AudioTrack.cpp revision 275e8e9de2e11b4b344f5a201f1f0e51fda02d9c
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 if (mCblk->mFlags & CBLK_INVALID) { 855 restoreTrack_l("getPosition"); 856 } 857 858 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 859 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 860 0 : updateAndGetPosition_l(); 861 } 862 return NO_ERROR; 863} 864 865status_t AudioTrack::getBufferPosition(uint32_t *position) 866{ 867 if (mSharedBuffer == 0 || mIsTimed) { 868 return INVALID_OPERATION; 869 } 870 if (position == NULL) { 871 return BAD_VALUE; 872 } 873 874 AutoMutex lock(mLock); 875 *position = mStaticProxy->getBufferPosition(); 876 return NO_ERROR; 877} 878 879status_t AudioTrack::reload() 880{ 881 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 882 return INVALID_OPERATION; 883 } 884 885 AutoMutex lock(mLock); 886 // See setPosition() regarding setting parameters such as loop points or position while active 887 if (mState == STATE_ACTIVE) { 888 return INVALID_OPERATION; 889 } 890 mNewPosition = mUpdatePeriod; 891 mLoopPeriod = 0; 892 // FIXME The new code cannot reload while keeping a loop specified. 893 // Need to check how the old code handled this, and whether it's a significant change. 894 mStaticProxy->setLoop(0, mFrameCount, 0); 895 return NO_ERROR; 896} 897 898audio_io_handle_t AudioTrack::getOutput() const 899{ 900 AutoMutex lock(mLock); 901 return mOutput; 902} 903 904status_t AudioTrack::attachAuxEffect(int effectId) 905{ 906 AutoMutex lock(mLock); 907 status_t status = mAudioTrack->attachAuxEffect(effectId); 908 if (status == NO_ERROR) { 909 mAuxEffectId = effectId; 910 } 911 return status; 912} 913 914audio_stream_type_t AudioTrack::streamType() const 915{ 916 if (mStreamType == AUDIO_STREAM_DEFAULT) { 917 return audio_attributes_to_stream_type(&mAttributes); 918 } 919 return mStreamType; 920} 921 922// ------------------------------------------------------------------------- 923 924// must be called with mLock held 925status_t AudioTrack::createTrack_l() 926{ 927 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 928 if (audioFlinger == 0) { 929 ALOGE("Could not get audioflinger"); 930 return NO_INIT; 931 } 932 933 audio_io_handle_t output; 934 audio_stream_type_t streamType = mStreamType; 935 audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL; 936 status_t status = AudioSystem::getOutputForAttr(attr, &output, 937 (audio_session_t)mSessionId, &streamType, 938 mSampleRate, mFormat, mChannelMask, 939 mFlags, mOffloadInfo); 940 941 942 if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) { 943 ALOGE("Could not get audio output for stream type %d, usage %d, sample rate %u, format %#x," 944 " channel mask %#x, flags %#x", 945 streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags); 946 return BAD_VALUE; 947 } 948 { 949 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 950 // we must release it ourselves if anything goes wrong. 951 952 // Not all of these values are needed under all conditions, but it is easier to get them all 953 954 uint32_t afLatency; 955 status = AudioSystem::getLatency(output, &afLatency); 956 if (status != NO_ERROR) { 957 ALOGE("getLatency(%d) failed status %d", output, status); 958 goto release; 959 } 960 961 size_t afFrameCount; 962 status = AudioSystem::getFrameCount(output, &afFrameCount); 963 if (status != NO_ERROR) { 964 ALOGE("getFrameCount(output=%d) status %d", output, status); 965 goto release; 966 } 967 968 uint32_t afSampleRate; 969 status = AudioSystem::getSamplingRate(output, &afSampleRate); 970 if (status != NO_ERROR) { 971 ALOGE("getSamplingRate(output=%d) status %d", output, status); 972 goto release; 973 } 974 if (mSampleRate == 0) { 975 mSampleRate = afSampleRate; 976 } 977 // Client decides whether the track is TIMED (see below), but can only express a preference 978 // for FAST. Server will perform additional tests. 979 if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( 980 // either of these use cases: 981 // use case 1: shared buffer 982 (mSharedBuffer != 0) || 983 // use case 2: callback transfer mode 984 (mTransfer == TRANSFER_CALLBACK)) && 985 // matching sample rate 986 (mSampleRate == afSampleRate))) { 987 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 988 // once denied, do not request again if IAudioTrack is re-created 989 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 990 } 991 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 992 993 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 994 // n = 1 fast track with single buffering; nBuffering is ignored 995 // n = 2 fast track with double buffering 996 // n = 2 normal track, no sample rate conversion 997 // n = 3 normal track, with sample rate conversion 998 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 999 // n > 3 very high latency or very small notification interval; nBuffering is ignored 1000 const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; 1001 1002 mNotificationFramesAct = mNotificationFramesReq; 1003 1004 size_t frameCount = mReqFrameCount; 1005 if (!audio_is_linear_pcm(mFormat)) { 1006 1007 if (mSharedBuffer != 0) { 1008 // Same comment as below about ignoring frameCount parameter for set() 1009 frameCount = mSharedBuffer->size(); 1010 } else if (frameCount == 0) { 1011 frameCount = afFrameCount; 1012 } 1013 if (mNotificationFramesAct != frameCount) { 1014 mNotificationFramesAct = frameCount; 1015 } 1016 } else if (mSharedBuffer != 0) { 1017 1018 // Ensure that buffer alignment matches channel count 1019 // 8-bit data in shared memory is not currently supported by AudioFlinger 1020 size_t alignment = audio_bytes_per_sample( 1021 mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat); 1022 if (alignment & 1) { 1023 alignment = 1; 1024 } 1025 if (mChannelCount > 1) { 1026 // More than 2 channels does not require stronger alignment than stereo 1027 alignment <<= 1; 1028 } 1029 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 1030 ALOGE("Invalid buffer alignment: address %p, channel count %u", 1031 mSharedBuffer->pointer(), mChannelCount); 1032 status = BAD_VALUE; 1033 goto release; 1034 } 1035 1036 // When initializing a shared buffer AudioTrack via constructors, 1037 // there's no frameCount parameter. 1038 // But when initializing a shared buffer AudioTrack via set(), 1039 // there _is_ a frameCount parameter. We silently ignore it. 1040 frameCount = mSharedBuffer->size() / mFrameSizeAF; 1041 1042 } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 1043 1044 // FIXME move these calculations and associated checks to server 1045 1046 // Ensure that buffer depth covers at least audio hardware latency 1047 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 1048 ALOGV("afFrameCount=%zu, minBufCount=%d, afSampleRate=%u, afLatency=%d", 1049 afFrameCount, minBufCount, afSampleRate, afLatency); 1050 if (minBufCount <= nBuffering) { 1051 minBufCount = nBuffering; 1052 } 1053 1054 size_t minFrameCount = afFrameCount * minBufCount * uint64_t(mSampleRate) / afSampleRate; 1055 ALOGV("minFrameCount: %zu, afFrameCount=%zu, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 1056 ", afLatency=%d", 1057 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); 1058 1059 if (frameCount == 0) { 1060 frameCount = minFrameCount; 1061 } else if (frameCount < minFrameCount) { 1062 // not ALOGW because it happens all the time when playing key clicks over A2DP 1063 ALOGV("Minimum buffer size corrected from %zu to %zu", 1064 frameCount, minFrameCount); 1065 frameCount = minFrameCount; 1066 } 1067 // Make sure that application is notified with sufficient margin before underrun 1068 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1069 mNotificationFramesAct = frameCount/nBuffering; 1070 } 1071 1072 } else { 1073 // For fast tracks, the frame count calculations and checks are done by server 1074 } 1075 1076 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 1077 if (mIsTimed) { 1078 trackFlags |= IAudioFlinger::TRACK_TIMED; 1079 } 1080 1081 pid_t tid = -1; 1082 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1083 trackFlags |= IAudioFlinger::TRACK_FAST; 1084 if (mAudioTrackThread != 0) { 1085 tid = mAudioTrackThread->getTid(); 1086 } 1087 } 1088 1089 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1090 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 1091 } 1092 1093 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1094 trackFlags |= IAudioFlinger::TRACK_DIRECT; 1095 } 1096 1097 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1098 // but we will still need the original value also 1099 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 1100 mSampleRate, 1101 // AudioFlinger only sees 16-bit PCM 1102 mFormat == AUDIO_FORMAT_PCM_8_BIT && 1103 !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT) ? 1104 AUDIO_FORMAT_PCM_16_BIT : mFormat, 1105 mChannelMask, 1106 &temp, 1107 &trackFlags, 1108 mSharedBuffer, 1109 output, 1110 tid, 1111 &mSessionId, 1112 mClientUid, 1113 &status); 1114 1115 if (status != NO_ERROR) { 1116 ALOGE("AudioFlinger could not create track, status: %d", status); 1117 goto release; 1118 } 1119 ALOG_ASSERT(track != 0); 1120 1121 // AudioFlinger now owns the reference to the I/O handle, 1122 // so we are no longer responsible for releasing it. 1123 1124 sp<IMemory> iMem = track->getCblk(); 1125 if (iMem == 0) { 1126 ALOGE("Could not get control block"); 1127 return NO_INIT; 1128 } 1129 void *iMemPointer = iMem->pointer(); 1130 if (iMemPointer == NULL) { 1131 ALOGE("Could not get control block pointer"); 1132 return NO_INIT; 1133 } 1134 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1135 if (mAudioTrack != 0) { 1136 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1137 mDeathNotifier.clear(); 1138 } 1139 mAudioTrack = track; 1140 mCblkMemory = iMem; 1141 IPCThreadState::self()->flushCommands(); 1142 1143 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1144 mCblk = cblk; 1145 // note that temp is the (possibly revised) value of frameCount 1146 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1147 // In current design, AudioTrack client checks and ensures frame count validity before 1148 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1149 // for fast track as it uses a special method of assigning frame count. 1150 ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp); 1151 } 1152 frameCount = temp; 1153 1154 mAwaitBoost = false; 1155 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1156 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1157 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount); 1158 mAwaitBoost = true; 1159 if (mSharedBuffer == 0) { 1160 // Theoretically double-buffering is not required for fast tracks, 1161 // due to tighter scheduling. But in practice, to accommodate kernels with 1162 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1163 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1164 mNotificationFramesAct = frameCount/nBuffering; 1165 } 1166 } 1167 } else { 1168 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount); 1169 // once denied, do not request again if IAudioTrack is re-created 1170 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1171 if (mSharedBuffer == 0) { 1172 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1173 mNotificationFramesAct = frameCount/nBuffering; 1174 } 1175 } 1176 } 1177 } 1178 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1179 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1180 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1181 } else { 1182 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1183 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1184 // FIXME This is a warning, not an error, so don't return error status 1185 //return NO_INIT; 1186 } 1187 } 1188 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1189 if (trackFlags & IAudioFlinger::TRACK_DIRECT) { 1190 ALOGV("AUDIO_OUTPUT_FLAG_DIRECT successful"); 1191 } else { 1192 ALOGW("AUDIO_OUTPUT_FLAG_DIRECT denied by server"); 1193 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT); 1194 // FIXME This is a warning, not an error, so don't return error status 1195 //return NO_INIT; 1196 } 1197 } 1198 1199 // We retain a copy of the I/O handle, but don't own the reference 1200 mOutput = output; 1201 mRefreshRemaining = true; 1202 1203 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1204 // is the value of pointer() for the shared buffer, otherwise buffers points 1205 // immediately after the control block. This address is for the mapping within client 1206 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1207 void* buffers; 1208 if (mSharedBuffer == 0) { 1209 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1210 } else { 1211 buffers = mSharedBuffer->pointer(); 1212 } 1213 1214 mAudioTrack->attachAuxEffect(mAuxEffectId); 1215 // FIXME don't believe this lie 1216 mLatency = afLatency + (1000*frameCount) / mSampleRate; 1217 1218 mFrameCount = frameCount; 1219 // If IAudioTrack is re-created, don't let the requested frameCount 1220 // decrease. This can confuse clients that cache frameCount(). 1221 if (frameCount > mReqFrameCount) { 1222 mReqFrameCount = frameCount; 1223 } 1224 1225 // update proxy 1226 if (mSharedBuffer == 0) { 1227 mStaticProxy.clear(); 1228 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1229 } else { 1230 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1231 mProxy = mStaticProxy; 1232 } 1233 1234 mProxy->setVolumeLR(gain_minifloat_pack( 1235 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]), 1236 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT]))); 1237 1238 mProxy->setSendLevel(mSendLevel); 1239 mProxy->setSampleRate(mSampleRate); 1240 mProxy->setMinimum(mNotificationFramesAct); 1241 1242 mDeathNotifier = new DeathNotifier(this); 1243 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1244 1245 return NO_ERROR; 1246 } 1247 1248release: 1249 AudioSystem::releaseOutput(output, streamType, (audio_session_t)mSessionId); 1250 if (status == NO_ERROR) { 1251 status = NO_INIT; 1252 } 1253 return status; 1254} 1255 1256status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1257{ 1258 if (audioBuffer == NULL) { 1259 return BAD_VALUE; 1260 } 1261 if (mTransfer != TRANSFER_OBTAIN) { 1262 audioBuffer->frameCount = 0; 1263 audioBuffer->size = 0; 1264 audioBuffer->raw = NULL; 1265 return INVALID_OPERATION; 1266 } 1267 1268 const struct timespec *requested; 1269 struct timespec timeout; 1270 if (waitCount == -1) { 1271 requested = &ClientProxy::kForever; 1272 } else if (waitCount == 0) { 1273 requested = &ClientProxy::kNonBlocking; 1274 } else if (waitCount > 0) { 1275 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1276 timeout.tv_sec = ms / 1000; 1277 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1278 requested = &timeout; 1279 } else { 1280 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1281 requested = NULL; 1282 } 1283 return obtainBuffer(audioBuffer, requested); 1284} 1285 1286status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1287 struct timespec *elapsed, size_t *nonContig) 1288{ 1289 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1290 uint32_t oldSequence = 0; 1291 uint32_t newSequence; 1292 1293 Proxy::Buffer buffer; 1294 status_t status = NO_ERROR; 1295 1296 static const int32_t kMaxTries = 5; 1297 int32_t tryCounter = kMaxTries; 1298 1299 do { 1300 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1301 // keep them from going away if another thread re-creates the track during obtainBuffer() 1302 sp<AudioTrackClientProxy> proxy; 1303 sp<IMemory> iMem; 1304 1305 { // start of lock scope 1306 AutoMutex lock(mLock); 1307 1308 newSequence = mSequence; 1309 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1310 if (status == DEAD_OBJECT) { 1311 // re-create track, unless someone else has already done so 1312 if (newSequence == oldSequence) { 1313 status = restoreTrack_l("obtainBuffer"); 1314 if (status != NO_ERROR) { 1315 buffer.mFrameCount = 0; 1316 buffer.mRaw = NULL; 1317 buffer.mNonContig = 0; 1318 break; 1319 } 1320 } 1321 } 1322 oldSequence = newSequence; 1323 1324 // Keep the extra references 1325 proxy = mProxy; 1326 iMem = mCblkMemory; 1327 1328 if (mState == STATE_STOPPING) { 1329 status = -EINTR; 1330 buffer.mFrameCount = 0; 1331 buffer.mRaw = NULL; 1332 buffer.mNonContig = 0; 1333 break; 1334 } 1335 1336 // Non-blocking if track is stopped or paused 1337 if (mState != STATE_ACTIVE) { 1338 requested = &ClientProxy::kNonBlocking; 1339 } 1340 1341 } // end of lock scope 1342 1343 buffer.mFrameCount = audioBuffer->frameCount; 1344 // FIXME starts the requested timeout and elapsed over from scratch 1345 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1346 1347 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1348 1349 audioBuffer->frameCount = buffer.mFrameCount; 1350 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1351 audioBuffer->raw = buffer.mRaw; 1352 if (nonContig != NULL) { 1353 *nonContig = buffer.mNonContig; 1354 } 1355 return status; 1356} 1357 1358void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1359{ 1360 if (mTransfer == TRANSFER_SHARED) { 1361 return; 1362 } 1363 1364 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1365 if (stepCount == 0) { 1366 return; 1367 } 1368 1369 Proxy::Buffer buffer; 1370 buffer.mFrameCount = stepCount; 1371 buffer.mRaw = audioBuffer->raw; 1372 1373 AutoMutex lock(mLock); 1374 mReleased += stepCount; 1375 mInUnderrun = false; 1376 mProxy->releaseBuffer(&buffer); 1377 1378 // restart track if it was disabled by audioflinger due to previous underrun 1379 if (mState == STATE_ACTIVE) { 1380 audio_track_cblk_t* cblk = mCblk; 1381 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1382 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1383 // FIXME ignoring status 1384 mAudioTrack->start(); 1385 } 1386 } 1387} 1388 1389// ------------------------------------------------------------------------- 1390 1391ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1392{ 1393 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1394 return INVALID_OPERATION; 1395 } 1396 1397 if (isDirect()) { 1398 AutoMutex lock(mLock); 1399 int32_t flags = android_atomic_and( 1400 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), 1401 &mCblk->mFlags); 1402 if (flags & CBLK_INVALID) { 1403 return DEAD_OBJECT; 1404 } 1405 } 1406 1407 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1408 // Sanity-check: user is most-likely passing an error code, and it would 1409 // make the return value ambiguous (actualSize vs error). 1410 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1411 return BAD_VALUE; 1412 } 1413 1414 size_t written = 0; 1415 Buffer audioBuffer; 1416 1417 while (userSize >= mFrameSize) { 1418 audioBuffer.frameCount = userSize / mFrameSize; 1419 1420 status_t err = obtainBuffer(&audioBuffer, 1421 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1422 if (err < 0) { 1423 if (written > 0) { 1424 break; 1425 } 1426 return ssize_t(err); 1427 } 1428 1429 size_t toWrite; 1430 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1431 // Divide capacity by 2 to take expansion into account 1432 toWrite = audioBuffer.size >> 1; 1433 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1434 } else { 1435 toWrite = audioBuffer.size; 1436 memcpy(audioBuffer.i8, buffer, toWrite); 1437 } 1438 buffer = ((const char *) buffer) + toWrite; 1439 userSize -= toWrite; 1440 written += toWrite; 1441 1442 releaseBuffer(&audioBuffer); 1443 } 1444 1445 return written; 1446} 1447 1448// ------------------------------------------------------------------------- 1449 1450TimedAudioTrack::TimedAudioTrack() { 1451 mIsTimed = true; 1452} 1453 1454status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1455{ 1456 AutoMutex lock(mLock); 1457 status_t result = UNKNOWN_ERROR; 1458 1459#if 1 1460 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1461 // while we are accessing the cblk 1462 sp<IAudioTrack> audioTrack = mAudioTrack; 1463 sp<IMemory> iMem = mCblkMemory; 1464#endif 1465 1466 // If the track is not invalid already, try to allocate a buffer. alloc 1467 // fails indicating that the server is dead, flag the track as invalid so 1468 // we can attempt to restore in just a bit. 1469 audio_track_cblk_t* cblk = mCblk; 1470 if (!(cblk->mFlags & CBLK_INVALID)) { 1471 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1472 if (result == DEAD_OBJECT) { 1473 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1474 } 1475 } 1476 1477 // If the track is invalid at this point, attempt to restore it. and try the 1478 // allocation one more time. 1479 if (cblk->mFlags & CBLK_INVALID) { 1480 result = restoreTrack_l("allocateTimedBuffer"); 1481 1482 if (result == NO_ERROR) { 1483 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1484 } 1485 } 1486 1487 return result; 1488} 1489 1490status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1491 int64_t pts) 1492{ 1493 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1494 { 1495 AutoMutex lock(mLock); 1496 audio_track_cblk_t* cblk = mCblk; 1497 // restart track if it was disabled by audioflinger due to previous underrun 1498 if (buffer->size() != 0 && status == NO_ERROR && 1499 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1500 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1501 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1502 // FIXME ignoring status 1503 mAudioTrack->start(); 1504 } 1505 } 1506 return status; 1507} 1508 1509status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1510 TargetTimeline target) 1511{ 1512 return mAudioTrack->setMediaTimeTransform(xform, target); 1513} 1514 1515// ------------------------------------------------------------------------- 1516 1517nsecs_t AudioTrack::processAudioBuffer() 1518{ 1519 // Currently the AudioTrack thread is not created if there are no callbacks. 1520 // Would it ever make sense to run the thread, even without callbacks? 1521 // If so, then replace this by checks at each use for mCbf != NULL. 1522 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1523 1524 mLock.lock(); 1525 if (mAwaitBoost) { 1526 mAwaitBoost = false; 1527 mLock.unlock(); 1528 static const int32_t kMaxTries = 5; 1529 int32_t tryCounter = kMaxTries; 1530 uint32_t pollUs = 10000; 1531 do { 1532 int policy = sched_getscheduler(0); 1533 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1534 break; 1535 } 1536 usleep(pollUs); 1537 pollUs <<= 1; 1538 } while (tryCounter-- > 0); 1539 if (tryCounter < 0) { 1540 ALOGE("did not receive expected priority boost on time"); 1541 } 1542 // Run again immediately 1543 return 0; 1544 } 1545 1546 // Can only reference mCblk while locked 1547 int32_t flags = android_atomic_and( 1548 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1549 1550 // Check for track invalidation 1551 if (flags & CBLK_INVALID) { 1552 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1553 // AudioSystem cache. We should not exit here but after calling the callback so 1554 // that the upper layers can recreate the track 1555 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) { 1556 status_t status = restoreTrack_l("processAudioBuffer"); 1557 mLock.unlock(); 1558 // Run again immediately, but with a new IAudioTrack 1559 return 0; 1560 } 1561 } 1562 1563 bool waitStreamEnd = mState == STATE_STOPPING; 1564 bool active = mState == STATE_ACTIVE; 1565 1566 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1567 bool newUnderrun = false; 1568 if (flags & CBLK_UNDERRUN) { 1569#if 0 1570 // Currently in shared buffer mode, when the server reaches the end of buffer, 1571 // the track stays active in continuous underrun state. It's up to the application 1572 // to pause or stop the track, or set the position to a new offset within buffer. 1573 // This was some experimental code to auto-pause on underrun. Keeping it here 1574 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1575 if (mTransfer == TRANSFER_SHARED) { 1576 mState = STATE_PAUSED; 1577 active = false; 1578 } 1579#endif 1580 if (!mInUnderrun) { 1581 mInUnderrun = true; 1582 newUnderrun = true; 1583 } 1584 } 1585 1586 // Get current position of server 1587 size_t position = updateAndGetPosition_l(); 1588 1589 // Manage marker callback 1590 bool markerReached = false; 1591 size_t markerPosition = mMarkerPosition; 1592 // FIXME fails for wraparound, need 64 bits 1593 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1594 mMarkerReached = markerReached = true; 1595 } 1596 1597 // Determine number of new position callback(s) that will be needed, while locked 1598 size_t newPosCount = 0; 1599 size_t newPosition = mNewPosition; 1600 size_t updatePeriod = mUpdatePeriod; 1601 // FIXME fails for wraparound, need 64 bits 1602 if (updatePeriod > 0 && position >= newPosition) { 1603 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1604 mNewPosition += updatePeriod * newPosCount; 1605 } 1606 1607 // Cache other fields that will be needed soon 1608 uint32_t loopPeriod = mLoopPeriod; 1609 uint32_t sampleRate = mSampleRate; 1610 uint32_t notificationFrames = mNotificationFramesAct; 1611 if (mRefreshRemaining) { 1612 mRefreshRemaining = false; 1613 mRemainingFrames = notificationFrames; 1614 mRetryOnPartialBuffer = false; 1615 } 1616 size_t misalignment = mProxy->getMisalignment(); 1617 uint32_t sequence = mSequence; 1618 sp<AudioTrackClientProxy> proxy = mProxy; 1619 1620 // These fields don't need to be cached, because they are assigned only by set(): 1621 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1622 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1623 1624 mLock.unlock(); 1625 1626 if (waitStreamEnd) { 1627 struct timespec timeout; 1628 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1629 timeout.tv_nsec = 0; 1630 1631 status_t status = proxy->waitStreamEndDone(&timeout); 1632 switch (status) { 1633 case NO_ERROR: 1634 case DEAD_OBJECT: 1635 case TIMED_OUT: 1636 mCbf(EVENT_STREAM_END, mUserData, NULL); 1637 { 1638 AutoMutex lock(mLock); 1639 // The previously assigned value of waitStreamEnd is no longer valid, 1640 // since the mutex has been unlocked and either the callback handler 1641 // or another thread could have re-started the AudioTrack during that time. 1642 waitStreamEnd = mState == STATE_STOPPING; 1643 if (waitStreamEnd) { 1644 mState = STATE_STOPPED; 1645 mReleased = 0; 1646 } 1647 } 1648 if (waitStreamEnd && status != DEAD_OBJECT) { 1649 return NS_INACTIVE; 1650 } 1651 break; 1652 } 1653 return 0; 1654 } 1655 1656 // perform callbacks while unlocked 1657 if (newUnderrun) { 1658 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1659 } 1660 // FIXME we will miss loops if loop cycle was signaled several times since last call 1661 // to processAudioBuffer() 1662 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1663 mCbf(EVENT_LOOP_END, mUserData, NULL); 1664 } 1665 if (flags & CBLK_BUFFER_END) { 1666 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1667 } 1668 if (markerReached) { 1669 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1670 } 1671 while (newPosCount > 0) { 1672 size_t temp = newPosition; 1673 mCbf(EVENT_NEW_POS, mUserData, &temp); 1674 newPosition += updatePeriod; 1675 newPosCount--; 1676 } 1677 1678 if (mObservedSequence != sequence) { 1679 mObservedSequence = sequence; 1680 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1681 // for offloaded tracks, just wait for the upper layers to recreate the track 1682 if (isOffloadedOrDirect()) { 1683 return NS_INACTIVE; 1684 } 1685 } 1686 1687 // if inactive, then don't run me again until re-started 1688 if (!active) { 1689 return NS_INACTIVE; 1690 } 1691 1692 // Compute the estimated time until the next timed event (position, markers, loops) 1693 // FIXME only for non-compressed audio 1694 uint32_t minFrames = ~0; 1695 if (!markerReached && position < markerPosition) { 1696 minFrames = markerPosition - position; 1697 } 1698 if (loopPeriod > 0 && loopPeriod < minFrames) { 1699 minFrames = loopPeriod; 1700 } 1701 if (updatePeriod > 0 && updatePeriod < minFrames) { 1702 minFrames = updatePeriod; 1703 } 1704 1705 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1706 static const uint32_t kPoll = 0; 1707 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1708 minFrames = kPoll * notificationFrames; 1709 } 1710 1711 // Convert frame units to time units 1712 nsecs_t ns = NS_WHENEVER; 1713 if (minFrames != (uint32_t) ~0) { 1714 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1715 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1716 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1717 } 1718 1719 // If not supplying data by EVENT_MORE_DATA, then we're done 1720 if (mTransfer != TRANSFER_CALLBACK) { 1721 return ns; 1722 } 1723 1724 struct timespec timeout; 1725 const struct timespec *requested = &ClientProxy::kForever; 1726 if (ns != NS_WHENEVER) { 1727 timeout.tv_sec = ns / 1000000000LL; 1728 timeout.tv_nsec = ns % 1000000000LL; 1729 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1730 requested = &timeout; 1731 } 1732 1733 while (mRemainingFrames > 0) { 1734 1735 Buffer audioBuffer; 1736 audioBuffer.frameCount = mRemainingFrames; 1737 size_t nonContig; 1738 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1739 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1740 "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount); 1741 requested = &ClientProxy::kNonBlocking; 1742 size_t avail = audioBuffer.frameCount + nonContig; 1743 ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d", 1744 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1745 if (err != NO_ERROR) { 1746 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1747 (isOffloaded() && (err == DEAD_OBJECT))) { 1748 return 0; 1749 } 1750 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1751 return NS_NEVER; 1752 } 1753 1754 if (mRetryOnPartialBuffer && !isOffloaded()) { 1755 mRetryOnPartialBuffer = false; 1756 if (avail < mRemainingFrames) { 1757 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1758 if (ns < 0 || myns < ns) { 1759 ns = myns; 1760 } 1761 return ns; 1762 } 1763 } 1764 1765 // Divide buffer size by 2 to take into account the expansion 1766 // due to 8 to 16 bit conversion: the callback must fill only half 1767 // of the destination buffer 1768 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1769 audioBuffer.size >>= 1; 1770 } 1771 1772 size_t reqSize = audioBuffer.size; 1773 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1774 size_t writtenSize = audioBuffer.size; 1775 1776 // Sanity check on returned size 1777 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1778 ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes", 1779 reqSize, ssize_t(writtenSize)); 1780 return NS_NEVER; 1781 } 1782 1783 if (writtenSize == 0) { 1784 // The callback is done filling buffers 1785 // Keep this thread going to handle timed events and 1786 // still try to get more data in intervals of WAIT_PERIOD_MS 1787 // but don't just loop and block the CPU, so wait 1788 return WAIT_PERIOD_MS * 1000000LL; 1789 } 1790 1791 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1792 // 8 to 16 bit conversion, note that source and destination are the same address 1793 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1794 audioBuffer.size <<= 1; 1795 } 1796 1797 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1798 audioBuffer.frameCount = releasedFrames; 1799 mRemainingFrames -= releasedFrames; 1800 if (misalignment >= releasedFrames) { 1801 misalignment -= releasedFrames; 1802 } else { 1803 misalignment = 0; 1804 } 1805 1806 releaseBuffer(&audioBuffer); 1807 1808 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1809 // if callback doesn't like to accept the full chunk 1810 if (writtenSize < reqSize) { 1811 continue; 1812 } 1813 1814 // There could be enough non-contiguous frames available to satisfy the remaining request 1815 if (mRemainingFrames <= nonContig) { 1816 continue; 1817 } 1818 1819#if 0 1820 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1821 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1822 // that total to a sum == notificationFrames. 1823 if (0 < misalignment && misalignment <= mRemainingFrames) { 1824 mRemainingFrames = misalignment; 1825 return (mRemainingFrames * 1100000000LL) / sampleRate; 1826 } 1827#endif 1828 1829 } 1830 mRemainingFrames = notificationFrames; 1831 mRetryOnPartialBuffer = true; 1832 1833 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1834 return 0; 1835} 1836 1837status_t AudioTrack::restoreTrack_l(const char *from) 1838{ 1839 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1840 isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from); 1841 ++mSequence; 1842 status_t result; 1843 1844 // refresh the audio configuration cache in this process to make sure we get new 1845 // output parameters and new IAudioFlinger in createTrack_l() 1846 AudioSystem::clearAudioConfigCache(); 1847 1848 if (isOffloadedOrDirect_l()) { 1849 // FIXME re-creation of offloaded tracks is not yet implemented 1850 return DEAD_OBJECT; 1851 } 1852 1853 // save the old static buffer position 1854 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1855 1856 // If a new IAudioTrack is successfully created, createTrack_l() will modify the 1857 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1858 // It will also delete the strong references on previous IAudioTrack and IMemory. 1859 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact. 1860 result = createTrack_l(); 1861 1862 // take the frames that will be lost by track recreation into account in saved position 1863 (void) updateAndGetPosition_l(); 1864 mPosition = mReleased; 1865 1866 if (result == NO_ERROR) { 1867 // continue playback from last known position, but 1868 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1869 if (mStaticProxy != NULL) { 1870 mLoopPeriod = 0; 1871 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1872 } 1873 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1874 // track destruction have been played? This is critical for SoundPool implementation 1875 // This must be broken, and needs to be tested/debugged. 1876#if 0 1877 // restore write index and set other indexes to reflect empty buffer status 1878 if (!strcmp(from, "start")) { 1879 // Make sure that a client relying on callback events indicating underrun or 1880 // the actual amount of audio frames played (e.g SoundPool) receives them. 1881 if (mSharedBuffer == 0) { 1882 // restart playback even if buffer is not completely filled. 1883 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1884 } 1885 } 1886#endif 1887 if (mState == STATE_ACTIVE) { 1888 result = mAudioTrack->start(); 1889 } 1890 } 1891 if (result != NO_ERROR) { 1892 ALOGW("restoreTrack_l() failed status %d", result); 1893 mState = STATE_STOPPED; 1894 mReleased = 0; 1895 } 1896 1897 return result; 1898} 1899 1900uint32_t AudioTrack::updateAndGetPosition_l() 1901{ 1902 // This is the sole place to read server consumed frames 1903 uint32_t newServer = mProxy->getPosition(); 1904 int32_t delta = newServer - mServer; 1905 mServer = newServer; 1906 // TODO There is controversy about whether there can be "negative jitter" in server position. 1907 // This should be investigated further, and if possible, it should be addressed. 1908 // A more definite failure mode is infrequent polling by client. 1909 // One could call (void)getPosition_l() in releaseBuffer(), 1910 // so mReleased and mPosition are always lock-step as best possible. 1911 // That should ensure delta never goes negative for infrequent polling 1912 // unless the server has more than 2^31 frames in its buffer, 1913 // in which case the use of uint32_t for these counters has bigger issues. 1914 if (delta < 0) { 1915 ALOGE("detected illegal retrograde motion by the server: mServer advanced by %d", delta); 1916 delta = 0; 1917 } 1918 return mPosition += (uint32_t) delta; 1919} 1920 1921status_t AudioTrack::setParameters(const String8& keyValuePairs) 1922{ 1923 AutoMutex lock(mLock); 1924 return mAudioTrack->setParameters(keyValuePairs); 1925} 1926 1927status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1928{ 1929 AutoMutex lock(mLock); 1930 // FIXME not implemented for fast tracks; should use proxy and SSQ 1931 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1932 return INVALID_OPERATION; 1933 } 1934 1935 switch (mState) { 1936 case STATE_ACTIVE: 1937 case STATE_PAUSED: 1938 break; // handle below 1939 case STATE_FLUSHED: 1940 case STATE_STOPPED: 1941 return WOULD_BLOCK; 1942 case STATE_STOPPING: 1943 case STATE_PAUSED_STOPPING: 1944 if (!isOffloaded_l()) { 1945 return INVALID_OPERATION; 1946 } 1947 break; // offloaded tracks handled below 1948 default: 1949 LOG_ALWAYS_FATAL("Invalid mState in getTimestamp(): %d", mState); 1950 break; 1951 } 1952 1953 if (mCblk->mFlags & CBLK_INVALID) { 1954 restoreTrack_l("getTimestamp"); 1955 } 1956 1957 // The presented frame count must always lag behind the consumed frame count. 1958 // To avoid a race, read the presented frames first. This ensures that presented <= consumed. 1959 status_t status = mAudioTrack->getTimestamp(timestamp); 1960 if (status != NO_ERROR) { 1961 ALOGV_IF(status != WOULD_BLOCK, "getTimestamp error:%#x", status); 1962 return status; 1963 } 1964 if (isOffloadedOrDirect_l()) { 1965 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) { 1966 // use cached paused position in case another offloaded track is running. 1967 timestamp.mPosition = mPausedPosition; 1968 clock_gettime(CLOCK_MONOTONIC, ×tamp.mTime); 1969 return NO_ERROR; 1970 } 1971 1972 // Check whether a pending flush or stop has completed, as those commands may 1973 // be asynchronous or return near finish. 1974 if (mStartUs != 0 && mSampleRate != 0) { 1975 static const int kTimeJitterUs = 100000; // 100 ms 1976 static const int k1SecUs = 1000000; 1977 1978 const int64_t timeNow = getNowUs(); 1979 1980 if (timeNow < mStartUs + k1SecUs) { // within first second of starting 1981 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime); 1982 if (timestampTimeUs < mStartUs) { 1983 return WOULD_BLOCK; // stale timestamp time, occurs before start. 1984 } 1985 const int64_t deltaTimeUs = timestampTimeUs - mStartUs; 1986 const int64_t deltaPositionByUs = timestamp.mPosition * 1000000LL / mSampleRate; 1987 1988 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) { 1989 // Verify that the counter can't count faster than the sample rate 1990 // since the start time. If greater, then that means we have failed 1991 // to completely flush or stop the previous playing track. 1992 ALOGW("incomplete flush or stop:" 1993 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)", 1994 (long long)deltaTimeUs, (long long)deltaPositionByUs, 1995 timestamp.mPosition); 1996 return WOULD_BLOCK; 1997 } 1998 } 1999 mStartUs = 0; // no need to check again, start timestamp has either expired or unneeded. 2000 } 2001 } else { 2002 // Update the mapping between local consumed (mPosition) and server consumed (mServer) 2003 (void) updateAndGetPosition_l(); 2004 // Server consumed (mServer) and presented both use the same server time base, 2005 // and server consumed is always >= presented. 2006 // The delta between these represents the number of frames in the buffer pipeline. 2007 // If this delta between these is greater than the client position, it means that 2008 // actually presented is still stuck at the starting line (figuratively speaking), 2009 // waiting for the first frame to go by. So we can't report a valid timestamp yet. 2010 if ((uint32_t) (mServer - timestamp.mPosition) > mPosition) { 2011 return INVALID_OPERATION; 2012 } 2013 // Convert timestamp position from server time base to client time base. 2014 // TODO The following code should work OK now because timestamp.mPosition is 32-bit. 2015 // But if we change it to 64-bit then this could fail. 2016 // If (mPosition - mServer) can be negative then should use: 2017 // (int32_t)(mPosition - mServer) 2018 timestamp.mPosition += mPosition - mServer; 2019 // Immediately after a call to getPosition_l(), mPosition and 2020 // mServer both represent the same frame position. mPosition is 2021 // in client's point of view, and mServer is in server's point of 2022 // view. So the difference between them is the "fudge factor" 2023 // between client and server views due to stop() and/or new 2024 // IAudioTrack. And timestamp.mPosition is initially in server's 2025 // point of view, so we need to apply the same fudge factor to it. 2026 } 2027 return status; 2028} 2029 2030String8 AudioTrack::getParameters(const String8& keys) 2031{ 2032 audio_io_handle_t output = getOutput(); 2033 if (output != AUDIO_IO_HANDLE_NONE) { 2034 return AudioSystem::getParameters(output, keys); 2035 } else { 2036 return String8::empty(); 2037 } 2038} 2039 2040bool AudioTrack::isOffloaded() const 2041{ 2042 AutoMutex lock(mLock); 2043 return isOffloaded_l(); 2044} 2045 2046bool AudioTrack::isDirect() const 2047{ 2048 AutoMutex lock(mLock); 2049 return isDirect_l(); 2050} 2051 2052bool AudioTrack::isOffloadedOrDirect() const 2053{ 2054 AutoMutex lock(mLock); 2055 return isOffloadedOrDirect_l(); 2056} 2057 2058 2059status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 2060{ 2061 2062 const size_t SIZE = 256; 2063 char buffer[SIZE]; 2064 String8 result; 2065 2066 result.append(" AudioTrack::dump\n"); 2067 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 2068 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 2069 result.append(buffer); 2070 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 2071 mChannelCount, mFrameCount); 2072 result.append(buffer); 2073 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 2074 result.append(buffer); 2075 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 2076 result.append(buffer); 2077 ::write(fd, result.string(), result.size()); 2078 return NO_ERROR; 2079} 2080 2081uint32_t AudioTrack::getUnderrunFrames() const 2082{ 2083 AutoMutex lock(mLock); 2084 return mProxy->getUnderrunFrames(); 2085} 2086 2087// ========================================================================= 2088 2089void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 2090{ 2091 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 2092 if (audioTrack != 0) { 2093 AutoMutex lock(audioTrack->mLock); 2094 audioTrack->mProxy->binderDied(); 2095 } 2096} 2097 2098// ========================================================================= 2099 2100AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 2101 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 2102 mIgnoreNextPausedInt(false) 2103{ 2104} 2105 2106AudioTrack::AudioTrackThread::~AudioTrackThread() 2107{ 2108} 2109 2110bool AudioTrack::AudioTrackThread::threadLoop() 2111{ 2112 { 2113 AutoMutex _l(mMyLock); 2114 if (mPaused) { 2115 mMyCond.wait(mMyLock); 2116 // caller will check for exitPending() 2117 return true; 2118 } 2119 if (mIgnoreNextPausedInt) { 2120 mIgnoreNextPausedInt = false; 2121 mPausedInt = false; 2122 } 2123 if (mPausedInt) { 2124 if (mPausedNs > 0) { 2125 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 2126 } else { 2127 mMyCond.wait(mMyLock); 2128 } 2129 mPausedInt = false; 2130 return true; 2131 } 2132 } 2133 if (exitPending()) { 2134 return false; 2135 } 2136 nsecs_t ns = mReceiver.processAudioBuffer(); 2137 switch (ns) { 2138 case 0: 2139 return true; 2140 case NS_INACTIVE: 2141 pauseInternal(); 2142 return true; 2143 case NS_NEVER: 2144 return false; 2145 case NS_WHENEVER: 2146 // FIXME increase poll interval, or make event-driven 2147 ns = 1000000000LL; 2148 // fall through 2149 default: 2150 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns); 2151 pauseInternal(ns); 2152 return true; 2153 } 2154} 2155 2156void AudioTrack::AudioTrackThread::requestExit() 2157{ 2158 // must be in this order to avoid a race condition 2159 Thread::requestExit(); 2160 resume(); 2161} 2162 2163void AudioTrack::AudioTrackThread::pause() 2164{ 2165 AutoMutex _l(mMyLock); 2166 mPaused = true; 2167} 2168 2169void AudioTrack::AudioTrackThread::resume() 2170{ 2171 AutoMutex _l(mMyLock); 2172 mIgnoreNextPausedInt = true; 2173 if (mPaused || mPausedInt) { 2174 mPaused = false; 2175 mPausedInt = false; 2176 mMyCond.signal(); 2177 } 2178} 2179 2180void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 2181{ 2182 AutoMutex _l(mMyLock); 2183 mPausedInt = true; 2184 mPausedNs = ns; 2185} 2186 2187}; // namespace android 2188