diff options
author | Jean-Baptiste Queru <jbq@google.com> | 2009-11-12 18:45:37 -0800 |
---|---|---|
committer | Jean-Baptiste Queru <jbq@google.com> | 2009-11-12 18:45:37 -0800 |
commit | a8c89077d78769bf4840fa91609edc51fe2fa02d (patch) | |
tree | 568e82009b7d4e9f69435e8b1503ebdd8d5cb7c3 /arm-fm-22k/lib_src/eas_fmengine.c | |
parent | 30eeb03a677a957edf0be3e3ccbf5d6d016ec23d (diff) | |
download | android_external_sonivox-a8c89077d78769bf4840fa91609edc51fe2fa02d.tar.gz android_external_sonivox-a8c89077d78769bf4840fa91609edc51fe2fa02d.tar.bz2 android_external_sonivox-a8c89077d78769bf4840fa91609edc51fe2fa02d.zip |
eclair snapshot
Diffstat (limited to 'arm-fm-22k/lib_src/eas_fmengine.c')
-rw-r--r-- | arm-fm-22k/lib_src/eas_fmengine.c | 1546 |
1 files changed, 773 insertions, 773 deletions
diff --git a/arm-fm-22k/lib_src/eas_fmengine.c b/arm-fm-22k/lib_src/eas_fmengine.c index 9c3da66..ea7f69c 100644 --- a/arm-fm-22k/lib_src/eas_fmengine.c +++ b/arm-fm-22k/lib_src/eas_fmengine.c @@ -1,12 +1,12 @@ -/*----------------------------------------------------------------------------
- *
- * File:
- * eas_fmengine.c
- *
- * Contents and purpose:
- * Implements the low-level FM synthesizer functions.
- *
- * Copyright Sonic Network Inc. 2004, 2005
+/*---------------------------------------------------------------------------- + * + * File: + * eas_fmengine.c + * + * Contents and purpose: + * Implements the low-level FM synthesizer functions. + * + * 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. @@ -19,767 +19,767 @@ * 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_types.h"
-#include "eas_math.h"
-#include "eas_audioconst.h"
-#include "eas_fmengine.h"
-
-#if defined(EAS_FM_SYNTH) || defined(EAS_HYBRID_SYNTH) || defined(EAS_SPLIT_HYBRID_SYNTH) || defined(EAS_SPLIT_FM_SYNTH)
-#include "eas_data.h"
-#endif
-
-/* externals */
-extern const EAS_I16 sineTable[];
-extern const EAS_U8 fmScaleTable[16];
-
-// saturation constants for 32-bit to 16-bit conversion
-#define _EAS_MAX_OUTPUT 32767
-#define _EAS_MIN_OUTPUT -32767
-
-static S_FM_ENG_VOICE voices[NUM_FM_VOICES];
-
-/* local prototypes */
-void FM_SynthMixVoice (S_FM_ENG_VOICE *p, EAS_U16 gainTarget, EAS_I32 numSamplesToAdd, EAS_PCM *pInputBuffer, EAS_I32 *pBuffer);
-
-/* used in development environment */
-#if defined(_SATURATION_MONITOR)
-static EAS_BOOL bSaturated = EAS_FALSE;
-
-/*----------------------------------------------------------------------------
- * FM_CheckSaturation()
- *----------------------------------------------------------------------------
- * Purpose:
- * Allows the sound development tool to check for saturation at the voice
- * level. Useful for tuning the level controls.
- *
- * Inputs:
- *
- * Outputs:
- * Returns true if saturation has occurred since the last time the function
- * was called.
- *
- * Side Effects:
- * Resets the saturation flag
- *----------------------------------------------------------------------------
-*/
-EAS_BOOL FM_CheckSaturation ()
-{
- EAS_BOOL bTemp;
- bTemp = bSaturated;
- bSaturated = EAS_FALSE;
- return bTemp;
-}
-#endif
-
-/*----------------------------------------------------------------------------
- * FM_Saturate()
- *----------------------------------------------------------------------------
- * Purpose:
- * This inline function saturates a 32-bit number to 16-bits
- *
- * Inputs:
- * psEASData - pointer to overall EAS data structure
- *
- * Outputs:
- * Returns a 16-bit integer
- *----------------------------------------------------------------------------
-*/
-EAS_INLINE EAS_I16 FM_Saturate (EAS_I32 nValue)
-{
- if (nValue > _EAS_MAX_OUTPUT)
- {
-#if defined(_SATURATION_MONITOR)
- bSaturated = EAS_TRUE;
-#endif
- return _EAS_MAX_OUTPUT;
- }
- if (nValue < _EAS_MIN_OUTPUT)
- {
-#if defined(_SATURATION_MONITOR)
- bSaturated = EAS_TRUE;
-#endif
- return _EAS_MIN_OUTPUT;
- }
- return (EAS_I16) nValue;
-}
-
-/*----------------------------------------------------------------------------
- * FM_Noise()
- *----------------------------------------------------------------------------
- * Purpose:
- * A 31-bit low-cost linear congruential PRNG algorithm used to
- * generate noise.
- *
- * Inputs:
- * pnSeed - pointer to 32-bit PRNG seed
- *
- * Outputs:
- * Returns a 16-bit integer
- *----------------------------------------------------------------------------
-*/
-EAS_INLINE EAS_I16 FM_Noise (EAS_U32 *pnSeed)
-{
- *pnSeed = *pnSeed * 214013L + 2531011L;
- return (EAS_I16) ((*pnSeed >> 15) & 0xffff);
-}
-
-/*----------------------------------------------------------------------------
- * FM_PhaseInc()
- *----------------------------------------------------------------------------
- * Purpose:
- * Transform pitch cents to linear phase increment
- *
- * Inputs:
- * nCents - measured in cents
- * psEASData - pointer to overall EAS data structure
- *
- * Outputs:
- * nResult - int.frac result (where frac has NUM_DENTS_FRAC_BITS)
- *
- * Side Effects:
- *
- *----------------------------------------------------------------------------
-*/
-static EAS_I32 FM_PhaseInc (EAS_I32 nCents)
-{
- EAS_I32 nDents;
- EAS_I32 nExponentInt, nExponentFrac;
- EAS_I32 nTemp1, nTemp2;
- EAS_I32 nResult;
-
- /* convert cents to dents */
- nDents = FMUL_15x15(nCents, CENTS_TO_DENTS);
- nExponentInt = GET_DENTS_INT_PART(nDents) + (32 - SINE_TABLE_SIZE_IN_BITS - NUM_EG1_FRAC_BITS);
- nExponentFrac = GET_DENTS_FRAC_PART(nDents);
-
- /* implement 2^(fracPart) as a power series */
- nTemp1 = GN2_TO_X2 + MULT_DENTS_COEF(nExponentFrac, GN2_TO_X3);
- nTemp2 = GN2_TO_X1 + MULT_DENTS_COEF(nExponentFrac, nTemp1);
- nTemp1 = GN2_TO_X0 + MULT_DENTS_COEF(nExponentFrac, nTemp2);
-
- /*
- implement 2^(intPart) as
- a left shift for intPart >= 0 or
- a left shift for intPart < 0
- */
- if (nExponentInt >= 0)
- {
- /* left shift for positive exponents */
- /*lint -e{703} <avoid multiply for performance>*/
- nResult = nTemp1 << nExponentInt;
- }
- else
- {
- /* right shift for negative exponents */
- nExponentInt = -nExponentInt;
- nResult = nTemp1 >> nExponentInt;
- }
-
- return nResult;
-}
-
-#if (NUM_OUTPUT_CHANNELS == 2)
-/*----------------------------------------------------------------------------
- * FM_CalculatePan()
- *----------------------------------------------------------------------------
- * Purpose:
- * Assign the left and right gain values corresponding to the given pan value.
- *
- * Inputs:
- * psVoice - ptr to the voice we have assigned for this channel
- * psArticulation - ptr to this voice's articulation
- * psEASData - pointer to overall EAS data structure
- *
- * Outputs:
- *
- * Side Effects:
- * the given voice's m_nGainLeft and m_nGainRight are assigned
- *----------------------------------------------------------------------------
-*/
-static void FM_CalculatePan (EAS_I16 pan, EAS_U16 *pGainLeft, EAS_U16 *pGainRight)
-{
- EAS_I32 nTemp;
- EAS_INT nNetAngle;
-
- /*
- Implement the following
- sin(x) = (2-4*c)*x^2 + c + x
- cos(x) = (2-4*c)*x^2 + c - x
-
- where c = 1/sqrt(2)
- using the a0 + x*(a1 + x*a2) approach
- */
-
- /*
- Get the Midi CC10 pan value for this voice's channel
- convert the pan value to an "angle" representation suitable for
- our sin, cos calculator. This representation is NOT necessarily the same
- as the transform in the GM manuals because of our sin, cos calculator.
- "angle" = (CC10 - 64)/128
- */
- /*lint -e{703} <avoid multiply for performance reasons>*/
- nNetAngle = ((EAS_I32) pan) << (NUM_EG1_FRAC_BITS -7);
-
- /* calculate sin */
- nTemp = EG1_ONE + FMUL_15x15(COEFF_PAN_G2, nNetAngle);
- nTemp = COEFF_PAN_G0 + FMUL_15x15(nTemp, nNetAngle);
-
- if (nTemp > SYNTH_FULL_SCALE_EG1_GAIN)
- nTemp = SYNTH_FULL_SCALE_EG1_GAIN;
- else if (nTemp < 0)
- nTemp = 0;
-
- *pGainRight = (EAS_U16) nTemp;
-
- /* calculate cos */
- nTemp = -EG1_ONE + FMUL_15x15(COEFF_PAN_G2, nNetAngle);
- nTemp = COEFF_PAN_G0 + FMUL_15x15(nTemp, nNetAngle);
-
- if (nTemp > SYNTH_FULL_SCALE_EG1_GAIN)
- nTemp = SYNTH_FULL_SCALE_EG1_GAIN;
- else if (nTemp < 0)
- nTemp = 0;
-
- *pGainLeft = (EAS_U16) nTemp;
-}
-#endif /* #if (NUM_OUTPUT_CHANNELS == 2) */
-
-/*----------------------------------------------------------------------------
- * FM_Operator()
- *----------------------------------------------------------------------------
- * Purpose:
- * Synthesizes a buffer of samples based on passed parameters.
- *
- * Inputs:
- * nNumSamplesToAdd - number of samples to synthesize
- * psEASData - pointer to overall EAS data structure
- *
- * Outputs:
- *
- * Side Effects:
- *
- *----------------------------------------------------------------------------
-*/
-void FM_Operator (
- S_FM_ENG_OPER *p,
- EAS_I32 numSamplesToAdd,
- EAS_PCM *pBuffer,
- EAS_PCM *pModBuffer,
- EAS_BOOL mix,
- EAS_U16 gainTarget,
- EAS_I16 pitch,
- EAS_U8 feedback,
- EAS_I16 *pLastOutput)
-{
- EAS_I32 gain;
- EAS_I32 gainInc;
- EAS_U32 phase;
- EAS_U32 phaseInc;
- EAS_U32 phaseTemp;
- EAS_I32 temp;
- EAS_I32 temp2;
-
- /* establish local gain variable */
- gain = (EAS_I32) p->gain << 16;
-
- /* calculate gain increment */
- /*lint -e{703} use shift for performance */
- gainInc = ((EAS_I32) gainTarget - (EAS_I32) p->gain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS);
-
- /* establish local phase variables */
- phase = p->phase;
-
- /* calculate the new phase increment */
- phaseInc = (EAS_U32) FM_PhaseInc(pitch);
-
- /* restore final output from previous frame for feedback loop */
- if (pLastOutput)
- temp = *pLastOutput;
- else
- temp = 0;
-
- /* generate a buffer of samples */
- while (numSamplesToAdd--)
- {
-
- /* incorporate modulation */
- if (pModBuffer)
- {
- /*lint -e{701} use shift for performance */
- temp = *pModBuffer++ << FM_MODULATOR_INPUT_SHIFT;
- }
-
- /* incorporate feedback */
- else
- {
- /*lint -e{703} use shift for performance */
- temp = (temp * (EAS_I32) feedback) << FM_FEEDBACK_SHIFT;
- }
-
- /*lint -e{737} <use this behavior to avoid extra mask step> */
- phaseTemp = phase + temp;
-
- /* fetch sample from wavetable */
- temp = sineTable[phaseTemp >> (32 - SINE_TABLE_SIZE_IN_BITS)];
-
- /* increment operator phase */
- phase += phaseInc;
-
- /* internal gain for modulation effects */
- temp = FMUL_15x15(temp, (gain >> 16));
-
- /* output gain calculation */
- temp2 = FMUL_15x15(temp, p->outputGain);
-
- /* saturating add to buffer */
- if (mix)
- {
- temp2 += *pBuffer;
- *pBuffer++ = FM_Saturate(temp2);
- }
-
- /* output to buffer */
- else
- *pBuffer++ = (EAS_I16) temp2;
-
- /* increment gain */
- gain += gainInc;
-
- }
-
- /* save phase and gain */
- p->phase = phase;
- p->gain = gainTarget;
-
- /* save last output for feedback in next frame */
- if (pLastOutput)
- *pLastOutput = (EAS_I16) temp;
-}
-
-/*----------------------------------------------------------------------------
- * FM_NoiseOperator()
- *----------------------------------------------------------------------------
- * Purpose:
- * Synthesizes a buffer of samples based on passed parameters.
- *
- * Inputs:
- * nNumSamplesToAdd - number of samples to synthesize
- * psEASData - pointer to overall EAS data structure
- *
- * Outputs:
- *
- * Side Effects:
- *
- *----------------------------------------------------------------------------
-*/
-void FM_NoiseOperator (
- S_FM_ENG_OPER *p,
- EAS_I32 numSamplesToAdd,
- EAS_PCM *pBuffer,
- EAS_BOOL mix,
- EAS_U16 gainTarget,
- EAS_U8 feedback,
- EAS_I16 *pLastOutput)
-{
- EAS_I32 gain;
- EAS_I32 gainInc;
- EAS_U32 phase;
- EAS_I32 temp;
- EAS_I32 temp2;
-
- /* establish local gain variable */
- gain = (EAS_I32) p->gain << 16;
-
- /* calculate gain increment */
- /*lint -e{703} use shift for performance */
- gainInc = ((EAS_I32) gainTarget - (EAS_I32) p->gain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS);
-
- /* establish local phase variables */
- phase = p->phase;
-
- /* establish local phase variables */
- phase = p->phase;
-
- /* recall last sample for filter Z-1 term */
- temp = 0;
- if (pLastOutput)
- temp = *pLastOutput;
-
- /* generate a buffer of samples */
- while (numSamplesToAdd--)
- {
-
- /* if using filter */
- if (pLastOutput)
- {
- /* use PRNG for noise */
- temp2 = FM_Noise(&phase);
-
- /*lint -e{704} use shift for performance */
- temp += ((temp2 -temp) * feedback) >> 8;
- }
- else
- {
- temp = FM_Noise(&phase);
- }
-
- /* internal gain for modulation effects */
- temp2 = FMUL_15x15(temp, (gain >> 16));
-
- /* output gain calculation */
- temp2 = FMUL_15x15(temp2, p->outputGain);
-
- /* saturating add to buffer */
- if (mix)
- {
- temp2 += *pBuffer;
- *pBuffer++ = FM_Saturate(temp2);
- }
-
- /* output to buffer */
- else
- *pBuffer++ = (EAS_I16) temp2;
-
- /* increment gain */
- gain += gainInc;
-
- }
-
- /* save phase and gain */
- p->phase = phase;
- p->gain = gainTarget;
-
- /* save last output for feedback in next frame */
- if (pLastOutput)
- *pLastOutput = (EAS_I16) temp;
-}
-
-/*----------------------------------------------------------------------------
- * FM_ConfigVoice()
- *----------------------------------------------------------------------------
- * Purpose:
- * Receives parameters to start a new voice.
- *
- * Inputs:
- * voiceNum - voice number to start
- * vCfg - configuration data
- * pMixBuffer - pointer to host supplied buffer
- *
- * Outputs:
- *
- * Side Effects:
- *
- * Notes:
- * pFrameBuffer is not used in the test version, but is passed as a
- * courtesy to split architecture implementations. It can be used as
- * as pointer to the interprocessor communications buffer when the
- * synthesis parameters are passed off to a DSP for synthesis.
- *----------------------------------------------------------------------------
-*/
-/*lint -esym(715, pFrameBuffer) pFrameBuffer not used in test version - see above */
-void FM_ConfigVoice (EAS_I32 voiceNum, S_FM_VOICE_CONFIG *vCfg, EAS_FRAME_BUFFER_HANDLE pFrameBuffer)
-{
- S_FM_ENG_VOICE *pVoice;
- EAS_INT i;
-
- /* establish pointer to voice data */
- pVoice = &voices[voiceNum];
-
- /* save data */
- pVoice->feedback = vCfg->feedback;
- pVoice->flags = vCfg->flags;
- pVoice->voiceGain = vCfg->voiceGain;
-
- /* initialize Z-1 terms */
- pVoice->op1Out = 0;
- pVoice->op3Out = 0;
-
- /* initialize operators */
- for (i = 0; i < 4; i++)
- {
- /* save operator data */
- pVoice->oper[i].gain = vCfg->gain[i];
- pVoice->oper[i].outputGain = vCfg->outputGain[i];
- pVoice->oper[i].outputGain = vCfg->outputGain[i];
-
- /* initalize operator */
- pVoice->oper[i].phase = 0;
- }
-
- /* calculate pan */
-#if NUM_OUTPUT_CHANNELS == 2
- FM_CalculatePan(vCfg->pan, &pVoice->gainLeft, &pVoice->gainRight);
-#endif
-}
-
-/*----------------------------------------------------------------------------
- * FM_ProcessVoice()
- *----------------------------------------------------------------------------
- * Purpose:
- * Synthesizes a buffer of samples based on calculated parameters.
- *
- * Inputs:
- * nNumSamplesToAdd - number of samples to synthesize
- * psEASData - pointer to overall EAS data structure
- *
- * Outputs:
- *
- * Side Effects:
- *
- * Notes:
- * pOut is not used in the test version, but is passed as a
- * courtesy to split architecture implementations. It can be used as
- * as pointer to the interprocessor communications buffer when the
- * synthesis parameters are passed off to a DSP for synthesis.
- *----------------------------------------------------------------------------
-*/
-/*lint -esym(715, pOut) pOut not used in test version - see above */
-void FM_ProcessVoice (
- EAS_I32 voiceNum,
- S_FM_VOICE_FRAME *pFrame,
- EAS_I32 numSamplesToAdd,
- EAS_PCM *pTempBuffer,
- EAS_PCM *pBuffer,
- EAS_I32 *pMixBuffer,
- EAS_FRAME_BUFFER_HANDLE pFrameBuffer)
-{
- S_FM_ENG_VOICE *p;
- EAS_PCM *pOutBuf;
- EAS_PCM *pMod;
- EAS_BOOL mix;
- EAS_U8 feedback1;
- EAS_U8 feedback3;
- EAS_U8 mode;
-
- /* establish pointer to voice data */
- p = &voices[voiceNum];
- mode = p->flags & 0x07;
-
- /* lookup feedback values */
- feedback1 = fmScaleTable[p->feedback >> 4];
- feedback3 = fmScaleTable[p->feedback & 0x0f];
-
- /* operator 3 is on output bus in modes 0, 1, and 3 */
- if ((mode == 0) || (mode == 1) || (mode == 3))
- pOutBuf = pBuffer;
- else
- pOutBuf = pTempBuffer;
-
- if (p->flags & FLAG_FM_ENG_VOICE_OP3_NOISE)
- {
- FM_NoiseOperator(
- p->oper + 2,
- numSamplesToAdd,
- pOutBuf,
- EAS_FALSE,
- pFrame->gain[2],
- feedback3,
- &p->op3Out);
- }
- else
- {
- FM_Operator(
- p->oper + 2,
- numSamplesToAdd,
- pOutBuf,
- 0,
- EAS_FALSE,
- pFrame->gain[2],
- pFrame->pitch[2],
- feedback3,
- &p->op3Out);
- }
-
- /* operator 4 is on output bus in modes 0, 1, and 2 */
- if (mode < 3)
- pOutBuf = pBuffer;
- else
- pOutBuf = pTempBuffer;
-
- /* operator 4 is modulated in modes 2, 4, and 5 */
- if ((mode == 2) || (mode == 4) || (mode == 5))
- pMod = pTempBuffer;
- else
- pMod = 0;
-
- /* operator 4 is in mix mode in modes 0 and 1 */
- mix = (mode < 2);
-
- if (p->flags & FLAG_FM_ENG_VOICE_OP4_NOISE)
- {
- FM_NoiseOperator(
- p->oper + 3,
- numSamplesToAdd,
- pOutBuf,
- mix,
- pFrame->gain[3],
- 0,
- 0);
- }
- else
- {
- FM_Operator(
- p->oper + 3,
- numSamplesToAdd,
- pOutBuf,
- pMod,
- mix,
- pFrame->gain[3],
- pFrame->pitch[3],
- 0,
- 0);
- }
-
- /* operator 1 is on output bus in mode 0 */
- if (mode == 0)
- pOutBuf = pBuffer;
- else
- pOutBuf = pTempBuffer;
-
- /* operator 1 is modulated in modes 3 and 4 */
- if ((mode == 3) || (mode == 4))
- pMod = pTempBuffer;
- else
- pMod = 0;
-
- /* operator 1 is in mix mode in modes 0 and 5 */
- mix = ((mode == 0) || (mode == 5));
-
- if (p->flags & FLAG_FM_ENG_VOICE_OP1_NOISE)
- {
- FM_NoiseOperator(
- p->oper,
- numSamplesToAdd,
- pOutBuf,
- mix,
- pFrame->gain[0],
- feedback1,
- &p->op1Out);
- }
- else
- {
- FM_Operator(
- p->oper,
- numSamplesToAdd,
- pOutBuf,
- pMod,
- mix,
- pFrame->gain[0],
- pFrame->pitch[0],
- feedback1,
- &p->op1Out);
- }
-
- /* operator 2 is modulated in all modes except 0 */
- if (mode != 0)
- pMod = pTempBuffer;
- else
- pMod = 0;
-
- /* operator 1 is in mix mode in modes 0 -3 */
- mix = (mode < 4);
-
- if (p->flags & FLAG_FM_ENG_VOICE_OP2_NOISE)
- {
- FM_NoiseOperator(
- p->oper + 1,
- numSamplesToAdd,
- pBuffer,
- mix,
- pFrame->gain[1],
- 0,
- 0);
- }
- else
- {
- FM_Operator(
- p->oper + 1,
- numSamplesToAdd,
- pBuffer,
- pMod,
- mix,
- pFrame->gain[1],
- pFrame->pitch[1],
- 0,
- 0);
- }
-
- /* mix voice output to synthesizer output buffer */
- FM_SynthMixVoice(p, pFrame->voiceGain, numSamplesToAdd, pBuffer, pMixBuffer);
-}
-
-/*----------------------------------------------------------------------------
- * FM_SynthMixVoice()
- *----------------------------------------------------------------------------
- * Purpose:
- * Mixes the voice output buffer into the final mix using an anti-zipper
- * filter.
- *
- * Inputs:
- * nNumSamplesToAdd - number of samples to synthesize
- * psEASData - pointer to overall EAS data structure
- *
- * Outputs:
- *
- * Side Effects:
- *
- *----------------------------------------------------------------------------
-*/
-void FM_SynthMixVoice(S_FM_ENG_VOICE *p, EAS_U16 nGainTarget, EAS_I32 numSamplesToAdd, EAS_PCM *pInputBuffer, EAS_I32 *pBuffer)
-{
- EAS_I32 nGain;
- EAS_I32 nGainInc;
- EAS_I32 nTemp;
-
- /* restore previous gain */
- /*lint -e{703} <use shift for performance> */
- nGain = (EAS_I32) p->voiceGain << 16;
-
- /* calculate gain increment */
- /*lint -e{703} <use shift for performance> */
- nGainInc = ((EAS_I32) nGainTarget - (EAS_I32) p->voiceGain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS);
-
- /* mix the output buffer */
- while (numSamplesToAdd--)
- {
- /* output gain calculation */
- nTemp = *pInputBuffer++;
-
- /* sum to output buffer */
-#if (NUM_OUTPUT_CHANNELS == 2)
-
- /*lint -e{704} <use shift for performance> */
- nTemp = ((EAS_I32) nTemp * (nGain >> 16)) >> FM_GAIN_SHIFT;
-
- /*lint -e{704} <use shift for performance> */
- {
- EAS_I32 nTemp2;
- nTemp = nTemp >> FM_STEREO_PRE_GAIN_SHIFT;
- nTemp2 = (nTemp * p->gainLeft) >> FM_STEREO_POST_GAIN_SHIFT;
- *pBuffer++ += nTemp2;
- nTemp2 = (nTemp * p->gainRight) >> FM_STEREO_POST_GAIN_SHIFT;
- *pBuffer++ += nTemp2;
- }
-#else
- /*lint -e{704} <use shift for performance> */
- nTemp = ((EAS_I32) nTemp * (nGain >> 16)) >> FM_MONO_GAIN_SHIFT;
- *pBuffer++ += nTemp;
-#endif
-
- /* increment gain for anti-zipper filter */
- nGain += nGainInc;
- }
-
- /* save gain */
- p->voiceGain = nGainTarget;
-}
-
+ * + *---------------------------------------------------------------------------- + * Revision Control: + * $Revision: 795 $ + * $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $ + *---------------------------------------------------------------------------- +*/ + +/* includes */ +#include "eas_types.h" +#include "eas_math.h" +#include "eas_audioconst.h" +#include "eas_fmengine.h" + +#if defined(EAS_FM_SYNTH) || defined(EAS_HYBRID_SYNTH) || defined(EAS_SPLIT_HYBRID_SYNTH) || defined(EAS_SPLIT_FM_SYNTH) +#include "eas_data.h" +#endif + +/* externals */ +extern const EAS_I16 sineTable[]; +extern const EAS_U8 fmScaleTable[16]; + +// saturation constants for 32-bit to 16-bit conversion +#define _EAS_MAX_OUTPUT 32767 +#define _EAS_MIN_OUTPUT -32767 + +static S_FM_ENG_VOICE voices[NUM_FM_VOICES]; + +/* local prototypes */ +void FM_SynthMixVoice (S_FM_ENG_VOICE *p, EAS_U16 gainTarget, EAS_I32 numSamplesToAdd, EAS_PCM *pInputBuffer, EAS_I32 *pBuffer); + +/* used in development environment */ +#if defined(_SATURATION_MONITOR) +static EAS_BOOL bSaturated = EAS_FALSE; + +/*---------------------------------------------------------------------------- + * FM_CheckSaturation() + *---------------------------------------------------------------------------- + * Purpose: + * Allows the sound development tool to check for saturation at the voice + * level. Useful for tuning the level controls. + * + * Inputs: + * + * Outputs: + * Returns true if saturation has occurred since the last time the function + * was called. + * + * Side Effects: + * Resets the saturation flag + *---------------------------------------------------------------------------- +*/ +EAS_BOOL FM_CheckSaturation () +{ + EAS_BOOL bTemp; + bTemp = bSaturated; + bSaturated = EAS_FALSE; + return bTemp; +} +#endif + +/*---------------------------------------------------------------------------- + * FM_Saturate() + *---------------------------------------------------------------------------- + * Purpose: + * This inline function saturates a 32-bit number to 16-bits + * + * Inputs: + * psEASData - pointer to overall EAS data structure + * + * Outputs: + * Returns a 16-bit integer + *---------------------------------------------------------------------------- +*/ +EAS_INLINE EAS_I16 FM_Saturate (EAS_I32 nValue) +{ + if (nValue > _EAS_MAX_OUTPUT) + { +#if defined(_SATURATION_MONITOR) + bSaturated = EAS_TRUE; +#endif + return _EAS_MAX_OUTPUT; + } + if (nValue < _EAS_MIN_OUTPUT) + { +#if defined(_SATURATION_MONITOR) + bSaturated = EAS_TRUE; +#endif + return _EAS_MIN_OUTPUT; + } + return (EAS_I16) nValue; +} + +/*---------------------------------------------------------------------------- + * FM_Noise() + *---------------------------------------------------------------------------- + * Purpose: + * A 31-bit low-cost linear congruential PRNG algorithm used to + * generate noise. + * + * Inputs: + * pnSeed - pointer to 32-bit PRNG seed + * + * Outputs: + * Returns a 16-bit integer + *---------------------------------------------------------------------------- +*/ +EAS_INLINE EAS_I16 FM_Noise (EAS_U32 *pnSeed) +{ + *pnSeed = *pnSeed * 214013L + 2531011L; + return (EAS_I16) ((*pnSeed >> 15) & 0xffff); +} + +/*---------------------------------------------------------------------------- + * FM_PhaseInc() + *---------------------------------------------------------------------------- + * Purpose: + * Transform pitch cents to linear phase increment + * + * Inputs: + * nCents - measured in cents + * psEASData - pointer to overall EAS data structure + * + * Outputs: + * nResult - int.frac result (where frac has NUM_DENTS_FRAC_BITS) + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +static EAS_I32 FM_PhaseInc (EAS_I32 nCents) +{ + EAS_I32 nDents; + EAS_I32 nExponentInt, nExponentFrac; + EAS_I32 nTemp1, nTemp2; + EAS_I32 nResult; + + /* convert cents to dents */ + nDents = FMUL_15x15(nCents, CENTS_TO_DENTS); + nExponentInt = GET_DENTS_INT_PART(nDents) + (32 - SINE_TABLE_SIZE_IN_BITS - NUM_EG1_FRAC_BITS); + nExponentFrac = GET_DENTS_FRAC_PART(nDents); + + /* implement 2^(fracPart) as a power series */ + nTemp1 = GN2_TO_X2 + MULT_DENTS_COEF(nExponentFrac, GN2_TO_X3); + nTemp2 = GN2_TO_X1 + MULT_DENTS_COEF(nExponentFrac, nTemp1); + nTemp1 = GN2_TO_X0 + MULT_DENTS_COEF(nExponentFrac, nTemp2); + + /* + implement 2^(intPart) as + a left shift for intPart >= 0 or + a left shift for intPart < 0 + */ + if (nExponentInt >= 0) + { + /* left shift for positive exponents */ + /*lint -e{703} <avoid multiply for performance>*/ + nResult = nTemp1 << nExponentInt; + } + else + { + /* right shift for negative exponents */ + nExponentInt = -nExponentInt; + nResult = nTemp1 >> nExponentInt; + } + + return nResult; +} + +#if (NUM_OUTPUT_CHANNELS == 2) +/*---------------------------------------------------------------------------- + * FM_CalculatePan() + *---------------------------------------------------------------------------- + * Purpose: + * Assign the left and right gain values corresponding to the given pan value. + * + * Inputs: + * psVoice - ptr to the voice we have assigned for this channel + * psArticulation - ptr to this voice's articulation + * psEASData - pointer to overall EAS data structure + * + * Outputs: + * + * Side Effects: + * the given voice's m_nGainLeft and m_nGainRight are assigned + *---------------------------------------------------------------------------- +*/ +static void FM_CalculatePan (EAS_I16 pan, EAS_U16 *pGainLeft, EAS_U16 *pGainRight) +{ + EAS_I32 nTemp; + EAS_INT nNetAngle; + + /* + Implement the following + sin(x) = (2-4*c)*x^2 + c + x + cos(x) = (2-4*c)*x^2 + c - x + + where c = 1/sqrt(2) + using the a0 + x*(a1 + x*a2) approach + */ + + /* + Get the Midi CC10 pan value for this voice's channel + convert the pan value to an "angle" representation suitable for + our sin, cos calculator. This representation is NOT necessarily the same + as the transform in the GM manuals because of our sin, cos calculator. + "angle" = (CC10 - 64)/128 + */ + /*lint -e{703} <avoid multiply for performance reasons>*/ + nNetAngle = ((EAS_I32) pan) << (NUM_EG1_FRAC_BITS -7); + + /* calculate sin */ + nTemp = EG1_ONE + FMUL_15x15(COEFF_PAN_G2, nNetAngle); + nTemp = COEFF_PAN_G0 + FMUL_15x15(nTemp, nNetAngle); + + if (nTemp > SYNTH_FULL_SCALE_EG1_GAIN) + nTemp = SYNTH_FULL_SCALE_EG1_GAIN; + else if (nTemp < 0) + nTemp = 0; + + *pGainRight = (EAS_U16) nTemp; + + /* calculate cos */ + nTemp = -EG1_ONE + FMUL_15x15(COEFF_PAN_G2, nNetAngle); + nTemp = COEFF_PAN_G0 + FMUL_15x15(nTemp, nNetAngle); + + if (nTemp > SYNTH_FULL_SCALE_EG1_GAIN) + nTemp = SYNTH_FULL_SCALE_EG1_GAIN; + else if (nTemp < 0) + nTemp = 0; + + *pGainLeft = (EAS_U16) nTemp; +} +#endif /* #if (NUM_OUTPUT_CHANNELS == 2) */ + +/*---------------------------------------------------------------------------- + * FM_Operator() + *---------------------------------------------------------------------------- + * Purpose: + * Synthesizes a buffer of samples based on passed parameters. + * + * Inputs: + * nNumSamplesToAdd - number of samples to synthesize + * psEASData - pointer to overall EAS data structure + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +void FM_Operator ( + S_FM_ENG_OPER *p, + EAS_I32 numSamplesToAdd, + EAS_PCM *pBuffer, + EAS_PCM *pModBuffer, + EAS_BOOL mix, + EAS_U16 gainTarget, + EAS_I16 pitch, + EAS_U8 feedback, + EAS_I16 *pLastOutput) +{ + EAS_I32 gain; + EAS_I32 gainInc; + EAS_U32 phase; + EAS_U32 phaseInc; + EAS_U32 phaseTemp; + EAS_I32 temp; + EAS_I32 temp2; + + /* establish local gain variable */ + gain = (EAS_I32) p->gain << 16; + + /* calculate gain increment */ + /*lint -e{703} use shift for performance */ + gainInc = ((EAS_I32) gainTarget - (EAS_I32) p->gain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS); + + /* establish local phase variables */ + phase = p->phase; + + /* calculate the new phase increment */ + phaseInc = (EAS_U32) FM_PhaseInc(pitch); + + /* restore final output from previous frame for feedback loop */ + if (pLastOutput) + temp = *pLastOutput; + else + temp = 0; + + /* generate a buffer of samples */ + while (numSamplesToAdd--) + { + + /* incorporate modulation */ + if (pModBuffer) + { + /*lint -e{701} use shift for performance */ + temp = *pModBuffer++ << FM_MODULATOR_INPUT_SHIFT; + } + + /* incorporate feedback */ + else + { + /*lint -e{703} use shift for performance */ + temp = (temp * (EAS_I32) feedback) << FM_FEEDBACK_SHIFT; + } + + /*lint -e{737} <use this behavior to avoid extra mask step> */ + phaseTemp = phase + temp; + + /* fetch sample from wavetable */ + temp = sineTable[phaseTemp >> (32 - SINE_TABLE_SIZE_IN_BITS)]; + + /* increment operator phase */ + phase += phaseInc; + + /* internal gain for modulation effects */ + temp = FMUL_15x15(temp, (gain >> 16)); + + /* output gain calculation */ + temp2 = FMUL_15x15(temp, p->outputGain); + + /* saturating add to buffer */ + if (mix) + { + temp2 += *pBuffer; + *pBuffer++ = FM_Saturate(temp2); + } + + /* output to buffer */ + else + *pBuffer++ = (EAS_I16) temp2; + + /* increment gain */ + gain += gainInc; + + } + + /* save phase and gain */ + p->phase = phase; + p->gain = gainTarget; + + /* save last output for feedback in next frame */ + if (pLastOutput) + *pLastOutput = (EAS_I16) temp; +} + +/*---------------------------------------------------------------------------- + * FM_NoiseOperator() + *---------------------------------------------------------------------------- + * Purpose: + * Synthesizes a buffer of samples based on passed parameters. + * + * Inputs: + * nNumSamplesToAdd - number of samples to synthesize + * psEASData - pointer to overall EAS data structure + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +void FM_NoiseOperator ( + S_FM_ENG_OPER *p, + EAS_I32 numSamplesToAdd, + EAS_PCM *pBuffer, + EAS_BOOL mix, + EAS_U16 gainTarget, + EAS_U8 feedback, + EAS_I16 *pLastOutput) +{ + EAS_I32 gain; + EAS_I32 gainInc; + EAS_U32 phase; + EAS_I32 temp; + EAS_I32 temp2; + + /* establish local gain variable */ + gain = (EAS_I32) p->gain << 16; + + /* calculate gain increment */ + /*lint -e{703} use shift for performance */ + gainInc = ((EAS_I32) gainTarget - (EAS_I32) p->gain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS); + + /* establish local phase variables */ + phase = p->phase; + + /* establish local phase variables */ + phase = p->phase; + + /* recall last sample for filter Z-1 term */ + temp = 0; + if (pLastOutput) + temp = *pLastOutput; + + /* generate a buffer of samples */ + while (numSamplesToAdd--) + { + + /* if using filter */ + if (pLastOutput) + { + /* use PRNG for noise */ + temp2 = FM_Noise(&phase); + + /*lint -e{704} use shift for performance */ + temp += ((temp2 -temp) * feedback) >> 8; + } + else + { + temp = FM_Noise(&phase); + } + + /* internal gain for modulation effects */ + temp2 = FMUL_15x15(temp, (gain >> 16)); + + /* output gain calculation */ + temp2 = FMUL_15x15(temp2, p->outputGain); + + /* saturating add to buffer */ + if (mix) + { + temp2 += *pBuffer; + *pBuffer++ = FM_Saturate(temp2); + } + + /* output to buffer */ + else + *pBuffer++ = (EAS_I16) temp2; + + /* increment gain */ + gain += gainInc; + + } + + /* save phase and gain */ + p->phase = phase; + p->gain = gainTarget; + + /* save last output for feedback in next frame */ + if (pLastOutput) + *pLastOutput = (EAS_I16) temp; +} + +/*---------------------------------------------------------------------------- + * FM_ConfigVoice() + *---------------------------------------------------------------------------- + * Purpose: + * Receives parameters to start a new voice. + * + * Inputs: + * voiceNum - voice number to start + * vCfg - configuration data + * pMixBuffer - pointer to host supplied buffer + * + * Outputs: + * + * Side Effects: + * + * Notes: + * pFrameBuffer is not used in the test version, but is passed as a + * courtesy to split architecture implementations. It can be used as + * as pointer to the interprocessor communications buffer when the + * synthesis parameters are passed off to a DSP for synthesis. + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pFrameBuffer) pFrameBuffer not used in test version - see above */ +void FM_ConfigVoice (EAS_I32 voiceNum, S_FM_VOICE_CONFIG *vCfg, EAS_FRAME_BUFFER_HANDLE pFrameBuffer) +{ + S_FM_ENG_VOICE *pVoice; + EAS_INT i; + + /* establish pointer to voice data */ + pVoice = &voices[voiceNum]; + + /* save data */ + pVoice->feedback = vCfg->feedback; + pVoice->flags = vCfg->flags; + pVoice->voiceGain = vCfg->voiceGain; + + /* initialize Z-1 terms */ + pVoice->op1Out = 0; + pVoice->op3Out = 0; + + /* initialize operators */ + for (i = 0; i < 4; i++) + { + /* save operator data */ + pVoice->oper[i].gain = vCfg->gain[i]; + pVoice->oper[i].outputGain = vCfg->outputGain[i]; + pVoice->oper[i].outputGain = vCfg->outputGain[i]; + + /* initalize operator */ + pVoice->oper[i].phase = 0; + } + + /* calculate pan */ +#if NUM_OUTPUT_CHANNELS == 2 + FM_CalculatePan(vCfg->pan, &pVoice->gainLeft, &pVoice->gainRight); +#endif +} + +/*---------------------------------------------------------------------------- + * FM_ProcessVoice() + *---------------------------------------------------------------------------- + * Purpose: + * Synthesizes a buffer of samples based on calculated parameters. + * + * Inputs: + * nNumSamplesToAdd - number of samples to synthesize + * psEASData - pointer to overall EAS data structure + * + * Outputs: + * + * Side Effects: + * + * Notes: + * pOut is not used in the test version, but is passed as a + * courtesy to split architecture implementations. It can be used as + * as pointer to the interprocessor communications buffer when the + * synthesis parameters are passed off to a DSP for synthesis. + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pOut) pOut not used in test version - see above */ +void FM_ProcessVoice ( + EAS_I32 voiceNum, + S_FM_VOICE_FRAME *pFrame, + EAS_I32 numSamplesToAdd, + EAS_PCM *pTempBuffer, + EAS_PCM *pBuffer, + EAS_I32 *pMixBuffer, + EAS_FRAME_BUFFER_HANDLE pFrameBuffer) +{ + S_FM_ENG_VOICE *p; + EAS_PCM *pOutBuf; + EAS_PCM *pMod; + EAS_BOOL mix; + EAS_U8 feedback1; + EAS_U8 feedback3; + EAS_U8 mode; + + /* establish pointer to voice data */ + p = &voices[voiceNum]; + mode = p->flags & 0x07; + + /* lookup feedback values */ + feedback1 = fmScaleTable[p->feedback >> 4]; + feedback3 = fmScaleTable[p->feedback & 0x0f]; + + /* operator 3 is on output bus in modes 0, 1, and 3 */ + if ((mode == 0) || (mode == 1) || (mode == 3)) + pOutBuf = pBuffer; + else + pOutBuf = pTempBuffer; + + if (p->flags & FLAG_FM_ENG_VOICE_OP3_NOISE) + { + FM_NoiseOperator( + p->oper + 2, + numSamplesToAdd, + pOutBuf, + EAS_FALSE, + pFrame->gain[2], + feedback3, + &p->op3Out); + } + else + { + FM_Operator( + p->oper + 2, + numSamplesToAdd, + pOutBuf, + 0, + EAS_FALSE, + pFrame->gain[2], + pFrame->pitch[2], + feedback3, + &p->op3Out); + } + + /* operator 4 is on output bus in modes 0, 1, and 2 */ + if (mode < 3) + pOutBuf = pBuffer; + else + pOutBuf = pTempBuffer; + + /* operator 4 is modulated in modes 2, 4, and 5 */ + if ((mode == 2) || (mode == 4) || (mode == 5)) + pMod = pTempBuffer; + else + pMod = 0; + + /* operator 4 is in mix mode in modes 0 and 1 */ + mix = (mode < 2); + + if (p->flags & FLAG_FM_ENG_VOICE_OP4_NOISE) + { + FM_NoiseOperator( + p->oper + 3, + numSamplesToAdd, + pOutBuf, + mix, + pFrame->gain[3], + 0, + 0); + } + else + { + FM_Operator( + p->oper + 3, + numSamplesToAdd, + pOutBuf, + pMod, + mix, + pFrame->gain[3], + pFrame->pitch[3], + 0, + 0); + } + + /* operator 1 is on output bus in mode 0 */ + if (mode == 0) + pOutBuf = pBuffer; + else + pOutBuf = pTempBuffer; + + /* operator 1 is modulated in modes 3 and 4 */ + if ((mode == 3) || (mode == 4)) + pMod = pTempBuffer; + else + pMod = 0; + + /* operator 1 is in mix mode in modes 0 and 5 */ + mix = ((mode == 0) || (mode == 5)); + + if (p->flags & FLAG_FM_ENG_VOICE_OP1_NOISE) + { + FM_NoiseOperator( + p->oper, + numSamplesToAdd, + pOutBuf, + mix, + pFrame->gain[0], + feedback1, + &p->op1Out); + } + else + { + FM_Operator( + p->oper, + numSamplesToAdd, + pOutBuf, + pMod, + mix, + pFrame->gain[0], + pFrame->pitch[0], + feedback1, + &p->op1Out); + } + + /* operator 2 is modulated in all modes except 0 */ + if (mode != 0) + pMod = pTempBuffer; + else + pMod = 0; + + /* operator 1 is in mix mode in modes 0 -3 */ + mix = (mode < 4); + + if (p->flags & FLAG_FM_ENG_VOICE_OP2_NOISE) + { + FM_NoiseOperator( + p->oper + 1, + numSamplesToAdd, + pBuffer, + mix, + pFrame->gain[1], + 0, + 0); + } + else + { + FM_Operator( + p->oper + 1, + numSamplesToAdd, + pBuffer, + pMod, + mix, + pFrame->gain[1], + pFrame->pitch[1], + 0, + 0); + } + + /* mix voice output to synthesizer output buffer */ + FM_SynthMixVoice(p, pFrame->voiceGain, numSamplesToAdd, pBuffer, pMixBuffer); +} + +/*---------------------------------------------------------------------------- + * FM_SynthMixVoice() + *---------------------------------------------------------------------------- + * Purpose: + * Mixes the voice output buffer into the final mix using an anti-zipper + * filter. + * + * Inputs: + * nNumSamplesToAdd - number of samples to synthesize + * psEASData - pointer to overall EAS data structure + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +void FM_SynthMixVoice(S_FM_ENG_VOICE *p, EAS_U16 nGainTarget, EAS_I32 numSamplesToAdd, EAS_PCM *pInputBuffer, EAS_I32 *pBuffer) +{ + EAS_I32 nGain; + EAS_I32 nGainInc; + EAS_I32 nTemp; + + /* restore previous gain */ + /*lint -e{703} <use shift for performance> */ + nGain = (EAS_I32) p->voiceGain << 16; + + /* calculate gain increment */ + /*lint -e{703} <use shift for performance> */ + nGainInc = ((EAS_I32) nGainTarget - (EAS_I32) p->voiceGain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS); + + /* mix the output buffer */ + while (numSamplesToAdd--) + { + /* output gain calculation */ + nTemp = *pInputBuffer++; + + /* sum to output buffer */ +#if (NUM_OUTPUT_CHANNELS == 2) + + /*lint -e{704} <use shift for performance> */ + nTemp = ((EAS_I32) nTemp * (nGain >> 16)) >> FM_GAIN_SHIFT; + + /*lint -e{704} <use shift for performance> */ + { + EAS_I32 nTemp2; + nTemp = nTemp >> FM_STEREO_PRE_GAIN_SHIFT; + nTemp2 = (nTemp * p->gainLeft) >> FM_STEREO_POST_GAIN_SHIFT; + *pBuffer++ += nTemp2; + nTemp2 = (nTemp * p->gainRight) >> FM_STEREO_POST_GAIN_SHIFT; + *pBuffer++ += nTemp2; + } +#else + /*lint -e{704} <use shift for performance> */ + nTemp = ((EAS_I32) nTemp * (nGain >> 16)) >> FM_MONO_GAIN_SHIFT; + *pBuffer++ += nTemp; +#endif + + /* increment gain for anti-zipper filter */ + nGain += nGainInc; + } + + /* save gain */ + p->voiceGain = nGainTarget; +} + |