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-rw-r--r--arm-wt-22k/lib_src/eas_wtsynth.c1428
1 files changed, 714 insertions, 714 deletions
diff --git a/arm-wt-22k/lib_src/eas_wtsynth.c b/arm-wt-22k/lib_src/eas_wtsynth.c
index 3eadf2d..45cf4b1 100644
--- a/arm-wt-22k/lib_src/eas_wtsynth.c
+++ b/arm-wt-22k/lib_src/eas_wtsynth.c
@@ -79,22 +79,22 @@ extern long statsSampleCount;
*----------------------------------------------------------------------------
*/
-const S_SYNTH_INTERFACE wtSynth =
+const S_SYNTH_INTERFACE wtSynth =
{
- WT_Initialize,
- WT_StartVoice,
- WT_UpdateVoice,
- WT_ReleaseVoice,
- WT_MuteVoice,
- WT_SustainPedal,
- WT_UpdateChannel
+ WT_Initialize,
+ WT_StartVoice,
+ WT_UpdateVoice,
+ WT_ReleaseVoice,
+ WT_MuteVoice,
+ WT_SustainPedal,
+ WT_UpdateChannel
};
#ifdef EAS_SPLIT_WT_SYNTH
const S_FRAME_INTERFACE wtFrameInterface =
{
- WTE_StartFrame,
- WTE_EndFrame
+ WTE_StartFrame,
+ WTE_EndFrame
};
#endif
@@ -112,37 +112,37 @@ const S_FRAME_INTERFACE wtFrameInterface =
*/
static EAS_RESULT WT_Initialize (S_VOICE_MGR *pVoiceMgr)
{
- EAS_INT i;
+ EAS_INT i;
- for (i = 0; i < NUM_WT_VOICES; i++)
- {
+ for (i = 0; i < NUM_WT_VOICES; i++)
+ {
- pVoiceMgr->wtVoices[i].artIndex = DEFAULT_ARTICULATION_INDEX;
+ pVoiceMgr->wtVoices[i].artIndex = DEFAULT_ARTICULATION_INDEX;
- pVoiceMgr->wtVoices[i].eg1State = DEFAULT_EG1_STATE;
- pVoiceMgr->wtVoices[i].eg1Value = DEFAULT_EG1_VALUE;
- pVoiceMgr->wtVoices[i].eg1Increment = DEFAULT_EG1_INCREMENT;
+ pVoiceMgr->wtVoices[i].eg1State = DEFAULT_EG1_STATE;
+ pVoiceMgr->wtVoices[i].eg1Value = DEFAULT_EG1_VALUE;
+ pVoiceMgr->wtVoices[i].eg1Increment = DEFAULT_EG1_INCREMENT;
- pVoiceMgr->wtVoices[i].eg2State = DEFAULT_EG2_STATE;
- pVoiceMgr->wtVoices[i].eg2Value = DEFAULT_EG2_VALUE;
- pVoiceMgr->wtVoices[i].eg2Increment = DEFAULT_EG2_INCREMENT;
+ pVoiceMgr->wtVoices[i].eg2State = DEFAULT_EG2_STATE;
+ pVoiceMgr->wtVoices[i].eg2Value = DEFAULT_EG2_VALUE;
+ pVoiceMgr->wtVoices[i].eg2Increment = DEFAULT_EG2_INCREMENT;
- /* left and right gain values are needed only if stereo output */
-#if (NUM_OUTPUT_CHANNELS == 2)
- pVoiceMgr->wtVoices[i].gainLeft = DEFAULT_VOICE_GAIN;
- pVoiceMgr->wtVoices[i].gainRight = DEFAULT_VOICE_GAIN;
+ /* left and right gain values are needed only if stereo output */
+#if (NUM_OUTPUT_CHANNELS == 2)
+ pVoiceMgr->wtVoices[i].gainLeft = DEFAULT_VOICE_GAIN;
+ pVoiceMgr->wtVoices[i].gainRight = DEFAULT_VOICE_GAIN;
#endif
- pVoiceMgr->wtVoices[i].phaseFrac = DEFAULT_PHASE_FRAC;
- pVoiceMgr->wtVoices[i].phaseAccum = DEFAULT_PHASE_INT;
+ pVoiceMgr->wtVoices[i].phaseFrac = DEFAULT_PHASE_FRAC;
+ pVoiceMgr->wtVoices[i].phaseAccum = DEFAULT_PHASE_INT;
#ifdef _FILTER_ENABLED
- pVoiceMgr->wtVoices[i].filter.z1 = DEFAULT_FILTER_ZERO;
- pVoiceMgr->wtVoices[i].filter.z2 = DEFAULT_FILTER_ZERO;
+ pVoiceMgr->wtVoices[i].filter.z1 = DEFAULT_FILTER_ZERO;
+ pVoiceMgr->wtVoices[i].filter.z2 = DEFAULT_FILTER_ZERO;
#endif
- }
+ }
- return EAS_TRUE;
+ return EAS_TRUE;
}
/*----------------------------------------------------------------------------
@@ -162,20 +162,20 @@ static EAS_RESULT WT_Initialize (S_VOICE_MGR *pVoiceMgr)
/*lint -esym(715, pVoice) used in some implementations */
static void WT_ReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum)
{
- S_WT_VOICE *pWTVoice;
- const S_ARTICULATION *pArticulation;
-
+ S_WT_VOICE *pWTVoice;
+ const S_ARTICULATION *pArticulation;
+
#ifdef DLS_SYNTHESIZER
- if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
- {
- DLS_ReleaseVoice(pVoiceMgr, pSynth, pVoice, voiceNum);
- return;
- }
+ if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
+ {
+ DLS_ReleaseVoice(pVoiceMgr, pSynth, pVoice, voiceNum);
+ return;
+ }
#endif
- pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
- pArticulation = &pSynth->pEAS->pArticulations[pWTVoice->artIndex];
-
+ pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
+ pArticulation = &pSynth->pEAS->pArticulations[pWTVoice->artIndex];
+
/* release EG1 */
pWTVoice->eg1State = eEnvelopeStateRelease;
pWTVoice->eg1Increment = pArticulation->eg1.releaseTime;
@@ -209,20 +209,20 @@ static void WT_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE
{
#ifdef DLS_SYNTHESIZER
- if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
- {
- DLS_MuteVoice(pVoiceMgr, pSynth, pVoice, voiceNum);
- return;
- }
+ if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
+ {
+ DLS_MuteVoice(pVoiceMgr, pSynth, pVoice, voiceNum);
+ return;
+ }
#endif
- /* clear deferred action flags */
- pVoice->voiceFlags &=
- ~(VOICE_FLAG_DEFER_MIDI_NOTE_OFF |
- VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF |
- VOICE_FLAG_DEFER_MUTE);
-
- /* set the envelope state */
+ /* clear deferred action flags */
+ pVoice->voiceFlags &=
+ ~(VOICE_FLAG_DEFER_MIDI_NOTE_OFF |
+ VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF |
+ VOICE_FLAG_DEFER_MUTE);
+
+ /* set the envelope state */
pVoiceMgr->wtVoices[voiceNum].eg1State = eEnvelopeStateMuted;
pVoiceMgr->wtVoices[voiceNum].eg2State = eEnvelopeStateMuted;
}
@@ -243,25 +243,25 @@ static void WT_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE
/*lint -esym(715, pChannel) used in some implementations */
static void WT_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum)
{
- S_WT_VOICE *pWTVoice;
+ S_WT_VOICE *pWTVoice;
#ifdef DLS_SYNTHESIZER
- if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
- {
- DLS_SustainPedal(pVoiceMgr, pSynth, pVoice, pChannel, voiceNum);
- return;
- }
+ if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
+ {
+ DLS_SustainPedal(pVoiceMgr, pSynth, pVoice, pChannel, voiceNum);
+ return;
+ }
#endif
- /* don't catch the voice if below the sustain level */
- pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
- if (pWTVoice->eg1Value < pSynth->pEAS->pArticulations[pWTVoice->artIndex].eg1.sustainLevel)
- return;
-
+ /* don't catch the voice if below the sustain level */
+ pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
+ if (pWTVoice->eg1Value < pSynth->pEAS->pArticulations[pWTVoice->artIndex].eg1.sustainLevel)
+ return;
+
/* sustain flag is set, damper pedal is on */
/* defer releasing this note until the damper pedal is off */
pWTVoice->eg1State = eEnvelopeStateDecay;
- pVoice->voiceState = eVoiceStatePlay;
+ pVoice->voiceState = eVoiceStatePlay;
/*
because sustain pedal is on, this voice
@@ -277,9 +277,9 @@ static void WT_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VO
/*----------------------------------------------------------------------------
* WT_StartVoice()
*----------------------------------------------------------------------------
- * Purpose:
+ * Purpose:
* Assign the region for the given instrument using the midi key number
- * and the RPN2 (coarse tuning) value. By using RPN2 as part of the
+ * and the RPN2 (coarse tuning) value. By using RPN2 as part of the
* region selection process, we reduce the amount a given sample has
* to be transposed by selecting the closest recorded root instead.
*
@@ -290,14 +290,14 @@ static void WT_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VO
* Setup and initialize the following voice parameters:
* m_nRegionIndex
*
- * Inputs:
+ * Inputs:
* pVoice - ptr to the voice we have assigned for this channel
* nRegionIndex - index of the region
* pEASData - pointer to overall EAS data structure
- *
+ *
* Outputs:
* success - could find and assign the region for this voice's note otherwise
- * failure - could not find nor assign the region for this voice's note
+ * failure - could not find nor assign the region for this voice's note
*
* Side Effects:
* psSynthObject->m_sVoice[].m_nRegionIndex is assigned
@@ -306,137 +306,137 @@ static void WT_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VO
*/
static EAS_RESULT WT_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex)
{
- S_WT_VOICE *pWTVoice;
- const S_WT_REGION *pRegion;
- const S_ARTICULATION *pArt;
- S_SYNTH_CHANNEL *pChannel;
-
+ S_WT_VOICE *pWTVoice;
+ const S_WT_REGION *pRegion;
+ const S_ARTICULATION *pArt;
+ S_SYNTH_CHANNEL *pChannel;
+
#if (NUM_OUTPUT_CHANNELS == 2)
- EAS_INT pan;
+ EAS_INT pan;
#endif
#ifdef EAS_SPLIT_WT_SYNTH
- S_WT_CONFIG wtConfig;
+ S_WT_CONFIG wtConfig;
#endif
-
- /* no samples have been synthesized for this note yet */
- pVoice->regionIndex = regionIndex;
- pVoice->voiceFlags = VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET;
- /* get the articulation index for this region */
- pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
- pChannel = &pSynth->channels[pVoice->channel & 15];
+ /* no samples have been synthesized for this note yet */
+ pVoice->regionIndex = regionIndex;
+ pVoice->voiceFlags = VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET;
+
+ /* get the articulation index for this region */
+ pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
+ pChannel = &pSynth->channels[pVoice->channel & 15];
- /* update static channel parameters */
- if (pChannel->channelFlags & CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS)
- WT_UpdateChannel(pVoiceMgr, pSynth, pVoice->channel & 15);
+ /* update static channel parameters */
+ if (pChannel->channelFlags & CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS)
+ WT_UpdateChannel(pVoiceMgr, pSynth, pVoice->channel & 15);
#ifdef DLS_SYNTHESIZER
- if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
- return DLS_StartVoice(pVoiceMgr, pSynth, pVoice, voiceNum, regionIndex);
+ if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
+ return DLS_StartVoice(pVoiceMgr, pSynth, pVoice, voiceNum, regionIndex);
#endif
- pRegion = &(pSynth->pEAS->pWTRegions[regionIndex]);
- pWTVoice->artIndex = pRegion->artIndex;
+ pRegion = &(pSynth->pEAS->pWTRegions[regionIndex]);
+ pWTVoice->artIndex = pRegion->artIndex;
#ifdef _DEBUG_SYNTH
- { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_StartVoice: Voice %ld; Region %d\n", (EAS_I32) (pVoice - pVoiceMgr->voices), regionIndex); */ }
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_StartVoice: Voice %ld; Region %d\n", (EAS_I32) (pVoice - pVoiceMgr->voices), regionIndex); */ }
#endif
- pArt = &pSynth->pEAS->pArticulations[pWTVoice->artIndex];
+ pArt = &pSynth->pEAS->pArticulations[pWTVoice->artIndex];
- /* MIDI note on puts this voice into attack state */
- pWTVoice->eg1State = eEnvelopeStateAttack;
- pWTVoice->eg1Value = 0;
- pWTVoice->eg1Increment = pArt->eg1.attackTime;
- pWTVoice->eg2State = eEnvelopeStateAttack;
- pWTVoice->eg2Value = 0;
- pWTVoice->eg2Increment = pArt->eg2.attackTime;
+ /* MIDI note on puts this voice into attack state */
+ pWTVoice->eg1State = eEnvelopeStateAttack;
+ pWTVoice->eg1Value = 0;
+ pWTVoice->eg1Increment = pArt->eg1.attackTime;
+ pWTVoice->eg2State = eEnvelopeStateAttack;
+ pWTVoice->eg2Value = 0;
+ pWTVoice->eg2Increment = pArt->eg2.attackTime;
- /* init the LFO */
- pWTVoice->modLFO.lfoValue = 0;
- pWTVoice->modLFO.lfoPhase = -pArt->lfoDelay;
+ /* init the LFO */
+ pWTVoice->modLFO.lfoValue = 0;
+ pWTVoice->modLFO.lfoPhase = -pArt->lfoDelay;
- pVoice->gain = 0;
+ pVoice->gain = 0;
#if (NUM_OUTPUT_CHANNELS == 2)
- /*
- 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
- */
- pan = (EAS_INT) pSynth->channels[pVoice->channel & 15].pan - 64;
- pan += pArt->pan;
- EAS_CalcPanControl(pan, &pWTVoice->gainLeft, &pWTVoice->gainRight);
-#endif
-
-#ifdef _FILTER_ENABLED
- /* clear out the filter states */
- pWTVoice->filter.z1 = 0;
- pWTVoice->filter.z2 = 0;
+ /*
+ 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
+ */
+ pan = (EAS_INT) pSynth->channels[pVoice->channel & 15].pan - 64;
+ pan += pArt->pan;
+ EAS_CalcPanControl(pan, &pWTVoice->gainLeft, &pWTVoice->gainRight);
+#endif
+
+#ifdef _FILTER_ENABLED
+ /* clear out the filter states */
+ pWTVoice->filter.z1 = 0;
+ pWTVoice->filter.z2 = 0;
#endif
- /* if this wave is to be generated using noise generator */
- if (pRegion->region.keyGroupAndFlags & REGION_FLAG_USE_WAVE_GENERATOR)
- {
- pWTVoice->phaseAccum = 4574296;
- pWTVoice->loopStart = WT_NOISE_GENERATOR;
- pWTVoice->loopEnd = 4574295;
- }
-
- /* normal sample */
- else
- {
-
+ /* if this wave is to be generated using noise generator */
+ if (pRegion->region.keyGroupAndFlags & REGION_FLAG_USE_WAVE_GENERATOR)
+ {
+ pWTVoice->phaseAccum = 4574296;
+ pWTVoice->loopStart = WT_NOISE_GENERATOR;
+ pWTVoice->loopEnd = 4574295;
+ }
+
+ /* normal sample */
+ else
+ {
+
#ifdef EAS_SPLIT_WT_SYNTH
- if (voiceNum < NUM_PRIMARY_VOICES)
- pWTVoice->phaseAccum = (EAS_U32) pSynth->pEAS->pSamples + pSynth->pEAS->pSampleOffsets[pRegion->waveIndex];
- else
- pWTVoice->phaseAccum = pSynth->pEAS->pSampleOffsets[pRegion->waveIndex];
+ if (voiceNum < NUM_PRIMARY_VOICES)
+ pWTVoice->phaseAccum = (EAS_U32) pSynth->pEAS->pSamples + pSynth->pEAS->pSampleOffsets[pRegion->waveIndex];
+ else
+ pWTVoice->phaseAccum = pSynth->pEAS->pSampleOffsets[pRegion->waveIndex];
#else
- pWTVoice->phaseAccum = (EAS_U32) pSynth->pEAS->pSamples + pSynth->pEAS->pSampleOffsets[pRegion->waveIndex];
+ pWTVoice->phaseAccum = (EAS_U32) pSynth->pEAS->pSamples + pSynth->pEAS->pSampleOffsets[pRegion->waveIndex];
#endif
- if (pRegion->region.keyGroupAndFlags & REGION_FLAG_IS_LOOPED)
- {
- pWTVoice->loopStart = pWTVoice->phaseAccum + pRegion->loopStart;
- pWTVoice->loopEnd = pWTVoice->phaseAccum + pRegion->loopEnd - 1;
- }
- else
- pWTVoice->loopStart = pWTVoice->loopEnd = pWTVoice->phaseAccum + pSynth->pEAS->pSampleLen[pRegion->waveIndex] - 1;
- }
+ if (pRegion->region.keyGroupAndFlags & REGION_FLAG_IS_LOOPED)
+ {
+ pWTVoice->loopStart = pWTVoice->phaseAccum + pRegion->loopStart;
+ pWTVoice->loopEnd = pWTVoice->phaseAccum + pRegion->loopEnd - 1;
+ }
+ else
+ pWTVoice->loopStart = pWTVoice->loopEnd = pWTVoice->phaseAccum + pSynth->pEAS->pSampleLen[pRegion->waveIndex] - 1;
+ }
#ifdef EAS_SPLIT_WT_SYNTH
- /* configure off-chip voices */
- if (voiceNum >= NUM_PRIMARY_VOICES)
- {
- wtConfig.phaseAccum = pWTVoice->phaseAccum;
- wtConfig.loopStart = pWTVoice->loopStart;
- wtConfig.loopEnd = pWTVoice->loopEnd;
- wtConfig.gain = pVoice->gain;
-
+ /* configure off-chip voices */
+ if (voiceNum >= NUM_PRIMARY_VOICES)
+ {
+ wtConfig.phaseAccum = pWTVoice->phaseAccum;
+ wtConfig.loopStart = pWTVoice->loopStart;
+ wtConfig.loopEnd = pWTVoice->loopEnd;
+ wtConfig.gain = pVoice->gain;
+
#if (NUM_OUTPUT_CHANNELS == 2)
- wtConfig.gainLeft = pWTVoice->gainLeft;
- wtConfig.gainRight = pWTVoice->gainRight;
+ wtConfig.gainLeft = pWTVoice->gainLeft;
+ wtConfig.gainRight = pWTVoice->gainRight;
#endif
- WTE_ConfigVoice(voiceNum - NUM_PRIMARY_VOICES, &wtConfig, pVoiceMgr->pFrameBuffer);
- }
+ WTE_ConfigVoice(voiceNum - NUM_PRIMARY_VOICES, &wtConfig, pVoiceMgr->pFrameBuffer);
+ }
#endif
- return EAS_SUCCESS;
+ return EAS_SUCCESS;
}
/*----------------------------------------------------------------------------
* WT_CheckSampleEnd
*----------------------------------------------------------------------------
- * Purpose:
+ * Purpose:
* Check for end of sample and calculate number of samples to synthesize
- *
- * Inputs:
- *
+ *
+ * Inputs:
+ *
* Outputs:
*
* Notes:
@@ -445,53 +445,53 @@ static EAS_RESULT WT_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNT
*/
EAS_BOOL WT_CheckSampleEnd (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame, EAS_BOOL update)
{
- EAS_U32 endPhaseAccum;
- EAS_U32 endPhaseFrac;
- EAS_I32 numSamples;
- EAS_BOOL done = EAS_FALSE;
-
- /* check to see if we hit the end of the waveform this time */
- /*lint -e{703} use shift for performance */
- endPhaseFrac = pWTVoice->phaseFrac + (pWTIntFrame->frame.phaseIncrement << SYNTH_UPDATE_PERIOD_IN_BITS);
- endPhaseAccum = pWTVoice->phaseAccum + GET_PHASE_INT_PART(endPhaseFrac);
- if (endPhaseAccum >= pWTVoice->loopEnd)
- {
- /* calculate how far current ptr is from end */
- numSamples = (EAS_I32) (pWTVoice->loopEnd - pWTVoice->phaseAccum);
-
- /* now account for the fractional portion */
- /*lint -e{703} use shift for performance */
- numSamples = (EAS_I32) ((numSamples << NUM_PHASE_FRAC_BITS) - pWTVoice->phaseFrac);
- if (pWTIntFrame->frame.phaseIncrement) {
- pWTIntFrame->numSamples = 1 + (numSamples / pWTIntFrame->frame.phaseIncrement);
- } else {
- pWTIntFrame->numSamples = numSamples;
- }
-
- /* sound will be done this frame */
- done = EAS_TRUE;
- }
-
- /* update data for off-chip synth */
- if (update)
- {
- pWTVoice->phaseFrac = endPhaseFrac;
- pWTVoice->phaseAccum = endPhaseAccum;
- }
-
- return done;
+ EAS_U32 endPhaseAccum;
+ EAS_U32 endPhaseFrac;
+ EAS_I32 numSamples;
+ EAS_BOOL done = EAS_FALSE;
+
+ /* check to see if we hit the end of the waveform this time */
+ /*lint -e{703} use shift for performance */
+ endPhaseFrac = pWTVoice->phaseFrac + (pWTIntFrame->frame.phaseIncrement << SYNTH_UPDATE_PERIOD_IN_BITS);
+ endPhaseAccum = pWTVoice->phaseAccum + GET_PHASE_INT_PART(endPhaseFrac);
+ if (endPhaseAccum >= pWTVoice->loopEnd)
+ {
+ /* calculate how far current ptr is from end */
+ numSamples = (EAS_I32) (pWTVoice->loopEnd - pWTVoice->phaseAccum);
+
+ /* now account for the fractional portion */
+ /*lint -e{703} use shift for performance */
+ numSamples = (EAS_I32) ((numSamples << NUM_PHASE_FRAC_BITS) - pWTVoice->phaseFrac);
+ if (pWTIntFrame->frame.phaseIncrement) {
+ pWTIntFrame->numSamples = 1 + (numSamples / pWTIntFrame->frame.phaseIncrement);
+ } else {
+ pWTIntFrame->numSamples = numSamples;
+ }
+
+ /* sound will be done this frame */
+ done = EAS_TRUE;
+ }
+
+ /* update data for off-chip synth */
+ if (update)
+ {
+ pWTVoice->phaseFrac = endPhaseFrac;
+ pWTVoice->phaseAccum = endPhaseAccum;
+ }
+
+ return done;
}
/*----------------------------------------------------------------------------
* WT_UpdateVoice()
*----------------------------------------------------------------------------
- * Purpose:
+ * Purpose:
* Synthesize a block of samples for the given voice.
* Use linear interpolation.
*
- * Inputs:
+ * Inputs:
* pEASData - pointer to overall EAS data structure
- *
+ *
* Outputs:
* number of samples actually written to buffer
*
@@ -502,108 +502,108 @@ EAS_BOOL WT_CheckSampleEnd (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame, E
*/
static EAS_BOOL WT_UpdateVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32 numSamples)
{
- S_WT_VOICE *pWTVoice;
- S_WT_INT_FRAME intFrame;
- S_SYNTH_CHANNEL *pChannel;
- const S_WT_REGION *pWTRegion;
- const S_ARTICULATION *pArt;
- EAS_I32 temp;
- EAS_BOOL done;
+ S_WT_VOICE *pWTVoice;
+ S_WT_INT_FRAME intFrame;
+ S_SYNTH_CHANNEL *pChannel;
+ const S_WT_REGION *pWTRegion;
+ const S_ARTICULATION *pArt;
+ EAS_I32 temp;
+ EAS_BOOL done;
#ifdef DLS_SYNTHESIZER
- if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
- return DLS_UpdateVoice(pVoiceMgr, pSynth, pVoice, voiceNum, pMixBuffer, numSamples);
-#endif
+ if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
+ return DLS_UpdateVoice(pVoiceMgr, pSynth, pVoice, voiceNum, pMixBuffer, numSamples);
+#endif
- /* establish pointers to critical data */
- pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
- pWTRegion = &pSynth->pEAS->pWTRegions[pVoice->regionIndex & REGION_INDEX_MASK];
- pArt = &pSynth->pEAS->pArticulations[pWTVoice->artIndex];
- pChannel = &pSynth->channels[pVoice->channel & 15];
- intFrame.prevGain = pVoice->gain;
+ /* establish pointers to critical data */
+ pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
+ pWTRegion = &pSynth->pEAS->pWTRegions[pVoice->regionIndex & REGION_INDEX_MASK];
+ pArt = &pSynth->pEAS->pArticulations[pWTVoice->artIndex];
+ pChannel = &pSynth->channels[pVoice->channel & 15];
+ intFrame.prevGain = pVoice->gain;
- /* update the envelopes */
- WT_UpdateEG1(pWTVoice, &pArt->eg1);
- WT_UpdateEG2(pWTVoice, &pArt->eg2);
+ /* update the envelopes */
+ WT_UpdateEG1(pWTVoice, &pArt->eg1);
+ WT_UpdateEG2(pWTVoice, &pArt->eg2);
- /* update the LFO */
- WT_UpdateLFO(&pWTVoice->modLFO, pArt->lfoFreq);
+ /* update the LFO */
+ WT_UpdateLFO(&pWTVoice->modLFO, pArt->lfoFreq);
#ifdef _FILTER_ENABLED
- /* calculate filter if library uses filter */
- if (pSynth->pEAS->libAttr & LIB_FORMAT_FILTER_ENABLED)
- WT_UpdateFilter(pWTVoice, &intFrame, pArt);
- else
- intFrame.frame.k = 0;
+ /* calculate filter if library uses filter */
+ if (pSynth->pEAS->libAttr & LIB_FORMAT_FILTER_ENABLED)
+ WT_UpdateFilter(pWTVoice, &intFrame, pArt);
+ else
+ intFrame.frame.k = 0;
#endif
- /* update the gain */
- intFrame.frame.gainTarget = WT_UpdateGain(pVoice, pWTVoice, pArt, pChannel, pWTRegion->gain);
+ /* update the gain */
+ intFrame.frame.gainTarget = WT_UpdateGain(pVoice, pWTVoice, pArt, pChannel, pWTRegion->gain);
- /* calculate base pitch*/
- temp = pChannel->staticPitch + pWTRegion->tuning;
+ /* calculate base pitch*/
+ temp = pChannel->staticPitch + pWTRegion->tuning;
- /* include global transpose */
- if (pChannel->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL)
- temp += pVoice->note * 100;
- else
- temp += (pVoice->note + pSynth->globalTranspose) * 100;
- intFrame.frame.phaseIncrement = WT_UpdatePhaseInc(pWTVoice, pArt, pChannel, temp);
+ /* include global transpose */
+ if (pChannel->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL)
+ temp += pVoice->note * 100;
+ else
+ temp += (pVoice->note + pSynth->globalTranspose) * 100;
+ intFrame.frame.phaseIncrement = WT_UpdatePhaseInc(pWTVoice, pArt, pChannel, temp);
- /* call into engine to generate samples */
- intFrame.pAudioBuffer = pVoiceMgr->voiceBuffer;
- intFrame.pMixBuffer = pMixBuffer;
- intFrame.numSamples = numSamples;
+ /* call into engine to generate samples */
+ intFrame.pAudioBuffer = pVoiceMgr->voiceBuffer;
+ intFrame.pMixBuffer = pMixBuffer;
+ intFrame.numSamples = numSamples;
- /* check for end of sample */
- if ((pWTVoice->loopStart != WT_NOISE_GENERATOR) && (pWTVoice->loopStart == pWTVoice->loopEnd))
- done = WT_CheckSampleEnd(pWTVoice, &intFrame, (EAS_BOOL) (voiceNum >= NUM_PRIMARY_VOICES));
- else
- done = EAS_FALSE;
+ /* check for end of sample */
+ if ((pWTVoice->loopStart != WT_NOISE_GENERATOR) && (pWTVoice->loopStart == pWTVoice->loopEnd))
+ done = WT_CheckSampleEnd(pWTVoice, &intFrame, (EAS_BOOL) (voiceNum >= NUM_PRIMARY_VOICES));
+ else
+ done = EAS_FALSE;
- if (intFrame.numSamples < 0) intFrame.numSamples = 0;
+ if (intFrame.numSamples < 0) intFrame.numSamples = 0;
#ifdef EAS_SPLIT_WT_SYNTH
- if (voiceNum < NUM_PRIMARY_VOICES)
- {
-#ifndef _SPLIT_WT_TEST_HARNESS
- WT_ProcessVoice(pWTVoice, &intFrame);
+ if (voiceNum < NUM_PRIMARY_VOICES)
+ {
+#ifndef _SPLIT_WT_TEST_HARNESS
+ WT_ProcessVoice(pWTVoice, &intFrame);
#endif
- }
- else
- WTE_ProcessVoice(voiceNum - NUM_PRIMARY_VOICES, &intFrame.frame, pVoiceMgr->pFrameBuffer);
+ }
+ else
+ WTE_ProcessVoice(voiceNum - NUM_PRIMARY_VOICES, &intFrame.frame, pVoiceMgr->pFrameBuffer);
#else
- WT_ProcessVoice(pWTVoice, &intFrame);
+ WT_ProcessVoice(pWTVoice, &intFrame);
#endif
- /* clear flag */
- pVoice->voiceFlags &= ~VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET;
+ /* clear flag */
+ pVoice->voiceFlags &= ~VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET;
/* if voice has finished, set flag for voice manager */
- if ((pVoice->voiceState != eVoiceStateStolen) && (pWTVoice->eg1State == eEnvelopeStateMuted))
- done = EAS_TRUE;
+ if ((pVoice->voiceState != eVoiceStateStolen) && (pWTVoice->eg1State == eEnvelopeStateMuted))
+ done = EAS_TRUE;
- /* if the update interval has elapsed, then force the current gain to the next
- * gain since we never actually reach the next gain when ramping -- we just get
- * very close to the target gain.
- */
- pVoice->gain = (EAS_I16) intFrame.frame.gainTarget;
+ /* if the update interval has elapsed, then force the current gain to the next
+ * gain since we never actually reach the next gain when ramping -- we just get
+ * very close to the target gain.
+ */
+ pVoice->gain = (EAS_I16) intFrame.frame.gainTarget;
- return done;
-}
+ return done;
+}
/*----------------------------------------------------------------------------
* WT_UpdatePhaseInc()
*----------------------------------------------------------------------------
- * Purpose:
+ * Purpose:
* Calculate the phase increment
*
- * Inputs:
+ * Inputs:
* pVoice - pointer to the voice being updated
* psRegion - pointer to the region
* psArticulation - pointer to the articulation
* nChannelPitchForThisVoice - the portion of the pitch that is fixed for this
- * voice during the duration of this synthesis
+ * voice during the duration of this synthesis
* pEASData - pointer to overall EAS data structure
*
* Outputs:
@@ -614,45 +614,45 @@ static EAS_BOOL WT_UpdateVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH
*/
static EAS_I32 WT_UpdatePhaseInc (S_WT_VOICE *pWTVoice, const S_ARTICULATION *pArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 pitchCents)
{
- EAS_I32 temp;
-
- /*pitchCents due to CC1 = LFO * (CC1 / 128) * DEFAULT_LFO_MOD_WHEEL_TO_PITCH_CENTS */
- temp = MULT_EG1_EG1(DEFAULT_LFO_MOD_WHEEL_TO_PITCH_CENTS,
- ((pChannel->modWheel) << (NUM_EG1_FRAC_BITS -7)));
-
- /* pitchCents due to channel pressure = LFO * (channel pressure / 128) * DEFAULT_LFO_CHANNEL_PRESSURE_TO_PITCH_CENTS */
- temp += MULT_EG1_EG1(DEFAULT_LFO_CHANNEL_PRESSURE_TO_PITCH_CENTS,
- ((pChannel->channelPressure) << (NUM_EG1_FRAC_BITS -7)));
-
- /* now multiply the (channel pressure + CC1) pitch values by the LFO value */
- temp = MULT_EG1_EG1(pWTVoice->modLFO.lfoValue, temp);
-
- /*
- add in the LFO pitch due to
- channel pressure and CC1 along with
- the LFO pitch, the EG2 pitch, and the
- "static" pitch for this voice on this channel
- */
- temp += pitchCents +
- (MULT_EG1_EG1(pWTVoice->eg2Value, pArt->eg2ToPitch)) +
- (MULT_EG1_EG1(pWTVoice->modLFO.lfoValue, pArt->lfoToPitch));
-
- /* convert from cents to linear phase increment */
- return EAS_Calculate2toX(temp);
+ EAS_I32 temp;
+
+ /*pitchCents due to CC1 = LFO * (CC1 / 128) * DEFAULT_LFO_MOD_WHEEL_TO_PITCH_CENTS */
+ temp = MULT_EG1_EG1(DEFAULT_LFO_MOD_WHEEL_TO_PITCH_CENTS,
+ ((pChannel->modWheel) << (NUM_EG1_FRAC_BITS -7)));
+
+ /* pitchCents due to channel pressure = LFO * (channel pressure / 128) * DEFAULT_LFO_CHANNEL_PRESSURE_TO_PITCH_CENTS */
+ temp += MULT_EG1_EG1(DEFAULT_LFO_CHANNEL_PRESSURE_TO_PITCH_CENTS,
+ ((pChannel->channelPressure) << (NUM_EG1_FRAC_BITS -7)));
+
+ /* now multiply the (channel pressure + CC1) pitch values by the LFO value */
+ temp = MULT_EG1_EG1(pWTVoice->modLFO.lfoValue, temp);
+
+ /*
+ add in the LFO pitch due to
+ channel pressure and CC1 along with
+ the LFO pitch, the EG2 pitch, and the
+ "static" pitch for this voice on this channel
+ */
+ temp += pitchCents +
+ (MULT_EG1_EG1(pWTVoice->eg2Value, pArt->eg2ToPitch)) +
+ (MULT_EG1_EG1(pWTVoice->modLFO.lfoValue, pArt->lfoToPitch));
+
+ /* convert from cents to linear phase increment */
+ return EAS_Calculate2toX(temp);
}
/*----------------------------------------------------------------------------
* WT_UpdateChannel()
*----------------------------------------------------------------------------
- * Purpose:
+ * Purpose:
* Calculate and assign static channel parameters
* These values only need to be updated if one of the controller values
* for this channel changes
*
- * Inputs:
+ * Inputs:
* nChannel - channel to update
* pEASData - pointer to overall EAS data structure
- *
+ *
* Outputs:
*
* Side Effects:
@@ -662,59 +662,59 @@ static EAS_I32 WT_UpdatePhaseInc (S_WT_VOICE *pWTVoice, const S_ARTICULATION *pA
/*lint -esym(715, pVoiceMgr) reserved for future use */
static void WT_UpdateChannel (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel)
{
- EAS_I32 staticGain;
- EAS_I32 pitchBend;
- S_SYNTH_CHANNEL *pChannel;
-
- pChannel = &pSynth->channels[channel];
-
- /*
- nChannelGain = (CC7 * CC11)^2 * master volume
- where CC7 == 100 by default, CC11 == 127, master volume == 32767
- */
- staticGain = MULT_EG1_EG1((pChannel->volume) << (NUM_EG1_FRAC_BITS - 7),
- (pChannel->expression) << (NUM_EG1_FRAC_BITS - 7));
-
- /* staticGain has to be squared */
- staticGain = MULT_EG1_EG1(staticGain, staticGain);
-
- pChannel->staticGain = (EAS_I16) MULT_EG1_EG1(staticGain, pSynth->masterVolume);
-
- /*
- calculate pitch bend: RPN0 * ((2*pitch wheel)/16384 -1)
- However, if we use the EG1 macros, remember that EG1 has a full
- scale value of 32768 (instead of 16384). So instead of multiplying
- by 2, multiply by 4 (left shift by 2), and subtract by 32768 instead
- of 16384. This utilizes the fact that the EG1 macro places a binary
- point 15 places to the left instead of 14 places.
- */
- /*lint -e{703} <avoid multiply for performance>*/
- pitchBend =
- (((EAS_I32)(pChannel->pitchBend) << 2)
- - 32768);
-
- pChannel->staticPitch =
- MULT_EG1_EG1(pitchBend, pChannel->pitchBendSensitivity);
-
- /* if this is not a drum channel, then add in the per-channel tuning */
- if (!(pChannel->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL))
- pChannel->staticPitch += pChannel->finePitch + (pChannel->coarsePitch * 100);
-
- /* clear update flag */
- pChannel->channelFlags &= ~CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS;
- return;
+ EAS_I32 staticGain;
+ EAS_I32 pitchBend;
+ S_SYNTH_CHANNEL *pChannel;
+
+ pChannel = &pSynth->channels[channel];
+
+ /*
+ nChannelGain = (CC7 * CC11)^2 * master volume
+ where CC7 == 100 by default, CC11 == 127, master volume == 32767
+ */
+ staticGain = MULT_EG1_EG1((pChannel->volume) << (NUM_EG1_FRAC_BITS - 7),
+ (pChannel->expression) << (NUM_EG1_FRAC_BITS - 7));
+
+ /* staticGain has to be squared */
+ staticGain = MULT_EG1_EG1(staticGain, staticGain);
+
+ pChannel->staticGain = (EAS_I16) MULT_EG1_EG1(staticGain, pSynth->masterVolume);
+
+ /*
+ calculate pitch bend: RPN0 * ((2*pitch wheel)/16384 -1)
+ However, if we use the EG1 macros, remember that EG1 has a full
+ scale value of 32768 (instead of 16384). So instead of multiplying
+ by 2, multiply by 4 (left shift by 2), and subtract by 32768 instead
+ of 16384. This utilizes the fact that the EG1 macro places a binary
+ point 15 places to the left instead of 14 places.
+ */
+ /*lint -e{703} <avoid multiply for performance>*/
+ pitchBend =
+ (((EAS_I32)(pChannel->pitchBend) << 2)
+ - 32768);
+
+ pChannel->staticPitch =
+ MULT_EG1_EG1(pitchBend, pChannel->pitchBendSensitivity);
+
+ /* if this is not a drum channel, then add in the per-channel tuning */
+ if (!(pChannel->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL))
+ pChannel->staticPitch += pChannel->finePitch + (pChannel->coarsePitch * 100);
+
+ /* clear update flag */
+ pChannel->channelFlags &= ~CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS;
+ return;
}
/*----------------------------------------------------------------------------
* WT_UpdateGain()
*----------------------------------------------------------------------------
- * Purpose:
+ * Purpose:
* Calculate and assign static voice parameters as part of WT_UpdateVoice()
*
- * Inputs:
- * pVoice - ptr to the synth voice that we want to synthesize
+ * Inputs:
+ * pVoice - ptr to the synth voice that we want to synthesize
* pEASData - pointer to overall EAS data structure
- *
+ *
* Outputs:
*
* Side Effects:
@@ -724,48 +724,48 @@ static void WT_UpdateChannel (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 ch
*/
static EAS_I32 WT_UpdateGain (S_SYNTH_VOICE *pVoice, S_WT_VOICE *pWTVoice, const S_ARTICULATION *pArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 gain)
{
- EAS_I32 lfoGain;
- EAS_I32 temp;
-
- /*
- If this voice was stolen, then the velocity is actually
- for the new note, not the note that we are currently ramping down.
- So we really shouldn't use this velocity. However, that would require
- more memory to store the velocity value, and the improvement may
- not be sufficient to warrant the added memory.
- */
- /* velocity is fixed at note start for a given voice and must be squared */
- temp = (pVoice->velocity) << (NUM_EG1_FRAC_BITS - 7);
- temp = MULT_EG1_EG1(temp, temp);
-
- /* region gain is fixed as part of the articulation */
- temp = MULT_EG1_EG1(temp, gain);
-
- /* include the channel gain */
- temp = MULT_EG1_EG1(temp, pChannel->staticGain);
-
- /* calculate LFO gain using an approximation for 10^x */
- lfoGain = MULT_EG1_EG1(pWTVoice->modLFO.lfoValue, pArt->lfoToGain);
- lfoGain = MULT_EG1_EG1(lfoGain, LFO_GAIN_TO_CENTS);
-
- /* convert from a dB-like value to linear gain */
- lfoGain = EAS_Calculate2toX(lfoGain);
- temp = MULT_EG1_EG1(temp, lfoGain);
-
- /* calculate the voice's gain */
- temp = (EAS_I16)MULT_EG1_EG1(temp, pWTVoice->eg1Value);
-
- return temp;
+ EAS_I32 lfoGain;
+ EAS_I32 temp;
+
+ /*
+ If this voice was stolen, then the velocity is actually
+ for the new note, not the note that we are currently ramping down.
+ So we really shouldn't use this velocity. However, that would require
+ more memory to store the velocity value, and the improvement may
+ not be sufficient to warrant the added memory.
+ */
+ /* velocity is fixed at note start for a given voice and must be squared */
+ temp = (pVoice->velocity) << (NUM_EG1_FRAC_BITS - 7);
+ temp = MULT_EG1_EG1(temp, temp);
+
+ /* region gain is fixed as part of the articulation */
+ temp = MULT_EG1_EG1(temp, gain);
+
+ /* include the channel gain */
+ temp = MULT_EG1_EG1(temp, pChannel->staticGain);
+
+ /* calculate LFO gain using an approximation for 10^x */
+ lfoGain = MULT_EG1_EG1(pWTVoice->modLFO.lfoValue, pArt->lfoToGain);
+ lfoGain = MULT_EG1_EG1(lfoGain, LFO_GAIN_TO_CENTS);
+
+ /* convert from a dB-like value to linear gain */
+ lfoGain = EAS_Calculate2toX(lfoGain);
+ temp = MULT_EG1_EG1(temp, lfoGain);
+
+ /* calculate the voice's gain */
+ temp = (EAS_I16)MULT_EG1_EG1(temp, pWTVoice->eg1Value);
+
+ return temp;
}
/*----------------------------------------------------------------------------
* WT_UpdateEG1()
*----------------------------------------------------------------------------
- * Purpose:
+ * Purpose:
* Calculate the EG1 envelope for the given voice (but do not update any
* state)
*
- * Inputs:
+ * Inputs:
* pVoice - ptr to the voice whose envelope we want to update
* nVoice - this voice's number - used only for debug
* pEASData - pointer to overall EAS data structure
@@ -779,88 +779,88 @@ static EAS_I32 WT_UpdateGain (S_SYNTH_VOICE *pVoice, S_WT_VOICE *pWTVoice, const
*/
static void WT_UpdateEG1 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv)
{
- EAS_I32 temp;
-
- switch (pWTVoice->eg1State)
- {
- case eEnvelopeStateAttack:
- temp = pWTVoice->eg1Value + pWTVoice->eg1Increment;
-
- /* check if we have reached peak amplitude */
- if (temp >= SYNTH_FULL_SCALE_EG1_GAIN)
- {
- /* limit the volume */
- temp = SYNTH_FULL_SCALE_EG1_GAIN;
-
- /* prepare to move to decay state */
- pWTVoice->eg1State = eEnvelopeStateDecay;
- pWTVoice->eg1Increment = pEnv->decayTime;
- }
-
- break;
-
- /* exponential decay */
- case eEnvelopeStateDecay:
- temp = MULT_EG1_EG1(pWTVoice->eg1Value, pWTVoice->eg1Increment);
-
- /* check if we have reached sustain level */
- if (temp <= pEnv->sustainLevel)
- {
- /* enforce the sustain level */
- temp = pEnv->sustainLevel;
-
- /* if sustain level is zero, skip sustain & release the voice */
- if (temp > 0)
- pWTVoice->eg1State = eEnvelopeStateSustain;
-
- /* move to sustain state */
- else
- pWTVoice->eg1State = eEnvelopeStateMuted;
- }
-
- break;
-
- case eEnvelopeStateSustain:
- return;
-
- case eEnvelopeStateRelease:
- temp = MULT_EG1_EG1(pWTVoice->eg1Value, pWTVoice->eg1Increment);
-
- /* if we hit zero, this voice isn't contributing any audio */
- if (temp <= 0)
- {
- temp = 0;
- pWTVoice->eg1State = eEnvelopeStateMuted;
- }
- break;
-
- /* voice is muted, set target to zero */
- case eEnvelopeStateMuted:
- temp = 0;
- break;
-
- case eEnvelopeStateInvalid:
- default:
- temp = 0;
-#ifdef _DEBUG_SYNTH
- { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_UpdateEG1: error, %d is an unrecognized state\n",
- pWTVoice->eg1State); */ }
+ EAS_I32 temp;
+
+ switch (pWTVoice->eg1State)
+ {
+ case eEnvelopeStateAttack:
+ temp = pWTVoice->eg1Value + pWTVoice->eg1Increment;
+
+ /* check if we have reached peak amplitude */
+ if (temp >= SYNTH_FULL_SCALE_EG1_GAIN)
+ {
+ /* limit the volume */
+ temp = SYNTH_FULL_SCALE_EG1_GAIN;
+
+ /* prepare to move to decay state */
+ pWTVoice->eg1State = eEnvelopeStateDecay;
+ pWTVoice->eg1Increment = pEnv->decayTime;
+ }
+
+ break;
+
+ /* exponential decay */
+ case eEnvelopeStateDecay:
+ temp = MULT_EG1_EG1(pWTVoice->eg1Value, pWTVoice->eg1Increment);
+
+ /* check if we have reached sustain level */
+ if (temp <= pEnv->sustainLevel)
+ {
+ /* enforce the sustain level */
+ temp = pEnv->sustainLevel;
+
+ /* if sustain level is zero, skip sustain & release the voice */
+ if (temp > 0)
+ pWTVoice->eg1State = eEnvelopeStateSustain;
+
+ /* move to sustain state */
+ else
+ pWTVoice->eg1State = eEnvelopeStateMuted;
+ }
+
+ break;
+
+ case eEnvelopeStateSustain:
+ return;
+
+ case eEnvelopeStateRelease:
+ temp = MULT_EG1_EG1(pWTVoice->eg1Value, pWTVoice->eg1Increment);
+
+ /* if we hit zero, this voice isn't contributing any audio */
+ if (temp <= 0)
+ {
+ temp = 0;
+ pWTVoice->eg1State = eEnvelopeStateMuted;
+ }
+ break;
+
+ /* voice is muted, set target to zero */
+ case eEnvelopeStateMuted:
+ temp = 0;
+ break;
+
+ case eEnvelopeStateInvalid:
+ default:
+ temp = 0;
+#ifdef _DEBUG_SYNTH
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_UpdateEG1: error, %d is an unrecognized state\n",
+ pWTVoice->eg1State); */ }
#endif
- break;
+ break;
- }
+ }
- pWTVoice->eg1Value = (EAS_I16) temp;
+ pWTVoice->eg1Value = (EAS_I16) temp;
}
/*----------------------------------------------------------------------------
* WT_UpdateEG2()
*----------------------------------------------------------------------------
- * Purpose:
+ * Purpose:
* Update the EG2 envelope for the given voice
*
- * Inputs:
- * pVoice - ptr to the voice whose envelope we want to update
+ * Inputs:
+ * pVoice - ptr to the voice whose envelope we want to update
* pEASData - pointer to overall EAS data structure
*
* Outputs:
@@ -872,83 +872,83 @@ static void WT_UpdateEG1 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv)
static void WT_UpdateEG2 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv)
{
- EAS_I32 temp;
-
- switch (pWTVoice->eg2State)
- {
- case eEnvelopeStateAttack:
- temp = pWTVoice->eg2Value + pWTVoice->eg2Increment;
-
- /* check if we have reached peak amplitude */
- if (temp >= SYNTH_FULL_SCALE_EG1_GAIN)
- {
- /* limit the volume */
- temp = SYNTH_FULL_SCALE_EG1_GAIN;
-
- /* prepare to move to decay state */
- pWTVoice->eg2State = eEnvelopeStateDecay;
-
- pWTVoice->eg2Increment = pEnv->decayTime;
- }
-
- break;
-
- /* implement linear pitch decay in cents */
- case eEnvelopeStateDecay:
- temp = pWTVoice->eg2Value -pWTVoice->eg2Increment;
-
- /* check if we have reached sustain level */
- if (temp <= pEnv->sustainLevel)
- {
- /* enforce the sustain level */
- temp = pEnv->sustainLevel;
-
- /* prepare to move to sustain state */
- pWTVoice->eg2State = eEnvelopeStateSustain;
- }
- break;
-
- case eEnvelopeStateSustain:
- return;
-
- case eEnvelopeStateRelease:
- temp = pWTVoice->eg2Value - pWTVoice->eg2Increment;
-
- if (temp <= 0)
- {
- temp = 0;
- pWTVoice->eg2State = eEnvelopeStateMuted;
- }
-
- break;
-
- /* voice is muted, set target to zero */
- case eEnvelopeStateMuted:
- temp = 0;
- break;
-
- case eEnvelopeStateInvalid:
- default:
- temp = 0;
-#ifdef _DEBUG_SYNTH
- { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_UpdateEG2: error, %d is an unrecognized state\n",
- pWTVoice->eg2State); */ }
+ EAS_I32 temp;
+
+ switch (pWTVoice->eg2State)
+ {
+ case eEnvelopeStateAttack:
+ temp = pWTVoice->eg2Value + pWTVoice->eg2Increment;
+
+ /* check if we have reached peak amplitude */
+ if (temp >= SYNTH_FULL_SCALE_EG1_GAIN)
+ {
+ /* limit the volume */
+ temp = SYNTH_FULL_SCALE_EG1_GAIN;
+
+ /* prepare to move to decay state */
+ pWTVoice->eg2State = eEnvelopeStateDecay;
+
+ pWTVoice->eg2Increment = pEnv->decayTime;
+ }
+
+ break;
+
+ /* implement linear pitch decay in cents */
+ case eEnvelopeStateDecay:
+ temp = pWTVoice->eg2Value -pWTVoice->eg2Increment;
+
+ /* check if we have reached sustain level */
+ if (temp <= pEnv->sustainLevel)
+ {
+ /* enforce the sustain level */
+ temp = pEnv->sustainLevel;
+
+ /* prepare to move to sustain state */
+ pWTVoice->eg2State = eEnvelopeStateSustain;
+ }
+ break;
+
+ case eEnvelopeStateSustain:
+ return;
+
+ case eEnvelopeStateRelease:
+ temp = pWTVoice->eg2Value - pWTVoice->eg2Increment;
+
+ if (temp <= 0)
+ {
+ temp = 0;
+ pWTVoice->eg2State = eEnvelopeStateMuted;
+ }
+
+ break;
+
+ /* voice is muted, set target to zero */
+ case eEnvelopeStateMuted:
+ temp = 0;
+ break;
+
+ case eEnvelopeStateInvalid:
+ default:
+ temp = 0;
+#ifdef _DEBUG_SYNTH
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_UpdateEG2: error, %d is an unrecognized state\n",
+ pWTVoice->eg2State); */ }
#endif
- break;
- }
-
- pWTVoice->eg2Value = (EAS_I16) temp;
+ break;
+ }
+
+ pWTVoice->eg2Value = (EAS_I16) temp;
}
/*----------------------------------------------------------------------------
* WT_UpdateLFO ()
*----------------------------------------------------------------------------
- * Purpose:
+ * Purpose:
* Calculate the LFO for the given voice
*
- * Inputs:
- * pLFO - ptr to the LFO data
- * phaseInc - phase increment
+ * Inputs:
+ * pLFO - ptr to the LFO data
+ * phaseInc - phase increment
*
* Outputs:
*
@@ -959,36 +959,36 @@ static void WT_UpdateEG2 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv)
void WT_UpdateLFO (S_LFO_CONTROL *pLFO, EAS_I16 phaseInc)
{
- /* To save memory, if m_nPhaseValue is negative, we are in the
- * delay phase, and m_nPhaseValue represents the time left
- * in the delay.
- */
- if (pLFO->lfoPhase < 0)
- {
- pLFO->lfoPhase++;
- return;
- }
-
- /* calculate LFO output from phase value */
- /*lint -e{701} Use shift for performance */
- pLFO->lfoValue = (EAS_I16) (pLFO->lfoPhase << 2);
- /*lint -e{502} <shortcut to turn sawtooth into triangle wave> */
- if ((pLFO->lfoPhase > 0x1fff) && (pLFO->lfoPhase < 0x6000))
- pLFO->lfoValue = ~pLFO->lfoValue;
-
- /* update LFO phase */
- pLFO->lfoPhase = (pLFO->lfoPhase + phaseInc) & 0x7fff;
+ /* To save memory, if m_nPhaseValue is negative, we are in the
+ * delay phase, and m_nPhaseValue represents the time left
+ * in the delay.
+ */
+ if (pLFO->lfoPhase < 0)
+ {
+ pLFO->lfoPhase++;
+ return;
+ }
+
+ /* calculate LFO output from phase value */
+ /*lint -e{701} Use shift for performance */
+ pLFO->lfoValue = (EAS_I16) (pLFO->lfoPhase << 2);
+ /*lint -e{502} <shortcut to turn sawtooth into triangle wave> */
+ if ((pLFO->lfoPhase > 0x1fff) && (pLFO->lfoPhase < 0x6000))
+ pLFO->lfoValue = ~pLFO->lfoValue;
+
+ /* update LFO phase */
+ pLFO->lfoPhase = (pLFO->lfoPhase + phaseInc) & 0x7fff;
}
#ifdef _FILTER_ENABLED
/*----------------------------------------------------------------------------
* WT_UpdateFilter()
*----------------------------------------------------------------------------
- * Purpose:
+ * Purpose:
* Update the Filter parameters
*
- * Inputs:
- * pVoice - ptr to the voice whose filter we want to update
+ * Inputs:
+ * pVoice - ptr to the voice whose filter we want to update
* pEASData - pointer to overall EAS data structure
*
* Outputs:
@@ -999,29 +999,29 @@ void WT_UpdateLFO (S_LFO_CONTROL *pLFO, EAS_I16 phaseInc)
*/
static void WT_UpdateFilter (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pIntFrame, const S_ARTICULATION *pArt)
{
- EAS_I32 cutoff;
+ EAS_I32 cutoff;
- /* no need to calculate filter coefficients if it is bypassed */
- if (pArt->filterCutoff == DEFAULT_EAS_FILTER_CUTOFF_FREQUENCY)
- {
- pIntFrame->frame.k = 0;
- return;
- }
+ /* no need to calculate filter coefficients if it is bypassed */
+ if (pArt->filterCutoff == DEFAULT_EAS_FILTER_CUTOFF_FREQUENCY)
+ {
+ pIntFrame->frame.k = 0;
+ return;
+ }
- /* determine the dynamic cutoff frequency */
- cutoff = MULT_EG1_EG1(pWTVoice->eg2Value, pArt->eg2ToFc);
- cutoff += pArt->filterCutoff;
+ /* determine the dynamic cutoff frequency */
+ cutoff = MULT_EG1_EG1(pWTVoice->eg2Value, pArt->eg2ToFc);
+ cutoff += pArt->filterCutoff;
- /* subtract the A5 offset and the sampling frequency */
- cutoff -= FILTER_CUTOFF_FREQ_ADJUST + A5_PITCH_OFFSET_IN_CENTS;
+ /* subtract the A5 offset and the sampling frequency */
+ cutoff -= FILTER_CUTOFF_FREQ_ADJUST + A5_PITCH_OFFSET_IN_CENTS;
- /* limit the cutoff frequency */
- if (cutoff > FILTER_CUTOFF_MAX_PITCH_CENTS)
- cutoff = FILTER_CUTOFF_MAX_PITCH_CENTS;
- else if (cutoff < FILTER_CUTOFF_MIN_PITCH_CENTS)
- cutoff = FILTER_CUTOFF_MIN_PITCH_CENTS;
+ /* limit the cutoff frequency */
+ if (cutoff > FILTER_CUTOFF_MAX_PITCH_CENTS)
+ cutoff = FILTER_CUTOFF_MAX_PITCH_CENTS;
+ else if (cutoff < FILTER_CUTOFF_MIN_PITCH_CENTS)
+ cutoff = FILTER_CUTOFF_MIN_PITCH_CENTS;
- WT_SetFilterCoeffs(pIntFrame, cutoff, pArt->filterQ);
+ WT_SetFilterCoeffs(pIntFrame, cutoff, pArt->filterQ);
}
#endif
@@ -1060,7 +1060,7 @@ static void WT_UpdateFilter (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pIntFrame, co
* n1g2*k^2
* n1g3*k^3
*
- * NOTE that n1g0 == n1g1 == 0, always, so we only need to store n1g2 and n1g3
+ * NOTE that n1g0 == n1g1 == 0, always, so we only need to store n1g2 and n1g3
*----------------------------------------------------------------------------
*/
@@ -1068,154 +1068,154 @@ static const EAS_I16 nk1g0 = -32768;
static const EAS_I16 nk1g2 = 1580;
static const EAS_I16 k2g0 = 32767;
-static const EAS_I16 k2g1[] =
+static const EAS_I16 k2g1[] =
{
- -11324, /* k2g1[0] = -0.3455751918948761 */
- -10387, /* k2g1[1] = -0.3169878073928751 */
- -9528, /* k2g1[2] = -0.29076528753345476 */
- -8740, /* k2g1[3] = -0.2667120011011279 */
- -8017, /* k2g1[4] = -0.24464850028971705 */
- -7353, /* k2g1[5] = -0.22441018194495696 */
- -6745, /* k2g1[6] = -0.20584605955455101 */
- -6187, /* k2g1[7] = -0.18881763682420102 */
- -5675, /* k2g1[8] = -0.1731978744360067 */
- -5206, /* k2g1[9] = -0.15887024228080968 */
- -4775, /* k2g1[10] = -0.14572785009373057 */
- -4380, /* k2g1[11] = -0.13367265000706827 */
- -4018, /* k2g1[12] = -0.1226147050712642 */
- -3685, /* k2g1[13] = -0.11247151828678581 */
- -3381, /* k2g1[14] = -0.10316741714122014 */
- -3101, /* k2g1[15] = -0.0946329890599603 */
- -2844, /* k2g1[16] = -0.08680456355870586 */
- -2609, /* k2g1[17] = -0.07962373723441349 */
- -2393, /* k2g1[18] = -0.07303693805092666 */
- -2195, /* k2g1[19] = -0.06699502566866912 */
- -2014, /* k2g1[20] = -0.06145292483669077 */
- -1847, /* k2g1[21] = -0.056369289112013346 */
- -1694, /* k2g1[22] = -0.05170619239747895 */
- -1554, /* k2g1[23] = -0.04742884599684141 */
- -1426, /* k2g1[24] = -0.043505339076210514 */
- -1308, /* k2g1[25] = -0.03990640059558053 */
- -1199, /* k2g1[26] = -0.03660518093435039 */
- -1100, /* k2g1[27] = -0.03357705158166837 */
- -1009, /* k2g1[28] = -0.030799421397205727 */
- -926, /* k2g1[29] = -0.028251568071585884 */
- -849 /* k2g1[30] = -0.025914483529091967 */
+ -11324, /* k2g1[0] = -0.3455751918948761 */
+ -10387, /* k2g1[1] = -0.3169878073928751 */
+ -9528, /* k2g1[2] = -0.29076528753345476 */
+ -8740, /* k2g1[3] = -0.2667120011011279 */
+ -8017, /* k2g1[4] = -0.24464850028971705 */
+ -7353, /* k2g1[5] = -0.22441018194495696 */
+ -6745, /* k2g1[6] = -0.20584605955455101 */
+ -6187, /* k2g1[7] = -0.18881763682420102 */
+ -5675, /* k2g1[8] = -0.1731978744360067 */
+ -5206, /* k2g1[9] = -0.15887024228080968 */
+ -4775, /* k2g1[10] = -0.14572785009373057 */
+ -4380, /* k2g1[11] = -0.13367265000706827 */
+ -4018, /* k2g1[12] = -0.1226147050712642 */
+ -3685, /* k2g1[13] = -0.11247151828678581 */
+ -3381, /* k2g1[14] = -0.10316741714122014 */
+ -3101, /* k2g1[15] = -0.0946329890599603 */
+ -2844, /* k2g1[16] = -0.08680456355870586 */
+ -2609, /* k2g1[17] = -0.07962373723441349 */
+ -2393, /* k2g1[18] = -0.07303693805092666 */
+ -2195, /* k2g1[19] = -0.06699502566866912 */
+ -2014, /* k2g1[20] = -0.06145292483669077 */
+ -1847, /* k2g1[21] = -0.056369289112013346 */
+ -1694, /* k2g1[22] = -0.05170619239747895 */
+ -1554, /* k2g1[23] = -0.04742884599684141 */
+ -1426, /* k2g1[24] = -0.043505339076210514 */
+ -1308, /* k2g1[25] = -0.03990640059558053 */
+ -1199, /* k2g1[26] = -0.03660518093435039 */
+ -1100, /* k2g1[27] = -0.03357705158166837 */
+ -1009, /* k2g1[28] = -0.030799421397205727 */
+ -926, /* k2g1[29] = -0.028251568071585884 */
+ -849 /* k2g1[30] = -0.025914483529091967 */
};
-static const EAS_I16 k2g2[] =
+static const EAS_I16 k2g2[] =
{
- 1957, /* k2g2[0] = 0.059711106626580836 */
- 1646, /* k2g2[1] = 0.05024063501786333 */
- 1385, /* k2g2[2] = 0.042272226217199664 */
- 1165, /* k2g2[3] = 0.03556764576567844 */
- 981, /* k2g2[4] = 0.029926444346999134 */
- 825, /* k2g2[5] = 0.025179964880280382 */
- 694, /* k2g2[6] = 0.02118630011706455 */
- 584, /* k2g2[7] = 0.01782604998793514 */
- 491, /* k2g2[8] = 0.014998751854573014 */
- 414, /* k2g2[9] = 0.012619876941179595 */
- 348, /* k2g2[10] = 0.010618303146468736 */
- 293, /* k2g2[11] = 0.008934188679954682 */
- 246, /* k2g2[12] = 0.007517182949855368 */
- 207, /* k2g2[13] = 0.006324921212866403 */
- 174, /* k2g2[14] = 0.005321757979794424 */
- 147, /* k2g2[15] = 0.004477701309210577 */
- 123, /* k2g2[16] = 0.00376751612730811 */
- 104, /* k2g2[17] = 0.0031699697655869644 */
- 87, /* k2g2[18] = 0.00266719715992703 */
- 74, /* k2g2[19] = 0.0022441667321724647 */
- 62, /* k2g2[20] = 0.0018882309854916855 */
- 52, /* k2g2[21] = 0.0015887483774966232 */
- 44, /* k2g2[22] = 0.0013367651661223448 */
- 37, /* k2g2[23] = 0.0011247477162958733 */
- 31, /* k2g2[24] = 0.0009463572640678758 */
- 26, /* k2g2[25] = 0.0007962604042473498 */
- 22, /* k2g2[26] = 0.0006699696356181593 */
- 18, /* k2g2[27] = 0.0005637091964589207 */
- 16, /* k2g2[28] = 0.00047430217920125243 */
- 13, /* k2g2[29] = 0.00039907554925166274 */
- 11 /* k2g2[30] = 0.00033578022828973666 */
+ 1957, /* k2g2[0] = 0.059711106626580836 */
+ 1646, /* k2g2[1] = 0.05024063501786333 */
+ 1385, /* k2g2[2] = 0.042272226217199664 */
+ 1165, /* k2g2[3] = 0.03556764576567844 */
+ 981, /* k2g2[4] = 0.029926444346999134 */
+ 825, /* k2g2[5] = 0.025179964880280382 */
+ 694, /* k2g2[6] = 0.02118630011706455 */
+ 584, /* k2g2[7] = 0.01782604998793514 */
+ 491, /* k2g2[8] = 0.014998751854573014 */
+ 414, /* k2g2[9] = 0.012619876941179595 */
+ 348, /* k2g2[10] = 0.010618303146468736 */
+ 293, /* k2g2[11] = 0.008934188679954682 */
+ 246, /* k2g2[12] = 0.007517182949855368 */
+ 207, /* k2g2[13] = 0.006324921212866403 */
+ 174, /* k2g2[14] = 0.005321757979794424 */
+ 147, /* k2g2[15] = 0.004477701309210577 */
+ 123, /* k2g2[16] = 0.00376751612730811 */
+ 104, /* k2g2[17] = 0.0031699697655869644 */
+ 87, /* k2g2[18] = 0.00266719715992703 */
+ 74, /* k2g2[19] = 0.0022441667321724647 */
+ 62, /* k2g2[20] = 0.0018882309854916855 */
+ 52, /* k2g2[21] = 0.0015887483774966232 */
+ 44, /* k2g2[22] = 0.0013367651661223448 */
+ 37, /* k2g2[23] = 0.0011247477162958733 */
+ 31, /* k2g2[24] = 0.0009463572640678758 */
+ 26, /* k2g2[25] = 0.0007962604042473498 */
+ 22, /* k2g2[26] = 0.0006699696356181593 */
+ 18, /* k2g2[27] = 0.0005637091964589207 */
+ 16, /* k2g2[28] = 0.00047430217920125243 */
+ 13, /* k2g2[29] = 0.00039907554925166274 */
+ 11 /* k2g2[30] = 0.00033578022828973666 */
};
-static const EAS_I16 n1g2[] =
+static const EAS_I16 n1g2[] =
{
- 3170, /* n1g2[0] = 0.0967319927350769 */
- 3036, /* n1g2[1] = 0.0926446051254155 */
- 2908, /* n1g2[2] = 0.08872992911818503 */
- 2785, /* n1g2[3] = 0.08498066682523227 */
- 2667, /* n1g2[4] = 0.08138982872895201 */
- 2554, /* n1g2[5] = 0.07795072065216213 */
- 2446, /* n1g2[6] = 0.0746569312785634 */
- 2343, /* n1g2[7] = 0.07150232020051943 */
- 2244, /* n1g2[8] = 0.06848100647187474 */
- 2149, /* n1g2[9] = 0.06558735764447099 */
- 2058, /* n1g2[10] = 0.06281597926792246 */
- 1971, /* n1g2[11] = 0.06016170483307614 */
- 1888, /* n1g2[12] = 0.05761958614040857 */
- 1808, /* n1g2[13] = 0.05518488407540374 */
- 1732, /* n1g2[14] = 0.052853059773715245 */
- 1659, /* n1g2[15] = 0.05061976615964251 */
- 1589, /* n1g2[16] = 0.04848083984214659 */
- 1521, /* n1g2[17] = 0.046432293353298 */
- 1457, /* n1g2[18] = 0.04447030771468711 */
- 1396, /* n1g2[19] = 0.04259122531793907 */
- 1337, /* n1g2[20] = 0.040791543106060944 */
- 1280, /* n1g2[21] = 0.03906790604290942 */
- 1226, /* n1g2[22] = 0.037417100858604564 */
- 1174, /* n1g2[23] = 0.035836050059229754 */
- 1125, /* n1g2[24] = 0.03432180618965023 */
- 1077, /* n1g2[25] = 0.03287154633875494 */
- 1032, /* n1g2[26] = 0.03148256687687814 */
- 988, /* n1g2[27] = 0.030152278415589925 */
- 946, /* n1g2[28] = 0.028878200980459685 */
- 906, /* n1g2[29] = 0.02765795938779331 */
- 868 /* n1g2[30] = 0.02648927881672521 */
+ 3170, /* n1g2[0] = 0.0967319927350769 */
+ 3036, /* n1g2[1] = 0.0926446051254155 */
+ 2908, /* n1g2[2] = 0.08872992911818503 */
+ 2785, /* n1g2[3] = 0.08498066682523227 */
+ 2667, /* n1g2[4] = 0.08138982872895201 */
+ 2554, /* n1g2[5] = 0.07795072065216213 */
+ 2446, /* n1g2[6] = 0.0746569312785634 */
+ 2343, /* n1g2[7] = 0.07150232020051943 */
+ 2244, /* n1g2[8] = 0.06848100647187474 */
+ 2149, /* n1g2[9] = 0.06558735764447099 */
+ 2058, /* n1g2[10] = 0.06281597926792246 */
+ 1971, /* n1g2[11] = 0.06016170483307614 */
+ 1888, /* n1g2[12] = 0.05761958614040857 */
+ 1808, /* n1g2[13] = 0.05518488407540374 */
+ 1732, /* n1g2[14] = 0.052853059773715245 */
+ 1659, /* n1g2[15] = 0.05061976615964251 */
+ 1589, /* n1g2[16] = 0.04848083984214659 */
+ 1521, /* n1g2[17] = 0.046432293353298 */
+ 1457, /* n1g2[18] = 0.04447030771468711 */
+ 1396, /* n1g2[19] = 0.04259122531793907 */
+ 1337, /* n1g2[20] = 0.040791543106060944 */
+ 1280, /* n1g2[21] = 0.03906790604290942 */
+ 1226, /* n1g2[22] = 0.037417100858604564 */
+ 1174, /* n1g2[23] = 0.035836050059229754 */
+ 1125, /* n1g2[24] = 0.03432180618965023 */
+ 1077, /* n1g2[25] = 0.03287154633875494 */
+ 1032, /* n1g2[26] = 0.03148256687687814 */
+ 988, /* n1g2[27] = 0.030152278415589925 */
+ 946, /* n1g2[28] = 0.028878200980459685 */
+ 906, /* n1g2[29] = 0.02765795938779331 */
+ 868 /* n1g2[30] = 0.02648927881672521 */
};
-static const EAS_I16 n1g3[] =
+static const EAS_I16 n1g3[] =
{
- -548, /* n1g3[0] = -0.016714088475899017 */
- -481, /* n1g3[1] = -0.014683605122742116 */
- -423, /* n1g3[2] = -0.012899791676436092 */
- -371, /* n1g3[3] = -0.01133268185193299 */
- -326, /* n1g3[4] = -0.00995594976868754 */
- -287, /* n1g3[5] = -0.008746467702146129 */
- -252, /* n1g3[6] = -0.00768391756106361 */
- -221, /* n1g3[7] = -0.006750449563854721 */
- -194, /* n1g3[8] = -0.005930382380083576 */
- -171, /* n1g3[9] = -0.005209939699767622 */
- -150, /* n1g3[10] = -0.004577018805123356 */
- -132, /* n1g3[11] = -0.004020987256990177 */
- -116, /* n1g3[12] = -0.003532504280467257 */
- -102, /* n1g3[13] = -0.00310336384922047 */
- -89, /* n1g3[14] = -0.002726356832432369 */
- -78, /* n1g3[15] = -0.002395149888601605 */
- -69, /* n1g3[16] = -0.0021041790717285314 */
- -61, /* n1g3[17] = -0.0018485563625771063 */
- -53, /* n1g3[18] = -0.001623987554831628 */
- -47, /* n1g3[19] = -0.0014267001167177025 */
- -41, /* n1g3[20] = -0.0012533798162347005 */
- -36, /* n1g3[21] = -0.0011011150453668693 */
- -32, /* n1g3[22] = -0.0009673479079754438 */
- -28, /* n1g3[23] = -0.0008498312496971563 */
- -24, /* n1g3[24] = -0.0007465909079943587 */
- -21, /* n1g3[25] = -0.0006558925481952733 */
- -19, /* n1g3[26] = -0.0005762125284029567 */
- -17, /* n1g3[27] = -0.0005062123038325457 */
- -15, /* n1g3[28] = -0.0004447159405951901 */
- -13, /* n1g3[29] = -0.00039069036118270117 */
- -11 /* n1g3[30] = -0.00034322798979677605 */
+ -548, /* n1g3[0] = -0.016714088475899017 */
+ -481, /* n1g3[1] = -0.014683605122742116 */
+ -423, /* n1g3[2] = -0.012899791676436092 */
+ -371, /* n1g3[3] = -0.01133268185193299 */
+ -326, /* n1g3[4] = -0.00995594976868754 */
+ -287, /* n1g3[5] = -0.008746467702146129 */
+ -252, /* n1g3[6] = -0.00768391756106361 */
+ -221, /* n1g3[7] = -0.006750449563854721 */
+ -194, /* n1g3[8] = -0.005930382380083576 */
+ -171, /* n1g3[9] = -0.005209939699767622 */
+ -150, /* n1g3[10] = -0.004577018805123356 */
+ -132, /* n1g3[11] = -0.004020987256990177 */
+ -116, /* n1g3[12] = -0.003532504280467257 */
+ -102, /* n1g3[13] = -0.00310336384922047 */
+ -89, /* n1g3[14] = -0.002726356832432369 */
+ -78, /* n1g3[15] = -0.002395149888601605 */
+ -69, /* n1g3[16] = -0.0021041790717285314 */
+ -61, /* n1g3[17] = -0.0018485563625771063 */
+ -53, /* n1g3[18] = -0.001623987554831628 */
+ -47, /* n1g3[19] = -0.0014267001167177025 */
+ -41, /* n1g3[20] = -0.0012533798162347005 */
+ -36, /* n1g3[21] = -0.0011011150453668693 */
+ -32, /* n1g3[22] = -0.0009673479079754438 */
+ -28, /* n1g3[23] = -0.0008498312496971563 */
+ -24, /* n1g3[24] = -0.0007465909079943587 */
+ -21, /* n1g3[25] = -0.0006558925481952733 */
+ -19, /* n1g3[26] = -0.0005762125284029567 */
+ -17, /* n1g3[27] = -0.0005062123038325457 */
+ -15, /* n1g3[28] = -0.0004447159405951901 */
+ -13, /* n1g3[29] = -0.00039069036118270117 */
+ -11 /* n1g3[30] = -0.00034322798979677605 */
};
/*----------------------------------------------------------------------------
* WT_SetFilterCoeffs()
*----------------------------------------------------------------------------
- * Purpose:
+ * Purpose:
* Update the Filter parameters
*
- * Inputs:
- * pVoice - ptr to the voice whose filter we want to update
+ * Inputs:
+ * pVoice - ptr to the voice whose filter we want to update
* pEASData - pointer to overall EAS data structure
*
* Outputs:
@@ -1226,32 +1226,32 @@ static const EAS_I16 n1g3[] =
*/
void WT_SetFilterCoeffs (S_WT_INT_FRAME *pIntFrame, EAS_I32 cutoff, EAS_I32 resonance)
{
- EAS_I32 temp;
-
- /*
- Convert the cutoff, which has had A5 subtracted, using the 2^x approx
- Note, this cutoff is related to theta cutoff by
- theta = k * 2^x
- We use 2^x and incorporate k in the power series coefs instead
- */
- cutoff = EAS_Calculate2toX(cutoff);
-
- /* calculate b2 coef */
- temp = k2g1[resonance] + MULT_AUDIO_COEF(cutoff, k2g2[resonance]);
- temp = k2g0 + MULT_AUDIO_COEF(cutoff, temp);
- pIntFrame->frame.b2 = temp;
-
- /* calculate b1 coef */
- temp = MULT_AUDIO_COEF(cutoff, nk1g2);
- temp = nk1g0 + MULT_AUDIO_COEF(cutoff, temp);
- temp += MULT_AUDIO_COEF(temp, pIntFrame->frame.b2);
- pIntFrame->frame.b1 = temp >> 1;
-
- /* calculate K coef */
- temp = n1g2[resonance] + MULT_AUDIO_COEF(cutoff, n1g3[resonance]);
- temp = MULT_AUDIO_COEF(cutoff, temp);
- temp = MULT_AUDIO_COEF(cutoff, temp);
- pIntFrame->frame.k = temp;
+ EAS_I32 temp;
+
+ /*
+ Convert the cutoff, which has had A5 subtracted, using the 2^x approx
+ Note, this cutoff is related to theta cutoff by
+ theta = k * 2^x
+ We use 2^x and incorporate k in the power series coefs instead
+ */
+ cutoff = EAS_Calculate2toX(cutoff);
+
+ /* calculate b2 coef */
+ temp = k2g1[resonance] + MULT_AUDIO_COEF(cutoff, k2g2[resonance]);
+ temp = k2g0 + MULT_AUDIO_COEF(cutoff, temp);
+ pIntFrame->frame.b2 = temp;
+
+ /* calculate b1 coef */
+ temp = MULT_AUDIO_COEF(cutoff, nk1g2);
+ temp = nk1g0 + MULT_AUDIO_COEF(cutoff, temp);
+ temp += MULT_AUDIO_COEF(temp, pIntFrame->frame.b2);
+ pIntFrame->frame.b1 = temp >> 1;
+
+ /* calculate K coef */
+ temp = n1g2[resonance] + MULT_AUDIO_COEF(cutoff, n1g3[resonance]);
+ temp = MULT_AUDIO_COEF(cutoff, temp);
+ temp = MULT_AUDIO_COEF(cutoff, temp);
+ pIntFrame->frame.k = temp;
}
#endif