AudioTrack.cpp revision 7f1bc8af1c46695191bf7e2aba6467f3616629c0
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18//#define LOG_NDEBUG 0 19#define LOG_TAG "AudioTrack" 20 21#include <inttypes.h> 22#include <math.h> 23#include <sys/resource.h> 24 25#include <audio_utils/primitives.h> 26#include <binder/IPCThreadState.h> 27#include <media/AudioTrack.h> 28#include <utils/Log.h> 29#include <private/media/AudioTrackShared.h> 30#include <media/IAudioFlinger.h> 31#include <media/AudioResamplerPublic.h> 32 33#define WAIT_PERIOD_MS 10 34#define WAIT_STREAM_END_TIMEOUT_SEC 120 35 36 37namespace android { 38// --------------------------------------------------------------------------- 39 40static int64_t convertTimespecToUs(const struct timespec &tv) 41{ 42 return tv.tv_sec * 1000000ll + tv.tv_nsec / 1000; 43} 44 45// current monotonic time in microseconds. 46static int64_t getNowUs() 47{ 48 struct timespec tv; 49 (void) clock_gettime(CLOCK_MONOTONIC, &tv); 50 return convertTimespecToUs(tv); 51} 52 53// static 54status_t AudioTrack::getMinFrameCount( 55 size_t* frameCount, 56 audio_stream_type_t streamType, 57 uint32_t sampleRate) 58{ 59 if (frameCount == NULL) { 60 return BAD_VALUE; 61 } 62 63 // FIXME merge with similar code in createTrack_l(), except we're missing 64 // some information here that is available in createTrack_l(): 65 // audio_io_handle_t output 66 // audio_format_t format 67 // audio_channel_mask_t channelMask 68 // audio_output_flags_t flags 69 uint32_t afSampleRate; 70 status_t status; 71 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 72 if (status != NO_ERROR) { 73 ALOGE("Unable to query output sample rate for stream type %d; status %d", 74 streamType, status); 75 return status; 76 } 77 size_t afFrameCount; 78 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 79 if (status != NO_ERROR) { 80 ALOGE("Unable to query output frame count for stream type %d; status %d", 81 streamType, status); 82 return status; 83 } 84 uint32_t afLatency; 85 status = AudioSystem::getOutputLatency(&afLatency, streamType); 86 if (status != NO_ERROR) { 87 ALOGE("Unable to query output latency for stream type %d; status %d", 88 streamType, status); 89 return status; 90 } 91 92 // Ensure that buffer depth covers at least audio hardware latency 93 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 94 if (minBufCount < 2) { 95 minBufCount = 2; 96 } 97 98 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 99 afFrameCount * minBufCount * uint64_t(sampleRate) / afSampleRate; 100 // The formula above should always produce a non-zero value, but return an error 101 // in the unlikely event that it does not, as that's part of the API contract. 102 if (*frameCount == 0) { 103 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", 104 streamType, sampleRate); 105 return BAD_VALUE; 106 } 107 ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%d, afSampleRate=%d, afLatency=%d", 108 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 109 return NO_ERROR; 110} 111 112// --------------------------------------------------------------------------- 113 114AudioTrack::AudioTrack() 115 : mStatus(NO_INIT), 116 mIsTimed(false), 117 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 118 mPreviousSchedulingGroup(SP_DEFAULT), 119 mPausedPosition(0) 120{ 121 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN; 122 mAttributes.usage = AUDIO_USAGE_UNKNOWN; 123 mAttributes.flags = 0x0; 124 strcpy(mAttributes.tags, ""); 125} 126 127AudioTrack::AudioTrack( 128 audio_stream_type_t streamType, 129 uint32_t sampleRate, 130 audio_format_t format, 131 audio_channel_mask_t channelMask, 132 size_t frameCount, 133 audio_output_flags_t flags, 134 callback_t cbf, 135 void* user, 136 uint32_t notificationFrames, 137 int sessionId, 138 transfer_type transferType, 139 const audio_offload_info_t *offloadInfo, 140 int uid, 141 pid_t pid, 142 const audio_attributes_t* pAttributes) 143 : mStatus(NO_INIT), 144 mIsTimed(false), 145 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 146 mPreviousSchedulingGroup(SP_DEFAULT), 147 mPausedPosition(0) 148{ 149 mStatus = set(streamType, sampleRate, format, channelMask, 150 frameCount, flags, cbf, user, notificationFrames, 151 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 152 offloadInfo, uid, pid, pAttributes); 153} 154 155AudioTrack::AudioTrack( 156 audio_stream_type_t streamType, 157 uint32_t sampleRate, 158 audio_format_t format, 159 audio_channel_mask_t channelMask, 160 const sp<IMemory>& sharedBuffer, 161 audio_output_flags_t flags, 162 callback_t cbf, 163 void* user, 164 uint32_t notificationFrames, 165 int sessionId, 166 transfer_type transferType, 167 const audio_offload_info_t *offloadInfo, 168 int uid, 169 pid_t pid, 170 const audio_attributes_t* pAttributes) 171 : mStatus(NO_INIT), 172 mIsTimed(false), 173 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 174 mPreviousSchedulingGroup(SP_DEFAULT), 175 mPausedPosition(0) 176{ 177 mStatus = set(streamType, sampleRate, format, channelMask, 178 0 /*frameCount*/, flags, cbf, user, notificationFrames, 179 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, 180 uid, pid, pAttributes); 181} 182 183AudioTrack::~AudioTrack() 184{ 185 if (mStatus == NO_ERROR) { 186 // Make sure that callback function exits in the case where 187 // it is looping on buffer full condition in obtainBuffer(). 188 // Otherwise the callback thread will never exit. 189 stop(); 190 if (mAudioTrackThread != 0) { 191 mProxy->interrupt(); 192 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 193 mAudioTrackThread->requestExitAndWait(); 194 mAudioTrackThread.clear(); 195 } 196 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 197 mAudioTrack.clear(); 198 mCblkMemory.clear(); 199 mSharedBuffer.clear(); 200 IPCThreadState::self()->flushCommands(); 201 ALOGV("~AudioTrack, releasing session id from %d on behalf of %d", 202 IPCThreadState::self()->getCallingPid(), mClientPid); 203 AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); 204 } 205} 206 207status_t AudioTrack::set( 208 audio_stream_type_t streamType, 209 uint32_t sampleRate, 210 audio_format_t format, 211 audio_channel_mask_t channelMask, 212 size_t frameCount, 213 audio_output_flags_t flags, 214 callback_t cbf, 215 void* user, 216 uint32_t notificationFrames, 217 const sp<IMemory>& sharedBuffer, 218 bool threadCanCallJava, 219 int sessionId, 220 transfer_type transferType, 221 const audio_offload_info_t *offloadInfo, 222 int uid, 223 pid_t pid, 224 const audio_attributes_t* pAttributes) 225{ 226 ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 227 "flags #%x, notificationFrames %u, sessionId %d, transferType %d", 228 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, 229 sessionId, transferType); 230 231 switch (transferType) { 232 case TRANSFER_DEFAULT: 233 if (sharedBuffer != 0) { 234 transferType = TRANSFER_SHARED; 235 } else if (cbf == NULL || threadCanCallJava) { 236 transferType = TRANSFER_SYNC; 237 } else { 238 transferType = TRANSFER_CALLBACK; 239 } 240 break; 241 case TRANSFER_CALLBACK: 242 if (cbf == NULL || sharedBuffer != 0) { 243 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 244 return BAD_VALUE; 245 } 246 break; 247 case TRANSFER_OBTAIN: 248 case TRANSFER_SYNC: 249 if (sharedBuffer != 0) { 250 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 251 return BAD_VALUE; 252 } 253 break; 254 case TRANSFER_SHARED: 255 if (sharedBuffer == 0) { 256 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 257 return BAD_VALUE; 258 } 259 break; 260 default: 261 ALOGE("Invalid transfer type %d", transferType); 262 return BAD_VALUE; 263 } 264 mSharedBuffer = sharedBuffer; 265 mTransfer = transferType; 266 267 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 268 sharedBuffer->size()); 269 270 ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags); 271 272 AutoMutex lock(mLock); 273 274 // invariant that mAudioTrack != 0 is true only after set() returns successfully 275 if (mAudioTrack != 0) { 276 ALOGE("Track already in use"); 277 return INVALID_OPERATION; 278 } 279 280 // handle default values first. 281 if (streamType == AUDIO_STREAM_DEFAULT) { 282 streamType = AUDIO_STREAM_MUSIC; 283 } 284 285 if (pAttributes == NULL) { 286 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 287 ALOGE("Invalid stream type %d", streamType); 288 return BAD_VALUE; 289 } 290 setAttributesFromStreamType(streamType); 291 mStreamType = streamType; 292 } else { 293 if (!isValidAttributes(pAttributes)) { 294 ALOGE("Invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]", 295 pAttributes->usage, pAttributes->content_type, pAttributes->flags, 296 pAttributes->tags); 297 } 298 // stream type shouldn't be looked at, this track has audio attributes 299 memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t)); 300 setStreamTypeFromAttributes(mAttributes); 301 ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]", 302 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags); 303 } 304 305 status_t status; 306 if (sampleRate == 0) { 307 status = AudioSystem::getOutputSamplingRateForAttr(&sampleRate, &mAttributes); 308 if (status != NO_ERROR) { 309 ALOGE("Could not get output sample rate for stream type %d; status %d", 310 mStreamType, status); 311 return status; 312 } 313 } 314 mSampleRate = sampleRate; 315 316 // these below should probably come from the audioFlinger too... 317 if (format == AUDIO_FORMAT_DEFAULT) { 318 format = AUDIO_FORMAT_PCM_16_BIT; 319 } 320 321 // validate parameters 322 if (!audio_is_valid_format(format)) { 323 ALOGE("Invalid format %#x", format); 324 return BAD_VALUE; 325 } 326 mFormat = format; 327 328 if (!audio_is_output_channel(channelMask)) { 329 ALOGE("Invalid channel mask %#x", channelMask); 330 return BAD_VALUE; 331 } 332 mChannelMask = channelMask; 333 uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 334 mChannelCount = channelCount; 335 336 // AudioFlinger does not currently support 8-bit data in shared memory 337 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 338 ALOGE("8-bit data in shared memory is not supported"); 339 return BAD_VALUE; 340 } 341 342 // force direct flag if format is not linear PCM 343 // or offload was requested 344 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 345 || !audio_is_linear_pcm(format)) { 346 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 347 ? "Offload request, forcing to Direct Output" 348 : "Not linear PCM, forcing to Direct Output"); 349 flags = (audio_output_flags_t) 350 // FIXME why can't we allow direct AND fast? 351 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 352 } 353 // only allow deep buffering for music stream type 354 if (mStreamType != AUDIO_STREAM_MUSIC) { 355 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 356 } 357 358 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { 359 if (audio_is_linear_pcm(format)) { 360 mFrameSize = channelCount * audio_bytes_per_sample(format); 361 } else { 362 mFrameSize = sizeof(uint8_t); 363 } 364 mFrameSizeAF = mFrameSize; 365 } else { 366 ALOG_ASSERT(audio_is_linear_pcm(format)); 367 mFrameSize = channelCount * audio_bytes_per_sample(format); 368 mFrameSizeAF = channelCount * audio_bytes_per_sample( 369 format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format); 370 // createTrack will return an error if PCM format is not supported by server, 371 // so no need to check for specific PCM formats here 372 } 373 374 // Make copy of input parameter offloadInfo so that in the future: 375 // (a) createTrack_l doesn't need it as an input parameter 376 // (b) we can support re-creation of offloaded tracks 377 if (offloadInfo != NULL) { 378 mOffloadInfoCopy = *offloadInfo; 379 mOffloadInfo = &mOffloadInfoCopy; 380 } else { 381 mOffloadInfo = NULL; 382 } 383 384 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f; 385 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f; 386 mSendLevel = 0.0f; 387 // mFrameCount is initialized in createTrack_l 388 mReqFrameCount = frameCount; 389 mNotificationFramesReq = notificationFrames; 390 mNotificationFramesAct = 0; 391 mSessionId = sessionId; 392 int callingpid = IPCThreadState::self()->getCallingPid(); 393 int mypid = getpid(); 394 if (uid == -1 || (callingpid != mypid)) { 395 mClientUid = IPCThreadState::self()->getCallingUid(); 396 } else { 397 mClientUid = uid; 398 } 399 if (pid == -1 || (callingpid != mypid)) { 400 mClientPid = callingpid; 401 } else { 402 mClientPid = pid; 403 } 404 mAuxEffectId = 0; 405 mFlags = flags; 406 mCbf = cbf; 407 408 if (cbf != NULL) { 409 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 410 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 411 } 412 413 // create the IAudioTrack 414 status = createTrack_l(); 415 416 if (status != NO_ERROR) { 417 if (mAudioTrackThread != 0) { 418 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 419 mAudioTrackThread->requestExitAndWait(); 420 mAudioTrackThread.clear(); 421 } 422 return status; 423 } 424 425 mStatus = NO_ERROR; 426 mState = STATE_STOPPED; 427 mUserData = user; 428 mLoopPeriod = 0; 429 mMarkerPosition = 0; 430 mMarkerReached = false; 431 mNewPosition = 0; 432 mUpdatePeriod = 0; 433 mServer = 0; 434 mPosition = 0; 435 mReleased = 0; 436 mStartUs = 0; 437 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 438 mSequence = 1; 439 mObservedSequence = mSequence; 440 mInUnderrun = false; 441 442 return NO_ERROR; 443} 444 445// ------------------------------------------------------------------------- 446 447status_t AudioTrack::start() 448{ 449 AutoMutex lock(mLock); 450 451 if (mState == STATE_ACTIVE) { 452 return INVALID_OPERATION; 453 } 454 455 mInUnderrun = true; 456 457 State previousState = mState; 458 if (previousState == STATE_PAUSED_STOPPING) { 459 mState = STATE_STOPPING; 460 } else { 461 mState = STATE_ACTIVE; 462 } 463 (void) updateAndGetPosition_l(); 464 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 465 // reset current position as seen by client to 0 466 mPosition = 0; 467 mReleased = 0; 468 // For offloaded tracks, we don't know if the hardware counters are really zero here, 469 // since the flush is asynchronous and stop may not fully drain. 470 // We save the time when the track is started to later verify whether 471 // the counters are realistic (i.e. start from zero after this time). 472 mStartUs = getNowUs(); 473 474 // force refresh of remaining frames by processAudioBuffer() as last 475 // write before stop could be partial. 476 mRefreshRemaining = true; 477 } 478 mNewPosition = mPosition + mUpdatePeriod; 479 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 480 481 sp<AudioTrackThread> t = mAudioTrackThread; 482 if (t != 0) { 483 if (previousState == STATE_STOPPING) { 484 mProxy->interrupt(); 485 } else { 486 t->resume(); 487 } 488 } else { 489 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 490 get_sched_policy(0, &mPreviousSchedulingGroup); 491 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 492 } 493 494 status_t status = NO_ERROR; 495 if (!(flags & CBLK_INVALID)) { 496 status = mAudioTrack->start(); 497 if (status == DEAD_OBJECT) { 498 flags |= CBLK_INVALID; 499 } 500 } 501 if (flags & CBLK_INVALID) { 502 status = restoreTrack_l("start"); 503 } 504 505 if (status != NO_ERROR) { 506 ALOGE("start() status %d", status); 507 mState = previousState; 508 if (t != 0) { 509 if (previousState != STATE_STOPPING) { 510 t->pause(); 511 } 512 } else { 513 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 514 set_sched_policy(0, mPreviousSchedulingGroup); 515 } 516 } 517 518 return status; 519} 520 521void AudioTrack::stop() 522{ 523 AutoMutex lock(mLock); 524 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 525 return; 526 } 527 528 if (isOffloaded_l()) { 529 mState = STATE_STOPPING; 530 } else { 531 mState = STATE_STOPPED; 532 } 533 534 mProxy->interrupt(); 535 mAudioTrack->stop(); 536 // the playback head position will reset to 0, so if a marker is set, we need 537 // to activate it again 538 mMarkerReached = false; 539#if 0 540 // Force flush if a shared buffer is used otherwise audioflinger 541 // will not stop before end of buffer is reached. 542 // It may be needed to make sure that we stop playback, likely in case looping is on. 543 if (mSharedBuffer != 0) { 544 flush_l(); 545 } 546#endif 547 548 sp<AudioTrackThread> t = mAudioTrackThread; 549 if (t != 0) { 550 if (!isOffloaded_l()) { 551 t->pause(); 552 } 553 } else { 554 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 555 set_sched_policy(0, mPreviousSchedulingGroup); 556 } 557} 558 559bool AudioTrack::stopped() const 560{ 561 AutoMutex lock(mLock); 562 return mState != STATE_ACTIVE; 563} 564 565void AudioTrack::flush() 566{ 567 if (mSharedBuffer != 0) { 568 return; 569 } 570 AutoMutex lock(mLock); 571 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 572 return; 573 } 574 flush_l(); 575} 576 577void AudioTrack::flush_l() 578{ 579 ALOG_ASSERT(mState != STATE_ACTIVE); 580 581 // clear playback marker and periodic update counter 582 mMarkerPosition = 0; 583 mMarkerReached = false; 584 mUpdatePeriod = 0; 585 mRefreshRemaining = true; 586 587 mState = STATE_FLUSHED; 588 if (isOffloaded_l()) { 589 mProxy->interrupt(); 590 } 591 mProxy->flush(); 592 mAudioTrack->flush(); 593} 594 595void AudioTrack::pause() 596{ 597 AutoMutex lock(mLock); 598 if (mState == STATE_ACTIVE) { 599 mState = STATE_PAUSED; 600 } else if (mState == STATE_STOPPING) { 601 mState = STATE_PAUSED_STOPPING; 602 } else { 603 return; 604 } 605 mProxy->interrupt(); 606 mAudioTrack->pause(); 607 608 if (isOffloaded_l()) { 609 if (mOutput != AUDIO_IO_HANDLE_NONE) { 610 // An offload output can be re-used between two audio tracks having 611 // the same configuration. A timestamp query for a paused track 612 // while the other is running would return an incorrect time. 613 // To fix this, cache the playback position on a pause() and return 614 // this time when requested until the track is resumed. 615 616 // OffloadThread sends HAL pause in its threadLoop. Time saved 617 // here can be slightly off. 618 619 // TODO: check return code for getRenderPosition. 620 621 uint32_t halFrames; 622 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); 623 ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); 624 } 625 } 626} 627 628status_t AudioTrack::setVolume(float left, float right) 629{ 630 // This duplicates a test by AudioTrack JNI, but that is not the only caller 631 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY || 632 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) { 633 return BAD_VALUE; 634 } 635 636 AutoMutex lock(mLock); 637 mVolume[AUDIO_INTERLEAVE_LEFT] = left; 638 mVolume[AUDIO_INTERLEAVE_RIGHT] = right; 639 640 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right))); 641 642 if (isOffloaded_l()) { 643 mAudioTrack->signal(); 644 } 645 return NO_ERROR; 646} 647 648status_t AudioTrack::setVolume(float volume) 649{ 650 return setVolume(volume, volume); 651} 652 653status_t AudioTrack::setAuxEffectSendLevel(float level) 654{ 655 // This duplicates a test by AudioTrack JNI, but that is not the only caller 656 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) { 657 return BAD_VALUE; 658 } 659 660 AutoMutex lock(mLock); 661 mSendLevel = level; 662 mProxy->setSendLevel(level); 663 664 return NO_ERROR; 665} 666 667void AudioTrack::getAuxEffectSendLevel(float* level) const 668{ 669 if (level != NULL) { 670 *level = mSendLevel; 671 } 672} 673 674status_t AudioTrack::setSampleRate(uint32_t rate) 675{ 676 if (mIsTimed || isOffloadedOrDirect()) { 677 return INVALID_OPERATION; 678 } 679 680 uint32_t afSamplingRate; 681 if (AudioSystem::getOutputSamplingRateForAttr(&afSamplingRate, &mAttributes) != 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 AutoMutex lock(mLock); 689 mSampleRate = rate; 690 mProxy->setSampleRate(rate); 691 692 return NO_ERROR; 693} 694 695uint32_t AudioTrack::getSampleRate() const 696{ 697 if (mIsTimed) { 698 return 0; 699 } 700 701 AutoMutex lock(mLock); 702 703 // sample rate can be updated during playback by the offloaded decoder so we need to 704 // query the HAL and update if needed. 705// FIXME use Proxy return channel to update the rate from server and avoid polling here 706 if (isOffloadedOrDirect_l()) { 707 if (mOutput != AUDIO_IO_HANDLE_NONE) { 708 uint32_t sampleRate = 0; 709 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate); 710 if (status == NO_ERROR) { 711 mSampleRate = sampleRate; 712 } 713 } 714 } 715 return mSampleRate; 716} 717 718status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 719{ 720 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 721 return INVALID_OPERATION; 722 } 723 724 if (loopCount == 0) { 725 ; 726 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 727 loopEnd - loopStart >= MIN_LOOP) { 728 ; 729 } else { 730 return BAD_VALUE; 731 } 732 733 AutoMutex lock(mLock); 734 // See setPosition() regarding setting parameters such as loop points or position while active 735 if (mState == STATE_ACTIVE) { 736 return INVALID_OPERATION; 737 } 738 setLoop_l(loopStart, loopEnd, loopCount); 739 return NO_ERROR; 740} 741 742void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 743{ 744 // FIXME If setting a loop also sets position to start of loop, then 745 // this is correct. Otherwise it should be removed. 746 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 747 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 748 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 749} 750 751status_t AudioTrack::setMarkerPosition(uint32_t marker) 752{ 753 // The only purpose of setting marker position is to get a callback 754 if (mCbf == NULL || isOffloadedOrDirect()) { 755 return INVALID_OPERATION; 756 } 757 758 AutoMutex lock(mLock); 759 mMarkerPosition = marker; 760 mMarkerReached = false; 761 762 return NO_ERROR; 763} 764 765status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 766{ 767 if (isOffloadedOrDirect()) { 768 return INVALID_OPERATION; 769 } 770 if (marker == NULL) { 771 return BAD_VALUE; 772 } 773 774 AutoMutex lock(mLock); 775 *marker = mMarkerPosition; 776 777 return NO_ERROR; 778} 779 780status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 781{ 782 // The only purpose of setting position update period is to get a callback 783 if (mCbf == NULL || isOffloadedOrDirect()) { 784 return INVALID_OPERATION; 785 } 786 787 AutoMutex lock(mLock); 788 mNewPosition = updateAndGetPosition_l() + updatePeriod; 789 mUpdatePeriod = updatePeriod; 790 791 return NO_ERROR; 792} 793 794status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 795{ 796 if (isOffloadedOrDirect()) { 797 return INVALID_OPERATION; 798 } 799 if (updatePeriod == NULL) { 800 return BAD_VALUE; 801 } 802 803 AutoMutex lock(mLock); 804 *updatePeriod = mUpdatePeriod; 805 806 return NO_ERROR; 807} 808 809status_t AudioTrack::setPosition(uint32_t position) 810{ 811 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 812 return INVALID_OPERATION; 813 } 814 if (position > mFrameCount) { 815 return BAD_VALUE; 816 } 817 818 AutoMutex lock(mLock); 819 // Currently we require that the player is inactive before setting parameters such as position 820 // or loop points. Otherwise, there could be a race condition: the application could read the 821 // current position, compute a new position or loop parameters, and then set that position or 822 // loop parameters but it would do the "wrong" thing since the position has continued to advance 823 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 824 // to specify how it wants to handle such scenarios. 825 if (mState == STATE_ACTIVE) { 826 return INVALID_OPERATION; 827 } 828 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 829 mLoopPeriod = 0; 830 // FIXME Check whether loops and setting position are incompatible in old code. 831 // If we use setLoop for both purposes we lose the capability to set the position while looping. 832 mStaticProxy->setLoop(position, mFrameCount, 0); 833 834 return NO_ERROR; 835} 836 837status_t AudioTrack::getPosition(uint32_t *position) 838{ 839 if (position == NULL) { 840 return BAD_VALUE; 841 } 842 843 AutoMutex lock(mLock); 844 if (isOffloadedOrDirect_l()) { 845 uint32_t dspFrames = 0; 846 847 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) { 848 ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); 849 *position = mPausedPosition; 850 return NO_ERROR; 851 } 852 853 if (mOutput != AUDIO_IO_HANDLE_NONE) { 854 uint32_t halFrames; 855 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 856 } 857 // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED) 858 // due to hardware latency. We leave this behavior for now. 859 *position = dspFrames; 860 } else { 861 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 862 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 863 0 : updateAndGetPosition_l(); 864 } 865 return NO_ERROR; 866} 867 868status_t AudioTrack::getBufferPosition(uint32_t *position) 869{ 870 if (mSharedBuffer == 0 || mIsTimed) { 871 return INVALID_OPERATION; 872 } 873 if (position == NULL) { 874 return BAD_VALUE; 875 } 876 877 AutoMutex lock(mLock); 878 *position = mStaticProxy->getBufferPosition(); 879 return NO_ERROR; 880} 881 882status_t AudioTrack::reload() 883{ 884 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 885 return INVALID_OPERATION; 886 } 887 888 AutoMutex lock(mLock); 889 // See setPosition() regarding setting parameters such as loop points or position while active 890 if (mState == STATE_ACTIVE) { 891 return INVALID_OPERATION; 892 } 893 mNewPosition = mUpdatePeriod; 894 mLoopPeriod = 0; 895 // FIXME The new code cannot reload while keeping a loop specified. 896 // Need to check how the old code handled this, and whether it's a significant change. 897 mStaticProxy->setLoop(0, mFrameCount, 0); 898 return NO_ERROR; 899} 900 901audio_io_handle_t AudioTrack::getOutput() const 902{ 903 AutoMutex lock(mLock); 904 return mOutput; 905} 906 907status_t AudioTrack::attachAuxEffect(int effectId) 908{ 909 AutoMutex lock(mLock); 910 status_t status = mAudioTrack->attachAuxEffect(effectId); 911 if (status == NO_ERROR) { 912 mAuxEffectId = effectId; 913 } 914 return status; 915} 916 917// ------------------------------------------------------------------------- 918 919// must be called with mLock held 920status_t AudioTrack::createTrack_l() 921{ 922 status_t status; 923 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 924 if (audioFlinger == 0) { 925 ALOGE("Could not get audioflinger"); 926 return NO_INIT; 927 } 928 929 audio_io_handle_t output = AudioSystem::getOutputForAttr(&mAttributes, mSampleRate, mFormat, 930 mChannelMask, mFlags, mOffloadInfo); 931 if (output == AUDIO_IO_HANDLE_NONE) { 932 ALOGE("Could not get audio output for stream type %d, usage %d, sample rate %u, format %#x," 933 " channel mask %#x, flags %#x", 934 mStreamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags); 935 return BAD_VALUE; 936 } 937 { 938 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 939 // we must release it ourselves if anything goes wrong. 940 941 // Not all of these values are needed under all conditions, but it is easier to get them all 942 943 uint32_t afLatency; 944 status = AudioSystem::getLatency(output, &afLatency); 945 if (status != NO_ERROR) { 946 ALOGE("getLatency(%d) failed status %d", output, status); 947 goto release; 948 } 949 950 size_t afFrameCount; 951 status = AudioSystem::getFrameCount(output, &afFrameCount); 952 if (status != NO_ERROR) { 953 ALOGE("getFrameCount(output=%d) status %d", output, status); 954 goto release; 955 } 956 957 uint32_t afSampleRate; 958 status = AudioSystem::getSamplingRate(output, &afSampleRate); 959 if (status != NO_ERROR) { 960 ALOGE("getSamplingRate(output=%d) status %d", output, status); 961 goto release; 962 } 963 964 // Client decides whether the track is TIMED (see below), but can only express a preference 965 // for FAST. Server will perform additional tests. 966 if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( 967 // either of these use cases: 968 // use case 1: shared buffer 969 (mSharedBuffer != 0) || 970 // use case 2: callback transfer mode 971 (mTransfer == TRANSFER_CALLBACK)) && 972 // matching sample rate 973 (mSampleRate == afSampleRate))) { 974 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 975 // once denied, do not request again if IAudioTrack is re-created 976 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 977 } 978 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 979 980 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 981 // n = 1 fast track with single buffering; nBuffering is ignored 982 // n = 2 fast track with double buffering 983 // n = 2 normal track, no sample rate conversion 984 // n = 3 normal track, with sample rate conversion 985 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 986 // n > 3 very high latency or very small notification interval; nBuffering is ignored 987 const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; 988 989 mNotificationFramesAct = mNotificationFramesReq; 990 991 size_t frameCount = mReqFrameCount; 992 if (!audio_is_linear_pcm(mFormat)) { 993 994 if (mSharedBuffer != 0) { 995 // Same comment as below about ignoring frameCount parameter for set() 996 frameCount = mSharedBuffer->size(); 997 } else if (frameCount == 0) { 998 frameCount = afFrameCount; 999 } 1000 if (mNotificationFramesAct != frameCount) { 1001 mNotificationFramesAct = frameCount; 1002 } 1003 } else if (mSharedBuffer != 0) { 1004 1005 // Ensure that buffer alignment matches channel count 1006 // 8-bit data in shared memory is not currently supported by AudioFlinger 1007 size_t alignment = audio_bytes_per_sample( 1008 mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat); 1009 if (alignment & 1) { 1010 alignment = 1; 1011 } 1012 if (mChannelCount > 1) { 1013 // More than 2 channels does not require stronger alignment than stereo 1014 alignment <<= 1; 1015 } 1016 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 1017 ALOGE("Invalid buffer alignment: address %p, channel count %u", 1018 mSharedBuffer->pointer(), mChannelCount); 1019 status = BAD_VALUE; 1020 goto release; 1021 } 1022 1023 // When initializing a shared buffer AudioTrack via constructors, 1024 // there's no frameCount parameter. 1025 // But when initializing a shared buffer AudioTrack via set(), 1026 // there _is_ a frameCount parameter. We silently ignore it. 1027 frameCount = mSharedBuffer->size() / mFrameSizeAF; 1028 1029 } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 1030 1031 // FIXME move these calculations and associated checks to server 1032 1033 // Ensure that buffer depth covers at least audio hardware latency 1034 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 1035 ALOGV("afFrameCount=%zu, minBufCount=%d, afSampleRate=%u, afLatency=%d", 1036 afFrameCount, minBufCount, afSampleRate, afLatency); 1037 if (minBufCount <= nBuffering) { 1038 minBufCount = nBuffering; 1039 } 1040 1041 size_t minFrameCount = afFrameCount * minBufCount * uint64_t(mSampleRate) / afSampleRate; 1042 ALOGV("minFrameCount: %zu, afFrameCount=%zu, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 1043 ", afLatency=%d", 1044 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); 1045 1046 if (frameCount == 0) { 1047 frameCount = minFrameCount; 1048 } else if (frameCount < minFrameCount) { 1049 // not ALOGW because it happens all the time when playing key clicks over A2DP 1050 ALOGV("Minimum buffer size corrected from %zu to %zu", 1051 frameCount, minFrameCount); 1052 frameCount = minFrameCount; 1053 } 1054 // Make sure that application is notified with sufficient margin before underrun 1055 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1056 mNotificationFramesAct = frameCount/nBuffering; 1057 } 1058 1059 } else { 1060 // For fast tracks, the frame count calculations and checks are done by server 1061 } 1062 1063 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 1064 if (mIsTimed) { 1065 trackFlags |= IAudioFlinger::TRACK_TIMED; 1066 } 1067 1068 pid_t tid = -1; 1069 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1070 trackFlags |= IAudioFlinger::TRACK_FAST; 1071 if (mAudioTrackThread != 0) { 1072 tid = mAudioTrackThread->getTid(); 1073 } 1074 } 1075 1076 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1077 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 1078 } 1079 1080 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1081 trackFlags |= IAudioFlinger::TRACK_DIRECT; 1082 } 1083 1084 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1085 // but we will still need the original value also 1086 sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType, 1087 mSampleRate, 1088 // AudioFlinger only sees 16-bit PCM 1089 mFormat == AUDIO_FORMAT_PCM_8_BIT && 1090 !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT) ? 1091 AUDIO_FORMAT_PCM_16_BIT : mFormat, 1092 mChannelMask, 1093 &temp, 1094 &trackFlags, 1095 mSharedBuffer, 1096 output, 1097 tid, 1098 &mSessionId, 1099 mClientUid, 1100 &status); 1101 1102 if (status != NO_ERROR) { 1103 ALOGE("AudioFlinger could not create track, status: %d", status); 1104 goto release; 1105 } 1106 ALOG_ASSERT(track != 0); 1107 1108 // AudioFlinger now owns the reference to the I/O handle, 1109 // so we are no longer responsible for releasing it. 1110 1111 sp<IMemory> iMem = track->getCblk(); 1112 if (iMem == 0) { 1113 ALOGE("Could not get control block"); 1114 return NO_INIT; 1115 } 1116 void *iMemPointer = iMem->pointer(); 1117 if (iMemPointer == NULL) { 1118 ALOGE("Could not get control block pointer"); 1119 return NO_INIT; 1120 } 1121 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1122 if (mAudioTrack != 0) { 1123 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1124 mDeathNotifier.clear(); 1125 } 1126 mAudioTrack = track; 1127 mCblkMemory = iMem; 1128 IPCThreadState::self()->flushCommands(); 1129 1130 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1131 mCblk = cblk; 1132 // note that temp is the (possibly revised) value of frameCount 1133 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1134 // In current design, AudioTrack client checks and ensures frame count validity before 1135 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1136 // for fast track as it uses a special method of assigning frame count. 1137 ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp); 1138 } 1139 frameCount = temp; 1140 1141 mAwaitBoost = false; 1142 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1143 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1144 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount); 1145 mAwaitBoost = true; 1146 if (mSharedBuffer == 0) { 1147 // Theoretically double-buffering is not required for fast tracks, 1148 // due to tighter scheduling. But in practice, to accommodate kernels with 1149 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1150 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1151 mNotificationFramesAct = frameCount/nBuffering; 1152 } 1153 } 1154 } else { 1155 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount); 1156 // once denied, do not request again if IAudioTrack is re-created 1157 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1158 if (mSharedBuffer == 0) { 1159 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1160 mNotificationFramesAct = frameCount/nBuffering; 1161 } 1162 } 1163 } 1164 } 1165 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1166 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1167 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1168 } else { 1169 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1170 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1171 // FIXME This is a warning, not an error, so don't return error status 1172 //return NO_INIT; 1173 } 1174 } 1175 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1176 if (trackFlags & IAudioFlinger::TRACK_DIRECT) { 1177 ALOGV("AUDIO_OUTPUT_FLAG_DIRECT successful"); 1178 } else { 1179 ALOGW("AUDIO_OUTPUT_FLAG_DIRECT denied by server"); 1180 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT); 1181 // FIXME This is a warning, not an error, so don't return error status 1182 //return NO_INIT; 1183 } 1184 } 1185 1186 // We retain a copy of the I/O handle, but don't own the reference 1187 mOutput = output; 1188 mRefreshRemaining = true; 1189 1190 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1191 // is the value of pointer() for the shared buffer, otherwise buffers points 1192 // immediately after the control block. This address is for the mapping within client 1193 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1194 void* buffers; 1195 if (mSharedBuffer == 0) { 1196 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1197 } else { 1198 buffers = mSharedBuffer->pointer(); 1199 } 1200 1201 mAudioTrack->attachAuxEffect(mAuxEffectId); 1202 // FIXME don't believe this lie 1203 mLatency = afLatency + (1000*frameCount) / mSampleRate; 1204 1205 mFrameCount = frameCount; 1206 // If IAudioTrack is re-created, don't let the requested frameCount 1207 // decrease. This can confuse clients that cache frameCount(). 1208 if (frameCount > mReqFrameCount) { 1209 mReqFrameCount = frameCount; 1210 } 1211 1212 // update proxy 1213 if (mSharedBuffer == 0) { 1214 mStaticProxy.clear(); 1215 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1216 } else { 1217 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1218 mProxy = mStaticProxy; 1219 } 1220 mProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); 1221 mProxy->setSendLevel(mSendLevel); 1222 mProxy->setSampleRate(mSampleRate); 1223 mProxy->setMinimum(mNotificationFramesAct); 1224 1225 mDeathNotifier = new DeathNotifier(this); 1226 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1227 1228 return NO_ERROR; 1229 } 1230 1231release: 1232 AudioSystem::releaseOutput(output); 1233 if (status == NO_ERROR) { 1234 status = NO_INIT; 1235 } 1236 return status; 1237} 1238 1239status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1240{ 1241 if (audioBuffer == NULL) { 1242 return BAD_VALUE; 1243 } 1244 if (mTransfer != TRANSFER_OBTAIN) { 1245 audioBuffer->frameCount = 0; 1246 audioBuffer->size = 0; 1247 audioBuffer->raw = NULL; 1248 return INVALID_OPERATION; 1249 } 1250 1251 const struct timespec *requested; 1252 struct timespec timeout; 1253 if (waitCount == -1) { 1254 requested = &ClientProxy::kForever; 1255 } else if (waitCount == 0) { 1256 requested = &ClientProxy::kNonBlocking; 1257 } else if (waitCount > 0) { 1258 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1259 timeout.tv_sec = ms / 1000; 1260 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1261 requested = &timeout; 1262 } else { 1263 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1264 requested = NULL; 1265 } 1266 return obtainBuffer(audioBuffer, requested); 1267} 1268 1269status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1270 struct timespec *elapsed, size_t *nonContig) 1271{ 1272 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1273 uint32_t oldSequence = 0; 1274 uint32_t newSequence; 1275 1276 Proxy::Buffer buffer; 1277 status_t status = NO_ERROR; 1278 1279 static const int32_t kMaxTries = 5; 1280 int32_t tryCounter = kMaxTries; 1281 1282 do { 1283 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1284 // keep them from going away if another thread re-creates the track during obtainBuffer() 1285 sp<AudioTrackClientProxy> proxy; 1286 sp<IMemory> iMem; 1287 1288 { // start of lock scope 1289 AutoMutex lock(mLock); 1290 1291 newSequence = mSequence; 1292 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1293 if (status == DEAD_OBJECT) { 1294 // re-create track, unless someone else has already done so 1295 if (newSequence == oldSequence) { 1296 status = restoreTrack_l("obtainBuffer"); 1297 if (status != NO_ERROR) { 1298 buffer.mFrameCount = 0; 1299 buffer.mRaw = NULL; 1300 buffer.mNonContig = 0; 1301 break; 1302 } 1303 } 1304 } 1305 oldSequence = newSequence; 1306 1307 // Keep the extra references 1308 proxy = mProxy; 1309 iMem = mCblkMemory; 1310 1311 if (mState == STATE_STOPPING) { 1312 status = -EINTR; 1313 buffer.mFrameCount = 0; 1314 buffer.mRaw = NULL; 1315 buffer.mNonContig = 0; 1316 break; 1317 } 1318 1319 // Non-blocking if track is stopped or paused 1320 if (mState != STATE_ACTIVE) { 1321 requested = &ClientProxy::kNonBlocking; 1322 } 1323 1324 } // end of lock scope 1325 1326 buffer.mFrameCount = audioBuffer->frameCount; 1327 // FIXME starts the requested timeout and elapsed over from scratch 1328 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1329 1330 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1331 1332 audioBuffer->frameCount = buffer.mFrameCount; 1333 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1334 audioBuffer->raw = buffer.mRaw; 1335 if (nonContig != NULL) { 1336 *nonContig = buffer.mNonContig; 1337 } 1338 return status; 1339} 1340 1341void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1342{ 1343 if (mTransfer == TRANSFER_SHARED) { 1344 return; 1345 } 1346 1347 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1348 if (stepCount == 0) { 1349 return; 1350 } 1351 1352 Proxy::Buffer buffer; 1353 buffer.mFrameCount = stepCount; 1354 buffer.mRaw = audioBuffer->raw; 1355 1356 AutoMutex lock(mLock); 1357 mReleased += stepCount; 1358 mInUnderrun = false; 1359 mProxy->releaseBuffer(&buffer); 1360 1361 // restart track if it was disabled by audioflinger due to previous underrun 1362 if (mState == STATE_ACTIVE) { 1363 audio_track_cblk_t* cblk = mCblk; 1364 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1365 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1366 // FIXME ignoring status 1367 mAudioTrack->start(); 1368 } 1369 } 1370} 1371 1372// ------------------------------------------------------------------------- 1373 1374ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1375{ 1376 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1377 return INVALID_OPERATION; 1378 } 1379 1380 if (isDirect()) { 1381 AutoMutex lock(mLock); 1382 int32_t flags = android_atomic_and( 1383 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), 1384 &mCblk->mFlags); 1385 if (flags & CBLK_INVALID) { 1386 return DEAD_OBJECT; 1387 } 1388 } 1389 1390 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1391 // Sanity-check: user is most-likely passing an error code, and it would 1392 // make the return value ambiguous (actualSize vs error). 1393 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1394 return BAD_VALUE; 1395 } 1396 1397 size_t written = 0; 1398 Buffer audioBuffer; 1399 1400 while (userSize >= mFrameSize) { 1401 audioBuffer.frameCount = userSize / mFrameSize; 1402 1403 status_t err = obtainBuffer(&audioBuffer, 1404 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1405 if (err < 0) { 1406 if (written > 0) { 1407 break; 1408 } 1409 return ssize_t(err); 1410 } 1411 1412 size_t toWrite; 1413 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1414 // Divide capacity by 2 to take expansion into account 1415 toWrite = audioBuffer.size >> 1; 1416 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1417 } else { 1418 toWrite = audioBuffer.size; 1419 memcpy(audioBuffer.i8, buffer, toWrite); 1420 } 1421 buffer = ((const char *) buffer) + toWrite; 1422 userSize -= toWrite; 1423 written += toWrite; 1424 1425 releaseBuffer(&audioBuffer); 1426 } 1427 1428 return written; 1429} 1430 1431// ------------------------------------------------------------------------- 1432 1433TimedAudioTrack::TimedAudioTrack() { 1434 mIsTimed = true; 1435} 1436 1437status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1438{ 1439 AutoMutex lock(mLock); 1440 status_t result = UNKNOWN_ERROR; 1441 1442#if 1 1443 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1444 // while we are accessing the cblk 1445 sp<IAudioTrack> audioTrack = mAudioTrack; 1446 sp<IMemory> iMem = mCblkMemory; 1447#endif 1448 1449 // If the track is not invalid already, try to allocate a buffer. alloc 1450 // fails indicating that the server is dead, flag the track as invalid so 1451 // we can attempt to restore in just a bit. 1452 audio_track_cblk_t* cblk = mCblk; 1453 if (!(cblk->mFlags & CBLK_INVALID)) { 1454 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1455 if (result == DEAD_OBJECT) { 1456 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1457 } 1458 } 1459 1460 // If the track is invalid at this point, attempt to restore it. and try the 1461 // allocation one more time. 1462 if (cblk->mFlags & CBLK_INVALID) { 1463 result = restoreTrack_l("allocateTimedBuffer"); 1464 1465 if (result == NO_ERROR) { 1466 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1467 } 1468 } 1469 1470 return result; 1471} 1472 1473status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1474 int64_t pts) 1475{ 1476 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1477 { 1478 AutoMutex lock(mLock); 1479 audio_track_cblk_t* cblk = mCblk; 1480 // restart track if it was disabled by audioflinger due to previous underrun 1481 if (buffer->size() != 0 && status == NO_ERROR && 1482 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1483 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1484 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1485 // FIXME ignoring status 1486 mAudioTrack->start(); 1487 } 1488 } 1489 return status; 1490} 1491 1492status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1493 TargetTimeline target) 1494{ 1495 return mAudioTrack->setMediaTimeTransform(xform, target); 1496} 1497 1498// ------------------------------------------------------------------------- 1499 1500nsecs_t AudioTrack::processAudioBuffer() 1501{ 1502 // Currently the AudioTrack thread is not created if there are no callbacks. 1503 // Would it ever make sense to run the thread, even without callbacks? 1504 // If so, then replace this by checks at each use for mCbf != NULL. 1505 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1506 1507 mLock.lock(); 1508 if (mAwaitBoost) { 1509 mAwaitBoost = false; 1510 mLock.unlock(); 1511 static const int32_t kMaxTries = 5; 1512 int32_t tryCounter = kMaxTries; 1513 uint32_t pollUs = 10000; 1514 do { 1515 int policy = sched_getscheduler(0); 1516 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1517 break; 1518 } 1519 usleep(pollUs); 1520 pollUs <<= 1; 1521 } while (tryCounter-- > 0); 1522 if (tryCounter < 0) { 1523 ALOGE("did not receive expected priority boost on time"); 1524 } 1525 // Run again immediately 1526 return 0; 1527 } 1528 1529 // Can only reference mCblk while locked 1530 int32_t flags = android_atomic_and( 1531 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1532 1533 // Check for track invalidation 1534 if (flags & CBLK_INVALID) { 1535 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1536 // AudioSystem cache. We should not exit here but after calling the callback so 1537 // that the upper layers can recreate the track 1538 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) { 1539 status_t status = restoreTrack_l("processAudioBuffer"); 1540 mLock.unlock(); 1541 // Run again immediately, but with a new IAudioTrack 1542 return 0; 1543 } 1544 } 1545 1546 bool waitStreamEnd = mState == STATE_STOPPING; 1547 bool active = mState == STATE_ACTIVE; 1548 1549 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1550 bool newUnderrun = false; 1551 if (flags & CBLK_UNDERRUN) { 1552#if 0 1553 // Currently in shared buffer mode, when the server reaches the end of buffer, 1554 // the track stays active in continuous underrun state. It's up to the application 1555 // to pause or stop the track, or set the position to a new offset within buffer. 1556 // This was some experimental code to auto-pause on underrun. Keeping it here 1557 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1558 if (mTransfer == TRANSFER_SHARED) { 1559 mState = STATE_PAUSED; 1560 active = false; 1561 } 1562#endif 1563 if (!mInUnderrun) { 1564 mInUnderrun = true; 1565 newUnderrun = true; 1566 } 1567 } 1568 1569 // Get current position of server 1570 size_t position = updateAndGetPosition_l(); 1571 1572 // Manage marker callback 1573 bool markerReached = false; 1574 size_t markerPosition = mMarkerPosition; 1575 // FIXME fails for wraparound, need 64 bits 1576 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1577 mMarkerReached = markerReached = true; 1578 } 1579 1580 // Determine number of new position callback(s) that will be needed, while locked 1581 size_t newPosCount = 0; 1582 size_t newPosition = mNewPosition; 1583 size_t updatePeriod = mUpdatePeriod; 1584 // FIXME fails for wraparound, need 64 bits 1585 if (updatePeriod > 0 && position >= newPosition) { 1586 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1587 mNewPosition += updatePeriod * newPosCount; 1588 } 1589 1590 // Cache other fields that will be needed soon 1591 uint32_t loopPeriod = mLoopPeriod; 1592 uint32_t sampleRate = mSampleRate; 1593 uint32_t notificationFrames = mNotificationFramesAct; 1594 if (mRefreshRemaining) { 1595 mRefreshRemaining = false; 1596 mRemainingFrames = notificationFrames; 1597 mRetryOnPartialBuffer = false; 1598 } 1599 size_t misalignment = mProxy->getMisalignment(); 1600 uint32_t sequence = mSequence; 1601 sp<AudioTrackClientProxy> proxy = mProxy; 1602 1603 // These fields don't need to be cached, because they are assigned only by set(): 1604 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1605 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1606 1607 mLock.unlock(); 1608 1609 if (waitStreamEnd) { 1610 struct timespec timeout; 1611 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1612 timeout.tv_nsec = 0; 1613 1614 status_t status = proxy->waitStreamEndDone(&timeout); 1615 switch (status) { 1616 case NO_ERROR: 1617 case DEAD_OBJECT: 1618 case TIMED_OUT: 1619 mCbf(EVENT_STREAM_END, mUserData, NULL); 1620 { 1621 AutoMutex lock(mLock); 1622 // The previously assigned value of waitStreamEnd is no longer valid, 1623 // since the mutex has been unlocked and either the callback handler 1624 // or another thread could have re-started the AudioTrack during that time. 1625 waitStreamEnd = mState == STATE_STOPPING; 1626 if (waitStreamEnd) { 1627 mState = STATE_STOPPED; 1628 } 1629 } 1630 if (waitStreamEnd && status != DEAD_OBJECT) { 1631 return NS_INACTIVE; 1632 } 1633 break; 1634 } 1635 return 0; 1636 } 1637 1638 // perform callbacks while unlocked 1639 if (newUnderrun) { 1640 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1641 } 1642 // FIXME we will miss loops if loop cycle was signaled several times since last call 1643 // to processAudioBuffer() 1644 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1645 mCbf(EVENT_LOOP_END, mUserData, NULL); 1646 } 1647 if (flags & CBLK_BUFFER_END) { 1648 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1649 } 1650 if (markerReached) { 1651 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1652 } 1653 while (newPosCount > 0) { 1654 size_t temp = newPosition; 1655 mCbf(EVENT_NEW_POS, mUserData, &temp); 1656 newPosition += updatePeriod; 1657 newPosCount--; 1658 } 1659 1660 if (mObservedSequence != sequence) { 1661 mObservedSequence = sequence; 1662 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1663 // for offloaded tracks, just wait for the upper layers to recreate the track 1664 if (isOffloadedOrDirect()) { 1665 return NS_INACTIVE; 1666 } 1667 } 1668 1669 // if inactive, then don't run me again until re-started 1670 if (!active) { 1671 return NS_INACTIVE; 1672 } 1673 1674 // Compute the estimated time until the next timed event (position, markers, loops) 1675 // FIXME only for non-compressed audio 1676 uint32_t minFrames = ~0; 1677 if (!markerReached && position < markerPosition) { 1678 minFrames = markerPosition - position; 1679 } 1680 if (loopPeriod > 0 && loopPeriod < minFrames) { 1681 minFrames = loopPeriod; 1682 } 1683 if (updatePeriod > 0 && updatePeriod < minFrames) { 1684 minFrames = updatePeriod; 1685 } 1686 1687 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1688 static const uint32_t kPoll = 0; 1689 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1690 minFrames = kPoll * notificationFrames; 1691 } 1692 1693 // Convert frame units to time units 1694 nsecs_t ns = NS_WHENEVER; 1695 if (minFrames != (uint32_t) ~0) { 1696 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1697 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1698 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1699 } 1700 1701 // If not supplying data by EVENT_MORE_DATA, then we're done 1702 if (mTransfer != TRANSFER_CALLBACK) { 1703 return ns; 1704 } 1705 1706 struct timespec timeout; 1707 const struct timespec *requested = &ClientProxy::kForever; 1708 if (ns != NS_WHENEVER) { 1709 timeout.tv_sec = ns / 1000000000LL; 1710 timeout.tv_nsec = ns % 1000000000LL; 1711 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1712 requested = &timeout; 1713 } 1714 1715 while (mRemainingFrames > 0) { 1716 1717 Buffer audioBuffer; 1718 audioBuffer.frameCount = mRemainingFrames; 1719 size_t nonContig; 1720 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1721 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1722 "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount); 1723 requested = &ClientProxy::kNonBlocking; 1724 size_t avail = audioBuffer.frameCount + nonContig; 1725 ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d", 1726 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1727 if (err != NO_ERROR) { 1728 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1729 (isOffloaded() && (err == DEAD_OBJECT))) { 1730 return 0; 1731 } 1732 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1733 return NS_NEVER; 1734 } 1735 1736 if (mRetryOnPartialBuffer && !isOffloaded()) { 1737 mRetryOnPartialBuffer = false; 1738 if (avail < mRemainingFrames) { 1739 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1740 if (ns < 0 || myns < ns) { 1741 ns = myns; 1742 } 1743 return ns; 1744 } 1745 } 1746 1747 // Divide buffer size by 2 to take into account the expansion 1748 // due to 8 to 16 bit conversion: the callback must fill only half 1749 // of the destination buffer 1750 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1751 audioBuffer.size >>= 1; 1752 } 1753 1754 size_t reqSize = audioBuffer.size; 1755 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1756 size_t writtenSize = audioBuffer.size; 1757 1758 // Sanity check on returned size 1759 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1760 ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes", 1761 reqSize, ssize_t(writtenSize)); 1762 return NS_NEVER; 1763 } 1764 1765 if (writtenSize == 0) { 1766 // The callback is done filling buffers 1767 // Keep this thread going to handle timed events and 1768 // still try to get more data in intervals of WAIT_PERIOD_MS 1769 // but don't just loop and block the CPU, so wait 1770 return WAIT_PERIOD_MS * 1000000LL; 1771 } 1772 1773 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1774 // 8 to 16 bit conversion, note that source and destination are the same address 1775 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1776 audioBuffer.size <<= 1; 1777 } 1778 1779 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1780 audioBuffer.frameCount = releasedFrames; 1781 mRemainingFrames -= releasedFrames; 1782 if (misalignment >= releasedFrames) { 1783 misalignment -= releasedFrames; 1784 } else { 1785 misalignment = 0; 1786 } 1787 1788 releaseBuffer(&audioBuffer); 1789 1790 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1791 // if callback doesn't like to accept the full chunk 1792 if (writtenSize < reqSize) { 1793 continue; 1794 } 1795 1796 // There could be enough non-contiguous frames available to satisfy the remaining request 1797 if (mRemainingFrames <= nonContig) { 1798 continue; 1799 } 1800 1801#if 0 1802 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1803 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1804 // that total to a sum == notificationFrames. 1805 if (0 < misalignment && misalignment <= mRemainingFrames) { 1806 mRemainingFrames = misalignment; 1807 return (mRemainingFrames * 1100000000LL) / sampleRate; 1808 } 1809#endif 1810 1811 } 1812 mRemainingFrames = notificationFrames; 1813 mRetryOnPartialBuffer = true; 1814 1815 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1816 return 0; 1817} 1818 1819status_t AudioTrack::restoreTrack_l(const char *from) 1820{ 1821 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1822 isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from); 1823 ++mSequence; 1824 status_t result; 1825 1826 // refresh the audio configuration cache in this process to make sure we get new 1827 // output parameters in createTrack_l() 1828 AudioSystem::clearAudioConfigCache(); 1829 1830 if (isOffloadedOrDirect_l()) { 1831 // FIXME re-creation of offloaded tracks is not yet implemented 1832 return DEAD_OBJECT; 1833 } 1834 1835 // save the old static buffer position 1836 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1837 1838 // If a new IAudioTrack is successfully created, createTrack_l() will modify the 1839 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1840 // It will also delete the strong references on previous IAudioTrack and IMemory. 1841 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact. 1842 result = createTrack_l(); 1843 1844 // take the frames that will be lost by track recreation into account in saved position 1845 (void) updateAndGetPosition_l(); 1846 mPosition = mReleased; 1847 1848 if (result == NO_ERROR) { 1849 // continue playback from last known position, but 1850 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1851 if (mStaticProxy != NULL) { 1852 mLoopPeriod = 0; 1853 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1854 } 1855 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1856 // track destruction have been played? This is critical for SoundPool implementation 1857 // This must be broken, and needs to be tested/debugged. 1858#if 0 1859 // restore write index and set other indexes to reflect empty buffer status 1860 if (!strcmp(from, "start")) { 1861 // Make sure that a client relying on callback events indicating underrun or 1862 // the actual amount of audio frames played (e.g SoundPool) receives them. 1863 if (mSharedBuffer == 0) { 1864 // restart playback even if buffer is not completely filled. 1865 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1866 } 1867 } 1868#endif 1869 if (mState == STATE_ACTIVE) { 1870 result = mAudioTrack->start(); 1871 } 1872 } 1873 if (result != NO_ERROR) { 1874 ALOGW("restoreTrack_l() failed status %d", result); 1875 mState = STATE_STOPPED; 1876 } 1877 1878 return result; 1879} 1880 1881uint32_t AudioTrack::updateAndGetPosition_l() 1882{ 1883 // This is the sole place to read server consumed frames 1884 uint32_t newServer = mProxy->getPosition(); 1885 int32_t delta = newServer - mServer; 1886 mServer = newServer; 1887 // TODO There is controversy about whether there can be "negative jitter" in server position. 1888 // This should be investigated further, and if possible, it should be addressed. 1889 // A more definite failure mode is infrequent polling by client. 1890 // One could call (void)getPosition_l() in releaseBuffer(), 1891 // so mReleased and mPosition are always lock-step as best possible. 1892 // That should ensure delta never goes negative for infrequent polling 1893 // unless the server has more than 2^31 frames in its buffer, 1894 // in which case the use of uint32_t for these counters has bigger issues. 1895 if (delta < 0) { 1896 ALOGE("detected illegal retrograde motion by the server: mServer advanced by %d", delta); 1897 delta = 0; 1898 } 1899 return mPosition += (uint32_t) delta; 1900} 1901 1902status_t AudioTrack::setParameters(const String8& keyValuePairs) 1903{ 1904 AutoMutex lock(mLock); 1905 return mAudioTrack->setParameters(keyValuePairs); 1906} 1907 1908status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1909{ 1910 AutoMutex lock(mLock); 1911 // FIXME not implemented for fast tracks; should use proxy and SSQ 1912 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1913 return INVALID_OPERATION; 1914 } 1915 1916 switch (mState) { 1917 case STATE_ACTIVE: 1918 case STATE_PAUSED: 1919 break; // handle below 1920 case STATE_FLUSHED: 1921 case STATE_STOPPED: 1922 return WOULD_BLOCK; 1923 case STATE_STOPPING: 1924 case STATE_PAUSED_STOPPING: 1925 if (!isOffloaded_l()) { 1926 return INVALID_OPERATION; 1927 } 1928 break; // offloaded tracks handled below 1929 default: 1930 LOG_ALWAYS_FATAL("Invalid mState in getTimestamp(): %d", mState); 1931 break; 1932 } 1933 1934 // The presented frame count must always lag behind the consumed frame count. 1935 // To avoid a race, read the presented frames first. This ensures that presented <= consumed. 1936 status_t status = mAudioTrack->getTimestamp(timestamp); 1937 if (status != NO_ERROR) { 1938 ALOGW_IF(status != WOULD_BLOCK, "getTimestamp error:%#x", status); 1939 return status; 1940 } 1941 if (isOffloadedOrDirect_l()) { 1942 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) { 1943 // use cached paused position in case another offloaded track is running. 1944 timestamp.mPosition = mPausedPosition; 1945 clock_gettime(CLOCK_MONOTONIC, ×tamp.mTime); 1946 return NO_ERROR; 1947 } 1948 1949 // Check whether a pending flush or stop has completed, as those commands may 1950 // be asynchronous or return near finish. 1951 if (mStartUs != 0 && mSampleRate != 0) { 1952 static const int kTimeJitterUs = 100000; // 100 ms 1953 static const int k1SecUs = 1000000; 1954 1955 const int64_t timeNow = getNowUs(); 1956 1957 if (timeNow < mStartUs + k1SecUs) { // within first second of starting 1958 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime); 1959 if (timestampTimeUs < mStartUs) { 1960 return WOULD_BLOCK; // stale timestamp time, occurs before start. 1961 } 1962 const int64_t deltaTimeUs = timestampTimeUs - mStartUs; 1963 const int64_t deltaPositionByUs = timestamp.mPosition * 1000000LL / mSampleRate; 1964 1965 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) { 1966 // Verify that the counter can't count faster than the sample rate 1967 // since the start time. If greater, then that means we have failed 1968 // to completely flush or stop the previous playing track. 1969 ALOGW("incomplete flush or stop:" 1970 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)", 1971 (long long)deltaTimeUs, (long long)deltaPositionByUs, 1972 timestamp.mPosition); 1973 return WOULD_BLOCK; 1974 } 1975 } 1976 mStartUs = 0; // no need to check again, start timestamp has either expired or unneeded. 1977 } 1978 } else { 1979 // Update the mapping between local consumed (mPosition) and server consumed (mServer) 1980 (void) updateAndGetPosition_l(); 1981 // Server consumed (mServer) and presented both use the same server time base, 1982 // and server consumed is always >= presented. 1983 // The delta between these represents the number of frames in the buffer pipeline. 1984 // If this delta between these is greater than the client position, it means that 1985 // actually presented is still stuck at the starting line (figuratively speaking), 1986 // waiting for the first frame to go by. So we can't report a valid timestamp yet. 1987 if ((uint32_t) (mServer - timestamp.mPosition) > mPosition) { 1988 return INVALID_OPERATION; 1989 } 1990 // Convert timestamp position from server time base to client time base. 1991 // TODO The following code should work OK now because timestamp.mPosition is 32-bit. 1992 // But if we change it to 64-bit then this could fail. 1993 // If (mPosition - mServer) can be negative then should use: 1994 // (int32_t)(mPosition - mServer) 1995 timestamp.mPosition += mPosition - mServer; 1996 // Immediately after a call to getPosition_l(), mPosition and 1997 // mServer both represent the same frame position. mPosition is 1998 // in client's point of view, and mServer is in server's point of 1999 // view. So the difference between them is the "fudge factor" 2000 // between client and server views due to stop() and/or new 2001 // IAudioTrack. And timestamp.mPosition is initially in server's 2002 // point of view, so we need to apply the same fudge factor to it. 2003 } 2004 return status; 2005} 2006 2007String8 AudioTrack::getParameters(const String8& keys) 2008{ 2009 audio_io_handle_t output = getOutput(); 2010 if (output != AUDIO_IO_HANDLE_NONE) { 2011 return AudioSystem::getParameters(output, keys); 2012 } else { 2013 return String8::empty(); 2014 } 2015} 2016 2017bool AudioTrack::isOffloaded() const 2018{ 2019 AutoMutex lock(mLock); 2020 return isOffloaded_l(); 2021} 2022 2023bool AudioTrack::isDirect() const 2024{ 2025 AutoMutex lock(mLock); 2026 return isDirect_l(); 2027} 2028 2029bool AudioTrack::isOffloadedOrDirect() const 2030{ 2031 AutoMutex lock(mLock); 2032 return isOffloadedOrDirect_l(); 2033} 2034 2035 2036status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 2037{ 2038 2039 const size_t SIZE = 256; 2040 char buffer[SIZE]; 2041 String8 result; 2042 2043 result.append(" AudioTrack::dump\n"); 2044 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 2045 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 2046 result.append(buffer); 2047 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 2048 mChannelCount, mFrameCount); 2049 result.append(buffer); 2050 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 2051 result.append(buffer); 2052 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 2053 result.append(buffer); 2054 ::write(fd, result.string(), result.size()); 2055 return NO_ERROR; 2056} 2057 2058uint32_t AudioTrack::getUnderrunFrames() const 2059{ 2060 AutoMutex lock(mLock); 2061 return mProxy->getUnderrunFrames(); 2062} 2063 2064void AudioTrack::setAttributesFromStreamType(audio_stream_type_t streamType) { 2065 mAttributes.flags = 0x0; 2066 2067 switch(streamType) { 2068 case AUDIO_STREAM_DEFAULT: 2069 case AUDIO_STREAM_MUSIC: 2070 mAttributes.content_type = AUDIO_CONTENT_TYPE_MUSIC; 2071 mAttributes.usage = AUDIO_USAGE_MEDIA; 2072 break; 2073 case AUDIO_STREAM_VOICE_CALL: 2074 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 2075 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION; 2076 break; 2077 case AUDIO_STREAM_ENFORCED_AUDIBLE: 2078 mAttributes.flags |= AUDIO_FLAG_AUDIBILITY_ENFORCED; 2079 // intended fall through, attributes in common with STREAM_SYSTEM 2080 case AUDIO_STREAM_SYSTEM: 2081 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2082 mAttributes.usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION; 2083 break; 2084 case AUDIO_STREAM_RING: 2085 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2086 mAttributes.usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE; 2087 break; 2088 case AUDIO_STREAM_ALARM: 2089 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2090 mAttributes.usage = AUDIO_USAGE_ALARM; 2091 break; 2092 case AUDIO_STREAM_NOTIFICATION: 2093 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2094 mAttributes.usage = AUDIO_USAGE_NOTIFICATION; 2095 break; 2096 case AUDIO_STREAM_BLUETOOTH_SCO: 2097 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 2098 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION; 2099 mAttributes.flags |= AUDIO_FLAG_SCO; 2100 break; 2101 case AUDIO_STREAM_DTMF: 2102 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2103 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING; 2104 break; 2105 case AUDIO_STREAM_TTS: 2106 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 2107 mAttributes.usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY; 2108 break; 2109 default: 2110 ALOGE("invalid stream type %d when converting to attributes", streamType); 2111 } 2112} 2113 2114void AudioTrack::setStreamTypeFromAttributes(audio_attributes_t& aa) { 2115 // flags to stream type mapping 2116 if ((aa.flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { 2117 mStreamType = AUDIO_STREAM_ENFORCED_AUDIBLE; 2118 return; 2119 } 2120 if ((aa.flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) { 2121 mStreamType = AUDIO_STREAM_BLUETOOTH_SCO; 2122 return; 2123 } 2124 2125 // usage to stream type mapping 2126 switch (aa.usage) { 2127 case AUDIO_USAGE_MEDIA: 2128 case AUDIO_USAGE_GAME: 2129 case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: 2130 case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: 2131 mStreamType = AUDIO_STREAM_MUSIC; 2132 return; 2133 case AUDIO_USAGE_ASSISTANCE_SONIFICATION: 2134 mStreamType = AUDIO_STREAM_SYSTEM; 2135 return; 2136 case AUDIO_USAGE_VOICE_COMMUNICATION: 2137 mStreamType = AUDIO_STREAM_VOICE_CALL; 2138 return; 2139 2140 case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: 2141 mStreamType = AUDIO_STREAM_DTMF; 2142 return; 2143 2144 case AUDIO_USAGE_ALARM: 2145 mStreamType = AUDIO_STREAM_ALARM; 2146 return; 2147 case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: 2148 mStreamType = AUDIO_STREAM_RING; 2149 return; 2150 2151 case AUDIO_USAGE_NOTIFICATION: 2152 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: 2153 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: 2154 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: 2155 case AUDIO_USAGE_NOTIFICATION_EVENT: 2156 mStreamType = AUDIO_STREAM_NOTIFICATION; 2157 return; 2158 2159 case AUDIO_USAGE_UNKNOWN: 2160 default: 2161 mStreamType = AUDIO_STREAM_MUSIC; 2162 } 2163} 2164 2165bool AudioTrack::isValidAttributes(const audio_attributes_t *paa) { 2166 // has flags that map to a strategy? 2167 if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO)) != 0) { 2168 return true; 2169 } 2170 2171 // has known usage? 2172 switch (paa->usage) { 2173 case AUDIO_USAGE_UNKNOWN: 2174 case AUDIO_USAGE_MEDIA: 2175 case AUDIO_USAGE_VOICE_COMMUNICATION: 2176 case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: 2177 case AUDIO_USAGE_ALARM: 2178 case AUDIO_USAGE_NOTIFICATION: 2179 case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: 2180 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: 2181 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: 2182 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: 2183 case AUDIO_USAGE_NOTIFICATION_EVENT: 2184 case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: 2185 case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: 2186 case AUDIO_USAGE_ASSISTANCE_SONIFICATION: 2187 case AUDIO_USAGE_GAME: 2188 break; 2189 default: 2190 return false; 2191 } 2192 return true; 2193} 2194// ========================================================================= 2195 2196void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 2197{ 2198 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 2199 if (audioTrack != 0) { 2200 AutoMutex lock(audioTrack->mLock); 2201 audioTrack->mProxy->binderDied(); 2202 } 2203} 2204 2205// ========================================================================= 2206 2207AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 2208 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 2209 mIgnoreNextPausedInt(false) 2210{ 2211} 2212 2213AudioTrack::AudioTrackThread::~AudioTrackThread() 2214{ 2215} 2216 2217bool AudioTrack::AudioTrackThread::threadLoop() 2218{ 2219 { 2220 AutoMutex _l(mMyLock); 2221 if (mPaused) { 2222 mMyCond.wait(mMyLock); 2223 // caller will check for exitPending() 2224 return true; 2225 } 2226 if (mIgnoreNextPausedInt) { 2227 mIgnoreNextPausedInt = false; 2228 mPausedInt = false; 2229 } 2230 if (mPausedInt) { 2231 if (mPausedNs > 0) { 2232 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 2233 } else { 2234 mMyCond.wait(mMyLock); 2235 } 2236 mPausedInt = false; 2237 return true; 2238 } 2239 } 2240 nsecs_t ns = mReceiver.processAudioBuffer(); 2241 switch (ns) { 2242 case 0: 2243 return true; 2244 case NS_INACTIVE: 2245 pauseInternal(); 2246 return true; 2247 case NS_NEVER: 2248 return false; 2249 case NS_WHENEVER: 2250 // FIXME increase poll interval, or make event-driven 2251 ns = 1000000000LL; 2252 // fall through 2253 default: 2254 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns); 2255 pauseInternal(ns); 2256 return true; 2257 } 2258} 2259 2260void AudioTrack::AudioTrackThread::requestExit() 2261{ 2262 // must be in this order to avoid a race condition 2263 Thread::requestExit(); 2264 resume(); 2265} 2266 2267void AudioTrack::AudioTrackThread::pause() 2268{ 2269 AutoMutex _l(mMyLock); 2270 mPaused = true; 2271} 2272 2273void AudioTrack::AudioTrackThread::resume() 2274{ 2275 AutoMutex _l(mMyLock); 2276 mIgnoreNextPausedInt = true; 2277 if (mPaused || mPausedInt) { 2278 mPaused = false; 2279 mPausedInt = false; 2280 mMyCond.signal(); 2281 } 2282} 2283 2284void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 2285{ 2286 AutoMutex _l(mMyLock); 2287 mPausedInt = true; 2288 mPausedNs = ns; 2289} 2290 2291}; // namespace android 2292