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