AudioTrack.cpp revision 1ab85ec401801ef9a9184650d0f5a1639b45eeb9
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 19//#define LOG_NDEBUG 0 20#define LOG_TAG "AudioTrack" 21 22#include <sys/resource.h> 23#include <audio_utils/primitives.h> 24#include <binder/IPCThreadState.h> 25#include <media/AudioTrack.h> 26#include <utils/Log.h> 27#include <private/media/AudioTrackShared.h> 28#include <media/IAudioFlinger.h> 29 30#define WAIT_PERIOD_MS 10 31 32namespace android { 33// --------------------------------------------------------------------------- 34 35// static 36status_t AudioTrack::getMinFrameCount( 37 size_t* frameCount, 38 audio_stream_type_t streamType, 39 uint32_t sampleRate) 40{ 41 if (frameCount == NULL) { 42 return BAD_VALUE; 43 } 44 45 // default to 0 in case of error 46 *frameCount = 0; 47 48 // FIXME merge with similar code in createTrack_l(), except we're missing 49 // some information here that is available in createTrack_l(): 50 // audio_io_handle_t output 51 // audio_format_t format 52 // audio_channel_mask_t channelMask 53 // audio_output_flags_t flags 54 uint32_t afSampleRate; 55 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 56 return NO_INIT; 57 } 58 size_t afFrameCount; 59 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { 60 return NO_INIT; 61 } 62 uint32_t afLatency; 63 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 64 return NO_INIT; 65 } 66 67 // Ensure that buffer depth covers at least audio hardware latency 68 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 69 if (minBufCount < 2) { 70 minBufCount = 2; 71 } 72 73 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 74 afFrameCount * minBufCount * sampleRate / afSampleRate; 75 ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 76 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 77 return NO_ERROR; 78} 79 80// --------------------------------------------------------------------------- 81 82AudioTrack::AudioTrack() 83 : mStatus(NO_INIT), 84 mIsTimed(false), 85 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 86 mPreviousSchedulingGroup(SP_DEFAULT) 87{ 88} 89 90AudioTrack::AudioTrack( 91 audio_stream_type_t streamType, 92 uint32_t sampleRate, 93 audio_format_t format, 94 audio_channel_mask_t channelMask, 95 int frameCount, 96 audio_output_flags_t flags, 97 callback_t cbf, 98 void* user, 99 int notificationFrames, 100 int sessionId, 101 transfer_type transferType, 102 const audio_offload_info_t *offloadInfo) 103 : mStatus(NO_INIT), 104 mIsTimed(false), 105 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 106 mPreviousSchedulingGroup(SP_DEFAULT) 107{ 108 mStatus = set(streamType, sampleRate, format, channelMask, 109 frameCount, flags, cbf, user, notificationFrames, 110 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo); 111} 112 113AudioTrack::AudioTrack( 114 audio_stream_type_t streamType, 115 uint32_t sampleRate, 116 audio_format_t format, 117 audio_channel_mask_t channelMask, 118 const sp<IMemory>& sharedBuffer, 119 audio_output_flags_t flags, 120 callback_t cbf, 121 void* user, 122 int notificationFrames, 123 int sessionId, 124 transfer_type transferType, 125 const audio_offload_info_t *offloadInfo) 126 : mStatus(NO_INIT), 127 mIsTimed(false), 128 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 129 mPreviousSchedulingGroup(SP_DEFAULT) 130{ 131 mStatus = set(streamType, sampleRate, format, channelMask, 132 0 /*frameCount*/, flags, cbf, user, notificationFrames, 133 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo); 134} 135 136AudioTrack::~AudioTrack() 137{ 138 if (mStatus == NO_ERROR) { 139 // Make sure that callback function exits in the case where 140 // it is looping on buffer full condition in obtainBuffer(). 141 // Otherwise the callback thread will never exit. 142 stop(); 143 if (mAudioTrackThread != 0) { 144 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 145 mAudioTrackThread->requestExitAndWait(); 146 mAudioTrackThread.clear(); 147 } 148 if (mAudioTrack != 0) { 149 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 150 mAudioTrack.clear(); 151 } 152 IPCThreadState::self()->flushCommands(); 153 AudioSystem::releaseAudioSessionId(mSessionId); 154 } 155} 156 157status_t AudioTrack::set( 158 audio_stream_type_t streamType, 159 uint32_t sampleRate, 160 audio_format_t format, 161 audio_channel_mask_t channelMask, 162 int frameCountInt, 163 audio_output_flags_t flags, 164 callback_t cbf, 165 void* user, 166 int notificationFrames, 167 const sp<IMemory>& sharedBuffer, 168 bool threadCanCallJava, 169 int sessionId, 170 transfer_type transferType, 171 const audio_offload_info_t *offloadInfo) 172{ 173 switch (transferType) { 174 case TRANSFER_DEFAULT: 175 if (sharedBuffer != 0) { 176 transferType = TRANSFER_SHARED; 177 } else if (cbf == NULL || threadCanCallJava) { 178 transferType = TRANSFER_SYNC; 179 } else { 180 transferType = TRANSFER_CALLBACK; 181 } 182 break; 183 case TRANSFER_CALLBACK: 184 if (cbf == NULL || sharedBuffer != 0) { 185 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 186 return BAD_VALUE; 187 } 188 break; 189 case TRANSFER_OBTAIN: 190 case TRANSFER_SYNC: 191 if (sharedBuffer != 0) { 192 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 193 return BAD_VALUE; 194 } 195 break; 196 case TRANSFER_SHARED: 197 if (sharedBuffer == 0) { 198 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 199 return BAD_VALUE; 200 } 201 break; 202 default: 203 ALOGE("Invalid transfer type %d", transferType); 204 return BAD_VALUE; 205 } 206 mTransfer = transferType; 207 208 // FIXME "int" here is legacy and will be replaced by size_t later 209 if (frameCountInt < 0) { 210 ALOGE("Invalid frame count %d", frameCountInt); 211 return BAD_VALUE; 212 } 213 size_t frameCount = frameCountInt; 214 215 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 216 sharedBuffer->size()); 217 218 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 219 220 AutoMutex lock(mLock); 221 222 if (mAudioTrack != 0) { 223 ALOGE("Track already in use"); 224 return INVALID_OPERATION; 225 } 226 227 // handle default values first. 228 if (streamType == AUDIO_STREAM_DEFAULT) { 229 streamType = AUDIO_STREAM_MUSIC; 230 } 231 232 if (sampleRate == 0) { 233 uint32_t afSampleRate; 234 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 235 return NO_INIT; 236 } 237 sampleRate = afSampleRate; 238 } 239 mSampleRate = sampleRate; 240 241 // these below should probably come from the audioFlinger too... 242 if (format == AUDIO_FORMAT_DEFAULT) { 243 format = AUDIO_FORMAT_PCM_16_BIT; 244 } 245 if (channelMask == 0) { 246 channelMask = AUDIO_CHANNEL_OUT_STEREO; 247 } 248 249 // validate parameters 250 if (!audio_is_valid_format(format)) { 251 ALOGE("Invalid format %d", format); 252 return BAD_VALUE; 253 } 254 255 // AudioFlinger does not currently support 8-bit data in shared memory 256 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 257 ALOGE("8-bit data in shared memory is not supported"); 258 return BAD_VALUE; 259 } 260 261 // force direct flag if format is not linear PCM 262 if (!audio_is_linear_pcm(format)) { 263 flags = (audio_output_flags_t) 264 // FIXME why can't we allow direct AND fast? 265 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 266 } 267 // only allow deep buffering for music stream type 268 if (streamType != AUDIO_STREAM_MUSIC) { 269 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 270 } 271 272 if (!audio_is_output_channel(channelMask)) { 273 ALOGE("Invalid channel mask %#x", channelMask); 274 return BAD_VALUE; 275 } 276 mChannelMask = channelMask; 277 uint32_t channelCount = popcount(channelMask); 278 mChannelCount = channelCount; 279 280 if (audio_is_linear_pcm(format)) { 281 mFrameSize = channelCount * audio_bytes_per_sample(format); 282 mFrameSizeAF = channelCount * sizeof(int16_t); 283 } else { 284 mFrameSize = sizeof(uint8_t); 285 mFrameSizeAF = sizeof(uint8_t); 286 } 287 288 audio_io_handle_t output = AudioSystem::getOutput( 289 streamType, 290 sampleRate, format, channelMask, 291 flags, 292 offloadInfo); 293 294 if (output == 0) { 295 ALOGE("Could not get audio output for stream type %d", streamType); 296 return BAD_VALUE; 297 } 298 299 mVolume[LEFT] = 1.0f; 300 mVolume[RIGHT] = 1.0f; 301 mSendLevel = 0.0f; 302 mFrameCount = frameCount; 303 mReqFrameCount = frameCount; 304 mNotificationFramesReq = notificationFrames; 305 mNotificationFramesAct = 0; 306 mSessionId = sessionId; 307 mAuxEffectId = 0; 308 mFlags = flags; 309 mCbf = cbf; 310 311 if (cbf != NULL) { 312 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 313 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 314 } 315 316 // create the IAudioTrack 317 status_t status = createTrack_l(streamType, 318 sampleRate, 319 format, 320 frameCount, 321 flags, 322 sharedBuffer, 323 output, 324 0 /*epoch*/); 325 326 if (status != NO_ERROR) { 327 if (mAudioTrackThread != 0) { 328 mAudioTrackThread->requestExit(); 329 mAudioTrackThread.clear(); 330 } 331 return status; 332 } 333 334 mStatus = NO_ERROR; 335 mStreamType = streamType; 336 mFormat = format; 337 mSharedBuffer = sharedBuffer; 338 mState = STATE_STOPPED; 339 mUserData = user; 340 mLoopPeriod = 0; 341 mMarkerPosition = 0; 342 mMarkerReached = false; 343 mNewPosition = 0; 344 mUpdatePeriod = 0; 345 AudioSystem::acquireAudioSessionId(mSessionId); 346 mSequence = 1; 347 mObservedSequence = mSequence; 348 mInUnderrun = false; 349 350 return NO_ERROR; 351} 352 353// ------------------------------------------------------------------------- 354 355void AudioTrack::start() 356{ 357 AutoMutex lock(mLock); 358 if (mState == STATE_ACTIVE) { 359 return; 360 } 361 362 mInUnderrun = true; 363 364 State previousState = mState; 365 mState = STATE_ACTIVE; 366 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 367 // reset current position as seen by client to 0 368 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 369 } 370 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 371 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->flags); 372 373 sp<AudioTrackThread> t = mAudioTrackThread; 374 if (t != 0) { 375 t->resume(); 376 } else { 377 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 378 get_sched_policy(0, &mPreviousSchedulingGroup); 379 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 380 } 381 382 status_t status = NO_ERROR; 383 if (!(flags & CBLK_INVALID)) { 384 status = mAudioTrack->start(); 385 if (status == DEAD_OBJECT) { 386 flags |= CBLK_INVALID; 387 } 388 } 389 if (flags & CBLK_INVALID) { 390 status = restoreTrack_l("start"); 391 } 392 393 if (status != NO_ERROR) { 394 ALOGE("start() status %d", status); 395 mState = previousState; 396 if (t != 0) { 397 t->pause(); 398 } else { 399 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 400 set_sched_policy(0, mPreviousSchedulingGroup); 401 } 402 } 403 404 // FIXME discarding status 405} 406 407void AudioTrack::stop() 408{ 409 AutoMutex lock(mLock); 410 // FIXME pause then stop should not be a nop 411 if (mState != STATE_ACTIVE) { 412 return; 413 } 414 415 mState = STATE_STOPPED; 416 mProxy->interrupt(); 417 mAudioTrack->stop(); 418 // the playback head position will reset to 0, so if a marker is set, we need 419 // to activate it again 420 mMarkerReached = false; 421#if 0 422 // Force flush if a shared buffer is used otherwise audioflinger 423 // will not stop before end of buffer is reached. 424 // It may be needed to make sure that we stop playback, likely in case looping is on. 425 if (mSharedBuffer != 0) { 426 flush_l(); 427 } 428#endif 429 sp<AudioTrackThread> t = mAudioTrackThread; 430 if (t != 0) { 431 t->pause(); 432 } else { 433 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 434 set_sched_policy(0, mPreviousSchedulingGroup); 435 } 436} 437 438bool AudioTrack::stopped() const 439{ 440 AutoMutex lock(mLock); 441 return mState != STATE_ACTIVE; 442} 443 444void AudioTrack::flush() 445{ 446 if (mSharedBuffer != 0) { 447 return; 448 } 449 AutoMutex lock(mLock); 450 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 451 return; 452 } 453 flush_l(); 454} 455 456void AudioTrack::flush_l() 457{ 458 ALOG_ASSERT(mState != STATE_ACTIVE); 459 460 // clear playback marker and periodic update counter 461 mMarkerPosition = 0; 462 mMarkerReached = false; 463 mUpdatePeriod = 0; 464 465 mState = STATE_FLUSHED; 466 mProxy->flush(); 467 mAudioTrack->flush(); 468} 469 470void AudioTrack::pause() 471{ 472 AutoMutex lock(mLock); 473 if (mState != STATE_ACTIVE) { 474 return; 475 } 476 mState = STATE_PAUSED; 477 mProxy->interrupt(); 478 mAudioTrack->pause(); 479} 480 481status_t AudioTrack::setVolume(float left, float right) 482{ 483 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 484 return BAD_VALUE; 485 } 486 487 AutoMutex lock(mLock); 488 mVolume[LEFT] = left; 489 mVolume[RIGHT] = right; 490 491 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 492 493 return NO_ERROR; 494} 495 496status_t AudioTrack::setVolume(float volume) 497{ 498 return setVolume(volume, volume); 499} 500 501status_t AudioTrack::setAuxEffectSendLevel(float level) 502{ 503 if (level < 0.0f || level > 1.0f) { 504 return BAD_VALUE; 505 } 506 507 AutoMutex lock(mLock); 508 mSendLevel = level; 509 mProxy->setSendLevel(level); 510 511 return NO_ERROR; 512} 513 514void AudioTrack::getAuxEffectSendLevel(float* level) const 515{ 516 if (level != NULL) { 517 *level = mSendLevel; 518 } 519} 520 521status_t AudioTrack::setSampleRate(uint32_t rate) 522{ 523 if (mIsTimed) { 524 return INVALID_OPERATION; 525 } 526 527 uint32_t afSamplingRate; 528 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 529 return NO_INIT; 530 } 531 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 532 if (rate == 0 || rate > afSamplingRate*2 ) { 533 return BAD_VALUE; 534 } 535 536 AutoMutex lock(mLock); 537 mSampleRate = rate; 538 mProxy->setSampleRate(rate); 539 540 return NO_ERROR; 541} 542 543uint32_t AudioTrack::getSampleRate() const 544{ 545 if (mIsTimed) { 546 return 0; 547 } 548 549 AutoMutex lock(mLock); 550 return mSampleRate; 551} 552 553status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 554{ 555 if (mSharedBuffer == 0 || mIsTimed) { 556 return INVALID_OPERATION; 557 } 558 559 if (loopCount == 0) { 560 ; 561 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 562 loopEnd - loopStart >= MIN_LOOP) { 563 ; 564 } else { 565 return BAD_VALUE; 566 } 567 568 AutoMutex lock(mLock); 569 // See setPosition() regarding setting parameters such as loop points or position while active 570 if (mState == STATE_ACTIVE) { 571 return INVALID_OPERATION; 572 } 573 setLoop_l(loopStart, loopEnd, loopCount); 574 return NO_ERROR; 575} 576 577void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 578{ 579 // FIXME If setting a loop also sets position to start of loop, then 580 // this is correct. Otherwise it should be removed. 581 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 582 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 583 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 584} 585 586status_t AudioTrack::setMarkerPosition(uint32_t marker) 587{ 588 if (mCbf == NULL) { 589 return INVALID_OPERATION; 590 } 591 592 AutoMutex lock(mLock); 593 mMarkerPosition = marker; 594 mMarkerReached = false; 595 596 return NO_ERROR; 597} 598 599status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 600{ 601 if (marker == NULL) { 602 return BAD_VALUE; 603 } 604 605 AutoMutex lock(mLock); 606 *marker = mMarkerPosition; 607 608 return NO_ERROR; 609} 610 611status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 612{ 613 if (mCbf == NULL) { 614 return INVALID_OPERATION; 615 } 616 617 AutoMutex lock(mLock); 618 mNewPosition = mProxy->getPosition() + updatePeriod; 619 mUpdatePeriod = updatePeriod; 620 621 return NO_ERROR; 622} 623 624status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 625{ 626 if (updatePeriod == NULL) { 627 return BAD_VALUE; 628 } 629 630 AutoMutex lock(mLock); 631 *updatePeriod = mUpdatePeriod; 632 633 return NO_ERROR; 634} 635 636status_t AudioTrack::setPosition(uint32_t position) 637{ 638 if (mSharedBuffer == 0 || mIsTimed) { 639 return INVALID_OPERATION; 640 } 641 if (position > mFrameCount) { 642 return BAD_VALUE; 643 } 644 645 AutoMutex lock(mLock); 646 // Currently we require that the player is inactive before setting parameters such as position 647 // or loop points. Otherwise, there could be a race condition: the application could read the 648 // current position, compute a new position or loop parameters, and then set that position or 649 // loop parameters but it would do the "wrong" thing since the position has continued to advance 650 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 651 // to specify how it wants to handle such scenarios. 652 if (mState == STATE_ACTIVE) { 653 return INVALID_OPERATION; 654 } 655 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 656 mLoopPeriod = 0; 657 // FIXME Check whether loops and setting position are incompatible in old code. 658 // If we use setLoop for both purposes we lose the capability to set the position while looping. 659 mStaticProxy->setLoop(position, mFrameCount, 0); 660 661 return NO_ERROR; 662} 663 664status_t AudioTrack::getPosition(uint32_t *position) const 665{ 666 if (position == NULL) { 667 return BAD_VALUE; 668 } 669 670 AutoMutex lock(mLock); 671 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 672 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 673 mProxy->getPosition(); 674 675 return NO_ERROR; 676} 677 678status_t AudioTrack::getBufferPosition(size_t *position) 679{ 680 if (mSharedBuffer == 0 || mIsTimed) { 681 return INVALID_OPERATION; 682 } 683 if (position == NULL) { 684 return BAD_VALUE; 685 } 686 687 AutoMutex lock(mLock); 688 *position = mStaticProxy->getBufferPosition(); 689 return NO_ERROR; 690} 691 692status_t AudioTrack::reload() 693{ 694 if (mSharedBuffer == 0 || mIsTimed) { 695 return INVALID_OPERATION; 696 } 697 698 AutoMutex lock(mLock); 699 // See setPosition() regarding setting parameters such as loop points or position while active 700 if (mState == STATE_ACTIVE) { 701 return INVALID_OPERATION; 702 } 703 mNewPosition = mUpdatePeriod; 704 mLoopPeriod = 0; 705 // FIXME The new code cannot reload while keeping a loop specified. 706 // Need to check how the old code handled this, and whether it's a significant change. 707 mStaticProxy->setLoop(0, mFrameCount, 0); 708 return NO_ERROR; 709} 710 711audio_io_handle_t AudioTrack::getOutput() 712{ 713 AutoMutex lock(mLock); 714 return getOutput_l(); 715} 716 717// must be called with mLock held 718audio_io_handle_t AudioTrack::getOutput_l() 719{ 720 return AudioSystem::getOutput(mStreamType, 721 mSampleRate, mFormat, mChannelMask, mFlags); 722} 723 724status_t AudioTrack::attachAuxEffect(int effectId) 725{ 726 AutoMutex lock(mLock); 727 status_t status = mAudioTrack->attachAuxEffect(effectId); 728 if (status == NO_ERROR) { 729 mAuxEffectId = effectId; 730 } 731 return status; 732} 733 734// ------------------------------------------------------------------------- 735 736// must be called with mLock held 737status_t AudioTrack::createTrack_l( 738 audio_stream_type_t streamType, 739 uint32_t sampleRate, 740 audio_format_t format, 741 size_t frameCount, 742 audio_output_flags_t flags, 743 const sp<IMemory>& sharedBuffer, 744 audio_io_handle_t output, 745 size_t epoch) 746{ 747 status_t status; 748 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 749 if (audioFlinger == 0) { 750 ALOGE("Could not get audioflinger"); 751 return NO_INIT; 752 } 753 754 uint32_t afLatency; 755 if ((status = AudioSystem::getLatency(output, streamType, &afLatency)) != NO_ERROR) { 756 ALOGE("getLatency(%d) failed status %d", output, status); 757 return NO_INIT; 758 } 759 760 // Client decides whether the track is TIMED (see below), but can only express a preference 761 // for FAST. Server will perform additional tests. 762 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 763 // either of these use cases: 764 // use case 1: shared buffer 765 (sharedBuffer != 0) || 766 // use case 2: callback handler 767 (mCbf != NULL))) { 768 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 769 // once denied, do not request again if IAudioTrack is re-created 770 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 771 mFlags = flags; 772 } 773 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 774 775 mNotificationFramesAct = mNotificationFramesReq; 776 777 if (!audio_is_linear_pcm(format)) { 778 779 if (sharedBuffer != 0) { 780 // Same comment as below about ignoring frameCount parameter for set() 781 frameCount = sharedBuffer->size(); 782 } else if (frameCount == 0) { 783 size_t afFrameCount; 784 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 785 if (status != NO_ERROR) { 786 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, 787 status); 788 return NO_INIT; 789 } 790 frameCount = afFrameCount; 791 } 792 793 } else if (sharedBuffer != 0) { 794 795 // Ensure that buffer alignment matches channel count 796 // 8-bit data in shared memory is not currently supported by AudioFlinger 797 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 798 if (mChannelCount > 1) { 799 // More than 2 channels does not require stronger alignment than stereo 800 alignment <<= 1; 801 } 802 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 803 ALOGE("Invalid buffer alignment: address %p, channel count %u", 804 sharedBuffer->pointer(), mChannelCount); 805 return BAD_VALUE; 806 } 807 808 // When initializing a shared buffer AudioTrack via constructors, 809 // there's no frameCount parameter. 810 // But when initializing a shared buffer AudioTrack via set(), 811 // there _is_ a frameCount parameter. We silently ignore it. 812 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 813 814 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 815 816 // FIXME move these calculations and associated checks to server 817 uint32_t afSampleRate; 818 status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); 819 if (status != NO_ERROR) { 820 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, 821 status); 822 return NO_INIT; 823 } 824 size_t afFrameCount; 825 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 826 if (status != NO_ERROR) { 827 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); 828 return NO_INIT; 829 } 830 831 // Ensure that buffer depth covers at least audio hardware latency 832 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 833 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 834 afFrameCount, minBufCount, afSampleRate, afLatency); 835 if (minBufCount <= 2) { 836 minBufCount = sampleRate == afSampleRate ? 2 : 3; 837 } 838 839 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 840 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 841 ", afLatency=%d", 842 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 843 844 if (frameCount == 0) { 845 frameCount = minFrameCount; 846 } 847 // Make sure that application is notified with sufficient margin 848 // before underrun 849 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) { 850 mNotificationFramesAct = frameCount/2; 851 } 852 if (frameCount < minFrameCount) { 853 // not ALOGW because it happens all the time when playing key clicks over A2DP 854 ALOGV("Minimum buffer size corrected from %d to %d", 855 frameCount, minFrameCount); 856 frameCount = minFrameCount; 857 } 858 859 } else { 860 // For fast tracks, the frame count calculations and checks are done by server 861 } 862 863 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 864 if (mIsTimed) { 865 trackFlags |= IAudioFlinger::TRACK_TIMED; 866 } 867 868 pid_t tid = -1; 869 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 870 trackFlags |= IAudioFlinger::TRACK_FAST; 871 if (mAudioTrackThread != 0) { 872 tid = mAudioTrackThread->getTid(); 873 } 874 } 875 876 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 877 sampleRate, 878 // AudioFlinger only sees 16-bit PCM 879 format == AUDIO_FORMAT_PCM_8_BIT ? 880 AUDIO_FORMAT_PCM_16_BIT : format, 881 mChannelMask, 882 frameCount, 883 &trackFlags, 884 sharedBuffer, 885 output, 886 tid, 887 &mSessionId, 888 &status); 889 890 if (track == 0) { 891 ALOGE("AudioFlinger could not create track, status: %d", status); 892 return status; 893 } 894 sp<IMemory> iMem = track->getCblk(); 895 if (iMem == 0) { 896 ALOGE("Could not get control block"); 897 return NO_INIT; 898 } 899 if (mAudioTrack != 0) { 900 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 901 mDeathNotifier.clear(); 902 } 903 mAudioTrack = track; 904 mCblkMemory = iMem; 905 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 906 mCblk = cblk; 907 size_t temp = cblk->frameCount_; 908 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 909 // In current design, AudioTrack client checks and ensures frame count validity before 910 // passing it to AudioFlinger so AudioFlinger should not return a different value except 911 // for fast track as it uses a special method of assigning frame count. 912 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 913 } 914 frameCount = temp; 915 mAwaitBoost = false; 916 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 917 if (trackFlags & IAudioFlinger::TRACK_FAST) { 918 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 919 mAwaitBoost = true; 920 if (sharedBuffer == 0) { 921 // double-buffering is not required for fast tracks, due to tighter scheduling 922 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount) { 923 mNotificationFramesAct = frameCount; 924 } 925 } 926 } else { 927 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 928 // once denied, do not request again if IAudioTrack is re-created 929 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 930 mFlags = flags; 931 if (sharedBuffer == 0) { 932 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) { 933 mNotificationFramesAct = frameCount/2; 934 } 935 } 936 } 937 } 938 mRefreshRemaining = true; 939 940 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 941 // is the value of pointer() for the shared buffer, otherwise buffers points 942 // immediately after the control block. This address is for the mapping within client 943 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 944 void* buffers; 945 if (sharedBuffer == 0) { 946 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 947 } else { 948 buffers = sharedBuffer->pointer(); 949 } 950 951 mAudioTrack->attachAuxEffect(mAuxEffectId); 952 // FIXME don't believe this lie 953 mLatency = afLatency + (1000*frameCount) / sampleRate; 954 mFrameCount = frameCount; 955 // If IAudioTrack is re-created, don't let the requested frameCount 956 // decrease. This can confuse clients that cache frameCount(). 957 if (frameCount > mReqFrameCount) { 958 mReqFrameCount = frameCount; 959 } 960 961 // update proxy 962 if (sharedBuffer == 0) { 963 mStaticProxy.clear(); 964 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 965 } else { 966 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 967 mProxy = mStaticProxy; 968 } 969 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 970 uint16_t(mVolume[LEFT] * 0x1000)); 971 mProxy->setSendLevel(mSendLevel); 972 mProxy->setSampleRate(mSampleRate); 973 mProxy->setEpoch(epoch); 974 mProxy->setMinimum(mNotificationFramesAct); 975 976 mDeathNotifier = new DeathNotifier(this); 977 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 978 979 return NO_ERROR; 980} 981 982status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 983{ 984 if (audioBuffer == NULL) { 985 return BAD_VALUE; 986 } 987 if (mTransfer != TRANSFER_OBTAIN) { 988 audioBuffer->frameCount = 0; 989 audioBuffer->size = 0; 990 audioBuffer->raw = NULL; 991 return INVALID_OPERATION; 992 } 993 994 const struct timespec *requested; 995 if (waitCount == -1) { 996 requested = &ClientProxy::kForever; 997 } else if (waitCount == 0) { 998 requested = &ClientProxy::kNonBlocking; 999 } else if (waitCount > 0) { 1000 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1001 struct timespec timeout; 1002 timeout.tv_sec = ms / 1000; 1003 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1004 requested = &timeout; 1005 } else { 1006 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1007 requested = NULL; 1008 } 1009 return obtainBuffer(audioBuffer, requested); 1010} 1011 1012status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1013 struct timespec *elapsed, size_t *nonContig) 1014{ 1015 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1016 uint32_t oldSequence = 0; 1017 uint32_t newSequence; 1018 1019 Proxy::Buffer buffer; 1020 status_t status = NO_ERROR; 1021 1022 static const int32_t kMaxTries = 5; 1023 int32_t tryCounter = kMaxTries; 1024 1025 do { 1026 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1027 // keep them from going away if another thread re-creates the track during obtainBuffer() 1028 sp<AudioTrackClientProxy> proxy; 1029 sp<IMemory> iMem; 1030 1031 { // start of lock scope 1032 AutoMutex lock(mLock); 1033 1034 newSequence = mSequence; 1035 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1036 if (status == DEAD_OBJECT) { 1037 // re-create track, unless someone else has already done so 1038 if (newSequence == oldSequence) { 1039 status = restoreTrack_l("obtainBuffer"); 1040 if (status != NO_ERROR) { 1041 break; 1042 } 1043 } 1044 } 1045 oldSequence = newSequence; 1046 1047 // Keep the extra references 1048 proxy = mProxy; 1049 iMem = mCblkMemory; 1050 1051 // Non-blocking if track is stopped or paused 1052 if (mState != STATE_ACTIVE) { 1053 requested = &ClientProxy::kNonBlocking; 1054 } 1055 1056 } // end of lock scope 1057 1058 buffer.mFrameCount = audioBuffer->frameCount; 1059 // FIXME starts the requested timeout and elapsed over from scratch 1060 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1061 1062 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1063 1064 audioBuffer->frameCount = buffer.mFrameCount; 1065 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1066 audioBuffer->raw = buffer.mRaw; 1067 if (nonContig != NULL) { 1068 *nonContig = buffer.mNonContig; 1069 } 1070 return status; 1071} 1072 1073void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1074{ 1075 if (mTransfer == TRANSFER_SHARED) { 1076 return; 1077 } 1078 1079 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1080 if (stepCount == 0) { 1081 return; 1082 } 1083 1084 Proxy::Buffer buffer; 1085 buffer.mFrameCount = stepCount; 1086 buffer.mRaw = audioBuffer->raw; 1087 1088 AutoMutex lock(mLock); 1089 mInUnderrun = false; 1090 mProxy->releaseBuffer(&buffer); 1091 1092 // restart track if it was disabled by audioflinger due to previous underrun 1093 if (mState == STATE_ACTIVE) { 1094 audio_track_cblk_t* cblk = mCblk; 1095 if (android_atomic_and(~CBLK_DISABLED, &cblk->flags) & CBLK_DISABLED) { 1096 ALOGW("releaseBuffer() track %p name=%#x disabled due to previous underrun, restarting", 1097 this, cblk->mName); 1098 // FIXME ignoring status 1099 mAudioTrack->start(); 1100 } 1101 } 1102} 1103 1104// ------------------------------------------------------------------------- 1105 1106ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1107{ 1108 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1109 return INVALID_OPERATION; 1110 } 1111 1112 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1113 // Sanity-check: user is most-likely passing an error code, and it would 1114 // make the return value ambiguous (actualSize vs error). 1115 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1116 return BAD_VALUE; 1117 } 1118 1119 size_t written = 0; 1120 Buffer audioBuffer; 1121 1122 while (userSize >= mFrameSize) { 1123 audioBuffer.frameCount = userSize / mFrameSize; 1124 1125 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1126 if (err < 0) { 1127 if (written > 0) { 1128 break; 1129 } 1130 return ssize_t(err); 1131 } 1132 1133 size_t toWrite; 1134 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1135 // Divide capacity by 2 to take expansion into account 1136 toWrite = audioBuffer.size >> 1; 1137 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1138 } else { 1139 toWrite = audioBuffer.size; 1140 memcpy(audioBuffer.i8, buffer, toWrite); 1141 } 1142 buffer = ((const char *) buffer) + toWrite; 1143 userSize -= toWrite; 1144 written += toWrite; 1145 1146 releaseBuffer(&audioBuffer); 1147 } 1148 1149 return written; 1150} 1151 1152// ------------------------------------------------------------------------- 1153 1154TimedAudioTrack::TimedAudioTrack() { 1155 mIsTimed = true; 1156} 1157 1158status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1159{ 1160 AutoMutex lock(mLock); 1161 status_t result = UNKNOWN_ERROR; 1162 1163#if 1 1164 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1165 // while we are accessing the cblk 1166 sp<IAudioTrack> audioTrack = mAudioTrack; 1167 sp<IMemory> iMem = mCblkMemory; 1168#endif 1169 1170 // If the track is not invalid already, try to allocate a buffer. alloc 1171 // fails indicating that the server is dead, flag the track as invalid so 1172 // we can attempt to restore in just a bit. 1173 audio_track_cblk_t* cblk = mCblk; 1174 if (!(cblk->flags & CBLK_INVALID)) { 1175 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1176 if (result == DEAD_OBJECT) { 1177 android_atomic_or(CBLK_INVALID, &cblk->flags); 1178 } 1179 } 1180 1181 // If the track is invalid at this point, attempt to restore it. and try the 1182 // allocation one more time. 1183 if (cblk->flags & CBLK_INVALID) { 1184 result = restoreTrack_l("allocateTimedBuffer"); 1185 1186 if (result == NO_ERROR) { 1187 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1188 } 1189 } 1190 1191 return result; 1192} 1193 1194status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1195 int64_t pts) 1196{ 1197 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1198 { 1199 AutoMutex lock(mLock); 1200 audio_track_cblk_t* cblk = mCblk; 1201 // restart track if it was disabled by audioflinger due to previous underrun 1202 if (buffer->size() != 0 && status == NO_ERROR && 1203 (mState == STATE_ACTIVE) && (cblk->flags & CBLK_DISABLED)) { 1204 android_atomic_and(~CBLK_DISABLED, &cblk->flags); 1205 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1206 // FIXME ignoring status 1207 mAudioTrack->start(); 1208 } 1209 } 1210 return status; 1211} 1212 1213status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1214 TargetTimeline target) 1215{ 1216 return mAudioTrack->setMediaTimeTransform(xform, target); 1217} 1218 1219// ------------------------------------------------------------------------- 1220 1221nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1222{ 1223 mLock.lock(); 1224 if (mAwaitBoost) { 1225 mAwaitBoost = false; 1226 mLock.unlock(); 1227 static const int32_t kMaxTries = 5; 1228 int32_t tryCounter = kMaxTries; 1229 uint32_t pollUs = 10000; 1230 do { 1231 int policy = sched_getscheduler(0); 1232 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1233 break; 1234 } 1235 usleep(pollUs); 1236 pollUs <<= 1; 1237 } while (tryCounter-- > 0); 1238 if (tryCounter < 0) { 1239 ALOGE("did not receive expected priority boost on time"); 1240 } 1241 return true; 1242 } 1243 1244 // Can only reference mCblk while locked 1245 int32_t flags = android_atomic_and( 1246 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->flags); 1247 1248 // Check for track invalidation 1249 if (flags & CBLK_INVALID) { 1250 (void) restoreTrack_l("processAudioBuffer"); 1251 mLock.unlock(); 1252 // Run again immediately, but with a new IAudioTrack 1253 return 0; 1254 } 1255 1256 bool active = mState == STATE_ACTIVE; 1257 1258 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1259 bool newUnderrun = false; 1260 if (flags & CBLK_UNDERRUN) { 1261#if 0 1262 // Currently in shared buffer mode, when the server reaches the end of buffer, 1263 // the track stays active in continuous underrun state. It's up to the application 1264 // to pause or stop the track, or set the position to a new offset within buffer. 1265 // This was some experimental code to auto-pause on underrun. Keeping it here 1266 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1267 if (mTransfer == TRANSFER_SHARED) { 1268 mState = STATE_PAUSED; 1269 active = false; 1270 } 1271#endif 1272 if (!mInUnderrun) { 1273 mInUnderrun = true; 1274 newUnderrun = true; 1275 } 1276 } 1277 1278 // Get current position of server 1279 size_t position = mProxy->getPosition(); 1280 1281 // Manage marker callback 1282 bool markerReached = false; 1283 size_t markerPosition = mMarkerPosition; 1284 // FIXME fails for wraparound, need 64 bits 1285 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1286 mMarkerReached = markerReached = true; 1287 } 1288 1289 // Determine number of new position callback(s) that will be needed, while locked 1290 size_t newPosCount = 0; 1291 size_t newPosition = mNewPosition; 1292 size_t updatePeriod = mUpdatePeriod; 1293 // FIXME fails for wraparound, need 64 bits 1294 if (updatePeriod > 0 && position >= newPosition) { 1295 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1296 mNewPosition += updatePeriod * newPosCount; 1297 } 1298 1299 // Cache other fields that will be needed soon 1300 uint32_t loopPeriod = mLoopPeriod; 1301 uint32_t sampleRate = mSampleRate; 1302 size_t notificationFrames = mNotificationFramesAct; 1303 if (mRefreshRemaining) { 1304 mRefreshRemaining = false; 1305 mRemainingFrames = notificationFrames; 1306 mRetryOnPartialBuffer = false; 1307 } 1308 size_t misalignment = mProxy->getMisalignment(); 1309 int32_t sequence = mSequence; 1310 1311 // These fields don't need to be cached, because they are assigned only by set(): 1312 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1313 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1314 1315 mLock.unlock(); 1316 1317 // perform callbacks while unlocked 1318 if (newUnderrun) { 1319 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1320 } 1321 // FIXME we will miss loops if loop cycle was signaled several times since last call 1322 // to processAudioBuffer() 1323 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1324 mCbf(EVENT_LOOP_END, mUserData, NULL); 1325 } 1326 if (flags & CBLK_BUFFER_END) { 1327 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1328 } 1329 if (markerReached) { 1330 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1331 } 1332 while (newPosCount > 0) { 1333 size_t temp = newPosition; 1334 mCbf(EVENT_NEW_POS, mUserData, &temp); 1335 newPosition += updatePeriod; 1336 newPosCount--; 1337 } 1338 if (mObservedSequence != sequence) { 1339 mObservedSequence = sequence; 1340 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1341 } 1342 1343 // if inactive, then don't run me again until re-started 1344 if (!active) { 1345 return NS_INACTIVE; 1346 } 1347 1348 // Compute the estimated time until the next timed event (position, markers, loops) 1349 // FIXME only for non-compressed audio 1350 uint32_t minFrames = ~0; 1351 if (!markerReached && position < markerPosition) { 1352 minFrames = markerPosition - position; 1353 } 1354 if (loopPeriod > 0 && loopPeriod < minFrames) { 1355 minFrames = loopPeriod; 1356 } 1357 if (updatePeriod > 0 && updatePeriod < minFrames) { 1358 minFrames = updatePeriod; 1359 } 1360 1361 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1362 static const uint32_t kPoll = 0; 1363 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1364 minFrames = kPoll * notificationFrames; 1365 } 1366 1367 // Convert frame units to time units 1368 nsecs_t ns = NS_WHENEVER; 1369 if (minFrames != (uint32_t) ~0) { 1370 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1371 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1372 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1373 } 1374 1375 // If not supplying data by EVENT_MORE_DATA, then we're done 1376 if (mTransfer != TRANSFER_CALLBACK) { 1377 return ns; 1378 } 1379 1380 struct timespec timeout; 1381 const struct timespec *requested = &ClientProxy::kForever; 1382 if (ns != NS_WHENEVER) { 1383 timeout.tv_sec = ns / 1000000000LL; 1384 timeout.tv_nsec = ns % 1000000000LL; 1385 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1386 requested = &timeout; 1387 } 1388 1389 while (mRemainingFrames > 0) { 1390 1391 Buffer audioBuffer; 1392 audioBuffer.frameCount = mRemainingFrames; 1393 size_t nonContig; 1394 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1395 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1396 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1397 requested = &ClientProxy::kNonBlocking; 1398 size_t avail = audioBuffer.frameCount + nonContig; 1399 ALOGV("obtainBuffer(%u) returned %u = %u + %u", 1400 mRemainingFrames, avail, audioBuffer.frameCount, nonContig); 1401 if (err != NO_ERROR) { 1402 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { 1403 return 0; 1404 } 1405 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1406 return NS_NEVER; 1407 } 1408 1409 if (mRetryOnPartialBuffer) { 1410 mRetryOnPartialBuffer = false; 1411 if (avail < mRemainingFrames) { 1412 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1413 if (ns < 0 || myns < ns) { 1414 ns = myns; 1415 } 1416 return ns; 1417 } 1418 } 1419 1420 // Divide buffer size by 2 to take into account the expansion 1421 // due to 8 to 16 bit conversion: the callback must fill only half 1422 // of the destination buffer 1423 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1424 audioBuffer.size >>= 1; 1425 } 1426 1427 size_t reqSize = audioBuffer.size; 1428 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1429 size_t writtenSize = audioBuffer.size; 1430 size_t writtenFrames = writtenSize / mFrameSize; 1431 1432 // Sanity check on returned size 1433 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1434 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1435 reqSize, (int) writtenSize); 1436 return NS_NEVER; 1437 } 1438 1439 if (writtenSize == 0) { 1440 // The callback is done filling buffers 1441 // Keep this thread going to handle timed events and 1442 // still try to get more data in intervals of WAIT_PERIOD_MS 1443 // but don't just loop and block the CPU, so wait 1444 return WAIT_PERIOD_MS * 1000000LL; 1445 } 1446 1447 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1448 // 8 to 16 bit conversion, note that source and destination are the same address 1449 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1450 audioBuffer.size <<= 1; 1451 } 1452 1453 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1454 audioBuffer.frameCount = releasedFrames; 1455 mRemainingFrames -= releasedFrames; 1456 if (misalignment >= releasedFrames) { 1457 misalignment -= releasedFrames; 1458 } else { 1459 misalignment = 0; 1460 } 1461 1462 releaseBuffer(&audioBuffer); 1463 1464 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1465 // if callback doesn't like to accept the full chunk 1466 if (writtenSize < reqSize) { 1467 continue; 1468 } 1469 1470 // There could be enough non-contiguous frames available to satisfy the remaining request 1471 if (mRemainingFrames <= nonContig) { 1472 continue; 1473 } 1474 1475#if 0 1476 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1477 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1478 // that total to a sum == notificationFrames. 1479 if (0 < misalignment && misalignment <= mRemainingFrames) { 1480 mRemainingFrames = misalignment; 1481 return (mRemainingFrames * 1100000000LL) / sampleRate; 1482 } 1483#endif 1484 1485 } 1486 mRemainingFrames = notificationFrames; 1487 mRetryOnPartialBuffer = true; 1488 1489 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1490 return 0; 1491} 1492 1493status_t AudioTrack::restoreTrack_l(const char *from) 1494{ 1495 ALOGW("dead IAudioTrack, creating a new one from %s()", from); 1496 ++mSequence; 1497 status_t result; 1498 1499 // refresh the audio configuration cache in this process to make sure we get new 1500 // output parameters in getOutput_l() and createTrack_l() 1501 AudioSystem::clearAudioConfigCache(); 1502 1503 // if the new IAudioTrack is created, createTrack_l() will modify the 1504 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1505 // It will also delete the strong references on previous IAudioTrack and IMemory 1506 size_t position = mProxy->getPosition(); 1507 mNewPosition = position + mUpdatePeriod; 1508 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1509 result = createTrack_l(mStreamType, 1510 mSampleRate, 1511 mFormat, 1512 mReqFrameCount, // so that frame count never goes down 1513 mFlags, 1514 mSharedBuffer, 1515 getOutput_l(), 1516 position /*epoch*/); 1517 1518 if (result == NO_ERROR) { 1519 // continue playback from last known position, but 1520 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1521 if (mStaticProxy != NULL) { 1522 mLoopPeriod = 0; 1523 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1524 } 1525 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1526 // track destruction have been played? This is critical for SoundPool implementation 1527 // This must be broken, and needs to be tested/debugged. 1528#if 0 1529 // restore write index and set other indexes to reflect empty buffer status 1530 if (!strcmp(from, "start")) { 1531 // Make sure that a client relying on callback events indicating underrun or 1532 // the actual amount of audio frames played (e.g SoundPool) receives them. 1533 if (mSharedBuffer == 0) { 1534 // restart playback even if buffer is not completely filled. 1535 android_atomic_or(CBLK_FORCEREADY, &mCblk->flags); 1536 } 1537 } 1538#endif 1539 if (mState == STATE_ACTIVE) { 1540 result = mAudioTrack->start(); 1541 } 1542 } 1543 if (result != NO_ERROR) { 1544 ALOGW("restoreTrack_l() failed status %d", result); 1545 mState = STATE_STOPPED; 1546 } 1547 1548 return result; 1549} 1550 1551status_t AudioTrack::setParameters(const String8& keyValuePairs) 1552{ 1553 AutoMutex lock(mLock); 1554 if (mAudioTrack != 0) { 1555 return mAudioTrack->setParameters(keyValuePairs); 1556 } else { 1557 return NO_INIT; 1558 } 1559} 1560 1561String8 AudioTrack::getParameters(const String8& keys) 1562{ 1563 return String8::empty(); 1564} 1565 1566status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1567{ 1568 1569 const size_t SIZE = 256; 1570 char buffer[SIZE]; 1571 String8 result; 1572 1573 result.append(" AudioTrack::dump\n"); 1574 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1575 mVolume[0], mVolume[1]); 1576 result.append(buffer); 1577 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1578 mChannelCount, mFrameCount); 1579 result.append(buffer); 1580 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1581 result.append(buffer); 1582 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1583 result.append(buffer); 1584 ::write(fd, result.string(), result.size()); 1585 return NO_ERROR; 1586} 1587 1588uint32_t AudioTrack::getUnderrunFrames() const 1589{ 1590 AutoMutex lock(mLock); 1591 return mProxy->getUnderrunFrames(); 1592} 1593 1594// ========================================================================= 1595 1596void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who) 1597{ 1598 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1599 if (audioTrack != 0) { 1600 AutoMutex lock(audioTrack->mLock); 1601 audioTrack->mProxy->binderDied(); 1602 } 1603} 1604 1605// ========================================================================= 1606 1607AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1608 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mResumeLatch(false) 1609{ 1610} 1611 1612AudioTrack::AudioTrackThread::~AudioTrackThread() 1613{ 1614} 1615 1616bool AudioTrack::AudioTrackThread::threadLoop() 1617{ 1618 { 1619 AutoMutex _l(mMyLock); 1620 if (mPaused) { 1621 mMyCond.wait(mMyLock); 1622 // caller will check for exitPending() 1623 return true; 1624 } 1625 } 1626 nsecs_t ns = mReceiver.processAudioBuffer(this); 1627 switch (ns) { 1628 case 0: 1629 return true; 1630 case NS_WHENEVER: 1631 sleep(1); 1632 return true; 1633 case NS_INACTIVE: 1634 pauseConditional(); 1635 return true; 1636 case NS_NEVER: 1637 return false; 1638 default: 1639 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1640 struct timespec req; 1641 req.tv_sec = ns / 1000000000LL; 1642 req.tv_nsec = ns % 1000000000LL; 1643 nanosleep(&req, NULL /*rem*/); 1644 return true; 1645 } 1646} 1647 1648void AudioTrack::AudioTrackThread::requestExit() 1649{ 1650 // must be in this order to avoid a race condition 1651 Thread::requestExit(); 1652 resume(); 1653} 1654 1655void AudioTrack::AudioTrackThread::pause() 1656{ 1657 AutoMutex _l(mMyLock); 1658 mPaused = true; 1659 mResumeLatch = false; 1660} 1661 1662void AudioTrack::AudioTrackThread::pauseConditional() 1663{ 1664 AutoMutex _l(mMyLock); 1665 if (mResumeLatch) { 1666 mResumeLatch = false; 1667 } else { 1668 mPaused = true; 1669 } 1670} 1671 1672void AudioTrack::AudioTrackThread::resume() 1673{ 1674 AutoMutex _l(mMyLock); 1675 if (mPaused) { 1676 mPaused = false; 1677 mResumeLatch = false; 1678 mMyCond.signal(); 1679 } else { 1680 mResumeLatch = true; 1681 } 1682} 1683 1684}; // namespace android 1685