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-rw-r--r--arm-fm-22k/lib_src/eas_fmsynth.c910
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diff --git a/arm-fm-22k/lib_src/eas_fmsynth.c b/arm-fm-22k/lib_src/eas_fmsynth.c
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--- a/arm-fm-22k/lib_src/eas_fmsynth.c
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@@ -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} <use shift for performance> */
- pFMVoice->oper[operIndex].outputGain = EAS_LogToLinear16(((EAS_I16) (pRegion->oper[operIndex].gain & 0xfc) - 0xfc) << 7);
-
- /* check for linear velocity flag */
- /*lint -e{703} <use shift for performance> */
- 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} <avoid multiply for performance>*/
- 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} <use shift for performance> */
- 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} <shortcut to turn sawtooth into sine wave> */
- 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} <use shift for performance> */
- 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;
-}
-