AudioTrack.cpp revision 200092b7f21d2b98f30b800e79d152636f9ba225
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18//#define LOG_NDEBUG 0 19#define LOG_TAG "AudioTrack" 20 21#include <inttypes.h> 22#include <math.h> 23#include <sys/resource.h> 24 25#include <audio_utils/primitives.h> 26#include <binder/IPCThreadState.h> 27#include <media/AudioTrack.h> 28#include <utils/Log.h> 29#include <private/media/AudioTrackShared.h> 30#include <media/IAudioFlinger.h> 31#include <media/AudioResamplerPublic.h> 32 33#define WAIT_PERIOD_MS 10 34#define WAIT_STREAM_END_TIMEOUT_SEC 120 35 36 37namespace android { 38// --------------------------------------------------------------------------- 39 40// static 41status_t AudioTrack::getMinFrameCount( 42 size_t* frameCount, 43 audio_stream_type_t streamType, 44 uint32_t sampleRate) 45{ 46 if (frameCount == NULL) { 47 return BAD_VALUE; 48 } 49 50 // FIXME merge with similar code in createTrack_l(), except we're missing 51 // some information here that is available in createTrack_l(): 52 // audio_io_handle_t output 53 // audio_format_t format 54 // audio_channel_mask_t channelMask 55 // audio_output_flags_t flags 56 uint32_t afSampleRate; 57 status_t status; 58 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 59 if (status != NO_ERROR) { 60 ALOGE("Unable to query output sample rate for stream type %d; status %d", 61 streamType, status); 62 return status; 63 } 64 size_t afFrameCount; 65 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 66 if (status != NO_ERROR) { 67 ALOGE("Unable to query output frame count for stream type %d; status %d", 68 streamType, status); 69 return status; 70 } 71 uint32_t afLatency; 72 status = AudioSystem::getOutputLatency(&afLatency, streamType); 73 if (status != NO_ERROR) { 74 ALOGE("Unable to query output latency for stream type %d; status %d", 75 streamType, status); 76 return status; 77 } 78 79 // Ensure that buffer depth covers at least audio hardware latency 80 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 81 if (minBufCount < 2) { 82 minBufCount = 2; 83 } 84 85 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 86 afFrameCount * minBufCount * uint64_t(sampleRate) / afSampleRate; 87 // The formula above should always produce a non-zero value, but return an error 88 // in the unlikely event that it does not, as that's part of the API contract. 89 if (*frameCount == 0) { 90 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", 91 streamType, sampleRate); 92 return BAD_VALUE; 93 } 94 ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%d, afSampleRate=%d, afLatency=%d", 95 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 96 return NO_ERROR; 97} 98 99// --------------------------------------------------------------------------- 100 101AudioTrack::AudioTrack() 102 : mStatus(NO_INIT), 103 mIsTimed(false), 104 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 105 mPreviousSchedulingGroup(SP_DEFAULT), 106 mPausedPosition(0) 107{ 108 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN; 109 mAttributes.usage = AUDIO_USAGE_UNKNOWN; 110 mAttributes.flags = 0x0; 111 strcpy(mAttributes.tags, ""); 112} 113 114AudioTrack::AudioTrack( 115 audio_stream_type_t streamType, 116 uint32_t sampleRate, 117 audio_format_t format, 118 audio_channel_mask_t channelMask, 119 size_t frameCount, 120 audio_output_flags_t flags, 121 callback_t cbf, 122 void* user, 123 uint32_t notificationFrames, 124 int sessionId, 125 transfer_type transferType, 126 const audio_offload_info_t *offloadInfo, 127 int uid, 128 pid_t pid, 129 const audio_attributes_t* pAttributes) 130 : mStatus(NO_INIT), 131 mIsTimed(false), 132 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 133 mPreviousSchedulingGroup(SP_DEFAULT), 134 mPausedPosition(0) 135{ 136 mStatus = set(streamType, sampleRate, format, channelMask, 137 frameCount, flags, cbf, user, notificationFrames, 138 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 139 offloadInfo, uid, pid, pAttributes); 140} 141 142AudioTrack::AudioTrack( 143 audio_stream_type_t streamType, 144 uint32_t sampleRate, 145 audio_format_t format, 146 audio_channel_mask_t channelMask, 147 const sp<IMemory>& sharedBuffer, 148 audio_output_flags_t flags, 149 callback_t cbf, 150 void* user, 151 uint32_t notificationFrames, 152 int sessionId, 153 transfer_type transferType, 154 const audio_offload_info_t *offloadInfo, 155 int uid, 156 pid_t pid, 157 const audio_attributes_t* pAttributes) 158 : mStatus(NO_INIT), 159 mIsTimed(false), 160 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 161 mPreviousSchedulingGroup(SP_DEFAULT), 162 mPausedPosition(0) 163{ 164 mStatus = set(streamType, sampleRate, format, channelMask, 165 0 /*frameCount*/, flags, cbf, user, notificationFrames, 166 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, 167 uid, pid, pAttributes); 168} 169 170AudioTrack::~AudioTrack() 171{ 172 if (mStatus == NO_ERROR) { 173 // Make sure that callback function exits in the case where 174 // it is looping on buffer full condition in obtainBuffer(). 175 // Otherwise the callback thread will never exit. 176 stop(); 177 if (mAudioTrackThread != 0) { 178 mProxy->interrupt(); 179 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 180 mAudioTrackThread->requestExitAndWait(); 181 mAudioTrackThread.clear(); 182 } 183 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 184 mAudioTrack.clear(); 185 mCblkMemory.clear(); 186 mSharedBuffer.clear(); 187 IPCThreadState::self()->flushCommands(); 188 ALOGV("~AudioTrack, releasing session id from %d on behalf of %d", 189 IPCThreadState::self()->getCallingPid(), mClientPid); 190 AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); 191 } 192} 193 194status_t AudioTrack::set( 195 audio_stream_type_t streamType, 196 uint32_t sampleRate, 197 audio_format_t format, 198 audio_channel_mask_t channelMask, 199 size_t frameCount, 200 audio_output_flags_t flags, 201 callback_t cbf, 202 void* user, 203 uint32_t notificationFrames, 204 const sp<IMemory>& sharedBuffer, 205 bool threadCanCallJava, 206 int sessionId, 207 transfer_type transferType, 208 const audio_offload_info_t *offloadInfo, 209 int uid, 210 pid_t pid, 211 const audio_attributes_t* pAttributes) 212{ 213 ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 214 "flags #%x, notificationFrames %u, sessionId %d, transferType %d", 215 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, 216 sessionId, transferType); 217 218 switch (transferType) { 219 case TRANSFER_DEFAULT: 220 if (sharedBuffer != 0) { 221 transferType = TRANSFER_SHARED; 222 } else if (cbf == NULL || threadCanCallJava) { 223 transferType = TRANSFER_SYNC; 224 } else { 225 transferType = TRANSFER_CALLBACK; 226 } 227 break; 228 case TRANSFER_CALLBACK: 229 if (cbf == NULL || sharedBuffer != 0) { 230 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 231 return BAD_VALUE; 232 } 233 break; 234 case TRANSFER_OBTAIN: 235 case TRANSFER_SYNC: 236 if (sharedBuffer != 0) { 237 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 238 return BAD_VALUE; 239 } 240 break; 241 case TRANSFER_SHARED: 242 if (sharedBuffer == 0) { 243 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 244 return BAD_VALUE; 245 } 246 break; 247 default: 248 ALOGE("Invalid transfer type %d", transferType); 249 return BAD_VALUE; 250 } 251 mSharedBuffer = sharedBuffer; 252 mTransfer = transferType; 253 254 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 255 sharedBuffer->size()); 256 257 ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags); 258 259 AutoMutex lock(mLock); 260 261 // invariant that mAudioTrack != 0 is true only after set() returns successfully 262 if (mAudioTrack != 0) { 263 ALOGE("Track already in use"); 264 return INVALID_OPERATION; 265 } 266 267 // handle default values first. 268 if (streamType == AUDIO_STREAM_DEFAULT) { 269 streamType = AUDIO_STREAM_MUSIC; 270 } 271 272 if (pAttributes == NULL) { 273 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 274 ALOGE("Invalid stream type %d", streamType); 275 return BAD_VALUE; 276 } 277 setAttributesFromStreamType(streamType); 278 mStreamType = streamType; 279 } else { 280 if (!isValidAttributes(pAttributes)) { 281 ALOGE("Invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]", 282 pAttributes->usage, pAttributes->content_type, pAttributes->flags, 283 pAttributes->tags); 284 } 285 // stream type shouldn't be looked at, this track has audio attributes 286 memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t)); 287 setStreamTypeFromAttributes(mAttributes); 288 ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]", 289 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags); 290 } 291 292 status_t status; 293 if (sampleRate == 0) { 294 status = AudioSystem::getOutputSamplingRateForAttr(&sampleRate, &mAttributes); 295 if (status != NO_ERROR) { 296 ALOGE("Could not get output sample rate for stream type %d; status %d", 297 mStreamType, status); 298 return status; 299 } 300 } 301 mSampleRate = sampleRate; 302 303 // these below should probably come from the audioFlinger too... 304 if (format == AUDIO_FORMAT_DEFAULT) { 305 format = AUDIO_FORMAT_PCM_16_BIT; 306 } 307 308 // validate parameters 309 if (!audio_is_valid_format(format)) { 310 ALOGE("Invalid format %#x", format); 311 return BAD_VALUE; 312 } 313 mFormat = format; 314 315 if (!audio_is_output_channel(channelMask)) { 316 ALOGE("Invalid channel mask %#x", channelMask); 317 return BAD_VALUE; 318 } 319 mChannelMask = channelMask; 320 uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 321 mChannelCount = channelCount; 322 323 // AudioFlinger does not currently support 8-bit data in shared memory 324 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 325 ALOGE("8-bit data in shared memory is not supported"); 326 return BAD_VALUE; 327 } 328 329 // force direct flag if format is not linear PCM 330 // or offload was requested 331 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 332 || !audio_is_linear_pcm(format)) { 333 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 334 ? "Offload request, forcing to Direct Output" 335 : "Not linear PCM, forcing to Direct Output"); 336 flags = (audio_output_flags_t) 337 // FIXME why can't we allow direct AND fast? 338 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 339 } 340 // only allow deep buffering for music stream type 341 if (mStreamType != AUDIO_STREAM_MUSIC) { 342 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 343 } 344 345 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { 346 if (audio_is_linear_pcm(format)) { 347 mFrameSize = channelCount * audio_bytes_per_sample(format); 348 } else { 349 mFrameSize = sizeof(uint8_t); 350 } 351 mFrameSizeAF = mFrameSize; 352 } else { 353 ALOG_ASSERT(audio_is_linear_pcm(format)); 354 mFrameSize = channelCount * audio_bytes_per_sample(format); 355 mFrameSizeAF = channelCount * audio_bytes_per_sample( 356 format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format); 357 // createTrack will return an error if PCM format is not supported by server, 358 // so no need to check for specific PCM formats here 359 } 360 361 // Make copy of input parameter offloadInfo so that in the future: 362 // (a) createTrack_l doesn't need it as an input parameter 363 // (b) we can support re-creation of offloaded tracks 364 if (offloadInfo != NULL) { 365 mOffloadInfoCopy = *offloadInfo; 366 mOffloadInfo = &mOffloadInfoCopy; 367 } else { 368 mOffloadInfo = NULL; 369 } 370 371 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f; 372 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f; 373 mSendLevel = 0.0f; 374 // mFrameCount is initialized in createTrack_l 375 mReqFrameCount = frameCount; 376 mNotificationFramesReq = notificationFrames; 377 mNotificationFramesAct = 0; 378 mSessionId = sessionId; 379 int callingpid = IPCThreadState::self()->getCallingPid(); 380 int mypid = getpid(); 381 if (uid == -1 || (callingpid != mypid)) { 382 mClientUid = IPCThreadState::self()->getCallingUid(); 383 } else { 384 mClientUid = uid; 385 } 386 if (pid == -1 || (callingpid != mypid)) { 387 mClientPid = callingpid; 388 } else { 389 mClientPid = pid; 390 } 391 mAuxEffectId = 0; 392 mFlags = flags; 393 mCbf = cbf; 394 395 if (cbf != NULL) { 396 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 397 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 398 } 399 400 // create the IAudioTrack 401 status = createTrack_l(); 402 403 if (status != NO_ERROR) { 404 if (mAudioTrackThread != 0) { 405 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 406 mAudioTrackThread->requestExitAndWait(); 407 mAudioTrackThread.clear(); 408 } 409 return status; 410 } 411 412 mStatus = NO_ERROR; 413 mState = STATE_STOPPED; 414 mUserData = user; 415 mLoopPeriod = 0; 416 mMarkerPosition = 0; 417 mMarkerReached = false; 418 mNewPosition = 0; 419 mUpdatePeriod = 0; 420 mServer = 0; 421 mPosition = 0; 422 mReleased = 0; 423 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 424 mSequence = 1; 425 mObservedSequence = mSequence; 426 mInUnderrun = false; 427 428 return NO_ERROR; 429} 430 431// ------------------------------------------------------------------------- 432 433status_t AudioTrack::start() 434{ 435 AutoMutex lock(mLock); 436 437 if (mState == STATE_ACTIVE) { 438 return INVALID_OPERATION; 439 } 440 441 mInUnderrun = true; 442 443 State previousState = mState; 444 if (previousState == STATE_PAUSED_STOPPING) { 445 mState = STATE_STOPPING; 446 } else { 447 mState = STATE_ACTIVE; 448 } 449 (void) updateAndGetPosition_l(); 450 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 451 // reset current position as seen by client to 0 452 mPosition = 0; 453 mReleased = 0; 454 // force refresh of remaining frames by processAudioBuffer() as last 455 // write before stop could be partial. 456 mRefreshRemaining = true; 457 } 458 mNewPosition = mPosition + mUpdatePeriod; 459 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 460 461 sp<AudioTrackThread> t = mAudioTrackThread; 462 if (t != 0) { 463 if (previousState == STATE_STOPPING) { 464 mProxy->interrupt(); 465 } else { 466 t->resume(); 467 } 468 } else { 469 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 470 get_sched_policy(0, &mPreviousSchedulingGroup); 471 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 472 } 473 474 status_t status = NO_ERROR; 475 if (!(flags & CBLK_INVALID)) { 476 status = mAudioTrack->start(); 477 if (status == DEAD_OBJECT) { 478 flags |= CBLK_INVALID; 479 } 480 } 481 if (flags & CBLK_INVALID) { 482 status = restoreTrack_l("start"); 483 } 484 485 if (status != NO_ERROR) { 486 ALOGE("start() status %d", status); 487 mState = previousState; 488 if (t != 0) { 489 if (previousState != STATE_STOPPING) { 490 t->pause(); 491 } 492 } else { 493 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 494 set_sched_policy(0, mPreviousSchedulingGroup); 495 } 496 } 497 498 return status; 499} 500 501void AudioTrack::stop() 502{ 503 AutoMutex lock(mLock); 504 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 505 return; 506 } 507 508 if (isOffloaded_l()) { 509 mState = STATE_STOPPING; 510 } else { 511 mState = STATE_STOPPED; 512 } 513 514 mProxy->interrupt(); 515 mAudioTrack->stop(); 516 // the playback head position will reset to 0, so if a marker is set, we need 517 // to activate it again 518 mMarkerReached = false; 519#if 0 520 // Force flush if a shared buffer is used otherwise audioflinger 521 // will not stop before end of buffer is reached. 522 // It may be needed to make sure that we stop playback, likely in case looping is on. 523 if (mSharedBuffer != 0) { 524 flush_l(); 525 } 526#endif 527 528 sp<AudioTrackThread> t = mAudioTrackThread; 529 if (t != 0) { 530 if (!isOffloaded_l()) { 531 t->pause(); 532 } 533 } else { 534 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 535 set_sched_policy(0, mPreviousSchedulingGroup); 536 } 537} 538 539bool AudioTrack::stopped() const 540{ 541 AutoMutex lock(mLock); 542 return mState != STATE_ACTIVE; 543} 544 545void AudioTrack::flush() 546{ 547 if (mSharedBuffer != 0) { 548 return; 549 } 550 AutoMutex lock(mLock); 551 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 552 return; 553 } 554 flush_l(); 555} 556 557void AudioTrack::flush_l() 558{ 559 ALOG_ASSERT(mState != STATE_ACTIVE); 560 561 // clear playback marker and periodic update counter 562 mMarkerPosition = 0; 563 mMarkerReached = false; 564 mUpdatePeriod = 0; 565 mRefreshRemaining = true; 566 567 mState = STATE_FLUSHED; 568 if (isOffloaded_l()) { 569 mProxy->interrupt(); 570 } 571 mProxy->flush(); 572 mAudioTrack->flush(); 573} 574 575void AudioTrack::pause() 576{ 577 AutoMutex lock(mLock); 578 if (mState == STATE_ACTIVE) { 579 mState = STATE_PAUSED; 580 } else if (mState == STATE_STOPPING) { 581 mState = STATE_PAUSED_STOPPING; 582 } else { 583 return; 584 } 585 mProxy->interrupt(); 586 mAudioTrack->pause(); 587 588 if (isOffloaded_l()) { 589 if (mOutput != AUDIO_IO_HANDLE_NONE) { 590 uint32_t halFrames; 591 // OffloadThread sends HAL pause in its threadLoop.. time saved 592 // here can be slightly off 593 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); 594 ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); 595 } 596 } 597} 598 599status_t AudioTrack::setVolume(float left, float right) 600{ 601 // This duplicates a test by AudioTrack JNI, but that is not the only caller 602 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY || 603 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) { 604 return BAD_VALUE; 605 } 606 607 AutoMutex lock(mLock); 608 mVolume[AUDIO_INTERLEAVE_LEFT] = left; 609 mVolume[AUDIO_INTERLEAVE_RIGHT] = right; 610 611 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right))); 612 613 if (isOffloaded_l()) { 614 mAudioTrack->signal(); 615 } 616 return NO_ERROR; 617} 618 619status_t AudioTrack::setVolume(float volume) 620{ 621 return setVolume(volume, volume); 622} 623 624status_t AudioTrack::setAuxEffectSendLevel(float level) 625{ 626 // This duplicates a test by AudioTrack JNI, but that is not the only caller 627 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) { 628 return BAD_VALUE; 629 } 630 631 AutoMutex lock(mLock); 632 mSendLevel = level; 633 mProxy->setSendLevel(level); 634 635 return NO_ERROR; 636} 637 638void AudioTrack::getAuxEffectSendLevel(float* level) const 639{ 640 if (level != NULL) { 641 *level = mSendLevel; 642 } 643} 644 645status_t AudioTrack::setSampleRate(uint32_t rate) 646{ 647 if (mIsTimed || isOffloadedOrDirect()) { 648 return INVALID_OPERATION; 649 } 650 651 uint32_t afSamplingRate; 652 if (AudioSystem::getOutputSamplingRateForAttr(&afSamplingRate, &mAttributes) != NO_ERROR) { 653 return NO_INIT; 654 } 655 if (rate == 0 || rate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 656 return BAD_VALUE; 657 } 658 659 AutoMutex lock(mLock); 660 mSampleRate = rate; 661 mProxy->setSampleRate(rate); 662 663 return NO_ERROR; 664} 665 666uint32_t AudioTrack::getSampleRate() const 667{ 668 if (mIsTimed) { 669 return 0; 670 } 671 672 AutoMutex lock(mLock); 673 674 // sample rate can be updated during playback by the offloaded decoder so we need to 675 // query the HAL and update if needed. 676// FIXME use Proxy return channel to update the rate from server and avoid polling here 677 if (isOffloadedOrDirect_l()) { 678 if (mOutput != AUDIO_IO_HANDLE_NONE) { 679 uint32_t sampleRate = 0; 680 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate); 681 if (status == NO_ERROR) { 682 mSampleRate = sampleRate; 683 } 684 } 685 } 686 return mSampleRate; 687} 688 689status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 690{ 691 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 692 return INVALID_OPERATION; 693 } 694 695 if (loopCount == 0) { 696 ; 697 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 698 loopEnd - loopStart >= MIN_LOOP) { 699 ; 700 } else { 701 return BAD_VALUE; 702 } 703 704 AutoMutex lock(mLock); 705 // See setPosition() regarding setting parameters such as loop points or position while active 706 if (mState == STATE_ACTIVE) { 707 return INVALID_OPERATION; 708 } 709 setLoop_l(loopStart, loopEnd, loopCount); 710 return NO_ERROR; 711} 712 713void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 714{ 715 // FIXME If setting a loop also sets position to start of loop, then 716 // this is correct. Otherwise it should be removed. 717 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 718 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 719 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 720} 721 722status_t AudioTrack::setMarkerPosition(uint32_t marker) 723{ 724 // The only purpose of setting marker position is to get a callback 725 if (mCbf == NULL || isOffloadedOrDirect()) { 726 return INVALID_OPERATION; 727 } 728 729 AutoMutex lock(mLock); 730 mMarkerPosition = marker; 731 mMarkerReached = false; 732 733 return NO_ERROR; 734} 735 736status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 737{ 738 if (isOffloadedOrDirect()) { 739 return INVALID_OPERATION; 740 } 741 if (marker == NULL) { 742 return BAD_VALUE; 743 } 744 745 AutoMutex lock(mLock); 746 *marker = mMarkerPosition; 747 748 return NO_ERROR; 749} 750 751status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 752{ 753 // The only purpose of setting position update period is to get a callback 754 if (mCbf == NULL || isOffloadedOrDirect()) { 755 return INVALID_OPERATION; 756 } 757 758 AutoMutex lock(mLock); 759 mNewPosition = updateAndGetPosition_l() + updatePeriod; 760 mUpdatePeriod = updatePeriod; 761 762 return NO_ERROR; 763} 764 765status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 766{ 767 if (isOffloadedOrDirect()) { 768 return INVALID_OPERATION; 769 } 770 if (updatePeriod == NULL) { 771 return BAD_VALUE; 772 } 773 774 AutoMutex lock(mLock); 775 *updatePeriod = mUpdatePeriod; 776 777 return NO_ERROR; 778} 779 780status_t AudioTrack::setPosition(uint32_t position) 781{ 782 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 783 return INVALID_OPERATION; 784 } 785 if (position > mFrameCount) { 786 return BAD_VALUE; 787 } 788 789 AutoMutex lock(mLock); 790 // Currently we require that the player is inactive before setting parameters such as position 791 // or loop points. Otherwise, there could be a race condition: the application could read the 792 // current position, compute a new position or loop parameters, and then set that position or 793 // loop parameters but it would do the "wrong" thing since the position has continued to advance 794 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 795 // to specify how it wants to handle such scenarios. 796 if (mState == STATE_ACTIVE) { 797 return INVALID_OPERATION; 798 } 799 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 800 mLoopPeriod = 0; 801 // FIXME Check whether loops and setting position are incompatible in old code. 802 // If we use setLoop for both purposes we lose the capability to set the position while looping. 803 mStaticProxy->setLoop(position, mFrameCount, 0); 804 805 return NO_ERROR; 806} 807 808status_t AudioTrack::getPosition(uint32_t *position) 809{ 810 if (position == NULL) { 811 return BAD_VALUE; 812 } 813 814 AutoMutex lock(mLock); 815 if (isOffloadedOrDirect_l()) { 816 uint32_t dspFrames = 0; 817 818 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) { 819 ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); 820 *position = mPausedPosition; 821 return NO_ERROR; 822 } 823 824 if (mOutput != AUDIO_IO_HANDLE_NONE) { 825 uint32_t halFrames; 826 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 827 } 828 *position = dspFrames; 829 } else { 830 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 831 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 832 0 : updateAndGetPosition_l(); 833 } 834 return NO_ERROR; 835} 836 837status_t AudioTrack::getBufferPosition(uint32_t *position) 838{ 839 if (mSharedBuffer == 0 || mIsTimed) { 840 return INVALID_OPERATION; 841 } 842 if (position == NULL) { 843 return BAD_VALUE; 844 } 845 846 AutoMutex lock(mLock); 847 *position = mStaticProxy->getBufferPosition(); 848 return NO_ERROR; 849} 850 851status_t AudioTrack::reload() 852{ 853 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 854 return INVALID_OPERATION; 855 } 856 857 AutoMutex lock(mLock); 858 // See setPosition() regarding setting parameters such as loop points or position while active 859 if (mState == STATE_ACTIVE) { 860 return INVALID_OPERATION; 861 } 862 mNewPosition = mUpdatePeriod; 863 mLoopPeriod = 0; 864 // FIXME The new code cannot reload while keeping a loop specified. 865 // Need to check how the old code handled this, and whether it's a significant change. 866 mStaticProxy->setLoop(0, mFrameCount, 0); 867 return NO_ERROR; 868} 869 870audio_io_handle_t AudioTrack::getOutput() const 871{ 872 AutoMutex lock(mLock); 873 return mOutput; 874} 875 876status_t AudioTrack::attachAuxEffect(int effectId) 877{ 878 AutoMutex lock(mLock); 879 status_t status = mAudioTrack->attachAuxEffect(effectId); 880 if (status == NO_ERROR) { 881 mAuxEffectId = effectId; 882 } 883 return status; 884} 885 886// ------------------------------------------------------------------------- 887 888// must be called with mLock held 889status_t AudioTrack::createTrack_l() 890{ 891 status_t status; 892 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 893 if (audioFlinger == 0) { 894 ALOGE("Could not get audioflinger"); 895 return NO_INIT; 896 } 897 898 audio_io_handle_t output = AudioSystem::getOutputForAttr(&mAttributes, mSampleRate, mFormat, 899 mChannelMask, mFlags, mOffloadInfo); 900 if (output == AUDIO_IO_HANDLE_NONE) { 901 ALOGE("Could not get audio output for stream type %d, usage %d, sample rate %u, format %#x," 902 " channel mask %#x, flags %#x", 903 mStreamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags); 904 return BAD_VALUE; 905 } 906 { 907 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 908 // we must release it ourselves if anything goes wrong. 909 910 // Not all of these values are needed under all conditions, but it is easier to get them all 911 912 uint32_t afLatency; 913 status = AudioSystem::getLatency(output, &afLatency); 914 if (status != NO_ERROR) { 915 ALOGE("getLatency(%d) failed status %d", output, status); 916 goto release; 917 } 918 919 size_t afFrameCount; 920 status = AudioSystem::getFrameCount(output, &afFrameCount); 921 if (status != NO_ERROR) { 922 ALOGE("getFrameCount(output=%d) status %d", output, status); 923 goto release; 924 } 925 926 uint32_t afSampleRate; 927 status = AudioSystem::getSamplingRate(output, &afSampleRate); 928 if (status != NO_ERROR) { 929 ALOGE("getSamplingRate(output=%d) status %d", output, status); 930 goto release; 931 } 932 933 // Client decides whether the track is TIMED (see below), but can only express a preference 934 // for FAST. Server will perform additional tests. 935 if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( 936 // either of these use cases: 937 // use case 1: shared buffer 938 (mSharedBuffer != 0) || 939 // use case 2: callback transfer mode 940 (mTransfer == TRANSFER_CALLBACK)) && 941 // matching sample rate 942 (mSampleRate == afSampleRate))) { 943 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 944 // once denied, do not request again if IAudioTrack is re-created 945 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 946 } 947 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 948 949 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 950 // n = 1 fast track with single buffering; nBuffering is ignored 951 // n = 2 fast track with double buffering 952 // n = 2 normal track, no sample rate conversion 953 // n = 3 normal track, with sample rate conversion 954 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 955 // n > 3 very high latency or very small notification interval; nBuffering is ignored 956 const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; 957 958 mNotificationFramesAct = mNotificationFramesReq; 959 960 size_t frameCount = mReqFrameCount; 961 if (!audio_is_linear_pcm(mFormat)) { 962 963 if (mSharedBuffer != 0) { 964 // Same comment as below about ignoring frameCount parameter for set() 965 frameCount = mSharedBuffer->size(); 966 } else if (frameCount == 0) { 967 frameCount = afFrameCount; 968 } 969 if (mNotificationFramesAct != frameCount) { 970 mNotificationFramesAct = frameCount; 971 } 972 } else if (mSharedBuffer != 0) { 973 974 // Ensure that buffer alignment matches channel count 975 // 8-bit data in shared memory is not currently supported by AudioFlinger 976 size_t alignment = audio_bytes_per_sample( 977 mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat); 978 if (alignment & 1) { 979 alignment = 1; 980 } 981 if (mChannelCount > 1) { 982 // More than 2 channels does not require stronger alignment than stereo 983 alignment <<= 1; 984 } 985 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 986 ALOGE("Invalid buffer alignment: address %p, channel count %u", 987 mSharedBuffer->pointer(), mChannelCount); 988 status = BAD_VALUE; 989 goto release; 990 } 991 992 // When initializing a shared buffer AudioTrack via constructors, 993 // there's no frameCount parameter. 994 // But when initializing a shared buffer AudioTrack via set(), 995 // there _is_ a frameCount parameter. We silently ignore it. 996 frameCount = mSharedBuffer->size() / mFrameSizeAF; 997 998 } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 999 1000 // FIXME move these calculations and associated checks to server 1001 1002 // Ensure that buffer depth covers at least audio hardware latency 1003 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 1004 ALOGV("afFrameCount=%zu, minBufCount=%d, afSampleRate=%u, afLatency=%d", 1005 afFrameCount, minBufCount, afSampleRate, afLatency); 1006 if (minBufCount <= nBuffering) { 1007 minBufCount = nBuffering; 1008 } 1009 1010 size_t minFrameCount = afFrameCount * minBufCount * uint64_t(mSampleRate) / afSampleRate; 1011 ALOGV("minFrameCount: %zu, afFrameCount=%zu, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 1012 ", afLatency=%d", 1013 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); 1014 1015 if (frameCount == 0) { 1016 frameCount = minFrameCount; 1017 } else if (frameCount < minFrameCount) { 1018 // not ALOGW because it happens all the time when playing key clicks over A2DP 1019 ALOGV("Minimum buffer size corrected from %zu to %zu", 1020 frameCount, minFrameCount); 1021 frameCount = minFrameCount; 1022 } 1023 // Make sure that application is notified with sufficient margin before underrun 1024 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1025 mNotificationFramesAct = frameCount/nBuffering; 1026 } 1027 1028 } else { 1029 // For fast tracks, the frame count calculations and checks are done by server 1030 } 1031 1032 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 1033 if (mIsTimed) { 1034 trackFlags |= IAudioFlinger::TRACK_TIMED; 1035 } 1036 1037 pid_t tid = -1; 1038 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1039 trackFlags |= IAudioFlinger::TRACK_FAST; 1040 if (mAudioTrackThread != 0) { 1041 tid = mAudioTrackThread->getTid(); 1042 } 1043 } 1044 1045 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1046 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 1047 } 1048 1049 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1050 trackFlags |= IAudioFlinger::TRACK_DIRECT; 1051 } 1052 1053 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1054 // but we will still need the original value also 1055 sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType, 1056 mSampleRate, 1057 // AudioFlinger only sees 16-bit PCM 1058 mFormat == AUDIO_FORMAT_PCM_8_BIT && 1059 !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT) ? 1060 AUDIO_FORMAT_PCM_16_BIT : mFormat, 1061 mChannelMask, 1062 &temp, 1063 &trackFlags, 1064 mSharedBuffer, 1065 output, 1066 tid, 1067 &mSessionId, 1068 mClientUid, 1069 &status); 1070 1071 if (status != NO_ERROR) { 1072 ALOGE("AudioFlinger could not create track, status: %d", status); 1073 goto release; 1074 } 1075 ALOG_ASSERT(track != 0); 1076 1077 // AudioFlinger now owns the reference to the I/O handle, 1078 // so we are no longer responsible for releasing it. 1079 1080 sp<IMemory> iMem = track->getCblk(); 1081 if (iMem == 0) { 1082 ALOGE("Could not get control block"); 1083 return NO_INIT; 1084 } 1085 void *iMemPointer = iMem->pointer(); 1086 if (iMemPointer == NULL) { 1087 ALOGE("Could not get control block pointer"); 1088 return NO_INIT; 1089 } 1090 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1091 if (mAudioTrack != 0) { 1092 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1093 mDeathNotifier.clear(); 1094 } 1095 mAudioTrack = track; 1096 mCblkMemory = iMem; 1097 IPCThreadState::self()->flushCommands(); 1098 1099 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1100 mCblk = cblk; 1101 // note that temp is the (possibly revised) value of frameCount 1102 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1103 // In current design, AudioTrack client checks and ensures frame count validity before 1104 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1105 // for fast track as it uses a special method of assigning frame count. 1106 ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp); 1107 } 1108 frameCount = temp; 1109 1110 mAwaitBoost = false; 1111 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1112 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1113 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount); 1114 mAwaitBoost = true; 1115 if (mSharedBuffer == 0) { 1116 // Theoretically double-buffering is not required for fast tracks, 1117 // due to tighter scheduling. But in practice, to accommodate kernels with 1118 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1119 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1120 mNotificationFramesAct = frameCount/nBuffering; 1121 } 1122 } 1123 } else { 1124 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount); 1125 // once denied, do not request again if IAudioTrack is re-created 1126 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1127 if (mSharedBuffer == 0) { 1128 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1129 mNotificationFramesAct = frameCount/nBuffering; 1130 } 1131 } 1132 } 1133 } 1134 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1135 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1136 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1137 } else { 1138 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1139 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1140 // FIXME This is a warning, not an error, so don't return error status 1141 //return NO_INIT; 1142 } 1143 } 1144 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1145 if (trackFlags & IAudioFlinger::TRACK_DIRECT) { 1146 ALOGV("AUDIO_OUTPUT_FLAG_DIRECT successful"); 1147 } else { 1148 ALOGW("AUDIO_OUTPUT_FLAG_DIRECT denied by server"); 1149 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT); 1150 // FIXME This is a warning, not an error, so don't return error status 1151 //return NO_INIT; 1152 } 1153 } 1154 1155 // We retain a copy of the I/O handle, but don't own the reference 1156 mOutput = output; 1157 mRefreshRemaining = true; 1158 1159 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1160 // is the value of pointer() for the shared buffer, otherwise buffers points 1161 // immediately after the control block. This address is for the mapping within client 1162 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1163 void* buffers; 1164 if (mSharedBuffer == 0) { 1165 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1166 } else { 1167 buffers = mSharedBuffer->pointer(); 1168 } 1169 1170 mAudioTrack->attachAuxEffect(mAuxEffectId); 1171 // FIXME don't believe this lie 1172 mLatency = afLatency + (1000*frameCount) / mSampleRate; 1173 1174 mFrameCount = frameCount; 1175 // If IAudioTrack is re-created, don't let the requested frameCount 1176 // decrease. This can confuse clients that cache frameCount(). 1177 if (frameCount > mReqFrameCount) { 1178 mReqFrameCount = frameCount; 1179 } 1180 1181 // update proxy 1182 if (mSharedBuffer == 0) { 1183 mStaticProxy.clear(); 1184 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1185 } else { 1186 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1187 mProxy = mStaticProxy; 1188 } 1189 mProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); 1190 mProxy->setSendLevel(mSendLevel); 1191 mProxy->setSampleRate(mSampleRate); 1192 mProxy->setMinimum(mNotificationFramesAct); 1193 1194 mDeathNotifier = new DeathNotifier(this); 1195 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1196 1197 return NO_ERROR; 1198 } 1199 1200release: 1201 AudioSystem::releaseOutput(output); 1202 if (status == NO_ERROR) { 1203 status = NO_INIT; 1204 } 1205 return status; 1206} 1207 1208status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1209{ 1210 if (audioBuffer == NULL) { 1211 return BAD_VALUE; 1212 } 1213 if (mTransfer != TRANSFER_OBTAIN) { 1214 audioBuffer->frameCount = 0; 1215 audioBuffer->size = 0; 1216 audioBuffer->raw = NULL; 1217 return INVALID_OPERATION; 1218 } 1219 1220 const struct timespec *requested; 1221 struct timespec timeout; 1222 if (waitCount == -1) { 1223 requested = &ClientProxy::kForever; 1224 } else if (waitCount == 0) { 1225 requested = &ClientProxy::kNonBlocking; 1226 } else if (waitCount > 0) { 1227 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1228 timeout.tv_sec = ms / 1000; 1229 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1230 requested = &timeout; 1231 } else { 1232 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1233 requested = NULL; 1234 } 1235 return obtainBuffer(audioBuffer, requested); 1236} 1237 1238status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1239 struct timespec *elapsed, size_t *nonContig) 1240{ 1241 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1242 uint32_t oldSequence = 0; 1243 uint32_t newSequence; 1244 1245 Proxy::Buffer buffer; 1246 status_t status = NO_ERROR; 1247 1248 static const int32_t kMaxTries = 5; 1249 int32_t tryCounter = kMaxTries; 1250 1251 do { 1252 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1253 // keep them from going away if another thread re-creates the track during obtainBuffer() 1254 sp<AudioTrackClientProxy> proxy; 1255 sp<IMemory> iMem; 1256 1257 { // start of lock scope 1258 AutoMutex lock(mLock); 1259 1260 newSequence = mSequence; 1261 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1262 if (status == DEAD_OBJECT) { 1263 // re-create track, unless someone else has already done so 1264 if (newSequence == oldSequence) { 1265 status = restoreTrack_l("obtainBuffer"); 1266 if (status != NO_ERROR) { 1267 buffer.mFrameCount = 0; 1268 buffer.mRaw = NULL; 1269 buffer.mNonContig = 0; 1270 break; 1271 } 1272 } 1273 } 1274 oldSequence = newSequence; 1275 1276 // Keep the extra references 1277 proxy = mProxy; 1278 iMem = mCblkMemory; 1279 1280 if (mState == STATE_STOPPING) { 1281 status = -EINTR; 1282 buffer.mFrameCount = 0; 1283 buffer.mRaw = NULL; 1284 buffer.mNonContig = 0; 1285 break; 1286 } 1287 1288 // Non-blocking if track is stopped or paused 1289 if (mState != STATE_ACTIVE) { 1290 requested = &ClientProxy::kNonBlocking; 1291 } 1292 1293 } // end of lock scope 1294 1295 buffer.mFrameCount = audioBuffer->frameCount; 1296 // FIXME starts the requested timeout and elapsed over from scratch 1297 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1298 1299 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1300 1301 audioBuffer->frameCount = buffer.mFrameCount; 1302 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1303 audioBuffer->raw = buffer.mRaw; 1304 if (nonContig != NULL) { 1305 *nonContig = buffer.mNonContig; 1306 } 1307 return status; 1308} 1309 1310void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1311{ 1312 if (mTransfer == TRANSFER_SHARED) { 1313 return; 1314 } 1315 1316 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1317 if (stepCount == 0) { 1318 return; 1319 } 1320 1321 Proxy::Buffer buffer; 1322 buffer.mFrameCount = stepCount; 1323 buffer.mRaw = audioBuffer->raw; 1324 1325 AutoMutex lock(mLock); 1326 mReleased += stepCount; 1327 mInUnderrun = false; 1328 mProxy->releaseBuffer(&buffer); 1329 1330 // restart track if it was disabled by audioflinger due to previous underrun 1331 if (mState == STATE_ACTIVE) { 1332 audio_track_cblk_t* cblk = mCblk; 1333 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1334 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1335 // FIXME ignoring status 1336 mAudioTrack->start(); 1337 } 1338 } 1339} 1340 1341// ------------------------------------------------------------------------- 1342 1343ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1344{ 1345 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1346 return INVALID_OPERATION; 1347 } 1348 1349 if (isDirect()) { 1350 AutoMutex lock(mLock); 1351 int32_t flags = android_atomic_and( 1352 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), 1353 &mCblk->mFlags); 1354 if (flags & CBLK_INVALID) { 1355 return DEAD_OBJECT; 1356 } 1357 } 1358 1359 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1360 // Sanity-check: user is most-likely passing an error code, and it would 1361 // make the return value ambiguous (actualSize vs error). 1362 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1363 return BAD_VALUE; 1364 } 1365 1366 size_t written = 0; 1367 Buffer audioBuffer; 1368 1369 while (userSize >= mFrameSize) { 1370 audioBuffer.frameCount = userSize / mFrameSize; 1371 1372 status_t err = obtainBuffer(&audioBuffer, 1373 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1374 if (err < 0) { 1375 if (written > 0) { 1376 break; 1377 } 1378 return ssize_t(err); 1379 } 1380 1381 size_t toWrite; 1382 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1383 // Divide capacity by 2 to take expansion into account 1384 toWrite = audioBuffer.size >> 1; 1385 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1386 } else { 1387 toWrite = audioBuffer.size; 1388 memcpy(audioBuffer.i8, buffer, toWrite); 1389 } 1390 buffer = ((const char *) buffer) + toWrite; 1391 userSize -= toWrite; 1392 written += toWrite; 1393 1394 releaseBuffer(&audioBuffer); 1395 } 1396 1397 return written; 1398} 1399 1400// ------------------------------------------------------------------------- 1401 1402TimedAudioTrack::TimedAudioTrack() { 1403 mIsTimed = true; 1404} 1405 1406status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1407{ 1408 AutoMutex lock(mLock); 1409 status_t result = UNKNOWN_ERROR; 1410 1411#if 1 1412 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1413 // while we are accessing the cblk 1414 sp<IAudioTrack> audioTrack = mAudioTrack; 1415 sp<IMemory> iMem = mCblkMemory; 1416#endif 1417 1418 // If the track is not invalid already, try to allocate a buffer. alloc 1419 // fails indicating that the server is dead, flag the track as invalid so 1420 // we can attempt to restore in just a bit. 1421 audio_track_cblk_t* cblk = mCblk; 1422 if (!(cblk->mFlags & CBLK_INVALID)) { 1423 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1424 if (result == DEAD_OBJECT) { 1425 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1426 } 1427 } 1428 1429 // If the track is invalid at this point, attempt to restore it. and try the 1430 // allocation one more time. 1431 if (cblk->mFlags & CBLK_INVALID) { 1432 result = restoreTrack_l("allocateTimedBuffer"); 1433 1434 if (result == NO_ERROR) { 1435 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1436 } 1437 } 1438 1439 return result; 1440} 1441 1442status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1443 int64_t pts) 1444{ 1445 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1446 { 1447 AutoMutex lock(mLock); 1448 audio_track_cblk_t* cblk = mCblk; 1449 // restart track if it was disabled by audioflinger due to previous underrun 1450 if (buffer->size() != 0 && status == NO_ERROR && 1451 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1452 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1453 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1454 // FIXME ignoring status 1455 mAudioTrack->start(); 1456 } 1457 } 1458 return status; 1459} 1460 1461status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1462 TargetTimeline target) 1463{ 1464 return mAudioTrack->setMediaTimeTransform(xform, target); 1465} 1466 1467// ------------------------------------------------------------------------- 1468 1469nsecs_t AudioTrack::processAudioBuffer() 1470{ 1471 // Currently the AudioTrack thread is not created if there are no callbacks. 1472 // Would it ever make sense to run the thread, even without callbacks? 1473 // If so, then replace this by checks at each use for mCbf != NULL. 1474 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1475 1476 mLock.lock(); 1477 if (mAwaitBoost) { 1478 mAwaitBoost = false; 1479 mLock.unlock(); 1480 static const int32_t kMaxTries = 5; 1481 int32_t tryCounter = kMaxTries; 1482 uint32_t pollUs = 10000; 1483 do { 1484 int policy = sched_getscheduler(0); 1485 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1486 break; 1487 } 1488 usleep(pollUs); 1489 pollUs <<= 1; 1490 } while (tryCounter-- > 0); 1491 if (tryCounter < 0) { 1492 ALOGE("did not receive expected priority boost on time"); 1493 } 1494 // Run again immediately 1495 return 0; 1496 } 1497 1498 // Can only reference mCblk while locked 1499 int32_t flags = android_atomic_and( 1500 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1501 1502 // Check for track invalidation 1503 if (flags & CBLK_INVALID) { 1504 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1505 // AudioSystem cache. We should not exit here but after calling the callback so 1506 // that the upper layers can recreate the track 1507 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) { 1508 status_t status = restoreTrack_l("processAudioBuffer"); 1509 mLock.unlock(); 1510 // Run again immediately, but with a new IAudioTrack 1511 return 0; 1512 } 1513 } 1514 1515 bool waitStreamEnd = mState == STATE_STOPPING; 1516 bool active = mState == STATE_ACTIVE; 1517 1518 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1519 bool newUnderrun = false; 1520 if (flags & CBLK_UNDERRUN) { 1521#if 0 1522 // Currently in shared buffer mode, when the server reaches the end of buffer, 1523 // the track stays active in continuous underrun state. It's up to the application 1524 // to pause or stop the track, or set the position to a new offset within buffer. 1525 // This was some experimental code to auto-pause on underrun. Keeping it here 1526 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1527 if (mTransfer == TRANSFER_SHARED) { 1528 mState = STATE_PAUSED; 1529 active = false; 1530 } 1531#endif 1532 if (!mInUnderrun) { 1533 mInUnderrun = true; 1534 newUnderrun = true; 1535 } 1536 } 1537 1538 // Get current position of server 1539 size_t position = updateAndGetPosition_l(); 1540 1541 // Manage marker callback 1542 bool markerReached = false; 1543 size_t markerPosition = mMarkerPosition; 1544 // FIXME fails for wraparound, need 64 bits 1545 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1546 mMarkerReached = markerReached = true; 1547 } 1548 1549 // Determine number of new position callback(s) that will be needed, while locked 1550 size_t newPosCount = 0; 1551 size_t newPosition = mNewPosition; 1552 size_t updatePeriod = mUpdatePeriod; 1553 // FIXME fails for wraparound, need 64 bits 1554 if (updatePeriod > 0 && position >= newPosition) { 1555 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1556 mNewPosition += updatePeriod * newPosCount; 1557 } 1558 1559 // Cache other fields that will be needed soon 1560 uint32_t loopPeriod = mLoopPeriod; 1561 uint32_t sampleRate = mSampleRate; 1562 uint32_t notificationFrames = mNotificationFramesAct; 1563 if (mRefreshRemaining) { 1564 mRefreshRemaining = false; 1565 mRemainingFrames = notificationFrames; 1566 mRetryOnPartialBuffer = false; 1567 } 1568 size_t misalignment = mProxy->getMisalignment(); 1569 uint32_t sequence = mSequence; 1570 sp<AudioTrackClientProxy> proxy = mProxy; 1571 1572 // These fields don't need to be cached, because they are assigned only by set(): 1573 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1574 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1575 1576 mLock.unlock(); 1577 1578 if (waitStreamEnd) { 1579 struct timespec timeout; 1580 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1581 timeout.tv_nsec = 0; 1582 1583 status_t status = proxy->waitStreamEndDone(&timeout); 1584 switch (status) { 1585 case NO_ERROR: 1586 case DEAD_OBJECT: 1587 case TIMED_OUT: 1588 mCbf(EVENT_STREAM_END, mUserData, NULL); 1589 { 1590 AutoMutex lock(mLock); 1591 // The previously assigned value of waitStreamEnd is no longer valid, 1592 // since the mutex has been unlocked and either the callback handler 1593 // or another thread could have re-started the AudioTrack during that time. 1594 waitStreamEnd = mState == STATE_STOPPING; 1595 if (waitStreamEnd) { 1596 mState = STATE_STOPPED; 1597 } 1598 } 1599 if (waitStreamEnd && status != DEAD_OBJECT) { 1600 return NS_INACTIVE; 1601 } 1602 break; 1603 } 1604 return 0; 1605 } 1606 1607 // perform callbacks while unlocked 1608 if (newUnderrun) { 1609 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1610 } 1611 // FIXME we will miss loops if loop cycle was signaled several times since last call 1612 // to processAudioBuffer() 1613 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1614 mCbf(EVENT_LOOP_END, mUserData, NULL); 1615 } 1616 if (flags & CBLK_BUFFER_END) { 1617 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1618 } 1619 if (markerReached) { 1620 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1621 } 1622 while (newPosCount > 0) { 1623 size_t temp = newPosition; 1624 mCbf(EVENT_NEW_POS, mUserData, &temp); 1625 newPosition += updatePeriod; 1626 newPosCount--; 1627 } 1628 1629 if (mObservedSequence != sequence) { 1630 mObservedSequence = sequence; 1631 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1632 // for offloaded tracks, just wait for the upper layers to recreate the track 1633 if (isOffloadedOrDirect()) { 1634 return NS_INACTIVE; 1635 } 1636 } 1637 1638 // if inactive, then don't run me again until re-started 1639 if (!active) { 1640 return NS_INACTIVE; 1641 } 1642 1643 // Compute the estimated time until the next timed event (position, markers, loops) 1644 // FIXME only for non-compressed audio 1645 uint32_t minFrames = ~0; 1646 if (!markerReached && position < markerPosition) { 1647 minFrames = markerPosition - position; 1648 } 1649 if (loopPeriod > 0 && loopPeriod < minFrames) { 1650 minFrames = loopPeriod; 1651 } 1652 if (updatePeriod > 0 && updatePeriod < minFrames) { 1653 minFrames = updatePeriod; 1654 } 1655 1656 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1657 static const uint32_t kPoll = 0; 1658 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1659 minFrames = kPoll * notificationFrames; 1660 } 1661 1662 // Convert frame units to time units 1663 nsecs_t ns = NS_WHENEVER; 1664 if (minFrames != (uint32_t) ~0) { 1665 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1666 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1667 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1668 } 1669 1670 // If not supplying data by EVENT_MORE_DATA, then we're done 1671 if (mTransfer != TRANSFER_CALLBACK) { 1672 return ns; 1673 } 1674 1675 struct timespec timeout; 1676 const struct timespec *requested = &ClientProxy::kForever; 1677 if (ns != NS_WHENEVER) { 1678 timeout.tv_sec = ns / 1000000000LL; 1679 timeout.tv_nsec = ns % 1000000000LL; 1680 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1681 requested = &timeout; 1682 } 1683 1684 while (mRemainingFrames > 0) { 1685 1686 Buffer audioBuffer; 1687 audioBuffer.frameCount = mRemainingFrames; 1688 size_t nonContig; 1689 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1690 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1691 "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount); 1692 requested = &ClientProxy::kNonBlocking; 1693 size_t avail = audioBuffer.frameCount + nonContig; 1694 ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d", 1695 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1696 if (err != NO_ERROR) { 1697 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1698 (isOffloaded() && (err == DEAD_OBJECT))) { 1699 return 0; 1700 } 1701 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1702 return NS_NEVER; 1703 } 1704 1705 if (mRetryOnPartialBuffer && !isOffloaded()) { 1706 mRetryOnPartialBuffer = false; 1707 if (avail < mRemainingFrames) { 1708 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1709 if (ns < 0 || myns < ns) { 1710 ns = myns; 1711 } 1712 return ns; 1713 } 1714 } 1715 1716 // Divide buffer size by 2 to take into account the expansion 1717 // due to 8 to 16 bit conversion: the callback must fill only half 1718 // of the destination buffer 1719 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1720 audioBuffer.size >>= 1; 1721 } 1722 1723 size_t reqSize = audioBuffer.size; 1724 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1725 size_t writtenSize = audioBuffer.size; 1726 1727 // Sanity check on returned size 1728 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1729 ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes", 1730 reqSize, ssize_t(writtenSize)); 1731 return NS_NEVER; 1732 } 1733 1734 if (writtenSize == 0) { 1735 // The callback is done filling buffers 1736 // Keep this thread going to handle timed events and 1737 // still try to get more data in intervals of WAIT_PERIOD_MS 1738 // but don't just loop and block the CPU, so wait 1739 return WAIT_PERIOD_MS * 1000000LL; 1740 } 1741 1742 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1743 // 8 to 16 bit conversion, note that source and destination are the same address 1744 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1745 audioBuffer.size <<= 1; 1746 } 1747 1748 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1749 audioBuffer.frameCount = releasedFrames; 1750 mRemainingFrames -= releasedFrames; 1751 if (misalignment >= releasedFrames) { 1752 misalignment -= releasedFrames; 1753 } else { 1754 misalignment = 0; 1755 } 1756 1757 releaseBuffer(&audioBuffer); 1758 1759 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1760 // if callback doesn't like to accept the full chunk 1761 if (writtenSize < reqSize) { 1762 continue; 1763 } 1764 1765 // There could be enough non-contiguous frames available to satisfy the remaining request 1766 if (mRemainingFrames <= nonContig) { 1767 continue; 1768 } 1769 1770#if 0 1771 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1772 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1773 // that total to a sum == notificationFrames. 1774 if (0 < misalignment && misalignment <= mRemainingFrames) { 1775 mRemainingFrames = misalignment; 1776 return (mRemainingFrames * 1100000000LL) / sampleRate; 1777 } 1778#endif 1779 1780 } 1781 mRemainingFrames = notificationFrames; 1782 mRetryOnPartialBuffer = true; 1783 1784 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1785 return 0; 1786} 1787 1788status_t AudioTrack::restoreTrack_l(const char *from) 1789{ 1790 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1791 isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from); 1792 ++mSequence; 1793 status_t result; 1794 1795 // refresh the audio configuration cache in this process to make sure we get new 1796 // output parameters in createTrack_l() 1797 AudioSystem::clearAudioConfigCache(); 1798 1799 if (isOffloadedOrDirect_l()) { 1800 // FIXME re-creation of offloaded tracks is not yet implemented 1801 return DEAD_OBJECT; 1802 } 1803 1804 // save the old static buffer position 1805 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1806 1807 // If a new IAudioTrack is successfully created, createTrack_l() will modify the 1808 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1809 // It will also delete the strong references on previous IAudioTrack and IMemory. 1810 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact. 1811 result = createTrack_l(); 1812 1813 // take the frames that will be lost by track recreation into account in saved position 1814 (void) updateAndGetPosition_l(); 1815 mPosition = mReleased; 1816 1817 if (result == NO_ERROR) { 1818 // continue playback from last known position, but 1819 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1820 if (mStaticProxy != NULL) { 1821 mLoopPeriod = 0; 1822 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1823 } 1824 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1825 // track destruction have been played? This is critical for SoundPool implementation 1826 // This must be broken, and needs to be tested/debugged. 1827#if 0 1828 // restore write index and set other indexes to reflect empty buffer status 1829 if (!strcmp(from, "start")) { 1830 // Make sure that a client relying on callback events indicating underrun or 1831 // the actual amount of audio frames played (e.g SoundPool) receives them. 1832 if (mSharedBuffer == 0) { 1833 // restart playback even if buffer is not completely filled. 1834 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1835 } 1836 } 1837#endif 1838 if (mState == STATE_ACTIVE) { 1839 result = mAudioTrack->start(); 1840 } 1841 } 1842 if (result != NO_ERROR) { 1843 ALOGW("restoreTrack_l() failed status %d", result); 1844 mState = STATE_STOPPED; 1845 } 1846 1847 return result; 1848} 1849 1850uint32_t AudioTrack::updateAndGetPosition_l() 1851{ 1852 // This is the sole place to read server consumed frames 1853 uint32_t newServer = mProxy->getPosition(); 1854 int32_t delta = newServer - mServer; 1855 mServer = newServer; 1856 // TODO There is controversy about whether there can be "negative jitter" in server position. 1857 // This should be investigated further, and if possible, it should be addressed. 1858 // A more definite failure mode is infrequent polling by client. 1859 // One could call (void)getPosition_l() in releaseBuffer(), 1860 // so mReleased and mPosition are always lock-step as best possible. 1861 // That should ensure delta never goes negative for infrequent polling 1862 // unless the server has more than 2^31 frames in its buffer, 1863 // in which case the use of uint32_t for these counters has bigger issues. 1864 if (delta < 0) { 1865 ALOGE("detected illegal retrograde motion by the server: mServer advanced by %d", delta); 1866 delta = 0; 1867 } 1868 return mPosition += (uint32_t) delta; 1869} 1870 1871status_t AudioTrack::setParameters(const String8& keyValuePairs) 1872{ 1873 AutoMutex lock(mLock); 1874 return mAudioTrack->setParameters(keyValuePairs); 1875} 1876 1877status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1878{ 1879 AutoMutex lock(mLock); 1880 // FIXME not implemented for fast tracks; should use proxy and SSQ 1881 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1882 return INVALID_OPERATION; 1883 } 1884 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1885 return INVALID_OPERATION; 1886 } 1887 // The presented frame count must always lag behind the consumed frame count. 1888 // To avoid a race, read the presented frames first. This ensures that presented <= consumed. 1889 status_t status = mAudioTrack->getTimestamp(timestamp); 1890 if (status == NO_ERROR) { 1891 // Update the mapping between local consumed (mPosition) and server consumed (mServer) 1892 (void) updateAndGetPosition_l(); 1893 // Server consumed (mServer) and presented both use the same server time base, 1894 // and server consumed is always >= presented. 1895 // The delta between these represents the number of frames in the buffer pipeline. 1896 // If this delta between these is greater than the client position, it means that 1897 // actually presented is still stuck at the starting line (figuratively speaking), 1898 // waiting for the first frame to go by. So we can't report a valid timestamp yet. 1899 if ((uint32_t) (mServer - timestamp.mPosition) > mPosition) { 1900 return INVALID_OPERATION; 1901 } 1902 // Convert timestamp position from server time base to client time base. 1903 // TODO The following code should work OK now because timestamp.mPosition is 32-bit. 1904 // But if we change it to 64-bit then this could fail. 1905 // If (mPosition - mServer) can be negative then should use: 1906 // (int32_t)(mPosition - mServer) 1907 timestamp.mPosition += mPosition - mServer; 1908 // Immediately after a call to getPosition_l(), mPosition and 1909 // mServer both represent the same frame position. mPosition is 1910 // in client's point of view, and mServer is in server's point of 1911 // view. So the difference between them is the "fudge factor" 1912 // between client and server views due to stop() and/or new 1913 // IAudioTrack. And timestamp.mPosition is initially in server's 1914 // point of view, so we need to apply the same fudge factor to it. 1915 } 1916 return status; 1917} 1918 1919String8 AudioTrack::getParameters(const String8& keys) 1920{ 1921 audio_io_handle_t output = getOutput(); 1922 if (output != AUDIO_IO_HANDLE_NONE) { 1923 return AudioSystem::getParameters(output, keys); 1924 } else { 1925 return String8::empty(); 1926 } 1927} 1928 1929bool AudioTrack::isOffloaded() const 1930{ 1931 AutoMutex lock(mLock); 1932 return isOffloaded_l(); 1933} 1934 1935bool AudioTrack::isDirect() const 1936{ 1937 AutoMutex lock(mLock); 1938 return isDirect_l(); 1939} 1940 1941bool AudioTrack::isOffloadedOrDirect() const 1942{ 1943 AutoMutex lock(mLock); 1944 return isOffloadedOrDirect_l(); 1945} 1946 1947 1948status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 1949{ 1950 1951 const size_t SIZE = 256; 1952 char buffer[SIZE]; 1953 String8 result; 1954 1955 result.append(" AudioTrack::dump\n"); 1956 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1957 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 1958 result.append(buffer); 1959 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 1960 mChannelCount, mFrameCount); 1961 result.append(buffer); 1962 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1963 result.append(buffer); 1964 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1965 result.append(buffer); 1966 ::write(fd, result.string(), result.size()); 1967 return NO_ERROR; 1968} 1969 1970uint32_t AudioTrack::getUnderrunFrames() const 1971{ 1972 AutoMutex lock(mLock); 1973 return mProxy->getUnderrunFrames(); 1974} 1975 1976void AudioTrack::setAttributesFromStreamType(audio_stream_type_t streamType) { 1977 mAttributes.flags = 0x0; 1978 1979 switch(streamType) { 1980 case AUDIO_STREAM_DEFAULT: 1981 case AUDIO_STREAM_MUSIC: 1982 mAttributes.content_type = AUDIO_CONTENT_TYPE_MUSIC; 1983 mAttributes.usage = AUDIO_USAGE_MEDIA; 1984 break; 1985 case AUDIO_STREAM_VOICE_CALL: 1986 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 1987 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION; 1988 break; 1989 case AUDIO_STREAM_ENFORCED_AUDIBLE: 1990 mAttributes.flags |= AUDIO_FLAG_AUDIBILITY_ENFORCED; 1991 // intended fall through, attributes in common with STREAM_SYSTEM 1992 case AUDIO_STREAM_SYSTEM: 1993 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 1994 mAttributes.usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION; 1995 break; 1996 case AUDIO_STREAM_RING: 1997 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 1998 mAttributes.usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE; 1999 break; 2000 case AUDIO_STREAM_ALARM: 2001 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2002 mAttributes.usage = AUDIO_USAGE_ALARM; 2003 break; 2004 case AUDIO_STREAM_NOTIFICATION: 2005 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2006 mAttributes.usage = AUDIO_USAGE_NOTIFICATION; 2007 break; 2008 case AUDIO_STREAM_BLUETOOTH_SCO: 2009 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 2010 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION; 2011 mAttributes.flags |= AUDIO_FLAG_SCO; 2012 break; 2013 case AUDIO_STREAM_DTMF: 2014 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2015 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING; 2016 break; 2017 case AUDIO_STREAM_TTS: 2018 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 2019 mAttributes.usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY; 2020 break; 2021 default: 2022 ALOGE("invalid stream type %d when converting to attributes", streamType); 2023 } 2024} 2025 2026void AudioTrack::setStreamTypeFromAttributes(audio_attributes_t& aa) { 2027 // flags to stream type mapping 2028 if ((aa.flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { 2029 mStreamType = AUDIO_STREAM_ENFORCED_AUDIBLE; 2030 return; 2031 } 2032 if ((aa.flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) { 2033 mStreamType = AUDIO_STREAM_BLUETOOTH_SCO; 2034 return; 2035 } 2036 2037 // usage to stream type mapping 2038 switch (aa.usage) { 2039 case AUDIO_USAGE_MEDIA: 2040 case AUDIO_USAGE_GAME: 2041 case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: 2042 case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: 2043 mStreamType = AUDIO_STREAM_MUSIC; 2044 return; 2045 case AUDIO_USAGE_ASSISTANCE_SONIFICATION: 2046 mStreamType = AUDIO_STREAM_SYSTEM; 2047 return; 2048 case AUDIO_USAGE_VOICE_COMMUNICATION: 2049 mStreamType = AUDIO_STREAM_VOICE_CALL; 2050 return; 2051 2052 case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: 2053 mStreamType = AUDIO_STREAM_DTMF; 2054 return; 2055 2056 case AUDIO_USAGE_ALARM: 2057 mStreamType = AUDIO_STREAM_ALARM; 2058 return; 2059 case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: 2060 mStreamType = AUDIO_STREAM_RING; 2061 return; 2062 2063 case AUDIO_USAGE_NOTIFICATION: 2064 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: 2065 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: 2066 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: 2067 case AUDIO_USAGE_NOTIFICATION_EVENT: 2068 mStreamType = AUDIO_STREAM_NOTIFICATION; 2069 return; 2070 2071 case AUDIO_USAGE_UNKNOWN: 2072 default: 2073 mStreamType = AUDIO_STREAM_MUSIC; 2074 } 2075} 2076 2077bool AudioTrack::isValidAttributes(const audio_attributes_t *paa) { 2078 // has flags that map to a strategy? 2079 if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO)) != 0) { 2080 return true; 2081 } 2082 2083 // has known usage? 2084 switch (paa->usage) { 2085 case AUDIO_USAGE_UNKNOWN: 2086 case AUDIO_USAGE_MEDIA: 2087 case AUDIO_USAGE_VOICE_COMMUNICATION: 2088 case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: 2089 case AUDIO_USAGE_ALARM: 2090 case AUDIO_USAGE_NOTIFICATION: 2091 case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: 2092 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: 2093 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: 2094 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: 2095 case AUDIO_USAGE_NOTIFICATION_EVENT: 2096 case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: 2097 case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: 2098 case AUDIO_USAGE_ASSISTANCE_SONIFICATION: 2099 case AUDIO_USAGE_GAME: 2100 break; 2101 default: 2102 return false; 2103 } 2104 return true; 2105} 2106// ========================================================================= 2107 2108void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 2109{ 2110 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 2111 if (audioTrack != 0) { 2112 AutoMutex lock(audioTrack->mLock); 2113 audioTrack->mProxy->binderDied(); 2114 } 2115} 2116 2117// ========================================================================= 2118 2119AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 2120 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 2121 mIgnoreNextPausedInt(false) 2122{ 2123} 2124 2125AudioTrack::AudioTrackThread::~AudioTrackThread() 2126{ 2127} 2128 2129bool AudioTrack::AudioTrackThread::threadLoop() 2130{ 2131 { 2132 AutoMutex _l(mMyLock); 2133 if (mPaused) { 2134 mMyCond.wait(mMyLock); 2135 // caller will check for exitPending() 2136 return true; 2137 } 2138 if (mIgnoreNextPausedInt) { 2139 mIgnoreNextPausedInt = false; 2140 mPausedInt = false; 2141 } 2142 if (mPausedInt) { 2143 if (mPausedNs > 0) { 2144 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 2145 } else { 2146 mMyCond.wait(mMyLock); 2147 } 2148 mPausedInt = false; 2149 return true; 2150 } 2151 } 2152 nsecs_t ns = mReceiver.processAudioBuffer(); 2153 switch (ns) { 2154 case 0: 2155 return true; 2156 case NS_INACTIVE: 2157 pauseInternal(); 2158 return true; 2159 case NS_NEVER: 2160 return false; 2161 case NS_WHENEVER: 2162 // FIXME increase poll interval, or make event-driven 2163 ns = 1000000000LL; 2164 // fall through 2165 default: 2166 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns); 2167 pauseInternal(ns); 2168 return true; 2169 } 2170} 2171 2172void AudioTrack::AudioTrackThread::requestExit() 2173{ 2174 // must be in this order to avoid a race condition 2175 Thread::requestExit(); 2176 resume(); 2177} 2178 2179void AudioTrack::AudioTrackThread::pause() 2180{ 2181 AutoMutex _l(mMyLock); 2182 mPaused = true; 2183} 2184 2185void AudioTrack::AudioTrackThread::resume() 2186{ 2187 AutoMutex _l(mMyLock); 2188 mIgnoreNextPausedInt = true; 2189 if (mPaused || mPausedInt) { 2190 mPaused = false; 2191 mPausedInt = false; 2192 mMyCond.signal(); 2193 } 2194} 2195 2196void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 2197{ 2198 AutoMutex _l(mMyLock); 2199 mPausedInt = true; 2200 mPausedNs = ns; 2201} 2202 2203}; // namespace android 2204