n"); */ } -#endif - - /* can't pause a stopped stream */ - pData = (S_IMELODY_DATA*) pInstData; - if (pData->state == EAS_STATE_STOPPED) - return EAS_ERROR_ALREADY_STOPPED; - - /* mute the synthesizer */ - VMMuteAllVoices(pEASData->pVoiceMgr, pData->pSynth); - pData->state = EAS_STATE_PAUSING; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * IMY_Resume() - *---------------------------------------------------------------------------- - * Purpose: - * Resume playing after a pause, sets state back to playing. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) common decoder interface - pEASData not used */ -static EAS_RESULT IMY_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_IMELODY_DATA *pData; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Resume: resume file\n"); */ } -#endif - - /* can't resume a stopped stream */ - pData = (S_IMELODY_DATA*) pInstData; - if (pData->state == EAS_STATE_STOPPED) - return EAS_ERROR_ALREADY_STOPPED; - - /* nothing to do but resume playback */ - pData->state = EAS_STATE_PLAY; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * IMY_SetData() - *---------------------------------------------------------------------------- - * Purpose: - * Adjust tempo relative to song tempo - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * pInstData - pointer to iMelody instance data - * rate - rate (28-bit fractional amount) - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) common decoder interface - pEASData not used */ -static EAS_RESULT IMY_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value) -{ - S_IMELODY_DATA *pData; - - pData = (S_IMELODY_DATA*) pInstData; - switch (param) - { - - /* set metadata callback */ - case PARSER_DATA_METADATA_CB: - EAS_HWMemCpy(&pData->metadata, (void*) value, sizeof(S_METADATA_CB)); - break; - - default: - return EAS_ERROR_INVALID_PARAMETER; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * IMY_GetData() - *---------------------------------------------------------------------------- - * Purpose: - * Return the file type - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * pInstData - pointer to iMelody instance data - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) common decoder interface - pEASData not used */ -static EAS_RESULT IMY_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue) -{ - S_IMELODY_DATA *pData; - - pData = (S_IMELODY_DATA*) pInstData; - - switch (param) - { - /* return file type as iMelody */ - case PARSER_DATA_FILE_TYPE: - *pValue = EAS_FILE_IMELODY; - break; - - case PARSER_DATA_SYNTH_HANDLE: - *pValue = (EAS_I32) pData->pSynth; - break; - - case PARSER_DATA_GAIN_OFFSET: - *pValue = IMELODY_GAIN_OFFSET; - break; - - default: - return EAS_ERROR_INVALID_PARAMETER; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * IMY_PlayNote() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL IMY_PlayNote (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData, EAS_I8 note, EAS_INT parserMode) -{ - EAS_I32 duration; - EAS_U8 velocity; - - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_PlayNote: start note %d\n", note); */ } -#endif - - /* get the duration */ - if (!IMY_GetDuration(pEASData->hwInstData, pData, &duration)) - return EAS_FALSE; - - /* save note value */ - pData->note = (EAS_U8) (pData->octave + noteTable[note - 'a'] + pData->noteModifier); - velocity = (EAS_U8) (pData->volume ? pData->volume * IMELODY_VEL_MUL + IMELODY_VEL_OFS : 0); - - /* start note only if in play mode */ - if (parserMode == eParserModePlay) - VMStartNote(pEASData->pVoiceMgr, pData->pSynth, IMELODY_CHANNEL, pData->note, velocity); - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_PlayNote: Start note %d, duration %d\n", pData->note, duration); */ } -#endif - - /* determine note length */ - switch (pData->style) - { - case 0: - /*lint -e{704} shift for performance */ - pData->restTicks = duration >> 4; - break; - case 1: - pData->restTicks = 0; - break; - case 2: - /*lint -e{704} shift for performance */ - pData->restTicks = duration >> 1; - break; - default: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "IMY_PlayNote: Note style out of range: %d\n", pData->style); */ } - /*lint -e{704} shift for performance */ - pData->restTicks = duration >> 4; - break; - } - - /* next event is at end of this note */ - pData->time += duration - pData->restTicks; - - /* reset the flat/sharp modifier */ - pData->noteModifier = 0; - - return EAS_TRUE; -} - -/*---------------------------------------------------------------------------- - * IMY_PlayRest() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL IMY_PlayRest (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData) -{ - EAS_I32 duration; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_PlayRest]n"); */ } -#endif - - /* get the duration */ - if (!IMY_GetDuration(pEASData->hwInstData, pData, &duration)) - return EAS_FALSE; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_PlayRest: note duration %d\n", duration); */ } -#endif - - /* next event is at end of this note */ - pData->time += duration; - return EAS_TRUE; -} - -/*---------------------------------------------------------------------------- - * IMY_GetDuration() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ - -static EAS_BOOL IMY_GetDuration (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_I32 *pDuration) -{ - EAS_I32 duration; - EAS_I8 c; - - /* get the duration */ - *pDuration = 0; - c = IMY_GetNextChar(hwInstData, pData, EAS_FALSE); - if (!c) - return EAS_FALSE; - if ((c < '0') || (c > '5')) - { -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetDuration: error in duration '%c'\n", c); */ } -#endif - return EAS_FALSE; - } - - /* calculate total length of note */ - duration = pData->tick * (1 << ('5' - c)); - - /* check for duration modifier */ - c = IMY_GetNextChar(hwInstData, pData, EAS_FALSE); - if (c) - { - if (c == '.') - /*lint -e{704} shift for performance */ - duration += duration >> 1; - else if (c == ':') - /*lint -e{704} shift for performance */ - duration += (duration >> 1) + (duration >> 2); - else if (c == ';') - /*lint -e{704} shift for performance */ - duration = (duration * TRIPLET_MULTIPLIER) >> TRIPLET_SHIFT; - else - PutBackChar(pData); - } - - *pDuration = duration; - return EAS_TRUE; -} - -/*---------------------------------------------------------------------------- - * IMY_GetLEDState() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL IMY_GetLEDState (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData) -{ - EAS_I8 c; - EAS_INT i; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_GetLEDState\n"); */ } -#endif - - for (i = 0; i < 5; i++) - { - c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_FALSE); - if (!c) - return EAS_FALSE; - switch (i) - { - case 3: - if (c == 'n') - { -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetLEDState: LED on\n"); */ } -#endif - EAS_HWLED(pEASData->hwInstData, EAS_TRUE); - return EAS_TRUE; - } - else if (c != 'f') - return EAS_FALSE; - break; - - case 4: - if (c == 'f') - { -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetLEDState: LED off\n"); */ } -#endif - EAS_HWLED(pEASData->hwInstData, EAS_FALSE); - return EAS_TRUE; - } - return EAS_FALSE; - - default: - if (c != ledStr[i]) - return EAS_FALSE; - break; - } - } - return EAS_FALSE; -} - -/*---------------------------------------------------------------------------- - * IMY_GetVibeState() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL IMY_GetVibeState (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData) -{ - EAS_I8 c; - EAS_INT i; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_GetVibeState\n"); */ } -#endif - - for (i = 0; i < 6; i++) - { - c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_FALSE); - if (!c) - return EAS_FALSE; - switch (i) - { - case 4: - if (c == 'n') - { -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetVibeState: vibrate on\n"); */ } -#endif - EAS_HWVibrate(pEASData->hwInstData, EAS_TRUE); - return EAS_TRUE; - } - else if (c != 'f') - return EAS_FALSE; - break; - - case 5: - if (c == 'f') - { -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetVibeState: vibrate off\n"); */ } -#endif - EAS_HWVibrate(pEASData->hwInstData, EAS_FALSE); - return EAS_TRUE; - } - return EAS_FALSE; - - default: - if (c != vibeStr[i]) - return EAS_FALSE; - break; - } - } - return EAS_FALSE; -} - -/*---------------------------------------------------------------------------- - * IMY_GetBackState() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL IMY_GetBackState (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData) -{ - EAS_I8 c; - EAS_INT i; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_GetBackState\n"); */ } -#endif - - for (i = 0; i < 5; i++) - { - c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_FALSE); - if (!c) - return EAS_FALSE; - switch (i) - { - case 3: - if (c == 'n') - { -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetBackState: backlight on\n"); */ } -#endif - EAS_HWBackLight(pEASData->hwInstData, EAS_TRUE); - return EAS_TRUE; - } - else if (c != 'f') - return EAS_FALSE; - break; - - case 4: - if (c == 'f') - { -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetBackState: backlight off\n"); */ } -#endif - EAS_HWBackLight(pEASData->hwInstData, EAS_FALSE); - return EAS_TRUE; - } - return EAS_FALSE; - - default: - if (c != backStr[i]) - return EAS_FALSE; - break; - } - } - return EAS_FALSE; -} - -/*---------------------------------------------------------------------------- - * IMY_GetVolume() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL IMY_GetVolume (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_BOOL inHeader) -{ - EAS_INT temp; - EAS_I8 c; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_GetVolume\n"); */ } -#endif - - c = IMY_GetNextChar(hwInstData, pData, inHeader); - if (c == '+') - { - if (pData->volume < 15) - pData->volume++; - return EAS_TRUE; - } - else if (c == '-') - { - if (pData->volume > 0) - pData->volume--; - return EAS_TRUE; - } - else if (IsDigit(c)) - temp = c - '0'; - else - return EAS_FALSE; - - c = IMY_GetNextChar(hwInstData, pData, inHeader); - if (IsDigit(c)) - temp = temp * 10 + c - '0'; - else if (c) - PutBackChar(pData); - if ((temp >= 0) && (temp <= 15)) - { - if (inHeader && (temp == 0)) - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Ignoring V0 encountered in header\n"); */ } - else - pData->volume = (EAS_U8) temp; - } - return EAS_TRUE; -} - -/*---------------------------------------------------------------------------- - * IMY_GetNumber() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL IMY_GetNumber (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_INT *temp, EAS_BOOL inHeader) -{ - EAS_BOOL ok; - EAS_I8 c; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_GetNumber\n"); */ } -#endif - - *temp = 0; - ok = EAS_FALSE; - for (;;) - { - c = IMY_GetNextChar(hwInstData, pData, inHeader); - if (IsDigit(c)) - { - *temp = *temp * 10 + c - '0'; - ok = EAS_TRUE; - } - else - { - if (c) - PutBackChar(pData); - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetNumber: value %d\n", *temp); */ } -#endif - - return ok; - } - } -} - -/*---------------------------------------------------------------------------- - * IMY_GetVersion() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL IMY_GetVersion (S_IMELODY_DATA *pData, EAS_INT *pVersion) -{ - EAS_I8 c; - EAS_INT temp; - EAS_INT version; - - version = temp = 0; - for (;;) - { - c = pData->buffer[pData->index++]; - if ((c == 0) || (c == '.')) - { - /*lint -e{701} use shift for performance */ - version = (version << 8) + temp; - if (c == 0) - { - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetVersion: version 0x%04x\n", version); */ } -#endif - - *pVersion = version; - return EAS_TRUE; - } - temp = 0; - } - else if (IsDigit(c)) - temp = (temp * 10) + c - '0'; - } -} - -/*---------------------------------------------------------------------------- - * IMY_MetaData() - *---------------------------------------------------------------------------- - * Purpose: - * Prepare to parse the file. Allocates instance data (or uses static allocation for - * static memory model). - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static void IMY_MetaData (S_IMELODY_DATA *pData, E_EAS_METADATA_TYPE metaType, EAS_I8 *buffer) -{ - EAS_I32 len; - - /* check for callback */ - if (!pData->metadata.callback) - return; - - /* copy data to host buffer */ - len = (EAS_I32) strlen((char*) buffer); - if (len >pData->metadata.bufferSize) - len = pData->metadata.bufferSize; - strncpy((char*) pData->metadata.buffer, (char*) buffer, (size_t) len); - pData->metadata.buffer[len] = 0; - - /* callback to host */ - pData->metadata.callback(metaType, pData->metadata.buffer, pData->metadata.pUserData); -} - -/*---------------------------------------------------------------------------- - * IMY_ParseHeader() - *---------------------------------------------------------------------------- - * Purpose: - * Prepare to parse the file. Allocates instance data (or uses static allocation for - * static memory model). - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT IMY_ParseHeader (S_EAS_DATA *pEASData, S_IMELODY_DATA* pData) -{ - EAS_RESULT result; - EAS_INT token; - EAS_INT temp; - EAS_I8 c; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_ParseHeader\n"); */ } -#endif - - /* initialize some defaults */ - pData->time = 0; - pData->tick = DEFAULT_TICK_CONV; - pData->note = 0; - pData->noteModifier = 0; - pData ->restTicks = 0; - pData->volume = 7; - pData->octave = 60; - pData->repeatOffset = -1; - pData->repeatCount = -1; - pData->style = 0; - - /* force the read of the first line */ - pData->index = 1; - - /* read data until we get to melody */ - for (;;) - { - /* read a line from the file and parse the token */ - if (pData->index != 0) - { - if ((result = IMY_ReadLine(pEASData->hwInstData, pData->fileHandle, pData->buffer, &pData->startLine)) != EAS_SUCCESS) - { -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_ParseHeader: IMY_ReadLine returned %d\n", result); */ } -#endif - return result; - } - } - token = IMY_ParseLine(pData->buffer, &pData->index); - - switch (token) - { - /* ignore these valid tokens */ - case TOKEN_BEGIN: - break; - - case TOKEN_FORMAT: - if (!IMY_GetVersion(pData, &temp)) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Invalid FORMAT field '%s'\n", pData->buffer); */ } - return EAS_ERROR_FILE_FORMAT; - } - if ((temp != 0x0100) && (temp != 0x0200)) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Unsupported FORMAT %02x\n", temp); */ } - return EAS_ERROR_UNRECOGNIZED_FORMAT; - } - break; - - case TOKEN_VERSION: - if (!IMY_GetVersion(pData, &temp)) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Invalid VERSION field '%s'\n", pData->buffer); */ } - return EAS_ERROR_FILE_FORMAT; - } - if ((temp != 0x0100) && (temp != 0x0102)) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Unsupported VERSION %02x\n", temp); */ } - return EAS_ERROR_UNRECOGNIZED_FORMAT; - } - break; - - case TOKEN_NAME: - IMY_MetaData(pData, EAS_METADATA_TITLE, pData->buffer + pData->index); - break; - - case TOKEN_COMPOSER: - IMY_MetaData(pData, EAS_METADATA_AUTHOR, pData->buffer + pData->index); - break; - - /* handle beat */ - case TOKEN_BEAT: - IMY_GetNumber(pEASData->hwInstData, pData, &temp, EAS_TRUE); - if ((temp >= 25) && (temp <= 900)) - pData->tick = TICK_CONVERT / temp; - break; - - /* handle style */ - case TOKEN_STYLE: - c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_TRUE); - if (c == 'S') - c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_TRUE); - if ((c >= '0') && (c <= '2')) - pData->style = (EAS_U8) (c - '0'); - else - { - PutBackChar(pData); - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Error in style command: %s\n", pData->buffer); */ } - } - break; - - /* handle volume */ - case TOKEN_VOLUME: - c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_TRUE); - if (c != 'V') - { - PutBackChar(pData); - if (!IsDigit(c)) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Error in volume command: %s\n", pData->buffer); */ } - break; - } - } - IMY_GetVolume(pEASData->hwInstData, pData, EAS_TRUE); - break; - - case TOKEN_MELODY: -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Header successfully parsed\n"); */ } -#endif - return EAS_SUCCESS; - - case TOKEN_END: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Unexpected END:IMELODY encountered\n"); */ } - return EAS_ERROR_FILE_FORMAT; - - default: - /* force a read of the next line */ - pData->index = 1; - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Ignoring unrecognized token in iMelody file: %s\n", pData->buffer); */ } - break; - } - } -} - -/*---------------------------------------------------------------------------- - * IMY_GetNextChar() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_I8 IMY_GetNextChar (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_BOOL inHeader) -{ - EAS_I8 c; - EAS_U8 index; - - for (;;) - { - /* get next character */ - c = pData->buffer[pData->index++]; - - /* buffer empty, read more */ - if (!c) - { - /* don't read the next line in the header */ - if (inHeader) - return 0; - - pData->index = 0; - pData->buffer[0] = 0; - if (IMY_ReadLine(hwInstData, pData->fileHandle, pData->buffer, &pData->startLine) != EAS_SUCCESS) - { -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetNextChar: EOF\n"); */ } -#endif - return 0; - } - - /* check for END:IMELODY token */ - if (IMY_ParseLine(pData->buffer, &index) == TOKEN_END) - { -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetNextChar: found END:IMELODY\n"); */ } -#endif - pData->buffer[0] = 0; - return 0; - } - continue; - } - - /* ignore white space */ - if (!IsSpace(c)) - { - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetNextChar returned '%c'\n", c); */ } -#endif - return c; - } - } -} - -/*---------------------------------------------------------------------------- - * IMY_ReadLine() - *---------------------------------------------------------------------------- - * Purpose: - * Reads a line of input from the file, discarding the CR/LF - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT IMY_ReadLine (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE fileHandle, EAS_I8 *buffer, EAS_I32 *pStartLine) -{ - EAS_RESULT result; - EAS_INT i; - EAS_I8 c; - - /* fetch current file position and save it */ - if (pStartLine != NULL) - { - if ((result = EAS_HWFilePos(hwInstData, fileHandle, pStartLine)) != EAS_SUCCESS) - { -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_ParseHeader: EAS_HWFilePos returned %d\n", result); */ } -#endif - return result; - } - } - - buffer[0] = 0; - for (i = 0; i < MAX_LINE_SIZE; ) - { - if ((result = EAS_HWGetByte(hwInstData, fileHandle, &c)) != EAS_SUCCESS) - { - if ((result == EAS_EOF) && (i > 0)) - break; - return result; - } - - /* return on LF or end of data */ - if (c == '\n') - break; - - /* store characters in buffer */ - if (c != '\r') - buffer[i++] = c; - } - buffer[i] = 0; - -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_ReadLine read %s\n", buffer); */ } -#endif - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * IMY_ParseLine() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_INT IMY_ParseLine (EAS_I8 *buffer, EAS_U8 *pIndex) -{ - EAS_INT i; - EAS_INT j; - - /* there's no strnicmp() in stdlib, so we have to roll our own */ - for (i = 0; i < TOKEN_INVALID; i++) - { - for (j = 0; ; j++) - { - /* end of token, must be a match */ - if (tokens[i][j] == 0) - { -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_ParseLine found token %d\n", i); */ } -#endif - *pIndex = (EAS_U8) j; - return i; - } - if (tokens[i][j] != ToUpper(buffer[j])) - break; - } - } -#ifdef _DEBUG_IMELODY - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_ParseLine: no token found\n"); */ } -#endif - return TOKEN_INVALID; -} - diff --git a/arm-hybrid-22k/lib_src/eas_imelodydata.c b/arm-hybrid-22k/lib_src/eas_imelodydata.c deleted file mode 100644 index e72dc0b..0000000 --- a/arm-hybrid-22k/lib_src/eas_imelodydata.c +++ /dev/null @@ -1,43 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_imelodydata.c - * - * Contents and purpose: - * SMF File Parser - * - * This file contains data definitions for the SMF parser. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 547 $ - * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_types.h" -#include "eas_imelodydata.h" - -/*---------------------------------------------------------------------------- - * - * eas_iMelodyData - * - * Static memory allocation for iMelody parser - *---------------------------------------------------------------------------- -*/ -S_IMELODY_DATA eas_iMelodyData; - diff --git a/arm-hybrid-22k/lib_src/eas_imelodydata.h b/arm-hybrid-22k/lib_src/eas_imelodydata.h deleted file mode 100644 index 303b8f6..0000000 --- a/arm-hybrid-22k/lib_src/eas_imelodydata.h +++ /dev/null @@ -1,73 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_imelodydata.h - * - * Contents and purpose: - * SMF File Parser - * - * This file contains data declarations for the iMelody parser. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 778 $ - * $Date: 2007-07-23 16:45:17 -0700 (Mon, 23 Jul 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef EAS_IMELODYDATA_H -#define EAS_IMELODYDATA_H - -#include "eas_data.h" - -/* maximum line size as specified in iMelody V1.2 spec */ -#define MAX_LINE_SIZE 75 - -/*---------------------------------------------------------------------------- - * - * S_IMELODY_DATA - * - * This structure contains the state data for the iMelody parser - *---------------------------------------------------------------------------- -*/ - -typedef struct -{ - EAS_FILE_HANDLE fileHandle; /* file handle */ - S_SYNTH *pSynth; /* pointer to synth */ - EAS_I32 fileOffset; /* offset to start of data */ - EAS_I32 time; /* current time in 256ths of a msec */ - EAS_I32 tickBase; /* basline length of 32nd note in 256th of a msec */ - EAS_I32 tick; /* actual length of 32nd note in 256th of a msec */ - EAS_I32 restTicks; /* ticks to rest after current note */ - EAS_I32 startLine; /* file offset at start of line (for repeats) */ - EAS_I32 repeatOffset; /* file offset to start of repeat section */ - S_METADATA_CB metadata; /* metadata callback */ - EAS_I16 repeatCount; /* repeat counter */ - EAS_U8 state; /* current state EAS_STATE_XXXX */ - EAS_U8 style; /* from STYLE */ - EAS_U8 index; /* index into buffer */ - EAS_U8 octave; /* octave prefix */ - EAS_U8 volume; /* current volume */ - EAS_U8 note; /* MIDI note number */ - EAS_I8 noteModifier; /* sharp or flat */ - EAS_I8 buffer[MAX_LINE_SIZE+1]; /* buffer for ASCII data */ -} S_IMELODY_DATA; - -#endif - - diff --git a/arm-hybrid-22k/lib_src/eas_math.c b/arm-hybrid-22k/lib_src/eas_math.c deleted file mode 100644 index 12d788e..0000000 --- a/arm-hybrid-22k/lib_src/eas_math.c +++ /dev/null @@ -1,168 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_math.c - * - * Contents and purpose: - * Contains common math routines for the various audio engines. - * - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 586 $ - * $Date: 2007-03-08 20:33:04 -0800 (Thu, 08 Mar 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas.h" -#include "eas_math.h" - -/* anything less than this converts to a fraction too small to represent in 32-bits */ -#define MIN_CENTS -18000 - -/*---------------------------------------------------------------------------- - * EAS_Calculate2toX() - *---------------------------------------------------------------------------- - * Purpose: - * Calculate 2^x - * - * Inputs: - * nCents - measured in cents - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * nResult - int.frac result (where frac has NUM_DENTS_FRAC_BITS) - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_I32 EAS_Calculate2toX (EAS_I32 nCents) -{ - EAS_I32 nDents; - EAS_I32 nExponentInt, nExponentFrac; - EAS_I32 nTemp1, nTemp2; - EAS_I32 nResult; - - /* check for minimum value */ - if (nCents < MIN_CENTS) - return 0; - - /* for the time being, convert cents to dents */ - nDents = FMUL_15x15(nCents, CENTS_TO_DENTS); - - nExponentInt = GET_DENTS_INT_PART(nDents); - nExponentFrac = GET_DENTS_FRAC_PART(nDents); - - /* - implement 2^(fracPart) as a power series - */ - nTemp1 = GN2_TO_X2 + MULT_DENTS_COEF(nExponentFrac, GN2_TO_X3); - nTemp2 = GN2_TO_X1 + MULT_DENTS_COEF(nExponentFrac, nTemp1); - nTemp1 = GN2_TO_X0 + MULT_DENTS_COEF(nExponentFrac, nTemp2); - - /* - implement 2^(intPart) as - a left shift for intPart >= 0 or - a left shift for intPart < 0 - */ - if (nExponentInt >= 0) - { - /* left shift for positive exponents */ - /*lint -e{703} */ - nResult = nTemp1 << nExponentInt; - } - else - { - /* right shift for negative exponents */ - nExponentInt = -nExponentInt; - nResult = nTemp1 >> nExponentInt; - } - - return nResult; -} - -/*---------------------------------------------------------------------------- - * EAS_LogToLinear16() - *---------------------------------------------------------------------------- - * Purpose: - * Transform log value to linear gain multiplier using piece-wise linear - * approximation - * - * Inputs: - * nGain - log scale value in 20.10 format. Even though gain is normally - * stored in 6.10 (16-bit) format we use 32-bit numbers here to eliminate - * the need for saturation checking when combining gain values. - * - * Outputs: - * Returns a 16-bit linear value approximately equal to 2^(nGain/1024) - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_U16 EAS_LogToLinear16 (EAS_I32 nGain) -{ - EAS_INT nExp; - EAS_U16 nTemp; - - /* bias to positive */ - nGain += 32767; - - /* check for infinite attenuation */ - if (nGain < 0) - return 0; - - /* extract the exponent */ - nExp = 31 - (nGain >> 10); - - /* check for maximum output */ - if (nExp < 0) - return 0x7fff; - - /* extract mantissa and restore implied 1 bit */ - nTemp = (EAS_U16)((((nGain & 0x3ff) << 4) | 0x4000) >> nExp); - - /* use shift to approximate power-of-2 operation */ - return nTemp; -} - -/*---------------------------------------------------------------------------- - * EAS_VolumeToGain() - *---------------------------------------------------------------------------- - * Purpose: - * Transform volume control in 1dB increments to gain multiplier - * - * Inputs: - * volume - 100 = 0dB, 99 = -1dB, 0 = -inf - * - * Outputs: - * Returns a 16-bit linear value - *---------------------------------------------------------------------------- -*/ -EAS_I16 EAS_VolumeToGain (EAS_INT volume) -{ - /* check for limits */ - if (volume <= 0) - return 0; - if (volume >= 100) - return 0x7fff; - - /*lint -e{702} use shift instead of division */ - return (EAS_I16) EAS_Calculate2toX((((volume - EAS_MAX_VOLUME) * 204099) >> 10) - 1); -} - diff --git a/arm-hybrid-22k/lib_src/eas_math.h b/arm-hybrid-22k/lib_src/eas_math.h deleted file mode 100644 index 719270b..0000000 --- a/arm-hybrid-22k/lib_src/eas_math.h +++ /dev/null @@ -1,412 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_math.h - * - * Contents and purpose: - * Contains common math routines for the various audio engines. - * - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 584 $ - * $Date: 2007-03-08 09:49:24 -0800 (Thu, 08 Mar 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_MATH_H -#define _EAS_MATH_H - - -/** coefs for pan, generates sin, cos */ -#define COEFF_PAN_G2 -27146 /* -0.82842712474619 = 2 - 4/sqrt(2) */ -#define COEFF_PAN_G0 23170 /* 0.707106781186547 = 1/sqrt(2) */ - -/* -coefficients for approximating -2^x = gn2toX0 + gn2toX1*x + gn2toX2*x^2 + gn2toX3*x^3 -where x is a int.frac number representing number of octaves. -Actually, we approximate only the 2^(frac) using the power series -and implement the 2^(int) as a shift, so that -2^x == 2^(int.frac) == 2^(int) * 2^(fract) - == (gn2toX0 + gn2toX1*x + gn2toX2*x^2 + gn2toX3*x^3) << (int) - -The gn2toX.. were generated using a best fit for a 3rd -order polynomial, instead of taking the coefficients from -a truncated Taylor (or Maclaurin?) series. -*/ - -#define GN2_TO_X0 32768 /* 1 */ -#define GN2_TO_X1 22833 /* 0.696807861328125 */ -#define GN2_TO_X2 7344 /* 0.22412109375 */ -#define GN2_TO_X3 2588 /* 0.0789794921875 */ - -/*---------------------------------------------------------------------------- - * Fixed Point Math - *---------------------------------------------------------------------------- - * These macros are used for fixed point multiplies. If the processor - * supports fixed point multiplies, replace these macros with inline - * assembly code to improve performance. - *---------------------------------------------------------------------------- -*/ - -/* Fixed point multiply 0.15 x 0.15 = 0.15 returned as 32-bits */ -#define FMUL_15x15(a,b) \ - /*lint -e(704) */ \ - (((EAS_I32)(a) * (EAS_I32)(b)) >> 15) - -/* Fixed point multiply 0.7 x 0.7 = 0.15 returned as 32-bits */ -#define FMUL_7x7(a,b) \ - /*lint -e(704) */ \ - (((EAS_I32)(a) * (EAS_I32)(b) ) << 1) - -/* Fixed point multiply 0.8 x 0.8 = 0.15 returned as 32-bits */ -#define FMUL_8x8(a,b) \ - /*lint -e(704) */ \ - (((EAS_I32)(a) * (EAS_I32)(b) ) >> 1) - -/* Fixed point multiply 0.8 x 1.15 = 0.15 returned as 32-bits */ -#define FMUL_8x15(a,b) \ - /*lint -e(704) */ \ - (((EAS_I32)((a) << 7) * (EAS_I32)(b)) >> 15) - -/* macros for fractional phase accumulator */ -/* -Note: changed the _U32 to _I32 on 03/14/02. This should not -affect the phase calculations, and should allow us to reuse these -macros for other audio sample related math. -*/ -#define HARDWARE_BIT_WIDTH 32 - -#define NUM_PHASE_INT_BITS 1 -#define NUM_PHASE_FRAC_BITS 15 - -#define PHASE_FRAC_MASK (EAS_U32) ((0x1L << NUM_PHASE_FRAC_BITS) -1) - -#define GET_PHASE_INT_PART(x) (EAS_U32)((EAS_U32)(x) >> NUM_PHASE_FRAC_BITS) -#define GET_PHASE_FRAC_PART(x) (EAS_U32)((EAS_U32)(x) & PHASE_FRAC_MASK) - -#define DEFAULT_PHASE_FRAC 0 -#define DEFAULT_PHASE_INT 0 - -/* -Linear interpolation calculates: -output = (1-frac) * sample[n] + (frac) * sample[n+1] - -where conceptually 0 <= frac < 1 - -For a fixed point implementation, frac is actually an integer value -with an implied binary point one position to the left. The value of -one (unity) is given by PHASE_ONE -one half and one quarter are useful for 4-point linear interp. -*/ -#define PHASE_ONE (EAS_I32) (0x1L << NUM_PHASE_FRAC_BITS) - -/* - Multiply the signed audio sample by the unsigned fraction. -- a is the signed audio sample -- b is the unsigned fraction (cast to signed int as long as coef - uses (n-1) or less bits, where n == hardware bit width) -*/ -#define MULT_AUDIO_COEF(audio,coef) /*lint -e704 */ \ - (EAS_I32)( \ - ( \ - ((EAS_I32)(audio)) * ((EAS_I32)(coef)) \ - ) \ - >> NUM_PHASE_FRAC_BITS \ - ) \ - /* lint +704 */ - -/* wet / dry calculation macros */ -#define NUM_WET_DRY_FRAC_BITS 7 // 15 -#define NUM_WET_DRY_INT_BITS 9 // 1 - -/* define a 1.0 */ -#define WET_DRY_ONE (EAS_I32) ((0x1L << NUM_WET_DRY_FRAC_BITS)) -#define WET_DRY_MINUS_ONE (EAS_I32) (~WET_DRY_ONE) -#define WET_DRY_FULL_SCALE (EAS_I32) (WET_DRY_ONE - 1) - -#define MULT_AUDIO_WET_DRY_COEF(audio,coef) /*lint -e(702) */ \ - (EAS_I32)( \ - ( \ - ((EAS_I32)(audio)) * ((EAS_I32)(coef)) \ - ) \ - >> NUM_WET_DRY_FRAC_BITS \ - ) - -/* Envelope 1 (EG1) calculation macros */ -#define NUM_EG1_INT_BITS 1 -#define NUM_EG1_FRAC_BITS 15 - -/* the max positive gain used in the synth for EG1 */ -/* SYNTH_FULL_SCALE_EG1_GAIN must match the value in the dls2eas -converter, otherwise, the values we read from the .eas file are bogus. */ -#define SYNTH_FULL_SCALE_EG1_GAIN (EAS_I32) ((0x1L << NUM_EG1_FRAC_BITS) -1) - -/* define a 1.0 */ -#define EG1_ONE (EAS_I32) ((0x1L << NUM_EG1_FRAC_BITS)) -#define EG1_MINUS_ONE (EAS_I32) (~SYNTH_FULL_SCALE_EG1_GAIN) - -#define EG1_HALF (EAS_I32) (EG1_ONE/2) -#define EG1_MINUS_HALF (EAS_I32) (EG1_MINUS_ONE/2) - -/* -We implement the EG1 using a linear gain value, which means that the -attack segment is handled by incrementing (adding) the linear gain. -However, EG1 treats the Decay, Sustain, and Release differently than -the Attack portion. For Decay, Sustain, and Release, the gain is -linear on dB scale, which is equivalent to exponential damping on -a linear scale. Because we use a linear gain for EG1, we implement -the Decay and Release as multiplication (instead of incrementing -as we did for the attack segment). -Therefore, we need the following macro to implement the multiplication -(i.e., exponential damping) during the Decay and Release segments of -the EG1 -*/ -#define MULT_EG1_EG1(gain,damping) /*lint -e(704) */ \ - (EAS_I32)( \ - ( \ - ((EAS_I32)(gain)) * ((EAS_I32)(damping)) \ - ) \ - >> NUM_EG1_FRAC_BITS \ - ) - -// Use the following macro specifically for the filter, when multiplying -// the b1 coefficient. The 0 <= |b1| < 2, which therefore might overflow -// in certain conditions because we store b1 as a 1.15 value. -// Instead, we could store b1 as b1p (b1' == b1 "prime") where -// b1p == b1/2, thus ensuring no potential overflow for b1p because -// 0 <= |b1p| < 1 -// However, during the filter calculation, we must account for the fact -// that we are using b1p instead of b1, and thereby multiply by -// an extra factor of 2. Rather than multiply by an extra factor of 2, -// we can instead shift the result right by one less, hence the -// modified shift right value of (NUM_EG1_FRAC_BITS -1) -#define MULT_EG1_EG1_X2(gain,damping) /*lint -e(702) */ \ - (EAS_I32)( \ - ( \ - ((EAS_I32)(gain)) * ((EAS_I32)(damping)) \ - ) \ - >> (NUM_EG1_FRAC_BITS -1) \ - ) - -#define SATURATE_EG1(x) /*lint -e{734} saturation operation */ \ - ((EAS_I32)(x) > SYNTH_FULL_SCALE_EG1_GAIN) ? (SYNTH_FULL_SCALE_EG1_GAIN) : \ - ((EAS_I32)(x) < EG1_MINUS_ONE) ? (EG1_MINUS_ONE) : (x); - - -/* use "digital cents" == "dents" instead of cents */ -/* we coudl re-use the phase frac macros, but if we do, -we must change the phase macros to cast to _I32 instead of _U32, -because using a _U32 cast causes problems when shifting the exponent -for the 2^x calculation, because right shift a negative values MUST -be sign extended, or else the 2^x calculation is wrong */ - -/* use "digital cents" == "dents" instead of cents */ -#define NUM_DENTS_FRAC_BITS 12 -#define NUM_DENTS_INT_BITS (HARDWARE_BIT_WIDTH - NUM_DENTS_FRAC_BITS) - -#define DENTS_FRAC_MASK (EAS_I32) ((0x1L << NUM_DENTS_FRAC_BITS) -1) - -#define GET_DENTS_INT_PART(x) /*lint -e(704) */ \ - (EAS_I32)((EAS_I32)(x) >> NUM_DENTS_FRAC_BITS) - -#define GET_DENTS_FRAC_PART(x) (EAS_I32)((EAS_I32)(x) & DENTS_FRAC_MASK) - -#define DENTS_ONE (EAS_I32) (0x1L << NUM_DENTS_FRAC_BITS) - -/* use CENTS_TO_DENTS to convert a value in cents to dents */ -#define CENTS_TO_DENTS (EAS_I32) (DENTS_ONE * (0x1L << NUM_EG1_FRAC_BITS) / 1200L) \ - - -/* -For gain, the LFO generates a value that modulates in terms -of dB. However, we use a linear gain value, so we must convert -the LFO value in dB to a linear gain. Normally, we would use -linear gain = 10^x, where x = LFO value in dB / 20. -Instead, we implement 10^x using our 2^x approximation. -because - - 10^x = 2^(log2(10^x)) = 2^(x * log2(10)) - -so we need to multiply by log2(10) which is just a constant. -Ah, but just wait -- our 2^x actually doesn't exactly implement -2^x, but it actually assumes that the input is in cents, and within -the 2^x approximation converts its input from cents to octaves -by dividing its input by 1200. - -So, in order to convert the LFO gain value in dB to something -that our existing 2^x approximation can use, multiply the LFO gain -by log2(10) * 1200 / 20 - -The divide by 20 helps convert dB to linear gain, and we might -as well incorporate that operation into this conversion. -Of course, we need to keep some fractional bits, so multiply -the constant by NUM_EG1_FRAC_BITS -*/ - -/* use LFO_GAIN_TO_CENTS to convert the LFO gain value to cents */ -#if 0 -#define DOUBLE_LOG2_10 (double) (3.32192809488736) /* log2(10) */ - -#define DOUBLE_LFO_GAIN_TO_CENTS (double) \ - ( \ - (DOUBLE_LOG2_10) * \ - 1200.0 / \ - 20.0 \ - ) - -#define LFO_GAIN_TO_CENTS (EAS_I32) \ - ( \ - DOUBLE_LFO_GAIN_TO_CENTS * \ - (0x1L << NUM_EG1_FRAC_BITS) \ - ) -#endif - -#define LFO_GAIN_TO_CENTS (EAS_I32) (1671981156L >> (23 - NUM_EG1_FRAC_BITS)) - - -#define MULT_DENTS_COEF(dents,coef) /*lint -e704 */ \ - (EAS_I32)( \ - ( \ - ((EAS_I32)(dents)) * ((EAS_I32)(coef)) \ - ) \ - >> NUM_DENTS_FRAC_BITS \ - ) \ - /* lint +e704 */ - -/* we use 16-bits in the PC per audio sample */ -#define BITS_PER_AUDIO_SAMPLE 16 - -/* we define 1 as 1.0 - 1 LSbit */ -#define DISTORTION_ONE (EAS_I32)((0x1L << (BITS_PER_AUDIO_SAMPLE-1)) -1) -#define DISTORTION_MINUS_ONE (EAS_I32)(~DISTORTION_ONE) - -/* drive coef is given as int.frac */ -#define NUM_DRIVE_COEF_INT_BITS 1 -#define NUM_DRIVE_COEF_FRAC_BITS 4 - -#define MULT_AUDIO_DRIVE(audio,drive) /*lint -e(702) */ \ - (EAS_I32) ( \ - ( \ - ((EAS_I32)(audio)) * ((EAS_I32)(drive)) \ - ) \ - >> NUM_DRIVE_COEF_FRAC_BITS \ - ) - -#define MULT_AUDIO_AUDIO(audio1,audio2) /*lint -e(702) */ \ - (EAS_I32) ( \ - ( \ - ((EAS_I32)(audio1)) * ((EAS_I32)(audio2)) \ - ) \ - >> (BITS_PER_AUDIO_SAMPLE-1) \ - ) - -#define SATURATE(x) \ - ((((EAS_I32)(x)) > DISTORTION_ONE) ? (DISTORTION_ONE) : \ - (((EAS_I32)(x)) < DISTORTION_MINUS_ONE) ? (DISTORTION_MINUS_ONE) : ((EAS_I32)(x))); - - - -/*---------------------------------------------------------------------------- - * EAS_Calculate2toX() - *---------------------------------------------------------------------------- - * Purpose: - * Calculate 2^x - * - * Inputs: - * nCents - measured in cents - * - * Outputs: - * nResult - int.frac result (where frac has NUM_DENTS_FRAC_BITS) - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_I32 EAS_Calculate2toX (EAS_I32 nCents); - -/*---------------------------------------------------------------------------- - * EAS_LogToLinear16() - *---------------------------------------------------------------------------- - * Purpose: - * Transform log value to linear gain multiplier using piece-wise linear - * approximation - * - * Inputs: - * nGain - log scale value in 20.10 format. Even though gain is normally - * stored in 6.10 (16-bit) format we use 32-bit numbers here to eliminate - * the need for saturation checking when combining gain values. - * - * Outputs: - * Returns a 16-bit linear value approximately equal to 2^(nGain/1024) - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_U16 EAS_LogToLinear16 (EAS_I32 nGain); - -/*---------------------------------------------------------------------------- - * EAS_VolumeToGain() - *---------------------------------------------------------------------------- - * Purpose: - * Transform volume control in 1dB increments to gain multiplier - * - * Inputs: - * volume - 100 = 0dB, 99 = -1dB, 0 = -inf - * - * Outputs: - * Returns a 16-bit linear value - *---------------------------------------------------------------------------- -*/ -EAS_I16 EAS_VolumeToGain (EAS_INT volume); - -/*---------------------------------------------------------------------------- - * EAS_fsqrt() - *---------------------------------------------------------------------------- - * Purpose: - * Calculates the square root of a 32-bit fixed point value - * - * Inputs: - * n = value of interest - * - * Outputs: - * returns the square root of n - * - *---------------------------------------------------------------------------- -*/ -EAS_U16 EAS_fsqrt (EAS_U32 n); - -/*---------------------------------------------------------------------------- - * EAS_flog2() - *---------------------------------------------------------------------------- - * Purpose: - * Calculates the log2 of a 32-bit fixed point value - * - * Inputs: - * n = value of interest - * - * Outputs: - * returns the log2 of n - * - *---------------------------------------------------------------------------- -*/ -EAS_I32 EAS_flog2 (EAS_U32 n); - -#endif - diff --git a/arm-hybrid-22k/lib_src/eas_midi.c b/arm-hybrid-22k/lib_src/eas_midi.c deleted file mode 100644 index 08aed72..0000000 --- a/arm-hybrid-22k/lib_src/eas_midi.c +++ /dev/null @@ -1,569 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_midi.c - * - * Contents and purpose: - * This file implements the MIDI stream parser. It is called by eas_smf.c to parse MIDI messages - * that are streamed out of the file. It can also parse live MIDI streams. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 794 $ - * $Date: 2007-08-01 00:08:48 -0700 (Wed, 01 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_data.h" -#include "eas_report.h" -#include "eas_miditypes.h" -#include "eas_midi.h" -#include "eas_vm_protos.h" -#include "eas_parser.h" - -#ifdef JET_INTERFACE -#include "jet_data.h" -#endif - - -/* state enumerations for ProcessSysExMessage */ -typedef enum -{ - eSysEx, - eSysExUnivNonRealTime, - eSysExUnivNrtTargetID, - eSysExGMControl, - eSysExUnivRealTime, - eSysExUnivRtTargetID, - eSysExDeviceControl, - eSysExMasterVolume, - eSysExMasterVolLSB, - eSysExSPMIDI, - eSysExSPMIDIchan, - eSysExSPMIDIMIP, - eSysExMfgID1, - eSysExMfgID2, - eSysExMfgID3, - eSysExEnhancer, - eSysExEnhancerSubID, - eSysExEnhancerFeedback1, - eSysExEnhancerFeedback2, - eSysExEnhancerDrive, - eSysExEnhancerWet, - eSysExEOX, - eSysExIgnore -} E_SYSEX_STATES; - -/* local prototypes */ -static EAS_RESULT ProcessMIDIMessage (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_INT parserMode); -static EAS_RESULT ProcessSysExMessage (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_U8 c, EAS_INT parserMode); - -/*---------------------------------------------------------------------------- - * EAS_InitMIDIStream() - *---------------------------------------------------------------------------- - * Purpose: - * Initializes the MIDI stream state for parsing. - * - * Inputs: - * - * Outputs: - * returns EAS_RESULT (EAS_SUCCESS is OK) - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void EAS_InitMIDIStream (S_MIDI_STREAM *pMIDIStream) -{ - pMIDIStream->byte3 = EAS_FALSE; - pMIDIStream->pending = EAS_FALSE; - pMIDIStream->runningStatus = 0; - pMIDIStream->status = 0; -} - -/*---------------------------------------------------------------------------- - * EAS_ParseMIDIStream() - *---------------------------------------------------------------------------- - * Purpose: - * Parses a MIDI input stream character by character. Characters are pushed (rather than pulled) - * so the interface works equally well for both file and stream I/O. - * - * Inputs: - * c - character from MIDI stream - * - * Outputs: - * returns EAS_RESULT (EAS_SUCCESS is OK) - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_ParseMIDIStream (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_U8 c, EAS_INT parserMode) -{ - - /* check for new status byte */ - if (c & 0x80) - { - /* save new running status */ - if (c < 0xf8) - { - pMIDIStream->runningStatus = c; - pMIDIStream->byte3 = EAS_FALSE; - - /* deal with SysEx */ - if ((c == 0xf7) || (c == 0xf0)) - { - if (parserMode == eParserModeMetaData) - return EAS_SUCCESS; - return ProcessSysExMessage(pEASData, pSynth, pMIDIStream, c, parserMode); - } - - /* inform the file parser that we're in the middle of a message */ - if ((c < 0xf4) || (c > 0xf6)) - pMIDIStream->pending = EAS_TRUE; - } - - /* real-time message - ignore it */ - return EAS_SUCCESS; - } - - /* 3rd byte of a 3-byte message? */ - if (pMIDIStream->byte3) - { - pMIDIStream->d2 = c; - pMIDIStream->byte3 = EAS_FALSE; - pMIDIStream->pending = EAS_FALSE; - if (parserMode == eParserModeMetaData) - return EAS_SUCCESS; - return ProcessMIDIMessage(pEASData, pSynth, pMIDIStream, parserMode); - } - - /* check for status received */ - if (pMIDIStream->runningStatus) - { - - /* save new status and data byte */ - pMIDIStream->status = pMIDIStream->runningStatus; - - /* check for 3-byte messages */ - if (pMIDIStream->status < 0xc0) - { - pMIDIStream->d1 = c; - pMIDIStream->pending = EAS_TRUE; - pMIDIStream->byte3 = EAS_TRUE; - return EAS_SUCCESS; - } - - /* check for 2-byte messages */ - if (pMIDIStream->status < 0xe0) - { - pMIDIStream->d1 = c; - pMIDIStream->pending = EAS_FALSE; - if (parserMode == eParserModeMetaData) - return EAS_SUCCESS; - return ProcessMIDIMessage(pEASData, pSynth, pMIDIStream, parserMode); - } - - /* check for more 3-bytes message */ - if (pMIDIStream->status < 0xf0) - { - pMIDIStream->d1 = c; - pMIDIStream->pending = EAS_TRUE; - pMIDIStream->byte3 = EAS_TRUE; - return EAS_SUCCESS; - } - - /* SysEx message? */ - if (pMIDIStream->status == 0xF0) - { - if (parserMode == eParserModeMetaData) - return EAS_SUCCESS; - return ProcessSysExMessage(pEASData, pSynth, pMIDIStream, c, parserMode); - } - - /* remaining messages all clear running status */ - pMIDIStream->runningStatus = 0; - - /* F2 is 3-byte message */ - if (pMIDIStream->status == 0xf2) - { - pMIDIStream->byte3 = EAS_TRUE; - return EAS_SUCCESS; - } - } - - /* no status byte received, provide a warning, but we should be able to recover */ - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Received MIDI data without a valid status byte: %d\n",c); */ } - pMIDIStream->pending = EAS_FALSE; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * ProcessMIDIMessage() - *---------------------------------------------------------------------------- - * Purpose: - * This function processes a typical MIDI message. All of the data has been received, just need - * to take appropriate action. - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ProcessMIDIMessage (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_INT parserMode) -{ - EAS_U8 channel; - - channel = pMIDIStream->status & 0x0f; - switch (pMIDIStream->status & 0xf0) - { - case 0x80: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"NoteOff: %02x %02x %02x\n", - pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ } - if (parserMode < eParserModeMute) - VMStopNote(pEASData->pVoiceMgr, pSynth, channel, pMIDIStream->d1, pMIDIStream->d2); - break; - - case 0x90: - if (pMIDIStream->d2) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"NoteOn: %02x %02x %02x\n", - pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ } - pMIDIStream->flags |= MIDI_FLAG_FIRST_NOTE; - if (parserMode == eParserModePlay) - VMStartNote(pEASData->pVoiceMgr, pSynth, channel, pMIDIStream->d1, pMIDIStream->d2); - } - else - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"NoteOff: %02x %02x %02x\n", - pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ } - if (parserMode < eParserModeMute) - VMStopNote(pEASData->pVoiceMgr, pSynth, channel, pMIDIStream->d1, pMIDIStream->d2); - } - break; - - case 0xa0: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"PolyPres: %02x %02x %02x\n", - pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ } - break; - - case 0xb0: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"Control: %02x %02x %02x\n", - pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ } - if (parserMode < eParserModeMute) - VMControlChange(pEASData->pVoiceMgr, pSynth, channel, pMIDIStream->d1, pMIDIStream->d2); -#ifdef JET_INTERFACE - if (pMIDIStream->jetData & MIDI_FLAGS_JET_CB) - { - JET_Event(pEASData, pMIDIStream->jetData & (JET_EVENT_SEG_MASK | JET_EVENT_TRACK_MASK), - channel, pMIDIStream->d1, pMIDIStream->d2); - } -#endif - break; - - case 0xc0: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"Program: %02x %02x\n", - pMIDIStream->status, pMIDIStream->d1); */ } - if (parserMode < eParserModeMute) - VMProgramChange(pEASData->pVoiceMgr, pSynth, channel, pMIDIStream->d1); - break; - - case 0xd0: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"ChanPres: %02x %02x\n", - pMIDIStream->status, pMIDIStream->d1); */ } - if (parserMode < eParserModeMute) - VMChannelPressure(pSynth, channel, pMIDIStream->d1); - break; - - case 0xe0: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"PBend: %02x %02x %02x\n", - pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ } - if (parserMode < eParserModeMute) - VMPitchBend(pSynth, channel, pMIDIStream->d1, pMIDIStream->d2); - break; - - default: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"Unknown: %02x %02x %02x\n", - pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ } - } - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * ProcessSysExMessage() - *---------------------------------------------------------------------------- - * Purpose: - * Process a SysEx character byte from the MIDI stream. Since we cannot - * simply wait for the next character to arrive, we are forced to save - * state after each character. It would be easier to parse at the file - * level, but then we lose the nice feature of being able to support - * these messages in a real-time MIDI stream. - * - * Inputs: - * pEASData - pointer to synthesizer instance data - * c - character to be processed - * locating - if true, the sequencer is relocating to a new position - * - * Outputs: - * - * - * Side Effects: - * - * Notes: - * These are the SysEx messages we can receive: - * - * SysEx messages - * { f0 7e 7f 09 01 f7 } GM 1 On - * { f0 7e 7f 09 02 f7 } GM 1/2 Off - * { f0 7e 7f 09 03 f7 } GM 2 On - * { f0 7f 7f 04 01 lsb msb } Master Volume - * { f0 7f 7f 0b 01 ch mip [ch mip ...] f7 } SP-MIDI - * { f0 00 01 3a 04 01 fdbk1 fdbk2 drive wet dry f7 } Enhancer - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ProcessSysExMessage (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_U8 c, EAS_INT parserMode) -{ - - /* check for start byte */ - if (c == 0xf0) - { - pMIDIStream->sysExState = eSysEx; - } - /* check for end byte */ - else if (c == 0xf7) - { - /* if this was a MIP message, update the MIP table */ - if ((pMIDIStream->sysExState == eSysExSPMIDIchan) && (parserMode != eParserModeMetaData)) - VMUpdateMIPTable(pEASData->pVoiceMgr, pSynth); - pMIDIStream->sysExState = eSysExIgnore; - } - - /* process SysEx message */ - else - { - switch (pMIDIStream->sysExState) - { - case eSysEx: - - /* first byte, determine message class */ - switch (c) - { - case 0x7e: - pMIDIStream->sysExState = eSysExUnivNonRealTime; - break; - case 0x7f: - pMIDIStream->sysExState = eSysExUnivRealTime; - break; - case 0x00: - pMIDIStream->sysExState = eSysExMfgID1; - break; - default: - pMIDIStream->sysExState = eSysExIgnore; - break; - } - break; - - /* process GM message */ - case eSysExUnivNonRealTime: - if (c == 0x7f) - pMIDIStream->sysExState = eSysExUnivNrtTargetID; - else - pMIDIStream->sysExState = eSysExIgnore; - break; - - case eSysExUnivNrtTargetID: - if (c == 0x09) - pMIDIStream->sysExState = eSysExGMControl; - else - pMIDIStream->sysExState = eSysExIgnore; - break; - - case eSysExGMControl: - if ((c == 1) || (c == 3)) - { - /* GM 1 or GM2 On, reset synth */ - if (parserMode != eParserModeMetaData) - { - pMIDIStream->flags |= MIDI_FLAG_GM_ON; - VMReset(pEASData->pVoiceMgr, pSynth, EAS_FALSE); - VMInitMIPTable(pSynth); - } - pMIDIStream->sysExState = eSysExEOX; - } - else - pMIDIStream->sysExState = eSysExIgnore; - break; - - /* Process Master Volume and SP-MIDI */ - case eSysExUnivRealTime: - if (c == 0x7f) - pMIDIStream->sysExState = eSysExUnivRtTargetID; - else - pMIDIStream->sysExState = eSysExIgnore; - break; - - case eSysExUnivRtTargetID: - if (c == 0x04) - pMIDIStream->sysExState = eSysExDeviceControl; - else if (c == 0x0b) - pMIDIStream->sysExState = eSysExSPMIDI; - else - pMIDIStream->sysExState = eSysExIgnore; - break; - - /* process master volume */ - case eSysExDeviceControl: - if (c == 0x01) - pMIDIStream->sysExState = eSysExMasterVolume; - else - pMIDIStream->sysExState = eSysExIgnore; - break; - - case eSysExMasterVolume: - /* save LSB */ - pMIDIStream->d1 = c; - pMIDIStream->sysExState = eSysExMasterVolLSB; - break; - - case eSysExMasterVolLSB: - if (parserMode != eParserModeMetaData) - { - EAS_I32 gain = ((EAS_I32) c << 8) | ((EAS_I32) pMIDIStream->d1 << 1); - gain = (gain * gain) >> 15; - VMSetVolume(pSynth, (EAS_U16) gain); - } - pMIDIStream->sysExState = eSysExEOX; - break; - - /* process SP-MIDI MIP message */ - case eSysExSPMIDI: - if (c == 0x01) - { - /* assume all channels are muted */ - if (parserMode != eParserModeMetaData) - VMInitMIPTable(pSynth); - pMIDIStream->d1 = 0; - pMIDIStream->sysExState = eSysExSPMIDIchan; - } - else - pMIDIStream->sysExState = eSysExIgnore; - break; - - case eSysExSPMIDIchan: - if (c < NUM_SYNTH_CHANNELS) - { - pMIDIStream->d2 = c; - pMIDIStream->sysExState = eSysExSPMIDIMIP; - } - else - { - /* bad MIP message - unmute channels */ - if (parserMode != eParserModeMetaData) - VMInitMIPTable(pSynth); - pMIDIStream->sysExState = eSysExIgnore; - } - break; - - case eSysExSPMIDIMIP: - /* process MIP entry here */ - if (parserMode != eParserModeMetaData) - VMSetMIPEntry(pEASData->pVoiceMgr, pSynth, pMIDIStream->d2, pMIDIStream->d1, c); - pMIDIStream->sysExState = eSysExSPMIDIchan; - - /* if 16 channels received, update MIP table */ - if (++pMIDIStream->d1 == NUM_SYNTH_CHANNELS) - { - if (parserMode != eParserModeMetaData) - VMUpdateMIPTable(pEASData->pVoiceMgr, pSynth); - pMIDIStream->sysExState = eSysExEOX; - } - break; - - /* process Enhancer */ - case eSysExMfgID1: - if (c == 0x01) - pMIDIStream->sysExState = eSysExMfgID1; - else - pMIDIStream->sysExState = eSysExIgnore; - break; - - case eSysExMfgID2: - if (c == 0x3a) - pMIDIStream->sysExState = eSysExMfgID1; - else - pMIDIStream->sysExState = eSysExIgnore; - break; - - case eSysExMfgID3: - if (c == 0x04) - pMIDIStream->sysExState = eSysExEnhancer; - else - pMIDIStream->sysExState = eSysExIgnore; - break; - - case eSysExEnhancer: - if (c == 0x01) - pMIDIStream->sysExState = eSysExEnhancerSubID; - else - pMIDIStream->sysExState = eSysExIgnore; - break; - - case eSysExEnhancerSubID: - pMIDIStream->sysExState = eSysExEnhancerFeedback1; - break; - - case eSysExEnhancerFeedback1: - pMIDIStream->sysExState = eSysExEnhancerFeedback2; - break; - - case eSysExEnhancerFeedback2: - pMIDIStream->sysExState = eSysExEnhancerDrive; - break; - - case eSysExEnhancerDrive: - pMIDIStream->sysExState = eSysExEnhancerWet; - break; - - case eSysExEnhancerWet: - pMIDIStream->sysExState = eSysExEOX; - break; - - case eSysExEOX: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Expected F7, received %02x\n", c); */ } - pMIDIStream->sysExState = eSysExIgnore; - break; - - case eSysExIgnore: - break; - - default: - pMIDIStream->sysExState = eSysExIgnore; - break; - } - } - - if (pMIDIStream->sysExState == eSysExIgnore) - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Ignoring SysEx byte %02x\n", c); */ } - return EAS_SUCCESS; -} /* end ProcessSysExMessage */ - diff --git a/arm-hybrid-22k/lib_src/eas_midi.h b/arm-hybrid-22k/lib_src/eas_midi.h deleted file mode 100644 index 37a03ee..0000000 --- a/arm-hybrid-22k/lib_src/eas_midi.h +++ /dev/null @@ -1,71 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_midi.h - * - * Contents and purpose: - * Prototypes for MIDI stream parsing functions - * - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_MIDI_H -#define _EAS_MIDI_H - -/*---------------------------------------------------------------------------- - * EAS_InitMIDIStream() - *---------------------------------------------------------------------------- - * Purpose: - * Initializes the MIDI stream state for parsing. - * - * Inputs: - * - * Outputs: - * returns EAS_RESULT (EAS_SUCCESS is OK) - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void EAS_InitMIDIStream (S_MIDI_STREAM *pMIDIStream); - -/*---------------------------------------------------------------------------- - * EAS_ParseMIDIStream() - *---------------------------------------------------------------------------- - * Purpose: - * Parses a MIDI input stream character by character. Characters are pushed (rather than pulled) - * so the interface works equally well for both file and stream I/O. - * - * Inputs: - * c - character from MIDI stream - * - * Outputs: - * returns EAS_RESULT (EAS_SUCCESS is OK) - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_ParseMIDIStream (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_U8 c, EAS_INT parserMode); - -#endif /* #define _EAS_MIDI_H */ - diff --git a/arm-hybrid-22k/lib_src/eas_midictrl.h b/arm-hybrid-22k/lib_src/eas_midictrl.h deleted file mode 100644 index 0c4217d..0000000 --- a/arm-hybrid-22k/lib_src/eas_midictrl.h +++ /dev/null @@ -1,64 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_midictrl.h - * - * Contents and purpose: - * MIDI controller definitions - * - * This header only contains declarations that are specific - * to this implementation. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_MIDICTRL_H -#define _EAS_MIDICTRL_H - -/* define controller types */ -/* - Note that these controller types are specified in base 10 (decimal) - and not in hexadecimal. The above midi messages are specified - in hexadecimal. -*/ -#define MIDI_CONTROLLER_BANK_SELECT 0 -#define MIDI_CONTROLLER_BANK_SELECT_MSB 0 -#define MIDI_CONTROLLER_MOD_WHEEL 1 -#define MIDI_CONTROLLER_ENTER_DATA_MSB 6 -#define MIDI_CONTROLLER_VOLUME 7 -#define MIDI_CONTROLLER_PAN 10 -#define MIDI_CONTROLLER_EXPRESSION 11 -#define MIDI_CONTROLLER_BANK_SELECT_LSB 32 -#define MIDI_CONTROLLER_ENTER_DATA_LSB 38 /* 0x26 */ -#define MIDI_CONTROLLER_SUSTAIN_PEDAL 64 -#define MIDI_CONTROLLER_SELECT_NRPN_LSB 98 -#define MIDI_CONTROLLER_SELECT_NRPN_MSB 99 -#define MIDI_CONTROLLER_SELECT_RPN_LSB 100 /* 0x64 */ -#define MIDI_CONTROLLER_SELECT_RPN_MSB 101 /* 0x65 */ -#define MIDI_CONTROLLER_ALL_SOUND_OFF 120 -#define MIDI_CONTROLLER_RESET_CONTROLLERS 121 -#define MIDI_CONTROLLER_ALL_NOTES_OFF 123 -#define MIDI_CONTROLLER_OMNI_OFF 124 -#define MIDI_CONTROLLER_OMNI_ON 125 -#define MIDI_CONTROLLER_MONO_ON_POLY_OFF 126 -#define MIDI_CONTROLLER_POLY_ON_MONO_OFF 127 - -#endif /* #ifndef _EAS_MIDICTRL_H */ diff --git a/arm-hybrid-22k/lib_src/eas_mididata.c b/arm-hybrid-22k/lib_src/eas_mididata.c deleted file mode 100644 index 2ee907e..0000000 --- a/arm-hybrid-22k/lib_src/eas_mididata.c +++ /dev/null @@ -1,34 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_mididata.c - * - * Contents and purpose: - * Data module for MIDI stream interface - * - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 547 $ - * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_miditypes.h" - -S_INTERACTIVE_MIDI eas_MIDIData; - diff --git a/arm-hybrid-22k/lib_src/eas_miditypes.h b/arm-hybrid-22k/lib_src/eas_miditypes.h deleted file mode 100644 index 0b7f96e..0000000 --- a/arm-hybrid-22k/lib_src/eas_miditypes.h +++ /dev/null @@ -1,138 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_miditypes.h - * - * Contents and purpose: - * Contains declarations for the MIDI stream parser. - * - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 778 $ - * $Date: 2007-07-23 16:45:17 -0700 (Mon, 23 Jul 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_MIDITYPES_H -#define _EAS_MIDITYPES_H - -#include "eas_data.h" -#include "eas_parser.h" - -/*---------------------------------------------------------------------------- - * S_MIDI_STREAM - * - * Maintains parser state for the MIDI stream parser - * - *---------------------------------------------------------------------------- -*/ - -typedef struct s_midi_stream_tag -{ - EAS_BOOL8 byte3; /* flag indicates 3rd byte expected */ - EAS_BOOL8 pending; /* flag indicates more data expected */ - EAS_U8 sysExState; /* maintains the SysEx state */ - EAS_U8 runningStatus; /* last running status received */ - EAS_U8 status; /* status byte */ - EAS_U8 d1; /* first data byte */ - EAS_U8 d2; /* second data byte */ - EAS_U8 flags; /* flags - see below for definition */ -#ifdef JET_INTERFACE - EAS_U32 jetData; /* JET data */ -#endif -} S_MIDI_STREAM; - -/* flags for S_MIDI_STREAM.flags */ -#define MIDI_FLAG_GM_ON 0x01 /* GM System On message received */ -#define MIDI_FLAG_FIRST_NOTE 0x02 /* first note received */ - -/* flags for S_MIDI_STREAM.jetFlags */ -#define MIDI_FLAGS_JET_MUTE 0x00000001 /* track is muted */ -#define MIDI_FLAGS_JET_CB 0x00000002 /* JET callback enabled */ - -/*---------------------------------------------------------------------------- - * - * S_SMF_STREAM - * - * This structure contains data required to parse an SMF stream. For SMF0 files, there - * will be a single instance of this per file. For SMF1 files, there will be multiple instance, - * one for each separate stream in the file. - * - *---------------------------------------------------------------------------- -*/ - -typedef struct s_smf_stream_tag -{ - EAS_FILE_HANDLE fileHandle; /* host wrapper file handle */ - EAS_U32 ticks; /* time of next event in stream */ - EAS_I32 startFilePos; /* start location of track within file */ - S_MIDI_STREAM midiStream; /* MIDI stream state */ -} S_SMF_STREAM; - -/*---------------------------------------------------------------------------- - * - * S_SMF_DATA - * - * This structure contains the instance data required to parse an SMF stream. - * - *---------------------------------------------------------------------------- -*/ - -typedef struct s_smf_data_tag -{ -#ifdef _CHECKED_BUILD - EAS_U32 handleCheck; /* signature check for checked build */ -#endif - S_SMF_STREAM *streams; /* pointer to individual streams in file */ - S_SMF_STREAM *nextStream; /* pointer to next stream with event */ - S_SYNTH *pSynth; /* pointer to synth */ - EAS_FILE_HANDLE fileHandle; /* file handle */ - S_METADATA_CB metadata; /* metadata callback */ - EAS_I32 fileOffset; /* for embedded files */ - EAS_I32 time; /* current time in milliseconds/256 */ - EAS_U16 numStreams; /* actual number of streams */ - EAS_U16 tickConv; /* current MIDI tick to msec conversion */ - EAS_U16 ppqn; /* ticks per quarter note */ - EAS_U8 state; /* current state EAS_STATE_XXXX */ - EAS_U8 flags; /* flags - see definitions below */ -} S_SMF_DATA; - -#define SMF_FLAGS_CHASE_MODE 0x01 /* chase mode - skip to first note */ -#define SMF_FLAGS_HAS_TIME_SIG 0x02 /* time signature encountered at time 0 */ -#define SMF_FLAGS_HAS_TEMPO 0x04 /* tempo encountered at time 0 */ -#define SMF_FLAGS_HAS_GM_ON 0x08 /* GM System On encountered at time 0 */ -#define SMF_FLAGS_JET_STREAM 0x80 /* JET in use - keep strict timing */ - -/* combo flags indicate setup bar */ -#define SMF_FLAGS_SETUP_BAR (SMF_FLAGS_HAS_TIME_SIG | SMF_FLAGS_HAS_TEMPO | SMF_FLAGS_HAS_GM_ON) - -/*---------------------------------------------------------------------------- - * Interactive MIDI structure - *---------------------------------------------------------------------------- -*/ -typedef struct s_interactive_midi_tag -{ -#ifdef _CHECKED_BUILD - EAS_U32 handleCheck; /* signature check for checked build */ -#endif - S_SYNTH *pSynth; /* pointer to synth */ - S_MIDI_STREAM stream; /* stream data */ -} S_INTERACTIVE_MIDI; - -#endif /* #ifndef _EAS_MIDITYPES_H */ - diff --git a/arm-hybrid-22k/lib_src/eas_mixbuf.c b/arm-hybrid-22k/lib_src/eas_mixbuf.c deleted file mode 100644 index 73e969a..0000000 --- a/arm-hybrid-22k/lib_src/eas_mixbuf.c +++ /dev/null @@ -1,36 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_mixbuf.c - * - * Contents and purpose: - * Contains a data allocation for synthesizer - * - * Copyright Sonic Network Inc. 2004 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -// includes -#include "eas_data.h" -#include "eas_mixer.h" - -// globals -EAS_I32 eas_MixBuffer[BUFFER_SIZE_IN_MONO_SAMPLES * NUM_OUTPUT_CHANNELS]; - diff --git a/arm-hybrid-22k/lib_src/eas_mixer.c b/arm-hybrid-22k/lib_src/eas_mixer.c deleted file mode 100644 index c4a2f9f..0000000 --- a/arm-hybrid-22k/lib_src/eas_mixer.c +++ /dev/null @@ -1,464 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_mixer.c - * - * Contents and purpose: - * This file contains the critical components of the mix engine that - * must be optimized for best performance. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 706 $ - * $Date: 2007-05-31 17:22:51 -0700 (Thu, 31 May 2007) $ - *---------------------------------------------------------------------------- -*/ - -//3 dls: This module is in the midst of being converted from a synth -//3 specific module to a general purpose mix engine - -/*------------------------------------ - * includes - *------------------------------------ -*/ -#include "eas_data.h" -#include "eas_host.h" -#include "eas_math.h" -#include "eas_mixer.h" -#include "eas_config.h" -#include "eas_report.h" - -#ifdef _MAXIMIZER_ENABLED -EAS_I32 MaximizerProcess (EAS_VOID_PTR pInstData, EAS_I32 *pSrc, EAS_I32 *pDst, EAS_I32 numSamples); -#endif - -/*------------------------------------ - * defines - *------------------------------------ -*/ - -/* need to boost stereo by ~3dB to compensate for the panner */ -#define STEREO_3DB_GAIN_BOOST 512 - -/*---------------------------------------------------------------------------- - * EAS_MixEngineInit() - *---------------------------------------------------------------------------- - * Purpose: - * Prepares the mix engine for work, allocates buffers, locates effects modules, etc. - * - * Inputs: - * pEASData - instance data - * pInstData - pointer to variable to receive instance data handle - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_MixEngineInit (S_EAS_DATA *pEASData) -{ - - /* check Configuration Module for mix buffer allocation */ - if (pEASData->staticMemoryModel) - pEASData->pMixBuffer = EAS_CMEnumData(EAS_CM_MIX_BUFFER); - else - pEASData->pMixBuffer = EAS_HWMalloc(pEASData->hwInstData, BUFFER_SIZE_IN_MONO_SAMPLES * NUM_OUTPUT_CHANNELS * sizeof(EAS_I32)); - if (pEASData->pMixBuffer == NULL) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Failed to allocate mix buffer memory\n"); */ } - return EAS_ERROR_MALLOC_FAILED; - } - EAS_HWMemSet((void *)(pEASData->pMixBuffer), 0, BUFFER_SIZE_IN_MONO_SAMPLES * NUM_OUTPUT_CHANNELS * sizeof(EAS_I32)); - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_MixEnginePrep() - *---------------------------------------------------------------------------- - * Purpose: - * Performs prep before synthesize a buffer of audio, such as clearing - * audio buffers, etc. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void EAS_MixEnginePrep (S_EAS_DATA *pEASData, EAS_I32 numSamples) -{ - - /* clear the mix buffer */ -#if (NUM_OUTPUT_CHANNELS == 2) - EAS_HWMemSet(pEASData->pMixBuffer, 0, numSamples * (EAS_I32) sizeof(long) * 2); -#else - EAS_HWMemSet(pEASData->pMixBuffer, 0, (EAS_I32) numSamples * (EAS_I32) sizeof(long)); -#endif - - /* need to clear other side-chain effect buffers (chorus & reverb) */ -} - -/*---------------------------------------------------------------------------- - * EAS_MixEnginePost - *---------------------------------------------------------------------------- - * Purpose: - * This routine does the post-processing after all voices have been - * synthesized. It calls any sweeteners and does the final mixdown to - * the output buffer. - * - * Inputs: - * - * Outputs: - * - * Notes: - *---------------------------------------------------------------------------- -*/ -void EAS_MixEnginePost (S_EAS_DATA *pEASData, EAS_I32 numSamples) -{ - EAS_U16 gain; - -//3 dls: Need to restore the mix engine metrics - - /* calculate the gain multiplier */ -#ifdef _MAXIMIZER_ENABLED - if (pEASData->effectsModules[EAS_MODULE_MAXIMIZER].effect) - { - EAS_I32 temp; - temp = MaximizerProcess(pEASData->effectsModules[EAS_MODULE_MAXIMIZER].effectData, pEASData->pMixBuffer, pEASData->pMixBuffer, numSamples); - temp = (temp * pEASData->masterGain) >> 15; - if (temp > 32767) - gain = 32767; - else - gain = (EAS_U16) temp; - } - else - gain = (EAS_U16) pEASData->masterGain; -#else - gain = (EAS_U16) pEASData->masterGain; -#endif - - /* Not using all the gain bits for now - * Reduce the input to the compressor by 6dB to prevent saturation - */ -#ifdef _COMPRESSOR_ENABLED - if (pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effectData) - gain = gain >> 5; - else - gain = gain >> 4; -#else - gain = gain >> 4; -#endif - - /* convert 32-bit mix buffer to 16-bit output format */ -#if (NUM_OUTPUT_CHANNELS == 2) - SynthMasterGain(pEASData->pMixBuffer, pEASData->pOutputAudioBuffer, gain, (EAS_U16) ((EAS_U16) numSamples * 2)); -#else - SynthMasterGain(pEASData->pMixBuffer, pEASData->pOutputAudioBuffer, gain, (EAS_U16) numSamples); -#endif - -#ifdef _ENHANCER_ENABLED - /* enhancer effect */ - if (pEASData->effectsModules[EAS_MODULE_ENHANCER].effectData) - (*pEASData->effectsModules[EAS_MODULE_ENHANCER].effect->pfProcess) - (pEASData->effectsModules[EAS_MODULE_ENHANCER].effectData, - pEASData->pOutputAudioBuffer, - pEASData->pOutputAudioBuffer, - numSamples); -#endif - -#ifdef _GRAPHIC_EQ_ENABLED - /* graphic EQ effect */ - if (pEASData->effectsModules[EAS_MODULE_GRAPHIC_EQ].effectData) - (*pEASData->effectsModules[EAS_MODULE_GRAPHIC_EQ].effect->pfProcess) - (pEASData->effectsModules[EAS_MODULE_GRAPHIC_EQ].effectData, - pEASData->pOutputAudioBuffer, - pEASData->pOutputAudioBuffer, - numSamples); -#endif - -#ifdef _COMPRESSOR_ENABLED - /* compressor effect */ - if (pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effectData) - (*pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effect->pfProcess) - (pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effectData, - pEASData->pOutputAudioBuffer, - pEASData->pOutputAudioBuffer, - numSamples); -#endif - -#ifdef _WOW_ENABLED - /* WOW requires a 32-bit buffer, borrow the mix buffer and - * pass it as the destination buffer - */ - /*lint -e{740} temporarily passing a parameter through an existing I/F */ - if (pEASData->effectsModules[EAS_MODULE_WOW].effectData) - (*pEASData->effectsModules[EAS_MODULE_WOW].effect->pfProcess) - (pEASData->effectsModules[EAS_MODULE_WOW].effectData, - pEASData->pOutputAudioBuffer, - (EAS_PCM*) pEASData->pMixBuffer, - numSamples); -#endif - -#ifdef _TONECONTROLEQ_ENABLED - /* ToneControlEQ effect */ - if (pEASData->effectsModules[EAS_MODULE_TONECONTROLEQ].effectData) - (*pEASData->effectsModules[EAS_MODULE_TONECONTROLEQ].effect->pfProcess) - (pEASData->effectsModules[EAS_MODULE_TONECONTROLEQ].effectData, - pEASData->pOutputAudioBuffer, - pEASData->pOutputAudioBuffer, - numSamples); -#endif - -#ifdef _REVERB_ENABLED - /* Reverb effect */ - if (pEASData->effectsModules[EAS_MODULE_REVERB].effectData) - (*pEASData->effectsModules[EAS_MODULE_REVERB].effect->pfProcess) - (pEASData->effectsModules[EAS_MODULE_REVERB].effectData, - pEASData->pOutputAudioBuffer, - pEASData->pOutputAudioBuffer, - numSamples); -#endif - -#ifdef _CHORUS_ENABLED - /* Chorus effect */ - if (pEASData->effectsModules[EAS_MODULE_CHORUS].effectData) - (*pEASData->effectsModules[EAS_MODULE_CHORUS].effect->pfProcess) - (pEASData->effectsModules[EAS_MODULE_CHORUS].effectData, - pEASData->pOutputAudioBuffer, - pEASData->pOutputAudioBuffer, - numSamples); -#endif - -} - -#ifndef NATIVE_EAS_KERNEL -/*---------------------------------------------------------------------------- - * SynthMasterGain - *---------------------------------------------------------------------------- - * Purpose: - * Mixes down audio from 32-bit to 16-bit target buffer - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -void SynthMasterGain (long *pInputBuffer, EAS_PCM *pOutputBuffer, EAS_U16 nGain, EAS_U16 numSamples) { - - /* loop through the buffer */ - while (numSamples--) { - long s; - - /* read a sample from the input buffer and add some guard bits */ - s = *pInputBuffer++; - - /* add some guard bits */ - /*lint -e{704} */ - s = s >> 7; - - /* apply master gain */ - s *= (long) nGain; - - /* shift to lower 16-bits */ - /*lint -e{704} */ - s = s >> 9; - - /* saturate */ - s = SATURATE(s); - - *pOutputBuffer++ = (EAS_PCM)s; - } -} -#endif - -/*---------------------------------------------------------------------------- - * EAS_MixEngineShutdown() - *---------------------------------------------------------------------------- - * Purpose: - * Shuts down effects modules and deallocates memory - * - * Inputs: - * pEASData - instance data - * pInstData - instance data handle - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_MixEngineShutdown (S_EAS_DATA *pEASData) -{ - - /* check Configuration Module for static memory allocation */ - if (!pEASData->staticMemoryModel && (pEASData->pMixBuffer != NULL)) - EAS_HWFree(pEASData->hwInstData, pEASData->pMixBuffer); - - return EAS_SUCCESS; -} - -#ifdef UNIFIED_MIXER -#ifndef NATIVE_MIX_STREAM -/*---------------------------------------------------------------------------- - * EAS_MixStream - *---------------------------------------------------------------------------- - * Mix a 16-bit stream into a 32-bit buffer - * - * pInputBuffer 16-bit input buffer - * pMixBuffer 32-bit mix buffer - * numSamples number of samples to mix - * gainLeft initial gain left or mono - * gainRight initial gain right - * gainLeft left gain increment per sample - * gainRight right gain increment per sample - * flags bit 0 = stereo source - * bit 1 = stereo output - *---------------------------------------------------------------------------- -*/ -void EAS_MixStream (EAS_PCM *pInputBuffer, EAS_I32 *pMixBuffer, EAS_I32 numSamples, EAS_I32 gainLeft, EAS_I32 gainRight, EAS_I32 gainIncLeft, EAS_I32 gainIncRight, EAS_I32 flags) -{ - EAS_I32 temp; - EAS_INT src, dest; - - /* NOTE: There are a lot of optimizations that can be done - * in the native implementations based on register - * availability, etc. For example, it may make sense to - * break this down into 8 separate routines: - * - * 1. Mono source to mono output - * 2. Mono source to stereo output - * 3. Stereo source to mono output - * 4. Stereo source to stereo output - * 5. Mono source to mono output - no gain change - * 6. Mono source to stereo output - no gain change - * 7. Stereo source to mono output - no gain change - * 8. Stereo source to stereo output - no gain change - * - * Other possibilities include loop unrolling, skipping - * a gain calculation every 2 or 4 samples, etc. - */ - - /* no gain change, use fast loops */ - if ((gainIncLeft == 0) && (gainIncRight == 0)) - { - switch (flags & (MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT)) - { - /* mono to mono */ - case 0: - gainLeft >>= 15; - for (src = dest = 0; src < numSamples; src++, dest++) - { - - pMixBuffer[dest] += (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS; - } - break; - - /* mono to stereo */ - case MIX_FLAGS_STEREO_OUTPUT: - gainLeft >>= 15; - gainRight >>= 15; - for (src = dest = 0; src < numSamples; src++, dest+=2) - { - pMixBuffer[dest] += (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS; - pMixBuffer[dest+1] += (pInputBuffer[src] * gainRight) >> NUM_MIXER_GUARD_BITS; - } - break; - - /* stereo to mono */ - case MIX_FLAGS_STEREO_SOURCE: - gainLeft >>= 15; - gainRight >>= 15; - for (src = dest = 0; src < numSamples; src+=2, dest++) - { - temp = (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS; - temp += ((pInputBuffer[src+1] * gainRight) >> NUM_MIXER_GUARD_BITS); - pMixBuffer[dest] += temp; - } - break; - - /* stereo to stereo */ - case MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT: - gainLeft >>= 15; - gainRight >>= 15; - for (src = dest = 0; src < numSamples; src+=2, dest+=2) - { - pMixBuffer[dest] += (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS; - pMixBuffer[dest+1] += (pInputBuffer[src+1] * gainRight) >> NUM_MIXER_GUARD_BITS; - } - break; - } - } - - /* gain change - do gain increment */ - else - { - switch (flags & (MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT)) - { - /* mono to mono */ - case 0: - for (src = dest = 0; src < numSamples; src++, dest++) - { - gainLeft += gainIncLeft; - pMixBuffer[dest] += (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS; - } - break; - - /* mono to stereo */ - case MIX_FLAGS_STEREO_OUTPUT: - for (src = dest = 0; src < numSamples; src++, dest+=2) - { - gainLeft += gainIncLeft; - gainRight += gainIncRight; - pMixBuffer[dest] += (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS; - pMixBuffer[dest+1] += (pInputBuffer[src] * (gainRight >> 15)) >> NUM_MIXER_GUARD_BITS; - } - break; - - /* stereo to mono */ - case MIX_FLAGS_STEREO_SOURCE: - for (src = dest = 0; src < numSamples; src+=2, dest++) - { - gainLeft += gainIncLeft; - gainRight += gainIncRight; - temp = (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS; - temp += ((pInputBuffer[src+1] * (gainRight >> 15)) >> NUM_MIXER_GUARD_BITS); - pMixBuffer[dest] += temp; - } - break; - - /* stereo to stereo */ - case MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT: - for (src = dest = 0; src < numSamples; src+=2, dest+=2) - { - gainLeft += gainIncLeft; - gainRight += gainIncRight; - pMixBuffer[dest] += (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS; - pMixBuffer[dest+1] += (pInputBuffer[src+1] * (gainRight >> 15)) >> NUM_MIXER_GUARD_BITS; - } - break; - } - } -} -#endif -#endif - diff --git a/arm-hybrid-22k/lib_src/eas_mixer.h b/arm-hybrid-22k/lib_src/eas_mixer.h deleted file mode 100644 index 2ba2d3d..0000000 --- a/arm-hybrid-22k/lib_src/eas_mixer.h +++ /dev/null @@ -1,137 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_mixer.h - * - * Contents and purpose: - * This file contains the critical components of the mix engine that - * must be optimized for best performance. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 706 $ - * $Date: 2007-05-31 17:22:51 -0700 (Thu, 31 May 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_MIXER_H -#define _EAS_MIXER_H - -//3 dls: This module is in the midst of being converted from a synth -//3 specific module to a general purpose mix engine - -#define MIX_FLAGS_STEREO_SOURCE 1 -#define MIX_FLAGS_STEREO_OUTPUT 2 -#define NUM_MIXER_GUARD_BITS 4 - -#include "eas_effects.h" - -extern void SynthMasterGain( long *pInputBuffer, EAS_PCM *pOutputBuffer, EAS_U16 nGain, EAS_U16 nNumLoopSamples); - -/*---------------------------------------------------------------------------- - * EAS_MixEngineInit() - *---------------------------------------------------------------------------- - * Purpose: - * Prepares the mix engine for work, allocates buffers, locates effects modules, etc. - * - * Inputs: - * pEASData - instance data - * pInstData - pointer to variable to receive instance data handle - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_MixEngineInit (EAS_DATA_HANDLE pEASData); - -/*---------------------------------------------------------------------------- - * EAS_MixEnginePrep() - *---------------------------------------------------------------------------- - * Purpose: - * Performs prep before synthesize a buffer of audio, such as clearing - * audio buffers, etc. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void EAS_MixEnginePrep (EAS_DATA_HANDLE pEASData, EAS_I32 nNumSamplesToAdd); - -/*---------------------------------------------------------------------------- - * EAS_MixEnginePost - *---------------------------------------------------------------------------- - * Purpose: - * This routine does the post-processing after all voices have been - * synthesized. It calls any sweeteners and does the final mixdown to - * the output buffer. - * - * Inputs: - * - * Outputs: - * - * Notes: - *---------------------------------------------------------------------------- -*/ -void EAS_MixEnginePost (EAS_DATA_HANDLE pEASData, EAS_I32 nNumSamplesToAdd); - -/*---------------------------------------------------------------------------- - * EAS_MixEngineShutdown() - *---------------------------------------------------------------------------- - * Purpose: - * Shuts down effects modules and deallocates memory - * - * Inputs: - * pEASData - instance data - * pInstData - instance data handle - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_MixEngineShutdown (EAS_DATA_HANDLE pEASData); - -#ifdef UNIFIED_MIXER -/*---------------------------------------------------------------------------- - * EAS_MixStream - *---------------------------------------------------------------------------- - * Mix a 16-bit stream into a 32-bit buffer - * - * pInputBuffer 16-bit input buffer - * pMixBuffer 32-bit mix buffer - * numSamples number of samples to mix - * gainLeft initial gain left or mono - * gainRight initial gain right - * gainLeft left gain increment per sample - * gainRight right gain increment per sample - * flags bit 0 = stereo source - * bit 1 = stereo output - *---------------------------------------------------------------------------- -*/ -void EAS_MixStream (EAS_PCM *pInputBuffer, EAS_I32 *pMixBuffer, EAS_I32 numSamples, EAS_I32 gainLeft, EAS_I32 gainRight, EAS_I32 gainIncLeft, EAS_I32 gainIncRight, EAS_I32 flags); -#endif - -#endif /* #ifndef _EAS_MIXER_H */ - diff --git a/arm-hybrid-22k/lib_src/eas_ota.c b/arm-hybrid-22k/lib_src/eas_ota.c deleted file mode 100644 index fb81d62..0000000 --- a/arm-hybrid-22k/lib_src/eas_ota.c +++ /dev/null @@ -1,1077 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_ota.c - * - * Contents and purpose: - * OTA parser - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 795 $ - * $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_data.h" -#include "eas_miditypes.h" -#include "eas_parser.h" -#include "eas_report.h" -#include "eas_host.h" -#include "eas_midi.h" -#include "eas_config.h" -#include "eas_vm_protos.h" -#include "eas_otadata.h" - -/* increase gain for mono ringtones */ -#define OTA_GAIN_OFFSET 8 - -/* file definitions */ -#define OTA_RINGTONE 0x25 -#define OTA_SOUND 0x1d -#define OTA_UNICODE 0x22 - -/* song type definitions */ -#define OTA_BASIC_SONG_TYPE 0x01 -#define OTA_TEMPORARY_SONG_TYPE 0x02 - -/* instruction ID coding */ -#define OTA_PATTERN_HEADER_ID 0x00 -#define OTA_NOTE_INST_ID 0x01 -#define OTA_SCALE_INST_ID 0x02 -#define OTA_STYLE_INST_ID 0x03 -#define OTA_TEMPO_INST_ID 0x04 -#define OTA_VOLUME_INST_ID 0x05 - -/* note durations */ -#define OTA_NORMAL_DURATION 0x00 -#define OTA_DOTTED_NOTE 0x01 -#define OTA_DOUBLE_DOTTED_NOTE 0x02 -#define OTA_TRIPLET_NOTE 0x03 - -/* loop count value for infinite loop */ -#define OTA_INFINITE_LOOP 0x0f - -/* length of 32nd note in 1/256ths of a msec for 63 BPM tempo */ -#define DEFAULT_TICK_CONV 30476 - -/* default channel and program for OTA playback */ -#define OTA_CHANNEL 0 -#define OTA_PROGRAM 80 -#define OTA_VEL_MUL 4 -#define OTA_VEL_OFS 67 -#define OTA_VEL_DEFAULT 95 - -/* multiplier for fixed point triplet conversion */ -#define TRIPLET_MULTIPLIER 683 -#define TRIPLET_SHIFT 10 - -/* local prototypes */ -static EAS_RESULT OTA_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset); -static EAS_RESULT OTA_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT OTA_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime); -static EAS_RESULT OTA_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode); -static EAS_RESULT OTA_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState); -static EAS_RESULT OTA_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT OTA_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT OTA_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT OTA_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT OTA_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value); -static EAS_RESULT OTA_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue); -static EAS_RESULT OTA_ParseHeader (S_EAS_DATA *pEASData, S_OTA_DATA* pData); -static EAS_RESULT OTA_FetchBitField (EAS_HW_DATA_HANDLE hwInstData, S_OTA_DATA *pData, EAS_I32 numBits, EAS_U8 *pValue); -static EAS_RESULT OTA_SavePosition (EAS_HW_DATA_HANDLE hwInstData, S_OTA_DATA *pData, S_OTA_LOC *pLoc); -static EAS_RESULT OTA_RestorePosition (EAS_HW_DATA_HANDLE hwInstData, S_OTA_DATA *pData, S_OTA_LOC *pLoc); - - -/*---------------------------------------------------------------------------- - * - * EAS_OTA_Parser - * - * This structure contains the functional interface for the OTA parser - *---------------------------------------------------------------------------- -*/ -const S_FILE_PARSER_INTERFACE EAS_OTA_Parser = -{ - OTA_CheckFileType, - OTA_Prepare, - OTA_Time, - OTA_Event, - OTA_State, - OTA_Close, - OTA_Reset, - OTA_Pause, - OTA_Resume, - NULL, - OTA_SetData, - OTA_GetData, - NULL -}; - -/*---------------------------------------------------------------------------- - * - * bpmTable - * - * BPM conversion table. Converts bpm values to 256ths of a millisecond for a 32nd note - *---------------------------------------------------------------------------- -*/ -static const EAS_U32 bpmTable[32] = -{ - 76800, 68571, 61935, 54857, - 48000, 42667, 38400, 34286, - 30476, 27429, 24000, 21333, - 19200, 17143, 15360, 13714, - 12000, 10667, 9600, 8533, - 7680, 6737, 6000, 5408, - 4800, 4267, 3840, 3398, - 3024, 2685, 2400, 2133 -}; - -/*---------------------------------------------------------------------------- - * OTA_CheckFileType() - *---------------------------------------------------------------------------- - * Purpose: - * Check the file type to see if we can parse it - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT OTA_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset) -{ - S_OTA_DATA* pData; - EAS_RESULT result; - EAS_INT cmdLen; - EAS_INT state; - EAS_U8 temp; - - /* read the first byte, should be command length */ - *ppHandle = NULL; - if ((result = EAS_HWGetByte(pEASData->hwInstData, fileHandle, &temp)) != EAS_SUCCESS) - return result; - - /* read all the commands */ - cmdLen = temp; - state = 0; - while (cmdLen--) - { - - /* read the command, upper 7 bits */ - if ((result = EAS_HWGetByte(pEASData->hwInstData, fileHandle, &temp)) != EAS_SUCCESS) - return result; - temp = temp >> 1; - - if (state == 0) - { - if (temp != OTA_RINGTONE) - break; - state++; - } - else - { - - if (temp == OTA_SOUND) - { - - /* check for static memory allocation */ - if (pEASData->staticMemoryModel) - pData = EAS_CMEnumData(EAS_CM_OTA_DATA); - else - pData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_OTA_DATA)); - if (!pData) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Malloc failed in OTA_Prepare\n"); */ } - return EAS_ERROR_MALLOC_FAILED; - } - EAS_HWMemSet(pData, 0, sizeof(S_OTA_DATA)); - - /* return a pointer to the instance data */ - pData->fileHandle = fileHandle; - pData->fileOffset = offset; - pData->state = EAS_STATE_OPEN; - *ppHandle = pData; - break; - } - - if (temp != OTA_UNICODE) - break; - } - } - - /* not recognized */ - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * OTA_Prepare() - *---------------------------------------------------------------------------- - * Purpose: - * Prepare to parse the file. Allocates instance data (or uses static allocation for - * static memory model). - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT OTA_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_OTA_DATA* pData; - EAS_RESULT result; - - /* check for valid state */ - pData = (S_OTA_DATA*) pInstData; - if (pData->state != EAS_STATE_OPEN) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - - /* instantiate a synthesizer */ - if ((result = VMInitMIDI(pEASData, &pData->pSynth)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitMIDI returned %d\n", result); */ } - return result; - } - - pData->state = EAS_STATE_ERROR; - if ((result = OTA_ParseHeader(pEASData, pData)) != EAS_SUCCESS) - return result; - - pData->state = EAS_STATE_READY; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * OTA_Time() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the time of the next event in msecs - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * pTime - pointer to variable to hold time of next event (in msecs) - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) common decoder interface - pEASData not used */ -static EAS_RESULT OTA_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime) -{ - S_OTA_DATA *pData; - - pData = (S_OTA_DATA*) pInstData; - - /* return time in milliseconds */ - /*lint -e{704} use shift instead of division */ - *pTime = pData->time >> 8; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * OTA_Event() - *---------------------------------------------------------------------------- - * Purpose: - * Parse the next event in the file - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT OTA_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode) -{ - S_OTA_DATA* pData; - EAS_RESULT result; - EAS_U32 duration; - EAS_U8 temp; - - pData = (S_OTA_DATA*) pInstData; - if (pData->state >= EAS_STATE_OPEN) - return EAS_SUCCESS; - - /* initialize MIDI channel when the track starts playing */ - if (pData->time == 0) - { - /* set program to square lead */ - if (parserMode != eParserModeMetaData) - VMProgramChange(pEASData->pVoiceMgr, pData->pSynth, OTA_CHANNEL, OTA_PROGRAM); - - /* set channel volume to max */ - if (parserMode != eParserModeMetaData) - VMControlChange(pEASData->pVoiceMgr, pData->pSynth, OTA_CHANNEL, 7, 127); - } - - /* check for end of note */ - if (pData->note) - { - /* stop the note */ - VMStopNote(pEASData->pVoiceMgr, pData->pSynth, OTA_CHANNEL, pData->note, 0); - pData->note = 0; - - /* check for rest between notes */ - if (pData->restTicks) - { - pData->time += (EAS_I32) pData->restTicks; - pData->restTicks = 0; - return EAS_SUCCESS; - } - } - - /* if not in a pattern, read the pattern header */ - while (pData->current.patternLen == 0) - { - - /* check for loop - don't do infinite loops when locating */ - if (pData->loopCount && ((parserMode == eParserModePlay) || (pData->loopCount != OTA_INFINITE_LOOP))) - { - /* if not infinite loop, decrement loop count */ - if (pData->loopCount != OTA_INFINITE_LOOP) - pData->loopCount--; - - /* back to start of pattern*/ - if ((result = OTA_RestorePosition(pEASData->hwInstData, pData, &pData->patterns[pData->currentPattern])) != EAS_SUCCESS) - return result; - } - - /* if no previous position to restore, continue forward */ - else if (pData->restore.fileOffset < 0) - { - - /* check for end of song */ - if (pData->numPatterns == 0) - { - pData->state = EAS_STATE_STOPPING; - VMReleaseAllVoices(pEASData->pVoiceMgr, pData->pSynth); - return EAS_SUCCESS; - } - - /* read the next pattern header */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 3, &temp)) != EAS_SUCCESS) - return result; - if (temp != OTA_PATTERN_HEADER_ID) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Expected OTA pattern header\n"); */ } - return EAS_ERROR_FILE_FORMAT; - } - - /* get the pattern ID */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 2, &pData->currentPattern)) != EAS_SUCCESS) - return result; - - /* get the loop count */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 4, &pData->loopCount)) != EAS_SUCCESS) - return result; - - /* get the pattern length */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 8, &pData->current.patternLen)) != EAS_SUCCESS) - return result; - - /* if pattern definition, save the current position */ - if (pData->current.patternLen) - { - if ((result = OTA_SavePosition(pEASData->hwInstData, pData, &pData->patterns[pData->currentPattern])) != EAS_SUCCESS) - return result; - } - - /* if pattern length is zero, repeat a previous pattern */ - else - { - /* make sure it's a valid pattern */ - if (pData->patterns[pData->currentPattern].fileOffset < 0) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "OTA pattern error, invalid pattern specified\n"); */ } - return EAS_ERROR_FILE_FORMAT; - } - - /* save current position and data */ - if ((result = OTA_SavePosition(pEASData->hwInstData, pData, &pData->restore)) != EAS_SUCCESS) - return result; - - /* seek to the pattern in the file */ - if ((result = OTA_RestorePosition(pEASData->hwInstData, pData, &pData->patterns[pData->currentPattern])) != EAS_SUCCESS) - return result; - } - - /* decrement pattern count */ - pData->numPatterns--; - } - - /* restore previous position */ - else - { - if ((result = OTA_RestorePosition(pEASData->hwInstData, pData, &pData->restore)) != EAS_SUCCESS) - return result; - } - } - - /* get the next event */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 3, &temp)) != EAS_SUCCESS) - return result; - - switch (temp) - { - case OTA_NOTE_INST_ID: - /* fetch note value */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 4, &pData->note)) != EAS_SUCCESS) - return result; - - /* fetch note duration */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 3, &temp)) != EAS_SUCCESS) - return result; - duration = pData->tick * (0x20 >> temp); - - /* fetch note duration modifier */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 2, &temp)) != EAS_SUCCESS) - return result; - switch (temp) - { - case OTA_NORMAL_DURATION: - break; - - case OTA_DOTTED_NOTE: - duration += duration >> 1; - break; - - case OTA_DOUBLE_DOTTED_NOTE: - duration += (duration >> 1) + (duration >> 2); - break; - - case OTA_TRIPLET_NOTE: - duration = (duration * TRIPLET_MULTIPLIER) >> TRIPLET_SHIFT; - break; - - default: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Unrecognized note duration ignored\n"); */ } - break; - } - - /* check for note */ - if (pData->note) - { - - /* determine note length based on style */ - switch (pData->style) - { - case 0: - pData->restTicks = duration >> 4; - break; - case 1: - pData->restTicks = 0; - break; - case 2: - pData->restTicks = duration >> 1; - break; - default: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Unrecognized note style ignored\n"); */ } - } - - /* add octave */ - pData->note += pData->octave; - if (parserMode == eParserModePlay) - VMStartNote(pEASData->pVoiceMgr, pData->pSynth, OTA_CHANNEL, pData->note, pData->velocity); - pData->time += (EAS_I32) duration - (EAS_I32) pData->restTicks; - } - - /* this is a rest */ - else - pData->time += (EAS_I32) duration; - break; - - case OTA_SCALE_INST_ID: - /* fetch octave */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 2, &temp)) != EAS_SUCCESS) - return result; - pData->octave = (EAS_U8) (temp * 12 + 59); - break; - - case OTA_STYLE_INST_ID: - /* fetch note style */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 2, &pData->style)) != EAS_SUCCESS) - return result; - break; - - case OTA_TEMPO_INST_ID: - /* fetch tempo */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 5, &temp)) != EAS_SUCCESS) - return result; - pData->tick = bpmTable[temp]; - break; - - case OTA_VOLUME_INST_ID: - /* fetch volume */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 4, &temp)) != EAS_SUCCESS) - return result; - pData->velocity = temp ? (EAS_U8) (temp * OTA_VEL_MUL + OTA_VEL_OFS) : 0; - break; - - default: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Unexpected instruction ID in OTA stream\n"); */ } - return EAS_ERROR_FILE_FORMAT; - } - - /* decrement pattern length */ - pData->current.patternLen--; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * OTA_State() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current state of the stream - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * pState - pointer to variable to store state - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) common decoder interface - pEASData not used */ -static EAS_RESULT OTA_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pState) -{ - S_OTA_DATA* pData; - - /* establish pointer to instance data */ - pData = (S_OTA_DATA*) pInstData; - - /* if stopping, check to see if synth voices are active */ - if (pData->state == EAS_STATE_STOPPING) - { - if (VMActiveVoices(pData->pSynth) == 0) - pData->state = EAS_STATE_STOPPED; - } - - if (pData->state == EAS_STATE_PAUSING) - { - if (VMActiveVoices(pData->pSynth) == 0) - pData->state = EAS_STATE_PAUSED; - } - - /* return current state */ - *pState = pData->state; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * OTA_Close() - *---------------------------------------------------------------------------- - * Purpose: - * Close the file and clean up - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT OTA_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_OTA_DATA* pData; - EAS_RESULT result; - - pData = (S_OTA_DATA*) pInstData; - - /* close the file */ - if ((result = EAS_HWCloseFile(pEASData->hwInstData, pData->fileHandle)) != EAS_SUCCESS) - return result; - - /* free the synth */ - if (pData->pSynth != NULL) - VMMIDIShutdown(pEASData, pData->pSynth); - - /* if using dynamic memory, free it */ - if (!pEASData->staticMemoryModel) - EAS_HWFree(pEASData->hwInstData, pData); - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * OTA_Reset() - *---------------------------------------------------------------------------- - * Purpose: - * Reset the sequencer. Used for locating backwards in the file. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT OTA_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_OTA_DATA* pData; - EAS_RESULT result; - - pData = (S_OTA_DATA*) pInstData; - - /* reset the synth */ - VMReset(pEASData->pVoiceMgr, pData->pSynth, EAS_TRUE); - pData->note = 0; - - /* reset file position and re-parse header */ - pData->state = EAS_STATE_ERROR; - if ((result = OTA_ParseHeader (pEASData, pData)) != EAS_SUCCESS) - return result; - - pData->state = EAS_STATE_READY; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * OTA_Pause() - *---------------------------------------------------------------------------- - * Purpose: - * Pauses the sequencer. Mutes all voices and sets state to pause. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT OTA_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_OTA_DATA *pData; - - /* can't pause a stopped stream */ - pData = (S_OTA_DATA*) pInstData; - if (pData->state == EAS_STATE_STOPPED) - return EAS_ERROR_ALREADY_STOPPED; - - /* mute the synthesizer */ - VMMuteAllVoices(pEASData->pVoiceMgr, pData->pSynth); - pData->state = EAS_STATE_PAUSING; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * OTA_Resume() - *---------------------------------------------------------------------------- - * Purpose: - * Resume playing after a pause, sets state back to playing. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) common decoder interface - pEASData not used */ -static EAS_RESULT OTA_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_OTA_DATA *pData; - - /* can't resume a stopped stream */ - pData = (S_OTA_DATA*) pInstData; - if (pData->state == EAS_STATE_STOPPED) - return EAS_ERROR_ALREADY_STOPPED; - - /* nothing to do but resume playback */ - pData->state = EAS_STATE_PLAY; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * OTA_SetData() - *---------------------------------------------------------------------------- - * Purpose: - * Return file type - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) common decoder interface - pEASData not used */ -static EAS_RESULT OTA_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value) -{ - S_OTA_DATA *pData; - - pData = (S_OTA_DATA *) pInstData; - switch (param) - { - - /* set metadata callback */ - case PARSER_DATA_METADATA_CB: - EAS_HWMemCpy(&pData->metadata, (void*) value, sizeof(S_METADATA_CB)); - break; - - default: - return EAS_ERROR_INVALID_PARAMETER; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * OTA_GetData() - *---------------------------------------------------------------------------- - * Purpose: - * Return file type - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) common decoder interface - pEASData not used */ -static EAS_RESULT OTA_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue) -{ - S_OTA_DATA *pData; - - pData = (S_OTA_DATA*) pInstData; - switch (param) - { - /* return file type as OTA */ - case PARSER_DATA_FILE_TYPE: - *pValue = EAS_FILE_OTA; - break; - -#if 0 - /* set transposition */ - case PARSER_DATA_TRANSPOSITION: - *pValue = pData->transposition; - break; -#endif - - case PARSER_DATA_SYNTH_HANDLE: - *pValue = (EAS_I32) pData->pSynth; - break; - - case PARSER_DATA_GAIN_OFFSET: - *pValue = OTA_GAIN_OFFSET; - break; - - default: - return EAS_ERROR_INVALID_PARAMETER; - } - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * OTA_ParseHeader() - *---------------------------------------------------------------------------- - * Purpose: - * Prepare to parse the file. Allocates instance data (or uses static allocation for - * static memory model). - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT OTA_ParseHeader (S_EAS_DATA *pEASData, S_OTA_DATA* pData) -{ - EAS_RESULT result; - EAS_INT i; - EAS_INT state; - EAS_U8 temp; - EAS_U8 titleLen; - - /* initialize some data */ - pData->flags = 0; - pData->time = 0; - pData->tick = DEFAULT_TICK_CONV; - pData->patterns[0].fileOffset = pData->patterns[1].fileOffset = - pData->patterns[2].fileOffset = pData->patterns[3].fileOffset = -1; - pData->current.bitCount = 0; - pData->current.patternLen = 0; - pData->loopCount = 0; - pData->restore.fileOffset = -1; - pData->note = 0; - pData->restTicks = 0; - pData->velocity = OTA_VEL_DEFAULT; - pData->style = 0; - pData->octave = 59; - - /* seek to start of data */ - if ((result = EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->fileOffset)) != EAS_SUCCESS) - return result; - - /* read the first byte, should be command length */ - if ((result = EAS_HWGetByte(pEASData->hwInstData, pData->fileHandle, &temp)) != EAS_SUCCESS) - return result; - - /* read all the commands */ - i = temp; - state = 0; - while (i--) - { - - /* fetch command, always starts on byte boundary */ - pData->current.bitCount = 0; - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 7, &temp)) != EAS_SUCCESS) - return result; - - if (state == 0) - { - if (temp != OTA_RINGTONE) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Expected OTA Ring Tone Programming type\n"); */ } - return EAS_ERROR_FILE_FORMAT; - } - state++; - } - else - { - - if (temp == OTA_SOUND) - break; - - if (temp == OTA_UNICODE) - pData->flags |= OTA_FLAGS_UNICODE; - else - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Expected OTA Sound or Unicode type\n"); */ } - return EAS_ERROR_FILE_FORMAT; - } - } - } - - /* get song type */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 3, &temp)) != EAS_SUCCESS) - return result; - - /* check for basic song type */ - if (temp == OTA_BASIC_SONG_TYPE) - { - /* fetch title length */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 4, &titleLen)) != EAS_SUCCESS) - return result; - - /* if unicode, double the length */ - if (pData->flags & OTA_FLAGS_UNICODE) - titleLen = (EAS_U8) (titleLen << 1); - - /* zero the metadata buffer */ - if (pData->metadata.buffer) - EAS_HWMemSet(pData->metadata.buffer, 0, pData->metadata.bufferSize); - - /* read the song title */ - for (i = 0; i < titleLen; i++) - { - /* fetch character */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 8, &temp)) != EAS_SUCCESS) - return result; - - /* check for metadata callback */ - if (pData->metadata.callback) - { - if (i < (pData->metadata.bufferSize - 1)) - pData->metadata.buffer[i] = (char) temp; - } - } - - /* if host has registered callback, call it now */ - if (pData->metadata.callback) - (*pData->metadata.callback)(EAS_METADATA_TITLE, pData->metadata.buffer, pData->metadata.pUserData); - } - - /* must be temporary song */ - else if (temp != OTA_TEMPORARY_SONG_TYPE) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Expected OTA basic or temporary song type\n"); */ } - return EAS_ERROR_FILE_FORMAT; - } - - /* get the song length */ - if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 8, &pData->numPatterns)) != EAS_SUCCESS) - return result; - - /* sanity check */ - if (pData->numPatterns == 0) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "OTA number of patterns is zero\n"); */ } - return EAS_ERROR_FILE_FORMAT; - } - - /* at start of first pattern */ - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * OTA_FetchBitField() - *---------------------------------------------------------------------------- - * Purpose: - * Fetch a specified number of bits from the input stream - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT OTA_FetchBitField (EAS_HW_DATA_HANDLE hwInstData, S_OTA_DATA *pData, EAS_I32 numBits, EAS_U8 *pValue) -{ - EAS_RESULT result; - EAS_I32 bitsLeft; - EAS_U8 value; - - value = 0; - - /* do we have enough bits? */ - bitsLeft = pData->current.bitCount - numBits; - - /* not enough bits, assemble them from 2 characters */ - if (bitsLeft < 0) - { - /* grab the remaining bits from the previous byte */ - if (pData->current.bitCount) - /*lint -e{504,734} this is a legitimate shift operation */ - value = pData->current.dataByte << -bitsLeft; - - /* read the next byte */ - if ((result = EAS_HWGetByte(hwInstData, pData->fileHandle, &pData->current.dataByte)) != EAS_SUCCESS) - return result; - bitsLeft += 8; - } - - /* more bits than needed? */ - if (bitsLeft > 0) - { - value |= pData->current.dataByte >> bitsLeft; - pData->current.bitCount = (EAS_U8) bitsLeft; - pData->current.dataByte = pData->current.dataByte & (0xff >> (8 - bitsLeft)); - } - - /* exactly the right number of bits */ - else - { - value |= pData->current.dataByte; - pData->current.bitCount = 0; - } - - *pValue = value; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * OTA_SavePosition() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT OTA_SavePosition (EAS_HW_DATA_HANDLE hwInstData, S_OTA_DATA *pData, S_OTA_LOC *pLoc) -{ - EAS_HWMemCpy(pLoc, &pData->current, sizeof(S_OTA_LOC)); - return EAS_HWFilePos(hwInstData, pData->fileHandle, &pLoc->fileOffset); -} - -/*---------------------------------------------------------------------------- - * OTA_RestorePosition() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT OTA_RestorePosition (EAS_HW_DATA_HANDLE hwInstData, S_OTA_DATA *pData, S_OTA_LOC *pLoc) -{ - EAS_HWMemCpy(&pData->current, pLoc, sizeof(S_OTA_LOC)); - pData->restore.fileOffset = -1; - return EAS_HWFileSeek(hwInstData, pData->fileHandle, pLoc->fileOffset); -} - diff --git a/arm-hybrid-22k/lib_src/eas_otadata.c b/arm-hybrid-22k/lib_src/eas_otadata.c deleted file mode 100644 index 237f832..0000000 --- a/arm-hybrid-22k/lib_src/eas_otadata.c +++ /dev/null @@ -1,41 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_otadata..c - * - * Contents and purpose: - * OTA Stream Parser data module for static memory model - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 547 $ - * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_types.h" -#include "eas_otadata.h" - -/*---------------------------------------------------------------------------- - * - * eas_OTAData - * - * Static memory allocation for OTA parser - *---------------------------------------------------------------------------- -*/ -S_OTA_DATA eas_OTAData; - diff --git a/arm-hybrid-22k/lib_src/eas_otadata.h b/arm-hybrid-22k/lib_src/eas_otadata.h deleted file mode 100644 index 63e963f..0000000 --- a/arm-hybrid-22k/lib_src/eas_otadata.h +++ /dev/null @@ -1,81 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_otadata.h - * - * Contents and purpose: - * OTA File Parser - * - * This file contains data declarations for the OTA parser. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef EAS_OTADATA_H -#define EAS_OTADATA_H - -#include "eas_data.h" - -/* definition for state flags */ -#define OTA_FLAGS_UNICODE 0x01 /* unicode text */ - -/*---------------------------------------------------------------------------- - * - * S_OTA_DATA - * - * This structure contains the state data for the OTA parser - *---------------------------------------------------------------------------- -*/ - -typedef struct -{ - EAS_I32 fileOffset; /* offset to location in file */ - EAS_U8 patternLen; /* length of current pattern */ - EAS_U8 dataByte; /* previous char from file */ - EAS_U8 bitCount; /* bit count in char */ -} S_OTA_LOC; - -typedef struct -{ - EAS_FILE_HANDLE fileHandle; /* file handle */ - S_SYNTH *pSynth; /* synth handle */ - EAS_I32 fileOffset; /* offset to start of data */ - EAS_I32 time; /* current time in 256ths of a msec */ - EAS_U32 tick; /* length of 32nd note in 256th of a msec */ - EAS_U32 restTicks; /* ticks to rest after current note */ - S_OTA_LOC patterns[4]; /* pattern locations */ - S_OTA_LOC current; /* current location */ - S_OTA_LOC restore; /* previous location */ - S_METADATA_CB metadata; /* metadata callback */ - EAS_U8 flags; /* bit flags */ - EAS_U8 numPatterns; /* number of patterns left in song */ - EAS_U8 currentPattern; /* current pattern for loop */ - EAS_U8 note; /* MIDI note number */ - EAS_U8 octave; /* octave modifier */ - EAS_U8 style; /* from STYLE */ - EAS_U8 velocity; /* current volume */ - EAS_U8 state; /* current state EAS_STATE_XXXX */ - EAS_U8 loopCount; /* loop count for pattern */ -} S_OTA_DATA; - -#endif - - diff --git a/arm-hybrid-22k/lib_src/eas_pan.c b/arm-hybrid-22k/lib_src/eas_pan.c deleted file mode 100644 index 373d90e..0000000 --- a/arm-hybrid-22k/lib_src/eas_pan.c +++ /dev/null @@ -1,98 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_pan.c - * - * Contents and purpose: - * Calculates left and right gain multipliers based on a pan value from -63 to +63 - * - * NOTES: - * The _CMX_PARSER and _MFI_PARSER preprocessor symbols determine - * whether the parser works for those particular file formats. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_pan.h" -#include "eas_math.h" - -/*---------------------------------------------------------------------------- - * EAS_CalcPanControl() - *---------------------------------------------------------------------------- - * Purpose: - * Assign the left and right gain values corresponding to the given pan value. - * - * This routine uses sin/cos approximations for an equal power curve: - * - * sin(x) = (2-4*c)*x^2 + c + x - * cos(x) = (2-4*c)*x^2 + c - x - * - * where c = 1/sqrt(2) - * using the a0 + x*(a1 + x*a2) approach - * - * Inputs: - * pan - pan value (-63 to + 63) - * - * Outputs: - * pGainLeft linear gain multiplier for left channel (15-bit fraction) - * pGainRight linear gain multiplier for left channel (15-bit fraction) - * - * Side Effects: - *---------------------------------------------------------------------------- -*/ -void EAS_CalcPanControl (EAS_INT pan, EAS_I16 *pGainLeft, EAS_I16 *pGainRight) -{ - EAS_INT temp; - EAS_INT netAngle; - - /* impose hard limit */ - if (pan < -63) - netAngle = -63; - else if (pan > 63) - netAngle = 63; - else - netAngle = pan; - - /*lint -e{701} */ - netAngle = netAngle << 8; - - /* calculate sin */ - temp = EG1_ONE + FMUL_15x15(COEFF_PAN_G2, netAngle); - temp = COEFF_PAN_G0 + FMUL_15x15(temp, netAngle); - - if (temp > SYNTH_FULL_SCALE_EG1_GAIN) - temp = SYNTH_FULL_SCALE_EG1_GAIN; - else if (temp < 0) - temp = 0; - - *pGainRight = (EAS_I16) temp; - - /* calculate cos */ - temp = -EG1_ONE + FMUL_15x15(COEFF_PAN_G2, netAngle); - temp = COEFF_PAN_G0 + FMUL_15x15(temp, netAngle); - if (temp > SYNTH_FULL_SCALE_EG1_GAIN) - temp = SYNTH_FULL_SCALE_EG1_GAIN; - else if (temp < 0) - temp = 0; - - *pGainLeft = (EAS_I16) temp; -} - diff --git a/arm-hybrid-22k/lib_src/eas_pan.h b/arm-hybrid-22k/lib_src/eas_pan.h deleted file mode 100644 index cefa650..0000000 --- a/arm-hybrid-22k/lib_src/eas_pan.h +++ /dev/null @@ -1,66 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_pan.h - * - * Contents and purpose: - * Calculates left and right gain multipliers based on a pan value from -63 to +63 - * - * NOTES: - * The _CMX_PARSER and _MFI_PARSER preprocessor symbols determine - * whether the parser works for those particular file formats. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef EAS_PAN_H -#define _EAS_PAN_H - -#include "eas_types.h" - -/*---------------------------------------------------------------------------- - * EAS_CalcPanControl() - *---------------------------------------------------------------------------- - * Purpose: - * Assign the left and right gain values corresponding to the given pan value. - * - * This routine uses sin/cos approximations for an equal power curve: - * - * sin(x) = (2-4*c)*x^2 + c + x - * cos(x) = (2-4*c)*x^2 + c - x - * - * where c = 1/sqrt(2) - * using the a0 + x*(a1 + x*a2) approach - * - * Inputs: - * pan - pan value (-63 to + 63) - * - * Outputs: - * pGainLeft linear gain multiplier for left channel (15-bit fraction) - * pGainRight linear gain multiplier for left channel (15-bit fraction) - * - * Side Effects: - *---------------------------------------------------------------------------- -*/ -void EAS_CalcPanControl (EAS_INT pan, EAS_I16 *pGainLeft, EAS_I16 *pGainRight); - -#endif - diff --git a/arm-hybrid-22k/lib_src/eas_parser.h b/arm-hybrid-22k/lib_src/eas_parser.h deleted file mode 100644 index 5512c82..0000000 --- a/arm-hybrid-22k/lib_src/eas_parser.h +++ /dev/null @@ -1,98 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_parser.h - * - * Contents and purpose: - * Interface declarations for the generic parser interface - * - * This header only contains declarations that are specific - * to this implementation. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 767 $ - * $Date: 2007-07-19 13:47:31 -0700 (Thu, 19 Jul 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_PARSER_H -#define _EAS_PARSER_H - -#include "eas_types.h" - - -/* metadata callback */ -typedef struct s_metadata_cb_tag -{ - EAS_METADATA_CBFUNC callback; - char *buffer; - EAS_VOID_PTR pUserData; - EAS_I32 bufferSize; -} S_METADATA_CB; - -/* generic parser interface */ -typedef struct -{ - EAS_RESULT (* EAS_CONST pfCheckFileType)(struct s_eas_data_tag *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset); - EAS_RESULT (* EAS_CONST pfPrepare)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData); - EAS_RESULT (* EAS_CONST pfTime)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime); - EAS_RESULT (* EAS_CONST pfEvent)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_INT parseMode); - EAS_RESULT (* EAS_CONST pfState)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState); - EAS_RESULT (* EAS_CONST pfClose)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData); - EAS_RESULT (* EAS_CONST pfReset)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData); - EAS_RESULT (* EAS_CONST pfPause)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData); - EAS_RESULT (* EAS_CONST pfResume)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData); - EAS_RESULT (* EAS_CONST pfLocate)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_I32 time, EAS_BOOL *pParserLocate); - EAS_RESULT (* EAS_CONST pfSetData)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value); - EAS_RESULT (* EAS_CONST pfGetData)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue); - EAS_RESULT (* EAS_CONST pfGetMetaData)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pMediaLength); -} S_FILE_PARSER_INTERFACE; - -typedef enum -{ - eParserModePlay, - eParserModeLocate, - eParserModeMute, - eParserModeMetaData -} E_PARSE_MODE; - -typedef enum -{ - PARSER_DATA_FILE_TYPE, - PARSER_DATA_PLAYBACK_RATE, - PARSER_DATA_TRANSPOSITION, - PARSER_DATA_VOLUME, - PARSER_DATA_SYNTH_HANDLE, - PARSER_DATA_METADATA_CB, - PARSER_DATA_DLS_COLLECTION, - PARSER_DATA_EAS_LIBRARY, - PARSER_DATA_POLYPHONY, - PARSER_DATA_PRIORITY, - PARSER_DATA_FORMAT, - PARSER_DATA_MEDIA_LENGTH, - PARSER_DATA_JET_CB, - PARSER_DATA_MUTE_FLAGS, - PARSER_DATA_SET_MUTE, - PARSER_DATA_CLEAR_MUTE, - PARSER_DATA_NOTE_COUNT, - PARSER_DATA_MAX_PCM_STREAMS, - PARSER_DATA_GAIN_OFFSET, - PARSER_DATA_PLAY_MODE -} E_PARSER_DATA; - -#endif /* #ifndef _EAS_PARSER_H */ diff --git a/arm-hybrid-22k/lib_src/eas_pcm.c b/arm-hybrid-22k/lib_src/eas_pcm.c deleted file mode 100644 index 64b8f71..0000000 --- a/arm-hybrid-22k/lib_src/eas_pcm.c +++ /dev/null @@ -1,1482 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_pcm.c - * - * Contents and purpose: - * Implements the PCM engine including ADPCM decode for SMAF and CMX audio playback. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 849 $ - * $Date: 2007-08-28 08:59:11 -0700 (Tue, 28 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_data.h" -#include "eas_report.h" -#include "eas_host.h" -#include "eas_config.h" -#include "eas_parser.h" -#include "eas_pcm.h" -#include "eas_math.h" -#include "eas_mixer.h" - -#define PCM_MIXER_GUARD_BITS (NUM_MIXER_GUARD_BITS + 1) - -/*---------------------------------------------------------------------------- - * Decoder interfaces - *---------------------------------------------------------------------------- -*/ - -static EAS_RESULT LinearPCMDecode (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState); -static EAS_RESULT LinearPCMLocate (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 time); - -static const S_DECODER_INTERFACE PCMDecoder = -{ - NULL, - LinearPCMDecode, - LinearPCMLocate, -}; - -/* SMAF ADPCM decoder */ -#ifdef _SMAF_PARSER -extern S_DECODER_INTERFACE SmafDecoder; -#define SMAF_DECODER &SmafDecoder -extern S_DECODER_INTERFACE Smaf7BitDecoder; -#define SMAF_7BIT_DECODER &Smaf7BitDecoder -#else -#define SMAF_DECODER NULL -#define SMAF_7BIT_DECODER NULL -#endif - -/* IMA ADPCM decoder */ -#ifdef _IMA_DECODER -extern S_DECODER_INTERFACE IMADecoder; -#define IMA_DECODER &IMADecoder -#else -#define IMA_DECODER NULL -#endif - -static const S_DECODER_INTERFACE * const decoders[] = -{ - &PCMDecoder, - SMAF_DECODER, - IMA_DECODER, - SMAF_7BIT_DECODER -}; - -/*---------------------------------------------------------------------------- - * Sample rate conversion - *---------------------------------------------------------------------------- -*/ - -#define SRC_RATE_MULTIPLER (0x40000000 / _OUTPUT_SAMPLE_RATE) - -#ifdef _LOOKUP_SAMPLE_RATE -static const EAS_U32 srcConvRate[][2] = -{ - 4000L, (4000L << 15) / _OUTPUT_SAMPLE_RATE, - 8000L, (8000L << 15) / _OUTPUT_SAMPLE_RATE, - 11025L, (11025L << 15) / _OUTPUT_SAMPLE_RATE, - 12000L, (12000L << 15) / _OUTPUT_SAMPLE_RATE, - 16000L, (16000L << 15) / _OUTPUT_SAMPLE_RATE, - 22050L, (22050L << 15) / _OUTPUT_SAMPLE_RATE, - 24000L, (24000L << 15) / _OUTPUT_SAMPLE_RATE, - 32000L, (32000L << 15) / _OUTPUT_SAMPLE_RATE -}; -static EAS_U32 CalcBaseFreq (EAS_U32 sampleRate); -#define SRC_CONV_RATE_ENTRIES (sizeof(srcConvRate)/sizeof(EAS_U32)/2) -#endif - - -/* interface prototypes */ -static EAS_RESULT RenderPCMStream (S_EAS_DATA *pEASData, S_PCM_STATE *pState, EAS_I32 numSamples); - - -/* local prototypes */ -static S_PCM_STATE *FindSlot (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_PCM_CALLBACK pCallbackFunc, EAS_VOID_PTR cbInstData); -static EAS_RESULT InitPCMStream (S_EAS_DATA *pEASData, S_PCM_STATE *pState); - -/*---------------------------------------------------------------------------- - * EAS_PEInit() - *---------------------------------------------------------------------------- - * Purpose: - * Initializes the PCM engine - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEInit (S_EAS_DATA *pEASData) -{ - S_PCM_STATE *pState; - EAS_INT i; - - /* check for static memory allocation */ - if (pEASData->staticMemoryModel) - pEASData->pPCMStreams = EAS_CMEnumData(EAS_CM_PCM_DATA); - /* allocate dynamic memory */ - else - pEASData->pPCMStreams = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_PCM_STATE) * MAX_PCM_STREAMS); - - if (!pEASData->pPCMStreams) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Failed to allocate memory for PCM streams\n"); */ } - return EAS_ERROR_MALLOC_FAILED; - } - - //zero the memory to insure complete initialization - EAS_HWMemSet((void *)(pEASData->pPCMStreams),0, sizeof(S_PCM_STATE) * MAX_PCM_STREAMS); - - /* initialize the state data */ - for (i = 0, pState = pEASData->pPCMStreams; i < MAX_PCM_STREAMS; i++, pState++) - pState->fileHandle = NULL; - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_PEShutdown() - *---------------------------------------------------------------------------- - * Purpose: - * Shuts down the PCM engine - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEShutdown (S_EAS_DATA *pEASData) -{ - - /* free any dynamic memory */ - if (!pEASData->staticMemoryModel) - { - if (pEASData->pPCMStreams) - { - EAS_HWFree(pEASData->hwInstData, pEASData->pPCMStreams); - pEASData->pPCMStreams = NULL; - } - } - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_PERender() - *---------------------------------------------------------------------------- - * Purpose: - * Render a buffer of PCM audio - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PERender (S_EAS_DATA* pEASData, EAS_I32 numSamples) -{ - S_PCM_STATE *pState; - EAS_RESULT result; - EAS_INT i; - - /* render all the active streams */ - for (i = 0, pState = pEASData->pPCMStreams; i < MAX_PCM_STREAMS; i++, pState++) - { - if ((pState->fileHandle) && (pState->state != EAS_STATE_STOPPED) && (pState->state != EAS_STATE_PAUSED)) - if ((result = RenderPCMStream(pEASData, pState, numSamples)) != EAS_SUCCESS) - return result; - } - return EAS_SUCCESS; -} - - -/*---------------------------------------------------------------------------- - * EAS_PEState() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current state of the stream - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * pState - pointer to variable to store state - * - * Outputs: - * - * - * Side Effects: - * - * Notes: - * This interface is also exposed in the internal library for use by the other modules. - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_RESULT EAS_PEState (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pInstData, EAS_STATE *pState) -{ - /* return current state */ - *pState = pInstData->state; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_PEClose() - *---------------------------------------------------------------------------- - * Purpose: - * Close the file and clean up - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEClose (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState) -{ - EAS_RESULT result; - - if ((result = EAS_HWCloseFile(pEASData->hwInstData, pState->fileHandle)) != EAS_SUCCESS) - return result; - - pState->fileHandle = NULL; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * PCM_Reset() - *---------------------------------------------------------------------------- - * Purpose: - * Reset the sequencer. Used for locating backwards in the file. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEReset (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState) -{ - EAS_RESULT result; - - /* reset file position to first byte of data in the stream */ - if ((result = EAS_HWFileSeek(pEASData->hwInstData, pState->fileHandle, pState->startPos)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %d seeking to start of PCM file\n", result); */ } - return result; - } - - /* re-initialize stream */ - return InitPCMStream(pEASData, pState); -} - -/*---------------------------------------------------------------------------- - * EAS_PEOpenStream() - *---------------------------------------------------------------------------- - * Purpose: - * Starts up a PCM playback - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEOpenStream (S_EAS_DATA *pEASData, S_PCM_OPEN_PARAMS *pParams, EAS_PCM_HANDLE *pHandle) -{ - EAS_RESULT result; - S_PCM_STATE *pState; - EAS_I32 filePos; - - /* make sure we support this decoder */ - if (pParams->decoder >= NUM_DECODER_MODULES) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Decoder selector out of range\n"); */ } - return EAS_ERROR_PARAMETER_RANGE; - } - if (decoders[pParams->decoder] == NULL) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Decoder module not available\n"); */ } - return EAS_ERROR_FEATURE_NOT_AVAILABLE; - } - - /* find a slot for the new stream */ - if ((pState = FindSlot(pEASData, pParams->fileHandle, pParams->pCallbackFunc, pParams->cbInstData)) == NULL) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Unable to open ADPCM stream, too many streams open\n"); */ } - return EAS_ERROR_MAX_PCM_STREAMS; - } - - /* get the current file position */ - if ((result = EAS_HWFilePos(pEASData->hwInstData, pState->fileHandle, &filePos)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_HWFilePos returned %ld\n",result); */ } - pState->fileHandle = NULL; - return result; - } - - pState->pDecoder = decoders[pParams->decoder]; - pState->startPos = filePos; - pState->bytesLeftLoop = pState->byteCount = pParams->size; - pState->loopStart = pParams->loopStart; - pState->samplesTilLoop = (EAS_I32) pState->loopStart; - pState->loopSamples = pParams->loopSamples; - pState->samplesInLoop = 0; - pState->blockSize = (EAS_U16) pParams->blockSize; - pState->flags = pParams->flags; - pState->envData = pParams->envData; - pState->volume = pParams->volume; - pState->sampleRate = (EAS_U16) pParams->sampleRate; - - /* set the base frequency */ - pState->basefreq = (SRC_RATE_MULTIPLER * (EAS_U32) pParams->sampleRate) >> 15; - - /* calculate shift for frequencies > 1.0 */ - pState->rateShift = 0; - while (pState->basefreq > 32767) - { - pState->basefreq = pState->basefreq >> 1; - pState->rateShift++; - } - - /* initialize */ - if ((result = InitPCMStream(pEASData, pState)) != EAS_SUCCESS) - return result; - - *pHandle = pState; - - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "EAS_PEOpenStream: StartPos=%d, byteCount = %d, loopSamples=%d\n", - pState->startPos, pState->byteCount, pState->loopSamples); */ } - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_PEContinueStream() - *---------------------------------------------------------------------------- - * Purpose: - * Continues a PCM stream - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -e{715} reserved for future use */ -EAS_RESULT EAS_PEContinueStream (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState, EAS_I32 size) -{ - - /* add new samples to count */ - pState->bytesLeft += size; - if (pState->bytesLeft > 0) - pState->flags &= ~PCM_FLAGS_EMPTY; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_PEGetFileHandle() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the file handle of a stream - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_RESULT EAS_PEGetFileHandle (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState, EAS_FILE_HANDLE *pFileHandle) -{ - *pFileHandle = pState->fileHandle; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_PEUpdateParams() - *---------------------------------------------------------------------------- - * Purpose: - * Update the pitch and volume parameters for a PCM stream - * - * Inputs: - * pEASData - pointer to EAS library instance data - * handle - pointer to S_PCM_STATE for this stream - * gainLeft - linear gain multipler in 1.15 fraction format - * gainRight - linear gain multipler in 1.15 fraction format - * pitch - pitch shift in cents - * initial - initial settings, set current gain - * - * Outputs: - * - * - * Side Effects: - * - * Notes - * In mono mode, leftGain controls the output gain and rightGain is ignored - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -/*lint -esym(715, gainRight) used only in 2-channel version */ -EAS_RESULT EAS_PEUpdateParams (S_EAS_DATA* pEASData, EAS_PCM_HANDLE pState, EAS_I16 pitch, EAS_I16 gainLeft, EAS_I16 gainRight) -{ - - pState->gainLeft = gainLeft; - -#if (NUM_OUTPUT_CHANNELS == 2) - pState->gainRight = gainRight; -#endif - - pState->pitch = pitch; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_PELocate() - *---------------------------------------------------------------------------- - * Purpose: - * This function seeks to the requested place in the file. Accuracy - * is dependent on the sample rate and block size. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * pState - stream handle - * time - media time in milliseconds - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PELocate (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState, EAS_I32 time) -{ - if (pState->pDecoder->pfLocate == NULL) - return EAS_ERROR_FEATURE_NOT_AVAILABLE; - - return pState->pDecoder->pfLocate(pEASData, pState, time); -} - -/*---------------------------------------------------------------------------- - * EAS_PEUpdateVolume() - *---------------------------------------------------------------------------- - * Purpose: - * Update the volume parameters for a PCM stream - * - * Inputs: - * pEASData - pointer to EAS library instance data - * handle - pointer to S_PCM_STATE for this stream - * gainLeft - linear gain multipler in 1.15 fraction format - * gainRight - linear gain multipler in 1.15 fraction format - * initial - initial settings, set current gain - * - * Outputs: - * - * - * Side Effects: - * - * Notes - * In mono mode, leftGain controls the output gain and rightGain is ignored - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_RESULT EAS_PEUpdateVolume (S_EAS_DATA* pEASData, EAS_PCM_HANDLE pState, EAS_I16 volume) -{ - pState->volume = volume; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_PEUpdatePitch() - *---------------------------------------------------------------------------- - * Purpose: - * Update the pitch parameter for a PCM stream - * - * Inputs: - * pEASData - pointer to EAS library instance data - * pState - pointer to S_PCM_STATE for this stream - * pitch - new pitch value in pitch cents - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_RESULT EAS_PEUpdatePitch (S_EAS_DATA* pEASData, EAS_PCM_HANDLE pState, EAS_I16 pitch) -{ - pState->pitch = pitch; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_PEPause() - *---------------------------------------------------------------------------- - * Purpose: - * Mute and stop rendering a PCM stream. Sets the gain target to zero and stops the playback - * at the end of the next audio frame. - * - * Inputs: - * pEASData - pointer to EAS library instance data - * handle - pointer to S_PCM_STATE for this stream - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_RESULT EAS_PEPause (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState) -{ - /* set state to stopping */ - pState->state = EAS_STATE_PAUSING; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_PEResume() - *---------------------------------------------------------------------------- - * Purpose: - * Resume rendering a PCM stream. Sets the gain target back to its - * previous setting and restarts playback at the end of the next audio - * frame. - * - * Inputs: - * pEASData - pointer to EAS library instance data - * handle - pointer to S_PCM_STATE for this stream - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_RESULT EAS_PEResume (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState) -{ - /* set state to stopping */ - pState->state = EAS_STATE_PLAY; - return EAS_SUCCESS; -} - -EAS_U32 getDecayScale(EAS_U32 index) -{ - EAS_U32 utemp; - - //envelope decay segment - switch (index) - { - case 0: //no decay - utemp = 512;//32768; - break; - case 1: //.0156 dB per update - utemp = 511;//32709; - break; - case 2: //.03125 - utemp = 510;//32649; - break; - case 3: //.0625 - utemp = 508;//32532; - break; - case 4: //.125 - utemp = 505;//32298; - break; - case 5: //.25 - utemp = 497;//31835; - break; - case 6: //.5 - utemp = 483;//30929; - break; - case 7: //1.0 - utemp = 456;//29193; - break; - case 8: //2.0 - utemp = 406;//26008; - break; - case 9: //4.0 - utemp = 323;//20642; - break; - case 10: //8.0 - utemp = 203;//13004; - break; - case 11: //16.0 - utemp = 81;//5160; - break; - case 12: //32.0 - utemp = 13;//813; - break; - case 13: //64.0 - utemp = 0;//20; - break; - case 14: //128.0 - utemp = 0; - break; - case 15: //256.0 - default: - utemp = 0; - break; - } - //printf("getdecayscale returned %d\n",utemp); - return utemp; -} - -EAS_U32 getAttackIncrement(EAS_U32 index) -{ - EAS_U32 utemp; - - //envelope decay segment - switch (index) - { - case 0: - utemp = 32; - break; - case 1: - utemp = 64; - break; - case 2: - utemp = 128; - break; - case 3: - utemp = 256; - break; - case 4: - utemp = 512; - break; - case 5: - utemp = 1024; - break; - case 6: - utemp = 2048; - break; - case 7: - utemp = 4096; - break; - case 8: - utemp = 8192; - break; - case 9: - utemp = 16384; - break; - case 10: - utemp = 32768; - break; - case 11: - utemp = 65536; - break; - case 12: - utemp = 65536; - break; - case 13: - utemp = 65536; - break; - case 14: - utemp = 65535; - break; - case 15: - default: - utemp = 0; - break; - } - //printf("getattackincrement returned %d\n",utemp); - return utemp; -} - -/*---------------------------------------------------------------------------- - * EAS_PERelease() - *---------------------------------------------------------------------------- - * Purpose: - * Put the PCM stream envelope into release. - * - * Inputs: - * pEASData - pointer to EAS library instance data - * handle - pointer to S_PCM_STATE for this stream - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_RESULT EAS_PERelease (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState) -{ - EAS_U32 utemp; - - //printf("handling note-off part of envelope\n"); - /*if the note is not ignore release or sustained*/ - if (((pState->envData >> 24) & 0x0F)==0) - { - /* set envelope state to release */ - pState->envState = PCM_ENV_RELEASE; - utemp = ((pState->envData >> 20) & 0x0F); - pState->envScale = getDecayScale(utemp); //getReleaseScale(utemp); - } - else - { - /*else change envelope state to sustain */ - pState->envState = PCM_ENV_SUSTAIN; - utemp = ((pState->envData >> 28) & 0x0F); - pState->envScale = getDecayScale(utemp); //getSustainScale(utemp); - } - //since we are in release, don't let anything hang around too long - //printf("checking env scale, val = %d\n",((S_PCM_STATE*) handle)->envScale); - if (pState->envScale > 505) - pState->envScale = 505; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * FindSlot() - *---------------------------------------------------------------------------- - * Purpose: - * Locates an empty stream slot and assigns the file handle - * - * Inputs: - * pEASData - pointer to EAS library instance data - * fileHandle - file handle - * pCallbackFunc - function to be called back upon EAS_STATE_STOPPED - * - * Outputs: - * returns handle to slot or NULL if all slots are used - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static S_PCM_STATE *FindSlot (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_PCM_CALLBACK pCallbackFunc, EAS_VOID_PTR cbInstData) -{ - EAS_INT i; - S_PCM_STATE *pState; - -#ifndef NO_PCM_STEAL - S_PCM_STATE *foundState = NULL; - EAS_INT count = 0; - EAS_U32 startOrder = 0xFFFFFFFF; - S_PCM_STATE *stealState = NULL; - EAS_U32 youngest = 0; - - /* find an empty slot, count total in use, and find oldest in use (lowest start order) */ - for (i = 0, pState = pEASData->pPCMStreams; i < MAX_PCM_STREAMS; i++, pState++) - { - /* if this one is available */ - if (pState->fileHandle == NULL) - { - foundState = pState; - } - /* else this one is in use, so see if it is the oldest, and count total in use */ - /* also find youngest */ - else - { - /*one more voice in use*/ - count++; - /* is this the oldest? (lowest start order) */ - if ((pState->state != EAS_STATE_STOPPING) && (pState->startOrder < startOrder)) - { - /* remember this one */ - stealState = pState; - /* remember the oldest so far */ - startOrder = pState->startOrder; - } - /* is this the youngest? (highest start order) */ - if (pState->startOrder >= youngest) - { - youngest = pState->startOrder; - } - } - } - - /* if there are too many voices active, stop the oldest one */ - if (count > PCM_STREAM_THRESHOLD) - { - //printf("stealing!!!\n"); - /* make sure we got one, although we should always have one at this point */ - if (stealState != NULL) - { - //flag this as stopping, so it will get shut off - stealState->state = EAS_STATE_STOPPING; - } - } - - /* if there are no available open streams (we won't likely see this, due to stealing) */ - if (foundState == NULL) - return NULL; - - /* save info */ - foundState->startOrder = youngest + 1; - foundState->fileHandle = fileHandle; - foundState->pCallback = pCallbackFunc; - foundState->cbInstData = cbInstData; - return foundState; -#else - /* find an empty slot*/ - for (i = 0; i < MAX_PCM_STREAMS; i++) - { - pState = &pEASData->pPCMStreams[i]; - if (pState->fileHandle != NULL) - continue; - - pState->fileHandle = fileHandle; - pState->pCallback = pCallbackFunc; - pState->cbInstData = cbInstData; - return pState; - } - return NULL; -#endif -} - -#ifdef _LOOKUP_SAMPLE_RATE -/*---------------------------------------------------------------------------- - * CalcBaseFreq() - *---------------------------------------------------------------------------- - * Purpose: - * Calculates the fractional phase increment for the sample rate converter - * - * Inputs: - * sampleRate - sample rate in samples/sec - * - * Outputs: - * Returns fractional sample rate with a 15-bit fraction - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_U32 CalcBaseFreq (EAS_U32 sampleRate) -{ - EAS_INT i; - - /* look up the conversion rate */ - for (i = 0; i < (EAS_INT)(SRC_CONV_RATE_ENTRIES); i ++) - { - if (srcConvRate[i][0] == sampleRate) - return srcConvRate[i][1]; - } - - /* if not found in table, do it the long way */ - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Sample rate %u not in table, calculating by division\n", sampleRate); */ } - - return (SRC_RATE_MULTIPLER * (EAS_U32) sampleRate) >> 15; -} -#endif - -/*---------------------------------------------------------------------------- - * InitPCMStream() - *---------------------------------------------------------------------------- - * Purpose: - * Start an ADPCM stream playback. Decodes the header, preps the engine. - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT InitPCMStream (S_EAS_DATA *pEASData, S_PCM_STATE *pState) -{ - - /* initialize the data structure */ - pState->bytesLeft = pState->byteCount; - pState->phase = 0; - pState->srcByte = 0; - pState->decoderL.acc = 0; - pState->decoderL.output = 0; - pState->decoderL.x0 = pState->decoderL.x1 = 0; - pState->decoderL.step = 0; - pState->decoderR.acc = 0; - pState->decoderR.output = 0; - pState->decoderR.x0 = pState->decoderR.x1 = 0; - pState->decoderR.step = 0; - pState->hiNibble = EAS_FALSE; - pState->pitch = 0; - pState->blockCount = 0; - pState->gainLeft = PCM_DEFAULT_GAIN_SETTING; -// pState->currentGainLeft = PCM_DEFAULT_GAIN_SETTING; - pState->envValue = 0; - pState->envState = PCM_ENV_START; - -#if (NUM_OUTPUT_CHANNELS == 2) - pState->gainRight = PCM_DEFAULT_GAIN_SETTING; -// pState->currentGainRight = PCM_DEFAULT_GAIN_SETTING; -#endif - pState->state = EAS_STATE_READY; - - /* initialize the decoder */ - if (pState->pDecoder->pfInit) - return (*pState->pDecoder->pfInit)(pEASData, pState); - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RenderPCMStream() - *---------------------------------------------------------------------------- - * Purpose: - * Decodes a buffer of ADPCM data. - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT RenderPCMStream (S_EAS_DATA *pEASData, S_PCM_STATE *pState, EAS_I32 numSamples) -{ - EAS_RESULT result; - EAS_U32 phaseInc; - EAS_I32 gainLeft, gainIncLeft; - EAS_I32 *pOut; - EAS_I32 temp; - EAS_U32 utemp; - -#if (NUM_OUTPUT_CHANNELS == 2) - EAS_I32 gainRight, gainIncRight; -#endif - -#if 0 - printf("env data: AR = %d, DR = %d, SL = %d, SR = %d, RR = %d\n", - ((pState->envData >> 12) & 0x0F), - ((pState->envData >> 16) & 0x0F), - ((pState->envData >> 8) & 0x0F), - ((pState->envData >> 28) & 0x0F), - ((pState->envData >> 20) & 0x0F)); -#endif - - if (pState->envState == PCM_ENV_START) - { - //printf("env start\n"); - utemp = ((pState->envData >> 12) & 0x0F); - //if fastest rate, attack is already completed - //do the same for slowest rate, since that allows zero to be passed for default envelope - if (utemp == 0x0F || utemp == 0x00) - { - //start envelope at full - pState->envValue = (32768<<7); - //jump right into decay - utemp = ((pState->envData >> 16) & 0x0F); - pState->envScale = getDecayScale(utemp); - pState->envState = PCM_ENV_DECAY; - pState->currentGainLeft = (EAS_I16) FMUL_15x15(pState->gainLeft, pState->volume); - pState->currentGainRight = (EAS_I16) FMUL_15x15(pState->gainRight, pState->volume); - } - //else attack has a ramp - else - { - //start the envelope very low - pState->envValue = (2<<7); - pState->currentGainLeft = 0; - pState->currentGainRight = 0; - //get envelope attack scaling value - pState->envScale = getAttackIncrement(utemp); - //go to attack state - pState->envState = PCM_ENV_ATTACK; - } - } - if (pState->envState == PCM_ENV_ATTACK) - { - //printf("env attack, env value = %d, env scale = %d\n",pState->envValue>>7,pState->envScale); - //update envelope value - pState->envValue = pState->envValue + (pState->envScale << 7); - //check envelope level and update state if needed - if (pState->envValue >= (32768<<7)) - { - pState->envValue = (32768<<7); - utemp = ((pState->envData >> 16) & 0x0F); - pState->envScale = getDecayScale(utemp); - pState->envState = PCM_ENV_DECAY; - } - } - else if (pState->envState == PCM_ENV_DECAY) - { - //printf("env decay, env value = %d, env scale = %d\n",pState->envValue>>7,pState->envScale); - //update envelope value - pState->envValue = (pState->envValue * pState->envScale)>>9; - //check envelope level against sustain level and update state if needed - utemp = ((pState->envData >> 8) & 0x0F); - if (utemp == (EAS_U32)0x0F) - utemp = (2<<7); - else - { - utemp = ((32769<<7) >> (utemp>>1)); - } - if (pState->envValue <= utemp) - { - utemp = ((pState->envData >> 28) & 0x0F); - pState->envScale = getDecayScale(utemp); //getSustainScale(utemp); - pState->envState = PCM_ENV_SUSTAIN; - } - } - else if (pState->envState == PCM_ENV_SUSTAIN) - { - //printf("env sustain, env value = %d, env scale = %d\n",pState->envValue>>7,pState->envScale); - //update envelope value - pState->envValue = (pState->envValue * pState->envScale)>>9; - //check envelope level against bottom level and update state if needed - if (pState->envValue <= (2<<7)) - { - //no more decay - pState->envScale = 512; - pState->envState = PCM_ENV_END; - } - } - else if (pState->envState == PCM_ENV_RELEASE) - { - //printf("env release, env value = %d, env scale = %d\n",pState->envValue>>7,pState->envScale); - //update envelope value - pState->envValue = (pState->envValue * pState->envScale)>>9; - //check envelope level against bottom level and update state if needed - if (pState->envValue <= (2<<7)) - { - //no more decay - pState->envScale = 512; - pState->envState = PCM_ENV_END; - } - } - else if (pState->envState == PCM_ENV_END) - { - //printf("env end\n"); - /* set state to stopping, already ramped down */ - pState->state = EAS_STATE_STOPPING; - } - - //pState->gainLeft = (EAS_U16)((pState->gainLeft * (pState->envValue>>7))>>15); - //pState->gainRight = (EAS_U16)((pState->gainRight * (pState->envValue>>7))>>15); - - /* gain to 32-bits to increase resolution on anti-zipper filter */ - /*lint -e{703} use shift for performance */ - gainLeft = (EAS_I32) pState->currentGainLeft << SYNTH_UPDATE_PERIOD_IN_BITS; -#if (NUM_OUTPUT_CHANNELS == 2) - /*lint -e{703} use shift for performance */ - gainRight = (EAS_I32) pState->currentGainRight << SYNTH_UPDATE_PERIOD_IN_BITS; -#endif - - /* calculate a new gain increment, gain target is zero if pausing */ - if ((pState->state == EAS_STATE_PAUSING) || (pState->state == EAS_STATE_PAUSED)) - { - gainIncLeft = -pState->currentGainLeft; -#if (NUM_OUTPUT_CHANNELS == 2) - gainIncRight= -pState->currentGainRight; -#endif - } - else - { - EAS_I32 gain = FMUL_15x15(pState->envValue >> 7, pState->volume); - gainIncLeft = FMUL_15x15(pState->gainLeft, gain) - pState->currentGainLeft; -#if (NUM_OUTPUT_CHANNELS == 2) - gainIncRight = FMUL_15x15(pState->gainRight, gain) - pState->currentGainRight; -#endif - } - - /* calculate phase increment */ - phaseInc = pState->basefreq; - - /* convert pitch cents to linear multiplier */ - if (pState->pitch) - { - temp = EAS_Calculate2toX(pState->pitch); - phaseInc = FMUL_15x15(phaseInc, temp); - } - phaseInc = phaseInc << pState->rateShift; - - /* pointer to mix buffer */ - pOut = pEASData->pMixBuffer; - - /* render a buffer of samples */ - while (numSamples--) - { - - /* interpolate an output sample */ - pState->decoderL.output = pState->decoderL.x0 + FMUL_15x15((pState->decoderL.x1 - pState->decoderL.x0), pState->phase & PHASE_FRAC_MASK); - - /* stereo output */ -#if (NUM_OUTPUT_CHANNELS == 2) - - /* stereo stream? */ - if (pState->flags & PCM_FLAGS_STEREO) - pState->decoderR.output = pState->decoderR.x0 + FMUL_15x15((pState->decoderR.x1 - pState->decoderR.x0), pState->phase & PHASE_FRAC_MASK); - - /* gain scale and mix */ - /*lint -e{704} use shift instead of division */ - *pOut++ += (pState->decoderL.output * (gainLeft >> SYNTH_UPDATE_PERIOD_IN_BITS)) >> PCM_MIXER_GUARD_BITS; - gainLeft += gainIncLeft; - - /*lint -e{704} use shift instead of division */ - if (pState->flags & PCM_FLAGS_STEREO) - *pOut++ += (pState->decoderR.output * (gainRight >> SYNTH_UPDATE_PERIOD_IN_BITS)) >> PCM_MIXER_GUARD_BITS; - else - *pOut++ += (pState->decoderL.output * (gainRight >> SYNTH_UPDATE_PERIOD_IN_BITS)) >> PCM_MIXER_GUARD_BITS; - - gainRight += gainIncRight; - - /* mono output */ -#else - /* if stereo stream, decode right channel and mix to mono */ - if (pState->flags & PCM_FLAGS_STEREO) - { - pState->decoderR.output= pState->decoderR.x0 + FMUL_15x15((pState->decoderR.x1 - pState->decoderR.x0), pState->phase & PHASE_FRAC_MASK); - - /* for mono, sum stereo ADPCM to mono */ - /*lint -e{704} use shift instead of division */ - *pOut++ += ((pState->decoderL.output + pState->decoderR.output) * (gainLeft >> SYNTH_UPDATE_PERIOD_IN_BITS)) >> PCM_MIXER_GUARD_BITS; - } - else - /*lint -e{704} use shift instead of division */ - *pOut++ += (pState->decoderL.output * (gainLeft >> SYNTH_UPDATE_PERIOD_IN_BITS)) >> PCM_MIXER_GUARD_BITS; - - gainLeft += gainIncLeft; -#endif - - /* advance phase accumulator */ - pState->phase += phaseInc; - - /* if integer part of phase accumulator is non-zero, advance to next sample */ - while (pState->phase & ~PHASE_FRAC_MASK) - { - pState->decoderL.x0 = pState->decoderL.x1; - pState->decoderR.x0 = pState->decoderR.x1; - - /* give the source a chance to continue the stream */ - if (!pState->bytesLeft && pState->pCallback && ((pState->flags & PCM_FLAGS_EMPTY) == 0)) - { - pState->flags |= PCM_FLAGS_EMPTY; - (*pState->pCallback)(pEASData, pState->cbInstData, pState, EAS_STATE_EMPTY); - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "RenderPCMStream: After empty callback, bytesLeft = %d\n", pState->bytesLeft); */ } - } - - /* decode the next sample */ - if ((result = (*pState->pDecoder->pfDecodeSample)(pEASData, pState)) != EAS_SUCCESS) - return result; - - /* adjust phase by one sample */ - pState->phase -= (1L << NUM_PHASE_FRAC_BITS); - } - - } - - /* save new gain */ - /*lint -e{704} use shift instead of division */ - pState->currentGainLeft = (EAS_I16) (gainLeft >> SYNTH_UPDATE_PERIOD_IN_BITS); - -#if (NUM_OUTPUT_CHANNELS == 2) - /*lint -e{704} use shift instead of division */ - pState->currentGainRight = (EAS_I16) (gainRight >> SYNTH_UPDATE_PERIOD_IN_BITS); -#endif - - /* if pausing, set new state and notify */ - if (pState->state == EAS_STATE_PAUSING) - { - pState->state = EAS_STATE_PAUSED; - if (pState->pCallback) - (*pState->pCallback)(pEASData, pState->cbInstData, pState, pState->state); - } - - /* if out of data, set stopped state and notify */ - if (pState->bytesLeft == 0 || pState->state == EAS_STATE_STOPPING) - { - pState->state = EAS_STATE_STOPPED; - - /* do callback unless the file has already been closed */ - if (pState->pCallback && pState->fileHandle) - (*pState->pCallback)(pEASData, pState->cbInstData, pState, pState->state); - } - - if (pState->state == EAS_STATE_READY) - pState->state = EAS_STATE_PLAY; - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * LinearPCMDecode() - *---------------------------------------------------------------------------- - * Purpose: - * Decodes a PCM sample - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT LinearPCMDecode (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState) -{ - EAS_RESULT result; - EAS_HW_DATA_HANDLE hwInstData; - - hwInstData = ((S_EAS_DATA*) pEASData)->hwInstData; - - /* if out of data, check for loop */ - if ((pState->bytesLeft == 0) && (pState->loopSamples != 0)) - { - if ((result = EAS_HWFileSeek(pEASData->hwInstData, pState->fileHandle, (EAS_I32) (pState->startPos + pState->loopLocation))) != EAS_SUCCESS) - return result; - pState->bytesLeft = pState->byteCount = (EAS_I32) pState->bytesLeftLoop; - pState->flags &= ~PCM_FLAGS_EMPTY; - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "LinearPCMDecode: Rewind file to %d, bytesLeft = %d\n", pState->startPos, pState->bytesLeft); */ } - } - - if (pState->bytesLeft) - { - - /* check format byte for 8-bit samples */ - if (pState->flags & PCM_FLAGS_8_BIT) - { - /* fetch left or mono sample */ - if ((result = EAS_HWGetByte(hwInstData, pState->fileHandle, &pState->srcByte)) != EAS_SUCCESS) - return result; - - /* if unsigned */ - if (pState->flags & PCM_FLAGS_UNSIGNED) - { - /*lint -e{734} converting unsigned 8-bit to signed 16-bit */ - pState->decoderL.x1 = (EAS_PCM)(((EAS_PCM) pState->srcByte << 8) ^ 0x8000); - } - else - { - /*lint -e{734} converting signed 8-bit to signed 16-bit */ - pState->decoderL.x1 = (EAS_PCM)((EAS_PCM) pState->srcByte << 8); - } - pState->bytesLeft--; - - /* fetch right sample */ - if(pState->flags & PCM_FLAGS_STEREO) - { - if ((result = EAS_HWGetByte(hwInstData, pState->fileHandle, &pState->srcByte)) != EAS_SUCCESS) - return result; - - /* if unsigned */ - if (pState->flags & PCM_FLAGS_UNSIGNED) - { - /*lint -e{734} converting unsigned 8-bit to signed 16-bit */ - pState->decoderR.x1 = (EAS_PCM)(((EAS_PCM) pState->srcByte << 8) ^ 0x8000); - } - else - { - /*lint -e{734} converting signed 8-bit to signed 16-bit */ - pState->decoderR.x1 = (EAS_PCM)((EAS_PCM) pState->srcByte << 8); - } - pState->bytesLeft--; - } - } - - /* must be 16-bit samples */ - else - { - //unsigned 16 bit currently not supported - if (pState->flags & PCM_FLAGS_UNSIGNED) - { - return EAS_ERROR_INVALID_PCM_TYPE; - } - - /* fetch left or mono sample */ - if ((result = EAS_HWGetWord(hwInstData, pState->fileHandle, &pState->decoderL.x1, EAS_FALSE)) != EAS_SUCCESS) - return result; - pState->bytesLeft -= 2; - - /* fetch right sample */ - if(pState->flags & PCM_FLAGS_STEREO) - { - if ((result = EAS_HWGetWord(hwInstData, pState->fileHandle, &pState->decoderR.x1, EAS_FALSE)) != EAS_SUCCESS) - return result; - pState->bytesLeft -= 2; - } - } - } - - /* no more data, force zero samples */ - else - pState->decoderL.x1 = pState->decoderR.x1 = 0; - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * LinearPCMLocate() - *---------------------------------------------------------------------------- - * Purpose: - * Locate in a linear PCM stream - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT LinearPCMLocate (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 time) -{ - EAS_RESULT result; - EAS_I32 temp; - EAS_I32 secs, msecs; - EAS_INT shift; - - /* calculate size of sample frame */ - if (pState->flags & PCM_FLAGS_8_BIT) - shift = 0; - else - shift = 1; - if (pState->flags & PCM_FLAGS_STEREO) - shift++; - - /* break down into secs and msecs */ - secs = time / 1000; - msecs = time - (secs * 1000); - - /* calculate sample number fraction from msecs */ - temp = (msecs * pState->sampleRate); - temp = (temp >> 10) + ((temp * 49) >> 21); - - /* add integer sample count */ - temp += secs * pState->sampleRate; - - /* calculate the position based on sample frame size */ - /*lint -e{703} use shift for performance */ - temp <<= shift; - - /* past end of sample? */ - if (temp > (EAS_I32) pState->loopStart) - { - /* if not looped, flag error */ - if (pState->loopSamples == 0) - { - pState->bytesLeft = 0; - pState->flags |= PCM_FLAGS_EMPTY; - return EAS_ERROR_LOCATE_BEYOND_END; - } - - /* looped sample - calculate position in loop */ - while (temp > (EAS_I32) pState->loopStart) - temp -= (EAS_I32) pState->loopStart; - } - - /* seek to new position */ - if ((result = EAS_PESeek(pEASData, pState, &temp)) != EAS_SUCCESS) - return result; - - /* reset state */ - if ((pState->state != EAS_STATE_PAUSING) && (pState->state != EAS_STATE_PAUSED)) - pState->state = EAS_STATE_READY; - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_PESeek - *---------------------------------------------------------------------------- - * Purpose: - * Locate to a particular byte in a PCM stream - *---------------------------------------------------------------------------- - * This bit is tricky because the chunks may not be contiguous, - * so we have to rely on the parser to position in the file. We - * do this by seeking to the end of each chunk and simulating an - * empty buffer condition until we get to where we want to go. - * - * A better solution would be a parser API for re-positioning, - * but there isn't time at the moment to re-factor all the - * parsers to support a new API. - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PESeek (S_EAS_DATA *pEASData, S_PCM_STATE *pState, EAS_I32 *pLocation) -{ - EAS_RESULT result; - - /* seek to start of audio */ - if ((result = EAS_HWFileSeek(pEASData->hwInstData, pState->fileHandle, pState->startPos)) != EAS_SUCCESS) - { - pState->state = EAS_STATE_ERROR; - return result; - } - pState->bytesLeft = pState->bytesLeftLoop; - - /* skip through chunks until we find the right chunk */ - while (*pLocation > (EAS_I32) pState->bytesLeft) - { - /* seek to end of audio chunk */ - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "EAS_PESeek: Seek to offset = %d\n", pState->bytesLeft); */ } - if ((result = EAS_HWFileSeekOfs(pEASData->hwInstData, pState->fileHandle, pState->bytesLeft)) != EAS_SUCCESS) - { - pState->state = EAS_STATE_ERROR; - return result; - } - *pLocation -= pState->bytesLeft; - pState->bytesLeft = 0; - pState->flags |= PCM_FLAGS_EMPTY; - - /* retrieve more data */ - if (pState->pCallback) - (*pState->pCallback)(pEASData, pState->cbInstData, pState, EAS_STATE_EMPTY); - - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "EAS_PESeek: bytesLeft=%d, byte location = %d\n", pState->bytesLeft, *pLocation); */ } - - /* no more samples */ - if (pState->bytesLeft == 0) - return EAS_ERROR_LOCATE_BEYOND_END; - } - - /* seek to new offset in current chunk */ - if (*pLocation > 0) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "EAS_PESeek: Seek to offset = %d\n", *pLocation); */ } - if ((result = EAS_HWFileSeekOfs(pEASData->hwInstData, pState->fileHandle, *pLocation)) != EAS_SUCCESS) - { - pState->state = EAS_STATE_ERROR; - return result; - } - - /* if not streamed, calculate number of bytes left */ - if (pState->flags & PCM_FLAGS_STREAMING) - pState->bytesLeft = 0x7fffffff; - else - pState->bytesLeft -= *pLocation; - } - return EAS_SUCCESS; -} - diff --git a/arm-hybrid-22k/lib_src/eas_pcm.h b/arm-hybrid-22k/lib_src/eas_pcm.h deleted file mode 100644 index c161757..0000000 --- a/arm-hybrid-22k/lib_src/eas_pcm.h +++ /dev/null @@ -1,359 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_pcm.h - * - * Contents and purpose: - * External function prototypes for eas_pcm.c module - * - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 847 $ - * $Date: 2007-08-27 21:30:08 -0700 (Mon, 27 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_PCM_H -#define _EAS_PCM_H - -/* default gain setting - roughly unity gain */ -#define PCM_DEFAULT_GAIN_SETTING 0x6000 - -typedef struct s_pcm_state_tag *EAS_PCM_HANDLE; -typedef void (*EAS_PCM_CALLBACK) (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR cbInstData, EAS_PCM_HANDLE pcmHandle, EAS_STATE state); - -/* parameters for EAS_PEOpenStream */ -typedef struct s_pcm_open_params_tag -{ - EAS_FILE_HANDLE fileHandle; - EAS_I32 decoder; - EAS_U32 sampleRate; - EAS_I32 size; - EAS_U32 loopStart; - EAS_U32 loopSamples; - EAS_I32 blockSize; - EAS_U32 flags; - EAS_U32 envData; - EAS_I16 volume; - EAS_PCM_CALLBACK pCallbackFunc; - EAS_VOID_PTR cbInstData; - } S_PCM_OPEN_PARAMS; - -/*---------------------------------------------------------------------------- - * EAS_PEInit() - *---------------------------------------------------------------------------- - * Purpose: - * Initializes the PCM engine - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEInit (EAS_DATA_HANDLE pEASData); - -/*---------------------------------------------------------------------------- - * EAS_PEShutdown() - *---------------------------------------------------------------------------- - * Purpose: - * Shuts down the PCM engine - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEShutdown (EAS_DATA_HANDLE pEASData); - -/*---------------------------------------------------------------------------- - * EAS_PEOpenStream() - *---------------------------------------------------------------------------- - * Purpose: - * Starts up a PCM playback - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEOpenStream (EAS_DATA_HANDLE pEASData, S_PCM_OPEN_PARAMS *pParams, EAS_PCM_HANDLE *pHandle); - -/*---------------------------------------------------------------------------- - * EAS_PEContinueStream() - *---------------------------------------------------------------------------- - * Purpose: - * Continues a PCM stream - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEContinueStream (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle, EAS_I32 size); - -/*---------------------------------------------------------------------------- - * EAS_PEGetFileHandle() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the file handle of a stream - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEGetFileHandle (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle, EAS_FILE_HANDLE *pFileHandle); - -/*---------------------------------------------------------------------------- - * EAS_PERender() - *---------------------------------------------------------------------------- - * Purpose: - * Render a buffer of PCM audio - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PERender (EAS_DATA_HANDLE pEASData, EAS_I32 numSamples); - -/*---------------------------------------------------------------------------- - * EAS_PEUpdateParams() - *---------------------------------------------------------------------------- - * Purpose: - * Update the pitch and volume parameters using MIDI controls - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEUpdateParams (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE pState, EAS_I16 pitch, EAS_I16 gainLeft, EAS_I16 gainRight); - -/*---------------------------------------------------------------------------- - * EAS_PELocate() - *---------------------------------------------------------------------------- - * Purpose: - * This function seeks to the requested place in the file. Accuracy - * is dependent on the sample rate and block size. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * pState - stream handle - * time - media time in milliseconds - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PELocate (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE pState, EAS_I32 time); - -/*---------------------------------------------------------------------------- - * EAS_PEUpdateVolume() - *---------------------------------------------------------------------------- - * Purpose: - * Update the volume parameters for a PCM stream - * - * Inputs: - * pEASData - pointer to EAS library instance data - * handle - pointer to S_PCM_STATE for this stream - * gainLeft - linear gain multipler in 1.15 fraction format - * gainRight - linear gain multipler in 1.15 fraction format - * initial - initial settings, set current gain - * - * Outputs: - * - * - * Side Effects: - * - * Notes - * In mono mode, leftGain controls the output gain and rightGain is ignored - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_RESULT EAS_PEUpdateVolume (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE pState, EAS_I16 volume); - -/*---------------------------------------------------------------------------- - * EAS_PEUpdatePitch() - *---------------------------------------------------------------------------- - * Purpose: - * Update the pitch parameter for a PCM stream - * - * Inputs: - * pEASData - pointer to EAS library instance data - * pState - pointer to S_PCM_STATE for this stream - * pitch - new pitch value in pitch cents - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_RESULT EAS_PEUpdatePitch (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE pState, EAS_I16 pitch); - -/*---------------------------------------------------------------------------- - * EAS_PEState() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current state of the stream - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * pState - pointer to variable to store state - * - * Outputs: - * - * - * Side Effects: - * - * Notes: - * This interface is also exposed in the internal library for use by the other modules. - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEState (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle, EAS_STATE *pState); - -/*---------------------------------------------------------------------------- - * EAS_PEClose() - *---------------------------------------------------------------------------- - * Purpose: - * Close the file and clean up - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEClose (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle); - -/*---------------------------------------------------------------------------- - * EAS_PEReset() - *---------------------------------------------------------------------------- - * Purpose: - * Reset the sequencer. Used for locating backwards in the file. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEReset (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle); - -/*---------------------------------------------------------------------------- - * EAS_PEPause() - *---------------------------------------------------------------------------- - * Purpose: - * Mute and pause rendering a PCM stream. Sets the gain target to zero and stops the playback - * at the end of the next audio frame. - * - * Inputs: - * pEASData - pointer to EAS library instance data - * handle - pointer to S_PCM_STATE for this stream - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEPause (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle); - -/*---------------------------------------------------------------------------- - * EAS_PEResume() - *---------------------------------------------------------------------------- - * Purpose: - * Resume rendering a PCM stream. Sets the gain target back to its - * previous setting and restarts playback at the end of the next audio - * frame. - * - * Inputs: - * pEASData - pointer to EAS library instance data - * handle - pointer to S_PCM_STATE for this stream - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PEResume (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle); - -/*---------------------------------------------------------------------------- - * EAS_PERelease() - *---------------------------------------------------------------------------- - * Purpose: - * Put the PCM stream envelope into release. - * - * Inputs: - * pEASData - pointer to EAS library instance data - * handle - pointer to S_PCM_STATE for this stream - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PERelease (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle); - -#endif /* end _EAS_PCM_H */ - diff --git a/arm-hybrid-22k/lib_src/eas_pcmdata.c b/arm-hybrid-22k/lib_src/eas_pcmdata.c deleted file mode 100644 index 5649f07..0000000 --- a/arm-hybrid-22k/lib_src/eas_pcmdata.c +++ /dev/null @@ -1,35 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_pcmdata.c - * - * Contents and purpose: - * Contains the static data for the PCM engine. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 547 $ - * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_data.h" - -/* static data allocation */ -S_PCM_STATE eas_PCMData[MAX_PCM_STREAMS]; - - diff --git a/arm-hybrid-22k/lib_src/eas_pcmdata.h b/arm-hybrid-22k/lib_src/eas_pcmdata.h deleted file mode 100644 index be2f8e5..0000000 --- a/arm-hybrid-22k/lib_src/eas_pcmdata.h +++ /dev/null @@ -1,157 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_pcmdata.h - * - * Contents and purpose: - * Data declarations for the PCM engine - * - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 847 $ - * $Date: 2007-08-27 21:30:08 -0700 (Mon, 27 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_PCMDATA_H -#define _EAS_PCMDATA_H - -/* sets the maximum number of simultaneous PCM streams */ -#ifndef MAX_PCM_STREAMS -#define MAX_PCM_STREAMS 16 -#define PCM_STREAM_THRESHOLD (MAX_PCM_STREAMS - 4) -#endif - -/* coefficents for high-pass filter in ADPCM */ -#define INTEGRATOR_COEFFICIENT 100 /* coefficient for leaky integrator */ - -/* additional flags in S_PCM_STATE.flags used internal to module */ -#define PCM_FLAGS_EMPTY 0x01000000 /* unsigned format */ - -/*---------------------------------------------------------------------------- - * S_PCM_STATE - * - * Retains state information for PCM streams. - *---------------------------------------------------------------------------- -*/ -typedef struct s_decoder_state_tag -{ - EAS_I32 output; /* last output for DC offset filter */ - EAS_I32 acc; /* accumulator for DC offset filter */ - EAS_I32 step; /* current ADPCM step size */ - EAS_PCM x1; /* current generated sample */ - EAS_PCM x0; /* previous generated sample */ -} S_DECODER_STATE; - -typedef enum -{ - PCM_ENV_START = 0, - PCM_ENV_ATTACK, - PCM_ENV_DECAY, - PCM_ENV_SUSTAIN, - PCM_ENV_RELEASE, - PCM_ENV_END -} E_PCM_ENV_STATE; - -typedef struct s_pcm_state_tag -{ -#ifdef _CHECKED_BUILD - EAS_U32 handleCheck; /* signature check for checked build */ -#endif - EAS_FILE_HANDLE fileHandle; /* pointer to input file */ - EAS_PCM_CALLBACK pCallback; /* pointer to callback function */ - EAS_VOID_PTR cbInstData; /* instance data for callback function */ - struct s_decoder_interface_tag EAS_CONST * pDecoder; /* pointer to decoder interface */ - EAS_STATE state; /* stream state */ - EAS_I32 time; /* media time */ - EAS_I32 startPos; /* start of PCM stream */ - EAS_I32 loopLocation; /* file location where loop starts */ - EAS_I32 byteCount; /* size of file */ - EAS_U32 loopStart; /* loop start, offset in samples from startPos */ - /* NOTE: For CMF, we use this to store total sample size */ - EAS_U32 loopSamples; /* total loop length, in samples, 0 means no loop */ - /* NOTE: For CMF, non-zero means looped */ - EAS_U32 samplesInLoop; /* samples left in the loop to play back */ - EAS_I32 samplesTilLoop; /* samples left to play until top of loop */ - EAS_I32 bytesLeft; /* count of bytes left in stream */ - EAS_I32 bytesLeftLoop; /* count of bytes left in stream, value at start of loop */ - EAS_U32 phase; /* current phase for interpolator */ - EAS_U32 basefreq; /* frequency multiplier */ - EAS_U32 flags; /* stream flags */ - EAS_U32 envData; /* envelope data (and LFO data) */ - EAS_U32 envValue; /* current envelope value */ - EAS_U32 envScale; /* current envelope scale */ - EAS_U32 startOrder; /* start order index, first is 0, next is 1, etc. */ - S_DECODER_STATE decoderL; /* left (mono) ADPCM state */ - S_DECODER_STATE decoderR; /* right ADPCM state */ - S_DECODER_STATE decoderLLoop; /* left (mono) ADPCM state, value at start of loop */ - S_DECODER_STATE decoderRLoop; /* right ADPCM state, value at start of loop */ - E_PCM_ENV_STATE envState; /* current envelope state */ - EAS_I16 volume; /* volume for stream */ - EAS_I16 pitch; /* relative pitch in cents - zero is unity playback */ - EAS_I16 gainLeft; /* requested gain */ - EAS_I16 gainRight; /* requested gain */ - EAS_I16 currentGainLeft; /* current gain for anti-zipper filter */ - EAS_I16 currentGainRight; /* current gain for anti-zipper filter */ - EAS_U16 blockSize; /* block size for ADPCM decoder */ - EAS_U16 blockCount; /* block counter for ADPCM decoder */ - EAS_U16 sampleRate; /* input sample rate */ - EAS_U8 srcByte; /* source byte */ - EAS_U8 msBitCount; /* count keeps track of MS bits */ - EAS_U8 msBitMask; /* mask keeps track of MS bits */ - EAS_U8 msBitValue; /* value keeps track of MS bits */ - EAS_U8 msBitCountLoop; /* count keeps track of MS bits, value at loop start */ - EAS_U8 msBitMaskLoop; /* mask keeps track of MS bits, value at loop start */ - EAS_U8 msBitValueLoop; /* value keeps track of MS bits, value at loop start */ - EAS_BOOL8 hiNibble; /* indicates high/low nibble is next */ - EAS_BOOL8 hiNibbleLoop; /* indicates high/low nibble is next, value loop start */ - EAS_U8 rateShift; /* for playback rate greater than 1.0 */ -} S_PCM_STATE; - -/*---------------------------------------------------------------------------- - * S_DECODER_INTERFACE - * - * Generic interface for audio decoders - *---------------------------------------------------------------------------- -*/ -typedef struct s_decoder_interface_tag -{ - EAS_RESULT (* EAS_CONST pfInit)(EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState); - EAS_RESULT (* EAS_CONST pfDecodeSample)(EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState); - EAS_RESULT (* EAS_CONST pfLocate)(EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 time); -} S_DECODER_INTERFACE; - - -/* header chunk for SMAF ADPCM */ -#define TAG_YAMAHA_ADPCM 0x4d776100 -#define TAG_MASK 0xffffff00 -#define TAG_RIFF_FILE 0x52494646 -#define TAG_WAVE_CHUNK 0x57415645 -#define TAG_FMT_CHUNK 0x666d7420 - -/*---------------------------------------------------------------------------- - * EAS_PESeek - *---------------------------------------------------------------------------- - * Purpose: - * Locate to a particular byte in a PCM stream - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_PESeek (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 *pLocation); - -#endif /* _EAS_PCMDATA_H */ - diff --git a/arm-hybrid-22k/lib_src/eas_public.c b/arm-hybrid-22k/lib_src/eas_public.c deleted file mode 100644 index ac43261..0000000 --- a/arm-hybrid-22k/lib_src/eas_public.c +++ /dev/null @@ -1,2597 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_public.c - * - * Contents and purpose: - * Contains EAS library public interface - * - * Copyright Sonic Network Inc. 2004 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 842 $ - * $Date: 2007-08-23 14:32:31 -0700 (Thu, 23 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_synthcfg.h" -#include "eas.h" -#include "eas_config.h" -#include "eas_host.h" -#include "eas_report.h" -#include "eas_data.h" -#include "eas_parser.h" -#include "eas_pcm.h" -#include "eas_midi.h" -#include "eas_mixer.h" -#include "eas_build.h" -#include "eas_vm_protos.h" -#include "eas_math.h" - -#ifdef JET_INTERFACE -#include "jet_data.h" -#endif - -#ifdef DLS_SYNTHESIZER -#include "eas_mdls.h" -#endif - -/* number of events to parse before calling EAS_HWYield function */ -#define YIELD_EVENT_COUNT 10 - -/*---------------------------------------------------------------------------- - * easLibConfig - * - * This structure is available through the EAS public interface to allow - * the user to check the configuration of the library. - *---------------------------------------------------------------------------- -*/ -static const S_EAS_LIB_CONFIG easLibConfig = -{ - LIB_VERSION, -#ifdef _CHECKED_BUILD - EAS_TRUE, -#else - EAS_FALSE, -#endif - MAX_SYNTH_VOICES, - NUM_OUTPUT_CHANNELS, - _OUTPUT_SAMPLE_RATE, - BUFFER_SIZE_IN_MONO_SAMPLES, -#ifdef _FILTER_ENABLED - EAS_TRUE, -#else - EAS_FALSE, -#endif - _BUILD_TIME_, - _BUILD_VERSION_ -}; - -/* local prototypes */ -static EAS_RESULT EAS_ParseEvents (S_EAS_DATA *pEASData, S_EAS_STREAM *pStream, EAS_U32 endTime, EAS_INT parseMode); - -/*---------------------------------------------------------------------------- - * EAS_SetStreamParameter - *---------------------------------------------------------------------------- - * Sets the specified parameter in the stream. Allows access to - * customizable settings within the individual file parsers. - *---------------------------------------------------------------------------- - * pEASData - pointer to EAS persistent data object - * pStream - stream handle - * param - enumerated parameter (see eas_parser.h) - * value - new value - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_SetStreamParameter (S_EAS_DATA *pEASData, EAS_HANDLE pStream, EAS_I32 param, EAS_I32 value) -{ - S_FILE_PARSER_INTERFACE *pParserModule; - - pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; - if (pParserModule->pfSetData) - return (*pParserModule->pfSetData)(pEASData, pStream->handle, param, value); - return EAS_ERROR_FEATURE_NOT_AVAILABLE; -} - -/*---------------------------------------------------------------------------- - * EAS_GetStreamParameter - *---------------------------------------------------------------------------- - * Sets the specified parameter in the stream. Allows access to - * customizable settings within the individual file parsers. - *---------------------------------------------------------------------------- - * pEASData - pointer to EAS persistent data object - * pStream - stream handle - * param - enumerated parameter (see eas_parser.h) - * pValue - pointer to variable to receive current setting - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_GetStreamParameter (S_EAS_DATA *pEASData, EAS_HANDLE pStream, EAS_I32 param, EAS_I32 *pValue) -{ - S_FILE_PARSER_INTERFACE *pParserModule; - - pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; - if (pParserModule->pfGetData) - return (*pParserModule->pfGetData)(pEASData, pStream->handle, param, pValue); - return EAS_ERROR_FEATURE_NOT_AVAILABLE; -} - -/*---------------------------------------------------------------------------- - * EAS_StreamReady() - *---------------------------------------------------------------------------- - * This routine sets common parameters like transpose, volume, etc. - * First, it attempts to use the parser EAS_SetStreamParameter interface. If that - * fails, it attempts to get the synth handle from the parser and - * set the parameter directly on the synth. This eliminates duplicate - * code in the parser. - *---------------------------------------------------------------------------- -*/ -EAS_BOOL EAS_StreamReady (S_EAS_DATA *pEASData, EAS_HANDLE pStream) -{ - S_FILE_PARSER_INTERFACE *pParserModule; - EAS_STATE state; - - pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; - if (pParserModule->pfState(pEASData, pStream->handle, &state) != EAS_SUCCESS) - return EAS_FALSE; - return (state < EAS_STATE_OPEN); -} - -/*---------------------------------------------------------------------------- - * EAS_IntSetStrmParam() - *---------------------------------------------------------------------------- - * This routine sets common parameters like transpose, volume, etc. - * First, it attempts to use the parser EAS_SetStreamParameter interface. If that - * fails, it attempts to get the synth handle from the parser and - * set the parameter directly on the synth. This eliminates duplicate - * code in the parser. - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_IntSetStrmParam (S_EAS_DATA *pEASData, EAS_HANDLE pStream, EAS_INT param, EAS_I32 value) -{ - S_SYNTH *pSynth; - - /* try to set the parameter using stream interface */ - if (EAS_SetStreamParameter(pEASData, pStream, param, value) == EAS_SUCCESS) - return EAS_SUCCESS; - - /* get a pointer to the synth object and set it directly */ - /*lint -e{740} we are cheating by passing a pointer through this interface */ - if (EAS_GetStreamParameter(pEASData, pStream, PARSER_DATA_SYNTH_HANDLE, (EAS_I32*) &pSynth) != EAS_SUCCESS) - return EAS_ERROR_INVALID_PARAMETER; - - if (pSynth == NULL) - return EAS_ERROR_INVALID_PARAMETER; - - switch (param) - { - -#ifdef DLS_SYNTHESIZER - case PARSER_DATA_DLS_COLLECTION: - { - EAS_RESULT result = VMSetDLSLib(pSynth, (EAS_DLSLIB_HANDLE) value); - if (result == EAS_SUCCESS) - { - DLSAddRef((S_DLS*) value); - VMInitializeAllChannels(pEASData->pVoiceMgr, pSynth); - } - return result; - } -#endif - - case PARSER_DATA_EAS_LIBRARY: - return VMSetEASLib(pSynth, (EAS_SNDLIB_HANDLE) value); - - case PARSER_DATA_POLYPHONY: - return VMSetPolyphony(pEASData->pVoiceMgr, pSynth, value); - - case PARSER_DATA_PRIORITY: - return VMSetPriority(pEASData->pVoiceMgr, pSynth, value); - - case PARSER_DATA_TRANSPOSITION: - VMSetTranposition(pSynth, value); - break; - - case PARSER_DATA_VOLUME: - VMSetVolume(pSynth, (EAS_U16) value); - break; - - default: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Invalid paramter %d in call to EAS_IntSetStrmParam", param); */ } - return EAS_ERROR_INVALID_PARAMETER; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_IntGetStrmParam() - *---------------------------------------------------------------------------- - * This routine gets common parameters like transpose, volume, etc. - * First, it attempts to use the parser EAS_GetStreamParameter interface. If that - * fails, it attempts to get the synth handle from the parser and - * get the parameter directly on the synth. - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_IntGetStrmParam (S_EAS_DATA *pEASData, EAS_HANDLE pStream, EAS_INT param, EAS_I32 *pValue) -{ - S_SYNTH *pSynth; - - /* try to set the parameter */ - if (EAS_GetStreamParameter(pEASData, pStream, param, pValue) == EAS_SUCCESS) - return EAS_SUCCESS; - - /* get a pointer to the synth object and retrieve data directly */ - /*lint -e{740} we are cheating by passing a pointer through this interface */ - if (EAS_GetStreamParameter(pEASData, pStream, PARSER_DATA_SYNTH_HANDLE, (EAS_I32*) &pSynth) != EAS_SUCCESS) - return EAS_ERROR_INVALID_PARAMETER; - - if (pSynth == NULL) - return EAS_ERROR_INVALID_PARAMETER; - - switch (param) - { - case PARSER_DATA_POLYPHONY: - return VMGetPolyphony(pEASData->pVoiceMgr, pSynth, pValue); - - case PARSER_DATA_PRIORITY: - return VMGetPriority(pEASData->pVoiceMgr, pSynth, pValue); - - case PARSER_DATA_TRANSPOSITION: - VMGetTranposition(pSynth, pValue); - break; - - case PARSER_DATA_NOTE_COUNT: - *pValue = VMGetNoteCount(pSynth); - break; - - default: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Invalid paramter %d in call to EAS_IntSetStrmParam", param); */ } - return EAS_ERROR_INVALID_PARAMETER; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_AllocateStream() - *---------------------------------------------------------------------------- - * Purpose: - * Allocates a stream handle - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -static EAS_INT EAS_AllocateStream (EAS_DATA_HANDLE pEASData) -{ - EAS_INT streamNum; - - /* check for static allocation, only one stream allowed */ - if (pEASData->staticMemoryModel) - { - if (pEASData->streams[0].handle != NULL) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Attempt to open multiple streams in static model\n"); */ } - return -1; - } - return 0; - } - - /* dynamic model */ - for (streamNum = 0; streamNum < MAX_NUMBER_STREAMS; streamNum++) - if (pEASData->streams[streamNum].handle == NULL) - break; - if (streamNum == MAX_NUMBER_STREAMS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Exceeded maximum number of open streams\n"); */ } - return -1; - } - return streamNum; -} - -/*---------------------------------------------------------------------------- - * EAS_InitStream() - *---------------------------------------------------------------------------- - * Purpose: - * Initialize a stream - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -static void EAS_InitStream (S_EAS_STREAM *pStream, EAS_VOID_PTR pParserModule, EAS_VOID_PTR streamHandle) -{ - pStream->pParserModule = pParserModule; - pStream->handle = streamHandle; - pStream->time = 0; - pStream->frameLength = AUDIO_FRAME_LENGTH; - pStream->repeatCount = 0; - pStream->volume = DEFAULT_STREAM_VOLUME; -} - -/*---------------------------------------------------------------------------- - * EAS_Config() - *---------------------------------------------------------------------------- - * Purpose: - * Returns a pointer to a structure containing the configuration options - * in this library build. - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC const S_EAS_LIB_CONFIG *EAS_Config (void) -{ - return &easLibConfig; -} - -/*---------------------------------------------------------------------------- - * EAS_Init() - *---------------------------------------------------------------------------- - * Purpose: - * Initialize the synthesizer library - * - * Inputs: - * ppEASData - pointer to data handle variable for this instance - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_Init (EAS_DATA_HANDLE *ppEASData) -{ - EAS_HW_DATA_HANDLE pHWInstData; - EAS_RESULT result; - S_EAS_DATA *pEASData; - EAS_INT module; - EAS_BOOL staticMemoryModel; - - /* get the memory model */ - staticMemoryModel = EAS_CMStaticMemoryModel(); - - /* initialize the host wrapper interface */ - *ppEASData = NULL; - if ((result = EAS_HWInit(&pHWInstData)) != EAS_SUCCESS) - return result; - - /* check Configuration Module for S_EAS_DATA allocation */ - if (staticMemoryModel) - pEASData = EAS_CMEnumData(EAS_CM_EAS_DATA); - else - pEASData = EAS_HWMalloc(pHWInstData, sizeof(S_EAS_DATA)); - if (!pEASData) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Failed to allocate EAS library memory\n"); */ } - return EAS_ERROR_MALLOC_FAILED; - } - - /* initialize some data */ - EAS_HWMemSet(pEASData, 0, sizeof(S_EAS_DATA)); - pEASData->staticMemoryModel = (EAS_BOOL8) staticMemoryModel; - pEASData->hwInstData = pHWInstData; - pEASData->renderTime = 0; - - /* set header search flag */ -#ifdef FILE_HEADER_SEARCH - pEASData->searchHeaderFlag = EAS_TRUE; -#endif - - /* initalize parameters */ - EAS_SetVolume(pEASData, NULL, DEFAULT_VOLUME); - -#ifdef _METRICS_ENABLED - /* initalize the metrics module */ - pEASData->pMetricsModule = EAS_CMEnumOptModules(EAS_MODULE_METRICS); - if (pEASData->pMetricsModule != NULL) - { - if ((result = (*pEASData->pMetricsModule->pfInit)(pEASData, &pEASData->pMetricsData)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %ld initializing metrics module\n", result); */ } - return result; - } - } -#endif - - /* initailize the voice manager & synthesizer */ - if ((result = VMInitialize(pEASData)) != EAS_SUCCESS) - return result; - - /* initialize mix engine */ - if ((result = EAS_MixEngineInit(pEASData)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %ld starting up mix engine\n", result); */ } - return result; - } - - /* initialize effects modules */ - for (module = 0; module < NUM_EFFECTS_MODULES; module++) - { - pEASData->effectsModules[module].effect = EAS_CMEnumFXModules(module); - if (pEASData->effectsModules[module].effect != NULL) - { - if ((result = (*pEASData->effectsModules[module].effect->pfInit)(pEASData, &pEASData->effectsModules[module].effectData)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Initialization of effects module %d returned %d\n", module, result); */ } - return result; - } - } - } - - /* initialize PCM engine */ - if ((result = EAS_PEInit(pEASData)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "EAS_PEInit failed with error code %ld\n", result); */ } - return result; - } - - /* return instance data pointer to host */ - *ppEASData = pEASData; - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_Shutdown() - *---------------------------------------------------------------------------- - * Purpose: - * Shuts down the library. Deallocates any memory associated with the - * synthesizer (dynamic memory model only) - * - * Inputs: - * pEASData - handle to data for this instance - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_Shutdown (EAS_DATA_HANDLE pEASData) -{ - EAS_HW_DATA_HANDLE hwInstData; - EAS_RESULT result, reportResult; - EAS_INT i; - - /* establish pointers */ - hwInstData = pEASData->hwInstData; - - /* check for NULL handle */ - if (!pEASData) - return EAS_ERROR_HANDLE_INTEGRITY; - - /* if there are streams open, close them */ - reportResult = EAS_SUCCESS; - for (i = 0; i < MAX_NUMBER_STREAMS; i++) - { - if (pEASData->streams[i].pParserModule && pEASData->streams[i].handle) - { - if ((result = (*((S_FILE_PARSER_INTERFACE*)(pEASData->streams[i].pParserModule))->pfClose)(pEASData, pEASData->streams[i].handle)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %ld shutting down parser module\n", result); */ } - reportResult = result; - } - } - } - - /* shutdown PCM engine */ - if ((result = EAS_PEShutdown(pEASData)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %ld shutting down PCM engine\n", result); */ } - if (reportResult == EAS_SUCCESS) - reportResult = result; - } - - /* shutdown mix engine */ - if ((result = EAS_MixEngineShutdown(pEASData)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %ld shutting down mix engine\n", result); */ } - if (reportResult == EAS_SUCCESS) - reportResult = result; - } - - /* shutdown effects modules */ - for (i = 0; i < NUM_EFFECTS_MODULES; i++) - { - if (pEASData->effectsModules[i].effect) - { - if ((result = (*pEASData->effectsModules[i].effect->pfShutdown)(pEASData, pEASData->effectsModules[i].effectData)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Shutdown of effects module %d returned %d\n", i, result); */ } - if (reportResult == EAS_SUCCESS) - reportResult = result; - } - } - } - - /* shutdown the voice manager & synthesizer */ - VMShutdown(pEASData); - -#ifdef _METRICS_ENABLED - /* shutdown the metrics module */ - if (pEASData->pMetricsModule != NULL) - { - if ((result = (*pEASData->pMetricsModule->pfShutdown)(pEASData, pEASData->pMetricsData)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %ld shutting down metrics module\n", result); */ } - if (reportResult == EAS_SUCCESS) - reportResult = result; - } - } -#endif - - /* release allocated memory */ - if (!pEASData->staticMemoryModel) - EAS_HWFree(hwInstData, pEASData); - - /* shutdown host wrappers */ - if (hwInstData) - { - if ((result = EAS_HWShutdown(hwInstData)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %ld shutting down host wrappers\n", result); */ } - if (reportResult == EAS_SUCCESS) - reportResult = result; - } - } - - return reportResult; -} - -#ifdef JET_INTERFACE -/*---------------------------------------------------------------------------- - * EAS_OpenJETStream() - *---------------------------------------------------------------------------- - * Private interface for JET to open an SMF stream with an offset - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_OpenJETStream (EAS_DATA_HANDLE pEASData, EAS_FILE_HANDLE fileHandle, EAS_I32 offset, EAS_HANDLE *ppStream) -{ - EAS_RESULT result; - EAS_VOID_PTR streamHandle; - S_FILE_PARSER_INTERFACE *pParserModule; - EAS_INT streamNum; - - /* allocate a stream */ - if ((streamNum = EAS_AllocateStream(pEASData)) < 0) - return EAS_ERROR_MAX_STREAMS_OPEN; - - /* check Configuration Module for SMF parser */ - *ppStream = NULL; - streamHandle = NULL; - pParserModule = (S_FILE_PARSER_INTERFACE *) EAS_CMEnumModules(0); - if (pParserModule == NULL) - return EAS_ERROR_UNRECOGNIZED_FORMAT; - - /* see if SMF parser recognizes the file */ - if ((result = (*pParserModule->pfCheckFileType)(pEASData, fileHandle, &streamHandle, offset)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "CheckFileType returned error %ld\n", result); */ } - return result; - } - - /* parser recognized the file, return the handle */ - if (streamHandle) - { - EAS_InitStream(&pEASData->streams[streamNum], pParserModule, streamHandle); - *ppStream = &pEASData->streams[streamNum]; - return EAS_SUCCESS; - } - - return EAS_ERROR_UNRECOGNIZED_FORMAT; -} -#endif - -/*---------------------------------------------------------------------------- - * EAS_OpenFile() - *---------------------------------------------------------------------------- - * Purpose: - * Opens a file for audio playback. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * pHandle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_OpenFile (EAS_DATA_HANDLE pEASData, EAS_FILE_LOCATOR locator, EAS_HANDLE *ppStream) -{ - EAS_RESULT result; - EAS_FILE_HANDLE fileHandle; - EAS_VOID_PTR streamHandle; - S_FILE_PARSER_INTERFACE *pParserModule; - EAS_INT streamNum; - EAS_INT moduleNum; - - /* open the file */ - if ((result = EAS_HWOpenFile(pEASData->hwInstData, locator, &fileHandle, EAS_FILE_READ)) != EAS_SUCCESS) - return result; - - /* allocate a stream */ - if ((streamNum = EAS_AllocateStream(pEASData)) < 0) - return EAS_ERROR_MAX_STREAMS_OPEN; - - /* check Configuration Module for file parsers */ - pParserModule = NULL; - *ppStream = NULL; - streamHandle = NULL; - for (moduleNum = 0; ; moduleNum++) - { - pParserModule = (S_FILE_PARSER_INTERFACE *) EAS_CMEnumModules(moduleNum); - if (pParserModule == NULL) - break; - - /* see if this parser recognizes it */ - if ((result = (*pParserModule->pfCheckFileType)(pEASData, fileHandle, &streamHandle, 0L)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "CheckFileType returned error %ld\n", result); */ } - return result; - } - - /* parser recognized the file, return the handle */ - if (streamHandle) - { - - /* save the parser pointer and file handle */ - EAS_InitStream(&pEASData->streams[streamNum], pParserModule, streamHandle); - *ppStream = &pEASData->streams[streamNum]; - return EAS_SUCCESS; - } - - /* rewind the file for the next parser */ - if ((result = EAS_HWFileSeek(pEASData->hwInstData, fileHandle, 0L)) != EAS_SUCCESS) - return result; - } - - /* no parser was able to recognize the file, close it and return an error */ - EAS_HWCloseFile(pEASData->hwInstData, fileHandle); - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "No parser recognized the requested file\n"); */ } - return EAS_ERROR_UNRECOGNIZED_FORMAT; -} - -#ifdef MMAPI_SUPPORT -/*---------------------------------------------------------------------------- - * EAS_MMAPIToneControl() - *---------------------------------------------------------------------------- - * Purpose: - * Opens a ToneControl file for audio playback. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * pHandle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_MMAPIToneControl (EAS_DATA_HANDLE pEASData, EAS_FILE_LOCATOR locator, EAS_HANDLE *ppStream) -{ - EAS_RESULT result; - EAS_FILE_HANDLE fileHandle; - EAS_VOID_PTR streamHandle; - S_FILE_PARSER_INTERFACE *pParserModule; - EAS_INT streamNum; - - /* check if the tone control parser is available */ - *ppStream = NULL; - streamHandle = NULL; - pParserModule = EAS_CMEnumOptModules(EAS_MODULE_MMAPI_TONE_CONTROL); - if (pParserModule == NULL) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_MMAPIToneControl: ToneControl parser not available\n"); */ } - return EAS_ERROR_FEATURE_NOT_AVAILABLE; - } - - /* open the file */ - if ((result = EAS_HWOpenFile(pEASData->hwInstData, locator, &fileHandle, EAS_FILE_READ)) != EAS_SUCCESS) - return result; - - /* allocate a stream */ - if ((streamNum = EAS_AllocateStream(pEASData)) < 0) - return EAS_ERROR_MAX_STREAMS_OPEN; - - /* see if ToneControl parser recognizes it */ - if ((result = (*pParserModule->pfCheckFileType)(pEASData, fileHandle, &streamHandle, 0L)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "CheckFileType returned error %ld\n", result); */ } - return result; - } - - /* parser accepted the file, return the handle */ - if (streamHandle) - { - - /* save the parser pointer and file handle */ - EAS_InitStream(&pEASData->streams[streamNum], pParserModule, streamHandle); - *ppStream = &pEASData->streams[streamNum]; - return EAS_SUCCESS; - } - - /* parser did not recognize the file, close it and return an error */ - EAS_HWCloseFile(pEASData->hwInstData, fileHandle); - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "No parser recognized the requested file\n"); */ } - return EAS_ERROR_UNRECOGNIZED_FORMAT; -} - -/*---------------------------------------------------------------------------- - * EAS_GetWaveFmtChunk - *---------------------------------------------------------------------------- - * Helper function to retrieve WAVE file fmt chunk for MMAPI - *---------------------------------------------------------------------------- - * pEASData - pointer to EAS persistent data object - * pStream - stream handle - * pFmtChunk - pointer to variable to receive current setting - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetWaveFmtChunk (S_EAS_DATA *pEASData, EAS_HANDLE pStream, EAS_VOID_PTR *ppFmtChunk) -{ - EAS_RESULT result; - EAS_I32 value; - - if ((result = EAS_GetStreamParameter(pEASData, pStream, PARSER_DATA_FORMAT, &value)) != EAS_SUCCESS) - return result; - *ppFmtChunk = (EAS_VOID_PTR) value; - return EAS_SUCCESS; -} -#endif - -/*---------------------------------------------------------------------------- - * EAS_GetFileType - *---------------------------------------------------------------------------- - * Returns the file type (see eas_types.h for enumerations) - *---------------------------------------------------------------------------- - * pEASData - pointer to EAS persistent data object - * pStream - stream handle - * pFileType - pointer to variable to receive file type - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetFileType (S_EAS_DATA *pEASData, EAS_HANDLE pStream, EAS_I32 *pFileType) -{ - if (!EAS_StreamReady (pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - return EAS_GetStreamParameter(pEASData, pStream, PARSER_DATA_FILE_TYPE, pFileType); -} - -/*---------------------------------------------------------------------------- - * EAS_Prepare() - *---------------------------------------------------------------------------- - * Purpose: - * Prepares the synthesizer to play the file or stream. Parses the first - * frame of data from the file and arms the synthesizer. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - file or stream handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_Prepare (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream) -{ - S_FILE_PARSER_INTERFACE *pParserModule; - EAS_STATE state; - EAS_RESULT result; - - pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; - if (pParserModule == NULL) - return EAS_ERROR_FEATURE_NOT_AVAILABLE; - - /* check for valid state */ - result = pParserModule->pfState(pEASData, pStream->handle, &state); - if (result == EAS_SUCCESS) - { - /* prepare the stream */ - if (state == EAS_STATE_OPEN) - { - pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; - result = (*pParserModule->pfPrepare)(pEASData, pStream->handle); - - /* set volume */ - if (result == EAS_SUCCESS) - result = EAS_SetVolume(pEASData, pStream, pStream->volume); - } - else - result = EAS_ERROR_NOT_VALID_IN_THIS_STATE; - - } - - return result; -} - -/*---------------------------------------------------------------------------- - * EAS_Render() - *---------------------------------------------------------------------------- - * Purpose: - * Parse the Midi data and render PCM audio data. - * - * Inputs: - * pEASData - buffer for internal EAS data - * pOut - output buffer pointer - * nNumRequested - requested num samples to generate - * pnNumGenerated - actual number of samples generated - * - * Outputs: - * EAS_SUCCESS if PCM data was successfully rendered - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_Render (EAS_DATA_HANDLE pEASData, EAS_PCM *pOut, EAS_I32 numRequested, EAS_I32 *pNumGenerated) -{ - S_FILE_PARSER_INTERFACE *pParserModule; - EAS_RESULT result; - EAS_I32 voicesRendered; - EAS_STATE parserState; - EAS_INT streamNum; - - /* assume no samples generated and reset workload */ - *pNumGenerated = 0; - VMInitWorkload(pEASData->pVoiceMgr); - - /* no support for other buffer sizes yet */ - if (numRequested != BUFFER_SIZE_IN_MONO_SAMPLES) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "This library supports only %ld samples in buffer, host requested %ld samples\n", - (EAS_I32) BUFFER_SIZE_IN_MONO_SAMPLES, numRequested); */ } - return EAS_BUFFER_SIZE_MISMATCH; - } - -#ifdef _METRICS_ENABLED - /* start performance counter */ - if (pEASData->pMetricsData) - (*pEASData->pMetricsModule->pfStartTimer)(pEASData->pMetricsData, EAS_PM_TOTAL_TIME); -#endif - - /* prep the frame buffer, do mix engine prep only if TRUE */ -#ifdef _SPLIT_ARCHITECTURE - if (VMStartFrame(pEASData)) - EAS_MixEnginePrep(pEASData, numRequested); -#else - /* prep the mix engine */ - EAS_MixEnginePrep(pEASData, numRequested); -#endif - - /* save the output buffer pointer */ - pEASData->pOutputAudioBuffer = pOut; - - -#ifdef _METRICS_ENABLED - /* start performance counter */ - if (pEASData->pMetricsData) - (*pEASData->pMetricsModule->pfStartTimer)(pEASData->pMetricsData, EAS_PM_PARSE_TIME); -#endif - - /* if we haven't finished parsing from last time, do it now */ - /* need to parse another frame of events before we render again */ - for (streamNum = 0; streamNum < MAX_NUMBER_STREAMS; streamNum++) - { - /* clear the locate flag */ - pEASData->streams[streamNum].streamFlags &= ~STREAM_FLAGS_LOCATE; - - if (pEASData->streams[streamNum].pParserModule) - { - - /* establish pointer to parser module */ - pParserModule = pEASData->streams[streamNum].pParserModule; - - /* handle pause */ - if (pEASData->streams[streamNum].streamFlags & STREAM_FLAGS_PAUSE) - { - if (pParserModule->pfPause) - result = pParserModule->pfPause(pEASData, pEASData->streams[streamNum].handle); - pEASData->streams[streamNum].streamFlags &= ~STREAM_FLAGS_PAUSE; - } - - /* get current state */ - if ((result = (*pParserModule->pfState)(pEASData, pEASData->streams[streamNum].handle, &parserState)) != EAS_SUCCESS) - return result; - - /* handle resume */ - if (parserState == EAS_STATE_PAUSED) - { - if (pEASData->streams[streamNum].streamFlags & STREAM_FLAGS_RESUME) - { - if (pParserModule->pfResume) - result = pParserModule->pfResume(pEASData, pEASData->streams[streamNum].handle); - pEASData->streams[streamNum].streamFlags &= ~STREAM_FLAGS_RESUME; - } - } - - /* if necessary, parse stream */ - if ((pEASData->streams[streamNum].streamFlags & STREAM_FLAGS_PARSED) == 0) - if ((result = EAS_ParseEvents(pEASData, &pEASData->streams[streamNum], pEASData->streams[streamNum].time + pEASData->streams[streamNum].frameLength, eParserModePlay)) != EAS_SUCCESS) - return result; - - /* check for an early abort */ - if ((pEASData->streams[streamNum].streamFlags) == 0) - { - -#ifdef _METRICS_ENABLED - /* stop performance counter */ - if (pEASData->pMetricsData) - (*pEASData->pMetricsModule->pfStartTimer)(pEASData->pMetricsData, EAS_PM_TOTAL_TIME); -#endif - - return EAS_SUCCESS; - } - - /* check for repeat */ - if (pEASData->streams[streamNum].repeatCount) - { - - /* check for stopped state */ - if ((result = (*pParserModule->pfState)(pEASData, pEASData->streams[streamNum].handle, &parserState)) != EAS_SUCCESS) - return result; - if (parserState == EAS_STATE_STOPPED) - { - - /* decrement repeat count, unless it is negative */ - if (pEASData->streams[streamNum].repeatCount > 0) - pEASData->streams[streamNum].repeatCount--; - - /* reset the parser */ - if ((result = (*pParserModule->pfReset)(pEASData, pEASData->streams[streamNum].handle)) != EAS_SUCCESS) - return result; - pEASData->streams[streamNum].time = 0; - } - } - } - } - -#ifdef _METRICS_ENABLED - /* stop performance counter */ - if (pEASData->pMetricsData) - (void)(*pEASData->pMetricsModule->pfStopTimer)(pEASData->pMetricsData, EAS_PM_PARSE_TIME); -#endif - -#ifdef _METRICS_ENABLED - /* start the render timer */ - if (pEASData->pMetricsData) - (*pEASData->pMetricsModule->pfStartTimer)(pEASData->pMetricsData, EAS_PM_RENDER_TIME); -#endif - - /* render audio */ - if ((result = VMRender(pEASData->pVoiceMgr, BUFFER_SIZE_IN_MONO_SAMPLES, pEASData->pMixBuffer, &voicesRendered)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "pfRender function returned error %ld\n", result); */ } - return result; - } - -#ifdef _METRICS_ENABLED - /* stop the render timer */ - if (pEASData->pMetricsData) { - (*pEASData->pMetricsModule->pfIncrementCounter)(pEASData->pMetricsData, EAS_PM_FRAME_COUNT, 1); - (void)(*pEASData->pMetricsModule->pfStopTimer)(pEASData->pMetricsData, EAS_PM_RENDER_TIME); - (*pEASData->pMetricsModule->pfIncrementCounter)(pEASData->pMetricsData, EAS_PM_TOTAL_VOICE_COUNT, (EAS_U32) voicesRendered); - (void)(*pEASData->pMetricsModule->pfRecordMaxValue)(pEASData->pMetricsData, EAS_PM_MAX_VOICES, (EAS_U32) voicesRendered); - } -#endif - - //2 Do we really need frameParsed? - /* need to parse another frame of events before we render again */ - for (streamNum = 0; streamNum < MAX_NUMBER_STREAMS; streamNum++) - if (pEASData->streams[streamNum].pParserModule != NULL) - pEASData->streams[streamNum].streamFlags &= ~STREAM_FLAGS_PARSED; - -#ifdef _METRICS_ENABLED - /* start performance counter */ - if (pEASData->pMetricsData) - (*pEASData->pMetricsModule->pfStartTimer)(pEASData->pMetricsData, EAS_PM_STREAM_TIME); -#endif - - /* render PCM audio */ - if ((result = EAS_PERender(pEASData, numRequested)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_PERender returned error %ld\n", result); */ } - return result; - } - -#ifdef _METRICS_ENABLED - /* stop the stream timer */ - if (pEASData->pMetricsData) - (void)(*pEASData->pMetricsModule->pfStopTimer)(pEASData->pMetricsData, EAS_PM_STREAM_TIME); -#endif - -#ifdef _METRICS_ENABLED - /* start the post timer */ - if (pEASData->pMetricsData) - (*pEASData->pMetricsModule->pfStartTimer)(pEASData->pMetricsData, EAS_PM_POST_TIME); -#endif - - /* for split architecture, send DSP vectors. Do post only if return is TRUE */ -#ifdef _SPLIT_ARCHITECTURE - if (VMEndFrame(pEASData)) - { - /* now do post-processing */ - EAS_MixEnginePost(pEASData, numRequested); - *pNumGenerated = numRequested; - } -#else - /* now do post-processing */ - EAS_MixEnginePost(pEASData, numRequested); - *pNumGenerated = numRequested; -#endif - -#ifdef _METRICS_ENABLED - /* stop the post timer */ - if (pEASData->pMetricsData) - (void)(*pEASData->pMetricsModule->pfStopTimer)(pEASData->pMetricsData, EAS_PM_POST_TIME); -#endif - - /* advance render time */ - pEASData->renderTime += AUDIO_FRAME_LENGTH; - -#if 0 - /* dump workload for debug */ - if (pEASData->pVoiceMgr->workload) - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Workload = %d\n", pEASData->pVoiceMgr->workload); */ } -#endif - -#ifdef _METRICS_ENABLED - /* stop performance counter */ - if (pEASData->pMetricsData) - { - PERF_TIMER temp; - temp = (*pEASData->pMetricsModule->pfStopTimer)(pEASData->pMetricsData, EAS_PM_TOTAL_TIME); - - /* if max render time, record the number of voices and time */ - if ((*pEASData->pMetricsModule->pfRecordMaxValue) - (pEASData->pMetricsData, EAS_PM_MAX_CYCLES, (EAS_U32) temp)) - { - (*pEASData->pMetricsModule->pfRecordValue)(pEASData->pMetricsData, EAS_PM_MAX_CYCLES_VOICES, (EAS_U32) voicesRendered); - (*pEASData->pMetricsModule->pfRecordValue)(pEASData->pMetricsData, EAS_PM_MAX_CYCLES_TIME, (EAS_I32) (pEASData->renderTime >> 8)); - } - } -#endif - -#ifdef JET_INTERFACE - /* let JET to do its thing */ - if (pEASData->jetHandle != NULL) - { - result = JET_Process(pEASData); - if (result != EAS_SUCCESS) - return result; - } -#endif - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_SetRepeat() - *---------------------------------------------------------------------------- - * Purpose: - * Set the selected stream to repeat. - * - * Inputs: - * pEASData - handle to data for this instance - * handle - handle to stream - * repeatCount - repeat count - * - * Outputs: - * - * Side Effects: - * - * Notes: - * 0 = no repeat - * 1 = repeat once, i.e. play through twice - * -1 = repeat forever - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_PUBLIC EAS_RESULT EAS_SetRepeat (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 repeatCount) -{ - pStream->repeatCount = repeatCount; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_GetRepeat() - *---------------------------------------------------------------------------- - * Purpose: - * Gets the current repeat count for the selected stream. - * - * Inputs: - * pEASData - handle to data for this instance - * handle - handle to stream - * pRrepeatCount - pointer to variable to hold repeat count - * - * Outputs: - * - * Side Effects: - * - * Notes: - * 0 = no repeat - * 1 = repeat once, i.e. play through twice - * -1 = repeat forever - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_PUBLIC EAS_RESULT EAS_GetRepeat (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 *pRepeatCount) -{ - *pRepeatCount = pStream->repeatCount; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_SetPlaybackRate() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the playback rate. - * - * Inputs: - * pEASData - handle to data for this instance - * handle - handle to stream - * rate - rate (28-bit fractional amount) - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_PUBLIC EAS_RESULT EAS_SetPlaybackRate (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_U32 rate) -{ - - /* check range */ - if ((rate < (1 << 27)) || (rate > (1 << 29))) - return EAS_ERROR_INVALID_PARAMETER; - - /* calculate new frame length - * - * NOTE: The maximum frame length we can accomodate based on a - * maximum rate of 2.0 (2^28) is 2047 (2^13-1). To accomodate a - * longer frame length or a higher maximum rate, the fixed point - * divide below will need to be adjusted - */ - pStream->frameLength = (AUDIO_FRAME_LENGTH * (rate >> 8)) >> 20; - - /* notify stream of new playback rate */ - EAS_SetStreamParameter(pEASData, pStream, PARSER_DATA_PLAYBACK_RATE, (EAS_I32) rate); - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_SetTransposition) - *---------------------------------------------------------------------------- - * Purpose: - * Sets the key tranposition for the synthesizer. Transposes all - * melodic instruments by the specified amount. Range is limited - * to +/-12 semitones. - * - * Inputs: - * pEASData - handle to data for this instance - * handle - handle to stream - * transposition - +/-12 semitones - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetTransposition (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 transposition) -{ - - /* check range */ - if ((transposition < -12) || (transposition > 12)) - return EAS_ERROR_INVALID_PARAMETER; - - if (!EAS_StreamReady(pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_TRANSPOSITION, transposition); -} - -/*---------------------------------------------------------------------------- - * EAS_ParseEvents() - *---------------------------------------------------------------------------- - * Purpose: - * Parse events in the current streams until the desired time is reached. - * - * Inputs: - * pEASData - buffer for internal EAS data - * endTime - stop parsing if this time is reached - * parseMode - play, locate, or metadata - * - * Outputs: - * EAS_SUCCESS if PCM data was successfully rendered - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT EAS_ParseEvents (S_EAS_DATA *pEASData, EAS_HANDLE pStream, EAS_U32 endTime, EAS_INT parseMode) -{ - S_FILE_PARSER_INTERFACE *pParserModule; - EAS_RESULT result; - EAS_I32 parserState; - EAS_BOOL done; - EAS_INT yieldCount = YIELD_EVENT_COUNT; - EAS_U32 time = 0; - - /* does this parser have a time function? */ - pParserModule = pStream->pParserModule; - if (pParserModule->pfTime == NULL) - { - /* check state */ - if ((result = (*pParserModule->pfState)(pEASData, pStream->handle, &parserState)) != EAS_SUCCESS) - return result; - /* if play state, advance time */ - if ((parserState >= EAS_STATE_READY) && (parserState <= EAS_STATE_PAUSING)) - pStream->time += pStream->frameLength; - done = EAS_TRUE; - } - - /* assume we're not done, in case we abort out */ - else - { - pStream->streamFlags &= ~STREAM_FLAGS_PARSED; - done = EAS_FALSE; - } - - while (!done) - { - - /* check for stopped state */ - if ((result = (*pParserModule->pfState)(pEASData, pStream->handle, &parserState)) != EAS_SUCCESS) - return result; - if (parserState > EAS_STATE_PLAY) - { - /* save current time if we're not in play mode */ - if (parseMode != eParserModePlay) - pStream->time = time << 8; - done = EAS_TRUE; - break; - } - - /* get the next event time */ - if (pParserModule->pfTime) - { - if ((result = (*pParserModule->pfTime)(pEASData, pStream->handle, &time)) != EAS_SUCCESS) - return result; - - /* if next event is within this frame, parse it */ - if (time < (endTime >> 8)) - { - - /* parse the next event */ - if (pParserModule->pfEvent) - if ((result = (*pParserModule->pfEvent)(pEASData, pStream->handle, parseMode)) != EAS_SUCCESS) - return result; - } - - /* no more events in this frame, advance time */ - else - { - pStream->time = endTime; - done = EAS_TRUE; - } - } - - /* check for max workload exceeded */ - if (VMCheckWorkload(pEASData->pVoiceMgr)) - { - /* stop even though we may not have parsed - * all the events in this frame. The parser will try to - * catch up on the next frame. - */ - break; - } - - /* give host a chance for an early abort */ - if (--yieldCount == 0) - { - if (EAS_HWYield(pEASData->hwInstData)) - break; - yieldCount = YIELD_EVENT_COUNT; - } - } - - /* if no early abort, parsing is complete for this frame */ - if (done) - pStream->streamFlags |= STREAM_FLAGS_PARSED; - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_ParseMetaData() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - file or stream handle - * playLength - pointer to variable to store the play length (in msecs) - * - * Outputs: - * - * - * Side Effects: - * - resets the parser to the start of the file - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_ParseMetaData (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 *playLength) -{ - S_FILE_PARSER_INTERFACE *pParserModule; - EAS_RESULT result; - EAS_STATE state; - - pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; - if (pParserModule == NULL) - return EAS_ERROR_FEATURE_NOT_AVAILABLE; - - /* check parser state */ - if ((result = (*pParserModule->pfState)(pEASData, pStream->handle, &state)) != EAS_SUCCESS) - return result; - if (state >= EAS_STATE_OPEN) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - - /* if parser has metadata function, use that */ - if (pParserModule->pfGetMetaData != NULL) - return pParserModule->pfGetMetaData(pEASData, pStream->handle, playLength); - - /* reset the parser to the beginning */ - if ((result = (*pParserModule->pfReset)(pEASData, pStream->handle)) != EAS_SUCCESS) - return result; - - /* parse the file to end */ - pStream->time = 0; - VMInitWorkload(pEASData->pVoiceMgr); - if ((result = EAS_ParseEvents(pEASData, pStream, 0x7fffffff, eParserModeMetaData)) != EAS_SUCCESS) - return result; - - /* get the parser time */ - if ((result = EAS_GetLocation(pEASData, pStream, playLength)) != EAS_SUCCESS) - return result; - - /* reset the parser to the beginning */ - pStream->time = 0; - return (*pParserModule->pfReset)(pEASData, pStream->handle); -} - -/*---------------------------------------------------------------------------- - * EAS_RegisterMetaDataCallback() - *---------------------------------------------------------------------------- - * Purpose: - * Registers a metadata callback function for parsed metadata. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - file or stream handle - * cbFunc - pointer to host callback function - * metaDataBuffer - pointer to metadata buffer - * metaDataBufSize - maximum size of the metadata buffer - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_RegisterMetaDataCallback ( - EAS_DATA_HANDLE pEASData, - EAS_HANDLE pStream, - EAS_METADATA_CBFUNC cbFunc, - char *metaDataBuffer, - EAS_I32 metaDataBufSize, - EAS_VOID_PTR pUserData) -{ - S_METADATA_CB metadata; - - if (!EAS_StreamReady(pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - - /* register callback function */ - metadata.callback = cbFunc; - metadata.buffer = metaDataBuffer; - metadata.bufferSize = metaDataBufSize; - metadata.pUserData = pUserData; - return EAS_SetStreamParameter(pEASData, pStream, PARSER_DATA_METADATA_CB, (EAS_I32) &metadata); -} - -/*---------------------------------------------------------------------------- - * EAS_GetNoteCount () - *---------------------------------------------------------------------------- - * Returns the total number of notes played in this stream - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetNoteCount (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 *pNoteCount) -{ - if (!EAS_StreamReady(pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - return EAS_IntGetStrmParam(pEASData, pStream, PARSER_DATA_NOTE_COUNT, pNoteCount); -} - -/*---------------------------------------------------------------------------- - * EAS_CloseFile() - *---------------------------------------------------------------------------- - * Purpose: - * Closes an audio file or stream. Playback should have either paused or - * completed (EAS_State returns EAS_PAUSED or EAS_STOPPED). - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - file or stream handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_CloseFile (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream) -{ - S_FILE_PARSER_INTERFACE *pParserModule; - EAS_RESULT result; - - /* call the close function */ - pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; - if (pParserModule == NULL) - return EAS_ERROR_FEATURE_NOT_AVAILABLE; - - result = (*pParserModule->pfClose)(pEASData, pStream->handle); - - /* clear the handle and parser interface pointer */ - pStream->handle = NULL; - pStream->pParserModule = NULL; - return result; -} - -/*---------------------------------------------------------------------------- - * EAS_OpenMIDIStream() - *---------------------------------------------------------------------------- - * Purpose: - * Opens a raw MIDI stream allowing the host to route MIDI cable data directly to the synthesizer - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * pHandle - pointer to variable to hold file or stream handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_OpenMIDIStream (EAS_DATA_HANDLE pEASData, EAS_HANDLE *ppStream, EAS_HANDLE streamHandle) -{ - EAS_RESULT result; - S_INTERACTIVE_MIDI *pMIDIStream; - EAS_INT streamNum; - - /* initialize some pointers */ - *ppStream = NULL; - - /* allocate a stream */ - if ((streamNum = EAS_AllocateStream(pEASData)) < 0) - return EAS_ERROR_MAX_STREAMS_OPEN; - - /* check Configuration Module for S_EAS_DATA allocation */ - if (pEASData->staticMemoryModel) - pMIDIStream = EAS_CMEnumData(EAS_CM_MIDI_STREAM_DATA); - else - pMIDIStream = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_INTERACTIVE_MIDI)); - - /* allocate dynamic memory */ - if (!pMIDIStream) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Failed to allocate MIDI stream data\n"); */ } - return EAS_ERROR_MALLOC_FAILED; - } - - /* zero the memory to insure complete initialization */ - EAS_HWMemSet(pMIDIStream, 0, sizeof(S_INTERACTIVE_MIDI)); - EAS_InitStream(&pEASData->streams[streamNum], NULL, pMIDIStream); - - /* instantiate a new synthesizer */ - if (streamHandle == NULL) - { - result = VMInitMIDI(pEASData, &pMIDIStream->pSynth); - } - - /* use an existing synthesizer */ - else - { - EAS_I32 value; - result = EAS_GetStreamParameter(pEASData, streamHandle, PARSER_DATA_SYNTH_HANDLE, &value); - pMIDIStream->pSynth = (S_SYNTH*) value; - VMIncRefCount(pMIDIStream->pSynth); - } - if (result != EAS_SUCCESS) - { - EAS_CloseMIDIStream(pEASData, &pEASData->streams[streamNum]); - return result; - } - - /* initialize the MIDI stream data */ - EAS_InitMIDIStream(&pMIDIStream->stream); - - *ppStream = (EAS_HANDLE) &pEASData->streams[streamNum]; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_WriteMIDIStream() - *---------------------------------------------------------------------------- - * Purpose: - * Send data to the MIDI stream device - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - stream handle - * pBuffer - pointer to buffer - * count - number of bytes to write - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_WriteMIDIStream (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_U8 *pBuffer, EAS_I32 count) -{ - S_INTERACTIVE_MIDI *pMIDIStream; - EAS_RESULT result; - - pMIDIStream = (S_INTERACTIVE_MIDI*) pStream->handle; - - /* send the entire buffer */ - while (count--) - { - if ((result = EAS_ParseMIDIStream(pEASData, pMIDIStream->pSynth, &pMIDIStream->stream, *pBuffer++, eParserModePlay)) != EAS_SUCCESS) - return result; - } - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_CloseMIDIStream() - *---------------------------------------------------------------------------- - * Purpose: - * Closes a raw MIDI stream - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - stream handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_CloseMIDIStream (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream) -{ - S_INTERACTIVE_MIDI *pMIDIStream; - - pMIDIStream = (S_INTERACTIVE_MIDI*) pStream->handle; - - /* close synth */ - if (pMIDIStream->pSynth != NULL) - { - VMMIDIShutdown(pEASData, pMIDIStream->pSynth); - pMIDIStream->pSynth = NULL; - } - - /* release allocated memory */ - if (!pEASData->staticMemoryModel) - EAS_HWFree(((S_EAS_DATA*) pEASData)->hwInstData, pMIDIStream); - - pStream->handle = NULL; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_State() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the state of an audio file or stream. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - file or stream handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_State (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_STATE *pState) -{ - S_FILE_PARSER_INTERFACE *pParserModule; - EAS_RESULT result; - - /* call the parser to return state */ - pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; - if (pParserModule == NULL) - return EAS_ERROR_FEATURE_NOT_AVAILABLE; - - if ((result = (*pParserModule->pfState)(pEASData, pStream->handle, pState)) != EAS_SUCCESS) - return result; - - /* if repeat count is set for this parser, mask the stopped state from the application */ - if (pStream->repeatCount && (*pState == EAS_STATE_STOPPED)) - *pState = EAS_STATE_PLAY; - - /* if we're not ready or playing, we don't need to hide state from host */ - if (*pState > EAS_STATE_PLAY) - return EAS_SUCCESS; - - /* if stream is about to be paused, report it as paused */ - if (pStream->streamFlags & STREAM_FLAGS_PAUSE) - { - if (pStream->streamFlags & STREAM_FLAGS_LOCATE) - *pState = EAS_STATE_PAUSED; - else - *pState = EAS_STATE_PAUSING; - } - - /* if stream is about to resume, report it as playing */ - if (pStream->streamFlags & STREAM_FLAGS_RESUME) - *pState = EAS_STATE_PLAY; - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_SetPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Set the polyphony of the stream. A value of 0 allows the stream - * to use all voices (set by EAS_SetSynthPolyphony). - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - handle returned by EAS_OpenFile - * polyphonyCount - the desired polyphony count - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetPolyphony (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 polyphonyCount) -{ - if (!EAS_StreamReady(pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_POLYPHONY, polyphonyCount); -} - -/*---------------------------------------------------------------------------- - * EAS_GetPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current polyphony setting of the stream - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - handle returned by EAS_OpenFile - * pPolyphonyCount - pointer to variable to receive polyphony count - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetPolyphony (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 *pPolyphonyCount) -{ - if (!EAS_StreamReady(pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - return EAS_IntGetStrmParam(pEASData, pStream, PARSER_DATA_POLYPHONY, pPolyphonyCount); -} - -/*---------------------------------------------------------------------------- - * EAS_SetSynthPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Set the polyphony of the synth . Value must be >= 1 and <= the - * maximum number of voices. This function will pin the polyphony - * at those limits - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * synthNum - synthesizer number (0 = onboard, 1 = DSP) - * polyphonyCount - the desired polyphony count - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetSynthPolyphony (EAS_DATA_HANDLE pEASData, EAS_I32 synthNum, EAS_I32 polyphonyCount) -{ - return VMSetSynthPolyphony(pEASData->pVoiceMgr, synthNum, polyphonyCount); -} - -/*---------------------------------------------------------------------------- - * EAS_GetSynthPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current polyphony setting of the synth - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * synthNum - synthesizer number (0 = onboard, 1 = DSP) - * pPolyphonyCount - pointer to variable to receive polyphony count - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetSynthPolyphony (EAS_DATA_HANDLE pEASData, EAS_I32 synthNum, EAS_I32 *pPolyphonyCount) -{ - return VMGetSynthPolyphony(pEASData->pVoiceMgr, synthNum, pPolyphonyCount); -} - -/*---------------------------------------------------------------------------- - * EAS_SetPriority() - *---------------------------------------------------------------------------- - * Purpose: - * Set the priority of the stream. Determines which stream's voices - * are stolen when there are insufficient voices for all notes. - * Value must be in the range of 1-15, lower values are higher - * priority. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - handle returned by EAS_OpenFile - * polyphonyCount - the desired polyphony count - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetPriority (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 priority) -{ - if (!EAS_StreamReady(pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_PRIORITY, priority); -} - -/*---------------------------------------------------------------------------- - * EAS_GetPriority() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current priority setting of the stream - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - handle returned by EAS_OpenFile - * pPriority - pointer to variable to receive priority - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetPriority (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 *pPriority) -{ - if (!EAS_StreamReady(pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - return EAS_IntGetStrmParam(pEASData, pStream, PARSER_DATA_PRIORITY, pPriority); -} - -/*---------------------------------------------------------------------------- - * EAS_SetVolume() - *---------------------------------------------------------------------------- - * Purpose: - * Set the master gain for the mix engine in 1dB increments - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * volume - the desired master gain (100 is max) - * handle - file or stream handle - * - * Outputs: - * - * - * Side Effects: - * overrides any previously set master volume from sysex - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetVolume (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 volume) -{ - EAS_I16 gain; - - /* check range */ - if ((volume < 0) || (volume > EAS_MAX_VOLUME)) - return EAS_ERROR_PARAMETER_RANGE; - - /* stream volume */ - if (pStream != NULL) - { - EAS_I32 gainOffset; - EAS_RESULT result; - - if (!EAS_StreamReady(pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - - /* get gain offset */ - pStream->volume = (EAS_U8) volume; - result = EAS_GetStreamParameter(pEASData, pStream, PARSER_DATA_GAIN_OFFSET, &gainOffset); - if (result == EAS_SUCCESS) - volume += gainOffset; - - /* set stream volume */ - gain = EAS_VolumeToGain(volume - STREAM_VOLUME_HEADROOM); - - /* convert to linear scalar */ - return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_VOLUME, gain); - } - - /* master volume */ - pEASData->masterVolume = (EAS_U8) volume; -#if (NUM_OUTPUT_CHANNELS == 1) - /* leave 3dB headroom for mono output */ - volume -= 3; -#endif - - gain = EAS_VolumeToGain(volume - STREAM_VOLUME_HEADROOM); - pEASData->masterGain = gain; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_GetVolume() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the master volume for the synthesizer. The default volume setting is - * 50. The volume range is 0 to 100; - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * volume - the desired master volume - * handle - file or stream handle - * - * Outputs: - * - * - * Side Effects: - * overrides any previously set master volume from sysex - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_I32 EAS_GetVolume (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream) -{ - if (pStream == NULL) - return pEASData->masterVolume; - - if (!EAS_StreamReady(pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - return pStream->volume; -} - -/*---------------------------------------------------------------------------- - * EAS_SetMaxLoad() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the maximum workload the parsers will do in a single call to - * EAS_Render. The units are currently arbitrary, but should correlate - * well to the actual CPU cycles consumed. The primary effect is to - * reduce the occasional peaks in CPU cycles consumed when parsing - * dense parts of a MIDI score. - * - * Inputs: - * pEASData - handle to data for this instance - * maxLoad - the desired maximum workload - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetMaxLoad (EAS_DATA_HANDLE pEASData, EAS_I32 maxLoad) -{ - VMSetWorkload(pEASData->pVoiceMgr, maxLoad); - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_SetMaxPCMStreams() - *---------------------------------------------------------------------------- - * Sets the maximum number of PCM streams allowed in parsers that - * use PCM streaming. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * streamHandle - handle returned by EAS_OpenFile - * maxNumStreams - maximum number of PCM streams - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetMaxPCMStreams (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 maxNumStreams) -{ - if (!EAS_StreamReady(pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_MAX_PCM_STREAMS, maxNumStreams); -} - -/*---------------------------------------------------------------------------- - * EAS_Locate() - *---------------------------------------------------------------------------- - * Purpose: - * Locate into the file associated with the handle. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - file handle - * milliseconds - playback offset from start of file in milliseconds - * - * Outputs: - * - * - * Side Effects: - * the actual offset will be quantized to the closest update period, typically - * a resolution of 5.9ms. Notes that are started prior to this time will not - * sound. Any notes currently playing will be shut off. - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_Locate (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 milliseconds, EAS_BOOL offset) -{ - S_FILE_PARSER_INTERFACE *pParserModule; - EAS_RESULT result; - EAS_U32 requestedTime; - EAS_STATE state; - - /* get pointer to parser function table */ - pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; - if (pParserModule == NULL) - return EAS_ERROR_FEATURE_NOT_AVAILABLE; - - if ((result = (*pParserModule->pfState)(pEASData, pStream->handle, &state)) != EAS_SUCCESS) - return result; - if (state >= EAS_STATE_OPEN) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - - /* handle offset and limit to start of file */ - /*lint -e{704} use shift for performance*/ - if (offset) - milliseconds += (EAS_I32) pStream->time >> 8; - if (milliseconds < 0) - milliseconds = 0; - - /* check to see if the request is different from the current time */ - requestedTime = (EAS_U32) milliseconds; - if (requestedTime == (pStream->time >> 8)) - return EAS_SUCCESS; - - /* set the locate flag */ - pStream->streamFlags |= STREAM_FLAGS_LOCATE; - - /* use the parser locate function, if available */ - if (pParserModule->pfLocate != NULL) - { - EAS_BOOL parserLocate = EAS_FALSE; - result = pParserModule->pfLocate(pEASData, pStream->handle, (EAS_I32) requestedTime, &parserLocate); - if (!parserLocate) - { - if (result == EAS_SUCCESS) - pStream->time = requestedTime << 8; - return result; - } - } - - /* if we were paused and not going to resume, set pause request flag */ - if (((state == EAS_STATE_PAUSING) || (state == EAS_STATE_PAUSED)) && ((pStream->streamFlags & STREAM_FLAGS_RESUME) == 0)) - pStream->streamFlags |= STREAM_FLAGS_PAUSE; - - /* reset the synth and parser */ - if ((result = (*pParserModule->pfReset)(pEASData, pStream->handle)) != EAS_SUCCESS) - return result; - pStream->time = 0; - - /* locating forward, clear parsed flag and parse data until we get to the requested location */ - if ((result = EAS_ParseEvents(pEASData, pStream, requestedTime << 8, eParserModeLocate)) != EAS_SUCCESS) - return result; - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_GetLocation() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current playback offset - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - file handle - * - * Outputs: - * The offset in milliseconds from the start of the current sequence, quantized - * to the nearest update period. Actual resolution is typically 5.9 ms. - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_PUBLIC EAS_RESULT EAS_GetLocation (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 *pTime) -{ - if (!EAS_StreamReady(pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - - *pTime = pStream->time >> 8; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_GetRenderTime() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current playback offset - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * - * Outputs: - * Gets the render time clock in msecs. - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetRenderTime (EAS_DATA_HANDLE pEASData, EAS_I32 *pTime) -{ - *pTime = pEASData->renderTime >> 8; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_Pause() - *---------------------------------------------------------------------------- - * Purpose: - * Pauses the playback of the data associated with this handle. The audio - * is gracefully ramped down to prevent clicks and pops. It may take several - * buffers of audio before the audio is muted. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * handle - file or stream handle - * - * Outputs: - * - * - * Side Effects: - * - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_Pause (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream) -{ - S_FILE_PARSER_INTERFACE *pParserModule; - EAS_STATE state; - EAS_RESULT result; - - pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; - if (pParserModule == NULL) - return EAS_ERROR_FEATURE_NOT_AVAILABLE; - - /* check for valid state */ - result = pParserModule->pfState(pEASData, pStream->handle, &state); - if (result == EAS_SUCCESS) - { - if ((state != EAS_STATE_PLAY) && (state != EAS_STATE_READY) && ((pStream->streamFlags & STREAM_FLAGS_RESUME) == 0)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - - /* make sure parser implements pause */ - if (pParserModule->pfPause == NULL) - result = EAS_ERROR_NOT_IMPLEMENTED; - - /* clear resume flag */ - pStream->streamFlags &= ~STREAM_FLAGS_RESUME; - - /* set pause flag */ - pStream->streamFlags |= STREAM_FLAGS_PAUSE; - -#if 0 - /* pause the stream */ - if (pParserModule->pfPause) - result = pParserModule->pfPause(pEASData, pStream->handle); - else - result = EAS_ERROR_NOT_IMPLEMENTED; -#endif - } - - return result; -} - -/*---------------------------------------------------------------------------- - * EAS_Resume() - *---------------------------------------------------------------------------- - * Purpose: - * Resumes the playback of the data associated with this handle. The audio - * is gracefully ramped up to prevent clicks and pops. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * handle - file or stream handle - * - * Outputs: - * - * - * Side Effects: - * - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_Resume (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream) -{ - S_FILE_PARSER_INTERFACE *pParserModule; - EAS_STATE state; - EAS_RESULT result; - - pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; - if (pParserModule == NULL) - return EAS_ERROR_FEATURE_NOT_AVAILABLE; - - /* check for valid state */ - result = pParserModule->pfState(pEASData, pStream->handle, &state); - if (result == EAS_SUCCESS) - { - if ((state != EAS_STATE_PAUSED) && (state != EAS_STATE_PAUSING) && ((pStream->streamFlags & STREAM_FLAGS_PAUSE) == 0)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - - /* make sure parser implements this function */ - if (pParserModule->pfResume == NULL) - result = EAS_ERROR_NOT_IMPLEMENTED; - - /* clear pause flag */ - pStream->streamFlags &= ~STREAM_FLAGS_PAUSE; - - /* set resume flag */ - pStream->streamFlags |= STREAM_FLAGS_RESUME; - -#if 0 - /* resume the stream */ - if (pParserModule->pfResume) - result = pParserModule->pfResume(pEASData, pStream->handle); - else - result = EAS_ERROR_NOT_IMPLEMENTED; -#endif - } - - return result; -} - -/*---------------------------------------------------------------------------- - * EAS_GetParameter() - *---------------------------------------------------------------------------- - * Purpose: - * Set the parameter of a module. See E_MODULES for a list of modules - * and the header files of the modules for a list of parameters. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * handle - file or stream handle - * module - enumerated module number - * param - enumerated parameter number - * pValue - pointer to variable to receive parameter value - * - * Outputs: - * - * - * Side Effects: - * - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetParameter (EAS_DATA_HANDLE pEASData, EAS_I32 module, EAS_I32 param, EAS_I32 *pValue) -{ - - if (module >= NUM_EFFECTS_MODULES) - return EAS_ERROR_INVALID_MODULE; - - if (pEASData->effectsModules[module].effectData == NULL) - return EAS_ERROR_INVALID_MODULE; - - return (*pEASData->effectsModules[module].effect->pFGetParam) - (pEASData->effectsModules[module].effectData, param, pValue); -} - -/*---------------------------------------------------------------------------- - * EAS_SetParameter() - *---------------------------------------------------------------------------- - * Purpose: - * Set the parameter of a module. See E_MODULES for a list of modules - * and the header files of the modules for a list of parameters. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * handle - file or stream handle - * module - enumerated module number - * param - enumerated parameter number - * value - new parameter value - * - * Outputs: - * - * - * Side Effects: - * - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetParameter (EAS_DATA_HANDLE pEASData, EAS_I32 module, EAS_I32 param, EAS_I32 value) -{ - - if (module >= NUM_EFFECTS_MODULES) - return EAS_ERROR_INVALID_MODULE; - - if (pEASData->effectsModules[module].effectData == NULL) - return EAS_ERROR_INVALID_MODULE; - - return (*pEASData->effectsModules[module].effect->pFSetParam) - (pEASData->effectsModules[module].effectData, param, value); -} - -#ifdef _METRICS_ENABLED -/*---------------------------------------------------------------------------- - * EAS_MetricsReport() - *---------------------------------------------------------------------------- - * Purpose: - * Displays the current metrics through the metrics interface. - * - * Inputs: - * p - instance data handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_MetricsReport (EAS_DATA_HANDLE pEASData) -{ - if (!pEASData->pMetricsModule) - return EAS_ERROR_INVALID_MODULE; - - return (*pEASData->pMetricsModule->pfReport)(pEASData->pMetricsData); -} - -/*---------------------------------------------------------------------------- - * EAS_MetricsReset() - *---------------------------------------------------------------------------- - * Purpose: - * Resets the metrics. - * - * Inputs: - * p - instance data handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_MetricsReset (EAS_DATA_HANDLE pEASData) -{ - - if (!pEASData->pMetricsModule) - return EAS_ERROR_INVALID_MODULE; - - return (*pEASData->pMetricsModule->pfReset)(pEASData->pMetricsData); -} -#endif - -/*---------------------------------------------------------------------------- - * EAS_SetSoundLibrary() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the location of the sound library. - * - * Inputs: - * pEASData - instance data handle - * pSoundLib - pointer to sound library - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetSoundLibrary (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_SNDLIB_HANDLE pSndLib) -{ - if (pStream) - { - if (!EAS_StreamReady(pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_EAS_LIBRARY, (EAS_I32) pSndLib); - } - - return VMSetGlobalEASLib(pEASData->pVoiceMgr, pSndLib); -} - -/*---------------------------------------------------------------------------- - * EAS_SetHeaderSearchFlag() - *---------------------------------------------------------------------------- - * By default, when EAS_OpenFile is called, the parsers check the - * first few bytes of the file looking for a specific header. Some - * mobile devices may add a header to the start of a file, which - * will prevent the parser from recognizing the file. If the - * searchFlag is set to EAS_TRUE, the parser will search the entire - * file looking for the header. This may enable EAS to recognize - * some files that it would ordinarily reject. The negative is that - * it make take slightly longer to process the EAS_OpenFile request. - * - * Inputs: - * pEASData - instance data handle - * searchFlag - search flag (EAS_TRUE or EAS_FALSE) - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetHeaderSearchFlag (EAS_DATA_HANDLE pEASData, EAS_BOOL searchFlag) -{ - pEASData->searchHeaderFlag = (EAS_BOOL8) searchFlag; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_SetPlayMode() - *---------------------------------------------------------------------------- - * Some file formats support special play modes, such as iMode partial - * play mode. This call can be used to change the play mode. The - * default play mode (usually straight playback) is always zero. - * - * Inputs: - * pEASData - instance data handle - * handle - file or stream handle - * playMode - play mode (see file parser for specifics) - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetPlayMode (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 playMode) -{ - return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_PLAY_MODE, playMode); -} - -#ifdef DLS_SYNTHESIZER -/*---------------------------------------------------------------------------- - * EAS_LoadDLSCollection() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the location of the sound library. - * - * Inputs: - * pEASData - instance data handle - * pSoundLib - pointer to sound library - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_LoadDLSCollection (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_FILE_LOCATOR locator) -{ - EAS_FILE_HANDLE fileHandle; - EAS_RESULT result; - EAS_DLSLIB_HANDLE pDLS; - - if (pStream != NULL) - { - if (!EAS_StreamReady(pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - } - - /* open the file */ - if ((result = EAS_HWOpenFile(pEASData->hwInstData, locator, &fileHandle, EAS_FILE_READ)) != EAS_SUCCESS) - return result; - - /* parse the file */ - result = DLSParser(pEASData->hwInstData, fileHandle, 0, &pDLS); - EAS_HWCloseFile(pEASData->hwInstData, fileHandle); - - if (result == EAS_SUCCESS) - { - - /* if a stream pStream is specified, point it to the DLS collection */ - if (pStream) - result = EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_DLS_COLLECTION, (EAS_I32) pDLS); - - /* global DLS load */ - else - result = VMSetGlobalDLSLib(pEASData, pDLS); - } - - return result; -} -#endif - -#ifdef EXTERNAL_AUDIO -/*---------------------------------------------------------------------------- - * EAS_RegExtAudioCallback() - *---------------------------------------------------------------------------- - * Purpose: - * Registers callback functions for audio events. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - file or stream handle - * cbProgChgFunc - pointer to host callback function for program change - * cbEventFunc - pointer to host callback functio for note events - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_RegExtAudioCallback (EAS_DATA_HANDLE pEASData, - EAS_HANDLE pStream, - EAS_VOID_PTR pInstData, - EAS_EXT_PRG_CHG_FUNC cbProgChgFunc, - EAS_EXT_EVENT_FUNC cbEventFunc) -{ - S_SYNTH *pSynth; - - if (!EAS_StreamReady(pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - - if (EAS_GetStreamParameter(pEASData, pStream, PARSER_DATA_SYNTH_HANDLE, (EAS_I32*) &pSynth) != EAS_SUCCESS) - return EAS_ERROR_INVALID_PARAMETER; - - if (pSynth == NULL) - return EAS_ERROR_INVALID_PARAMETER; - - VMRegExtAudioCallback(pSynth, pInstData, cbProgChgFunc, cbEventFunc); - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * EAS_GetMIDIControllers() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current state of MIDI controllers on the requested channel. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - file or stream handle - * pControl - pointer to structure to receive data - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_GetMIDIControllers (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_U8 channel, S_MIDI_CONTROLLERS *pControl) -{ - S_SYNTH *pSynth; - - if (!EAS_StreamReady(pEASData, pStream)) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - - if (EAS_GetStreamParameter(pEASData, pStream, PARSER_DATA_SYNTH_HANDLE, (EAS_I32*) &pSynth) != EAS_SUCCESS) - return EAS_ERROR_INVALID_PARAMETER; - - if (pSynth == NULL) - return EAS_ERROR_INVALID_PARAMETER; - - VMGetMIDIControllers(pSynth, channel, pControl); - return EAS_SUCCESS; -} -#endif - -#ifdef _SPLIT_ARCHITECTURE -/*---------------------------------------------------------------------------- - * EAS_SetFrameBuffer() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the frame buffer pointer passed to the IPC communications functions - * - * Inputs: - * pEASData - instance data handle - * locator - file locator - * - * Outputs: - * - * - * Side Effects: - * May overlay instruments in the GM sound set - * - *---------------------------------------------------------------------------- -*/ -EAS_PUBLIC EAS_RESULT EAS_SetFrameBuffer (EAS_DATA_HANDLE pEASData, EAS_FRAME_BUFFER_HANDLE pFrameBuffer) -{ - if (pEASData->pVoiceMgr) - pEASData->pVoiceMgr->pFrameBuffer = pFrameBuffer; - return EAS_SUCCESS; -} -#endif - -/*---------------------------------------------------------------------------- - * EAS_SearchFile - *---------------------------------------------------------------------------- - * Search file for specific sequence starting at current file - * position. Returns offset to start of sequence. - * - * Inputs: - * pEASData - pointer to EAS persistent data object - * fileHandle - file handle - * searchString - pointer to search sequence - * len - length of search sequence - * pOffset - pointer to variable to store offset to sequence - * - * Returns EAS_EOF if end-of-file is reached - *---------------------------------------------------------------------------- -*/ -EAS_RESULT EAS_SearchFile (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, const EAS_U8 *searchString, EAS_I32 len, EAS_I32 *pOffset) -{ - EAS_RESULT result; - EAS_INT index; - EAS_U8 c; - - *pOffset = -1; - index = 0; - for (;;) - { - result = EAS_HWGetByte(pEASData->hwInstData, fileHandle, &c); - if (result != EAS_SUCCESS) - return result; - if (c == searchString[index]) - { - index++; - if (index == 4) - { - result = EAS_HWFilePos(pEASData->hwInstData, fileHandle, pOffset); - if (result != EAS_SUCCESS) - return result; - *pOffset -= len; - break; - } - } - else - index = 0; - } - return EAS_SUCCESS; -} - - diff --git a/arm-hybrid-22k/lib_src/eas_reverb.c b/arm-hybrid-22k/lib_src/eas_reverb.c deleted file mode 100644 index 6d99862..0000000 --- a/arm-hybrid-22k/lib_src/eas_reverb.c +++ /dev/null @@ -1,1154 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_reverb.c - * - * Contents and purpose: - * Contains the implementation of the Reverb effect. - * - * - * Copyright Sonic Network Inc. 2006 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 510 $ - * $Date: 2006-12-19 01:47:33 -0800 (Tue, 19 Dec 2006) $ - *---------------------------------------------------------------------------- -*/ - -/*------------------------------------ - * includes - *------------------------------------ -*/ - -#include "eas_data.h" -#include "eas_effects.h" -#include "eas_math.h" -#include "eas_reverbdata.h" -#include "eas_reverb.h" -#include "eas_config.h" -#include "eas_host.h" -#include "eas_report.h" - -/* prototypes for effects interface */ -static EAS_RESULT ReverbInit (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData); -static void ReverbProcess (EAS_VOID_PTR pInstData, EAS_PCM *pSrc, EAS_PCM *pDst, EAS_I32 numSamples); -static EAS_RESULT ReverbShutdown (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT ReverbGetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue); -static EAS_RESULT ReverbSetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value); - -/* common effects interface for configuration module */ -const S_EFFECTS_INTERFACE EAS_Reverb = -{ - ReverbInit, - ReverbProcess, - ReverbShutdown, - ReverbGetParam, - ReverbSetParam -}; - - - -/*---------------------------------------------------------------------------- - * InitializeReverb() - *---------------------------------------------------------------------------- - * Purpose: - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ReverbInit(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData) -{ - EAS_I32 i; - EAS_U16 nOffset; - EAS_INT temp; - - S_REVERB_OBJECT *pReverbData; - S_REVERB_PRESET *pPreset; - - /* check Configuration Module for data allocation */ - if (pEASData->staticMemoryModel) - pReverbData = EAS_CMEnumFXData(EAS_MODULE_REVERB); - - /* allocate dynamic memory */ - else - pReverbData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_REVERB_OBJECT)); - - if (pReverbData == NULL) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Failed to allocate Reverb memory\n"); */ } - return EAS_ERROR_MALLOC_FAILED; - } - - /* clear the structure */ - EAS_HWMemSet(pReverbData, 0, sizeof(S_REVERB_OBJECT)); - - ReverbReadInPresets(pReverbData); - - pReverbData->m_nMinSamplesToAdd = REVERB_UPDATE_PERIOD_IN_SAMPLES; - - pReverbData->m_nRevOutFbkR = 0; - pReverbData->m_nRevOutFbkL = 0; - - pReverbData->m_sAp0.m_zApIn = AP0_IN; - pReverbData->m_sAp0.m_zApOut = AP0_IN + DEFAULT_AP0_LENGTH; - pReverbData->m_sAp0.m_nApGain = DEFAULT_AP0_GAIN; - - pReverbData->m_zD0In = DELAY0_IN; - - pReverbData->m_sAp1.m_zApIn = AP1_IN; - pReverbData->m_sAp1.m_zApOut = AP1_IN + DEFAULT_AP1_LENGTH; - pReverbData->m_sAp1.m_nApGain = DEFAULT_AP1_GAIN; - - pReverbData->m_zD1In = DELAY1_IN; - - pReverbData->m_zLpf0 = 0; - pReverbData->m_zLpf1 = 0; - pReverbData->m_nLpfFwd = 8837; - pReverbData->m_nLpfFbk = 6494; - - pReverbData->m_nSin = 0; - pReverbData->m_nCos = 0; - pReverbData->m_nSinIncrement = 0; - pReverbData->m_nCosIncrement = 0; - - // set xfade parameters - pReverbData->m_nXfadeInterval = (EAS_U16)REVERB_XFADE_PERIOD_IN_SAMPLES; - pReverbData->m_nXfadeCounter = pReverbData->m_nXfadeInterval + 1; // force update on first iteration - pReverbData->m_nPhase = -32768; - pReverbData->m_nPhaseIncrement = REVERB_XFADE_PHASE_INCREMENT; - - pReverbData->m_nNoise = (EAS_I16)0xABCD; - - pReverbData->m_nMaxExcursion = 0x007F; - - // set delay tap lengths - nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion, - &pReverbData->m_nNoise ); - - pReverbData->m_zD1Cross = - DELAY1_OUT - pReverbData->m_nMaxExcursion + nOffset; - - nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion, - &pReverbData->m_nNoise ); - - pReverbData->m_zD0Cross = - DELAY1_OUT - pReverbData->m_nMaxExcursion - nOffset; - - nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion, - &pReverbData->m_nNoise ); - - pReverbData->m_zD0Self = - DELAY0_OUT - pReverbData->m_nMaxExcursion - nOffset; - - nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion, - &pReverbData->m_nNoise ); - - pReverbData->m_zD1Self = - DELAY1_OUT - pReverbData->m_nMaxExcursion + nOffset; - - // for debugging purposes, allow noise generator - pReverbData->m_bUseNoise = EAS_FALSE; - - // for debugging purposes, allow bypass - pReverbData->m_bBypass = EAS_TRUE; //EAS_FALSE; - - pReverbData->m_nNextRoom = 1; - - pReverbData->m_nCurrentRoom = pReverbData->m_nNextRoom + 1; // force update on first iteration - - pReverbData->m_nWet = REVERB_DEFAULT_WET; - - pReverbData->m_nDry = REVERB_DEFAULT_DRY; - - // set base index into circular buffer - pReverbData->m_nBaseIndex = 0; - - // set the early reflections, L - pReverbData->m_sEarlyL.m_nLpfFbk = 4915; - pReverbData->m_sEarlyL.m_nLpfFwd = 27852; - pReverbData->m_sEarlyL.m_zLpf = 0; - - for (i=0; i < REVERB_MAX_NUM_REFLECTIONS; i++) - { - pReverbData->m_sEarlyL.m_nGain[i] = 0; - pReverbData->m_sEarlyL.m_zDelay[i] = 0; - } - - // set the early reflections, R - pReverbData->m_sEarlyR.m_nLpfFbk = 4915; - pReverbData->m_sEarlyR.m_nLpfFwd = 27852; - pReverbData->m_sEarlyR.m_zLpf = 0; - - for (i=0; i < REVERB_MAX_NUM_REFLECTIONS; i++) - { - pReverbData->m_sEarlyR.m_nGain[i] = 0; - pReverbData->m_sEarlyR.m_zDelay[i] = 0; - } - - // clear the reverb delay line - for (i=0; i < REVERB_BUFFER_SIZE_IN_SAMPLES; i++) - { - pReverbData->m_nDelayLine[i] = 0; - } - - //////////////////////////////// - ///code from the EAS DEMO Reverb - //now copy from the new preset into the reverb - pPreset = &pReverbData->m_sPreset.m_sPreset[pReverbData->m_nNextRoom]; - - pReverbData->m_nLpfFbk = pPreset->m_nLpfFbk; - pReverbData->m_nLpfFwd = pPreset->m_nLpfFwd; - - pReverbData->m_nEarly = pPreset->m_nEarly; - pReverbData->m_nWet = pPreset->m_nWet; - pReverbData->m_nDry = pPreset->m_nDry; - - pReverbData->m_nMaxExcursion = pPreset->m_nMaxExcursion; - //stored as time based, convert to sample based - temp = pPreset->m_nXfadeInterval; - /*lint -e{702} shift for performance */ - temp = (temp * _OUTPUT_SAMPLE_RATE) >> 16; - pReverbData->m_nXfadeInterval = (EAS_U16) temp; - //gsReverbObject.m_nXfadeInterval = pPreset->m_nXfadeInterval; - - pReverbData->m_sAp0.m_nApGain = pPreset->m_nAp0_ApGain; - //stored as time based, convert to absolute sample value - temp = pPreset->m_nAp0_ApOut; - /*lint -e{702} shift for performance */ - temp = (temp * _OUTPUT_SAMPLE_RATE) >> 16; - pReverbData->m_sAp0.m_zApOut = (EAS_U16) (pReverbData->m_sAp0.m_zApIn + temp); - //gsReverbObject.m_sAp0.m_zApOut = pPreset->m_nAp0_ApOut; - - pReverbData->m_sAp1.m_nApGain = pPreset->m_nAp1_ApGain; - //stored as time based, convert to absolute sample value - temp = pPreset->m_nAp1_ApOut; - /*lint -e{702} shift for performance */ - temp = (temp * _OUTPUT_SAMPLE_RATE) >> 16; - pReverbData->m_sAp1.m_zApOut = (EAS_U16) (pReverbData->m_sAp1.m_zApIn + temp); - //gsReverbObject.m_sAp1.m_zApOut = pPreset->m_nAp1_ApOut; - ///code from the EAS DEMO Reverb - //////////////////////////////// - - *pInstData = pReverbData; - - return EAS_SUCCESS; - -} /* end InitializeReverb */ - - - -/*---------------------------------------------------------------------------- - * ReverbProcess() - *---------------------------------------------------------------------------- - * Purpose: - * Reverberate the requested number of samples (block based processing) - * - * Inputs: - * pInputBuffer - src buffer - * pOutputBuffer - dst buffer - * nNumSamplesToAdd - number of samples to write to buffer - * - * Outputs: - * number of samples actually written to buffer - * - * Side Effects: - * - samples are added to the presently free buffer - * - *---------------------------------------------------------------------------- -*/ -static void ReverbProcess(EAS_VOID_PTR pInstData, EAS_PCM *pSrc, EAS_PCM *pDst, EAS_I32 numSamples) -{ - S_REVERB_OBJECT *pReverbData; - - pReverbData = (S_REVERB_OBJECT*) pInstData; - - //if bypassed or the preset forces the signal to be completely dry - if (pReverbData->m_bBypass || - (pReverbData->m_nWet == 0 && pReverbData->m_nDry == 32767)) - { - if (pSrc != pDst) - EAS_HWMemCpy(pSrc, pDst, numSamples * NUM_OUTPUT_CHANNELS * (EAS_I32) sizeof(EAS_PCM)); - return; - } - - if (pReverbData->m_nNextRoom != pReverbData->m_nCurrentRoom) - { - ReverbUpdateRoom(pReverbData); - } - - ReverbUpdateXfade(pReverbData, numSamples); - - Reverb(pReverbData, numSamples, pDst, pSrc); - - /* check if update counter needs to be reset */ - if (pReverbData->m_nUpdateCounter >= REVERB_MODULO_UPDATE_PERIOD_IN_SAMPLES) - { - /* update interval has elapsed, so reset counter */ - pReverbData->m_nUpdateCounter = 0; - } /* end if m_nUpdateCounter >= update interval */ - - /* increment update counter */ - pReverbData->m_nUpdateCounter += (EAS_I16)numSamples; - -} /* end ComputeReverb */ - -/*---------------------------------------------------------------------------- - * ReverbUpdateXfade - *---------------------------------------------------------------------------- - * Purpose: - * Update the xfade parameters as required - * - * Inputs: - * nNumSamplesToAdd - number of samples to write to buffer - * - * Outputs: - * - * - * Side Effects: - * - xfade parameters will be changed - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ReverbUpdateXfade(S_REVERB_OBJECT *pReverbData, EAS_INT nNumSamplesToAdd) -{ - EAS_U16 nOffset; - EAS_I16 tempCos; - EAS_I16 tempSin; - - if (pReverbData->m_nXfadeCounter >= pReverbData->m_nXfadeInterval) - { - /* update interval has elapsed, so reset counter */ - pReverbData->m_nXfadeCounter = 0; - - // Pin the sin,cos values to min / max values to ensure that the - // modulated taps' coefs are zero (thus no clicks) - if (pReverbData->m_nPhaseIncrement > 0) - { - // if phase increment > 0, then sin -> 1, cos -> 0 - pReverbData->m_nSin = 32767; - pReverbData->m_nCos = 0; - - // reset the phase to match the sin, cos values - pReverbData->m_nPhase = 32767; - - // modulate the cross taps because their tap coefs are zero - nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion, &pReverbData->m_nNoise ); - - pReverbData->m_zD1Cross = - DELAY1_OUT - pReverbData->m_nMaxExcursion + nOffset; - - nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion, &pReverbData->m_nNoise ); - - pReverbData->m_zD0Cross = - DELAY0_OUT - pReverbData->m_nMaxExcursion - nOffset; - } - else - { - // if phase increment < 0, then sin -> 0, cos -> 1 - pReverbData->m_nSin = 0; - pReverbData->m_nCos = 32767; - - // reset the phase to match the sin, cos values - pReverbData->m_nPhase = -32768; - - // modulate the self taps because their tap coefs are zero - nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion, &pReverbData->m_nNoise ); - - pReverbData->m_zD0Self = - DELAY0_OUT - pReverbData->m_nMaxExcursion - nOffset; - - nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion, &pReverbData->m_nNoise ); - - pReverbData->m_zD1Self = - DELAY1_OUT - pReverbData->m_nMaxExcursion + nOffset; - - } // end if-else (pReverbData->m_nPhaseIncrement > 0) - - // Reverse the direction of the sin,cos so that the - // tap whose coef was previously increasing now decreases - // and vice versa - pReverbData->m_nPhaseIncrement = -pReverbData->m_nPhaseIncrement; - - } // end if counter >= update interval - - //compute what phase will be next time - pReverbData->m_nPhase += pReverbData->m_nPhaseIncrement; - - //calculate what the new sin and cos need to reach by the next update - ReverbCalculateSinCos(pReverbData->m_nPhase, &tempSin, &tempCos); - - //calculate the per-sample increment required to get there by the next update - /*lint -e{702} shift for performance */ - pReverbData->m_nSinIncrement = - (tempSin - pReverbData->m_nSin) >> REVERB_UPDATE_PERIOD_IN_BITS; - - /*lint -e{702} shift for performance */ - pReverbData->m_nCosIncrement = - (tempCos - pReverbData->m_nCos) >> REVERB_UPDATE_PERIOD_IN_BITS; - - - /* increment update counter */ - pReverbData->m_nXfadeCounter += (EAS_U16) nNumSamplesToAdd; - - return EAS_SUCCESS; - -} /* end ReverbUpdateXfade */ - - -/*---------------------------------------------------------------------------- - * ReverbCalculateNoise - *---------------------------------------------------------------------------- - * Purpose: - * Calculate a noise sample and limit its value - * - * Inputs: - * nMaxExcursion - noise value is limited to this value - * pnNoise - return new noise sample in this (not limited) - * - * Outputs: - * new limited noise value - * - * Side Effects: - * - *pnNoise noise value is updated - * - *---------------------------------------------------------------------------- -*/ -static EAS_U16 ReverbCalculateNoise(EAS_U16 nMaxExcursion, EAS_I16 *pnNoise) -{ - // calculate new noise value - *pnNoise = (EAS_I16) (*pnNoise * 5 + 1); - -#if 0 // 1xxx, test - *pnNoise = 0; -#endif // 1xxx, test - - // return the limited noise value - return (nMaxExcursion & (*pnNoise)); - -} /* end ReverbCalculateNoise */ - -/*---------------------------------------------------------------------------- - * ReverbCalculateSinCos - *---------------------------------------------------------------------------- - * Purpose: - * Calculate a new sin and cosine value based on the given phase - * - * Inputs: - * nPhase - phase angle - * pnSin - input old value, output new value - * pnCos - input old value, output new value - * - * Outputs: - * - * Side Effects: - * - *pnSin, *pnCos are updated - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ReverbCalculateSinCos(EAS_I16 nPhase, EAS_I16 *pnSin, EAS_I16 *pnCos) -{ - EAS_I32 nTemp; - EAS_I32 nNetAngle; - - // -1 <= nPhase < 1 - // However, for the calculation, we need a value - // that ranges from -1/2 to +1/2, so divide the phase by 2 - /*lint -e{702} shift for performance */ - nNetAngle = nPhase >> 1; - - /* - Implement the following - sin(x) = (2-4*c)*x^2 + c + x - cos(x) = (2-4*c)*x^2 + c - x - - where c = 1/sqrt(2) - using the a0 + x*(a1 + x*a2) approach - */ - - /* limit the input "angle" to be between -0.5 and +0.5 */ - if (nNetAngle > EG1_HALF) - { - nNetAngle = EG1_HALF; - } - else if (nNetAngle < EG1_MINUS_HALF) - { - nNetAngle = EG1_MINUS_HALF; - } - - /* calculate sin */ - nTemp = EG1_ONE + MULT_EG1_EG1(REVERB_PAN_G2, nNetAngle); - nTemp = REVERB_PAN_G0 + MULT_EG1_EG1(nTemp, nNetAngle); - *pnSin = (EAS_I16) SATURATE_EG1(nTemp); - - /* calculate cos */ - nTemp = -EG1_ONE + MULT_EG1_EG1(REVERB_PAN_G2, nNetAngle); - nTemp = REVERB_PAN_G0 + MULT_EG1_EG1(nTemp, nNetAngle); - *pnCos = (EAS_I16) SATURATE_EG1(nTemp); - - return EAS_SUCCESS; -} /* end ReverbCalculateSinCos */ - -/*---------------------------------------------------------------------------- - * Reverb - *---------------------------------------------------------------------------- - * Purpose: - * apply reverb to the given signal - * - * Inputs: - * nNu - * pnSin - input old value, output new value - * pnCos - input old value, output new value - * - * Outputs: - * number of samples actually reverberated - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT Reverb(S_REVERB_OBJECT *pReverbData, EAS_INT nNumSamplesToAdd, EAS_PCM *pOutputBuffer, EAS_PCM *pInputBuffer) -{ - EAS_I32 i; - EAS_I32 nDelayOut; - EAS_U16 nBase; - - EAS_U32 nAddr; - EAS_I32 nTemp1; - EAS_I32 nTemp2; - EAS_I32 nApIn; - EAS_I32 nApOut; - - EAS_I32 j; - EAS_I32 nEarlyOut; - - EAS_I32 tempValue; - - - // get the base address - nBase = pReverbData->m_nBaseIndex; - - for (i=0; i < nNumSamplesToAdd; i++) - { - // ********** Left Allpass - start - // left input = (left dry/4) + right feedback from previous period - /*lint -e{702} use shift for performance */ - nApIn = ((*pInputBuffer++)>>2) + pReverbData->m_nRevOutFbkR; -// nApIn = *pInputBuffer++; // 1xxx test and debug ap - - // fetch allpass delay line out - //nAddr = CIRCULAR(nBase, psAp0->m_zApOut, REVERB_BUFFER_MASK); - nAddr = CIRCULAR(nBase, pReverbData->m_sAp0.m_zApOut, REVERB_BUFFER_MASK); - nDelayOut = pReverbData->m_nDelayLine[nAddr]; - - // calculate allpass feedforward; subtract the feedforward result - nTemp1 = MULT_EG1_EG1(nApIn, pReverbData->m_sAp0.m_nApGain); - nApOut = SATURATE(nDelayOut - nTemp1); // allpass output - - // calculate allpass feedback; add the feedback result - nTemp1 = MULT_EG1_EG1(nApOut, pReverbData->m_sAp0.m_nApGain); - nTemp1 = SATURATE(nApIn + nTemp1); - - // inject into allpass delay - nAddr = CIRCULAR(nBase, pReverbData->m_sAp0.m_zApIn, REVERB_BUFFER_MASK); - pReverbData->m_nDelayLine[nAddr] = (EAS_PCM) nTemp1; - - // inject allpass output into delay line - nAddr = CIRCULAR(nBase, pReverbData->m_zD0In, REVERB_BUFFER_MASK); - pReverbData->m_nDelayLine[nAddr] = (EAS_PCM) nApOut; - - // ********** Left Allpass - end - - // ********** Right Allpass - start - // right input = (right dry/4) + left feedback from previous period - /*lint -e{702} use shift for performance */ - nApIn = ((*pInputBuffer++)>>2) + pReverbData->m_nRevOutFbkL; -// nApIn = *pInputBuffer++; // 1xxx test and debug ap - - // fetch allpass delay line out - nAddr = CIRCULAR(nBase, pReverbData->m_sAp1.m_zApOut, REVERB_BUFFER_MASK); - nDelayOut = pReverbData->m_nDelayLine[nAddr]; - - // calculate allpass feedforward; subtract the feedforward result - nTemp1 = MULT_EG1_EG1(nApIn, pReverbData->m_sAp1.m_nApGain); - nApOut = SATURATE(nDelayOut - nTemp1); // allpass output - - // calculate allpass feedback; add the feedback result - nTemp1 = MULT_EG1_EG1(nApOut, pReverbData->m_sAp1.m_nApGain); - nTemp1 = SATURATE(nApIn + nTemp1); - - // inject into allpass delay - nAddr = CIRCULAR(nBase, pReverbData->m_sAp1.m_zApIn, REVERB_BUFFER_MASK); - pReverbData->m_nDelayLine[nAddr] = (EAS_PCM) nTemp1; - - // inject allpass output into delay line - nAddr = CIRCULAR(nBase, pReverbData->m_zD1In, REVERB_BUFFER_MASK); - pReverbData->m_nDelayLine[nAddr] = (EAS_PCM) nApOut; - - // ********** Right Allpass - end - - // ********** D0 output - start - // fetch delay line self out - nAddr = CIRCULAR(nBase, pReverbData->m_zD0Self, REVERB_BUFFER_MASK); - nDelayOut = pReverbData->m_nDelayLine[nAddr]; - - // calculate delay line self out - nTemp1 = MULT_EG1_EG1(nDelayOut, pReverbData->m_nSin); - - // fetch delay line cross out - nAddr = CIRCULAR(nBase, pReverbData->m_zD1Cross, REVERB_BUFFER_MASK); - nDelayOut = pReverbData->m_nDelayLine[nAddr]; - - // calculate delay line self out - nTemp2 = MULT_EG1_EG1(nDelayOut, pReverbData->m_nCos); - - // calculate unfiltered delay out - nDelayOut = SATURATE(nTemp1 + nTemp2); - - // calculate lowpass filter (mixer scale factor included in LPF feedforward) - nTemp1 = MULT_EG1_EG1(nDelayOut, pReverbData->m_nLpfFwd); - - nTemp2 = MULT_EG1_EG1(pReverbData->m_zLpf0, pReverbData->m_nLpfFbk); - - // calculate filtered delay out and simultaneously update LPF state variable - // filtered delay output is stored in m_zLpf0 - pReverbData->m_zLpf0 = (EAS_PCM) SATURATE(nTemp1 + nTemp2); - - // ********** D0 output - end - - // ********** D1 output - start - // fetch delay line self out - nAddr = CIRCULAR(nBase, pReverbData->m_zD1Self, REVERB_BUFFER_MASK); - nDelayOut = pReverbData->m_nDelayLine[nAddr]; - - // calculate delay line self out - nTemp1 = MULT_EG1_EG1(nDelayOut, pReverbData->m_nSin); - - // fetch delay line cross out - nAddr = CIRCULAR(nBase, pReverbData->m_zD0Cross, REVERB_BUFFER_MASK); - nDelayOut = pReverbData->m_nDelayLine[nAddr]; - - // calculate delay line self out - nTemp2 = MULT_EG1_EG1(nDelayOut, pReverbData->m_nCos); - - // calculate unfiltered delay out - nDelayOut = SATURATE(nTemp1 + nTemp2); - - // calculate lowpass filter (mixer scale factor included in LPF feedforward) - nTemp1 = MULT_EG1_EG1(nDelayOut, pReverbData->m_nLpfFwd); - - nTemp2 = MULT_EG1_EG1(pReverbData->m_zLpf1, pReverbData->m_nLpfFbk); - - // calculate filtered delay out and simultaneously update LPF state variable - // filtered delay output is stored in m_zLpf1 - pReverbData->m_zLpf1 = (EAS_PCM)SATURATE(nTemp1 + nTemp2); - - // ********** D1 output - end - - // ********** mixer and feedback - start - // sum is fedback to right input (R + L) - pReverbData->m_nRevOutFbkL = - (EAS_PCM)SATURATE((EAS_I32)pReverbData->m_zLpf1 + (EAS_I32)pReverbData->m_zLpf0); - - // difference is feedback to left input (R - L) - /*lint -e{685} lint complains that it can't saturate negative */ - pReverbData->m_nRevOutFbkR = - (EAS_PCM)SATURATE((EAS_I32)pReverbData->m_zLpf1 - (EAS_I32)pReverbData->m_zLpf0); - - // ********** mixer and feedback - end - - // ********** start early reflection generator, left - //psEarly = &(pReverbData->m_sEarlyL); - - nEarlyOut = 0; - - for (j=0; j < REVERB_MAX_NUM_REFLECTIONS; j++) - { - // fetch delay line out - //nAddr = CIRCULAR(nBase, psEarly->m_zDelay[j], REVERB_BUFFER_MASK); - nAddr = CIRCULAR(nBase, pReverbData->m_sEarlyL.m_zDelay[j], REVERB_BUFFER_MASK); - - nDelayOut = pReverbData->m_nDelayLine[nAddr]; - - // calculate reflection - //nTemp1 = MULT_EG1_EG1(nDelayOut, psEarly->m_nGain[j]); - nTemp1 = MULT_EG1_EG1(nDelayOut, pReverbData->m_sEarlyL.m_nGain[j]); - - nEarlyOut = SATURATE(nEarlyOut + nTemp1); - - } // end for (j=0; j < REVERB_MAX_NUM_REFLECTIONS; j++) - - // apply lowpass to early reflections - //nTemp1 = MULT_EG1_EG1(nEarlyOut, psEarly->m_nLpfFwd); - nTemp1 = MULT_EG1_EG1(nEarlyOut, pReverbData->m_sEarlyL.m_nLpfFwd); - - //nTemp2 = MULT_EG1_EG1(psEarly->m_zLpf, psEarly->m_nLpfFbk); - nTemp2 = MULT_EG1_EG1(pReverbData->m_sEarlyL.m_zLpf, pReverbData->m_sEarlyL.m_nLpfFbk); - - - // calculate filtered out and simultaneously update LPF state variable - // filtered output is stored in m_zLpf1 - //psEarly->m_zLpf = SATURATE(nTemp1 + nTemp2); - pReverbData->m_sEarlyL.m_zLpf = (EAS_PCM) SATURATE(nTemp1 + nTemp2); - - // combine filtered early and late reflections for output - //*pOutputBuffer++ = inL; - //tempValue = SATURATE(psEarly->m_zLpf + pReverbData->m_nRevOutFbkL); - tempValue = SATURATE((EAS_I32)pReverbData->m_sEarlyL.m_zLpf + (EAS_I32)pReverbData->m_nRevOutFbkL); - //scale reverb output by wet level - /*lint -e{701} use shift for performance */ - tempValue = MULT_EG1_EG1(tempValue, (pReverbData->m_nWet<<1)); - //sum with output buffer - tempValue += *pOutputBuffer; - *pOutputBuffer++ = (EAS_PCM)SATURATE(tempValue); - - // ********** end early reflection generator, left - - // ********** start early reflection generator, right - //psEarly = &(pReverbData->m_sEarlyR); - - nEarlyOut = 0; - - for (j=0; j < REVERB_MAX_NUM_REFLECTIONS; j++) - { - // fetch delay line out - nAddr = CIRCULAR(nBase, pReverbData->m_sEarlyR.m_zDelay[j], REVERB_BUFFER_MASK); - nDelayOut = pReverbData->m_nDelayLine[nAddr]; - - // calculate reflection - nTemp1 = MULT_EG1_EG1(nDelayOut, pReverbData->m_sEarlyR.m_nGain[j]); - - nEarlyOut = SATURATE(nEarlyOut + nTemp1); - - } // end for (j=0; j < REVERB_MAX_NUM_REFLECTIONS; j++) - - // apply lowpass to early reflections - nTemp1 = MULT_EG1_EG1(nEarlyOut, pReverbData->m_sEarlyR.m_nLpfFwd); - - nTemp2 = MULT_EG1_EG1(pReverbData->m_sEarlyR.m_zLpf, pReverbData->m_sEarlyR.m_nLpfFbk); - - // calculate filtered out and simultaneously update LPF state variable - // filtered output is stored in m_zLpf1 - pReverbData->m_sEarlyR.m_zLpf = (EAS_PCM)SATURATE(nTemp1 + nTemp2); - - // combine filtered early and late reflections for output - //*pOutputBuffer++ = inR; - tempValue = SATURATE((EAS_I32)pReverbData->m_sEarlyR.m_zLpf + (EAS_I32)pReverbData->m_nRevOutFbkR); - //scale reverb output by wet level - /*lint -e{701} use shift for performance */ - tempValue = MULT_EG1_EG1(tempValue, (pReverbData->m_nWet << 1)); - //sum with output buffer - tempValue = tempValue + *pOutputBuffer; - *pOutputBuffer++ = (EAS_PCM)SATURATE(tempValue); - - // ********** end early reflection generator, right - - // decrement base addr for next sample period - nBase--; - - pReverbData->m_nSin += pReverbData->m_nSinIncrement; - pReverbData->m_nCos += pReverbData->m_nCosIncrement; - - } // end for (i=0; i < nNumSamplesToAdd; i++) - - // store the most up to date version - pReverbData->m_nBaseIndex = nBase; - - return EAS_SUCCESS; -} /* end Reverb */ - - - -/*---------------------------------------------------------------------------- - * ReverbShutdown() - *---------------------------------------------------------------------------- - * Purpose: - * Initializes the Reverb effect. - * - * Inputs: - * pInstData - handle to instance data - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ReverbShutdown (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData) -{ - /* check Configuration Module for static memory allocation */ - if (!pEASData->staticMemoryModel) - EAS_HWFree(pEASData->hwInstData, pInstData); - return EAS_SUCCESS; -} /* end ReverbShutdown */ - -/*---------------------------------------------------------------------------- - * ReverbGetParam() - *---------------------------------------------------------------------------- - * Purpose: - * Get a Reverb parameter - * - * Inputs: - * pInstData - handle to instance data - * param - parameter index - * *pValue - pointer to variable to hold retrieved value - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ReverbGetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue) -{ - S_REVERB_OBJECT *p; - - p = (S_REVERB_OBJECT*) pInstData; - - switch (param) - { - case EAS_PARAM_REVERB_BYPASS: - *pValue = (EAS_I32) p->m_bBypass; - break; - case EAS_PARAM_REVERB_PRESET: - *pValue = (EAS_I8) p->m_nCurrentRoom; - break; - case EAS_PARAM_REVERB_WET: - *pValue = p->m_nWet; - break; - case EAS_PARAM_REVERB_DRY: - *pValue = p->m_nDry; - break; - default: - return EAS_ERROR_INVALID_PARAMETER; - } - return EAS_SUCCESS; -} /* end ReverbGetParam */ - - -/*---------------------------------------------------------------------------- - * ReverbSetParam() - *---------------------------------------------------------------------------- - * Purpose: - * Set a Reverb parameter - * - * Inputs: - * pInstData - handle to instance data - * param - parameter index - * *pValue - new paramter value - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ReverbSetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value) -{ - S_REVERB_OBJECT *p; - - p = (S_REVERB_OBJECT*) pInstData; - - switch (param) - { - case EAS_PARAM_REVERB_BYPASS: - p->m_bBypass = (EAS_BOOL) value; - break; - case EAS_PARAM_REVERB_PRESET: - if(value!=EAS_PARAM_REVERB_LARGE_HALL && value!=EAS_PARAM_REVERB_HALL && - value!=EAS_PARAM_REVERB_CHAMBER && value!=EAS_PARAM_REVERB_ROOM) - return EAS_ERROR_INVALID_PARAMETER; - p->m_nNextRoom = (EAS_I16)value; - break; - case EAS_PARAM_REVERB_WET: - if(value>EAS_REVERB_WET_MAX || valuem_nWet = (EAS_I16)value; - break; - case EAS_PARAM_REVERB_DRY: - if(value>EAS_REVERB_DRY_MAX || valuem_nDry = (EAS_I16)value; - break; - default: - return EAS_ERROR_INVALID_PARAMETER; - } - return EAS_SUCCESS; -} /* end ReverbSetParam */ - - -/*---------------------------------------------------------------------------- - * ReverbUpdateRoom - *---------------------------------------------------------------------------- - * Purpose: - * Update the room's preset parameters as required - * - * Inputs: - * - * Outputs: - * - * - * Side Effects: - * - reverb paramters (fbk, fwd, etc) will be changed - * - m_nCurrentRoom := m_nNextRoom - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ReverbUpdateRoom(S_REVERB_OBJECT *pReverbData) -{ - EAS_INT temp; - - S_REVERB_PRESET *pPreset = &pReverbData->m_sPreset.m_sPreset[pReverbData->m_nNextRoom]; - - pReverbData->m_nLpfFwd = pPreset->m_nLpfFwd; - pReverbData->m_nLpfFbk = pPreset->m_nLpfFbk; - - pReverbData->m_nEarly = pPreset->m_nEarly; - pReverbData->m_nWet = pPreset->m_nWet; - pReverbData->m_nDry = pPreset->m_nDry; - - - pReverbData->m_nMaxExcursion = pPreset->m_nMaxExcursion; - //stored as time based, convert to sample based - temp = pPreset->m_nXfadeInterval; - /*lint -e{702} shift for performance */ - temp = (temp * _OUTPUT_SAMPLE_RATE) >> 16; - pReverbData->m_nXfadeInterval = (EAS_U16) temp; - //gpsReverbObject->m_nXfadeInterval = pPreset->m_nXfadeInterval; - pReverbData->m_sAp0.m_nApGain = pPreset->m_nAp0_ApGain; - //stored as time based, convert to absolute sample value - temp = pPreset->m_nAp0_ApOut; - /*lint -e{702} shift for performance */ - temp = (temp * _OUTPUT_SAMPLE_RATE) >> 16; - pReverbData->m_sAp0.m_zApOut = (EAS_U16) (pReverbData->m_sAp0.m_zApIn + temp); - //gpsReverbObject->m_sAp0.m_zApOut = pPreset->m_nAp0_ApOut; - pReverbData->m_sAp1.m_nApGain = pPreset->m_nAp1_ApGain; - //stored as time based, convert to absolute sample value - temp = pPreset->m_nAp1_ApOut; - /*lint -e{702} shift for performance */ - temp = (temp * _OUTPUT_SAMPLE_RATE) >> 16; - pReverbData->m_sAp1.m_zApOut = (EAS_U16) (pReverbData->m_sAp1.m_zApIn + temp); - //gpsReverbObject->m_sAp1.m_zApOut = pPreset->m_nAp1_ApOut; - - pReverbData->m_nCurrentRoom = pReverbData->m_nNextRoom; - - return EAS_SUCCESS; - -} /* end ReverbUpdateRoom */ - - -/*---------------------------------------------------------------------------- - * ReverbReadInPresets() - *---------------------------------------------------------------------------- - * Purpose: sets global reverb preset bank to defaults - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ReverbReadInPresets(S_REVERB_OBJECT *pReverbData) -{ - - int preset = 0; - int defaultPreset = 0; - - //now init any remaining presets to defaults - for (defaultPreset = preset; defaultPreset < REVERB_MAX_ROOM_TYPE; defaultPreset++) - { - S_REVERB_PRESET *pPreset = &pReverbData->m_sPreset.m_sPreset[defaultPreset]; - if (defaultPreset == 0 || defaultPreset > REVERB_MAX_ROOM_TYPE-1) - { - pPreset->m_nLpfFbk = 8307; - pPreset->m_nLpfFwd = 14768; - pPreset->m_nEarly = 0; - pPreset->m_nWet = 27690; - pPreset->m_nDry = 32767; - pPreset->m_nEarlyL_LpfFbk = 3692; - pPreset->m_nEarlyL_LpfFwd = 29075; - pPreset->m_nEarlyL_Delay0 = 922; - pPreset->m_nEarlyL_Gain0 = 22152; - pPreset->m_nEarlyL_Delay1 = 1462; - pPreset->m_nEarlyL_Gain1 = 17537; - pPreset->m_nEarlyL_Delay2 = 0; - pPreset->m_nEarlyL_Gain2 = 14768; - pPreset->m_nEarlyL_Delay3 = 1221; - pPreset->m_nEarlyL_Gain3 = 14307; - pPreset->m_nEarlyL_Delay4 = 0; - pPreset->m_nEarlyL_Gain4 = 13384; - pPreset->m_nEarlyR_Delay0 = 502; - pPreset->m_nEarlyR_Gain0 = 20306; - pPreset->m_nEarlyR_Delay1 = 1762; - pPreset->m_nEarlyR_Gain1 = 17537; - pPreset->m_nEarlyR_Delay2 = 0; - pPreset->m_nEarlyR_Gain2 = 14768; - pPreset->m_nEarlyR_Delay3 = 0; - pPreset->m_nEarlyR_Gain3 = 16153; - pPreset->m_nEarlyR_Delay4 = 0; - pPreset->m_nEarlyR_Gain4 = 13384; - pPreset->m_nMaxExcursion = 127; - pPreset->m_nXfadeInterval = 6388; - pPreset->m_nAp0_ApGain = 15691; - pPreset->m_nAp0_ApOut = 711; - pPreset->m_nAp1_ApGain = 17999; - pPreset->m_nAp1_ApOut = 1113; - pPreset->m_rfu4 = 0; - pPreset->m_rfu5 = 0; - pPreset->m_rfu6 = 0; - pPreset->m_rfu7 = 0; - pPreset->m_rfu8 = 0; - pPreset->m_rfu9 = 0; - pPreset->m_rfu10 = 0; - } - else if (defaultPreset == 1) - { - pPreset->m_nLpfFbk = 6461; - pPreset->m_nLpfFwd = 14307; - pPreset->m_nEarly = 0; - pPreset->m_nWet = 27690; - pPreset->m_nDry = 32767; - pPreset->m_nEarlyL_LpfFbk = 3692; - pPreset->m_nEarlyL_LpfFwd = 29075; - pPreset->m_nEarlyL_Delay0 = 922; - pPreset->m_nEarlyL_Gain0 = 22152; - pPreset->m_nEarlyL_Delay1 = 1462; - pPreset->m_nEarlyL_Gain1 = 17537; - pPreset->m_nEarlyL_Delay2 = 0; - pPreset->m_nEarlyL_Gain2 = 14768; - pPreset->m_nEarlyL_Delay3 = 1221; - pPreset->m_nEarlyL_Gain3 = 14307; - pPreset->m_nEarlyL_Delay4 = 0; - pPreset->m_nEarlyL_Gain4 = 13384; - pPreset->m_nEarlyR_Delay0 = 502; - pPreset->m_nEarlyR_Gain0 = 20306; - pPreset->m_nEarlyR_Delay1 = 1762; - pPreset->m_nEarlyR_Gain1 = 17537; - pPreset->m_nEarlyR_Delay2 = 0; - pPreset->m_nEarlyR_Gain2 = 14768; - pPreset->m_nEarlyR_Delay3 = 0; - pPreset->m_nEarlyR_Gain3 = 16153; - pPreset->m_nEarlyR_Delay4 = 0; - pPreset->m_nEarlyR_Gain4 = 13384; - pPreset->m_nMaxExcursion = 127; - pPreset->m_nXfadeInterval = 6391; - pPreset->m_nAp0_ApGain = 15230; - pPreset->m_nAp0_ApOut = 708; - pPreset->m_nAp1_ApGain = 9692; - pPreset->m_nAp1_ApOut = 1113; - pPreset->m_rfu4 = 0; - pPreset->m_rfu5 = 0; - pPreset->m_rfu6 = 0; - pPreset->m_rfu7 = 0; - pPreset->m_rfu8 = 0; - pPreset->m_rfu9 = 0; - pPreset->m_rfu10 = 0; - } - else if (defaultPreset == 2) - { - pPreset->m_nLpfFbk = 5077; - pPreset->m_nLpfFwd = 12922; - pPreset->m_nEarly = 0; - pPreset->m_nWet = 24460; - pPreset->m_nDry = 32767; - pPreset->m_nEarlyL_LpfFbk = 3692; - pPreset->m_nEarlyL_LpfFwd = 29075; - pPreset->m_nEarlyL_Delay0 = 922; - pPreset->m_nEarlyL_Gain0 = 22152; - pPreset->m_nEarlyL_Delay1 = 1462; - pPreset->m_nEarlyL_Gain1 = 17537; - pPreset->m_nEarlyL_Delay2 = 0; - pPreset->m_nEarlyL_Gain2 = 14768; - pPreset->m_nEarlyL_Delay3 = 1221; - pPreset->m_nEarlyL_Gain3 = 14307; - pPreset->m_nEarlyL_Delay4 = 0; - pPreset->m_nEarlyL_Gain4 = 13384; - pPreset->m_nEarlyR_Delay0 = 502; - pPreset->m_nEarlyR_Gain0 = 20306; - pPreset->m_nEarlyR_Delay1 = 1762; - pPreset->m_nEarlyR_Gain1 = 17537; - pPreset->m_nEarlyR_Delay2 = 0; - pPreset->m_nEarlyR_Gain2 = 14768; - pPreset->m_nEarlyR_Delay3 = 0; - pPreset->m_nEarlyR_Gain3 = 16153; - pPreset->m_nEarlyR_Delay4 = 0; - pPreset->m_nEarlyR_Gain4 = 13384; - pPreset->m_nMaxExcursion = 127; - pPreset->m_nXfadeInterval = 6449; - pPreset->m_nAp0_ApGain = 15691; - pPreset->m_nAp0_ApOut = 774; - pPreset->m_nAp1_ApGain = 15691; - pPreset->m_nAp1_ApOut = 1113; - pPreset->m_rfu4 = 0; - pPreset->m_rfu5 = 0; - pPreset->m_rfu6 = 0; - pPreset->m_rfu7 = 0; - pPreset->m_rfu8 = 0; - pPreset->m_rfu9 = 0; - pPreset->m_rfu10 = 0; - } - else if (defaultPreset == 3) - { - pPreset->m_nLpfFbk = 5077; - pPreset->m_nLpfFwd = 11076; - pPreset->m_nEarly = 0; - pPreset->m_nWet = 23075; - pPreset->m_nDry = 32767; - pPreset->m_nEarlyL_LpfFbk = 3692; - pPreset->m_nEarlyL_LpfFwd = 29075; - pPreset->m_nEarlyL_Delay0 = 922; - pPreset->m_nEarlyL_Gain0 = 22152; - pPreset->m_nEarlyL_Delay1 = 1462; - pPreset->m_nEarlyL_Gain1 = 17537; - pPreset->m_nEarlyL_Delay2 = 0; - pPreset->m_nEarlyL_Gain2 = 14768; - pPreset->m_nEarlyL_Delay3 = 1221; - pPreset->m_nEarlyL_Gain3 = 14307; - pPreset->m_nEarlyL_Delay4 = 0; - pPreset->m_nEarlyL_Gain4 = 13384; - pPreset->m_nEarlyR_Delay0 = 502; - pPreset->m_nEarlyR_Gain0 = 20306; - pPreset->m_nEarlyR_Delay1 = 1762; - pPreset->m_nEarlyR_Gain1 = 17537; - pPreset->m_nEarlyR_Delay2 = 0; - pPreset->m_nEarlyR_Gain2 = 14768; - pPreset->m_nEarlyR_Delay3 = 0; - pPreset->m_nEarlyR_Gain3 = 16153; - pPreset->m_nEarlyR_Delay4 = 0; - pPreset->m_nEarlyR_Gain4 = 13384; - pPreset->m_nMaxExcursion = 127; - pPreset->m_nXfadeInterval = 6470; //6483; - pPreset->m_nAp0_ApGain = 14768; - pPreset->m_nAp0_ApOut = 792; - pPreset->m_nAp1_ApGain = 15783; - pPreset->m_nAp1_ApOut = 1113; - pPreset->m_rfu4 = 0; - pPreset->m_rfu5 = 0; - pPreset->m_rfu6 = 0; - pPreset->m_rfu7 = 0; - pPreset->m_rfu8 = 0; - pPreset->m_rfu9 = 0; - pPreset->m_rfu10 = 0; - - } - } - - return EAS_SUCCESS; -} diff --git a/arm-hybrid-22k/lib_src/eas_reverbdata.c b/arm-hybrid-22k/lib_src/eas_reverbdata.c deleted file mode 100644 index 5d48c1b..0000000 --- a/arm-hybrid-22k/lib_src/eas_reverbdata.c +++ /dev/null @@ -1,34 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_reverbdata.c - * - * Contents and purpose: - * Contains the static data allocation for the Reverb effect - * - * - * Copyright Sonic Network Inc. 2006 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 550 $ - * $Date: 2007-02-02 09:37:03 -0800 (Fri, 02 Feb 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_reverbdata.h" - -S_REVERB_OBJECT eas_ReverbData; - diff --git a/arm-hybrid-22k/lib_src/eas_reverbdata.h b/arm-hybrid-22k/lib_src/eas_reverbdata.h deleted file mode 100644 index ef424da..0000000 --- a/arm-hybrid-22k/lib_src/eas_reverbdata.h +++ /dev/null @@ -1,486 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_reverbdata.h - * - * Contents and purpose: - * Contains the prototypes for the Reverb effect. - * - * - * Copyright Sonic Network Inc. 2006 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 499 $ - * $Date: 2006-12-11 16:07:20 -0800 (Mon, 11 Dec 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_REVERBDATA_H -#define _EAS_REVERBDATA_H - -#include "eas_types.h" -#include "eas_audioconst.h" - -/*------------------------------------ - * defines - *------------------------------------ -*/ - -/* -CIRCULAR() calculates the array index using modulo arithmetic. -The "trick" is that modulo arithmetic is simplified by masking -the effective address where the mask is (2^n)-1. This only works -if the buffer size is a power of two. -*/ -#define CIRCULAR(base,offset,size) (EAS_U32)( \ - ( \ - ((EAS_I32)(base)) + ((EAS_I32)(offset)) \ - ) \ - & size \ - ) - -/* reverb parameters are updated every 2^(REVERB_UPDATE_PERIOD_IN_BITS) samples */ -#if defined (_SAMPLE_RATE_8000) - -#define REVERB_UPDATE_PERIOD_IN_BITS 5 -#define REVERB_BUFFER_SIZE_IN_SAMPLES 2048 - -#elif defined (_SAMPLE_RATE_16000) - -#define REVERB_UPDATE_PERIOD_IN_BITS 6 -#define REVERB_BUFFER_SIZE_IN_SAMPLES 4096 - -#elif defined (_SAMPLE_RATE_22050) - -#define REVERB_UPDATE_PERIOD_IN_BITS 7 -#define REVERB_BUFFER_SIZE_IN_SAMPLES 4096 - -#elif defined (_SAMPLE_RATE_32000) - -#define REVERB_UPDATE_PERIOD_IN_BITS 7 -#define REVERB_BUFFER_SIZE_IN_SAMPLES 8192 - -#elif defined (_SAMPLE_RATE_44100) - -#define REVERB_UPDATE_PERIOD_IN_BITS 8 -#define REVERB_BUFFER_SIZE_IN_SAMPLES 8192 - -#elif defined (_SAMPLE_RATE_48000) - -#define REVERB_UPDATE_PERIOD_IN_BITS 8 -#define REVERB_BUFFER_SIZE_IN_SAMPLES 8192 - -#endif - -// Define a mask for circular addressing, so that array index -// can wraparound and stay in array boundary of 0, 1, ..., (buffer size -1) -// The buffer size MUST be a power of two -#define REVERB_BUFFER_MASK (REVERB_BUFFER_SIZE_IN_SAMPLES -1) - -#define REVERB_MAX_ROOM_TYPE 4 // any room numbers larger than this are invalid -#define REVERB_MAX_NUM_REFLECTIONS 5 // max num reflections per channel - -/* synth parameters are updated every SYNTH_UPDATE_PERIOD_IN_SAMPLES */ -#define REVERB_UPDATE_PERIOD_IN_SAMPLES (EAS_I32)(0x1L << REVERB_UPDATE_PERIOD_IN_BITS) - -/* -calculate the update counter by bitwise ANDING with this value to -generate a 2^n modulo value -*/ -#define REVERB_MODULO_UPDATE_PERIOD_IN_SAMPLES (EAS_I32)(REVERB_UPDATE_PERIOD_IN_SAMPLES -1) - -/* synth parameters are updated every SYNTH_UPDATE_PERIOD_IN_SECONDS seconds */ -#define REVERB_UPDATE_PERIOD_IN_SECONDS (REVERB_UPDATE_PERIOD_IN_SAMPLES / _OUTPUT_SAMPLE_RATE) - -// xfade parameters -#define REVERB_XFADE_PERIOD_IN_SECONDS (100.0 / 1000.0) // xfade once every this many seconds - -#define REVERB_XFADE_PERIOD_IN_SAMPLES (REVERB_XFADE_PERIOD_IN_SECONDS * _OUTPUT_SAMPLE_RATE) - -#define REVERB_XFADE_PHASE_INCREMENT (EAS_I16)(65536 / ((EAS_I16)REVERB_XFADE_PERIOD_IN_SAMPLES/(EAS_I16)REVERB_UPDATE_PERIOD_IN_SAMPLES)) - -/**********/ -/* the entire synth uses various flags in a bit field */ - -/* if flag is set, synth reset has been requested */ -#define REVERB_FLAG_RESET_IS_REQUESTED 0x01 /* bit 0 */ -#define MASK_REVERB_RESET_IS_REQUESTED 0x01 -#define MASK_REVERB_RESET_IS_NOT_REQUESTED (EAS_U32)(~MASK_REVERB_RESET_IS_REQUESTED) - -/* -by default, we always want to update ALL channel parameters -when we reset the synth (e.g., during GM ON) -*/ -#define DEFAULT_REVERB_FLAGS 0x0 - -/* coefficients for generating sin, cos */ -#define REVERB_PAN_G2 4294940151 /* -0.82842712474619 = 2 - 4/sqrt(2) */ -/* -EAS_I32 nPanG1 = +1.0 for sin -EAS_I32 nPanG1 = -1.0 for cos -*/ -#define REVERB_PAN_G0 23170 /* 0.707106781186547 = 1/sqrt(2) */ - -/*************************************************************/ -// define the input injection points -#define GUARD 5 // safety guard of this many samples - -#define MAX_AP_TIME (double) (20.0/1000.0) // delay time in milliseconds -#define MAX_DELAY_TIME (double) (65.0/1000.0) // delay time in milliseconds - -#define MAX_AP_SAMPLES (int)(((double) MAX_AP_TIME) * ((double) _OUTPUT_SAMPLE_RATE)) -#define MAX_DELAY_SAMPLES (int)(((double) MAX_DELAY_TIME) * ((double) _OUTPUT_SAMPLE_RATE)) - -#define AP0_IN 0 -#define AP1_IN (AP0_IN + MAX_AP_SAMPLES + GUARD) -#define DELAY0_IN (AP1_IN + MAX_AP_SAMPLES + GUARD) -#define DELAY1_IN (DELAY0_IN + MAX_DELAY_SAMPLES + GUARD) - -// Define the max offsets for the end points of each section -// i.e., we don't expect a given section's taps to go beyond -// the following limits -#define AP0_OUT (AP0_IN + MAX_AP_SAMPLES -1) -#define AP1_OUT (AP1_IN + MAX_AP_SAMPLES -1) -#define DELAY0_OUT (DELAY0_IN + MAX_DELAY_SAMPLES -1) -#define DELAY1_OUT (DELAY1_IN + MAX_DELAY_SAMPLES -1) - -#define REVERB_DEFAULT_ROOM_NUMBER 1 // default preset number -#define DEFAULT_AP0_LENGTH (int)(((double) (17.0/1000.0)) * ((double) _OUTPUT_SAMPLE_RATE)) -#define DEFAULT_AP0_GAIN 19400 -#define DEFAULT_AP1_LENGTH (int)(((double) (16.5/1000.0)) * ((double) _OUTPUT_SAMPLE_RATE)) -#define DEFAULT_AP1_GAIN -19400 - -#define REVERB_DEFAULT_WET 32767 -#define REVERB_DEFAULT_DRY 0 - -#define EAS_REVERB_WET_MAX 32767 -#define EAS_REVERB_WET_MIN 0 -#define EAS_REVERB_DRY_MAX 32767 -#define EAS_REVERB_DRY_MIN 0 - -/* parameters for each allpass */ -typedef struct -{ - EAS_U16 m_zApOut; // delay offset for ap out - - EAS_I16 m_nApGain; // gain for ap - - EAS_U16 m_zApIn; // delay offset for ap in - -} S_ALLPASS_OBJECT; - - -/* parameters for each allpass */ -typedef struct -{ - EAS_PCM m_zLpf; // actual state variable, not a length - - EAS_I16 m_nLpfFwd; // lpf forward gain - - EAS_I16 m_nLpfFbk; // lpf feedback gain - - EAS_U16 m_zDelay[REVERB_MAX_NUM_REFLECTIONS]; // delay offset for ap out - - EAS_I16 m_nGain[REVERB_MAX_NUM_REFLECTIONS]; // gain for ap - -} S_EARLY_REFLECTION_OBJECT; - -//demo -typedef struct -{ - EAS_I16 m_nLpfFbk; - EAS_I16 m_nLpfFwd; - - EAS_I16 m_nEarly; - EAS_I16 m_nWet; - EAS_I16 m_nDry; - - EAS_I16 m_nEarlyL_LpfFbk; - EAS_I16 m_nEarlyL_LpfFwd; - - EAS_I16 m_nEarlyL_Delay0; //8 - EAS_I16 m_nEarlyL_Gain0; - EAS_I16 m_nEarlyL_Delay1; - EAS_I16 m_nEarlyL_Gain1; - EAS_I16 m_nEarlyL_Delay2; - EAS_I16 m_nEarlyL_Gain2; - EAS_I16 m_nEarlyL_Delay3; - EAS_I16 m_nEarlyL_Gain3; - EAS_I16 m_nEarlyL_Delay4; - EAS_I16 m_nEarlyL_Gain4; - - EAS_I16 m_nEarlyR_Delay0; //18 - EAS_I16 m_nEarlyR_Gain0; - EAS_I16 m_nEarlyR_Delay1; - EAS_I16 m_nEarlyR_Gain1; - EAS_I16 m_nEarlyR_Delay2; - EAS_I16 m_nEarlyR_Gain2; - EAS_I16 m_nEarlyR_Delay3; - EAS_I16 m_nEarlyR_Gain3; - EAS_I16 m_nEarlyR_Delay4; - EAS_I16 m_nEarlyR_Gain4; - - EAS_U16 m_nMaxExcursion; //28 - EAS_I16 m_nXfadeInterval; - - EAS_I16 m_nAp0_ApGain; //30 - EAS_I16 m_nAp0_ApOut; - EAS_I16 m_nAp1_ApGain; - EAS_I16 m_nAp1_ApOut; - - EAS_I16 m_rfu4; - EAS_I16 m_rfu5; - EAS_I16 m_rfu6; - EAS_I16 m_rfu7; - EAS_I16 m_rfu8; - EAS_I16 m_rfu9; - EAS_I16 m_rfu10; //43 - -} S_REVERB_PRESET; - -typedef struct -{ - S_REVERB_PRESET m_sPreset[REVERB_MAX_ROOM_TYPE]; //array of presets - -} S_REVERB_PRESET_BANK; - -/* parameters for each reverb */ -typedef struct -{ - /* controls entire reverb playback volume */ - /* to conserve memory, use the MSB and ignore the LSB */ - EAS_U8 m_nMasterVolume; - - /* update counter keeps track of when synth params need updating */ - /* only needs to be as large as REVERB_UPDATE_PERIOD_IN_SAMPLES */ - EAS_I16 m_nUpdateCounter; - - EAS_U16 m_nMinSamplesToAdd; /* ComputeReverb() generates this many samples */ - - EAS_U8 m_nFlags; /* misc flags/bit fields */ - - EAS_PCM *m_pOutputBuffer; - EAS_PCM *m_pInputBuffer; - - EAS_U16 m_nNumSamplesInOutputBuffer; - EAS_U16 m_nNumSamplesInInputBuffer; - - EAS_U16 m_nNumInputSamplesRead; // if m_nNumInputSamplesRead >= NumSamplesInInputBuffer - // then get a new input buffer - EAS_PCM *m_pNextInputSample; - - EAS_U16 m_nBaseIndex; // base index for circular buffer - - // reverb delay line offsets, allpass parameters, etc: - - EAS_PCM m_nRevOutFbkR; // combine feedback reverb right out with dry left in - - S_ALLPASS_OBJECT m_sAp0; // allpass 0 (left channel) - - EAS_U16 m_zD0In; // delay offset for delay line D0 in - - EAS_PCM m_nRevOutFbkL; // combine feedback reverb left out with dry right in - - S_ALLPASS_OBJECT m_sAp1; // allpass 1 (right channel) - - EAS_U16 m_zD1In; // delay offset for delay line D1 in - - // delay output taps, notice criss cross order - EAS_U16 m_zD0Self; // self feeds forward d0 --> d0 - - EAS_U16 m_zD1Cross; // cross feeds across d1 --> d0 - - EAS_PCM m_zLpf0; // actual state variable, not a length - - EAS_U16 m_zD1Self; // self feeds forward d1 --> d1 - - EAS_U16 m_zD0Cross; // cross feeds across d0 --> d1 - - EAS_PCM m_zLpf1; // actual state variable, not a length - - EAS_I16 m_nSin; // gain for self taps - - EAS_I16 m_nCos; // gain for cross taps - - EAS_I16 m_nSinIncrement; // increment for gain - - EAS_I16 m_nCosIncrement; // increment for gain - - EAS_I16 m_nLpfFwd; // lpf forward gain (includes scaling for mixer) - - EAS_I16 m_nLpfFbk; // lpf feedback gain - - EAS_U16 m_nXfadeInterval; // update/xfade after this many samples - - EAS_U16 m_nXfadeCounter; // keep track of when to xfade - - EAS_I16 m_nPhase; // -1 <= m_nPhase < 1 - // but during sin,cos calculations - // use m_nPhase/2 - - EAS_I16 m_nPhaseIncrement; // add this to m_nPhase each frame - - EAS_I16 m_nNoise; // random noise sample - - EAS_U16 m_nMaxExcursion; // the taps can excurse +/- this amount - - EAS_BOOL m_bUseNoise; // if EAS_TRUE, use noise as input signal - - EAS_BOOL m_bBypass; // if EAS_TRUE, then bypass reverb and copy input to output - - EAS_I16 m_nCurrentRoom; // preset number for current room - - EAS_I16 m_nNextRoom; // preset number for next room - - EAS_I16 m_nWet; // gain for wet (processed) signal - - EAS_I16 m_nDry; // gain for dry (unprocessed) signal - - EAS_I16 m_nEarly; // gain for early (widen) signal - - S_EARLY_REFLECTION_OBJECT m_sEarlyL; // left channel early reflections - S_EARLY_REFLECTION_OBJECT m_sEarlyR; // right channel early reflections - - EAS_PCM m_nDelayLine[REVERB_BUFFER_SIZE_IN_SAMPLES]; // one large delay line for all reverb elements - - S_REVERB_PRESET pPreset; - - S_REVERB_PRESET_BANK m_sPreset; - - //EAS_I8 preset; - -} S_REVERB_OBJECT; - - -/*------------------------------------ - * prototypes - *------------------------------------ -*/ - -/*---------------------------------------------------------------------------- - * ReverbUpdateXfade - *---------------------------------------------------------------------------- - * Purpose: - * Update the xfade parameters as required - * - * Inputs: - * nNumSamplesToAdd - number of samples to write to buffer - * - * Outputs: - * - * - * Side Effects: - * - xfade parameters will be changed - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ReverbUpdateXfade(S_REVERB_OBJECT* pReverbData, EAS_INT nNumSamplesToAdd); - -/*---------------------------------------------------------------------------- - * ReverbCalculateNoise - *---------------------------------------------------------------------------- - * Purpose: - * Calculate a noise sample and limit its value - * - * Inputs: - * nMaxExcursion - noise value is limited to this value - * pnNoise - return new noise sample in this (not limited) - * - * Outputs: - * new limited noise value - * - * Side Effects: - * - *pnNoise noise value is updated - * - *---------------------------------------------------------------------------- -*/ -static EAS_U16 ReverbCalculateNoise(EAS_U16 nMaxExcursion, EAS_I16 *pnNoise); - -/*---------------------------------------------------------------------------- - * ReverbCalculateSinCos - *---------------------------------------------------------------------------- - * Purpose: - * Calculate a new sin and cosine value based on the given phase - * - * Inputs: - * nPhase - phase angle - * pnSin - input old value, output new value - * pnCos - input old value, output new value - * - * Outputs: - * - * Side Effects: - * - *pnSin, *pnCos are updated - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ReverbCalculateSinCos(EAS_I16 nPhase, EAS_I16 *pnSin, EAS_I16 *pnCos); - -/*---------------------------------------------------------------------------- - * Reverb - *---------------------------------------------------------------------------- - * Purpose: - * apply reverb to the given signal - * - * Inputs: - * nNu - * pnSin - input old value, output new value - * pnCos - input old value, output new value - * - * Outputs: - * number of samples actually reverberated - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT Reverb(S_REVERB_OBJECT* pReverbData, EAS_INT nNumSamplesToAdd, EAS_PCM *pOutputBuffer, EAS_PCM *pInputBuffer); - -/*---------------------------------------------------------------------------- - * ReverbReadInPresets() - *---------------------------------------------------------------------------- - * Purpose: sets global reverb preset bank to defaults - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ReverbReadInPresets(S_REVERB_OBJECT* pReverbData); - - -/*---------------------------------------------------------------------------- - * ReverbUpdateRoom - *---------------------------------------------------------------------------- - * Purpose: - * Update the room's preset parameters as required - * - * Inputs: - * - * Outputs: - * - * - * Side Effects: - * - reverb paramters (fbk, fwd, etc) will be changed - * - m_nCurrentRoom := m_nNextRoom - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT ReverbUpdateRoom(S_REVERB_OBJECT* pReverbData); - -#endif /* #ifndef _EAS_REVERBDATA_H */ - - diff --git a/arm-hybrid-22k/lib_src/eas_rtttl.c b/arm-hybrid-22k/lib_src/eas_rtttl.c deleted file mode 100644 index 486ad60..0000000 --- a/arm-hybrid-22k/lib_src/eas_rtttl.c +++ /dev/null @@ -1,1197 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_rtttl.c - * - * Contents and purpose: - * RTTTL parser - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 795 $ - * $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_data.h" -#include "eas_miditypes.h" -#include "eas_parser.h" -#include "eas_report.h" -#include "eas_host.h" -#include "eas_midi.h" -#include "eas_config.h" -#include "eas_vm_protos.h" -#include "eas_rtttldata.h" -#include "eas_ctype.h" - -/* increase gain for mono ringtones */ -#define RTTTL_GAIN_OFFSET 8 - -/* maximum title length including colon separator */ -#define RTTTL_MAX_TITLE_LEN 32 -#define RTTTL_INFINITE_LOOP 15 - -/* length of 32nd note in 1/256ths of a msec for 63 BPM tempo */ -#define DEFAULT_TICK_CONV 30476 -#define TICK_CONVERT 1920000 - -/* default channel and program for RTTTL playback */ -#define RTTTL_CHANNEL 0 -#define RTTTL_PROGRAM 80 -#define RTTTL_VELOCITY 127 - -/* note used for rest */ -#define RTTTL_REST 1 - -/* multiplier for fixed point triplet conversion */ -#define TRIPLET_MULTIPLIER 683 -#define TRIPLET_SHIFT 10 - -/* local prototypes */ -static EAS_RESULT RTTTL_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset); -static EAS_RESULT RTTTL_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT RTTTL_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime); -static EAS_RESULT RTTTL_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode); -static EAS_RESULT RTTTL_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState); -static EAS_RESULT RTTTL_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT RTTTL_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT RTTTL_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT RTTTL_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT RTTTL_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value); -static EAS_RESULT RTTTL_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue); -static EAS_RESULT RTTTL_GetStyle (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData); -static EAS_RESULT RTTTL_GetDuration (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I8 *pDuration); -static EAS_RESULT RTTTL_GetOctave (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_U8 *pOctave); -static EAS_RESULT RTTTL_GetTempo (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData); -static EAS_RESULT RTTTL_GetNumber (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I32 *pValue); -static EAS_RESULT RTTTL_ParseHeader (S_EAS_DATA *pEASData, S_RTTTL_DATA* pData, EAS_BOOL metaData); -static EAS_RESULT RTTTL_GetNextChar (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I8 *pValue); -static EAS_RESULT RTTTL_PeekNextChar (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I8 *pValue); - -/* inline functions */ -EAS_INLINE void RTTTL_PutBackChar (S_RTTTL_DATA *pData, EAS_I8 value) { pData->dataByte = value; } - - -/* lookup table for note values */ -static const EAS_U8 noteTable[] = { 21, 23, 12, 14, 16, 17, 19, 23 }; - -/*---------------------------------------------------------------------------- - * - * EAS_RTTTL_Parser - * - * This structure contains the functional interface for the iMelody parser - *---------------------------------------------------------------------------- -*/ -const S_FILE_PARSER_INTERFACE EAS_RTTTL_Parser = -{ - RTTTL_CheckFileType, - RTTTL_Prepare, - RTTTL_Time, - RTTTL_Event, - RTTTL_State, - RTTTL_Close, - RTTTL_Reset, - RTTTL_Pause, - RTTTL_Resume, - NULL, - RTTTL_SetData, - RTTTL_GetData, - NULL -}; - -/*---------------------------------------------------------------------------- - * RTTTL_CheckFileType() - *---------------------------------------------------------------------------- - * Purpose: - * Check the file type to see if we can parse it - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT RTTTL_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset) -{ - S_RTTTL_DATA data; - S_RTTTL_DATA *pData; - - /* see if we can parse the header */ - data.fileHandle = fileHandle; - data.fileOffset = offset; - *ppHandle= NULL; - if (RTTTL_ParseHeader (pEASData, &data, EAS_FALSE) == EAS_SUCCESS) - { - - /* check for static memory allocation */ - if (pEASData->staticMemoryModel) - pData = EAS_CMEnumData(EAS_CM_RTTTL_DATA); - else - pData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_RTTTL_DATA)); - if (!pData) - return EAS_ERROR_MALLOC_FAILED; - EAS_HWMemSet(pData, 0, sizeof(S_RTTTL_DATA)); - - /* return a pointer to the instance data */ - pData->fileHandle = fileHandle; - pData->fileOffset = offset; - pData->state = EAS_STATE_OPEN; - *ppHandle = pData; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_Prepare() - *---------------------------------------------------------------------------- - * Purpose: - * Prepare to parse the file. Allocates instance data (or uses static allocation for - * static memory model). - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT RTTTL_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_RTTTL_DATA* pData; - EAS_RESULT result; - - /* check for valid state */ - pData = (S_RTTTL_DATA*) pInstData; - if (pData->state != EAS_STATE_OPEN) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - - /* instantiate a synthesizer */ - if ((result = VMInitMIDI(pEASData, &pData->pSynth)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitMIDI returned %d\n", result); */ } - return result; - } - - pData->state = EAS_STATE_ERROR; - if ((result = RTTTL_ParseHeader (pEASData, pData, (EAS_BOOL) (pData->metadata.callback != NULL))) != EAS_SUCCESS) - { - /* if using dynamic memory, free it */ - if (!pEASData->staticMemoryModel) - EAS_HWFree(pEASData->hwInstData, pData); - return result; - } - - pData->state = EAS_STATE_READY; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_Time() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the time of the next event in msecs - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * pTime - pointer to variable to hold time of next event (in msecs) - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -static EAS_RESULT RTTTL_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime) -{ - S_RTTTL_DATA *pData; - - pData = (S_RTTTL_DATA*) pInstData; - - /* return time in milliseconds */ - /*lint -e{704} use shift instead of division */ - *pTime = pData->time >> 8; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_Event() - *---------------------------------------------------------------------------- - * Purpose: - * Parse the next event in the file - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT RTTTL_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode) -{ - S_RTTTL_DATA* pData; - EAS_RESULT result; - EAS_I32 ticks; - EAS_I32 temp; - EAS_I8 c; - EAS_U8 note; - EAS_U8 octave; - - pData = (S_RTTTL_DATA*) pInstData; - if (pData->state >= EAS_STATE_OPEN) - return EAS_SUCCESS; - - /* initialize MIDI channel when the track starts playing */ - if (pData->time == 0) - { - /* set program to square lead */ - VMProgramChange(pEASData->pVoiceMgr, pData->pSynth, RTTTL_CHANNEL, RTTTL_PROGRAM); - - /* set channel volume to max */ - VMControlChange(pEASData->pVoiceMgr, pData->pSynth, RTTTL_CHANNEL, 7, 127); - } - - /* check for end of note */ - if (pData->note) - { - /* stop the note */ - VMStopNote(pEASData->pVoiceMgr, pData->pSynth, RTTTL_CHANNEL, pData->note, 0); - pData->note = 0; - - /* check for rest between notes */ - if (pData->restTicks) - { - pData->time += pData->restTicks; - pData->restTicks = 0; - return EAS_SUCCESS; - } - } - - /* parse the next event */ - octave = pData->octave; - note = 0; - ticks = pData->duration * pData->tick; - for (;;) - { - - /* get next character */ - if ((result = RTTTL_GetNextChar(pEASData->hwInstData, pData, &c)) != EAS_SUCCESS) - { - if (result != EAS_EOF) - return result; - - /* end of file, if no notes to process, check for looping */ - if (!note) - { - /* if no loop set state to stopping */ - if (pData->repeatCount == 0) - { - pData->state = EAS_STATE_STOPPING; - VMReleaseAllVoices(pEASData->pVoiceMgr, pData->pSynth); - return EAS_SUCCESS; - } - - /* decrement loop count */ - if (pData->repeatCount != RTTTL_INFINITE_LOOP) - pData->repeatCount--; - - /* if locating, ignore infinite loops */ - else if (parserMode != eParserModePlay) - { - pData->state = EAS_STATE_STOPPING; - VMReleaseAllVoices(pEASData->pVoiceMgr, pData->pSynth); - return EAS_SUCCESS; - } - - /* loop back to start of notes */ - if ((result = EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->repeatOffset)) != EAS_SUCCESS) - return result; - continue; - } - - /* still have a note to process */ - else - c = ','; - } - - /* bpm */ - if (c == 'b') - { - /* peek at next character */ - if ((result = RTTTL_PeekNextChar(pEASData->hwInstData, pData, &c)) != EAS_SUCCESS) - return result; - - /* if a number, must be octave or tempo */ - if (IsDigit(c)) - { - if ((result = RTTTL_GetNumber(pEASData->hwInstData, pData, &temp)) != EAS_SUCCESS) - return result; - - /* check for octave first */ - if ((temp >= 4) && (temp <= 7)) - { - octave = (EAS_U8) temp; - } - - /* check for tempo */ - else if ((temp >= 25) && (temp <= 900)) - { - pData->tick = TICK_CONVERT / (EAS_U32) temp; - } - - /* don't know what it was */ - else - return EAS_ERROR_FILE_FORMAT; - } - - /* must be a note */ - else - { - note = noteTable[1]; - } - } - - /* octave */ - else if (c == 'o') - { - if ((result = RTTTL_GetOctave(pEASData->hwInstData, pData, &pData->octave)) != EAS_SUCCESS) - return result; - } - - /* style */ - else if (c == 's') - { - if ((result = RTTTL_GetStyle(pEASData->hwInstData, pData)) != EAS_SUCCESS) - return result; - } - - /* duration or octave */ - else if (IsDigit(c)) - { - RTTTL_PutBackChar(pData, c); - - /* duration comes before note */ - if (!note) - { - if ((result = RTTTL_GetDuration(pEASData->hwInstData, pData, &c)) != EAS_SUCCESS) - return result; - ticks = c * pData->tick; - } - - /* octave comes after note */ - else - { - if ((result = RTTTL_GetOctave(pEASData->hwInstData, pData, &octave)) != EAS_SUCCESS) - return result; - } - } - - /* note or rest */ - else if ((c >= 'a') && (c <= 'h')) - { - note = noteTable[c - 'a']; - } - - else if (c == 'p') - { - note = RTTTL_REST; - } - - /* dotted note */ - else if (c == '.') - { - /*lint -e{704} shift for performance */ - ticks += ticks >> 1; - } - - /* accidental */ - else if (c == '#') - { - if (note) - note++; - } - - /* end of event */ - else if ((c == ',') && note) - { - - /* handle note events */ - if (note != RTTTL_REST) - { - - /* save note and start it */ - pData->note = note + octave; - if (parserMode == eParserModePlay) - VMStartNote(pEASData->pVoiceMgr, pData->pSynth, RTTTL_CHANNEL, pData->note, RTTTL_VELOCITY); - - /* determine note length */ - switch (pData->style) - { - /* natural */ - case 'n': - /*lint -e{704} shift for performance */ - pData->restTicks = ticks >> 4; - break; - /* continuous */ - - case 'c': - pData->restTicks = 0; - break; - - /* staccato */ - case 's': - /*lint -e{704} shift for performance */ - pData->restTicks = ticks >> 1; - break; - - default: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "RTTTL_Event: Unexpected style type %c\n", pData->style); */ } - break; - } - - /* next event is at end of this note */ - pData->time += ticks - pData->restTicks; - } - - /* rest */ - else - pData->time += ticks; - - /* event found, return to caller */ - break; - } - } - - pData->state = EAS_STATE_PLAY; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_State() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current state of the stream - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * pState - pointer to variable to store state - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -static EAS_RESULT RTTTL_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pState) -{ - S_RTTTL_DATA* pData; - - /* establish pointer to instance data */ - pData = (S_RTTTL_DATA*) pInstData; - - /* if stopping, check to see if synth voices are active */ - if (pData->state == EAS_STATE_STOPPING) - { - if (VMActiveVoices(pData->pSynth) == 0) - pData->state = EAS_STATE_STOPPED; - } - - if (pData->state == EAS_STATE_PAUSING) - { - if (VMActiveVoices(pData->pSynth) == 0) - pData->state = EAS_STATE_PAUSED; - } - - /* return current state */ - *pState = pData->state; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_Close() - *---------------------------------------------------------------------------- - * Purpose: - * Close the file and clean up - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT RTTTL_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_RTTTL_DATA* pData; - EAS_RESULT result; - - pData = (S_RTTTL_DATA*) pInstData; - - /* close the file */ - if ((result = EAS_HWCloseFile(pEASData->hwInstData, pData->fileHandle)) != EAS_SUCCESS) - return result; - - /* free the synth */ - if (pData->pSynth != NULL) - VMMIDIShutdown(pEASData, pData->pSynth); - - /* if using dynamic memory, free it */ - if (!pEASData->staticMemoryModel) - EAS_HWFree(pEASData->hwInstData, pData); - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_Reset() - *---------------------------------------------------------------------------- - * Purpose: - * Reset the sequencer. Used for locating backwards in the file. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT RTTTL_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_RTTTL_DATA* pData; - EAS_RESULT result; - - pData = (S_RTTTL_DATA*) pInstData; - - /* reset the synth */ - VMReset(pEASData->pVoiceMgr, pData->pSynth, EAS_TRUE); - - /* reset time to zero */ - pData->time = 0; - pData->note = 0; - - /* reset file position and re-parse header */ - pData->state = EAS_STATE_ERROR; - if ((result = EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->fileOffset)) != EAS_SUCCESS) - return result; - if ((result = RTTTL_ParseHeader (pEASData, pData, EAS_TRUE)) != EAS_SUCCESS) - return result; - - pData->state = EAS_STATE_READY; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_Pause() - *---------------------------------------------------------------------------- - * Purpose: - * Pauses the sequencer. Mutes all voices and sets state to pause. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT RTTTL_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_RTTTL_DATA *pData; - - /* can't pause a stopped stream */ - pData = (S_RTTTL_DATA*) pInstData; - if (pData->state == EAS_STATE_STOPPED) - return EAS_ERROR_ALREADY_STOPPED; - - /* mute the synthesizer */ - VMMuteAllVoices(pEASData->pVoiceMgr, pData->pSynth); - pData->state = EAS_STATE_PAUSING; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_Resume() - *---------------------------------------------------------------------------- - * Purpose: - * Resume playing after a pause, sets state back to playing. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -static EAS_RESULT RTTTL_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_RTTTL_DATA *pData; - - /* can't resume a stopped stream */ - pData = (S_RTTTL_DATA*) pInstData; - if (pData->state == EAS_STATE_STOPPED) - return EAS_ERROR_ALREADY_STOPPED; - - /* nothing to do but resume playback */ - pData->state = EAS_STATE_PLAY; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_SetData() - *---------------------------------------------------------------------------- - * Purpose: - * Return file type - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -static EAS_RESULT RTTTL_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value) -{ - S_RTTTL_DATA *pData; - - pData = (S_RTTTL_DATA *) pInstData; - switch (param) - { - - /* set metadata callback */ - case PARSER_DATA_METADATA_CB: - EAS_HWMemCpy(&pData->metadata, (void*) value, sizeof(S_METADATA_CB)); - break; - - default: - return EAS_ERROR_INVALID_PARAMETER; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_GetData() - *---------------------------------------------------------------------------- - * Purpose: - * Return file type - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -static EAS_RESULT RTTTL_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue) -{ - S_RTTTL_DATA *pData; - - pData = (S_RTTTL_DATA *) pInstData; - switch (param) - { - /* return file type as RTTTL */ - case PARSER_DATA_FILE_TYPE: - *pValue = EAS_FILE_RTTTL; - break; - -#if 0 - /* set transposition */ - case PARSER_DATA_TRANSPOSITION: - *pValue = pData->transposition; - break; -#endif - - case PARSER_DATA_SYNTH_HANDLE: - *pValue = (EAS_I32) pData->pSynth; - break; - - case PARSER_DATA_GAIN_OFFSET: - *pValue = RTTTL_GAIN_OFFSET; - break; - - default: - return EAS_ERROR_INVALID_PARAMETER; - } - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_GetStyle() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT RTTTL_GetStyle (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData) -{ - EAS_RESULT result; - EAS_I8 style; - - /* get style */ - if ((result = RTTTL_GetNextChar(hwInstData, pData, &style)) != EAS_SUCCESS) - return result; - - if ((style != 's') && (style != 'n') && (style != 'c')) - return EAS_ERROR_FILE_FORMAT; - - pData->style = style; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_GetDuration() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT RTTTL_GetDuration (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I8 *pDuration) -{ - EAS_RESULT result; - EAS_I32 duration; - EAS_I8 temp; - - /* get the duration */ - if ((result = RTTTL_GetNumber(hwInstData, pData, &duration)) != EAS_SUCCESS) - return result; - - if ((duration != 1) && (duration != 2) && (duration != 4) && (duration != 8) && (duration != 16) && (duration != 32)) - return EAS_ERROR_FILE_FORMAT; - - temp = 64; - while (duration) - { - /*lint -e{704} shift for performance */ - duration = duration >> 1; - /*lint -e{702} use shift for performance */ - temp = temp >> 1; - } - - *pDuration = temp; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_GetOctave() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT RTTTL_GetOctave (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_U8 *pOctave) -{ - EAS_RESULT result; - EAS_I32 octave; - - /* get the tempo */ - if ((result = RTTTL_GetNumber(hwInstData, pData, &octave)) != EAS_SUCCESS) - return result; - - if ((octave < 4) || (octave > 7)) - return EAS_ERROR_FILE_FORMAT; - - *pOctave = (EAS_U8) (octave * 12); - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_GetTempo() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT RTTTL_GetTempo (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData) -{ - EAS_RESULT result; - EAS_I32 tempo; - - /* get the tempo */ - if ((result = RTTTL_GetNumber(hwInstData, pData, &tempo)) != EAS_SUCCESS) - return result; - - if ((tempo < 25) || (tempo > 900)) - return EAS_ERROR_FILE_FORMAT; - - pData->tick = TICK_CONVERT / (EAS_U32) tempo; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_GetNumber() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT RTTTL_GetNumber (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I32 *pValue) -{ - EAS_RESULT result; - EAS_INT temp; - EAS_I8 c; - - *pValue = -1; - temp = 0; - for (;;) - { - if ((result = RTTTL_PeekNextChar(hwInstData, pData, &c)) != EAS_SUCCESS) - { - if ((result == EAS_EOF) && (*pValue != -1)) - return EAS_SUCCESS; - return result; - } - - if (IsDigit(c)) - { - pData->dataByte = 0; - temp = temp * 10 + c - '0'; - *pValue = temp; - } - else - return EAS_SUCCESS; - } -} - -/*---------------------------------------------------------------------------- - * RTTTL_ParseHeader() - *---------------------------------------------------------------------------- - * Purpose: - * Prepare to parse the file. Allocates instance data (or uses static allocation for - * static memory model). - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT RTTTL_ParseHeader (S_EAS_DATA *pEASData, S_RTTTL_DATA* pData, EAS_BOOL metaData) -{ - EAS_RESULT result; - EAS_I32 i; - EAS_I8 temp; - EAS_I8 control; - - /* initialize some defaults */ - pData->time = 0; - pData->tick = DEFAULT_TICK_CONV; - pData->note = 0; - pData->duration = 4; - pData ->restTicks = 0; - pData->octave = 60; - pData->repeatOffset = -1; - pData->repeatCount = 0; - pData->style = 'n'; - pData->dataByte = 0; - - metaData = metaData && (pData->metadata.buffer != NULL) && (pData->metadata.callback != NULL); - - /* seek to start of data */ - if ((result = EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->fileOffset)) != EAS_SUCCESS) - return result; - - /* zero the metadata buffer */ - if (metaData) - EAS_HWMemSet(pData->metadata.buffer, 0, pData->metadata.bufferSize); - - /* read the title */ - for (i = 0; i < RTTTL_MAX_TITLE_LEN; i++) - { - if ((result = EAS_HWGetByte(pEASData->hwInstData, pData->fileHandle, &temp)) != EAS_SUCCESS) - return result; - - if (temp == ':') - break; - - /* pass along metadata */ - if (metaData) - { - if (i < (pData->metadata.bufferSize- 1)) - pData->metadata.buffer[i] = (char) temp; - } - } - - /* check for error in title */ - if (i == RTTTL_MAX_TITLE_LEN) - return EAS_ERROR_FILE_FORMAT; - - /* pass along metadata */ - if (metaData) - (*pData->metadata.callback)(EAS_METADATA_TITLE, pData->metadata.buffer, pData->metadata.pUserData); - - /* control fields */ - for (;;) - { - - /* get control type */ - if ((result = RTTTL_GetNextChar(pEASData->hwInstData, pData, &control)) != EAS_SUCCESS) - return result; - - /* next char should be equal sign */ - if ((result = RTTTL_GetNextChar(pEASData->hwInstData, pData, &temp)) != EAS_SUCCESS) - return result; - if (temp != '=') - return EAS_ERROR_FILE_FORMAT; - - /* get the control value */ - switch (control) - { - - /* bpm */ - case 'b': - if ((result = RTTTL_GetTempo(pEASData->hwInstData, pData)) != EAS_SUCCESS) - return result; - break; - - /* duration */ - case 'd': - if ((result = RTTTL_GetDuration(pEASData->hwInstData, pData, &temp)) != EAS_SUCCESS) - return result; - pData->duration = temp; - break; - - /* loop */ - case 'l': - if ((result = RTTTL_GetNumber(pEASData->hwInstData, pData, &i)) != EAS_SUCCESS) - return result; - if ((i < 0) || (i > 15)) - return EAS_ERROR_FILE_FORMAT; - pData->repeatCount = (EAS_U8) i; - break; - - /* octave */ - case 'o': - if ((result = RTTTL_GetOctave(pEASData->hwInstData, pData, &pData->octave)) != EAS_SUCCESS) - return result; - break; - - /* get style */ - case 's': - if ((result = RTTTL_GetStyle(pEASData->hwInstData, pData)) != EAS_SUCCESS) - return result; - break; - - /* unrecognized control */ - default: - return EAS_ERROR_FILE_FORMAT; - } - - /* next character should be comma or colon */ - if ((result = RTTTL_GetNextChar(pEASData->hwInstData, pData, &temp)) != EAS_SUCCESS) - return result; - - /* check for end of control field */ - if (temp == ':') - break; - - /* must be a comma */ - if (temp != ',') - return EAS_ERROR_FILE_FORMAT; - } - - /* should be at the start of the music block */ - if ((result = EAS_HWFilePos(pEASData->hwInstData, pData->fileHandle, &pData->repeatOffset)) != EAS_SUCCESS) - return result; - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * RTTTL_GetNextChar() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT RTTTL_GetNextChar (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I8 *pValue) -{ - EAS_RESULT result; - EAS_I8 temp; - - *pValue = 0; - for(;;) - { - - /* check for character that has been put back */ - if (pData->dataByte) - { - temp = pData->dataByte; - pData->dataByte = 0; - } - else - { - if ((result = EAS_HWGetByte(hwInstData, pData->fileHandle, &temp)) != EAS_SUCCESS) - return result; - } - - /* ignore white space */ - if (!IsSpace(temp)) - { - *pValue = ToLower(temp); - return EAS_SUCCESS; - } - } -} - -/*---------------------------------------------------------------------------- - * RTTTL_PeekNextChar() - *---------------------------------------------------------------------------- - * Purpose: - * - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT RTTTL_PeekNextChar (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I8 *pValue) -{ - EAS_RESULT result; - EAS_I8 temp; - - *pValue = 0; - for(;;) - { - - /* read a character from the file, if necessary */ - if (!pData->dataByte) - { - if ((result = EAS_HWGetByte(hwInstData, pData->fileHandle, &pData->dataByte)) != EAS_SUCCESS) - return result; - - } - temp = pData->dataByte; - - /* ignore white space */ - if (!IsSpace(temp)) - { - *pValue = ToLower(temp); - return EAS_SUCCESS; - } - pData->dataByte = 0; - } -} - diff --git a/arm-hybrid-22k/lib_src/eas_rtttldata.c b/arm-hybrid-22k/lib_src/eas_rtttldata.c deleted file mode 100644 index 7a500bd..0000000 --- a/arm-hybrid-22k/lib_src/eas_rtttldata.c +++ /dev/null @@ -1,41 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_rtttldata.c - * - * Contents and purpose: - * RTTTL File Parser data module for static memory models - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 547 $ - * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_types.h" -#include "eas_rtttldata.h" - -/*---------------------------------------------------------------------------- - * - * eas_RTTTLData - * - * Static memory allocation for RTTTL parser - *---------------------------------------------------------------------------- -*/ -S_RTTTL_DATA eas_RTTTLData; - diff --git a/arm-hybrid-22k/lib_src/eas_rtttldata.h b/arm-hybrid-22k/lib_src/eas_rtttldata.h deleted file mode 100644 index bf4c38b..0000000 --- a/arm-hybrid-22k/lib_src/eas_rtttldata.h +++ /dev/null @@ -1,70 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_rtttldata.h - * - * Contents and purpose: - * SMF File Parser - * - * This file contains data declarations for the RTTTL parser. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef EAS_RTTTLDATA_H -#define EAS_RTTTLDATA_H - -#include "eas_data.h" - - -/* maximum line size as specified in iMelody V1.2 spec */ -#define MAX_LINE_SIZE 75 - -/*---------------------------------------------------------------------------- - * - * S_RTTTL_DATA - * - * This structure contains the state data for the iMelody parser - *---------------------------------------------------------------------------- -*/ - -typedef struct -{ - EAS_FILE_HANDLE fileHandle; /* file handle */ - S_SYNTH *pSynth; /* synthesizer handle */ - S_METADATA_CB metadata; /* metadata callback */ - EAS_I32 fileOffset; /* offset to start of data */ - EAS_I32 time; /* current time in 256ths of a msec */ - EAS_I32 tick; /* length of 32nd note in 256th of a msec */ - EAS_I32 restTicks; /* ticks to rest after current note */ - EAS_I32 repeatOffset; /* file offset to start of repeat section */ - EAS_U8 repeatCount; /* repeat counter */ - EAS_I8 dataByte; /* storage for characters that are "put back" */ - EAS_U8 state; /* current state EAS_STATE_XXXX */ - EAS_I8 style; /* from STYLE */ - EAS_U8 note; /* MIDI note number */ - EAS_U8 octave; /* decault octave prefix */ - EAS_I8 duration; /* default note duration */ -} S_RTTTL_DATA; - -#endif - - diff --git a/arm-hybrid-22k/lib_src/eas_smf.c b/arm-hybrid-22k/lib_src/eas_smf.c deleted file mode 100644 index 7b56e97..0000000 --- a/arm-hybrid-22k/lib_src/eas_smf.c +++ /dev/null @@ -1,1203 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_smf.c - * - * Contents and purpose: - * SMF Type 0 and 1 File Parser - * - * For SMF timebase analysis, see "MIDI Sequencer Analysis.xls". - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 803 $ - * $Date: 2007-08-01 09:57:00 -0700 (Wed, 01 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_data.h" -#include "eas_miditypes.h" -#include "eas_parser.h" -#include "eas_report.h" -#include "eas_host.h" -#include "eas_midi.h" -#include "eas_config.h" -#include "eas_vm_protos.h" -#include "eas_smfdata.h" -#include "eas_smf.h" - -#ifdef JET_INTERFACE -#include "jet_data.h" -#endif - -//3 dls: The timebase for this module is adequate to keep MIDI and -//3 digital audio synchronized for only a few minutes. It should be -//3 sufficient for most mobile applications. If better accuracy is -//3 required, more fractional bits should be added to the timebase. - -static const EAS_U8 smfHeader[] = { 'M', 'T', 'h', 'd' }; - -/* local prototypes */ -static EAS_RESULT SMF_GetVarLenData (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE fileHandle, EAS_U32 *pData); -static EAS_RESULT SMF_ParseMetaEvent (S_EAS_DATA *pEASData, S_SMF_DATA *pSMFData, S_SMF_STREAM *pSMFStream); -static EAS_RESULT SMF_ParseSysEx (S_EAS_DATA *pEASData, S_SMF_DATA *pSMFData, S_SMF_STREAM *pSMFStream, EAS_U8 f0, EAS_INT parserMode); -static EAS_RESULT SMF_ParseEvent (S_EAS_DATA *pEASData, S_SMF_DATA *pSMFData, S_SMF_STREAM *pSMFStream, EAS_INT parserMode); -static EAS_RESULT SMF_GetDeltaTime (EAS_HW_DATA_HANDLE hwInstData, S_SMF_STREAM *pSMFStream); -static void SMF_UpdateTime (S_SMF_DATA *pSMFData, EAS_U32 ticks); - - -/*---------------------------------------------------------------------------- - * - * SMF_Parser - * - * This structure contains the functional interface for the SMF parser - *---------------------------------------------------------------------------- -*/ -const S_FILE_PARSER_INTERFACE EAS_SMF_Parser = -{ - SMF_CheckFileType, - SMF_Prepare, - SMF_Time, - SMF_Event, - SMF_State, - SMF_Close, - SMF_Reset, - SMF_Pause, - SMF_Resume, - NULL, - SMF_SetData, - SMF_GetData, - NULL -}; - -/*---------------------------------------------------------------------------- - * SMF_CheckFileType() - *---------------------------------------------------------------------------- - * Purpose: - * Check the file type to see if we can parse it - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT SMF_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset) -{ - S_SMF_DATA* pSMFData; - EAS_RESULT result; - - /* seek to starting offset - usually 0 */ - *ppHandle = NULL; - if ((result = EAS_HWFileSeek(pEASData->hwInstData, fileHandle, offset)) != EAS_SUCCESS) - return result; - - /* search through file for header - slow method */ - if (pEASData->searchHeaderFlag) - { - result = EAS_SearchFile(pEASData, fileHandle, smfHeader, sizeof(smfHeader), &offset); - if (result != EAS_SUCCESS) - return (result == EAS_EOF) ? EAS_SUCCESS : result; - } - - /* read the first 4 bytes of the file - quick method */ - else { - EAS_U8 header[4]; - EAS_I32 count; - if ((result = EAS_HWReadFile(pEASData->hwInstData, fileHandle, header, sizeof(header), &count)) != EAS_SUCCESS) - return result; - - /* check for 'MTrk' - return if no match */ - if ((header[0] != 'M') || (header[1] != 'T') || (header[2] != 'h') || (header[3] != 'd')) - return EAS_SUCCESS; - } - - /* check for static memory allocation */ - if (pEASData->staticMemoryModel) - pSMFData = EAS_CMEnumData(EAS_CM_SMF_DATA); - else - { - pSMFData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_SMF_DATA)); - EAS_HWMemSet((void *)pSMFData,0, sizeof(S_SMF_DATA)); - } - if (!pSMFData) - return EAS_ERROR_MALLOC_FAILED; - - /* initialize some critical data */ - pSMFData->fileHandle = fileHandle; - pSMFData->fileOffset = offset; - pSMFData->pSynth = NULL; - pSMFData->time = 0; - pSMFData->state = EAS_STATE_OPEN; - *ppHandle = pSMFData; - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_Prepare() - *---------------------------------------------------------------------------- - * Purpose: - * Prepare to parse the file. Allocates instance data (or uses static allocation for - * static memory model). - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT SMF_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_SMF_DATA* pSMFData; - EAS_RESULT result; - - /* check for valid state */ - pSMFData = (S_SMF_DATA *) pInstData; - if (pSMFData->state != EAS_STATE_OPEN) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - - /* instantiate a synthesizer */ - if ((result = VMInitMIDI(pEASData, &pSMFData->pSynth)) != EAS_SUCCESS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitMIDI returned %d\n", result); */ } - return result; - } - - /* parse the file header and setup the individual stream parsers */ - if ((result = SMF_ParseHeader(pEASData->hwInstData, pSMFData)) != EAS_SUCCESS) - return result; - - /* ready to play */ - pSMFData->state = EAS_STATE_READY; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_Time() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the time of the next event in msecs - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * pTime - pointer to variable to hold time of next event (in msecs) - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_RESULT SMF_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime) -{ - S_SMF_DATA *pSMFData; - - pSMFData = (S_SMF_DATA*) pInstData; - - /* sanity check */ -#ifdef _CHECKED_BUILD - if (pSMFData->state == EAS_STATE_STOPPED) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Can't ask for time on a stopped stream\n"); */ } - } - - if (pSMFData->nextStream == NULL) - { - { /* dpp: EAS_ReportEx( _EAS_SEVERITY_ERROR, "no is NULL\n"); */ } - } -#endif - -#if 0 - /* return time in milliseconds */ - /* if chase mode, lie about time */ - if (pSMFData->flags & SMF_FLAGS_CHASE_MODE) - *pTime = 0; - - else -#endif - - /*lint -e{704} use shift instead of division */ - *pTime = pSMFData->time >> 8; - - *pTime = pSMFData->time >> 8; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_Event() - *---------------------------------------------------------------------------- - * Purpose: - * Parse the next event in the file - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT SMF_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode) -{ - S_SMF_DATA* pSMFData; - EAS_RESULT result; - EAS_I32 i; - EAS_U32 ticks; - EAS_U32 temp; - - /* establish pointer to instance data */ - pSMFData = (S_SMF_DATA*) pInstData; - if (pSMFData->state >= EAS_STATE_OPEN) - return EAS_SUCCESS; - - /* get current ticks */ - ticks = pSMFData->nextStream->ticks; - - /* assume that an error occurred */ - pSMFData->state = EAS_STATE_ERROR; - -#ifdef JET_INTERFACE - /* if JET has track muted, set parser mode to mute */ - if (pSMFData->nextStream->midiStream.jetData & MIDI_FLAGS_JET_MUTE) - parserMode = eParserModeMute; -#endif - - /* parse the next event from all the streams */ - if ((result = SMF_ParseEvent(pEASData, pSMFData, pSMFData->nextStream, parserMode)) != EAS_SUCCESS) - { - /* check for unexpected end-of-file */ - if (result != EAS_EOF) - return result; - - /* indicate end of track for this stream */ - pSMFData->nextStream->ticks = SMF_END_OF_TRACK; - } - - /* get next delta time, unless already at end of track */ - else if (pSMFData->nextStream->ticks != SMF_END_OF_TRACK) - { - if ((result = SMF_GetDeltaTime(pEASData->hwInstData, pSMFData->nextStream)) != EAS_SUCCESS) - { - /* check for unexpected end-of-file */ - if (result != EAS_EOF) - return result; - - /* indicate end of track for this stream */ - pSMFData->nextStream->ticks = SMF_END_OF_TRACK; - } - - /* if zero delta to next event, stay with this stream */ - else if (pSMFData->nextStream->ticks == ticks) - { - pSMFData->state = EAS_STATE_PLAY; - return EAS_SUCCESS; - } - } - - /* find next event in all streams */ - temp = 0x7ffffff; - pSMFData->nextStream = NULL; - for (i = 0; i < pSMFData->numStreams; i++) - { - if (pSMFData->streams[i].ticks < temp) - { - temp = pSMFData->streams[i].ticks; - pSMFData->nextStream = &pSMFData->streams[i]; - } - } - - /* are there any more events to parse? */ - if (pSMFData->nextStream) - { - pSMFData->state = EAS_STATE_PLAY; - - /* update the time of the next event */ - SMF_UpdateTime(pSMFData, pSMFData->nextStream->ticks - ticks); - } - else - { - pSMFData->state = EAS_STATE_STOPPING; - VMReleaseAllVoices(pEASData->pVoiceMgr, pSMFData->pSynth); - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_State() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current state of the stream - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * pState - pointer to variable to store state - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_RESULT SMF_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pState) -{ - S_SMF_DATA* pSMFData; - - /* establish pointer to instance data */ - pSMFData = (S_SMF_DATA*) pInstData; - - /* if stopping, check to see if synth voices are active */ - if (pSMFData->state == EAS_STATE_STOPPING) - { - if (VMActiveVoices(pSMFData->pSynth) == 0) - pSMFData->state = EAS_STATE_STOPPED; - } - - if (pSMFData->state == EAS_STATE_PAUSING) - { - if (VMActiveVoices(pSMFData->pSynth) == 0) - pSMFData->state = EAS_STATE_PAUSED; - } - - /* return current state */ - *pState = pSMFData->state; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_Close() - *---------------------------------------------------------------------------- - * Purpose: - * Close the file and clean up - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT SMF_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_SMF_DATA* pSMFData; - EAS_I32 i; - EAS_RESULT result; - - pSMFData = (S_SMF_DATA*) pInstData; - - /* close all the streams */ - for (i = 0; i < pSMFData->numStreams; i++) - { - if (pSMFData->streams[i].fileHandle != NULL) - { - if ((result = EAS_HWCloseFile(pEASData->hwInstData, pSMFData->streams[i].fileHandle)) != EAS_SUCCESS) - return result; - } - } - if (pSMFData->fileHandle != NULL) - if ((result = EAS_HWCloseFile(pEASData->hwInstData, pSMFData->fileHandle)) != EAS_SUCCESS) - return result; - - /* free the synth */ - if (pSMFData->pSynth != NULL) - VMMIDIShutdown(pEASData, pSMFData->pSynth); - - /* if using dynamic memory, free it */ - if (!pEASData->staticMemoryModel) - { - if (pSMFData->streams) - EAS_HWFree(pEASData->hwInstData, pSMFData->streams); - - /* free the instance data */ - EAS_HWFree(pEASData->hwInstData, pSMFData); - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_Reset() - *---------------------------------------------------------------------------- - * Purpose: - * Reset the sequencer. Used for locating backwards in the file. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT SMF_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_SMF_DATA* pSMFData; - EAS_I32 i; - EAS_RESULT result; - EAS_U32 ticks; - - pSMFData = (S_SMF_DATA*) pInstData; - - /* reset time to zero */ - pSMFData->time = 0; - - /* reset the synth */ - VMReset(pEASData->pVoiceMgr, pSMFData->pSynth, EAS_TRUE); - - /* find the start of each track */ - ticks = 0x7fffffffL; - pSMFData->nextStream = NULL; - for (i = 0; i < pSMFData->numStreams; i++) - { - - /* reset file position to first byte of data in track */ - if ((result = EAS_HWFileSeek(pEASData->hwInstData, pSMFData->streams[i].fileHandle, pSMFData->streams[i].startFilePos)) != EAS_SUCCESS) - return result; - - /* initalize some data */ - pSMFData->streams[i].ticks = 0; - - /* initalize the MIDI parser data */ - EAS_InitMIDIStream(&pSMFData->streams[i].midiStream); - - /* parse the first delta time in each stream */ - if ((result = SMF_GetDeltaTime(pEASData->hwInstData,&pSMFData->streams[i])) != EAS_SUCCESS) - return result; - if (pSMFData->streams[i].ticks < ticks) - { - ticks = pSMFData->streams[i].ticks; - pSMFData->nextStream = &pSMFData->streams[i]; - } - } - - - pSMFData->state = EAS_STATE_READY; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_Pause() - *---------------------------------------------------------------------------- - * Purpose: - * Pauses the sequencer. Mutes all voices and sets state to pause. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT SMF_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_SMF_DATA *pSMFData; - - /* can't pause a stopped stream */ - pSMFData = (S_SMF_DATA*) pInstData; - if (pSMFData->state == EAS_STATE_STOPPED) - return EAS_ERROR_ALREADY_STOPPED; - - /* mute the synthesizer */ - VMMuteAllVoices(pEASData->pVoiceMgr, pSMFData->pSynth); - pSMFData->state = EAS_STATE_PAUSING; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_Resume() - *---------------------------------------------------------------------------- - * Purpose: - * Resume playing after a pause, sets state back to playing. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_RESULT SMF_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_SMF_DATA *pSMFData; - - /* can't resume a stopped stream */ - pSMFData = (S_SMF_DATA*) pInstData; - if (pSMFData->state == EAS_STATE_STOPPED) - return EAS_ERROR_ALREADY_STOPPED; - - /* nothing to do but resume playback */ - pSMFData->state = EAS_STATE_PLAY; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_SetData() - *---------------------------------------------------------------------------- - * Purpose: - * Sets parser parameters - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_RESULT SMF_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value) -{ - S_SMF_DATA *pSMFData; - - pSMFData = (S_SMF_DATA*) pInstData; - switch (param) - { - - /* set metadata callback */ - case PARSER_DATA_METADATA_CB: - EAS_HWMemCpy(&pSMFData->metadata, (void*) value, sizeof(S_METADATA_CB)); - break; - -#ifdef JET_INTERFACE - /* set jet segment and track ID of all tracks for callback function */ - case PARSER_DATA_JET_CB: - { - EAS_U32 i; - EAS_U32 bit = (EAS_U32) value; - bit = (bit << JET_EVENT_SEG_SHIFT) & JET_EVENT_SEG_MASK; - for (i = 0; i < pSMFData->numStreams; i++) - pSMFData->streams[i].midiStream.jetData = - (pSMFData->streams[i].midiStream.jetData & - ~(JET_EVENT_TRACK_MASK | JET_EVENT_SEG_MASK)) | - i << JET_EVENT_TRACK_SHIFT | bit | MIDI_FLAGS_JET_CB; - pSMFData->flags |= SMF_FLAGS_JET_STREAM; - } - break; - - /* set state of all mute flags at once */ - case PARSER_DATA_MUTE_FLAGS: - { - EAS_INT i; - EAS_U32 bit = (EAS_U32) value; - for (i = 0; i < pSMFData->numStreams; i++) - { - if (bit & 1) - pSMFData->streams[i].midiStream.jetData |= MIDI_FLAGS_JET_MUTE; - else - pSMFData->streams[i].midiStream.jetData &= ~MIDI_FLAGS_JET_MUTE; - bit >>= 1; - } - } - break; - - /* set track mute */ - case PARSER_DATA_SET_MUTE: - if (value < pSMFData->numStreams) - pSMFData->streams[value].midiStream.jetData |= MIDI_FLAGS_JET_MUTE; - else - return EAS_ERROR_PARAMETER_RANGE; - break; - - /* clear track mute */ - case PARSER_DATA_CLEAR_MUTE: - if (value < pSMFData->numStreams) - pSMFData->streams[value].midiStream.jetData &= ~MIDI_FLAGS_JET_MUTE; - else - return EAS_ERROR_PARAMETER_RANGE; - break; -#endif - - default: - return EAS_ERROR_INVALID_PARAMETER; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_GetData() - *---------------------------------------------------------------------------- - * Purpose: - * Retrieves parser parameters - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -EAS_RESULT SMF_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue) -{ - S_SMF_DATA *pSMFData; - - pSMFData = (S_SMF_DATA*) pInstData; - switch (param) - { - /* return file type */ - case PARSER_DATA_FILE_TYPE: - if (pSMFData->numStreams == 1) - *pValue = EAS_FILE_SMF0; - else - *pValue = EAS_FILE_SMF1; - break; - -/* now handled in eas_public.c */ -#if 0 - case PARSER_DATA_POLYPHONY: - if (pSMFData->pSynth) - VMGetPolyphony(pEASData->pVoiceMgr, pSMFData->pSynth, pValue); - else - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - break; - - case PARSER_DATA_PRIORITY: - if (pSMFData->pSynth) - VMGetPriority(pEASData->pVoiceMgr, pSMFData->pSynth, pValue); - break; - - /* set transposition */ - case PARSER_DATA_TRANSPOSITION: - *pValue = pSMFData->transposition; - break; -#endif - - case PARSER_DATA_SYNTH_HANDLE: - *pValue = (EAS_I32) pSMFData->pSynth; - break; - - default: - return EAS_ERROR_INVALID_PARAMETER; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_GetVarLenData() - *---------------------------------------------------------------------------- - * Purpose: - * Reads a varible length quantity from an SMF file - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT SMF_GetVarLenData (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE fileHandle, EAS_U32 *pData) -{ - EAS_RESULT result; - EAS_U32 data; - EAS_U8 c; - - /* read until bit 7 is zero */ - data = 0; - do - { - if ((result = EAS_HWGetByte(hwInstData, fileHandle,&c)) != EAS_SUCCESS) - return result; - data = (data << 7) | (c & 0x7f); - } while (c & 0x80); - *pData = data; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_GetDeltaTime() - *---------------------------------------------------------------------------- - * Purpose: - * Reads a varible length quantity from an SMF file - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT SMF_GetDeltaTime (EAS_HW_DATA_HANDLE hwInstData, S_SMF_STREAM *pSMFStream) -{ - EAS_RESULT result; - EAS_U32 ticks; - - if ((result = SMF_GetVarLenData(hwInstData, pSMFStream->fileHandle, &ticks)) != EAS_SUCCESS) - return result; - - pSMFStream->ticks += ticks; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_ParseMetaEvent() - *---------------------------------------------------------------------------- - * Purpose: - * Reads a varible length quantity from an SMF file - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT SMF_ParseMetaEvent (S_EAS_DATA *pEASData, S_SMF_DATA *pSMFData, S_SMF_STREAM *pSMFStream) -{ - EAS_RESULT result; - EAS_U32 len; - EAS_I32 pos; - EAS_U32 temp; - EAS_U8 c; - - /* get the meta-event type */ - if ((result = EAS_HWGetByte(pEASData->hwInstData, pSMFStream->fileHandle, &c)) != EAS_SUCCESS) - return result; - - /* get the length */ - if ((result = SMF_GetVarLenData(pEASData->hwInstData, pSMFStream->fileHandle, &len)) != EAS_SUCCESS) - return result; - - /* get the current file position so we can skip the event */ - if ((result = EAS_HWFilePos(pEASData->hwInstData, pSMFStream->fileHandle, &pos)) != EAS_SUCCESS) - return result; - pos += (EAS_I32) len; - - /* end of track? */ - if (c == SMF_META_END_OF_TRACK) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Meta-event: end of track\n", c, len); */ } - pSMFStream->ticks = SMF_END_OF_TRACK; - } - - /* tempo event? */ - else if (c == SMF_META_TEMPO) - { - /* read the 3-byte timebase value */ - temp = 0; - while (len--) - { - if ((result = EAS_HWGetByte(pEASData->hwInstData, pSMFStream->fileHandle, &c)) != EAS_SUCCESS) - return result; - temp = (temp << 8) | c; - } - - pSMFData->tickConv = (EAS_U16) (((temp * 1024) / pSMFData->ppqn + 500) / 1000); - pSMFData->flags |= SMF_FLAGS_HAS_TEMPO; - } - - /* check for time signature - see iMelody spec V1.4 section 4.1.2.2.3.6 */ - else if (c == SMF_META_TIME_SIGNATURE) - { - pSMFData->flags |= SMF_FLAGS_HAS_TIME_SIG; - } - - /* if the host has registered a metadata callback return the metadata */ - else if (pSMFData->metadata.callback) - { - EAS_I32 readLen; - E_EAS_METADATA_TYPE metaType; - - metaType = EAS_METADATA_UNKNOWN; - - /* only process title on the first track */ - if (c == SMF_META_SEQTRK_NAME) - metaType = EAS_METADATA_TITLE; - else if (c == SMF_META_TEXT) - metaType = EAS_METADATA_TEXT; - else if (c == SMF_META_COPYRIGHT) - metaType = EAS_METADATA_COPYRIGHT; - else if (c == SMF_META_LYRIC) - metaType = EAS_METADATA_LYRIC; - - if (metaType != EAS_METADATA_UNKNOWN) - { - readLen = pSMFData->metadata.bufferSize - 1; - if ((EAS_I32) len < readLen) - readLen = (EAS_I32) len; - if ((result = EAS_HWReadFile(pEASData->hwInstData, pSMFStream->fileHandle, pSMFData->metadata.buffer, readLen, &readLen)) != EAS_SUCCESS) - return result; - pSMFData->metadata.buffer[readLen] = 0; - pSMFData->metadata.callback(metaType, pSMFData->metadata.buffer, pSMFData->metadata.pUserData); - } - } - - /* position file to next event - in case we ignored all or part of the meta-event */ - if ((result = EAS_HWFileSeek(pEASData->hwInstData, pSMFStream->fileHandle, pos)) != EAS_SUCCESS) - return result; - - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Meta-event: type=%02x, len=%d\n", c, len); */ } - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_ParseSysEx() - *---------------------------------------------------------------------------- - * Purpose: - * Reads a varible length quantity from an SMF file - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT SMF_ParseSysEx (S_EAS_DATA *pEASData, S_SMF_DATA *pSMFData, S_SMF_STREAM *pSMFStream, EAS_U8 f0, EAS_INT parserMode) -{ - EAS_RESULT result; - EAS_U32 len; - EAS_U8 c; - - /* get the length */ - if ((result = SMF_GetVarLenData(pEASData->hwInstData, pSMFStream->fileHandle, &len)) != EAS_SUCCESS) - return result; - - /* start of SysEx message? */ - if (f0 == 0xf0) - { - if ((result = EAS_ParseMIDIStream(pEASData, pSMFData->pSynth, &pSMFStream->midiStream, f0, parserMode)) != EAS_SUCCESS) - return result; - } - - /* feed the SysEx to the stream parser */ - while (len--) - { - if ((result = EAS_HWGetByte(pEASData->hwInstData, pSMFStream->fileHandle, &c)) != EAS_SUCCESS) - return result; - if ((result = EAS_ParseMIDIStream(pEASData, pSMFData->pSynth, &pSMFStream->midiStream, c, parserMode)) != EAS_SUCCESS) - return result; - - /* check for GM system ON */ - if (pSMFStream->midiStream.flags & MIDI_FLAG_GM_ON) - pSMFData->flags |= SMF_FLAGS_HAS_GM_ON; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_ParseEvent() - *---------------------------------------------------------------------------- - * Purpose: - * Reads a varible length quantity from an SMF file - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT SMF_ParseEvent (S_EAS_DATA *pEASData, S_SMF_DATA *pSMFData, S_SMF_STREAM *pSMFStream, EAS_INT parserMode) -{ - EAS_RESULT result; - EAS_U8 c; - - /* get the event type */ - if ((result = EAS_HWGetByte(pEASData->hwInstData, pSMFStream->fileHandle, &c)) != EAS_SUCCESS) - return result; - - /* parse meta-event */ - if (c == 0xff) - { - if ((result = SMF_ParseMetaEvent(pEASData, pSMFData, pSMFStream)) != EAS_SUCCESS) - return result; - } - - /* parse SysEx */ - else if ((c == 0xf0) || (c == 0xf7)) - { - if ((result = SMF_ParseSysEx(pEASData, pSMFData, pSMFStream, c, parserMode)) != EAS_SUCCESS) - return result; - } - - /* parse MIDI message */ - else - { - if ((result = EAS_ParseMIDIStream(pEASData, pSMFData->pSynth, &pSMFStream->midiStream, c, parserMode)) != EAS_SUCCESS) - return result; - - /* keep streaming data to the MIDI parser until the message is complete */ - while (pSMFStream->midiStream.pending) - { - if ((result = EAS_HWGetByte(pEASData->hwInstData, pSMFStream->fileHandle, &c)) != EAS_SUCCESS) - return result; - if ((result = EAS_ParseMIDIStream(pEASData, pSMFData->pSynth, &pSMFStream->midiStream, c, parserMode)) != EAS_SUCCESS) - return result; - } - - } - - /* chase mode logic */ - if (pSMFData->time == 0) - { - if (pSMFData->flags & SMF_FLAGS_CHASE_MODE) - { - if (pSMFStream->midiStream.flags & MIDI_FLAG_FIRST_NOTE) - pSMFData->flags &= ~SMF_FLAGS_CHASE_MODE; - } - else if ((pSMFData->flags & SMF_FLAGS_SETUP_BAR) == SMF_FLAGS_SETUP_BAR) - pSMFData->flags = (pSMFData->flags & ~SMF_FLAGS_SETUP_BAR) | SMF_FLAGS_CHASE_MODE; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * SMF_ParseHeader() - *---------------------------------------------------------------------------- - * Purpose: - * Parses the header of an SMF file, allocates memory the stream parsers and initializes the - * stream parsers. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * pSMFData - pointer to parser instance data - * fileHandle - file handle - * fileOffset - offset in the file where the header data starts, usually 0 - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -e{801} we know that 'goto' is deprecated - but it's cleaner in this case */ -EAS_RESULT SMF_ParseHeader (EAS_HW_DATA_HANDLE hwInstData, S_SMF_DATA *pSMFData) -{ - EAS_RESULT result; - EAS_I32 i; - EAS_U16 division; - EAS_U32 chunkSize; - EAS_U32 chunkStart; - EAS_U32 temp; - EAS_U32 ticks; - - /* rewind the file and find the end of the header chunk */ - if ((result = EAS_HWFileSeek(hwInstData, pSMFData->fileHandle, pSMFData->fileOffset + SMF_OFS_HEADER_SIZE)) != EAS_SUCCESS) - goto ReadError; - if ((result = EAS_HWGetDWord(hwInstData, pSMFData->fileHandle, &chunkSize, EAS_TRUE)) != EAS_SUCCESS) - goto ReadError; - - /* determine the number of tracks */ - if ((result = EAS_HWFileSeek(hwInstData, pSMFData->fileHandle, pSMFData->fileOffset + SMF_OFS_NUM_TRACKS)) != EAS_SUCCESS) - goto ReadError; - if ((result = EAS_HWGetWord(hwInstData, pSMFData->fileHandle, &pSMFData->numStreams, EAS_TRUE)) != EAS_SUCCESS) - goto ReadError; - - /* limit the number of tracks */ - if (pSMFData->numStreams > MAX_SMF_STREAMS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "SMF file contains %u tracks, playing %d tracks\n", pSMFData->numStreams, MAX_SMF_STREAMS); */ } - pSMFData->numStreams = MAX_SMF_STREAMS; - } - - /* get the time division */ - if ((result = EAS_HWGetWord(hwInstData, pSMFData->fileHandle, &division, EAS_TRUE)) != EAS_SUCCESS) - goto ReadError; - - /* setup default timebase for 120 bpm */ - pSMFData->ppqn = 192; - if (division & 0x8000) - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING,"No support for SMPTE code timebase\n"); */ } - else - pSMFData->ppqn = (division & 0x7fff); - pSMFData->tickConv = (EAS_U16) (((SMF_DEFAULT_TIMEBASE * 1024) / pSMFData->ppqn + 500) / 1000); - - /* dynamic memory allocation, allocate memory for streams */ - if (pSMFData->streams == NULL) - { - pSMFData->streams = EAS_HWMalloc(hwInstData,sizeof(S_SMF_STREAM) * pSMFData->numStreams); - if (pSMFData->streams == NULL) - return EAS_ERROR_MALLOC_FAILED; - - /* zero the memory to insure complete initialization */ - EAS_HWMemSet((void *)(pSMFData->streams), 0, sizeof(S_SMF_STREAM) * pSMFData->numStreams); - } - - /* find the start of each track */ - chunkStart = (EAS_U32) pSMFData->fileOffset; - ticks = 0x7fffffffL; - pSMFData->nextStream = NULL; - for (i = 0; i < pSMFData->numStreams; i++) - { - - for (;;) - { - - /* calculate start of next chunk - checking for errors */ - temp = chunkStart + SMF_CHUNK_INFO_SIZE + chunkSize; - if (temp <= chunkStart) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING,"Error in chunk size at offset %d\n", chunkStart); */ } - return EAS_ERROR_FILE_FORMAT; - } - chunkStart = temp; - - /* seek to the start of the next chunk */ - if ((result = EAS_HWFileSeek(hwInstData, pSMFData->fileHandle, (EAS_I32) chunkStart)) != EAS_SUCCESS) - goto ReadError; - - /* read the chunk identifier */ - if ((result = EAS_HWGetDWord(hwInstData, pSMFData->fileHandle, &temp, EAS_TRUE)) != EAS_SUCCESS) - goto ReadError; - - /* read the chunk size */ - if ((result = EAS_HWGetDWord(hwInstData, pSMFData->fileHandle, &chunkSize, EAS_TRUE)) != EAS_SUCCESS) - goto ReadError; - - /* make sure this is an 'MTrk' chunk */ - if (temp == SMF_CHUNK_TYPE_TRACK) - break; - - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING,"Unexpected chunk type: 0x%08x\n", temp); */ } - } - - /* initalize some data */ - pSMFData->streams[i].ticks = 0; - pSMFData->streams[i].fileHandle = pSMFData->fileHandle; - - /* NULL the file handle so we don't try to close it twice */ - pSMFData->fileHandle = NULL; - - /* save this file position as the start of the track */ - pSMFData->streams[i].startFilePos = (EAS_I32) chunkStart + SMF_CHUNK_INFO_SIZE; - - /* initalize the MIDI parser data */ - EAS_InitMIDIStream(&pSMFData->streams[i].midiStream); - - /* parse the first delta time in each stream */ - if ((result = SMF_GetDeltaTime(hwInstData, &pSMFData->streams[i])) != EAS_SUCCESS) - goto ReadError; - - if (pSMFData->streams[i].ticks < ticks) - { - ticks = pSMFData->streams[i].ticks; - pSMFData->nextStream = &pSMFData->streams[i]; - } - - /* more tracks to do, create a duplicate file handle */ - if (i < (pSMFData->numStreams - 1)) - { - if ((result = EAS_HWDupHandle(hwInstData, pSMFData->streams[i].fileHandle, &pSMFData->fileHandle)) != EAS_SUCCESS) - goto ReadError; - } - } - - /* update the time of the next event */ - if (pSMFData->nextStream) - SMF_UpdateTime(pSMFData, pSMFData->nextStream->ticks); - - return EAS_SUCCESS; - - /* ugly goto: but simpler than structured */ - ReadError: - if (result == EAS_EOF) - return EAS_ERROR_FILE_FORMAT; - return result; -} - -/*---------------------------------------------------------------------------- - * SMF_UpdateTime() - *---------------------------------------------------------------------------- - * Purpose: - * Update the millisecond time base by converting the ticks into millieconds - * - * Inputs: - * - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static void SMF_UpdateTime (S_SMF_DATA *pSMFData, EAS_U32 ticks) -{ - EAS_U32 temp1, temp2; - - if (pSMFData->flags & SMF_FLAGS_CHASE_MODE) - return; - - temp1 = (ticks >> 10) * pSMFData->tickConv; - temp2 = (ticks & 0x3ff) * pSMFData->tickConv; - pSMFData->time += (EAS_I32)((temp1 << 8) + (temp2 >> 2)); -} - diff --git a/arm-hybrid-22k/lib_src/eas_smf.h b/arm-hybrid-22k/lib_src/eas_smf.h deleted file mode 100644 index 9f66ab9..0000000 --- a/arm-hybrid-22k/lib_src/eas_smf.h +++ /dev/null @@ -1,49 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_smf.h - * - * Contents and purpose: - * SMF Type 0 and 1 File Parser - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_SMF_H -#define _EAS_SMF_H - -/* prototypes for private interface to SMF parser */ -EAS_RESULT SMF_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset); -EAS_RESULT SMF_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -EAS_RESULT SMF_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime); -EAS_RESULT SMF_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode); -EAS_RESULT SMF_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState); -EAS_RESULT SMF_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -EAS_RESULT SMF_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -EAS_RESULT SMF_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -EAS_RESULT SMF_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -EAS_RESULT SMF_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value); -EAS_RESULT SMF_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue); -EAS_RESULT SMF_ParseHeader (EAS_HW_DATA_HANDLE hwInstData, S_SMF_DATA *pSMFData); - -#endif /* end _EAS_SMF_H */ - - diff --git a/arm-hybrid-22k/lib_src/eas_smfdata.c b/arm-hybrid-22k/lib_src/eas_smfdata.c deleted file mode 100644 index 5c27551..0000000 --- a/arm-hybrid-22k/lib_src/eas_smfdata.c +++ /dev/null @@ -1,66 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_smfdata.c - * - * Contents and purpose: - * SMF File Parser - * - * This file contains data definitions for the SMF parser. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 778 $ - * $Date: 2007-07-23 16:45:17 -0700 (Mon, 23 Jul 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_miditypes.h" -#include "eas_smfdata.h" - -/*---------------------------------------------------------------------------- - * - * S_SMF_STREAM - * - * Static memory allocation for SMF parser - *---------------------------------------------------------------------------- -*/ -static S_SMF_STREAM eas_SMFStreams[MAX_SMF_STREAMS]; - -/*---------------------------------------------------------------------------- - * - * eas_SMFData - * - * Static memory allocation for SMF parser - *---------------------------------------------------------------------------- -*/ -S_SMF_DATA eas_SMFData = -{ - eas_SMFStreams, /* pointer to individual streams in file */ - 0, /* pointer to next stream with event */ - 0, /* pointer to synth */ - 0, /* file handle */ - { 0, 0, 0, 0}, /* metadata callback */ - 0, /* file offset */ - 0, /* current time in milliseconds/256 */ - 0, /* actual number of streams */ - 0, /* current MIDI tick to msec conversion */ - 0, /* ticks per quarter note */ - 0, /* current state EAS_STATE_XXXX */ - 0 /* flags */ -}; - diff --git a/arm-hybrid-22k/lib_src/eas_smfdata.h b/arm-hybrid-22k/lib_src/eas_smfdata.h deleted file mode 100644 index cf59cdc..0000000 --- a/arm-hybrid-22k/lib_src/eas_smfdata.h +++ /dev/null @@ -1,66 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_smfdata.h - * - * Contents and purpose: - * SMF File Parser - * - * This file contains data definitions for the SMF parser. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 686 $ - * $Date: 2007-05-03 14:10:54 -0700 (Thu, 03 May 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_SMF_DATA_H -#define _EAS_SMF_DATA_H - -#ifndef MAX_SMF_STREAMS -#define MAX_SMF_STREAMS 17 -#endif - -/* offsets in to the SMF file */ -#define SMF_OFS_HEADER_SIZE 4 -#define SMF_OFS_FILE_TYPE 8 -#define SMF_OFS_NUM_TRACKS 10 - -/* size of chunk info (chunk ID + chunk size) */ -#define SMF_CHUNK_INFO_SIZE 8 - -/* 'MTrk' track chunk ID */ -#define SMF_CHUNK_TYPE_TRACK 0x4d54726bL - -/* some useful meta-events */ -#define SMF_META_TEXT 0x01 -#define SMF_META_COPYRIGHT 0x02 -#define SMF_META_SEQTRK_NAME 0x03 -#define SMF_META_LYRIC 0x05 -#define SMF_META_END_OF_TRACK 0x2f -#define SMF_META_TEMPO 0x51 -#define SMF_META_TIME_SIGNATURE 0x58 - -/* default timebase (120BPM) */ -#define SMF_DEFAULT_TIMEBASE 500000L - -/* value for pSMFStream->ticks to signify end of track */ -#define SMF_END_OF_TRACK 0xffffffff - -#endif - diff --git a/arm-hybrid-22k/lib_src/eas_sndlib.h b/arm-hybrid-22k/lib_src/eas_sndlib.h deleted file mode 100644 index e05bee0..0000000 --- a/arm-hybrid-22k/lib_src/eas_sndlib.h +++ /dev/null @@ -1,406 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_sndlib.h - * - * Contents and purpose: - * Declarations for the sound library - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 550 $ - * $Date: 2007-02-02 09:37:03 -0800 (Fri, 02 Feb 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_SNDLIB_H -#define _EAS_SNDLIB_H - -#include "eas_types.h" -#include "eas_synthcfg.h" - -#ifdef _WT_SYNTH -#include "eas_wtengine.h" -#endif - -/*---------------------------------------------------------------------------- - * This is bit of a hack to allow us to keep the same structure - * declarations for the DLS parser. Normally, the data is located - * in read-only memory, but for DLS, we store the data in RW - * memory. - *---------------------------------------------------------------------------- -*/ -#ifndef SCNST -#define SCNST const -#endif - -/*---------------------------------------------------------------------------- - * sample size - *---------------------------------------------------------------------------- -*/ -#ifdef _16_BIT_SAMPLES -typedef EAS_I16 EAS_SAMPLE; -#else -typedef EAS_I8 EAS_SAMPLE; -#endif - -/*---------------------------------------------------------------------------- - * EAS Library ID - quick check for valid library and version - *---------------------------------------------------------------------------- -*/ -#define _EAS_LIBRARY_VERSION 0x01534145 - -#define NUM_PROGRAMS_IN_BANK 128 -#define INVALID_REGION_INDEX 0xffff - -/* this bit in region index indicates that region is for secondary synth */ -#define FLAG_RGN_IDX_FM_SYNTH 0x8000 -#define FLAG_RGN_IDX_DLS_SYNTH 0x4000 -#define REGION_INDEX_MASK 0x3fff - -/*---------------------------------------------------------------------------- - * Generic region data structure - * - * This must be the first element in each region structure - *---------------------------------------------------------------------------- -*/ -typedef struct s_region_tag -{ - EAS_U16 keyGroupAndFlags; - EAS_U8 rangeLow; - EAS_U8 rangeHigh; -} S_REGION; - -/* - * Bit fields for m_nKeyGroupAndFlags - * Bits 0-2 are mode bits in FM synth - * Bits 8-11 are the key group - */ -#define REGION_FLAG_IS_LOOPED 0x01 -#define REGION_FLAG_USE_WAVE_GENERATOR 0x02 -#define REGION_FLAG_USE_ADPCM 0x04 -#define REGION_FLAG_ONE_SHOT 0x08 -#define REGION_FLAG_SQUARE_WAVE 0x10 -#define REGION_FLAG_OFF_CHIP 0x20 -#define REGION_FLAG_NON_SELF_EXCLUSIVE 0x40 -#define REGION_FLAG_LAST_REGION 0x8000 - -/*---------------------------------------------------------------------------- - * Envelope data structure - *---------------------------------------------------------------------------- -*/ -typedef struct s_envelope_tag -{ - EAS_I16 attackTime; - EAS_I16 decayTime; - EAS_I16 sustainLevel; - EAS_I16 releaseTime; -} S_ENVELOPE; - -/*---------------------------------------------------------------------------- - * DLS envelope data structure - *---------------------------------------------------------------------------- -*/ -typedef struct s_dls_envelope_tag -{ - EAS_I16 delayTime; - EAS_I16 attackTime; - EAS_I16 holdTime; - EAS_I16 decayTime; - EAS_I16 sustainLevel; - EAS_I16 releaseTime; - EAS_I16 velToAttack; - EAS_I16 keyNumToDecay; - EAS_I16 keyNumToHold; -} S_DLS_ENVELOPE; - -/*---------------------------------------------------------------------------- - * LFO data structure - *---------------------------------------------------------------------------- -*/ -typedef struct s_lfo_params_tag -{ - EAS_I16 lfoFreq; - EAS_I16 lfoDelay; -} S_LFO_PARAMS; - -/*---------------------------------------------------------------------------- - * Articulation data structure - *---------------------------------------------------------------------------- -*/ -typedef struct s_articulation_tag -{ - S_ENVELOPE eg1; - S_ENVELOPE eg2; - EAS_I16 lfoToPitch; - EAS_I16 lfoDelay; - EAS_I16 lfoFreq; - EAS_I16 eg2ToPitch; - EAS_I16 eg2ToFc; - EAS_I16 filterCutoff; - EAS_I8 lfoToGain; - EAS_U8 filterQ; - EAS_I8 pan; -} S_ARTICULATION; - -/*---------------------------------------------------------------------------- - * DLS articulation data structure - *---------------------------------------------------------------------------- -*/ - -typedef struct s_dls_articulation_tag -{ - S_LFO_PARAMS modLFO; - S_LFO_PARAMS vibLFO; - - S_DLS_ENVELOPE eg1; - S_DLS_ENVELOPE eg2; - - EAS_I16 eg1ShutdownTime; - - EAS_I16 filterCutoff; - EAS_I16 modLFOToFc; - EAS_I16 modLFOCC1ToFc; - EAS_I16 modLFOChanPressToFc; - EAS_I16 eg2ToFc; - EAS_I16 velToFc; - EAS_I16 keyNumToFc; - - EAS_I16 modLFOToGain; - EAS_I16 modLFOCC1ToGain; - EAS_I16 modLFOChanPressToGain; - - EAS_I16 tuning; - EAS_I16 keyNumToPitch; - EAS_I16 vibLFOToPitch; - EAS_I16 vibLFOCC1ToPitch; - EAS_I16 vibLFOChanPressToPitch; - EAS_I16 modLFOToPitch; - EAS_I16 modLFOCC1ToPitch; - EAS_I16 modLFOChanPressToPitch; - EAS_I16 eg2ToPitch; - - /* pad to 4-byte boundary */ - EAS_U16 pad; - - EAS_I8 pan; - EAS_U8 filterQandFlags; - -#ifdef _REVERB - EAS_I16 reverbSend; - EAS_I16 cc91ToReverbSend; -#endif - -#ifdef _CHORUS - EAS_I16 chorusSend; - EAS_I16 cc93ToChorusSend; -#endif -} S_DLS_ARTICULATION; - -/* flags in filterQandFlags - * NOTE: Q is stored in bottom 5 bits - */ -#define FLAG_DLS_VELOCITY_SENSITIVE 0x80 -#define FILTER_Q_MASK 0x1f - -/*---------------------------------------------------------------------------- - * Wavetable region data structure - *---------------------------------------------------------------------------- -*/ -typedef struct s_wt_region_tag -{ - S_REGION region; - EAS_I16 tuning; - EAS_I16 gain; - EAS_U32 loopStart; - EAS_U32 loopEnd; - EAS_U16 waveIndex; - EAS_U16 artIndex; -} S_WT_REGION; - -/*---------------------------------------------------------------------------- - * DLS region data structure - *---------------------------------------------------------------------------- -*/ -typedef struct s_dls_region_tag -{ - S_WT_REGION wtRegion; - EAS_U8 velLow; - EAS_U8 velHigh; -} S_DLS_REGION; - -/*---------------------------------------------------------------------------- - * FM synthesizer data structures - *---------------------------------------------------------------------------- -*/ -typedef struct s_fm_oper_tag -{ - EAS_I16 tuning; - EAS_U8 attackDecay; - EAS_U8 velocityRelease; - EAS_U8 egKeyScale; - EAS_U8 sustain; - EAS_U8 gain; - EAS_U8 flags; -} S_FM_OPER; - -/* defines for S_FM_OPER.m_nFlags */ -#define FM_OPER_FLAG_MONOTONE 0x01 -#define FM_OPER_FLAG_NO_VIBRATO 0x02 -#define FM_OPER_FLAG_NOISE 0x04 -#define FM_OPER_FLAG_LINEAR_VELOCITY 0x08 - -/* NOTE: The first two structure elements are common with S_WT_REGION - * and we will rely on that in the voice management code and must - * remain there unless the voice management code is revisited. - */ -typedef struct s_fm_region_tag -{ - S_REGION region; - EAS_U8 vibTrem; - EAS_U8 lfoFreqDelay; - EAS_U8 feedback; - EAS_I8 pan; - S_FM_OPER oper[4]; -} S_FM_REGION; - -/*---------------------------------------------------------------------------- - * Common data structures - *---------------------------------------------------------------------------- -*/ - -/*---------------------------------------------------------------------------- - * Program data structure - * Used for individual programs not stored as a complete bank. - *---------------------------------------------------------------------------- -*/ -typedef struct s_program_tag -{ - EAS_U32 locale; - EAS_U16 regionIndex; -} S_PROGRAM; - -/*---------------------------------------------------------------------------- - * Bank data structure - * - * A bank always consists of 128 programs. If a bank is less than 128 - * programs, it should be stored as a spare matrix in the pPrograms - * array. - * - * bankNum: MSB/LSB of MIDI bank select controller - * regionIndex: Index of first region in program - *---------------------------------------------------------------------------- -*/ -typedef struct s_bank_tag -{ - EAS_U16 locale; - EAS_U16 regionIndex[NUM_PROGRAMS_IN_BANK]; -} S_BANK; - - -/* defines for libFormat field - * bits 0-17 are the sample rate - * bit 18 is true if wavetable is present - * bit 19 is true if FM is present - * bit 20 is true if filter is enabled - * bit 21 is sample depth (0 = 8-bits, 1 = 16-bits) - * bits 22-31 are reserved - */ -#define LIBFORMAT_SAMPLE_RATE_MASK 0x0003ffff -#define LIB_FORMAT_TYPE_MASK 0x000c0000 -#define LIB_FORMAT_WAVETABLE 0x00000000 -#define LIB_FORMAT_FM 0x00040000 -#define LIB_FORMAT_HYBRID 0x00080000 -#define LIB_FORMAT_FILTER_ENABLED 0x00100000 -#define LIB_FORMAT_16_BIT_SAMPLES 0x00200000 - -#ifdef DLS_SYNTHESIZER -/*---------------------------------------------------------------------------- - * DLS data structure - * - * pDLSPrograms pointer to array of DLS programs - * pDLSRegions pointer to array of DLS regions - * pDLSArticulations pointer to array of DLS articulations - * pSampleLen pointer to array of sample lengths - * ppSamples pointer to array of sample pointers - * numDLSPrograms number of DLS programs - * numDLSRegions number of DLS regions - * numDLSArticulations number of DLS articulations - * numDLSSamples number of DLS samples - *---------------------------------------------------------------------------- -*/ -typedef struct s_eas_dls_tag -{ - S_PROGRAM *pDLSPrograms; - S_DLS_REGION *pDLSRegions; - S_DLS_ARTICULATION *pDLSArticulations; - EAS_U32 *pDLSSampleLen; - EAS_U32 *pDLSSampleOffsets; - EAS_SAMPLE *pDLSSamples; - EAS_U16 numDLSPrograms; - EAS_U16 numDLSRegions; - EAS_U16 numDLSArticulations; - EAS_U16 numDLSSamples; - EAS_U8 refCount; -} S_DLS; -#endif - -/*---------------------------------------------------------------------------- - * Sound library data structure - * - * pBanks pointer to array of banks - * pPrograms pointer to array of programs - * pWTRegions pointer to array of wavetable regions - * pFMRegions pointer to array of FM regions - * pArticulations pointer to array of articulations - * pSampleLen pointer to array of sample lengths - * ppSamples pointer to array of sample pointers - * numBanks number of banks - * numPrograms number of individual program - * numRegions number of regions - * numArticulations number of articulations - * numSamples number of samples - *---------------------------------------------------------------------------- -*/ -typedef struct s_eas_sndlib_tag -{ - SCNST EAS_U32 identifier; - SCNST EAS_U32 libAttr; - - SCNST S_BANK *pBanks; - SCNST S_PROGRAM *pPrograms; - - SCNST S_WT_REGION *pWTRegions; - SCNST S_ARTICULATION *pArticulations; - SCNST EAS_U32 *pSampleLen; - SCNST EAS_U32 *pSampleOffsets; - SCNST EAS_SAMPLE *pSamples; - - SCNST S_FM_REGION *pFMRegions; - - SCNST EAS_U16 numBanks; - SCNST EAS_U16 numPrograms; - - SCNST EAS_U16 numWTRegions; - SCNST EAS_U16 numArticulations; - SCNST EAS_U16 numSamples; - - SCNST EAS_U16 numFMRegions; -} S_EAS; - -#endif - diff --git a/arm-hybrid-22k/lib_src/eas_synth.h b/arm-hybrid-22k/lib_src/eas_synth.h deleted file mode 100644 index b242b03..0000000 --- a/arm-hybrid-22k/lib_src/eas_synth.h +++ /dev/null @@ -1,395 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_synth.h - * - * Contents and purpose: - * Declarations, interfaces, and prototypes for synth. - * - * Copyright Sonic Network Inc. 2004, 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 718 $ - * $Date: 2007-06-08 16:43:16 -0700 (Fri, 08 Jun 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_SYNTH_H -#define _EAS_SYNTH_H - -#include "eas_types.h" -#include "eas_sndlib.h" - -#ifdef _WT_SYNTH -#include "eas_wtsynth.h" -#endif - -#ifdef _FM_SYNTH -#include "eas_fmsynth.h" -#endif - -#ifndef NUM_OUTPUT_CHANNELS -#define NUM_OUTPUT_CHANNELS 2 -#endif - -#ifndef MAX_SYNTH_VOICES -#define MAX_SYNTH_VOICES 64 -#endif - -#ifndef MAX_VIRTUAL_SYNTHESIZERS -#define MAX_VIRTUAL_SYNTHESIZERS 4 -#endif - -/* defines */ -#ifndef NUM_PRIMARY_VOICES -#define NUM_PRIMARY_VOICES MAX_SYNTH_VOICES -#elif !defined(NUM_SECONDARY_VOICES) -#define NUM_SECONDARY_VOICES (MAX_SYNTH_VOICES - NUM_PRIMARY_VOICES) -#endif - -#if defined(EAS_WT_SYNTH) -#define NUM_WT_VOICES MAX_SYNTH_VOICES - -/* FM on MCU */ -#elif defined(EAS_FM_SYNTH) -#define NUM_FM_VOICES MAX_SYNTH_VOICES - -/* wavetable drums on MCU, wavetable melodic on DSP */ -#elif defined(EAS_SPLIT_WT_SYNTH) -#define NUM_WT_VOICES MAX_SYNTH_VOICES - -/* wavetable drums and FM melodic on MCU */ -#elif defined(EAS_HYBRID_SYNTH) -#define NUM_WT_VOICES NUM_PRIMARY_VOICES -#define NUM_FM_VOICES NUM_SECONDARY_VOICES - -/* wavetable drums on MCU, FM melodic on DSP */ -#elif defined(EAS_SPLIT_HYBRID_SYNTH) -#define NUM_WT_VOICES NUM_PRIMARY_VOICES -#define NUM_FM_VOICES NUM_SECONDARY_VOICES - -/* FM synth on DSP */ -#elif defined(EAS_SPLIT_FM_SYNTH) -#define NUM_FM_VOICES MAX_SYNTH_VOICES - -#else -#error "Unrecognized architecture option" -#endif - -#define NUM_SYNTH_CHANNELS 16 - -#define DEFAULT_SYNTH_VOICES MAX_SYNTH_VOICES - -/* use the following values to specify unassigned channels or voices */ -#define UNASSIGNED_SYNTH_CHANNEL NUM_SYNTH_CHANNELS -#define UNASSIGNED_SYNTH_VOICE MAX_SYNTH_VOICES - - -/* synth parameters are updated every SYNTH_UPDATE_PERIOD_IN_SAMPLES */ -#define SYNTH_UPDATE_PERIOD_IN_SAMPLES (EAS_I32)(0x1L << SYNTH_UPDATE_PERIOD_IN_BITS) - -/* stealing weighting factors */ -#define NOTE_AGE_STEAL_WEIGHT 1 -#define NOTE_GAIN_STEAL_WEIGHT 4 -#define CHANNEL_POLY_STEAL_WEIGHT 12 -#define CHANNEL_PRIORITY_STEAL_WEIGHT 2 -#define NOTE_MATCH_PENALTY 128 -#define SYNTH_PRIORITY_WEIGHT 8 - -/* default synth master volume */ -#define DEFAULT_SYNTH_MASTER_VOLUME 0x7fff - -#define DEFAULT_SYNTH_PRIORITY 5 - -/* default tuning values */ -#define DEFAULT_PITCH_BEND_SENSITIVITY 200 /* 2 semitones */ -#define DEFAULT_FINE_PITCH 0 /* 0 cents */ -#define DEFAULT_COARSE_PITCH 0 /* 0 semitones */ - -/* default drum channel is 10, but is internally 9 due to unit offset */ -#define DEFAULT_DRUM_CHANNEL 9 - -/* drum channel can simultaneously play this many voices at most */ -#define DEFAULT_CHANNEL_POLYPHONY_LIMIT 2 - -/* default instrument is acoustic piano */ -#define DEFAULT_MELODY_BANK_MSB 0x79 -#define DEFAULT_RHYTHM_BANK_MSB 0x78 -#define DEFAULT_MELODY_BANK_NUMBER (DEFAULT_MELODY_BANK_MSB << 8) -#define DEFAULT_RHYTHM_BANK_NUMBER (DEFAULT_RHYTHM_BANK_MSB << 8) -#define DEFAULT_SYNTH_PROGRAM_NUMBER 0 - -#define DEFAULT_PITCH_BEND 0x2000 /* 0x2000 == (0x40 << 7) | 0x00 */ -#define DEFAULT_MOD_WHEEL 0 -#define DEFAULT_CHANNEL_VOLUME 0x64 -#define DEFAULT_PAN 0x40 /* decimal 64, center */ - -#ifdef _REVERB -#define DEFAULT_REVERB_SEND 40 /* some reverb */ -#endif - -#ifdef _CHORUS -#define DEFAULT_CHORUS_SEND 0 /* no chorus */ -#endif - -#define DEFAULT_EAS_FILTER_CUTOFF_FREQUENCY 0 /* EAS synth uses a different default */ -#define DEFAULT_FILTER_RESONANCE 0 -#define DEFAULT_EXPRESSION 0x7F - -#define DEFAULT_CHANNEL_PRESSURE 0 - -#define DEFAULT_REGISTERED_PARAM 0x3FFF - -#define DEFAULT_CHANNEL_STATIC_GAIN 0 -#define DEFAULT_CHANNEL_STATIC_PITCH 0 - -#define DEFAULT_LFO_MOD_WHEEL_TO_PITCH_CENTS 50 -#define DEFAULT_LFO_CHANNEL_PRESSURE_TO_PITCH_CENTS 50 - -#define DEFAULT_KEY_NUMBER 0x69 -#define DEFAULT_VELOCITY 0x64 -#define DEFAULT_REGION_INDEX 0 -#define DEFAULT_ARTICULATION_INDEX 0 -#define DEFAULT_VOICE_GAIN 0 -#define DEFAULT_AGE 0 -#define DEFAULT_SP_MIDI_PRIORITY 16 - - -/* filter defines */ -#define DEFAULT_FILTER_ZERO 0 -#define FILTER_CUTOFF_MAX_PITCH_CENTS 1919 -#define FILTER_CUTOFF_MIN_PITCH_CENTS -4467 -#define A5_PITCH_OFFSET_IN_CENTS 6900 - -/*------------------------------------ - * S_SYNTH_CHANNEL data structure - *------------------------------------ -*/ - -/* S_SYNTH_CHANNEL.m_nFlags */ -#define CHANNEL_FLAG_SUSTAIN_PEDAL 0x01 -#define CHANNEL_FLAG_MUTE 0x02 -#define CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS 0x04 -#define CHANNEL_FLAG_RHYTHM_CHANNEL 0x08 -#define CHANNEL_FLAG_EXTERNAL_AUDIO 0x10 -#define DEFAULT_CHANNEL_FLAGS 0 - -/* macros for extracting virtual synth and channel numbers */ -#define GET_VSYNTH(a) ((a) >> 4) -#define GET_CHANNEL(a) ((a) & 15) - -typedef struct s_synth_channel_tag -{ - /* use static channel parameters to reduce MIPs */ - /* parameters shared by multiple voices assigned to same channel */ - EAS_I32 staticPitch; /* (pitch bend * pitch sens) + fine pitch */ - EAS_I16 staticGain; /* (CC7 * CC11 * master vol)^2 */ - - EAS_U16 regionIndex; /* index of first region in program */ - - EAS_U16 bankNum; /* play programs from this bank */ - EAS_I16 pitchBend; /* pitch wheel value */ - EAS_I16 pitchBendSensitivity; - EAS_I16 registeredParam; /* currently selected registered param */ - - -#if defined(_FM_SYNTH) - EAS_I16 lfoAmt; /* amount of LFO to apply to voice */ -#endif - - EAS_U8 programNum; /* play this instrument number */ - EAS_U8 modWheel; /* CC1 */ - EAS_U8 volume; /* CC7 */ - EAS_U8 pan; /* CC10 */ - - EAS_U8 expression; /* CC11 */ - - /* the following parameters are controlled by RPNs */ - EAS_I8 finePitch; - EAS_I8 coarsePitch; - - EAS_U8 channelPressure; /* applied to all voices on a given channel */ - - EAS_U8 channelFlags; /* bit field channelFlags for */ - /* CC64, SP-MIDI channel masking */ - - EAS_U8 pool; /* SPMIDI channel voice pool */ - EAS_U8 mip; /* SPMIDI MIP setting */ - -#ifdef _REVERB - EAS_U8 reverbSend; /* CC91 */ -#endif - -#ifdef _CHORUS - EAS_U8 chorusSend; /* CC93 */ -#endif -} S_SYNTH_CHANNEL; - -/*------------------------------------ - * S_SYNTH_VOICE data structure - *------------------------------------ -*/ - -/* S_SYNTH_VOICE.m_nFlags */ -#define VOICE_FLAG_UPDATE_VOICE_PARAMETERS 0x01 -#define VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF 0x02 -#define VOICE_FLAG_DEFER_MIDI_NOTE_OFF 0x04 -#define VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET 0x08 -#define VOICE_FLAG_DEFER_MUTE 0x40 -#define DEFAULT_VOICE_FLAGS 0 - -/* S_SYNTH_VOICE.m_eState */ -typedef enum { - - eVoiceStateFree = 0, - eVoiceStateStart, - eVoiceStatePlay, - eVoiceStateRelease, - eVoiceStateMuting, - eVoiceStateStolen, - eVoiceStateInvalid /* should never be in this state! */ - -} E_VOICE_STATE; -#define DEFAULT_VOICE_STATE eVoiceStateFree - -typedef struct s_synth_voice_tag -{ - -/* These parameters are common to both wavetable and FM - * synthesizers. The voice manager should only access this data. - * Any other data should be manipulated by the code that is - * specific to that synthesizer and reflected back through the - * common state data available here. - */ - EAS_U16 regionIndex; /* index to wave and playback params */ - EAS_I16 gain; /* current gain */ - EAS_U16 age; /* large value means old note */ - EAS_U16 nextRegionIndex; /* index to wave and playback params */ - EAS_U8 voiceState; /* current voice state */ - EAS_U8 voiceFlags; /* misc flags/bit fields */ - EAS_U8 channel; /* this voice plays on this synth channel */ - EAS_U8 note; /* 12 <= key number <= 108 */ - EAS_U8 velocity; /* 0 <= velocity <= 127 */ - EAS_U8 nextChannel; /* play stolen voice on this channel */ - EAS_U8 nextNote; /* 12 <= key number <= 108 */ - EAS_U8 nextVelocity; /* 0 <= velocity <= 127 */ -} S_SYNTH_VOICE; - -/*------------------------------------ - * S_SYNTH data structure - * - * One instance for each MIDI stream - *------------------------------------ -*/ - -/* S_SYNTH.m_nFlags */ -#define SYNTH_FLAG_RESET_IS_REQUESTED 0x01 -#define SYNTH_FLAG_SP_MIDI_ON 0x02 -#define SYNTH_FLAG_UPDATE_ALL_CHANNEL_PARAMETERS 0x04 -#define SYNTH_FLAG_DEFERRED_MIDI_NOTE_OFF_PENDING 0x08 -#define DEFAULT_SYNTH_FLAGS SYNTH_FLAG_UPDATE_ALL_CHANNEL_PARAMETERS - -typedef struct s_synth_tag -{ - struct s_eas_data_tag *pEASData; - const S_EAS *pEAS; - -#ifdef DLS_SYNTHESIZER - S_DLS *pDLS; -#endif - -#ifdef EXTERNAL_AUDIO - EAS_EXT_PRG_CHG_FUNC cbProgChgFunc; - EAS_EXT_EVENT_FUNC cbEventFunc; - EAS_VOID_PTR *pExtAudioInstData; -#endif - - S_SYNTH_CHANNEL channels[NUM_SYNTH_CHANNELS]; - EAS_I32 totalNoteCount; - EAS_U16 maxPolyphony; - EAS_U16 numActiveVoices; - EAS_U16 masterVolume; - EAS_U8 channelsByPriority[NUM_SYNTH_CHANNELS]; - EAS_U8 poolCount[NUM_SYNTH_CHANNELS]; - EAS_U8 poolAlloc[NUM_SYNTH_CHANNELS]; - EAS_U8 synthFlags; - EAS_I8 globalTranspose; - EAS_U8 vSynthNum; - EAS_U8 refCount; - EAS_U8 priority; -} S_SYNTH; - -/*------------------------------------ - * S_VOICE_MGR data structure - * - * One instance for each EAS library instance - *------------------------------------ -*/ -typedef struct s_voice_mgr_tag -{ - S_SYNTH *pSynth[MAX_VIRTUAL_SYNTHESIZERS]; - EAS_PCM voiceBuffer[SYNTH_UPDATE_PERIOD_IN_SAMPLES]; - -#ifdef _FM_SYNTH - EAS_PCM operMixBuffer[SYNTH_UPDATE_PERIOD_IN_SAMPLES]; - S_FM_VOICE fmVoices[NUM_FM_VOICES]; -#endif - -#ifdef _WT_SYNTH - S_WT_VOICE wtVoices[NUM_WT_VOICES]; -#endif - -#ifdef _REVERB - EAS_PCM reverbSendBuffer[NUM_OUTPUT_CHANNELS * SYNTH_UPDATE_PERIOD_IN_SAMPLES]; -#endif - -#ifdef _CHORUS - EAS_PCM chorusSendBuffer[NUM_OUTPUT_CHANNELS * SYNTH_UPDATE_PERIOD_IN_SAMPLES]; -#endif - S_SYNTH_VOICE voices[MAX_SYNTH_VOICES]; - - EAS_SNDLIB_HANDLE pGlobalEAS; - -#ifdef DLS_SYNTHESIZER - S_DLS *pGlobalDLS; -#endif - -#ifdef _SPLIT_ARCHITECTURE - EAS_FRAME_BUFFER_HANDLE pFrameBuffer; -#endif - -#if defined(_SECONDARY_SYNTH) || defined(EAS_SPLIT_WT_SYNTH) - EAS_U16 maxPolyphonyPrimary; - EAS_U16 maxPolyphonySecondary; -#endif - - EAS_I32 workload; - EAS_I32 maxWorkLoad; - - EAS_U16 activeVoices; - EAS_U16 maxPolyphony; - - EAS_U16 age; - -/* limits the number of voice starts in a frame for split architecture */ -#ifdef MAX_VOICE_STARTS - EAS_U16 numVoiceStarts; -#endif -} S_VOICE_MGR; - -#endif /* #ifdef _EAS_SYNTH_H */ - - diff --git a/arm-hybrid-22k/lib_src/eas_synth_protos.h b/arm-hybrid-22k/lib_src/eas_synth_protos.h deleted file mode 100644 index a2ef10d..0000000 --- a/arm-hybrid-22k/lib_src/eas_synth_protos.h +++ /dev/null @@ -1,60 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_synth_protos.h - * - * Contents and purpose: - * Declarations, interfaces, and prototypes for synth. - * - * Copyright Sonic Network Inc. 2004 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_SYNTH_PROTOS_H -#define _EAS_SYNTH_PROTOS_H - -/* includes */ -#include "eas_data.h" -#include "eas_sndlib.h" - -#ifdef _SPLIT_ARCHITECTURE -typedef struct s_frame_interface_tag -{ - EAS_BOOL (* EAS_CONST pfStartFrame)(EAS_FRAME_BUFFER_HANDLE pFrameBuffer); - EAS_BOOL (* EAS_CONST pfEndFrame)(EAS_FRAME_BUFFER_HANDLE pFrameBuffer, EAS_I32 *pMixBuffer, EAS_I16 masterGain); -} S_FRAME_INTERFACE; -#endif - -/* generic synthesizer interface */ -typedef struct -{ - EAS_RESULT (* EAS_CONST pfInitialize)(S_VOICE_MGR *pVoiceMgr); - EAS_RESULT (* EAS_CONST pfStartVoice)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex); - EAS_BOOL (* EAS_CONST pfUpdateVoice)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32 numSamples); - void (* EAS_CONST pfReleaseVoice)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum); - void (* EAS_CONST pfMuteVoice)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum); - void (* EAS_CONST pfSustainPedal)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum); - void (* EAS_CONST pfUpdateChannel)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel); -} S_SYNTH_INTERFACE; - -#endif - - - diff --git a/arm-hybrid-22k/lib_src/eas_synthcfg.h b/arm-hybrid-22k/lib_src/eas_synthcfg.h deleted file mode 100644 index 2491e6d..0000000 --- a/arm-hybrid-22k/lib_src/eas_synthcfg.h +++ /dev/null @@ -1,70 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_synthcfg.h - * - * Contents and purpose: - * Defines for various synth configurations - * - * Copyright Sonic Network Inc. 2004, 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 664 $ - * $Date: 2007-04-25 13:11:22 -0700 (Wed, 25 Apr 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_SYNTHCFG_H -#define _EAS_SYNTHCFG_H - -#if defined(EAS_WT_SYNTH) -#define _WT_SYNTH - -/* FM on MCU */ -#elif defined(EAS_FM_SYNTH) -#define _FM_SYNTH - -/* wavetable drums and FM melodic on MCU */ -#elif defined(EAS_HYBRID_SYNTH) -#define _WT_SYNTH -#define _FM_SYNTH -#define _SECONDARY_SYNTH -#define _HYBRID_SYNTH - -/* wavetable drums on MCU, wavetable melodic on DSP */ -#elif defined(EAS_SPLIT_WT_SYNTH) -#define _WT_SYNTH -#define _SPLIT_ARCHITECTURE - -/* wavetable drums on MCU, FM melodic on DSP */ -#elif defined(EAS_SPLIT_HYBRID_SYNTH) -#define _WT_SYNTH -#define _FM_SYNTH -#define _SECONDARY_SYNTH -#define _SPLIT_ARCHITECTURE -#define _HYBRID_SYNTH - -/* FM synth on DSP */ -#elif defined(EAS_SPLIT_FM_SYNTH) -#define _FM_SYNTH -#define _SPLIT_ARCHITECTURE - -#else -#error "Unrecognized architecture option" -#endif - -#endif - diff --git a/arm-hybrid-22k/lib_src/eas_vm_protos.h b/arm-hybrid-22k/lib_src/eas_vm_protos.h deleted file mode 100644 index eb49ba8..0000000 --- a/arm-hybrid-22k/lib_src/eas_vm_protos.h +++ /dev/null @@ -1,1086 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_vm_protos.h - * - * Contents and purpose: - * Declarations, interfaces, and prototypes for voice manager. - * - * Copyright Sonic Network Inc. 2004 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 736 $ - * $Date: 2007-06-22 13:51:24 -0700 (Fri, 22 Jun 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_VM_PROTOS_H -#define _EAS_VM_PROTOS_H - -// includes -#include "eas_data.h" -#include "eas_sndlib.h" - -/*---------------------------------------------------------------------------- - * VMInitialize() - *---------------------------------------------------------------------------- - * Purpose: - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMInitialize (S_EAS_DATA *pEASData); - -/*---------------------------------------------------------------------------- - * VMInitMIDI() - *---------------------------------------------------------------------------- - * Purpose: - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMInitMIDI (S_EAS_DATA *pEASData, S_SYNTH **ppSynth); - -/*---------------------------------------------------------------------------- - * VMInitializeAllChannels() - *---------------------------------------------------------------------------- - * Purpose: - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -void VMInitializeAllChannels (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth); - -/*---------------------------------------------------------------------------- - * VMResetControllers() - *---------------------------------------------------------------------------- - * Purpose: - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -void VMResetControllers (S_SYNTH *pSynth); - -/*---------------------------------------------------------------------------- - * VMInitMIPTable() - *---------------------------------------------------------------------------- - * Purpose: - * Initialize the SP-MIDI MIP table - * - * Inputs: - * pEASData - pointer to synthesizer instance data - * mute - EAS_FALSE to unmute channels, EAS_TRUE to mute - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void VMInitMIPTable (S_SYNTH *pSynth); - -/*---------------------------------------------------------------------------- - * VMSetMIPEntry() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the priority and MIP level for a MIDI channel - * - * Inputs: - * pEASData - pointer to synthesizer instance data - * channel - MIDI channel number - * priority - priority (0-15 with 0 = highest priority) - * mip - maximum instantaneous polyphony - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void VMSetMIPEntry (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 priority, EAS_U8 mip); - -/*---------------------------------------------------------------------------- - * VMUpdateMIPTable() - *---------------------------------------------------------------------------- - * Purpose: - * This routine is called when the polyphony count in the synthesizer changes - * - * Inputs: - * pEASData - pointer to synthesizer instance data - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void VMUpdateMIPTable (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth); - -/*---------------------------------------------------------------------------- - * VMInitializeAllVoices() - *---------------------------------------------------------------------------- - * Purpose: - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -void VMInitializeAllVoices (S_VOICE_MGR *pVoiceMgr, EAS_INT vSynthNum); - -/*---------------------------------------------------------------------------- - * VMStartNote() - *---------------------------------------------------------------------------- - * Purpose: - * Update the synth's state to play the requested note on the requested - * channel if possible. - * - * Inputs: - * nChannel - the MIDI channel - * nKeyNumber - the MIDI key number for this note - * nNoteVelocity - the key velocity for this note - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -void VMStartNote (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 note, EAS_U8 velocity); - -/*---------------------------------------------------------------------------- - * VMCheckKeyGroup() - *---------------------------------------------------------------------------- - * Purpose: - * If the note that we've been asked to start is in the same key group as - * any currently playing notes, then we must shut down the currently playing - * note in the same key group and then start the newly requested note. - * - * Inputs: - * nChannel - synth channel that wants to start a new note - * nKeyNumber - new note's midi note number - * nRegionIndex - calling routine finds this index and gives to us - * nNoteVelocity - new note's velocity - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * pbVoiceStealingRequired - flag: this routine sets true if we needed to - * steal a voice - * - * Side Effects: - * gsSynthObject.m_sVoice[free voice num].m_nKeyNumber may be assigned - * gsSynthObject.m_sVoice[free voice num].m_nVelocity may be assigned - *---------------------------------------------------------------------------- -*/ -void VMCheckKeyGroup (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U16 keyGroup, EAS_U8 channel); - -/*---------------------------------------------------------------------------- - * VMCheckPolyphonyLimiting() - *---------------------------------------------------------------------------- - * Purpose: - * We only play at most 2 of the same note on a MIDI channel. - * E.g., if we are asked to start note 36, and there are already two voices - * that are playing note 36, then we must steal the voice playing - * the oldest note 36 and use that stolen voice to play the new note 36. - * - * Inputs: - * nChannel - synth channel that wants to start a new note - * nKeyNumber - new note's midi note number - * nNoteVelocity - new note's velocity - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * pbVoiceStealingRequired - flag: this routine sets true if we needed to - * steal a voice - * * - * Side Effects: - * psSynthObject->m_sVoice[free voice num].m_nKeyNumber may be assigned - * psSynthObject->m_sVoice[free voice num].m_nVelocity may be assigned - *---------------------------------------------------------------------------- -*/ -EAS_BOOL VMCheckPolyphonyLimiting (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 note, EAS_U8 velocity, EAS_U16 regionIndex, EAS_I32 lowVoice, EAS_I32 highVoice); - -/*---------------------------------------------------------------------------- - * VMStopNote() - *---------------------------------------------------------------------------- - * Purpose: - * Update the synth's state to end the requested note on the requested - * channel. - * - * Inputs: - * nChannel - the MIDI channel - * nKeyNumber - the key number of the note to stop - * nNoteVelocity - the note-off velocity - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * Side Effects: - * gsSynthObject.m_sVoice[free voice num].m_nSynthChannel may be assigned - * gsSynthObject.m_sVoice[free voice num].m_nKeyNumber is assigned - * gsSynthObject.m_sVoice[free voice num].m_nVelocity is assigned - *---------------------------------------------------------------------------- -*/ -void VMStopNote (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 key, EAS_U8 velocity); - -/*---------------------------------------------------------------------------- - * VMFindAvailableVoice() - *---------------------------------------------------------------------------- - * Purpose: - * Find an available voice and return the voice number if available. - * - * Inputs: - * pnVoiceNumber - really an output, see below - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * pnVoiceNumber - returns the voice number of available voice if found - * success - if there is an available voice - * failure - otherwise - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMFindAvailableVoice (S_VOICE_MGR *pVoiceMgr, EAS_INT *pVoiceNumber, EAS_I32 lowVoice, EAS_I32 highVoice); - -/*---------------------------------------------------------------------------- - * VMStealVoice() - *---------------------------------------------------------------------------- - * Purpose: - * Steal a voice and return the voice number - * - * Stealing algorithm: steal the best choice with minimal work, taking into - * account SP-Midi channel priorities and polyphony allocation. - * - * In one pass through all the voices, figure out which voice to steal - * taking into account a number of different factors: - * Priority of the voice's MIDI channel - * Number of voices over the polyphony allocation for voice's MIDI channel - * Amplitude of the voice - * Note age - * Key velocity (for voices that haven't been started yet) - * If any matching notes are found - * - * Inputs: - * nChannel - the channel that this voice wants to be started on - * nKeyNumber - the key number for this new voice - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * pnVoiceNumber - voice stolen - * EAS_RESULT EAS_SUCCESS - always successful - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMStealVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_INT *pVoiceNumber, EAS_U8 channel, EAS_U8 note, EAS_I32 lowVoice, EAS_I32 highVoice); - -/*---------------------------------------------------------------------------- - * VMAddSamples() - *---------------------------------------------------------------------------- - * Purpose: - * Synthesize the requested number of samples. - * - * Inputs: - * nNumSamplesToAdd - number of samples to write to buffer - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * number of samples actually written to buffer - * - * Side Effects: - * - samples are added to the presently free buffer - * - *---------------------------------------------------------------------------- -*/ -EAS_I32 VMAddSamples (S_VOICE_MGR *pVoiceMgr, EAS_I32 *pMixBuffer, EAS_I32 numSamplesToAdd); - -/*---------------------------------------------------------------------------- - * VMProgramChange() - *---------------------------------------------------------------------------- - * Purpose: - * Change the instrument (program) for the given channel. - * - * Depending on the program number, and the bank selected for this channel, the - * program may be in ROM, RAM (from SMAF or CMX related RAM wavetable), or - * Alternate wavetable (from mobile DLS or other DLS file) - * - * This function figures out what wavetable should be used, and sets it up as the - * wavetable to use for this channel. Also the channel may switch from a melodic - * channel to a rhythm channel, or vice versa. - * - * Inputs: - * - * Outputs: - * Side Effects: - * gsSynthObject.m_sChannel[nChannel].m_nProgramNumber is likely changed - * gsSynthObject.m_sChannel[nChannel].m_psEAS may be changed - * gsSynthObject.m_sChannel[nChannel].m_bRhythmChannel may be changed - * - *---------------------------------------------------------------------------- -*/ -void VMProgramChange (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 program); - -/*---------------------------------------------------------------------------- - * VMChannelPressure() - *---------------------------------------------------------------------------- - * Purpose: - * Change the channel pressure for the given channel - * - * Inputs: - * nChannel - the MIDI channel - * nVelocity - the channel pressure value - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * Side Effects: - * gsSynthObject.m_sChannel[nChannel].m_nChannelPressure is updated - *---------------------------------------------------------------------------- -*/ -void VMChannelPressure (S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 value); - -/*---------------------------------------------------------------------------- - * VMPitchBend() - *---------------------------------------------------------------------------- - * Purpose: - * Change the pitch wheel value for the given channel. - * This routine constructs the proper 14-bit argument when the calling routine - * passes the pitch LSB and MSB. - * - * Note: some midi disassemblers display a bipolar pitch bend value. - * We can display the bipolar value using - * if m_nPitchBend >= 0x2000 - * bipolar pitch bend = postive (m_nPitchBend - 0x2000) - * else - * bipolar pitch bend = negative (0x2000 - m_nPitchBend) - * - * Inputs: - * nChannel - the MIDI channel - * nPitchLSB - the LSB byte from the pitch bend message - * nPitchMSB - the MSB byte from the message - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * gsSynthObject.m_sChannel[nChannel].m_nPitchBend is changed - * - *---------------------------------------------------------------------------- -*/ -void VMPitchBend (S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 pitchLSB, EAS_U8 pitchMSB); - -/*---------------------------------------------------------------------------- - * VMControlChange() - *---------------------------------------------------------------------------- - * Purpose: - * Change the controller (or mode) for the given channel. - * - * Inputs: - * nChannel - the MIDI channel - * nControllerNumber - the controller number - * nControlValue - the controller number for this control change - * nControlValue - the value for this control change - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * Side Effects: - * gsSynthObject.m_sChannel[nChannel] controller is changed - * - *---------------------------------------------------------------------------- -*/ -void VMControlChange (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 controller, EAS_U8 value); - -/*---------------------------------------------------------------------------- - * VMUpdateRPNStateMachine() - *---------------------------------------------------------------------------- - * Purpose: - * Call this function when we want to parse a stream of RPN messages. - * NOTE: The synth has only one set of global RPN data instead of RPN data - * per channel. - * So actually, we don't really need to look at the nChannel parameter, - * but we pass it to facilitate future upgrades. Furthermore, we only - * support RPN0 (pitch bend sensitivity), RPN1 (fine tuning) and - * RPN2 (coarse tuning). Any other RPNs are rejected. - * - * Inputs: - * nChannel - the MIDI channel - * nControllerNumber - the RPN controller number - * nControlValue - the value for this control change - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * gsSynthObject.m_RPN0 (or m_RPN1 or m_RPN2) may be updated if the - * proper RPN message sequence is parsed. - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMUpdateRPNStateMachine (S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 controller, EAS_U8 value); - -/*---------------------------------------------------------------------------- - * VMUpdateStaticChannelParameters() - *---------------------------------------------------------------------------- - * Purpose: - * Update all of the static channel parameters for channels that have had - * a controller change values - * Or if the synth has signalled that all channels must forcibly - * be updated - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * none - * - * Side Effects: - * - psSynthObject->m_sChannel[].m_nStaticGain and m_nStaticPitch - * are updated for channels whose controller values have changed - * or if the synth has signalled that all channels must forcibly - * be updated - *---------------------------------------------------------------------------- -*/ -void VMUpdateStaticChannelParameters (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth); - -/*---------------------------------------------------------------------------- - * VMReleaseAllDeferredNoteOffs() - *---------------------------------------------------------------------------- - * Purpose: - * Call this functin when the sustain flag is presently set but - * we are now transitioning from damper pedal on to - * damper pedal off. This means all notes in this channel - * that received a note off while the damper pedal was on, and - * had their note-off requests deferred, should now proceed to - * the release state. - * - * Inputs: - * nChannel - this channel has its sustain pedal transitioning from on to off - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * Side Effects: - * any voice with deferred note offs on this channel are updated such that - * - * - *---------------------------------------------------------------------------- -*/ -void VMReleaseAllDeferredNoteOffs (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel); - -/*---------------------------------------------------------------------------- - * VMCatchNotesForSustainPedal() - *---------------------------------------------------------------------------- - * Purpose: - * Call this function when the sustain flag is presently clear and - * the damper pedal is off and we are transitioning from damper pedal OFF to - * damper pedal ON. Currently sounding notes should be left - * unchanged. However, we should try to "catch" notes if possible. - * If any notes have levels >= sustain level, catch them, - * otherwise, let them continue to release. - * - * Inputs: - * nChannel - this channel has its sustain pedal transitioning from on to off - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * Side Effects: - * any voice with deferred note offs on this channel are updated such that - * psVoice->m_sEG1.m_eState = eEnvelopeStateSustainPedal - *---------------------------------------------------------------------------- -*/ -void VMCatchNotesForSustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel); - -/*---------------------------------------------------------------------------- - * VMUpdateAllNotesAge() - *---------------------------------------------------------------------------- - * Purpose: - * Increment the note age for all voices older than the age of the voice - * that is stopping, effectively making the voices "younger". - * - * Inputs: - * nAge - age of voice that is going away - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * m_nAge for some voices is incremented - *---------------------------------------------------------------------------- -*/ -void VMUpdateAllNotesAge (S_VOICE_MGR *pVoiceMgr, EAS_U16 nAge); - -/*---------------------------------------------------------------------------- - * VMFindRegionIndex() - *---------------------------------------------------------------------------- - * Purpose: - * Find the region index for the given instrument using the midi key number - * and the RPN2 (coarse tuning) value. By using RPN2 as part of the - * region selection process, we reduce the amount a given sample has - * to be transposed by selecting the closest recorded root instead. - * - * Inputs: - * nChannel - current channel for this note - * nKeyNumber - current midi note number - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * pnRegionIndex - valid only if we returned success - * success if we found the region index number, otherwise - * failure - * - * Side Effects: - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMFindRegionIndex (S_VOICE_MGR *pVoiceMgr, EAS_U8 channel, EAS_U8 note, EAS_U16 *pRegionIndex); - -/*---------------------------------------------------------------------------- - * VMIncRefCount() - *---------------------------------------------------------------------------- - * Increment reference count for virtual synth - *---------------------------------------------------------------------------- -*/ -void VMIncRefCount (S_SYNTH *pSynth); - -/*---------------------------------------------------------------------------- - * VMReset() - *---------------------------------------------------------------------------- - * Purpose: - * We call this routine to start the process of reseting the synth. - * This routine sets a flag for the entire synth indicating that we want - * to reset. - * We also force all voices to mute quickly. - * However, we do not actually perform any synthesis in this routine. That - * is, we do not ramp the voices down from this routine, but instead, we - * let the "regular" synth processing steps take care of adding the ramp - * down samples to the output buffer. After we are sure that all voices - * have completed ramping down, we continue the process of resetting the - * synth (from another routine). - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - set a flag (in gsSynthObject.m_nFlags) indicating synth reset requested. - * - force all voices to update their envelope states to mute - * - *---------------------------------------------------------------------------- -*/ -void VMReset (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_BOOL force); - -/*---------------------------------------------------------------------------- - * VMMuteAllVoices() - *---------------------------------------------------------------------------- - * Purpose: - * We call this in an emergency reset situation. - * This forces all voices to mute quickly. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - forces all voices to update their envelope states to mute - * - *---------------------------------------------------------------------------- -*/ -void VMMuteVoice (S_VOICE_MGR *pVoiceMgr, EAS_I32 voiceNum); -void VMMuteAllVoices (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth); - -/*---------------------------------------------------------------------------- - * VMReleaseAllVoices() - *---------------------------------------------------------------------------- - * Purpose: - * We call this after we've encountered the end of the Midi file. - * This ensures all voice are either in release (because we received their - * note off already) or forces them to mute quickly. - * We use this as a safety to prevent bad midi files from playing forever. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - forces all voices to update their envelope states to release or mute - * - *---------------------------------------------------------------------------- -*/ -void VMReleaseAllVoices (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth); - -/*---------------------------------------------------------------------------- - * VMAllNotesOff() - *---------------------------------------------------------------------------- - * Purpose: - * Quickly mute all notes on the given channel. - * - * Inputs: - * nChannel - quickly turn off all notes on this channel - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - forces all voices on this channel to update their envelope states to mute - * - *---------------------------------------------------------------------------- -*/ -void VMAllNotesOff (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel); - -/*---------------------------------------------------------------------------- - * VMDeferredStopNote() - *---------------------------------------------------------------------------- - * Purpose: - * Stop the notes that had deferred note-off requests. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * None. - * - * Side Effects: - * voices that have had deferred note-off requests are now put into release - * gsSynthObject.m_sVoice[i].m_nFlags has the VOICE_FLAG_DEFER_MIDI_NOTE_OFF - * cleared - *---------------------------------------------------------------------------- -*/ -void VMDeferredStopNote (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth); - -/*---------------------------------------------------------------------------- - * VMSetSynthPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Set the synth to a new polyphony value. Value must be >= 1 and - * <= MAX_SYNTH_VOICES. This function will pin the polyphony at those limits - * - * Inputs: - * pVoiceMgr pointer to synthesizer data - * synth synthesizer number (0 = onboard, 1 = DSP) - * polyphonyCount desired polyphony count - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMSetSynthPolyphony (S_VOICE_MGR *pVoiceMgr, EAS_I32 synth, EAS_I32 polyphonyCount); - -/*---------------------------------------------------------------------------- - * VMGetSynthPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Set the synth to a new polyphony value. Value must be >= 1 and - * <= MAX_SYNTH_VOICES. This function will pin the polyphony at those limits - * - * Inputs: - * pVoiceMgr pointer to synthesizer data - * synth synthesizer number (0 = onboard, 1 = DSP) - * polyphonyCount desired polyphony count - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMGetSynthPolyphony (S_VOICE_MGR *pVoiceMgr, EAS_I32 synth, EAS_I32 *pPolyphonyCount); - -/*---------------------------------------------------------------------------- - * VMSetPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Set the virtual synth polyphony. 0 = no limit (i.e. can use - * all available voices). - * - * Inputs: - * pVoiceMgr pointer to synthesizer data - * polyphonyCount desired polyphony count - * pSynth pointer to virtual synth - * - * Outputs: - * Returns error code - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMSetPolyphony (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 polyphonyCount); - -/*---------------------------------------------------------------------------- - * VMGetPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current polyphony setting - * - * Inputs: - * pVoiceMgr pointer to synthesizer data - * pSynth pointer to virtual synth - * pPolyphonyCount pointer to variable to receive data - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMGetPolyphony (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 *pPolyphonyCount); - -/*---------------------------------------------------------------------------- - * VMSetPriority() - *---------------------------------------------------------------------------- - * Purpose: - * Set the virtual synth priority - * - * Inputs: - * pVoiceMgr pointer to synthesizer data - * priority new priority - * pSynth pointer to virtual synth - * - * Outputs: - * Returns error code - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMSetPriority (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 priority); - -/*---------------------------------------------------------------------------- - * VMGetPriority() - *---------------------------------------------------------------------------- - * Purpose: - * Get the virtual synth priority - * - * Inputs: - * pVoiceMgr pointer to synthesizer data - * pPriority pointer to variable to hold priority - * pSynth pointer to virtual synth - * - * Outputs: - * Returns error code - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMGetPriority (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 *pPriority); - -/*---------------------------------------------------------------------------- - * VMSetVolume() - *---------------------------------------------------------------------------- - * Purpose: - * Set the master volume for this sequence - * - * Inputs: - * nSynthVolume - the desired master volume - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * - * Side Effects: - * overrides any previously set master volume from sysex - * - *---------------------------------------------------------------------------- -*/ -void VMSetVolume (S_SYNTH *pSynth, EAS_U16 masterVolume); - -/*---------------------------------------------------------------------------- - * VMSetPitchBendRange() - *---------------------------------------------------------------------------- - * Set the pitch bend range for the given channel. - *---------------------------------------------------------------------------- -*/ -void VMSetPitchBendRange (S_SYNTH *pSynth, EAS_INT channel, EAS_I16 pitchBendRange); - -/*---------------------------------------------------------------------------- - * VMSetEASLib() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the pointer to the sound library - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMSetGlobalEASLib (S_VOICE_MGR *pVoiceMgr, EAS_SNDLIB_HANDLE pEAS); -EAS_RESULT VMSetEASLib (S_SYNTH *pSynth, EAS_SNDLIB_HANDLE pEAS); - -#ifdef DLS_SYNTHESIZER -/*---------------------------------------------------------------------------- - * VMSetDLSLib() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the pointer to the sound library - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMSetGlobalDLSLib (EAS_DATA_HANDLE pEASData, EAS_DLSLIB_HANDLE pDLS); -EAS_RESULT VMSetDLSLib (S_SYNTH *pSynth, EAS_DLSLIB_HANDLE pDLS); -#endif - -/*---------------------------------------------------------------------------- - * VMSetTranposition() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the global key transposition used by the synthesizer. - * Transposes all melodic instruments up or down by the specified - * amount. Range is limited to +/-12 semitones. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * transposition - transpose amount (+/-12) - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void VMSetTranposition (S_SYNTH *pSynth, EAS_I32 transposition); - -/*---------------------------------------------------------------------------- - * VMGetTranposition() - *---------------------------------------------------------------------------- - * Purpose: - * Gets the global key transposition used by the synthesizer. - * Transposes all melodic instruments up or down by the specified - * amount. Range is limited to +/-12 semitones. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void VMGetTranposition (S_SYNTH *pSynth, EAS_I32 *pTransposition); - -/*---------------------------------------------------------------------------- - * VMGetNoteCount() - *---------------------------------------------------------------------------- -* Returns the total note count -*---------------------------------------------------------------------------- -*/ -EAS_I32 VMGetNoteCount (S_SYNTH *pSynth); - -/*---------------------------------------------------------------------------- - * VMRender() - *---------------------------------------------------------------------------- - * Purpose: - * This routine renders a frame of audio - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * pVoicesRendered - number of voices rendered this frame - * - * Side Effects: - * sets psMidiObject->m_nMaxWorkloadPerFrame - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMRender (S_VOICE_MGR *pVoiceMgr, EAS_I32 numSamples, EAS_I32 *pMixBuffer, EAS_I32 *pVoicesRendered); - -/*---------------------------------------------------------------------------- - * VMInitWorkload() - *---------------------------------------------------------------------------- - * Purpose: - * Clears the workload counter - * - * Inputs: - * pVoiceMgr - pointer to instance data - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void VMInitWorkload (S_VOICE_MGR *pVoiceMgr); - -/*---------------------------------------------------------------------------- - * VMSetWorkload() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the max workload for a single frame. - * - * Inputs: - * pVoiceMgr - pointer to instance data - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void VMSetWorkload (S_VOICE_MGR *pVoiceMgr, EAS_I32 maxWorkLoad); - -/*---------------------------------------------------------------------------- - * VMCheckWorkload() - *---------------------------------------------------------------------------- - * Purpose: - * Checks to see if work load has been exceeded on this frame. - * - * Inputs: - * pVoiceMgr - pointer to instance data - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_BOOL VMCheckWorkload (S_VOICE_MGR *pVoiceMgr); - -/*---------------------------------------------------------------------------- - * VMActiveVoices() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the number of active voices in the synthesizer. - * - * Inputs: - * pEASData - pointer to instance data - * - * Outputs: - * Returns the number of active voices - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_I32 VMActiveVoices (S_SYNTH *pSynth); - -/*---------------------------------------------------------------------------- - * VMMIDIShutdown() - *---------------------------------------------------------------------------- - * Purpose: - * Clean up any Synth related system issues. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * None - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void VMMIDIShutdown (S_EAS_DATA *pEASData, S_SYNTH *pSynth); - -/*---------------------------------------------------------------------------- - * VMShutdown() - *---------------------------------------------------------------------------- - * Purpose: - * Clean up any Synth related system issues. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * None - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void VMShutdown (S_EAS_DATA *pEASData); - -#ifdef EXTERNAL_AUDIO -/*---------------------------------------------------------------------------- - * EAS_RegExtAudioCallback() - *---------------------------------------------------------------------------- - * Register a callback for external audio processing - *---------------------------------------------------------------------------- -*/ -void VMRegExtAudioCallback (S_SYNTH *pSynth, EAS_VOID_PTR pInstData, EAS_EXT_PRG_CHG_FUNC cbProgChgFunc, EAS_EXT_EVENT_FUNC cbEventFunc); - -/*---------------------------------------------------------------------------- - * VMGetMIDIControllers() - *---------------------------------------------------------------------------- - * Returns the MIDI controller values on the specified channel - *---------------------------------------------------------------------------- -*/ -void VMGetMIDIControllers (S_SYNTH *pSynth, EAS_U8 channel, S_MIDI_CONTROLLERS *pControl); -#endif - -#ifdef _SPLIT_ARCHITECTURE -/*---------------------------------------------------------------------------- - * VMStartFrame() - *---------------------------------------------------------------------------- - * Purpose: - * Starts an audio frame - * - * Inputs: - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_BOOL VMStartFrame (S_EAS_DATA *pEASData); - -/*---------------------------------------------------------------------------- - * VMEndFrame() - *---------------------------------------------------------------------------- - * Purpose: - * Stops an audio frame - * - * Inputs: - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_BOOL VMEndFrame (S_EAS_DATA *pEASData); -#endif - -#endif /* #ifdef _EAS_VM_PROTOS_H */ - diff --git a/arm-hybrid-22k/lib_src/eas_voicemgt.c b/arm-hybrid-22k/lib_src/eas_voicemgt.c deleted file mode 100644 index 873f29d..0000000 --- a/arm-hybrid-22k/lib_src/eas_voicemgt.c +++ /dev/null @@ -1,3971 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_voicemgt.c - * - * Contents and purpose: - * Implements the synthesizer functions. - * - * Copyright Sonic Network Inc. 2004 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 794 $ - * $Date: 2007-08-01 00:08:48 -0700 (Wed, 01 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -/* includes */ -#include "eas.h" -#include "eas_data.h" -#include "eas_config.h" -#include "eas_report.h" -#include "eas_midictrl.h" -#include "eas_host.h" -#include "eas_synth_protos.h" -#include "eas_vm_protos.h" - -#ifdef DLS_SYNTHESIZER -#include "eas_mdls.h" -#endif - -// #define _DEBUG_VM - -/* some defines for workload */ -#define WORKLOAD_AMOUNT_SMALL_INCREMENT 5 -#define WORKLOAD_AMOUNT_START_NOTE 10 -#define WORKLOAD_AMOUNT_STOP_NOTE 10 -#define WORKLOAD_AMOUNT_KEY_GROUP 10 -#define WORKLOAD_AMOUNT_POLY_LIMIT 10 - -/* pointer to base sound library */ -extern S_EAS easSoundLib; - -#ifdef TEST_HARNESS -extern S_EAS easTestLib; -EAS_SNDLIB_HANDLE VMGetLibHandle(EAS_INT libNum) -{ - switch (libNum) - { - case 0: - return &easSoundLib; -#ifdef _WT_SYNTH - case 1: - return &easTestLib; -#endif - default: - return NULL; - } -} -#endif - -/* pointer to synthesizer interface(s) */ -#ifdef _WT_SYNTH -extern const S_SYNTH_INTERFACE wtSynth; -#endif - -#ifdef _FM_SYNTH -extern const S_SYNTH_INTERFACE fmSynth; -#endif - -typedef S_SYNTH_INTERFACE *S_SYNTH_INTERFACE_HANDLE; - -/* wavetable on MCU */ -#if defined(EAS_WT_SYNTH) -const S_SYNTH_INTERFACE *const pPrimarySynth = &wtSynth; - -/* FM on MCU */ -#elif defined(EAS_FM_SYNTH) -const S_SYNTH_INTERFACE *const pPrimarySynth = &fmSynth; - -/* wavetable drums on MCU, FM melodic on DSP */ -#elif defined(EAS_HYBRID_SYNTH) -const S_SYNTH_INTERFACE *const pPrimarySynth = &wtSynth; -const S_SYNTH_INTERFACE *const pSecondarySynth = &fmSynth; - -/* wavetable drums on MCU, wavetable melodic on DSP */ -#elif defined(EAS_SPLIT_WT_SYNTH) -const S_SYNTH_INTERFACE *const pPrimarySynth = &wtSynth; -extern const S_FRAME_INTERFACE wtFrameInterface; -const S_FRAME_INTERFACE *const pFrameInterface = &wtFrameInterface; - -/* wavetable drums on MCU, FM melodic on DSP */ -#elif defined(EAS_SPLIT_HYBRID_SYNTH) -const S_SYNTH_INTERFACE *const pPrimarySynth = &wtSynth; -const S_SYNTH_INTERFACE *const pSecondarySynth = &fmSynth; -extern const S_FRAME_INTERFACE fmFrameInterface; -const S_FRAME_INTERFACE *const pFrameInterface = &fmFrameInterface; - -/* FM on DSP */ -#elif defined(EAS_SPLIT_FM_SYNTH) -const S_SYNTH_INTERFACE *const pPrimarySynth = &fmSynth; -extern const S_FRAME_INTERFACE fmFrameInterface; -const S_FRAME_INTERFACE *const pFrameInterface = &fmFrameInterface; - -#else -#error "Undefined architecture option" -#endif - -/*---------------------------------------------------------------------------- - * inline functions - *---------------------------------------------------------------------------- -*/ -EAS_INLINE const S_REGION* GetRegionPtr (S_SYNTH *pSynth, EAS_U16 regionIndex) -{ -#if defined(DLS_SYNTHESIZER) - if (regionIndex & FLAG_RGN_IDX_DLS_SYNTH) - return &pSynth->pDLS->pDLSRegions[regionIndex & REGION_INDEX_MASK].wtRegion.region; -#endif -#if defined(_HYBRID_SYNTH) - if (regionIndex & FLAG_RGN_IDX_FM_SYNTH) - return &pSynth->pEAS->pFMRegions[regionIndex & REGION_INDEX_MASK].region; - else - return &pSynth->pEAS->pWTRegions[regionIndex].region; -#elif defined(_WT_SYNTH) - return &pSynth->pEAS->pWTRegions[regionIndex].region; -#elif defined(_FM_SYNTH) - return &pSynth->pEAS->pFMRegions[regionIndex].region; -#endif -} - -/*lint -esym(715, voiceNum) used in some implementation */ -EAS_INLINE const S_SYNTH_INTERFACE* GetSynthPtr (EAS_INT voiceNum) -{ -#if defined(_HYBRID_SYNTH) - if (voiceNum < NUM_PRIMARY_VOICES) - return pPrimarySynth; - else - return pSecondarySynth; -#else - return pPrimarySynth; -#endif -} - -EAS_INLINE EAS_INT GetAdjustedVoiceNum (EAS_INT voiceNum) -{ -#if defined(_HYBRID_SYNTH) - if (voiceNum >= NUM_PRIMARY_VOICES) - return voiceNum - NUM_PRIMARY_VOICES; -#endif - return voiceNum; -} - -EAS_INLINE EAS_U8 VSynthToChannel (S_SYNTH *pSynth, EAS_U8 channel) -{ - /*lint -e{734} synthNum is always 0-15 */ - return channel | (pSynth->vSynthNum << 4); -} - -/*---------------------------------------------------------------------------- - * InitVoice() - *---------------------------------------------------------------------------- - * Initialize a synthesizer voice - *---------------------------------------------------------------------------- -*/ -void InitVoice (S_SYNTH_VOICE *pVoice) -{ - pVoice->channel = UNASSIGNED_SYNTH_CHANNEL; - pVoice->nextChannel = UNASSIGNED_SYNTH_CHANNEL; - pVoice->note = pVoice->nextNote = DEFAULT_KEY_NUMBER; - pVoice->velocity = pVoice->nextVelocity = DEFAULT_VELOCITY; - pVoice->regionIndex = DEFAULT_REGION_INDEX; - pVoice->age = DEFAULT_AGE; - pVoice->voiceFlags = DEFAULT_VOICE_FLAGS; - pVoice->voiceState = DEFAULT_VOICE_STATE; -} - -/*---------------------------------------------------------------------------- - * IncVoicePoolCount() - *---------------------------------------------------------------------------- - * Updates the voice pool count when a voice changes state - *---------------------------------------------------------------------------- -*/ -static void IncVoicePoolCount (S_VOICE_MGR *pVoiceMgr, S_SYNTH_VOICE *pVoice) -{ - S_SYNTH *pSynth; - EAS_INT pool; - - /* ignore muting voices */ - if (pVoice->voiceState == eVoiceStateMuting) - return; - - if (pVoice->voiceState == eVoiceStateStolen) - { - pSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->nextChannel)]; - pool = pSynth->channels[GET_CHANNEL(pVoice->nextChannel)].pool; - } - else - { - pSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->channel)]; - pool = pSynth->channels[GET_CHANNEL(pVoice->channel)].pool; - } - - pSynth->poolCount[pool]++; - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IncVoicePoolCount: Synth=%d pool=%d\n", pSynth->vSynthNum, pool); */ } -#endif -} - -/*---------------------------------------------------------------------------- - * DecVoicePoolCount() - *---------------------------------------------------------------------------- - * Updates the voice pool count when a voice changes state - *---------------------------------------------------------------------------- -*/ -static void DecVoicePoolCount (S_VOICE_MGR *pVoiceMgr, S_SYNTH_VOICE *pVoice) -{ - S_SYNTH *pSynth; - EAS_INT pool; - - /* ignore muting voices */ - if (pVoice->voiceState == eVoiceStateMuting) - return; - - if (pVoice->voiceState == eVoiceStateStolen) - { - pSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->nextChannel)]; - pool = pSynth->channels[GET_CHANNEL(pVoice->nextChannel)].pool; - } - else - { - pSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->channel)]; - pool = pSynth->channels[GET_CHANNEL(pVoice->channel)].pool; - } - - pSynth->poolCount[pool]--; - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "DecVoicePoolCount: Synth=%d pool=%d\n", pSynth->vSynthNum, pool); */ } -#endif -} - -/*---------------------------------------------------------------------------- - * VMInitialize() - *---------------------------------------------------------------------------- - * Purpose: - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMInitialize (S_EAS_DATA *pEASData) -{ - S_VOICE_MGR *pVoiceMgr; - EAS_INT i; - - /* check Configuration Module for data allocation */ - if (pEASData->staticMemoryModel) - pVoiceMgr = EAS_CMEnumData(EAS_CM_SYNTH_DATA); - else - pVoiceMgr = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_VOICE_MGR)); - if (!pVoiceMgr) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitialize: Failed to allocate synthesizer memory\n"); */ } - return EAS_ERROR_MALLOC_FAILED; - } - EAS_HWMemSet(pVoiceMgr, 0, sizeof(S_VOICE_MGR)); - - /* initialize non-zero variables */ - pVoiceMgr->pGlobalEAS = (S_EAS*) &easSoundLib; - pVoiceMgr->maxPolyphony = (EAS_U16) MAX_SYNTH_VOICES; - -#if defined(_SECONDARY_SYNTH) || defined(EAS_SPLIT_WT_SYNTH) - pVoiceMgr->maxPolyphonyPrimary = NUM_PRIMARY_VOICES; - pVoiceMgr->maxPolyphonySecondary = NUM_SECONDARY_VOICES; -#endif - - /* set max workload to zero */ - pVoiceMgr->maxWorkLoad = 0; - - /* initialize the voice manager parameters */ - for (i = 0; i < MAX_SYNTH_VOICES; i++) - InitVoice(&pVoiceMgr->voices[i]); - - /* initialize the synth */ - /*lint -e{522} return unused at this time */ - pPrimarySynth->pfInitialize(pVoiceMgr); - - /* initialize the off-chip synth */ -#ifdef _HYBRID_SYNTH - /*lint -e{522} return unused at this time */ - pSecondarySynth->pfInitialize(pVoiceMgr); -#endif - - pEASData->pVoiceMgr = pVoiceMgr; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * VMInitMIDI() - *---------------------------------------------------------------------------- - * Purpose: - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMInitMIDI (S_EAS_DATA *pEASData, S_SYNTH **ppSynth) -{ - EAS_RESULT result; - S_SYNTH *pSynth; - EAS_INT virtualSynthNum; - - *ppSynth = NULL; - - /* static memory model only allows one synth */ - if (pEASData->staticMemoryModel) - { - if (pEASData->pVoiceMgr->pSynth[0] != NULL) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitMIDI: No virtual synthesizer support for static memory model\n"); */ } - return EAS_ERROR_NO_VIRTUAL_SYNTHESIZER; - } - - /* check Configuration Module for data allocation */ - pSynth = EAS_CMEnumData(EAS_CM_MIDI_DATA); - virtualSynthNum = 0; - } - - /* dynamic memory model */ - else - { - for (virtualSynthNum = 0; virtualSynthNum < MAX_VIRTUAL_SYNTHESIZERS; virtualSynthNum++) - if (pEASData->pVoiceMgr->pSynth[virtualSynthNum] == NULL) - break; - if (virtualSynthNum == MAX_VIRTUAL_SYNTHESIZERS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitMIDI: Exceeded number of active virtual synthesizers"); */ } - return EAS_ERROR_NO_VIRTUAL_SYNTHESIZER; - } - pSynth = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_SYNTH)); - } - - /* make sure we have a valid memory pointer */ - if (pSynth == NULL) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitMIDI: Failed to allocate synthesizer memory\n"); */ } - return EAS_ERROR_MALLOC_FAILED; - } - EAS_HWMemSet(pSynth, 0, sizeof(S_SYNTH)); - - /* set the sound library pointer */ - if ((result = VMSetEASLib(pSynth, pEASData->pVoiceMgr->pGlobalEAS)) != EAS_SUCCESS) - { - VMMIDIShutdown(pEASData, pSynth); - return result; - } - - /* link in DLS bank if downloaded */ -#ifdef DLS_SYNTHESIZER - if (pEASData->pVoiceMgr->pGlobalDLS) - { - pSynth->pDLS = pEASData->pVoiceMgr->pGlobalDLS; - DLSAddRef(pSynth->pDLS); - } -#endif - - /* initialize MIDI state variables */ - pSynth->synthFlags = DEFAULT_SYNTH_FLAGS; - pSynth->masterVolume = DEFAULT_SYNTH_MASTER_VOLUME; - pSynth->refCount = 1; - pSynth->priority = DEFAULT_SYNTH_PRIORITY; - pSynth->poolAlloc[0] = (EAS_U8) pEASData->pVoiceMgr->maxPolyphony; - - VMInitializeAllChannels(pEASData->pVoiceMgr, pSynth); - - pSynth->vSynthNum = (EAS_U8) virtualSynthNum; - pEASData->pVoiceMgr->pSynth[virtualSynthNum] = pSynth; - - *ppSynth = pSynth; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * VMIncRefCount() - *---------------------------------------------------------------------------- - * Increment reference count for virtual synth - *---------------------------------------------------------------------------- -*/ -void VMIncRefCount (S_SYNTH *pSynth) -{ - pSynth->refCount++; -} - -/*---------------------------------------------------------------------------- - * VMReset() - *---------------------------------------------------------------------------- - * Purpose: - * We call this routine to start the process of reseting the synth. - * This routine sets a flag for the entire synth indicating that we want - * to reset. - * We also force all voices to mute quickly. - * However, we do not actually perform any synthesis in this routine. That - * is, we do not ramp the voices down from this routine, but instead, we - * let the "regular" synth processing steps take care of adding the ramp - * down samples to the output buffer. After we are sure that all voices - * have completed ramping down, we continue the process of resetting the - * synth (from another routine). - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * force - force reset even if voices are active - * - * Outputs: - * - * Side Effects: - * - set a flag (in psSynthObject->m_nFlags) indicating synth reset requested. - * - force all voices to update their envelope states to mute - * - *---------------------------------------------------------------------------- -*/ -void VMReset (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_BOOL force) -{ - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMReset: request to reset synth. Force = %d\n", force); */ } -#endif - - /* force voices to off state - may cause audio artifacts */ - if (force) - { - pVoiceMgr->activeVoices -= pSynth->numActiveVoices; - pSynth->numActiveVoices = 0; - VMInitializeAllVoices(pVoiceMgr, pSynth->vSynthNum); - } - else - VMMuteAllVoices(pVoiceMgr, pSynth); - - /* don't reset if voices are still playing */ - if (pSynth->numActiveVoices == 0) - { - EAS_INT i; - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMReset: complete the reset process\n"); */ } -#endif - - VMInitializeAllChannels(pVoiceMgr, pSynth); - for (i = 0; i < NUM_SYNTH_CHANNELS; i++) - pSynth->poolCount[i] = 0; - - /* set polyphony */ - if (pSynth->maxPolyphony < pVoiceMgr->maxPolyphony) - pSynth->poolAlloc[0] = (EAS_U8) pVoiceMgr->maxPolyphony; - else - pSynth->poolAlloc[0] = (EAS_U8) pSynth->maxPolyphony; - - /* clear reset flag */ - pSynth->synthFlags &= ~SYNTH_FLAG_RESET_IS_REQUESTED; - } - - /* handle reset after voices are muted */ - else - pSynth->synthFlags |= SYNTH_FLAG_RESET_IS_REQUESTED; -} - -/*---------------------------------------------------------------------------- - * VMInitializeAllChannels() - *---------------------------------------------------------------------------- - * Purpose: - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -void VMInitializeAllChannels (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth) -{ - S_SYNTH_CHANNEL *pChannel; - EAS_INT i; - - VMResetControllers(pSynth); - - /* init each channel */ - pChannel = pSynth->channels; - - for (i = 0; i < NUM_SYNTH_CHANNELS; i++, pChannel++) - { - pChannel->channelFlags = DEFAULT_CHANNEL_FLAGS; - pChannel->staticGain = DEFAULT_CHANNEL_STATIC_GAIN; - pChannel->staticPitch = DEFAULT_CHANNEL_STATIC_PITCH; - pChannel->pool = 0; - - /* the drum channel needs a different init */ - if (i == DEFAULT_DRUM_CHANNEL) - { - pChannel->bankNum = DEFAULT_RHYTHM_BANK_NUMBER; - pChannel->channelFlags |= CHANNEL_FLAG_RHYTHM_CHANNEL; - } - else - pChannel->bankNum = DEFAULT_MELODY_BANK_NUMBER; - - VMProgramChange(pVoiceMgr, pSynth, (EAS_U8) i, DEFAULT_SYNTH_PROGRAM_NUMBER); - } - -} - -/*---------------------------------------------------------------------------- - * VMResetControllers() - *---------------------------------------------------------------------------- - * Purpose: - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -void VMResetControllers (S_SYNTH *pSynth) -{ - S_SYNTH_CHANNEL *pChannel; - EAS_INT i; - - pChannel = pSynth->channels; - - for (i = 0; i < NUM_SYNTH_CHANNELS; i++, pChannel++) - { - pChannel->pitchBend = DEFAULT_PITCH_BEND; - pChannel->modWheel = DEFAULT_MOD_WHEEL; - pChannel->volume = DEFAULT_CHANNEL_VOLUME; - pChannel->pan = DEFAULT_PAN; - pChannel->expression = DEFAULT_EXPRESSION; - -#ifdef _REVERB - pSynth->channels[i].reverbSend = DEFAULT_REVERB_SEND; -#endif - -#ifdef _CHORUS - pSynth->channels[i].chorusSend = DEFAULT_CHORUS_SEND; -#endif - - pChannel->channelPressure = DEFAULT_CHANNEL_PRESSURE; - pChannel->registeredParam = DEFAULT_REGISTERED_PARAM; - pChannel->pitchBendSensitivity = DEFAULT_PITCH_BEND_SENSITIVITY; - pChannel->finePitch = DEFAULT_FINE_PITCH; - pChannel->coarsePitch = DEFAULT_COARSE_PITCH; - - /* update all voices on this channel */ - pChannel->channelFlags |= CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS; - } -} - -/*---------------------------------------------------------------------------- - * VMInitializeAllVoices() - *---------------------------------------------------------------------------- - * Purpose: - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -void VMInitializeAllVoices (S_VOICE_MGR *pVoiceMgr, EAS_INT vSynthNum) -{ - EAS_INT i; - - /* initialize the voice manager parameters */ - for (i = 0; i < MAX_SYNTH_VOICES; i++) - { - if (pVoiceMgr->voices[i].voiceState != eVoiceStateStolen) - { - if (GET_VSYNTH(pVoiceMgr->voices[i].channel) == vSynthNum) - InitVoice(&pVoiceMgr->voices[i]); - } - else - { - if (GET_VSYNTH(pVoiceMgr->voices[i].nextChannel) == vSynthNum) - InitVoice(&pVoiceMgr->voices[i]); - } - } -} - -/*---------------------------------------------------------------------------- - * VMMuteVoice() - *---------------------------------------------------------------------------- - * Mute the selected voice - *---------------------------------------------------------------------------- -*/ -void VMMuteVoice (S_VOICE_MGR *pVoiceMgr, EAS_I32 voiceNum) -{ - S_SYNTH *pSynth; - S_SYNTH_VOICE *pVoice; - - /* take no action if voice is already muted */ - pVoice = &pVoiceMgr->voices[voiceNum]; - if ((pVoice->voiceState == eVoiceStateMuting) || (pVoice->voiceState == eVoiceStateFree)) - return; - - /* one less voice in pool */ - DecVoicePoolCount(pVoiceMgr, pVoice); - - pSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->channel)]; - GetSynthPtr(voiceNum)->pfMuteVoice(pVoiceMgr, pSynth, pVoice, GetAdjustedVoiceNum(voiceNum)); - pVoice->voiceState = eVoiceStateMuting; - -} - -/*---------------------------------------------------------------------------- - * VMReleaseVoice() - *---------------------------------------------------------------------------- - * Release the selected voice - *---------------------------------------------------------------------------- -*/ -void VMReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 voiceNum) -{ - S_SYNTH_VOICE *pVoice = &pVoiceMgr->voices[voiceNum]; - - /* take no action if voice is already free, muting, or releasing */ - if (( pVoice->voiceState == eVoiceStateMuting) || - (pVoice->voiceState == eVoiceStateFree) || - (pVoice->voiceState == eVoiceStateRelease)) - return; - - /* stolen voices should just be muted */ - if (pVoice->voiceState == eVoiceStateStolen) - VMMuteVoice(pVoiceMgr, voiceNum); - - /* release this voice */ - GetSynthPtr(voiceNum)->pfReleaseVoice(pVoiceMgr, pSynth, &pVoiceMgr->voices[voiceNum], GetAdjustedVoiceNum(voiceNum)); - pVoice->voiceState = eVoiceStateRelease; -} - -/*---------------------------------------------------------------------------- - * VMInitMIPTable() - *---------------------------------------------------------------------------- - * Initialize the SP-MIDI MIP table in preparation for receiving MIP message - *---------------------------------------------------------------------------- -*/ -void VMInitMIPTable (S_SYNTH *pSynth) -{ - EAS_INT i; - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMInitMIPTable\n"); */ } -#endif - - /* clear SP-MIDI flag */ - pSynth->synthFlags &= ~SYNTH_FLAG_SP_MIDI_ON; - for (i = 0; i < NUM_SYNTH_CHANNELS; i++) - { - pSynth->channels[i].pool = 0; - pSynth->channels[i].mip = 0; - } -} - -/*---------------------------------------------------------------------------- - * VMSetMIPEntry() - *---------------------------------------------------------------------------- - * Sets the priority and MIP level for a MIDI channel - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pVoiceMgr) reserved for future use */ -void VMSetMIPEntry (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 priority, EAS_U8 mip) -{ - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMSetMIPEntry: channel=%d, priority=%d, MIP=%d\n", channel, priority, mip); */ } -#endif - - /* save data for use by MIP message processing */ - if (priority < NUM_SYNTH_CHANNELS) - { - pSynth->channels[channel].pool = priority; - pSynth->channels[channel].mip = mip; - } -} - -/*---------------------------------------------------------------------------- - * VMMIPUpdateChannelMuting() - *---------------------------------------------------------------------------- - * This routine is called after an SP-MIDI message is received and - * any time the allocated polyphony changes. It mutes or unmutes - * channels based on polyphony. - *---------------------------------------------------------------------------- -*/ -void VMMIPUpdateChannelMuting (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth) -{ - EAS_INT i; - EAS_INT maxPolyphony; - EAS_INT channel; - EAS_INT vSynthNum; - EAS_INT pool; - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMUpdateMIPTable\n"); */ } -#endif - - /* determine max polyphony */ - if (pSynth->maxPolyphony) - maxPolyphony = pSynth->maxPolyphony; - else - maxPolyphony = pVoiceMgr->maxPolyphony; - - /* process channels */ - for (i = 0; i < NUM_SYNTH_CHANNELS; i++) - { - - /* channel must be in MIP message and must meet allocation target */ - if ((pSynth->channels[i].mip != 0) && (pSynth->channels[i].mip <= maxPolyphony)) - pSynth->channels[i].channelFlags &= ~CHANNEL_FLAG_MUTE; - else - pSynth->channels[i].channelFlags |= CHANNEL_FLAG_MUTE; - - /* reset voice pool count */ - pSynth->poolCount[i] = 0; - } - - /* mute any voices on muted channels, and count unmuted voices */ - for (i = 0; i < MAX_SYNTH_VOICES; i++) - { - - /* ignore free voices */ - if (pVoiceMgr->voices[i].voiceState == eVoiceStateFree) - continue; - - /* get channel and virtual synth */ - if (pVoiceMgr->voices[i].voiceState != eVoiceStateStolen) - { - vSynthNum = GET_VSYNTH(pVoiceMgr->voices[i].channel); - channel = GET_CHANNEL(pVoiceMgr->voices[i].channel); - } - else - { - vSynthNum = GET_VSYNTH(pVoiceMgr->voices[i].nextChannel); - channel = GET_CHANNEL(pVoiceMgr->voices[i].nextChannel); - } - - /* ignore voices on other synths */ - if (vSynthNum != pSynth->vSynthNum) - continue; - - /* count voices */ - pool = pSynth->channels[channel].pool; - - /* deal with muted channels */ - if (pSynth->channels[channel].channelFlags & CHANNEL_FLAG_MUTE) - { - /* mute stolen voices scheduled to play on this channel */ - if (pVoiceMgr->voices[i].voiceState == eVoiceStateStolen) - pVoiceMgr->voices[i].voiceState = eVoiceStateMuting; - - /* release voices that aren't already muting */ - else if (pVoiceMgr->voices[i].voiceState != eVoiceStateMuting) - { - VMReleaseVoice(pVoiceMgr, pSynth, i); - pSynth->poolCount[pool]++; - } - } - - /* not muted, count this voice */ - else - pSynth->poolCount[pool]++; - } -} - -/*---------------------------------------------------------------------------- - * VMUpdateMIPTable() - *---------------------------------------------------------------------------- - * This routine is called at the end of the SysEx message to allow - * the Voice Manager to complete the initialization of the MIP - * table. It assigns channels to the appropriate voice pool based - * on the MIP setting and calculates the voices allocated for each - * pool. - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pVoiceMgr) reserved for future use */ -void VMUpdateMIPTable (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth) -{ - S_SYNTH_CHANNEL *pChannel; - EAS_INT i; - EAS_INT currentMIP; - EAS_INT currentPool; - EAS_INT priority[NUM_SYNTH_CHANNELS]; - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMUpdateMIPTable\n"); */ } -#endif - - /* set SP-MIDI flag */ - pSynth->synthFlags |= SYNTH_FLAG_SP_MIDI_ON; - - /* sort channels into priority order */ - for (i = 0; i < NUM_SYNTH_CHANNELS; i++) - priority[i] = -1; - for (i = 0; i < NUM_SYNTH_CHANNELS; i++) - { - if (pSynth->channels[i].pool != DEFAULT_SP_MIDI_PRIORITY) - priority[pSynth->channels[i].pool] = i; - } - - /* process channels in priority order */ - currentMIP = 0; - currentPool = -1; - for (i = 0; i < NUM_SYNTH_CHANNELS; i++) - { - /* stop when we run out of channels */ - if (priority[i] == -1) - break; - - pChannel = &pSynth->channels[priority[i]]; - - /* when 2 or more channels have the same MIP setting, they - * share a common voice pool - */ - if (pChannel->mip == currentMIP) - pChannel->pool = (EAS_U8) currentPool; - - /* new voice pool */ - else - { - currentPool++; - pSynth->poolAlloc[currentPool] = (EAS_U8) (pChannel->mip - currentMIP); - currentMIP = pChannel->mip; - } - } - - /* set SP-MIDI flag */ - pSynth->synthFlags |= SYNTH_FLAG_SP_MIDI_ON; - - /* update channel muting */ - VMMIPUpdateChannelMuting (pVoiceMgr, pSynth); -} - -/*---------------------------------------------------------------------------- - * VMMuteAllVoices() - *---------------------------------------------------------------------------- - * Purpose: - * We call this in an emergency reset situation. - * This forces all voices to mute quickly. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - forces all voices to update their envelope states to mute - * - *---------------------------------------------------------------------------- -*/ -void VMMuteAllVoices (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth) -{ - EAS_INT i; - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMMuteAllVoices: about to mute all voices!!\n"); */ } -#endif - - for (i = 0; i < MAX_SYNTH_VOICES; i++) - { - /* for stolen voices, check new channel */ - if (pVoiceMgr->voices[i].voiceState == eVoiceStateStolen) - { - if (GET_VSYNTH(pVoiceMgr->voices[i].nextChannel) == pSynth->vSynthNum) - VMMuteVoice(pVoiceMgr, i); - } - - else if (pSynth->vSynthNum == GET_VSYNTH(pVoiceMgr->voices[i].channel)) - VMMuteVoice(pVoiceMgr, i); - } -} - -/*---------------------------------------------------------------------------- - * VMReleaseAllVoices() - *---------------------------------------------------------------------------- - * Purpose: - * We call this after we've encountered the end of the Midi file. - * This ensures all voices are either in release (because we received their - * note off already) or forces them to mute quickly. - * We use this as a safety to prevent bad midi files from playing forever. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - forces all voices to update their envelope states to release or mute - * - *---------------------------------------------------------------------------- -*/ -void VMReleaseAllVoices (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth) -{ - EAS_INT i; - - /* release sustain pedal on all channels */ - for (i = 0; i < NUM_SYNTH_CHANNELS; i++) - { - if (pSynth->channels[ i ].channelFlags & CHANNEL_FLAG_SUSTAIN_PEDAL) - { - VMReleaseAllDeferredNoteOffs(pVoiceMgr, pSynth, (EAS_U8) i); - pSynth->channels[i].channelFlags &= ~CHANNEL_FLAG_SUSTAIN_PEDAL; - } - } - - /* release all voices */ - for (i = 0; i < MAX_SYNTH_VOICES; i++) - { - - switch (pVoiceMgr->voices[i].voiceState) - { - case eVoiceStateStart: - case eVoiceStatePlay: - /* only release voices on this synth */ - if (GET_VSYNTH(pVoiceMgr->voices[i].channel) == pSynth->vSynthNum) - VMReleaseVoice(pVoiceMgr, pSynth, i); - break; - - case eVoiceStateStolen: - if (GET_VSYNTH(pVoiceMgr->voices[i].nextChannel) == pSynth->vSynthNum) - VMMuteVoice(pVoiceMgr, i); - break; - - case eVoiceStateFree: - case eVoiceStateRelease: - case eVoiceStateMuting: - break; - - case eVoiceStateInvalid: - default: -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMReleaseAllVoices: error, %d is an unrecognized state\n", - pVoiceMgr->voices[i].voiceState); */ } -#endif - break; - } - } -} - -/*---------------------------------------------------------------------------- - * VMAllNotesOff() - *---------------------------------------------------------------------------- - * Purpose: - * Quickly mute all notes on the given channel. - * - * Inputs: - * nChannel - quickly turn off all notes on this channel - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - forces all voices on this channel to update their envelope states to mute - * - *---------------------------------------------------------------------------- -*/ -void VMAllNotesOff (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel) -{ - EAS_INT voiceNum; - S_SYNTH_VOICE *pVoice; - -#ifdef _DEBUG_VM - if (channel >= NUM_SYNTH_CHANNELS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMAllNotesOff: error, %d invalid channel number\n", - channel); */ } - return; - } -#endif - - /* increment workload */ - pVoiceMgr->workload += WORKLOAD_AMOUNT_SMALL_INCREMENT; - - /* check each voice */ - channel = VSynthToChannel(pSynth, channel); - for (voiceNum = 0; voiceNum < MAX_SYNTH_VOICES; voiceNum++) - { - pVoice = &pVoiceMgr->voices[voiceNum]; - if (pVoice->voiceState != eVoiceStateFree) - { - if (((pVoice->voiceState != eVoiceStateStolen) && (channel == pVoice->channel)) || - ((pVoice->voiceState == eVoiceStateStolen) && (channel == pVoice->nextChannel))) - { - /* this voice is assigned to the requested channel */ - GetSynthPtr(voiceNum)->pfMuteVoice(pVoiceMgr, pSynth, pVoice, GetAdjustedVoiceNum(voiceNum)); - pVoice->voiceState = eVoiceStateMuting; - } - } - } -} - -/*---------------------------------------------------------------------------- - * VMDeferredStopNote() - *---------------------------------------------------------------------------- - * Purpose: - * Stop the notes that had deferred note-off requests. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * None. - * - * Side Effects: - * voices that have had deferred note-off requests are now put into release - * psSynthObject->m_sVoice[i].m_nFlags has the VOICE_FLAG_DEFER_MIDI_NOTE_OFF - * cleared - *---------------------------------------------------------------------------- -*/ -void VMDeferredStopNote (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth) -{ - EAS_INT voiceNum; - EAS_INT channel; - EAS_BOOL deferredNoteOff; - - deferredNoteOff = EAS_FALSE; - - /* check each voice to see if it requires a deferred note off */ - for (voiceNum=0; voiceNum < MAX_SYNTH_VOICES; voiceNum++) - { - if (pVoiceMgr->voices[voiceNum].voiceFlags & VOICE_FLAG_DEFER_MIDI_NOTE_OFF) - { - /* check if this voice was stolen */ - if (pVoiceMgr->voices[voiceNum].voiceState == eVoiceStateStolen) - { - /* - This voice was stolen, AND it also has a deferred note-off. - The stolen note must be completely ramped down at this point. - The note that caused the stealing to occur, however, must - have received a note-off request before the note that caused - stealing ever had a chance to even start. We want to give - the note that caused the stealing a chance to play, so we - start it on the next update interval, and we defer sending - the note-off request until the subsequent update interval. - So do not send the note-off request for this voice because - this voice was stolen and should have completed ramping down, - Also, do not clear the global flag nor this voice's flag - because we must indicate that the subsequent update interval, - after the note that caused stealing has started, should - then send the deferred note-off request. - */ - deferredNoteOff = EAS_TRUE; - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMDeferredStopNote: defer request to stop voice %d (channel=%d note=%d) - voice not started\n", - voiceNum, - pVoiceMgr->voices[voiceNum].nextChannel, - pVoiceMgr->voices[voiceNum].note); */ } - - /* sanity check: this stolen voice better be ramped to zero */ - if (0 != pVoiceMgr->voices[voiceNum].gain) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMDeferredStopNote: warning, this voice did not complete its ramp to zero\n"); */ } - } -#endif // #ifdef _DEBUG_VM - - } - else - { - /* clear the flag using exor */ - pVoiceMgr->voices[voiceNum].voiceFlags ^= - VOICE_FLAG_DEFER_MIDI_NOTE_OFF; - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMDeferredStopNote: Stop voice %d (channel=%d note=%d)\n", - voiceNum, - pVoiceMgr->voices[voiceNum].nextChannel, - pVoiceMgr->voices[voiceNum].note); */ } -#endif - - channel = pVoiceMgr->voices[voiceNum].channel & 15; - - /* check if sustain pedal is on */ - if (pSynth->channels[channel].channelFlags & CHANNEL_FLAG_SUSTAIN_PEDAL) - { - GetSynthPtr(voiceNum)->pfSustainPedal(pVoiceMgr, pSynth, &pVoiceMgr->voices[voiceNum], &pSynth->channels[channel], GetAdjustedVoiceNum(voiceNum)); - } - - /* release this voice */ - else - VMReleaseVoice(pVoiceMgr, pSynth, voiceNum); - - } - - } - - } - - /* clear the deferred note-off flag, unless there's another one pending */ - if (deferredNoteOff == EAS_FALSE) - pSynth->synthFlags ^= SYNTH_FLAG_DEFERRED_MIDI_NOTE_OFF_PENDING; -} - -/*---------------------------------------------------------------------------- - * VMReleaseAllDeferredNoteOffs() - *---------------------------------------------------------------------------- - * Purpose: - * Call this functin when the sustain flag is presently set but - * we are now transitioning from damper pedal on to - * damper pedal off. This means all notes in this channel - * that received a note off while the damper pedal was on, and - * had their note-off requests deferred, should now proceed to - * the release state. - * - * Inputs: - * nChannel - this channel has its sustain pedal transitioning from on to off - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * Side Effects: - * any voice with deferred note offs on this channel are updated such that - * pVoice->m_sEG1.m_eState = eEnvelopeStateRelease - * pVoice->m_sEG1.m_nIncrement = release increment - * pVoice->m_nFlags = clear the deferred note off flag - *---------------------------------------------------------------------------- -*/ -void VMReleaseAllDeferredNoteOffs (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel) -{ - S_SYNTH_VOICE *pVoice; - EAS_INT voiceNum; - -#ifdef _DEBUG_VM - if (channel >= NUM_SYNTH_CHANNELS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMReleaseAllDeferredNoteOffs: error, %d invalid channel number\n", - channel); */ } - return; - } -#endif /* #ifdef _DEBUG_VM */ - - /* increment workload */ - pVoiceMgr->workload += WORKLOAD_AMOUNT_SMALL_INCREMENT; - - /* find all the voices assigned to this channel */ - channel = VSynthToChannel(pSynth, channel); - for (voiceNum = 0; voiceNum < MAX_SYNTH_VOICES; voiceNum++) - { - - pVoice = &pVoiceMgr->voices[voiceNum]; - if (channel == pVoice->channel) - { - - /* does this voice have a deferred note off? */ - if (pVoice->voiceFlags & VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF) - { - /* release voice */ - VMReleaseVoice(pVoiceMgr, pSynth, voiceNum); - - /* use exor to flip bit, clear the flag */ - pVoice->voiceFlags &= ~VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF; - - } - - } - } - - return; -} - -/*---------------------------------------------------------------------------- - * VMCatchNotesForSustainPedal() - *---------------------------------------------------------------------------- - * Purpose: - * Call this function when the sustain flag is presently clear and - * the damper pedal is off and we are transitioning from damper pedal OFF to - * damper pedal ON. Currently sounding notes should be left - * unchanged. However, we should try to "catch" notes if possible. - * If any notes are in release and have levels >= sustain level, catch them, - * otherwise, let them continue to release. - * - * Inputs: - * nChannel - this channel has its sustain pedal transitioning from on to off - * psEASData - pointer to overall EAS data structure - * - * Outputs: - *---------------------------------------------------------------------------- -*/ -void VMCatchNotesForSustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel) -{ - EAS_INT voiceNum; - -#ifdef _DEBUG_VM - if (channel >= NUM_SYNTH_CHANNELS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMCatchNotesForSustainPedal: error, %d invalid channel number\n", - channel); */ } - return; - } -#endif - - pVoiceMgr->workload += WORKLOAD_AMOUNT_SMALL_INCREMENT; - channel = VSynthToChannel(pSynth, channel); - - /* find all the voices assigned to this channel */ - for (voiceNum = 0; voiceNum < MAX_SYNTH_VOICES; voiceNum++) - { - if (channel == pVoiceMgr->voices[voiceNum].channel) - { - if (eVoiceStateRelease == pVoiceMgr->voices[voiceNum].voiceState) - GetSynthPtr(voiceNum)->pfSustainPedal(pVoiceMgr, pSynth, &pVoiceMgr->voices[voiceNum], &pSynth->channels[channel], GetAdjustedVoiceNum(voiceNum)); - } - } -} - -/*---------------------------------------------------------------------------- - * VMUpdateAllNotesAge() - *---------------------------------------------------------------------------- - * Purpose: - * Increment the note age for all of the active voices. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * m_nAge for all voices is incremented - *---------------------------------------------------------------------------- -*/ -void VMUpdateAllNotesAge (S_VOICE_MGR *pVoiceMgr, EAS_U16 age) -{ - EAS_INT i; - - for (i = 0; i < MAX_SYNTH_VOICES; i++) - { - if (age - pVoiceMgr->voices[i].age > 0) - pVoiceMgr->voices[i].age++; - } -} - -/*---------------------------------------------------------------------------- - * VMStolenVoice() - *---------------------------------------------------------------------------- - * Purpose: - * The selected voice is being stolen. Sets the parameters so that the - * voice will begin playing the new sound on the next buffer. - * - * Inputs: - * pVoice - pointer to voice to steal - * nChannel - the channel to start a note on - * nKeyNumber - the key number to start a note for - * nNoteVelocity - the key velocity from this note - * - * Outputs: - * None - *---------------------------------------------------------------------------- -*/ -static void VMStolenVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 voiceNum, EAS_U8 channel, EAS_U8 note, EAS_U8 velocity, EAS_U16 regionIndex) -{ - S_SYNTH_VOICE *pVoice = &pVoiceMgr->voices[voiceNum]; - - /* one less voice in old pool */ - DecVoicePoolCount(pVoiceMgr, pVoice); - - /* mute the sound that is currently playing */ - GetSynthPtr(voiceNum)->pfMuteVoice(pVoiceMgr, pVoiceMgr->pSynth[GET_VSYNTH(pVoice->channel)], &pVoiceMgr->voices[voiceNum], GetAdjustedVoiceNum(voiceNum)); - pVoice->voiceState = eVoiceStateStolen; - - /* set new note data */ - pVoice->nextChannel = VSynthToChannel(pSynth, channel); - pVoice->nextNote = note; - pVoice->nextVelocity = velocity; - pVoice->nextRegionIndex = regionIndex; - - /* one more voice in new pool */ - IncVoicePoolCount(pVoiceMgr, pVoice); - - /* clear the deferred flags */ - pVoice->voiceFlags &= - ~(VOICE_FLAG_DEFER_MIDI_NOTE_OFF | - VOICE_FLAG_DEFER_MUTE | - VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF); - - /* all notes older than this one get "younger" */ - VMUpdateAllNotesAge(pVoiceMgr, pVoice->age); - - /* assign current age to this note and increment for the next note */ - pVoice->age = pVoiceMgr->age++; -} - -/*---------------------------------------------------------------------------- - * VMFreeVoice() - *---------------------------------------------------------------------------- - * Purpose: - * The selected voice is done playing and being returned to the - * pool of free voices - * - * Inputs: - * pVoice - pointer to voice to free - * - * Outputs: - * None - *---------------------------------------------------------------------------- -*/ -static void VMFreeVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice) -{ - - /* do nothing if voice is already free */ - if (pVoice->voiceState == eVoiceStateFree) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "VMFreeVoice: Attempt to free voice that is already free\n"); */ } - return; - } - - /* if we jump directly to free without passing muting stage, - * we need to adjust the voice count */ - DecVoicePoolCount(pVoiceMgr, pVoice); - - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "VMFreeVoice: Synth=%d\n", pSynth->vSynthNum); */ } -#endif - - /* return to free voice pool */ - pVoiceMgr->activeVoices--; - pSynth->numActiveVoices--; - InitVoice(pVoice); - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMFreeVoice: free voice %d\n", pVoice - pVoiceMgr->voices); */ } -#endif - - /* all notes older than this one get "younger" */ - VMUpdateAllNotesAge(pVoiceMgr, pVoice->age); - } - -/*---------------------------------------------------------------------------- - * VMRetargetStolenVoice() - *---------------------------------------------------------------------------- - * Purpose: - * The selected voice has been stolen and needs to be initalized with - * the paramters of its new note. - * - * Inputs: - * pVoice - pointer to voice to retarget - * - * Outputs: - * None - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL VMRetargetStolenVoice (S_VOICE_MGR *pVoiceMgr, EAS_I32 voiceNum) -{ - EAS_U8 flags; - S_SYNTH_CHANNEL *pMIDIChannel; - S_SYNTH_VOICE *pVoice; - S_SYNTH *pSynth; - S_SYNTH *pNextSynth; - - /* establish some pointers */ - pVoice = &pVoiceMgr->voices[voiceNum]; - pSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->channel)]; - pMIDIChannel = &pSynth->channels[pVoice->channel & 15]; - pNextSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->nextChannel)]; - -#ifdef _DEBUG_VM -{ /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMRetargetStolenVoice: retargeting stolen voice %d on channel %d\n", - voiceNum, pVoice->channel); */ } - - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\to channel %d note: %d velocity: %d\n", - pVoice->nextChannel, pVoice->nextNote, pVoice->nextVelocity); */ } -#endif - - /* make sure new channel hasn't been muted by SP-MIDI since the voice was stolen */ - if ((pSynth->synthFlags & SYNTH_FLAG_SP_MIDI_ON) && - (pMIDIChannel->channelFlags & CHANNEL_FLAG_MUTE)) - { - VMFreeVoice(pVoiceMgr, pSynth, &pVoiceMgr->voices[voiceNum]); - return EAS_FALSE; - } - - /* if assigned to a new synth, correct the active voice count */ - if (pVoice->channel != pVoice->nextChannel) - { -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMRetargetStolenVoice: Note assigned to different virtual synth, adjusting numActiveVoices\n"); */ } -#endif - pSynth->numActiveVoices--; - pNextSynth->numActiveVoices++; - } - - /* assign new channel number, and increase channel voice count */ - pVoice->channel = pVoice->nextChannel; - pMIDIChannel = &pNextSynth->channels[pVoice->channel & 15]; - - /* assign other data */ - pVoice->note = pVoice->nextNote; - pVoice->velocity = pVoice->nextVelocity; - pVoice->nextChannel = UNASSIGNED_SYNTH_CHANNEL; - pVoice->regionIndex = pVoice->nextRegionIndex; - - /* save the flags, pfStartVoice() will clear them */ - flags = pVoice->voiceFlags; - - /* keep track of the note-start related workload */ - pVoiceMgr->workload += WORKLOAD_AMOUNT_START_NOTE; - - /* setup the voice parameters */ - pVoice->voiceState = eVoiceStateStart; - - /*lint -e{522} return not used at this time */ - GetSynthPtr(voiceNum)->pfStartVoice(pVoiceMgr, pNextSynth, &pVoiceMgr->voices[voiceNum], GetAdjustedVoiceNum(voiceNum), pVoice->regionIndex); - - /* did the new note already receive a MIDI note-off request? */ - if (flags & VOICE_FLAG_DEFER_MIDI_NOTE_OFF) - { -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMRetargetVoice: stolen note note-off request deferred\n"); */ } -#endif - pVoice->voiceFlags |= VOICE_FLAG_DEFER_MIDI_NOTE_OFF; - pNextSynth->synthFlags |= SYNTH_FLAG_DEFERRED_MIDI_NOTE_OFF_PENDING; - } - - return EAS_TRUE; -} - -/*---------------------------------------------------------------------------- - * VMCheckKeyGroup() - *---------------------------------------------------------------------------- - * If the note that we've been asked to start is in the same key group as - * any currently playing notes, then we must shut down the currently playing - * note in the same key group - *---------------------------------------------------------------------------- -*/ -void VMCheckKeyGroup (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U16 keyGroup, EAS_U8 channel) -{ - const S_REGION *pRegion; - EAS_INT voiceNum; - - /* increment frame workload */ - pVoiceMgr->workload += WORKLOAD_AMOUNT_KEY_GROUP; - - /* need to check all voices in case this is a layered sound */ - channel = VSynthToChannel(pSynth, channel); - for (voiceNum = 0; voiceNum < MAX_SYNTH_VOICES; voiceNum++) - { - if (pVoiceMgr->voices[voiceNum].voiceState != eVoiceStateStolen) - { - /* voice must be on the same channel */ - if (channel == pVoiceMgr->voices[voiceNum].channel) - { - /* check key group */ - pRegion = GetRegionPtr(pSynth, pVoiceMgr->voices[voiceNum].regionIndex); - if (keyGroup == (pRegion->keyGroupAndFlags & 0x0f00)) - { -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMCheckKeyGroup: voice %d matches key group %d\n", voiceNum, keyGroup >> 8); */ } -#endif - - /* if this voice was just started, set it to mute on the next buffer */ - if (pVoiceMgr->voices[voiceNum].voiceFlags & VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET) - pVoiceMgr->voices[voiceNum].voiceFlags |= VOICE_FLAG_DEFER_MUTE; - - /* mute immediately */ - else - VMMuteVoice(pVoiceMgr, voiceNum); - } - } - } - - /* for stolen voice, check new values */ - else - { - /* voice must be on the same channel */ - if (channel == pVoiceMgr->voices[voiceNum].nextChannel) - { - /* check key group */ - pRegion = GetRegionPtr(pSynth, pVoiceMgr->voices[voiceNum].nextRegionIndex); - if (keyGroup == (pRegion->keyGroupAndFlags & 0x0f00)) - { -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMCheckKeyGroup: voice %d matches key group %d\n", voiceNum, keyGroup >> 8); */ } -#endif - - /* if this voice was just started, set it to mute on the next buffer */ - if (pVoiceMgr->voices[voiceNum].voiceFlags & VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET) - pVoiceMgr->voices[voiceNum].voiceFlags |= VOICE_FLAG_DEFER_MUTE; - - /* mute immediately */ - else - VMMuteVoice(pVoiceMgr, voiceNum); - } - } - - } - } -} - -/*---------------------------------------------------------------------------- - * VMCheckPolyphonyLimiting() - *---------------------------------------------------------------------------- - * Purpose: - * We only play at most 2 of the same note on a MIDI channel. - * E.g., if we are asked to start note 36, and there are already two voices - * that are playing note 36, then we must steal the voice playing - * the oldest note 36 and use that stolen voice to play the new note 36. - * - * Inputs: - * nChannel - synth channel that wants to start a new note - * nKeyNumber - new note's midi note number - * nNoteVelocity - new note's velocity - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * pbVoiceStealingRequired - flag: this routine sets true if we needed to - * steal a voice - * * - * Side Effects: - * psSynthObject->m_sVoice[free voice num].m_nKeyNumber may be assigned - * psSynthObject->m_sVoice[free voice num].m_nVelocity may be assigned - *---------------------------------------------------------------------------- -*/ -EAS_BOOL VMCheckPolyphonyLimiting (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 note, EAS_U8 velocity, EAS_U16 regionIndex, EAS_I32 lowVoice, EAS_I32 highVoice) -{ - EAS_INT voiceNum; - EAS_INT oldestVoiceNum; - EAS_INT numVoicesPlayingNote; - EAS_U16 age; - EAS_U16 oldestNoteAge; - - pVoiceMgr->workload += WORKLOAD_AMOUNT_POLY_LIMIT; - - numVoicesPlayingNote = 0; - oldestVoiceNum = MAX_SYNTH_VOICES; - oldestNoteAge = 0; - channel = VSynthToChannel(pSynth, channel); - - /* examine each voice on this channel playing this note */ - for (voiceNum = lowVoice; voiceNum <= highVoice; voiceNum++) - { - /* check stolen notes separately */ - if (pVoiceMgr->voices[voiceNum].voiceState != eVoiceStateStolen) - { - - /* same channel and note ? */ - if ((channel == pVoiceMgr->voices[voiceNum].channel) && (note == pVoiceMgr->voices[voiceNum].note)) - { - numVoicesPlayingNote++; - age = pVoiceMgr->age - pVoiceMgr->voices[voiceNum].age; - - /* is this the oldest voice for this note? */ - if (age >= oldestNoteAge) - { - oldestNoteAge = age; - oldestVoiceNum = voiceNum; - } - } - } - - /* handle stolen voices */ - else - { - /* same channel and note ? */ - if ((channel == pVoiceMgr->voices[voiceNum].nextChannel) && (note == pVoiceMgr->voices[voiceNum].nextNote)) - { - numVoicesPlayingNote++; - } - } - } - - /* check to see if we exceeded poly limit */ - if (numVoicesPlayingNote < DEFAULT_CHANNEL_POLYPHONY_LIMIT) - return EAS_FALSE; - - /* make sure we have a voice to steal */ - if (oldestVoiceNum != MAX_SYNTH_VOICES) - { -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMCheckPolyphonyLimiting: voice %d has the oldest note\n", oldestVoiceNum); */ } - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "VMCheckPolyphonyLimiting: polyphony limiting requires shutting down note %d \n", pVoiceMgr->voices[oldestVoiceNum].note); */ } -#endif - VMStolenVoice(pVoiceMgr, pSynth, oldestVoiceNum, channel, note, velocity, regionIndex); - return EAS_TRUE; - } - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "VMCheckPolyphonyLimiting: No oldest voice to steal\n"); */ } -#endif - return EAS_FALSE; -} - -/*---------------------------------------------------------------------------- - * VMStartVoice() - *---------------------------------------------------------------------------- - * Starts a voice given a region index - *---------------------------------------------------------------------------- -*/ -void VMStartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 note, EAS_U8 velocity, EAS_U16 regionIndex) -{ - const S_REGION *pRegion; - S_SYNTH_CHANNEL *pChannel; - EAS_INT voiceNum; - EAS_INT maxSynthPoly; - EAS_I32 lowVoice, highVoice; - EAS_U16 keyGroup; - - pChannel = &pSynth->channels[channel]; - pRegion = GetRegionPtr(pSynth, regionIndex); - - /* select correct synth */ -#if defined(_SECONDARY_SYNTH) || defined(EAS_SPLIT_WT_SYNTH) - { -#ifdef EAS_SPLIT_WT_SYNTH - if ((pRegion->keyGroupAndFlags & REGION_FLAG_OFF_CHIP) == 0) -#else - if ((regionIndex & FLAG_RGN_IDX_FM_SYNTH) == 0) -#endif - { - lowVoice = 0; - highVoice = NUM_PRIMARY_VOICES - 1; - } - else - { - lowVoice = NUM_PRIMARY_VOICES; - highVoice = MAX_SYNTH_VOICES - 1; - } - } -#else - lowVoice = 0; - highVoice = MAX_SYNTH_VOICES - 1; -#endif - - /* keep track of the note-start related workload */ - pVoiceMgr->workload+= WORKLOAD_AMOUNT_START_NOTE; - - /* other voices in pool, check for key group and poly limiting */ - if (pSynth->poolCount[pChannel->pool] != 0) - { - - /* check for key group exclusivity */ - keyGroup = pRegion->keyGroupAndFlags & 0x0f00; - if (keyGroup!= 0) - VMCheckKeyGroup(pVoiceMgr, pSynth, keyGroup, channel); - - /* check polyphony limit and steal a voice if necessary */ - if ((pRegion->keyGroupAndFlags & REGION_FLAG_NON_SELF_EXCLUSIVE) == 0) - { - if (VMCheckPolyphonyLimiting(pVoiceMgr, pSynth, channel, note, velocity, regionIndex, lowVoice, highVoice) == EAS_TRUE) - return; - } - } - - /* check max poly allocation */ - if ((pSynth->maxPolyphony == 0) || (pVoiceMgr->maxPolyphony < pSynth->maxPolyphony)) - maxSynthPoly = pVoiceMgr->maxPolyphony; - else - maxSynthPoly = pSynth->maxPolyphony; - - /* any free voices? */ - if ((pVoiceMgr->activeVoices < pVoiceMgr->maxPolyphony) && - (pSynth->numActiveVoices < maxSynthPoly) && - (EAS_SUCCESS == VMFindAvailableVoice(pVoiceMgr, &voiceNum, lowVoice, highVoice))) - { - S_SYNTH_VOICE *pVoice = &pVoiceMgr->voices[voiceNum]; - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "VMStartVoice: Synth=%d\n", pSynth->vSynthNum); */ } -#endif - - /* bump voice counts */ - pVoiceMgr->activeVoices++; - pSynth->numActiveVoices++; - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMStartVoice: voice %d assigned to channel %d note %d velocity %d\n", - voiceNum, channel, note, velocity); */ } -#endif - - /* save parameters */ - pVoiceMgr->voices[voiceNum].channel = VSynthToChannel(pSynth, channel); - pVoiceMgr->voices[voiceNum].note = note; - pVoiceMgr->voices[voiceNum].velocity = velocity; - - /* establish note age for voice stealing */ - pVoiceMgr->voices[voiceNum].age = pVoiceMgr->age++; - - /* setup the synthesis parameters */ - pVoiceMgr->voices[voiceNum].voiceState = eVoiceStateStart; - - /* increment voice pool count */ - IncVoicePoolCount(pVoiceMgr, pVoice); - - /* start voice on correct synth */ - /*lint -e{522} return not used at this time */ - GetSynthPtr(voiceNum)->pfStartVoice(pVoiceMgr, pSynth, &pVoiceMgr->voices[voiceNum], GetAdjustedVoiceNum(voiceNum), regionIndex); - return; - } - - /* no free voices, we have to steal one using appropriate algorithm */ - if (VMStealVoice(pVoiceMgr, pSynth, &voiceNum, channel, note, lowVoice, highVoice) == EAS_SUCCESS) - VMStolenVoice(pVoiceMgr, pSynth, voiceNum, channel, note, velocity, regionIndex); - -#ifdef _DEBUG_VM - else - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMStartVoice: Could not steal a voice for channel %d note %d velocity %d\n", - channel, note, velocity); */ } - } -#endif - - return; -} - -/*---------------------------------------------------------------------------- - * VMStartNote() - *---------------------------------------------------------------------------- - * Purpose: - * Update the synth's state to play the requested note on the requested - * channel if possible. - * - * Inputs: - * nChannel - the channel to start a note on - * nKeyNumber - the key number to start a note for - * nNoteVelocity - the key velocity from this note - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * Side Effects: - * psSynthObject->m_nNumActiveVoices may be incremented - * psSynthObject->m_sVoice[free voice num].m_nSynthChannel may be assigned - * psSynthObject->m_sVoice[free voice num].m_nKeyNumber is assigned - * psSynthObject->m_sVoice[free voice num].m_nVelocity is assigned - *---------------------------------------------------------------------------- -*/ -void VMStartNote (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 note, EAS_U8 velocity) -{ - S_SYNTH_CHANNEL *pChannel; - EAS_U16 regionIndex; - EAS_I16 adjustedNote; - - /* bump note count */ - pSynth->totalNoteCount++; - - pChannel = &pSynth->channels[channel]; - - /* check channel mute */ - if (pChannel->channelFlags & CHANNEL_FLAG_MUTE) - return; - -#ifdef EXTERNAL_AUDIO - /* pass event to external audio when requested */ - if ((pChannel->channelFlags & CHANNEL_FLAG_EXTERNAL_AUDIO) && (pSynth->cbEventFunc != NULL)) - { - S_EXT_AUDIO_EVENT event; - event.channel = channel; - event.note = note; - event.velocity = velocity; - event.noteOn = EAS_TRUE; - if (pSynth->cbEventFunc(pSynth->pExtAudioInstData, &event)) - return; - } -#endif - - /* start search at first region */ - regionIndex = pChannel->regionIndex; - - /* handle transposition */ - adjustedNote = note; - if (pChannel->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL) - adjustedNote += pChannel->coarsePitch; - else - adjustedNote += pChannel->coarsePitch + pSynth->globalTranspose; - - /* limit adjusted key number so it does not wraparound, over/underflow */ - if (adjustedNote < 0) - { - adjustedNote = 0; - } - else if (adjustedNote > 127) - { - adjustedNote = 127; - } - -#if defined(DLS_SYNTHESIZER) - if (regionIndex & FLAG_RGN_IDX_DLS_SYNTH) - { - /* DLS voice */ - for (;;) - { - /*lint -e{740,826} cast OK, we know this is actually a DLS region */ - const S_DLS_REGION *pDLSRegion = (S_DLS_REGION*) GetRegionPtr(pSynth, regionIndex); - - /* check key against this region's key and velocity range */ - if (((adjustedNote >= pDLSRegion->wtRegion.region.rangeLow) && (adjustedNote <= pDLSRegion->wtRegion.region.rangeHigh)) && - ((velocity >= pDLSRegion->velLow) && (velocity <= pDLSRegion->velHigh))) - { - VMStartVoice(pVoiceMgr, pSynth, channel, note, velocity, regionIndex); - } - - /* last region in program? */ - if (pDLSRegion->wtRegion.region.keyGroupAndFlags & REGION_FLAG_LAST_REGION) - break; - - /* advance to next region */ - regionIndex++; - } - } - else -#endif - - /* braces here for #if clause */ - { - /* EAS voice */ - for (;;) - { - const S_REGION *pRegion = GetRegionPtr(pSynth, regionIndex); - - /* check key against this region's keyrange */ - if ((adjustedNote >= pRegion->rangeLow) && (adjustedNote <= pRegion->rangeHigh)) - { - VMStartVoice(pVoiceMgr, pSynth, channel, note, velocity, regionIndex); - break; - } - - /* last region in program? */ - if (pRegion->keyGroupAndFlags & REGION_FLAG_LAST_REGION) - break; - - /* advance to next region */ - regionIndex++; - } - } -} - -/*---------------------------------------------------------------------------- - * VMStopNote() - *---------------------------------------------------------------------------- - * Purpose: - * Update the synth's state to end the requested note on the requested - * channel. - * - * Inputs: - * nChannel - the channel to stop a note on - * nKeyNumber - the key number for this note off - * nNoteVelocity - the note-off velocity - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * Side Effects: - * psSynthObject->m_sVoice[free voice num].m_nSynthChannel may be assigned - * psSynthObject->m_sVoice[free voice num].m_nKeyNumber is assigned - * psSynthObject->m_sVoice[free voice num].m_nVelocity is assigned - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, velocity) reserved for future use */ -void VMStopNote (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 note, EAS_U8 velocity) -{ - S_SYNTH_CHANNEL *pChannel; - EAS_INT voiceNum; - - pChannel = &(pSynth->channels[channel]); - -#ifdef EXTERNAL_AUDIO - if ((pChannel->channelFlags & CHANNEL_FLAG_EXTERNAL_AUDIO) && (pSynth->cbEventFunc != NULL)) - { - S_EXT_AUDIO_EVENT event; - event.channel = channel; - event.note = note; - event.velocity = velocity; - event.noteOn = EAS_FALSE; - if (pSynth->cbEventFunc(pSynth->pExtAudioInstData, &event)) - return; - } -#endif - - /* keep track of the note-start workload */ - pVoiceMgr->workload += WORKLOAD_AMOUNT_STOP_NOTE; - - channel = VSynthToChannel(pSynth, channel); - - for (voiceNum=0; voiceNum < MAX_SYNTH_VOICES; voiceNum++) - { - - /* stolen notes are handled separately */ - if (eVoiceStateStolen != pVoiceMgr->voices[voiceNum].voiceState) - { - - /* channel and key number must match */ - if ((channel == pVoiceMgr->voices[voiceNum].channel) && (note == pVoiceMgr->voices[voiceNum].note)) - { -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMStopNote: voice %d channel %d note %d\n", - voiceNum, channel, note); */ } -#endif - - /* if sustain pedal is down, set deferred note-off flag */ - if (pChannel->channelFlags & CHANNEL_FLAG_SUSTAIN_PEDAL) - { - pVoiceMgr->voices[voiceNum].voiceFlags |= VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF; - continue; - } - - /* if this note just started, wait before we stop it */ - if (pVoiceMgr->voices[voiceNum].voiceFlags & VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET) - { -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tDeferred: Not started yet\n"); */ } -#endif - pVoiceMgr->voices[voiceNum].voiceFlags |= VOICE_FLAG_DEFER_MIDI_NOTE_OFF; - pSynth->synthFlags |= SYNTH_FLAG_DEFERRED_MIDI_NOTE_OFF_PENDING; - } - - /* release voice */ - else - VMReleaseVoice(pVoiceMgr, pSynth, voiceNum); - - } - } - - /* process stolen notes, new channel and key number must match */ - else if ((channel == pVoiceMgr->voices[voiceNum].nextChannel) && (note == pVoiceMgr->voices[voiceNum].nextNote)) - { - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMStopNote: voice %d channel %d note %d\n\tDeferred: Stolen voice\n", - voiceNum, channel, note); */ } -#endif - pVoiceMgr->voices[voiceNum].voiceFlags |= VOICE_FLAG_DEFER_MIDI_NOTE_OFF; - } - } -} - -/*---------------------------------------------------------------------------- - * VMFindAvailableVoice() - *---------------------------------------------------------------------------- - * Purpose: - * Find an available voice and return the voice number if available. - * - * Inputs: - * pnVoiceNumber - really an output, returns the voice number found - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * success - if there is an available voice - * failure - otherwise - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMFindAvailableVoice (S_VOICE_MGR *pVoiceMgr, EAS_INT *pVoiceNumber, EAS_I32 lowVoice, EAS_I32 highVoice) -{ - EAS_INT voiceNum; - - /* Check each voice to see if it has been assigned to a synth channel */ - for (voiceNum = lowVoice; voiceNum <= highVoice; voiceNum++) - { - /* check if this voice has been assigned to a synth channel */ - if ( pVoiceMgr->voices[voiceNum].voiceState == eVoiceStateFree) - { - *pVoiceNumber = voiceNum; /* this voice is available */ - return EAS_SUCCESS; - } - } - - /* if we reach here, we have not found a free voice */ - *pVoiceNumber = UNASSIGNED_SYNTH_VOICE; - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMFindAvailableVoice: error, could not find an available voice\n"); */ } -#endif - return EAS_FAILURE; -} - -/*---------------------------------------------------------------------------- - * VMStealVoice() - *---------------------------------------------------------------------------- - * Purpose: - * Steal a voice and return the voice number - * - * Stealing algorithm: steal the best choice with minimal work, taking into - * account SP-Midi channel priorities and polyphony allocation. - * - * In one pass through all the voices, figure out which voice to steal - * taking into account a number of different factors: - * Priority of the voice's MIDI channel - * Number of voices over the polyphony allocation for voice's MIDI channel - * Amplitude of the voice - * Note age - * Key velocity (for voices that haven't been started yet) - * If any matching notes are found - * - * Inputs: - * pnVoiceNumber - really an output, see below - * nChannel - the channel that this voice wants to be started on - * nKeyNumber - the key number for this new voice - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * pnVoiceNumber - voice number of the voice that was stolen - * EAS_RESULT EAS_SUCCESS - always successful - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMStealVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_INT *pVoiceNumber, EAS_U8 channel, EAS_U8 note, EAS_I32 lowVoice, EAS_I32 highVoice) -{ - S_SYNTH_VOICE *pCurrVoice; - S_SYNTH *pCurrSynth; - EAS_INT voiceNum; - EAS_INT bestCandidate; - EAS_U8 currChannel; - EAS_U8 currNote; - EAS_I32 bestPriority; - EAS_I32 currentPriority; - - /* determine which voice to steal */ - bestPriority = 0; - bestCandidate = MAX_SYNTH_VOICES; - - for (voiceNum = lowVoice; voiceNum <= highVoice; voiceNum++) - { - pCurrVoice = &pVoiceMgr->voices[voiceNum]; - - /* ignore free voices */ - if (pCurrVoice->voiceState == eVoiceStateFree) - continue; - - /* for stolen voices, use the new parameters, not the old */ - if (pCurrVoice->voiceState == eVoiceStateStolen) - { - pCurrSynth = pVoiceMgr->pSynth[GET_VSYNTH(pCurrVoice->nextChannel)]; - currChannel = pCurrVoice->nextChannel; - currNote = pCurrVoice->nextNote; - } - else - { - pCurrSynth = pVoiceMgr->pSynth[GET_VSYNTH(pCurrVoice->channel)]; - currChannel = pCurrVoice->channel; - currNote = pCurrVoice->note; - } - - /* ignore voices that are higher priority */ - if (pSynth->priority > pCurrSynth->priority) - continue; -#ifdef _DEBUG_VM -// { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMStealVoice: New priority = %d exceeds old priority = %d\n", pSynth->priority, pCurrSynth->priority); */ } -#endif - - /* if voice is stolen or just started, reduce the likelihood it will be stolen */ - if (( pCurrVoice->voiceState == eVoiceStateStolen) || (pCurrVoice->voiceFlags & VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET)) - { - currentPriority = 128 - pCurrVoice->nextVelocity; - } - else - { - /* compute the priority of this voice, higher means better for stealing */ - /* use not age */ - currentPriority = (EAS_I32) pCurrVoice->age << NOTE_AGE_STEAL_WEIGHT; - - /* include note gain -higher gain is lower steal value */ - /*lint -e{704} use shift for performance */ - currentPriority += ((32768 >> (12 - NOTE_GAIN_STEAL_WEIGHT)) + 256) - - ((EAS_I32) pCurrVoice->gain >> (12 - NOTE_GAIN_STEAL_WEIGHT)); - } - - /* in SP-MIDI mode, include over poly allocation and channel priority */ - if (pSynth->synthFlags & SYNTH_FLAG_SP_MIDI_ON) - { - S_SYNTH_CHANNEL *pChannel = &pCurrSynth->channels[GET_CHANNEL(currChannel)]; - /*lint -e{701} use shift for performance */ - if (pSynth->poolCount[pChannel->pool] >= pSynth->poolAlloc[pChannel->pool]) - currentPriority += (pSynth->poolCount[pChannel->pool] -pSynth->poolAlloc[pChannel->pool] + 1) << CHANNEL_POLY_STEAL_WEIGHT; - - /* include channel priority */ - currentPriority += (EAS_I32)(pChannel->pool << CHANNEL_PRIORITY_STEAL_WEIGHT); - } - - /* if a note is already playing that matches this note, consider stealing it more readily */ - if ((note == currNote) && (channel == currChannel)) - currentPriority += NOTE_MATCH_PENALTY; - - /* is this the best choice so far? */ - if (currentPriority >= bestPriority) - { - bestPriority = currentPriority; - bestCandidate = voiceNum; - } - } - - /* may happen if all voices are allocated to a higher priority virtual synth */ - if (bestCandidate == MAX_SYNTH_VOICES) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMStealVoice: Unable to allocate a voice\n"); */ } - return EAS_ERROR_NO_VOICE_ALLOCATED; - } - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMStealVoice: Voice %d stolen\n", bestCandidate); */ } - - /* are we stealing a stolen voice? */ - if (pVoiceMgr->voices[bestCandidate].voiceState == eVoiceStateStolen) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "VMStealVoice: Voice %d is already marked as stolen and was scheduled to play ch: %d note: %d vel: %d\n", - bestCandidate, - pVoiceMgr->voices[bestCandidate].nextChannel, - pVoiceMgr->voices[bestCandidate].nextNote, - pVoiceMgr->voices[bestCandidate].nextVelocity); */ } - } -#endif - - *pVoiceNumber = (EAS_U16) bestCandidate; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * VMChannelPressure() - *---------------------------------------------------------------------------- - * Purpose: - * Change the channel pressure for the given channel - * - * Inputs: - * nChannel - the MIDI channel - * nVelocity - the channel pressure value - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * Side Effects: - * psSynthObject->m_sChannel[nChannel].m_nChannelPressure is updated - *---------------------------------------------------------------------------- -*/ -void VMChannelPressure (S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 value) -{ - S_SYNTH_CHANNEL *pChannel; - - pChannel = &(pSynth->channels[channel]); - pChannel->channelPressure = value; - - /* - set a channel flag to request parameter updates - for all the voices associated with this channel - */ - pChannel->channelFlags |= CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS; -} - -/*---------------------------------------------------------------------------- - * VMPitchBend() - *---------------------------------------------------------------------------- - * Purpose: - * Change the pitch wheel value for the given channel. - * This routine constructs the proper 14-bit argument when the calling routine - * passes the pitch LSB and MSB. - * - * Note: some midi disassemblers display a bipolar pitch bend value. - * We can display the bipolar value using - * if m_nPitchBend >= 0x2000 - * bipolar pitch bend = postive (m_nPitchBend - 0x2000) - * else - * bipolar pitch bend = negative (0x2000 - m_nPitchBend) - * - * Inputs: - * nChannel - the MIDI channel - * nPitchLSB - the LSB byte of the pitch bend message - * nPitchMSB - the MSB byte of the pitch bend message - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * psSynthObject->m_sChannel[nChannel].m_nPitchBend is changed - * - *---------------------------------------------------------------------------- -*/ -void VMPitchBend (S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 nPitchLSB, EAS_U8 nPitchMSB) -{ - S_SYNTH_CHANNEL *pChannel; - - pChannel = &(pSynth->channels[channel]); - pChannel->pitchBend = (EAS_I16) ((nPitchMSB << 7) | nPitchLSB); - - /* - set a channel flag to request parameter updates - for all the voices associated with this channel - */ - pChannel->channelFlags |= CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS; -} - -/*---------------------------------------------------------------------------- - * VMControlChange() - *---------------------------------------------------------------------------- - * Purpose: - * Change the controller (or mode) for the given channel. - * - * Inputs: - * nChannel - the MIDI channel - * nControllerNumber - the MIDI controller number - * nControlValue - the value for this controller message - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * Side Effects: - * psSynthObject->m_sChannel[nChannel] controller is changed - * - *---------------------------------------------------------------------------- -*/ -void VMControlChange (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 controller, EAS_U8 value) -{ - S_SYNTH_CHANNEL *pChannel; - - pChannel = &(pSynth->channels[channel]); - - /* - set a channel flag to request parameter updates - for all the voices associated with this channel - */ - pChannel->channelFlags |= CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS; - - switch ( controller ) - { - case MIDI_CONTROLLER_BANK_SELECT_MSB: -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMControlChange: Bank Select MSB: msb 0x%X\n", value); */ } -#endif - /* use this MSB with a zero LSB, until we get an LSB message */ - pChannel->bankNum = value << 8; - break; - - case MIDI_CONTROLLER_MOD_WHEEL: - /* we treat mod wheel as a 7-bit controller and only use the MSB */ - pChannel->modWheel = value; - break; - - case MIDI_CONTROLLER_VOLUME: - /* we treat volume as a 7-bit controller and only use the MSB */ - pChannel->volume = value; - break; - - case MIDI_CONTROLLER_PAN: - /* we treat pan as a 7-bit controller and only use the MSB */ - pChannel->pan = value; - break; - - case MIDI_CONTROLLER_EXPRESSION: - /* we treat expression as a 7-bit controller and only use the MSB */ - pChannel->expression = value; - break; - - case MIDI_CONTROLLER_BANK_SELECT_LSB: -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMControlChange: Bank Select LSB: lsb 0x%X\n", value); */ } -#endif - /* - construct bank number as 7-bits (stored as 8) of existing MSB - and 7-bits of new LSB (also stored as 8( - */ - pChannel->bankNum = - (pChannel->bankNum & 0xFF00) | value; - - break; - - case MIDI_CONTROLLER_SUSTAIN_PEDAL: - /* we treat sustain pedal as a boolean on/off bit flag */ - if (value < 64) - { - /* - we are requested to turn the pedal off, but first check - if the pedal is already on - */ - if (0 != - (pChannel->channelFlags & CHANNEL_FLAG_SUSTAIN_PEDAL) - ) - { - /* - The sustain flag is presently set and the damper pedal is on. - We are therefore transitioning from damper pedal ON to - damper pedal OFF. This means all notes in this channel - that received a note off while the damper pedal was on, and - had their note-off requests deferred, should now proceed to - the release state. - */ - VMReleaseAllDeferredNoteOffs(pVoiceMgr, pSynth, channel); - } /* end if sustain pedal is already on */ - - /* turn the sustain pedal off */ - pChannel->channelFlags &= ~CHANNEL_FLAG_SUSTAIN_PEDAL; - } - else - { - /* - we are requested to turn the pedal on, but first check - if the pedal is already off - */ - if (0 == - (pChannel->channelFlags & CHANNEL_FLAG_SUSTAIN_PEDAL) - ) - { - /* - The sustain flag is presently clear and the damper pedal is off. - We are therefore transitioning from damper pedal OFF to - damper pedal ON. Currently sounding notes should be left - unchanged. However, we should try to "catch" notes if possible. - If any notes have levels >= sustain level, catch them, - otherwise, let them continue to release. - */ - VMCatchNotesForSustainPedal(pVoiceMgr, pSynth, channel); - } - - /* turn the sustain pedal on */ - pChannel->channelFlags |= CHANNEL_FLAG_SUSTAIN_PEDAL; - } - - break; -#ifdef _REVERB - case MIDI_CONTROLLER_REVERB_SEND: - /* we treat send as a 7-bit controller and only use the MSB */ - pSynth->channels[channel].reverbSend = value; - break; -#endif -#ifdef _CHORUS - case MIDI_CONTROLLER_CHORUS_SEND: - /* we treat send as a 7-bit controller and only use the MSB */ - pSynth->channels[channel].chorusSend = value; - break; -#endif - case MIDI_CONTROLLER_RESET_CONTROLLERS: - /* despite the Midi message name, not ALL controllers are reset */ - pChannel->modWheel = DEFAULT_MOD_WHEEL; - pChannel->expression = DEFAULT_EXPRESSION; - - /* turn the sustain pedal off as default/reset */ - pChannel->channelFlags &= ~CHANNEL_FLAG_SUSTAIN_PEDAL; - pChannel->pitchBend = DEFAULT_PITCH_BEND; - - /* reset channel pressure */ - pChannel->channelPressure = DEFAULT_CHANNEL_PRESSURE; - - /* reset RPN values */ - pChannel->registeredParam = DEFAULT_REGISTERED_PARAM; - pChannel->pitchBendSensitivity = DEFAULT_PITCH_BEND_SENSITIVITY; - pChannel->finePitch = DEFAULT_FINE_PITCH; - pChannel->coarsePitch = DEFAULT_COARSE_PITCH; - - /* - program change, bank select, channel volume CC7, pan CC10 - are NOT reset - */ - break; - - /* - For logical reasons, the RPN data entry are grouped together. - However, keep in mind that these cases are not necessarily in - ascending order. - e.g., MIDI_CONTROLLER_DATA_ENTRY_MSB == 6, - whereas MIDI_CONTROLLER_SUSTAIN_PEDAL == 64. - So arrange these case statements in whatever manner is more efficient for - the processor / compiler. - */ - case MIDI_CONTROLLER_ENTER_DATA_MSB: - case MIDI_CONTROLLER_ENTER_DATA_LSB: - case MIDI_CONTROLLER_SELECT_RPN_LSB: - case MIDI_CONTROLLER_SELECT_RPN_MSB: - case MIDI_CONTROLLER_SELECT_NRPN_MSB: - case MIDI_CONTROLLER_SELECT_NRPN_LSB: - VMUpdateRPNStateMachine(pSynth, channel, controller, value); - break; - - case MIDI_CONTROLLER_ALL_SOUND_OFF: - case MIDI_CONTROLLER_ALL_NOTES_OFF: - case MIDI_CONTROLLER_OMNI_OFF: - case MIDI_CONTROLLER_OMNI_ON: - case MIDI_CONTROLLER_MONO_ON_POLY_OFF: - case MIDI_CONTROLLER_POLY_ON_MONO_OFF: - /* NOTE: we treat all sounds off the same as all notes off */ - VMAllNotesOff(pVoiceMgr, pSynth, channel); - break; - - default: -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMControlChange: controller %d not yet implemented\n", controller); */ } -#endif - break; - - } - - return; -} - -/*---------------------------------------------------------------------------- - * VMUpdateRPNStateMachine() - *---------------------------------------------------------------------------- - * Purpose: - * Call this function when we want to parse RPN related controller messages. - * We only support RPN0 (pitch bend sensitivity), RPN1 (fine tuning) and - * RPN2 (coarse tuning). Any other RPNs or NRPNs are ignored for now. - *. - * Supports any order, so not a state machine anymore. This function was - * rewritten to work correctly regardless of order. - * - * Inputs: - * nChannel - the channel this controller message is coming from - * nControllerNumber - which RPN related controller - * nControlValue - the value of the RPN related controller - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * returns EAS_RESULT, which is typically EAS_SUCCESS, since there are - * few possible errors - * - * Side Effects: - * gsSynthObject.m_sChannel[nChannel].m_nPitchBendSensitivity - * (or m_nFinePitch or m_nCoarsePitch) - * will be updated if the proper RPN message is received. - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMUpdateRPNStateMachine (S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 controller, EAS_U8 value) -{ - S_SYNTH_CHANNEL *pChannel; - -#ifdef _DEBUG_VM - if (channel >= NUM_SYNTH_CHANNELS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMUpdateRPNStateMachines: error, %d invalid channel number\n", - channel); */ } - return EAS_FAILURE; - } -#endif - - pChannel = &(pSynth->channels[channel]); - - switch (controller) - { - case MIDI_CONTROLLER_SELECT_NRPN_MSB: - case MIDI_CONTROLLER_SELECT_NRPN_LSB: - pChannel->registeredParam = DEFAULT_REGISTERED_PARAM; - break; - case MIDI_CONTROLLER_SELECT_RPN_MSB: - pChannel->registeredParam = - (pChannel->registeredParam & 0x7F) | (value<<7); - break; - case MIDI_CONTROLLER_SELECT_RPN_LSB: - pChannel->registeredParam = - (pChannel->registeredParam & 0x7F00) | value; - break; - case MIDI_CONTROLLER_ENTER_DATA_MSB: - switch (pChannel->registeredParam) - { - case 0: - pChannel->pitchBendSensitivity = value * 100; - break; - case 1: - /*lint -e{702} */ - pChannel->finePitch = (EAS_I8)((((value << 7) - 8192) * 100) >> 13); - break; - case 2: - pChannel->coarsePitch = (EAS_I8)(value - 64); - break; - default: - break; - } - break; - case MIDI_CONTROLLER_ENTER_DATA_LSB: - switch (pChannel->registeredParam) - { - case 0: - //ignore lsb - break; - case 1: - //ignore lsb - break; - case 2: - //ignore lsb - break; - default: - break; - } - break; - default: - return EAS_FAILURE; //not a RPN related controller - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * VMUpdateStaticChannelParameters() - *---------------------------------------------------------------------------- - * Purpose: - * Update all of the static channel parameters for channels that have had - * a controller change values - * Or if the synth has signalled that all channels must forcibly - * be updated - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * none - * - * Side Effects: - * - psSynthObject->m_sChannel[].m_nStaticGain and m_nStaticPitch - * are updated for channels whose controller values have changed - * or if the synth has signalled that all channels must forcibly - * be updated - *---------------------------------------------------------------------------- -*/ -void VMUpdateStaticChannelParameters (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth) -{ - EAS_INT channel; - - if (pSynth->synthFlags & SYNTH_FLAG_UPDATE_ALL_CHANNEL_PARAMETERS) - { - /* - the synth wants us to forcibly update all channel - parameters. This event occurs when we are about to - finish resetting the synth - */ - for (channel = 0; channel < NUM_SYNTH_CHANNELS; channel++) - { -#ifdef _HYBRID_SYNTH - if (pSynth->channels[channel].regionIndex & FLAG_RGN_IDX_FM_SYNTH) - pSecondarySynth->pfUpdateChannel(pVoiceMgr, pSynth, (EAS_U8) channel); - else - pPrimarySynth->pfUpdateChannel(pVoiceMgr, pSynth, (EAS_U8) channel); -#else - pPrimarySynth->pfUpdateChannel(pVoiceMgr, pSynth, (EAS_U8) channel); -#endif - } - - /* - clear the flag to indicates we have now forcibly - updated all channel parameters - */ - pSynth->synthFlags &= ~SYNTH_FLAG_UPDATE_ALL_CHANNEL_PARAMETERS; - } - else - { - - /* only update channel params if signalled by a channel flag */ - for (channel = 0; channel < NUM_SYNTH_CHANNELS; channel++) - { - if ( 0 != (pSynth->channels[channel].channelFlags & CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS)) - { -#ifdef _HYBRID_SYNTH - if (pSynth->channels[channel].regionIndex & FLAG_RGN_IDX_FM_SYNTH) - pSecondarySynth->pfUpdateChannel(pVoiceMgr, pSynth, (EAS_U8) channel); - else - pPrimarySynth->pfUpdateChannel(pVoiceMgr, pSynth, (EAS_U8) channel); -#else - pPrimarySynth->pfUpdateChannel(pVoiceMgr, pSynth, (EAS_U8) channel); -#endif - } - } - - } - - return; -} - -/*---------------------------------------------------------------------------- - * VMFindProgram() - *---------------------------------------------------------------------------- - * Purpose: - * Look up an individual program in sound library. This function - * searches the bank list for a program, then the individual program - * list. - * - * Inputs: - * - * Outputs: - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT VMFindProgram (const S_EAS *pEAS, EAS_U32 bank, EAS_U8 programNum, EAS_U16 *pRegionIndex) -{ - EAS_U32 locale; - const S_PROGRAM *p; - EAS_U16 i; - EAS_U16 regionIndex; - - /* make sure we have a valid sound library */ - if (pEAS == NULL) - return EAS_FAILURE; - - /* search the banks */ - for (i = 0; i < pEAS->numBanks; i++) - { - if (bank == (EAS_U32) pEAS->pBanks[i].locale) - { - regionIndex = pEAS->pBanks[i].regionIndex[programNum]; - if (regionIndex != INVALID_REGION_INDEX) - { - *pRegionIndex = regionIndex; - return EAS_SUCCESS; - } - break; - } - } - - /* establish locale */ - locale = ( bank << 8) | programNum; - - /* search for program */ - for (i = 0, p = pEAS->pPrograms; i < pEAS->numPrograms; i++, p++) - { - if (p->locale == locale) - { - *pRegionIndex = p->regionIndex; - return EAS_SUCCESS; - } - } - - return EAS_FAILURE; -} - -#ifdef DLS_SYNTHESIZER -/*---------------------------------------------------------------------------- - * VMFindDLSProgram() - *---------------------------------------------------------------------------- - * Purpose: - * Look up an individual program in sound library. This function - * searches the bank list for a program, then the individual program - * list. - * - * Inputs: - * - * Outputs: - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT VMFindDLSProgram (const S_DLS *pDLS, EAS_U32 bank, EAS_U8 programNum, EAS_U16 *pRegionIndex) -{ - EAS_U32 locale; - const S_PROGRAM *p; - EAS_U16 i; - - /* make sure we have a valid sound library */ - if (pDLS == NULL) - return EAS_FAILURE; - - /* establish locale */ - locale = (bank << 8) | programNum; - - /* search for program */ - for (i = 0, p = pDLS->pDLSPrograms; i < pDLS->numDLSPrograms; i++, p++) - { - if (p->locale == locale) - { - *pRegionIndex = p->regionIndex; - return EAS_SUCCESS; - } - } - - return EAS_FAILURE; -} -#endif - -/*---------------------------------------------------------------------------- - * VMProgramChange() - *---------------------------------------------------------------------------- - * Purpose: - * Change the instrument (program) for the given channel. - * - * Depending on the program number, and the bank selected for this channel, the - * program may be in ROM, RAM (from SMAF or CMX related RAM wavetable), or - * Alternate wavetable (from mobile DLS or other DLS file) - * - * This function figures out what wavetable should be used, and sets it up as the - * wavetable to use for this channel. Also the channel may switch from a melodic - * channel to a rhythm channel, or vice versa. - * - * Inputs: - * - * Outputs: - * Side Effects: - * gsSynthObject.m_sChannel[nChannel].m_nProgramNumber is likely changed - * gsSynthObject.m_sChannel[nChannel].m_psEAS may be changed - * gsSynthObject.m_sChannel[nChannel].m_bRhythmChannel may be changed - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pVoiceMgr) reserved for future use */ -void VMProgramChange (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 program) -{ - S_SYNTH_CHANNEL *pChannel; - EAS_U32 bank; - EAS_U16 regionIndex; - -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "VMProgramChange: vSynthNum=%d, channel=%d, program=%d\n", pSynth->vSynthNum, channel, program); */ } -#endif - - /* setup pointer to MIDI channel data */ - pChannel = &pSynth->channels[channel]; - bank = pChannel->bankNum; - - /* allow channels to switch between being melodic or rhythm channels, using GM2 CC values */ - if ((bank & 0xFF00) == DEFAULT_RHYTHM_BANK_NUMBER) - { - /* make it a rhythm channel */ - pChannel->channelFlags |= CHANNEL_FLAG_RHYTHM_CHANNEL; - } - else if ((bank & 0xFF00) == DEFAULT_MELODY_BANK_NUMBER) - { - /* make it a melody channel */ - pChannel->channelFlags &= ~CHANNEL_FLAG_RHYTHM_CHANNEL; - } - - regionIndex = DEFAULT_REGION_INDEX; - -#ifdef EXTERNAL_AUDIO - /* give the external audio interface a chance to handle it */ - if (pSynth->cbProgChgFunc != NULL) - { - S_EXT_AUDIO_PRG_CHG prgChg; - prgChg.channel = channel; - prgChg.bank = (EAS_U16) bank; - prgChg.program = program; - if (pSynth->cbProgChgFunc(pSynth->pExtAudioInstData, &prgChg)) - pChannel->channelFlags |= CHANNEL_FLAG_EXTERNAL_AUDIO; - } - -#endif - - -#ifdef DLS_SYNTHESIZER - /* first check for DLS program that may overlay the internal instrument */ - if (VMFindDLSProgram(pSynth->pDLS, bank, program, ®ionIndex) != EAS_SUCCESS) -#endif - - /* braces to support 'if' clause above */ - { - - /* look in the internal banks */ - if (VMFindProgram(pSynth->pEAS, bank, program, ®ionIndex) != EAS_SUCCESS) - - /* fall back to default bank */ - { - if (pSynth->channels[channel].channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL) - bank = DEFAULT_RHYTHM_BANK_NUMBER; - else - bank = DEFAULT_MELODY_BANK_NUMBER; - - if (VMFindProgram(pSynth->pEAS, bank, program, ®ionIndex) != EAS_SUCCESS) - - /* switch to program 0 in the default bank */ - { - if (VMFindProgram(pSynth->pEAS, bank, 0, ®ionIndex) != EAS_SUCCESS) - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "VMProgramChange: No program @ %03d:%03d:%03d\n", - (bank >> 8) & 0x7f, bank & 0x7f, program); */ } - } - } - } - - /* we have our new program change for this channel */ - pChannel->programNum = program; - pChannel->regionIndex = regionIndex; - - /* - set a channel flag to request parameter updates - for all the voices associated with this channel - */ - pChannel->channelFlags |= CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS; - - return; -} - -/*---------------------------------------------------------------------------- - * VMAddSamples() - *---------------------------------------------------------------------------- - * Purpose: - * Synthesize the requested number of samples (block based processing) - * - * Inputs: - * nNumSamplesToAdd - number of samples to write to buffer - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * number of voices rendered - * - * Side Effects: - * - samples are added to the presently free buffer - * - *---------------------------------------------------------------------------- -*/ -EAS_I32 VMAddSamples (S_VOICE_MGR *pVoiceMgr, EAS_I32 *pMixBuffer, EAS_I32 numSamples) -{ - S_SYNTH *pSynth; - EAS_INT voicesRendered; - EAS_INT voiceNum; - EAS_BOOL done; - -#ifdef _REVERB - EAS_PCM *pReverbSendBuffer; -#endif // ifdef _REVERB - -#ifdef _CHORUS - EAS_PCM *pChorusSendBuffer; -#endif // ifdef _CHORUS - - voicesRendered = 0; - for (voiceNum = 0; voiceNum < MAX_SYNTH_VOICES; voiceNum++) - { - - /* retarget stolen voices */ - if ((pVoiceMgr->voices[voiceNum].voiceState == eVoiceStateStolen) && (pVoiceMgr->voices[voiceNum].gain <= 0)) - VMRetargetStolenVoice(pVoiceMgr, voiceNum); - - /* get pointer to virtual synth */ - pSynth = pVoiceMgr->pSynth[pVoiceMgr->voices[voiceNum].channel >> 4]; - - /* synthesize active voices */ - if (pVoiceMgr->voices[voiceNum].voiceState != eVoiceStateFree) - { - done = GetSynthPtr(voiceNum)->pfUpdateVoice(pVoiceMgr, pSynth, &pVoiceMgr->voices[voiceNum], GetAdjustedVoiceNum(voiceNum), pMixBuffer, numSamples); - voicesRendered++; - - /* voice is finished */ - if (done == EAS_TRUE) - { - /* set gain of stolen voice to zero so it will be restarted */ - if (pVoiceMgr->voices[voiceNum].voiceState == eVoiceStateStolen) - pVoiceMgr->voices[voiceNum].gain = 0; - - /* or return it to the free voice pool */ - else - VMFreeVoice(pVoiceMgr, pSynth, &pVoiceMgr->voices[voiceNum]); - } - - /* if this voice is scheduled to be muted, set the mute flag */ - if (pVoiceMgr->voices[voiceNum].voiceFlags & VOICE_FLAG_DEFER_MUTE) - { - pVoiceMgr->voices[voiceNum].voiceFlags &= ~(VOICE_FLAG_DEFER_MUTE | VOICE_FLAG_DEFER_MIDI_NOTE_OFF); - VMMuteVoice(pVoiceMgr, voiceNum); - } - - /* if voice just started, advance state to play */ - if (pVoiceMgr->voices[voiceNum].voiceState == eVoiceStateStart) - pVoiceMgr->voices[voiceNum].voiceState = eVoiceStatePlay; - } - } - - return voicesRendered; -} - -/*---------------------------------------------------------------------------- - * VMRender() - *---------------------------------------------------------------------------- - * Purpose: - * This routine renders a frame of audio - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * pVoicesRendered - number of voices rendered this frame - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMRender (S_VOICE_MGR *pVoiceMgr, EAS_I32 numSamples, EAS_I32 *pMixBuffer, EAS_I32 *pVoicesRendered) -{ - S_SYNTH *pSynth; - EAS_INT i; - EAS_INT channel; - -#ifdef _CHECKED_BUILD - SanityCheck(pVoiceMgr); -#endif - - /* update MIDI channel parameters */ - *pVoicesRendered = 0; - for (i = 0; i < MAX_VIRTUAL_SYNTHESIZERS; i++) - { - if (pVoiceMgr->pSynth[i] != NULL) - VMUpdateStaticChannelParameters(pVoiceMgr, pVoiceMgr->pSynth[i]); - } - - /* synthesize a buffer of audio */ - *pVoicesRendered = VMAddSamples(pVoiceMgr, pMixBuffer, numSamples); - - /* - * check for deferred note-off messages - * If flag is set, that means one or more voices are expecting deferred - * midi note-off messages because the midi note-on and corresponding midi - * note-off requests occurred during the same update interval. The goal - * is the defer the note-off request so that the note can at least start. - */ - for (i = 0; i < MAX_VIRTUAL_SYNTHESIZERS; i++) - { - pSynth = pVoiceMgr->pSynth[i]; - - if (pSynth== NULL) - continue; - - if (pSynth->synthFlags & SYNTH_FLAG_DEFERRED_MIDI_NOTE_OFF_PENDING) - VMDeferredStopNote(pVoiceMgr, pSynth); - - /* check if we need to reset the synth */ - if ((pSynth->synthFlags & SYNTH_FLAG_RESET_IS_REQUESTED) && - (pSynth->numActiveVoices == 0)) - { - /* - complete the process of resetting the synth now that - all voices have muted - */ -#ifdef _DEBUG_VM - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMAddSamples: complete the reset process\n"); */ } -#endif - - VMInitializeAllChannels(pVoiceMgr, pSynth); - VMInitializeAllVoices(pVoiceMgr, pSynth->vSynthNum); - - /* clear the reset flag */ - pSynth->synthFlags &= ~SYNTH_FLAG_RESET_IS_REQUESTED; - } - - /* clear channel update flags */ - for (channel = 0; channel < NUM_SYNTH_CHANNELS; channel++) - pSynth->channels[channel].channelFlags &= ~CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS; - - } - -#ifdef _CHECKED_BUILD - SanityCheck(pVoiceMgr); -#endif - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * VMInitWorkload() - *---------------------------------------------------------------------------- - * Purpose: - * Clears the workload counter - * - * Inputs: - * pVoiceMgr - pointer to instance data - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void VMInitWorkload (S_VOICE_MGR *pVoiceMgr) -{ - pVoiceMgr->workload = 0; -} - -/*---------------------------------------------------------------------------- - * VMSetWorkload() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the max workload for a single frame. - * - * Inputs: - * pVoiceMgr - pointer to instance data - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void VMSetWorkload (S_VOICE_MGR *pVoiceMgr, EAS_I32 maxWorkLoad) -{ - pVoiceMgr->maxWorkLoad = maxWorkLoad; -} - -/*---------------------------------------------------------------------------- - * VMCheckWorkload() - *---------------------------------------------------------------------------- - * Purpose: - * Checks to see if work load has been exceeded on this frame. - * - * Inputs: - * pVoiceMgr - pointer to instance data - * - * Outputs: - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_BOOL VMCheckWorkload (S_VOICE_MGR *pVoiceMgr) -{ - if (pVoiceMgr->maxWorkLoad > 0) - return (EAS_BOOL) (pVoiceMgr->workload >= pVoiceMgr->maxWorkLoad); - return EAS_FALSE; -} - -/*---------------------------------------------------------------------------- - * VMActiveVoices() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the number of active voices in the synthesizer. - * - * Inputs: - * pEASData - pointer to instance data - * - * Outputs: - * Returns the number of active voices - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_I32 VMActiveVoices (S_SYNTH *pSynth) -{ - return pSynth->numActiveVoices; -} - -/*---------------------------------------------------------------------------- - * VMSetSynthPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Set the synth to a new polyphony value. Value must be >= 1 and - * <= MAX_SYNTH_VOICES. This function will pin the polyphony at those limits - * - * Inputs: - * pVoiceMgr pointer to synthesizer data - * polyphonyCount desired polyphony count - * synth synthesizer number (0 = onboard, 1 = DSP) - * - * Outputs: - * Returns error code - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMSetSynthPolyphony (S_VOICE_MGR *pVoiceMgr, EAS_I32 synth, EAS_I32 polyphonyCount) -{ - EAS_INT i; - EAS_INT activeVoices; - - /* lower limit */ - if (polyphonyCount < 1) - polyphonyCount = 1; - - /* split architecture */ -#if defined(_SECONDARY_SYNTH) || defined(EAS_SPLIT_WT_SYNTH) - if (synth == EAS_MCU_SYNTH) - { - if (polyphonyCount > NUM_PRIMARY_VOICES) - polyphonyCount = NUM_PRIMARY_VOICES; - if (pVoiceMgr->maxPolyphonyPrimary == polyphonyCount) - return EAS_SUCCESS; - pVoiceMgr->maxPolyphonyPrimary = (EAS_U16) polyphonyCount; - } - else if (synth == EAS_DSP_SYNTH) - { - if (polyphonyCount > NUM_SECONDARY_VOICES) - polyphonyCount = NUM_SECONDARY_VOICES; - if (pVoiceMgr->maxPolyphonySecondary == polyphonyCount) - return EAS_SUCCESS; - pVoiceMgr->maxPolyphonySecondary = (EAS_U16) polyphonyCount; - } - else - return EAS_ERROR_PARAMETER_RANGE; - - /* setting for SP-MIDI */ - pVoiceMgr->maxPolyphony = pVoiceMgr->maxPolyphonyPrimary + pVoiceMgr->maxPolyphonySecondary; - - /* standard architecture */ -#else - if (synth != EAS_MCU_SYNTH) - return EAS_ERROR_PARAMETER_RANGE; - - /* pin desired value to possible limits */ - if (polyphonyCount > MAX_SYNTH_VOICES) - polyphonyCount = MAX_SYNTH_VOICES; - - /* set polyphony, if value is different than current value */ - if (pVoiceMgr->maxPolyphony == polyphonyCount) - return EAS_SUCCESS; - - pVoiceMgr->maxPolyphony = (EAS_U16) polyphonyCount; -#endif - - /* if SPMIDI enabled, update channel masking based on new polyphony */ - for (i = 0; i < MAX_VIRTUAL_SYNTHESIZERS; i++) - { - if (pVoiceMgr->pSynth[i]) - { - if (pVoiceMgr->pSynth[i]->synthFlags & SYNTH_FLAG_SP_MIDI_ON) - VMMIPUpdateChannelMuting(pVoiceMgr, pVoiceMgr->pSynth[i]); - else - pVoiceMgr->pSynth[i]->poolAlloc[0] = (EAS_U8) polyphonyCount; - } - } - - /* are we under polyphony limit? */ - if (pVoiceMgr->activeVoices <= polyphonyCount) - return EAS_SUCCESS; - - /* count the number of active voices */ - activeVoices = 0; - for (i = 0; i < MAX_SYNTH_VOICES; i++) - { - - /* is voice active? */ - if ((pVoiceMgr->voices[i].voiceState != eVoiceStateFree) && (pVoiceMgr->voices[i].voiceState != eVoiceStateMuting)) - activeVoices++; - } - - /* we may have to mute voices to reach new target */ - while (activeVoices > polyphonyCount) - { - S_SYNTH *pSynth; - S_SYNTH_VOICE *pVoice; - EAS_I32 currentPriority, bestPriority; - EAS_INT bestCandidate; - - /* find the lowest priority voice */ - bestPriority = bestCandidate = -1; - for (i = 0; i < MAX_SYNTH_VOICES; i++) - { - - pVoice = &pVoiceMgr->voices[i]; - - /* ignore free and muting voices */ - if ((pVoice->voiceState == eVoiceStateFree) || (pVoice->voiceState == eVoiceStateMuting)) - continue; - - pSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->channel)]; - - /* if voice is stolen or just started, reduce the likelihood it will be stolen */ - if (( pVoice->voiceState == eVoiceStateStolen) || (pVoice->voiceFlags & VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET)) - { - /* include velocity */ - currentPriority = 128 - pVoice->nextVelocity; - - /* include channel priority */ - currentPriority += pSynth->channels[GET_CHANNEL(pVoice->nextChannel)].pool << CHANNEL_PRIORITY_STEAL_WEIGHT; - } - else - { - /* include age */ - currentPriority = (EAS_I32) pVoice->age << NOTE_AGE_STEAL_WEIGHT; - - /* include note gain -higher gain is lower steal value */ - /*lint -e{704} use shift for performance */ - currentPriority += ((32768 >> (12 - NOTE_GAIN_STEAL_WEIGHT)) + 256) - - ((EAS_I32) pVoice->gain >> (12 - NOTE_GAIN_STEAL_WEIGHT)); - - /* include channel priority */ - currentPriority += pSynth->channels[GET_CHANNEL(pVoice->channel)].pool << CHANNEL_PRIORITY_STEAL_WEIGHT; - } - - /* include synth priority */ - currentPriority += pSynth->priority << SYNTH_PRIORITY_WEIGHT; - - /* is this the best choice so far? */ - if (currentPriority > bestPriority) - { - bestPriority = currentPriority; - bestCandidate = i; - } - } - - /* shutdown best candidate */ - if (bestCandidate < 0) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "VMSetPolyphony: Unable to reduce polyphony\n"); */ } - break; - } - - /* shut down this voice */ - /*lint -e{771} pSynth is initialized if bestCandidate >= 0 */ - VMMuteVoice(pVoiceMgr, bestCandidate); - activeVoices--; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * VMGetSynthPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current polyphony setting - * - * Inputs: - * pVoiceMgr pointer to synthesizer data - * synth synthesizer number (0 = onboard, 1 = DSP) - * - * Outputs: - * Returns actual polyphony value set, as pinned by limits - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMGetSynthPolyphony (S_VOICE_MGR *pVoiceMgr, EAS_I32 synth, EAS_I32 *pPolyphonyCount) -{ - -#if defined(_SECONDARY_SYNTH) || defined(EAS_SPLIT_WT_SYNTH) - if (synth == EAS_MCU_SYNTH) - *pPolyphonyCount = pVoiceMgr->maxPolyphonyPrimary; - else if (synth == EAS_DSP_SYNTH) - *pPolyphonyCount = pVoiceMgr->maxPolyphonySecondary; - else - return EAS_ERROR_PARAMETER_RANGE; -#else - if (synth != EAS_MCU_SYNTH) - return EAS_ERROR_PARAMETER_RANGE; - *pPolyphonyCount = pVoiceMgr->maxPolyphony; -#endif - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * VMSetPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Set the virtual synth polyphony. 0 = no limit (i.e. can use - * all available voices). - * - * Inputs: - * pVoiceMgr pointer to synthesizer data - * polyphonyCount desired polyphony count - * pSynth pointer to virtual synth - * - * Outputs: - * Returns error code - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMSetPolyphony (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 polyphonyCount) -{ - EAS_INT i; - EAS_INT activeVoices; - - /* check limits */ - if (polyphonyCount < 0) - return EAS_ERROR_PARAMETER_RANGE; - - /* zero is max polyphony */ - if ((polyphonyCount == 0) || (polyphonyCount > MAX_SYNTH_VOICES)) - { - pSynth->maxPolyphony = 0; - return EAS_SUCCESS; - } - - /* set new polyphony */ - pSynth->maxPolyphony = (EAS_U16) polyphonyCount; - - /* max polyphony is minimum of virtual synth and actual synth */ - if (polyphonyCount > pVoiceMgr->maxPolyphony) - polyphonyCount = pVoiceMgr->maxPolyphony; - - /* if SP-MIDI mode, update the channel muting */ - if (pSynth->synthFlags & SYNTH_FLAG_SP_MIDI_ON) - VMMIPUpdateChannelMuting(pVoiceMgr, pSynth); - else - pSynth->poolAlloc[0] = (EAS_U8) polyphonyCount; - - /* are we under polyphony limit? */ - if (pSynth->numActiveVoices <= polyphonyCount) - return EAS_SUCCESS; - - /* count the number of active voices */ - activeVoices = 0; - for (i = 0; i < MAX_SYNTH_VOICES; i++) - { - /* this synth? */ - if (GET_VSYNTH(pVoiceMgr->voices[i].nextChannel) != pSynth->vSynthNum) - continue; - - /* is voice active? */ - if ((pVoiceMgr->voices[i].voiceState != eVoiceStateFree) && (pVoiceMgr->voices[i].voiceState != eVoiceStateMuting)) - activeVoices++; - } - - /* we may have to mute voices to reach new target */ - while (activeVoices > polyphonyCount) - { - S_SYNTH_VOICE *pVoice; - EAS_I32 currentPriority, bestPriority; - EAS_INT bestCandidate; - - /* find the lowest priority voice */ - bestPriority = bestCandidate = -1; - for (i = 0; i < MAX_SYNTH_VOICES; i++) - { - pVoice = &pVoiceMgr->voices[i]; - - /* this synth? */ - if (GET_VSYNTH(pVoice->nextChannel) != pSynth->vSynthNum) - continue; - - /* if voice is stolen or just started, reduce the likelihood it will be stolen */ - if (( pVoice->voiceState == eVoiceStateStolen) || (pVoice->voiceFlags & VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET)) - { - /* include velocity */ - currentPriority = 128 - pVoice->nextVelocity; - - /* include channel priority */ - currentPriority += pSynth->channels[GET_CHANNEL(pVoice->nextChannel)].pool << CHANNEL_PRIORITY_STEAL_WEIGHT; - } - else - { - /* include age */ - currentPriority = (EAS_I32) pVoice->age << NOTE_AGE_STEAL_WEIGHT; - - /* include note gain -higher gain is lower steal value */ - /*lint -e{704} use shift for performance */ - currentPriority += ((32768 >> (12 - NOTE_GAIN_STEAL_WEIGHT)) + 256) - - ((EAS_I32) pVoice->gain >> (12 - NOTE_GAIN_STEAL_WEIGHT)); - - /* include channel priority */ - currentPriority += pSynth->channels[GET_CHANNEL(pVoice->nextChannel)].pool << CHANNEL_PRIORITY_STEAL_WEIGHT; - } - - /* is this the best choice so far? */ - if (currentPriority > bestPriority) - { - bestPriority = currentPriority; - bestCandidate = i; - } - } - - /* shutdown best candidate */ - if (bestCandidate < 0) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "VMSetPolyphony: Unable to reduce polyphony\n"); */ } - break; - } - - /* shut down this voice */ - VMMuteVoice(pVoiceMgr, bestCandidate); - activeVoices--; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * VMGetPolyphony() - *---------------------------------------------------------------------------- - * Purpose: - * Get the virtual synth polyphony - * - * Inputs: - * pVoiceMgr pointer to synthesizer data - * pPolyphonyCount pointer to variable to hold polyphony count - * pSynth pointer to virtual synth - * - * Outputs: - * Returns error code - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pVoiceMgr) reserved for future use */ -EAS_RESULT VMGetPolyphony (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 *pPolyphonyCount) -{ - *pPolyphonyCount = (EAS_U16) pSynth->maxPolyphony; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * VMSetPriority() - *---------------------------------------------------------------------------- - * Purpose: - * Set the virtual synth priority - * - * Inputs: - * pVoiceMgr pointer to synthesizer data - * priority new priority - * pSynth pointer to virtual synth - * - * Outputs: - * Returns error code - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pVoiceMgr) reserved for future use */ -EAS_RESULT VMSetPriority (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 priority) -{ - pSynth->priority = (EAS_U8) priority ; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * VMGetPriority() - *---------------------------------------------------------------------------- - * Purpose: - * Get the virtual synth priority - * - * Inputs: - * pVoiceMgr pointer to synthesizer data - * pPriority pointer to variable to hold priority - * pSynth pointer to virtual synth - * - * Outputs: - * Returns error code - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pVoiceMgr) reserved for future use */ -EAS_RESULT VMGetPriority (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 *pPriority) -{ - *pPriority = pSynth->priority; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * VMSetVolume() - *---------------------------------------------------------------------------- - * Purpose: - * Set the master volume for this synthesizer for this sequence. - * - * Inputs: - * nSynthVolume - the desired master volume - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * - * Side Effects: - * overrides any previously set master volume from sysex - * - *---------------------------------------------------------------------------- -*/ -void VMSetVolume (S_SYNTH *pSynth, EAS_U16 masterVolume) -{ - pSynth->masterVolume = masterVolume; - pSynth->synthFlags |= SYNTH_FLAG_UPDATE_ALL_CHANNEL_PARAMETERS; -} - -/*---------------------------------------------------------------------------- - * VMSetPitchBendRange() - *---------------------------------------------------------------------------- - * Set the pitch bend range for the given channel. - *---------------------------------------------------------------------------- -*/ -void VMSetPitchBendRange (S_SYNTH *pSynth, EAS_INT channel, EAS_I16 pitchBendRange) -{ - pSynth->channels[channel].pitchBendSensitivity = pitchBendRange; -} - -/*---------------------------------------------------------------------------- - * VMValidateEASLib() - *---------------------------------------------------------------------------- - * Validates an EAS library - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMValidateEASLib (EAS_SNDLIB_HANDLE pEAS) -{ - /* validate the sound library */ - if (pEAS) - { - if (pEAS->identifier != _EAS_LIBRARY_VERSION) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMValidateEASLib: Sound library mismatch in sound library: Read 0x%08x, expected 0x%08x\n", - pEAS->identifier, _EAS_LIBRARY_VERSION); */ } - return EAS_ERROR_SOUND_LIBRARY; - } - - /* check sample rate */ - if ((pEAS->libAttr & LIBFORMAT_SAMPLE_RATE_MASK) != _OUTPUT_SAMPLE_RATE) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMValidateEASLib: Sample rate mismatch in sound library: Read %lu, expected %lu\n", - pEAS->libAttr & LIBFORMAT_SAMPLE_RATE_MASK, _OUTPUT_SAMPLE_RATE); */ } - return EAS_ERROR_SOUND_LIBRARY; - } - -#ifdef _WT_SYNTH - /* check sample bit depth */ -#ifdef _8_BIT_SAMPLES - if (pEAS->libAttr & LIB_FORMAT_16_BIT_SAMPLES) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMValidateEASLib: Expected 8-bit samples and found 16-bit\n", - pEAS->libAttr & LIBFORMAT_SAMPLE_RATE_MASK, _OUTPUT_SAMPLE_RATE); */ } - return EAS_ERROR_SOUND_LIBRARY; - } -#endif -#ifdef _16_BIT_SAMPLES - if ((pEAS->libAttr & LIB_FORMAT_16_BIT_SAMPLES) == 0) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMValidateEASLib: Expected 16-bit samples and found 8-bit\n", - pEAS->libAttr & LIBFORMAT_SAMPLE_RATE_MASK, _OUTPUT_SAMPLE_RATE); */ } - return EAS_ERROR_SOUND_LIBRARY; - } -#endif -#endif - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * VMSetGlobalEASLib() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the EAS library to be used by the synthesizer - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMSetGlobalEASLib (S_VOICE_MGR *pVoiceMgr, EAS_SNDLIB_HANDLE pEAS) -{ - EAS_RESULT result; - - result = VMValidateEASLib(pEAS); - if (result != EAS_SUCCESS) - return result; - - pVoiceMgr->pGlobalEAS = pEAS; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * VMSetEASLib() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the EAS library to be used by the synthesizer - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMSetEASLib (S_SYNTH *pSynth, EAS_SNDLIB_HANDLE pEAS) -{ - EAS_RESULT result; - - result = VMValidateEASLib(pEAS); - if (result != EAS_SUCCESS) - return result; - - pSynth->pEAS = pEAS; - return EAS_SUCCESS; -} - -#ifdef DLS_SYNTHESIZER -/*---------------------------------------------------------------------------- - * VMSetGlobalDLSLib() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the DLS library to be used by the synthesizer - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMSetGlobalDLSLib (EAS_DATA_HANDLE pEASData, EAS_DLSLIB_HANDLE pDLS) -{ - - if (pEASData->pVoiceMgr->pGlobalDLS) - DLSCleanup(pEASData->hwInstData, pEASData->pVoiceMgr->pGlobalDLS); - - pEASData->pVoiceMgr->pGlobalDLS = pDLS; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * VMSetDLSLib() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the DLS library to be used by the synthesizer - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMSetDLSLib (S_SYNTH *pSynth, EAS_DLSLIB_HANDLE pDLS) -{ - pSynth->pDLS = pDLS; - return EAS_SUCCESS; -} -#endif - -/*---------------------------------------------------------------------------- - * VMSetTranposition() - *---------------------------------------------------------------------------- - * Purpose: - * Sets the global key transposition used by the synthesizer. - * Transposes all melodic instruments up or down by the specified - * amount. Range is limited to +/-12 semitones. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void VMSetTranposition (S_SYNTH *pSynth, EAS_I32 transposition) -{ - pSynth->globalTranspose = (EAS_I8) transposition; -} - -/*---------------------------------------------------------------------------- - * VMGetTranposition() - *---------------------------------------------------------------------------- - * Purpose: - * Gets the global key transposition used by the synthesizer. - * Transposes all melodic instruments up or down by the specified - * amount. Range is limited to +/-12 semitones. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void VMGetTranposition (S_SYNTH *pSynth, EAS_I32 *pTransposition) -{ - *pTransposition = pSynth->globalTranspose; -} - -/*---------------------------------------------------------------------------- - * VMGetNoteCount() - *---------------------------------------------------------------------------- -* Returns the total note count -*---------------------------------------------------------------------------- -*/ -EAS_I32 VMGetNoteCount (S_SYNTH *pSynth) -{ - return pSynth->totalNoteCount; -} - -/*---------------------------------------------------------------------------- - * VMMIDIShutdown() - *---------------------------------------------------------------------------- - * Purpose: - * Clean up any Synth related system issues. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * None - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void VMMIDIShutdown (S_EAS_DATA *pEASData, S_SYNTH *pSynth) -{ - EAS_INT vSynthNum; - - /* decrement reference count, free if all references are gone */ - if (--pSynth->refCount > 0) - return; - - vSynthNum = pSynth->vSynthNum; - - /* cleanup DLS load */ -#ifdef DLS_SYNTHESIZER - /*lint -e{550} result used only in debugging code */ - if (pSynth->pDLS != NULL) - { - EAS_RESULT result; - if ((result = DLSCleanup(pEASData->hwInstData, pSynth->pDLS)) != EAS_SUCCESS) - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMMIDIShutdown: Error %ld cleaning up DLS collection\n", result); */ } - pSynth->pDLS = NULL; - } -#endif - - VMReset(pEASData->pVoiceMgr, pSynth, EAS_TRUE); - - /* check Configuration Module for static memory allocation */ - if (!pEASData->staticMemoryModel) - EAS_HWFree(pEASData->hwInstData, pSynth); - - /* clear pointer to MIDI state */ - pEASData->pVoiceMgr->pSynth[vSynthNum] = NULL; -} - -/*---------------------------------------------------------------------------- - * VMShutdown() - *---------------------------------------------------------------------------- - * Purpose: - * Clean up any Synth related system issues. - * - * Inputs: - * psEASData - pointer to overall EAS data structure - * - * Outputs: - * None - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -void VMShutdown (S_EAS_DATA *pEASData) -{ - - /* don't free a NULL pointer */ - if (pEASData->pVoiceMgr == NULL) - return; - -#ifdef DLS_SYNTHESIZER - /* if we have a global DLS collection, clean it up */ - if (pEASData->pVoiceMgr->pGlobalDLS) - { - DLSCleanup(pEASData->hwInstData, pEASData->pVoiceMgr->pGlobalDLS); - pEASData->pVoiceMgr->pGlobalDLS = NULL; - } -#endif - - /* check Configuration Module for static memory allocation */ - if (!pEASData->staticMemoryModel) - EAS_HWFree(pEASData->hwInstData, pEASData->pVoiceMgr); - pEASData->pVoiceMgr = NULL; -} - -#ifdef EXTERNAL_AUDIO -/*---------------------------------------------------------------------------- - * EAS_RegExtAudioCallback() - *---------------------------------------------------------------------------- - * Register a callback for external audio processing - *---------------------------------------------------------------------------- -*/ -void VMRegExtAudioCallback (S_SYNTH *pSynth, EAS_VOID_PTR pInstData, EAS_EXT_PRG_CHG_FUNC cbProgChgFunc, EAS_EXT_EVENT_FUNC cbEventFunc) -{ - pSynth->pExtAudioInstData = pInstData; - pSynth->cbProgChgFunc = cbProgChgFunc; - pSynth->cbEventFunc = cbEventFunc; -} - -/*---------------------------------------------------------------------------- - * VMGetMIDIControllers() - *---------------------------------------------------------------------------- - * Returns the MIDI controller values on the specified channel - *---------------------------------------------------------------------------- -*/ -void VMGetMIDIControllers (S_SYNTH *pSynth, EAS_U8 channel, S_MIDI_CONTROLLERS *pControl) -{ - pControl->modWheel = pSynth->channels[channel].modWheel; - pControl->volume = pSynth->channels[channel].volume; - pControl->pan = pSynth->channels[channel].pan; - pControl->expression = pSynth->channels[channel].expression; - pControl->channelPressure = pSynth->channels[channel].channelPressure; - -#ifdef _REVERB - pControl->reverbSend = pSynth->channels[channel].reverbSend; -#endif - -#ifdef _CHORUSE - pControl->chorusSend = pSynth->channels[channel].chorusSend; -#endif -} -#endif - -#ifdef _SPLIT_ARCHITECTURE -/*---------------------------------------------------------------------------- - * VMStartFrame() - *---------------------------------------------------------------------------- - * Purpose: - * Starts an audio frame - * - * Inputs: - * - * Outputs: - * Returns true if EAS_MixEnginePrep should be called (onboard mixing) - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_BOOL VMStartFrame (S_EAS_DATA *pEASData) -{ - - /* init counter for voices starts in split architecture */ -#ifdef MAX_VOICE_STARTS - pVoiceMgr->numVoiceStarts = 0; -#endif - - return pFrameInterface->pfStartFrame(pEASData->pVoiceMgr->pFrameBuffer); -} - -/*---------------------------------------------------------------------------- - * VMEndFrame() - *---------------------------------------------------------------------------- - * Purpose: - * Stops an audio frame - * - * Inputs: - * - * Outputs: - * Returns true if EAS_MixEnginePost should be called (onboard mixing) - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -EAS_BOOL VMEndFrame (S_EAS_DATA *pEASData) -{ - - return pFrameInterface->pfEndFrame(pEASData->pVoiceMgr->pFrameBuffer, pEASData->pMixBuffer, pEASData->masterGain); -} -#endif - -#ifdef TEST_HARNESS -/*---------------------------------------------------------------------------- - * SanityCheck() - *---------------------------------------------------------------------------- -*/ -EAS_RESULT VMSanityCheck (EAS_DATA_HANDLE pEASData) -{ - S_SYNTH_VOICE *pVoice; - S_SYNTH *pSynth; - EAS_INT i; - EAS_INT j; - EAS_INT freeVoices; - EAS_INT activeVoices; - EAS_INT playingVoices; - EAS_INT stolenVoices; - EAS_INT releasingVoices; - EAS_INT mutingVoices; - EAS_INT poolCount[MAX_VIRTUAL_SYNTHESIZERS][NUM_SYNTH_CHANNELS]; - EAS_INT vSynthNum; - EAS_RESULT result = EAS_SUCCESS; - - /* initialize counts */ - EAS_HWMemSet(poolCount, 0, sizeof(poolCount)); - freeVoices = activeVoices = playingVoices = stolenVoices = releasingVoices = mutingVoices = 0; - - /* iterate through all voices */ - for (i = 0; i < MAX_SYNTH_VOICES; i++) - { - pVoice = &pEASData->pVoiceMgr->voices[i]; - if (pVoice->voiceState != eVoiceStateFree) - { - vSynthNum = GET_VSYNTH(pVoice->channel); - if (vSynthNum >= MAX_VIRTUAL_SYNTHESIZERS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMSanityCheck: Voice %d has invalid virtual synth number %d\n", i, vSynthNum); */ } - result = EAS_FAILURE; - continue; - } - pSynth = pEASData->pVoiceMgr->pSynth[vSynthNum]; - - switch (pVoice->voiceState) - { - case eVoiceStateMuting: - activeVoices++; - mutingVoices++; - break; - - case eVoiceStateStolen: - vSynthNum = GET_VSYNTH(pVoice->nextChannel); - if (vSynthNum >= MAX_VIRTUAL_SYNTHESIZERS) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMSanityCheck: Voice %d has invalid virtual synth number %d\n", i, vSynthNum); */ } - result = EAS_FAILURE; - continue; - } - pSynth = pEASData->pVoiceMgr->pSynth[vSynthNum]; - activeVoices++; - stolenVoices++; - poolCount[vSynthNum][pSynth->channels[GET_CHANNEL(pVoice->nextChannel)].pool]++; - break; - - case eVoiceStateStart: - case eVoiceStatePlay: - activeVoices++; - playingVoices++; - poolCount[vSynthNum][pSynth->channels[GET_CHANNEL(pVoice->channel)].pool]++; - break; - - case eVoiceStateRelease: - activeVoices++; - releasingVoices++; - poolCount[vSynthNum][pSynth->channels[GET_CHANNEL(pVoice->channel)].pool]++; - break; - - default: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMSanityCheck : voice %d in invalid state\n", i); */ } - result = EAS_FAILURE; - break; - } - } - - /* count free voices */ - else - freeVoices++; - } - - /* dump state info */ - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "%d free\n", freeVoices); */ } - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "%d active\n", activeVoices); */ } - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "%d playing\n", playingVoices); */ } - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "%d releasing\n", releasingVoices); */ } - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "%d muting\n", mutingVoices); */ } - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "%d stolen\n", stolenVoices); */ } - - if (pEASData->pVoiceMgr->activeVoices != activeVoices) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Active voice mismatch was %d should be %d\n", - pEASData->pVoiceMgr->activeVoices, activeVoices); */ } - result = EAS_FAILURE; - } - - /* check virtual synth status */ - for (i = 0; i < MAX_VIRTUAL_SYNTHESIZERS; i++) - { - if (pEASData->pVoiceMgr->pSynth[i] == NULL) - continue; - - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Synth %d numActiveVoices: %d\n", i, pEASData->pVoiceMgr->pSynth[i]->numActiveVoices); */ } - if (pEASData->pVoiceMgr->pSynth[i]->numActiveVoices > MAX_SYNTH_VOICES) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMSanityCheck: Synth %d illegal count for numActiveVoices: %d\n", i, pEASData->pVoiceMgr->pSynth[i]->numActiveVoices); */ } - result = EAS_FAILURE; - } - for (j = 0; j < NUM_SYNTH_CHANNELS; j++) - { - if (poolCount[i][j] != pEASData->pVoiceMgr->pSynth[i]->poolCount[j]) - { - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Pool count mismatch synth %d pool %d, was %d, should be %d\n", - i, j, pEASData->pVoiceMgr->pSynth[i]->poolCount[j], poolCount[i][j]); */ } - result = EAS_FAILURE; - } - } - } - - return result; -} -#endif - - diff --git a/arm-hybrid-22k/lib_src/eas_wavefile.c b/arm-hybrid-22k/lib_src/eas_wavefile.c deleted file mode 100644 index d3f3ba0..0000000 --- a/arm-hybrid-22k/lib_src/eas_wavefile.c +++ /dev/null @@ -1,867 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_wavefile.c - * - * Contents and purpose: - * This file implements the wave file parser. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 852 $ - * $Date: 2007-09-04 11:43:49 -0700 (Tue, 04 Sep 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_data.h" -#include "eas_report.h" -#include "eas_host.h" -#include "eas_config.h" -#include "eas_parser.h" -#include "eas_pcm.h" -#include "eas_wavefile.h" - -/* lint is choking on the ARM math.h file, so we declare the log10 function here */ -extern double log10(double x); - -/* increase gain to compensate for loss in mixer */ -#define WAVE_GAIN_OFFSET 6 - -/* constant for 1200 / log10(2.0) */ -#define PITCH_CENTS_CONVERSION 3986.313714 - -/*---------------------------------------------------------------------------- - * WAVE file defines - *---------------------------------------------------------------------------- -*/ -/* RIFF chunks */ -#define CHUNK_TYPE(a,b,c,d) ( \ - ( ((EAS_U32)(a) & 0xFF) << 24 ) \ - + ( ((EAS_U32)(b) & 0xFF) << 16 ) \ - + ( ((EAS_U32)(c) & 0xFF) << 8 ) \ - + ( ((EAS_U32)(d) & 0xFF) ) ) - -#define CHUNK_RIFF CHUNK_TYPE('R','I','F','F') -#define CHUNK_WAVE CHUNK_TYPE('W','A','V','E') -#define CHUNK_FMT CHUNK_TYPE('f','m','t',' ') -#define CHUNK_DATA CHUNK_TYPE('d','a','t','a') -#define CHUNK_LIST CHUNK_TYPE('L','I','S','T') -#define CHUNK_INFO CHUNK_TYPE('I','N','F','O') -#define CHUNK_INAM CHUNK_TYPE('I','N','A','M') -#define CHUNK_ICOP CHUNK_TYPE('I','C','O','P') -#define CHUNK_IART CHUNK_TYPE('I','A','R','T') - -/* wave file format identifiers */ -#define WAVE_FORMAT_PCM 0x0001 -#define WAVE_FORMAT_IMA_ADPCM 0x0011 - -/* file size for streamed file */ -#define FILE_SIZE_STREAMING 0x80000000 - -/*---------------------------------------------------------------------------- - * prototypes - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT WaveCheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *pHandle, EAS_I32 offset); -static EAS_RESULT WavePrepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT WaveState (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState); -static EAS_RESULT WaveClose (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT WaveReset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT WaveLocate (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 time, EAS_BOOL *pParserLocate); -static EAS_RESULT WavePause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT WaveResume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData); -static EAS_RESULT WaveSetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value); -static EAS_RESULT WaveGetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue); -static EAS_RESULT WaveParseHeader (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, S_WAVE_STATE *pWaveData); -static EAS_RESULT WaveGetMetaData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pMediaLength); - -#ifdef MMAPI_SUPPORT -static EAS_RESULT SaveFmtChunk (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, S_WAVE_STATE *pWaveData, EAS_I32 size); -#endif - -/*---------------------------------------------------------------------------- - * - * EAS_Wave_Parser - * - * This structure contains the functional interface for the Wave file parser - *---------------------------------------------------------------------------- -*/ -const S_FILE_PARSER_INTERFACE EAS_Wave_Parser = -{ - WaveCheckFileType, - WavePrepare, - NULL, - NULL, - WaveState, - WaveClose, - WaveReset, - WavePause, - WaveResume, - WaveLocate, - WaveSetData, - WaveGetData, - WaveGetMetaData -}; - -/*---------------------------------------------------------------------------- - * WaveCheckFileType() - *---------------------------------------------------------------------------- - * Purpose: - * Check the file type to see if we can parse it - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT WaveCheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *pHandle, EAS_I32 offset) -{ - S_WAVE_STATE *pWaveData; - - /* zero the memory to insure complete initialization */ - *pHandle = NULL; - - /* read the file header */ - if (WaveParseHeader(pEASData, fileHandle, NULL) == EAS_SUCCESS) - { - - /* check for static memory allocation */ - if (pEASData->staticMemoryModel) - pWaveData = EAS_CMEnumData(EAS_CM_WAVE_DATA); - else - pWaveData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_WAVE_STATE)); - if (!pWaveData) - return EAS_ERROR_MALLOC_FAILED; - EAS_HWMemSet(pWaveData, 0, sizeof(S_WAVE_STATE)); - - /* return a pointer to the instance data */ - pWaveData->fileHandle = fileHandle; - pWaveData->fileOffset = offset; - *pHandle = pWaveData; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * WavePrepare() - *---------------------------------------------------------------------------- - * Purpose: - * Prepare to parse the file. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT WavePrepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_WAVE_STATE *pWaveData; - EAS_RESULT result; - - /* validate parser state */ - pWaveData = (S_WAVE_STATE*) pInstData; - if (pWaveData->streamHandle != NULL) - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; - - /* back to start of file */ - pWaveData->time = 0; - if ((result = EAS_HWFileSeek(pEASData->hwInstData, pWaveData->fileHandle, pWaveData->fileOffset)) != EAS_SUCCESS) - return result; - - /* parse the file header */ - if ((result = WaveParseHeader(pEASData, pWaveData->fileHandle, pWaveData)) != EAS_SUCCESS) - return result; - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * WaveState() - *---------------------------------------------------------------------------- - * Purpose: - * Returns the current state of the stream - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * pState - pointer to variable to store state - * - * Outputs: - * - * - * Side Effects: - * - * Notes: - * This interface is also exposed in the internal library for use by the other modules. - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT WaveState (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState) -{ - S_WAVE_STATE *pWaveData; - - /* return current state */ - pWaveData = (S_WAVE_STATE*) pInstData; - if (pWaveData->streamHandle) - return EAS_PEState(pEASData, pWaveData->streamHandle, pState); - - /* if no stream handle, and time is not zero, we are done */ - if (pWaveData->time > 0) - *pState = EAS_STATE_STOPPED; - else - *pState = EAS_STATE_OPEN; - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * WaveClose() - *---------------------------------------------------------------------------- - * Purpose: - * Close the file and clean up - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT WaveClose (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - S_WAVE_STATE *pWaveData; - EAS_RESULT result; - - pWaveData = (S_WAVE_STATE*) pInstData; - - /* close the stream */ - if (pWaveData->streamHandle) - { - if ((result = EAS_PEClose(pEASData, pWaveData->streamHandle)) != EAS_SUCCESS) - return result; - pWaveData->streamHandle = NULL; - } - - /* if using dynamic memory, free it */ - if (!pEASData->staticMemoryModel) - { - -#ifdef MMAPI_SUPPORT - /* need to free the fmt chunk */ - if (pWaveData->fmtChunk != NULL) - EAS_HWFree(pEASData->hwInstData, pWaveData->fmtChunk); -#endif - - /* free the instance data */ - EAS_HWFree(pEASData->hwInstData, pWaveData); - - } - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * WaveReset() - *---------------------------------------------------------------------------- - * Purpose: - * Reset the sequencer. Used for locating backwards in the file. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT WaveReset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - EAS_PCM_HANDLE streamHandle; - - /* reset to first byte of data in the stream */ - streamHandle = ((S_WAVE_STATE*)pInstData)->streamHandle; - if (streamHandle) - return EAS_PEReset(pEASData, streamHandle); - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; -} - -/*---------------------------------------------------------------------------- - * WaveLocate() - *---------------------------------------------------------------------------- - * Purpose: - * Rewind/fast-forward in file. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * handle - pointer to file handle - * time - time (in msecs) - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pParserLocate) reserved for future use */ -static EAS_RESULT WaveLocate (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 time, EAS_BOOL *pParserLocate) -{ - EAS_PCM_HANDLE streamHandle; - - /* reset to first byte of data in the stream */ - streamHandle = ((S_WAVE_STATE*)pInstData)->streamHandle; - if (streamHandle) - return EAS_PELocate(pEASData, streamHandle, time); - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; -} - -/*---------------------------------------------------------------------------- - * WavePause() - *---------------------------------------------------------------------------- - * Purpose: - * Mute and stop rendering a PCM stream. Sets the gain target to zero and stops the playback - * at the end of the next audio frame. - * - * Inputs: - * pEASData - pointer to EAS library instance data - * handle - pointer to S_WAVE_STATE for this stream - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -static EAS_RESULT WavePause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - EAS_PCM_HANDLE streamHandle; - - /* pause the stream */ - streamHandle = ((S_WAVE_STATE*)pInstData)->streamHandle; - if (streamHandle) - return EAS_PEPause(pEASData, streamHandle); - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; -} - -/*---------------------------------------------------------------------------- - * WaveResume() - *---------------------------------------------------------------------------- - * Purpose: - * Resume rendering a PCM stream. Sets the gain target back to its - * previous setting and restarts playback at the end of the next audio - * frame. - * - * Inputs: - * pEASData - pointer to EAS library instance data - * handle - pointer to S_WAVE_STATE for this stream - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -static EAS_RESULT WaveResume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) -{ - EAS_PCM_HANDLE streamHandle; - - /* resume the stream */ - streamHandle = ((S_WAVE_STATE*)pInstData)->streamHandle; - if (streamHandle) - return EAS_PEResume(pEASData, streamHandle); - return EAS_ERROR_NOT_VALID_IN_THIS_STATE; -} - -/*---------------------------------------------------------------------------- - * WaveSetData() - *---------------------------------------------------------------------------- - * Purpose: - * - * Inputs: - * pEASData - pointer to EAS library instance data - * handle - pointer to S_WAVE_STATE for this stream - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT WaveSetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value) -{ - S_WAVE_STATE *pWaveData = (S_WAVE_STATE*) pInstData; - - switch (param) - { - /* set metadata callback */ - case PARSER_DATA_METADATA_CB: - EAS_HWMemCpy(&pWaveData->metadata, (void*) value, sizeof(S_METADATA_CB)); - return EAS_SUCCESS; - - case PARSER_DATA_PLAYBACK_RATE: - value = (EAS_I32) (PITCH_CENTS_CONVERSION * log10((double) value / (double) (1 << 28))); - return EAS_PEUpdatePitch(pEASData, pWaveData->streamHandle, (EAS_I16) value); - - case PARSER_DATA_VOLUME: - return EAS_PEUpdateVolume(pEASData, pWaveData->streamHandle, (EAS_I16) value); - - default: - return EAS_ERROR_INVALID_PARAMETER; - } -} - -/*---------------------------------------------------------------------------- - * WaveGetData() - *---------------------------------------------------------------------------- - * Purpose: - * - * Inputs: - * pEASData - pointer to EAS library instance data - * handle - pointer to S_WAVE_STATE for this stream - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pEASData) reserved for future use */ -static EAS_RESULT WaveGetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue) -{ - S_WAVE_STATE *pWaveData; - - pWaveData = (S_WAVE_STATE*) pInstData; - switch (param) - { - /* return file type as WAVE */ - case PARSER_DATA_FILE_TYPE: - *pValue = pWaveData->fileType; - break; - -#ifdef MMAPI_SUPPORT - /* return pointer to 'fmt' chunk */ - case PARSER_DATA_FORMAT: - *pValue = (EAS_I32) pWaveData->fmtChunk; - break; -#endif - - case PARSER_DATA_GAIN_OFFSET: - *pValue = WAVE_GAIN_OFFSET; - break; - - default: - return EAS_ERROR_INVALID_PARAMETER; - } - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * WaveParseHeader() - *---------------------------------------------------------------------------- - * Purpose: - * Parse the WAVE file header. - * - * Inputs: - * pEASData - pointer to EAS library instance data - * handle - pointer to S_WAVE_STATE for this stream - * - * Outputs: - * - * - * Side Effects: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT WaveParseHeader (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, S_WAVE_STATE *pWaveData) -{ - S_PCM_OPEN_PARAMS params; - EAS_RESULT result; - EAS_U32 tag; - EAS_U32 fileSize; - EAS_U32 size; - EAS_I32 pos; - EAS_I32 audioOffset; - EAS_U16 usTemp; - EAS_BOOL parseDone; - EAS_U32 avgBytesPerSec; - - /* init some data (and keep lint happy) */ - params.sampleRate = 0; - params.size = 0; - audioOffset = 0; - params.decoder = 0; - params.blockSize = 0; - params.pCallbackFunc = NULL; - params.cbInstData = NULL; - params.loopSamples = 0; - params.fileHandle = fileHandle; - params.volume = 0x7fff; - params.envData = 0; - avgBytesPerSec = 8000; - - /* check for 'RIFF' tag */ - if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &tag, EAS_TRUE)) != EAS_FALSE) - return result; - if (tag != CHUNK_RIFF) - return EAS_ERROR_UNRECOGNIZED_FORMAT; - - /* get size */ - if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &fileSize, EAS_FALSE)) != EAS_FALSE) - return result; - - /* check for 'WAVE' tag */ - if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &tag, EAS_TRUE)) != EAS_FALSE) - return result; - if (tag != CHUNK_WAVE) - return EAS_ERROR_UNRECOGNIZED_FORMAT; - - /* this is enough to say we recognize the file */ - if (pWaveData == NULL) - return EAS_SUCCESS; - - /* check for streaming mode */ - pWaveData->flags = 0; - pWaveData->mediaLength = -1; - pWaveData->infoChunkPos = -1; - pWaveData->infoChunkSize = -1; - if (fileSize== FILE_SIZE_STREAMING) - { - pWaveData->flags |= PCM_FLAGS_STREAMING; - fileSize = 0x7fffffff; - } - - /* find out where we're at */ - if ((result = EAS_HWFilePos(pEASData->hwInstData, fileHandle, &pos)) != EAS_SUCCESS) - return result; - fileSize -= 4; - - parseDone = EAS_FALSE; - for (;;) - { - /* get tag and size for next chunk */ - if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &tag, EAS_TRUE)) != EAS_FALSE) - return result; - if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &size, EAS_FALSE)) != EAS_FALSE) - return result; - - /* process chunk */ - pos += 8; - switch (tag) - { - case CHUNK_FMT: - -#ifdef MMAPI_SUPPORT - if ((result = SaveFmtChunk(pEASData, fileHandle, pWaveData, (EAS_I32) size)) != EAS_SUCCESS) - return result; -#endif - - /* get audio format */ - if ((result = EAS_HWGetWord(pEASData->hwInstData, fileHandle, &usTemp, EAS_FALSE)) != EAS_FALSE) - return result; - if (usTemp == WAVE_FORMAT_PCM) - { - params.decoder = EAS_DECODER_PCM; - pWaveData->fileType = EAS_FILE_WAVE_PCM; - } - else if (usTemp == WAVE_FORMAT_IMA_ADPCM) - { - params.decoder = EAS_DECODER_IMA_ADPCM; - pWaveData->fileType = EAS_FILE_WAVE_IMA_ADPCM; - } - else - return EAS_ERROR_UNRECOGNIZED_FORMAT; - - /* get number of channels */ - if ((result = EAS_HWGetWord(pEASData->hwInstData, fileHandle, &usTemp, EAS_FALSE)) != EAS_FALSE) - return result; - if (usTemp == 2) - pWaveData->flags |= PCM_FLAGS_STEREO; - else if (usTemp != 1) - return EAS_ERROR_UNRECOGNIZED_FORMAT; - - /* get sample rate */ - if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, ¶ms.sampleRate, EAS_FALSE)) != EAS_FALSE) - return result; - - /* get stream rate */ - if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &avgBytesPerSec, EAS_FALSE)) != EAS_FALSE) - return result; - - /* get block alignment */ - if ((result = EAS_HWGetWord(pEASData->hwInstData, fileHandle, &usTemp, EAS_FALSE)) != EAS_FALSE) - return result; - params.blockSize = usTemp; - - /* get bits per sample */ - if ((result = EAS_HWGetWord(pEASData->hwInstData, fileHandle, &usTemp, EAS_FALSE)) != EAS_FALSE) - return result; - - /* PCM, must be 8 or 16 bit samples */ - if (params.decoder == EAS_DECODER_PCM) - { - if (usTemp == 8) - pWaveData->flags |= PCM_FLAGS_8_BIT | PCM_FLAGS_UNSIGNED; - else if (usTemp != 16) - return EAS_ERROR_UNRECOGNIZED_FORMAT; - } - - /* for IMA ADPCM, we only support mono 4-bit ADPCM */ - else - { - if ((usTemp != 4) || (pWaveData->flags & PCM_FLAGS_STEREO)) - return EAS_ERROR_UNRECOGNIZED_FORMAT; - } - - break; - - case CHUNK_DATA: - audioOffset = pos; - if (pWaveData->flags & PCM_FLAGS_STREAMING) - { - params.size = 0x7fffffff; - parseDone = EAS_TRUE; - } - else - { - params.size = (EAS_I32) size; - params.loopStart = size; - /* use more accurate method if possible */ - if (size <= (0x7fffffff / 1000)) - pWaveData->mediaLength = (EAS_I32) ((size * 1000) / avgBytesPerSec); - else - pWaveData->mediaLength = (EAS_I32) (size / (avgBytesPerSec / 1000)); - } - break; - - case CHUNK_LIST: - /* get the list type */ - if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &tag, EAS_TRUE)) != EAS_FALSE) - return result; - if (tag == CHUNK_INFO) - { - pWaveData->infoChunkPos = pos + 4; - pWaveData->infoChunkSize = (EAS_I32) size - 4; - } - break; - - default: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WaveParseHeader: %c%c%c%c chunk - %d byte(s) ignored\n", - (char) (tag >> 24), (char) (tag >> 16), (char) (tag >> 8), (char) tag, size); */ } - break; - } - - if (parseDone) - break; - - /* subtract header size */ - fileSize -= 8; - - /* account for zero-padding on odd length chunks */ - if (size & 1) - size++; - - /* this check works for files with odd length last chunk and no zero-pad */ - if (size >= fileSize) - { - if (size > fileSize) - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WaveParseHeader: '%c%c%c%c' chunk size exceeds length of file or is not zero-padded\n", - (char) (tag >> 24), (char) (tag >> 16), (char) (tag >> 8), (char) tag, size); */ } - break; - } - - /* subtract size of data chunk (including any zero-pad) */ - fileSize -= size; - - /* seek to next chunk */ - pos += (EAS_I32) size; - if ((result = EAS_HWFileSeek(pEASData->hwInstData, fileHandle, pos)) != EAS_SUCCESS) - return result; - } - - /* check for valid header */ - if ((params.sampleRate == 0) || (params.size == 0)) - return EAS_ERROR_UNRECOGNIZED_FORMAT; - - /* save the pertinent information */ - pWaveData->audioOffset = audioOffset; - params.flags = pWaveData->flags; - - /* seek to data */ - if ((result = EAS_HWFileSeek(pEASData->hwInstData, fileHandle, audioOffset)) != EAS_SUCCESS) - return result; - - /* open a stream in the PCM engine */ - return EAS_PEOpenStream(pEASData, ¶ms, &pWaveData->streamHandle); -} - -/*---------------------------------------------------------------------------- - * WaveGetMetaData() - *---------------------------------------------------------------------------- - * Purpose: - * Process the INFO chunk and return metadata to host - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT WaveGetMetaData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pMediaLength) -{ - S_WAVE_STATE *pWaveData; - EAS_RESULT result; - EAS_I32 pos; - EAS_U32 size; - EAS_I32 infoSize; - EAS_U32 tag; - EAS_I32 restorePos; - E_EAS_METADATA_TYPE metaType; - EAS_I32 metaLen; - - /* get current position so we can restore it */ - pWaveData = (S_WAVE_STATE*) pInstData; - - /* return media length */ - *pMediaLength = pWaveData->mediaLength; - - /* did we encounter an INFO chunk? */ - if (pWaveData->infoChunkPos < 0) - return EAS_SUCCESS; - - if ((result = EAS_HWFilePos(pEASData->hwInstData, pWaveData->fileHandle, &restorePos)) != EAS_SUCCESS) - return result; - - /* offset to start of first chunk in INFO chunk */ - pos = pWaveData->infoChunkPos; - infoSize = pWaveData->infoChunkSize; - - /* read all the chunks in the INFO chunk */ - for (;;) - { - - /* seek to next chunk */ - if ((result = EAS_HWFileSeek(pEASData->hwInstData, pWaveData->fileHandle, pos)) != EAS_SUCCESS) - return result; - - /* get tag and size for next chunk */ - if ((result = EAS_HWGetDWord(pEASData->hwInstData, pWaveData->fileHandle, &tag, EAS_TRUE)) != EAS_FALSE) - return result; - if ((result = EAS_HWGetDWord(pEASData->hwInstData, pWaveData->fileHandle, &size, EAS_FALSE)) != EAS_FALSE) - return result; - - /* process chunk */ - pos += 8; - metaType = EAS_METADATA_UNKNOWN; - switch (tag) - { - case CHUNK_INAM: - metaType = EAS_METADATA_TITLE; - break; - - case CHUNK_IART: - metaType = EAS_METADATA_AUTHOR; - break; - - case CHUNK_ICOP: - metaType = EAS_METADATA_COPYRIGHT; - break; - - default: - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WaveParseHeader: %c%c%c%c chunk - %d byte(s) ignored\n", - (char) (tag >> 24), (char) (tag >> 16), (char) (tag >> 8), (char) tag, size); */ } - break; - } - - /* process known metadata */ - if (metaType != EAS_METADATA_UNKNOWN) - { - metaLen = pWaveData->metadata.bufferSize - 1; - if (metaLen > (EAS_I32) size) - metaLen = (EAS_I32) size; - if ((result = EAS_HWReadFile(pEASData->hwInstData, pWaveData->fileHandle, pWaveData->metadata.buffer, metaLen, &metaLen)) != EAS_SUCCESS) - return result; - pWaveData->metadata.buffer[metaLen] = 0; - pWaveData->metadata.callback(metaType, pWaveData->metadata.buffer, pWaveData->metadata.pUserData); - } - - /* subtract this block */ - if (size & 1) - size++; - infoSize -= (EAS_I32) size + 8; - if (infoSize == 0) - break; - pos += (EAS_I32) size; - } - - - /* restore original position */ - return EAS_HWFileSeek(pEASData->hwInstData, pWaveData->fileHandle, restorePos); -} - -#ifdef MMAPI_SUPPORT -/*---------------------------------------------------------------------------- - * SaveFmtChunk() - *---------------------------------------------------------------------------- - * Purpose: - * Save the fmt chunk for the MMAPI library - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT SaveFmtChunk (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, S_WAVE_STATE *pWaveData, EAS_I32 fmtSize) -{ - EAS_RESULT result; - EAS_I32 pos; - EAS_I32 count; - - /* save current file position */ - if ((result = EAS_HWFilePos(pEASData->hwInstData, fileHandle, &pos)) != EAS_SUCCESS) - return result; - - /* allocate a chunk of memory */ - pWaveData->fmtChunk = EAS_HWMalloc(pEASData->hwInstData, fmtSize); - if (!pWaveData->fmtChunk) - return EAS_ERROR_MALLOC_FAILED; - - /* read the fmt chunk into memory */ - if ((result = EAS_HWReadFile(pEASData->hwInstData, fileHandle, pWaveData->fmtChunk, fmtSize, &count)) != EAS_SUCCESS) - return result; - if (count != fmtSize) - return EAS_ERROR_FILE_READ_FAILED; - - /* restore file position */ - return EAS_HWFileSeek(pEASData->hwInstData, fileHandle, pos); -} -#endif - diff --git a/arm-hybrid-22k/lib_src/eas_wavefile.h b/arm-hybrid-22k/lib_src/eas_wavefile.h deleted file mode 100644 index b8b76df..0000000 --- a/arm-hybrid-22k/lib_src/eas_wavefile.h +++ /dev/null @@ -1,63 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_wavefile.h - * - * Contents and purpose: - * Static data block for wave file parser. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 439 $ - * $Date: 2006-10-26 11:53:18 -0700 (Thu, 26 Oct 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_WAVEFILE_H -#define _EAS_WAVEFILE_H - -#include "eas_data.h" -#include "eas_pcm.h" - -/*---------------------------------------------------------------------------- - * - * S_WAVE_STATE - * - * This structure contains the WAVE file parser state information - *---------------------------------------------------------------------------- -*/ -typedef struct s_wave_state_tag -{ - EAS_FILE_HANDLE fileHandle; - EAS_PCM_HANDLE streamHandle; - S_METADATA_CB metadata; - EAS_U32 time; - EAS_I32 fileOffset; - EAS_I32 audioOffset; - EAS_I32 mediaLength; - EAS_U32 audioSize; - EAS_U32 flags; - EAS_I16 fileType; -#ifdef MMAPI_SUPPORT - EAS_VOID_PTR fmtChunk; -#endif - EAS_I32 infoChunkPos; - EAS_I32 infoChunkSize; -} S_WAVE_STATE; - -#endif - diff --git a/arm-hybrid-22k/lib_src/eas_wavefiledata.c b/arm-hybrid-22k/lib_src/eas_wavefiledata.c deleted file mode 100644 index 3742aa6..0000000 --- a/arm-hybrid-22k/lib_src/eas_wavefiledata.c +++ /dev/null @@ -1,33 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_wavefiledata.c - * - * Contents and purpose: - * Static data block for wave file parser. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 547 $ - * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $ - *---------------------------------------------------------------------------- -*/ - -#include "eas_wavefile.h" - -S_WAVE_STATE eas_WaveData; - diff --git a/arm-hybrid-22k/lib_src/eas_wt_IPC_frame.h b/arm-hybrid-22k/lib_src/eas_wt_IPC_frame.h deleted file mode 100644 index 376fd3a..0000000 --- a/arm-hybrid-22k/lib_src/eas_wt_IPC_frame.h +++ /dev/null @@ -1,82 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_wt_IPC_frame.h - * - * Contents and purpose: - * This module contains data definitions for the interprocessor - * communications framework for a split-architecture synthesizer. - * - * This sample version writes IPC data to a file that can be used - * as a test vector for the DSP simulator. For a real-time system - * the file I/O is replaced with an IPC protocol in the hardware. - * - * Synchronization with the DSP is accomplished at the API level, - * i.e. the host code should call EAS_Render when it is ready to - * buffer another block of data for transmission to the DSP. - * - * Copyright Sonic Network Inc. 2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 818 $ - * $Date: 2007-08-02 15:19:41 -0700 (Thu, 02 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_WT_IPC_FRAME_H -#define _EAS_WT_IPC_FRAME_H - -/*---------------------------------------------------------------------------- - * S_WT_FRAME - * - * This structure contains the common parameters that are updated - *for each frame of audio. - *---------------------------------------------------------------------------- -*/ -typedef struct s_wt_frame_tag -{ - EAS_I32 gainTarget; - EAS_I32 phaseIncrement; - -#if defined(_FILTER_ENABLED) - EAS_I32 k; - EAS_I32 b1; - EAS_I32 b2; -#endif -} S_WT_FRAME; - -/*---------------------------------------------------------------------------- - * S_WT_CONFIG - * - * This structure contains state data for the wavetable engine - *---------------------------------------------------------------------------- -*/ -typedef struct s_wt_config_tag -{ - EAS_U32 loopEnd; /* points to last PCM sample (not 1 beyond last) */ - EAS_U32 loopStart; /* points to first sample at start of loop */ - EAS_U32 phaseAccum; /* current sample, integer portion of phase */ - -#if (NUM_OUTPUT_CHANNELS == 2) - EAS_I16 gainLeft; /* left channel gain */ - EAS_I16 gainRight; /* right channel gain */ -#endif - - EAS_I16 gain; /* current voice gain */ -} S_WT_CONFIG; - -#endif - diff --git a/arm-hybrid-22k/lib_src/eas_wtengine.c b/arm-hybrid-22k/lib_src/eas_wtengine.c deleted file mode 100644 index 4215924..0000000 --- a/arm-hybrid-22k/lib_src/eas_wtengine.c +++ /dev/null @@ -1,661 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_wtengine.c - * - * Contents and purpose: - * This file contains the critical synthesizer components that need to - * be optimized for best performance. - * - * Copyright Sonic Network Inc. 2004-2005 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 844 $ - * $Date: 2007-08-23 14:33:32 -0700 (Thu, 23 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -/*------------------------------------ - * includes - *------------------------------------ -*/ -#include "eas_types.h" -#include "eas_math.h" -#include "eas_audioconst.h" -#include "eas_sndlib.h" -#include "eas_wtengine.h" -#include "eas_mixer.h" - -/*---------------------------------------------------------------------------- - * prototypes - *---------------------------------------------------------------------------- -*/ -extern void WT_NoiseGenerator (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame); -extern void WT_VoiceGain (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame); - -#if defined(_OPTIMIZED_MONO) -extern void WT_InterpolateMono (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame); -#else -extern void WT_InterpolateNoLoop (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame); -extern void WT_Interpolate (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame); -#endif - -#if defined(_FILTER_ENABLED) -extern void WT_VoiceFilter (S_FILTER_CONTROL*pFilter, S_WT_INT_FRAME *pWTIntFrame); -#endif - -#if defined(_OPTIMIZED_MONO) || !defined(NATIVE_EAS_KERNEL) -/*---------------------------------------------------------------------------- - * WT_VoiceGain - *---------------------------------------------------------------------------- - * Purpose: - * Output gain for individual voice - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pWTVoice) reserved for future use */ -void WT_VoiceGain (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame) -{ - EAS_I32 *pMixBuffer; - EAS_PCM *pInputBuffer; - EAS_I32 gain; - EAS_I32 gainIncrement; - EAS_I32 tmp0; - EAS_I32 tmp1; - EAS_I32 tmp2; - EAS_I32 numSamples; - -#if (NUM_OUTPUT_CHANNELS == 2) - EAS_I32 gainLeft, gainRight; -#endif - - /* initialize some local variables */ - numSamples = pWTIntFrame->numSamples; - pMixBuffer = pWTIntFrame->pMixBuffer; - pInputBuffer = pWTIntFrame->pAudioBuffer; - - /*lint -e{703} */ - gainIncrement = (pWTIntFrame->frame.gainTarget - pWTIntFrame->prevGain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS); - if (gainIncrement < 0) - gainIncrement++; - /*lint -e{703} */ - gain = pWTIntFrame->prevGain << 16; - -#if (NUM_OUTPUT_CHANNELS == 2) - gainLeft = pWTVoice->gainLeft; - gainRight = pWTVoice->gainRight; -#endif - - while (numSamples--) { - - /* incremental gain step to prevent zipper noise */ - tmp0 = *pInputBuffer++; - gain += gainIncrement; - /*lint -e{704} */ - tmp2 = gain >> 16; - - /* scale sample by gain */ - tmp2 *= tmp0; - - - /* stereo output */ -#if (NUM_OUTPUT_CHANNELS == 2) - /*lint -e{704} */ - tmp2 = tmp2 >> 14; - - /* get the current sample in the final mix buffer */ - tmp1 = *pMixBuffer; - - /* left channel */ - tmp0 = tmp2 * gainLeft; - /*lint -e{704} */ - tmp0 = tmp0 >> NUM_MIXER_GUARD_BITS; - tmp1 += tmp0; - *pMixBuffer++ = tmp1; - - /* get the current sample in the final mix buffer */ - tmp1 = *pMixBuffer; - - /* right channel */ - tmp0 = tmp2 * gainRight; - /*lint -e{704} */ - tmp0 = tmp0 >> NUM_MIXER_GUARD_BITS; - tmp1 += tmp0; - *pMixBuffer++ = tmp1; - - /* mono output */ -#else - - /* get the current sample in the final mix buffer */ - tmp1 = *pMixBuffer; - /*lint -e{704} */ - tmp2 = tmp2 >> (NUM_MIXER_GUARD_BITS - 1); - tmp1 += tmp2; - *pMixBuffer++ = tmp1; -#endif - - } -} -#endif - -#ifndef NATIVE_EAS_KERNEL -/*---------------------------------------------------------------------------- - * WT_Interpolate - *---------------------------------------------------------------------------- - * Purpose: - * Interpolation engine for wavetable synth - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -void WT_Interpolate (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame) -{ - EAS_PCM *pOutputBuffer; - EAS_I32 phaseInc; - EAS_I32 phaseFrac; - EAS_I32 acc0; - const EAS_SAMPLE *pSamples; - const EAS_SAMPLE *loopEnd; - EAS_I32 samp1; - EAS_I32 samp2; - EAS_I32 numSamples; - - /* initialize some local variables */ - numSamples = pWTIntFrame->numSamples; - pOutputBuffer = pWTIntFrame->pAudioBuffer; - - loopEnd = (const EAS_SAMPLE*) pWTVoice->loopEnd + 1; - pSamples = (const EAS_SAMPLE*) pWTVoice->phaseAccum; - /*lint -e{713} truncation is OK */ - phaseFrac = pWTVoice->phaseFrac; - phaseInc = pWTIntFrame->frame.phaseIncrement; - - /* fetch adjacent samples */ -#if defined(_8_BIT_SAMPLES) - /*lint -e{701} */ - samp1 = pSamples[0] << 8; - /*lint -e{701} */ - samp2 = pSamples[1] << 8; -#else - samp1 = pSamples[0]; - samp2 = pSamples[1]; -#endif - - while (numSamples--) { - - /* linear interpolation */ - acc0 = samp2 - samp1; - acc0 = acc0 * phaseFrac; - /*lint -e{704} */ - acc0 = samp1 + (acc0 >> NUM_PHASE_FRAC_BITS); - - /* save new output sample in buffer */ - /*lint -e{704} */ - *pOutputBuffer++ = (EAS_I16)(acc0 >> 2); - - /* increment phase */ - phaseFrac += phaseInc; - /*lint -e{704} */ - acc0 = phaseFrac >> NUM_PHASE_FRAC_BITS; - - /* next sample */ - if (acc0 > 0) { - - /* advance sample pointer */ - pSamples += acc0; - phaseFrac = (EAS_I32)((EAS_U32)phaseFrac & PHASE_FRAC_MASK); - - /* check for loop end */ - acc0 = (EAS_I32) (pSamples - loopEnd); - if (acc0 >= 0) - pSamples = (const EAS_SAMPLE*) pWTVoice->loopStart + acc0; - - /* fetch new samples */ -#if defined(_8_BIT_SAMPLES) - /*lint -e{701} */ - samp1 = pSamples[0] << 8; - /*lint -e{701} */ - samp2 = pSamples[1] << 8; -#else - samp1 = pSamples[0]; - samp2 = pSamples[1]; -#endif - } - } - - /* save pointer and phase */ - pWTVoice->phaseAccum = (EAS_U32) pSamples; - pWTVoice->phaseFrac = (EAS_U32) phaseFrac; -} -#endif - -#ifndef NATIVE_EAS_KERNEL -/*---------------------------------------------------------------------------- - * WT_InterpolateNoLoop - *---------------------------------------------------------------------------- - * Purpose: - * Interpolation engine for wavetable synth - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -void WT_InterpolateNoLoop (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame) -{ - EAS_PCM *pOutputBuffer; - EAS_I32 phaseInc; - EAS_I32 phaseFrac; - EAS_I32 acc0; - const EAS_SAMPLE *pSamples; - EAS_I32 samp1; - EAS_I32 samp2; - EAS_I32 numSamples; - - /* initialize some local variables */ - numSamples = pWTIntFrame->numSamples; - pOutputBuffer = pWTIntFrame->pAudioBuffer; - - phaseInc = pWTIntFrame->frame.phaseIncrement; - pSamples = (const EAS_SAMPLE*) pWTVoice->phaseAccum; - phaseFrac = (EAS_I32)pWTVoice->phaseFrac; - - /* fetch adjacent samples */ -#if defined(_8_BIT_SAMPLES) - /*lint -e{701} */ - samp1 = pSamples[0] << 8; - /*lint -e{701} */ - samp2 = pSamples[1] << 8; -#else - samp1 = pSamples[0]; - samp2 = pSamples[1]; -#endif - - while (numSamples--) { - - - /* linear interpolation */ - acc0 = samp2 - samp1; - acc0 = acc0 * phaseFrac; - /*lint -e{704} */ - acc0 = samp1 + (acc0 >> NUM_PHASE_FRAC_BITS); - - /* save new output sample in buffer */ - /*lint -e{704} */ - *pOutputBuffer++ = (EAS_I16)(acc0 >> 2); - - /* increment phase */ - phaseFrac += phaseInc; - /*lint -e{704} */ - acc0 = phaseFrac >> NUM_PHASE_FRAC_BITS; - - /* next sample */ - if (acc0 > 0) { - - /* advance sample pointer */ - pSamples += acc0; - phaseFrac = (EAS_I32)((EAS_U32)phaseFrac & PHASE_FRAC_MASK); - - /* fetch new samples */ -#if defined(_8_BIT_SAMPLES) - /*lint -e{701} */ - samp1 = pSamples[0] << 8; - /*lint -e{701} */ - samp2 = pSamples[1] << 8; -#else - samp1 = pSamples[0]; - samp2 = pSamples[1]; -#endif - } - } - - /* save pointer and phase */ - pWTVoice->phaseAccum = (EAS_U32) pSamples; - pWTVoice->phaseFrac = (EAS_U32) phaseFrac; -} -#endif - -#if defined(_FILTER_ENABLED) && !defined(NATIVE_EAS_KERNEL) -/*---------------------------------------------------------------------------- - * WT_VoiceFilter - *---------------------------------------------------------------------------- - * Purpose: - * Implements a 2-pole filter - * - * Inputs: - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -void WT_VoiceFilter (S_FILTER_CONTROL *pFilter, S_WT_INT_FRAME *pWTIntFrame) -{ - EAS_PCM *pAudioBuffer; - EAS_I32 k; - EAS_I32 b1; - EAS_I32 b2; - EAS_I32 z1; - EAS_I32 z2; - EAS_I32 acc0; - EAS_I32 acc1; - EAS_I32 numSamples; - - /* initialize some local variables */ - numSamples = pWTIntFrame->numSamples; - pAudioBuffer = pWTIntFrame->pAudioBuffer; - - z1 = pFilter->z1; - z2 = pFilter->z2; - b1 = -pWTIntFrame->frame.b1; - - /*lint -e{702} */ - b2 = -pWTIntFrame->frame.b2 >> 1; - - /*lint -e{702} */ - k = pWTIntFrame->frame.k >> 1; - - while (numSamples--) - { - - /* do filter calculations */ - acc0 = *pAudioBuffer; - acc1 = z1 * b1; - acc1 += z2 * b2; - acc0 = acc1 + k * acc0; - z2 = z1; - - /*lint -e{702} */ - z1 = acc0 >> 14; - *pAudioBuffer++ = (EAS_I16) z1; - } - - /* save delay values */ - pFilter->z1 = (EAS_I16) z1; - pFilter->z2 = (EAS_I16) z2; -} -#endif - -/*---------------------------------------------------------------------------- - * WT_NoiseGenerator - *---------------------------------------------------------------------------- - * Purpose: - * Generate pseudo-white noise using PRNG and interpolation engine - * - * Inputs: - * - * Outputs: - * - * Notes: - * This output is scaled -12dB to prevent saturation in the filter. For a - * high quality synthesizer, the output can be set to full scale, however - * if the filter is used, it can overflow with certain coefficients. In this - * case, either a saturation operation should take in the filter before - * scaling back to 16 bits or the signal path should be increased to 18 bits - * or more. - *---------------------------------------------------------------------------- -*/ - void WT_NoiseGenerator (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame) - { - EAS_PCM *pOutputBuffer; - EAS_I32 phaseInc; - EAS_I32 tmp0; - EAS_I32 tmp1; - EAS_I32 nInterpolatedSample; - EAS_I32 numSamples; - - /* initialize some local variables */ - numSamples = pWTIntFrame->numSamples; - pOutputBuffer = pWTIntFrame->pAudioBuffer; - phaseInc = pWTIntFrame->frame.phaseIncrement; - - /* get last two samples generated */ - /*lint -e{704} */ - tmp0 = (EAS_I32) (pWTVoice->phaseAccum) >> 18; - /*lint -e{704} */ - tmp1 = (EAS_I32) (pWTVoice->loopEnd) >> 18; - - /* generate a buffer of noise */ - while (numSamples--) { - nInterpolatedSample = MULT_AUDIO_COEF( tmp0, (PHASE_ONE - pWTVoice->phaseFrac)); - nInterpolatedSample += MULT_AUDIO_COEF( tmp1, pWTVoice->phaseFrac); - *pOutputBuffer++ = (EAS_PCM) nInterpolatedSample; - - /* update PRNG */ - pWTVoice->phaseFrac += (EAS_U32) phaseInc; - if (GET_PHASE_INT_PART(pWTVoice->phaseFrac)) { - tmp0 = tmp1; - pWTVoice->phaseAccum = pWTVoice->loopEnd; - pWTVoice->loopEnd = (5 * pWTVoice->loopEnd + 1); - tmp1 = (EAS_I32) (pWTVoice->loopEnd) >> 18; - pWTVoice->phaseFrac = GET_PHASE_FRAC_PART(pWTVoice->phaseFrac); - } - - } -} - -#ifndef _OPTIMIZED_MONO -/*---------------------------------------------------------------------------- - * WT_ProcessVoice - *---------------------------------------------------------------------------- - * Purpose: - * This routine does the block processing for one voice. It is isolated - * from the main synth code to allow for various implementation-specific - * optimizations. It calls the interpolator, filter, and gain routines - * appropriate for a particular configuration. - * - * Inputs: - * - * Outputs: - * - * Notes: - *---------------------------------------------------------------------------- -*/ -void WT_ProcessVoice (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame) -{ - - /* use noise generator */ - if (pWTVoice->loopStart == WT_NOISE_GENERATOR) - WT_NoiseGenerator(pWTVoice, pWTIntFrame); - - /* generate interpolated samples for looped waves */ - else if (pWTVoice->loopStart != pWTVoice->loopEnd) - WT_Interpolate(pWTVoice, pWTIntFrame); - - /* generate interpolated samples for unlooped waves */ - else - { - WT_InterpolateNoLoop(pWTVoice, pWTIntFrame); - } - -#ifdef _FILTER_ENABLED - if (pWTIntFrame->frame.k != 0) - WT_VoiceFilter(&pWTVoice->filter, pWTIntFrame); -#endif - -//2 TEST NEW MIXER FUNCTION -#ifdef UNIFIED_MIXER - { - EAS_I32 gainLeft, gainIncLeft; - -#if (NUM_OUTPUT_CHANNELS == 2) - EAS_I32 gainRight, gainIncRight; -#endif - - gainLeft = (pWTIntFrame->prevGain * pWTVoice->gainLeft) << 1; - gainIncLeft = (((pWTIntFrame->frame.gainTarget * pWTVoice->gainLeft) << 1) - gainLeft) >> SYNTH_UPDATE_PERIOD_IN_BITS; - -#if (NUM_OUTPUT_CHANNELS == 2) - gainRight = (pWTIntFrame->prevGain * pWTVoice->gainRight) << 1; - gainIncRight = (((pWTIntFrame->frame.gainTarget * pWTVoice->gainRight) << 1) - gainRight) >> SYNTH_UPDATE_PERIOD_IN_BITS; - EAS_MixStream( - pWTIntFrame->pAudioBuffer, - pWTIntFrame->pMixBuffer, - pWTIntFrame->numSamples, - gainLeft, - gainRight, - gainIncLeft, - gainIncRight, - MIX_FLAGS_STEREO_OUTPUT); - -#else - EAS_MixStream( - pWTIntFrame->pAudioBuffer, - pWTIntFrame->pMixBuffer, - pWTIntFrame->numSamples, - gainLeft, - 0, - gainIncLeft, - 0, - 0); -#endif - } - -#else - /* apply gain, and left and right gain */ - WT_VoiceGain(pWTVoice, pWTIntFrame); -#endif -} -#endif - -#if defined(_OPTIMIZED_MONO) && !defined(NATIVE_EAS_KERNEL) -/*---------------------------------------------------------------------------- - * WT_InterpolateMono - *---------------------------------------------------------------------------- - * Purpose: - * A C version of the sample interpolation + gain routine, optimized for mono. - * It's not pretty, but it matches the assembly code exactly. - * - * Inputs: - * - * Outputs: - * - * Notes: - *---------------------------------------------------------------------------- -*/ -void WT_InterpolateMono (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame) -{ - EAS_I32 *pMixBuffer; - const EAS_I8 *pLoopEnd; - const EAS_I8 *pCurrentPhaseInt; - EAS_I32 numSamples; - EAS_I32 gain; - EAS_I32 gainIncrement; - EAS_I32 currentPhaseFrac; - EAS_I32 phaseInc; - EAS_I32 tmp0; - EAS_I32 tmp1; - EAS_I32 tmp2; - EAS_I8 *pLoopStart; - - numSamples = pWTIntFrame->numSamples; - pMixBuffer = pWTIntFrame->pMixBuffer; - - /* calculate gain increment */ - gainIncrement = (pWTIntFrame->gainTarget - pWTIntFrame->prevGain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS); - if (gainIncrement < 0) - gainIncrement++; - gain = pWTIntFrame->prevGain << 16; - - pCurrentPhaseInt = pWTVoice->pPhaseAccum; - currentPhaseFrac = pWTVoice->phaseFrac; - phaseInc = pWTIntFrame->phaseIncrement; - - pLoopStart = pWTVoice->pLoopStart; - pLoopEnd = pWTVoice->pLoopEnd + 1; - -InterpolationLoop: - tmp0 = (EAS_I32)(pCurrentPhaseInt - pLoopEnd); - if (tmp0 >= 0) - pCurrentPhaseInt = pLoopStart + tmp0; - - tmp0 = *pCurrentPhaseInt; - tmp1 = *(pCurrentPhaseInt + 1); - - tmp2 = phaseInc + currentPhaseFrac; - - tmp1 = tmp1 - tmp0; - tmp1 = tmp1 * currentPhaseFrac; - - tmp1 = tmp0 + (tmp1 >> NUM_EG1_FRAC_BITS); - - pCurrentPhaseInt += (tmp2 >> NUM_PHASE_FRAC_BITS); - currentPhaseFrac = tmp2 & PHASE_FRAC_MASK; - - gain += gainIncrement; - tmp2 = (gain >> SYNTH_UPDATE_PERIOD_IN_BITS); - - tmp0 = *pMixBuffer; - tmp2 = tmp1 * tmp2; - tmp2 = (tmp2 >> 9); - tmp0 = tmp2 + tmp0; - *pMixBuffer++ = tmp0; - - numSamples--; - if (numSamples) - goto InterpolationLoop; - - pWTVoice->pPhaseAccum = pCurrentPhaseInt; - pWTVoice->phaseFrac = currentPhaseFrac; - /*lint -e{702} */ - pWTVoice->gain = (EAS_I16)(gain >> SYNTH_UPDATE_PERIOD_IN_BITS); -} -#endif - -#ifdef _OPTIMIZED_MONO -/*---------------------------------------------------------------------------- - * WT_ProcessVoice - *---------------------------------------------------------------------------- - * Purpose: - * This routine does the block processing for one voice. It is isolated - * from the main synth code to allow for various implementation-specific - * optimizations. It calls the interpolator, filter, and gain routines - * appropriate for a particular configuration. - * - * Inputs: - * - * Outputs: - * - * Notes: - * This special version works handles an optimized mono-only signal - * without filters - *---------------------------------------------------------------------------- -*/ -void WT_ProcessVoice (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame) -{ - - /* use noise generator */ - if (pWTVoice->loopStart== WT_NOISE_GENERATOR) - { - WT_NoiseGenerator(pWTVoice, pWTIntFrame); - WT_VoiceGain(pWTVoice, pWTIntFrame); - } - - /* or generate interpolated samples */ - else - { - WT_InterpolateMono(pWTVoice, pWTIntFrame); - } -} -#endif - diff --git a/arm-hybrid-22k/lib_src/eas_wtengine.h b/arm-hybrid-22k/lib_src/eas_wtengine.h deleted file mode 100644 index 6401322..0000000 --- a/arm-hybrid-22k/lib_src/eas_wtengine.h +++ /dev/null @@ -1,171 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_wtengine.h - * - * Contents and purpose: - * This file defines the interface for wavetable synthesis engine - * - * Copyright Sonic Network Inc. 2004 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 818 $ - * $Date: 2007-08-02 15:19:41 -0700 (Thu, 02 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_WTENGINE_H -#define _EAS_WTENGINE_H - -/* option sanity check */ -#if defined(_OPTIMIZED_MONO) && defined(_FILTER_ENABLED) -#error "Incompatible build settings: _OPTIMIZED_MONO cannot be used with _FILTER_ENABLED" -#endif - -#if defined(_OPTIMIZED_MONO) && (NUM_OUTPUT_CHANNELS != 1) -#error "Incompatible build settings: _OPTIMIZED_MONO can only be used with NUM_OUTPUT_CHANNELS = 1" -#endif - -#include "eas_wt_IPC_frame.h" - -/*---------------------------------------------------------------------------- - * defines - *---------------------------------------------------------------------------- -*/ -#define WT_NOISE_GENERATOR 0xffffffff - -/*---------------------------------------------------------------------------- - * typedefs - *---------------------------------------------------------------------------- -*/ - -/*---------------------------------------------------------------------------- - * S_WT_INT_FRAME - * - * This structure includes S_WT_FRAME plus the bus mixing - * parameters for the internal voices. - *---------------------------------------------------------------------------- -*/ -typedef struct s_wt_int_frame_tag -{ - S_WT_FRAME frame; - EAS_PCM *pAudioBuffer; - EAS_I32 *pMixBuffer; - EAS_I32 numSamples; - EAS_I32 prevGain; -} S_WT_INT_FRAME; - -#if defined(_FILTER_ENABLED) -/*---------------------------------------------------------------------------- - * S_FILTER_CONTROL data structure - *---------------------------------------------------------------------------- -*/ -typedef struct s_filter_control_tag -{ - EAS_I16 z1; /* 1 sample delay state variable */ - EAS_I16 z2; /* 2 sample delay state variable */ -} S_FILTER_CONTROL; -#endif - -/*------------------------------------ - * S_LFO_CONTROL data structure - *------------------------------------ -*/ -typedef struct s_lfo_control_tag -{ - EAS_I16 lfoValue; /* LFO current output value */ - EAS_I16 lfoPhase; /* LFO current phase */ -} S_LFO_CONTROL; - -/* bit definitions for S_WT_VOICE:flags */ -#define WT_FLAGS_ADPCM_NIBBLE 1 /* high/low nibble flag */ -#define WT_FLAGS_ADPCM_READY 2 /* first 2 samples are decoded */ -#define WT_FLAGS_USE_ADPCM 4 /* sample is ADPCM encoded */ - -/* eg1State and eg2State */ -typedef enum { - eEnvelopeStateInit = 0, - eEnvelopeStateDelay, - eEnvelopeStateAttack, - eEnvelopeStateHold, - eEnvelopeStateDecay, - eEnvelopeStateSustain, - eEnvelopeStateRelease, - eEnvelopeStateMuting, - eEnvelopeStateMuted, - eEnvelopeStateInvalid /* should never be in this state! */ -} E_ENVELOPE_STATE; - -#define DEFAULT_EG1_STATE eEnvelopeStateAttack -#define DEFAULT_EG1_VALUE 0 -#define DEFAULT_EG1_INCREMENT 0 -#define DEFAULT_EG2_STATE eEnvelopeStateAttack -#define DEFAULT_EG2_VALUE 0 -#define DEFAULT_EG2_INCREMENT 0 - -/*---------------------------------------------------------------------------- - * S_WT_VOICE - * - * This structure contains state data for the wavetable engine - *---------------------------------------------------------------------------- -*/ -typedef struct s_wt_voice_tag -{ - EAS_U32 loopEnd; /* points to last PCM sample (not 1 beyond last) */ - EAS_U32 loopStart; /* points to first sample at start of loop */ - EAS_U32 phaseAccum; /* current sample, integer portion of phase */ - EAS_U32 phaseFrac; /* fractional portion of phase */ - -#if (NUM_OUTPUT_CHANNELS == 2) - EAS_I16 gainLeft; /* current gain, left ch */ - EAS_I16 gainRight; /* current gain, right ch */ -#endif - -#if defined(_FILTER_ENABLED) - S_FILTER_CONTROL filter; /* low pass filter */ -#endif - - S_LFO_CONTROL modLFO; /* modulator LFO */ - -#ifdef DLS_SYNTHESIZER - S_LFO_CONTROL vibLFO; /* vibrato LFO */ -#endif - - /* envelope control */ - EAS_I16 eg1Value; - EAS_I16 eg2Value; - EAS_I16 eg1Increment; - EAS_I16 eg2Increment; - EAS_U8 eg1State; - EAS_U8 eg2State; - - EAS_U16 artIndex; /* index to articulation params */ - -} S_WT_VOICE; - -/*---------------------------------------------------------------------------- - * prototypes - *---------------------------------------------------------------------------- -*/ -EAS_BOOL WT_CheckSampleEnd (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame, EAS_BOOL update); -void WT_ProcessVoice (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame); - -#ifdef EAS_SPLIT_WT_SYNTH -void WTE_ConfigVoice (EAS_I32 voiceNum, S_WT_CONFIG *pWTConfig, EAS_FRAME_BUFFER_HANDLE pFrameBuffer); -void WTE_ProcessVoice (EAS_I32 voiceNum, S_WT_FRAME *pWTParams, EAS_FRAME_BUFFER_HANDLE pFrameBuffer); -#endif - -#endif diff --git a/arm-hybrid-22k/lib_src/eas_wtsynth.c b/arm-hybrid-22k/lib_src/eas_wtsynth.c deleted file mode 100644 index 8098e09..0000000 --- a/arm-hybrid-22k/lib_src/eas_wtsynth.c +++ /dev/null @@ -1,1251 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_wtsynth.c - * - * Contents and purpose: - * Implements the synthesizer functions. - * - * Copyright Sonic Network Inc. 2004 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 795 $ - * $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $ - *---------------------------------------------------------------------------- -*/ - -// includes -#include "eas_data.h" -#include "eas_report.h" -#include "eas_host.h" -#include "eas_math.h" -#include "eas_synth_protos.h" -#include "eas_wtsynth.h" -#include "eas_pan.h" - -#ifdef DLS_SYNTHESIZER -#include "eas_dlssynth.h" -#endif - -#ifdef _METRICS_ENABLED -#include "eas_perf.h" -#endif - -/* local prototypes */ -static EAS_RESULT WT_Initialize(S_VOICE_MGR *pVoiceMgr); -static void WT_ReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum); -static void WT_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum); -static void WT_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum); -static EAS_RESULT WT_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex); -static EAS_BOOL WT_UpdateVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32 numSamples); -static void WT_UpdateChannel (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel); -static EAS_I32 WT_UpdatePhaseInc (S_WT_VOICE *pWTVoice, const S_ARTICULATION *pArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 pitchCents); -static EAS_I32 WT_UpdateGain (S_SYNTH_VOICE *pVoice, S_WT_VOICE *pWTVoice, const S_ARTICULATION *pArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 gain); -static void WT_UpdateEG1 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv); -static void WT_UpdateEG2 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv); - -#ifdef EAS_SPLIT_WT_SYNTH -extern EAS_BOOL WTE_StartFrame (EAS_FRAME_BUFFER_HANDLE pFrameBuffer); -extern EAS_BOOL WTE_EndFrame (EAS_FRAME_BUFFER_HANDLE pFrameBuffer, EAS_I32 *pMixBuffer, EAS_I16 masterGain); -#endif - -#ifdef _FILTER_ENABLED -static void WT_UpdateFilter (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pIntFrame, const S_ARTICULATION *pArt); -#endif - -#ifdef _STATS -extern double statsPhaseIncrement; -extern double statsMaxPhaseIncrement; -extern long statsPhaseSampleCount; -extern double statsSampleSize; -extern long statsSampleCount; -#endif - -/*---------------------------------------------------------------------------- - * Synthesizer interface - *---------------------------------------------------------------------------- -*/ - -const S_SYNTH_INTERFACE wtSynth = -{ - WT_Initialize, - WT_StartVoice, - WT_UpdateVoice, - WT_ReleaseVoice, - WT_MuteVoice, - WT_SustainPedal, - WT_UpdateChannel -}; - -#ifdef EAS_SPLIT_WT_SYNTH -const S_FRAME_INTERFACE wtFrameInterface = -{ - WTE_StartFrame, - WTE_EndFrame -}; -#endif - -/*---------------------------------------------------------------------------- - * WT_Initialize() - *---------------------------------------------------------------------------- - * Purpose: - * - * Inputs: - * pVoice - pointer to voice to initialize - * - * Outputs: - * - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT WT_Initialize (S_VOICE_MGR *pVoiceMgr) -{ - EAS_INT i; - - for (i = 0; i < NUM_WT_VOICES; i++) - { - - pVoiceMgr->wtVoices[i].artIndex = DEFAULT_ARTICULATION_INDEX; - - pVoiceMgr->wtVoices[i].eg1State = DEFAULT_EG1_STATE; - pVoiceMgr->wtVoices[i].eg1Value = DEFAULT_EG1_VALUE; - pVoiceMgr->wtVoices[i].eg1Increment = DEFAULT_EG1_INCREMENT; - - pVoiceMgr->wtVoices[i].eg2State = DEFAULT_EG2_STATE; - pVoiceMgr->wtVoices[i].eg2Value = DEFAULT_EG2_VALUE; - pVoiceMgr->wtVoices[i].eg2Increment = DEFAULT_EG2_INCREMENT; - - /* left and right gain values are needed only if stereo output */ -#if (NUM_OUTPUT_CHANNELS == 2) - pVoiceMgr->wtVoices[i].gainLeft = DEFAULT_VOICE_GAIN; - pVoiceMgr->wtVoices[i].gainRight = DEFAULT_VOICE_GAIN; -#endif - - pVoiceMgr->wtVoices[i].phaseFrac = DEFAULT_PHASE_FRAC; - pVoiceMgr->wtVoices[i].phaseAccum = DEFAULT_PHASE_INT; - -#ifdef _FILTER_ENABLED - pVoiceMgr->wtVoices[i].filter.z1 = DEFAULT_FILTER_ZERO; - pVoiceMgr->wtVoices[i].filter.z2 = DEFAULT_FILTER_ZERO; -#endif - } - - return EAS_TRUE; -} - -/*---------------------------------------------------------------------------- - * WT_ReleaseVoice() - *---------------------------------------------------------------------------- - * Purpose: - * The selected voice is being released. - * - * Inputs: - * pEASData - pointer to S_EAS_DATA - * pVoice - pointer to voice to release - * - * Outputs: - * None - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pVoice) used in some implementations */ -static void WT_ReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum) -{ - S_WT_VOICE *pWTVoice; - const S_ARTICULATION *pArticulation; - -#ifdef DLS_SYNTHESIZER - if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH) - { - DLS_ReleaseVoice(pVoiceMgr, pSynth, pVoice, voiceNum); - return; - } -#endif - - pWTVoice = &pVoiceMgr->wtVoices[voiceNum]; - pArticulation = &pSynth->pEAS->pArticulations[pWTVoice->artIndex]; - - /* release EG1 */ - pWTVoice->eg1State = eEnvelopeStateRelease; - pWTVoice->eg1Increment = pArticulation->eg1.releaseTime; - - /* - The spec says we should release EG2, but doing so with the current - voicing is causing clicks. This fix will need to be coordinated with - a new sound library release - */ - - /* release EG2 */ - pWTVoice->eg2State = eEnvelopeStateRelease; - pWTVoice->eg2Increment = pArticulation->eg2.releaseTime; -} - -/*---------------------------------------------------------------------------- - * WT_MuteVoice() - *---------------------------------------------------------------------------- - * Purpose: - * The selected voice is being muted. - * - * Inputs: - * pVoice - pointer to voice to release - * - * Outputs: - * None - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pSynth) used in some implementations */ -static void WT_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum) -{ - -#ifdef DLS_SYNTHESIZER - if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH) - { - DLS_MuteVoice(pVoiceMgr, pSynth, pVoice, voiceNum); - return; - } -#endif - - /* clear deferred action flags */ - pVoice->voiceFlags &= - ~(VOICE_FLAG_DEFER_MIDI_NOTE_OFF | - VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF | - VOICE_FLAG_DEFER_MUTE); - - /* set the envelope state */ - pVoiceMgr->wtVoices[voiceNum].eg1State = eEnvelopeStateMuted; - pVoiceMgr->wtVoices[voiceNum].eg2State = eEnvelopeStateMuted; -} - -/*---------------------------------------------------------------------------- - * WT_SustainPedal() - *---------------------------------------------------------------------------- - * Purpose: - * The selected voice is held due to sustain pedal - * - * Inputs: - * pVoice - pointer to voice to sustain - * - * Outputs: - * None - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pChannel) used in some implementations */ -static void WT_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum) -{ - S_WT_VOICE *pWTVoice; - -#ifdef DLS_SYNTHESIZER - if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH) - { - DLS_SustainPedal(pVoiceMgr, pSynth, pVoice, pChannel, voiceNum); - return; - } -#endif - - /* don't catch the voice if below the sustain level */ - pWTVoice = &pVoiceMgr->wtVoices[voiceNum]; - if (pWTVoice->eg1Value < pSynth->pEAS->pArticulations[pWTVoice->artIndex].eg1.sustainLevel) - return; - - /* sustain flag is set, damper pedal is on */ - /* defer releasing this note until the damper pedal is off */ - pWTVoice->eg1State = eEnvelopeStateDecay; - pVoice->voiceState = eVoiceStatePlay; - - /* - because sustain pedal is on, this voice - should defer releasing its note - */ - pVoice->voiceFlags |= VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF; - -#ifdef _DEBUG_SYNTH - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_SustainPedal: defer note off because sustain pedal is on\n"); */ } -#endif -} - -/*---------------------------------------------------------------------------- - * WT_StartVoice() - *---------------------------------------------------------------------------- - * Purpose: - * Assign the region for the given instrument using the midi key number - * and the RPN2 (coarse tuning) value. By using RPN2 as part of the - * region selection process, we reduce the amount a given sample has - * to be transposed by selecting the closest recorded root instead. - * - * This routine is the second half of SynthAssignRegion(). - * If the region was successfully found by SynthFindRegionIndex(), - * then assign the region's parameters to the voice. - * - * Setup and initialize the following voice parameters: - * m_nRegionIndex - * - * Inputs: - * pVoice - ptr to the voice we have assigned for this channel - * nRegionIndex - index of the region - * pEASData - pointer to overall EAS data structure - * - * Outputs: - * success - could find and assign the region for this voice's note otherwise - * failure - could not find nor assign the region for this voice's note - * - * Side Effects: - * psSynthObject->m_sVoice[].m_nRegionIndex is assigned - * psSynthObject->m_sVoice[] parameters are assigned - *---------------------------------------------------------------------------- -*/ -static EAS_RESULT WT_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex) -{ - S_WT_VOICE *pWTVoice; - const S_WT_REGION *pRegion; - const S_ARTICULATION *pArt; - S_SYNTH_CHANNEL *pChannel; - -#if (NUM_OUTPUT_CHANNELS == 2) - EAS_INT pan; -#endif - -#ifdef EAS_SPLIT_WT_SYNTH - S_WT_CONFIG wtConfig; -#endif - - /* no samples have been synthesized for this note yet */ - pVoice->regionIndex = regionIndex; - pVoice->voiceFlags = VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET; - - /* get the articulation index for this region */ - pWTVoice = &pVoiceMgr->wtVoices[voiceNum]; - pChannel = &pSynth->channels[pVoice->channel & 15]; - - /* update static channel parameters */ - if (pChannel->channelFlags & CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS) - WT_UpdateChannel(pVoiceMgr, pSynth, pVoice->channel & 15); - -#ifdef DLS_SYNTHESIZER - if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH) - return DLS_StartVoice(pVoiceMgr, pSynth, pVoice, voiceNum, regionIndex); -#endif - - pRegion = &(pSynth->pEAS->pWTRegions[regionIndex]); - pWTVoice->artIndex = pRegion->artIndex; - -#ifdef _DEBUG_SYNTH - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_StartVoice: Voice %ld; Region %d\n", (EAS_I32) (pVoice - pVoiceMgr->voices), regionIndex); */ } -#endif - - pArt = &pSynth->pEAS->pArticulations[pWTVoice->artIndex]; - - /* MIDI note on puts this voice into attack state */ - pWTVoice->eg1State = eEnvelopeStateAttack; - pWTVoice->eg1Value = 0; - pWTVoice->eg1Increment = pArt->eg1.attackTime; - pWTVoice->eg2State = eEnvelopeStateAttack; - pWTVoice->eg2Value = 0; - pWTVoice->eg2Increment = pArt->eg2.attackTime; - - /* init the LFO */ - pWTVoice->modLFO.lfoValue = 0; - pWTVoice->modLFO.lfoPhase = -pArt->lfoDelay; - - pVoice->gain = 0; - -#if (NUM_OUTPUT_CHANNELS == 2) - /* - Get the Midi CC10 pan value for this voice's channel - convert the pan value to an "angle" representation suitable for - our sin, cos calculator. This representation is NOT necessarily the same - as the transform in the GM manuals because of our sin, cos calculator. - "angle" = (CC10 - 64)/128 - */ - pan = (EAS_INT) pSynth->channels[pVoice->channel & 15].pan - 64; - pan += pArt->pan; - EAS_CalcPanControl(pan, &pWTVoice->gainLeft, &pWTVoice->gainRight); -#endif - -#ifdef _FILTER_ENABLED - /* clear out the filter states */ - pWTVoice->filter.z1 = 0; - pWTVoice->filter.z2 = 0; -#endif - - /* if this wave is to be generated using noise generator */ - if (pRegion->region.keyGroupAndFlags & REGION_FLAG_USE_WAVE_GENERATOR) - { - pWTVoice->phaseAccum = 4574296; - pWTVoice->loopStart = WT_NOISE_GENERATOR; - pWTVoice->loopEnd = 4574295; - } - - /* normal sample */ - else - { - -#ifdef EAS_SPLIT_WT_SYNTH - if (voiceNum < NUM_PRIMARY_VOICES) - pWTVoice->phaseAccum = (EAS_U32) pSynth->pEAS->pSamples + pSynth->pEAS->pSampleOffsets[pRegion->waveIndex]; - else - pWTVoice->phaseAccum = pSynth->pEAS->pSampleOffsets[pRegion->waveIndex]; -#else - pWTVoice->phaseAccum = (EAS_U32) pSynth->pEAS->pSamples + pSynth->pEAS->pSampleOffsets[pRegion->waveIndex]; -#endif - - if (pRegion->region.keyGroupAndFlags & REGION_FLAG_IS_LOOPED) - { - pWTVoice->loopStart = pWTVoice->phaseAccum + pRegion->loopStart; - pWTVoice->loopEnd = pWTVoice->phaseAccum + pRegion->loopEnd - 1; - } - else - pWTVoice->loopStart = pWTVoice->loopEnd = pWTVoice->phaseAccum + pSynth->pEAS->pSampleLen[pRegion->waveIndex] - 1; - } - -#ifdef EAS_SPLIT_WT_SYNTH - /* configure off-chip voices */ - if (voiceNum >= NUM_PRIMARY_VOICES) - { - wtConfig.phaseAccum = pWTVoice->phaseAccum; - wtConfig.loopStart = pWTVoice->loopStart; - wtConfig.loopEnd = pWTVoice->loopEnd; - wtConfig.gain = pVoice->gain; - -#if (NUM_OUTPUT_CHANNELS == 2) - wtConfig.gainLeft = pWTVoice->gainLeft; - wtConfig.gainRight = pWTVoice->gainRight; -#endif - - WTE_ConfigVoice(voiceNum - NUM_PRIMARY_VOICES, &wtConfig, pVoiceMgr->pFrameBuffer); - } -#endif - - return EAS_SUCCESS; -} - -/*---------------------------------------------------------------------------- - * WT_CheckSampleEnd - *---------------------------------------------------------------------------- - * Purpose: - * Check for end of sample and calculate number of samples to synthesize - * - * Inputs: - * - * Outputs: - * - * Notes: - * - *---------------------------------------------------------------------------- -*/ -EAS_BOOL WT_CheckSampleEnd (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame, EAS_BOOL update) -{ - EAS_U32 endPhaseAccum; - EAS_U32 endPhaseFrac; - EAS_I32 numSamples; - EAS_BOOL done = EAS_FALSE; - - /* check to see if we hit the end of the waveform this time */ - /*lint -e{703} use shift for performance */ - endPhaseFrac = pWTVoice->phaseFrac + (pWTIntFrame->frame.phaseIncrement << SYNTH_UPDATE_PERIOD_IN_BITS); - endPhaseAccum = pWTVoice->phaseAccum + GET_PHASE_INT_PART(endPhaseFrac); - if (endPhaseAccum >= pWTVoice->loopEnd) - { - /* calculate how far current ptr is from end */ - numSamples = (EAS_I32) (pWTVoice->loopEnd - pWTVoice->phaseAccum); - - /* now account for the fractional portion */ - /*lint -e{703} use shift for performance */ - numSamples = (EAS_I32) ((numSamples << NUM_PHASE_FRAC_BITS) - pWTVoice->phaseFrac); - pWTIntFrame->numSamples = 1 + (numSamples / pWTIntFrame->frame.phaseIncrement); - - /* sound will be done this frame */ - done = EAS_TRUE; - } - - /* update data for off-chip synth */ - if (update) - { - pWTVoice->phaseFrac = endPhaseFrac; - pWTVoice->phaseAccum = endPhaseAccum; - } - - return done; -} - -/*---------------------------------------------------------------------------- - * WT_UpdateVoice() - *---------------------------------------------------------------------------- - * Purpose: - * Synthesize a block of samples for the given voice. - * Use linear interpolation. - * - * Inputs: - * pEASData - pointer to overall EAS data structure - * - * Outputs: - * number of samples actually written to buffer - * - * Side Effects: - * - samples are added to the presently free buffer - * - *---------------------------------------------------------------------------- -*/ -static EAS_BOOL WT_UpdateVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32 numSamples) -{ - S_WT_VOICE *pWTVoice; - S_WT_INT_FRAME intFrame; - S_SYNTH_CHANNEL *pChannel; - const S_WT_REGION *pWTRegion; - const S_ARTICULATION *pArt; - EAS_I32 temp; - EAS_BOOL done; - -#ifdef DLS_SYNTHESIZER - if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH) - return DLS_UpdateVoice(pVoiceMgr, pSynth, pVoice, voiceNum, pMixBuffer, numSamples); -#endif - - /* establish pointers to critical data */ - pWTVoice = &pVoiceMgr->wtVoices[voiceNum]; - pWTRegion = &pSynth->pEAS->pWTRegions[pVoice->regionIndex & REGION_INDEX_MASK]; - pArt = &pSynth->pEAS->pArticulations[pWTVoice->artIndex]; - pChannel = &pSynth->channels[pVoice->channel & 15]; - intFrame.prevGain = pVoice->gain; - - /* update the envelopes */ - WT_UpdateEG1(pWTVoice, &pArt->eg1); - WT_UpdateEG2(pWTVoice, &pArt->eg2); - - /* update the LFO */ - WT_UpdateLFO(&pWTVoice->modLFO, pArt->lfoFreq); - -#ifdef _FILTER_ENABLED - /* calculate filter if library uses filter */ - if (pSynth->pEAS->libAttr & LIB_FORMAT_FILTER_ENABLED) - WT_UpdateFilter(pWTVoice, &intFrame, pArt); - else - intFrame.frame.k = 0; -#endif - - /* update the gain */ - intFrame.frame.gainTarget = WT_UpdateGain(pVoice, pWTVoice, pArt, pChannel, pWTRegion->gain); - - /* calculate base pitch*/ - temp = pChannel->staticPitch + pWTRegion->tuning; - - /* include global transpose */ - if (pChannel->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL) - temp += pVoice->note * 100; - else - temp += (pVoice->note + pSynth->globalTranspose) * 100; - intFrame.frame.phaseIncrement = WT_UpdatePhaseInc(pWTVoice, pArt, pChannel, temp); - - /* call into engine to generate samples */ - intFrame.pAudioBuffer = pVoiceMgr->voiceBuffer; - intFrame.pMixBuffer = pMixBuffer; - intFrame.numSamples = numSamples; - - /* check for end of sample */ - if ((pWTVoice->loopStart != WT_NOISE_GENERATOR) && (pWTVoice->loopStart == pWTVoice->loopEnd)) - done = WT_CheckSampleEnd(pWTVoice, &intFrame, (EAS_BOOL) (voiceNum >= NUM_PRIMARY_VOICES)); - else - done = EAS_FALSE; - -#ifdef EAS_SPLIT_WT_SYNTH - if (voiceNum < NUM_PRIMARY_VOICES) - { -#ifndef _SPLIT_WT_TEST_HARNESS - WT_ProcessVoice(pWTVoice, &intFrame); -#endif - } - else - WTE_ProcessVoice(voiceNum - NUM_PRIMARY_VOICES, &intFrame.frame, pVoiceMgr->pFrameBuffer); -#else - WT_ProcessVoice(pWTVoice, &intFrame); -#endif - - /* clear flag */ - pVoice->voiceFlags &= ~VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET; - - /* if voice has finished, set flag for voice manager */ - if ((pVoice->voiceState != eVoiceStateStolen) && (pWTVoice->eg1State == eEnvelopeStateMuted)) - done = EAS_TRUE; - - /* if the update interval has elapsed, then force the current gain to the next - * gain since we never actually reach the next gain when ramping -- we just get - * very close to the target gain. - */ - pVoice->gain = (EAS_I16) intFrame.frame.gainTarget; - - return done; -} - -/*---------------------------------------------------------------------------- - * WT_UpdatePhaseInc() - *---------------------------------------------------------------------------- - * Purpose: - * Calculate the phase increment - * - * Inputs: - * pVoice - pointer to the voice being updated - * psRegion - pointer to the region - * psArticulation - pointer to the articulation - * nChannelPitchForThisVoice - the portion of the pitch that is fixed for this - * voice during the duration of this synthesis - * pEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * set the phase increment for this voice - *---------------------------------------------------------------------------- -*/ -static EAS_I32 WT_UpdatePhaseInc (S_WT_VOICE *pWTVoice, const S_ARTICULATION *pArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 pitchCents) -{ - EAS_I32 temp; - - /*pitchCents due to CC1 = LFO * (CC1 / 128) * DEFAULT_LFO_MOD_WHEEL_TO_PITCH_CENTS */ - temp = MULT_EG1_EG1(DEFAULT_LFO_MOD_WHEEL_TO_PITCH_CENTS, - ((pChannel->modWheel) << (NUM_EG1_FRAC_BITS -7))); - - /* pitchCents due to channel pressure = LFO * (channel pressure / 128) * DEFAULT_LFO_CHANNEL_PRESSURE_TO_PITCH_CENTS */ - temp += MULT_EG1_EG1(DEFAULT_LFO_CHANNEL_PRESSURE_TO_PITCH_CENTS, - ((pChannel->channelPressure) << (NUM_EG1_FRAC_BITS -7))); - - /* now multiply the (channel pressure + CC1) pitch values by the LFO value */ - temp = MULT_EG1_EG1(pWTVoice->modLFO.lfoValue, temp); - - /* - add in the LFO pitch due to - channel pressure and CC1 along with - the LFO pitch, the EG2 pitch, and the - "static" pitch for this voice on this channel - */ - temp += pitchCents + - (MULT_EG1_EG1(pWTVoice->eg2Value, pArt->eg2ToPitch)) + - (MULT_EG1_EG1(pWTVoice->modLFO.lfoValue, pArt->lfoToPitch)); - - /* convert from cents to linear phase increment */ - return EAS_Calculate2toX(temp); -} - -/*---------------------------------------------------------------------------- - * WT_UpdateChannel() - *---------------------------------------------------------------------------- - * Purpose: - * Calculate and assign static channel parameters - * These values only need to be updated if one of the controller values - * for this channel changes - * - * Inputs: - * nChannel - channel to update - * pEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - the given channel's static gain and static pitch are updated - *---------------------------------------------------------------------------- -*/ -/*lint -esym(715, pVoiceMgr) reserved for future use */ -static void WT_UpdateChannel (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel) -{ - EAS_I32 staticGain; - EAS_I32 pitchBend; - S_SYNTH_CHANNEL *pChannel; - - pChannel = &pSynth->channels[channel]; - - /* - nChannelGain = (CC7 * CC11)^2 * master volume - where CC7 == 100 by default, CC11 == 127, master volume == 32767 - */ - staticGain = MULT_EG1_EG1((pChannel->volume) << (NUM_EG1_FRAC_BITS - 7), - (pChannel->expression) << (NUM_EG1_FRAC_BITS - 7)); - - /* staticGain has to be squared */ - staticGain = MULT_EG1_EG1(staticGain, staticGain); - - pChannel->staticGain = (EAS_I16) MULT_EG1_EG1(staticGain, pSynth->masterVolume); - - /* - calculate pitch bend: RPN0 * ((2*pitch wheel)/16384 -1) - However, if we use the EG1 macros, remember that EG1 has a full - scale value of 32768 (instead of 16384). So instead of multiplying - by 2, multiply by 4 (left shift by 2), and subtract by 32768 instead - of 16384. This utilizes the fact that the EG1 macro places a binary - point 15 places to the left instead of 14 places. - */ - /*lint -e{703} */ - pitchBend = - (((EAS_I32)(pChannel->pitchBend) << 2) - - 32768); - - pChannel->staticPitch = - MULT_EG1_EG1(pitchBend, pChannel->pitchBendSensitivity); - - /* if this is not a drum channel, then add in the per-channel tuning */ - if (!(pChannel->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL)) - pChannel->staticPitch += pChannel->finePitch + (pChannel->coarsePitch * 100); - - /* clear update flag */ - pChannel->channelFlags &= ~CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS; - return; -} - -/*---------------------------------------------------------------------------- - * WT_UpdateGain() - *---------------------------------------------------------------------------- - * Purpose: - * Calculate and assign static voice parameters as part of WT_UpdateVoice() - * - * Inputs: - * pVoice - ptr to the synth voice that we want to synthesize - * pEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - various voice parameters are calculated and assigned - * - *---------------------------------------------------------------------------- -*/ -static EAS_I32 WT_UpdateGain (S_SYNTH_VOICE *pVoice, S_WT_VOICE *pWTVoice, const S_ARTICULATION *pArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 gain) -{ - EAS_I32 lfoGain; - EAS_I32 temp; - - /* - If this voice was stolen, then the velocity is actually - for the new note, not the note that we are currently ramping down. - So we really shouldn't use this velocity. However, that would require - more memory to store the velocity value, and the improvement may - not be sufficient to warrant the added memory. - */ - /* velocity is fixed at note start for a given voice and must be squared */ - temp = (pVoice->velocity) << (NUM_EG1_FRAC_BITS - 7); - temp = MULT_EG1_EG1(temp, temp); - - /* region gain is fixed as part of the articulation */ - temp = MULT_EG1_EG1(temp, gain); - - /* include the channel gain */ - temp = MULT_EG1_EG1(temp, pChannel->staticGain); - - /* calculate LFO gain using an approximation for 10^x */ - lfoGain = MULT_EG1_EG1(pWTVoice->modLFO.lfoValue, pArt->lfoToGain); - lfoGain = MULT_EG1_EG1(lfoGain, LFO_GAIN_TO_CENTS); - - /* convert from a dB-like value to linear gain */ - lfoGain = EAS_Calculate2toX(lfoGain); - temp = MULT_EG1_EG1(temp, lfoGain); - - /* calculate the voice's gain */ - temp = (EAS_I16)MULT_EG1_EG1(temp, pWTVoice->eg1Value); - - return temp; -} - -/*---------------------------------------------------------------------------- - * WT_UpdateEG1() - *---------------------------------------------------------------------------- - * Purpose: - * Calculate the EG1 envelope for the given voice (but do not update any - * state) - * - * Inputs: - * pVoice - ptr to the voice whose envelope we want to update - * nVoice - this voice's number - used only for debug - * pEASData - pointer to overall EAS data structure - * - * Outputs: - * nValue - the envelope value - * - * Side Effects: - * - updates EG1 state value for the given voice - *---------------------------------------------------------------------------- -*/ -static void WT_UpdateEG1 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv) -{ - EAS_I32 temp; - - switch (pWTVoice->eg1State) - { - case eEnvelopeStateAttack: - temp = pWTVoice->eg1Value + pWTVoice->eg1Increment; - - /* check if we have reached peak amplitude */ - if (temp >= SYNTH_FULL_SCALE_EG1_GAIN) - { - /* limit the volume */ - temp = SYNTH_FULL_SCALE_EG1_GAIN; - - /* prepare to move to decay state */ - pWTVoice->eg1State = eEnvelopeStateDecay; - pWTVoice->eg1Increment = pEnv->decayTime; - } - - break; - - /* exponential decay */ - case eEnvelopeStateDecay: - temp = MULT_EG1_EG1(pWTVoice->eg1Value, pWTVoice->eg1Increment); - - /* check if we have reached sustain level */ - if (temp <= pEnv->sustainLevel) - { - /* enforce the sustain level */ - temp = pEnv->sustainLevel; - - /* if sustain level is zero, skip sustain & release the voice */ - if (temp > 0) - pWTVoice->eg1State = eEnvelopeStateSustain; - - /* move to sustain state */ - else - pWTVoice->eg1State = eEnvelopeStateMuted; - } - - break; - - case eEnvelopeStateSustain: - return; - - case eEnvelopeStateRelease: - temp = MULT_EG1_EG1(pWTVoice->eg1Value, pWTVoice->eg1Increment); - - /* if we hit zero, this voice isn't contributing any audio */ - if (temp <= 0) - { - temp = 0; - pWTVoice->eg1State = eEnvelopeStateMuted; - } - break; - - /* voice is muted, set target to zero */ - case eEnvelopeStateMuted: - temp = 0; - break; - - case eEnvelopeStateInvalid: - default: - temp = 0; -#ifdef _DEBUG_SYNTH - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_UpdateEG1: error, %d is an unrecognized state\n", - pWTVoice->eg1State); */ } -#endif - break; - - } - - pWTVoice->eg1Value = (EAS_I16) temp; -} - -/*---------------------------------------------------------------------------- - * WT_UpdateEG2() - *---------------------------------------------------------------------------- - * Purpose: - * Update the EG2 envelope for the given voice - * - * Inputs: - * pVoice - ptr to the voice whose envelope we want to update - * pEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - updates EG2 values for the given voice - *---------------------------------------------------------------------------- -*/ - -static void WT_UpdateEG2 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv) -{ - EAS_I32 temp; - - switch (pWTVoice->eg2State) - { - case eEnvelopeStateAttack: - temp = pWTVoice->eg2Value + pWTVoice->eg2Increment; - - /* check if we have reached peak amplitude */ - if (temp >= SYNTH_FULL_SCALE_EG1_GAIN) - { - /* limit the volume */ - temp = SYNTH_FULL_SCALE_EG1_GAIN; - - /* prepare to move to decay state */ - pWTVoice->eg2State = eEnvelopeStateDecay; - - pWTVoice->eg2Increment = pEnv->decayTime; - } - - break; - - /* implement linear pitch decay in cents */ - case eEnvelopeStateDecay: - temp = pWTVoice->eg2Value -pWTVoice->eg2Increment; - - /* check if we have reached sustain level */ - if (temp <= pEnv->sustainLevel) - { - /* enforce the sustain level */ - temp = pEnv->sustainLevel; - - /* prepare to move to sustain state */ - pWTVoice->eg2State = eEnvelopeStateSustain; - } - break; - - case eEnvelopeStateSustain: - return; - - case eEnvelopeStateRelease: - temp = pWTVoice->eg2Value - pWTVoice->eg2Increment; - - if (temp <= 0) - { - temp = 0; - pWTVoice->eg2State = eEnvelopeStateMuted; - } - - break; - - /* voice is muted, set target to zero */ - case eEnvelopeStateMuted: - temp = 0; - break; - - case eEnvelopeStateInvalid: - default: - temp = 0; -#ifdef _DEBUG_SYNTH - { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_UpdateEG2: error, %d is an unrecognized state\n", - pWTVoice->eg2State); */ } -#endif - break; - } - - pWTVoice->eg2Value = (EAS_I16) temp; -} - -/*---------------------------------------------------------------------------- - * WT_UpdateLFO () - *---------------------------------------------------------------------------- - * Purpose: - * Calculate the LFO for the given voice - * - * Inputs: - * pLFO - ptr to the LFO data - * phaseInc - phase increment - * - * Outputs: - * - * Side Effects: - * - updates LFO values for the given voice - *---------------------------------------------------------------------------- -*/ -void WT_UpdateLFO (S_LFO_CONTROL *pLFO, EAS_I16 phaseInc) -{ - - /* To save memory, if m_nPhaseValue is negative, we are in the - * delay phase, and m_nPhaseValue represents the time left - * in the delay. - */ - if (pLFO->lfoPhase < 0) - { - pLFO->lfoPhase++; - return; - } - - /* calculate LFO output from phase value */ - /*lint -e{701} Use shift for performance */ - pLFO->lfoValue = (EAS_I16) (pLFO->lfoPhase << 2); - /*lint -e{502} */ - if ((pLFO->lfoPhase > 0x1fff) && (pLFO->lfoPhase < 0x6000)) - pLFO->lfoValue = ~pLFO->lfoValue; - - /* update LFO phase */ - pLFO->lfoPhase = (pLFO->lfoPhase + phaseInc) & 0x7fff; -} - -#ifdef _FILTER_ENABLED -/*---------------------------------------------------------------------------- - * WT_UpdateFilter() - *---------------------------------------------------------------------------- - * Purpose: - * Update the Filter parameters - * - * Inputs: - * pVoice - ptr to the voice whose filter we want to update - * pEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - updates Filter values for the given voice - *---------------------------------------------------------------------------- -*/ -static void WT_UpdateFilter (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pIntFrame, const S_ARTICULATION *pArt) -{ - EAS_I32 cutoff; - - /* no need to calculate filter coefficients if it is bypassed */ - if (pArt->filterCutoff == DEFAULT_EAS_FILTER_CUTOFF_FREQUENCY) - { - pIntFrame->frame.k = 0; - return; - } - - /* determine the dynamic cutoff frequency */ - cutoff = MULT_EG1_EG1(pWTVoice->eg2Value, pArt->eg2ToFc); - cutoff += pArt->filterCutoff; - - /* subtract the A5 offset and the sampling frequency */ - cutoff -= FILTER_CUTOFF_FREQ_ADJUST + A5_PITCH_OFFSET_IN_CENTS; - - /* limit the cutoff frequency */ - if (cutoff > FILTER_CUTOFF_MAX_PITCH_CENTS) - cutoff = FILTER_CUTOFF_MAX_PITCH_CENTS; - else if (cutoff < FILTER_CUTOFF_MIN_PITCH_CENTS) - cutoff = FILTER_CUTOFF_MIN_PITCH_CENTS; - - WT_SetFilterCoeffs(pIntFrame, cutoff, pArt->filterQ); -} -#endif - -#if defined(_FILTER_ENABLED) || defined(DLS_SYNTHESIZER) -/*---------------------------------------------------------------------------- - * coef - *---------------------------------------------------------------------------- - * Table of filter coefficients for low-pass filter - *---------------------------------------------------------------------------- - * - * polynomial coefficients are based on 8kHz sampling frequency - * filter coef b2 = k2 = k2g0*k^0 + k2g1*k^1*(2^x) + k2g2*k^2*(2^x) - * - *where k2g0, k2g1, k2g2 are from the truncated power series expansion on theta - *(k*2^x = theta, but we incorporate the k along with the k2g0, k2g1, k2g2) - *note: this is a power series in 2^x, not k*2^x - *where k = (2*pi*440)/8kHz == convert octaves to radians - * - * so actually, the following coefs listed as k2g0, k2g1, k2g2 are really - * k2g0*k^0 = k2g0 - * k2g1*k^1 - * k2g2*k^2 - * - * - * filter coef n1 = numerator = n1g0*k^0 + n1g1*k^1*(2^x) + n1g2*k^2*(2^x) + n1g3*k^3*(2^x) - * - *where n1g0, n1g1, n1g2, n1g3 are from the truncated power series expansion on theta - *(k*2^x = theta, but we incorporate the k along with the n1g0, n1g1, n1g2, n2g3) - *note: this is a power series in 2^x, not k*2^x - *where k = (2*pi*440)/8kHz == convert octaves to radians - *we also include the optimization factor of 0.81 - * - * so actually, the following coefs listed as n1g0, n1g1, n1g2, n2g3 are really - * n1g0*k^0 = n1g0 - * n1g1*k^1 - * n1g2*k^2 - * n1g3*k^3 - * - * NOTE that n1g0 == n1g1 == 0, always, so we only need to store n1g2 and n1g3 - *---------------------------------------------------------------------------- -*/ - -static const EAS_I16 nk1g0 = -32768; -static const EAS_I16 nk1g2 = 1580; -static const EAS_I16 k2g0 = 32767; - -static const EAS_I16 k2g1[] = -{ - -11324, /* k2g1[0] = -0.3455751918948761 */ - -10387, /* k2g1[1] = -0.3169878073928751 */ - -9528, /* k2g1[2] = -0.29076528753345476 */ - -8740, /* k2g1[3] = -0.2667120011011279 */ - -8017, /* k2g1[4] = -0.24464850028971705 */ - -7353, /* k2g1[5] = -0.22441018194495696 */ - -6745, /* k2g1[6] = -0.20584605955455101 */ - -6187, /* k2g1[7] = -0.18881763682420102 */ - -5675, /* k2g1[8] = -0.1731978744360067 */ - -5206, /* k2g1[9] = -0.15887024228080968 */ - -4775, /* k2g1[10] = -0.14572785009373057 */ - -4380, /* k2g1[11] = -0.13367265000706827 */ - -4018, /* k2g1[12] = -0.1226147050712642 */ - -3685, /* k2g1[13] = -0.11247151828678581 */ - -3381, /* k2g1[14] = -0.10316741714122014 */ - -3101, /* k2g1[15] = -0.0946329890599603 */ - -2844, /* k2g1[16] = -0.08680456355870586 */ - -2609, /* k2g1[17] = -0.07962373723441349 */ - -2393, /* k2g1[18] = -0.07303693805092666 */ - -2195, /* k2g1[19] = -0.06699502566866912 */ - -2014, /* k2g1[20] = -0.06145292483669077 */ - -1847, /* k2g1[21] = -0.056369289112013346 */ - -1694, /* k2g1[22] = -0.05170619239747895 */ - -1554, /* k2g1[23] = -0.04742884599684141 */ - -1426, /* k2g1[24] = -0.043505339076210514 */ - -1308, /* k2g1[25] = -0.03990640059558053 */ - -1199, /* k2g1[26] = -0.03660518093435039 */ - -1100, /* k2g1[27] = -0.03357705158166837 */ - -1009, /* k2g1[28] = -0.030799421397205727 */ - -926, /* k2g1[29] = -0.028251568071585884 */ - -849 /* k2g1[30] = -0.025914483529091967 */ -}; - -static const EAS_I16 k2g2[] = -{ - 1957, /* k2g2[0] = 0.059711106626580836 */ - 1646, /* k2g2[1] = 0.05024063501786333 */ - 1385, /* k2g2[2] = 0.042272226217199664 */ - 1165, /* k2g2[3] = 0.03556764576567844 */ - 981, /* k2g2[4] = 0.029926444346999134 */ - 825, /* k2g2[5] = 0.025179964880280382 */ - 694, /* k2g2[6] = 0.02118630011706455 */ - 584, /* k2g2[7] = 0.01782604998793514 */ - 491, /* k2g2[8] = 0.014998751854573014 */ - 414, /* k2g2[9] = 0.012619876941179595 */ - 348, /* k2g2[10] = 0.010618303146468736 */ - 293, /* k2g2[11] = 0.008934188679954682 */ - 246, /* k2g2[12] = 0.007517182949855368 */ - 207, /* k2g2[13] = 0.006324921212866403 */ - 174, /* k2g2[14] = 0.005321757979794424 */ - 147, /* k2g2[15] = 0.004477701309210577 */ - 123, /* k2g2[16] = 0.00376751612730811 */ - 104, /* k2g2[17] = 0.0031699697655869644 */ - 87, /* k2g2[18] = 0.00266719715992703 */ - 74, /* k2g2[19] = 0.0022441667321724647 */ - 62, /* k2g2[20] = 0.0018882309854916855 */ - 52, /* k2g2[21] = 0.0015887483774966232 */ - 44, /* k2g2[22] = 0.0013367651661223448 */ - 37, /* k2g2[23] = 0.0011247477162958733 */ - 31, /* k2g2[24] = 0.0009463572640678758 */ - 26, /* k2g2[25] = 0.0007962604042473498 */ - 22, /* k2g2[26] = 0.0006699696356181593 */ - 18, /* k2g2[27] = 0.0005637091964589207 */ - 16, /* k2g2[28] = 0.00047430217920125243 */ - 13, /* k2g2[29] = 0.00039907554925166274 */ - 11 /* k2g2[30] = 0.00033578022828973666 */ -}; - -static const EAS_I16 n1g2[] = -{ - 3170, /* n1g2[0] = 0.0967319927350769 */ - 3036, /* n1g2[1] = 0.0926446051254155 */ - 2908, /* n1g2[2] = 0.08872992911818503 */ - 2785, /* n1g2[3] = 0.08498066682523227 */ - 2667, /* n1g2[4] = 0.08138982872895201 */ - 2554, /* n1g2[5] = 0.07795072065216213 */ - 2446, /* n1g2[6] = 0.0746569312785634 */ - 2343, /* n1g2[7] = 0.07150232020051943 */ - 2244, /* n1g2[8] = 0.06848100647187474 */ - 2149, /* n1g2[9] = 0.06558735764447099 */ - 2058, /* n1g2[10] = 0.06281597926792246 */ - 1971, /* n1g2[11] = 0.06016170483307614 */ - 1888, /* n1g2[12] = 0.05761958614040857 */ - 1808, /* n1g2[13] = 0.05518488407540374 */ - 1732, /* n1g2[14] = 0.052853059773715245 */ - 1659, /* n1g2[15] = 0.05061976615964251 */ - 1589, /* n1g2[16] = 0.04848083984214659 */ - 1521, /* n1g2[17] = 0.046432293353298 */ - 1457, /* n1g2[18] = 0.04447030771468711 */ - 1396, /* n1g2[19] = 0.04259122531793907 */ - 1337, /* n1g2[20] = 0.040791543106060944 */ - 1280, /* n1g2[21] = 0.03906790604290942 */ - 1226, /* n1g2[22] = 0.037417100858604564 */ - 1174, /* n1g2[23] = 0.035836050059229754 */ - 1125, /* n1g2[24] = 0.03432180618965023 */ - 1077, /* n1g2[25] = 0.03287154633875494 */ - 1032, /* n1g2[26] = 0.03148256687687814 */ - 988, /* n1g2[27] = 0.030152278415589925 */ - 946, /* n1g2[28] = 0.028878200980459685 */ - 906, /* n1g2[29] = 0.02765795938779331 */ - 868 /* n1g2[30] = 0.02648927881672521 */ -}; - -static const EAS_I16 n1g3[] = -{ - -548, /* n1g3[0] = -0.016714088475899017 */ - -481, /* n1g3[1] = -0.014683605122742116 */ - -423, /* n1g3[2] = -0.012899791676436092 */ - -371, /* n1g3[3] = -0.01133268185193299 */ - -326, /* n1g3[4] = -0.00995594976868754 */ - -287, /* n1g3[5] = -0.008746467702146129 */ - -252, /* n1g3[6] = -0.00768391756106361 */ - -221, /* n1g3[7] = -0.006750449563854721 */ - -194, /* n1g3[8] = -0.005930382380083576 */ - -171, /* n1g3[9] = -0.005209939699767622 */ - -150, /* n1g3[10] = -0.004577018805123356 */ - -132, /* n1g3[11] = -0.004020987256990177 */ - -116, /* n1g3[12] = -0.003532504280467257 */ - -102, /* n1g3[13] = -0.00310336384922047 */ - -89, /* n1g3[14] = -0.002726356832432369 */ - -78, /* n1g3[15] = -0.002395149888601605 */ - -69, /* n1g3[16] = -0.0021041790717285314 */ - -61, /* n1g3[17] = -0.0018485563625771063 */ - -53, /* n1g3[18] = -0.001623987554831628 */ - -47, /* n1g3[19] = -0.0014267001167177025 */ - -41, /* n1g3[20] = -0.0012533798162347005 */ - -36, /* n1g3[21] = -0.0011011150453668693 */ - -32, /* n1g3[22] = -0.0009673479079754438 */ - -28, /* n1g3[23] = -0.0008498312496971563 */ - -24, /* n1g3[24] = -0.0007465909079943587 */ - -21, /* n1g3[25] = -0.0006558925481952733 */ - -19, /* n1g3[26] = -0.0005762125284029567 */ - -17, /* n1g3[27] = -0.0005062123038325457 */ - -15, /* n1g3[28] = -0.0004447159405951901 */ - -13, /* n1g3[29] = -0.00039069036118270117 */ - -11 /* n1g3[30] = -0.00034322798979677605 */ -}; - -/*---------------------------------------------------------------------------- - * WT_SetFilterCoeffs() - *---------------------------------------------------------------------------- - * Purpose: - * Update the Filter parameters - * - * Inputs: - * pVoice - ptr to the voice whose filter we want to update - * pEASData - pointer to overall EAS data structure - * - * Outputs: - * - * Side Effects: - * - updates Filter values for the given voice - *---------------------------------------------------------------------------- -*/ -void WT_SetFilterCoeffs (S_WT_INT_FRAME *pIntFrame, EAS_I32 cutoff, EAS_I32 resonance) -{ - EAS_I32 temp; - - /* - Convert the cutoff, which has had A5 subtracted, using the 2^x approx - Note, this cutoff is related to theta cutoff by - theta = k * 2^x - We use 2^x and incorporate k in the power series coefs instead - */ - cutoff = EAS_Calculate2toX(cutoff); - - /* calculate b2 coef */ - temp = k2g1[resonance] + MULT_AUDIO_COEF(cutoff, k2g2[resonance]); - temp = k2g0 + MULT_AUDIO_COEF(cutoff, temp); - pIntFrame->frame.b2 = temp; - - /* calculate b1 coef */ - temp = MULT_AUDIO_COEF(cutoff, nk1g2); - temp = nk1g0 + MULT_AUDIO_COEF(cutoff, temp); - temp += MULT_AUDIO_COEF(temp, pIntFrame->frame.b2); - pIntFrame->frame.b1 = temp >> 1; - - /* calculate K coef */ - temp = n1g2[resonance] + MULT_AUDIO_COEF(cutoff, n1g3[resonance]); - temp = MULT_AUDIO_COEF(cutoff, temp); - temp = MULT_AUDIO_COEF(cutoff, temp); - pIntFrame->frame.k = temp; -} -#endif - diff --git a/arm-hybrid-22k/lib_src/eas_wtsynth.h b/arm-hybrid-22k/lib_src/eas_wtsynth.h deleted file mode 100644 index 106ab40..0000000 --- a/arm-hybrid-22k/lib_src/eas_wtsynth.h +++ /dev/null @@ -1,66 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * File: - * eas_wtsynth.h - * - * Contents and purpose: - * This file defines the interface for synthesizer engine - * - * Copyright Sonic Network Inc. 2004 - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision: 82 $ - * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $ - *---------------------------------------------------------------------------- -*/ - -#ifndef _EAS_WTSYNTH_H -#define _EAS_WTSYNTH_H - -#include "eas_sndlib.h" -#include "eas_wtengine.h" - -/* adjust the filter cutoff frequency to the sample rate */ -#if defined (_SAMPLE_RATE_8000) -#define FILTER_CUTOFF_FREQ_ADJUST 0 -#elif defined (_SAMPLE_RATE_16000) -#define FILTER_CUTOFF_FREQ_ADJUST 1200 -#elif defined (_SAMPLE_RATE_20000) -#define FILTER_CUTOFF_FREQ_ADJUST 1586 -#elif defined (_SAMPLE_RATE_22050) -#define FILTER_CUTOFF_FREQ_ADJUST 1756 -#elif defined (_SAMPLE_RATE_24000) -#define FILTER_CUTOFF_FREQ_ADJUST 1902 -#elif defined (_SAMPLE_RATE_32000) -#define FILTER_CUTOFF_FREQ_ADJUST 2400 -#elif defined (_SAMPLE_RATE_44100) -#define FILTER_CUTOFF_FREQ_ADJUST 2956 -#elif defined (_SAMPLE_RATE_48000) -#define FILTER_CUTOFF_FREQ_ADJUST 3102 -#else -#error "_SAMPLE_RATE_XXXXX must be defined to valid rate" -#endif - -/* function prototypes */ -void WT_UpdateLFO (S_LFO_CONTROL *pLFO, EAS_I16 phaseInc); - -#if defined(_FILTER_ENABLED) || defined(DLS_SYNTHESIZER) -void WT_SetFilterCoeffs (S_WT_INT_FRAME *pIntFrame, EAS_I32 cutoff, EAS_I32 resonance); -#endif - -#endif - - diff --git a/arm-hybrid-22k/lib_src/hybrid_22khz_mcu.c b/arm-hybrid-22k/lib_src/hybrid_22khz_mcu.c deleted file mode 100644 index 880819c..0000000 --- a/arm-hybrid-22k/lib_src/hybrid_22khz_mcu.c +++ /dev/null @@ -1,5149 +0,0 @@ -/*---------------------------------------------------------------------------- - * - * Filename: C:\Sonic\Trunk\EASLib\WTLibrary\hybrid_22khz_mcu.c - * Source: C:\Sonic\Trunk\Wavetables\Sonic_20Khz_Drums.dls - * CmdLine: -w C:\Sonic\Trunk\EASLib\WTLibrary\hybrid_22khz_mcu.c -l C:\Sonic\Trunk\EASLib\WTLibrary\hybrid_22khz.log -d 0 -c -f C:\Sonic\Release3-5\EASLib\FMSynth\GMdblib-3.fml C:\Sonic\Trunk\Wavetables\Sonic_20Khz_Drums.dls -w -l -d -c -f C:\Sonic\Trunk\Wavetables\Sonic_20Khz_Drums.dls - * Purpose: Wavetable sound libary - * - * Copyright (c) 2006 Sonic Network Inc. - - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - * - *---------------------------------------------------------------------------- - * Revision Control: - * $Revision:$ - * $Date:$ - *---------------------------------------------------------------------------- -*/ - -#include "eas_sndlib.h" - -/*---------------------------------------------------------------------------- - * Articulations - *---------------------------------------------------------------------------- -*/ -const S_ARTICULATION eas_articulations[] = -{ - { /* articulation 0 */ - { 32767, 30725, 0, 30725 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 0 - }, - { /* articulation 1 */ - { 32767, 26863, 0, 26863 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 0 - }, - { /* articulation 2 */ - { 32767, 30484, 0, 30668 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 0 - }, - { /* articulation 3 */ - { 32767, 26439, 0, 26439 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 0 - }, - { /* articulation 4 */ - { 32767, 0, 32767, 32715 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 0 - }, - { /* articulation 5 */ - { 32767, 21333, 0, 21333 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 0 - }, - { /* articulation 6 */ - { 32767, 31882, 0, 31938 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 0 - }, - { /* articulation 7 */ - { 32767, 32663, 32767, 32663 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 0 - }, - { /* articulation 8 */ - { 32767, 0, 32767, 0 }, - { 32767, 1902, 0, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 0 - }, - { /* articulation 9 */ - { 32767, 32349, 0, 32349 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 0 - }, - { /* articulation 10 */ - { 32767, 0, 32767, 17213 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -1 - }, - { /* articulation 11 */ - { 32767, 32072, 0, 32072 }, - { 32767, 761, 0, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -56 - }, - { /* articulation 12 */ - { 32767, 23749, 0, 23749 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 56 - }, - { /* articulation 13 */ - { 32767, 32010, 0, 32010 }, - { 32767, 761, 0, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -31 - }, - { /* articulation 14 */ - { 9511, 21333, 0, 21333 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 56 - }, - { /* articulation 15 */ - { 32767, 31844, 0, 31844 }, - { 32767, 761, 0, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -6 - }, - { /* articulation 16 */ - { 32767, 32123, 0, 32194 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 56 - }, - { /* articulation 17 */ - { 32767, 31730, 0, 31730 }, - { 32767, 761, 0, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 6 - }, - { /* articulation 18 */ - { 32767, 31391, 0, 31391 }, - { 32767, 951, 0, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 31 - }, - { /* articulation 19 */ - { 32767, 31964, 0, 31964 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 25 - }, - { /* articulation 20 */ - { 32767, 31056, 0, 31056 }, - { 32767, 951, 0, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 63 - }, - { /* articulation 21 */ - { 32767, 32289, 0, 32271 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -37 - }, - { /* articulation 22 */ - { 19021, 31882, 0, 31911 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -37 - }, - { /* articulation 23 */ - { 32767, 31988, 0, 32032 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -37 - }, - { /* articulation 24 */ - { 32767, 0, 32767, 32663 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 12 - }, - { /* articulation 25 */ - { 32767, 31352, 0, 31352 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -25 - }, - { /* articulation 26 */ - { 32767, 0, 32767, 32663 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 25 - }, - { /* articulation 27 */ - { 32767, 31817, 0, 31781 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -25 - }, - { /* articulation 28 */ - { 32767, 30725, 0, 30725 }, - { 32767, 95, 0, 0 }, - 0, 0, 951, 240, 0, 0, 0, 0, -56 - }, - { /* articulation 29 */ - { 32767, 32230, 0, 32218 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -37 - }, - { /* articulation 30 */ - { 32767, 26439, 0, 26439 }, - { 32767, 3804, 0, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 50 - }, - { /* articulation 31 */ - { 32767, 23749, 0, 23749 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -50 - }, - { /* articulation 32 */ - { 32767, 29434, 0, 29434 }, - { 32767, 3804, 0, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -50 - }, - { /* articulation 33 */ - { 32767, 30240, 0, 30234 }, - { 32767, 3804, 0, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -44 - }, - { /* articulation 34 */ - { 32767, 32558, 0, 32558 }, - { 32767, 254, 0, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 25 - }, - { /* articulation 35 */ - { 32767, 0, 32767, 32663 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -63 - }, - { /* articulation 36 */ - { 3804, 23749, 0, 23749 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -63 - }, - { /* articulation 37 */ - { 32767, 23749, 0, 23749 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -59 - }, - { /* articulation 38 */ - { 32767, 30725, 0, 30725 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 50 - }, - { /* articulation 39 */ - { 32767, 28809, 0, 28809 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 44 - }, - { /* articulation 40 */ - { 1902, 30725, 0, 30725 }, - { 32767, 380, 0, 0 }, - 0, 0, 951, -100, 0, 0, 0, 0, 44 - }, - { /* articulation 41 */ - { 32767, 9042, 0, 9042 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 25 - }, - { /* articulation 42 */ - { 32767, 29889, 0, 29889 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 63 - }, - { /* articulation 43 */ - { 32767, 30240, 0, 30234 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 63 - }, - { /* articulation 44 */ - { 19021, 19970, 0, 19970 }, - { 951, 32767, 32767, 0 }, - 0, 0, 951, 100, 0, 0, 0, 0, -25 - }, - { /* articulation 45 */ - { 3804, 17213, 0, 17213 }, - { 951, 32767, 32767, 0 }, - 0, 0, 951, 500, 0, 0, 0, 0, -25 - }, - { /* articulation 46 */ - { 32767, 17213, 0, 17213 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -56 - }, - { /* articulation 47 */ - { 32767, 30725, 0, 30725 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, -56 - }, - { /* articulation 48 */ - { 32767, 0, 32767, 0 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 0 - }, - { /* articulation 49 */ - { 32767, 31180, 0, 31180 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 0 - }, - { /* articulation 50 */ - { 19021, 31964, 0, 32071 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 0 - }, - { /* articulation 51 */ - { 32767, 29669, 0, 29669 }, - { 32767, 32767, 32767, 0 }, - 0, 0, 951, 0, 0, 0, 0, 0, 0 - }, - { /* articulation 52 */ - { 32767, 31742, 0, 31352 }, - { 32767, 294, 0, 0 }, - 0, 0, 951, 0, 10000, 7121, 0, 0, 0 - } -}; /*end Articulations */ - -/*---------------------------------------------------------------------------- - * Regions - *---------------------------------------------------------------------------- -*/ -const S_WT_REGION eas_regions[] = -{ - { { 0, 27, 27 }, -2868, 16422, 0, 0, 19, 0 }, /* region 0 */ - { { 0, 28, 28 }, -3568, 32767, 0, 0, 13, 0 }, /* region 1 */ - { { 0, 29, 29 }, -4553, 32767, 0, 0, 9, 1 }, /* region 2 */ - { { 0, 30, 30 }, -4853, 32767, 0, 0, 9, 2 }, /* region 3 */ - { { 0, 31, 31 }, -3868, 23197, 0, 0, 15, 3 }, /* region 4 */ - { { 1536, 32, 32 }, -3368, 20675, 0, 0, 29, 4 }, /* region 5 */ - { { 1537, 33, 33 }, -3868, 20675, 792, 800, 17, 5 }, /* region 6 */ - { { 1537, 34, 34 }, -3968, 16422, 792, 800, 17, 6 }, /* region 7 */ - { { 0, 35, 35 }, -4968, 32767, 0, 0, 20, 7 }, /* region 8 */ - { { 0, 36, 36 }, -4968, 32767, 0, 0, 20, 7 }, /* region 9 */ - { { 0, 37, 37 }, -4051, 18426, 0, 0, 16, 8 }, /* region 10 */ - { { 0, 38, 38 }, -4151, 23197, 0, 0, 5, 9 }, /* region 11 */ - { { 0, 39, 39 }, -4668, 23197, 0, 0, 12, 10 }, /* region 12 */ - { { 0, 40, 40 }, -4151, 23197, 0, 0, 5, 4 }, /* region 13 */ - { { 1, 41, 41 }, -5855, 26028, 798, 993, 14, 11 }, /* region 14 */ - { { 257, 42, 42 }, -4368, 26028, 4288, 6792, 2, 12 }, /* region 15 */ - { { 1, 43, 43 }, -5755, 26028, 798, 993, 14, 13 }, /* region 16 */ - { { 257, 44, 44 }, -4568, 26028, 4288, 6792, 2, 14 }, /* region 17 */ - { { 1, 45, 45 }, -5755, 26028, 798, 993, 14, 15 }, /* region 18 */ - { { 257, 46, 46 }, -4768, 26028, 4288, 6792, 2, 16 }, /* region 19 */ - { { 1, 47, 47 }, -5455, 26028, 798, 993, 14, 17 }, /* region 20 */ - { { 1, 48, 48 }, -5355, 26028, 798, 993, 14, 18 }, /* region 21 */ - { { 1, 49, 49 }, -5368, 16422, 1294, 5241, 3, 19 }, /* region 22 */ - { { 1, 50, 50 }, -5255, 26028, 798, 993, 14, 20 }, /* region 23 */ - { { 1, 51, 51 }, -5268, 16422, 6592, 9921, 0, 21 }, /* region 24 */ - { { 1, 52, 52 }, -5768, 32767, 1294, 5241, 3, 22 }, /* region 25 */ - { { 1, 53, 53 }, -5418, 14636, 6592, 9921, 0, 23 }, /* region 26 */ - { { 0, 54, 54 }, -5751, 26028, 0, 0, 11, 24 }, /* region 27 */ - { { 1, 55, 55 }, -5468, 32767, 1294, 5241, 3, 25 }, /* region 28 */ - { { 0, 56, 56 }, -7255, 32767, 0, 0, 24, 26 }, /* region 29 */ - { { 1, 57, 57 }, -5868, 32767, 1294, 5241, 3, 27 }, /* region 30 */ - { { 1, 58, 58 }, -7053, 23197, 0, 166, 26, 28 }, /* region 31 */ - { { 1, 59, 59 }, -5968, 16422, 6592, 9921, 0, 29 }, /* region 32 */ - { { 1, 60, 60 }, -6453, 23197, 432, 582, 18, 30 }, /* region 33 */ - { { 1, 61, 61 }, -6853, 16422, 432, 582, 18, 30 }, /* region 34 */ - { { 1, 62, 62 }, -7253, 20675, 432, 582, 18, 31 }, /* region 35 */ - { { 1, 63, 63 }, -7353, 23197, 432, 582, 18, 32 }, /* region 36 */ - { { 1, 64, 64 }, -7953, 23197, 432, 582, 18, 33 }, /* region 37 */ - { { 0, 65, 65 }, -7555, 32767, 0, 0, 4, 34 }, /* region 38 */ - { { 0, 66, 66 }, -7955, 20675, 0, 0, 4, 34 }, /* region 39 */ - { { 512, 67, 67 }, -7155, 18426, 0, 0, 24, 35 }, /* region 40 */ - { { 512, 68, 68 }, -7755, 18426, 0, 0, 24, 35 }, /* region 41 */ - { { 0, 69, 69 }, -7755, 32767, 0, 0, 21, 36 }, /* region 42 */ - { { 0, 70, 70 }, -6855, 21900, 0, 0, 21, 37 }, /* region 43 */ - { { 769, 71, 71 }, -6355, 23197, 0, 1226, 10, 38 }, /* region 44 */ - { { 769, 72, 72 }, -6955, 26028, 0, 1226, 10, 38 }, /* region 45 */ - { { 1024, 73, 73 }, -7955, 32767, 0, 0, 7, 39 }, /* region 46 */ - { { 1024, 74, 74 }, -8455, 32767, 0, 0, 7, 40 }, /* region 47 */ - { { 1, 75, 75 }, -8068, 23197, 0, 29, 30, 41 }, /* region 48 */ - { { 0, 76, 76 }, -10455, 23197, 0, 0, 28, 42 }, /* region 49 */ - { { 0, 77, 77 }, -10055, 23197, 0, 0, 28, 43 }, /* region 50 */ - { { 0, 78, 78 }, -8853, 16422, 0, 0, 23, 44 }, /* region 51 */ - { { 0, 79, 79 }, -10253, 16422, 0, 0, 23, 45 }, /* region 52 */ - { { 1280, 80, 80 }, -6468, 13045, 0, 0, 25, 46 }, /* region 53 */ - { { 1280, 81, 81 }, -6568, 16422, 0, 0, 25, 47 }, /* region 54 */ - { { 0, 82, 82 }, -8455, 20675, 0, 0, 22, 48 }, /* region 55 */ - { { 0, 83, 83 }, -9068, 32767, 0, 0, 6, 49 }, /* region 56 */ - { { 1, 84, 84 }, -8568, 23197, 0, 9337, 1, 50 }, /* region 57 */ - { { 0, 85, 85 }, -9655, 32767, 0, 0, 27, 0 }, /* region 58 */ - { { 0, 86, 86 }, -9068, 16422, 0, 0, 8, 51 }, /* region 59 */ - { { 32769, 87, 87 }, -9168, 32767, 1335, 1603, 8, 52 } /* region 60 */ -}; /* end Regions */ - -/*---------------------------------------------------------------------------- - * FM Regions - *---------------------------------------------------------------------------- -*/ -const S_FM_REGION eas_fmRegions[] = -{ - - { /* FM region 0 */ - { 37, 0, 127 }, 0, 255, 8, 0, - { - { 514, 239, 47, 97, 0, 184, 3 }, - { 1, 244, 89, 114, 0, 248, 2 }, - { 3370, 244, 49, 76, 40, 192, 2 }, - { -1, 227, 97, 51, 160, 212, 2 } - } - }, - { /* FM region 1 */ - { 37, 0, 127 }, 160, 255, 8, 0, - { - { 2514, 223, 95, 72, 0, 176, 3 }, - { 1, 244, 73, 145, 0, 244, 2 }, - { 3600, 245, 81, 198, 40, 192, 2 }, - { 3, 246, 81, 163, 108, 212, 2 } - } - }, - { /* FM region 2 */ - { 37, 0, 127 }, 160, 255, 119, 0, - { - { 0, 216, 79, 72, 0, 216, 2 }, - { 2, 244, 73, 145, 0, 244, 2 }, - { 3370, 247, 33, 182, 60, 204, 2 }, - { 1200, 246, 65, 163, 108, 204, 2 } - } - }, - { /* FM region 3 */ - { 37, 0, 127 }, 160, 255, 1, 0, - { - { 3369, 248, 65, 71, 40, 208, 2 }, - { -3, 245, 88, 113, 0, 244, 2 }, - { 2784, 225, 65, 133, 80, 192, 2 }, - { 3, 241, 81, 113, 80, 216, 2 } - } - }, - { /* FM region 4 */ - { 34, 0, 127 }, 0, 255, 128, 0, - { - { 0, 229, 155, 183, 0, 228, 2 }, - { -3, 243, 90, 81, 0, 244, 2 }, - { 4800, 248, 109, 180, 36, 192, 2 }, - { 3, 245, 90, 85, 16, 244, 2 } - } - }, - { /* FM region 5 */ - { 34, 0, 127 }, 9, 96, 192, 0, - { - { 1200, 229, 157, 180, 0, 216, 2 }, - { -3, 244, 90, 81, 0, 244, 2 }, - { 1902, 255, 111, 182, 80, 208, 2 }, - { 3, 246, 92, 83, 0, 244, 2 } - } - }, - { /* FM region 6 */ - { 34, 0, 127 }, 0, 255, 154, 0, - { - { 3102, 244, 63, 102, 228, 228, 2 }, - { 1200, 247, 93, 97, 0, 236, 2 }, - { 1902, 255, 63, 98, 156, 220, 2 }, - { 1200, 244, 92, 98, 0, 236, 2 } - } - }, - { /* FM region 7 */ - { 37, 0, 127 }, 0, 255, 202, 0, - { - { 0, 251, 131, 19, 216, 220, 2 }, - { 1201, 247, 62, 113, 0, 240, 2 }, - { 0, 243, 154, 36, 240, 224, 2 }, - { 2784, 250, 61, 36, 240, 208, 2 } - } - }, - { /* FM region 8 */ - { 33, 0, 127 }, 0, 255, 80, 0, - { - { -1, 213, 191, 183, 0, 204, 2 }, - { 1, 245, 154, 129, 0, 244, 2 }, - { 3831, 252, 159, 100, 0, 200, 2 }, - { 1197, 246, 91, 182, 0, 244, 2 } - } - }, - { /* FM region 9 */ - { 34, 0, 127 }, 48, 80, 21, 0, - { - { 2982, 255, 43, 96, 0, 196, 3 }, - { 3, 247, 71, 130, 0, 244, 2 }, - { 3358, 253, 40, 98, 144, 208, 2 }, - { -2, 246, 70, 130, 0, 236, 2 } - } - }, - { /* FM region 10 */ - { 34, 0, 127 }, 48, 80, 26, 0, - { - { 3096, 249, 72, 100, 0, 208, 2 }, - { 2185, 249, 102, 130, 0, 240, 2 }, - { 3386, 247, 66, 100, 144, 212, 2 }, - { -2, 247, 102, 130, 0, 240, 2 } - } - }, - { /* FM region 11 */ - { 34, 0, 127 }, 92, 67, 21, 0, - { - { 2982, 255, 27, 146, 0, 200, 3 }, - { 3, 246, 68, 146, 0, 240, 2 }, - { 3358, 250, 149, 116, 144, 208, 2 }, - { -3, 245, 68, 146, 0, 240, 0 } - } - }, - { /* FM region 12 */ - { 34, 0, 127 }, 0, 67, 0, 0, - { - { 1500, 239, 60, 151, 0, 220, 2 }, - { 0, 247, 76, 146, 0, 240, 2 }, - { 2398, 234, 156, 151, 0, 212, 2 }, - { 0, 246, 105, 146, 0, 244, 2 } - } - }, - { /* FM region 13 */ - { 34, 0, 127 }, 0, 67, 0, 0, - { - { 2500, 255, 60, 151, 0, 220, 2 }, - { 0, 249, 92, 146, 0, 244, 2 }, - { 3369, 250, 156, 151, 0, 196, 2 }, - { 0, 248, 89, 146, 0, 244, 2 } - } - }, - { /* FM region 14 */ - { 37, 0, 127 }, 160, 255, 0, 0, - { - { 2300, 229, 112, 49, 0, 208, 2 }, - { -3, 247, 67, 50, 0, 248, 2 }, - { 1074, 255, 41, 49, 0, 196, 2 }, - { 686, 240, 97, 18, 0, 196, 2 } - } - }, - { /* FM region 15 */ - { 37, 0, 127 }, 160, 255, 219, 0, - { - { 3369, 255, 65, 70, 40, 216, 2 }, - { 1, 246, 72, 113, 0, 240, 2 }, - { 1902, 225, 33, 129, 80, 204, 2 }, - 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} - }, - { /* FM region 92 */ - { 34, 0, 127 }, 82, 67, 71, 0, - { - { 2982, 228, 22, 146, 88, 192, 3 }, - { 3, 102, 84, 146, 196, 240, 2 }, - { 3358, 50, 149, 116, 144, 208, 2 }, - { -3, 85, 84, 146, 120, 240, 0 } - } - }, - { /* FM region 93 */ - { 37, 0, 127 }, 48, 126, 219, 0, - { - { -3, 49, 19, 33, 120, 200, 0 }, - { 0, 81, 70, 33, 220, 240, 0 }, - { 3804, 242, 18, 50, 200, 200, 0 }, - { 1203, 82, 19, 82, 200, 176, 0 } - } - }, - { /* FM region 94 */ - { 35, 0, 127 }, 32, 138, 13, 0, - { - { 2786, 116, 67, 34, 204, 184, 0 }, - { 1902, 114, 69, 33, 192, 232, 2 }, - { -3, 178, 69, 33, 188, 232, 2 }, - { 3804, 254, 82, 34, 164, 228, 2 } - } - }, - { /* FM region 95 */ - { 34, 0, 127 }, 48, 135, 238, 0, - { - { -2, 34, 85, 34, 184, 224, 0 }, - { 1, 113, 70, 33, 228, 236, 0 }, - { 2, 19, 85, 34, 156, 224, 2 }, - { -1, 129, 70, 33, 224, 236, 2 } - } - }, - { /* FM region 96 */ - { 50, 0, 127 }, 240, 112, 221, 0, - { - { 3369, 213, 69, 32, 0, 204, 0 }, - { 0, 193, 70, 33, 112, 232, 2 }, - { 0, 145, 69, 34, 244, 208, 2 }, - { -9000, 145, 70, 33, 224, 236, 3 } - } - }, - { /* FM region 97 */ - { 34, 0, 127 }, 96, 122, 168, 0, - { - { -1, 99, 51, 33, 200, 208, 0 }, - { -8500, 81, 83, 33, 232, 240, 3 }, - { 702, 99, 52, 33, 200, 208, 2 }, - { -9500, 65, 83, 34, 224, 240, 3 } - } - }, - { /* FM region 98 */ - { 34, 0, 127 }, 0, 67, 0, 0, - { - { 1500, 217, 55, 151, 20, 224, 2 }, - { 3, 231, 70, 146, 88, 220, 2 }, - { 2369, 115, 148, 151, 32, 196, 2 }, - { -3, 118, 36, 146, 64, 244, 2 } - } - }, - { /* FM region 99 */ - { 34, 0, 127 }, 64, 169, 204, 0, - { - { -3, 228, 69, 34, 148, 220, 0 }, - { -7448, 243, 69, 33, 200, 240, 3 }, - { 3, 81, 68, 34, 212, 212, 2 }, - { -8526, 65, 68, 33, 196, 240, 3 } - } - }, - { /* FM region 100 */ - { 34, 0, 127 }, 64, 119, 187, 0, - { - { 2786, 228, 22, 146, 176, 192, 0 }, - { 3, 102, 68, 146, 196, 236, 2 }, - { 3369, 178, 149, 116, 176, 208, 2 }, - { -3, 231, 68, 146, 120, 240, 0 } - } - }, - { /* FM region 101 */ - { 34, 0, 127 }, 240, 144, 239, 0, - { - { -2, 49, 69, 34, 236, 208, 2 }, - { -9000, 113, 102, 33, 228, 236, 3 }, - { 2400, 149, 69, 34, 12, 216, 1 }, - { 0, 145, 102, 33, 224, 236, 2 } - } - }, - { /* FM region 102 */ - { 50, 0, 127 }, 241, 176, 6, 0, - { - { 1200, 247, 49, 64, 252, 204, 0 }, - { 3804, 246, 101, 32, 0, 232, 2 }, - { 1902, 247, 32, 32, 112, 188, 2 }, - { 0, 228, 84, 32, 0, 240, 2 } - } - }, - { /* FM region 103 */ - { 37, 0, 127 }, 64, 101, 221, 0, - { - { 1, 194, 68, 97, 196, 200, 2 }, - { -10001, 247, 100, 114, 176, 240, 3 }, - { 3370, 213, 33, 70, 52, 200, 2 }, - { -1, 178, 68, 49, 208, 212, 0 } - } - }, - { /* FM region 104 */ - { 34, 0, 127 }, 0, 255, 203, 0, - { - { -3, 245, 82, 99, 200, 232, 2 }, - { 2787, 244, 84, 96, 0, 236, 2 }, - { 1198, 133, 81, 100, 196, 220, 2 }, - { 1902, 147, 67, 80, 0, 232, 2 } - } - }, - { /* FM region 105 */ - { 37, 0, 127 }, 0, 255, 140, 0, - { - { 500, 255, 137, 179, 0, 200, 3 }, - { 1902, 248, 90, 160, 0, 244, 2 }, - { 3804, 245, 57, 35, 164, 204, 2 }, - { 0, 245, 38, 51, 196, 208, 2 } - } - }, - { /* FM region 106 */ - { 37, 0, 127 }, 0, 255, 72, 0, - { - { 1000, 238, 57, 65, 0, 188, 3 }, - { 1902, 247, 103, 112, 0, 244, 2 }, - { 2786, 250, 36, 81, 68, 212, 2 }, - { 0, 249, 50, 49, 172, 204, 2 } - } - }, - { /* FM region 107 */ - { 37, 0, 127 }, 16, 119, 72, 0, - { - { 1500, 255, 89, 65, 0, 196, 3 }, - { 2790, 246, 39, 112, 0, 240, 0 }, - { 1905, 246, 36, 81, 168, 208, 0 }, - { 0, 249, 114, 49, 172, 212, 0 } - } - }, - { /* FM region 108 */ - { 37, 0, 127 }, 0, 255, 237, 0, - { - { 1902, 254, 89, 65, 0, 212, 2 }, - { 0, 248, 87, 112, 0, 240, 2 }, - { 3369, 231, 62, 81, 0, 208, 2 }, - { 3, 245, 118, 49, 96, 196, 2 } - } - }, - { /* FM region 109 */ - { 34, 0, 127 }, 16, 188, 205, 0, - { - { -2, 179, 47, 50, 244, 224, 2 }, - { 1900, 145, 94, 49, 248, 232, 2 }, - { 3, 210, 46, 2, 244, 208, 2 }, - { 2789, 133, 93, 4, 180, 244, 2 } - } - }, - { /* FM region 110 */ - { 37, 0, 127 }, 48, 135, 220, 0, - { - { 1901, 162, 25, 35, 144, 208, 0 }, - { 0, 113, 105, 65, 220, 240, 0 }, - { 3369, 233, 88, 51, 120, 212, 0 }, - { 0, 229, 24, 84, 200, 208, 0 } - } - }, - { /* FM region 111 */ - { 34, 0, 127 }, 112, 32, 190, 0, - { - { 0, 53, 79, 66, 152, 212, 2 }, - { 1200, 53, 75, 64, 136, 244, 2 }, - { 500, 149, 60, 66, 16, 208, 2 }, - { 1902, 200, 78, 64, 0, 248, 0 } - } - }, - { /* FM region 112 */ - { 37, 0, 127 }, 0, 144, 130, 0, - { - { 2514, 255, 68, 53, 0, 204, 2 }, - { 2400, 247, 133, 48, 0, 240, 2 }, - { 4151, 243, 67, 50, 0, 212, 2 }, - { 3369, 243, 66, 56, 0, 204, 2 } - } - }, - { /* FM region 113 */ - { 37, 0, 127 }, 0, 255, 0, 0, - { - { 514, 253, 79, 51, 0, 196, 3 }, - { 1905, 252, 89, 51, 0, 244, 2 }, - { 4349, 245, 35, 51, 0, 208, 2 }, - { 1205, 247, 34, 51, 0, 208, 2 } - } - }, - { /* FM region 114 */ - { 37, 0, 127 }, 0, 255, 0, 0, - { - { 514, 221, 69, 35, 0, 204, 3 }, - { 0, 250, 86, 115, 0, 252, 2 }, - { 1884, 244, 116, 51, 0, 200, 2 }, - { 1208, 210, 35, 51, 0, 208, 2 } - } - }, - { /* FM region 115 */ - { 37, 0, 127 }, 0, 255, 16, 0, - { - { 514, 222, 85, 163, 0, 192, 3 }, - { 0, 254, 108, 163, 0, 252, 2 }, - { 3800, 255, 143, 160, 0, 176, 2 }, - { 1200, 250, 105, 163, 0, 212, 2 } - } - }, - { /* FM region 116 */ - { 37, 0, 127 }, 0, 255, 16, 0, - { - { 1514, 249, 101, 163, 0, 204, 3 }, - { -1200, 249, 87, 160, 0, 252, 2 }, - { 0, 235, 143, 160, 0, 204, 2 }, - { 1200, 234, 73, 163, 0, 204, 2 } - } - }, - { /* FM region 117 */ - { 37, 0, 127 }, 0, 255, 16, 0, - { - { 500, 239, 101, 160, 0, 204, 3 }, - { -1195, 248, 104, 160, 0, 252, 2 }, - { 1898, 252, 72, 163, 0, 216, 2 }, - { 1239, 248, 87, 163, 0, 196, 2 } - } - }, - { /* FM region 118 */ - { 37, 0, 127 }, 0, 255, 255, 0, - { - { 500, 255, 98, 160, 0, 196, 3 }, - { -1, 249, 105, 160, 0, 252, 2 }, - { 1907, 250, 71, 160, 0, 252, 2 }, - { 1182, 249, 87, 161, 0, 192, 2 } - } - }, - { /* FM region 119 */ - { 37, 0, 127 }, 0, 0, 100, 0, - { - { 600, 32, 15, 0, 252, 224, 6 }, - { 0, 47, 111, 65, 0, 244, 2 }, - { 1826, 16, 47, 0, 252, 216, 2 }, - { 3551, 240, 47, 0, 252, 212, 2 } - } - }, - { /* FM region 120 */ - { 52, 0, 127 }, 240, 128, 235, 0, - { - { 1228, 161, 47, 17, 196, 200, 3 }, - { 3000, 123, 75, 17, 0, 240, 2 }, - { 7022, 72, 43, 17, 0, 216, 0 }, - { 4000, 150, 79, 17, 48, 196, 3 } - } - }, - { /* FM region 121 */ - { 37, 0, 127 }, 224, 16, 86, 0, - { - { 275, 251, 6, 0, 36, 200, 2 }, - { 0, 101, 104, 65, 56, 240, 2 }, - { 0, 240, 6, 0, 252, 208, 6 }, - { 1000, 195, 8, 0, 248, 200, 2 } - } - }, - { /* FM region 122 */ - { 34, 0, 127 }, 0, 0, 185, 0, - { - { 600, 35, 66, 17, 72, 224, 4 }, - { -13000, 81, 67, 17, 228, 244, 2 }, - { 702, 97, 38, 17, 212, 196, 6 }, - { -14000, 81, 65, 17, 224, 244, 3 } - } - }, - { /* FM region 123 */ - { 50, 0, 127 }, 240, 112, 237, 0, - { - { -6528, 153, 127, 16, 0, 252, 3 }, - { 1200, 105, 109, 16, 0, 216, 2 }, - { -6022, 179, 139, 17, 0, 248, 3 }, - { 2000, 104, 79, 17, 0, 240, 0 } - } - }, - { /* FM region 124 */ - { 50, 0, 127 }, 240, 240, 16, 0, - { - { 1914, 240, 64, 160, 240, 208, 2 }, - { 1200, 240, 73, 163, 240, 244, 0 }, - { 1900, 240, 64, 160, 240, 148, 2 }, - { 4151, 240, 73, 163, 240, 244, 0 } - } - }, - { /* FM region 125 */ - { 34, 0, 127 }, 240, 56, 235, 0, - { - { -5522, 97, 32, 17, 196, 240, 3 }, - { 0, 84, 75, 17, 180, 248, 3 }, - { 702, 65, 38, 17, 224, 212, 6 }, - { -4000, 161, 73, 17, 224, 252, 1 } - } - }, - { /* FM region 126 */ - { 53, 0, 127 }, 240, 248, 37, 0, - { - { 1050, 243, 0, 0, 252, 224, 7 }, - { 2000, 49, 68, 0, 224, 236, 3 }, - { 350, 240, 0, 0, 252, 216, 1 }, - { 700, 240, 0, 0, 252, 212, 3 } - } - }, - { /* FM region 127 */ - { 53, 0, 127 }, 240, 248, 37, 0, - { - { 1050, 245, 85, 0, 0, 244, 7 }, - { -5000, 247, 71, 0, 0, 252, 3 }, - { 350, 240, 0, 0, 0, 164, 0 }, - { 700, 32, 0, 0, 0, 252, 2 } - } - }}; /* end FM Regions */ - -/*---------------------------------------------------------------------------- - * Programs - *---------------------------------------------------------------------------- -*/ -const S_PROGRAM eas_programs[] = -{ - { 7864320, 0 } /* program 0 */ -}; /* end Programs */ - -/*---------------------------------------------------------------------------- - * Banks - *---------------------------------------------------------------------------- -*/ -const S_BANK eas_banks[] = -{ - { /* bank 0 */ - 30976, - { - 32768, 32769, 32770, 32771, 32772, 32773, 32774, 32775, - 32776, 32777, 32778, 32779, 32780, 32781, 32782, 32783, - 32784, 32785, 32786, 32787, 32788, 32789, 32790, 32791, - 32792, 32793, 32794, 32795, 32796, 32797, 32798, 32799, - 32800, 32801, 32802, 32803, 32804, 32805, 32806, 32807, - 32808, 32809, 32810, 32811, 32812, 32813, 32814, 32815, - 32816, 32817, 32818, 32819, 32820, 32821, 32822, 32823, - 32824, 32825, 32826, 32827, 32828, 32829, 32830, 32831, - 32832, 32833, 32834, 32835, 32836, 32837, 32838, 32839, - 32840, 32841, 32842, 32843, 32844, 32845, 32846, 32847, - 32848, 32849, 32850, 32851, 32852, 32853, 32854, 32855, - 32856, 32857, 32858, 32859, 32860, 32861, 32862, 32863, - 32864, 32865, 32866, 32867, 32868, 32869, 32870, 32871, - 32872, 32873, 32874, 32875, 32876, 32877, 32878, 32879, - 32880, 32881, 32882, 32883, 32884, 32885, 32886, 32887, - 32888, 32889, 32890, 32891, 32892, 32893, 32894, 32895 - } - } -}; /* end Banks */ - -/*---------------------------------------------------------------------------- - * Samples - *---------------------------------------------------------------------------- -*/ - -const EAS_SAMPLE eas_samples[] = -{ - 13, -24, 28, -32, 33, -37, 39, -61, 119, -76, 120, -70, 99, -122, 89, -113, - 91, -123, 122, -123, 77, 86, -116, 6, -118, 123, -23, 64, -93, 17, 24, -125, - 124, -125, 124, -24, -12, 56, 87, -54, 38, -91, 64, -2, -41, 126, -127, 20, - 8, -48, -62, 127, -128, 88, -43, -18, 86, -100, 44, -32, -26, 71, -13, 6, - 51, -33, -50, 106, -59, 33, 5, -20, 69, -56, 54, -48, -6, 38, -5, -38, - 35, -1, 12, -1, 4, 23, -56, 19, -7, 5, -4, 31, -38, 3, -12, -9, - -25, -7, 24, -32, 17, -21, -22, 24, -20, 6, -18, 0, -15, 5, -16, 15, - 3, 2, -10, 3, 27, -31, 37, -12, 32, -5, 2, -21, 27, -14, 20, -8, - 5, 15, -7, -11, 20, -37, 7, -23, 17, -22, 11, -14, 8, -44, 39, -52, - 26, 3, 14, -10, 51, -45, 29, -19, 19, -3, 41, -21, 46, -23, 10, 16, - 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37, -7, 122, 72, -57, -71, -88, -28, 48, 127, 36, -30, -53, -126, -16, 71, - 92, 51, -1, -83, -128, 14, 43, 83, 86, -14, -106, -87, -7, 28, 1, 122, - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 0, 0, 0, 0 -}; - -const EAS_U32 eas_sampleLengths[] = -{ - 9922, 9338, 6793, 5242, 3057, 2818, 2806, 1835, - 1603, 1262, 1227, 1168, 1132, 1132, 994, 907, - 817, 801, 583, 418, 402, 387, 387, 357, - 347, 212, 167, 40, 40, 32, 30 -}; - -const EAS_U32 eas_sampleOffsets[] = -{ - 0x00000000, 0x000026c2, 0x00004b3c, 0x000065c5, 0x00007a3f, 0x00008630, 0x00009132, 0x00009c28, - 0x0000a353, 0x0000a996, 0x0000ae84, 0x0000b34f, 0x0000b7df, 0x0000bc4b, 0x0000c0b7, 0x0000c499, - 0x0000c824, 0x0000cb55, 0x0000ce76, 0x0000d0bd, 0x0000d25f, 0x0000d3f1, 0x0000d574, 0x0000d6f7, - 0x0000d85c, 0x0000d9b7, 0x0000da8b, 0x0000db32, 0x0000db5a, 0x0000db82, 0x0000dba2 -}; - -/*---------------------------------------------------------------------------- - * S_EAS (hybrid) - *---------------------------------------------------------------------------- -*/ -const S_EAS easSoundLib = -{ - 0x01534145, - 0x00105622, - eas_banks, - eas_programs, - eas_regions, - eas_articulations, - eas_sampleLengths, - eas_sampleOffsets, - eas_samples, - eas_fmRegions, - 1, - 1, - 61, - 53, - 31, - 128 -}; /* end S_EAS */ - -/*---------------------------------------------------------------------------- - * Statistics - * - * Number of banks: 1 - * Number of programs: 1 - * Number of regions: 61 - * Number of articulations: 53 - * Number of samples: 31 - * Size of sample pool: 56276 - *---------------------------------------------------------------------------- -*/ -/* end C:\Sonic\Trunk\EASLib\WTLibrary\hybrid_22khz_mcu.c */ -- cgit v1.2.3