diff options
Diffstat (limited to 'arm-fm-22k/lib_src/eas_mixer.c')
| -rw-r--r-- | arm-fm-22k/lib_src/eas_mixer.c | 464 |
1 files changed, 464 insertions, 0 deletions
diff --git a/arm-fm-22k/lib_src/eas_mixer.c b/arm-fm-22k/lib_src/eas_mixer.c new file mode 100644 index 0000000..c4a2f9f --- /dev/null +++ b/arm-fm-22k/lib_src/eas_mixer.c @@ -0,0 +1,464 @@ +/*----------------------------------------------------------------------------
+ *
+ * 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} <avoid divide for performance>*/
+ s = s >> 7;
+
+ /* apply master gain */
+ s *= (long) nGain;
+
+ /* shift to lower 16-bits */
+ /*lint -e{704} <avoid divide for performance>*/
+ 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
+
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