blob: 87092d0a5bd1ea86733d82a72256ad0e38334422 [file] [log] [blame]
Marco Nelissen372be892014-12-04 08:59:22 -08001/*
2 * Copyright (C) 2007 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_NDEBUG 0
18#define LOG_TAG "SoundPool"
19
20#include <inttypes.h>
21
22#include <utils/Log.h>
23
24#define USE_SHARED_MEM_BUFFER
25
26#include <media/AudioTrack.h>
27#include <media/IMediaHTTPService.h>
28#include <media/mediaplayer.h>
29#include <media/stagefright/MediaExtractor.h>
30#include "SoundPool.h"
31#include "SoundPoolThread.h"
32#include <media/AudioPolicyHelper.h>
33#include <ndk/NdkMediaCodec.h>
34#include <ndk/NdkMediaExtractor.h>
35#include <ndk/NdkMediaFormat.h>
36
37namespace android
38{
39
40int kDefaultBufferCount = 4;
41uint32_t kMaxSampleRate = 48000;
42uint32_t kDefaultSampleRate = 44100;
43uint32_t kDefaultFrameCount = 1200;
44size_t kDefaultHeapSize = 1024 * 1024; // 1MB
45
46
47SoundPool::SoundPool(int maxChannels, const audio_attributes_t* pAttributes)
48{
49 ALOGV("SoundPool constructor: maxChannels=%d, attr.usage=%d, attr.flags=0x%x, attr.tags=%s",
50 maxChannels, pAttributes->usage, pAttributes->flags, pAttributes->tags);
51
52 // check limits
53 mMaxChannels = maxChannels;
54 if (mMaxChannels < 1) {
55 mMaxChannels = 1;
56 }
57 else if (mMaxChannels > 32) {
58 mMaxChannels = 32;
59 }
60 ALOGW_IF(maxChannels != mMaxChannels, "App requested %d channels", maxChannels);
61
62 mQuit = false;
Jean-Michel Trivi8e48c692016-10-19 11:52:08 -070063 mMuted = false;
Marco Nelissen372be892014-12-04 08:59:22 -080064 mDecodeThread = 0;
65 memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
66 mAllocated = 0;
67 mNextSampleID = 0;
68 mNextChannelID = 0;
69
70 mCallback = 0;
71 mUserData = 0;
72
73 mChannelPool = new SoundChannel[mMaxChannels];
74 for (int i = 0; i < mMaxChannels; ++i) {
75 mChannelPool[i].init(this);
76 mChannels.push_back(&mChannelPool[i]);
77 }
78
79 // start decode thread
80 startThreads();
81}
82
83SoundPool::~SoundPool()
84{
85 ALOGV("SoundPool destructor");
86 mDecodeThread->quit();
87 quit();
88
89 Mutex::Autolock lock(&mLock);
90
91 mChannels.clear();
92 if (mChannelPool)
93 delete [] mChannelPool;
94 // clean up samples
95 ALOGV("clear samples");
96 mSamples.clear();
97
98 if (mDecodeThread)
99 delete mDecodeThread;
100}
101
102void SoundPool::addToRestartList(SoundChannel* channel)
103{
104 Mutex::Autolock lock(&mRestartLock);
105 if (!mQuit) {
106 mRestart.push_back(channel);
107 mCondition.signal();
108 }
109}
110
111void SoundPool::addToStopList(SoundChannel* channel)
112{
113 Mutex::Autolock lock(&mRestartLock);
114 if (!mQuit) {
115 mStop.push_back(channel);
116 mCondition.signal();
117 }
118}
119
120int SoundPool::beginThread(void* arg)
121{
122 SoundPool* p = (SoundPool*)arg;
123 return p->run();
124}
125
126int SoundPool::run()
127{
128 mRestartLock.lock();
129 while (!mQuit) {
130 mCondition.wait(mRestartLock);
131 ALOGV("awake");
132 if (mQuit) break;
133
134 while (!mStop.empty()) {
135 SoundChannel* channel;
136 ALOGV("Getting channel from stop list");
137 List<SoundChannel* >::iterator iter = mStop.begin();
138 channel = *iter;
139 mStop.erase(iter);
140 mRestartLock.unlock();
141 if (channel != 0) {
142 Mutex::Autolock lock(&mLock);
143 channel->stop();
144 }
145 mRestartLock.lock();
146 if (mQuit) break;
147 }
148
149 while (!mRestart.empty()) {
150 SoundChannel* channel;
151 ALOGV("Getting channel from list");
152 List<SoundChannel*>::iterator iter = mRestart.begin();
153 channel = *iter;
154 mRestart.erase(iter);
155 mRestartLock.unlock();
156 if (channel != 0) {
157 Mutex::Autolock lock(&mLock);
158 channel->nextEvent();
159 }
160 mRestartLock.lock();
161 if (mQuit) break;
162 }
163 }
164
165 mStop.clear();
166 mRestart.clear();
167 mCondition.signal();
168 mRestartLock.unlock();
169 ALOGV("goodbye");
170 return 0;
171}
172
173void SoundPool::quit()
174{
175 mRestartLock.lock();
176 mQuit = true;
177 mCondition.signal();
178 mCondition.wait(mRestartLock);
179 ALOGV("return from quit");
180 mRestartLock.unlock();
181}
182
183bool SoundPool::startThreads()
184{
185 createThreadEtc(beginThread, this, "SoundPool");
186 if (mDecodeThread == NULL)
187 mDecodeThread = new SoundPoolThread(this);
188 return mDecodeThread != NULL;
189}
190
Andy Hung0275a982015-11-30 16:09:55 -0800191sp<Sample> SoundPool::findSample(int sampleID)
192{
193 Mutex::Autolock lock(&mLock);
194 return findSample_l(sampleID);
195}
196
197sp<Sample> SoundPool::findSample_l(int sampleID)
198{
199 return mSamples.valueFor(sampleID);
200}
201
Marco Nelissen372be892014-12-04 08:59:22 -0800202SoundChannel* SoundPool::findChannel(int channelID)
203{
204 for (int i = 0; i < mMaxChannels; ++i) {
205 if (mChannelPool[i].channelID() == channelID) {
206 return &mChannelPool[i];
207 }
208 }
209 return NULL;
210}
211
212SoundChannel* SoundPool::findNextChannel(int channelID)
213{
214 for (int i = 0; i < mMaxChannels; ++i) {
215 if (mChannelPool[i].nextChannelID() == channelID) {
216 return &mChannelPool[i];
217 }
218 }
219 return NULL;
220}
221
222int SoundPool::load(int fd, int64_t offset, int64_t length, int priority __unused)
223{
224 ALOGV("load: fd=%d, offset=%" PRId64 ", length=%" PRId64 ", priority=%d",
225 fd, offset, length, priority);
Andy Hung0275a982015-11-30 16:09:55 -0800226 int sampleID;
227 {
228 Mutex::Autolock lock(&mLock);
229 sampleID = ++mNextSampleID;
230 sp<Sample> sample = new Sample(sampleID, fd, offset, length);
231 mSamples.add(sampleID, sample);
232 sample->startLoad();
233 }
234 // mDecodeThread->loadSample() must be called outside of mLock.
235 // mDecodeThread->loadSample() may block on mDecodeThread message queue space;
236 // the message queue emptying may block on SoundPool::findSample().
237 //
238 // It theoretically possible that sample loads might decode out-of-order.
239 mDecodeThread->loadSample(sampleID);
240 return sampleID;
Marco Nelissen372be892014-12-04 08:59:22 -0800241}
242
243bool SoundPool::unload(int sampleID)
244{
245 ALOGV("unload: sampleID=%d", sampleID);
246 Mutex::Autolock lock(&mLock);
Andy Hunga6238ef2015-05-15 18:39:09 -0700247 return mSamples.removeItem(sampleID) >= 0; // removeItem() returns index or BAD_VALUE
Marco Nelissen372be892014-12-04 08:59:22 -0800248}
249
250int SoundPool::play(int sampleID, float leftVolume, float rightVolume,
251 int priority, int loop, float rate)
252{
253 ALOGV("play sampleID=%d, leftVolume=%f, rightVolume=%f, priority=%d, loop=%d, rate=%f",
254 sampleID, leftVolume, rightVolume, priority, loop, rate);
Marco Nelissen372be892014-12-04 08:59:22 -0800255 SoundChannel* channel;
256 int channelID;
257
258 Mutex::Autolock lock(&mLock);
259
260 if (mQuit) {
261 return 0;
262 }
263 // is sample ready?
Andy Hung0275a982015-11-30 16:09:55 -0800264 sp<Sample> sample(findSample_l(sampleID));
Marco Nelissen372be892014-12-04 08:59:22 -0800265 if ((sample == 0) || (sample->state() != Sample::READY)) {
266 ALOGW(" sample %d not READY", sampleID);
267 return 0;
268 }
269
270 dump();
271
272 // allocate a channel
Andy Hung0c4b81b2015-03-17 23:02:00 +0000273 channel = allocateChannel_l(priority, sampleID);
Marco Nelissen372be892014-12-04 08:59:22 -0800274
275 // no channel allocated - return 0
276 if (!channel) {
277 ALOGV("No channel allocated");
278 return 0;
279 }
280
281 channelID = ++mNextChannelID;
282
283 ALOGV("play channel %p state = %d", channel, channel->state());
284 channel->play(sample, channelID, leftVolume, rightVolume, priority, loop, rate);
285 return channelID;
286}
287
Andy Hung0c4b81b2015-03-17 23:02:00 +0000288SoundChannel* SoundPool::allocateChannel_l(int priority, int sampleID)
Marco Nelissen372be892014-12-04 08:59:22 -0800289{
290 List<SoundChannel*>::iterator iter;
291 SoundChannel* channel = NULL;
292
Andy Hung0c4b81b2015-03-17 23:02:00 +0000293 // check if channel for given sampleID still available
Marco Nelissen372be892014-12-04 08:59:22 -0800294 if (!mChannels.empty()) {
Andy Hung0c4b81b2015-03-17 23:02:00 +0000295 for (iter = mChannels.begin(); iter != mChannels.end(); ++iter) {
296 if (sampleID == (*iter)->getPrevSampleID() && (*iter)->state() == SoundChannel::IDLE) {
297 channel = *iter;
298 mChannels.erase(iter);
299 ALOGV("Allocated recycled channel for same sampleID");
300 break;
301 }
302 }
303 }
304
305 // allocate any channel
306 if (!channel && !mChannels.empty()) {
Marco Nelissen372be892014-12-04 08:59:22 -0800307 iter = mChannels.begin();
308 if (priority >= (*iter)->priority()) {
309 channel = *iter;
310 mChannels.erase(iter);
311 ALOGV("Allocated active channel");
312 }
313 }
314
315 // update priority and put it back in the list
316 if (channel) {
317 channel->setPriority(priority);
318 for (iter = mChannels.begin(); iter != mChannels.end(); ++iter) {
319 if (priority < (*iter)->priority()) {
320 break;
321 }
322 }
323 mChannels.insert(iter, channel);
324 }
325 return channel;
326}
327
328// move a channel from its current position to the front of the list
329void SoundPool::moveToFront_l(SoundChannel* channel)
330{
331 for (List<SoundChannel*>::iterator iter = mChannels.begin(); iter != mChannels.end(); ++iter) {
332 if (*iter == channel) {
333 mChannels.erase(iter);
334 mChannels.push_front(channel);
335 break;
336 }
337 }
338}
339
340void SoundPool::pause(int channelID)
341{
342 ALOGV("pause(%d)", channelID);
343 Mutex::Autolock lock(&mLock);
344 SoundChannel* channel = findChannel(channelID);
345 if (channel) {
346 channel->pause();
347 }
348}
349
350void SoundPool::autoPause()
351{
352 ALOGV("autoPause()");
353 Mutex::Autolock lock(&mLock);
354 for (int i = 0; i < mMaxChannels; ++i) {
355 SoundChannel* channel = &mChannelPool[i];
356 channel->autoPause();
357 }
358}
359
360void SoundPool::resume(int channelID)
361{
362 ALOGV("resume(%d)", channelID);
363 Mutex::Autolock lock(&mLock);
364 SoundChannel* channel = findChannel(channelID);
365 if (channel) {
366 channel->resume();
367 }
368}
369
Jean-Michel Trivi8e48c692016-10-19 11:52:08 -0700370void SoundPool::mute(bool muting)
371{
372 ALOGV("mute(%d)", muting);
373 Mutex::Autolock lock(&mLock);
374 mMuted = muting;
375 if (!mChannels.empty()) {
376 for (List<SoundChannel*>::iterator iter = mChannels.begin();
377 iter != mChannels.end(); ++iter) {
378 (*iter)->mute(muting);
379 }
380 }
381}
382
Marco Nelissen372be892014-12-04 08:59:22 -0800383void SoundPool::autoResume()
384{
385 ALOGV("autoResume()");
386 Mutex::Autolock lock(&mLock);
387 for (int i = 0; i < mMaxChannels; ++i) {
388 SoundChannel* channel = &mChannelPool[i];
389 channel->autoResume();
390 }
391}
392
393void SoundPool::stop(int channelID)
394{
395 ALOGV("stop(%d)", channelID);
396 Mutex::Autolock lock(&mLock);
397 SoundChannel* channel = findChannel(channelID);
398 if (channel) {
399 channel->stop();
400 } else {
401 channel = findNextChannel(channelID);
402 if (channel)
403 channel->clearNextEvent();
404 }
405}
406
407void SoundPool::setVolume(int channelID, float leftVolume, float rightVolume)
408{
409 Mutex::Autolock lock(&mLock);
410 SoundChannel* channel = findChannel(channelID);
411 if (channel) {
412 channel->setVolume(leftVolume, rightVolume);
413 }
414}
415
416void SoundPool::setPriority(int channelID, int priority)
417{
418 ALOGV("setPriority(%d, %d)", channelID, priority);
419 Mutex::Autolock lock(&mLock);
420 SoundChannel* channel = findChannel(channelID);
421 if (channel) {
422 channel->setPriority(priority);
423 }
424}
425
426void SoundPool::setLoop(int channelID, int loop)
427{
428 ALOGV("setLoop(%d, %d)", channelID, loop);
429 Mutex::Autolock lock(&mLock);
430 SoundChannel* channel = findChannel(channelID);
431 if (channel) {
432 channel->setLoop(loop);
433 }
434}
435
436void SoundPool::setRate(int channelID, float rate)
437{
438 ALOGV("setRate(%d, %f)", channelID, rate);
439 Mutex::Autolock lock(&mLock);
440 SoundChannel* channel = findChannel(channelID);
441 if (channel) {
442 channel->setRate(rate);
443 }
444}
445
446// call with lock held
447void SoundPool::done_l(SoundChannel* channel)
448{
449 ALOGV("done_l(%d)", channel->channelID());
450 // if "stolen", play next event
451 if (channel->nextChannelID() != 0) {
452 ALOGV("add to restart list");
453 addToRestartList(channel);
454 }
455
456 // return to idle state
457 else {
458 ALOGV("move to front");
459 moveToFront_l(channel);
460 }
461}
462
463void SoundPool::setCallback(SoundPoolCallback* callback, void* user)
464{
465 Mutex::Autolock lock(&mCallbackLock);
466 mCallback = callback;
467 mUserData = user;
468}
469
470void SoundPool::notify(SoundPoolEvent event)
471{
472 Mutex::Autolock lock(&mCallbackLock);
473 if (mCallback != NULL) {
474 mCallback(event, this, mUserData);
475 }
476}
477
478void SoundPool::dump()
479{
480 for (int i = 0; i < mMaxChannels; ++i) {
481 mChannelPool[i].dump();
482 }
483}
484
485
486Sample::Sample(int sampleID, int fd, int64_t offset, int64_t length)
487{
488 init();
489 mSampleID = sampleID;
490 mFd = dup(fd);
491 mOffset = offset;
492 mLength = length;
493 ALOGV("create sampleID=%d, fd=%d, offset=%" PRId64 " length=%" PRId64,
494 mSampleID, mFd, mLength, mOffset);
495}
496
497void Sample::init()
498{
499 mSize = 0;
500 mRefCount = 0;
501 mSampleID = 0;
502 mState = UNLOADED;
503 mFd = -1;
504 mOffset = 0;
505 mLength = 0;
506}
507
508Sample::~Sample()
509{
510 ALOGV("Sample::destructor sampleID=%d, fd=%d", mSampleID, mFd);
511 if (mFd > 0) {
512 ALOGV("close(%d)", mFd);
513 ::close(mFd);
514 }
515}
516
517static status_t decode(int fd, int64_t offset, int64_t length,
518 uint32_t *rate, int *numChannels, audio_format_t *audioFormat,
519 sp<MemoryHeapBase> heap, size_t *memsize) {
520
Marco Nelissen6cd61102015-01-27 12:17:48 -0800521 ALOGV("fd %d, offset %" PRId64 ", size %" PRId64, fd, offset, length);
Marco Nelissen372be892014-12-04 08:59:22 -0800522 AMediaExtractor *ex = AMediaExtractor_new();
523 status_t err = AMediaExtractor_setDataSourceFd(ex, fd, offset, length);
524
525 if (err != AMEDIA_OK) {
Marco Nelissen06524dc2015-02-10 15:45:23 -0800526 AMediaExtractor_delete(ex);
Marco Nelissen372be892014-12-04 08:59:22 -0800527 return err;
528 }
529
530 *audioFormat = AUDIO_FORMAT_PCM_16_BIT;
531
532 size_t numTracks = AMediaExtractor_getTrackCount(ex);
533 for (size_t i = 0; i < numTracks; i++) {
534 AMediaFormat *format = AMediaExtractor_getTrackFormat(ex, i);
535 const char *mime;
536 if (!AMediaFormat_getString(format, AMEDIAFORMAT_KEY_MIME, &mime)) {
537 AMediaExtractor_delete(ex);
538 AMediaFormat_delete(format);
539 return UNKNOWN_ERROR;
540 }
541 if (strncmp(mime, "audio/", 6) == 0) {
542
543 AMediaCodec *codec = AMediaCodec_createDecoderByType(mime);
Andy Hung26eca012015-04-28 18:43:03 -0700544 if (codec == NULL
545 || AMediaCodec_configure(codec, format,
546 NULL /* window */, NULL /* drm */, 0 /* flags */) != AMEDIA_OK
547 || AMediaCodec_start(codec) != AMEDIA_OK
548 || AMediaExtractor_selectTrack(ex, i) != AMEDIA_OK) {
Marco Nelissen372be892014-12-04 08:59:22 -0800549 AMediaExtractor_delete(ex);
550 AMediaCodec_delete(codec);
551 AMediaFormat_delete(format);
552 return UNKNOWN_ERROR;
553 }
554
555 bool sawInputEOS = false;
556 bool sawOutputEOS = false;
557 uint8_t* writePos = static_cast<uint8_t*>(heap->getBase());
558 size_t available = heap->getSize();
559 size_t written = 0;
560
561 AMediaFormat_delete(format);
562 format = AMediaCodec_getOutputFormat(codec);
563
564 while (!sawOutputEOS) {
565 if (!sawInputEOS) {
566 ssize_t bufidx = AMediaCodec_dequeueInputBuffer(codec, 5000);
Marco Nelissen6cd61102015-01-27 12:17:48 -0800567 ALOGV("input buffer %zd", bufidx);
Marco Nelissen372be892014-12-04 08:59:22 -0800568 if (bufidx >= 0) {
569 size_t bufsize;
570 uint8_t *buf = AMediaCodec_getInputBuffer(codec, bufidx, &bufsize);
Andy Hung04747732016-03-29 15:13:08 -0700571 if (buf == nullptr) {
572 ALOGE("AMediaCodec_getInputBuffer returned nullptr, short decode");
573 break;
574 }
Marco Nelissen372be892014-12-04 08:59:22 -0800575 int sampleSize = AMediaExtractor_readSampleData(ex, buf, bufsize);
576 ALOGV("read %d", sampleSize);
577 if (sampleSize < 0) {
578 sampleSize = 0;
579 sawInputEOS = true;
580 ALOGV("EOS");
581 }
582 int64_t presentationTimeUs = AMediaExtractor_getSampleTime(ex);
583
Andy Hung04747732016-03-29 15:13:08 -0700584 media_status_t mstatus = AMediaCodec_queueInputBuffer(codec, bufidx,
Marco Nelissen372be892014-12-04 08:59:22 -0800585 0 /* offset */, sampleSize, presentationTimeUs,
586 sawInputEOS ? AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM : 0);
Andy Hung04747732016-03-29 15:13:08 -0700587 if (mstatus != AMEDIA_OK) {
588 // AMEDIA_ERROR_UNKNOWN == { -ERANGE -EINVAL -EACCES }
589 ALOGE("AMediaCodec_queueInputBuffer returned status %d, short decode",
590 (int)mstatus);
591 break;
592 }
593 (void)AMediaExtractor_advance(ex);
Marco Nelissen372be892014-12-04 08:59:22 -0800594 }
595 }
596
597 AMediaCodecBufferInfo info;
598 int status = AMediaCodec_dequeueOutputBuffer(codec, &info, 1);
599 ALOGV("dequeueoutput returned: %d", status);
600 if (status >= 0) {
601 if (info.flags & AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM) {
602 ALOGV("output EOS");
603 sawOutputEOS = true;
604 }
605 ALOGV("got decoded buffer size %d", info.size);
606
607 uint8_t *buf = AMediaCodec_getOutputBuffer(codec, status, NULL /* out_size */);
Andy Hung04747732016-03-29 15:13:08 -0700608 if (buf == nullptr) {
609 ALOGE("AMediaCodec_getOutputBuffer returned nullptr, short decode");
610 break;
611 }
Marco Nelissen372be892014-12-04 08:59:22 -0800612 size_t dataSize = info.size;
613 if (dataSize > available) {
614 dataSize = available;
615 }
616 memcpy(writePos, buf + info.offset, dataSize);
617 writePos += dataSize;
618 written += dataSize;
619 available -= dataSize;
Andy Hung04747732016-03-29 15:13:08 -0700620 media_status_t mstatus = AMediaCodec_releaseOutputBuffer(
621 codec, status, false /* render */);
622 if (mstatus != AMEDIA_OK) {
623 // AMEDIA_ERROR_UNKNOWN == { -ERANGE -EINVAL -EACCES }
624 ALOGE("AMediaCodec_releaseOutputBuffer returned status %d, short decode",
625 (int)mstatus);
626 break;
627 }
Marco Nelissen372be892014-12-04 08:59:22 -0800628 if (available == 0) {
629 // there might be more data, but there's no space for it
630 sawOutputEOS = true;
631 }
632 } else if (status == AMEDIACODEC_INFO_OUTPUT_BUFFERS_CHANGED) {
633 ALOGV("output buffers changed");
634 } else if (status == AMEDIACODEC_INFO_OUTPUT_FORMAT_CHANGED) {
635 AMediaFormat_delete(format);
636 format = AMediaCodec_getOutputFormat(codec);
637 ALOGV("format changed to: %s", AMediaFormat_toString(format));
638 } else if (status == AMEDIACODEC_INFO_TRY_AGAIN_LATER) {
639 ALOGV("no output buffer right now");
Marco Nelissen2f01c802015-09-15 12:49:07 -0700640 } else if (status <= AMEDIA_ERROR_BASE) {
641 ALOGE("decode error: %d", status);
642 break;
Marco Nelissen372be892014-12-04 08:59:22 -0800643 } else {
644 ALOGV("unexpected info code: %d", status);
645 }
646 }
647
Andy Hung04747732016-03-29 15:13:08 -0700648 (void)AMediaCodec_stop(codec);
649 (void)AMediaCodec_delete(codec);
650 (void)AMediaExtractor_delete(ex);
Marco Nelissen372be892014-12-04 08:59:22 -0800651 if (!AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_SAMPLE_RATE, (int32_t*) rate) ||
652 !AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_CHANNEL_COUNT, numChannels)) {
Andy Hung04747732016-03-29 15:13:08 -0700653 (void)AMediaFormat_delete(format);
Marco Nelissen372be892014-12-04 08:59:22 -0800654 return UNKNOWN_ERROR;
655 }
Andy Hung04747732016-03-29 15:13:08 -0700656 (void)AMediaFormat_delete(format);
Marco Nelissen372be892014-12-04 08:59:22 -0800657 *memsize = written;
658 return OK;
659 }
Andy Hung04747732016-03-29 15:13:08 -0700660 (void)AMediaFormat_delete(format);
Marco Nelissen372be892014-12-04 08:59:22 -0800661 }
Andy Hung04747732016-03-29 15:13:08 -0700662 (void)AMediaExtractor_delete(ex);
Marco Nelissen372be892014-12-04 08:59:22 -0800663 return UNKNOWN_ERROR;
664}
665
666status_t Sample::doLoad()
667{
668 uint32_t sampleRate;
669 int numChannels;
670 audio_format_t format;
671 status_t status;
672 mHeap = new MemoryHeapBase(kDefaultHeapSize);
673
674 ALOGV("Start decode");
675 status = decode(mFd, mOffset, mLength, &sampleRate, &numChannels, &format,
676 mHeap, &mSize);
677 ALOGV("close(%d)", mFd);
678 ::close(mFd);
679 mFd = -1;
680 if (status != NO_ERROR) {
681 ALOGE("Unable to load sample");
682 goto error;
683 }
684 ALOGV("pointer = %p, size = %zu, sampleRate = %u, numChannels = %d",
685 mHeap->getBase(), mSize, sampleRate, numChannels);
686
687 if (sampleRate > kMaxSampleRate) {
688 ALOGE("Sample rate (%u) out of range", sampleRate);
689 status = BAD_VALUE;
690 goto error;
691 }
692
Glenn Kastenbd2c3d62015-12-14 12:21:03 -0800693 if ((numChannels < 1) || (numChannels > FCC_8)) {
Marco Nelissen372be892014-12-04 08:59:22 -0800694 ALOGE("Sample channel count (%d) out of range", numChannels);
695 status = BAD_VALUE;
696 goto error;
697 }
698
699 mData = new MemoryBase(mHeap, 0, mSize);
700 mSampleRate = sampleRate;
701 mNumChannels = numChannels;
702 mFormat = format;
703 mState = READY;
704 return NO_ERROR;
705
706error:
707 mHeap.clear();
708 return status;
709}
710
711
712void SoundChannel::init(SoundPool* soundPool)
713{
714 mSoundPool = soundPool;
Andy Hung0c4b81b2015-03-17 23:02:00 +0000715 mPrevSampleID = -1;
Marco Nelissen372be892014-12-04 08:59:22 -0800716}
717
718// call with sound pool lock held
719void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftVolume,
720 float rightVolume, int priority, int loop, float rate)
721{
722 sp<AudioTrack> oldTrack;
723 sp<AudioTrack> newTrack;
Andy Hung0c4b81b2015-03-17 23:02:00 +0000724 status_t status = NO_ERROR;
Marco Nelissen372be892014-12-04 08:59:22 -0800725
726 { // scope for the lock
727 Mutex::Autolock lock(&mLock);
728
729 ALOGV("SoundChannel::play %p: sampleID=%d, channelID=%d, leftVolume=%f, rightVolume=%f,"
730 " priority=%d, loop=%d, rate=%f",
731 this, sample->sampleID(), nextChannelID, leftVolume, rightVolume,
732 priority, loop, rate);
733
734 // if not idle, this voice is being stolen
735 if (mState != IDLE) {
736 ALOGV("channel %d stolen - event queued for channel %d", channelID(), nextChannelID);
737 mNextEvent.set(sample, nextChannelID, leftVolume, rightVolume, priority, loop, rate);
738 stop_l();
739 return;
740 }
741
742 // initialize track
743 size_t afFrameCount;
744 uint32_t afSampleRate;
745 audio_stream_type_t streamType = audio_attributes_to_stream_type(mSoundPool->attributes());
746 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
747 afFrameCount = kDefaultFrameCount;
748 }
749 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
750 afSampleRate = kDefaultSampleRate;
751 }
752 int numChannels = sample->numChannels();
753 uint32_t sampleRate = uint32_t(float(sample->sampleRate()) * rate + 0.5);
754 size_t frameCount = 0;
755
756 if (loop) {
Andy Hunga1c35162015-03-06 15:00:42 -0800757 const audio_format_t format = sample->format();
758 const size_t frameSize = audio_is_linear_pcm(format)
759 ? numChannels * audio_bytes_per_sample(format) : 1;
760 frameCount = sample->size() / frameSize;
Marco Nelissen372be892014-12-04 08:59:22 -0800761 }
762
763#ifndef USE_SHARED_MEM_BUFFER
764 uint32_t totalFrames = (kDefaultBufferCount * afFrameCount * sampleRate) / afSampleRate;
765 // Ensure minimum audio buffer size in case of short looped sample
766 if(frameCount < totalFrames) {
767 frameCount = totalFrames;
768 }
769#endif
770
Andy Hung32ccb692015-03-27 18:27:27 -0700771 // check if the existing track has the same sample id.
772 if (mAudioTrack != 0 && mPrevSampleID == sample->sampleID()) {
773 // the sample rate may fail to change if the audio track is a fast track.
774 if (mAudioTrack->setSampleRate(sampleRate) == NO_ERROR) {
775 newTrack = mAudioTrack;
776 ALOGV("reusing track %p for sample %d", mAudioTrack.get(), sample->sampleID());
777 }
778 }
779 if (newTrack == 0) {
Andy Hung0c4b81b2015-03-17 23:02:00 +0000780 // mToggle toggles each time a track is started on a given channel.
781 // The toggle is concatenated with the SoundChannel address and passed to AudioTrack
782 // as callback user data. This enables the detection of callbacks received from the old
783 // audio track while the new one is being started and avoids processing them with
784 // wrong audio audio buffer size (mAudioBufferSize)
785 unsigned long toggle = mToggle ^ 1;
786 void *userData = (void *)((unsigned long)this | toggle);
787 audio_channel_mask_t channelMask = audio_channel_out_mask_from_count(numChannels);
Marco Nelissen372be892014-12-04 08:59:22 -0800788
Andy Hung0c4b81b2015-03-17 23:02:00 +0000789 // do not create a new audio track if current track is compatible with sample parameters
790 #ifdef USE_SHARED_MEM_BUFFER
791 newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
Jean-Michel Trivi6c307872015-05-13 19:03:21 -0700792 channelMask, sample->getIMemory(), AUDIO_OUTPUT_FLAG_FAST, callback, userData,
793 0 /*default notification frames*/, AUDIO_SESSION_ALLOCATE,
794 AudioTrack::TRANSFER_DEFAULT,
795 NULL /*offloadInfo*/, -1 /*uid*/, -1 /*pid*/, mSoundPool->attributes());
Andy Hung0c4b81b2015-03-17 23:02:00 +0000796 #else
797 uint32_t bufferFrames = (totalFrames + (kDefaultBufferCount - 1)) / kDefaultBufferCount;
798 newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
799 channelMask, frameCount, AUDIO_OUTPUT_FLAG_FAST, callback, userData,
Jean-Michel Trivi6c307872015-05-13 19:03:21 -0700800 bufferFrames, AUDIO_SESSION_ALLOCATE, AudioTrack::TRANSFER_DEFAULT,
801 NULL /*offloadInfo*/, -1 /*uid*/, -1 /*pid*/, mSoundPool->attributes());
Andy Hung0c4b81b2015-03-17 23:02:00 +0000802 #endif
803 oldTrack = mAudioTrack;
804 status = newTrack->initCheck();
805 if (status != NO_ERROR) {
806 ALOGE("Error creating AudioTrack");
Glenn Kasten14d226a2015-05-18 13:53:39 -0700807 // newTrack goes out of scope, so reference count drops to zero
Andy Hung0c4b81b2015-03-17 23:02:00 +0000808 goto exit;
809 }
810 // From now on, AudioTrack callbacks received with previous toggle value will be ignored.
811 mToggle = toggle;
812 mAudioTrack = newTrack;
Andy Hungbc453732015-03-17 23:05:12 +0000813 ALOGV("using new track %p for sample %d", newTrack.get(), sample->sampleID());
Marco Nelissen372be892014-12-04 08:59:22 -0800814 }
Marco Nelissen372be892014-12-04 08:59:22 -0800815 newTrack->setVolume(leftVolume, rightVolume);
816 newTrack->setLoop(0, frameCount, loop);
Marco Nelissen372be892014-12-04 08:59:22 -0800817 mPos = 0;
818 mSample = sample;
819 mChannelID = nextChannelID;
820 mPriority = priority;
821 mLoop = loop;
822 mLeftVolume = leftVolume;
823 mRightVolume = rightVolume;
824 mNumChannels = numChannels;
825 mRate = rate;
826 clearNextEvent();
827 mState = PLAYING;
828 mAudioTrack->start();
829 mAudioBufferSize = newTrack->frameCount()*newTrack->frameSize();
830 }
831
832exit:
833 ALOGV("delete oldTrack %p", oldTrack.get());
834 if (status != NO_ERROR) {
835 mAudioTrack.clear();
836 }
837}
838
839void SoundChannel::nextEvent()
840{
841 sp<Sample> sample;
842 int nextChannelID;
843 float leftVolume;
844 float rightVolume;
845 int priority;
846 int loop;
847 float rate;
848
849 // check for valid event
850 {
851 Mutex::Autolock lock(&mLock);
852 nextChannelID = mNextEvent.channelID();
853 if (nextChannelID == 0) {
854 ALOGV("stolen channel has no event");
855 return;
856 }
857
858 sample = mNextEvent.sample();
859 leftVolume = mNextEvent.leftVolume();
860 rightVolume = mNextEvent.rightVolume();
861 priority = mNextEvent.priority();
862 loop = mNextEvent.loop();
863 rate = mNextEvent.rate();
864 }
865
866 ALOGV("Starting stolen channel %d -> %d", channelID(), nextChannelID);
867 play(sample, nextChannelID, leftVolume, rightVolume, priority, loop, rate);
868}
869
870void SoundChannel::callback(int event, void* user, void *info)
871{
872 SoundChannel* channel = static_cast<SoundChannel*>((void *)((unsigned long)user & ~1));
873
874 channel->process(event, info, (unsigned long)user & 1);
875}
876
877void SoundChannel::process(int event, void *info, unsigned long toggle)
878{
879 //ALOGV("process(%d)", mChannelID);
880
881 Mutex::Autolock lock(&mLock);
882
883 AudioTrack::Buffer* b = NULL;
884 if (event == AudioTrack::EVENT_MORE_DATA) {
885 b = static_cast<AudioTrack::Buffer *>(info);
886 }
887
888 if (mToggle != toggle) {
889 ALOGV("process wrong toggle %p channel %d", this, mChannelID);
890 if (b != NULL) {
891 b->size = 0;
892 }
893 return;
894 }
895
896 sp<Sample> sample = mSample;
897
898// ALOGV("SoundChannel::process event %d", event);
899
900 if (event == AudioTrack::EVENT_MORE_DATA) {
901
902 // check for stop state
903 if (b->size == 0) return;
904
905 if (mState == IDLE) {
906 b->size = 0;
907 return;
908 }
909
910 if (sample != 0) {
911 // fill buffer
912 uint8_t* q = (uint8_t*) b->i8;
913 size_t count = 0;
914
915 if (mPos < (int)sample->size()) {
916 uint8_t* p = sample->data() + mPos;
917 count = sample->size() - mPos;
918 if (count > b->size) {
919 count = b->size;
920 }
921 memcpy(q, p, count);
922// ALOGV("fill: q=%p, p=%p, mPos=%u, b->size=%u, count=%d", q, p, mPos, b->size,
923// count);
924 } else if (mPos < mAudioBufferSize) {
925 count = mAudioBufferSize - mPos;
926 if (count > b->size) {
927 count = b->size;
928 }
929 memset(q, 0, count);
930// ALOGV("fill extra: q=%p, mPos=%u, b->size=%u, count=%d", q, mPos, b->size, count);
931 }
932
933 mPos += count;
934 b->size = count;
935 //ALOGV("buffer=%p, [0]=%d", b->i16, b->i16[0]);
936 }
937 } else if (event == AudioTrack::EVENT_UNDERRUN || event == AudioTrack::EVENT_BUFFER_END) {
938 ALOGV("process %p channel %d event %s",
939 this, mChannelID, (event == AudioTrack::EVENT_UNDERRUN) ? "UNDERRUN" :
940 "BUFFER_END");
941 mSoundPool->addToStopList(this);
942 } else if (event == AudioTrack::EVENT_LOOP_END) {
943 ALOGV("End loop %p channel %d", this, mChannelID);
944 } else if (event == AudioTrack::EVENT_NEW_IAUDIOTRACK) {
945 ALOGV("process %p channel %d NEW_IAUDIOTRACK", this, mChannelID);
946 } else {
947 ALOGW("SoundChannel::process unexpected event %d", event);
948 }
949}
950
951
952// call with lock held
953bool SoundChannel::doStop_l()
954{
955 if (mState != IDLE) {
956 setVolume_l(0, 0);
957 ALOGV("stop");
958 mAudioTrack->stop();
Andy Hung0c4b81b2015-03-17 23:02:00 +0000959 mPrevSampleID = mSample->sampleID();
Marco Nelissen372be892014-12-04 08:59:22 -0800960 mSample.clear();
961 mState = IDLE;
962 mPriority = IDLE_PRIORITY;
963 return true;
964 }
965 return false;
966}
967
968// call with lock held and sound pool lock held
969void SoundChannel::stop_l()
970{
971 if (doStop_l()) {
972 mSoundPool->done_l(this);
973 }
974}
975
976// call with sound pool lock held
977void SoundChannel::stop()
978{
979 bool stopped;
980 {
981 Mutex::Autolock lock(&mLock);
982 stopped = doStop_l();
983 }
984
985 if (stopped) {
986 mSoundPool->done_l(this);
987 }
988}
989
990//FIXME: Pause is a little broken right now
991void SoundChannel::pause()
992{
993 Mutex::Autolock lock(&mLock);
994 if (mState == PLAYING) {
995 ALOGV("pause track");
996 mState = PAUSED;
997 mAudioTrack->pause();
998 }
999}
1000
1001void SoundChannel::autoPause()
1002{
1003 Mutex::Autolock lock(&mLock);
1004 if (mState == PLAYING) {
1005 ALOGV("pause track");
1006 mState = PAUSED;
1007 mAutoPaused = true;
1008 mAudioTrack->pause();
1009 }
1010}
1011
1012void SoundChannel::resume()
1013{
1014 Mutex::Autolock lock(&mLock);
1015 if (mState == PAUSED) {
1016 ALOGV("resume track");
1017 mState = PLAYING;
1018 mAutoPaused = false;
1019 mAudioTrack->start();
1020 }
1021}
1022
1023void SoundChannel::autoResume()
1024{
1025 Mutex::Autolock lock(&mLock);
1026 if (mAutoPaused && (mState == PAUSED)) {
1027 ALOGV("resume track");
1028 mState = PLAYING;
1029 mAutoPaused = false;
1030 mAudioTrack->start();
1031 }
1032}
1033
1034void SoundChannel::setRate(float rate)
1035{
1036 Mutex::Autolock lock(&mLock);
1037 if (mAudioTrack != NULL && mSample != 0) {
1038 uint32_t sampleRate = uint32_t(float(mSample->sampleRate()) * rate + 0.5);
1039 mAudioTrack->setSampleRate(sampleRate);
1040 mRate = rate;
1041 }
1042}
1043
1044// call with lock held
1045void SoundChannel::setVolume_l(float leftVolume, float rightVolume)
1046{
1047 mLeftVolume = leftVolume;
1048 mRightVolume = rightVolume;
Jean-Michel Trivi8e48c692016-10-19 11:52:08 -07001049 if (mAudioTrack != NULL && !mMuted)
Marco Nelissen372be892014-12-04 08:59:22 -08001050 mAudioTrack->setVolume(leftVolume, rightVolume);
1051}
1052
1053void SoundChannel::setVolume(float leftVolume, float rightVolume)
1054{
1055 Mutex::Autolock lock(&mLock);
1056 setVolume_l(leftVolume, rightVolume);
1057}
1058
Jean-Michel Trivi8e48c692016-10-19 11:52:08 -07001059void SoundChannel::mute(bool muting)
1060{
1061 Mutex::Autolock lock(&mLock);
1062 mMuted = muting;
1063 if (mAudioTrack != NULL) {
1064 if (mMuted) {
1065 mAudioTrack->setVolume(0.0f, 0.0f);
1066 } else {
1067 mAudioTrack->setVolume(mLeftVolume, mRightVolume);
1068 }
1069 }
1070}
1071
Marco Nelissen372be892014-12-04 08:59:22 -08001072void SoundChannel::setLoop(int loop)
1073{
1074 Mutex::Autolock lock(&mLock);
1075 if (mAudioTrack != NULL && mSample != 0) {
1076 uint32_t loopEnd = mSample->size()/mNumChannels/
1077 ((mSample->format() == AUDIO_FORMAT_PCM_16_BIT) ? sizeof(int16_t) : sizeof(uint8_t));
1078 mAudioTrack->setLoop(0, loopEnd, loop);
1079 mLoop = loop;
1080 }
1081}
1082
1083SoundChannel::~SoundChannel()
1084{
1085 ALOGV("SoundChannel destructor %p", this);
1086 {
1087 Mutex::Autolock lock(&mLock);
1088 clearNextEvent();
1089 doStop_l();
1090 }
1091 // do not call AudioTrack destructor with mLock held as it will wait for the AudioTrack
1092 // callback thread to exit which may need to execute process() and acquire the mLock.
1093 mAudioTrack.clear();
1094}
1095
1096void SoundChannel::dump()
1097{
1098 ALOGV("mState = %d mChannelID=%d, mNumChannels=%d, mPos = %d, mPriority=%d, mLoop=%d",
1099 mState, mChannelID, mNumChannels, mPos, mPriority, mLoop);
1100}
1101
1102void SoundEvent::set(const sp<Sample>& sample, int channelID, float leftVolume,
1103 float rightVolume, int priority, int loop, float rate)
1104{
1105 mSample = sample;
1106 mChannelID = channelID;
1107 mLeftVolume = leftVolume;
1108 mRightVolume = rightVolume;
1109 mPriority = priority;
1110 mLoop = loop;
1111 mRate =rate;
1112}
1113
1114} // end namespace android