mirror of
https://github.com/id-Software/DOOM-IOS2.git
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Initial Commit
This commit is contained in:
Jeff Farrand
464
common/embeddedaudiosynthesis/arm-fm-22k/lib_src/eas_mixer.c
Executable file
464
common/embeddedaudiosynthesis/arm-fm-22k/lib_src/eas_mixer.c
Executable file
@@ -0,0 +1,464 @@
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/*----------------------------------------------------------------------------
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*
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* File:
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* eas_mixer.c
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*
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* Contents and purpose:
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* This file contains the critical components of the mix engine that
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* must be optimized for best performance.
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*
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* Copyright Sonic Network Inc. 2005
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*
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*----------------------------------------------------------------------------
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* Revision Control:
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* $Revision: 706 $
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* $Date: 2007-05-31 17:22:51 -0700 (Thu, 31 May 2007) $
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*----------------------------------------------------------------------------
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*/
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//3 dls: This module is in the midst of being converted from a synth
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//3 specific module to a general purpose mix engine
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/*------------------------------------
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* includes
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*------------------------------------
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*/
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#include "eas_data.h"
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#include "eas_host.h"
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#include "eas_math.h"
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#include "eas_mixer.h"
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#include "eas_config.h"
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#include "eas_report.h"
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#ifdef _MAXIMIZER_ENABLED
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EAS_I32 MaximizerProcess (EAS_VOID_PTR pInstData, EAS_I32 *pSrc, EAS_I32 *pDst, EAS_I32 numSamples);
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#endif
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/*------------------------------------
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* defines
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*------------------------------------
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*/
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/* need to boost stereo by ~3dB to compensate for the panner */
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#define STEREO_3DB_GAIN_BOOST 512
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/*----------------------------------------------------------------------------
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* EAS_MixEngineInit()
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*----------------------------------------------------------------------------
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* Purpose:
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* Prepares the mix engine for work, allocates buffers, locates effects modules, etc.
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*
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* Inputs:
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* pEASData - instance data
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* pInstData - pointer to variable to receive instance data handle
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*
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* Outputs:
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*
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* Side Effects:
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*
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*----------------------------------------------------------------------------
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*/
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EAS_RESULT EAS_MixEngineInit (S_EAS_DATA *pEASData)
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{
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/* check Configuration Module for mix buffer allocation */
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if (pEASData->staticMemoryModel)
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pEASData->pMixBuffer = EAS_CMEnumData(EAS_CM_MIX_BUFFER);
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else
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pEASData->pMixBuffer = EAS_HWMalloc(pEASData->hwInstData, BUFFER_SIZE_IN_MONO_SAMPLES * NUM_OUTPUT_CHANNELS * sizeof(EAS_I32));
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if (pEASData->pMixBuffer == NULL)
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{
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{ /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Failed to allocate mix buffer memory\n"); */ }
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return EAS_ERROR_MALLOC_FAILED;
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}
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EAS_HWMemSet((void *)(pEASData->pMixBuffer), 0, BUFFER_SIZE_IN_MONO_SAMPLES * NUM_OUTPUT_CHANNELS * sizeof(EAS_I32));
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return EAS_SUCCESS;
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}
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/*----------------------------------------------------------------------------
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* EAS_MixEnginePrep()
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*----------------------------------------------------------------------------
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* Purpose:
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* Performs prep before synthesize a buffer of audio, such as clearing
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* audio buffers, etc.
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*
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* Inputs:
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* psEASData - pointer to overall EAS data structure
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*
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* Outputs:
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*
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* Side Effects:
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*
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*----------------------------------------------------------------------------
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*/
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void EAS_MixEnginePrep (S_EAS_DATA *pEASData, EAS_I32 numSamples)
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{
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/* clear the mix buffer */
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#if (NUM_OUTPUT_CHANNELS == 2)
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EAS_HWMemSet(pEASData->pMixBuffer, 0, numSamples * (EAS_I32) sizeof(long) * 2);
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#else
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EAS_HWMemSet(pEASData->pMixBuffer, 0, (EAS_I32) numSamples * (EAS_I32) sizeof(long));
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#endif
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/* need to clear other side-chain effect buffers (chorus & reverb) */
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}
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/*----------------------------------------------------------------------------
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* EAS_MixEnginePost
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*----------------------------------------------------------------------------
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* Purpose:
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* This routine does the post-processing after all voices have been
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* synthesized. It calls any sweeteners and does the final mixdown to
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* the output buffer.
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*
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* Inputs:
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*
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* Outputs:
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*
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* Notes:
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*----------------------------------------------------------------------------
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*/
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void EAS_MixEnginePost (S_EAS_DATA *pEASData, EAS_I32 numSamples)
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{
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EAS_U16 gain;
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//3 dls: Need to restore the mix engine metrics
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/* calculate the gain multiplier */
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#ifdef _MAXIMIZER_ENABLED
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if (pEASData->effectsModules[EAS_MODULE_MAXIMIZER].effect)
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{
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EAS_I32 temp;
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temp = MaximizerProcess(pEASData->effectsModules[EAS_MODULE_MAXIMIZER].effectData, pEASData->pMixBuffer, pEASData->pMixBuffer, numSamples);
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temp = (temp * pEASData->masterGain) >> 15;
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if (temp > 32767)
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gain = 32767;
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else
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gain = (EAS_U16) temp;
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}
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else
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gain = (EAS_U16) pEASData->masterGain;
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#else
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gain = (EAS_U16) pEASData->masterGain;
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#endif
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/* Not using all the gain bits for now
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* Reduce the input to the compressor by 6dB to prevent saturation
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*/
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#ifdef _COMPRESSOR_ENABLED
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if (pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effectData)
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gain = gain >> 5;
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else
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gain = gain >> 4;
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#else
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gain = gain >> 4;
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#endif
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/* convert 32-bit mix buffer to 16-bit output format */
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#if (NUM_OUTPUT_CHANNELS == 2)
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SynthMasterGain(pEASData->pMixBuffer, pEASData->pOutputAudioBuffer, gain, (EAS_U16) ((EAS_U16) numSamples * 2));
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#else
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SynthMasterGain(pEASData->pMixBuffer, pEASData->pOutputAudioBuffer, gain, (EAS_U16) numSamples);
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#endif
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#ifdef _ENHANCER_ENABLED
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/* enhancer effect */
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if (pEASData->effectsModules[EAS_MODULE_ENHANCER].effectData)
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(*pEASData->effectsModules[EAS_MODULE_ENHANCER].effect->pfProcess)
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(pEASData->effectsModules[EAS_MODULE_ENHANCER].effectData,
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pEASData->pOutputAudioBuffer,
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pEASData->pOutputAudioBuffer,
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numSamples);
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#endif
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#ifdef _GRAPHIC_EQ_ENABLED
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/* graphic EQ effect */
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if (pEASData->effectsModules[EAS_MODULE_GRAPHIC_EQ].effectData)
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(*pEASData->effectsModules[EAS_MODULE_GRAPHIC_EQ].effect->pfProcess)
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(pEASData->effectsModules[EAS_MODULE_GRAPHIC_EQ].effectData,
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pEASData->pOutputAudioBuffer,
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pEASData->pOutputAudioBuffer,
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numSamples);
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#endif
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#ifdef _COMPRESSOR_ENABLED
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/* compressor effect */
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if (pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effectData)
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(*pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effect->pfProcess)
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(pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effectData,
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pEASData->pOutputAudioBuffer,
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pEASData->pOutputAudioBuffer,
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numSamples);
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#endif
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#ifdef _WOW_ENABLED
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/* WOW requires a 32-bit buffer, borrow the mix buffer and
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* pass it as the destination buffer
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*/
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/*lint -e{740} temporarily passing a parameter through an existing I/F */
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if (pEASData->effectsModules[EAS_MODULE_WOW].effectData)
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(*pEASData->effectsModules[EAS_MODULE_WOW].effect->pfProcess)
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(pEASData->effectsModules[EAS_MODULE_WOW].effectData,
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pEASData->pOutputAudioBuffer,
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(EAS_PCM*) pEASData->pMixBuffer,
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numSamples);
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#endif
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#ifdef _TONECONTROLEQ_ENABLED
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/* ToneControlEQ effect */
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if (pEASData->effectsModules[EAS_MODULE_TONECONTROLEQ].effectData)
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(*pEASData->effectsModules[EAS_MODULE_TONECONTROLEQ].effect->pfProcess)
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(pEASData->effectsModules[EAS_MODULE_TONECONTROLEQ].effectData,
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pEASData->pOutputAudioBuffer,
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pEASData->pOutputAudioBuffer,
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numSamples);
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#endif
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#ifdef _REVERB_ENABLED
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/* Reverb effect */
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if (pEASData->effectsModules[EAS_MODULE_REVERB].effectData)
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(*pEASData->effectsModules[EAS_MODULE_REVERB].effect->pfProcess)
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(pEASData->effectsModules[EAS_MODULE_REVERB].effectData,
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pEASData->pOutputAudioBuffer,
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pEASData->pOutputAudioBuffer,
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numSamples);
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#endif
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#ifdef _CHORUS_ENABLED
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/* Chorus effect */
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if (pEASData->effectsModules[EAS_MODULE_CHORUS].effectData)
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(*pEASData->effectsModules[EAS_MODULE_CHORUS].effect->pfProcess)
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(pEASData->effectsModules[EAS_MODULE_CHORUS].effectData,
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pEASData->pOutputAudioBuffer,
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pEASData->pOutputAudioBuffer,
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numSamples);
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#endif
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}
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#ifndef NATIVE_EAS_KERNEL
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/*----------------------------------------------------------------------------
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* SynthMasterGain
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*----------------------------------------------------------------------------
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* Purpose:
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* Mixes down audio from 32-bit to 16-bit target buffer
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*
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* Inputs:
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*
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* Outputs:
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*
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*----------------------------------------------------------------------------
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*/
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void SynthMasterGain (long *pInputBuffer, EAS_PCM *pOutputBuffer, EAS_U16 nGain, EAS_U16 numSamples) {
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/* loop through the buffer */
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while (numSamples--) {
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long s;
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/* read a sample from the input buffer and add some guard bits */
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s = *pInputBuffer++;
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/* add some guard bits */
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/*lint -e{704} <avoid divide for performance>*/
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s = s >> 7;
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/* apply master gain */
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s *= (long) nGain;
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/* shift to lower 16-bits */
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/*lint -e{704} <avoid divide for performance>*/
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s = s >> 9;
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/* saturate */
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s = SATURATE(s);
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*pOutputBuffer++ = (EAS_PCM)s;
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}
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}
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#endif
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/*----------------------------------------------------------------------------
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* EAS_MixEngineShutdown()
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*----------------------------------------------------------------------------
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* Purpose:
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* Shuts down effects modules and deallocates memory
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*
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* Inputs:
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* pEASData - instance data
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* pInstData - instance data handle
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*
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* Outputs:
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*
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* Side Effects:
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*
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*----------------------------------------------------------------------------
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*/
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EAS_RESULT EAS_MixEngineShutdown (S_EAS_DATA *pEASData)
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{
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/* check Configuration Module for static memory allocation */
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if (!pEASData->staticMemoryModel && (pEASData->pMixBuffer != NULL))
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EAS_HWFree(pEASData->hwInstData, pEASData->pMixBuffer);
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return EAS_SUCCESS;
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}
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#ifdef UNIFIED_MIXER
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#ifndef NATIVE_MIX_STREAM
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/*----------------------------------------------------------------------------
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* EAS_MixStream
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*----------------------------------------------------------------------------
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* Mix a 16-bit stream into a 32-bit buffer
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*
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* pInputBuffer 16-bit input buffer
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* pMixBuffer 32-bit mix buffer
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* numSamples number of samples to mix
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* gainLeft initial gain left or mono
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* gainRight initial gain right
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* gainLeft left gain increment per sample
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* gainRight right gain increment per sample
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* flags bit 0 = stereo source
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* bit 1 = stereo output
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*----------------------------------------------------------------------------
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*/
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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)
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{
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EAS_I32 temp;
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EAS_INT src, dest;
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/* NOTE: There are a lot of optimizations that can be done
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* in the native implementations based on register
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* availability, etc. For example, it may make sense to
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* break this down into 8 separate routines:
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*
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* 1. Mono source to mono output
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* 2. Mono source to stereo output
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* 3. Stereo source to mono output
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* 4. Stereo source to stereo output
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* 5. Mono source to mono output - no gain change
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* 6. Mono source to stereo output - no gain change
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* 7. Stereo source to mono output - no gain change
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* 8. Stereo source to stereo output - no gain change
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*
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* Other possibilities include loop unrolling, skipping
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* a gain calculation every 2 or 4 samples, etc.
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*/
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/* no gain change, use fast loops */
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if ((gainIncLeft == 0) && (gainIncRight == 0))
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{
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switch (flags & (MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT))
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{
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/* mono to mono */
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case 0:
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gainLeft >>= 15;
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for (src = dest = 0; src < numSamples; src++, dest++)
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{
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pMixBuffer[dest] += (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS;
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}
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break;
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/* mono to stereo */
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case MIX_FLAGS_STEREO_OUTPUT:
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gainLeft >>= 15;
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gainRight >>= 15;
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for (src = dest = 0; src < numSamples; src++, dest+=2)
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{
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pMixBuffer[dest] += (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS;
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pMixBuffer[dest+1] += (pInputBuffer[src] * gainRight) >> NUM_MIXER_GUARD_BITS;
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}
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break;
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/* stereo to mono */
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case MIX_FLAGS_STEREO_SOURCE:
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gainLeft >>= 15;
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gainRight >>= 15;
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for (src = dest = 0; src < numSamples; src+=2, dest++)
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{
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temp = (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS;
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temp += ((pInputBuffer[src+1] * gainRight) >> NUM_MIXER_GUARD_BITS);
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pMixBuffer[dest] += temp;
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}
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break;
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/* stereo to stereo */
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case MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT:
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gainLeft >>= 15;
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gainRight >>= 15;
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for (src = dest = 0; src < numSamples; src+=2, dest+=2)
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{
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pMixBuffer[dest] += (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS;
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pMixBuffer[dest+1] += (pInputBuffer[src+1] * gainRight) >> NUM_MIXER_GUARD_BITS;
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}
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break;
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}
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}
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/* gain change - do gain increment */
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else
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{
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switch (flags & (MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT))
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{
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/* mono to mono */
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case 0:
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for (src = dest = 0; src < numSamples; src++, dest++)
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{
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gainLeft += gainIncLeft;
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pMixBuffer[dest] += (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS;
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}
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break;
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/* mono to stereo */
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case MIX_FLAGS_STEREO_OUTPUT:
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for (src = dest = 0; src < numSamples; src++, dest+=2)
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{
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gainLeft += gainIncLeft;
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gainRight += gainIncRight;
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pMixBuffer[dest] += (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS;
|
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pMixBuffer[dest+1] += (pInputBuffer[src] * (gainRight >> 15)) >> NUM_MIXER_GUARD_BITS;
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}
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break;
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/* stereo to mono */
|
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case MIX_FLAGS_STEREO_SOURCE:
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for (src = dest = 0; src < numSamples; src+=2, dest++)
|
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{
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gainLeft += gainIncLeft;
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gainRight += gainIncRight;
|
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temp = (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS;
|
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temp += ((pInputBuffer[src+1] * (gainRight >> 15)) >> NUM_MIXER_GUARD_BITS);
|
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pMixBuffer[dest] += temp;
|
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}
|
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break;
|
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|
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/* stereo to stereo */
|
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case MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT:
|
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for (src = dest = 0; src < numSamples; src+=2, dest+=2)
|
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{
|
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gainLeft += gainIncLeft;
|
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gainRight += gainIncRight;
|
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pMixBuffer[dest] += (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS;
|
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pMixBuffer[dest+1] += (pInputBuffer[src+1] * (gainRight >> 15)) >> NUM_MIXER_GUARD_BITS;
|
||||
}
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
#endif
|
||||
#endif
|
||||
|
||||
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