1 /*
2 * Copyright (C) 2015 The Android Open Source Project
3 *
4 * Licensed under the Apache License, Version 2.0 (the "License");
5 * you may not use this file except in compliance with the License.
6 * You may obtain a copy of the License at
7 *
8 * http://www.apache.org/licenses/LICENSE-2.0
9 *
10 * Unless required by applicable law or agreed to in writing, software
11 * distributed under the License is distributed on an "AS IS" BASIS,
12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
13 * See the License for the specific language governing permissions and
14 * limitations under the License.
15 */
16
17 #define LOG_TAG "BufferProvider"
18 //#define LOG_NDEBUG 0
19
20 #include <audio_utils/primitives.h>
21 #include <audio_utils/format.h>
22 #include <external/sonic/sonic.h>
23 #include <media/audiohal/EffectBufferHalInterface.h>
24 #include <media/audiohal/EffectHalInterface.h>
25 #include <media/audiohal/EffectsFactoryHalInterface.h>
26 #include <media/AudioResamplerPublic.h>
27 #include <media/BufferProviders.h>
28 #include <system/audio_effects/effect_downmix.h>
29 #include <utils/Log.h>
30
31 #ifndef ARRAY_SIZE
32 #define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
33 #endif
34
35 namespace android {
36
37 // ----------------------------------------------------------------------------
38
39 template <typename T>
min(const T & a,const T & b)40 static inline T min(const T& a, const T& b)
41 {
42 return a < b ? a : b;
43 }
44
CopyBufferProvider(size_t inputFrameSize,size_t outputFrameSize,size_t bufferFrameCount)45 CopyBufferProvider::CopyBufferProvider(size_t inputFrameSize,
46 size_t outputFrameSize, size_t bufferFrameCount) :
47 mInputFrameSize(inputFrameSize),
48 mOutputFrameSize(outputFrameSize),
49 mLocalBufferFrameCount(bufferFrameCount),
50 mLocalBufferData(NULL),
51 mConsumed(0)
52 {
53 ALOGV("CopyBufferProvider(%p)(%zu, %zu, %zu)", this,
54 inputFrameSize, outputFrameSize, bufferFrameCount);
55 LOG_ALWAYS_FATAL_IF(inputFrameSize < outputFrameSize && bufferFrameCount == 0,
56 "Requires local buffer if inputFrameSize(%zu) < outputFrameSize(%zu)",
57 inputFrameSize, outputFrameSize);
58 if (mLocalBufferFrameCount) {
59 (void)posix_memalign(&mLocalBufferData, 32, mLocalBufferFrameCount * mOutputFrameSize);
60 }
61 mBuffer.frameCount = 0;
62 }
63
~CopyBufferProvider()64 CopyBufferProvider::~CopyBufferProvider()
65 {
66 ALOGV("~CopyBufferProvider(%p)", this);
67 if (mBuffer.frameCount != 0) {
68 mTrackBufferProvider->releaseBuffer(&mBuffer);
69 }
70 free(mLocalBufferData);
71 }
72
getNextBuffer(AudioBufferProvider::Buffer * pBuffer)73 status_t CopyBufferProvider::getNextBuffer(AudioBufferProvider::Buffer *pBuffer)
74 {
75 //ALOGV("CopyBufferProvider(%p)::getNextBuffer(%p (%zu))",
76 // this, pBuffer, pBuffer->frameCount);
77 if (mLocalBufferFrameCount == 0) {
78 status_t res = mTrackBufferProvider->getNextBuffer(pBuffer);
79 if (res == OK) {
80 copyFrames(pBuffer->raw, pBuffer->raw, pBuffer->frameCount);
81 }
82 return res;
83 }
84 if (mBuffer.frameCount == 0) {
85 mBuffer.frameCount = pBuffer->frameCount;
86 status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer);
87 // At one time an upstream buffer provider had
88 // res == OK and mBuffer.frameCount == 0, doesn't seem to happen now 7/18/2014.
89 //
90 // By API spec, if res != OK, then mBuffer.frameCount == 0.
91 // but there may be improper implementations.
92 ALOG_ASSERT(res == OK || mBuffer.frameCount == 0);
93 if (res != OK || mBuffer.frameCount == 0) { // not needed by API spec, but to be safe.
94 pBuffer->raw = NULL;
95 pBuffer->frameCount = 0;
96 return res;
97 }
98 mConsumed = 0;
99 }
100 ALOG_ASSERT(mConsumed < mBuffer.frameCount);
101 size_t count = min(mLocalBufferFrameCount, mBuffer.frameCount - mConsumed);
102 count = min(count, pBuffer->frameCount);
103 pBuffer->raw = mLocalBufferData;
104 pBuffer->frameCount = count;
105 copyFrames(pBuffer->raw, (uint8_t*)mBuffer.raw + mConsumed * mInputFrameSize,
106 pBuffer->frameCount);
107 return OK;
108 }
109
releaseBuffer(AudioBufferProvider::Buffer * pBuffer)110 void CopyBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer)
111 {
112 //ALOGV("CopyBufferProvider(%p)::releaseBuffer(%p(%zu))",
113 // this, pBuffer, pBuffer->frameCount);
114 if (mLocalBufferFrameCount == 0) {
115 mTrackBufferProvider->releaseBuffer(pBuffer);
116 return;
117 }
118 // LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount");
119 mConsumed += pBuffer->frameCount; // TODO: update for efficiency to reuse existing content
120 if (mConsumed != 0 && mConsumed >= mBuffer.frameCount) {
121 mTrackBufferProvider->releaseBuffer(&mBuffer);
122 ALOG_ASSERT(mBuffer.frameCount == 0);
123 }
124 pBuffer->raw = NULL;
125 pBuffer->frameCount = 0;
126 }
127
reset()128 void CopyBufferProvider::reset()
129 {
130 if (mBuffer.frameCount != 0) {
131 mTrackBufferProvider->releaseBuffer(&mBuffer);
132 }
133 mConsumed = 0;
134 }
135
DownmixerBufferProvider(audio_channel_mask_t inputChannelMask,audio_channel_mask_t outputChannelMask,audio_format_t format,uint32_t sampleRate,int32_t sessionId,size_t bufferFrameCount)136 DownmixerBufferProvider::DownmixerBufferProvider(
137 audio_channel_mask_t inputChannelMask,
138 audio_channel_mask_t outputChannelMask, audio_format_t format,
139 uint32_t sampleRate, int32_t sessionId, size_t bufferFrameCount) :
140 CopyBufferProvider(
141 audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(inputChannelMask),
142 audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(outputChannelMask),
143 bufferFrameCount) // set bufferFrameCount to 0 to do in-place
144 {
145 ALOGV("DownmixerBufferProvider(%p)(%#x, %#x, %#x %u %d %d)",
146 this, inputChannelMask, outputChannelMask, format,
147 sampleRate, sessionId, (int)bufferFrameCount);
148 if (!sIsMultichannelCapable) {
149 ALOGE("DownmixerBufferProvider() error: not multichannel capable");
150 return;
151 }
152 mEffectsFactory = EffectsFactoryHalInterface::create();
153 if (mEffectsFactory == 0) {
154 ALOGE("DownmixerBufferProvider() error: could not obtain the effects factory");
155 return;
156 }
157 if (mEffectsFactory->createEffect(&sDwnmFxDesc.uuid,
158 sessionId,
159 SESSION_ID_INVALID_AND_IGNORED,
160 &mDownmixInterface) != 0) {
161 ALOGE("DownmixerBufferProvider() error creating downmixer effect");
162 mDownmixInterface.clear();
163 mEffectsFactory.clear();
164 return;
165 }
166 // channel input configuration will be overridden per-track
167 mDownmixConfig.inputCfg.channels = inputChannelMask; // FIXME: Should be bits
168 mDownmixConfig.outputCfg.channels = outputChannelMask; // FIXME: should be bits
169 mDownmixConfig.inputCfg.format = format;
170 mDownmixConfig.outputCfg.format = format;
171 mDownmixConfig.inputCfg.samplingRate = sampleRate;
172 mDownmixConfig.outputCfg.samplingRate = sampleRate;
173 mDownmixConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
174 mDownmixConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE;
175 // input and output buffer provider, and frame count will not be used as the downmix effect
176 // process() function is called directly (see DownmixerBufferProvider::getNextBuffer())
177 mDownmixConfig.inputCfg.mask = EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS |
178 EFFECT_CONFIG_FORMAT | EFFECT_CONFIG_ACC_MODE;
179 mDownmixConfig.outputCfg.mask = mDownmixConfig.inputCfg.mask;
180
181 mInFrameSize =
182 audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(inputChannelMask);
183 mOutFrameSize =
184 audio_bytes_per_sample(format) * audio_channel_count_from_out_mask(outputChannelMask);
185 status_t status;
186 status = EffectBufferHalInterface::mirror(
187 nullptr, mInFrameSize * bufferFrameCount, &mInBuffer);
188 if (status != 0) {
189 ALOGE("DownmixerBufferProvider() error %d while creating input buffer", status);
190 mDownmixInterface.clear();
191 mEffectsFactory.clear();
192 return;
193 }
194 status = EffectBufferHalInterface::mirror(
195 nullptr, mOutFrameSize * bufferFrameCount, &mOutBuffer);
196 if (status != 0) {
197 ALOGE("DownmixerBufferProvider() error %d while creating output buffer", status);
198 mInBuffer.clear();
199 mDownmixInterface.clear();
200 mEffectsFactory.clear();
201 return;
202 }
203 mDownmixInterface->setInBuffer(mInBuffer);
204 mDownmixInterface->setOutBuffer(mOutBuffer);
205
206 int cmdStatus;
207 uint32_t replySize = sizeof(int);
208
209 // Configure downmixer
210 status = mDownmixInterface->command(
211 EFFECT_CMD_SET_CONFIG /*cmdCode*/, sizeof(effect_config_t) /*cmdSize*/,
212 &mDownmixConfig /*pCmdData*/,
213 &replySize, &cmdStatus /*pReplyData*/);
214 if (status != 0 || cmdStatus != 0) {
215 ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while configuring downmixer",
216 status, cmdStatus);
217 mOutBuffer.clear();
218 mInBuffer.clear();
219 mDownmixInterface.clear();
220 mEffectsFactory.clear();
221 return;
222 }
223
224 // Enable downmixer
225 replySize = sizeof(int);
226 status = mDownmixInterface->command(
227 EFFECT_CMD_ENABLE /*cmdCode*/, 0 /*cmdSize*/, NULL /*pCmdData*/,
228 &replySize, &cmdStatus /*pReplyData*/);
229 if (status != 0 || cmdStatus != 0) {
230 ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while enabling downmixer",
231 status, cmdStatus);
232 mOutBuffer.clear();
233 mInBuffer.clear();
234 mDownmixInterface.clear();
235 mEffectsFactory.clear();
236 return;
237 }
238
239 // Set downmix type
240 // parameter size rounded for padding on 32bit boundary
241 const int psizePadded = ((sizeof(downmix_params_t) - 1)/sizeof(int) + 1) * sizeof(int);
242 const int downmixParamSize =
243 sizeof(effect_param_t) + psizePadded + sizeof(downmix_type_t);
244 effect_param_t * const param = (effect_param_t *) malloc(downmixParamSize);
245 param->psize = sizeof(downmix_params_t);
246 const downmix_params_t downmixParam = DOWNMIX_PARAM_TYPE;
247 memcpy(param->data, &downmixParam, param->psize);
248 const downmix_type_t downmixType = DOWNMIX_TYPE_FOLD;
249 param->vsize = sizeof(downmix_type_t);
250 memcpy(param->data + psizePadded, &downmixType, param->vsize);
251 replySize = sizeof(int);
252 status = mDownmixInterface->command(
253 EFFECT_CMD_SET_PARAM /* cmdCode */, downmixParamSize /* cmdSize */,
254 param /*pCmdData*/, &replySize, &cmdStatus /*pReplyData*/);
255 free(param);
256 if (status != 0 || cmdStatus != 0) {
257 ALOGE("DownmixerBufferProvider() error %d cmdStatus %d while setting downmix type",
258 status, cmdStatus);
259 mOutBuffer.clear();
260 mInBuffer.clear();
261 mDownmixInterface.clear();
262 mEffectsFactory.clear();
263 return;
264 }
265 ALOGV("DownmixerBufferProvider() downmix type set to %d", (int) downmixType);
266 }
267
~DownmixerBufferProvider()268 DownmixerBufferProvider::~DownmixerBufferProvider()
269 {
270 ALOGV("~DownmixerBufferProvider (%p)", this);
271 if (mDownmixInterface != 0) {
272 mDownmixInterface->close();
273 }
274 }
275
copyFrames(void * dst,const void * src,size_t frames)276 void DownmixerBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
277 {
278 mInBuffer->setExternalData(const_cast<void*>(src));
279 mInBuffer->setFrameCount(frames);
280 mInBuffer->update(mInFrameSize * frames);
281 mOutBuffer->setFrameCount(frames);
282 mOutBuffer->setExternalData(dst);
283 if (dst != src) {
284 // Downmix may be accumulating, need to populate the output buffer
285 // with the dst data.
286 mOutBuffer->update(mOutFrameSize * frames);
287 }
288 // may be in-place if src == dst.
289 status_t res = mDownmixInterface->process();
290 if (res == OK) {
291 mOutBuffer->commit(mOutFrameSize * frames);
292 } else {
293 ALOGE("DownmixBufferProvider error %d", res);
294 }
295 }
296
297 /* call once in a pthread_once handler. */
init()298 /*static*/ status_t DownmixerBufferProvider::init()
299 {
300 // find multichannel downmix effect if we have to play multichannel content
301 sp<EffectsFactoryHalInterface> effectsFactory = EffectsFactoryHalInterface::create();
302 if (effectsFactory == 0) {
303 ALOGE("AudioMixer() error: could not obtain the effects factory");
304 return NO_INIT;
305 }
306 uint32_t numEffects = 0;
307 int ret = effectsFactory->queryNumberEffects(&numEffects);
308 if (ret != 0) {
309 ALOGE("AudioMixer() error %d querying number of effects", ret);
310 return NO_INIT;
311 }
312 ALOGV("EffectQueryNumberEffects() numEffects=%d", numEffects);
313
314 for (uint32_t i = 0 ; i < numEffects ; i++) {
315 if (effectsFactory->getDescriptor(i, &sDwnmFxDesc) == 0) {
316 ALOGV("effect %d is called %s", i, sDwnmFxDesc.name);
317 if (memcmp(&sDwnmFxDesc.type, EFFECT_UIID_DOWNMIX, sizeof(effect_uuid_t)) == 0) {
318 ALOGI("found effect \"%s\" from %s",
319 sDwnmFxDesc.name, sDwnmFxDesc.implementor);
320 sIsMultichannelCapable = true;
321 break;
322 }
323 }
324 }
325 ALOGW_IF(!sIsMultichannelCapable, "unable to find downmix effect");
326 return NO_INIT;
327 }
328
329 /*static*/ bool DownmixerBufferProvider::sIsMultichannelCapable = false;
330 /*static*/ effect_descriptor_t DownmixerBufferProvider::sDwnmFxDesc;
331
RemixBufferProvider(audio_channel_mask_t inputChannelMask,audio_channel_mask_t outputChannelMask,audio_format_t format,size_t bufferFrameCount)332 RemixBufferProvider::RemixBufferProvider(audio_channel_mask_t inputChannelMask,
333 audio_channel_mask_t outputChannelMask, audio_format_t format,
334 size_t bufferFrameCount) :
335 CopyBufferProvider(
336 audio_bytes_per_sample(format)
337 * audio_channel_count_from_out_mask(inputChannelMask),
338 audio_bytes_per_sample(format)
339 * audio_channel_count_from_out_mask(outputChannelMask),
340 bufferFrameCount),
341 mFormat(format),
342 mSampleSize(audio_bytes_per_sample(format)),
343 mInputChannels(audio_channel_count_from_out_mask(inputChannelMask)),
344 mOutputChannels(audio_channel_count_from_out_mask(outputChannelMask))
345 {
346 ALOGV("RemixBufferProvider(%p)(%#x, %#x, %#x) %zu %zu",
347 this, format, inputChannelMask, outputChannelMask,
348 mInputChannels, mOutputChannels);
349 (void) memcpy_by_index_array_initialization_from_channel_mask(
350 mIdxAry, ARRAY_SIZE(mIdxAry), outputChannelMask, inputChannelMask);
351 }
352
copyFrames(void * dst,const void * src,size_t frames)353 void RemixBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
354 {
355 memcpy_by_index_array(dst, mOutputChannels,
356 src, mInputChannels, mIdxAry, mSampleSize, frames);
357 }
358
ReformatBufferProvider(int32_t channelCount,audio_format_t inputFormat,audio_format_t outputFormat,size_t bufferFrameCount)359 ReformatBufferProvider::ReformatBufferProvider(int32_t channelCount,
360 audio_format_t inputFormat, audio_format_t outputFormat,
361 size_t bufferFrameCount) :
362 CopyBufferProvider(
363 channelCount * audio_bytes_per_sample(inputFormat),
364 channelCount * audio_bytes_per_sample(outputFormat),
365 bufferFrameCount),
366 mChannelCount(channelCount),
367 mInputFormat(inputFormat),
368 mOutputFormat(outputFormat)
369 {
370 ALOGV("ReformatBufferProvider(%p)(%u, %#x, %#x)",
371 this, channelCount, inputFormat, outputFormat);
372 }
373
copyFrames(void * dst,const void * src,size_t frames)374 void ReformatBufferProvider::copyFrames(void *dst, const void *src, size_t frames)
375 {
376 memcpy_by_audio_format(dst, mOutputFormat, src, mInputFormat, frames * mChannelCount);
377 }
378
TimestretchBufferProvider(int32_t channelCount,audio_format_t format,uint32_t sampleRate,const AudioPlaybackRate & playbackRate)379 TimestretchBufferProvider::TimestretchBufferProvider(int32_t channelCount,
380 audio_format_t format, uint32_t sampleRate, const AudioPlaybackRate &playbackRate) :
381 mChannelCount(channelCount),
382 mFormat(format),
383 mSampleRate(sampleRate),
384 mFrameSize(channelCount * audio_bytes_per_sample(format)),
385 mLocalBufferFrameCount(0),
386 mLocalBufferData(NULL),
387 mRemaining(0),
388 mSonicStream(sonicCreateStream(sampleRate, mChannelCount)),
389 mFallbackFailErrorShown(false),
390 mAudioPlaybackRateValid(false)
391 {
392 LOG_ALWAYS_FATAL_IF(mSonicStream == NULL,
393 "TimestretchBufferProvider can't allocate Sonic stream");
394
395 setPlaybackRate(playbackRate);
396 ALOGV("TimestretchBufferProvider(%p)(%u, %#x, %u %f %f %d %d)",
397 this, channelCount, format, sampleRate, playbackRate.mSpeed,
398 playbackRate.mPitch, playbackRate.mStretchMode, playbackRate.mFallbackMode);
399 mBuffer.frameCount = 0;
400 }
401
~TimestretchBufferProvider()402 TimestretchBufferProvider::~TimestretchBufferProvider()
403 {
404 ALOGV("~TimestretchBufferProvider(%p)", this);
405 sonicDestroyStream(mSonicStream);
406 if (mBuffer.frameCount != 0) {
407 mTrackBufferProvider->releaseBuffer(&mBuffer);
408 }
409 free(mLocalBufferData);
410 }
411
getNextBuffer(AudioBufferProvider::Buffer * pBuffer)412 status_t TimestretchBufferProvider::getNextBuffer(
413 AudioBufferProvider::Buffer *pBuffer)
414 {
415 ALOGV("TimestretchBufferProvider(%p)::getNextBuffer(%p (%zu))",
416 this, pBuffer, pBuffer->frameCount);
417
418 // BYPASS
419 //return mTrackBufferProvider->getNextBuffer(pBuffer);
420
421 // check if previously processed data is sufficient.
422 if (pBuffer->frameCount <= mRemaining) {
423 ALOGV("previous sufficient");
424 pBuffer->raw = mLocalBufferData;
425 return OK;
426 }
427
428 // do we need to resize our buffer?
429 if (pBuffer->frameCount > mLocalBufferFrameCount) {
430 void *newmem;
431 if (posix_memalign(&newmem, 32, pBuffer->frameCount * mFrameSize) == OK) {
432 if (mRemaining != 0) {
433 memcpy(newmem, mLocalBufferData, mRemaining * mFrameSize);
434 }
435 free(mLocalBufferData);
436 mLocalBufferData = newmem;
437 mLocalBufferFrameCount = pBuffer->frameCount;
438 }
439 }
440
441 // need to fetch more data
442 const size_t outputDesired = pBuffer->frameCount - mRemaining;
443 size_t dstAvailable;
444 do {
445 mBuffer.frameCount = mPlaybackRate.mSpeed == AUDIO_TIMESTRETCH_SPEED_NORMAL
446 ? outputDesired : outputDesired * mPlaybackRate.mSpeed + 1;
447
448 status_t res = mTrackBufferProvider->getNextBuffer(&mBuffer);
449
450 ALOG_ASSERT(res == OK || mBuffer.frameCount == 0);
451 if (res != OK || mBuffer.frameCount == 0) { // not needed by API spec, but to be safe.
452 ALOGV("upstream provider cannot provide data");
453 if (mRemaining == 0) {
454 pBuffer->raw = NULL;
455 pBuffer->frameCount = 0;
456 return res;
457 } else { // return partial count
458 pBuffer->raw = mLocalBufferData;
459 pBuffer->frameCount = mRemaining;
460 return OK;
461 }
462 }
463
464 // time-stretch the data
465 dstAvailable = min(mLocalBufferFrameCount - mRemaining, outputDesired);
466 size_t srcAvailable = mBuffer.frameCount;
467 processFrames((uint8_t*)mLocalBufferData + mRemaining * mFrameSize, &dstAvailable,
468 mBuffer.raw, &srcAvailable);
469
470 // release all data consumed
471 mBuffer.frameCount = srcAvailable;
472 mTrackBufferProvider->releaseBuffer(&mBuffer);
473 } while (dstAvailable == 0); // try until we get output data or upstream provider fails.
474
475 // update buffer vars with the actual data processed and return with buffer
476 mRemaining += dstAvailable;
477
478 pBuffer->raw = mLocalBufferData;
479 pBuffer->frameCount = mRemaining;
480
481 return OK;
482 }
483
releaseBuffer(AudioBufferProvider::Buffer * pBuffer)484 void TimestretchBufferProvider::releaseBuffer(AudioBufferProvider::Buffer *pBuffer)
485 {
486 ALOGV("TimestretchBufferProvider(%p)::releaseBuffer(%p (%zu))",
487 this, pBuffer, pBuffer->frameCount);
488
489 // BYPASS
490 //return mTrackBufferProvider->releaseBuffer(pBuffer);
491
492 // LOG_ALWAYS_FATAL_IF(pBuffer->frameCount == 0, "Invalid framecount");
493 if (pBuffer->frameCount < mRemaining) {
494 memcpy(mLocalBufferData,
495 (uint8_t*)mLocalBufferData + pBuffer->frameCount * mFrameSize,
496 (mRemaining - pBuffer->frameCount) * mFrameSize);
497 mRemaining -= pBuffer->frameCount;
498 } else if (pBuffer->frameCount == mRemaining) {
499 mRemaining = 0;
500 } else {
501 LOG_ALWAYS_FATAL("Releasing more frames(%zu) than available(%zu)",
502 pBuffer->frameCount, mRemaining);
503 }
504
505 pBuffer->raw = NULL;
506 pBuffer->frameCount = 0;
507 }
508
reset()509 void TimestretchBufferProvider::reset()
510 {
511 mRemaining = 0;
512 }
513
setPlaybackRate(const AudioPlaybackRate & playbackRate)514 status_t TimestretchBufferProvider::setPlaybackRate(const AudioPlaybackRate &playbackRate)
515 {
516 mPlaybackRate = playbackRate;
517 mFallbackFailErrorShown = false;
518 sonicSetSpeed(mSonicStream, mPlaybackRate.mSpeed);
519 //TODO: pitch is ignored for now
520 //TODO: optimize: if parameters are the same, don't do any extra computation.
521
522 mAudioPlaybackRateValid = isAudioPlaybackRateValid(mPlaybackRate);
523 return OK;
524 }
525
processFrames(void * dstBuffer,size_t * dstFrames,const void * srcBuffer,size_t * srcFrames)526 void TimestretchBufferProvider::processFrames(void *dstBuffer, size_t *dstFrames,
527 const void *srcBuffer, size_t *srcFrames)
528 {
529 ALOGV("processFrames(%zu %zu) remaining(%zu)", *dstFrames, *srcFrames, mRemaining);
530 // Note dstFrames is the required number of frames.
531
532 if (!mAudioPlaybackRateValid) {
533 //fallback mode
534 // Ensure consumption from src is as expected.
535 // TODO: add logic to track "very accurate" consumption related to speed, original sampling
536 // rate, actual frames processed.
537
538 const size_t targetSrc = *dstFrames * mPlaybackRate.mSpeed;
539 if (*srcFrames < targetSrc) { // limit dst frames to that possible
540 *dstFrames = *srcFrames / mPlaybackRate.mSpeed;
541 } else if (*srcFrames > targetSrc + 1) {
542 *srcFrames = targetSrc + 1;
543 }
544 if (*dstFrames > 0) {
545 switch(mPlaybackRate.mFallbackMode) {
546 case AUDIO_TIMESTRETCH_FALLBACK_CUT_REPEAT:
547 if (*dstFrames <= *srcFrames) {
548 size_t copySize = mFrameSize * *dstFrames;
549 memcpy(dstBuffer, srcBuffer, copySize);
550 } else {
551 // cyclically repeat the source.
552 for (size_t count = 0; count < *dstFrames; count += *srcFrames) {
553 size_t remaining = min(*srcFrames, *dstFrames - count);
554 memcpy((uint8_t*)dstBuffer + mFrameSize * count,
555 srcBuffer, mFrameSize * remaining);
556 }
557 }
558 break;
559 case AUDIO_TIMESTRETCH_FALLBACK_DEFAULT:
560 case AUDIO_TIMESTRETCH_FALLBACK_MUTE:
561 memset(dstBuffer,0, mFrameSize * *dstFrames);
562 break;
563 case AUDIO_TIMESTRETCH_FALLBACK_FAIL:
564 default:
565 if(!mFallbackFailErrorShown) {
566 ALOGE("invalid parameters in TimestretchBufferProvider fallbackMode:%d",
567 mPlaybackRate.mFallbackMode);
568 mFallbackFailErrorShown = true;
569 }
570 break;
571 }
572 }
573 } else {
574 switch (mFormat) {
575 case AUDIO_FORMAT_PCM_FLOAT:
576 if (sonicWriteFloatToStream(mSonicStream, (float*)srcBuffer, *srcFrames) != 1) {
577 ALOGE("sonicWriteFloatToStream cannot realloc");
578 *srcFrames = 0; // cannot consume all of srcBuffer
579 }
580 *dstFrames = sonicReadFloatFromStream(mSonicStream, (float*)dstBuffer, *dstFrames);
581 break;
582 case AUDIO_FORMAT_PCM_16_BIT:
583 if (sonicWriteShortToStream(mSonicStream, (short*)srcBuffer, *srcFrames) != 1) {
584 ALOGE("sonicWriteShortToStream cannot realloc");
585 *srcFrames = 0; // cannot consume all of srcBuffer
586 }
587 *dstFrames = sonicReadShortFromStream(mSonicStream, (short*)dstBuffer, *dstFrames);
588 break;
589 default:
590 // could also be caught on construction
591 LOG_ALWAYS_FATAL("invalid format %#x for TimestretchBufferProvider", mFormat);
592 }
593 }
594 }
595 // ----------------------------------------------------------------------------
596 } // namespace android
597