AudioTrack.cpp revision be837c328ae1ea2b193d05aaa3d4214c263b5b77
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 1230 mProxy->setVolumeLR(gain_minifloat_pack( 1231 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]), 1232 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT]))); 1233 1234 mProxy->setSendLevel(mSendLevel); 1235 mProxy->setSampleRate(mSampleRate); 1236 mProxy->setMinimum(mNotificationFramesAct); 1237 1238 mDeathNotifier = new DeathNotifier(this); 1239 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1240 1241 return NO_ERROR; 1242 } 1243 1244release: 1245 AudioSystem::releaseOutput(output, streamType, (audio_session_t)mSessionId); 1246 if (status == NO_ERROR) { 1247 status = NO_INIT; 1248 } 1249 return status; 1250} 1251 1252status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1253{ 1254 if (audioBuffer == NULL) { 1255 return BAD_VALUE; 1256 } 1257 if (mTransfer != TRANSFER_OBTAIN) { 1258 audioBuffer->frameCount = 0; 1259 audioBuffer->size = 0; 1260 audioBuffer->raw = NULL; 1261 return INVALID_OPERATION; 1262 } 1263 1264 const struct timespec *requested; 1265 struct timespec timeout; 1266 if (waitCount == -1) { 1267 requested = &ClientProxy::kForever; 1268 } else if (waitCount == 0) { 1269 requested = &ClientProxy::kNonBlocking; 1270 } else if (waitCount > 0) { 1271 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1272 timeout.tv_sec = ms / 1000; 1273 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1274 requested = &timeout; 1275 } else { 1276 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1277 requested = NULL; 1278 } 1279 return obtainBuffer(audioBuffer, requested); 1280} 1281 1282status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1283 struct timespec *elapsed, size_t *nonContig) 1284{ 1285 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1286 uint32_t oldSequence = 0; 1287 uint32_t newSequence; 1288 1289 Proxy::Buffer buffer; 1290 status_t status = NO_ERROR; 1291 1292 static const int32_t kMaxTries = 5; 1293 int32_t tryCounter = kMaxTries; 1294 1295 do { 1296 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1297 // keep them from going away if another thread re-creates the track during obtainBuffer() 1298 sp<AudioTrackClientProxy> proxy; 1299 sp<IMemory> iMem; 1300 1301 { // start of lock scope 1302 AutoMutex lock(mLock); 1303 1304 newSequence = mSequence; 1305 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1306 if (status == DEAD_OBJECT) { 1307 // re-create track, unless someone else has already done so 1308 if (newSequence == oldSequence) { 1309 status = restoreTrack_l("obtainBuffer"); 1310 if (status != NO_ERROR) { 1311 buffer.mFrameCount = 0; 1312 buffer.mRaw = NULL; 1313 buffer.mNonContig = 0; 1314 break; 1315 } 1316 } 1317 } 1318 oldSequence = newSequence; 1319 1320 // Keep the extra references 1321 proxy = mProxy; 1322 iMem = mCblkMemory; 1323 1324 if (mState == STATE_STOPPING) { 1325 status = -EINTR; 1326 buffer.mFrameCount = 0; 1327 buffer.mRaw = NULL; 1328 buffer.mNonContig = 0; 1329 break; 1330 } 1331 1332 // Non-blocking if track is stopped or paused 1333 if (mState != STATE_ACTIVE) { 1334 requested = &ClientProxy::kNonBlocking; 1335 } 1336 1337 } // end of lock scope 1338 1339 buffer.mFrameCount = audioBuffer->frameCount; 1340 // FIXME starts the requested timeout and elapsed over from scratch 1341 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1342 1343 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1344 1345 audioBuffer->frameCount = buffer.mFrameCount; 1346 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1347 audioBuffer->raw = buffer.mRaw; 1348 if (nonContig != NULL) { 1349 *nonContig = buffer.mNonContig; 1350 } 1351 return status; 1352} 1353 1354void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1355{ 1356 if (mTransfer == TRANSFER_SHARED) { 1357 return; 1358 } 1359 1360 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1361 if (stepCount == 0) { 1362 return; 1363 } 1364 1365 Proxy::Buffer buffer; 1366 buffer.mFrameCount = stepCount; 1367 buffer.mRaw = audioBuffer->raw; 1368 1369 AutoMutex lock(mLock); 1370 mReleased += stepCount; 1371 mInUnderrun = false; 1372 mProxy->releaseBuffer(&buffer); 1373 1374 // restart track if it was disabled by audioflinger due to previous underrun 1375 if (mState == STATE_ACTIVE) { 1376 audio_track_cblk_t* cblk = mCblk; 1377 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1378 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1379 // FIXME ignoring status 1380 mAudioTrack->start(); 1381 } 1382 } 1383} 1384 1385// ------------------------------------------------------------------------- 1386 1387ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1388{ 1389 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1390 return INVALID_OPERATION; 1391 } 1392 1393 if (isDirect()) { 1394 AutoMutex lock(mLock); 1395 int32_t flags = android_atomic_and( 1396 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), 1397 &mCblk->mFlags); 1398 if (flags & CBLK_INVALID) { 1399 return DEAD_OBJECT; 1400 } 1401 } 1402 1403 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1404 // Sanity-check: user is most-likely passing an error code, and it would 1405 // make the return value ambiguous (actualSize vs error). 1406 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1407 return BAD_VALUE; 1408 } 1409 1410 size_t written = 0; 1411 Buffer audioBuffer; 1412 1413 while (userSize >= mFrameSize) { 1414 audioBuffer.frameCount = userSize / mFrameSize; 1415 1416 status_t err = obtainBuffer(&audioBuffer, 1417 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1418 if (err < 0) { 1419 if (written > 0) { 1420 break; 1421 } 1422 return ssize_t(err); 1423 } 1424 1425 size_t toWrite; 1426 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1427 // Divide capacity by 2 to take expansion into account 1428 toWrite = audioBuffer.size >> 1; 1429 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1430 } else { 1431 toWrite = audioBuffer.size; 1432 memcpy(audioBuffer.i8, buffer, toWrite); 1433 } 1434 buffer = ((const char *) buffer) + toWrite; 1435 userSize -= toWrite; 1436 written += toWrite; 1437 1438 releaseBuffer(&audioBuffer); 1439 } 1440 1441 return written; 1442} 1443 1444// ------------------------------------------------------------------------- 1445 1446TimedAudioTrack::TimedAudioTrack() { 1447 mIsTimed = true; 1448} 1449 1450status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1451{ 1452 AutoMutex lock(mLock); 1453 status_t result = UNKNOWN_ERROR; 1454 1455#if 1 1456 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1457 // while we are accessing the cblk 1458 sp<IAudioTrack> audioTrack = mAudioTrack; 1459 sp<IMemory> iMem = mCblkMemory; 1460#endif 1461 1462 // If the track is not invalid already, try to allocate a buffer. alloc 1463 // fails indicating that the server is dead, flag the track as invalid so 1464 // we can attempt to restore in just a bit. 1465 audio_track_cblk_t* cblk = mCblk; 1466 if (!(cblk->mFlags & CBLK_INVALID)) { 1467 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1468 if (result == DEAD_OBJECT) { 1469 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1470 } 1471 } 1472 1473 // If the track is invalid at this point, attempt to restore it. and try the 1474 // allocation one more time. 1475 if (cblk->mFlags & CBLK_INVALID) { 1476 result = restoreTrack_l("allocateTimedBuffer"); 1477 1478 if (result == NO_ERROR) { 1479 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1480 } 1481 } 1482 1483 return result; 1484} 1485 1486status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1487 int64_t pts) 1488{ 1489 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1490 { 1491 AutoMutex lock(mLock); 1492 audio_track_cblk_t* cblk = mCblk; 1493 // restart track if it was disabled by audioflinger due to previous underrun 1494 if (buffer->size() != 0 && status == NO_ERROR && 1495 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1496 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1497 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1498 // FIXME ignoring status 1499 mAudioTrack->start(); 1500 } 1501 } 1502 return status; 1503} 1504 1505status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1506 TargetTimeline target) 1507{ 1508 return mAudioTrack->setMediaTimeTransform(xform, target); 1509} 1510 1511// ------------------------------------------------------------------------- 1512 1513nsecs_t AudioTrack::processAudioBuffer() 1514{ 1515 // Currently the AudioTrack thread is not created if there are no callbacks. 1516 // Would it ever make sense to run the thread, even without callbacks? 1517 // If so, then replace this by checks at each use for mCbf != NULL. 1518 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1519 1520 mLock.lock(); 1521 if (mAwaitBoost) { 1522 mAwaitBoost = false; 1523 mLock.unlock(); 1524 static const int32_t kMaxTries = 5; 1525 int32_t tryCounter = kMaxTries; 1526 uint32_t pollUs = 10000; 1527 do { 1528 int policy = sched_getscheduler(0); 1529 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1530 break; 1531 } 1532 usleep(pollUs); 1533 pollUs <<= 1; 1534 } while (tryCounter-- > 0); 1535 if (tryCounter < 0) { 1536 ALOGE("did not receive expected priority boost on time"); 1537 } 1538 // Run again immediately 1539 return 0; 1540 } 1541 1542 // Can only reference mCblk while locked 1543 int32_t flags = android_atomic_and( 1544 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1545 1546 // Check for track invalidation 1547 if (flags & CBLK_INVALID) { 1548 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1549 // AudioSystem cache. We should not exit here but after calling the callback so 1550 // that the upper layers can recreate the track 1551 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) { 1552 status_t status = restoreTrack_l("processAudioBuffer"); 1553 mLock.unlock(); 1554 // Run again immediately, but with a new IAudioTrack 1555 return 0; 1556 } 1557 } 1558 1559 bool waitStreamEnd = mState == STATE_STOPPING; 1560 bool active = mState == STATE_ACTIVE; 1561 1562 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1563 bool newUnderrun = false; 1564 if (flags & CBLK_UNDERRUN) { 1565#if 0 1566 // Currently in shared buffer mode, when the server reaches the end of buffer, 1567 // the track stays active in continuous underrun state. It's up to the application 1568 // to pause or stop the track, or set the position to a new offset within buffer. 1569 // This was some experimental code to auto-pause on underrun. Keeping it here 1570 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1571 if (mTransfer == TRANSFER_SHARED) { 1572 mState = STATE_PAUSED; 1573 active = false; 1574 } 1575#endif 1576 if (!mInUnderrun) { 1577 mInUnderrun = true; 1578 newUnderrun = true; 1579 } 1580 } 1581 1582 // Get current position of server 1583 size_t position = updateAndGetPosition_l(); 1584 1585 // Manage marker callback 1586 bool markerReached = false; 1587 size_t markerPosition = mMarkerPosition; 1588 // FIXME fails for wraparound, need 64 bits 1589 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1590 mMarkerReached = markerReached = true; 1591 } 1592 1593 // Determine number of new position callback(s) that will be needed, while locked 1594 size_t newPosCount = 0; 1595 size_t newPosition = mNewPosition; 1596 size_t updatePeriod = mUpdatePeriod; 1597 // FIXME fails for wraparound, need 64 bits 1598 if (updatePeriod > 0 && position >= newPosition) { 1599 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1600 mNewPosition += updatePeriod * newPosCount; 1601 } 1602 1603 // Cache other fields that will be needed soon 1604 uint32_t loopPeriod = mLoopPeriod; 1605 uint32_t sampleRate = mSampleRate; 1606 uint32_t notificationFrames = mNotificationFramesAct; 1607 if (mRefreshRemaining) { 1608 mRefreshRemaining = false; 1609 mRemainingFrames = notificationFrames; 1610 mRetryOnPartialBuffer = false; 1611 } 1612 size_t misalignment = mProxy->getMisalignment(); 1613 uint32_t sequence = mSequence; 1614 sp<AudioTrackClientProxy> proxy = mProxy; 1615 1616 // These fields don't need to be cached, because they are assigned only by set(): 1617 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1618 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1619 1620 mLock.unlock(); 1621 1622 if (waitStreamEnd) { 1623 struct timespec timeout; 1624 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1625 timeout.tv_nsec = 0; 1626 1627 status_t status = proxy->waitStreamEndDone(&timeout); 1628 switch (status) { 1629 case NO_ERROR: 1630 case DEAD_OBJECT: 1631 case TIMED_OUT: 1632 mCbf(EVENT_STREAM_END, mUserData, NULL); 1633 { 1634 AutoMutex lock(mLock); 1635 // The previously assigned value of waitStreamEnd is no longer valid, 1636 // since the mutex has been unlocked and either the callback handler 1637 // or another thread could have re-started the AudioTrack during that time. 1638 waitStreamEnd = mState == STATE_STOPPING; 1639 if (waitStreamEnd) { 1640 mState = STATE_STOPPED; 1641 mReleased = 0; 1642 } 1643 } 1644 if (waitStreamEnd && status != DEAD_OBJECT) { 1645 return NS_INACTIVE; 1646 } 1647 break; 1648 } 1649 return 0; 1650 } 1651 1652 // perform callbacks while unlocked 1653 if (newUnderrun) { 1654 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1655 } 1656 // FIXME we will miss loops if loop cycle was signaled several times since last call 1657 // to processAudioBuffer() 1658 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1659 mCbf(EVENT_LOOP_END, mUserData, NULL); 1660 } 1661 if (flags & CBLK_BUFFER_END) { 1662 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1663 } 1664 if (markerReached) { 1665 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1666 } 1667 while (newPosCount > 0) { 1668 size_t temp = newPosition; 1669 mCbf(EVENT_NEW_POS, mUserData, &temp); 1670 newPosition += updatePeriod; 1671 newPosCount--; 1672 } 1673 1674 if (mObservedSequence != sequence) { 1675 mObservedSequence = sequence; 1676 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1677 // for offloaded tracks, just wait for the upper layers to recreate the track 1678 if (isOffloadedOrDirect()) { 1679 return NS_INACTIVE; 1680 } 1681 } 1682 1683 // if inactive, then don't run me again until re-started 1684 if (!active) { 1685 return NS_INACTIVE; 1686 } 1687 1688 // Compute the estimated time until the next timed event (position, markers, loops) 1689 // FIXME only for non-compressed audio 1690 uint32_t minFrames = ~0; 1691 if (!markerReached && position < markerPosition) { 1692 minFrames = markerPosition - position; 1693 } 1694 if (loopPeriod > 0 && loopPeriod < minFrames) { 1695 minFrames = loopPeriod; 1696 } 1697 if (updatePeriod > 0 && updatePeriod < minFrames) { 1698 minFrames = updatePeriod; 1699 } 1700 1701 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1702 static const uint32_t kPoll = 0; 1703 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1704 minFrames = kPoll * notificationFrames; 1705 } 1706 1707 // Convert frame units to time units 1708 nsecs_t ns = NS_WHENEVER; 1709 if (minFrames != (uint32_t) ~0) { 1710 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1711 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1712 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1713 } 1714 1715 // If not supplying data by EVENT_MORE_DATA, then we're done 1716 if (mTransfer != TRANSFER_CALLBACK) { 1717 return ns; 1718 } 1719 1720 struct timespec timeout; 1721 const struct timespec *requested = &ClientProxy::kForever; 1722 if (ns != NS_WHENEVER) { 1723 timeout.tv_sec = ns / 1000000000LL; 1724 timeout.tv_nsec = ns % 1000000000LL; 1725 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1726 requested = &timeout; 1727 } 1728 1729 while (mRemainingFrames > 0) { 1730 1731 Buffer audioBuffer; 1732 audioBuffer.frameCount = mRemainingFrames; 1733 size_t nonContig; 1734 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1735 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1736 "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount); 1737 requested = &ClientProxy::kNonBlocking; 1738 size_t avail = audioBuffer.frameCount + nonContig; 1739 ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d", 1740 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1741 if (err != NO_ERROR) { 1742 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1743 (isOffloaded() && (err == DEAD_OBJECT))) { 1744 return 0; 1745 } 1746 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1747 return NS_NEVER; 1748 } 1749 1750 if (mRetryOnPartialBuffer && !isOffloaded()) { 1751 mRetryOnPartialBuffer = false; 1752 if (avail < mRemainingFrames) { 1753 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1754 if (ns < 0 || myns < ns) { 1755 ns = myns; 1756 } 1757 return ns; 1758 } 1759 } 1760 1761 // Divide buffer size by 2 to take into account the expansion 1762 // due to 8 to 16 bit conversion: the callback must fill only half 1763 // of the destination buffer 1764 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1765 audioBuffer.size >>= 1; 1766 } 1767 1768 size_t reqSize = audioBuffer.size; 1769 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1770 size_t writtenSize = audioBuffer.size; 1771 1772 // Sanity check on returned size 1773 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1774 ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes", 1775 reqSize, ssize_t(writtenSize)); 1776 return NS_NEVER; 1777 } 1778 1779 if (writtenSize == 0) { 1780 // The callback is done filling buffers 1781 // Keep this thread going to handle timed events and 1782 // still try to get more data in intervals of WAIT_PERIOD_MS 1783 // but don't just loop and block the CPU, so wait 1784 return WAIT_PERIOD_MS * 1000000LL; 1785 } 1786 1787 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1788 // 8 to 16 bit conversion, note that source and destination are the same address 1789 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1790 audioBuffer.size <<= 1; 1791 } 1792 1793 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1794 audioBuffer.frameCount = releasedFrames; 1795 mRemainingFrames -= releasedFrames; 1796 if (misalignment >= releasedFrames) { 1797 misalignment -= releasedFrames; 1798 } else { 1799 misalignment = 0; 1800 } 1801 1802 releaseBuffer(&audioBuffer); 1803 1804 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1805 // if callback doesn't like to accept the full chunk 1806 if (writtenSize < reqSize) { 1807 continue; 1808 } 1809 1810 // There could be enough non-contiguous frames available to satisfy the remaining request 1811 if (mRemainingFrames <= nonContig) { 1812 continue; 1813 } 1814 1815#if 0 1816 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1817 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1818 // that total to a sum == notificationFrames. 1819 if (0 < misalignment && misalignment <= mRemainingFrames) { 1820 mRemainingFrames = misalignment; 1821 return (mRemainingFrames * 1100000000LL) / sampleRate; 1822 } 1823#endif 1824 1825 } 1826 mRemainingFrames = notificationFrames; 1827 mRetryOnPartialBuffer = true; 1828 1829 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1830 return 0; 1831} 1832 1833status_t AudioTrack::restoreTrack_l(const char *from) 1834{ 1835 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1836 isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from); 1837 ++mSequence; 1838 status_t result; 1839 1840 // refresh the audio configuration cache in this process to make sure we get new 1841 // output parameters and new IAudioFlinger in createTrack_l() 1842 AudioSystem::clearAudioConfigCache(); 1843 1844 if (isOffloadedOrDirect_l()) { 1845 // FIXME re-creation of offloaded tracks is not yet implemented 1846 return DEAD_OBJECT; 1847 } 1848 1849 // save the old static buffer position 1850 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1851 1852 // If a new IAudioTrack is successfully created, createTrack_l() will modify the 1853 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1854 // It will also delete the strong references on previous IAudioTrack and IMemory. 1855 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact. 1856 result = createTrack_l(); 1857 1858 // take the frames that will be lost by track recreation into account in saved position 1859 (void) updateAndGetPosition_l(); 1860 mPosition = mReleased; 1861 1862 if (result == NO_ERROR) { 1863 // continue playback from last known position, but 1864 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1865 if (mStaticProxy != NULL) { 1866 mLoopPeriod = 0; 1867 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1868 } 1869 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1870 // track destruction have been played? This is critical for SoundPool implementation 1871 // This must be broken, and needs to be tested/debugged. 1872#if 0 1873 // restore write index and set other indexes to reflect empty buffer status 1874 if (!strcmp(from, "start")) { 1875 // Make sure that a client relying on callback events indicating underrun or 1876 // the actual amount of audio frames played (e.g SoundPool) receives them. 1877 if (mSharedBuffer == 0) { 1878 // restart playback even if buffer is not completely filled. 1879 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1880 } 1881 } 1882#endif 1883 if (mState == STATE_ACTIVE) { 1884 result = mAudioTrack->start(); 1885 } 1886 } 1887 if (result != NO_ERROR) { 1888 ALOGW("restoreTrack_l() failed status %d", result); 1889 mState = STATE_STOPPED; 1890 mReleased = 0; 1891 } 1892 1893 return result; 1894} 1895 1896uint32_t AudioTrack::updateAndGetPosition_l() 1897{ 1898 // This is the sole place to read server consumed frames 1899 uint32_t newServer = mProxy->getPosition(); 1900 int32_t delta = newServer - mServer; 1901 mServer = newServer; 1902 // TODO There is controversy about whether there can be "negative jitter" in server position. 1903 // This should be investigated further, and if possible, it should be addressed. 1904 // A more definite failure mode is infrequent polling by client. 1905 // One could call (void)getPosition_l() in releaseBuffer(), 1906 // so mReleased and mPosition are always lock-step as best possible. 1907 // That should ensure delta never goes negative for infrequent polling 1908 // unless the server has more than 2^31 frames in its buffer, 1909 // in which case the use of uint32_t for these counters has bigger issues. 1910 if (delta < 0) { 1911 ALOGE("detected illegal retrograde motion by the server: mServer advanced by %d", delta); 1912 delta = 0; 1913 } 1914 return mPosition += (uint32_t) delta; 1915} 1916 1917status_t AudioTrack::setParameters(const String8& keyValuePairs) 1918{ 1919 AutoMutex lock(mLock); 1920 return mAudioTrack->setParameters(keyValuePairs); 1921} 1922 1923status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1924{ 1925 AutoMutex lock(mLock); 1926 // FIXME not implemented for fast tracks; should use proxy and SSQ 1927 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1928 return INVALID_OPERATION; 1929 } 1930 1931 switch (mState) { 1932 case STATE_ACTIVE: 1933 case STATE_PAUSED: 1934 break; // handle below 1935 case STATE_FLUSHED: 1936 case STATE_STOPPED: 1937 return WOULD_BLOCK; 1938 case STATE_STOPPING: 1939 case STATE_PAUSED_STOPPING: 1940 if (!isOffloaded_l()) { 1941 return INVALID_OPERATION; 1942 } 1943 break; // offloaded tracks handled below 1944 default: 1945 LOG_ALWAYS_FATAL("Invalid mState in getTimestamp(): %d", mState); 1946 break; 1947 } 1948 1949 // The presented frame count must always lag behind the consumed frame count. 1950 // To avoid a race, read the presented frames first. This ensures that presented <= consumed. 1951 status_t status = mAudioTrack->getTimestamp(timestamp); 1952 if (status != NO_ERROR) { 1953 ALOGV_IF(status != WOULD_BLOCK, "getTimestamp error:%#x", status); 1954 return status; 1955 } 1956 if (isOffloadedOrDirect_l()) { 1957 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) { 1958 // use cached paused position in case another offloaded track is running. 1959 timestamp.mPosition = mPausedPosition; 1960 clock_gettime(CLOCK_MONOTONIC, ×tamp.mTime); 1961 return NO_ERROR; 1962 } 1963 1964 // Check whether a pending flush or stop has completed, as those commands may 1965 // be asynchronous or return near finish. 1966 if (mStartUs != 0 && mSampleRate != 0) { 1967 static const int kTimeJitterUs = 100000; // 100 ms 1968 static const int k1SecUs = 1000000; 1969 1970 const int64_t timeNow = getNowUs(); 1971 1972 if (timeNow < mStartUs + k1SecUs) { // within first second of starting 1973 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime); 1974 if (timestampTimeUs < mStartUs) { 1975 return WOULD_BLOCK; // stale timestamp time, occurs before start. 1976 } 1977 const int64_t deltaTimeUs = timestampTimeUs - mStartUs; 1978 const int64_t deltaPositionByUs = timestamp.mPosition * 1000000LL / mSampleRate; 1979 1980 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) { 1981 // Verify that the counter can't count faster than the sample rate 1982 // since the start time. If greater, then that means we have failed 1983 // to completely flush or stop the previous playing track. 1984 ALOGW("incomplete flush or stop:" 1985 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)", 1986 (long long)deltaTimeUs, (long long)deltaPositionByUs, 1987 timestamp.mPosition); 1988 return WOULD_BLOCK; 1989 } 1990 } 1991 mStartUs = 0; // no need to check again, start timestamp has either expired or unneeded. 1992 } 1993 } else { 1994 // Update the mapping between local consumed (mPosition) and server consumed (mServer) 1995 (void) updateAndGetPosition_l(); 1996 // Server consumed (mServer) and presented both use the same server time base, 1997 // and server consumed is always >= presented. 1998 // The delta between these represents the number of frames in the buffer pipeline. 1999 // If this delta between these is greater than the client position, it means that 2000 // actually presented is still stuck at the starting line (figuratively speaking), 2001 // waiting for the first frame to go by. So we can't report a valid timestamp yet. 2002 if ((uint32_t) (mServer - timestamp.mPosition) > mPosition) { 2003 return INVALID_OPERATION; 2004 } 2005 // Convert timestamp position from server time base to client time base. 2006 // TODO The following code should work OK now because timestamp.mPosition is 32-bit. 2007 // But if we change it to 64-bit then this could fail. 2008 // If (mPosition - mServer) can be negative then should use: 2009 // (int32_t)(mPosition - mServer) 2010 timestamp.mPosition += mPosition - mServer; 2011 // Immediately after a call to getPosition_l(), mPosition and 2012 // mServer both represent the same frame position. mPosition is 2013 // in client's point of view, and mServer is in server's point of 2014 // view. So the difference between them is the "fudge factor" 2015 // between client and server views due to stop() and/or new 2016 // IAudioTrack. And timestamp.mPosition is initially in server's 2017 // point of view, so we need to apply the same fudge factor to it. 2018 } 2019 return status; 2020} 2021 2022String8 AudioTrack::getParameters(const String8& keys) 2023{ 2024 audio_io_handle_t output = getOutput(); 2025 if (output != AUDIO_IO_HANDLE_NONE) { 2026 return AudioSystem::getParameters(output, keys); 2027 } else { 2028 return String8::empty(); 2029 } 2030} 2031 2032bool AudioTrack::isOffloaded() const 2033{ 2034 AutoMutex lock(mLock); 2035 return isOffloaded_l(); 2036} 2037 2038bool AudioTrack::isDirect() const 2039{ 2040 AutoMutex lock(mLock); 2041 return isDirect_l(); 2042} 2043 2044bool AudioTrack::isOffloadedOrDirect() const 2045{ 2046 AutoMutex lock(mLock); 2047 return isOffloadedOrDirect_l(); 2048} 2049 2050 2051status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 2052{ 2053 2054 const size_t SIZE = 256; 2055 char buffer[SIZE]; 2056 String8 result; 2057 2058 result.append(" AudioTrack::dump\n"); 2059 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 2060 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 2061 result.append(buffer); 2062 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 2063 mChannelCount, mFrameCount); 2064 result.append(buffer); 2065 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 2066 result.append(buffer); 2067 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 2068 result.append(buffer); 2069 ::write(fd, result.string(), result.size()); 2070 return NO_ERROR; 2071} 2072 2073uint32_t AudioTrack::getUnderrunFrames() const 2074{ 2075 AutoMutex lock(mLock); 2076 return mProxy->getUnderrunFrames(); 2077} 2078 2079// ========================================================================= 2080 2081void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 2082{ 2083 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 2084 if (audioTrack != 0) { 2085 AutoMutex lock(audioTrack->mLock); 2086 audioTrack->mProxy->binderDied(); 2087 } 2088} 2089 2090// ========================================================================= 2091 2092AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 2093 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 2094 mIgnoreNextPausedInt(false) 2095{ 2096} 2097 2098AudioTrack::AudioTrackThread::~AudioTrackThread() 2099{ 2100} 2101 2102bool AudioTrack::AudioTrackThread::threadLoop() 2103{ 2104 { 2105 AutoMutex _l(mMyLock); 2106 if (mPaused) { 2107 mMyCond.wait(mMyLock); 2108 // caller will check for exitPending() 2109 return true; 2110 } 2111 if (mIgnoreNextPausedInt) { 2112 mIgnoreNextPausedInt = false; 2113 mPausedInt = false; 2114 } 2115 if (mPausedInt) { 2116 if (mPausedNs > 0) { 2117 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 2118 } else { 2119 mMyCond.wait(mMyLock); 2120 } 2121 mPausedInt = false; 2122 return true; 2123 } 2124 } 2125 if (exitPending()) { 2126 return false; 2127 } 2128 nsecs_t ns = mReceiver.processAudioBuffer(); 2129 switch (ns) { 2130 case 0: 2131 return true; 2132 case NS_INACTIVE: 2133 pauseInternal(); 2134 return true; 2135 case NS_NEVER: 2136 return false; 2137 case NS_WHENEVER: 2138 // FIXME increase poll interval, or make event-driven 2139 ns = 1000000000LL; 2140 // fall through 2141 default: 2142 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns); 2143 pauseInternal(ns); 2144 return true; 2145 } 2146} 2147 2148void AudioTrack::AudioTrackThread::requestExit() 2149{ 2150 // must be in this order to avoid a race condition 2151 Thread::requestExit(); 2152 resume(); 2153} 2154 2155void AudioTrack::AudioTrackThread::pause() 2156{ 2157 AutoMutex _l(mMyLock); 2158 mPaused = true; 2159} 2160 2161void AudioTrack::AudioTrackThread::resume() 2162{ 2163 AutoMutex _l(mMyLock); 2164 mIgnoreNextPausedInt = true; 2165 if (mPaused || mPausedInt) { 2166 mPaused = false; 2167 mPausedInt = false; 2168 mMyCond.signal(); 2169 } 2170} 2171 2172void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 2173{ 2174 AutoMutex _l(mMyLock); 2175 mPausedInt = true; 2176 mPausedNs = ns; 2177} 2178 2179}; // namespace android 2180