arm-hybrid-22k/lib_src/eas_wtengine.c

/*----------------------------------------------------------------------------
 *
 * 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