AudioTrack.cpp revision 34fb29696b0f3abf61b10f8d053b1f33d501de0a
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18//#define LOG_NDEBUG 0 19#define LOG_TAG "AudioTrack" 20 21#include <inttypes.h> 22#include <math.h> 23#include <sys/resource.h> 24 25#include <audio_utils/primitives.h> 26#include <binder/IPCThreadState.h> 27#include <media/AudioTrack.h> 28#include <utils/Log.h> 29#include <private/media/AudioTrackShared.h> 30#include <media/IAudioFlinger.h> 31 32#define WAIT_PERIOD_MS 10 33#define WAIT_STREAM_END_TIMEOUT_SEC 120 34 35 36namespace android { 37// --------------------------------------------------------------------------- 38 39// static 40status_t AudioTrack::getMinFrameCount( 41 size_t* frameCount, 42 audio_stream_type_t streamType, 43 uint32_t sampleRate) 44{ 45 if (frameCount == NULL) { 46 return BAD_VALUE; 47 } 48 49 // FIXME merge with similar code in createTrack_l(), except we're missing 50 // some information here that is available in createTrack_l(): 51 // audio_io_handle_t output 52 // audio_format_t format 53 // audio_channel_mask_t channelMask 54 // audio_output_flags_t flags 55 uint32_t afSampleRate; 56 status_t status; 57 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 58 if (status != NO_ERROR) { 59 ALOGE("Unable to query output sample rate for stream type %d; status %d", 60 streamType, status); 61 return status; 62 } 63 size_t afFrameCount; 64 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 65 if (status != NO_ERROR) { 66 ALOGE("Unable to query output frame count for stream type %d; status %d", 67 streamType, status); 68 return status; 69 } 70 uint32_t afLatency; 71 status = AudioSystem::getOutputLatency(&afLatency, streamType); 72 if (status != NO_ERROR) { 73 ALOGE("Unable to query output latency for stream type %d; status %d", 74 streamType, status); 75 return status; 76 } 77 78 // Ensure that buffer depth covers at least audio hardware latency 79 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 80 if (minBufCount < 2) { 81 minBufCount = 2; 82 } 83 84 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 85 afFrameCount * minBufCount * sampleRate / afSampleRate; 86 // The formula above should always produce a non-zero value, but return an error 87 // in the unlikely event that it does not, as that's part of the API contract. 88 if (*frameCount == 0) { 89 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", 90 streamType, sampleRate); 91 return BAD_VALUE; 92 } 93 ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%d, afSampleRate=%d, afLatency=%d", 94 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 95 return NO_ERROR; 96} 97 98// --------------------------------------------------------------------------- 99 100AudioTrack::AudioTrack() 101 : mStatus(NO_INIT), 102 mIsTimed(false), 103 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 104 mPreviousSchedulingGroup(SP_DEFAULT), 105 mPausedPosition(0) 106{ 107 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN; 108 mAttributes.usage = AUDIO_USAGE_UNKNOWN; 109 mAttributes.flags = 0x0; 110 strcpy(mAttributes.tags, ""); 111} 112 113AudioTrack::AudioTrack( 114 audio_stream_type_t streamType, 115 uint32_t sampleRate, 116 audio_format_t format, 117 audio_channel_mask_t channelMask, 118 size_t frameCount, 119 audio_output_flags_t flags, 120 callback_t cbf, 121 void* user, 122 uint32_t notificationFrames, 123 int sessionId, 124 transfer_type transferType, 125 const audio_offload_info_t *offloadInfo, 126 int uid, 127 pid_t pid) 128 : mStatus(NO_INIT), 129 mIsTimed(false), 130 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 131 mPreviousSchedulingGroup(SP_DEFAULT), 132 mPausedPosition(0) 133{ 134 mStatus = set(streamType, sampleRate, format, channelMask, 135 frameCount, flags, cbf, user, notificationFrames, 136 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 137 offloadInfo, uid, pid, NULL /*no audio attributes*/); 138} 139 140AudioTrack::AudioTrack( 141 audio_stream_type_t streamType, 142 uint32_t sampleRate, 143 audio_format_t format, 144 audio_channel_mask_t channelMask, 145 const sp<IMemory>& sharedBuffer, 146 audio_output_flags_t flags, 147 callback_t cbf, 148 void* user, 149 uint32_t notificationFrames, 150 int sessionId, 151 transfer_type transferType, 152 const audio_offload_info_t *offloadInfo, 153 int uid, 154 pid_t pid) 155 : mStatus(NO_INIT), 156 mIsTimed(false), 157 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 158 mPreviousSchedulingGroup(SP_DEFAULT), 159 mPausedPosition(0) 160{ 161 mStatus = set(streamType, sampleRate, format, channelMask, 162 0 /*frameCount*/, flags, cbf, user, notificationFrames, 163 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, 164 uid, pid, NULL /*no audio attributes*/); 165} 166 167AudioTrack::~AudioTrack() 168{ 169 if (mStatus == NO_ERROR) { 170 // Make sure that callback function exits in the case where 171 // it is looping on buffer full condition in obtainBuffer(). 172 // Otherwise the callback thread will never exit. 173 stop(); 174 if (mAudioTrackThread != 0) { 175 mProxy->interrupt(); 176 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 177 mAudioTrackThread->requestExitAndWait(); 178 mAudioTrackThread.clear(); 179 } 180 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 181 mAudioTrack.clear(); 182 mCblkMemory.clear(); 183 mSharedBuffer.clear(); 184 IPCThreadState::self()->flushCommands(); 185 ALOGV("~AudioTrack, releasing session id from %d on behalf of %d", 186 IPCThreadState::self()->getCallingPid(), mClientPid); 187 AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); 188 } 189} 190 191status_t AudioTrack::set( 192 audio_stream_type_t streamType, 193 uint32_t sampleRate, 194 audio_format_t format, 195 audio_channel_mask_t channelMask, 196 size_t frameCount, 197 audio_output_flags_t flags, 198 callback_t cbf, 199 void* user, 200 uint32_t notificationFrames, 201 const sp<IMemory>& sharedBuffer, 202 bool threadCanCallJava, 203 int sessionId, 204 transfer_type transferType, 205 const audio_offload_info_t *offloadInfo, 206 int uid, 207 pid_t pid, 208 audio_attributes_t* pAttributes) 209{ 210 ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 211 "flags #%x, notificationFrames %u, sessionId %d, transferType %d", 212 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, 213 sessionId, transferType); 214 215 switch (transferType) { 216 case TRANSFER_DEFAULT: 217 if (sharedBuffer != 0) { 218 transferType = TRANSFER_SHARED; 219 } else if (cbf == NULL || threadCanCallJava) { 220 transferType = TRANSFER_SYNC; 221 } else { 222 transferType = TRANSFER_CALLBACK; 223 } 224 break; 225 case TRANSFER_CALLBACK: 226 if (cbf == NULL || sharedBuffer != 0) { 227 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 228 return BAD_VALUE; 229 } 230 break; 231 case TRANSFER_OBTAIN: 232 case TRANSFER_SYNC: 233 if (sharedBuffer != 0) { 234 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 235 return BAD_VALUE; 236 } 237 break; 238 case TRANSFER_SHARED: 239 if (sharedBuffer == 0) { 240 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 241 return BAD_VALUE; 242 } 243 break; 244 default: 245 ALOGE("Invalid transfer type %d", transferType); 246 return BAD_VALUE; 247 } 248 mSharedBuffer = sharedBuffer; 249 mTransfer = transferType; 250 251 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 252 sharedBuffer->size()); 253 254 ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags); 255 256 AutoMutex lock(mLock); 257 258 // invariant that mAudioTrack != 0 is true only after set() returns successfully 259 if (mAudioTrack != 0) { 260 ALOGE("Track already in use"); 261 return INVALID_OPERATION; 262 } 263 264 // handle default values first. 265 if (streamType == AUDIO_STREAM_DEFAULT) { 266 streamType = AUDIO_STREAM_MUSIC; 267 } 268 269 if (pAttributes == NULL) { 270 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 271 ALOGE("Invalid stream type %d", streamType); 272 return BAD_VALUE; 273 } 274 setAttributesFromStreamType(streamType); 275 mStreamType = streamType; 276 } else { 277 if (!isValidAttributes(pAttributes)) { 278 ALOGE("Invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]", 279 pAttributes->usage, pAttributes->content_type, pAttributes->flags, 280 pAttributes->tags); 281 } 282 // stream type shouldn't be looked at, this track has audio attributes 283 memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t)); 284 setStreamTypeFromAttributes(mAttributes); 285 ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]", 286 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags); 287 } 288 289 status_t status; 290 if (sampleRate == 0) { 291 status = AudioSystem::getOutputSamplingRateForAttr(&sampleRate, &mAttributes); 292 if (status != NO_ERROR) { 293 ALOGE("Could not get output sample rate for stream type %d; status %d", 294 mStreamType, status); 295 return status; 296 } 297 } 298 mSampleRate = sampleRate; 299 300 // these below should probably come from the audioFlinger too... 301 if (format == AUDIO_FORMAT_DEFAULT) { 302 format = AUDIO_FORMAT_PCM_16_BIT; 303 } 304 305 // validate parameters 306 if (!audio_is_valid_format(format)) { 307 ALOGE("Invalid format %#x", format); 308 return BAD_VALUE; 309 } 310 mFormat = format; 311 312 if (!audio_is_output_channel(channelMask)) { 313 ALOGE("Invalid channel mask %#x", channelMask); 314 return BAD_VALUE; 315 } 316 mChannelMask = channelMask; 317 uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 318 mChannelCount = channelCount; 319 320 // AudioFlinger does not currently support 8-bit data in shared memory 321 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 322 ALOGE("8-bit data in shared memory is not supported"); 323 return BAD_VALUE; 324 } 325 326 // force direct flag if format is not linear PCM 327 // or offload was requested 328 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 329 || !audio_is_linear_pcm(format)) { 330 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 331 ? "Offload request, forcing to Direct Output" 332 : "Not linear PCM, forcing to Direct Output"); 333 flags = (audio_output_flags_t) 334 // FIXME why can't we allow direct AND fast? 335 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 336 } 337 // only allow deep buffering for music stream type 338 if (mStreamType != AUDIO_STREAM_MUSIC) { 339 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 340 } 341 342 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { 343 if (audio_is_linear_pcm(format)) { 344 mFrameSize = channelCount * audio_bytes_per_sample(format); 345 } else { 346 mFrameSize = sizeof(uint8_t); 347 } 348 mFrameSizeAF = mFrameSize; 349 } else { 350 ALOG_ASSERT(audio_is_linear_pcm(format)); 351 mFrameSize = channelCount * audio_bytes_per_sample(format); 352 mFrameSizeAF = channelCount * audio_bytes_per_sample( 353 format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format); 354 // createTrack will return an error if PCM format is not supported by server, 355 // so no need to check for specific PCM formats here 356 } 357 358 // Make copy of input parameter offloadInfo so that in the future: 359 // (a) createTrack_l doesn't need it as an input parameter 360 // (b) we can support re-creation of offloaded tracks 361 if (offloadInfo != NULL) { 362 mOffloadInfoCopy = *offloadInfo; 363 mOffloadInfo = &mOffloadInfoCopy; 364 } else { 365 mOffloadInfo = NULL; 366 } 367 368 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f; 369 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f; 370 mSendLevel = 0.0f; 371 // mFrameCount is initialized in createTrack_l 372 mReqFrameCount = frameCount; 373 mNotificationFramesReq = notificationFrames; 374 mNotificationFramesAct = 0; 375 mSessionId = sessionId; 376 int callingpid = IPCThreadState::self()->getCallingPid(); 377 int mypid = getpid(); 378 if (uid == -1 || (callingpid != mypid)) { 379 mClientUid = IPCThreadState::self()->getCallingUid(); 380 } else { 381 mClientUid = uid; 382 } 383 if (pid == -1 || (callingpid != mypid)) { 384 mClientPid = callingpid; 385 } else { 386 mClientPid = pid; 387 } 388 mAuxEffectId = 0; 389 mFlags = flags; 390 mCbf = cbf; 391 392 if (cbf != NULL) { 393 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 394 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 395 } 396 397 // create the IAudioTrack 398 status = createTrack_l(0 /*epoch*/); 399 400 if (status != NO_ERROR) { 401 if (mAudioTrackThread != 0) { 402 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 403 mAudioTrackThread->requestExitAndWait(); 404 mAudioTrackThread.clear(); 405 } 406 return status; 407 } 408 409 mStatus = NO_ERROR; 410 mState = STATE_STOPPED; 411 mUserData = user; 412 mLoopPeriod = 0; 413 mMarkerPosition = 0; 414 mMarkerReached = false; 415 mNewPosition = 0; 416 mUpdatePeriod = 0; 417 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 418 mSequence = 1; 419 mObservedSequence = mSequence; 420 mInUnderrun = false; 421 422 return NO_ERROR; 423} 424 425// ------------------------------------------------------------------------- 426 427status_t AudioTrack::start() 428{ 429 AutoMutex lock(mLock); 430 431 if (mState == STATE_ACTIVE) { 432 return INVALID_OPERATION; 433 } 434 435 mInUnderrun = true; 436 437 State previousState = mState; 438 if (previousState == STATE_PAUSED_STOPPING) { 439 mState = STATE_STOPPING; 440 } else { 441 mState = STATE_ACTIVE; 442 } 443 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 444 // reset current position as seen by client to 0 445 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 446 // force refresh of remaining frames by processAudioBuffer() as last 447 // write before stop could be partial. 448 mRefreshRemaining = true; 449 } 450 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 451 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 452 453 sp<AudioTrackThread> t = mAudioTrackThread; 454 if (t != 0) { 455 if (previousState == STATE_STOPPING) { 456 mProxy->interrupt(); 457 } else { 458 t->resume(); 459 } 460 } else { 461 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 462 get_sched_policy(0, &mPreviousSchedulingGroup); 463 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 464 } 465 466 status_t status = NO_ERROR; 467 if (!(flags & CBLK_INVALID)) { 468 status = mAudioTrack->start(); 469 if (status == DEAD_OBJECT) { 470 flags |= CBLK_INVALID; 471 } 472 } 473 if (flags & CBLK_INVALID) { 474 status = restoreTrack_l("start"); 475 } 476 477 if (status != NO_ERROR) { 478 ALOGE("start() status %d", status); 479 mState = previousState; 480 if (t != 0) { 481 if (previousState != STATE_STOPPING) { 482 t->pause(); 483 } 484 } else { 485 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 486 set_sched_policy(0, mPreviousSchedulingGroup); 487 } 488 } 489 490 return status; 491} 492 493void AudioTrack::stop() 494{ 495 AutoMutex lock(mLock); 496 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 497 return; 498 } 499 500 if (isOffloaded_l()) { 501 mState = STATE_STOPPING; 502 } else { 503 mState = STATE_STOPPED; 504 } 505 506 mProxy->interrupt(); 507 mAudioTrack->stop(); 508 // the playback head position will reset to 0, so if a marker is set, we need 509 // to activate it again 510 mMarkerReached = false; 511#if 0 512 // Force flush if a shared buffer is used otherwise audioflinger 513 // will not stop before end of buffer is reached. 514 // It may be needed to make sure that we stop playback, likely in case looping is on. 515 if (mSharedBuffer != 0) { 516 flush_l(); 517 } 518#endif 519 520 sp<AudioTrackThread> t = mAudioTrackThread; 521 if (t != 0) { 522 if (!isOffloaded_l()) { 523 t->pause(); 524 } 525 } else { 526 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 527 set_sched_policy(0, mPreviousSchedulingGroup); 528 } 529} 530 531bool AudioTrack::stopped() const 532{ 533 AutoMutex lock(mLock); 534 return mState != STATE_ACTIVE; 535} 536 537void AudioTrack::flush() 538{ 539 if (mSharedBuffer != 0) { 540 return; 541 } 542 AutoMutex lock(mLock); 543 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 544 return; 545 } 546 flush_l(); 547} 548 549void AudioTrack::flush_l() 550{ 551 ALOG_ASSERT(mState != STATE_ACTIVE); 552 553 // clear playback marker and periodic update counter 554 mMarkerPosition = 0; 555 mMarkerReached = false; 556 mUpdatePeriod = 0; 557 mRefreshRemaining = true; 558 559 mState = STATE_FLUSHED; 560 if (isOffloaded_l()) { 561 mProxy->interrupt(); 562 } 563 mProxy->flush(); 564 mAudioTrack->flush(); 565} 566 567void AudioTrack::pause() 568{ 569 AutoMutex lock(mLock); 570 if (mState == STATE_ACTIVE) { 571 mState = STATE_PAUSED; 572 } else if (mState == STATE_STOPPING) { 573 mState = STATE_PAUSED_STOPPING; 574 } else { 575 return; 576 } 577 mProxy->interrupt(); 578 mAudioTrack->pause(); 579 580 if (isOffloaded_l()) { 581 if (mOutput != AUDIO_IO_HANDLE_NONE) { 582 uint32_t halFrames; 583 // OffloadThread sends HAL pause in its threadLoop.. time saved 584 // here can be slightly off 585 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); 586 ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); 587 } 588 } 589} 590 591status_t AudioTrack::setVolume(float left, float right) 592{ 593 // This duplicates a test by AudioTrack JNI, but that is not the only caller 594 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY || 595 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) { 596 return BAD_VALUE; 597 } 598 599 AutoMutex lock(mLock); 600 mVolume[AUDIO_INTERLEAVE_LEFT] = left; 601 mVolume[AUDIO_INTERLEAVE_RIGHT] = right; 602 603 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right))); 604 605 if (isOffloaded_l()) { 606 mAudioTrack->signal(); 607 } 608 return NO_ERROR; 609} 610 611status_t AudioTrack::setVolume(float volume) 612{ 613 return setVolume(volume, volume); 614} 615 616status_t AudioTrack::setAuxEffectSendLevel(float level) 617{ 618 // This duplicates a test by AudioTrack JNI, but that is not the only caller 619 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) { 620 return BAD_VALUE; 621 } 622 623 AutoMutex lock(mLock); 624 mSendLevel = level; 625 mProxy->setSendLevel(level); 626 627 return NO_ERROR; 628} 629 630void AudioTrack::getAuxEffectSendLevel(float* level) const 631{ 632 if (level != NULL) { 633 *level = mSendLevel; 634 } 635} 636 637status_t AudioTrack::setSampleRate(uint32_t rate) 638{ 639 if (mIsTimed || isOffloadedOrDirect()) { 640 return INVALID_OPERATION; 641 } 642 643 uint32_t afSamplingRate; 644 if (AudioSystem::getOutputSamplingRateForAttr(&afSamplingRate, &mAttributes) != NO_ERROR) { 645 return NO_INIT; 646 } 647 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 648 if (rate == 0 || rate > afSamplingRate*2 ) { 649 return BAD_VALUE; 650 } 651 652 AutoMutex lock(mLock); 653 mSampleRate = rate; 654 mProxy->setSampleRate(rate); 655 656 return NO_ERROR; 657} 658 659uint32_t AudioTrack::getSampleRate() const 660{ 661 if (mIsTimed) { 662 return 0; 663 } 664 665 AutoMutex lock(mLock); 666 667 // sample rate can be updated during playback by the offloaded decoder so we need to 668 // query the HAL and update if needed. 669// FIXME use Proxy return channel to update the rate from server and avoid polling here 670 if (isOffloadedOrDirect_l()) { 671 if (mOutput != AUDIO_IO_HANDLE_NONE) { 672 uint32_t sampleRate = 0; 673 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate); 674 if (status == NO_ERROR) { 675 mSampleRate = sampleRate; 676 } 677 } 678 } 679 return mSampleRate; 680} 681 682status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 683{ 684 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 685 return INVALID_OPERATION; 686 } 687 688 if (loopCount == 0) { 689 ; 690 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 691 loopEnd - loopStart >= MIN_LOOP) { 692 ; 693 } else { 694 return BAD_VALUE; 695 } 696 697 AutoMutex lock(mLock); 698 // See setPosition() regarding setting parameters such as loop points or position while active 699 if (mState == STATE_ACTIVE) { 700 return INVALID_OPERATION; 701 } 702 setLoop_l(loopStart, loopEnd, loopCount); 703 return NO_ERROR; 704} 705 706void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 707{ 708 // FIXME If setting a loop also sets position to start of loop, then 709 // this is correct. Otherwise it should be removed. 710 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 711 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 712 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 713} 714 715status_t AudioTrack::setMarkerPosition(uint32_t marker) 716{ 717 // The only purpose of setting marker position is to get a callback 718 if (mCbf == NULL || isOffloadedOrDirect()) { 719 return INVALID_OPERATION; 720 } 721 722 AutoMutex lock(mLock); 723 mMarkerPosition = marker; 724 mMarkerReached = false; 725 726 return NO_ERROR; 727} 728 729status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 730{ 731 if (isOffloadedOrDirect()) { 732 return INVALID_OPERATION; 733 } 734 if (marker == NULL) { 735 return BAD_VALUE; 736 } 737 738 AutoMutex lock(mLock); 739 *marker = mMarkerPosition; 740 741 return NO_ERROR; 742} 743 744status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 745{ 746 // The only purpose of setting position update period is to get a callback 747 if (mCbf == NULL || isOffloadedOrDirect()) { 748 return INVALID_OPERATION; 749 } 750 751 AutoMutex lock(mLock); 752 mNewPosition = mProxy->getPosition() + updatePeriod; 753 mUpdatePeriod = updatePeriod; 754 755 return NO_ERROR; 756} 757 758status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 759{ 760 if (isOffloadedOrDirect()) { 761 return INVALID_OPERATION; 762 } 763 if (updatePeriod == NULL) { 764 return BAD_VALUE; 765 } 766 767 AutoMutex lock(mLock); 768 *updatePeriod = mUpdatePeriod; 769 770 return NO_ERROR; 771} 772 773status_t AudioTrack::setPosition(uint32_t position) 774{ 775 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 776 return INVALID_OPERATION; 777 } 778 if (position > mFrameCount) { 779 return BAD_VALUE; 780 } 781 782 AutoMutex lock(mLock); 783 // Currently we require that the player is inactive before setting parameters such as position 784 // or loop points. Otherwise, there could be a race condition: the application could read the 785 // current position, compute a new position or loop parameters, and then set that position or 786 // loop parameters but it would do the "wrong" thing since the position has continued to advance 787 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 788 // to specify how it wants to handle such scenarios. 789 if (mState == STATE_ACTIVE) { 790 return INVALID_OPERATION; 791 } 792 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 793 mLoopPeriod = 0; 794 // FIXME Check whether loops and setting position are incompatible in old code. 795 // If we use setLoop for both purposes we lose the capability to set the position while looping. 796 mStaticProxy->setLoop(position, mFrameCount, 0); 797 798 return NO_ERROR; 799} 800 801status_t AudioTrack::getPosition(uint32_t *position) const 802{ 803 if (position == NULL) { 804 return BAD_VALUE; 805 } 806 807 AutoMutex lock(mLock); 808 if (isOffloadedOrDirect_l()) { 809 uint32_t dspFrames = 0; 810 811 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) { 812 ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); 813 *position = mPausedPosition; 814 return NO_ERROR; 815 } 816 817 if (mOutput != AUDIO_IO_HANDLE_NONE) { 818 uint32_t halFrames; 819 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 820 } 821 *position = dspFrames; 822 } else { 823 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 824 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 825 mProxy->getPosition(); 826 } 827 return NO_ERROR; 828} 829 830status_t AudioTrack::getBufferPosition(uint32_t *position) 831{ 832 if (mSharedBuffer == 0 || mIsTimed) { 833 return INVALID_OPERATION; 834 } 835 if (position == NULL) { 836 return BAD_VALUE; 837 } 838 839 AutoMutex lock(mLock); 840 *position = mStaticProxy->getBufferPosition(); 841 return NO_ERROR; 842} 843 844status_t AudioTrack::reload() 845{ 846 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 847 return INVALID_OPERATION; 848 } 849 850 AutoMutex lock(mLock); 851 // See setPosition() regarding setting parameters such as loop points or position while active 852 if (mState == STATE_ACTIVE) { 853 return INVALID_OPERATION; 854 } 855 mNewPosition = mUpdatePeriod; 856 mLoopPeriod = 0; 857 // FIXME The new code cannot reload while keeping a loop specified. 858 // Need to check how the old code handled this, and whether it's a significant change. 859 mStaticProxy->setLoop(0, mFrameCount, 0); 860 return NO_ERROR; 861} 862 863audio_io_handle_t AudioTrack::getOutput() const 864{ 865 AutoMutex lock(mLock); 866 return mOutput; 867} 868 869status_t AudioTrack::attachAuxEffect(int effectId) 870{ 871 AutoMutex lock(mLock); 872 status_t status = mAudioTrack->attachAuxEffect(effectId); 873 if (status == NO_ERROR) { 874 mAuxEffectId = effectId; 875 } 876 return status; 877} 878 879// ------------------------------------------------------------------------- 880 881// must be called with mLock held 882status_t AudioTrack::createTrack_l(size_t epoch) 883{ 884 status_t status; 885 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 886 if (audioFlinger == 0) { 887 ALOGE("Could not get audioflinger"); 888 return NO_INIT; 889 } 890 891 audio_io_handle_t output = AudioSystem::getOutputForAttr(&mAttributes, mSampleRate, mFormat, 892 mChannelMask, mFlags, mOffloadInfo); 893 if (output == AUDIO_IO_HANDLE_NONE) { 894 ALOGE("Could not get audio output for stream type %d, usage %d, sample rate %u, format %#x," 895 " channel mask %#x, flags %#x", 896 mStreamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags); 897 return BAD_VALUE; 898 } 899 { 900 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 901 // we must release it ourselves if anything goes wrong. 902 903 // Not all of these values are needed under all conditions, but it is easier to get them all 904 905 uint32_t afLatency; 906 status = AudioSystem::getLatency(output, &afLatency); 907 if (status != NO_ERROR) { 908 ALOGE("getLatency(%d) failed status %d", output, status); 909 goto release; 910 } 911 912 size_t afFrameCount; 913 status = AudioSystem::getFrameCount(output, &afFrameCount); 914 if (status != NO_ERROR) { 915 ALOGE("getFrameCount(output=%d) status %d", output, status); 916 goto release; 917 } 918 919 uint32_t afSampleRate; 920 status = AudioSystem::getSamplingRate(output, &afSampleRate); 921 if (status != NO_ERROR) { 922 ALOGE("getSamplingRate(output=%d) status %d", output, status); 923 goto release; 924 } 925 926 // Client decides whether the track is TIMED (see below), but can only express a preference 927 // for FAST. Server will perform additional tests. 928 if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( 929 // either of these use cases: 930 // use case 1: shared buffer 931 (mSharedBuffer != 0) || 932 // use case 2: callback transfer mode 933 (mTransfer == TRANSFER_CALLBACK)) && 934 // matching sample rate 935 (mSampleRate == afSampleRate))) { 936 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 937 // once denied, do not request again if IAudioTrack is re-created 938 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 939 } 940 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 941 942 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 943 // n = 1 fast track with single buffering; nBuffering is ignored 944 // n = 2 fast track with double buffering 945 // n = 2 normal track, no sample rate conversion 946 // n = 3 normal track, with sample rate conversion 947 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 948 // n > 3 very high latency or very small notification interval; nBuffering is ignored 949 const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; 950 951 mNotificationFramesAct = mNotificationFramesReq; 952 953 size_t frameCount = mReqFrameCount; 954 if (!audio_is_linear_pcm(mFormat)) { 955 956 if (mSharedBuffer != 0) { 957 // Same comment as below about ignoring frameCount parameter for set() 958 frameCount = mSharedBuffer->size(); 959 } else if (frameCount == 0) { 960 frameCount = afFrameCount; 961 } 962 if (mNotificationFramesAct != frameCount) { 963 mNotificationFramesAct = frameCount; 964 } 965 } else if (mSharedBuffer != 0) { 966 967 // Ensure that buffer alignment matches channel count 968 // 8-bit data in shared memory is not currently supported by AudioFlinger 969 size_t alignment = audio_bytes_per_sample( 970 mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat); 971 if (alignment & 1) { 972 alignment = 1; 973 } 974 if (mChannelCount > 1) { 975 // More than 2 channels does not require stronger alignment than stereo 976 alignment <<= 1; 977 } 978 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 979 ALOGE("Invalid buffer alignment: address %p, channel count %u", 980 mSharedBuffer->pointer(), mChannelCount); 981 status = BAD_VALUE; 982 goto release; 983 } 984 985 // When initializing a shared buffer AudioTrack via constructors, 986 // there's no frameCount parameter. 987 // But when initializing a shared buffer AudioTrack via set(), 988 // there _is_ a frameCount parameter. We silently ignore it. 989 frameCount = mSharedBuffer->size() / mFrameSizeAF; 990 991 } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 992 993 // FIXME move these calculations and associated checks to server 994 995 // Ensure that buffer depth covers at least audio hardware latency 996 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 997 ALOGV("afFrameCount=%zu, minBufCount=%d, afSampleRate=%u, afLatency=%d", 998 afFrameCount, minBufCount, afSampleRate, afLatency); 999 if (minBufCount <= nBuffering) { 1000 minBufCount = nBuffering; 1001 } 1002 1003 size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate; 1004 ALOGV("minFrameCount: %zu, afFrameCount=%zu, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 1005 ", afLatency=%d", 1006 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); 1007 1008 if (frameCount == 0) { 1009 frameCount = minFrameCount; 1010 } else if (frameCount < minFrameCount) { 1011 // not ALOGW because it happens all the time when playing key clicks over A2DP 1012 ALOGV("Minimum buffer size corrected from %zu to %zu", 1013 frameCount, minFrameCount); 1014 frameCount = minFrameCount; 1015 } 1016 // Make sure that application is notified with sufficient margin before underrun 1017 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1018 mNotificationFramesAct = frameCount/nBuffering; 1019 } 1020 1021 } else { 1022 // For fast tracks, the frame count calculations and checks are done by server 1023 } 1024 1025 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 1026 if (mIsTimed) { 1027 trackFlags |= IAudioFlinger::TRACK_TIMED; 1028 } 1029 1030 pid_t tid = -1; 1031 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1032 trackFlags |= IAudioFlinger::TRACK_FAST; 1033 if (mAudioTrackThread != 0) { 1034 tid = mAudioTrackThread->getTid(); 1035 } 1036 } 1037 1038 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1039 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 1040 } 1041 1042 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1043 trackFlags |= IAudioFlinger::TRACK_DIRECT; 1044 } 1045 1046 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1047 // but we will still need the original value also 1048 sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType, 1049 mSampleRate, 1050 // AudioFlinger only sees 16-bit PCM 1051 mFormat == AUDIO_FORMAT_PCM_8_BIT && 1052 !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT) ? 1053 AUDIO_FORMAT_PCM_16_BIT : mFormat, 1054 mChannelMask, 1055 &temp, 1056 &trackFlags, 1057 mSharedBuffer, 1058 output, 1059 tid, 1060 &mSessionId, 1061 mClientUid, 1062 &status); 1063 1064 if (status != NO_ERROR) { 1065 ALOGE("AudioFlinger could not create track, status: %d", status); 1066 goto release; 1067 } 1068 ALOG_ASSERT(track != 0); 1069 1070 // AudioFlinger now owns the reference to the I/O handle, 1071 // so we are no longer responsible for releasing it. 1072 1073 sp<IMemory> iMem = track->getCblk(); 1074 if (iMem == 0) { 1075 ALOGE("Could not get control block"); 1076 return NO_INIT; 1077 } 1078 void *iMemPointer = iMem->pointer(); 1079 if (iMemPointer == NULL) { 1080 ALOGE("Could not get control block pointer"); 1081 return NO_INIT; 1082 } 1083 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1084 if (mAudioTrack != 0) { 1085 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1086 mDeathNotifier.clear(); 1087 } 1088 mAudioTrack = track; 1089 mCblkMemory = iMem; 1090 IPCThreadState::self()->flushCommands(); 1091 1092 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1093 mCblk = cblk; 1094 // note that temp is the (possibly revised) value of frameCount 1095 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1096 // In current design, AudioTrack client checks and ensures frame count validity before 1097 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1098 // for fast track as it uses a special method of assigning frame count. 1099 ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp); 1100 } 1101 frameCount = temp; 1102 1103 mAwaitBoost = false; 1104 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1105 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1106 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount); 1107 mAwaitBoost = true; 1108 if (mSharedBuffer == 0) { 1109 // Theoretically double-buffering is not required for fast tracks, 1110 // due to tighter scheduling. But in practice, to accommodate kernels with 1111 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1112 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1113 mNotificationFramesAct = frameCount/nBuffering; 1114 } 1115 } 1116 } else { 1117 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount); 1118 // once denied, do not request again if IAudioTrack is re-created 1119 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1120 if (mSharedBuffer == 0) { 1121 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1122 mNotificationFramesAct = frameCount/nBuffering; 1123 } 1124 } 1125 } 1126 } 1127 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1128 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1129 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1130 } else { 1131 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1132 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1133 // FIXME This is a warning, not an error, so don't return error status 1134 //return NO_INIT; 1135 } 1136 } 1137 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1138 if (trackFlags & IAudioFlinger::TRACK_DIRECT) { 1139 ALOGV("AUDIO_OUTPUT_FLAG_DIRECT successful"); 1140 } else { 1141 ALOGW("AUDIO_OUTPUT_FLAG_DIRECT denied by server"); 1142 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT); 1143 // FIXME This is a warning, not an error, so don't return error status 1144 //return NO_INIT; 1145 } 1146 } 1147 1148 // We retain a copy of the I/O handle, but don't own the reference 1149 mOutput = output; 1150 mRefreshRemaining = true; 1151 1152 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1153 // is the value of pointer() for the shared buffer, otherwise buffers points 1154 // immediately after the control block. This address is for the mapping within client 1155 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1156 void* buffers; 1157 if (mSharedBuffer == 0) { 1158 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1159 } else { 1160 buffers = mSharedBuffer->pointer(); 1161 } 1162 1163 mAudioTrack->attachAuxEffect(mAuxEffectId); 1164 // FIXME don't believe this lie 1165 mLatency = afLatency + (1000*frameCount) / mSampleRate; 1166 1167 mFrameCount = frameCount; 1168 // If IAudioTrack is re-created, don't let the requested frameCount 1169 // decrease. This can confuse clients that cache frameCount(). 1170 if (frameCount > mReqFrameCount) { 1171 mReqFrameCount = frameCount; 1172 } 1173 1174 // update proxy 1175 if (mSharedBuffer == 0) { 1176 mStaticProxy.clear(); 1177 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1178 } else { 1179 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1180 mProxy = mStaticProxy; 1181 } 1182 mProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); 1183 mProxy->setSendLevel(mSendLevel); 1184 mProxy->setSampleRate(mSampleRate); 1185 mProxy->setEpoch(epoch); 1186 mProxy->setMinimum(mNotificationFramesAct); 1187 1188 mDeathNotifier = new DeathNotifier(this); 1189 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1190 1191 return NO_ERROR; 1192 } 1193 1194release: 1195 AudioSystem::releaseOutput(output); 1196 if (status == NO_ERROR) { 1197 status = NO_INIT; 1198 } 1199 return status; 1200} 1201 1202status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1203{ 1204 if (audioBuffer == NULL) { 1205 return BAD_VALUE; 1206 } 1207 if (mTransfer != TRANSFER_OBTAIN) { 1208 audioBuffer->frameCount = 0; 1209 audioBuffer->size = 0; 1210 audioBuffer->raw = NULL; 1211 return INVALID_OPERATION; 1212 } 1213 1214 const struct timespec *requested; 1215 struct timespec timeout; 1216 if (waitCount == -1) { 1217 requested = &ClientProxy::kForever; 1218 } else if (waitCount == 0) { 1219 requested = &ClientProxy::kNonBlocking; 1220 } else if (waitCount > 0) { 1221 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1222 timeout.tv_sec = ms / 1000; 1223 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1224 requested = &timeout; 1225 } else { 1226 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1227 requested = NULL; 1228 } 1229 return obtainBuffer(audioBuffer, requested); 1230} 1231 1232status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1233 struct timespec *elapsed, size_t *nonContig) 1234{ 1235 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1236 uint32_t oldSequence = 0; 1237 uint32_t newSequence; 1238 1239 Proxy::Buffer buffer; 1240 status_t status = NO_ERROR; 1241 1242 static const int32_t kMaxTries = 5; 1243 int32_t tryCounter = kMaxTries; 1244 1245 do { 1246 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1247 // keep them from going away if another thread re-creates the track during obtainBuffer() 1248 sp<AudioTrackClientProxy> proxy; 1249 sp<IMemory> iMem; 1250 1251 { // start of lock scope 1252 AutoMutex lock(mLock); 1253 1254 newSequence = mSequence; 1255 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1256 if (status == DEAD_OBJECT) { 1257 // re-create track, unless someone else has already done so 1258 if (newSequence == oldSequence) { 1259 status = restoreTrack_l("obtainBuffer"); 1260 if (status != NO_ERROR) { 1261 buffer.mFrameCount = 0; 1262 buffer.mRaw = NULL; 1263 buffer.mNonContig = 0; 1264 break; 1265 } 1266 } 1267 } 1268 oldSequence = newSequence; 1269 1270 // Keep the extra references 1271 proxy = mProxy; 1272 iMem = mCblkMemory; 1273 1274 if (mState == STATE_STOPPING) { 1275 status = -EINTR; 1276 buffer.mFrameCount = 0; 1277 buffer.mRaw = NULL; 1278 buffer.mNonContig = 0; 1279 break; 1280 } 1281 1282 // Non-blocking if track is stopped or paused 1283 if (mState != STATE_ACTIVE) { 1284 requested = &ClientProxy::kNonBlocking; 1285 } 1286 1287 } // end of lock scope 1288 1289 buffer.mFrameCount = audioBuffer->frameCount; 1290 // FIXME starts the requested timeout and elapsed over from scratch 1291 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1292 1293 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1294 1295 audioBuffer->frameCount = buffer.mFrameCount; 1296 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1297 audioBuffer->raw = buffer.mRaw; 1298 if (nonContig != NULL) { 1299 *nonContig = buffer.mNonContig; 1300 } 1301 return status; 1302} 1303 1304void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1305{ 1306 if (mTransfer == TRANSFER_SHARED) { 1307 return; 1308 } 1309 1310 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1311 if (stepCount == 0) { 1312 return; 1313 } 1314 1315 Proxy::Buffer buffer; 1316 buffer.mFrameCount = stepCount; 1317 buffer.mRaw = audioBuffer->raw; 1318 1319 AutoMutex lock(mLock); 1320 mInUnderrun = false; 1321 mProxy->releaseBuffer(&buffer); 1322 1323 // restart track if it was disabled by audioflinger due to previous underrun 1324 if (mState == STATE_ACTIVE) { 1325 audio_track_cblk_t* cblk = mCblk; 1326 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1327 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1328 // FIXME ignoring status 1329 mAudioTrack->start(); 1330 } 1331 } 1332} 1333 1334// ------------------------------------------------------------------------- 1335 1336ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1337{ 1338 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1339 return INVALID_OPERATION; 1340 } 1341 1342 if (isDirect()) { 1343 AutoMutex lock(mLock); 1344 int32_t flags = android_atomic_and( 1345 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), 1346 &mCblk->mFlags); 1347 if (flags & CBLK_INVALID) { 1348 return DEAD_OBJECT; 1349 } 1350 } 1351 1352 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1353 // Sanity-check: user is most-likely passing an error code, and it would 1354 // make the return value ambiguous (actualSize vs error). 1355 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1356 return BAD_VALUE; 1357 } 1358 1359 size_t written = 0; 1360 Buffer audioBuffer; 1361 1362 while (userSize >= mFrameSize) { 1363 audioBuffer.frameCount = userSize / mFrameSize; 1364 1365 status_t err = obtainBuffer(&audioBuffer, 1366 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1367 if (err < 0) { 1368 if (written > 0) { 1369 break; 1370 } 1371 return ssize_t(err); 1372 } 1373 1374 size_t toWrite; 1375 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1376 // Divide capacity by 2 to take expansion into account 1377 toWrite = audioBuffer.size >> 1; 1378 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1379 } else { 1380 toWrite = audioBuffer.size; 1381 memcpy(audioBuffer.i8, buffer, toWrite); 1382 } 1383 buffer = ((const char *) buffer) + toWrite; 1384 userSize -= toWrite; 1385 written += toWrite; 1386 1387 releaseBuffer(&audioBuffer); 1388 } 1389 1390 return written; 1391} 1392 1393// ------------------------------------------------------------------------- 1394 1395TimedAudioTrack::TimedAudioTrack() { 1396 mIsTimed = true; 1397} 1398 1399status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1400{ 1401 AutoMutex lock(mLock); 1402 status_t result = UNKNOWN_ERROR; 1403 1404#if 1 1405 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1406 // while we are accessing the cblk 1407 sp<IAudioTrack> audioTrack = mAudioTrack; 1408 sp<IMemory> iMem = mCblkMemory; 1409#endif 1410 1411 // If the track is not invalid already, try to allocate a buffer. alloc 1412 // fails indicating that the server is dead, flag the track as invalid so 1413 // we can attempt to restore in just a bit. 1414 audio_track_cblk_t* cblk = mCblk; 1415 if (!(cblk->mFlags & CBLK_INVALID)) { 1416 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1417 if (result == DEAD_OBJECT) { 1418 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1419 } 1420 } 1421 1422 // If the track is invalid at this point, attempt to restore it. and try the 1423 // allocation one more time. 1424 if (cblk->mFlags & CBLK_INVALID) { 1425 result = restoreTrack_l("allocateTimedBuffer"); 1426 1427 if (result == NO_ERROR) { 1428 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1429 } 1430 } 1431 1432 return result; 1433} 1434 1435status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1436 int64_t pts) 1437{ 1438 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1439 { 1440 AutoMutex lock(mLock); 1441 audio_track_cblk_t* cblk = mCblk; 1442 // restart track if it was disabled by audioflinger due to previous underrun 1443 if (buffer->size() != 0 && status == NO_ERROR && 1444 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1445 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1446 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1447 // FIXME ignoring status 1448 mAudioTrack->start(); 1449 } 1450 } 1451 return status; 1452} 1453 1454status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1455 TargetTimeline target) 1456{ 1457 return mAudioTrack->setMediaTimeTransform(xform, target); 1458} 1459 1460// ------------------------------------------------------------------------- 1461 1462nsecs_t AudioTrack::processAudioBuffer() 1463{ 1464 // Currently the AudioTrack thread is not created if there are no callbacks. 1465 // Would it ever make sense to run the thread, even without callbacks? 1466 // If so, then replace this by checks at each use for mCbf != NULL. 1467 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1468 1469 mLock.lock(); 1470 if (mAwaitBoost) { 1471 mAwaitBoost = false; 1472 mLock.unlock(); 1473 static const int32_t kMaxTries = 5; 1474 int32_t tryCounter = kMaxTries; 1475 uint32_t pollUs = 10000; 1476 do { 1477 int policy = sched_getscheduler(0); 1478 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1479 break; 1480 } 1481 usleep(pollUs); 1482 pollUs <<= 1; 1483 } while (tryCounter-- > 0); 1484 if (tryCounter < 0) { 1485 ALOGE("did not receive expected priority boost on time"); 1486 } 1487 // Run again immediately 1488 return 0; 1489 } 1490 1491 // Can only reference mCblk while locked 1492 int32_t flags = android_atomic_and( 1493 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1494 1495 // Check for track invalidation 1496 if (flags & CBLK_INVALID) { 1497 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1498 // AudioSystem cache. We should not exit here but after calling the callback so 1499 // that the upper layers can recreate the track 1500 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) { 1501 status_t status = restoreTrack_l("processAudioBuffer"); 1502 mLock.unlock(); 1503 // Run again immediately, but with a new IAudioTrack 1504 return 0; 1505 } 1506 } 1507 1508 bool waitStreamEnd = mState == STATE_STOPPING; 1509 bool active = mState == STATE_ACTIVE; 1510 1511 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1512 bool newUnderrun = false; 1513 if (flags & CBLK_UNDERRUN) { 1514#if 0 1515 // Currently in shared buffer mode, when the server reaches the end of buffer, 1516 // the track stays active in continuous underrun state. It's up to the application 1517 // to pause or stop the track, or set the position to a new offset within buffer. 1518 // This was some experimental code to auto-pause on underrun. Keeping it here 1519 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1520 if (mTransfer == TRANSFER_SHARED) { 1521 mState = STATE_PAUSED; 1522 active = false; 1523 } 1524#endif 1525 if (!mInUnderrun) { 1526 mInUnderrun = true; 1527 newUnderrun = true; 1528 } 1529 } 1530 1531 // Get current position of server 1532 size_t position = mProxy->getPosition(); 1533 1534 // Manage marker callback 1535 bool markerReached = false; 1536 size_t markerPosition = mMarkerPosition; 1537 // FIXME fails for wraparound, need 64 bits 1538 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1539 mMarkerReached = markerReached = true; 1540 } 1541 1542 // Determine number of new position callback(s) that will be needed, while locked 1543 size_t newPosCount = 0; 1544 size_t newPosition = mNewPosition; 1545 size_t updatePeriod = mUpdatePeriod; 1546 // FIXME fails for wraparound, need 64 bits 1547 if (updatePeriod > 0 && position >= newPosition) { 1548 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1549 mNewPosition += updatePeriod * newPosCount; 1550 } 1551 1552 // Cache other fields that will be needed soon 1553 uint32_t loopPeriod = mLoopPeriod; 1554 uint32_t sampleRate = mSampleRate; 1555 uint32_t notificationFrames = mNotificationFramesAct; 1556 if (mRefreshRemaining) { 1557 mRefreshRemaining = false; 1558 mRemainingFrames = notificationFrames; 1559 mRetryOnPartialBuffer = false; 1560 } 1561 size_t misalignment = mProxy->getMisalignment(); 1562 uint32_t sequence = mSequence; 1563 sp<AudioTrackClientProxy> proxy = mProxy; 1564 1565 // These fields don't need to be cached, because they are assigned only by set(): 1566 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1567 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1568 1569 mLock.unlock(); 1570 1571 if (waitStreamEnd) { 1572 struct timespec timeout; 1573 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1574 timeout.tv_nsec = 0; 1575 1576 status_t status = proxy->waitStreamEndDone(&timeout); 1577 switch (status) { 1578 case NO_ERROR: 1579 case DEAD_OBJECT: 1580 case TIMED_OUT: 1581 mCbf(EVENT_STREAM_END, mUserData, NULL); 1582 { 1583 AutoMutex lock(mLock); 1584 // The previously assigned value of waitStreamEnd is no longer valid, 1585 // since the mutex has been unlocked and either the callback handler 1586 // or another thread could have re-started the AudioTrack during that time. 1587 waitStreamEnd = mState == STATE_STOPPING; 1588 if (waitStreamEnd) { 1589 mState = STATE_STOPPED; 1590 } 1591 } 1592 if (waitStreamEnd && status != DEAD_OBJECT) { 1593 return NS_INACTIVE; 1594 } 1595 break; 1596 } 1597 return 0; 1598 } 1599 1600 // perform callbacks while unlocked 1601 if (newUnderrun) { 1602 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1603 } 1604 // FIXME we will miss loops if loop cycle was signaled several times since last call 1605 // to processAudioBuffer() 1606 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1607 mCbf(EVENT_LOOP_END, mUserData, NULL); 1608 } 1609 if (flags & CBLK_BUFFER_END) { 1610 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1611 } 1612 if (markerReached) { 1613 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1614 } 1615 while (newPosCount > 0) { 1616 size_t temp = newPosition; 1617 mCbf(EVENT_NEW_POS, mUserData, &temp); 1618 newPosition += updatePeriod; 1619 newPosCount--; 1620 } 1621 1622 if (mObservedSequence != sequence) { 1623 mObservedSequence = sequence; 1624 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1625 // for offloaded tracks, just wait for the upper layers to recreate the track 1626 if (isOffloadedOrDirect()) { 1627 return NS_INACTIVE; 1628 } 1629 } 1630 1631 // if inactive, then don't run me again until re-started 1632 if (!active) { 1633 return NS_INACTIVE; 1634 } 1635 1636 // Compute the estimated time until the next timed event (position, markers, loops) 1637 // FIXME only for non-compressed audio 1638 uint32_t minFrames = ~0; 1639 if (!markerReached && position < markerPosition) { 1640 minFrames = markerPosition - position; 1641 } 1642 if (loopPeriod > 0 && loopPeriod < minFrames) { 1643 minFrames = loopPeriod; 1644 } 1645 if (updatePeriod > 0 && updatePeriod < minFrames) { 1646 minFrames = updatePeriod; 1647 } 1648 1649 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1650 static const uint32_t kPoll = 0; 1651 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1652 minFrames = kPoll * notificationFrames; 1653 } 1654 1655 // Convert frame units to time units 1656 nsecs_t ns = NS_WHENEVER; 1657 if (minFrames != (uint32_t) ~0) { 1658 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1659 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1660 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1661 } 1662 1663 // If not supplying data by EVENT_MORE_DATA, then we're done 1664 if (mTransfer != TRANSFER_CALLBACK) { 1665 return ns; 1666 } 1667 1668 struct timespec timeout; 1669 const struct timespec *requested = &ClientProxy::kForever; 1670 if (ns != NS_WHENEVER) { 1671 timeout.tv_sec = ns / 1000000000LL; 1672 timeout.tv_nsec = ns % 1000000000LL; 1673 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1674 requested = &timeout; 1675 } 1676 1677 while (mRemainingFrames > 0) { 1678 1679 Buffer audioBuffer; 1680 audioBuffer.frameCount = mRemainingFrames; 1681 size_t nonContig; 1682 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1683 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1684 "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount); 1685 requested = &ClientProxy::kNonBlocking; 1686 size_t avail = audioBuffer.frameCount + nonContig; 1687 ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d", 1688 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1689 if (err != NO_ERROR) { 1690 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1691 (isOffloaded() && (err == DEAD_OBJECT))) { 1692 return 0; 1693 } 1694 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1695 return NS_NEVER; 1696 } 1697 1698 if (mRetryOnPartialBuffer && !isOffloaded()) { 1699 mRetryOnPartialBuffer = false; 1700 if (avail < mRemainingFrames) { 1701 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1702 if (ns < 0 || myns < ns) { 1703 ns = myns; 1704 } 1705 return ns; 1706 } 1707 } 1708 1709 // Divide buffer size by 2 to take into account the expansion 1710 // due to 8 to 16 bit conversion: the callback must fill only half 1711 // of the destination buffer 1712 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1713 audioBuffer.size >>= 1; 1714 } 1715 1716 size_t reqSize = audioBuffer.size; 1717 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1718 size_t writtenSize = audioBuffer.size; 1719 1720 // Sanity check on returned size 1721 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1722 ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes", 1723 reqSize, ssize_t(writtenSize)); 1724 return NS_NEVER; 1725 } 1726 1727 if (writtenSize == 0) { 1728 // The callback is done filling buffers 1729 // Keep this thread going to handle timed events and 1730 // still try to get more data in intervals of WAIT_PERIOD_MS 1731 // but don't just loop and block the CPU, so wait 1732 return WAIT_PERIOD_MS * 1000000LL; 1733 } 1734 1735 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1736 // 8 to 16 bit conversion, note that source and destination are the same address 1737 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1738 audioBuffer.size <<= 1; 1739 } 1740 1741 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1742 audioBuffer.frameCount = releasedFrames; 1743 mRemainingFrames -= releasedFrames; 1744 if (misalignment >= releasedFrames) { 1745 misalignment -= releasedFrames; 1746 } else { 1747 misalignment = 0; 1748 } 1749 1750 releaseBuffer(&audioBuffer); 1751 1752 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1753 // if callback doesn't like to accept the full chunk 1754 if (writtenSize < reqSize) { 1755 continue; 1756 } 1757 1758 // There could be enough non-contiguous frames available to satisfy the remaining request 1759 if (mRemainingFrames <= nonContig) { 1760 continue; 1761 } 1762 1763#if 0 1764 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1765 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1766 // that total to a sum == notificationFrames. 1767 if (0 < misalignment && misalignment <= mRemainingFrames) { 1768 mRemainingFrames = misalignment; 1769 return (mRemainingFrames * 1100000000LL) / sampleRate; 1770 } 1771#endif 1772 1773 } 1774 mRemainingFrames = notificationFrames; 1775 mRetryOnPartialBuffer = true; 1776 1777 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1778 return 0; 1779} 1780 1781status_t AudioTrack::restoreTrack_l(const char *from) 1782{ 1783 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1784 isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from); 1785 ++mSequence; 1786 status_t result; 1787 1788 // refresh the audio configuration cache in this process to make sure we get new 1789 // output parameters in createTrack_l() 1790 AudioSystem::clearAudioConfigCache(); 1791 1792 if (isOffloadedOrDirect_l()) { 1793 // FIXME re-creation of offloaded tracks is not yet implemented 1794 return DEAD_OBJECT; 1795 } 1796 1797 // if the new IAudioTrack is created, createTrack_l() will modify the 1798 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1799 // It will also delete the strong references on previous IAudioTrack and IMemory 1800 1801 // take the frames that will be lost by track recreation into account in saved position 1802 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1803 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1804 result = createTrack_l(position /*epoch*/); 1805 1806 if (result == NO_ERROR) { 1807 // continue playback from last known position, but 1808 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1809 if (mStaticProxy != NULL) { 1810 mLoopPeriod = 0; 1811 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1812 } 1813 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1814 // track destruction have been played? This is critical for SoundPool implementation 1815 // This must be broken, and needs to be tested/debugged. 1816#if 0 1817 // restore write index and set other indexes to reflect empty buffer status 1818 if (!strcmp(from, "start")) { 1819 // Make sure that a client relying on callback events indicating underrun or 1820 // the actual amount of audio frames played (e.g SoundPool) receives them. 1821 if (mSharedBuffer == 0) { 1822 // restart playback even if buffer is not completely filled. 1823 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1824 } 1825 } 1826#endif 1827 if (mState == STATE_ACTIVE) { 1828 result = mAudioTrack->start(); 1829 } 1830 } 1831 if (result != NO_ERROR) { 1832 ALOGW("restoreTrack_l() failed status %d", result); 1833 mState = STATE_STOPPED; 1834 } 1835 1836 return result; 1837} 1838 1839status_t AudioTrack::setParameters(const String8& keyValuePairs) 1840{ 1841 AutoMutex lock(mLock); 1842 return mAudioTrack->setParameters(keyValuePairs); 1843} 1844 1845status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1846{ 1847 AutoMutex lock(mLock); 1848 // FIXME not implemented for fast tracks; should use proxy and SSQ 1849 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1850 return INVALID_OPERATION; 1851 } 1852 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1853 return INVALID_OPERATION; 1854 } 1855 status_t status = mAudioTrack->getTimestamp(timestamp); 1856 if (status == NO_ERROR) { 1857 timestamp.mPosition += mProxy->getEpoch(); 1858 } 1859 return status; 1860} 1861 1862String8 AudioTrack::getParameters(const String8& keys) 1863{ 1864 audio_io_handle_t output = getOutput(); 1865 if (output != AUDIO_IO_HANDLE_NONE) { 1866 return AudioSystem::getParameters(output, keys); 1867 } else { 1868 return String8::empty(); 1869 } 1870} 1871 1872bool AudioTrack::isOffloaded() const 1873{ 1874 AutoMutex lock(mLock); 1875 return isOffloaded_l(); 1876} 1877 1878bool AudioTrack::isDirect() const 1879{ 1880 AutoMutex lock(mLock); 1881 return isDirect_l(); 1882} 1883 1884bool AudioTrack::isOffloadedOrDirect() const 1885{ 1886 AutoMutex lock(mLock); 1887 return isOffloadedOrDirect_l(); 1888} 1889 1890 1891status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 1892{ 1893 1894 const size_t SIZE = 256; 1895 char buffer[SIZE]; 1896 String8 result; 1897 1898 result.append(" AudioTrack::dump\n"); 1899 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1900 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 1901 result.append(buffer); 1902 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 1903 mChannelCount, mFrameCount); 1904 result.append(buffer); 1905 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1906 result.append(buffer); 1907 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1908 result.append(buffer); 1909 ::write(fd, result.string(), result.size()); 1910 return NO_ERROR; 1911} 1912 1913uint32_t AudioTrack::getUnderrunFrames() const 1914{ 1915 AutoMutex lock(mLock); 1916 return mProxy->getUnderrunFrames(); 1917} 1918 1919void AudioTrack::setAttributesFromStreamType(audio_stream_type_t streamType) { 1920 mAttributes.flags = 0x0; 1921 1922 switch(streamType) { 1923 case AUDIO_STREAM_DEFAULT: 1924 case AUDIO_STREAM_MUSIC: 1925 mAttributes.content_type = AUDIO_CONTENT_TYPE_MUSIC; 1926 mAttributes.usage = AUDIO_USAGE_MEDIA; 1927 break; 1928 case AUDIO_STREAM_VOICE_CALL: 1929 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 1930 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION; 1931 break; 1932 case AUDIO_STREAM_ENFORCED_AUDIBLE: 1933 mAttributes.flags |= AUDIO_FLAG_AUDIBILITY_ENFORCED; 1934 // intended fall through, attributes in common with STREAM_SYSTEM 1935 case AUDIO_STREAM_SYSTEM: 1936 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 1937 mAttributes.usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION; 1938 break; 1939 case AUDIO_STREAM_RING: 1940 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 1941 mAttributes.usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE; 1942 break; 1943 case AUDIO_STREAM_ALARM: 1944 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 1945 mAttributes.usage = AUDIO_USAGE_ALARM; 1946 break; 1947 case AUDIO_STREAM_NOTIFICATION: 1948 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 1949 mAttributes.usage = AUDIO_USAGE_NOTIFICATION; 1950 break; 1951 case AUDIO_STREAM_BLUETOOTH_SCO: 1952 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 1953 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION; 1954 mAttributes.flags |= AUDIO_FLAG_SCO; 1955 break; 1956 case AUDIO_STREAM_DTMF: 1957 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 1958 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING; 1959 break; 1960 case AUDIO_STREAM_TTS: 1961 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 1962 mAttributes.usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY; 1963 break; 1964 default: 1965 ALOGE("invalid stream type %d when converting to attributes", streamType); 1966 } 1967} 1968 1969void AudioTrack::setStreamTypeFromAttributes(audio_attributes_t& aa) { 1970 // flags to stream type mapping 1971 if ((aa.flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { 1972 mStreamType = AUDIO_STREAM_ENFORCED_AUDIBLE; 1973 return; 1974 } 1975 if ((aa.flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) { 1976 mStreamType = AUDIO_STREAM_BLUETOOTH_SCO; 1977 return; 1978 } 1979 1980 // usage to stream type mapping 1981 switch (aa.usage) { 1982 case AUDIO_USAGE_MEDIA: 1983 case AUDIO_USAGE_GAME: 1984 case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: 1985 case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: 1986 mStreamType = AUDIO_STREAM_MUSIC; 1987 return; 1988 case AUDIO_USAGE_ASSISTANCE_SONIFICATION: 1989 mStreamType = AUDIO_STREAM_SYSTEM; 1990 return; 1991 case AUDIO_USAGE_VOICE_COMMUNICATION: 1992 mStreamType = AUDIO_STREAM_VOICE_CALL; 1993 return; 1994 1995 case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: 1996 mStreamType = AUDIO_STREAM_DTMF; 1997 return; 1998 1999 case AUDIO_USAGE_ALARM: 2000 mStreamType = AUDIO_STREAM_ALARM; 2001 return; 2002 case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: 2003 mStreamType = AUDIO_STREAM_RING; 2004 return; 2005 2006 case AUDIO_USAGE_NOTIFICATION: 2007 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: 2008 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: 2009 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: 2010 case AUDIO_USAGE_NOTIFICATION_EVENT: 2011 mStreamType = AUDIO_STREAM_NOTIFICATION; 2012 return; 2013 2014 case AUDIO_USAGE_UNKNOWN: 2015 default: 2016 mStreamType = AUDIO_STREAM_MUSIC; 2017 } 2018} 2019 2020bool AudioTrack::isValidAttributes(const audio_attributes_t *paa) { 2021 // has flags that map to a strategy? 2022 if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO)) != 0) { 2023 return true; 2024 } 2025 2026 // has known usage? 2027 switch (paa->usage) { 2028 case AUDIO_USAGE_UNKNOWN: 2029 case AUDIO_USAGE_MEDIA: 2030 case AUDIO_USAGE_VOICE_COMMUNICATION: 2031 case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: 2032 case AUDIO_USAGE_ALARM: 2033 case AUDIO_USAGE_NOTIFICATION: 2034 case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: 2035 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: 2036 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: 2037 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: 2038 case AUDIO_USAGE_NOTIFICATION_EVENT: 2039 case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: 2040 case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: 2041 case AUDIO_USAGE_ASSISTANCE_SONIFICATION: 2042 case AUDIO_USAGE_GAME: 2043 break; 2044 default: 2045 return false; 2046 } 2047 return true; 2048} 2049// ========================================================================= 2050 2051void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 2052{ 2053 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 2054 if (audioTrack != 0) { 2055 AutoMutex lock(audioTrack->mLock); 2056 audioTrack->mProxy->binderDied(); 2057 } 2058} 2059 2060// ========================================================================= 2061 2062AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 2063 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 2064 mIgnoreNextPausedInt(false) 2065{ 2066} 2067 2068AudioTrack::AudioTrackThread::~AudioTrackThread() 2069{ 2070} 2071 2072bool AudioTrack::AudioTrackThread::threadLoop() 2073{ 2074 { 2075 AutoMutex _l(mMyLock); 2076 if (mPaused) { 2077 mMyCond.wait(mMyLock); 2078 // caller will check for exitPending() 2079 return true; 2080 } 2081 if (mIgnoreNextPausedInt) { 2082 mIgnoreNextPausedInt = false; 2083 mPausedInt = false; 2084 } 2085 if (mPausedInt) { 2086 if (mPausedNs > 0) { 2087 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 2088 } else { 2089 mMyCond.wait(mMyLock); 2090 } 2091 mPausedInt = false; 2092 return true; 2093 } 2094 } 2095 nsecs_t ns = mReceiver.processAudioBuffer(); 2096 switch (ns) { 2097 case 0: 2098 return true; 2099 case NS_INACTIVE: 2100 pauseInternal(); 2101 return true; 2102 case NS_NEVER: 2103 return false; 2104 case NS_WHENEVER: 2105 // FIXME increase poll interval, or make event-driven 2106 ns = 1000000000LL; 2107 // fall through 2108 default: 2109 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns); 2110 pauseInternal(ns); 2111 return true; 2112 } 2113} 2114 2115void AudioTrack::AudioTrackThread::requestExit() 2116{ 2117 // must be in this order to avoid a race condition 2118 Thread::requestExit(); 2119 resume(); 2120} 2121 2122void AudioTrack::AudioTrackThread::pause() 2123{ 2124 AutoMutex _l(mMyLock); 2125 mPaused = true; 2126} 2127 2128void AudioTrack::AudioTrackThread::resume() 2129{ 2130 AutoMutex _l(mMyLock); 2131 mIgnoreNextPausedInt = true; 2132 if (mPaused || mPausedInt) { 2133 mPaused = false; 2134 mPausedInt = false; 2135 mMyCond.signal(); 2136 } 2137} 2138 2139void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 2140{ 2141 AutoMutex _l(mMyLock); 2142 mPausedInt = true; 2143 mPausedNs = ns; 2144} 2145 2146}; // namespace android 2147