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#include <media/AudioResamplerPublic.h> 32 33#define WAIT_PERIOD_MS 10 34#define WAIT_STREAM_END_TIMEOUT_SEC 120 35 36 37namespace android { 38// --------------------------------------------------------------------------- 39 40static int64_t convertTimespecToUs(const struct timespec &tv) 41{ 42 return tv.tv_sec * 1000000ll + tv.tv_nsec / 1000; 43} 44 45// current monotonic time in microseconds. 46static int64_t getNowUs() 47{ 48 struct timespec tv; 49 (void) clock_gettime(CLOCK_MONOTONIC, &tv); 50 return convertTimespecToUs(tv); 51} 52 53// static 54status_t AudioTrack::getMinFrameCount( 55 size_t* frameCount, 56 audio_stream_type_t streamType, 57 uint32_t sampleRate) 58{ 59 if (frameCount == NULL) { 60 return BAD_VALUE; 61 } 62 63 // FIXME merge with similar code in createTrack_l(), except we're missing 64 // some information here that is available in createTrack_l(): 65 // audio_io_handle_t output 66 // audio_format_t format 67 // audio_channel_mask_t channelMask 68 // audio_output_flags_t flags 69 uint32_t afSampleRate; 70 status_t status; 71 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 72 if (status != NO_ERROR) { 73 ALOGE("Unable to query output sample rate for stream type %d; status %d", 74 streamType, status); 75 return status; 76 } 77 size_t afFrameCount; 78 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 79 if (status != NO_ERROR) { 80 ALOGE("Unable to query output frame count for stream type %d; status %d", 81 streamType, status); 82 return status; 83 } 84 uint32_t afLatency; 85 status = AudioSystem::getOutputLatency(&afLatency, streamType); 86 if (status != NO_ERROR) { 87 ALOGE("Unable to query output latency for stream type %d; status %d", 88 streamType, status); 89 return status; 90 } 91 92 // Ensure that buffer depth covers at least audio hardware latency 93 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 94 if (minBufCount < 2) { 95 minBufCount = 2; 96 } 97 98 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 99 afFrameCount * minBufCount * uint64_t(sampleRate) / afSampleRate; 100 // The formula above should always produce a non-zero value, but return an error 101 // in the unlikely event that it does not, as that's part of the API contract. 102 if (*frameCount == 0) { 103 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", 104 streamType, sampleRate); 105 return BAD_VALUE; 106 } 107 ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%d, afSampleRate=%d, afLatency=%d", 108 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 109 return NO_ERROR; 110} 111 112// --------------------------------------------------------------------------- 113 114AudioTrack::AudioTrack() 115 : mStatus(NO_INIT), 116 mIsTimed(false), 117 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 118 mPreviousSchedulingGroup(SP_DEFAULT), 119 mPausedPosition(0) 120{ 121 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN; 122 mAttributes.usage = AUDIO_USAGE_UNKNOWN; 123 mAttributes.flags = 0x0; 124 strcpy(mAttributes.tags, ""); 125} 126 127AudioTrack::AudioTrack( 128 audio_stream_type_t streamType, 129 uint32_t sampleRate, 130 audio_format_t format, 131 audio_channel_mask_t channelMask, 132 size_t frameCount, 133 audio_output_flags_t flags, 134 callback_t cbf, 135 void* user, 136 uint32_t notificationFrames, 137 int sessionId, 138 transfer_type transferType, 139 const audio_offload_info_t *offloadInfo, 140 int uid, 141 pid_t pid, 142 const audio_attributes_t* pAttributes) 143 : mStatus(NO_INIT), 144 mIsTimed(false), 145 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 146 mPreviousSchedulingGroup(SP_DEFAULT), 147 mPausedPosition(0) 148{ 149 mStatus = set(streamType, sampleRate, format, channelMask, 150 frameCount, flags, cbf, user, notificationFrames, 151 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 152 offloadInfo, uid, pid, pAttributes); 153} 154 155AudioTrack::AudioTrack( 156 audio_stream_type_t streamType, 157 uint32_t sampleRate, 158 audio_format_t format, 159 audio_channel_mask_t channelMask, 160 const sp<IMemory>& sharedBuffer, 161 audio_output_flags_t flags, 162 callback_t cbf, 163 void* user, 164 uint32_t notificationFrames, 165 int sessionId, 166 transfer_type transferType, 167 const audio_offload_info_t *offloadInfo, 168 int uid, 169 pid_t pid, 170 const audio_attributes_t* pAttributes) 171 : mStatus(NO_INIT), 172 mIsTimed(false), 173 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 174 mPreviousSchedulingGroup(SP_DEFAULT), 175 mPausedPosition(0) 176{ 177 mStatus = set(streamType, sampleRate, format, channelMask, 178 0 /*frameCount*/, flags, cbf, user, notificationFrames, 179 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, 180 uid, pid, pAttributes); 181} 182 183AudioTrack::~AudioTrack() 184{ 185 if (mStatus == NO_ERROR) { 186 // Make sure that callback function exits in the case where 187 // it is looping on buffer full condition in obtainBuffer(). 188 // Otherwise the callback thread will never exit. 189 stop(); 190 if (mAudioTrackThread != 0) { 191 mProxy->interrupt(); 192 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 193 mAudioTrackThread->requestExitAndWait(); 194 mAudioTrackThread.clear(); 195 } 196 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 197 mAudioTrack.clear(); 198 mCblkMemory.clear(); 199 mSharedBuffer.clear(); 200 IPCThreadState::self()->flushCommands(); 201 ALOGV("~AudioTrack, releasing session id from %d on behalf of %d", 202 IPCThreadState::self()->getCallingPid(), mClientPid); 203 AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); 204 } 205} 206 207status_t AudioTrack::set( 208 audio_stream_type_t streamType, 209 uint32_t sampleRate, 210 audio_format_t format, 211 audio_channel_mask_t channelMask, 212 size_t frameCount, 213 audio_output_flags_t flags, 214 callback_t cbf, 215 void* user, 216 uint32_t notificationFrames, 217 const sp<IMemory>& sharedBuffer, 218 bool threadCanCallJava, 219 int sessionId, 220 transfer_type transferType, 221 const audio_offload_info_t *offloadInfo, 222 int uid, 223 pid_t pid, 224 const audio_attributes_t* pAttributes) 225{ 226 ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 227 "flags #%x, notificationFrames %u, sessionId %d, transferType %d", 228 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, 229 sessionId, transferType); 230 231 switch (transferType) { 232 case TRANSFER_DEFAULT: 233 if (sharedBuffer != 0) { 234 transferType = TRANSFER_SHARED; 235 } else if (cbf == NULL || threadCanCallJava) { 236 transferType = TRANSFER_SYNC; 237 } else { 238 transferType = TRANSFER_CALLBACK; 239 } 240 break; 241 case TRANSFER_CALLBACK: 242 if (cbf == NULL || sharedBuffer != 0) { 243 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 244 return BAD_VALUE; 245 } 246 break; 247 case TRANSFER_OBTAIN: 248 case TRANSFER_SYNC: 249 if (sharedBuffer != 0) { 250 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 251 return BAD_VALUE; 252 } 253 break; 254 case TRANSFER_SHARED: 255 if (sharedBuffer == 0) { 256 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 257 return BAD_VALUE; 258 } 259 break; 260 default: 261 ALOGE("Invalid transfer type %d", transferType); 262 return BAD_VALUE; 263 } 264 mSharedBuffer = sharedBuffer; 265 mTransfer = transferType; 266 267 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 268 sharedBuffer->size()); 269 270 ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags); 271 272 AutoMutex lock(mLock); 273 274 // invariant that mAudioTrack != 0 is true only after set() returns successfully 275 if (mAudioTrack != 0) { 276 ALOGE("Track already in use"); 277 return INVALID_OPERATION; 278 } 279 280 // handle default values first. 281 if (streamType == AUDIO_STREAM_DEFAULT) { 282 streamType = AUDIO_STREAM_MUSIC; 283 } 284 285 if (pAttributes == NULL) { 286 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 287 ALOGE("Invalid stream type %d", streamType); 288 return BAD_VALUE; 289 } 290 setAttributesFromStreamType(streamType); 291 mStreamType = streamType; 292 } else { 293 if (!isValidAttributes(pAttributes)) { 294 ALOGE("Invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]", 295 pAttributes->usage, pAttributes->content_type, pAttributes->flags, 296 pAttributes->tags); 297 } 298 // stream type shouldn't be looked at, this track has audio attributes 299 memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t)); 300 setStreamTypeFromAttributes(mAttributes); 301 ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]", 302 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags); 303 } 304 305 status_t status; 306 if (sampleRate == 0) { 307 status = AudioSystem::getOutputSamplingRateForAttr(&sampleRate, &mAttributes); 308 if (status != NO_ERROR) { 309 ALOGE("Could not get output sample rate for stream type %d; status %d", 310 mStreamType, status); 311 return status; 312 } 313 } 314 mSampleRate = sampleRate; 315 316 // these below should probably come from the audioFlinger too... 317 if (format == AUDIO_FORMAT_DEFAULT) { 318 format = AUDIO_FORMAT_PCM_16_BIT; 319 } 320 321 // validate parameters 322 if (!audio_is_valid_format(format)) { 323 ALOGE("Invalid format %#x", format); 324 return BAD_VALUE; 325 } 326 mFormat = format; 327 328 if (!audio_is_output_channel(channelMask)) { 329 ALOGE("Invalid channel mask %#x", channelMask); 330 return BAD_VALUE; 331 } 332 mChannelMask = channelMask; 333 uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 334 mChannelCount = channelCount; 335 336 // AudioFlinger does not currently support 8-bit data in shared memory 337 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 338 ALOGE("8-bit data in shared memory is not supported"); 339 return BAD_VALUE; 340 } 341 342 // force direct flag if format is not linear PCM 343 // or offload was requested 344 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 345 || !audio_is_linear_pcm(format)) { 346 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 347 ? "Offload request, forcing to Direct Output" 348 : "Not linear PCM, forcing to Direct Output"); 349 flags = (audio_output_flags_t) 350 // FIXME why can't we allow direct AND fast? 351 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 352 } 353 // only allow deep buffering for music stream type 354 if (mStreamType != AUDIO_STREAM_MUSIC) { 355 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 356 } 357 358 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { 359 if (audio_is_linear_pcm(format)) { 360 mFrameSize = channelCount * audio_bytes_per_sample(format); 361 } else { 362 mFrameSize = sizeof(uint8_t); 363 } 364 mFrameSizeAF = mFrameSize; 365 } else { 366 ALOG_ASSERT(audio_is_linear_pcm(format)); 367 mFrameSize = channelCount * audio_bytes_per_sample(format); 368 mFrameSizeAF = channelCount * audio_bytes_per_sample( 369 format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format); 370 // createTrack will return an error if PCM format is not supported by server, 371 // so no need to check for specific PCM formats here 372 } 373 374 // Make copy of input parameter offloadInfo so that in the future: 375 // (a) createTrack_l doesn't need it as an input parameter 376 // (b) we can support re-creation of offloaded tracks 377 if (offloadInfo != NULL) { 378 mOffloadInfoCopy = *offloadInfo; 379 mOffloadInfo = &mOffloadInfoCopy; 380 } else { 381 mOffloadInfo = NULL; 382 } 383 384 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f; 385 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f; 386 mSendLevel = 0.0f; 387 // mFrameCount is initialized in createTrack_l 388 mReqFrameCount = frameCount; 389 mNotificationFramesReq = notificationFrames; 390 mNotificationFramesAct = 0; 391 mSessionId = sessionId; 392 int callingpid = IPCThreadState::self()->getCallingPid(); 393 int mypid = getpid(); 394 if (uid == -1 || (callingpid != mypid)) { 395 mClientUid = IPCThreadState::self()->getCallingUid(); 396 } else { 397 mClientUid = uid; 398 } 399 if (pid == -1 || (callingpid != mypid)) { 400 mClientPid = callingpid; 401 } else { 402 mClientPid = pid; 403 } 404 mAuxEffectId = 0; 405 mFlags = flags; 406 mCbf = cbf; 407 408 if (cbf != NULL) { 409 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 410 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 411 } 412 413 // create the IAudioTrack 414 status = createTrack_l(); 415 416 if (status != NO_ERROR) { 417 if (mAudioTrackThread != 0) { 418 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 419 mAudioTrackThread->requestExitAndWait(); 420 mAudioTrackThread.clear(); 421 } 422 return status; 423 } 424 425 mStatus = NO_ERROR; 426 mState = STATE_STOPPED; 427 mUserData = user; 428 mLoopPeriod = 0; 429 mMarkerPosition = 0; 430 mMarkerReached = false; 431 mNewPosition = 0; 432 mUpdatePeriod = 0; 433 mServer = 0; 434 mPosition = 0; 435 mReleased = 0; 436 mStartUs = 0; 437 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 438 mSequence = 1; 439 mObservedSequence = mSequence; 440 mInUnderrun = false; 441 442 return NO_ERROR; 443} 444 445// ------------------------------------------------------------------------- 446 447status_t AudioTrack::start() 448{ 449 AutoMutex lock(mLock); 450 451 if (mState == STATE_ACTIVE) { 452 return INVALID_OPERATION; 453 } 454 455 mInUnderrun = true; 456 457 State previousState = mState; 458 if (previousState == STATE_PAUSED_STOPPING) { 459 mState = STATE_STOPPING; 460 } else { 461 mState = STATE_ACTIVE; 462 } 463 (void) updateAndGetPosition_l(); 464 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 465 // reset current position as seen by client to 0 466 mPosition = 0; 467 // For offloaded tracks, we don't know if the hardware counters are really zero here, 468 // since the flush is asynchronous and stop may not fully drain. 469 // We save the time when the track is started to later verify whether 470 // the counters are realistic (i.e. start from zero after this time). 471 mStartUs = getNowUs(); 472 473 // force refresh of remaining frames by processAudioBuffer() as last 474 // write before stop could be partial. 475 mRefreshRemaining = true; 476 } 477 mNewPosition = mPosition + mUpdatePeriod; 478 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 479 480 sp<AudioTrackThread> t = mAudioTrackThread; 481 if (t != 0) { 482 if (previousState == STATE_STOPPING) { 483 mProxy->interrupt(); 484 } else { 485 t->resume(); 486 } 487 } else { 488 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 489 get_sched_policy(0, &mPreviousSchedulingGroup); 490 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 491 } 492 493 status_t status = NO_ERROR; 494 if (!(flags & CBLK_INVALID)) { 495 status = mAudioTrack->start(); 496 if (status == DEAD_OBJECT) { 497 flags |= CBLK_INVALID; 498 } 499 } 500 if (flags & CBLK_INVALID) { 501 status = restoreTrack_l("start"); 502 } 503 504 if (status != NO_ERROR) { 505 ALOGE("start() status %d", status); 506 mState = previousState; 507 if (t != 0) { 508 if (previousState != STATE_STOPPING) { 509 t->pause(); 510 } 511 } else { 512 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 513 set_sched_policy(0, mPreviousSchedulingGroup); 514 } 515 } 516 517 return status; 518} 519 520void AudioTrack::stop() 521{ 522 AutoMutex lock(mLock); 523 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 524 return; 525 } 526 527 if (isOffloaded_l()) { 528 mState = STATE_STOPPING; 529 } else { 530 mState = STATE_STOPPED; 531 mReleased = 0; 532 } 533 534 mProxy->interrupt(); 535 mAudioTrack->stop(); 536 // the playback head position will reset to 0, so if a marker is set, we need 537 // to activate it again 538 mMarkerReached = false; 539#if 0 540 // Force flush if a shared buffer is used otherwise audioflinger 541 // will not stop before end of buffer is reached. 542 // It may be needed to make sure that we stop playback, likely in case looping is on. 543 if (mSharedBuffer != 0) { 544 flush_l(); 545 } 546#endif 547 548 sp<AudioTrackThread> t = mAudioTrackThread; 549 if (t != 0) { 550 if (!isOffloaded_l()) { 551 t->pause(); 552 } 553 } else { 554 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 555 set_sched_policy(0, mPreviousSchedulingGroup); 556 } 557} 558 559bool AudioTrack::stopped() const 560{ 561 AutoMutex lock(mLock); 562 return mState != STATE_ACTIVE; 563} 564 565void AudioTrack::flush() 566{ 567 if (mSharedBuffer != 0) { 568 return; 569 } 570 AutoMutex lock(mLock); 571 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 572 return; 573 } 574 flush_l(); 575} 576 577void AudioTrack::flush_l() 578{ 579 ALOG_ASSERT(mState != STATE_ACTIVE); 580 581 // clear playback marker and periodic update counter 582 mMarkerPosition = 0; 583 mMarkerReached = false; 584 mUpdatePeriod = 0; 585 mRefreshRemaining = true; 586 587 mState = STATE_FLUSHED; 588 mReleased = 0; 589 if (isOffloaded_l()) { 590 mProxy->interrupt(); 591 } 592 mProxy->flush(); 593 mAudioTrack->flush(); 594} 595 596void AudioTrack::pause() 597{ 598 AutoMutex lock(mLock); 599 if (mState == STATE_ACTIVE) { 600 mState = STATE_PAUSED; 601 } else if (mState == STATE_STOPPING) { 602 mState = STATE_PAUSED_STOPPING; 603 } else { 604 return; 605 } 606 mProxy->interrupt(); 607 mAudioTrack->pause(); 608 609 if (isOffloaded_l()) { 610 if (mOutput != AUDIO_IO_HANDLE_NONE) { 611 // An offload output can be re-used between two audio tracks having 612 // the same configuration. A timestamp query for a paused track 613 // while the other is running would return an incorrect time. 614 // To fix this, cache the playback position on a pause() and return 615 // this time when requested until the track is resumed. 616 617 // OffloadThread sends HAL pause in its threadLoop. Time saved 618 // here can be slightly off. 619 620 // TODO: check return code for getRenderPosition. 621 622 uint32_t halFrames; 623 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); 624 ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); 625 } 626 } 627} 628 629status_t AudioTrack::setVolume(float left, float right) 630{ 631 // This duplicates a test by AudioTrack JNI, but that is not the only caller 632 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY || 633 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) { 634 return BAD_VALUE; 635 } 636 637 AutoMutex lock(mLock); 638 mVolume[AUDIO_INTERLEAVE_LEFT] = left; 639 mVolume[AUDIO_INTERLEAVE_RIGHT] = right; 640 641 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right))); 642 643 if (isOffloaded_l()) { 644 mAudioTrack->signal(); 645 } 646 return NO_ERROR; 647} 648 649status_t AudioTrack::setVolume(float volume) 650{ 651 return setVolume(volume, volume); 652} 653 654status_t AudioTrack::setAuxEffectSendLevel(float level) 655{ 656 // This duplicates a test by AudioTrack JNI, but that is not the only caller 657 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) { 658 return BAD_VALUE; 659 } 660 661 AutoMutex lock(mLock); 662 mSendLevel = level; 663 mProxy->setSendLevel(level); 664 665 return NO_ERROR; 666} 667 668void AudioTrack::getAuxEffectSendLevel(float* level) const 669{ 670 if (level != NULL) { 671 *level = mSendLevel; 672 } 673} 674 675status_t AudioTrack::setSampleRate(uint32_t rate) 676{ 677 if (mIsTimed || isOffloadedOrDirect()) { 678 return INVALID_OPERATION; 679 } 680 681 uint32_t afSamplingRate; 682 if (AudioSystem::getOutputSamplingRateForAttr(&afSamplingRate, &mAttributes) != NO_ERROR) { 683 return NO_INIT; 684 } 685 if (rate == 0 || rate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 686 return BAD_VALUE; 687 } 688 689 AutoMutex lock(mLock); 690 mSampleRate = rate; 691 mProxy->setSampleRate(rate); 692 693 return NO_ERROR; 694} 695 696uint32_t AudioTrack::getSampleRate() const 697{ 698 if (mIsTimed) { 699 return 0; 700 } 701 702 AutoMutex lock(mLock); 703 704 // sample rate can be updated during playback by the offloaded decoder so we need to 705 // query the HAL and update if needed. 706// FIXME use Proxy return channel to update the rate from server and avoid polling here 707 if (isOffloadedOrDirect_l()) { 708 if (mOutput != AUDIO_IO_HANDLE_NONE) { 709 uint32_t sampleRate = 0; 710 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate); 711 if (status == NO_ERROR) { 712 mSampleRate = sampleRate; 713 } 714 } 715 } 716 return mSampleRate; 717} 718 719status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 720{ 721 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 722 return INVALID_OPERATION; 723 } 724 725 if (loopCount == 0) { 726 ; 727 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 728 loopEnd - loopStart >= MIN_LOOP) { 729 ; 730 } else { 731 return BAD_VALUE; 732 } 733 734 AutoMutex lock(mLock); 735 // See setPosition() regarding setting parameters such as loop points or position while active 736 if (mState == STATE_ACTIVE) { 737 return INVALID_OPERATION; 738 } 739 setLoop_l(loopStart, loopEnd, loopCount); 740 return NO_ERROR; 741} 742 743void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 744{ 745 // FIXME If setting a loop also sets position to start of loop, then 746 // this is correct. Otherwise it should be removed. 747 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 748 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 749 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 750} 751 752status_t AudioTrack::setMarkerPosition(uint32_t marker) 753{ 754 // The only purpose of setting marker position is to get a callback 755 if (mCbf == NULL || isOffloadedOrDirect()) { 756 return INVALID_OPERATION; 757 } 758 759 AutoMutex lock(mLock); 760 mMarkerPosition = marker; 761 mMarkerReached = false; 762 763 return NO_ERROR; 764} 765 766status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 767{ 768 if (isOffloadedOrDirect()) { 769 return INVALID_OPERATION; 770 } 771 if (marker == NULL) { 772 return BAD_VALUE; 773 } 774 775 AutoMutex lock(mLock); 776 *marker = mMarkerPosition; 777 778 return NO_ERROR; 779} 780 781status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 782{ 783 // The only purpose of setting position update period is to get a callback 784 if (mCbf == NULL || isOffloadedOrDirect()) { 785 return INVALID_OPERATION; 786 } 787 788 AutoMutex lock(mLock); 789 mNewPosition = updateAndGetPosition_l() + updatePeriod; 790 mUpdatePeriod = updatePeriod; 791 792 return NO_ERROR; 793} 794 795status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 796{ 797 if (isOffloadedOrDirect()) { 798 return INVALID_OPERATION; 799 } 800 if (updatePeriod == NULL) { 801 return BAD_VALUE; 802 } 803 804 AutoMutex lock(mLock); 805 *updatePeriod = mUpdatePeriod; 806 807 return NO_ERROR; 808} 809 810status_t AudioTrack::setPosition(uint32_t position) 811{ 812 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 813 return INVALID_OPERATION; 814 } 815 if (position > mFrameCount) { 816 return BAD_VALUE; 817 } 818 819 AutoMutex lock(mLock); 820 // Currently we require that the player is inactive before setting parameters such as position 821 // or loop points. Otherwise, there could be a race condition: the application could read the 822 // current position, compute a new position or loop parameters, and then set that position or 823 // loop parameters but it would do the "wrong" thing since the position has continued to advance 824 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 825 // to specify how it wants to handle such scenarios. 826 if (mState == STATE_ACTIVE) { 827 return INVALID_OPERATION; 828 } 829 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 830 mLoopPeriod = 0; 831 // FIXME Check whether loops and setting position are incompatible in old code. 832 // If we use setLoop for both purposes we lose the capability to set the position while looping. 833 mStaticProxy->setLoop(position, mFrameCount, 0); 834 835 return NO_ERROR; 836} 837 838status_t AudioTrack::getPosition(uint32_t *position) 839{ 840 if (position == NULL) { 841 return BAD_VALUE; 842 } 843 844 AutoMutex lock(mLock); 845 if (isOffloadedOrDirect_l()) { 846 uint32_t dspFrames = 0; 847 848 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) { 849 ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); 850 *position = mPausedPosition; 851 return NO_ERROR; 852 } 853 854 if (mOutput != AUDIO_IO_HANDLE_NONE) { 855 uint32_t halFrames; 856 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 857 } 858 // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED) 859 // due to hardware latency. We leave this behavior for now. 860 *position = dspFrames; 861 } else { 862 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 863 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 864 0 : updateAndGetPosition_l(); 865 } 866 return NO_ERROR; 867} 868 869status_t AudioTrack::getBufferPosition(uint32_t *position) 870{ 871 if (mSharedBuffer == 0 || mIsTimed) { 872 return INVALID_OPERATION; 873 } 874 if (position == NULL) { 875 return BAD_VALUE; 876 } 877 878 AutoMutex lock(mLock); 879 *position = mStaticProxy->getBufferPosition(); 880 return NO_ERROR; 881} 882 883status_t AudioTrack::reload() 884{ 885 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 886 return INVALID_OPERATION; 887 } 888 889 AutoMutex lock(mLock); 890 // See setPosition() regarding setting parameters such as loop points or position while active 891 if (mState == STATE_ACTIVE) { 892 return INVALID_OPERATION; 893 } 894 mNewPosition = mUpdatePeriod; 895 mLoopPeriod = 0; 896 // FIXME The new code cannot reload while keeping a loop specified. 897 // Need to check how the old code handled this, and whether it's a significant change. 898 mStaticProxy->setLoop(0, mFrameCount, 0); 899 return NO_ERROR; 900} 901 902audio_io_handle_t AudioTrack::getOutput() const 903{ 904 AutoMutex lock(mLock); 905 return mOutput; 906} 907 908status_t AudioTrack::attachAuxEffect(int effectId) 909{ 910 AutoMutex lock(mLock); 911 status_t status = mAudioTrack->attachAuxEffect(effectId); 912 if (status == NO_ERROR) { 913 mAuxEffectId = effectId; 914 } 915 return status; 916} 917 918// ------------------------------------------------------------------------- 919 920// must be called with mLock held 921status_t AudioTrack::createTrack_l() 922{ 923 status_t status; 924 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 925 if (audioFlinger == 0) { 926 ALOGE("Could not get audioflinger"); 927 return NO_INIT; 928 } 929 930 audio_io_handle_t output = AudioSystem::getOutputForAttr(&mAttributes, mSampleRate, mFormat, 931 mChannelMask, mFlags, mOffloadInfo); 932 if (output == AUDIO_IO_HANDLE_NONE) { 933 ALOGE("Could not get audio output for stream type %d, usage %d, sample rate %u, format %#x," 934 " channel mask %#x, flags %#x", 935 mStreamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags); 936 return BAD_VALUE; 937 } 938 { 939 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 940 // we must release it ourselves if anything goes wrong. 941 942 // Not all of these values are needed under all conditions, but it is easier to get them all 943 944 uint32_t afLatency; 945 status = AudioSystem::getLatency(output, &afLatency); 946 if (status != NO_ERROR) { 947 ALOGE("getLatency(%d) failed status %d", output, status); 948 goto release; 949 } 950 951 size_t afFrameCount; 952 status = AudioSystem::getFrameCount(output, &afFrameCount); 953 if (status != NO_ERROR) { 954 ALOGE("getFrameCount(output=%d) status %d", output, status); 955 goto release; 956 } 957 958 uint32_t afSampleRate; 959 status = AudioSystem::getSamplingRate(output, &afSampleRate); 960 if (status != NO_ERROR) { 961 ALOGE("getSamplingRate(output=%d) status %d", output, status); 962 goto release; 963 } 964 965 // Client decides whether the track is TIMED (see below), but can only express a preference 966 // for FAST. Server will perform additional tests. 967 if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( 968 // either of these use cases: 969 // use case 1: shared buffer 970 (mSharedBuffer != 0) || 971 // use case 2: callback transfer mode 972 (mTransfer == TRANSFER_CALLBACK)) && 973 // matching sample rate 974 (mSampleRate == afSampleRate))) { 975 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 976 // once denied, do not request again if IAudioTrack is re-created 977 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 978 } 979 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 980 981 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 982 // n = 1 fast track with single buffering; nBuffering is ignored 983 // n = 2 fast track with double buffering 984 // n = 2 normal track, no sample rate conversion 985 // n = 3 normal track, with sample rate conversion 986 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 987 // n > 3 very high latency or very small notification interval; nBuffering is ignored 988 const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; 989 990 mNotificationFramesAct = mNotificationFramesReq; 991 992 size_t frameCount = mReqFrameCount; 993 if (!audio_is_linear_pcm(mFormat)) { 994 995 if (mSharedBuffer != 0) { 996 // Same comment as below about ignoring frameCount parameter for set() 997 frameCount = mSharedBuffer->size(); 998 } else if (frameCount == 0) { 999 frameCount = afFrameCount; 1000 } 1001 if (mNotificationFramesAct != frameCount) { 1002 mNotificationFramesAct = frameCount; 1003 } 1004 } else if (mSharedBuffer != 0) { 1005 1006 // Ensure that buffer alignment matches channel count 1007 // 8-bit data in shared memory is not currently supported by AudioFlinger 1008 size_t alignment = audio_bytes_per_sample( 1009 mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat); 1010 if (alignment & 1) { 1011 alignment = 1; 1012 } 1013 if (mChannelCount > 1) { 1014 // More than 2 channels does not require stronger alignment than stereo 1015 alignment <<= 1; 1016 } 1017 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 1018 ALOGE("Invalid buffer alignment: address %p, channel count %u", 1019 mSharedBuffer->pointer(), mChannelCount); 1020 status = BAD_VALUE; 1021 goto release; 1022 } 1023 1024 // When initializing a shared buffer AudioTrack via constructors, 1025 // there's no frameCount parameter. 1026 // But when initializing a shared buffer AudioTrack via set(), 1027 // there _is_ a frameCount parameter. We silently ignore it. 1028 frameCount = mSharedBuffer->size() / mFrameSizeAF; 1029 1030 } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 1031 1032 // FIXME move these calculations and associated checks to server 1033 1034 // Ensure that buffer depth covers at least audio hardware latency 1035 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 1036 ALOGV("afFrameCount=%zu, minBufCount=%d, afSampleRate=%u, afLatency=%d", 1037 afFrameCount, minBufCount, afSampleRate, afLatency); 1038 if (minBufCount <= nBuffering) { 1039 minBufCount = nBuffering; 1040 } 1041 1042 size_t minFrameCount = afFrameCount * minBufCount * uint64_t(mSampleRate) / afSampleRate; 1043 ALOGV("minFrameCount: %zu, afFrameCount=%zu, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 1044 ", afLatency=%d", 1045 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); 1046 1047 if (frameCount == 0) { 1048 frameCount = minFrameCount; 1049 } else if (frameCount < minFrameCount) { 1050 // not ALOGW because it happens all the time when playing key clicks over A2DP 1051 ALOGV("Minimum buffer size corrected from %zu to %zu", 1052 frameCount, minFrameCount); 1053 frameCount = minFrameCount; 1054 } 1055 // Make sure that application is notified with sufficient margin before underrun 1056 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1057 mNotificationFramesAct = frameCount/nBuffering; 1058 } 1059 1060 } else { 1061 // For fast tracks, the frame count calculations and checks are done by server 1062 } 1063 1064 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 1065 if (mIsTimed) { 1066 trackFlags |= IAudioFlinger::TRACK_TIMED; 1067 } 1068 1069 pid_t tid = -1; 1070 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1071 trackFlags |= IAudioFlinger::TRACK_FAST; 1072 if (mAudioTrackThread != 0) { 1073 tid = mAudioTrackThread->getTid(); 1074 } 1075 } 1076 1077 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1078 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 1079 } 1080 1081 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1082 trackFlags |= IAudioFlinger::TRACK_DIRECT; 1083 } 1084 1085 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1086 // but we will still need the original value also 1087 sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType, 1088 mSampleRate, 1089 // AudioFlinger only sees 16-bit PCM 1090 mFormat == AUDIO_FORMAT_PCM_8_BIT && 1091 !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT) ? 1092 AUDIO_FORMAT_PCM_16_BIT : mFormat, 1093 mChannelMask, 1094 &temp, 1095 &trackFlags, 1096 mSharedBuffer, 1097 output, 1098 tid, 1099 &mSessionId, 1100 mClientUid, 1101 &status); 1102 1103 if (status != NO_ERROR) { 1104 ALOGE("AudioFlinger could not create track, status: %d", status); 1105 goto release; 1106 } 1107 ALOG_ASSERT(track != 0); 1108 1109 // AudioFlinger now owns the reference to the I/O handle, 1110 // so we are no longer responsible for releasing it. 1111 1112 sp<IMemory> iMem = track->getCblk(); 1113 if (iMem == 0) { 1114 ALOGE("Could not get control block"); 1115 return NO_INIT; 1116 } 1117 void *iMemPointer = iMem->pointer(); 1118 if (iMemPointer == NULL) { 1119 ALOGE("Could not get control block pointer"); 1120 return NO_INIT; 1121 } 1122 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1123 if (mAudioTrack != 0) { 1124 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1125 mDeathNotifier.clear(); 1126 } 1127 mAudioTrack = track; 1128 mCblkMemory = iMem; 1129 IPCThreadState::self()->flushCommands(); 1130 1131 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1132 mCblk = cblk; 1133 // note that temp is the (possibly revised) value of frameCount 1134 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1135 // In current design, AudioTrack client checks and ensures frame count validity before 1136 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1137 // for fast track as it uses a special method of assigning frame count. 1138 ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp); 1139 } 1140 frameCount = temp; 1141 1142 mAwaitBoost = false; 1143 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1144 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1145 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount); 1146 mAwaitBoost = true; 1147 if (mSharedBuffer == 0) { 1148 // Theoretically double-buffering is not required for fast tracks, 1149 // due to tighter scheduling. But in practice, to accommodate kernels with 1150 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1151 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1152 mNotificationFramesAct = frameCount/nBuffering; 1153 } 1154 } 1155 } else { 1156 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount); 1157 // once denied, do not request again if IAudioTrack is re-created 1158 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1159 if (mSharedBuffer == 0) { 1160 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1161 mNotificationFramesAct = frameCount/nBuffering; 1162 } 1163 } 1164 } 1165 } 1166 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1167 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1168 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1169 } else { 1170 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1171 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1172 // FIXME This is a warning, not an error, so don't return error status 1173 //return NO_INIT; 1174 } 1175 } 1176 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1177 if (trackFlags & IAudioFlinger::TRACK_DIRECT) { 1178 ALOGV("AUDIO_OUTPUT_FLAG_DIRECT successful"); 1179 } else { 1180 ALOGW("AUDIO_OUTPUT_FLAG_DIRECT denied by server"); 1181 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT); 1182 // FIXME This is a warning, not an error, so don't return error status 1183 //return NO_INIT; 1184 } 1185 } 1186 1187 // We retain a copy of the I/O handle, but don't own the reference 1188 mOutput = output; 1189 mRefreshRemaining = true; 1190 1191 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1192 // is the value of pointer() for the shared buffer, otherwise buffers points 1193 // immediately after the control block. This address is for the mapping within client 1194 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1195 void* buffers; 1196 if (mSharedBuffer == 0) { 1197 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1198 } else { 1199 buffers = mSharedBuffer->pointer(); 1200 } 1201 1202 mAudioTrack->attachAuxEffect(mAuxEffectId); 1203 // FIXME don't believe this lie 1204 mLatency = afLatency + (1000*frameCount) / mSampleRate; 1205 1206 mFrameCount = frameCount; 1207 // If IAudioTrack is re-created, don't let the requested frameCount 1208 // decrease. This can confuse clients that cache frameCount(). 1209 if (frameCount > mReqFrameCount) { 1210 mReqFrameCount = frameCount; 1211 } 1212 1213 // update proxy 1214 if (mSharedBuffer == 0) { 1215 mStaticProxy.clear(); 1216 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1217 } else { 1218 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1219 mProxy = mStaticProxy; 1220 } 1221 mProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); 1222 mProxy->setSendLevel(mSendLevel); 1223 mProxy->setSampleRate(mSampleRate); 1224 mProxy->setMinimum(mNotificationFramesAct); 1225 1226 mDeathNotifier = new DeathNotifier(this); 1227 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1228 1229 return NO_ERROR; 1230 } 1231 1232release: 1233 AudioSystem::releaseOutput(output); 1234 if (status == NO_ERROR) { 1235 status = NO_INIT; 1236 } 1237 return status; 1238} 1239 1240status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1241{ 1242 if (audioBuffer == NULL) { 1243 return BAD_VALUE; 1244 } 1245 if (mTransfer != TRANSFER_OBTAIN) { 1246 audioBuffer->frameCount = 0; 1247 audioBuffer->size = 0; 1248 audioBuffer->raw = NULL; 1249 return INVALID_OPERATION; 1250 } 1251 1252 const struct timespec *requested; 1253 struct timespec timeout; 1254 if (waitCount == -1) { 1255 requested = &ClientProxy::kForever; 1256 } else if (waitCount == 0) { 1257 requested = &ClientProxy::kNonBlocking; 1258 } else if (waitCount > 0) { 1259 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1260 timeout.tv_sec = ms / 1000; 1261 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1262 requested = &timeout; 1263 } else { 1264 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1265 requested = NULL; 1266 } 1267 return obtainBuffer(audioBuffer, requested); 1268} 1269 1270status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1271 struct timespec *elapsed, size_t *nonContig) 1272{ 1273 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1274 uint32_t oldSequence = 0; 1275 uint32_t newSequence; 1276 1277 Proxy::Buffer buffer; 1278 status_t status = NO_ERROR; 1279 1280 static const int32_t kMaxTries = 5; 1281 int32_t tryCounter = kMaxTries; 1282 1283 do { 1284 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1285 // keep them from going away if another thread re-creates the track during obtainBuffer() 1286 sp<AudioTrackClientProxy> proxy; 1287 sp<IMemory> iMem; 1288 1289 { // start of lock scope 1290 AutoMutex lock(mLock); 1291 1292 newSequence = mSequence; 1293 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1294 if (status == DEAD_OBJECT) { 1295 // re-create track, unless someone else has already done so 1296 if (newSequence == oldSequence) { 1297 status = restoreTrack_l("obtainBuffer"); 1298 if (status != NO_ERROR) { 1299 buffer.mFrameCount = 0; 1300 buffer.mRaw = NULL; 1301 buffer.mNonContig = 0; 1302 break; 1303 } 1304 } 1305 } 1306 oldSequence = newSequence; 1307 1308 // Keep the extra references 1309 proxy = mProxy; 1310 iMem = mCblkMemory; 1311 1312 if (mState == STATE_STOPPING) { 1313 status = -EINTR; 1314 buffer.mFrameCount = 0; 1315 buffer.mRaw = NULL; 1316 buffer.mNonContig = 0; 1317 break; 1318 } 1319 1320 // Non-blocking if track is stopped or paused 1321 if (mState != STATE_ACTIVE) { 1322 requested = &ClientProxy::kNonBlocking; 1323 } 1324 1325 } // end of lock scope 1326 1327 buffer.mFrameCount = audioBuffer->frameCount; 1328 // FIXME starts the requested timeout and elapsed over from scratch 1329 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1330 1331 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1332 1333 audioBuffer->frameCount = buffer.mFrameCount; 1334 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1335 audioBuffer->raw = buffer.mRaw; 1336 if (nonContig != NULL) { 1337 *nonContig = buffer.mNonContig; 1338 } 1339 return status; 1340} 1341 1342void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1343{ 1344 if (mTransfer == TRANSFER_SHARED) { 1345 return; 1346 } 1347 1348 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1349 if (stepCount == 0) { 1350 return; 1351 } 1352 1353 Proxy::Buffer buffer; 1354 buffer.mFrameCount = stepCount; 1355 buffer.mRaw = audioBuffer->raw; 1356 1357 AutoMutex lock(mLock); 1358 mReleased += stepCount; 1359 mInUnderrun = false; 1360 mProxy->releaseBuffer(&buffer); 1361 1362 // restart track if it was disabled by audioflinger due to previous underrun 1363 if (mState == STATE_ACTIVE) { 1364 audio_track_cblk_t* cblk = mCblk; 1365 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1366 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1367 // FIXME ignoring status 1368 mAudioTrack->start(); 1369 } 1370 } 1371} 1372 1373// ------------------------------------------------------------------------- 1374 1375ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1376{ 1377 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1378 return INVALID_OPERATION; 1379 } 1380 1381 if (isDirect()) { 1382 AutoMutex lock(mLock); 1383 int32_t flags = android_atomic_and( 1384 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), 1385 &mCblk->mFlags); 1386 if (flags & CBLK_INVALID) { 1387 return DEAD_OBJECT; 1388 } 1389 } 1390 1391 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1392 // Sanity-check: user is most-likely passing an error code, and it would 1393 // make the return value ambiguous (actualSize vs error). 1394 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1395 return BAD_VALUE; 1396 } 1397 1398 size_t written = 0; 1399 Buffer audioBuffer; 1400 1401 while (userSize >= mFrameSize) { 1402 audioBuffer.frameCount = userSize / mFrameSize; 1403 1404 status_t err = obtainBuffer(&audioBuffer, 1405 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1406 if (err < 0) { 1407 if (written > 0) { 1408 break; 1409 } 1410 return ssize_t(err); 1411 } 1412 1413 size_t toWrite; 1414 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1415 // Divide capacity by 2 to take expansion into account 1416 toWrite = audioBuffer.size >> 1; 1417 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1418 } else { 1419 toWrite = audioBuffer.size; 1420 memcpy(audioBuffer.i8, buffer, toWrite); 1421 } 1422 buffer = ((const char *) buffer) + toWrite; 1423 userSize -= toWrite; 1424 written += toWrite; 1425 1426 releaseBuffer(&audioBuffer); 1427 } 1428 1429 return written; 1430} 1431 1432// ------------------------------------------------------------------------- 1433 1434TimedAudioTrack::TimedAudioTrack() { 1435 mIsTimed = true; 1436} 1437 1438status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1439{ 1440 AutoMutex lock(mLock); 1441 status_t result = UNKNOWN_ERROR; 1442 1443#if 1 1444 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1445 // while we are accessing the cblk 1446 sp<IAudioTrack> audioTrack = mAudioTrack; 1447 sp<IMemory> iMem = mCblkMemory; 1448#endif 1449 1450 // If the track is not invalid already, try to allocate a buffer. alloc 1451 // fails indicating that the server is dead, flag the track as invalid so 1452 // we can attempt to restore in just a bit. 1453 audio_track_cblk_t* cblk = mCblk; 1454 if (!(cblk->mFlags & CBLK_INVALID)) { 1455 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1456 if (result == DEAD_OBJECT) { 1457 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1458 } 1459 } 1460 1461 // If the track is invalid at this point, attempt to restore it. and try the 1462 // allocation one more time. 1463 if (cblk->mFlags & CBLK_INVALID) { 1464 result = restoreTrack_l("allocateTimedBuffer"); 1465 1466 if (result == NO_ERROR) { 1467 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1468 } 1469 } 1470 1471 return result; 1472} 1473 1474status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1475 int64_t pts) 1476{ 1477 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1478 { 1479 AutoMutex lock(mLock); 1480 audio_track_cblk_t* cblk = mCblk; 1481 // restart track if it was disabled by audioflinger due to previous underrun 1482 if (buffer->size() != 0 && status == NO_ERROR && 1483 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1484 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1485 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1486 // FIXME ignoring status 1487 mAudioTrack->start(); 1488 } 1489 } 1490 return status; 1491} 1492 1493status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1494 TargetTimeline target) 1495{ 1496 return mAudioTrack->setMediaTimeTransform(xform, target); 1497} 1498 1499// ------------------------------------------------------------------------- 1500 1501nsecs_t AudioTrack::processAudioBuffer() 1502{ 1503 // Currently the AudioTrack thread is not created if there are no callbacks. 1504 // Would it ever make sense to run the thread, even without callbacks? 1505 // If so, then replace this by checks at each use for mCbf != NULL. 1506 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1507 1508 mLock.lock(); 1509 if (mAwaitBoost) { 1510 mAwaitBoost = false; 1511 mLock.unlock(); 1512 static const int32_t kMaxTries = 5; 1513 int32_t tryCounter = kMaxTries; 1514 uint32_t pollUs = 10000; 1515 do { 1516 int policy = sched_getscheduler(0); 1517 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1518 break; 1519 } 1520 usleep(pollUs); 1521 pollUs <<= 1; 1522 } while (tryCounter-- > 0); 1523 if (tryCounter < 0) { 1524 ALOGE("did not receive expected priority boost on time"); 1525 } 1526 // Run again immediately 1527 return 0; 1528 } 1529 1530 // Can only reference mCblk while locked 1531 int32_t flags = android_atomic_and( 1532 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1533 1534 // Check for track invalidation 1535 if (flags & CBLK_INVALID) { 1536 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1537 // AudioSystem cache. We should not exit here but after calling the callback so 1538 // that the upper layers can recreate the track 1539 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) { 1540 status_t status = restoreTrack_l("processAudioBuffer"); 1541 mLock.unlock(); 1542 // Run again immediately, but with a new IAudioTrack 1543 return 0; 1544 } 1545 } 1546 1547 bool waitStreamEnd = mState == STATE_STOPPING; 1548 bool active = mState == STATE_ACTIVE; 1549 1550 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1551 bool newUnderrun = false; 1552 if (flags & CBLK_UNDERRUN) { 1553#if 0 1554 // Currently in shared buffer mode, when the server reaches the end of buffer, 1555 // the track stays active in continuous underrun state. It's up to the application 1556 // to pause or stop the track, or set the position to a new offset within buffer. 1557 // This was some experimental code to auto-pause on underrun. Keeping it here 1558 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1559 if (mTransfer == TRANSFER_SHARED) { 1560 mState = STATE_PAUSED; 1561 active = false; 1562 } 1563#endif 1564 if (!mInUnderrun) { 1565 mInUnderrun = true; 1566 newUnderrun = true; 1567 } 1568 } 1569 1570 // Get current position of server 1571 size_t position = updateAndGetPosition_l(); 1572 1573 // Manage marker callback 1574 bool markerReached = false; 1575 size_t markerPosition = mMarkerPosition; 1576 // FIXME fails for wraparound, need 64 bits 1577 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1578 mMarkerReached = markerReached = true; 1579 } 1580 1581 // Determine number of new position callback(s) that will be needed, while locked 1582 size_t newPosCount = 0; 1583 size_t newPosition = mNewPosition; 1584 size_t updatePeriod = mUpdatePeriod; 1585 // FIXME fails for wraparound, need 64 bits 1586 if (updatePeriod > 0 && position >= newPosition) { 1587 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1588 mNewPosition += updatePeriod * newPosCount; 1589 } 1590 1591 // Cache other fields that will be needed soon 1592 uint32_t loopPeriod = mLoopPeriod; 1593 uint32_t sampleRate = mSampleRate; 1594 uint32_t notificationFrames = mNotificationFramesAct; 1595 if (mRefreshRemaining) { 1596 mRefreshRemaining = false; 1597 mRemainingFrames = notificationFrames; 1598 mRetryOnPartialBuffer = false; 1599 } 1600 size_t misalignment = mProxy->getMisalignment(); 1601 uint32_t sequence = mSequence; 1602 sp<AudioTrackClientProxy> proxy = mProxy; 1603 1604 // These fields don't need to be cached, because they are assigned only by set(): 1605 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1606 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1607 1608 mLock.unlock(); 1609 1610 if (waitStreamEnd) { 1611 struct timespec timeout; 1612 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1613 timeout.tv_nsec = 0; 1614 1615 status_t status = proxy->waitStreamEndDone(&timeout); 1616 switch (status) { 1617 case NO_ERROR: 1618 case DEAD_OBJECT: 1619 case TIMED_OUT: 1620 mCbf(EVENT_STREAM_END, mUserData, NULL); 1621 { 1622 AutoMutex lock(mLock); 1623 // The previously assigned value of waitStreamEnd is no longer valid, 1624 // since the mutex has been unlocked and either the callback handler 1625 // or another thread could have re-started the AudioTrack during that time. 1626 waitStreamEnd = mState == STATE_STOPPING; 1627 if (waitStreamEnd) { 1628 mState = STATE_STOPPED; 1629 mReleased = 0; 1630 } 1631 } 1632 if (waitStreamEnd && status != DEAD_OBJECT) { 1633 return NS_INACTIVE; 1634 } 1635 break; 1636 } 1637 return 0; 1638 } 1639 1640 // perform callbacks while unlocked 1641 if (newUnderrun) { 1642 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1643 } 1644 // FIXME we will miss loops if loop cycle was signaled several times since last call 1645 // to processAudioBuffer() 1646 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1647 mCbf(EVENT_LOOP_END, mUserData, NULL); 1648 } 1649 if (flags & CBLK_BUFFER_END) { 1650 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1651 } 1652 if (markerReached) { 1653 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1654 } 1655 while (newPosCount > 0) { 1656 size_t temp = newPosition; 1657 mCbf(EVENT_NEW_POS, mUserData, &temp); 1658 newPosition += updatePeriod; 1659 newPosCount--; 1660 } 1661 1662 if (mObservedSequence != sequence) { 1663 mObservedSequence = sequence; 1664 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1665 // for offloaded tracks, just wait for the upper layers to recreate the track 1666 if (isOffloadedOrDirect()) { 1667 return NS_INACTIVE; 1668 } 1669 } 1670 1671 // if inactive, then don't run me again until re-started 1672 if (!active) { 1673 return NS_INACTIVE; 1674 } 1675 1676 // Compute the estimated time until the next timed event (position, markers, loops) 1677 // FIXME only for non-compressed audio 1678 uint32_t minFrames = ~0; 1679 if (!markerReached && position < markerPosition) { 1680 minFrames = markerPosition - position; 1681 } 1682 if (loopPeriod > 0 && loopPeriod < minFrames) { 1683 minFrames = loopPeriod; 1684 } 1685 if (updatePeriod > 0 && updatePeriod < minFrames) { 1686 minFrames = updatePeriod; 1687 } 1688 1689 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1690 static const uint32_t kPoll = 0; 1691 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1692 minFrames = kPoll * notificationFrames; 1693 } 1694 1695 // Convert frame units to time units 1696 nsecs_t ns = NS_WHENEVER; 1697 if (minFrames != (uint32_t) ~0) { 1698 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1699 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1700 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1701 } 1702 1703 // If not supplying data by EVENT_MORE_DATA, then we're done 1704 if (mTransfer != TRANSFER_CALLBACK) { 1705 return ns; 1706 } 1707 1708 struct timespec timeout; 1709 const struct timespec *requested = &ClientProxy::kForever; 1710 if (ns != NS_WHENEVER) { 1711 timeout.tv_sec = ns / 1000000000LL; 1712 timeout.tv_nsec = ns % 1000000000LL; 1713 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1714 requested = &timeout; 1715 } 1716 1717 while (mRemainingFrames > 0) { 1718 1719 Buffer audioBuffer; 1720 audioBuffer.frameCount = mRemainingFrames; 1721 size_t nonContig; 1722 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1723 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1724 "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount); 1725 requested = &ClientProxy::kNonBlocking; 1726 size_t avail = audioBuffer.frameCount + nonContig; 1727 ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d", 1728 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1729 if (err != NO_ERROR) { 1730 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1731 (isOffloaded() && (err == DEAD_OBJECT))) { 1732 return 0; 1733 } 1734 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1735 return NS_NEVER; 1736 } 1737 1738 if (mRetryOnPartialBuffer && !isOffloaded()) { 1739 mRetryOnPartialBuffer = false; 1740 if (avail < mRemainingFrames) { 1741 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1742 if (ns < 0 || myns < ns) { 1743 ns = myns; 1744 } 1745 return ns; 1746 } 1747 } 1748 1749 // Divide buffer size by 2 to take into account the expansion 1750 // due to 8 to 16 bit conversion: the callback must fill only half 1751 // of the destination buffer 1752 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1753 audioBuffer.size >>= 1; 1754 } 1755 1756 size_t reqSize = audioBuffer.size; 1757 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1758 size_t writtenSize = audioBuffer.size; 1759 1760 // Sanity check on returned size 1761 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1762 ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes", 1763 reqSize, ssize_t(writtenSize)); 1764 return NS_NEVER; 1765 } 1766 1767 if (writtenSize == 0) { 1768 // The callback is done filling buffers 1769 // Keep this thread going to handle timed events and 1770 // still try to get more data in intervals of WAIT_PERIOD_MS 1771 // but don't just loop and block the CPU, so wait 1772 return WAIT_PERIOD_MS * 1000000LL; 1773 } 1774 1775 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1776 // 8 to 16 bit conversion, note that source and destination are the same address 1777 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1778 audioBuffer.size <<= 1; 1779 } 1780 1781 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1782 audioBuffer.frameCount = releasedFrames; 1783 mRemainingFrames -= releasedFrames; 1784 if (misalignment >= releasedFrames) { 1785 misalignment -= releasedFrames; 1786 } else { 1787 misalignment = 0; 1788 } 1789 1790 releaseBuffer(&audioBuffer); 1791 1792 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1793 // if callback doesn't like to accept the full chunk 1794 if (writtenSize < reqSize) { 1795 continue; 1796 } 1797 1798 // There could be enough non-contiguous frames available to satisfy the remaining request 1799 if (mRemainingFrames <= nonContig) { 1800 continue; 1801 } 1802 1803#if 0 1804 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1805 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1806 // that total to a sum == notificationFrames. 1807 if (0 < misalignment && misalignment <= mRemainingFrames) { 1808 mRemainingFrames = misalignment; 1809 return (mRemainingFrames * 1100000000LL) / sampleRate; 1810 } 1811#endif 1812 1813 } 1814 mRemainingFrames = notificationFrames; 1815 mRetryOnPartialBuffer = true; 1816 1817 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1818 return 0; 1819} 1820 1821status_t AudioTrack::restoreTrack_l(const char *from) 1822{ 1823 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1824 isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from); 1825 ++mSequence; 1826 status_t result; 1827 1828 // refresh the audio configuration cache in this process to make sure we get new 1829 // output parameters in createTrack_l() 1830 AudioSystem::clearAudioConfigCache(); 1831 1832 if (isOffloadedOrDirect_l()) { 1833 // FIXME re-creation of offloaded tracks is not yet implemented 1834 return DEAD_OBJECT; 1835 } 1836 1837 // save the old static buffer position 1838 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1839 1840 // If a new IAudioTrack is successfully created, createTrack_l() will modify the 1841 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1842 // It will also delete the strong references on previous IAudioTrack and IMemory. 1843 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact. 1844 result = createTrack_l(); 1845 1846 // take the frames that will be lost by track recreation into account in saved position 1847 (void) updateAndGetPosition_l(); 1848 mPosition = mReleased; 1849 1850 if (result == NO_ERROR) { 1851 // continue playback from last known position, but 1852 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1853 if (mStaticProxy != NULL) { 1854 mLoopPeriod = 0; 1855 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1856 } 1857 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1858 // track destruction have been played? This is critical for SoundPool implementation 1859 // This must be broken, and needs to be tested/debugged. 1860#if 0 1861 // restore write index and set other indexes to reflect empty buffer status 1862 if (!strcmp(from, "start")) { 1863 // Make sure that a client relying on callback events indicating underrun or 1864 // the actual amount of audio frames played (e.g SoundPool) receives them. 1865 if (mSharedBuffer == 0) { 1866 // restart playback even if buffer is not completely filled. 1867 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1868 } 1869 } 1870#endif 1871 if (mState == STATE_ACTIVE) { 1872 result = mAudioTrack->start(); 1873 } 1874 } 1875 if (result != NO_ERROR) { 1876 ALOGW("restoreTrack_l() failed status %d", result); 1877 mState = STATE_STOPPED; 1878 mReleased = 0; 1879 } 1880 1881 return result; 1882} 1883 1884uint32_t AudioTrack::updateAndGetPosition_l() 1885{ 1886 // This is the sole place to read server consumed frames 1887 uint32_t newServer = mProxy->getPosition(); 1888 int32_t delta = newServer - mServer; 1889 mServer = newServer; 1890 // TODO There is controversy about whether there can be "negative jitter" in server position. 1891 // This should be investigated further, and if possible, it should be addressed. 1892 // A more definite failure mode is infrequent polling by client. 1893 // One could call (void)getPosition_l() in releaseBuffer(), 1894 // so mReleased and mPosition are always lock-step as best possible. 1895 // That should ensure delta never goes negative for infrequent polling 1896 // unless the server has more than 2^31 frames in its buffer, 1897 // in which case the use of uint32_t for these counters has bigger issues. 1898 if (delta < 0) { 1899 ALOGE("detected illegal retrograde motion by the server: mServer advanced by %d", delta); 1900 delta = 0; 1901 } 1902 return mPosition += (uint32_t) delta; 1903} 1904 1905status_t AudioTrack::setParameters(const String8& keyValuePairs) 1906{ 1907 AutoMutex lock(mLock); 1908 return mAudioTrack->setParameters(keyValuePairs); 1909} 1910 1911status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1912{ 1913 AutoMutex lock(mLock); 1914 // FIXME not implemented for fast tracks; should use proxy and SSQ 1915 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1916 return INVALID_OPERATION; 1917 } 1918 1919 switch (mState) { 1920 case STATE_ACTIVE: 1921 case STATE_PAUSED: 1922 break; // handle below 1923 case STATE_FLUSHED: 1924 case STATE_STOPPED: 1925 return WOULD_BLOCK; 1926 case STATE_STOPPING: 1927 case STATE_PAUSED_STOPPING: 1928 if (!isOffloaded_l()) { 1929 return INVALID_OPERATION; 1930 } 1931 break; // offloaded tracks handled below 1932 default: 1933 LOG_ALWAYS_FATAL("Invalid mState in getTimestamp(): %d", mState); 1934 break; 1935 } 1936 1937 // The presented frame count must always lag behind the consumed frame count. 1938 // To avoid a race, read the presented frames first. This ensures that presented <= consumed. 1939 status_t status = mAudioTrack->getTimestamp(timestamp); 1940 if (status != NO_ERROR) { 1941 ALOGV_IF(status != WOULD_BLOCK, "getTimestamp error:%#x", status); 1942 return status; 1943 } 1944 if (isOffloadedOrDirect_l()) { 1945 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) { 1946 // use cached paused position in case another offloaded track is running. 1947 timestamp.mPosition = mPausedPosition; 1948 clock_gettime(CLOCK_MONOTONIC, ×tamp.mTime); 1949 return NO_ERROR; 1950 } 1951 1952 // Check whether a pending flush or stop has completed, as those commands may 1953 // be asynchronous or return near finish. 1954 if (mStartUs != 0 && mSampleRate != 0) { 1955 static const int kTimeJitterUs = 100000; // 100 ms 1956 static const int k1SecUs = 1000000; 1957 1958 const int64_t timeNow = getNowUs(); 1959 1960 if (timeNow < mStartUs + k1SecUs) { // within first second of starting 1961 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime); 1962 if (timestampTimeUs < mStartUs) { 1963 return WOULD_BLOCK; // stale timestamp time, occurs before start. 1964 } 1965 const int64_t deltaTimeUs = timestampTimeUs - mStartUs; 1966 const int64_t deltaPositionByUs = timestamp.mPosition * 1000000LL / mSampleRate; 1967 1968 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) { 1969 // Verify that the counter can't count faster than the sample rate 1970 // since the start time. If greater, then that means we have failed 1971 // to completely flush or stop the previous playing track. 1972 ALOGW("incomplete flush or stop:" 1973 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)", 1974 (long long)deltaTimeUs, (long long)deltaPositionByUs, 1975 timestamp.mPosition); 1976 return WOULD_BLOCK; 1977 } 1978 } 1979 mStartUs = 0; // no need to check again, start timestamp has either expired or unneeded. 1980 } 1981 } else { 1982 // Update the mapping between local consumed (mPosition) and server consumed (mServer) 1983 (void) updateAndGetPosition_l(); 1984 // Server consumed (mServer) and presented both use the same server time base, 1985 // and server consumed is always >= presented. 1986 // The delta between these represents the number of frames in the buffer pipeline. 1987 // If this delta between these is greater than the client position, it means that 1988 // actually presented is still stuck at the starting line (figuratively speaking), 1989 // waiting for the first frame to go by. So we can't report a valid timestamp yet. 1990 if ((uint32_t) (mServer - timestamp.mPosition) > mPosition) { 1991 return INVALID_OPERATION; 1992 } 1993 // Convert timestamp position from server time base to client time base. 1994 // TODO The following code should work OK now because timestamp.mPosition is 32-bit. 1995 // But if we change it to 64-bit then this could fail. 1996 // If (mPosition - mServer) can be negative then should use: 1997 // (int32_t)(mPosition - mServer) 1998 timestamp.mPosition += mPosition - mServer; 1999 // Immediately after a call to getPosition_l(), mPosition and 2000 // mServer both represent the same frame position. mPosition is 2001 // in client's point of view, and mServer is in server's point of 2002 // view. So the difference between them is the "fudge factor" 2003 // between client and server views due to stop() and/or new 2004 // IAudioTrack. And timestamp.mPosition is initially in server's 2005 // point of view, so we need to apply the same fudge factor to it. 2006 } 2007 return status; 2008} 2009 2010String8 AudioTrack::getParameters(const String8& keys) 2011{ 2012 audio_io_handle_t output = getOutput(); 2013 if (output != AUDIO_IO_HANDLE_NONE) { 2014 return AudioSystem::getParameters(output, keys); 2015 } else { 2016 return String8::empty(); 2017 } 2018} 2019 2020bool AudioTrack::isOffloaded() const 2021{ 2022 AutoMutex lock(mLock); 2023 return isOffloaded_l(); 2024} 2025 2026bool AudioTrack::isDirect() const 2027{ 2028 AutoMutex lock(mLock); 2029 return isDirect_l(); 2030} 2031 2032bool AudioTrack::isOffloadedOrDirect() const 2033{ 2034 AutoMutex lock(mLock); 2035 return isOffloadedOrDirect_l(); 2036} 2037 2038 2039status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 2040{ 2041 2042 const size_t SIZE = 256; 2043 char buffer[SIZE]; 2044 String8 result; 2045 2046 result.append(" AudioTrack::dump\n"); 2047 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 2048 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 2049 result.append(buffer); 2050 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 2051 mChannelCount, mFrameCount); 2052 result.append(buffer); 2053 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 2054 result.append(buffer); 2055 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 2056 result.append(buffer); 2057 ::write(fd, result.string(), result.size()); 2058 return NO_ERROR; 2059} 2060 2061uint32_t AudioTrack::getUnderrunFrames() const 2062{ 2063 AutoMutex lock(mLock); 2064 return mProxy->getUnderrunFrames(); 2065} 2066 2067void AudioTrack::setAttributesFromStreamType(audio_stream_type_t streamType) { 2068 mAttributes.flags = 0x0; 2069 2070 switch(streamType) { 2071 case AUDIO_STREAM_DEFAULT: 2072 case AUDIO_STREAM_MUSIC: 2073 mAttributes.content_type = AUDIO_CONTENT_TYPE_MUSIC; 2074 mAttributes.usage = AUDIO_USAGE_MEDIA; 2075 break; 2076 case AUDIO_STREAM_VOICE_CALL: 2077 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 2078 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION; 2079 break; 2080 case AUDIO_STREAM_ENFORCED_AUDIBLE: 2081 mAttributes.flags |= AUDIO_FLAG_AUDIBILITY_ENFORCED; 2082 // intended fall through, attributes in common with STREAM_SYSTEM 2083 case AUDIO_STREAM_SYSTEM: 2084 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2085 mAttributes.usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION; 2086 break; 2087 case AUDIO_STREAM_RING: 2088 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2089 mAttributes.usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE; 2090 break; 2091 case AUDIO_STREAM_ALARM: 2092 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2093 mAttributes.usage = AUDIO_USAGE_ALARM; 2094 break; 2095 case AUDIO_STREAM_NOTIFICATION: 2096 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2097 mAttributes.usage = AUDIO_USAGE_NOTIFICATION; 2098 break; 2099 case AUDIO_STREAM_BLUETOOTH_SCO: 2100 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 2101 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION; 2102 mAttributes.flags |= AUDIO_FLAG_SCO; 2103 break; 2104 case AUDIO_STREAM_DTMF: 2105 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2106 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING; 2107 break; 2108 case AUDIO_STREAM_TTS: 2109 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 2110 mAttributes.usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY; 2111 break; 2112 default: 2113 ALOGE("invalid stream type %d when converting to attributes", streamType); 2114 } 2115} 2116 2117void AudioTrack::setStreamTypeFromAttributes(audio_attributes_t& aa) { 2118 // flags to stream type mapping 2119 if ((aa.flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { 2120 mStreamType = AUDIO_STREAM_ENFORCED_AUDIBLE; 2121 return; 2122 } 2123 if ((aa.flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) { 2124 mStreamType = AUDIO_STREAM_BLUETOOTH_SCO; 2125 return; 2126 } 2127 2128 // usage to stream type mapping 2129 switch (aa.usage) { 2130 case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: 2131 // TODO once AudioPolicyManager fully supports audio_attributes_t, 2132 // remove stream change based on phone state 2133 if (AudioSystem::getPhoneState() == AUDIO_MODE_RINGTONE) { 2134 mStreamType = AUDIO_STREAM_RING; 2135 break; 2136 } 2137 /// FALL THROUGH 2138 case AUDIO_USAGE_MEDIA: 2139 case AUDIO_USAGE_GAME: 2140 case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: 2141 mStreamType = AUDIO_STREAM_MUSIC; 2142 return; 2143 case AUDIO_USAGE_ASSISTANCE_SONIFICATION: 2144 mStreamType = AUDIO_STREAM_SYSTEM; 2145 return; 2146 case AUDIO_USAGE_VOICE_COMMUNICATION: 2147 mStreamType = AUDIO_STREAM_VOICE_CALL; 2148 return; 2149 2150 case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: 2151 mStreamType = AUDIO_STREAM_DTMF; 2152 return; 2153 2154 case AUDIO_USAGE_ALARM: 2155 mStreamType = AUDIO_STREAM_ALARM; 2156 return; 2157 case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: 2158 mStreamType = AUDIO_STREAM_RING; 2159 return; 2160 2161 case AUDIO_USAGE_NOTIFICATION: 2162 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: 2163 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: 2164 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: 2165 case AUDIO_USAGE_NOTIFICATION_EVENT: 2166 mStreamType = AUDIO_STREAM_NOTIFICATION; 2167 return; 2168 2169 case AUDIO_USAGE_UNKNOWN: 2170 default: 2171 mStreamType = AUDIO_STREAM_MUSIC; 2172 } 2173} 2174 2175bool AudioTrack::isValidAttributes(const audio_attributes_t *paa) { 2176 // has flags that map to a strategy? 2177 if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO)) != 0) { 2178 return true; 2179 } 2180 2181 // has known usage? 2182 switch (paa->usage) { 2183 case AUDIO_USAGE_UNKNOWN: 2184 case AUDIO_USAGE_MEDIA: 2185 case AUDIO_USAGE_VOICE_COMMUNICATION: 2186 case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: 2187 case AUDIO_USAGE_ALARM: 2188 case AUDIO_USAGE_NOTIFICATION: 2189 case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: 2190 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: 2191 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: 2192 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: 2193 case AUDIO_USAGE_NOTIFICATION_EVENT: 2194 case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: 2195 case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: 2196 case AUDIO_USAGE_ASSISTANCE_SONIFICATION: 2197 case AUDIO_USAGE_GAME: 2198 break; 2199 default: 2200 return false; 2201 } 2202 return true; 2203} 2204// ========================================================================= 2205 2206void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 2207{ 2208 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 2209 if (audioTrack != 0) { 2210 AutoMutex lock(audioTrack->mLock); 2211 audioTrack->mProxy->binderDied(); 2212 } 2213} 2214 2215// ========================================================================= 2216 2217AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 2218 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 2219 mIgnoreNextPausedInt(false) 2220{ 2221} 2222 2223AudioTrack::AudioTrackThread::~AudioTrackThread() 2224{ 2225} 2226 2227bool AudioTrack::AudioTrackThread::threadLoop() 2228{ 2229 { 2230 AutoMutex _l(mMyLock); 2231 if (mPaused) { 2232 mMyCond.wait(mMyLock); 2233 // caller will check for exitPending() 2234 return true; 2235 } 2236 if (mIgnoreNextPausedInt) { 2237 mIgnoreNextPausedInt = false; 2238 mPausedInt = false; 2239 } 2240 if (mPausedInt) { 2241 if (mPausedNs > 0) { 2242 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 2243 } else { 2244 mMyCond.wait(mMyLock); 2245 } 2246 mPausedInt = false; 2247 return true; 2248 } 2249 } 2250 if (exitPending()) { 2251 return false; 2252 } 2253 nsecs_t ns = mReceiver.processAudioBuffer(); 2254 switch (ns) { 2255 case 0: 2256 return true; 2257 case NS_INACTIVE: 2258 pauseInternal(); 2259 return true; 2260 case NS_NEVER: 2261 return false; 2262 case NS_WHENEVER: 2263 // FIXME increase poll interval, or make event-driven 2264 ns = 1000000000LL; 2265 // fall through 2266 default: 2267 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns); 2268 pauseInternal(ns); 2269 return true; 2270 } 2271} 2272 2273void AudioTrack::AudioTrackThread::requestExit() 2274{ 2275 // must be in this order to avoid a race condition 2276 Thread::requestExit(); 2277 resume(); 2278} 2279 2280void AudioTrack::AudioTrackThread::pause() 2281{ 2282 AutoMutex _l(mMyLock); 2283 mPaused = true; 2284} 2285 2286void AudioTrack::AudioTrackThread::resume() 2287{ 2288 AutoMutex _l(mMyLock); 2289 mIgnoreNextPausedInt = true; 2290 if (mPaused || mPausedInt) { 2291 mPaused = false; 2292 mPausedInt = false; 2293 mMyCond.signal(); 2294 } 2295} 2296 2297void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 2298{ 2299 AutoMutex _l(mMyLock); 2300 mPausedInt = true; 2301 mPausedNs = ns; 2302} 2303 2304}; // namespace android 2305