AudioTrack.cpp revision 9f80dd223d83d9bb9077fb6baee056cee4eaf7e5
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 : mStatus(NO_INIT), 102 mIsTimed(false), 103 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 104 mPreviousSchedulingGroup(SP_DEFAULT) 105{ 106 mStatus = set(streamType, sampleRate, format, channelMask, 107 frameCount, flags, cbf, user, notificationFrames, 108 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType); 109} 110 111AudioTrack::AudioTrack( 112 audio_stream_type_t streamType, 113 uint32_t sampleRate, 114 audio_format_t format, 115 audio_channel_mask_t channelMask, 116 const sp<IMemory>& sharedBuffer, 117 audio_output_flags_t flags, 118 callback_t cbf, 119 void* user, 120 int notificationFrames, 121 int sessionId, 122 transfer_type transferType) 123 : mStatus(NO_INIT), 124 mIsTimed(false), 125 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 126 mPreviousSchedulingGroup(SP_DEFAULT) 127{ 128 mStatus = set(streamType, sampleRate, format, channelMask, 129 0 /*frameCount*/, flags, cbf, user, notificationFrames, 130 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType); 131} 132 133AudioTrack::~AudioTrack() 134{ 135 if (mStatus == NO_ERROR) { 136 // Make sure that callback function exits in the case where 137 // it is looping on buffer full condition in obtainBuffer(). 138 // Otherwise the callback thread will never exit. 139 stop(); 140 if (mAudioTrackThread != 0) { 141 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 142 mAudioTrackThread->requestExitAndWait(); 143 mAudioTrackThread.clear(); 144 } 145 if (mAudioTrack != 0) { 146 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 147 mAudioTrack.clear(); 148 } 149 IPCThreadState::self()->flushCommands(); 150 AudioSystem::releaseAudioSessionId(mSessionId); 151 } 152} 153 154status_t AudioTrack::set( 155 audio_stream_type_t streamType, 156 uint32_t sampleRate, 157 audio_format_t format, 158 audio_channel_mask_t channelMask, 159 int frameCountInt, 160 audio_output_flags_t flags, 161 callback_t cbf, 162 void* user, 163 int notificationFrames, 164 const sp<IMemory>& sharedBuffer, 165 bool threadCanCallJava, 166 int sessionId, 167 transfer_type transferType) 168{ 169 switch (transferType) { 170 case TRANSFER_DEFAULT: 171 if (sharedBuffer != 0) { 172 transferType = TRANSFER_SHARED; 173 } else if (cbf == NULL || threadCanCallJava) { 174 transferType = TRANSFER_SYNC; 175 } else { 176 transferType = TRANSFER_CALLBACK; 177 } 178 break; 179 case TRANSFER_CALLBACK: 180 if (cbf == NULL || sharedBuffer != 0) { 181 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 182 return BAD_VALUE; 183 } 184 break; 185 case TRANSFER_OBTAIN: 186 case TRANSFER_SYNC: 187 if (sharedBuffer != 0) { 188 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 189 return BAD_VALUE; 190 } 191 break; 192 case TRANSFER_SHARED: 193 if (sharedBuffer == 0) { 194 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 195 return BAD_VALUE; 196 } 197 break; 198 default: 199 ALOGE("Invalid transfer type %d", transferType); 200 return BAD_VALUE; 201 } 202 mTransfer = transferType; 203 204 // FIXME "int" here is legacy and will be replaced by size_t later 205 if (frameCountInt < 0) { 206 ALOGE("Invalid frame count %d", frameCountInt); 207 return BAD_VALUE; 208 } 209 size_t frameCount = frameCountInt; 210 211 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 212 sharedBuffer->size()); 213 214 ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags); 215 216 AutoMutex lock(mLock); 217 218 if (mAudioTrack != 0) { 219 ALOGE("Track already in use"); 220 return INVALID_OPERATION; 221 } 222 223 // handle default values first. 224 if (streamType == AUDIO_STREAM_DEFAULT) { 225 streamType = AUDIO_STREAM_MUSIC; 226 } 227 228 if (sampleRate == 0) { 229 uint32_t afSampleRate; 230 if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) { 231 return NO_INIT; 232 } 233 sampleRate = afSampleRate; 234 } 235 mSampleRate = sampleRate; 236 237 // these below should probably come from the audioFlinger too... 238 if (format == AUDIO_FORMAT_DEFAULT) { 239 format = AUDIO_FORMAT_PCM_16_BIT; 240 } 241 if (channelMask == 0) { 242 channelMask = AUDIO_CHANNEL_OUT_STEREO; 243 } 244 245 // validate parameters 246 if (!audio_is_valid_format(format)) { 247 ALOGE("Invalid format %d", format); 248 return BAD_VALUE; 249 } 250 251 // AudioFlinger does not currently support 8-bit data in shared memory 252 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 253 ALOGE("8-bit data in shared memory is not supported"); 254 return BAD_VALUE; 255 } 256 257 // force direct flag if format is not linear PCM 258 if (!audio_is_linear_pcm(format)) { 259 flags = (audio_output_flags_t) 260 // FIXME why can't we allow direct AND fast? 261 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 262 } 263 // only allow deep buffering for music stream type 264 if (streamType != AUDIO_STREAM_MUSIC) { 265 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 266 } 267 268 if (!audio_is_output_channel(channelMask)) { 269 ALOGE("Invalid channel mask %#x", channelMask); 270 return BAD_VALUE; 271 } 272 mChannelMask = channelMask; 273 uint32_t channelCount = popcount(channelMask); 274 mChannelCount = channelCount; 275 276 if (audio_is_linear_pcm(format)) { 277 mFrameSize = channelCount * audio_bytes_per_sample(format); 278 mFrameSizeAF = channelCount * sizeof(int16_t); 279 } else { 280 mFrameSize = sizeof(uint8_t); 281 mFrameSizeAF = sizeof(uint8_t); 282 } 283 284 audio_io_handle_t output = AudioSystem::getOutput( 285 streamType, 286 sampleRate, format, channelMask, 287 flags); 288 289 if (output == 0) { 290 ALOGE("Could not get audio output for stream type %d", streamType); 291 return BAD_VALUE; 292 } 293 294 mVolume[LEFT] = 1.0f; 295 mVolume[RIGHT] = 1.0f; 296 mSendLevel = 0.0f; 297 mFrameCount = frameCount; 298 mReqFrameCount = frameCount; 299 mNotificationFramesReq = notificationFrames; 300 mNotificationFramesAct = 0; 301 mSessionId = sessionId; 302 mAuxEffectId = 0; 303 mFlags = flags; 304 mCbf = cbf; 305 306 if (cbf != NULL) { 307 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 308 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 309 } 310 311 // create the IAudioTrack 312 status_t status = createTrack_l(streamType, 313 sampleRate, 314 format, 315 frameCount, 316 flags, 317 sharedBuffer, 318 output, 319 0 /*epoch*/); 320 321 if (status != NO_ERROR) { 322 if (mAudioTrackThread != 0) { 323 mAudioTrackThread->requestExit(); 324 mAudioTrackThread.clear(); 325 } 326 return status; 327 } 328 329 mStatus = NO_ERROR; 330 mStreamType = streamType; 331 mFormat = format; 332 mSharedBuffer = sharedBuffer; 333 mState = STATE_STOPPED; 334 mUserData = user; 335 mLoopPeriod = 0; 336 mMarkerPosition = 0; 337 mMarkerReached = false; 338 mNewPosition = 0; 339 mUpdatePeriod = 0; 340 AudioSystem::acquireAudioSessionId(mSessionId); 341 mSequence = 1; 342 mObservedSequence = mSequence; 343 mInUnderrun = false; 344 345 return NO_ERROR; 346} 347 348// ------------------------------------------------------------------------- 349 350void AudioTrack::start() 351{ 352 AutoMutex lock(mLock); 353 if (mState == STATE_ACTIVE) { 354 return; 355 } 356 357 mInUnderrun = true; 358 359 State previousState = mState; 360 mState = STATE_ACTIVE; 361 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 362 // reset current position as seen by client to 0 363 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 364 } 365 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 366 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->flags); 367 368 sp<AudioTrackThread> t = mAudioTrackThread; 369 if (t != 0) { 370 t->resume(); 371 } else { 372 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 373 get_sched_policy(0, &mPreviousSchedulingGroup); 374 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 375 } 376 377 status_t status = NO_ERROR; 378 if (!(flags & CBLK_INVALID)) { 379 status = mAudioTrack->start(); 380 if (status == DEAD_OBJECT) { 381 flags |= CBLK_INVALID; 382 } 383 } 384 if (flags & CBLK_INVALID) { 385 status = restoreTrack_l("start"); 386 } 387 388 if (status != NO_ERROR) { 389 ALOGE("start() status %d", status); 390 mState = previousState; 391 if (t != 0) { 392 t->pause(); 393 } else { 394 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 395 set_sched_policy(0, mPreviousSchedulingGroup); 396 } 397 } 398 399 // FIXME discarding status 400} 401 402void AudioTrack::stop() 403{ 404 AutoMutex lock(mLock); 405 // FIXME pause then stop should not be a nop 406 if (mState != STATE_ACTIVE) { 407 return; 408 } 409 410 mState = STATE_STOPPED; 411 mProxy->interrupt(); 412 mAudioTrack->stop(); 413 // the playback head position will reset to 0, so if a marker is set, we need 414 // to activate it again 415 mMarkerReached = false; 416#if 0 417 // Force flush if a shared buffer is used otherwise audioflinger 418 // will not stop before end of buffer is reached. 419 // It may be needed to make sure that we stop playback, likely in case looping is on. 420 if (mSharedBuffer != 0) { 421 flush_l(); 422 } 423#endif 424 sp<AudioTrackThread> t = mAudioTrackThread; 425 if (t != 0) { 426 t->pause(); 427 } else { 428 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 429 set_sched_policy(0, mPreviousSchedulingGroup); 430 } 431} 432 433bool AudioTrack::stopped() const 434{ 435 AutoMutex lock(mLock); 436 return mState != STATE_ACTIVE; 437} 438 439void AudioTrack::flush() 440{ 441 if (mSharedBuffer != 0) { 442 return; 443 } 444 AutoMutex lock(mLock); 445 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 446 return; 447 } 448 flush_l(); 449} 450 451void AudioTrack::flush_l() 452{ 453 ALOG_ASSERT(mState != STATE_ACTIVE); 454 455 // clear playback marker and periodic update counter 456 mMarkerPosition = 0; 457 mMarkerReached = false; 458 mUpdatePeriod = 0; 459 460 mState = STATE_FLUSHED; 461 mProxy->flush(); 462 mAudioTrack->flush(); 463} 464 465void AudioTrack::pause() 466{ 467 AutoMutex lock(mLock); 468 if (mState != STATE_ACTIVE) { 469 return; 470 } 471 mState = STATE_PAUSED; 472 mProxy->interrupt(); 473 mAudioTrack->pause(); 474} 475 476status_t AudioTrack::setVolume(float left, float right) 477{ 478 if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) { 479 return BAD_VALUE; 480 } 481 482 AutoMutex lock(mLock); 483 mVolume[LEFT] = left; 484 mVolume[RIGHT] = right; 485 486 mProxy->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000)); 487 488 return NO_ERROR; 489} 490 491status_t AudioTrack::setVolume(float volume) 492{ 493 return setVolume(volume, volume); 494} 495 496status_t AudioTrack::setAuxEffectSendLevel(float level) 497{ 498 if (level < 0.0f || level > 1.0f) { 499 return BAD_VALUE; 500 } 501 502 AutoMutex lock(mLock); 503 mSendLevel = level; 504 mProxy->setSendLevel(level); 505 506 return NO_ERROR; 507} 508 509void AudioTrack::getAuxEffectSendLevel(float* level) const 510{ 511 if (level != NULL) { 512 *level = mSendLevel; 513 } 514} 515 516status_t AudioTrack::setSampleRate(uint32_t rate) 517{ 518 if (mIsTimed) { 519 return INVALID_OPERATION; 520 } 521 522 uint32_t afSamplingRate; 523 if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) { 524 return NO_INIT; 525 } 526 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 527 if (rate == 0 || rate > afSamplingRate*2 ) { 528 return BAD_VALUE; 529 } 530 531 AutoMutex lock(mLock); 532 mSampleRate = rate; 533 mProxy->setSampleRate(rate); 534 535 return NO_ERROR; 536} 537 538uint32_t AudioTrack::getSampleRate() const 539{ 540 if (mIsTimed) { 541 return 0; 542 } 543 544 AutoMutex lock(mLock); 545 return mSampleRate; 546} 547 548status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 549{ 550 if (mSharedBuffer == 0 || mIsTimed) { 551 return INVALID_OPERATION; 552 } 553 554 if (loopCount == 0) { 555 ; 556 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 557 loopEnd - loopStart >= MIN_LOOP) { 558 ; 559 } else { 560 return BAD_VALUE; 561 } 562 563 AutoMutex lock(mLock); 564 // See setPosition() regarding setting parameters such as loop points or position while active 565 if (mState == STATE_ACTIVE) { 566 return INVALID_OPERATION; 567 } 568 setLoop_l(loopStart, loopEnd, loopCount); 569 return NO_ERROR; 570} 571 572void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 573{ 574 // FIXME If setting a loop also sets position to start of loop, then 575 // this is correct. Otherwise it should be removed. 576 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 577 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 578 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 579} 580 581status_t AudioTrack::setMarkerPosition(uint32_t marker) 582{ 583 if (mCbf == NULL) { 584 return INVALID_OPERATION; 585 } 586 587 AutoMutex lock(mLock); 588 mMarkerPosition = marker; 589 mMarkerReached = false; 590 591 return NO_ERROR; 592} 593 594status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 595{ 596 if (marker == NULL) { 597 return BAD_VALUE; 598 } 599 600 AutoMutex lock(mLock); 601 *marker = mMarkerPosition; 602 603 return NO_ERROR; 604} 605 606status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 607{ 608 if (mCbf == NULL) { 609 return INVALID_OPERATION; 610 } 611 612 AutoMutex lock(mLock); 613 mNewPosition = mProxy->getPosition() + updatePeriod; 614 mUpdatePeriod = updatePeriod; 615 616 return NO_ERROR; 617} 618 619status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 620{ 621 if (updatePeriod == NULL) { 622 return BAD_VALUE; 623 } 624 625 AutoMutex lock(mLock); 626 *updatePeriod = mUpdatePeriod; 627 628 return NO_ERROR; 629} 630 631status_t AudioTrack::setPosition(uint32_t position) 632{ 633 if (mSharedBuffer == 0 || mIsTimed) { 634 return INVALID_OPERATION; 635 } 636 if (position > mFrameCount) { 637 return BAD_VALUE; 638 } 639 640 AutoMutex lock(mLock); 641 // Currently we require that the player is inactive before setting parameters such as position 642 // or loop points. Otherwise, there could be a race condition: the application could read the 643 // current position, compute a new position or loop parameters, and then set that position or 644 // loop parameters but it would do the "wrong" thing since the position has continued to advance 645 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 646 // to specify how it wants to handle such scenarios. 647 if (mState == STATE_ACTIVE) { 648 return INVALID_OPERATION; 649 } 650 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 651 mLoopPeriod = 0; 652 // FIXME Check whether loops and setting position are incompatible in old code. 653 // If we use setLoop for both purposes we lose the capability to set the position while looping. 654 mStaticProxy->setLoop(position, mFrameCount, 0); 655 656 return NO_ERROR; 657} 658 659status_t AudioTrack::getPosition(uint32_t *position) const 660{ 661 if (position == NULL) { 662 return BAD_VALUE; 663 } 664 665 AutoMutex lock(mLock); 666 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 667 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 668 mProxy->getPosition(); 669 670 return NO_ERROR; 671} 672 673status_t AudioTrack::getBufferPosition(size_t *position) 674{ 675 if (mSharedBuffer == 0 || mIsTimed) { 676 return INVALID_OPERATION; 677 } 678 if (position == NULL) { 679 return BAD_VALUE; 680 } 681 682 AutoMutex lock(mLock); 683 *position = mStaticProxy->getBufferPosition(); 684 return NO_ERROR; 685} 686 687status_t AudioTrack::reload() 688{ 689 if (mSharedBuffer == 0 || mIsTimed) { 690 return INVALID_OPERATION; 691 } 692 693 AutoMutex lock(mLock); 694 // See setPosition() regarding setting parameters such as loop points or position while active 695 if (mState == STATE_ACTIVE) { 696 return INVALID_OPERATION; 697 } 698 mNewPosition = mUpdatePeriod; 699 mLoopPeriod = 0; 700 // FIXME The new code cannot reload while keeping a loop specified. 701 // Need to check how the old code handled this, and whether it's a significant change. 702 mStaticProxy->setLoop(0, mFrameCount, 0); 703 return NO_ERROR; 704} 705 706audio_io_handle_t AudioTrack::getOutput() 707{ 708 AutoMutex lock(mLock); 709 return getOutput_l(); 710} 711 712// must be called with mLock held 713audio_io_handle_t AudioTrack::getOutput_l() 714{ 715 return AudioSystem::getOutput(mStreamType, 716 mSampleRate, mFormat, mChannelMask, mFlags); 717} 718 719status_t AudioTrack::attachAuxEffect(int effectId) 720{ 721 AutoMutex lock(mLock); 722 status_t status = mAudioTrack->attachAuxEffect(effectId); 723 if (status == NO_ERROR) { 724 mAuxEffectId = effectId; 725 } 726 return status; 727} 728 729// ------------------------------------------------------------------------- 730 731// must be called with mLock held 732status_t AudioTrack::createTrack_l( 733 audio_stream_type_t streamType, 734 uint32_t sampleRate, 735 audio_format_t format, 736 size_t frameCount, 737 audio_output_flags_t flags, 738 const sp<IMemory>& sharedBuffer, 739 audio_io_handle_t output, 740 size_t epoch) 741{ 742 status_t status; 743 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 744 if (audioFlinger == 0) { 745 ALOGE("Could not get audioflinger"); 746 return NO_INIT; 747 } 748 749 uint32_t afLatency; 750 if ((status = AudioSystem::getLatency(output, streamType, &afLatency)) != NO_ERROR) { 751 ALOGE("getLatency(%d) failed status %d", output, status); 752 return NO_INIT; 753 } 754 755 // Client decides whether the track is TIMED (see below), but can only express a preference 756 // for FAST. Server will perform additional tests. 757 if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !( 758 // either of these use cases: 759 // use case 1: shared buffer 760 (sharedBuffer != 0) || 761 // use case 2: callback handler 762 (mCbf != NULL))) { 763 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 764 // once denied, do not request again if IAudioTrack is re-created 765 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 766 mFlags = flags; 767 } 768 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 769 770 mNotificationFramesAct = mNotificationFramesReq; 771 772 if (!audio_is_linear_pcm(format)) { 773 774 if (sharedBuffer != 0) { 775 // Same comment as below about ignoring frameCount parameter for set() 776 frameCount = sharedBuffer->size(); 777 } else if (frameCount == 0) { 778 size_t afFrameCount; 779 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 780 if (status != NO_ERROR) { 781 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, 782 status); 783 return NO_INIT; 784 } 785 frameCount = afFrameCount; 786 } 787 788 } else if (sharedBuffer != 0) { 789 790 // Ensure that buffer alignment matches channel count 791 // 8-bit data in shared memory is not currently supported by AudioFlinger 792 size_t alignment = /* format == AUDIO_FORMAT_PCM_8_BIT ? 1 : */ 2; 793 if (mChannelCount > 1) { 794 // More than 2 channels does not require stronger alignment than stereo 795 alignment <<= 1; 796 } 797 if (((size_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 798 ALOGE("Invalid buffer alignment: address %p, channel count %u", 799 sharedBuffer->pointer(), mChannelCount); 800 return BAD_VALUE; 801 } 802 803 // When initializing a shared buffer AudioTrack via constructors, 804 // there's no frameCount parameter. 805 // But when initializing a shared buffer AudioTrack via set(), 806 // there _is_ a frameCount parameter. We silently ignore it. 807 frameCount = sharedBuffer->size()/mChannelCount/sizeof(int16_t); 808 809 } else if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 810 811 // FIXME move these calculations and associated checks to server 812 uint32_t afSampleRate; 813 status = AudioSystem::getSamplingRate(output, streamType, &afSampleRate); 814 if (status != NO_ERROR) { 815 ALOGE("getSamplingRate(output=%d, streamType=%d) status %d", output, streamType, 816 status); 817 return NO_INIT; 818 } 819 size_t afFrameCount; 820 status = AudioSystem::getFrameCount(output, streamType, &afFrameCount); 821 if (status != NO_ERROR) { 822 ALOGE("getFrameCount(output=%d, streamType=%d) status %d", output, streamType, status); 823 return NO_INIT; 824 } 825 826 // Ensure that buffer depth covers at least audio hardware latency 827 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 828 ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d", 829 afFrameCount, minBufCount, afSampleRate, afLatency); 830 if (minBufCount <= 2) { 831 minBufCount = sampleRate == afSampleRate ? 2 : 3; 832 } 833 834 size_t minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate; 835 ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 836 ", afLatency=%d", 837 minFrameCount, afFrameCount, minBufCount, sampleRate, afSampleRate, afLatency); 838 839 if (frameCount == 0) { 840 frameCount = minFrameCount; 841 } 842 // Make sure that application is notified with sufficient margin 843 // before underrun 844 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) { 845 mNotificationFramesAct = frameCount/2; 846 } 847 if (frameCount < minFrameCount) { 848 // not ALOGW because it happens all the time when playing key clicks over A2DP 849 ALOGV("Minimum buffer size corrected from %d to %d", 850 frameCount, minFrameCount); 851 frameCount = minFrameCount; 852 } 853 854 } else { 855 // For fast tracks, the frame count calculations and checks are done by server 856 } 857 858 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 859 if (mIsTimed) { 860 trackFlags |= IAudioFlinger::TRACK_TIMED; 861 } 862 863 pid_t tid = -1; 864 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 865 trackFlags |= IAudioFlinger::TRACK_FAST; 866 if (mAudioTrackThread != 0) { 867 tid = mAudioTrackThread->getTid(); 868 } 869 } 870 871 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 872 sampleRate, 873 // AudioFlinger only sees 16-bit PCM 874 format == AUDIO_FORMAT_PCM_8_BIT ? 875 AUDIO_FORMAT_PCM_16_BIT : format, 876 mChannelMask, 877 frameCount, 878 &trackFlags, 879 sharedBuffer, 880 output, 881 tid, 882 &mSessionId, 883 &status); 884 885 if (track == 0) { 886 ALOGE("AudioFlinger could not create track, status: %d", status); 887 return status; 888 } 889 sp<IMemory> iMem = track->getCblk(); 890 if (iMem == 0) { 891 ALOGE("Could not get control block"); 892 return NO_INIT; 893 } 894 if (mAudioTrack != 0) { 895 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 896 mDeathNotifier.clear(); 897 } 898 mAudioTrack = track; 899 mCblkMemory = iMem; 900 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMem->pointer()); 901 mCblk = cblk; 902 size_t temp = cblk->frameCount_; 903 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 904 // In current design, AudioTrack client checks and ensures frame count validity before 905 // passing it to AudioFlinger so AudioFlinger should not return a different value except 906 // for fast track as it uses a special method of assigning frame count. 907 ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp); 908 } 909 frameCount = temp; 910 mAwaitBoost = false; 911 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 912 if (trackFlags & IAudioFlinger::TRACK_FAST) { 913 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount); 914 mAwaitBoost = true; 915 if (sharedBuffer == 0) { 916 // double-buffering is not required for fast tracks, due to tighter scheduling 917 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount) { 918 mNotificationFramesAct = frameCount; 919 } 920 } 921 } else { 922 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount); 923 // once denied, do not request again if IAudioTrack is re-created 924 flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST); 925 mFlags = flags; 926 if (sharedBuffer == 0) { 927 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/2) { 928 mNotificationFramesAct = frameCount/2; 929 } 930 } 931 } 932 } 933 mRefreshRemaining = true; 934 935 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 936 // is the value of pointer() for the shared buffer, otherwise buffers points 937 // immediately after the control block. This address is for the mapping within client 938 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 939 void* buffers; 940 if (sharedBuffer == 0) { 941 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 942 } else { 943 buffers = sharedBuffer->pointer(); 944 } 945 946 mAudioTrack->attachAuxEffect(mAuxEffectId); 947 // FIXME don't believe this lie 948 mLatency = afLatency + (1000*frameCount) / sampleRate; 949 mFrameCount = frameCount; 950 // If IAudioTrack is re-created, don't let the requested frameCount 951 // decrease. This can confuse clients that cache frameCount(). 952 if (frameCount > mReqFrameCount) { 953 mReqFrameCount = frameCount; 954 } 955 956 // update proxy 957 if (sharedBuffer == 0) { 958 mStaticProxy.clear(); 959 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 960 } else { 961 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 962 mProxy = mStaticProxy; 963 } 964 mProxy->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | 965 uint16_t(mVolume[LEFT] * 0x1000)); 966 mProxy->setSendLevel(mSendLevel); 967 mProxy->setSampleRate(mSampleRate); 968 mProxy->setEpoch(epoch); 969 mProxy->setMinimum(mNotificationFramesAct); 970 971 mDeathNotifier = new DeathNotifier(this); 972 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 973 974 return NO_ERROR; 975} 976 977status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 978{ 979 if (audioBuffer == NULL) { 980 return BAD_VALUE; 981 } 982 if (mTransfer != TRANSFER_OBTAIN) { 983 audioBuffer->frameCount = 0; 984 audioBuffer->size = 0; 985 audioBuffer->raw = NULL; 986 return INVALID_OPERATION; 987 } 988 989 const struct timespec *requested; 990 if (waitCount == -1) { 991 requested = &ClientProxy::kForever; 992 } else if (waitCount == 0) { 993 requested = &ClientProxy::kNonBlocking; 994 } else if (waitCount > 0) { 995 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 996 struct timespec timeout; 997 timeout.tv_sec = ms / 1000; 998 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 999 requested = &timeout; 1000 } else { 1001 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1002 requested = NULL; 1003 } 1004 return obtainBuffer(audioBuffer, requested); 1005} 1006 1007status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1008 struct timespec *elapsed, size_t *nonContig) 1009{ 1010 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1011 uint32_t oldSequence = 0; 1012 uint32_t newSequence; 1013 1014 Proxy::Buffer buffer; 1015 status_t status = NO_ERROR; 1016 1017 static const int32_t kMaxTries = 5; 1018 int32_t tryCounter = kMaxTries; 1019 1020 do { 1021 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1022 // keep them from going away if another thread re-creates the track during obtainBuffer() 1023 sp<AudioTrackClientProxy> proxy; 1024 sp<IMemory> iMem; 1025 1026 { // start of lock scope 1027 AutoMutex lock(mLock); 1028 1029 newSequence = mSequence; 1030 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1031 if (status == DEAD_OBJECT) { 1032 // re-create track, unless someone else has already done so 1033 if (newSequence == oldSequence) { 1034 status = restoreTrack_l("obtainBuffer"); 1035 if (status != NO_ERROR) { 1036 break; 1037 } 1038 } 1039 } 1040 oldSequence = newSequence; 1041 1042 // Keep the extra references 1043 proxy = mProxy; 1044 iMem = mCblkMemory; 1045 1046 // Non-blocking if track is stopped or paused 1047 if (mState != STATE_ACTIVE) { 1048 requested = &ClientProxy::kNonBlocking; 1049 } 1050 1051 } // end of lock scope 1052 1053 buffer.mFrameCount = audioBuffer->frameCount; 1054 // FIXME starts the requested timeout and elapsed over from scratch 1055 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1056 1057 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1058 1059 audioBuffer->frameCount = buffer.mFrameCount; 1060 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1061 audioBuffer->raw = buffer.mRaw; 1062 if (nonContig != NULL) { 1063 *nonContig = buffer.mNonContig; 1064 } 1065 return status; 1066} 1067 1068void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1069{ 1070 if (mTransfer == TRANSFER_SHARED) { 1071 return; 1072 } 1073 1074 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1075 if (stepCount == 0) { 1076 return; 1077 } 1078 1079 Proxy::Buffer buffer; 1080 buffer.mFrameCount = stepCount; 1081 buffer.mRaw = audioBuffer->raw; 1082 1083 AutoMutex lock(mLock); 1084 mInUnderrun = false; 1085 mProxy->releaseBuffer(&buffer); 1086 1087 // restart track if it was disabled by audioflinger due to previous underrun 1088 if (mState == STATE_ACTIVE) { 1089 audio_track_cblk_t* cblk = mCblk; 1090 if (android_atomic_and(~CBLK_DISABLED, &cblk->flags) & CBLK_DISABLED) { 1091 ALOGW("releaseBuffer() track %p name=%#x disabled due to previous underrun, restarting", 1092 this, cblk->mName); 1093 // FIXME ignoring status 1094 mAudioTrack->start(); 1095 } 1096 } 1097} 1098 1099// ------------------------------------------------------------------------- 1100 1101ssize_t AudioTrack::write(const void* buffer, size_t userSize) 1102{ 1103 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1104 return INVALID_OPERATION; 1105 } 1106 1107 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1108 // Sanity-check: user is most-likely passing an error code, and it would 1109 // make the return value ambiguous (actualSize vs error). 1110 ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)", buffer, userSize, userSize); 1111 return BAD_VALUE; 1112 } 1113 1114 size_t written = 0; 1115 Buffer audioBuffer; 1116 1117 while (userSize >= mFrameSize) { 1118 audioBuffer.frameCount = userSize / mFrameSize; 1119 1120 status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever); 1121 if (err < 0) { 1122 if (written > 0) { 1123 break; 1124 } 1125 return ssize_t(err); 1126 } 1127 1128 size_t toWrite; 1129 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1130 // Divide capacity by 2 to take expansion into account 1131 toWrite = audioBuffer.size >> 1; 1132 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1133 } else { 1134 toWrite = audioBuffer.size; 1135 memcpy(audioBuffer.i8, buffer, toWrite); 1136 } 1137 buffer = ((const char *) buffer) + toWrite; 1138 userSize -= toWrite; 1139 written += toWrite; 1140 1141 releaseBuffer(&audioBuffer); 1142 } 1143 1144 return written; 1145} 1146 1147// ------------------------------------------------------------------------- 1148 1149TimedAudioTrack::TimedAudioTrack() { 1150 mIsTimed = true; 1151} 1152 1153status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1154{ 1155 AutoMutex lock(mLock); 1156 status_t result = UNKNOWN_ERROR; 1157 1158#if 1 1159 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1160 // while we are accessing the cblk 1161 sp<IAudioTrack> audioTrack = mAudioTrack; 1162 sp<IMemory> iMem = mCblkMemory; 1163#endif 1164 1165 // If the track is not invalid already, try to allocate a buffer. alloc 1166 // fails indicating that the server is dead, flag the track as invalid so 1167 // we can attempt to restore in just a bit. 1168 audio_track_cblk_t* cblk = mCblk; 1169 if (!(cblk->flags & CBLK_INVALID)) { 1170 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1171 if (result == DEAD_OBJECT) { 1172 android_atomic_or(CBLK_INVALID, &cblk->flags); 1173 } 1174 } 1175 1176 // If the track is invalid at this point, attempt to restore it. and try the 1177 // allocation one more time. 1178 if (cblk->flags & CBLK_INVALID) { 1179 result = restoreTrack_l("allocateTimedBuffer"); 1180 1181 if (result == NO_ERROR) { 1182 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1183 } 1184 } 1185 1186 return result; 1187} 1188 1189status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1190 int64_t pts) 1191{ 1192 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1193 { 1194 AutoMutex lock(mLock); 1195 audio_track_cblk_t* cblk = mCblk; 1196 // restart track if it was disabled by audioflinger due to previous underrun 1197 if (buffer->size() != 0 && status == NO_ERROR && 1198 (mState == STATE_ACTIVE) && (cblk->flags & CBLK_DISABLED)) { 1199 android_atomic_and(~CBLK_DISABLED, &cblk->flags); 1200 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1201 // FIXME ignoring status 1202 mAudioTrack->start(); 1203 } 1204 } 1205 return status; 1206} 1207 1208status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1209 TargetTimeline target) 1210{ 1211 return mAudioTrack->setMediaTimeTransform(xform, target); 1212} 1213 1214// ------------------------------------------------------------------------- 1215 1216nsecs_t AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread) 1217{ 1218 mLock.lock(); 1219 if (mAwaitBoost) { 1220 mAwaitBoost = false; 1221 mLock.unlock(); 1222 static const int32_t kMaxTries = 5; 1223 int32_t tryCounter = kMaxTries; 1224 uint32_t pollUs = 10000; 1225 do { 1226 int policy = sched_getscheduler(0); 1227 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1228 break; 1229 } 1230 usleep(pollUs); 1231 pollUs <<= 1; 1232 } while (tryCounter-- > 0); 1233 if (tryCounter < 0) { 1234 ALOGE("did not receive expected priority boost on time"); 1235 } 1236 return true; 1237 } 1238 1239 // Can only reference mCblk while locked 1240 int32_t flags = android_atomic_and( 1241 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->flags); 1242 1243 // Check for track invalidation 1244 if (flags & CBLK_INVALID) { 1245 (void) restoreTrack_l("processAudioBuffer"); 1246 mLock.unlock(); 1247 // Run again immediately, but with a new IAudioTrack 1248 return 0; 1249 } 1250 1251 bool active = mState == STATE_ACTIVE; 1252 1253 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1254 bool newUnderrun = false; 1255 if (flags & CBLK_UNDERRUN) { 1256#if 0 1257 // Currently in shared buffer mode, when the server reaches the end of buffer, 1258 // the track stays active in continuous underrun state. It's up to the application 1259 // to pause or stop the track, or set the position to a new offset within buffer. 1260 // This was some experimental code to auto-pause on underrun. Keeping it here 1261 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1262 if (mTransfer == TRANSFER_SHARED) { 1263 mState = STATE_PAUSED; 1264 active = false; 1265 } 1266#endif 1267 if (!mInUnderrun) { 1268 mInUnderrun = true; 1269 newUnderrun = true; 1270 } 1271 } 1272 1273 // Get current position of server 1274 size_t position = mProxy->getPosition(); 1275 1276 // Manage marker callback 1277 bool markerReached = false; 1278 size_t markerPosition = mMarkerPosition; 1279 // FIXME fails for wraparound, need 64 bits 1280 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1281 mMarkerReached = markerReached = true; 1282 } 1283 1284 // Determine number of new position callback(s) that will be needed, while locked 1285 size_t newPosCount = 0; 1286 size_t newPosition = mNewPosition; 1287 size_t updatePeriod = mUpdatePeriod; 1288 // FIXME fails for wraparound, need 64 bits 1289 if (updatePeriod > 0 && position >= newPosition) { 1290 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1291 mNewPosition += updatePeriod * newPosCount; 1292 } 1293 1294 // Cache other fields that will be needed soon 1295 uint32_t loopPeriod = mLoopPeriod; 1296 uint32_t sampleRate = mSampleRate; 1297 size_t notificationFrames = mNotificationFramesAct; 1298 if (mRefreshRemaining) { 1299 mRefreshRemaining = false; 1300 mRemainingFrames = notificationFrames; 1301 mRetryOnPartialBuffer = false; 1302 } 1303 size_t misalignment = mProxy->getMisalignment(); 1304 int32_t sequence = mSequence; 1305 1306 // These fields don't need to be cached, because they are assigned only by set(): 1307 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1308 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1309 1310 mLock.unlock(); 1311 1312 // perform callbacks while unlocked 1313 if (newUnderrun) { 1314 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1315 } 1316 // FIXME we will miss loops if loop cycle was signaled several times since last call 1317 // to processAudioBuffer() 1318 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1319 mCbf(EVENT_LOOP_END, mUserData, NULL); 1320 } 1321 if (flags & CBLK_BUFFER_END) { 1322 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1323 } 1324 if (markerReached) { 1325 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1326 } 1327 while (newPosCount > 0) { 1328 size_t temp = newPosition; 1329 mCbf(EVENT_NEW_POS, mUserData, &temp); 1330 newPosition += updatePeriod; 1331 newPosCount--; 1332 } 1333 if (mObservedSequence != sequence) { 1334 mObservedSequence = sequence; 1335 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1336 } 1337 1338 // if inactive, then don't run me again until re-started 1339 if (!active) { 1340 return NS_INACTIVE; 1341 } 1342 1343 // Compute the estimated time until the next timed event (position, markers, loops) 1344 // FIXME only for non-compressed audio 1345 uint32_t minFrames = ~0; 1346 if (!markerReached && position < markerPosition) { 1347 minFrames = markerPosition - position; 1348 } 1349 if (loopPeriod > 0 && loopPeriod < minFrames) { 1350 minFrames = loopPeriod; 1351 } 1352 if (updatePeriod > 0 && updatePeriod < minFrames) { 1353 minFrames = updatePeriod; 1354 } 1355 1356 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1357 static const uint32_t kPoll = 0; 1358 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1359 minFrames = kPoll * notificationFrames; 1360 } 1361 1362 // Convert frame units to time units 1363 nsecs_t ns = NS_WHENEVER; 1364 if (minFrames != (uint32_t) ~0) { 1365 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1366 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1367 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1368 } 1369 1370 // If not supplying data by EVENT_MORE_DATA, then we're done 1371 if (mTransfer != TRANSFER_CALLBACK) { 1372 return ns; 1373 } 1374 1375 struct timespec timeout; 1376 const struct timespec *requested = &ClientProxy::kForever; 1377 if (ns != NS_WHENEVER) { 1378 timeout.tv_sec = ns / 1000000000LL; 1379 timeout.tv_nsec = ns % 1000000000LL; 1380 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1381 requested = &timeout; 1382 } 1383 1384 while (mRemainingFrames > 0) { 1385 1386 Buffer audioBuffer; 1387 audioBuffer.frameCount = mRemainingFrames; 1388 size_t nonContig; 1389 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1390 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1391 "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount); 1392 requested = &ClientProxy::kNonBlocking; 1393 size_t avail = audioBuffer.frameCount + nonContig; 1394 ALOGV("obtainBuffer(%u) returned %u = %u + %u", 1395 mRemainingFrames, avail, audioBuffer.frameCount, nonContig); 1396 if (err != NO_ERROR) { 1397 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) { 1398 return 0; 1399 } 1400 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1401 return NS_NEVER; 1402 } 1403 1404 if (mRetryOnPartialBuffer) { 1405 mRetryOnPartialBuffer = false; 1406 if (avail < mRemainingFrames) { 1407 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1408 if (ns < 0 || myns < ns) { 1409 ns = myns; 1410 } 1411 return ns; 1412 } 1413 } 1414 1415 // Divide buffer size by 2 to take into account the expansion 1416 // due to 8 to 16 bit conversion: the callback must fill only half 1417 // of the destination buffer 1418 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1419 audioBuffer.size >>= 1; 1420 } 1421 1422 size_t reqSize = audioBuffer.size; 1423 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1424 size_t writtenSize = audioBuffer.size; 1425 size_t writtenFrames = writtenSize / mFrameSize; 1426 1427 // Sanity check on returned size 1428 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1429 ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes", 1430 reqSize, (int) writtenSize); 1431 return NS_NEVER; 1432 } 1433 1434 if (writtenSize == 0) { 1435 // The callback is done filling buffers 1436 // Keep this thread going to handle timed events and 1437 // still try to get more data in intervals of WAIT_PERIOD_MS 1438 // but don't just loop and block the CPU, so wait 1439 return WAIT_PERIOD_MS * 1000000LL; 1440 } 1441 1442 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1443 // 8 to 16 bit conversion, note that source and destination are the same address 1444 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1445 audioBuffer.size <<= 1; 1446 } 1447 1448 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1449 audioBuffer.frameCount = releasedFrames; 1450 mRemainingFrames -= releasedFrames; 1451 if (misalignment >= releasedFrames) { 1452 misalignment -= releasedFrames; 1453 } else { 1454 misalignment = 0; 1455 } 1456 1457 releaseBuffer(&audioBuffer); 1458 1459 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1460 // if callback doesn't like to accept the full chunk 1461 if (writtenSize < reqSize) { 1462 continue; 1463 } 1464 1465 // There could be enough non-contiguous frames available to satisfy the remaining request 1466 if (mRemainingFrames <= nonContig) { 1467 continue; 1468 } 1469 1470#if 0 1471 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1472 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1473 // that total to a sum == notificationFrames. 1474 if (0 < misalignment && misalignment <= mRemainingFrames) { 1475 mRemainingFrames = misalignment; 1476 return (mRemainingFrames * 1100000000LL) / sampleRate; 1477 } 1478#endif 1479 1480 } 1481 mRemainingFrames = notificationFrames; 1482 mRetryOnPartialBuffer = true; 1483 1484 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1485 return 0; 1486} 1487 1488status_t AudioTrack::restoreTrack_l(const char *from) 1489{ 1490 ALOGW("dead IAudioTrack, creating a new one from %s()", from); 1491 ++mSequence; 1492 status_t result; 1493 1494 // refresh the audio configuration cache in this process to make sure we get new 1495 // output parameters in getOutput_l() and createTrack_l() 1496 AudioSystem::clearAudioConfigCache(); 1497 1498 // if the new IAudioTrack is created, createTrack_l() will modify the 1499 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1500 // It will also delete the strong references on previous IAudioTrack and IMemory 1501 size_t position = mProxy->getPosition(); 1502 mNewPosition = position + mUpdatePeriod; 1503 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1504 result = createTrack_l(mStreamType, 1505 mSampleRate, 1506 mFormat, 1507 mReqFrameCount, // so that frame count never goes down 1508 mFlags, 1509 mSharedBuffer, 1510 getOutput_l(), 1511 position /*epoch*/); 1512 1513 if (result == NO_ERROR) { 1514 // continue playback from last known position, but 1515 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1516 if (mStaticProxy != NULL) { 1517 mLoopPeriod = 0; 1518 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1519 } 1520 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1521 // track destruction have been played? This is critical for SoundPool implementation 1522 // This must be broken, and needs to be tested/debugged. 1523#if 0 1524 // restore write index and set other indexes to reflect empty buffer status 1525 if (!strcmp(from, "start")) { 1526 // Make sure that a client relying on callback events indicating underrun or 1527 // the actual amount of audio frames played (e.g SoundPool) receives them. 1528 if (mSharedBuffer == 0) { 1529 // restart playback even if buffer is not completely filled. 1530 android_atomic_or(CBLK_FORCEREADY, &mCblk->flags); 1531 } 1532 } 1533#endif 1534 if (mState == STATE_ACTIVE) { 1535 result = mAudioTrack->start(); 1536 } 1537 } 1538 if (result != NO_ERROR) { 1539 ALOGW("restoreTrack_l() failed status %d", result); 1540 mState = STATE_STOPPED; 1541 } 1542 1543 return result; 1544} 1545 1546status_t AudioTrack::dump(int fd, const Vector<String16>& args) const 1547{ 1548 1549 const size_t SIZE = 256; 1550 char buffer[SIZE]; 1551 String8 result; 1552 1553 result.append(" AudioTrack::dump\n"); 1554 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1555 mVolume[0], mVolume[1]); 1556 result.append(buffer); 1557 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, 1558 mChannelCount, mFrameCount); 1559 result.append(buffer); 1560 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1561 result.append(buffer); 1562 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1563 result.append(buffer); 1564 ::write(fd, result.string(), result.size()); 1565 return NO_ERROR; 1566} 1567 1568uint32_t AudioTrack::getUnderrunFrames() const 1569{ 1570 AutoMutex lock(mLock); 1571 return mProxy->getUnderrunFrames(); 1572} 1573 1574// ========================================================================= 1575 1576void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who) 1577{ 1578 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 1579 if (audioTrack != 0) { 1580 AutoMutex lock(audioTrack->mLock); 1581 audioTrack->mProxy->binderDied(); 1582 } 1583} 1584 1585// ========================================================================= 1586 1587AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 1588 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mResumeLatch(false) 1589{ 1590} 1591 1592AudioTrack::AudioTrackThread::~AudioTrackThread() 1593{ 1594} 1595 1596bool AudioTrack::AudioTrackThread::threadLoop() 1597{ 1598 { 1599 AutoMutex _l(mMyLock); 1600 if (mPaused) { 1601 mMyCond.wait(mMyLock); 1602 // caller will check for exitPending() 1603 return true; 1604 } 1605 } 1606 nsecs_t ns = mReceiver.processAudioBuffer(this); 1607 switch (ns) { 1608 case 0: 1609 return true; 1610 case NS_WHENEVER: 1611 sleep(1); 1612 return true; 1613 case NS_INACTIVE: 1614 pauseConditional(); 1615 return true; 1616 case NS_NEVER: 1617 return false; 1618 default: 1619 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns); 1620 struct timespec req; 1621 req.tv_sec = ns / 1000000000LL; 1622 req.tv_nsec = ns % 1000000000LL; 1623 nanosleep(&req, NULL /*rem*/); 1624 return true; 1625 } 1626} 1627 1628void AudioTrack::AudioTrackThread::requestExit() 1629{ 1630 // must be in this order to avoid a race condition 1631 Thread::requestExit(); 1632 resume(); 1633} 1634 1635void AudioTrack::AudioTrackThread::pause() 1636{ 1637 AutoMutex _l(mMyLock); 1638 mPaused = true; 1639 mResumeLatch = false; 1640} 1641 1642void AudioTrack::AudioTrackThread::pauseConditional() 1643{ 1644 AutoMutex _l(mMyLock); 1645 if (mResumeLatch) { 1646 mResumeLatch = false; 1647 } else { 1648 mPaused = true; 1649 } 1650} 1651 1652void AudioTrack::AudioTrackThread::resume() 1653{ 1654 AutoMutex _l(mMyLock); 1655 if (mPaused) { 1656 mPaused = false; 1657 mResumeLatch = false; 1658 mMyCond.signal(); 1659 } else { 1660 mResumeLatch = true; 1661 } 1662} 1663 1664}; // namespace android 1665