AudioTrack.cpp revision ad3af3305f024bcbbd55c894a4995e449498e1ba
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 29#define WAIT_PERIOD_MS 10 30 31namespace android { 32// --------------------------------------------------------------------------- 33 34// static 35status_t AudioTrack::getMinFrameCount( 36 size_t* frameCount, 37 audio_stream_type_t streamType, 38 uint32_t sampleRate) 39{ 40 if (frameCount == NULL) { 41 return BAD_VALUE; 42 } 43 44 // default to 0 in case of error 45 *frameCount = 0; 46 47 // FIXME merge with similar code in createTrack_l(), except we're missing 48 // some information here that is available in createTrack_l(): 49 // audio_io_handle_t output 50 // audio_format_t format 51 // audio_channel_mask_t channelMask 52 // audio_output_flags_t flags 53 uint32_t afSampleRate; 54 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 55 return NO_INIT; 56 } 57 size_t afFrameCount; 58 if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) { 59 return NO_INIT; 60 } 61 uint32_t afLatency; 62 if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) { 63 return NO_INIT; 64 } 65 66 // Ensure that buffer depth covers at least audio hardware latency 67 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 68 if (minBufCount < 2) { 69 minBufCount = 2; 70 } 71 72 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 73 afFrameCount * minBufCount * sampleRate / afSampleRate; 74 ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d", 75 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 76 return NO_ERROR; 77} 78 79// --------------------------------------------------------------------------- 80 81AudioTrack::AudioTrack() 82 : mStatus(NO_INIT), 83 mIsTimed(false), 84 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 85 mPreviousSchedulingGroup(SP_DEFAULT) 86{ 87} 88 89AudioTrack::AudioTrack( 90 audio_stream_type_t streamType, 91 uint32_t sampleRate, 92 audio_format_t format, 93 audio_channel_mask_t channelMask, 94 int frameCount, 95 audio_output_flags_t flags, 96 callback_t cbf, 97 void* user, 98 int notificationFrames, 99 int sessionId, 100 transfer_type transferType, 101 const audio_offload_info_t *offloadInfo) 102 : mStatus(NO_INIT), 103 mIsTimed(false), 104 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 105 mPreviousSchedulingGroup(SP_DEFAULT) 106{ 107 mStatus = set(streamType, sampleRate, format, channelMask, 108 frameCount, flags, cbf, user, notificationFrames, 109 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo); 110} 111 112AudioTrack::AudioTrack( 113 audio_stream_type_t streamType, 114 uint32_t sampleRate, 115 audio_format_t format, 116 audio_channel_mask_t channelMask, 117 const sp<IMemory>& sharedBuffer, 118 audio_output_flags_t flags, 119 callback_t cbf, 120 void* user, 121 int notificationFrames, 122 int sessionId, 123 transfer_type transferType, 124 const audio_offload_info_t *offloadInfo) 125 : mStatus(NO_INIT), 126 mIsTimed(false), 127 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 128 mPreviousSchedulingGroup(SP_DEFAULT) 129{ 130 mStatus = set(streamType, sampleRate, format, channelMask, 131 0 /*frameCount*/, flags, cbf, user, notificationFrames, 132 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo); 133} 134 135AudioTrack::~AudioTrack() 136{ 137 if (mStatus == NO_ERROR) { 138 // Make sure that callback function exits in the case where 139 // it is looping on buffer full condition in obtainBuffer(). 140 // Otherwise the callback thread will never exit. 141 stop(); 142 if (mAudioTrackThread != 0) { 143 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 144 mAudioTrackThread->requestExitAndWait(); 145 mAudioTrackThread.clear(); 146 } 147 if (mAudioTrack != 0) { 148 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 149 mAudioTrack.clear(); 150 } 151 IPCThreadState::self()->flushCommands(); 152 AudioSystem::releaseAudioSessionId(mSessionId); 153 } 154} 155 156status_t AudioTrack::set( 157 audio_stream_type_t streamType, 158 uint32_t sampleRate, 159 audio_format_t format, 160 audio_channel_mask_t channelMask, 161 int frameCountInt, 162 audio_output_flags_t flags, 163 callback_t cbf, 164 void* user, 165 int notificationFrames, 166 const sp<IMemory>& sharedBuffer, 167 bool threadCanCallJava, 168 int sessionId, 169 transfer_type transferType, 170 const audio_offload_info_t *offloadInfo) 171{ 172 switch (transferType) { 173 case TRANSFER_DEFAULT: 174 if (sharedBuffer != 0) { 175 transferType = TRANSFER_SHARED; 176 } else if (cbf == NULL || threadCanCallJava) { 177 transferType = TRANSFER_SYNC; 178 } else { 179 transferType = TRANSFER_CALLBACK; 180 } 181 break; 182 case TRANSFER_CALLBACK: 183 if (cbf == NULL || sharedBuffer != 0) { 184 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 185 return BAD_VALUE; 186 } 187 break; 188 case TRANSFER_OBTAIN: 189 case TRANSFER_SYNC: 190 if (sharedBuffer != 0) { 191 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 192 return BAD_VALUE; 193 } 194 break; 195 case TRANSFER_SHARED: 196 if (sharedBuffer == 0) { 197 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 198 return BAD_VALUE; 199 } 200 break; 201 default: 202 ALOGE("Invalid transfer type %d", transferType); 203 return BAD_VALUE; 204 } 205 mTransfer = transferType; 206 207 // FIXME "int" here is legacy and will be replaced by size_t later 208 if (frameCountInt < 0) { 209 ALOGE("Invalid frame count %d", frameCountInt); 210 return BAD_VALUE; 211 } 212 size_t frameCount = frameCountInt; 213 214 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 215 sharedBuffer->size()); 216 217 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 218 219 AutoMutex lock(mLock); 220 221 if (mAudioTrack != 0) { 222 ALOGE("Track already in use"); 223 return INVALID_OPERATION; 224 } 225 226 // handle default values first. 227 if (streamType == AUDIO_STREAM_DEFAULT) { 228 streamType = AUDIO_STREAM_MUSIC; 229 } 230 231 if (sampleRate == 0) { 232 uint32_t afSampleRate; 233 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 234 return NO_INIT; 235 } 236 sampleRate = afSampleRate; 237 } 238 mSampleRate = sampleRate; 239 240 // these below should probably come from the audioFlinger too... 241 if (format == AUDIO_FORMAT_DEFAULT) { 242 format = AUDIO_FORMAT_PCM_16_BIT; 243 } 244 if (channelMask == 0) { 245 channelMask = AUDIO_CHANNEL_OUT_STEREO; 246 } 247 248 // validate parameters 249 if (!audio_is_valid_format(format)) { 250 ALOGE("Invalid format %d", format); 251 return BAD_VALUE; 252 } 253 254 // AudioFlinger does not currently support 8-bit data in shared memory 255 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 256 ALOGE("8-bit data in shared memory is not supported"); 257 return BAD_VALUE; 258 } 259 260 // force direct flag if format is not linear PCM 261 if (!audio_is_linear_pcm(format)) { 262 flags = (audio_output_flags_t) 263 // FIXME why can't we allow direct AND fast? 264 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 265 } 266 // only allow deep buffering for music stream type 267 if (streamType != AUDIO_STREAM_MUSIC) { 268 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 269 } 270 271 if (!audio_is_output_channel(channelMask)) { 272 ALOGE("Invalid channel mask %#x", channelMask); 273 return BAD_VALUE; 274 } 275 mChannelMask = channelMask; 276 uint32_t channelCount = popcount(channelMask); 277 mChannelCount = channelCount; 278 279 if (audio_is_linear_pcm(format)) { 280 mFrameSize = channelCount * audio_bytes_per_sample(format); 281 mFrameSizeAF = channelCount * sizeof(int16_t); 282 } else { 283 mFrameSize = sizeof(uint8_t); 284 mFrameSizeAF = sizeof(uint8_t); 285 } 286 287 audio_io_handle_t output = AudioSystem::getOutput( 288 streamType, 289 sampleRate, format, channelMask, 290 flags, 291 offloadInfo); 292 293 if (output == 0) { 294 ALOGE("Could not get audio output for stream type %d", streamType); 295 return BAD_VALUE; 296 } 297 298 mVolume[LEFT] = 1.0f; 299 mVolume[RIGHT] = 1.0f; 300 mSendLevel = 0.0f; 301 mFrameCount = frameCount; 302 mReqFrameCount = frameCount; 303 mNotificationFramesReq = notificationFrames; 304 mNotificationFramesAct = 0; 305 mSessionId = sessionId; 306 mAuxEffectId = 0; 307 mFlags = flags; 308 mCbf = cbf; 309 310 if (cbf != NULL) { 311 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 312 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 313 } 314 315 // create the IAudioTrack 316 status_t status = createTrack_l(streamType, 317 sampleRate, 318 format, 319 frameCount, 320 flags, 321 sharedBuffer, 322 output, 323 0 /*epoch*/); 324 325 if (status != NO_ERROR) { 326 if (mAudioTrackThread != 0) { 327 mAudioTrackThread->requestExit(); 328 mAudioTrackThread.clear(); 329 } 330 return status; 331 } 332 333 mStatus = NO_ERROR; 334 mStreamType = streamType; 335 mFormat = format; 336 mSharedBuffer = sharedBuffer; 337 mState = STATE_STOPPED; 338 mUserData = user; 339 mLoopPeriod = 0; 340 mMarkerPosition = 0; 341 mMarkerReached = false; 342 mNewPosition = 0; 343 mUpdatePeriod = 0; 344 AudioSystem::acquireAudioSessionId(mSessionId); 345 mSequence = 1; 346 mObservedSequence = mSequence; 347 mInUnderrun = false; 348 349 return NO_ERROR; 350} 351 352// ------------------------------------------------------------------------- 353 354void AudioTrack::start() 355{ 356 AutoMutex lock(mLock); 357 if (mState == STATE_ACTIVE) { 358 return; 359 } 360 361 mInUnderrun = true; 362 363 State previousState = mState; 364 mState = STATE_ACTIVE; 365 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 366 // reset current position as seen by client to 0 367 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 368 } 369 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 370 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->flags); 371 372 sp<AudioTrackThread> t = mAudioTrackThread; 373 if (t != 0) { 374 t->resume(); 375 } else { 376 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 377 get_sched_policy(0, &mPreviousSchedulingGroup); 378 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 379 } 380 381 status_t status = NO_ERROR; 382 if (!(flags & CBLK_INVALID)) { 383 status = mAudioTrack->start(); 384 if (status == DEAD_OBJECT) { 385 flags |= CBLK_INVALID; 386 } 387 } 388 if (flags & CBLK_INVALID) { 389 status = restoreTrack_l("start"); 390 } 391 392 if (status != NO_ERROR) { 393 ALOGE("start() status %d", status); 394 mState = previousState; 395 if (t != 0) { 396 t->pause(); 397 } else { 398 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 399 set_sched_policy(0, mPreviousSchedulingGroup); 400 } 401 } 402 403 // FIXME discarding status 404} 405 406void AudioTrack::stop() 407{ 408 AutoMutex lock(mLock); 409 // FIXME pause then stop should not be a nop 410 if (mState != STATE_ACTIVE) { 411 return; 412 } 413 414 mState = STATE_STOPPED; 415 mProxy->interrupt(); 416 mAudioTrack->stop(); 417 // the playback head position will reset to 0, so if a marker is set, we need 418 // to activate it again 419 mMarkerReached = false; 420#if 0 421 // Force flush if a shared buffer is used otherwise audioflinger 422 // will not stop before end of buffer is reached. 423 // It may be needed to make sure that we stop playback, likely in case looping is on. 424 if (mSharedBuffer != 0) { 425 flush_l(); 426 } 427#endif 428 sp<AudioTrackThread> t = mAudioTrackThread; 429 if (t != 0) { 430 t->pause(); 431 } else { 432 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 433 set_sched_policy(0, mPreviousSchedulingGroup); 434 } 435} 436 437bool AudioTrack::stopped() const 438{ 439 AutoMutex lock(mLock); 440 return mState != STATE_ACTIVE; 441} 442 443void AudioTrack::flush() 444{ 445 if (mSharedBuffer != 0) { 446 return; 447 } 448 AutoMutex lock(mLock); 449 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 450 return; 451 } 452 flush_l(); 453} 454 455void AudioTrack::flush_l() 456{ 457 ALOG_ASSERT(mState != STATE_ACTIVE); 458 459 // clear playback marker and periodic update counter 460 mMarkerPosition = 0; 461 mMarkerReached = false; 462 mUpdatePeriod = 0; 463 464 mState = STATE_FLUSHED; 465 mProxy->flush(); 466 mAudioTrack->flush(); 467} 468 469void AudioTrack::pause() 470{ 471 AutoMutex lock(mLock); 472 if (mState != STATE_ACTIVE) { 473 return; 474 } 475 mState = STATE_PAUSED; 476 mProxy->interrupt(); 477 mAudioTrack->pause(); 478} 479 480status_t AudioTrack::setVolume(float left, float right) 481{ 482 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 483 return BAD_VALUE; 484 } 485 486 AutoMutex lock(mLock); 487 mVolume[LEFT] = left; 488 mVolume[RIGHT] = right; 489 490 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 491 492 return NO_ERROR; 493} 494 495status_t AudioTrack::setVolume(float volume) 496{ 497 return setVolume(volume, volume); 498} 499 500status_t AudioTrack::setAuxEffectSendLevel(float level) 501{ 502 if (level < 0.0f || level > 1.0f) { 503 return BAD_VALUE; 504 } 505 506 AutoMutex lock(mLock); 507 mSendLevel = level; 508 mProxy->setSendLevel(level); 509 510 return NO_ERROR; 511} 512 513void AudioTrack::getAuxEffectSendLevel(float* level) const 514{ 515 if (level != NULL) { 516 *level = mSendLevel; 517 } 518} 519 520status_t AudioTrack::setSampleRate(uint32_t rate) 521{ 522 if (mIsTimed) { 523 return INVALID_OPERATION; 524 } 525 526 uint32_t afSamplingRate; 527 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 528 return NO_INIT; 529 } 530 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 531 if (rate == 0 || rate > afSamplingRate*2 ) { 532 return BAD_VALUE; 533 } 534 535 AutoMutex lock(mLock); 536 mSampleRate = rate; 537 mProxy->setSampleRate(rate); 538 539 return NO_ERROR; 540} 541 542uint32_t AudioTrack::getSampleRate() const 543{ 544 if (mIsTimed) { 545 return 0; 546 } 547 548 AutoMutex lock(mLock); 549 return mSampleRate; 550} 551 552status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 553{ 554 if (mSharedBuffer == 0 || mIsTimed) { 555 return INVALID_OPERATION; 556 } 557 558 if (loopCount == 0) { 559 ; 560 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 561 loopEnd - loopStart >= MIN_LOOP) { 562 ; 563 } else { 564 return BAD_VALUE; 565 } 566 567 AutoMutex lock(mLock); 568 // See setPosition() regarding setting parameters such as loop points or position while active 569 if (mState == STATE_ACTIVE) { 570 return INVALID_OPERATION; 571 } 572 setLoop_l(loopStart, loopEnd, loopCount); 573 return NO_ERROR; 574} 575 576void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 577{ 578 // FIXME If setting a loop also sets position to start of loop, then 579 // this is correct. Otherwise it should be removed. 580 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 581 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 582 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 583} 584 585status_t AudioTrack::setMarkerPosition(uint32_t marker) 586{ 587 if (mCbf == NULL) { 588 return INVALID_OPERATION; 589 } 590 591 AutoMutex lock(mLock); 592 mMarkerPosition = marker; 593 mMarkerReached = false; 594 595 return NO_ERROR; 596} 597 598status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 599{ 600 if (marker == NULL) { 601 return BAD_VALUE; 602 } 603 604 AutoMutex lock(mLock); 605 *marker = mMarkerPosition; 606 607 return NO_ERROR; 608} 609 610status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 611{ 612 if (mCbf == NULL) { 613 return INVALID_OPERATION; 614 } 615 616 AutoMutex lock(mLock); 617 mNewPosition = mProxy->getPosition() + updatePeriod; 618 mUpdatePeriod = updatePeriod; 619 620 return NO_ERROR; 621} 622 623status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 624{ 625 if (updatePeriod == NULL) { 626 return BAD_VALUE; 627 } 628 629 AutoMutex lock(mLock); 630 *updatePeriod = mUpdatePeriod; 631 632 return NO_ERROR; 633} 634 635status_t AudioTrack::setPosition(uint32_t position) 636{ 637 if (mSharedBuffer == 0 || mIsTimed) { 638 return INVALID_OPERATION; 639 } 640 if (position > mFrameCount) { 641 return BAD_VALUE; 642 } 643 644 AutoMutex lock(mLock); 645 // Currently we require that the player is inactive before setting parameters such as position 646 // or loop points. Otherwise, there could be a race condition: the application could read the 647 // current position, compute a new position or loop parameters, and then set that position or 648 // loop parameters but it would do the "wrong" thing since the position has continued to advance 649 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 650 // to specify how it wants to handle such scenarios. 651 if (mState == STATE_ACTIVE) { 652 return INVALID_OPERATION; 653 } 654 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 655 mLoopPeriod = 0; 656 // FIXME Check whether loops and setting position are incompatible in old code. 657 // If we use setLoop for both purposes we lose the capability to set the position while looping. 658 mStaticProxy->setLoop(position, mFrameCount, 0); 659 660 return NO_ERROR; 661} 662 663status_t AudioTrack::getPosition(uint32_t *position) const 664{ 665 if (position == NULL) { 666 return BAD_VALUE; 667 } 668 669 AutoMutex lock(mLock); 670 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 671 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 672 mProxy->getPosition(); 673 674 return NO_ERROR; 675} 676 677status_t AudioTrack::getBufferPosition(size_t *position) 678{ 679 if (mSharedBuffer == 0 || mIsTimed) { 680 return INVALID_OPERATION; 681 } 682 if (position == NULL) { 683 return BAD_VALUE; 684 } 685 686 AutoMutex lock(mLock); 687 *position = mStaticProxy->getBufferPosition(); 688 return NO_ERROR; 689} 690 691status_t AudioTrack::reload() 692{ 693 if (mSharedBuffer == 0 || mIsTimed) { 694 return INVALID_OPERATION; 695 } 696 697 AutoMutex lock(mLock); 698 // See setPosition() regarding setting parameters such as loop points or position while active 699 if (mState == STATE_ACTIVE) { 700 return INVALID_OPERATION; 701 } 702 mNewPosition = mUpdatePeriod; 703 mLoopPeriod = 0; 704 // FIXME The new code cannot reload while keeping a loop specified. 705 // Need to check how the old code handled this, and whether it's a significant change. 706 mStaticProxy->setLoop(0, mFrameCount, 0); 707 return NO_ERROR; 708} 709 710audio_io_handle_t AudioTrack::getOutput() 711{ 712 AutoMutex lock(mLock); 713 return getOutput_l(); 714} 715 716// must be called with mLock held 717audio_io_handle_t AudioTrack::getOutput_l() 718{ 719 return AudioSystem::getOutput(mStreamType, 720 mSampleRate, mFormat, mChannelMask, mFlags); 721} 722 723status_t AudioTrack::attachAuxEffect(int effectId) 724{ 725 AutoMutex lock(mLock); 726 status_t status = mAudioTrack->attachAuxEffect(effectId); 727 if (status == NO_ERROR) { 728 mAuxEffectId = effectId; 729 } 730 return status; 731} 732 733// ------------------------------------------------------------------------- 734 735// must be called with mLock held 736status_t AudioTrack::createTrack_l( 737 audio_stream_type_t streamType, 738 uint32_t sampleRate, 739 audio_format_t format, 740 size_t frameCount, 741 audio_output_flags_t flags, 742 const sp<IMemory>& sharedBuffer, 743 audio_io_handle_t output, 744 size_t epoch) 745{ 746 status_t status; 747 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 748 if (audioFlinger == 0) { 749 ALOGE("Could not get audioflinger"); 750 return NO_INIT; 751 } 752 753 uint32_t afLatency; 754 if ((status = AudioSystem::getLatency(output, streamType, &afLatency)) != NO_ERROR) { 755 ALOGE("getLatency(%d) failed status %d", output, status); 756 return NO_INIT; 757 } 758 759 // Client decides whether the track is TIMED (see below), but can only express a preference 760 // for FAST. Server will perform additional tests. 761 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 762 // either of these use cases: 763 // use case 1: shared buffer 764 (sharedBuffer != 0) || 765 // use case 2: callback handler 766 (mCbf != NULL))) { 767 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 768 // once denied, do not request again if IAudioTrack is re-created 769 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 770 mFlags = flags; 771 } 772 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 773 774 mNotificationFramesAct = mNotificationFramesReq; 775 776 if (!audio_is_linear_pcm(format)) { 777 778 if (sharedBuffer != 0) { 779 // Same comment as below about ignoring frameCount parameter for set() 780 frameCount = sharedBuffer->size(); 781 } else if (frameCount == 0) { 782 size_t afFrameCount; 783 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 784 if (status != NO_ERROR) { 785 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, 786 status); 787 return NO_INIT; 788 } 789 frameCount = afFrameCount; 790 } 791 792 } else if (sharedBuffer != 0) { 793 794 // Ensure that buffer alignment matches channel count 795 // 8-bit data in shared memory is not currently supported by AudioFlinger 796 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 797 if (mChannelCount > 1) { 798 // More than 2 channels does not require stronger alignment than stereo 799 alignment <<= 1; 800 } 801 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 802 ALOGE("Invalid buffer alignment: address %p, channel count %u", 803 sharedBuffer->pointer(), mChannelCount); 804 return BAD_VALUE; 805 } 806 807 // When initializing a shared buffer AudioTrack via constructors, 808 // there's no frameCount parameter. 809 // But when initializing a shared buffer AudioTrack via set(), 810 // there _is_ a frameCount parameter. We silently ignore it. 811 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 812 813 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 814 815 // FIXME move these calculations and associated checks to server 816 uint32_t afSampleRate; 817 status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); 818 if (status != NO_ERROR) { 819 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, 820 status); 821 return NO_INIT; 822 } 823 size_t afFrameCount; 824 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 825 if (status != NO_ERROR) { 826 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); 827 return NO_INIT; 828 } 829 830 // Ensure that buffer depth covers at least audio hardware latency 831 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 832 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 833 afFrameCount, minBufCount, afSampleRate, afLatency); 834 if (minBufCount <= 2) { 835 minBufCount = sampleRate == afSampleRate ? 2 : 3; 836 } 837 838 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 839 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 840 ", afLatency=%d", 841 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 842 843 if (frameCount == 0) { 844 frameCount = minFrameCount; 845 } 846 // Make sure that application is notified with sufficient margin 847 // before underrun 848 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) { 849 mNotificationFramesAct = frameCount/2; 850 } 851 if (frameCount < minFrameCount) { 852 // not ALOGW because it happens all the time when playing key clicks over A2DP 853 ALOGV("Minimum buffer size corrected from %d to %d", 854 frameCount, minFrameCount); 855 frameCount = minFrameCount; 856 } 857 858 } else { 859 // For fast tracks, the frame count calculations and checks are done by server 860 } 861 862 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 863 if (mIsTimed) { 864 trackFlags |= IAudioFlinger::TRACK_TIMED; 865 } 866 867 pid_t tid = -1; 868 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 869 trackFlags |= IAudioFlinger::TRACK_FAST; 870 if (mAudioTrackThread != 0) { 871 tid = mAudioTrackThread->getTid(); 872 } 873 } 874 875 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 876 sampleRate, 877 // AudioFlinger only sees 16-bit PCM 878 format == AUDIO_FORMAT_PCM_8_BIT ? 879 AUDIO_FORMAT_PCM_16_BIT : format, 880 mChannelMask, 881 frameCount, 882 &trackFlags, 883 sharedBuffer, 884 output, 885 tid, 886 &mSessionId, 887 &status); 888 889 if (track == 0) { 890 ALOGE("AudioFlinger could not create track, status: %d", status); 891 return status; 892 } 893 sp<IMemory> iMem = track->getCblk(); 894 if (iMem == 0) { 895 ALOGE("Could not get control block"); 896 return NO_INIT; 897 } 898 if (mAudioTrack != 0) { 899 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 900 mDeathNotifier.clear(); 901 } 902 mAudioTrack = track; 903 mCblkMemory = iMem; 904 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 905 mCblk = cblk; 906 size_t temp = cblk->frameCount_; 907 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 908 // In current design, AudioTrack client checks and ensures frame count validity before 909 // passing it to AudioFlinger so AudioFlinger should not return a different value except 910 // for fast track as it uses a special method of assigning frame count. 911 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 912 } 913 frameCount = temp; 914 mAwaitBoost = false; 915 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 916 if (trackFlags & IAudioFlinger::TRACK_FAST) { 917 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 918 mAwaitBoost = true; 919 if (sharedBuffer == 0) { 920 // double-buffering is not required for fast tracks, due to tighter scheduling 921 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount) { 922 mNotificationFramesAct = frameCount; 923 } 924 } 925 } else { 926 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 927 // once denied, do not request again if IAudioTrack is re-created 928 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 929 mFlags = flags; 930 if (sharedBuffer == 0) { 931 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) { 932 mNotificationFramesAct = frameCount/2; 933 } 934 } 935 } 936 } 937 mRefreshRemaining = true; 938 939 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 940 // is the value of pointer() for the shared buffer, otherwise buffers points 941 // immediately after the control block. This address is for the mapping within client 942 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 943 void* buffers; 944 if (sharedBuffer == 0) { 945 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 946 } else { 947 buffers = sharedBuffer->pointer(); 948 } 949 950 mAudioTrack->attachAuxEffect(mAuxEffectId); 951 // FIXME don't believe this lie 952 mLatency = afLatency + (1000*frameCount) / sampleRate; 953 mFrameCount = frameCount; 954 // If IAudioTrack is re-created, don't let the requested frameCount 955 // decrease. This can confuse clients that cache frameCount(). 956 if (frameCount > mReqFrameCount) { 957 mReqFrameCount = frameCount; 958 } 959 960 // update proxy 961 if (sharedBuffer == 0) { 962 mStaticProxy.clear(); 963 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 964 } else { 965 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 966 mProxy = mStaticProxy; 967 } 968 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 969 uint16_t(mVolume[LEFT] * 0x1000)); 970 mProxy->setSendLevel(mSendLevel); 971 mProxy->setSampleRate(mSampleRate); 972 mProxy->setEpoch(epoch); 973 mProxy->setMinimum(mNotificationFramesAct); 974 975 mDeathNotifier = new DeathNotifier(this); 976 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 977 978 return NO_ERROR; 979} 980 981status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 982{ 983 if (audioBuffer == NULL) { 984 return BAD_VALUE; 985 } 986 if (mTransfer != TRANSFER_OBTAIN) { 987 audioBuffer->frameCount = 0; 988 audioBuffer->size = 0; 989 audioBuffer->raw = NULL; 990 return INVALID_OPERATION; 991 } 992 993 const struct timespec *requested; 994 if (waitCount == -1) { 995 requested = &ClientProxy::kForever; 996 } else if (waitCount == 0) { 997 requested = &ClientProxy::kNonBlocking; 998 } else if (waitCount > 0) { 999 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1000 struct timespec timeout; 1001 timeout.tv_sec = ms / 1000; 1002 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1003 requested = &timeout; 1004 } else { 1005 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1006 requested = NULL; 1007 } 1008 return obtainBuffer(audioBuffer, requested); 1009} 1010 1011status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1012 struct timespec *elapsed, size_t *nonContig) 1013{ 1014 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1015 uint32_t oldSequence = 0; 1016 uint32_t newSequence; 1017 1018 Proxy::Buffer buffer; 1019 status_t status = NO_ERROR; 1020 1021 static const int32_t kMaxTries = 5; 1022 int32_t tryCounter = kMaxTries; 1023 1024 do { 1025 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1026 // keep them from going away if another thread re-creates the track during obtainBuffer() 1027 sp<AudioTrackClientProxy> proxy; 1028 sp<IMemory> iMem; 1029 1030 { // start of lock scope 1031 AutoMutex lock(mLock); 1032 1033 newSequence = mSequence; 1034 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1035 if (status == DEAD_OBJECT) { 1036 // re-create track, unless someone else has already done so 1037 if (newSequence == oldSequence) { 1038 status = restoreTrack_l("obtainBuffer"); 1039 if (status != NO_ERROR) { 1040 break; 1041 } 1042 } 1043 } 1044 oldSequence = newSequence; 1045 1046 // Keep the extra references 1047 proxy = mProxy; 1048 iMem = mCblkMemory; 1049 1050 // Non-blocking if track is stopped or paused 1051 if (mState != STATE_ACTIVE) { 1052 requested = &ClientProxy::kNonBlocking; 1053 } 1054 1055 } // end of lock scope 1056 1057 buffer.mFrameCount = audioBuffer->frameCount; 1058 // FIXME starts the requested timeout and elapsed over from scratch 1059 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1060 1061 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1062 1063 audioBuffer->frameCount = buffer.mFrameCount; 1064 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1065 audioBuffer->raw = buffer.mRaw; 1066 if (nonContig != NULL) { 1067 *nonContig = buffer.mNonContig; 1068 } 1069 return status; 1070} 1071 1072void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1073{ 1074 if (mTransfer == TRANSFER_SHARED) { 1075 return; 1076 } 1077 1078 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1079 if (stepCount == 0) { 1080 return; 1081 } 1082 1083 Proxy::Buffer buffer; 1084 buffer.mFrameCount = stepCount; 1085 buffer.mRaw = audioBuffer->raw; 1086 1087 AutoMutex lock(mLock); 1088 mInUnderrun = false; 1089 mProxy->releaseBuffer(&buffer); 1090 1091 // restart track if it was disabled by audioflinger due to previous underrun 1092 if (mState == STATE_ACTIVE) { 1093 audio_track_cblk_t* cblk = mCblk; 1094 if (android_atomic_and(~CBLK_DISABLED, &cblk->flags) & CBLK_DISABLED) { 1095 ALOGW("releaseBuffer() track %p name=%#x disabled due to previous underrun, restarting", 1096 this, cblk->mName); 1097 // FIXME ignoring status 1098 mAudioTrack->start(); 1099 } 1100 } 1101} 1102 1103// ------------------------------------------------------------------------- 1104 1105ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1106{ 1107 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1108 return INVALID_OPERATION; 1109 } 1110 1111 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1112 // Sanity-check: user is most-likely passing an error code, and it would 1113 // make the return value ambiguous (actualSize vs error). 1114 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1115 return BAD_VALUE; 1116 } 1117 1118 size_t written = 0; 1119 Buffer audioBuffer; 1120 1121 while (userSize >= mFrameSize) { 1122 audioBuffer.frameCount = userSize / mFrameSize; 1123 1124 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1125 if (err < 0) { 1126 if (written > 0) { 1127 break; 1128 } 1129 return ssize_t(err); 1130 } 1131 1132 size_t toWrite; 1133 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1134 // Divide capacity by 2 to take expansion into account 1135 toWrite = audioBuffer.size >> 1; 1136 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1137 } else { 1138 toWrite = audioBuffer.size; 1139 memcpy(audioBuffer.i8, buffer, toWrite); 1140 } 1141 buffer = ((const char *) buffer) + toWrite; 1142 userSize -= toWrite; 1143 written += toWrite; 1144 1145 releaseBuffer(&audioBuffer); 1146 } 1147 1148 return written; 1149} 1150 1151// ------------------------------------------------------------------------- 1152 1153TimedAudioTrack::TimedAudioTrack() { 1154 mIsTimed = true; 1155} 1156 1157status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1158{ 1159 AutoMutex lock(mLock); 1160 status_t result = UNKNOWN_ERROR; 1161 1162#if 1 1163 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1164 // while we are accessing the cblk 1165 sp<IAudioTrack> audioTrack = mAudioTrack; 1166 sp<IMemory> iMem = mCblkMemory; 1167#endif 1168 1169 // If the track is not invalid already, try to allocate a buffer. alloc 1170 // fails indicating that the server is dead, flag the track as invalid so 1171 // we can attempt to restore in just a bit. 1172 audio_track_cblk_t* cblk = mCblk; 1173 if (!(cblk->flags & CBLK_INVALID)) { 1174 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1175 if (result == DEAD_OBJECT) { 1176 android_atomic_or(CBLK_INVALID, &cblk->flags); 1177 } 1178 } 1179 1180 // If the track is invalid at this point, attempt to restore it. and try the 1181 // allocation one more time. 1182 if (cblk->flags & CBLK_INVALID) { 1183 result = restoreTrack_l("allocateTimedBuffer"); 1184 1185 if (result == NO_ERROR) { 1186 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1187 } 1188 } 1189 1190 return result; 1191} 1192 1193status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1194 int64_t pts) 1195{ 1196 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1197 { 1198 AutoMutex lock(mLock); 1199 audio_track_cblk_t* cblk = mCblk; 1200 // restart track if it was disabled by audioflinger due to previous underrun 1201 if (buffer->size() != 0 && status == NO_ERROR && 1202 (mState == STATE_ACTIVE) && (cblk->flags & CBLK_DISABLED)) { 1203 android_atomic_and(~CBLK_DISABLED, &cblk->flags); 1204 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1205 // FIXME ignoring status 1206 mAudioTrack->start(); 1207 } 1208 } 1209 return status; 1210} 1211 1212status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1213 TargetTimeline target) 1214{ 1215 return mAudioTrack->setMediaTimeTransform(xform, target); 1216} 1217 1218// ------------------------------------------------------------------------- 1219 1220nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1221{ 1222 mLock.lock(); 1223 if (mAwaitBoost) { 1224 mAwaitBoost = false; 1225 mLock.unlock(); 1226 static const int32_t kMaxTries = 5; 1227 int32_t tryCounter = kMaxTries; 1228 uint32_t pollUs = 10000; 1229 do { 1230 int policy = sched_getscheduler(0); 1231 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1232 break; 1233 } 1234 usleep(pollUs); 1235 pollUs <<= 1; 1236 } while (tryCounter-- > 0); 1237 if (tryCounter < 0) { 1238 ALOGE("did not receive expected priority boost on time"); 1239 } 1240 return true; 1241 } 1242 1243 // Can only reference mCblk while locked 1244 int32_t flags = android_atomic_and( 1245 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->flags); 1246 1247 // Check for track invalidation 1248 if (flags & CBLK_INVALID) { 1249 (void) restoreTrack_l("processAudioBuffer"); 1250 mLock.unlock(); 1251 // Run again immediately, but with a new IAudioTrack 1252 return 0; 1253 } 1254 1255 bool active = mState == STATE_ACTIVE; 1256 1257 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1258 bool newUnderrun = false; 1259 if (flags & CBLK_UNDERRUN) { 1260#if 0 1261 // Currently in shared buffer mode, when the server reaches the end of buffer, 1262 // the track stays active in continuous underrun state. It's up to the application 1263 // to pause or stop the track, or set the position to a new offset within buffer. 1264 // This was some experimental code to auto-pause on underrun. Keeping it here 1265 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1266 if (mTransfer == TRANSFER_SHARED) { 1267 mState = STATE_PAUSED; 1268 active = false; 1269 } 1270#endif 1271 if (!mInUnderrun) { 1272 mInUnderrun = true; 1273 newUnderrun = true; 1274 } 1275 } 1276 1277 // Get current position of server 1278 size_t position = mProxy->getPosition(); 1279 1280 // Manage marker callback 1281 bool markerReached = false; 1282 size_t markerPosition = mMarkerPosition; 1283 // FIXME fails for wraparound, need 64 bits 1284 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1285 mMarkerReached = markerReached = true; 1286 } 1287 1288 // Determine number of new position callback(s) that will be needed, while locked 1289 size_t newPosCount = 0; 1290 size_t newPosition = mNewPosition; 1291 size_t updatePeriod = mUpdatePeriod; 1292 // FIXME fails for wraparound, need 64 bits 1293 if (updatePeriod > 0 && position >= newPosition) { 1294 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1295 mNewPosition += updatePeriod * newPosCount; 1296 } 1297 1298 // Cache other fields that will be needed soon 1299 uint32_t loopPeriod = mLoopPeriod; 1300 uint32_t sampleRate = mSampleRate; 1301 size_t notificationFrames = mNotificationFramesAct; 1302 if (mRefreshRemaining) { 1303 mRefreshRemaining = false; 1304 mRemainingFrames = notificationFrames; 1305 mRetryOnPartialBuffer = false; 1306 } 1307 size_t misalignment = mProxy->getMisalignment(); 1308 int32_t sequence = mSequence; 1309 1310 // These fields don't need to be cached, because they are assigned only by set(): 1311 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1312 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1313 1314 mLock.unlock(); 1315 1316 // perform callbacks while unlocked 1317 if (newUnderrun) { 1318 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1319 } 1320 // FIXME we will miss loops if loop cycle was signaled several times since last call 1321 // to processAudioBuffer() 1322 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1323 mCbf(EVENT_LOOP_END, mUserData, NULL); 1324 } 1325 if (flags & CBLK_BUFFER_END) { 1326 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1327 } 1328 if (markerReached) { 1329 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1330 } 1331 while (newPosCount > 0) { 1332 size_t temp = newPosition; 1333 mCbf(EVENT_NEW_POS, mUserData, &temp); 1334 newPosition += updatePeriod; 1335 newPosCount--; 1336 } 1337 if (mObservedSequence != sequence) { 1338 mObservedSequence = sequence; 1339 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1340 } 1341 1342 // if inactive, then don't run me again until re-started 1343 if (!active) { 1344 return NS_INACTIVE; 1345 } 1346 1347 // Compute the estimated time until the next timed event (position, markers, loops) 1348 // FIXME only for non-compressed audio 1349 uint32_t minFrames = ~0; 1350 if (!markerReached && position < markerPosition) { 1351 minFrames = markerPosition - position; 1352 } 1353 if (loopPeriod > 0 && loopPeriod < minFrames) { 1354 minFrames = loopPeriod; 1355 } 1356 if (updatePeriod > 0 && updatePeriod < minFrames) { 1357 minFrames = updatePeriod; 1358 } 1359 1360 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1361 static const uint32_t kPoll = 0; 1362 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1363 minFrames = kPoll * notificationFrames; 1364 } 1365 1366 // Convert frame units to time units 1367 nsecs_t ns = NS_WHENEVER; 1368 if (minFrames != (uint32_t) ~0) { 1369 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1370 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1371 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1372 } 1373 1374 // If not supplying data by EVENT_MORE_DATA, then we're done 1375 if (mTransfer != TRANSFER_CALLBACK) { 1376 return ns; 1377 } 1378 1379 struct timespec timeout; 1380 const struct timespec *requested = &ClientProxy::kForever; 1381 if (ns != NS_WHENEVER) { 1382 timeout.tv_sec = ns / 1000000000LL; 1383 timeout.tv_nsec = ns % 1000000000LL; 1384 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1385 requested = &timeout; 1386 } 1387 1388 while (mRemainingFrames > 0) { 1389 1390 Buffer audioBuffer; 1391 audioBuffer.frameCount = mRemainingFrames; 1392 size_t nonContig; 1393 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1394 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1395 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1396 requested = &ClientProxy::kNonBlocking; 1397 size_t avail = audioBuffer.frameCount + nonContig; 1398 ALOGV("obtainBuffer(%u) returned %u = %u + %u", 1399 mRemainingFrames, avail, audioBuffer.frameCount, nonContig); 1400 if (err != NO_ERROR) { 1401 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { 1402 return 0; 1403 } 1404 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1405 return NS_NEVER; 1406 } 1407 1408 if (mRetryOnPartialBuffer) { 1409 mRetryOnPartialBuffer = false; 1410 if (avail < mRemainingFrames) { 1411 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1412 if (ns < 0 || myns < ns) { 1413 ns = myns; 1414 } 1415 return ns; 1416 } 1417 } 1418 1419 // Divide buffer size by 2 to take into account the expansion 1420 // due to 8 to 16 bit conversion: the callback must fill only half 1421 // of the destination buffer 1422 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1423 audioBuffer.size >>= 1; 1424 } 1425 1426 size_t reqSize = audioBuffer.size; 1427 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1428 size_t writtenSize = audioBuffer.size; 1429 size_t writtenFrames = writtenSize / mFrameSize; 1430 1431 // Sanity check on returned size 1432 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1433 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1434 reqSize, (int) writtenSize); 1435 return NS_NEVER; 1436 } 1437 1438 if (writtenSize == 0) { 1439 // The callback is done filling buffers 1440 // Keep this thread going to handle timed events and 1441 // still try to get more data in intervals of WAIT_PERIOD_MS 1442 // but don't just loop and block the CPU, so wait 1443 return WAIT_PERIOD_MS * 1000000LL; 1444 } 1445 1446 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1447 // 8 to 16 bit conversion, note that source and destination are the same address 1448 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1449 audioBuffer.size <<= 1; 1450 } 1451 1452 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1453 audioBuffer.frameCount = releasedFrames; 1454 mRemainingFrames -= releasedFrames; 1455 if (misalignment >= releasedFrames) { 1456 misalignment -= releasedFrames; 1457 } else { 1458 misalignment = 0; 1459 } 1460 1461 releaseBuffer(&audioBuffer); 1462 1463 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1464 // if callback doesn't like to accept the full chunk 1465 if (writtenSize < reqSize) { 1466 continue; 1467 } 1468 1469 // There could be enough non-contiguous frames available to satisfy the remaining request 1470 if (mRemainingFrames <= nonContig) { 1471 continue; 1472 } 1473 1474#if 0 1475 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1476 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1477 // that total to a sum == notificationFrames. 1478 if (0 < misalignment && misalignment <= mRemainingFrames) { 1479 mRemainingFrames = misalignment; 1480 return (mRemainingFrames * 1100000000LL) / sampleRate; 1481 } 1482#endif 1483 1484 } 1485 mRemainingFrames = notificationFrames; 1486 mRetryOnPartialBuffer = true; 1487 1488 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1489 return 0; 1490} 1491 1492status_t AudioTrack::restoreTrack_l(const char *from) 1493{ 1494 ALOGW("dead IAudioTrack, creating a new one from %s()", from); 1495 ++mSequence; 1496 status_t result; 1497 1498 // refresh the audio configuration cache in this process to make sure we get new 1499 // output parameters in getOutput_l() and createTrack_l() 1500 AudioSystem::clearAudioConfigCache(); 1501 1502 // if the new IAudioTrack is created, createTrack_l() will modify the 1503 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1504 // It will also delete the strong references on previous IAudioTrack and IMemory 1505 size_t position = mProxy->getPosition(); 1506 mNewPosition = position + mUpdatePeriod; 1507 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1508 result = createTrack_l(mStreamType, 1509 mSampleRate, 1510 mFormat, 1511 mReqFrameCount, // so that frame count never goes down 1512 mFlags, 1513 mSharedBuffer, 1514 getOutput_l(), 1515 position /*epoch*/); 1516 1517 if (result == NO_ERROR) { 1518 // continue playback from last known position, but 1519 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1520 if (mStaticProxy != NULL) { 1521 mLoopPeriod = 0; 1522 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1523 } 1524 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1525 // track destruction have been played? This is critical for SoundPool implementation 1526 // This must be broken, and needs to be tested/debugged. 1527#if 0 1528 // restore write index and set other indexes to reflect empty buffer status 1529 if (!strcmp(from, "start")) { 1530 // Make sure that a client relying on callback events indicating underrun or 1531 // the actual amount of audio frames played (e.g SoundPool) receives them. 1532 if (mSharedBuffer == 0) { 1533 // restart playback even if buffer is not completely filled. 1534 android_atomic_or(CBLK_FORCEREADY, &mCblk->flags); 1535 } 1536 } 1537#endif 1538 if (mState == STATE_ACTIVE) { 1539 result = mAudioTrack->start(); 1540 } 1541 } 1542 if (result != NO_ERROR) { 1543 ALOGW("restoreTrack_l() failed status %d", result); 1544 mState = STATE_STOPPED; 1545 } 1546 1547 return result; 1548} 1549 1550status_t AudioTrack::setParameters(const String8& keyValuePairs) 1551{ 1552 AutoMutex lock(mLock); 1553 if (mAudioTrack != 0) { 1554 return mAudioTrack->setParameters(keyValuePairs); 1555 } else { 1556 return NO_INIT; 1557 } 1558} 1559 1560String8 AudioTrack::getParameters(const String8& keys) 1561{ 1562 return String8::empty(); 1563} 1564 1565status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1566{ 1567 1568 const size_t SIZE = 256; 1569 char buffer[SIZE]; 1570 String8 result; 1571 1572 result.append(" AudioTrack::dump\n"); 1573 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1574 mVolume[0], mVolume[1]); 1575 result.append(buffer); 1576 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1577 mChannelCount, mFrameCount); 1578 result.append(buffer); 1579 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1580 result.append(buffer); 1581 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1582 result.append(buffer); 1583 ::write(fd, result.string(), result.size()); 1584 return NO_ERROR; 1585} 1586 1587uint32_t AudioTrack::getUnderrunFrames() const 1588{ 1589 AutoMutex lock(mLock); 1590 return mProxy->getUnderrunFrames(); 1591} 1592 1593// ========================================================================= 1594 1595void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who) 1596{ 1597 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1598 if (audioTrack != 0) { 1599 AutoMutex lock(audioTrack->mLock); 1600 audioTrack->mProxy->binderDied(); 1601 } 1602} 1603 1604// ========================================================================= 1605 1606AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1607 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mResumeLatch(false) 1608{ 1609} 1610 1611AudioTrack::AudioTrackThread::~AudioTrackThread() 1612{ 1613} 1614 1615bool AudioTrack::AudioTrackThread::threadLoop() 1616{ 1617 { 1618 AutoMutex _l(mMyLock); 1619 if (mPaused) { 1620 mMyCond.wait(mMyLock); 1621 // caller will check for exitPending() 1622 return true; 1623 } 1624 } 1625 nsecs_t ns = mReceiver.processAudioBuffer(this); 1626 switch (ns) { 1627 case 0: 1628 return true; 1629 case NS_WHENEVER: 1630 sleep(1); 1631 return true; 1632 case NS_INACTIVE: 1633 pauseConditional(); 1634 return true; 1635 case NS_NEVER: 1636 return false; 1637 default: 1638 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1639 struct timespec req; 1640 req.tv_sec = ns / 1000000000LL; 1641 req.tv_nsec = ns % 1000000000LL; 1642 nanosleep(&req, NULL /*rem*/); 1643 return true; 1644 } 1645} 1646 1647void AudioTrack::AudioTrackThread::requestExit() 1648{ 1649 // must be in this order to avoid a race condition 1650 Thread::requestExit(); 1651 resume(); 1652} 1653 1654void AudioTrack::AudioTrackThread::pause() 1655{ 1656 AutoMutex _l(mMyLock); 1657 mPaused = true; 1658 mResumeLatch = false; 1659} 1660 1661void AudioTrack::AudioTrackThread::pauseConditional() 1662{ 1663 AutoMutex _l(mMyLock); 1664 if (mResumeLatch) { 1665 mResumeLatch = false; 1666 } else { 1667 mPaused = true; 1668 } 1669} 1670 1671void AudioTrack::AudioTrackThread::resume() 1672{ 1673 AutoMutex _l(mMyLock); 1674 if (mPaused) { 1675 mPaused = false; 1676 mResumeLatch = false; 1677 mMyCond.signal(); 1678 } else { 1679 mResumeLatch = true; 1680 } 1681} 1682 1683}; // namespace android 1684