/*----------------------------------------------------------------------------
*
* 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} <avoid multiply for performance>*/
gainIncrement = (pWTIntFrame->frame.gainTarget - pWTIntFrame->prevGain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS);
if (gainIncrement < 0)
gainIncrement++;
/*lint -e{703} <avoid multiply for performance>*/
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} <avoid divide>*/
tmp2 = gain >> 16;
/* scale sample by gain */
tmp2 *= tmp0;
/* stereo output */
#if (NUM_OUTPUT_CHANNELS == 2)
/*lint -e{704} <avoid divide>*/
tmp2 = tmp2 >> 14;
/* get the current sample in the final mix buffer */
tmp1 = *pMixBuffer;
/* left channel */
tmp0 = tmp2 * gainLeft;
/*lint -e{704} <avoid divide>*/
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} <avoid divide>*/
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} <avoid divide>*/
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} <avoid multiply for performance>*/
samp1 = pSamples[0] << 8;
/*lint -e{701} <avoid multiply for performance>*/
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} <avoid divide>*/
acc0 = samp1 + (acc0 >> NUM_PHASE_FRAC_BITS);
/* save new output sample in buffer */
/*lint -e{704} <avoid divide>*/
*pOutputBuffer++ = (EAS_I16)(acc0 >> 2);
/* increment phase */
phaseFrac += phaseInc;
/*lint -e{704} <avoid divide>*/
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} <avoid multiply for performance>*/
samp1 = pSamples[0] << 8;
/*lint -e{701} <avoid multiply for performance>*/
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} <avoid multiply for performance>*/
samp1 = pSamples[0] << 8;
/*lint -e{701} <avoid multiply for performance>*/
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} <avoid divide>*/
acc0 = samp1 + (acc0 >> NUM_PHASE_FRAC_BITS);
/* save new output sample in buffer */
/*lint -e{704} <avoid divide>*/
*pOutputBuffer++ = (EAS_I16)(acc0 >> 2);
/* increment phase */
phaseFrac += phaseInc;
/*lint -e{704} <avoid divide>*/
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} <avoid multiply for performance>*/
samp1 = pSamples[0] << 8;
/*lint -e{701} <avoid multiply for performance>*/
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} <avoid divide> */
b2 = -pWTIntFrame->frame.b2 >> 1;
/*lint -e{702} <avoid divide> */
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} <avoid divide> */
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} <avoid divide for performance>*/
tmp0 = (EAS_I32) (pWTVoice->phaseAccum) >> 18;
/*lint -e{704} <avoid divide for performance>*/
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} <avoid divide>*/
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