1 /*----------------------------------------------------------------------------
2 *
3 * File:
4 * eas_wtsynth.c
5 *
6 * Contents and purpose:
7 * Implements the synthesizer functions.
8 *
9 * Copyright Sonic Network Inc. 2004
10
11 * Licensed under the Apache License, Version 2.0 (the "License");
12 * you may not use this file except in compliance with the License.
13 * You may obtain a copy of the License at
14 *
15 * http://www.apache.org/licenses/LICENSE-2.0
16 *
17 * Unless required by applicable law or agreed to in writing, software
18 * distributed under the License is distributed on an "AS IS" BASIS,
19 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
20 * See the License for the specific language governing permissions and
21 * limitations under the License.
22 *
23 *----------------------------------------------------------------------------
24 * Revision Control:
25 * $Revision: 795 $
26 * $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $
27 *----------------------------------------------------------------------------
28 */
29
30 // includes
31 #define LOG_TAG "SYNTH"
32 #include "log/log.h"
33 #include <cutils/log.h>
34
35 #include "eas_data.h"
36 #include "eas_report.h"
37 #include "eas_host.h"
38 #include "eas_math.h"
39 #include "eas_synth_protos.h"
40 #include "eas_wtsynth.h"
41 #include "eas_pan.h"
42
43 #ifdef DLS_SYNTHESIZER
44 #include "eas_dlssynth.h"
45 #endif
46
47 #ifdef _METRICS_ENABLED
48 #include "eas_perf.h"
49 #endif
50
51 /* local prototypes */
52 static EAS_RESULT WT_Initialize(S_VOICE_MGR *pVoiceMgr);
53 static void WT_ReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum);
54 static void WT_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum);
55 static void WT_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum);
56 static EAS_RESULT WT_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex);
57 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);
58 static void WT_UpdateChannel (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel);
59 static EAS_I32 WT_UpdatePhaseInc (S_WT_VOICE *pWTVoice, const S_ARTICULATION *pArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 pitchCents);
60 static EAS_I32 WT_UpdateGain (S_SYNTH_VOICE *pVoice, S_WT_VOICE *pWTVoice, const S_ARTICULATION *pArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 gain);
61 static void WT_UpdateEG1 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv);
62 static void WT_UpdateEG2 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv);
63
64 #ifdef EAS_SPLIT_WT_SYNTH
65 extern EAS_BOOL WTE_StartFrame (EAS_FRAME_BUFFER_HANDLE pFrameBuffer);
66 extern EAS_BOOL WTE_EndFrame (EAS_FRAME_BUFFER_HANDLE pFrameBuffer, EAS_I32 *pMixBuffer, EAS_I16 masterGain);
67 #endif
68
69 #ifdef _FILTER_ENABLED
70 static void WT_UpdateFilter (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pIntFrame, const S_ARTICULATION *pArt);
71 #endif
72
73 #ifdef _STATS
74 extern double statsPhaseIncrement;
75 extern double statsMaxPhaseIncrement;
76 extern long statsPhaseSampleCount;
77 extern double statsSampleSize;
78 extern long statsSampleCount;
79 #endif
80
81 /*----------------------------------------------------------------------------
82 * Synthesizer interface
83 *----------------------------------------------------------------------------
84 */
85
86 const S_SYNTH_INTERFACE wtSynth =
87 {
88 WT_Initialize,
89 WT_StartVoice,
90 WT_UpdateVoice,
91 WT_ReleaseVoice,
92 WT_MuteVoice,
93 WT_SustainPedal,
94 WT_UpdateChannel
95 };
96
97 #ifdef EAS_SPLIT_WT_SYNTH
98 const S_FRAME_INTERFACE wtFrameInterface =
99 {
100 WTE_StartFrame,
101 WTE_EndFrame
102 };
103 #endif
104
105 /*----------------------------------------------------------------------------
106 * WT_Initialize()
107 *----------------------------------------------------------------------------
108 * Purpose:
109 *
110 * Inputs:
111 * pVoice - pointer to voice to initialize
112 *
113 * Outputs:
114 *
115 *----------------------------------------------------------------------------
116 */
WT_Initialize(S_VOICE_MGR * pVoiceMgr)117 static EAS_RESULT WT_Initialize (S_VOICE_MGR *pVoiceMgr)
118 {
119 EAS_INT i;
120
121 for (i = 0; i < NUM_WT_VOICES; i++)
122 {
123
124 pVoiceMgr->wtVoices[i].artIndex = DEFAULT_ARTICULATION_INDEX;
125
126 pVoiceMgr->wtVoices[i].eg1State = DEFAULT_EG1_STATE;
127 pVoiceMgr->wtVoices[i].eg1Value = DEFAULT_EG1_VALUE;
128 pVoiceMgr->wtVoices[i].eg1Increment = DEFAULT_EG1_INCREMENT;
129
130 pVoiceMgr->wtVoices[i].eg2State = DEFAULT_EG2_STATE;
131 pVoiceMgr->wtVoices[i].eg2Value = DEFAULT_EG2_VALUE;
132 pVoiceMgr->wtVoices[i].eg2Increment = DEFAULT_EG2_INCREMENT;
133
134 /* left and right gain values are needed only if stereo output */
135 #if (NUM_OUTPUT_CHANNELS == 2)
136 pVoiceMgr->wtVoices[i].gainLeft = DEFAULT_VOICE_GAIN;
137 pVoiceMgr->wtVoices[i].gainRight = DEFAULT_VOICE_GAIN;
138 #endif
139
140 pVoiceMgr->wtVoices[i].phaseFrac = DEFAULT_PHASE_FRAC;
141 pVoiceMgr->wtVoices[i].phaseAccum = DEFAULT_PHASE_INT;
142
143 #ifdef _FILTER_ENABLED
144 pVoiceMgr->wtVoices[i].filter.z1 = DEFAULT_FILTER_ZERO;
145 pVoiceMgr->wtVoices[i].filter.z2 = DEFAULT_FILTER_ZERO;
146 #endif
147 }
148
149 return EAS_TRUE;
150 }
151
152 /*----------------------------------------------------------------------------
153 * WT_ReleaseVoice()
154 *----------------------------------------------------------------------------
155 * Purpose:
156 * The selected voice is being released.
157 *
158 * Inputs:
159 * pEASData - pointer to S_EAS_DATA
160 * pVoice - pointer to voice to release
161 *
162 * Outputs:
163 * None
164 *----------------------------------------------------------------------------
165 */
166 /*lint -esym(715, pVoice) used in some implementations */
WT_ReleaseVoice(S_VOICE_MGR * pVoiceMgr,S_SYNTH * pSynth,S_SYNTH_VOICE * pVoice,EAS_I32 voiceNum)167 static void WT_ReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum)
168 {
169 S_WT_VOICE *pWTVoice;
170 const S_ARTICULATION *pArticulation;
171
172 #ifdef DLS_SYNTHESIZER
173 if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
174 {
175 DLS_ReleaseVoice(pVoiceMgr, pSynth, pVoice, voiceNum);
176 return;
177 }
178 #endif
179
180 pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
181 pArticulation = &pSynth->pEAS->pArticulations[pWTVoice->artIndex];
182
183 /* release EG1 */
184 pWTVoice->eg1State = eEnvelopeStateRelease;
185 pWTVoice->eg1Increment = pArticulation->eg1.releaseTime;
186
187 /*
188 The spec says we should release EG2, but doing so with the current
189 voicing is causing clicks. This fix will need to be coordinated with
190 a new sound library release
191 */
192
193 /* release EG2 */
194 pWTVoice->eg2State = eEnvelopeStateRelease;
195 pWTVoice->eg2Increment = pArticulation->eg2.releaseTime;
196 }
197
198 /*----------------------------------------------------------------------------
199 * WT_MuteVoice()
200 *----------------------------------------------------------------------------
201 * Purpose:
202 * The selected voice is being muted.
203 *
204 * Inputs:
205 * pVoice - pointer to voice to release
206 *
207 * Outputs:
208 * None
209 *----------------------------------------------------------------------------
210 */
211 /*lint -esym(715, pSynth) used in some implementations */
WT_MuteVoice(S_VOICE_MGR * pVoiceMgr,S_SYNTH * pSynth,S_SYNTH_VOICE * pVoice,EAS_I32 voiceNum)212 static void WT_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum)
213 {
214
215 #ifdef DLS_SYNTHESIZER
216 if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
217 {
218 DLS_MuteVoice(pVoiceMgr, pSynth, pVoice, voiceNum);
219 return;
220 }
221 #endif
222
223 /* clear deferred action flags */
224 pVoice->voiceFlags &=
225 ~(VOICE_FLAG_DEFER_MIDI_NOTE_OFF |
226 VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF |
227 VOICE_FLAG_DEFER_MUTE);
228
229 /* set the envelope state */
230 pVoiceMgr->wtVoices[voiceNum].eg1State = eEnvelopeStateMuted;
231 pVoiceMgr->wtVoices[voiceNum].eg2State = eEnvelopeStateMuted;
232 }
233
234 /*----------------------------------------------------------------------------
235 * WT_SustainPedal()
236 *----------------------------------------------------------------------------
237 * Purpose:
238 * The selected voice is held due to sustain pedal
239 *
240 * Inputs:
241 * pVoice - pointer to voice to sustain
242 *
243 * Outputs:
244 * None
245 *----------------------------------------------------------------------------
246 */
247 /*lint -esym(715, pChannel) used in some implementations */
WT_SustainPedal(S_VOICE_MGR * pVoiceMgr,S_SYNTH * pSynth,S_SYNTH_VOICE * pVoice,S_SYNTH_CHANNEL * pChannel,EAS_I32 voiceNum)248 static void WT_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum)
249 {
250 S_WT_VOICE *pWTVoice;
251
252 #ifdef DLS_SYNTHESIZER
253 if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
254 {
255 DLS_SustainPedal(pVoiceMgr, pSynth, pVoice, pChannel, voiceNum);
256 return;
257 }
258 #endif
259
260 /* don't catch the voice if below the sustain level */
261 pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
262 if (pWTVoice->eg1Value < pSynth->pEAS->pArticulations[pWTVoice->artIndex].eg1.sustainLevel)
263 return;
264
265 /* sustain flag is set, damper pedal is on */
266 /* defer releasing this note until the damper pedal is off */
267 pWTVoice->eg1State = eEnvelopeStateDecay;
268 pVoice->voiceState = eVoiceStatePlay;
269
270 /*
271 because sustain pedal is on, this voice
272 should defer releasing its note
273 */
274 pVoice->voiceFlags |= VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF;
275
276 #ifdef _DEBUG_SYNTH
277 { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_SustainPedal: defer note off because sustain pedal is on\n"); */ }
278 #endif
279 }
280
281 /*----------------------------------------------------------------------------
282 * WT_StartVoice()
283 *----------------------------------------------------------------------------
284 * Purpose:
285 * Assign the region for the given instrument using the midi key number
286 * and the RPN2 (coarse tuning) value. By using RPN2 as part of the
287 * region selection process, we reduce the amount a given sample has
288 * to be transposed by selecting the closest recorded root instead.
289 *
290 * This routine is the second half of SynthAssignRegion().
291 * If the region was successfully found by SynthFindRegionIndex(),
292 * then assign the region's parameters to the voice.
293 *
294 * Setup and initialize the following voice parameters:
295 * m_nRegionIndex
296 *
297 * Inputs:
298 * pVoice - ptr to the voice we have assigned for this channel
299 * nRegionIndex - index of the region
300 * pEASData - pointer to overall EAS data structure
301 *
302 * Outputs:
303 * success - could find and assign the region for this voice's note otherwise
304 * failure - could not find nor assign the region for this voice's note
305 *
306 * Side Effects:
307 * psSynthObject->m_sVoice[].m_nRegionIndex is assigned
308 * psSynthObject->m_sVoice[] parameters are assigned
309 *----------------------------------------------------------------------------
310 */
WT_StartVoice(S_VOICE_MGR * pVoiceMgr,S_SYNTH * pSynth,S_SYNTH_VOICE * pVoice,EAS_I32 voiceNum,EAS_U16 regionIndex)311 static EAS_RESULT WT_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex)
312 {
313 S_WT_VOICE *pWTVoice;
314 const S_WT_REGION *pRegion;
315 const S_ARTICULATION *pArt;
316 S_SYNTH_CHANNEL *pChannel;
317
318 #if (NUM_OUTPUT_CHANNELS == 2)
319 EAS_INT pan;
320 #endif
321
322 #ifdef EAS_SPLIT_WT_SYNTH
323 S_WT_CONFIG wtConfig;
324 #endif
325
326 /* no samples have been synthesized for this note yet */
327 pVoice->regionIndex = regionIndex;
328 pVoice->voiceFlags = VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET;
329
330 /* get the articulation index for this region */
331 pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
332 pChannel = &pSynth->channels[pVoice->channel & 15];
333
334 /* update static channel parameters */
335 if (pChannel->channelFlags & CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS)
336 WT_UpdateChannel(pVoiceMgr, pSynth, pVoice->channel & 15);
337
338 #ifdef DLS_SYNTHESIZER
339 if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
340 return DLS_StartVoice(pVoiceMgr, pSynth, pVoice, voiceNum, regionIndex);
341 #endif
342
343 pRegion = &(pSynth->pEAS->pWTRegions[regionIndex]);
344 pWTVoice->artIndex = pRegion->artIndex;
345
346 #ifdef _DEBUG_SYNTH
347 { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_StartVoice: Voice %ld; Region %d\n", (EAS_I32) (pVoice - pVoiceMgr->voices), regionIndex); */ }
348 #endif
349
350 pArt = &pSynth->pEAS->pArticulations[pWTVoice->artIndex];
351
352 /* MIDI note on puts this voice into attack state */
353 pWTVoice->eg1State = eEnvelopeStateAttack;
354 pWTVoice->eg1Value = 0;
355 pWTVoice->eg1Increment = pArt->eg1.attackTime;
356 pWTVoice->eg2State = eEnvelopeStateAttack;
357 pWTVoice->eg2Value = 0;
358 pWTVoice->eg2Increment = pArt->eg2.attackTime;
359
360 /* init the LFO */
361 pWTVoice->modLFO.lfoValue = 0;
362 pWTVoice->modLFO.lfoPhase = -pArt->lfoDelay;
363
364 pVoice->gain = 0;
365
366 #if (NUM_OUTPUT_CHANNELS == 2)
367 /*
368 Get the Midi CC10 pan value for this voice's channel
369 convert the pan value to an "angle" representation suitable for
370 our sin, cos calculator. This representation is NOT necessarily the same
371 as the transform in the GM manuals because of our sin, cos calculator.
372 "angle" = (CC10 - 64)/128
373 */
374 pan = (EAS_INT) pSynth->channels[pVoice->channel & 15].pan - 64;
375 pan += pArt->pan;
376 EAS_CalcPanControl(pan, &pWTVoice->gainLeft, &pWTVoice->gainRight);
377 #endif
378
379 #ifdef _FILTER_ENABLED
380 /* clear out the filter states */
381 pWTVoice->filter.z1 = 0;
382 pWTVoice->filter.z2 = 0;
383 #endif
384
385 /* if this wave is to be generated using noise generator */
386 if (pRegion->region.keyGroupAndFlags & REGION_FLAG_USE_WAVE_GENERATOR)
387 {
388 pWTVoice->phaseAccum = 4574296;
389 pWTVoice->loopStart = WT_NOISE_GENERATOR;
390 pWTVoice->loopEnd = 4574295;
391 }
392
393 /* normal sample */
394 else
395 {
396
397 #ifdef EAS_SPLIT_WT_SYNTH
398 if (voiceNum < NUM_PRIMARY_VOICES)
399 pWTVoice->phaseAccum = (EAS_U32) pSynth->pEAS->pSamples + pSynth->pEAS->pSampleOffsets[pRegion->waveIndex];
400 else
401 pWTVoice->phaseAccum = pSynth->pEAS->pSampleOffsets[pRegion->waveIndex];
402 #else
403 pWTVoice->phaseAccum = (EAS_U32) pSynth->pEAS->pSamples + pSynth->pEAS->pSampleOffsets[pRegion->waveIndex];
404 #endif
405
406 if (pRegion->region.keyGroupAndFlags & REGION_FLAG_IS_LOOPED)
407 {
408 #if defined (_8_BIT_SAMPLES)
409 pWTVoice->loopStart = pWTVoice->phaseAccum + pRegion->loopStart;
410 pWTVoice->loopEnd = pWTVoice->phaseAccum + pRegion->loopEnd - 1;
411 #else //_16_BIT_SAMPLES
412 pWTVoice->loopStart = pWTVoice->phaseAccum + (pRegion->loopStart<<1);
413 pWTVoice->loopEnd = pWTVoice->phaseAccum + (pRegion->loopEnd<<1) - 2;
414 #endif
415 }
416 else {
417 #if defined (_8_BIT_SAMPLES)
418 pWTVoice->loopStart = pWTVoice->loopEnd = pWTVoice->phaseAccum + pSynth->pEAS->pSampleLen[pRegion->waveIndex] - 1;
419 #else //_16_BIT_SAMPLES
420 pWTVoice->loopStart = pWTVoice->loopEnd = pWTVoice->phaseAccum + pSynth->pEAS->pSampleLen[pRegion->waveIndex] - 2;
421 #endif
422 }
423 }
424
425 #ifdef EAS_SPLIT_WT_SYNTH
426 /* configure off-chip voices */
427 if (voiceNum >= NUM_PRIMARY_VOICES)
428 {
429 wtConfig.phaseAccum = pWTVoice->phaseAccum;
430 wtConfig.loopStart = pWTVoice->loopStart;
431 wtConfig.loopEnd = pWTVoice->loopEnd;
432 wtConfig.gain = pVoice->gain;
433
434 #if (NUM_OUTPUT_CHANNELS == 2)
435 wtConfig.gainLeft = pWTVoice->gainLeft;
436 wtConfig.gainRight = pWTVoice->gainRight;
437 #endif
438
439 WTE_ConfigVoice(voiceNum - NUM_PRIMARY_VOICES, &wtConfig, pVoiceMgr->pFrameBuffer);
440 }
441 #endif
442
443 return EAS_SUCCESS;
444 }
445
446 /*----------------------------------------------------------------------------
447 * WT_CheckSampleEnd
448 *----------------------------------------------------------------------------
449 * Purpose:
450 * Check for end of sample and calculate number of samples to synthesize
451 *
452 * Inputs:
453 *
454 * Outputs:
455 *
456 * Notes:
457 *
458 *----------------------------------------------------------------------------
459 */
WT_CheckSampleEnd(S_WT_VOICE * pWTVoice,S_WT_INT_FRAME * pWTIntFrame,EAS_BOOL update)460 EAS_BOOL WT_CheckSampleEnd (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame, EAS_BOOL update)
461 {
462 EAS_U32 endPhaseAccum;
463 EAS_U32 endPhaseFrac;
464 EAS_I32 numSamples;
465 EAS_BOOL done = EAS_FALSE;
466
467 /* check to see if we hit the end of the waveform this time */
468 /*lint -e{703} use shift for performance */
469 endPhaseFrac = pWTVoice->phaseFrac + (pWTIntFrame->frame.phaseIncrement << SYNTH_UPDATE_PERIOD_IN_BITS);
470 #if defined (_8_BIT_SAMPLES)
471 endPhaseAccum = pWTVoice->phaseAccum + GET_PHASE_INT_PART(endPhaseFrac);
472 #else //_16_BIT_SAMPLES
473 // Multiply by 2 for 16 bit processing module implementation
474 endPhaseAccum = pWTVoice->phaseAccum + (EAS_U32)(endPhaseFrac >> 14);
475 #endif
476 if (endPhaseAccum >= pWTVoice->loopEnd)
477 {
478 /* calculate how far current ptr is from end */
479 numSamples = (EAS_I32) (pWTVoice->loopEnd - pWTVoice->phaseAccum);
480 #if defined (_16_BIT_SAMPLES)
481 numSamples >>= 1; // Divide by 2 for 16 bit processing module implementation
482 #endif
483 /* now account for the fractional portion */
484 /*lint -e{703} use shift for performance */
485 numSamples = (EAS_I32) ((numSamples << NUM_PHASE_FRAC_BITS) - pWTVoice->phaseFrac);
486 if (pWTIntFrame->frame.phaseIncrement) {
487 pWTIntFrame->numSamples = 1 + (numSamples / pWTIntFrame->frame.phaseIncrement);
488 } else {
489 pWTIntFrame->numSamples = numSamples;
490 }
491 if (pWTIntFrame->numSamples < 0) {
492 ALOGE("b/26366256");
493 android_errorWriteLog(0x534e4554, "26366256");
494 pWTIntFrame->numSamples = 0;
495 }
496
497 /* sound will be done this frame */
498 done = EAS_TRUE;
499 }
500
501 /* update data for off-chip synth */
502 if (update)
503 {
504 pWTVoice->phaseFrac = endPhaseFrac;
505 pWTVoice->phaseAccum = endPhaseAccum;
506 }
507
508 return done;
509 }
510
511 /*----------------------------------------------------------------------------
512 * WT_UpdateVoice()
513 *----------------------------------------------------------------------------
514 * Purpose:
515 * Synthesize a block of samples for the given voice.
516 * Use linear interpolation.
517 *
518 * Inputs:
519 * pEASData - pointer to overall EAS data structure
520 *
521 * Outputs:
522 * number of samples actually written to buffer
523 *
524 * Side Effects:
525 * - samples are added to the presently free buffer
526 *
527 *----------------------------------------------------------------------------
528 */
WT_UpdateVoice(S_VOICE_MGR * pVoiceMgr,S_SYNTH * pSynth,S_SYNTH_VOICE * pVoice,EAS_I32 voiceNum,EAS_I32 * pMixBuffer,EAS_I32 numSamples)529 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)
530 {
531 S_WT_VOICE *pWTVoice;
532 S_WT_INT_FRAME intFrame;
533 S_SYNTH_CHANNEL *pChannel;
534 const S_WT_REGION *pWTRegion;
535 const S_ARTICULATION *pArt;
536 EAS_I32 temp;
537 EAS_BOOL done;
538
539 #ifdef DLS_SYNTHESIZER
540 if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
541 return DLS_UpdateVoice(pVoiceMgr, pSynth, pVoice, voiceNum, pMixBuffer, numSamples);
542 #endif
543
544 /* establish pointers to critical data */
545 pWTVoice = &pVoiceMgr->wtVoices[voiceNum];
546 pWTRegion = &pSynth->pEAS->pWTRegions[pVoice->regionIndex & REGION_INDEX_MASK];
547 pArt = &pSynth->pEAS->pArticulations[pWTVoice->artIndex];
548 pChannel = &pSynth->channels[pVoice->channel & 15];
549 intFrame.prevGain = pVoice->gain;
550
551 /* update the envelopes */
552 WT_UpdateEG1(pWTVoice, &pArt->eg1);
553 WT_UpdateEG2(pWTVoice, &pArt->eg2);
554
555 /* update the LFO */
556 WT_UpdateLFO(&pWTVoice->modLFO, pArt->lfoFreq);
557
558 #ifdef _FILTER_ENABLED
559 /* calculate filter if library uses filter */
560 if (pSynth->pEAS->libAttr & LIB_FORMAT_FILTER_ENABLED)
561 WT_UpdateFilter(pWTVoice, &intFrame, pArt);
562 else
563 intFrame.frame.k = 0;
564 #endif
565
566 /* update the gain */
567 intFrame.frame.gainTarget = WT_UpdateGain(pVoice, pWTVoice, pArt, pChannel, pWTRegion->gain);
568
569 /* calculate base pitch*/
570 temp = pChannel->staticPitch + pWTRegion->tuning;
571
572 /* include global transpose */
573 if (pChannel->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL)
574 temp += pVoice->note * 100;
575 else
576 temp += (pVoice->note + pSynth->globalTranspose) * 100;
577 intFrame.frame.phaseIncrement = WT_UpdatePhaseInc(pWTVoice, pArt, pChannel, temp);
578 temp = pWTVoice->loopEnd - pWTVoice->loopStart;
579 #ifdef _16_BIT_SAMPLES
580 temp >>= 1;
581 #endif
582 if (temp != 0) {
583 temp = temp << NUM_PHASE_FRAC_BITS;
584 if (intFrame.frame.phaseIncrement > temp) {
585 ALOGW("%p phaseIncrement=%d", pWTVoice, (int)intFrame.frame.phaseIncrement);
586 intFrame.frame.phaseIncrement %= temp;
587 }
588 }
589
590 /* call into engine to generate samples */
591 intFrame.pAudioBuffer = pVoiceMgr->voiceBuffer;
592 intFrame.pMixBuffer = pMixBuffer;
593 intFrame.numSamples = numSamples;
594
595 /* check for end of sample */
596 if ((pWTVoice->loopStart != WT_NOISE_GENERATOR) && (pWTVoice->loopStart == pWTVoice->loopEnd))
597 done = WT_CheckSampleEnd(pWTVoice, &intFrame, (EAS_BOOL) (voiceNum >= NUM_PRIMARY_VOICES));
598 else
599 done = EAS_FALSE;
600
601 if (intFrame.numSamples < 0) intFrame.numSamples = 0;
602
603 if (intFrame.numSamples > BUFFER_SIZE_IN_MONO_SAMPLES)
604 intFrame.numSamples = BUFFER_SIZE_IN_MONO_SAMPLES;
605
606 #ifdef EAS_SPLIT_WT_SYNTH
607 if (voiceNum < NUM_PRIMARY_VOICES)
608 {
609 #ifndef _SPLIT_WT_TEST_HARNESS
610 WT_ProcessVoice(pWTVoice, &intFrame);
611 #endif
612 }
613 else
614 WTE_ProcessVoice(voiceNum - NUM_PRIMARY_VOICES, &intFrame.frame, pVoiceMgr->pFrameBuffer);
615 #else
616 WT_ProcessVoice(pWTVoice, &intFrame);
617 #endif
618
619 /* clear flag */
620 pVoice->voiceFlags &= ~VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET;
621
622 /* if voice has finished, set flag for voice manager */
623 if ((pVoice->voiceState != eVoiceStateStolen) && (pWTVoice->eg1State == eEnvelopeStateMuted))
624 done = EAS_TRUE;
625
626 /* if the update interval has elapsed, then force the current gain to the next
627 * gain since we never actually reach the next gain when ramping -- we just get
628 * very close to the target gain.
629 */
630 pVoice->gain = (EAS_I16) intFrame.frame.gainTarget;
631
632 return done;
633 }
634
635 /*----------------------------------------------------------------------------
636 * WT_UpdatePhaseInc()
637 *----------------------------------------------------------------------------
638 * Purpose:
639 * Calculate the phase increment
640 *
641 * Inputs:
642 * pVoice - pointer to the voice being updated
643 * psRegion - pointer to the region
644 * psArticulation - pointer to the articulation
645 * nChannelPitchForThisVoice - the portion of the pitch that is fixed for this
646 * voice during the duration of this synthesis
647 * pEASData - pointer to overall EAS data structure
648 *
649 * Outputs:
650 *
651 * Side Effects:
652 * set the phase increment for this voice
653 *----------------------------------------------------------------------------
654 */
WT_UpdatePhaseInc(S_WT_VOICE * pWTVoice,const S_ARTICULATION * pArt,S_SYNTH_CHANNEL * pChannel,EAS_I32 pitchCents)655 static EAS_I32 WT_UpdatePhaseInc (S_WT_VOICE *pWTVoice, const S_ARTICULATION *pArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 pitchCents)
656 {
657 EAS_I32 temp;
658
659 /*pitchCents due to CC1 = LFO * (CC1 / 128) * DEFAULT_LFO_MOD_WHEEL_TO_PITCH_CENTS */
660 temp = MULT_EG1_EG1(DEFAULT_LFO_MOD_WHEEL_TO_PITCH_CENTS,
661 ((pChannel->modWheel) << (NUM_EG1_FRAC_BITS -7)));
662
663 /* pitchCents due to channel pressure = LFO * (channel pressure / 128) * DEFAULT_LFO_CHANNEL_PRESSURE_TO_PITCH_CENTS */
664 temp += MULT_EG1_EG1(DEFAULT_LFO_CHANNEL_PRESSURE_TO_PITCH_CENTS,
665 ((pChannel->channelPressure) << (NUM_EG1_FRAC_BITS -7)));
666
667 /* now multiply the (channel pressure + CC1) pitch values by the LFO value */
668 temp = MULT_EG1_EG1(pWTVoice->modLFO.lfoValue, temp);
669
670 /*
671 add in the LFO pitch due to
672 channel pressure and CC1 along with
673 the LFO pitch, the EG2 pitch, and the
674 "static" pitch for this voice on this channel
675 */
676 temp += pitchCents +
677 (MULT_EG1_EG1(pWTVoice->eg2Value, pArt->eg2ToPitch)) +
678 (MULT_EG1_EG1(pWTVoice->modLFO.lfoValue, pArt->lfoToPitch));
679
680 /* convert from cents to linear phase increment */
681 return EAS_Calculate2toX(temp);
682 }
683
684 /*----------------------------------------------------------------------------
685 * WT_UpdateChannel()
686 *----------------------------------------------------------------------------
687 * Purpose:
688 * Calculate and assign static channel parameters
689 * These values only need to be updated if one of the controller values
690 * for this channel changes
691 *
692 * Inputs:
693 * nChannel - channel to update
694 * pEASData - pointer to overall EAS data structure
695 *
696 * Outputs:
697 *
698 * Side Effects:
699 * - the given channel's static gain and static pitch are updated
700 *----------------------------------------------------------------------------
701 */
702 /*lint -esym(715, pVoiceMgr) reserved for future use */
WT_UpdateChannel(S_VOICE_MGR * pVoiceMgr,S_SYNTH * pSynth,EAS_U8 channel)703 static void WT_UpdateChannel (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel)
704 {
705 EAS_I32 staticGain;
706 EAS_I32 pitchBend;
707 S_SYNTH_CHANNEL *pChannel;
708
709 pChannel = &pSynth->channels[channel];
710
711 /*
712 nChannelGain = (CC7 * CC11)^2 * master volume
713 where CC7 == 100 by default, CC11 == 127, master volume == 32767
714 */
715 staticGain = MULT_EG1_EG1((pChannel->volume) << (NUM_EG1_FRAC_BITS - 7),
716 (pChannel->expression) << (NUM_EG1_FRAC_BITS - 7));
717
718 /* staticGain has to be squared */
719 staticGain = MULT_EG1_EG1(staticGain, staticGain);
720
721 pChannel->staticGain = (EAS_I16) MULT_EG1_EG1(staticGain, pSynth->masterVolume);
722
723 /*
724 calculate pitch bend: RPN0 * ((2*pitch wheel)/16384 -1)
725 However, if we use the EG1 macros, remember that EG1 has a full
726 scale value of 32768 (instead of 16384). So instead of multiplying
727 by 2, multiply by 4 (left shift by 2), and subtract by 32768 instead
728 of 16384. This utilizes the fact that the EG1 macro places a binary
729 point 15 places to the left instead of 14 places.
730 */
731 /*lint -e{703} <avoid multiply for performance>*/
732 pitchBend =
733 (((EAS_I32)(pChannel->pitchBend) << 2)
734 - 32768);
735
736 pChannel->staticPitch =
737 MULT_EG1_EG1(pitchBend, pChannel->pitchBendSensitivity);
738
739 /* if this is not a drum channel, then add in the per-channel tuning */
740 if (!(pChannel->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL))
741 pChannel->staticPitch += pChannel->finePitch + (pChannel->coarsePitch * 100);
742
743 /* clear update flag */
744 pChannel->channelFlags &= ~CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS;
745 return;
746 }
747
748 /*----------------------------------------------------------------------------
749 * WT_UpdateGain()
750 *----------------------------------------------------------------------------
751 * Purpose:
752 * Calculate and assign static voice parameters as part of WT_UpdateVoice()
753 *
754 * Inputs:
755 * pVoice - ptr to the synth voice that we want to synthesize
756 * pEASData - pointer to overall EAS data structure
757 *
758 * Outputs:
759 *
760 * Side Effects:
761 * - various voice parameters are calculated and assigned
762 *
763 *----------------------------------------------------------------------------
764 */
WT_UpdateGain(S_SYNTH_VOICE * pVoice,S_WT_VOICE * pWTVoice,const S_ARTICULATION * pArt,S_SYNTH_CHANNEL * pChannel,EAS_I32 gain)765 static EAS_I32 WT_UpdateGain (S_SYNTH_VOICE *pVoice, S_WT_VOICE *pWTVoice, const S_ARTICULATION *pArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 gain)
766 {
767 EAS_I32 lfoGain;
768 EAS_I32 temp;
769
770 /*
771 If this voice was stolen, then the velocity is actually
772 for the new note, not the note that we are currently ramping down.
773 So we really shouldn't use this velocity. However, that would require
774 more memory to store the velocity value, and the improvement may
775 not be sufficient to warrant the added memory.
776 */
777 /* velocity is fixed at note start for a given voice and must be squared */
778 temp = (pVoice->velocity) << (NUM_EG1_FRAC_BITS - 7);
779 temp = MULT_EG1_EG1(temp, temp);
780
781 /* region gain is fixed as part of the articulation */
782 temp = MULT_EG1_EG1(temp, gain);
783
784 /* include the channel gain */
785 temp = MULT_EG1_EG1(temp, pChannel->staticGain);
786
787 /* calculate LFO gain using an approximation for 10^x */
788 lfoGain = MULT_EG1_EG1(pWTVoice->modLFO.lfoValue, pArt->lfoToGain);
789 lfoGain = MULT_EG1_EG1(lfoGain, LFO_GAIN_TO_CENTS);
790
791 /* convert from a dB-like value to linear gain */
792 lfoGain = EAS_Calculate2toX(lfoGain);
793 temp = MULT_EG1_EG1(temp, lfoGain);
794
795 /* calculate the voice's gain */
796 temp = (EAS_I16)MULT_EG1_EG1(temp, pWTVoice->eg1Value);
797
798 return temp;
799 }
800
801 /*----------------------------------------------------------------------------
802 * WT_UpdateEG1()
803 *----------------------------------------------------------------------------
804 * Purpose:
805 * Calculate the EG1 envelope for the given voice (but do not update any
806 * state)
807 *
808 * Inputs:
809 * pVoice - ptr to the voice whose envelope we want to update
810 * nVoice - this voice's number - used only for debug
811 * pEASData - pointer to overall EAS data structure
812 *
813 * Outputs:
814 * nValue - the envelope value
815 *
816 * Side Effects:
817 * - updates EG1 state value for the given voice
818 *----------------------------------------------------------------------------
819 */
WT_UpdateEG1(S_WT_VOICE * pWTVoice,const S_ENVELOPE * pEnv)820 static void WT_UpdateEG1 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv)
821 {
822 EAS_I32 temp;
823
824 switch (pWTVoice->eg1State)
825 {
826 case eEnvelopeStateAttack:
827 temp = pWTVoice->eg1Value + pWTVoice->eg1Increment;
828
829 /* check if we have reached peak amplitude */
830 if (temp >= SYNTH_FULL_SCALE_EG1_GAIN)
831 {
832 /* limit the volume */
833 temp = SYNTH_FULL_SCALE_EG1_GAIN;
834
835 /* prepare to move to decay state */
836 pWTVoice->eg1State = eEnvelopeStateDecay;
837 pWTVoice->eg1Increment = pEnv->decayTime;
838 }
839
840 break;
841
842 /* exponential decay */
843 case eEnvelopeStateDecay:
844 temp = MULT_EG1_EG1(pWTVoice->eg1Value, pWTVoice->eg1Increment);
845
846 /* check if we have reached sustain level */
847 if (temp <= pEnv->sustainLevel)
848 {
849 /* enforce the sustain level */
850 temp = pEnv->sustainLevel;
851
852 /* if sustain level is zero, skip sustain & release the voice */
853 if (temp > 0)
854 pWTVoice->eg1State = eEnvelopeStateSustain;
855
856 /* move to sustain state */
857 else
858 pWTVoice->eg1State = eEnvelopeStateMuted;
859 }
860
861 break;
862
863 case eEnvelopeStateSustain:
864 return;
865
866 case eEnvelopeStateRelease:
867 temp = MULT_EG1_EG1(pWTVoice->eg1Value, pWTVoice->eg1Increment);
868
869 /* if we hit zero, this voice isn't contributing any audio */
870 if (temp <= 0)
871 {
872 temp = 0;
873 pWTVoice->eg1State = eEnvelopeStateMuted;
874 }
875 break;
876
877 /* voice is muted, set target to zero */
878 case eEnvelopeStateMuted:
879 temp = 0;
880 break;
881
882 case eEnvelopeStateInvalid:
883 default:
884 temp = 0;
885 #ifdef _DEBUG_SYNTH
886 { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_UpdateEG1: error, %d is an unrecognized state\n",
887 pWTVoice->eg1State); */ }
888 #endif
889 break;
890
891 }
892
893 pWTVoice->eg1Value = (EAS_I16) temp;
894 }
895
896 /*----------------------------------------------------------------------------
897 * WT_UpdateEG2()
898 *----------------------------------------------------------------------------
899 * Purpose:
900 * Update the EG2 envelope for the given voice
901 *
902 * Inputs:
903 * pVoice - ptr to the voice whose envelope we want to update
904 * pEASData - pointer to overall EAS data structure
905 *
906 * Outputs:
907 *
908 * Side Effects:
909 * - updates EG2 values for the given voice
910 *----------------------------------------------------------------------------
911 */
912
WT_UpdateEG2(S_WT_VOICE * pWTVoice,const S_ENVELOPE * pEnv)913 static void WT_UpdateEG2 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv)
914 {
915 EAS_I32 temp;
916
917 switch (pWTVoice->eg2State)
918 {
919 case eEnvelopeStateAttack:
920 temp = pWTVoice->eg2Value + pWTVoice->eg2Increment;
921
922 /* check if we have reached peak amplitude */
923 if (temp >= SYNTH_FULL_SCALE_EG1_GAIN)
924 {
925 /* limit the volume */
926 temp = SYNTH_FULL_SCALE_EG1_GAIN;
927
928 /* prepare to move to decay state */
929 pWTVoice->eg2State = eEnvelopeStateDecay;
930
931 pWTVoice->eg2Increment = pEnv->decayTime;
932 }
933
934 break;
935
936 /* implement linear pitch decay in cents */
937 case eEnvelopeStateDecay:
938 temp = pWTVoice->eg2Value -pWTVoice->eg2Increment;
939
940 /* check if we have reached sustain level */
941 if (temp <= pEnv->sustainLevel)
942 {
943 /* enforce the sustain level */
944 temp = pEnv->sustainLevel;
945
946 /* prepare to move to sustain state */
947 pWTVoice->eg2State = eEnvelopeStateSustain;
948 }
949 break;
950
951 case eEnvelopeStateSustain:
952 return;
953
954 case eEnvelopeStateRelease:
955 temp = pWTVoice->eg2Value - pWTVoice->eg2Increment;
956
957 if (temp <= 0)
958 {
959 temp = 0;
960 pWTVoice->eg2State = eEnvelopeStateMuted;
961 }
962
963 break;
964
965 /* voice is muted, set target to zero */
966 case eEnvelopeStateMuted:
967 temp = 0;
968 break;
969
970 case eEnvelopeStateInvalid:
971 default:
972 temp = 0;
973 #ifdef _DEBUG_SYNTH
974 { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_UpdateEG2: error, %d is an unrecognized state\n",
975 pWTVoice->eg2State); */ }
976 #endif
977 break;
978 }
979
980 pWTVoice->eg2Value = (EAS_I16) temp;
981 }
982
983 /*----------------------------------------------------------------------------
984 * WT_UpdateLFO ()
985 *----------------------------------------------------------------------------
986 * Purpose:
987 * Calculate the LFO for the given voice
988 *
989 * Inputs:
990 * pLFO - ptr to the LFO data
991 * phaseInc - phase increment
992 *
993 * Outputs:
994 *
995 * Side Effects:
996 * - updates LFO values for the given voice
997 *----------------------------------------------------------------------------
998 */
WT_UpdateLFO(S_LFO_CONTROL * pLFO,EAS_I16 phaseInc)999 void WT_UpdateLFO (S_LFO_CONTROL *pLFO, EAS_I16 phaseInc)
1000 {
1001
1002 /* To save memory, if m_nPhaseValue is negative, we are in the
1003 * delay phase, and m_nPhaseValue represents the time left
1004 * in the delay.
1005 */
1006 if (pLFO->lfoPhase < 0)
1007 {
1008 pLFO->lfoPhase++;
1009 return;
1010 }
1011
1012 /* calculate LFO output from phase value */
1013 /*lint -e{701} Use shift for performance */
1014 pLFO->lfoValue = (EAS_I16) (pLFO->lfoPhase << 2);
1015 /*lint -e{502} <shortcut to turn sawtooth into triangle wave> */
1016 if ((pLFO->lfoPhase > 0x1fff) && (pLFO->lfoPhase < 0x6000))
1017 pLFO->lfoValue = ~pLFO->lfoValue;
1018
1019 /* update LFO phase */
1020 pLFO->lfoPhase = (pLFO->lfoPhase + phaseInc) & 0x7fff;
1021 }
1022
1023 #ifdef _FILTER_ENABLED
1024 /*----------------------------------------------------------------------------
1025 * WT_UpdateFilter()
1026 *----------------------------------------------------------------------------
1027 * Purpose:
1028 * Update the Filter parameters
1029 *
1030 * Inputs:
1031 * pVoice - ptr to the voice whose filter we want to update
1032 * pEASData - pointer to overall EAS data structure
1033 *
1034 * Outputs:
1035 *
1036 * Side Effects:
1037 * - updates Filter values for the given voice
1038 *----------------------------------------------------------------------------
1039 */
WT_UpdateFilter(S_WT_VOICE * pWTVoice,S_WT_INT_FRAME * pIntFrame,const S_ARTICULATION * pArt)1040 static void WT_UpdateFilter (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pIntFrame, const S_ARTICULATION *pArt)
1041 {
1042 EAS_I32 cutoff;
1043
1044 /* no need to calculate filter coefficients if it is bypassed */
1045 if (pArt->filterCutoff == DEFAULT_EAS_FILTER_CUTOFF_FREQUENCY)
1046 {
1047 pIntFrame->frame.k = 0;
1048 return;
1049 }
1050
1051 /* determine the dynamic cutoff frequency */
1052 cutoff = MULT_EG1_EG1(pWTVoice->eg2Value, pArt->eg2ToFc);
1053 cutoff += pArt->filterCutoff;
1054
1055 /* subtract the A5 offset and the sampling frequency */
1056 cutoff -= FILTER_CUTOFF_FREQ_ADJUST + A5_PITCH_OFFSET_IN_CENTS;
1057
1058 /* limit the cutoff frequency */
1059 if (cutoff > FILTER_CUTOFF_MAX_PITCH_CENTS)
1060 cutoff = FILTER_CUTOFF_MAX_PITCH_CENTS;
1061 else if (cutoff < FILTER_CUTOFF_MIN_PITCH_CENTS)
1062 cutoff = FILTER_CUTOFF_MIN_PITCH_CENTS;
1063
1064 WT_SetFilterCoeffs(pIntFrame, cutoff, pArt->filterQ);
1065 }
1066 #endif
1067
1068 #if defined(_FILTER_ENABLED) || defined(DLS_SYNTHESIZER)
1069 /*----------------------------------------------------------------------------
1070 * coef
1071 *----------------------------------------------------------------------------
1072 * Table of filter coefficients for low-pass filter
1073 *----------------------------------------------------------------------------
1074 *
1075 * polynomial coefficients are based on 8kHz sampling frequency
1076 * filter coef b2 = k2 = k2g0*k^0 + k2g1*k^1*(2^x) + k2g2*k^2*(2^x)
1077 *
1078 *where k2g0, k2g1, k2g2 are from the truncated power series expansion on theta
1079 *(k*2^x = theta, but we incorporate the k along with the k2g0, k2g1, k2g2)
1080 *note: this is a power series in 2^x, not k*2^x
1081 *where k = (2*pi*440)/8kHz == convert octaves to radians
1082 *
1083 * so actually, the following coefs listed as k2g0, k2g1, k2g2 are really
1084 * k2g0*k^0 = k2g0
1085 * k2g1*k^1
1086 * k2g2*k^2
1087 *
1088 *
1089 * filter coef n1 = numerator = n1g0*k^0 + n1g1*k^1*(2^x) + n1g2*k^2*(2^x) + n1g3*k^3*(2^x)
1090 *
1091 *where n1g0, n1g1, n1g2, n1g3 are from the truncated power series expansion on theta
1092 *(k*2^x = theta, but we incorporate the k along with the n1g0, n1g1, n1g2, n2g3)
1093 *note: this is a power series in 2^x, not k*2^x
1094 *where k = (2*pi*440)/8kHz == convert octaves to radians
1095 *we also include the optimization factor of 0.81
1096 *
1097 * so actually, the following coefs listed as n1g0, n1g1, n1g2, n2g3 are really
1098 * n1g0*k^0 = n1g0
1099 * n1g1*k^1
1100 * n1g2*k^2
1101 * n1g3*k^3
1102 *
1103 * NOTE that n1g0 == n1g1 == 0, always, so we only need to store n1g2 and n1g3
1104 *----------------------------------------------------------------------------
1105 */
1106
1107 static const EAS_I16 nk1g0 = -32768;
1108 static const EAS_I16 nk1g2 = 1580;
1109 static const EAS_I16 k2g0 = 32767;
1110
1111 static const EAS_I16 k2g1[] =
1112 {
1113 -11324, /* k2g1[0] = -0.3455751918948761 */
1114 -10387, /* k2g1[1] = -0.3169878073928751 */
1115 -9528, /* k2g1[2] = -0.29076528753345476 */
1116 -8740, /* k2g1[3] = -0.2667120011011279 */
1117 -8017, /* k2g1[4] = -0.24464850028971705 */
1118 -7353, /* k2g1[5] = -0.22441018194495696 */
1119 -6745, /* k2g1[6] = -0.20584605955455101 */
1120 -6187, /* k2g1[7] = -0.18881763682420102 */
1121 -5675, /* k2g1[8] = -0.1731978744360067 */
1122 -5206, /* k2g1[9] = -0.15887024228080968 */
1123 -4775, /* k2g1[10] = -0.14572785009373057 */
1124 -4380, /* k2g1[11] = -0.13367265000706827 */
1125 -4018, /* k2g1[12] = -0.1226147050712642 */
1126 -3685, /* k2g1[13] = -0.11247151828678581 */
1127 -3381, /* k2g1[14] = -0.10316741714122014 */
1128 -3101, /* k2g1[15] = -0.0946329890599603 */
1129 -2844, /* k2g1[16] = -0.08680456355870586 */
1130 -2609, /* k2g1[17] = -0.07962373723441349 */
1131 -2393, /* k2g1[18] = -0.07303693805092666 */
1132 -2195, /* k2g1[19] = -0.06699502566866912 */
1133 -2014, /* k2g1[20] = -0.06145292483669077 */
1134 -1847, /* k2g1[21] = -0.056369289112013346 */
1135 -1694, /* k2g1[22] = -0.05170619239747895 */
1136 -1554, /* k2g1[23] = -0.04742884599684141 */
1137 -1426, /* k2g1[24] = -0.043505339076210514 */
1138 -1308, /* k2g1[25] = -0.03990640059558053 */
1139 -1199, /* k2g1[26] = -0.03660518093435039 */
1140 -1100, /* k2g1[27] = -0.03357705158166837 */
1141 -1009, /* k2g1[28] = -0.030799421397205727 */
1142 -926, /* k2g1[29] = -0.028251568071585884 */
1143 -849 /* k2g1[30] = -0.025914483529091967 */
1144 };
1145
1146 static const EAS_I16 k2g2[] =
1147 {
1148 1957, /* k2g2[0] = 0.059711106626580836 */
1149 1646, /* k2g2[1] = 0.05024063501786333 */
1150 1385, /* k2g2[2] = 0.042272226217199664 */
1151 1165, /* k2g2[3] = 0.03556764576567844 */
1152 981, /* k2g2[4] = 0.029926444346999134 */
1153 825, /* k2g2[5] = 0.025179964880280382 */
1154 694, /* k2g2[6] = 0.02118630011706455 */
1155 584, /* k2g2[7] = 0.01782604998793514 */
1156 491, /* k2g2[8] = 0.014998751854573014 */
1157 414, /* k2g2[9] = 0.012619876941179595 */
1158 348, /* k2g2[10] = 0.010618303146468736 */
1159 293, /* k2g2[11] = 0.008934188679954682 */
1160 246, /* k2g2[12] = 0.007517182949855368 */
1161 207, /* k2g2[13] = 0.006324921212866403 */
1162 174, /* k2g2[14] = 0.005321757979794424 */
1163 147, /* k2g2[15] = 0.004477701309210577 */
1164 123, /* k2g2[16] = 0.00376751612730811 */
1165 104, /* k2g2[17] = 0.0031699697655869644 */
1166 87, /* k2g2[18] = 0.00266719715992703 */
1167 74, /* k2g2[19] = 0.0022441667321724647 */
1168 62, /* k2g2[20] = 0.0018882309854916855 */
1169 52, /* k2g2[21] = 0.0015887483774966232 */
1170 44, /* k2g2[22] = 0.0013367651661223448 */
1171 37, /* k2g2[23] = 0.0011247477162958733 */
1172 31, /* k2g2[24] = 0.0009463572640678758 */
1173 26, /* k2g2[25] = 0.0007962604042473498 */
1174 22, /* k2g2[26] = 0.0006699696356181593 */
1175 18, /* k2g2[27] = 0.0005637091964589207 */
1176 16, /* k2g2[28] = 0.00047430217920125243 */
1177 13, /* k2g2[29] = 0.00039907554925166274 */
1178 11 /* k2g2[30] = 0.00033578022828973666 */
1179 };
1180
1181 static const EAS_I16 n1g2[] =
1182 {
1183 3170, /* n1g2[0] = 0.0967319927350769 */
1184 3036, /* n1g2[1] = 0.0926446051254155 */
1185 2908, /* n1g2[2] = 0.08872992911818503 */
1186 2785, /* n1g2[3] = 0.08498066682523227 */
1187 2667, /* n1g2[4] = 0.08138982872895201 */
1188 2554, /* n1g2[5] = 0.07795072065216213 */
1189 2446, /* n1g2[6] = 0.0746569312785634 */
1190 2343, /* n1g2[7] = 0.07150232020051943 */
1191 2244, /* n1g2[8] = 0.06848100647187474 */
1192 2149, /* n1g2[9] = 0.06558735764447099 */
1193 2058, /* n1g2[10] = 0.06281597926792246 */
1194 1971, /* n1g2[11] = 0.06016170483307614 */
1195 1888, /* n1g2[12] = 0.05761958614040857 */
1196 1808, /* n1g2[13] = 0.05518488407540374 */
1197 1732, /* n1g2[14] = 0.052853059773715245 */
1198 1659, /* n1g2[15] = 0.05061976615964251 */
1199 1589, /* n1g2[16] = 0.04848083984214659 */
1200 1521, /* n1g2[17] = 0.046432293353298 */
1201 1457, /* n1g2[18] = 0.04447030771468711 */
1202 1396, /* n1g2[19] = 0.04259122531793907 */
1203 1337, /* n1g2[20] = 0.040791543106060944 */
1204 1280, /* n1g2[21] = 0.03906790604290942 */
1205 1226, /* n1g2[22] = 0.037417100858604564 */
1206 1174, /* n1g2[23] = 0.035836050059229754 */
1207 1125, /* n1g2[24] = 0.03432180618965023 */
1208 1077, /* n1g2[25] = 0.03287154633875494 */
1209 1032, /* n1g2[26] = 0.03148256687687814 */
1210 988, /* n1g2[27] = 0.030152278415589925 */
1211 946, /* n1g2[28] = 0.028878200980459685 */
1212 906, /* n1g2[29] = 0.02765795938779331 */
1213 868 /* n1g2[30] = 0.02648927881672521 */
1214 };
1215
1216 static const EAS_I16 n1g3[] =
1217 {
1218 -548, /* n1g3[0] = -0.016714088475899017 */
1219 -481, /* n1g3[1] = -0.014683605122742116 */
1220 -423, /* n1g3[2] = -0.012899791676436092 */
1221 -371, /* n1g3[3] = -0.01133268185193299 */
1222 -326, /* n1g3[4] = -0.00995594976868754 */
1223 -287, /* n1g3[5] = -0.008746467702146129 */
1224 -252, /* n1g3[6] = -0.00768391756106361 */
1225 -221, /* n1g3[7] = -0.006750449563854721 */
1226 -194, /* n1g3[8] = -0.005930382380083576 */
1227 -171, /* n1g3[9] = -0.005209939699767622 */
1228 -150, /* n1g3[10] = -0.004577018805123356 */
1229 -132, /* n1g3[11] = -0.004020987256990177 */
1230 -116, /* n1g3[12] = -0.003532504280467257 */
1231 -102, /* n1g3[13] = -0.00310336384922047 */
1232 -89, /* n1g3[14] = -0.002726356832432369 */
1233 -78, /* n1g3[15] = -0.002395149888601605 */
1234 -69, /* n1g3[16] = -0.0021041790717285314 */
1235 -61, /* n1g3[17] = -0.0018485563625771063 */
1236 -53, /* n1g3[18] = -0.001623987554831628 */
1237 -47, /* n1g3[19] = -0.0014267001167177025 */
1238 -41, /* n1g3[20] = -0.0012533798162347005 */
1239 -36, /* n1g3[21] = -0.0011011150453668693 */
1240 -32, /* n1g3[22] = -0.0009673479079754438 */
1241 -28, /* n1g3[23] = -0.0008498312496971563 */
1242 -24, /* n1g3[24] = -0.0007465909079943587 */
1243 -21, /* n1g3[25] = -0.0006558925481952733 */
1244 -19, /* n1g3[26] = -0.0005762125284029567 */
1245 -17, /* n1g3[27] = -0.0005062123038325457 */
1246 -15, /* n1g3[28] = -0.0004447159405951901 */
1247 -13, /* n1g3[29] = -0.00039069036118270117 */
1248 -11 /* n1g3[30] = -0.00034322798979677605 */
1249 };
1250
1251 /*----------------------------------------------------------------------------
1252 * WT_SetFilterCoeffs()
1253 *----------------------------------------------------------------------------
1254 * Purpose:
1255 * Update the Filter parameters
1256 *
1257 * Inputs:
1258 * pVoice - ptr to the voice whose filter we want to update
1259 * pEASData - pointer to overall EAS data structure
1260 *
1261 * Outputs:
1262 *
1263 * Side Effects:
1264 * - updates Filter values for the given voice
1265 *----------------------------------------------------------------------------
1266 */
WT_SetFilterCoeffs(S_WT_INT_FRAME * pIntFrame,EAS_I32 cutoff,EAS_I32 resonance)1267 void WT_SetFilterCoeffs (S_WT_INT_FRAME *pIntFrame, EAS_I32 cutoff, EAS_I32 resonance)
1268 {
1269 EAS_I32 temp;
1270
1271 /*
1272 Convert the cutoff, which has had A5 subtracted, using the 2^x approx
1273 Note, this cutoff is related to theta cutoff by
1274 theta = k * 2^x
1275 We use 2^x and incorporate k in the power series coefs instead
1276 */
1277 cutoff = EAS_Calculate2toX(cutoff);
1278
1279 /* calculate b2 coef */
1280 temp = k2g1[resonance] + MULT_AUDIO_COEF(cutoff, k2g2[resonance]);
1281 temp = k2g0 + MULT_AUDIO_COEF(cutoff, temp);
1282 pIntFrame->frame.b2 = temp;
1283
1284 /* calculate b1 coef */
1285 temp = MULT_AUDIO_COEF(cutoff, nk1g2);
1286 temp = nk1g0 + MULT_AUDIO_COEF(cutoff, temp);
1287 temp += MULT_AUDIO_COEF(temp, pIntFrame->frame.b2);
1288 pIntFrame->frame.b1 = temp >> 1;
1289
1290 /* calculate K coef */
1291 temp = n1g2[resonance] + MULT_AUDIO_COEF(cutoff, n1g3[resonance]);
1292 temp = MULT_AUDIO_COEF(cutoff, temp);
1293 temp = MULT_AUDIO_COEF(cutoff, temp);
1294 pIntFrame->frame.k = temp;
1295 }
1296 #endif
1297
1298