AudioTrack.cpp revision caf7f48a0ef558689d39aafd187c1571ff4128b4
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/AudioPolicyHelper.h> 32#include <media/AudioResamplerPublic.h> 33 34#define WAIT_PERIOD_MS 10 35#define WAIT_STREAM_END_TIMEOUT_SEC 120 36 37 38namespace android { 39// --------------------------------------------------------------------------- 40 41static int64_t convertTimespecToUs(const struct timespec &tv) 42{ 43 return tv.tv_sec * 1000000ll + tv.tv_nsec / 1000; 44} 45 46// current monotonic time in microseconds. 47static int64_t getNowUs() 48{ 49 struct timespec tv; 50 (void) clock_gettime(CLOCK_MONOTONIC, &tv); 51 return convertTimespecToUs(tv); 52} 53 54// static 55status_t AudioTrack::getMinFrameCount( 56 size_t* frameCount, 57 audio_stream_type_t streamType, 58 uint32_t sampleRate) 59{ 60 if (frameCount == NULL) { 61 return BAD_VALUE; 62 } 63 64 // FIXME merge with similar code in createTrack_l(), except we're missing 65 // some information here that is available in createTrack_l(): 66 // audio_io_handle_t output 67 // audio_format_t format 68 // audio_channel_mask_t channelMask 69 // audio_output_flags_t flags 70 uint32_t afSampleRate; 71 status_t status; 72 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 73 if (status != NO_ERROR) { 74 ALOGE("Unable to query output sample rate for stream type %d; status %d", 75 streamType, status); 76 return status; 77 } 78 size_t afFrameCount; 79 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 80 if (status != NO_ERROR) { 81 ALOGE("Unable to query output frame count for stream type %d; status %d", 82 streamType, status); 83 return status; 84 } 85 uint32_t afLatency; 86 status = AudioSystem::getOutputLatency(&afLatency, streamType); 87 if (status != NO_ERROR) { 88 ALOGE("Unable to query output latency for stream type %d; status %d", 89 streamType, status); 90 return status; 91 } 92 93 // Ensure that buffer depth covers at least audio hardware latency 94 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 95 if (minBufCount < 2) { 96 minBufCount = 2; 97 } 98 99 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 100 afFrameCount * minBufCount * uint64_t(sampleRate) / afSampleRate; 101 // The formula above should always produce a non-zero value, but return an error 102 // in the unlikely event that it does not, as that's part of the API contract. 103 if (*frameCount == 0) { 104 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", 105 streamType, sampleRate); 106 return BAD_VALUE; 107 } 108 ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%d, afSampleRate=%d, afLatency=%d", 109 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 110 return NO_ERROR; 111} 112 113// --------------------------------------------------------------------------- 114 115AudioTrack::AudioTrack() 116 : mStatus(NO_INIT), 117 mIsTimed(false), 118 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 119 mPreviousSchedulingGroup(SP_DEFAULT), 120 mPausedPosition(0) 121{ 122 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN; 123 mAttributes.usage = AUDIO_USAGE_UNKNOWN; 124 mAttributes.flags = 0x0; 125 strcpy(mAttributes.tags, ""); 126} 127 128AudioTrack::AudioTrack( 129 audio_stream_type_t streamType, 130 uint32_t sampleRate, 131 audio_format_t format, 132 audio_channel_mask_t channelMask, 133 size_t frameCount, 134 audio_output_flags_t flags, 135 callback_t cbf, 136 void* user, 137 uint32_t notificationFrames, 138 int sessionId, 139 transfer_type transferType, 140 const audio_offload_info_t *offloadInfo, 141 int uid, 142 pid_t pid, 143 const audio_attributes_t* pAttributes) 144 : mStatus(NO_INIT), 145 mIsTimed(false), 146 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 147 mPreviousSchedulingGroup(SP_DEFAULT), 148 mPausedPosition(0) 149{ 150 mStatus = set(streamType, sampleRate, format, channelMask, 151 frameCount, flags, cbf, user, notificationFrames, 152 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 153 offloadInfo, uid, pid, pAttributes); 154} 155 156AudioTrack::AudioTrack( 157 audio_stream_type_t streamType, 158 uint32_t sampleRate, 159 audio_format_t format, 160 audio_channel_mask_t channelMask, 161 const sp<IMemory>& sharedBuffer, 162 audio_output_flags_t flags, 163 callback_t cbf, 164 void* user, 165 uint32_t notificationFrames, 166 int sessionId, 167 transfer_type transferType, 168 const audio_offload_info_t *offloadInfo, 169 int uid, 170 pid_t pid, 171 const audio_attributes_t* pAttributes) 172 : mStatus(NO_INIT), 173 mIsTimed(false), 174 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 175 mPreviousSchedulingGroup(SP_DEFAULT), 176 mPausedPosition(0) 177{ 178 mStatus = set(streamType, sampleRate, format, channelMask, 179 0 /*frameCount*/, flags, cbf, user, notificationFrames, 180 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, 181 uid, pid, pAttributes); 182} 183 184AudioTrack::~AudioTrack() 185{ 186 if (mStatus == NO_ERROR) { 187 // Make sure that callback function exits in the case where 188 // it is looping on buffer full condition in obtainBuffer(). 189 // Otherwise the callback thread will never exit. 190 stop(); 191 if (mAudioTrackThread != 0) { 192 mProxy->interrupt(); 193 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 194 mAudioTrackThread->requestExitAndWait(); 195 mAudioTrackThread.clear(); 196 } 197 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 198 mAudioTrack.clear(); 199 mCblkMemory.clear(); 200 mSharedBuffer.clear(); 201 IPCThreadState::self()->flushCommands(); 202 ALOGV("~AudioTrack, releasing session id from %d on behalf of %d", 203 IPCThreadState::self()->getCallingPid(), mClientPid); 204 AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); 205 } 206} 207 208status_t AudioTrack::set( 209 audio_stream_type_t streamType, 210 uint32_t sampleRate, 211 audio_format_t format, 212 audio_channel_mask_t channelMask, 213 size_t frameCount, 214 audio_output_flags_t flags, 215 callback_t cbf, 216 void* user, 217 uint32_t notificationFrames, 218 const sp<IMemory>& sharedBuffer, 219 bool threadCanCallJava, 220 int sessionId, 221 transfer_type transferType, 222 const audio_offload_info_t *offloadInfo, 223 int uid, 224 pid_t pid, 225 const audio_attributes_t* pAttributes) 226{ 227 ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 228 "flags #%x, notificationFrames %u, sessionId %d, transferType %d", 229 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, 230 sessionId, transferType); 231 232 switch (transferType) { 233 case TRANSFER_DEFAULT: 234 if (sharedBuffer != 0) { 235 transferType = TRANSFER_SHARED; 236 } else if (cbf == NULL || threadCanCallJava) { 237 transferType = TRANSFER_SYNC; 238 } else { 239 transferType = TRANSFER_CALLBACK; 240 } 241 break; 242 case TRANSFER_CALLBACK: 243 if (cbf == NULL || sharedBuffer != 0) { 244 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 245 return BAD_VALUE; 246 } 247 break; 248 case TRANSFER_OBTAIN: 249 case TRANSFER_SYNC: 250 if (sharedBuffer != 0) { 251 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 252 return BAD_VALUE; 253 } 254 break; 255 case TRANSFER_SHARED: 256 if (sharedBuffer == 0) { 257 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 258 return BAD_VALUE; 259 } 260 break; 261 default: 262 ALOGE("Invalid transfer type %d", transferType); 263 return BAD_VALUE; 264 } 265 mSharedBuffer = sharedBuffer; 266 mTransfer = transferType; 267 268 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 269 sharedBuffer->size()); 270 271 ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags); 272 273 AutoMutex lock(mLock); 274 275 // invariant that mAudioTrack != 0 is true only after set() returns successfully 276 if (mAudioTrack != 0) { 277 ALOGE("Track already in use"); 278 return INVALID_OPERATION; 279 } 280 281 // handle default values first. 282 if (streamType == AUDIO_STREAM_DEFAULT) { 283 streamType = AUDIO_STREAM_MUSIC; 284 } 285 if (pAttributes == NULL) { 286 if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) { 287 ALOGE("Invalid stream type %d", streamType); 288 return BAD_VALUE; 289 } 290 mStreamType = streamType; 291 292 } else { 293 // stream type shouldn't be looked at, this track has audio attributes 294 memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t)); 295 ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]", 296 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags); 297 mStreamType = AUDIO_STREAM_DEFAULT; 298 } 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 } 337 338 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { 339 if (audio_is_linear_pcm(format)) { 340 mFrameSize = channelCount * audio_bytes_per_sample(format); 341 } else { 342 mFrameSize = sizeof(uint8_t); 343 } 344 mFrameSizeAF = mFrameSize; 345 } else { 346 ALOG_ASSERT(audio_is_linear_pcm(format)); 347 mFrameSize = channelCount * audio_bytes_per_sample(format); 348 mFrameSizeAF = channelCount * audio_bytes_per_sample( 349 format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format); 350 // createTrack will return an error if PCM format is not supported by server, 351 // so no need to check for specific PCM formats here 352 } 353 354 // sampling rate must be specified for direct outputs 355 if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) { 356 return BAD_VALUE; 357 } 358 mSampleRate = sampleRate; 359 360 // Make copy of input parameter offloadInfo so that in the future: 361 // (a) createTrack_l doesn't need it as an input parameter 362 // (b) we can support re-creation of offloaded tracks 363 if (offloadInfo != NULL) { 364 mOffloadInfoCopy = *offloadInfo; 365 mOffloadInfo = &mOffloadInfoCopy; 366 } else { 367 mOffloadInfo = NULL; 368 } 369 370 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f; 371 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f; 372 mSendLevel = 0.0f; 373 // mFrameCount is initialized in createTrack_l 374 mReqFrameCount = frameCount; 375 mNotificationFramesReq = notificationFrames; 376 mNotificationFramesAct = 0; 377 if (sessionId == AUDIO_SESSION_ALLOCATE) { 378 mSessionId = AudioSystem::newAudioUniqueId(); 379 } else { 380 mSessionId = sessionId; 381 } 382 int callingpid = IPCThreadState::self()->getCallingPid(); 383 int mypid = getpid(); 384 if (uid == -1 || (callingpid != mypid)) { 385 mClientUid = IPCThreadState::self()->getCallingUid(); 386 } else { 387 mClientUid = uid; 388 } 389 if (pid == -1 || (callingpid != mypid)) { 390 mClientPid = callingpid; 391 } else { 392 mClientPid = pid; 393 } 394 mAuxEffectId = 0; 395 mFlags = flags; 396 mCbf = cbf; 397 398 if (cbf != NULL) { 399 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 400 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 401 } 402 403 // create the IAudioTrack 404 status_t status = createTrack_l(); 405 406 if (status != NO_ERROR) { 407 if (mAudioTrackThread != 0) { 408 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 409 mAudioTrackThread->requestExitAndWait(); 410 mAudioTrackThread.clear(); 411 } 412 return status; 413 } 414 415 mStatus = NO_ERROR; 416 mState = STATE_STOPPED; 417 mUserData = user; 418 mLoopPeriod = 0; 419 mMarkerPosition = 0; 420 mMarkerReached = false; 421 mNewPosition = 0; 422 mUpdatePeriod = 0; 423 mServer = 0; 424 mPosition = 0; 425 mReleased = 0; 426 mStartUs = 0; 427 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 428 mSequence = 1; 429 mObservedSequence = mSequence; 430 mInUnderrun = false; 431 432 return NO_ERROR; 433} 434 435// ------------------------------------------------------------------------- 436 437status_t AudioTrack::start() 438{ 439 AutoMutex lock(mLock); 440 441 if (mState == STATE_ACTIVE) { 442 return INVALID_OPERATION; 443 } 444 445 mInUnderrun = true; 446 447 State previousState = mState; 448 if (previousState == STATE_PAUSED_STOPPING) { 449 mState = STATE_STOPPING; 450 } else { 451 mState = STATE_ACTIVE; 452 } 453 (void) updateAndGetPosition_l(); 454 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 455 // reset current position as seen by client to 0 456 mPosition = 0; 457 // For offloaded tracks, we don't know if the hardware counters are really zero here, 458 // since the flush is asynchronous and stop may not fully drain. 459 // We save the time when the track is started to later verify whether 460 // the counters are realistic (i.e. start from zero after this time). 461 mStartUs = getNowUs(); 462 463 // force refresh of remaining frames by processAudioBuffer() as last 464 // write before stop could be partial. 465 mRefreshRemaining = true; 466 } 467 mNewPosition = mPosition + mUpdatePeriod; 468 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 469 470 sp<AudioTrackThread> t = mAudioTrackThread; 471 if (t != 0) { 472 if (previousState == STATE_STOPPING) { 473 mProxy->interrupt(); 474 } else { 475 t->resume(); 476 } 477 } else { 478 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 479 get_sched_policy(0, &mPreviousSchedulingGroup); 480 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 481 } 482 483 status_t status = NO_ERROR; 484 if (!(flags & CBLK_INVALID)) { 485 status = mAudioTrack->start(); 486 if (status == DEAD_OBJECT) { 487 flags |= CBLK_INVALID; 488 } 489 } 490 if (flags & CBLK_INVALID) { 491 status = restoreTrack_l("start"); 492 } 493 494 if (status != NO_ERROR) { 495 ALOGE("start() status %d", status); 496 mState = previousState; 497 if (t != 0) { 498 if (previousState != STATE_STOPPING) { 499 t->pause(); 500 } 501 } else { 502 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 503 set_sched_policy(0, mPreviousSchedulingGroup); 504 } 505 } 506 507 return status; 508} 509 510void AudioTrack::stop() 511{ 512 AutoMutex lock(mLock); 513 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 514 return; 515 } 516 517 if (isOffloaded_l()) { 518 mState = STATE_STOPPING; 519 } else { 520 mState = STATE_STOPPED; 521 mReleased = 0; 522 } 523 524 mProxy->interrupt(); 525 mAudioTrack->stop(); 526 // the playback head position will reset to 0, so if a marker is set, we need 527 // to activate it again 528 mMarkerReached = false; 529#if 0 530 // Force flush if a shared buffer is used otherwise audioflinger 531 // will not stop before end of buffer is reached. 532 // It may be needed to make sure that we stop playback, likely in case looping is on. 533 if (mSharedBuffer != 0) { 534 flush_l(); 535 } 536#endif 537 538 sp<AudioTrackThread> t = mAudioTrackThread; 539 if (t != 0) { 540 if (!isOffloaded_l()) { 541 t->pause(); 542 } 543 } else { 544 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 545 set_sched_policy(0, mPreviousSchedulingGroup); 546 } 547} 548 549bool AudioTrack::stopped() const 550{ 551 AutoMutex lock(mLock); 552 return mState != STATE_ACTIVE; 553} 554 555void AudioTrack::flush() 556{ 557 if (mSharedBuffer != 0) { 558 return; 559 } 560 AutoMutex lock(mLock); 561 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 562 return; 563 } 564 flush_l(); 565} 566 567void AudioTrack::flush_l() 568{ 569 ALOG_ASSERT(mState != STATE_ACTIVE); 570 571 // clear playback marker and periodic update counter 572 mMarkerPosition = 0; 573 mMarkerReached = false; 574 mUpdatePeriod = 0; 575 mRefreshRemaining = true; 576 577 mState = STATE_FLUSHED; 578 mReleased = 0; 579 if (isOffloaded_l()) { 580 mProxy->interrupt(); 581 } 582 mProxy->flush(); 583 mAudioTrack->flush(); 584} 585 586void AudioTrack::pause() 587{ 588 AutoMutex lock(mLock); 589 if (mState == STATE_ACTIVE) { 590 mState = STATE_PAUSED; 591 } else if (mState == STATE_STOPPING) { 592 mState = STATE_PAUSED_STOPPING; 593 } else { 594 return; 595 } 596 mProxy->interrupt(); 597 mAudioTrack->pause(); 598 599 if (isOffloaded_l()) { 600 if (mOutput != AUDIO_IO_HANDLE_NONE) { 601 // An offload output can be re-used between two audio tracks having 602 // the same configuration. A timestamp query for a paused track 603 // while the other is running would return an incorrect time. 604 // To fix this, cache the playback position on a pause() and return 605 // this time when requested until the track is resumed. 606 607 // OffloadThread sends HAL pause in its threadLoop. Time saved 608 // here can be slightly off. 609 610 // TODO: check return code for getRenderPosition. 611 612 uint32_t halFrames; 613 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); 614 ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); 615 } 616 } 617} 618 619status_t AudioTrack::setVolume(float left, float right) 620{ 621 // This duplicates a test by AudioTrack JNI, but that is not the only caller 622 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY || 623 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) { 624 return BAD_VALUE; 625 } 626 627 AutoMutex lock(mLock); 628 mVolume[AUDIO_INTERLEAVE_LEFT] = left; 629 mVolume[AUDIO_INTERLEAVE_RIGHT] = right; 630 631 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right))); 632 633 if (isOffloaded_l()) { 634 mAudioTrack->signal(); 635 } 636 return NO_ERROR; 637} 638 639status_t AudioTrack::setVolume(float volume) 640{ 641 return setVolume(volume, volume); 642} 643 644status_t AudioTrack::setAuxEffectSendLevel(float level) 645{ 646 // This duplicates a test by AudioTrack JNI, but that is not the only caller 647 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) { 648 return BAD_VALUE; 649 } 650 651 AutoMutex lock(mLock); 652 mSendLevel = level; 653 mProxy->setSendLevel(level); 654 655 return NO_ERROR; 656} 657 658void AudioTrack::getAuxEffectSendLevel(float* level) const 659{ 660 if (level != NULL) { 661 *level = mSendLevel; 662 } 663} 664 665status_t AudioTrack::setSampleRate(uint32_t rate) 666{ 667 if (mIsTimed || isOffloadedOrDirect()) { 668 return INVALID_OPERATION; 669 } 670 671 AutoMutex lock(mLock); 672 if (mOutput == AUDIO_IO_HANDLE_NONE) { 673 return NO_INIT; 674 } 675 uint32_t afSamplingRate; 676 if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) { 677 return NO_INIT; 678 } 679 if (rate == 0 || rate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 680 return BAD_VALUE; 681 } 682 683 mSampleRate = rate; 684 mProxy->setSampleRate(rate); 685 686 return NO_ERROR; 687} 688 689uint32_t AudioTrack::getSampleRate() const 690{ 691 if (mIsTimed) { 692 return 0; 693 } 694 695 AutoMutex lock(mLock); 696 697 // sample rate can be updated during playback by the offloaded decoder so we need to 698 // query the HAL and update if needed. 699// FIXME use Proxy return channel to update the rate from server and avoid polling here 700 if (isOffloadedOrDirect_l()) { 701 if (mOutput != AUDIO_IO_HANDLE_NONE) { 702 uint32_t sampleRate = 0; 703 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate); 704 if (status == NO_ERROR) { 705 mSampleRate = sampleRate; 706 } 707 } 708 } 709 return mSampleRate; 710} 711 712status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 713{ 714 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 715 return INVALID_OPERATION; 716 } 717 718 if (loopCount == 0) { 719 ; 720 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 721 loopEnd - loopStart >= MIN_LOOP) { 722 ; 723 } else { 724 return BAD_VALUE; 725 } 726 727 AutoMutex lock(mLock); 728 // See setPosition() regarding setting parameters such as loop points or position while active 729 if (mState == STATE_ACTIVE) { 730 return INVALID_OPERATION; 731 } 732 setLoop_l(loopStart, loopEnd, loopCount); 733 return NO_ERROR; 734} 735 736void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 737{ 738 // Setting the loop will reset next notification update period (like setPosition). 739 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 740 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 741 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 742} 743 744status_t AudioTrack::setMarkerPosition(uint32_t marker) 745{ 746 // The only purpose of setting marker position is to get a callback 747 if (mCbf == NULL || isOffloadedOrDirect()) { 748 return INVALID_OPERATION; 749 } 750 751 AutoMutex lock(mLock); 752 mMarkerPosition = marker; 753 mMarkerReached = false; 754 755 return NO_ERROR; 756} 757 758status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 759{ 760 if (isOffloadedOrDirect()) { 761 return INVALID_OPERATION; 762 } 763 if (marker == NULL) { 764 return BAD_VALUE; 765 } 766 767 AutoMutex lock(mLock); 768 *marker = mMarkerPosition; 769 770 return NO_ERROR; 771} 772 773status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 774{ 775 // The only purpose of setting position update period is to get a callback 776 if (mCbf == NULL || isOffloadedOrDirect()) { 777 return INVALID_OPERATION; 778 } 779 780 AutoMutex lock(mLock); 781 mNewPosition = updateAndGetPosition_l() + updatePeriod; 782 mUpdatePeriod = updatePeriod; 783 784 return NO_ERROR; 785} 786 787status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 788{ 789 if (isOffloadedOrDirect()) { 790 return INVALID_OPERATION; 791 } 792 if (updatePeriod == NULL) { 793 return BAD_VALUE; 794 } 795 796 AutoMutex lock(mLock); 797 *updatePeriod = mUpdatePeriod; 798 799 return NO_ERROR; 800} 801 802status_t AudioTrack::setPosition(uint32_t position) 803{ 804 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 805 return INVALID_OPERATION; 806 } 807 if (position > mFrameCount) { 808 return BAD_VALUE; 809 } 810 811 AutoMutex lock(mLock); 812 // Currently we require that the player is inactive before setting parameters such as position 813 // or loop points. Otherwise, there could be a race condition: the application could read the 814 // current position, compute a new position or loop parameters, and then set that position or 815 // loop parameters but it would do the "wrong" thing since the position has continued to advance 816 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 817 // to specify how it wants to handle such scenarios. 818 if (mState == STATE_ACTIVE) { 819 return INVALID_OPERATION; 820 } 821 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 822 mLoopPeriod = 0; 823 // FIXME Check whether loops and setting position are incompatible in old code. 824 // If we use setLoop for both purposes we lose the capability to set the position while looping. 825 mStaticProxy->setLoop(position, mFrameCount, 0); 826 827 return NO_ERROR; 828} 829 830status_t AudioTrack::getPosition(uint32_t *position) 831{ 832 if (position == NULL) { 833 return BAD_VALUE; 834 } 835 836 AutoMutex lock(mLock); 837 if (isOffloadedOrDirect_l()) { 838 uint32_t dspFrames = 0; 839 840 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) { 841 ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); 842 *position = mPausedPosition; 843 return NO_ERROR; 844 } 845 846 if (mOutput != AUDIO_IO_HANDLE_NONE) { 847 uint32_t halFrames; 848 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 849 } 850 // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED) 851 // due to hardware latency. We leave this behavior for now. 852 *position = dspFrames; 853 } else { 854 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 855 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 856 0 : updateAndGetPosition_l(); 857 } 858 return NO_ERROR; 859} 860 861status_t AudioTrack::getBufferPosition(uint32_t *position) 862{ 863 if (mSharedBuffer == 0 || mIsTimed) { 864 return INVALID_OPERATION; 865 } 866 if (position == NULL) { 867 return BAD_VALUE; 868 } 869 870 AutoMutex lock(mLock); 871 *position = mStaticProxy->getBufferPosition(); 872 return NO_ERROR; 873} 874 875status_t AudioTrack::reload() 876{ 877 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 878 return INVALID_OPERATION; 879 } 880 881 AutoMutex lock(mLock); 882 // See setPosition() regarding setting parameters such as loop points or position while active 883 if (mState == STATE_ACTIVE) { 884 return INVALID_OPERATION; 885 } 886 mNewPosition = mUpdatePeriod; 887 mLoopPeriod = 0; 888 // FIXME The new code cannot reload while keeping a loop specified. 889 // Need to check how the old code handled this, and whether it's a significant change. 890 mStaticProxy->setLoop(0, mFrameCount, 0); 891 return NO_ERROR; 892} 893 894audio_io_handle_t AudioTrack::getOutput() const 895{ 896 AutoMutex lock(mLock); 897 return mOutput; 898} 899 900status_t AudioTrack::attachAuxEffect(int effectId) 901{ 902 AutoMutex lock(mLock); 903 status_t status = mAudioTrack->attachAuxEffect(effectId); 904 if (status == NO_ERROR) { 905 mAuxEffectId = effectId; 906 } 907 return status; 908} 909 910audio_stream_type_t AudioTrack::streamType() const 911{ 912 if (mStreamType == AUDIO_STREAM_DEFAULT) { 913 return audio_attributes_to_stream_type(&mAttributes); 914 } 915 return mStreamType; 916} 917 918// ------------------------------------------------------------------------- 919 920// must be called with mLock held 921status_t AudioTrack::createTrack_l() 922{ 923 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 924 if (audioFlinger == 0) { 925 ALOGE("Could not get audioflinger"); 926 return NO_INIT; 927 } 928 929 audio_io_handle_t output; 930 audio_stream_type_t streamType = mStreamType; 931 audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL; 932 status_t status = AudioSystem::getOutputForAttr(attr, &output, 933 (audio_session_t)mSessionId, &streamType, 934 mSampleRate, mFormat, mChannelMask, 935 mFlags, mOffloadInfo); 936 937 938 if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) { 939 ALOGE("Could not get audio output for stream type %d, usage %d, sample rate %u, format %#x," 940 " channel mask %#x, flags %#x", 941 streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags); 942 return BAD_VALUE; 943 } 944 { 945 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 946 // we must release it ourselves if anything goes wrong. 947 948 // Not all of these values are needed under all conditions, but it is easier to get them all 949 950 uint32_t afLatency; 951 status = AudioSystem::getLatency(output, &afLatency); 952 if (status != NO_ERROR) { 953 ALOGE("getLatency(%d) failed status %d", output, status); 954 goto release; 955 } 956 957 size_t afFrameCount; 958 status = AudioSystem::getFrameCount(output, &afFrameCount); 959 if (status != NO_ERROR) { 960 ALOGE("getFrameCount(output=%d) status %d", output, status); 961 goto release; 962 } 963 964 uint32_t afSampleRate; 965 status = AudioSystem::getSamplingRate(output, &afSampleRate); 966 if (status != NO_ERROR) { 967 ALOGE("getSamplingRate(output=%d) status %d", output, status); 968 goto release; 969 } 970 if (mSampleRate == 0) { 971 mSampleRate = afSampleRate; 972 } 973 // Client decides whether the track is TIMED (see below), but can only express a preference 974 // for FAST. Server will perform additional tests. 975 if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( 976 // either of these use cases: 977 // use case 1: shared buffer 978 (mSharedBuffer != 0) || 979 // use case 2: callback transfer mode 980 (mTransfer == TRANSFER_CALLBACK)) && 981 // matching sample rate 982 (mSampleRate == afSampleRate))) { 983 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 984 // once denied, do not request again if IAudioTrack is re-created 985 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 986 } 987 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 988 989 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 990 // n = 1 fast track with single buffering; nBuffering is ignored 991 // n = 2 fast track with double buffering 992 // n = 2 normal track, no sample rate conversion 993 // n = 3 normal track, with sample rate conversion 994 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 995 // n > 3 very high latency or very small notification interval; nBuffering is ignored 996 const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; 997 998 mNotificationFramesAct = mNotificationFramesReq; 999 1000 size_t frameCount = mReqFrameCount; 1001 if (!audio_is_linear_pcm(mFormat)) { 1002 1003 if (mSharedBuffer != 0) { 1004 // Same comment as below about ignoring frameCount parameter for set() 1005 frameCount = mSharedBuffer->size(); 1006 } else if (frameCount == 0) { 1007 frameCount = afFrameCount; 1008 } 1009 if (mNotificationFramesAct != frameCount) { 1010 mNotificationFramesAct = frameCount; 1011 } 1012 } else if (mSharedBuffer != 0) { 1013 1014 // Ensure that buffer alignment matches channel count 1015 // 8-bit data in shared memory is not currently supported by AudioFlinger 1016 size_t alignment = audio_bytes_per_sample( 1017 mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat); 1018 if (alignment & 1) { 1019 alignment = 1; 1020 } 1021 if (mChannelCount > 1) { 1022 // More than 2 channels does not require stronger alignment than stereo 1023 alignment <<= 1; 1024 } 1025 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 1026 ALOGE("Invalid buffer alignment: address %p, channel count %u", 1027 mSharedBuffer->pointer(), mChannelCount); 1028 status = BAD_VALUE; 1029 goto release; 1030 } 1031 1032 // When initializing a shared buffer AudioTrack via constructors, 1033 // there's no frameCount parameter. 1034 // But when initializing a shared buffer AudioTrack via set(), 1035 // there _is_ a frameCount parameter. We silently ignore it. 1036 frameCount = mSharedBuffer->size() / mFrameSizeAF; 1037 1038 } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 1039 1040 // FIXME move these calculations and associated checks to server 1041 1042 // Ensure that buffer depth covers at least audio hardware latency 1043 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 1044 ALOGV("afFrameCount=%zu, minBufCount=%d, afSampleRate=%u, afLatency=%d", 1045 afFrameCount, minBufCount, afSampleRate, afLatency); 1046 if (minBufCount <= nBuffering) { 1047 minBufCount = nBuffering; 1048 } 1049 1050 size_t minFrameCount = afFrameCount * minBufCount * uint64_t(mSampleRate) / afSampleRate; 1051 ALOGV("minFrameCount: %zu, afFrameCount=%zu, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 1052 ", afLatency=%d", 1053 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); 1054 1055 if (frameCount == 0) { 1056 frameCount = minFrameCount; 1057 } else if (frameCount < minFrameCount) { 1058 // not ALOGW because it happens all the time when playing key clicks over A2DP 1059 ALOGV("Minimum buffer size corrected from %zu to %zu", 1060 frameCount, minFrameCount); 1061 frameCount = minFrameCount; 1062 } 1063 // Make sure that application is notified with sufficient margin before underrun 1064 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1065 mNotificationFramesAct = frameCount/nBuffering; 1066 } 1067 1068 } else { 1069 // For fast tracks, the frame count calculations and checks are done by server 1070 } 1071 1072 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 1073 if (mIsTimed) { 1074 trackFlags |= IAudioFlinger::TRACK_TIMED; 1075 } 1076 1077 pid_t tid = -1; 1078 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1079 trackFlags |= IAudioFlinger::TRACK_FAST; 1080 if (mAudioTrackThread != 0) { 1081 tid = mAudioTrackThread->getTid(); 1082 } 1083 } 1084 1085 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1086 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 1087 } 1088 1089 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1090 trackFlags |= IAudioFlinger::TRACK_DIRECT; 1091 } 1092 1093 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1094 // but we will still need the original value also 1095 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 1096 mSampleRate, 1097 // AudioFlinger only sees 16-bit PCM 1098 mFormat == AUDIO_FORMAT_PCM_8_BIT && 1099 !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT) ? 1100 AUDIO_FORMAT_PCM_16_BIT : mFormat, 1101 mChannelMask, 1102 &temp, 1103 &trackFlags, 1104 mSharedBuffer, 1105 output, 1106 tid, 1107 &mSessionId, 1108 mClientUid, 1109 &status); 1110 1111 if (status != NO_ERROR) { 1112 ALOGE("AudioFlinger could not create track, status: %d", status); 1113 goto release; 1114 } 1115 ALOG_ASSERT(track != 0); 1116 1117 // AudioFlinger now owns the reference to the I/O handle, 1118 // so we are no longer responsible for releasing it. 1119 1120 sp<IMemory> iMem = track->getCblk(); 1121 if (iMem == 0) { 1122 ALOGE("Could not get control block"); 1123 return NO_INIT; 1124 } 1125 void *iMemPointer = iMem->pointer(); 1126 if (iMemPointer == NULL) { 1127 ALOGE("Could not get control block pointer"); 1128 return NO_INIT; 1129 } 1130 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1131 if (mAudioTrack != 0) { 1132 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1133 mDeathNotifier.clear(); 1134 } 1135 mAudioTrack = track; 1136 mCblkMemory = iMem; 1137 IPCThreadState::self()->flushCommands(); 1138 1139 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1140 mCblk = cblk; 1141 // note that temp is the (possibly revised) value of frameCount 1142 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1143 // In current design, AudioTrack client checks and ensures frame count validity before 1144 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1145 // for fast track as it uses a special method of assigning frame count. 1146 ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp); 1147 } 1148 frameCount = temp; 1149 1150 mAwaitBoost = false; 1151 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1152 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1153 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount); 1154 mAwaitBoost = true; 1155 if (mSharedBuffer == 0) { 1156 // Theoretically double-buffering is not required for fast tracks, 1157 // due to tighter scheduling. But in practice, to accommodate kernels with 1158 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1159 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1160 mNotificationFramesAct = frameCount/nBuffering; 1161 } 1162 } 1163 } else { 1164 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount); 1165 // once denied, do not request again if IAudioTrack is re-created 1166 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1167 if (mSharedBuffer == 0) { 1168 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1169 mNotificationFramesAct = frameCount/nBuffering; 1170 } 1171 } 1172 } 1173 } 1174 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1175 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1176 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1177 } else { 1178 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1179 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1180 // FIXME This is a warning, not an error, so don't return error status 1181 //return NO_INIT; 1182 } 1183 } 1184 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1185 if (trackFlags & IAudioFlinger::TRACK_DIRECT) { 1186 ALOGV("AUDIO_OUTPUT_FLAG_DIRECT successful"); 1187 } else { 1188 ALOGW("AUDIO_OUTPUT_FLAG_DIRECT denied by server"); 1189 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT); 1190 // FIXME This is a warning, not an error, so don't return error status 1191 //return NO_INIT; 1192 } 1193 } 1194 1195 // We retain a copy of the I/O handle, but don't own the reference 1196 mOutput = output; 1197 mRefreshRemaining = true; 1198 1199 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1200 // is the value of pointer() for the shared buffer, otherwise buffers points 1201 // immediately after the control block. This address is for the mapping within client 1202 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1203 void* buffers; 1204 if (mSharedBuffer == 0) { 1205 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1206 } else { 1207 buffers = mSharedBuffer->pointer(); 1208 } 1209 1210 mAudioTrack->attachAuxEffect(mAuxEffectId); 1211 // FIXME don't believe this lie 1212 mLatency = afLatency + (1000*frameCount) / mSampleRate; 1213 1214 mFrameCount = frameCount; 1215 // If IAudioTrack is re-created, don't let the requested frameCount 1216 // decrease. This can confuse clients that cache frameCount(). 1217 if (frameCount > mReqFrameCount) { 1218 mReqFrameCount = frameCount; 1219 } 1220 1221 // update proxy 1222 if (mSharedBuffer == 0) { 1223 mStaticProxy.clear(); 1224 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1225 } else { 1226 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1227 mProxy = mStaticProxy; 1228 } 1229 mProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); 1230 mProxy->setSendLevel(mSendLevel); 1231 mProxy->setSampleRate(mSampleRate); 1232 mProxy->setMinimum(mNotificationFramesAct); 1233 1234 mDeathNotifier = new DeathNotifier(this); 1235 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1236 1237 return NO_ERROR; 1238 } 1239 1240release: 1241 AudioSystem::releaseOutput(output, streamType, (audio_session_t)mSessionId); 1242 if (status == NO_ERROR) { 1243 status = NO_INIT; 1244 } 1245 return status; 1246} 1247 1248status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1249{ 1250 if (audioBuffer == NULL) { 1251 return BAD_VALUE; 1252 } 1253 if (mTransfer != TRANSFER_OBTAIN) { 1254 audioBuffer->frameCount = 0; 1255 audioBuffer->size = 0; 1256 audioBuffer->raw = NULL; 1257 return INVALID_OPERATION; 1258 } 1259 1260 const struct timespec *requested; 1261 struct timespec timeout; 1262 if (waitCount == -1) { 1263 requested = &ClientProxy::kForever; 1264 } else if (waitCount == 0) { 1265 requested = &ClientProxy::kNonBlocking; 1266 } else if (waitCount > 0) { 1267 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1268 timeout.tv_sec = ms / 1000; 1269 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1270 requested = &timeout; 1271 } else { 1272 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1273 requested = NULL; 1274 } 1275 return obtainBuffer(audioBuffer, requested); 1276} 1277 1278status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1279 struct timespec *elapsed, size_t *nonContig) 1280{ 1281 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1282 uint32_t oldSequence = 0; 1283 uint32_t newSequence; 1284 1285 Proxy::Buffer buffer; 1286 status_t status = NO_ERROR; 1287 1288 static const int32_t kMaxTries = 5; 1289 int32_t tryCounter = kMaxTries; 1290 1291 do { 1292 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1293 // keep them from going away if another thread re-creates the track during obtainBuffer() 1294 sp<AudioTrackClientProxy> proxy; 1295 sp<IMemory> iMem; 1296 1297 { // start of lock scope 1298 AutoMutex lock(mLock); 1299 1300 newSequence = mSequence; 1301 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1302 if (status == DEAD_OBJECT) { 1303 // re-create track, unless someone else has already done so 1304 if (newSequence == oldSequence) { 1305 status = restoreTrack_l("obtainBuffer"); 1306 if (status != NO_ERROR) { 1307 buffer.mFrameCount = 0; 1308 buffer.mRaw = NULL; 1309 buffer.mNonContig = 0; 1310 break; 1311 } 1312 } 1313 } 1314 oldSequence = newSequence; 1315 1316 // Keep the extra references 1317 proxy = mProxy; 1318 iMem = mCblkMemory; 1319 1320 if (mState == STATE_STOPPING) { 1321 status = -EINTR; 1322 buffer.mFrameCount = 0; 1323 buffer.mRaw = NULL; 1324 buffer.mNonContig = 0; 1325 break; 1326 } 1327 1328 // Non-blocking if track is stopped or paused 1329 if (mState != STATE_ACTIVE) { 1330 requested = &ClientProxy::kNonBlocking; 1331 } 1332 1333 } // end of lock scope 1334 1335 buffer.mFrameCount = audioBuffer->frameCount; 1336 // FIXME starts the requested timeout and elapsed over from scratch 1337 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1338 1339 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1340 1341 audioBuffer->frameCount = buffer.mFrameCount; 1342 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1343 audioBuffer->raw = buffer.mRaw; 1344 if (nonContig != NULL) { 1345 *nonContig = buffer.mNonContig; 1346 } 1347 return status; 1348} 1349 1350void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1351{ 1352 if (mTransfer == TRANSFER_SHARED) { 1353 return; 1354 } 1355 1356 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1357 if (stepCount == 0) { 1358 return; 1359 } 1360 1361 Proxy::Buffer buffer; 1362 buffer.mFrameCount = stepCount; 1363 buffer.mRaw = audioBuffer->raw; 1364 1365 AutoMutex lock(mLock); 1366 mReleased += stepCount; 1367 mInUnderrun = false; 1368 mProxy->releaseBuffer(&buffer); 1369 1370 // restart track if it was disabled by audioflinger due to previous underrun 1371 if (mState == STATE_ACTIVE) { 1372 audio_track_cblk_t* cblk = mCblk; 1373 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1374 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1375 // FIXME ignoring status 1376 mAudioTrack->start(); 1377 } 1378 } 1379} 1380 1381// ------------------------------------------------------------------------- 1382 1383ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1384{ 1385 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1386 return INVALID_OPERATION; 1387 } 1388 1389 if (isDirect()) { 1390 AutoMutex lock(mLock); 1391 int32_t flags = android_atomic_and( 1392 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), 1393 &mCblk->mFlags); 1394 if (flags & CBLK_INVALID) { 1395 return DEAD_OBJECT; 1396 } 1397 } 1398 1399 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1400 // Sanity-check: user is most-likely passing an error code, and it would 1401 // make the return value ambiguous (actualSize vs error). 1402 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1403 return BAD_VALUE; 1404 } 1405 1406 size_t written = 0; 1407 Buffer audioBuffer; 1408 1409 while (userSize >= mFrameSize) { 1410 audioBuffer.frameCount = userSize / mFrameSize; 1411 1412 status_t err = obtainBuffer(&audioBuffer, 1413 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1414 if (err < 0) { 1415 if (written > 0) { 1416 break; 1417 } 1418 return ssize_t(err); 1419 } 1420 1421 size_t toWrite; 1422 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1423 // Divide capacity by 2 to take expansion into account 1424 toWrite = audioBuffer.size >> 1; 1425 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1426 } else { 1427 toWrite = audioBuffer.size; 1428 memcpy(audioBuffer.i8, buffer, toWrite); 1429 } 1430 buffer = ((const char *) buffer) + toWrite; 1431 userSize -= toWrite; 1432 written += toWrite; 1433 1434 releaseBuffer(&audioBuffer); 1435 } 1436 1437 return written; 1438} 1439 1440// ------------------------------------------------------------------------- 1441 1442TimedAudioTrack::TimedAudioTrack() { 1443 mIsTimed = true; 1444} 1445 1446status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1447{ 1448 AutoMutex lock(mLock); 1449 status_t result = UNKNOWN_ERROR; 1450 1451#if 1 1452 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1453 // while we are accessing the cblk 1454 sp<IAudioTrack> audioTrack = mAudioTrack; 1455 sp<IMemory> iMem = mCblkMemory; 1456#endif 1457 1458 // If the track is not invalid already, try to allocate a buffer. alloc 1459 // fails indicating that the server is dead, flag the track as invalid so 1460 // we can attempt to restore in just a bit. 1461 audio_track_cblk_t* cblk = mCblk; 1462 if (!(cblk->mFlags & CBLK_INVALID)) { 1463 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1464 if (result == DEAD_OBJECT) { 1465 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1466 } 1467 } 1468 1469 // If the track is invalid at this point, attempt to restore it. and try the 1470 // allocation one more time. 1471 if (cblk->mFlags & CBLK_INVALID) { 1472 result = restoreTrack_l("allocateTimedBuffer"); 1473 1474 if (result == NO_ERROR) { 1475 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1476 } 1477 } 1478 1479 return result; 1480} 1481 1482status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1483 int64_t pts) 1484{ 1485 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1486 { 1487 AutoMutex lock(mLock); 1488 audio_track_cblk_t* cblk = mCblk; 1489 // restart track if it was disabled by audioflinger due to previous underrun 1490 if (buffer->size() != 0 && status == NO_ERROR && 1491 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1492 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1493 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1494 // FIXME ignoring status 1495 mAudioTrack->start(); 1496 } 1497 } 1498 return status; 1499} 1500 1501status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1502 TargetTimeline target) 1503{ 1504 return mAudioTrack->setMediaTimeTransform(xform, target); 1505} 1506 1507// ------------------------------------------------------------------------- 1508 1509nsecs_t AudioTrack::processAudioBuffer() 1510{ 1511 // Currently the AudioTrack thread is not created if there are no callbacks. 1512 // Would it ever make sense to run the thread, even without callbacks? 1513 // If so, then replace this by checks at each use for mCbf != NULL. 1514 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1515 1516 mLock.lock(); 1517 if (mAwaitBoost) { 1518 mAwaitBoost = false; 1519 mLock.unlock(); 1520 static const int32_t kMaxTries = 5; 1521 int32_t tryCounter = kMaxTries; 1522 uint32_t pollUs = 10000; 1523 do { 1524 int policy = sched_getscheduler(0); 1525 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1526 break; 1527 } 1528 usleep(pollUs); 1529 pollUs <<= 1; 1530 } while (tryCounter-- > 0); 1531 if (tryCounter < 0) { 1532 ALOGE("did not receive expected priority boost on time"); 1533 } 1534 // Run again immediately 1535 return 0; 1536 } 1537 1538 // Can only reference mCblk while locked 1539 int32_t flags = android_atomic_and( 1540 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1541 1542 // Check for track invalidation 1543 if (flags & CBLK_INVALID) { 1544 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1545 // AudioSystem cache. We should not exit here but after calling the callback so 1546 // that the upper layers can recreate the track 1547 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) { 1548 status_t status = restoreTrack_l("processAudioBuffer"); 1549 mLock.unlock(); 1550 // Run again immediately, but with a new IAudioTrack 1551 return 0; 1552 } 1553 } 1554 1555 bool waitStreamEnd = mState == STATE_STOPPING; 1556 bool active = mState == STATE_ACTIVE; 1557 1558 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1559 bool newUnderrun = false; 1560 if (flags & CBLK_UNDERRUN) { 1561#if 0 1562 // Currently in shared buffer mode, when the server reaches the end of buffer, 1563 // the track stays active in continuous underrun state. It's up to the application 1564 // to pause or stop the track, or set the position to a new offset within buffer. 1565 // This was some experimental code to auto-pause on underrun. Keeping it here 1566 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1567 if (mTransfer == TRANSFER_SHARED) { 1568 mState = STATE_PAUSED; 1569 active = false; 1570 } 1571#endif 1572 if (!mInUnderrun) { 1573 mInUnderrun = true; 1574 newUnderrun = true; 1575 } 1576 } 1577 1578 // Get current position of server 1579 size_t position = updateAndGetPosition_l(); 1580 1581 // Manage marker callback 1582 bool markerReached = false; 1583 size_t markerPosition = mMarkerPosition; 1584 // FIXME fails for wraparound, need 64 bits 1585 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1586 mMarkerReached = markerReached = true; 1587 } 1588 1589 // Determine number of new position callback(s) that will be needed, while locked 1590 size_t newPosCount = 0; 1591 size_t newPosition = mNewPosition; 1592 size_t updatePeriod = mUpdatePeriod; 1593 // FIXME fails for wraparound, need 64 bits 1594 if (updatePeriod > 0 && position >= newPosition) { 1595 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1596 mNewPosition += updatePeriod * newPosCount; 1597 } 1598 1599 // Cache other fields that will be needed soon 1600 uint32_t loopPeriod = mLoopPeriod; 1601 uint32_t sampleRate = mSampleRate; 1602 uint32_t notificationFrames = mNotificationFramesAct; 1603 if (mRefreshRemaining) { 1604 mRefreshRemaining = false; 1605 mRemainingFrames = notificationFrames; 1606 mRetryOnPartialBuffer = false; 1607 } 1608 size_t misalignment = mProxy->getMisalignment(); 1609 uint32_t sequence = mSequence; 1610 sp<AudioTrackClientProxy> proxy = mProxy; 1611 1612 // These fields don't need to be cached, because they are assigned only by set(): 1613 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1614 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1615 1616 mLock.unlock(); 1617 1618 if (waitStreamEnd) { 1619 struct timespec timeout; 1620 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1621 timeout.tv_nsec = 0; 1622 1623 status_t status = proxy->waitStreamEndDone(&timeout); 1624 switch (status) { 1625 case NO_ERROR: 1626 case DEAD_OBJECT: 1627 case TIMED_OUT: 1628 mCbf(EVENT_STREAM_END, mUserData, NULL); 1629 { 1630 AutoMutex lock(mLock); 1631 // The previously assigned value of waitStreamEnd is no longer valid, 1632 // since the mutex has been unlocked and either the callback handler 1633 // or another thread could have re-started the AudioTrack during that time. 1634 waitStreamEnd = mState == STATE_STOPPING; 1635 if (waitStreamEnd) { 1636 mState = STATE_STOPPED; 1637 mReleased = 0; 1638 } 1639 } 1640 if (waitStreamEnd && status != DEAD_OBJECT) { 1641 return NS_INACTIVE; 1642 } 1643 break; 1644 } 1645 return 0; 1646 } 1647 1648 // perform callbacks while unlocked 1649 if (newUnderrun) { 1650 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1651 } 1652 // FIXME we will miss loops if loop cycle was signaled several times since last call 1653 // to processAudioBuffer() 1654 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1655 mCbf(EVENT_LOOP_END, mUserData, NULL); 1656 } 1657 if (flags & CBLK_BUFFER_END) { 1658 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1659 } 1660 if (markerReached) { 1661 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1662 } 1663 while (newPosCount > 0) { 1664 size_t temp = newPosition; 1665 mCbf(EVENT_NEW_POS, mUserData, &temp); 1666 newPosition += updatePeriod; 1667 newPosCount--; 1668 } 1669 1670 if (mObservedSequence != sequence) { 1671 mObservedSequence = sequence; 1672 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1673 // for offloaded tracks, just wait for the upper layers to recreate the track 1674 if (isOffloadedOrDirect()) { 1675 return NS_INACTIVE; 1676 } 1677 } 1678 1679 // if inactive, then don't run me again until re-started 1680 if (!active) { 1681 return NS_INACTIVE; 1682 } 1683 1684 // Compute the estimated time until the next timed event (position, markers, loops) 1685 // FIXME only for non-compressed audio 1686 uint32_t minFrames = ~0; 1687 if (!markerReached && position < markerPosition) { 1688 minFrames = markerPosition - position; 1689 } 1690 if (loopPeriod > 0 && loopPeriod < minFrames) { 1691 minFrames = loopPeriod; 1692 } 1693 if (updatePeriod > 0 && updatePeriod < minFrames) { 1694 minFrames = updatePeriod; 1695 } 1696 1697 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1698 static const uint32_t kPoll = 0; 1699 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1700 minFrames = kPoll * notificationFrames; 1701 } 1702 1703 // Convert frame units to time units 1704 nsecs_t ns = NS_WHENEVER; 1705 if (minFrames != (uint32_t) ~0) { 1706 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1707 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1708 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1709 } 1710 1711 // If not supplying data by EVENT_MORE_DATA, then we're done 1712 if (mTransfer != TRANSFER_CALLBACK) { 1713 return ns; 1714 } 1715 1716 struct timespec timeout; 1717 const struct timespec *requested = &ClientProxy::kForever; 1718 if (ns != NS_WHENEVER) { 1719 timeout.tv_sec = ns / 1000000000LL; 1720 timeout.tv_nsec = ns % 1000000000LL; 1721 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1722 requested = &timeout; 1723 } 1724 1725 while (mRemainingFrames > 0) { 1726 1727 Buffer audioBuffer; 1728 audioBuffer.frameCount = mRemainingFrames; 1729 size_t nonContig; 1730 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1731 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1732 "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount); 1733 requested = &ClientProxy::kNonBlocking; 1734 size_t avail = audioBuffer.frameCount + nonContig; 1735 ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d", 1736 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1737 if (err != NO_ERROR) { 1738 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1739 (isOffloaded() && (err == DEAD_OBJECT))) { 1740 return 0; 1741 } 1742 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1743 return NS_NEVER; 1744 } 1745 1746 if (mRetryOnPartialBuffer && !isOffloaded()) { 1747 mRetryOnPartialBuffer = false; 1748 if (avail < mRemainingFrames) { 1749 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1750 if (ns < 0 || myns < ns) { 1751 ns = myns; 1752 } 1753 return ns; 1754 } 1755 } 1756 1757 // Divide buffer size by 2 to take into account the expansion 1758 // due to 8 to 16 bit conversion: the callback must fill only half 1759 // of the destination buffer 1760 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1761 audioBuffer.size >>= 1; 1762 } 1763 1764 size_t reqSize = audioBuffer.size; 1765 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1766 size_t writtenSize = audioBuffer.size; 1767 1768 // Sanity check on returned size 1769 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1770 ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes", 1771 reqSize, ssize_t(writtenSize)); 1772 return NS_NEVER; 1773 } 1774 1775 if (writtenSize == 0) { 1776 // The callback is done filling buffers 1777 // Keep this thread going to handle timed events and 1778 // still try to get more data in intervals of WAIT_PERIOD_MS 1779 // but don't just loop and block the CPU, so wait 1780 return WAIT_PERIOD_MS * 1000000LL; 1781 } 1782 1783 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1784 // 8 to 16 bit conversion, note that source and destination are the same address 1785 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1786 audioBuffer.size <<= 1; 1787 } 1788 1789 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1790 audioBuffer.frameCount = releasedFrames; 1791 mRemainingFrames -= releasedFrames; 1792 if (misalignment >= releasedFrames) { 1793 misalignment -= releasedFrames; 1794 } else { 1795 misalignment = 0; 1796 } 1797 1798 releaseBuffer(&audioBuffer); 1799 1800 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1801 // if callback doesn't like to accept the full chunk 1802 if (writtenSize < reqSize) { 1803 continue; 1804 } 1805 1806 // There could be enough non-contiguous frames available to satisfy the remaining request 1807 if (mRemainingFrames <= nonContig) { 1808 continue; 1809 } 1810 1811#if 0 1812 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1813 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1814 // that total to a sum == notificationFrames. 1815 if (0 < misalignment && misalignment <= mRemainingFrames) { 1816 mRemainingFrames = misalignment; 1817 return (mRemainingFrames * 1100000000LL) / sampleRate; 1818 } 1819#endif 1820 1821 } 1822 mRemainingFrames = notificationFrames; 1823 mRetryOnPartialBuffer = true; 1824 1825 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1826 return 0; 1827} 1828 1829status_t AudioTrack::restoreTrack_l(const char *from) 1830{ 1831 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1832 isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from); 1833 ++mSequence; 1834 status_t result; 1835 1836 // refresh the audio configuration cache in this process to make sure we get new 1837 // output parameters and new IAudioFlinger in createTrack_l() 1838 AudioSystem::clearAudioConfigCache(); 1839 1840 if (isOffloadedOrDirect_l()) { 1841 // FIXME re-creation of offloaded tracks is not yet implemented 1842 return DEAD_OBJECT; 1843 } 1844 1845 // save the old static buffer position 1846 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1847 1848 // If a new IAudioTrack is successfully created, createTrack_l() will modify the 1849 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1850 // It will also delete the strong references on previous IAudioTrack and IMemory. 1851 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact. 1852 result = createTrack_l(); 1853 1854 // take the frames that will be lost by track recreation into account in saved position 1855 (void) updateAndGetPosition_l(); 1856 mPosition = mReleased; 1857 1858 if (result == NO_ERROR) { 1859 // continue playback from last known position, but 1860 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1861 if (mStaticProxy != NULL) { 1862 mLoopPeriod = 0; 1863 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1864 } 1865 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1866 // track destruction have been played? This is critical for SoundPool implementation 1867 // This must be broken, and needs to be tested/debugged. 1868#if 0 1869 // restore write index and set other indexes to reflect empty buffer status 1870 if (!strcmp(from, "start")) { 1871 // Make sure that a client relying on callback events indicating underrun or 1872 // the actual amount of audio frames played (e.g SoundPool) receives them. 1873 if (mSharedBuffer == 0) { 1874 // restart playback even if buffer is not completely filled. 1875 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1876 } 1877 } 1878#endif 1879 if (mState == STATE_ACTIVE) { 1880 result = mAudioTrack->start(); 1881 } 1882 } 1883 if (result != NO_ERROR) { 1884 ALOGW("restoreTrack_l() failed status %d", result); 1885 mState = STATE_STOPPED; 1886 mReleased = 0; 1887 } 1888 1889 return result; 1890} 1891 1892uint32_t AudioTrack::updateAndGetPosition_l() 1893{ 1894 // This is the sole place to read server consumed frames 1895 uint32_t newServer = mProxy->getPosition(); 1896 int32_t delta = newServer - mServer; 1897 mServer = newServer; 1898 // TODO There is controversy about whether there can be "negative jitter" in server position. 1899 // This should be investigated further, and if possible, it should be addressed. 1900 // A more definite failure mode is infrequent polling by client. 1901 // One could call (void)getPosition_l() in releaseBuffer(), 1902 // so mReleased and mPosition are always lock-step as best possible. 1903 // That should ensure delta never goes negative for infrequent polling 1904 // unless the server has more than 2^31 frames in its buffer, 1905 // in which case the use of uint32_t for these counters has bigger issues. 1906 if (delta < 0) { 1907 ALOGE("detected illegal retrograde motion by the server: mServer advanced by %d", delta); 1908 delta = 0; 1909 } 1910 return mPosition += (uint32_t) delta; 1911} 1912 1913status_t AudioTrack::setParameters(const String8& keyValuePairs) 1914{ 1915 AutoMutex lock(mLock); 1916 return mAudioTrack->setParameters(keyValuePairs); 1917} 1918 1919status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1920{ 1921 AutoMutex lock(mLock); 1922 // FIXME not implemented for fast tracks; should use proxy and SSQ 1923 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1924 return INVALID_OPERATION; 1925 } 1926 1927 switch (mState) { 1928 case STATE_ACTIVE: 1929 case STATE_PAUSED: 1930 break; // handle below 1931 case STATE_FLUSHED: 1932 case STATE_STOPPED: 1933 return WOULD_BLOCK; 1934 case STATE_STOPPING: 1935 case STATE_PAUSED_STOPPING: 1936 if (!isOffloaded_l()) { 1937 return INVALID_OPERATION; 1938 } 1939 break; // offloaded tracks handled below 1940 default: 1941 LOG_ALWAYS_FATAL("Invalid mState in getTimestamp(): %d", mState); 1942 break; 1943 } 1944 1945 // The presented frame count must always lag behind the consumed frame count. 1946 // To avoid a race, read the presented frames first. This ensures that presented <= consumed. 1947 status_t status = mAudioTrack->getTimestamp(timestamp); 1948 if (status != NO_ERROR) { 1949 ALOGV_IF(status != WOULD_BLOCK, "getTimestamp error:%#x", status); 1950 return status; 1951 } 1952 if (isOffloadedOrDirect_l()) { 1953 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) { 1954 // use cached paused position in case another offloaded track is running. 1955 timestamp.mPosition = mPausedPosition; 1956 clock_gettime(CLOCK_MONOTONIC, ×tamp.mTime); 1957 return NO_ERROR; 1958 } 1959 1960 // Check whether a pending flush or stop has completed, as those commands may 1961 // be asynchronous or return near finish. 1962 if (mStartUs != 0 && mSampleRate != 0) { 1963 static const int kTimeJitterUs = 100000; // 100 ms 1964 static const int k1SecUs = 1000000; 1965 1966 const int64_t timeNow = getNowUs(); 1967 1968 if (timeNow < mStartUs + k1SecUs) { // within first second of starting 1969 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime); 1970 if (timestampTimeUs < mStartUs) { 1971 return WOULD_BLOCK; // stale timestamp time, occurs before start. 1972 } 1973 const int64_t deltaTimeUs = timestampTimeUs - mStartUs; 1974 const int64_t deltaPositionByUs = timestamp.mPosition * 1000000LL / mSampleRate; 1975 1976 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) { 1977 // Verify that the counter can't count faster than the sample rate 1978 // since the start time. If greater, then that means we have failed 1979 // to completely flush or stop the previous playing track. 1980 ALOGW("incomplete flush or stop:" 1981 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)", 1982 (long long)deltaTimeUs, (long long)deltaPositionByUs, 1983 timestamp.mPosition); 1984 return WOULD_BLOCK; 1985 } 1986 } 1987 mStartUs = 0; // no need to check again, start timestamp has either expired or unneeded. 1988 } 1989 } else { 1990 // Update the mapping between local consumed (mPosition) and server consumed (mServer) 1991 (void) updateAndGetPosition_l(); 1992 // Server consumed (mServer) and presented both use the same server time base, 1993 // and server consumed is always >= presented. 1994 // The delta between these represents the number of frames in the buffer pipeline. 1995 // If this delta between these is greater than the client position, it means that 1996 // actually presented is still stuck at the starting line (figuratively speaking), 1997 // waiting for the first frame to go by. So we can't report a valid timestamp yet. 1998 if ((uint32_t) (mServer - timestamp.mPosition) > mPosition) { 1999 return INVALID_OPERATION; 2000 } 2001 // Convert timestamp position from server time base to client time base. 2002 // TODO The following code should work OK now because timestamp.mPosition is 32-bit. 2003 // But if we change it to 64-bit then this could fail. 2004 // If (mPosition - mServer) can be negative then should use: 2005 // (int32_t)(mPosition - mServer) 2006 timestamp.mPosition += mPosition - mServer; 2007 // Immediately after a call to getPosition_l(), mPosition and 2008 // mServer both represent the same frame position. mPosition is 2009 // in client's point of view, and mServer is in server's point of 2010 // view. So the difference between them is the "fudge factor" 2011 // between client and server views due to stop() and/or new 2012 // IAudioTrack. And timestamp.mPosition is initially in server's 2013 // point of view, so we need to apply the same fudge factor to it. 2014 } 2015 return status; 2016} 2017 2018String8 AudioTrack::getParameters(const String8& keys) 2019{ 2020 audio_io_handle_t output = getOutput(); 2021 if (output != AUDIO_IO_HANDLE_NONE) { 2022 return AudioSystem::getParameters(output, keys); 2023 } else { 2024 return String8::empty(); 2025 } 2026} 2027 2028bool AudioTrack::isOffloaded() const 2029{ 2030 AutoMutex lock(mLock); 2031 return isOffloaded_l(); 2032} 2033 2034bool AudioTrack::isDirect() const 2035{ 2036 AutoMutex lock(mLock); 2037 return isDirect_l(); 2038} 2039 2040bool AudioTrack::isOffloadedOrDirect() const 2041{ 2042 AutoMutex lock(mLock); 2043 return isOffloadedOrDirect_l(); 2044} 2045 2046 2047status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 2048{ 2049 2050 const size_t SIZE = 256; 2051 char buffer[SIZE]; 2052 String8 result; 2053 2054 result.append(" AudioTrack::dump\n"); 2055 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 2056 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 2057 result.append(buffer); 2058 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 2059 mChannelCount, mFrameCount); 2060 result.append(buffer); 2061 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 2062 result.append(buffer); 2063 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 2064 result.append(buffer); 2065 ::write(fd, result.string(), result.size()); 2066 return NO_ERROR; 2067} 2068 2069uint32_t AudioTrack::getUnderrunFrames() const 2070{ 2071 AutoMutex lock(mLock); 2072 return mProxy->getUnderrunFrames(); 2073} 2074 2075// ========================================================================= 2076 2077void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 2078{ 2079 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 2080 if (audioTrack != 0) { 2081 AutoMutex lock(audioTrack->mLock); 2082 audioTrack->mProxy->binderDied(); 2083 } 2084} 2085 2086// ========================================================================= 2087 2088AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 2089 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 2090 mIgnoreNextPausedInt(false) 2091{ 2092} 2093 2094AudioTrack::AudioTrackThread::~AudioTrackThread() 2095{ 2096} 2097 2098bool AudioTrack::AudioTrackThread::threadLoop() 2099{ 2100 { 2101 AutoMutex _l(mMyLock); 2102 if (mPaused) { 2103 mMyCond.wait(mMyLock); 2104 // caller will check for exitPending() 2105 return true; 2106 } 2107 if (mIgnoreNextPausedInt) { 2108 mIgnoreNextPausedInt = false; 2109 mPausedInt = false; 2110 } 2111 if (mPausedInt) { 2112 if (mPausedNs > 0) { 2113 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 2114 } else { 2115 mMyCond.wait(mMyLock); 2116 } 2117 mPausedInt = false; 2118 return true; 2119 } 2120 } 2121 if (exitPending()) { 2122 return false; 2123 } 2124 nsecs_t ns = mReceiver.processAudioBuffer(); 2125 switch (ns) { 2126 case 0: 2127 return true; 2128 case NS_INACTIVE: 2129 pauseInternal(); 2130 return true; 2131 case NS_NEVER: 2132 return false; 2133 case NS_WHENEVER: 2134 // FIXME increase poll interval, or make event-driven 2135 ns = 1000000000LL; 2136 // fall through 2137 default: 2138 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns); 2139 pauseInternal(ns); 2140 return true; 2141 } 2142} 2143 2144void AudioTrack::AudioTrackThread::requestExit() 2145{ 2146 // must be in this order to avoid a race condition 2147 Thread::requestExit(); 2148 resume(); 2149} 2150 2151void AudioTrack::AudioTrackThread::pause() 2152{ 2153 AutoMutex _l(mMyLock); 2154 mPaused = true; 2155} 2156 2157void AudioTrack::AudioTrackThread::resume() 2158{ 2159 AutoMutex _l(mMyLock); 2160 mIgnoreNextPausedInt = true; 2161 if (mPaused || mPausedInt) { 2162 mPaused = false; 2163 mPausedInt = false; 2164 mMyCond.signal(); 2165 } 2166} 2167 2168void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 2169{ 2170 AutoMutex _l(mMyLock); 2171 mPausedInt = true; 2172 mPausedNs = ns; 2173} 2174 2175}; // namespace android 2176