Marco Nelissen | 372be89 | 2014-12-04 08:59:22 -0800 | [diff] [blame] | 1 | /* |
| 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 | |
| 37 | namespace android |
| 38 | { |
| 39 | |
| 40 | int kDefaultBufferCount = 4; |
| 41 | uint32_t kMaxSampleRate = 48000; |
| 42 | uint32_t kDefaultSampleRate = 44100; |
| 43 | uint32_t kDefaultFrameCount = 1200; |
| 44 | size_t kDefaultHeapSize = 1024 * 1024; // 1MB |
| 45 | |
| 46 | |
| 47 | SoundPool::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; |
| 63 | mDecodeThread = 0; |
| 64 | memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t)); |
| 65 | mAllocated = 0; |
| 66 | mNextSampleID = 0; |
| 67 | mNextChannelID = 0; |
| 68 | |
| 69 | mCallback = 0; |
| 70 | mUserData = 0; |
| 71 | |
| 72 | mChannelPool = new SoundChannel[mMaxChannels]; |
| 73 | for (int i = 0; i < mMaxChannels; ++i) { |
| 74 | mChannelPool[i].init(this); |
| 75 | mChannels.push_back(&mChannelPool[i]); |
| 76 | } |
| 77 | |
| 78 | // start decode thread |
| 79 | startThreads(); |
| 80 | } |
| 81 | |
| 82 | SoundPool::~SoundPool() |
| 83 | { |
| 84 | ALOGV("SoundPool destructor"); |
| 85 | mDecodeThread->quit(); |
| 86 | quit(); |
| 87 | |
| 88 | Mutex::Autolock lock(&mLock); |
| 89 | |
| 90 | mChannels.clear(); |
| 91 | if (mChannelPool) |
| 92 | delete [] mChannelPool; |
| 93 | // clean up samples |
| 94 | ALOGV("clear samples"); |
| 95 | mSamples.clear(); |
| 96 | |
| 97 | if (mDecodeThread) |
| 98 | delete mDecodeThread; |
| 99 | } |
| 100 | |
| 101 | void SoundPool::addToRestartList(SoundChannel* channel) |
| 102 | { |
| 103 | Mutex::Autolock lock(&mRestartLock); |
| 104 | if (!mQuit) { |
| 105 | mRestart.push_back(channel); |
| 106 | mCondition.signal(); |
| 107 | } |
| 108 | } |
| 109 | |
| 110 | void SoundPool::addToStopList(SoundChannel* channel) |
| 111 | { |
| 112 | Mutex::Autolock lock(&mRestartLock); |
| 113 | if (!mQuit) { |
| 114 | mStop.push_back(channel); |
| 115 | mCondition.signal(); |
| 116 | } |
| 117 | } |
| 118 | |
| 119 | int SoundPool::beginThread(void* arg) |
| 120 | { |
| 121 | SoundPool* p = (SoundPool*)arg; |
| 122 | return p->run(); |
| 123 | } |
| 124 | |
| 125 | int SoundPool::run() |
| 126 | { |
| 127 | mRestartLock.lock(); |
| 128 | while (!mQuit) { |
| 129 | mCondition.wait(mRestartLock); |
| 130 | ALOGV("awake"); |
| 131 | if (mQuit) break; |
| 132 | |
| 133 | while (!mStop.empty()) { |
| 134 | SoundChannel* channel; |
| 135 | ALOGV("Getting channel from stop list"); |
| 136 | List<SoundChannel* >::iterator iter = mStop.begin(); |
| 137 | channel = *iter; |
| 138 | mStop.erase(iter); |
| 139 | mRestartLock.unlock(); |
| 140 | if (channel != 0) { |
| 141 | Mutex::Autolock lock(&mLock); |
| 142 | channel->stop(); |
| 143 | } |
| 144 | mRestartLock.lock(); |
| 145 | if (mQuit) break; |
| 146 | } |
| 147 | |
| 148 | while (!mRestart.empty()) { |
| 149 | SoundChannel* channel; |
| 150 | ALOGV("Getting channel from list"); |
| 151 | List<SoundChannel*>::iterator iter = mRestart.begin(); |
| 152 | channel = *iter; |
| 153 | mRestart.erase(iter); |
| 154 | mRestartLock.unlock(); |
| 155 | if (channel != 0) { |
| 156 | Mutex::Autolock lock(&mLock); |
| 157 | channel->nextEvent(); |
| 158 | } |
| 159 | mRestartLock.lock(); |
| 160 | if (mQuit) break; |
| 161 | } |
| 162 | } |
| 163 | |
| 164 | mStop.clear(); |
| 165 | mRestart.clear(); |
| 166 | mCondition.signal(); |
| 167 | mRestartLock.unlock(); |
| 168 | ALOGV("goodbye"); |
| 169 | return 0; |
| 170 | } |
| 171 | |
| 172 | void SoundPool::quit() |
| 173 | { |
| 174 | mRestartLock.lock(); |
| 175 | mQuit = true; |
| 176 | mCondition.signal(); |
| 177 | mCondition.wait(mRestartLock); |
| 178 | ALOGV("return from quit"); |
| 179 | mRestartLock.unlock(); |
| 180 | } |
| 181 | |
| 182 | bool SoundPool::startThreads() |
| 183 | { |
| 184 | createThreadEtc(beginThread, this, "SoundPool"); |
| 185 | if (mDecodeThread == NULL) |
| 186 | mDecodeThread = new SoundPoolThread(this); |
| 187 | return mDecodeThread != NULL; |
| 188 | } |
| 189 | |
| 190 | SoundChannel* SoundPool::findChannel(int channelID) |
| 191 | { |
| 192 | for (int i = 0; i < mMaxChannels; ++i) { |
| 193 | if (mChannelPool[i].channelID() == channelID) { |
| 194 | return &mChannelPool[i]; |
| 195 | } |
| 196 | } |
| 197 | return NULL; |
| 198 | } |
| 199 | |
| 200 | SoundChannel* SoundPool::findNextChannel(int channelID) |
| 201 | { |
| 202 | for (int i = 0; i < mMaxChannels; ++i) { |
| 203 | if (mChannelPool[i].nextChannelID() == channelID) { |
| 204 | return &mChannelPool[i]; |
| 205 | } |
| 206 | } |
| 207 | return NULL; |
| 208 | } |
| 209 | |
| 210 | int SoundPool::load(int fd, int64_t offset, int64_t length, int priority __unused) |
| 211 | { |
| 212 | ALOGV("load: fd=%d, offset=%" PRId64 ", length=%" PRId64 ", priority=%d", |
| 213 | fd, offset, length, priority); |
| 214 | Mutex::Autolock lock(&mLock); |
| 215 | sp<Sample> sample = new Sample(++mNextSampleID, fd, offset, length); |
| 216 | mSamples.add(sample->sampleID(), sample); |
| 217 | doLoad(sample); |
| 218 | return sample->sampleID(); |
| 219 | } |
| 220 | |
| 221 | void SoundPool::doLoad(sp<Sample>& sample) |
| 222 | { |
| 223 | ALOGV("doLoad: loading sample sampleID=%d", sample->sampleID()); |
| 224 | sample->startLoad(); |
| 225 | mDecodeThread->loadSample(sample->sampleID()); |
| 226 | } |
| 227 | |
| 228 | bool SoundPool::unload(int sampleID) |
| 229 | { |
| 230 | ALOGV("unload: sampleID=%d", sampleID); |
| 231 | Mutex::Autolock lock(&mLock); |
| 232 | return mSamples.removeItem(sampleID); |
| 233 | } |
| 234 | |
| 235 | int SoundPool::play(int sampleID, float leftVolume, float rightVolume, |
| 236 | int priority, int loop, float rate) |
| 237 | { |
| 238 | ALOGV("play sampleID=%d, leftVolume=%f, rightVolume=%f, priority=%d, loop=%d, rate=%f", |
| 239 | sampleID, leftVolume, rightVolume, priority, loop, rate); |
| 240 | sp<Sample> sample; |
| 241 | SoundChannel* channel; |
| 242 | int channelID; |
| 243 | |
| 244 | Mutex::Autolock lock(&mLock); |
| 245 | |
| 246 | if (mQuit) { |
| 247 | return 0; |
| 248 | } |
| 249 | // is sample ready? |
| 250 | sample = findSample(sampleID); |
| 251 | if ((sample == 0) || (sample->state() != Sample::READY)) { |
| 252 | ALOGW(" sample %d not READY", sampleID); |
| 253 | return 0; |
| 254 | } |
| 255 | |
| 256 | dump(); |
| 257 | |
| 258 | // allocate a channel |
Glenn Kasten | 18c225d | 2015-03-09 08:36:35 -0700 | [diff] [blame] | 259 | channel = allocateChannel_l(priority); |
Marco Nelissen | 372be89 | 2014-12-04 08:59:22 -0800 | [diff] [blame] | 260 | |
| 261 | // no channel allocated - return 0 |
| 262 | if (!channel) { |
| 263 | ALOGV("No channel allocated"); |
| 264 | return 0; |
| 265 | } |
| 266 | |
| 267 | channelID = ++mNextChannelID; |
| 268 | |
| 269 | ALOGV("play channel %p state = %d", channel, channel->state()); |
| 270 | channel->play(sample, channelID, leftVolume, rightVolume, priority, loop, rate); |
| 271 | return channelID; |
| 272 | } |
| 273 | |
Glenn Kasten | 18c225d | 2015-03-09 08:36:35 -0700 | [diff] [blame] | 274 | SoundChannel* SoundPool::allocateChannel_l(int priority) |
Marco Nelissen | 372be89 | 2014-12-04 08:59:22 -0800 | [diff] [blame] | 275 | { |
| 276 | List<SoundChannel*>::iterator iter; |
| 277 | SoundChannel* channel = NULL; |
| 278 | |
Glenn Kasten | 18c225d | 2015-03-09 08:36:35 -0700 | [diff] [blame] | 279 | // allocate a channel |
Marco Nelissen | 372be89 | 2014-12-04 08:59:22 -0800 | [diff] [blame] | 280 | if (!mChannels.empty()) { |
| 281 | iter = mChannels.begin(); |
| 282 | if (priority >= (*iter)->priority()) { |
| 283 | channel = *iter; |
| 284 | mChannels.erase(iter); |
| 285 | ALOGV("Allocated active channel"); |
| 286 | } |
| 287 | } |
| 288 | |
| 289 | // update priority and put it back in the list |
| 290 | if (channel) { |
| 291 | channel->setPriority(priority); |
| 292 | for (iter = mChannels.begin(); iter != mChannels.end(); ++iter) { |
| 293 | if (priority < (*iter)->priority()) { |
| 294 | break; |
| 295 | } |
| 296 | } |
| 297 | mChannels.insert(iter, channel); |
| 298 | } |
| 299 | return channel; |
| 300 | } |
| 301 | |
| 302 | // move a channel from its current position to the front of the list |
| 303 | void SoundPool::moveToFront_l(SoundChannel* channel) |
| 304 | { |
| 305 | for (List<SoundChannel*>::iterator iter = mChannels.begin(); iter != mChannels.end(); ++iter) { |
| 306 | if (*iter == channel) { |
| 307 | mChannels.erase(iter); |
| 308 | mChannels.push_front(channel); |
| 309 | break; |
| 310 | } |
| 311 | } |
| 312 | } |
| 313 | |
| 314 | void SoundPool::pause(int channelID) |
| 315 | { |
| 316 | ALOGV("pause(%d)", channelID); |
| 317 | Mutex::Autolock lock(&mLock); |
| 318 | SoundChannel* channel = findChannel(channelID); |
| 319 | if (channel) { |
| 320 | channel->pause(); |
| 321 | } |
| 322 | } |
| 323 | |
| 324 | void SoundPool::autoPause() |
| 325 | { |
| 326 | ALOGV("autoPause()"); |
| 327 | Mutex::Autolock lock(&mLock); |
| 328 | for (int i = 0; i < mMaxChannels; ++i) { |
| 329 | SoundChannel* channel = &mChannelPool[i]; |
| 330 | channel->autoPause(); |
| 331 | } |
| 332 | } |
| 333 | |
| 334 | void SoundPool::resume(int channelID) |
| 335 | { |
| 336 | ALOGV("resume(%d)", channelID); |
| 337 | Mutex::Autolock lock(&mLock); |
| 338 | SoundChannel* channel = findChannel(channelID); |
| 339 | if (channel) { |
| 340 | channel->resume(); |
| 341 | } |
| 342 | } |
| 343 | |
| 344 | void SoundPool::autoResume() |
| 345 | { |
| 346 | ALOGV("autoResume()"); |
| 347 | Mutex::Autolock lock(&mLock); |
| 348 | for (int i = 0; i < mMaxChannels; ++i) { |
| 349 | SoundChannel* channel = &mChannelPool[i]; |
| 350 | channel->autoResume(); |
| 351 | } |
| 352 | } |
| 353 | |
| 354 | void SoundPool::stop(int channelID) |
| 355 | { |
| 356 | ALOGV("stop(%d)", channelID); |
| 357 | Mutex::Autolock lock(&mLock); |
| 358 | SoundChannel* channel = findChannel(channelID); |
| 359 | if (channel) { |
| 360 | channel->stop(); |
| 361 | } else { |
| 362 | channel = findNextChannel(channelID); |
| 363 | if (channel) |
| 364 | channel->clearNextEvent(); |
| 365 | } |
| 366 | } |
| 367 | |
| 368 | void SoundPool::setVolume(int channelID, float leftVolume, float rightVolume) |
| 369 | { |
| 370 | Mutex::Autolock lock(&mLock); |
| 371 | SoundChannel* channel = findChannel(channelID); |
| 372 | if (channel) { |
| 373 | channel->setVolume(leftVolume, rightVolume); |
| 374 | } |
| 375 | } |
| 376 | |
| 377 | void SoundPool::setPriority(int channelID, int priority) |
| 378 | { |
| 379 | ALOGV("setPriority(%d, %d)", channelID, priority); |
| 380 | Mutex::Autolock lock(&mLock); |
| 381 | SoundChannel* channel = findChannel(channelID); |
| 382 | if (channel) { |
| 383 | channel->setPriority(priority); |
| 384 | } |
| 385 | } |
| 386 | |
| 387 | void SoundPool::setLoop(int channelID, int loop) |
| 388 | { |
| 389 | ALOGV("setLoop(%d, %d)", channelID, loop); |
| 390 | Mutex::Autolock lock(&mLock); |
| 391 | SoundChannel* channel = findChannel(channelID); |
| 392 | if (channel) { |
| 393 | channel->setLoop(loop); |
| 394 | } |
| 395 | } |
| 396 | |
| 397 | void SoundPool::setRate(int channelID, float rate) |
| 398 | { |
| 399 | ALOGV("setRate(%d, %f)", channelID, rate); |
| 400 | Mutex::Autolock lock(&mLock); |
| 401 | SoundChannel* channel = findChannel(channelID); |
| 402 | if (channel) { |
| 403 | channel->setRate(rate); |
| 404 | } |
| 405 | } |
| 406 | |
| 407 | // call with lock held |
| 408 | void SoundPool::done_l(SoundChannel* channel) |
| 409 | { |
| 410 | ALOGV("done_l(%d)", channel->channelID()); |
| 411 | // if "stolen", play next event |
| 412 | if (channel->nextChannelID() != 0) { |
| 413 | ALOGV("add to restart list"); |
| 414 | addToRestartList(channel); |
| 415 | } |
| 416 | |
| 417 | // return to idle state |
| 418 | else { |
| 419 | ALOGV("move to front"); |
| 420 | moveToFront_l(channel); |
| 421 | } |
| 422 | } |
| 423 | |
| 424 | void SoundPool::setCallback(SoundPoolCallback* callback, void* user) |
| 425 | { |
| 426 | Mutex::Autolock lock(&mCallbackLock); |
| 427 | mCallback = callback; |
| 428 | mUserData = user; |
| 429 | } |
| 430 | |
| 431 | void SoundPool::notify(SoundPoolEvent event) |
| 432 | { |
| 433 | Mutex::Autolock lock(&mCallbackLock); |
| 434 | if (mCallback != NULL) { |
| 435 | mCallback(event, this, mUserData); |
| 436 | } |
| 437 | } |
| 438 | |
| 439 | void SoundPool::dump() |
| 440 | { |
| 441 | for (int i = 0; i < mMaxChannels; ++i) { |
| 442 | mChannelPool[i].dump(); |
| 443 | } |
| 444 | } |
| 445 | |
| 446 | |
| 447 | Sample::Sample(int sampleID, int fd, int64_t offset, int64_t length) |
| 448 | { |
| 449 | init(); |
| 450 | mSampleID = sampleID; |
| 451 | mFd = dup(fd); |
| 452 | mOffset = offset; |
| 453 | mLength = length; |
| 454 | ALOGV("create sampleID=%d, fd=%d, offset=%" PRId64 " length=%" PRId64, |
| 455 | mSampleID, mFd, mLength, mOffset); |
| 456 | } |
| 457 | |
| 458 | void Sample::init() |
| 459 | { |
| 460 | mSize = 0; |
| 461 | mRefCount = 0; |
| 462 | mSampleID = 0; |
| 463 | mState = UNLOADED; |
| 464 | mFd = -1; |
| 465 | mOffset = 0; |
| 466 | mLength = 0; |
| 467 | } |
| 468 | |
| 469 | Sample::~Sample() |
| 470 | { |
| 471 | ALOGV("Sample::destructor sampleID=%d, fd=%d", mSampleID, mFd); |
| 472 | if (mFd > 0) { |
| 473 | ALOGV("close(%d)", mFd); |
| 474 | ::close(mFd); |
| 475 | } |
| 476 | } |
| 477 | |
| 478 | static status_t decode(int fd, int64_t offset, int64_t length, |
| 479 | uint32_t *rate, int *numChannels, audio_format_t *audioFormat, |
| 480 | sp<MemoryHeapBase> heap, size_t *memsize) { |
| 481 | |
Marco Nelissen | 6cd6110 | 2015-01-27 12:17:48 -0800 | [diff] [blame] | 482 | ALOGV("fd %d, offset %" PRId64 ", size %" PRId64, fd, offset, length); |
Marco Nelissen | 372be89 | 2014-12-04 08:59:22 -0800 | [diff] [blame] | 483 | AMediaExtractor *ex = AMediaExtractor_new(); |
| 484 | status_t err = AMediaExtractor_setDataSourceFd(ex, fd, offset, length); |
| 485 | |
| 486 | if (err != AMEDIA_OK) { |
Marco Nelissen | 06524dc | 2015-02-10 15:45:23 -0800 | [diff] [blame] | 487 | AMediaExtractor_delete(ex); |
Marco Nelissen | 372be89 | 2014-12-04 08:59:22 -0800 | [diff] [blame] | 488 | return err; |
| 489 | } |
| 490 | |
| 491 | *audioFormat = AUDIO_FORMAT_PCM_16_BIT; |
| 492 | |
| 493 | size_t numTracks = AMediaExtractor_getTrackCount(ex); |
| 494 | for (size_t i = 0; i < numTracks; i++) { |
| 495 | AMediaFormat *format = AMediaExtractor_getTrackFormat(ex, i); |
| 496 | const char *mime; |
| 497 | if (!AMediaFormat_getString(format, AMEDIAFORMAT_KEY_MIME, &mime)) { |
| 498 | AMediaExtractor_delete(ex); |
| 499 | AMediaFormat_delete(format); |
| 500 | return UNKNOWN_ERROR; |
| 501 | } |
| 502 | if (strncmp(mime, "audio/", 6) == 0) { |
| 503 | |
| 504 | AMediaCodec *codec = AMediaCodec_createDecoderByType(mime); |
| 505 | if (AMediaCodec_configure(codec, format, |
| 506 | NULL /* window */, NULL /* drm */, 0 /* flags */) != AMEDIA_OK |
| 507 | || AMediaCodec_start(codec) != AMEDIA_OK |
| 508 | || AMediaExtractor_selectTrack(ex, i) != AMEDIA_OK) { |
| 509 | AMediaExtractor_delete(ex); |
| 510 | AMediaCodec_delete(codec); |
| 511 | AMediaFormat_delete(format); |
| 512 | return UNKNOWN_ERROR; |
| 513 | } |
| 514 | |
| 515 | bool sawInputEOS = false; |
| 516 | bool sawOutputEOS = false; |
| 517 | uint8_t* writePos = static_cast<uint8_t*>(heap->getBase()); |
| 518 | size_t available = heap->getSize(); |
| 519 | size_t written = 0; |
| 520 | |
| 521 | AMediaFormat_delete(format); |
| 522 | format = AMediaCodec_getOutputFormat(codec); |
| 523 | |
| 524 | while (!sawOutputEOS) { |
| 525 | if (!sawInputEOS) { |
| 526 | ssize_t bufidx = AMediaCodec_dequeueInputBuffer(codec, 5000); |
Marco Nelissen | 6cd6110 | 2015-01-27 12:17:48 -0800 | [diff] [blame] | 527 | ALOGV("input buffer %zd", bufidx); |
Marco Nelissen | 372be89 | 2014-12-04 08:59:22 -0800 | [diff] [blame] | 528 | if (bufidx >= 0) { |
| 529 | size_t bufsize; |
| 530 | uint8_t *buf = AMediaCodec_getInputBuffer(codec, bufidx, &bufsize); |
| 531 | int sampleSize = AMediaExtractor_readSampleData(ex, buf, bufsize); |
| 532 | ALOGV("read %d", sampleSize); |
| 533 | if (sampleSize < 0) { |
| 534 | sampleSize = 0; |
| 535 | sawInputEOS = true; |
| 536 | ALOGV("EOS"); |
| 537 | } |
| 538 | int64_t presentationTimeUs = AMediaExtractor_getSampleTime(ex); |
| 539 | |
| 540 | AMediaCodec_queueInputBuffer(codec, bufidx, |
| 541 | 0 /* offset */, sampleSize, presentationTimeUs, |
| 542 | sawInputEOS ? AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM : 0); |
| 543 | AMediaExtractor_advance(ex); |
| 544 | } |
| 545 | } |
| 546 | |
| 547 | AMediaCodecBufferInfo info; |
| 548 | int status = AMediaCodec_dequeueOutputBuffer(codec, &info, 1); |
| 549 | ALOGV("dequeueoutput returned: %d", status); |
| 550 | if (status >= 0) { |
| 551 | if (info.flags & AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM) { |
| 552 | ALOGV("output EOS"); |
| 553 | sawOutputEOS = true; |
| 554 | } |
| 555 | ALOGV("got decoded buffer size %d", info.size); |
| 556 | |
| 557 | uint8_t *buf = AMediaCodec_getOutputBuffer(codec, status, NULL /* out_size */); |
| 558 | size_t dataSize = info.size; |
| 559 | if (dataSize > available) { |
| 560 | dataSize = available; |
| 561 | } |
| 562 | memcpy(writePos, buf + info.offset, dataSize); |
| 563 | writePos += dataSize; |
| 564 | written += dataSize; |
| 565 | available -= dataSize; |
| 566 | AMediaCodec_releaseOutputBuffer(codec, status, false /* render */); |
| 567 | if (available == 0) { |
| 568 | // there might be more data, but there's no space for it |
| 569 | sawOutputEOS = true; |
| 570 | } |
| 571 | } else if (status == AMEDIACODEC_INFO_OUTPUT_BUFFERS_CHANGED) { |
| 572 | ALOGV("output buffers changed"); |
| 573 | } else if (status == AMEDIACODEC_INFO_OUTPUT_FORMAT_CHANGED) { |
| 574 | AMediaFormat_delete(format); |
| 575 | format = AMediaCodec_getOutputFormat(codec); |
| 576 | ALOGV("format changed to: %s", AMediaFormat_toString(format)); |
| 577 | } else if (status == AMEDIACODEC_INFO_TRY_AGAIN_LATER) { |
| 578 | ALOGV("no output buffer right now"); |
| 579 | } else { |
| 580 | ALOGV("unexpected info code: %d", status); |
| 581 | } |
| 582 | } |
| 583 | |
| 584 | AMediaCodec_stop(codec); |
| 585 | AMediaCodec_delete(codec); |
| 586 | AMediaExtractor_delete(ex); |
| 587 | if (!AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_SAMPLE_RATE, (int32_t*) rate) || |
| 588 | !AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_CHANNEL_COUNT, numChannels)) { |
| 589 | AMediaFormat_delete(format); |
| 590 | return UNKNOWN_ERROR; |
| 591 | } |
| 592 | AMediaFormat_delete(format); |
| 593 | *memsize = written; |
| 594 | return OK; |
| 595 | } |
| 596 | AMediaFormat_delete(format); |
| 597 | } |
| 598 | AMediaExtractor_delete(ex); |
| 599 | return UNKNOWN_ERROR; |
| 600 | } |
| 601 | |
| 602 | status_t Sample::doLoad() |
| 603 | { |
| 604 | uint32_t sampleRate; |
| 605 | int numChannels; |
| 606 | audio_format_t format; |
| 607 | status_t status; |
| 608 | mHeap = new MemoryHeapBase(kDefaultHeapSize); |
| 609 | |
| 610 | ALOGV("Start decode"); |
| 611 | status = decode(mFd, mOffset, mLength, &sampleRate, &numChannels, &format, |
| 612 | mHeap, &mSize); |
| 613 | ALOGV("close(%d)", mFd); |
| 614 | ::close(mFd); |
| 615 | mFd = -1; |
| 616 | if (status != NO_ERROR) { |
| 617 | ALOGE("Unable to load sample"); |
| 618 | goto error; |
| 619 | } |
| 620 | ALOGV("pointer = %p, size = %zu, sampleRate = %u, numChannels = %d", |
| 621 | mHeap->getBase(), mSize, sampleRate, numChannels); |
| 622 | |
| 623 | if (sampleRate > kMaxSampleRate) { |
| 624 | ALOGE("Sample rate (%u) out of range", sampleRate); |
| 625 | status = BAD_VALUE; |
| 626 | goto error; |
| 627 | } |
| 628 | |
Andy Hung | a1c3516 | 2015-03-06 15:00:42 -0800 | [diff] [blame] | 629 | if ((numChannels < 1) || (numChannels > 8)) { |
Marco Nelissen | 372be89 | 2014-12-04 08:59:22 -0800 | [diff] [blame] | 630 | ALOGE("Sample channel count (%d) out of range", numChannels); |
| 631 | status = BAD_VALUE; |
| 632 | goto error; |
| 633 | } |
| 634 | |
| 635 | mData = new MemoryBase(mHeap, 0, mSize); |
| 636 | mSampleRate = sampleRate; |
| 637 | mNumChannels = numChannels; |
| 638 | mFormat = format; |
| 639 | mState = READY; |
| 640 | return NO_ERROR; |
| 641 | |
| 642 | error: |
| 643 | mHeap.clear(); |
| 644 | return status; |
| 645 | } |
| 646 | |
| 647 | |
| 648 | void SoundChannel::init(SoundPool* soundPool) |
| 649 | { |
| 650 | mSoundPool = soundPool; |
| 651 | } |
| 652 | |
| 653 | // call with sound pool lock held |
| 654 | void SoundChannel::play(const sp<Sample>& sample, int nextChannelID, float leftVolume, |
| 655 | float rightVolume, int priority, int loop, float rate) |
| 656 | { |
| 657 | sp<AudioTrack> oldTrack; |
| 658 | sp<AudioTrack> newTrack; |
Glenn Kasten | 18c225d | 2015-03-09 08:36:35 -0700 | [diff] [blame] | 659 | status_t status; |
Marco Nelissen | 372be89 | 2014-12-04 08:59:22 -0800 | [diff] [blame] | 660 | |
| 661 | { // scope for the lock |
| 662 | Mutex::Autolock lock(&mLock); |
| 663 | |
| 664 | ALOGV("SoundChannel::play %p: sampleID=%d, channelID=%d, leftVolume=%f, rightVolume=%f," |
| 665 | " priority=%d, loop=%d, rate=%f", |
| 666 | this, sample->sampleID(), nextChannelID, leftVolume, rightVolume, |
| 667 | priority, loop, rate); |
| 668 | |
| 669 | // if not idle, this voice is being stolen |
| 670 | if (mState != IDLE) { |
| 671 | ALOGV("channel %d stolen - event queued for channel %d", channelID(), nextChannelID); |
| 672 | mNextEvent.set(sample, nextChannelID, leftVolume, rightVolume, priority, loop, rate); |
| 673 | stop_l(); |
| 674 | return; |
| 675 | } |
| 676 | |
| 677 | // initialize track |
| 678 | size_t afFrameCount; |
| 679 | uint32_t afSampleRate; |
| 680 | audio_stream_type_t streamType = audio_attributes_to_stream_type(mSoundPool->attributes()); |
| 681 | if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { |
| 682 | afFrameCount = kDefaultFrameCount; |
| 683 | } |
| 684 | if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { |
| 685 | afSampleRate = kDefaultSampleRate; |
| 686 | } |
| 687 | int numChannels = sample->numChannels(); |
| 688 | uint32_t sampleRate = uint32_t(float(sample->sampleRate()) * rate + 0.5); |
| 689 | size_t frameCount = 0; |
| 690 | |
| 691 | if (loop) { |
Andy Hung | a1c3516 | 2015-03-06 15:00:42 -0800 | [diff] [blame] | 692 | const audio_format_t format = sample->format(); |
| 693 | const size_t frameSize = audio_is_linear_pcm(format) |
| 694 | ? numChannels * audio_bytes_per_sample(format) : 1; |
| 695 | frameCount = sample->size() / frameSize; |
Marco Nelissen | 372be89 | 2014-12-04 08:59:22 -0800 | [diff] [blame] | 696 | } |
| 697 | |
| 698 | #ifndef USE_SHARED_MEM_BUFFER |
| 699 | uint32_t totalFrames = (kDefaultBufferCount * afFrameCount * sampleRate) / afSampleRate; |
| 700 | // Ensure minimum audio buffer size in case of short looped sample |
| 701 | if(frameCount < totalFrames) { |
| 702 | frameCount = totalFrames; |
| 703 | } |
| 704 | #endif |
| 705 | |
Glenn Kasten | 18c225d | 2015-03-09 08:36:35 -0700 | [diff] [blame] | 706 | // mToggle toggles each time a track is started on a given channel. |
| 707 | // The toggle is concatenated with the SoundChannel address and passed to AudioTrack |
| 708 | // as callback user data. This enables the detection of callbacks received from the old |
| 709 | // audio track while the new one is being started and avoids processing them with |
| 710 | // wrong audio audio buffer size (mAudioBufferSize) |
| 711 | unsigned long toggle = mToggle ^ 1; |
| 712 | void *userData = (void *)((unsigned long)this | toggle); |
| 713 | audio_channel_mask_t channelMask = audio_channel_out_mask_from_count(numChannels); |
Marco Nelissen | 372be89 | 2014-12-04 08:59:22 -0800 | [diff] [blame] | 714 | |
Glenn Kasten | 18c225d | 2015-03-09 08:36:35 -0700 | [diff] [blame] | 715 | // do not create a new audio track if current track is compatible with sample parameters |
| 716 | #ifdef USE_SHARED_MEM_BUFFER |
| 717 | newTrack = new AudioTrack(streamType, sampleRate, sample->format(), |
| 718 | channelMask, sample->getIMemory(), AUDIO_OUTPUT_FLAG_FAST, callback, userData); |
| 719 | #else |
| 720 | uint32_t bufferFrames = (totalFrames + (kDefaultBufferCount - 1)) / kDefaultBufferCount; |
| 721 | newTrack = new AudioTrack(streamType, sampleRate, sample->format(), |
| 722 | channelMask, frameCount, AUDIO_OUTPUT_FLAG_FAST, callback, userData, |
| 723 | bufferFrames); |
| 724 | #endif |
| 725 | oldTrack = mAudioTrack; |
| 726 | status = newTrack->initCheck(); |
| 727 | if (status != NO_ERROR) { |
| 728 | ALOGE("Error creating AudioTrack"); |
| 729 | goto exit; |
Marco Nelissen | 372be89 | 2014-12-04 08:59:22 -0800 | [diff] [blame] | 730 | } |
Glenn Kasten | 18c225d | 2015-03-09 08:36:35 -0700 | [diff] [blame] | 731 | ALOGV("setVolume %p", newTrack.get()); |
Marco Nelissen | 372be89 | 2014-12-04 08:59:22 -0800 | [diff] [blame] | 732 | newTrack->setVolume(leftVolume, rightVolume); |
| 733 | newTrack->setLoop(0, frameCount, loop); |
Glenn Kasten | 18c225d | 2015-03-09 08:36:35 -0700 | [diff] [blame] | 734 | |
| 735 | // From now on, AudioTrack callbacks received with previous toggle value will be ignored. |
| 736 | mToggle = toggle; |
| 737 | mAudioTrack = newTrack; |
Marco Nelissen | 372be89 | 2014-12-04 08:59:22 -0800 | [diff] [blame] | 738 | mPos = 0; |
| 739 | mSample = sample; |
| 740 | mChannelID = nextChannelID; |
| 741 | mPriority = priority; |
| 742 | mLoop = loop; |
| 743 | mLeftVolume = leftVolume; |
| 744 | mRightVolume = rightVolume; |
| 745 | mNumChannels = numChannels; |
| 746 | mRate = rate; |
| 747 | clearNextEvent(); |
| 748 | mState = PLAYING; |
| 749 | mAudioTrack->start(); |
| 750 | mAudioBufferSize = newTrack->frameCount()*newTrack->frameSize(); |
| 751 | } |
| 752 | |
| 753 | exit: |
| 754 | ALOGV("delete oldTrack %p", oldTrack.get()); |
| 755 | if (status != NO_ERROR) { |
| 756 | mAudioTrack.clear(); |
| 757 | } |
| 758 | } |
| 759 | |
| 760 | void SoundChannel::nextEvent() |
| 761 | { |
| 762 | sp<Sample> sample; |
| 763 | int nextChannelID; |
| 764 | float leftVolume; |
| 765 | float rightVolume; |
| 766 | int priority; |
| 767 | int loop; |
| 768 | float rate; |
| 769 | |
| 770 | // check for valid event |
| 771 | { |
| 772 | Mutex::Autolock lock(&mLock); |
| 773 | nextChannelID = mNextEvent.channelID(); |
| 774 | if (nextChannelID == 0) { |
| 775 | ALOGV("stolen channel has no event"); |
| 776 | return; |
| 777 | } |
| 778 | |
| 779 | sample = mNextEvent.sample(); |
| 780 | leftVolume = mNextEvent.leftVolume(); |
| 781 | rightVolume = mNextEvent.rightVolume(); |
| 782 | priority = mNextEvent.priority(); |
| 783 | loop = mNextEvent.loop(); |
| 784 | rate = mNextEvent.rate(); |
| 785 | } |
| 786 | |
| 787 | ALOGV("Starting stolen channel %d -> %d", channelID(), nextChannelID); |
| 788 | play(sample, nextChannelID, leftVolume, rightVolume, priority, loop, rate); |
| 789 | } |
| 790 | |
| 791 | void SoundChannel::callback(int event, void* user, void *info) |
| 792 | { |
| 793 | SoundChannel* channel = static_cast<SoundChannel*>((void *)((unsigned long)user & ~1)); |
| 794 | |
| 795 | channel->process(event, info, (unsigned long)user & 1); |
| 796 | } |
| 797 | |
| 798 | void SoundChannel::process(int event, void *info, unsigned long toggle) |
| 799 | { |
| 800 | //ALOGV("process(%d)", mChannelID); |
| 801 | |
| 802 | Mutex::Autolock lock(&mLock); |
| 803 | |
| 804 | AudioTrack::Buffer* b = NULL; |
| 805 | if (event == AudioTrack::EVENT_MORE_DATA) { |
| 806 | b = static_cast<AudioTrack::Buffer *>(info); |
| 807 | } |
| 808 | |
| 809 | if (mToggle != toggle) { |
| 810 | ALOGV("process wrong toggle %p channel %d", this, mChannelID); |
| 811 | if (b != NULL) { |
| 812 | b->size = 0; |
| 813 | } |
| 814 | return; |
| 815 | } |
| 816 | |
| 817 | sp<Sample> sample = mSample; |
| 818 | |
| 819 | // ALOGV("SoundChannel::process event %d", event); |
| 820 | |
| 821 | if (event == AudioTrack::EVENT_MORE_DATA) { |
| 822 | |
| 823 | // check for stop state |
| 824 | if (b->size == 0) return; |
| 825 | |
| 826 | if (mState == IDLE) { |
| 827 | b->size = 0; |
| 828 | return; |
| 829 | } |
| 830 | |
| 831 | if (sample != 0) { |
| 832 | // fill buffer |
| 833 | uint8_t* q = (uint8_t*) b->i8; |
| 834 | size_t count = 0; |
| 835 | |
| 836 | if (mPos < (int)sample->size()) { |
| 837 | uint8_t* p = sample->data() + mPos; |
| 838 | count = sample->size() - mPos; |
| 839 | if (count > b->size) { |
| 840 | count = b->size; |
| 841 | } |
| 842 | memcpy(q, p, count); |
| 843 | // ALOGV("fill: q=%p, p=%p, mPos=%u, b->size=%u, count=%d", q, p, mPos, b->size, |
| 844 | // count); |
| 845 | } else if (mPos < mAudioBufferSize) { |
| 846 | count = mAudioBufferSize - mPos; |
| 847 | if (count > b->size) { |
| 848 | count = b->size; |
| 849 | } |
| 850 | memset(q, 0, count); |
| 851 | // ALOGV("fill extra: q=%p, mPos=%u, b->size=%u, count=%d", q, mPos, b->size, count); |
| 852 | } |
| 853 | |
| 854 | mPos += count; |
| 855 | b->size = count; |
| 856 | //ALOGV("buffer=%p, [0]=%d", b->i16, b->i16[0]); |
| 857 | } |
| 858 | } else if (event == AudioTrack::EVENT_UNDERRUN || event == AudioTrack::EVENT_BUFFER_END) { |
| 859 | ALOGV("process %p channel %d event %s", |
| 860 | this, mChannelID, (event == AudioTrack::EVENT_UNDERRUN) ? "UNDERRUN" : |
| 861 | "BUFFER_END"); |
| 862 | mSoundPool->addToStopList(this); |
| 863 | } else if (event == AudioTrack::EVENT_LOOP_END) { |
| 864 | ALOGV("End loop %p channel %d", this, mChannelID); |
| 865 | } else if (event == AudioTrack::EVENT_NEW_IAUDIOTRACK) { |
| 866 | ALOGV("process %p channel %d NEW_IAUDIOTRACK", this, mChannelID); |
| 867 | } else { |
| 868 | ALOGW("SoundChannel::process unexpected event %d", event); |
| 869 | } |
| 870 | } |
| 871 | |
| 872 | |
| 873 | // call with lock held |
| 874 | bool SoundChannel::doStop_l() |
| 875 | { |
| 876 | if (mState != IDLE) { |
| 877 | setVolume_l(0, 0); |
| 878 | ALOGV("stop"); |
| 879 | mAudioTrack->stop(); |
Marco Nelissen | 372be89 | 2014-12-04 08:59:22 -0800 | [diff] [blame] | 880 | mSample.clear(); |
| 881 | mState = IDLE; |
| 882 | mPriority = IDLE_PRIORITY; |
| 883 | return true; |
| 884 | } |
| 885 | return false; |
| 886 | } |
| 887 | |
| 888 | // call with lock held and sound pool lock held |
| 889 | void SoundChannel::stop_l() |
| 890 | { |
| 891 | if (doStop_l()) { |
| 892 | mSoundPool->done_l(this); |
| 893 | } |
| 894 | } |
| 895 | |
| 896 | // call with sound pool lock held |
| 897 | void SoundChannel::stop() |
| 898 | { |
| 899 | bool stopped; |
| 900 | { |
| 901 | Mutex::Autolock lock(&mLock); |
| 902 | stopped = doStop_l(); |
| 903 | } |
| 904 | |
| 905 | if (stopped) { |
| 906 | mSoundPool->done_l(this); |
| 907 | } |
| 908 | } |
| 909 | |
| 910 | //FIXME: Pause is a little broken right now |
| 911 | void SoundChannel::pause() |
| 912 | { |
| 913 | Mutex::Autolock lock(&mLock); |
| 914 | if (mState == PLAYING) { |
| 915 | ALOGV("pause track"); |
| 916 | mState = PAUSED; |
| 917 | mAudioTrack->pause(); |
| 918 | } |
| 919 | } |
| 920 | |
| 921 | void SoundChannel::autoPause() |
| 922 | { |
| 923 | Mutex::Autolock lock(&mLock); |
| 924 | if (mState == PLAYING) { |
| 925 | ALOGV("pause track"); |
| 926 | mState = PAUSED; |
| 927 | mAutoPaused = true; |
| 928 | mAudioTrack->pause(); |
| 929 | } |
| 930 | } |
| 931 | |
| 932 | void SoundChannel::resume() |
| 933 | { |
| 934 | Mutex::Autolock lock(&mLock); |
| 935 | if (mState == PAUSED) { |
| 936 | ALOGV("resume track"); |
| 937 | mState = PLAYING; |
| 938 | mAutoPaused = false; |
| 939 | mAudioTrack->start(); |
| 940 | } |
| 941 | } |
| 942 | |
| 943 | void SoundChannel::autoResume() |
| 944 | { |
| 945 | Mutex::Autolock lock(&mLock); |
| 946 | if (mAutoPaused && (mState == PAUSED)) { |
| 947 | ALOGV("resume track"); |
| 948 | mState = PLAYING; |
| 949 | mAutoPaused = false; |
| 950 | mAudioTrack->start(); |
| 951 | } |
| 952 | } |
| 953 | |
| 954 | void SoundChannel::setRate(float rate) |
| 955 | { |
| 956 | Mutex::Autolock lock(&mLock); |
| 957 | if (mAudioTrack != NULL && mSample != 0) { |
| 958 | uint32_t sampleRate = uint32_t(float(mSample->sampleRate()) * rate + 0.5); |
| 959 | mAudioTrack->setSampleRate(sampleRate); |
| 960 | mRate = rate; |
| 961 | } |
| 962 | } |
| 963 | |
| 964 | // call with lock held |
| 965 | void SoundChannel::setVolume_l(float leftVolume, float rightVolume) |
| 966 | { |
| 967 | mLeftVolume = leftVolume; |
| 968 | mRightVolume = rightVolume; |
| 969 | if (mAudioTrack != NULL) |
| 970 | mAudioTrack->setVolume(leftVolume, rightVolume); |
| 971 | } |
| 972 | |
| 973 | void SoundChannel::setVolume(float leftVolume, float rightVolume) |
| 974 | { |
| 975 | Mutex::Autolock lock(&mLock); |
| 976 | setVolume_l(leftVolume, rightVolume); |
| 977 | } |
| 978 | |
| 979 | void SoundChannel::setLoop(int loop) |
| 980 | { |
| 981 | Mutex::Autolock lock(&mLock); |
| 982 | if (mAudioTrack != NULL && mSample != 0) { |
| 983 | uint32_t loopEnd = mSample->size()/mNumChannels/ |
| 984 | ((mSample->format() == AUDIO_FORMAT_PCM_16_BIT) ? sizeof(int16_t) : sizeof(uint8_t)); |
| 985 | mAudioTrack->setLoop(0, loopEnd, loop); |
| 986 | mLoop = loop; |
| 987 | } |
| 988 | } |
| 989 | |
| 990 | SoundChannel::~SoundChannel() |
| 991 | { |
| 992 | ALOGV("SoundChannel destructor %p", this); |
| 993 | { |
| 994 | Mutex::Autolock lock(&mLock); |
| 995 | clearNextEvent(); |
| 996 | doStop_l(); |
| 997 | } |
| 998 | // do not call AudioTrack destructor with mLock held as it will wait for the AudioTrack |
| 999 | // callback thread to exit which may need to execute process() and acquire the mLock. |
| 1000 | mAudioTrack.clear(); |
| 1001 | } |
| 1002 | |
| 1003 | void SoundChannel::dump() |
| 1004 | { |
| 1005 | ALOGV("mState = %d mChannelID=%d, mNumChannels=%d, mPos = %d, mPriority=%d, mLoop=%d", |
| 1006 | mState, mChannelID, mNumChannels, mPos, mPriority, mLoop); |
| 1007 | } |
| 1008 | |
| 1009 | void SoundEvent::set(const sp<Sample>& sample, int channelID, float leftVolume, |
| 1010 | float rightVolume, int priority, int loop, float rate) |
| 1011 | { |
| 1012 | mSample = sample; |
| 1013 | mChannelID = channelID; |
| 1014 | mLeftVolume = leftVolume; |
| 1015 | mRightVolume = rightVolume; |
| 1016 | mPriority = priority; |
| 1017 | mLoop = loop; |
| 1018 | mRate =rate; |
| 1019 | } |
| 1020 | |
| 1021 | } // end namespace android |