AudioTrack.cpp revision 551b5355d34aa42890811fc3606d3b63429296cd
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/AudioPolicyHelper.h> 32#include <media/AudioResamplerPublic.h> 33 34#define WAIT_PERIOD_MS 10 35#define WAIT_STREAM_END_TIMEOUT_SEC 120 36static const int kMaxLoopCountNotifications = 32; 37 38namespace android { 39// --------------------------------------------------------------------------- 40 41template <typename T> 42const T &min(const T &x, const T &y) { 43 return x < y ? x : y; 44} 45 46static int64_t convertTimespecToUs(const struct timespec &tv) 47{ 48 return tv.tv_sec * 1000000ll + tv.tv_nsec / 1000; 49} 50 51// current monotonic time in microseconds. 52static int64_t getNowUs() 53{ 54 struct timespec tv; 55 (void) clock_gettime(CLOCK_MONOTONIC, &tv); 56 return convertTimespecToUs(tv); 57} 58 59// static 60status_t AudioTrack::getMinFrameCount( 61 size_t* frameCount, 62 audio_stream_type_t streamType, 63 uint32_t sampleRate) 64{ 65 if (frameCount == NULL) { 66 return BAD_VALUE; 67 } 68 69 // FIXME handle in server, like createTrack_l(), possible missing info: 70 // audio_io_handle_t output 71 // audio_format_t format 72 // audio_channel_mask_t channelMask 73 // audio_output_flags_t flags (FAST) 74 uint32_t afSampleRate; 75 status_t status; 76 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 77 if (status != NO_ERROR) { 78 ALOGE("Unable to query output sample rate for stream type %d; status %d", 79 streamType, status); 80 return status; 81 } 82 size_t afFrameCount; 83 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 84 if (status != NO_ERROR) { 85 ALOGE("Unable to query output frame count for stream type %d; status %d", 86 streamType, status); 87 return status; 88 } 89 uint32_t afLatency; 90 status = AudioSystem::getOutputLatency(&afLatency, streamType); 91 if (status != NO_ERROR) { 92 ALOGE("Unable to query output latency for stream type %d; status %d", 93 streamType, status); 94 return status; 95 } 96 97 // Ensure that buffer depth covers at least audio hardware latency 98 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 99 if (minBufCount < 2) { 100 minBufCount = 2; 101 } 102 103 *frameCount = minBufCount * sourceFramesNeeded(sampleRate, afFrameCount, afSampleRate); 104 // The formula above should always produce a non-zero value under normal circumstances: 105 // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX. 106 // Return error in the unlikely event that it does not, as that's part of the API contract. 107 if (*frameCount == 0) { 108 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %u", 109 streamType, sampleRate); 110 return BAD_VALUE; 111 } 112 ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%u, afSampleRate=%u, afLatency=%u", 113 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 114 return NO_ERROR; 115} 116 117// --------------------------------------------------------------------------- 118 119AudioTrack::AudioTrack() 120 : mStatus(NO_INIT), 121 mIsTimed(false), 122 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 123 mPreviousSchedulingGroup(SP_DEFAULT), 124 mPausedPosition(0) 125{ 126 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN; 127 mAttributes.usage = AUDIO_USAGE_UNKNOWN; 128 mAttributes.flags = 0x0; 129 strcpy(mAttributes.tags, ""); 130} 131 132AudioTrack::AudioTrack( 133 audio_stream_type_t streamType, 134 uint32_t sampleRate, 135 audio_format_t format, 136 audio_channel_mask_t channelMask, 137 size_t frameCount, 138 audio_output_flags_t flags, 139 callback_t cbf, 140 void* user, 141 uint32_t notificationFrames, 142 int sessionId, 143 transfer_type transferType, 144 const audio_offload_info_t *offloadInfo, 145 int uid, 146 pid_t pid, 147 const audio_attributes_t* pAttributes) 148 : mStatus(NO_INIT), 149 mIsTimed(false), 150 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 151 mPreviousSchedulingGroup(SP_DEFAULT), 152 mPausedPosition(0) 153{ 154 mStatus = set(streamType, sampleRate, format, channelMask, 155 frameCount, flags, cbf, user, notificationFrames, 156 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 157 offloadInfo, uid, pid, pAttributes); 158} 159 160AudioTrack::AudioTrack( 161 audio_stream_type_t streamType, 162 uint32_t sampleRate, 163 audio_format_t format, 164 audio_channel_mask_t channelMask, 165 const sp<IMemory>& sharedBuffer, 166 audio_output_flags_t flags, 167 callback_t cbf, 168 void* user, 169 uint32_t notificationFrames, 170 int sessionId, 171 transfer_type transferType, 172 const audio_offload_info_t *offloadInfo, 173 int uid, 174 pid_t pid, 175 const audio_attributes_t* pAttributes) 176 : mStatus(NO_INIT), 177 mIsTimed(false), 178 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 179 mPreviousSchedulingGroup(SP_DEFAULT), 180 mPausedPosition(0) 181{ 182 mStatus = set(streamType, sampleRate, format, channelMask, 183 0 /*frameCount*/, flags, cbf, user, notificationFrames, 184 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, 185 uid, pid, pAttributes); 186} 187 188AudioTrack::~AudioTrack() 189{ 190 if (mStatus == NO_ERROR) { 191 // Make sure that callback function exits in the case where 192 // it is looping on buffer full condition in obtainBuffer(). 193 // Otherwise the callback thread will never exit. 194 stop(); 195 if (mAudioTrackThread != 0) { 196 mProxy->interrupt(); 197 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 198 mAudioTrackThread->requestExitAndWait(); 199 mAudioTrackThread.clear(); 200 } 201 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this); 202 mAudioTrack.clear(); 203 mCblkMemory.clear(); 204 mSharedBuffer.clear(); 205 IPCThreadState::self()->flushCommands(); 206 ALOGV("~AudioTrack, releasing session id %d from %d on behalf of %d", 207 mSessionId, IPCThreadState::self()->getCallingPid(), mClientPid); 208 AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); 209 } 210} 211 212status_t AudioTrack::set( 213 audio_stream_type_t streamType, 214 uint32_t sampleRate, 215 audio_format_t format, 216 audio_channel_mask_t channelMask, 217 size_t frameCount, 218 audio_output_flags_t flags, 219 callback_t cbf, 220 void* user, 221 uint32_t notificationFrames, 222 const sp<IMemory>& sharedBuffer, 223 bool threadCanCallJava, 224 int sessionId, 225 transfer_type transferType, 226 const audio_offload_info_t *offloadInfo, 227 int uid, 228 pid_t pid, 229 const audio_attributes_t* pAttributes) 230{ 231 ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 232 "flags #%x, notificationFrames %u, sessionId %d, transferType %d, uid %d, pid %d", 233 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, 234 sessionId, transferType, uid, pid); 235 236 switch (transferType) { 237 case TRANSFER_DEFAULT: 238 if (sharedBuffer != 0) { 239 transferType = TRANSFER_SHARED; 240 } else if (cbf == NULL || threadCanCallJava) { 241 transferType = TRANSFER_SYNC; 242 } else { 243 transferType = TRANSFER_CALLBACK; 244 } 245 break; 246 case TRANSFER_CALLBACK: 247 if (cbf == NULL || sharedBuffer != 0) { 248 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 249 return BAD_VALUE; 250 } 251 break; 252 case TRANSFER_OBTAIN: 253 case TRANSFER_SYNC: 254 if (sharedBuffer != 0) { 255 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 256 return BAD_VALUE; 257 } 258 break; 259 case TRANSFER_SHARED: 260 if (sharedBuffer == 0) { 261 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 262 return BAD_VALUE; 263 } 264 break; 265 default: 266 ALOGE("Invalid transfer type %d", transferType); 267 return BAD_VALUE; 268 } 269 mSharedBuffer = sharedBuffer; 270 mTransfer = transferType; 271 272 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 273 sharedBuffer->size()); 274 275 ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags); 276 277 // invariant that mAudioTrack != 0 is true only after set() returns successfully 278 if (mAudioTrack != 0) { 279 ALOGE("Track already in use"); 280 return INVALID_OPERATION; 281 } 282 283 // handle default values first. 284 if (streamType == AUDIO_STREAM_DEFAULT) { 285 streamType = AUDIO_STREAM_MUSIC; 286 } 287 if (pAttributes == NULL) { 288 if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) { 289 ALOGE("Invalid stream type %d", streamType); 290 return BAD_VALUE; 291 } 292 mStreamType = streamType; 293 294 } else { 295 // stream type shouldn't be looked at, this track has audio attributes 296 memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t)); 297 ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]", 298 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags); 299 mStreamType = AUDIO_STREAM_DEFAULT; 300 if ((mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) { 301 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC); 302 } 303 } 304 305 // these below should probably come from the audioFlinger too... 306 if (format == AUDIO_FORMAT_DEFAULT) { 307 format = AUDIO_FORMAT_PCM_16_BIT; 308 } 309 310 // validate parameters 311 if (!audio_is_valid_format(format)) { 312 ALOGE("Invalid format %#x", format); 313 return BAD_VALUE; 314 } 315 mFormat = format; 316 317 if (!audio_is_output_channel(channelMask)) { 318 ALOGE("Invalid channel mask %#x", channelMask); 319 return BAD_VALUE; 320 } 321 mChannelMask = channelMask; 322 uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 323 mChannelCount = channelCount; 324 325 // force direct flag if format is not linear PCM 326 // or offload was requested 327 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 328 || !audio_is_linear_pcm(format)) { 329 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 330 ? "Offload request, forcing to Direct Output" 331 : "Not linear PCM, forcing to Direct Output"); 332 flags = (audio_output_flags_t) 333 // FIXME why can't we allow direct AND fast? 334 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 335 } 336 337 // force direct flag if HW A/V sync requested 338 if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) { 339 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT); 340 } 341 342 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { 343 if (audio_is_linear_pcm(format)) { 344 mFrameSize = channelCount * audio_bytes_per_sample(format); 345 } else { 346 mFrameSize = sizeof(uint8_t); 347 } 348 } else { 349 ALOG_ASSERT(audio_is_linear_pcm(format)); 350 mFrameSize = channelCount * audio_bytes_per_sample(format); 351 // createTrack will return an error if PCM format is not supported by server, 352 // so no need to check for specific PCM formats here 353 } 354 355 // sampling rate must be specified for direct outputs 356 if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) { 357 return BAD_VALUE; 358 } 359 mSampleRate = sampleRate; 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 if (sessionId == AUDIO_SESSION_ALLOCATE) { 379 mSessionId = AudioSystem::newAudioUniqueId(); 380 } else { 381 mSessionId = sessionId; 382 } 383 int callingpid = IPCThreadState::self()->getCallingPid(); 384 int mypid = getpid(); 385 if (uid == -1 || (callingpid != mypid)) { 386 mClientUid = IPCThreadState::self()->getCallingUid(); 387 } else { 388 mClientUid = uid; 389 } 390 if (pid == -1 || (callingpid != mypid)) { 391 mClientPid = callingpid; 392 } else { 393 mClientPid = pid; 394 } 395 mAuxEffectId = 0; 396 mFlags = flags; 397 mCbf = cbf; 398 399 if (cbf != NULL) { 400 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 401 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 402 // thread begins in paused state, and will not reference us until start() 403 } 404 405 // create the IAudioTrack 406 status_t status = createTrack_l(); 407 408 if (status != NO_ERROR) { 409 if (mAudioTrackThread != 0) { 410 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 411 mAudioTrackThread->requestExitAndWait(); 412 mAudioTrackThread.clear(); 413 } 414 return status; 415 } 416 417 mStatus = NO_ERROR; 418 mState = STATE_STOPPED; 419 mUserData = user; 420 mLoopCount = 0; 421 mLoopStart = 0; 422 mLoopEnd = 0; 423 mLoopCountNotified = 0; 424 mMarkerPosition = 0; 425 mMarkerReached = false; 426 mNewPosition = 0; 427 mUpdatePeriod = 0; 428 mServer = 0; 429 mPosition = 0; 430 mReleased = 0; 431 mStartUs = 0; 432 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 433 mSequence = 1; 434 mObservedSequence = mSequence; 435 mInUnderrun = false; 436 437 return NO_ERROR; 438} 439 440// ------------------------------------------------------------------------- 441 442status_t AudioTrack::start() 443{ 444 AutoMutex lock(mLock); 445 446 if (mState == STATE_ACTIVE) { 447 return INVALID_OPERATION; 448 } 449 450 mInUnderrun = true; 451 452 State previousState = mState; 453 if (previousState == STATE_PAUSED_STOPPING) { 454 mState = STATE_STOPPING; 455 } else { 456 mState = STATE_ACTIVE; 457 } 458 (void) updateAndGetPosition_l(); 459 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 460 // reset current position as seen by client to 0 461 mPosition = 0; 462 // For offloaded tracks, we don't know if the hardware counters are really zero here, 463 // since the flush is asynchronous and stop may not fully drain. 464 // We save the time when the track is started to later verify whether 465 // the counters are realistic (i.e. start from zero after this time). 466 mStartUs = getNowUs(); 467 468 // force refresh of remaining frames by processAudioBuffer() as last 469 // write before stop could be partial. 470 mRefreshRemaining = true; 471 } 472 mNewPosition = mPosition + mUpdatePeriod; 473 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 474 475 sp<AudioTrackThread> t = mAudioTrackThread; 476 if (t != 0) { 477 if (previousState == STATE_STOPPING) { 478 mProxy->interrupt(); 479 } else { 480 t->resume(); 481 } 482 } else { 483 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 484 get_sched_policy(0, &mPreviousSchedulingGroup); 485 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 486 } 487 488 status_t status = NO_ERROR; 489 if (!(flags & CBLK_INVALID)) { 490 status = mAudioTrack->start(); 491 if (status == DEAD_OBJECT) { 492 flags |= CBLK_INVALID; 493 } 494 } 495 if (flags & CBLK_INVALID) { 496 status = restoreTrack_l("start"); 497 } 498 499 if (status != NO_ERROR) { 500 ALOGE("start() status %d", status); 501 mState = previousState; 502 if (t != 0) { 503 if (previousState != STATE_STOPPING) { 504 t->pause(); 505 } 506 } else { 507 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 508 set_sched_policy(0, mPreviousSchedulingGroup); 509 } 510 } 511 512 return status; 513} 514 515void AudioTrack::stop() 516{ 517 AutoMutex lock(mLock); 518 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 519 return; 520 } 521 522 if (isOffloaded_l()) { 523 mState = STATE_STOPPING; 524 } else { 525 mState = STATE_STOPPED; 526 mReleased = 0; 527 } 528 529 mProxy->interrupt(); 530 mAudioTrack->stop(); 531 // the playback head position will reset to 0, so if a marker is set, we need 532 // to activate it again 533 mMarkerReached = false; 534 535 if (mSharedBuffer != 0) { 536 // clear buffer position and loop count. 537 mStaticProxy->setBufferPositionAndLoop(0 /* position */, 538 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */); 539 } 540 541 sp<AudioTrackThread> t = mAudioTrackThread; 542 if (t != 0) { 543 if (!isOffloaded_l()) { 544 t->pause(); 545 } 546 } else { 547 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 548 set_sched_policy(0, mPreviousSchedulingGroup); 549 } 550} 551 552bool AudioTrack::stopped() const 553{ 554 AutoMutex lock(mLock); 555 return mState != STATE_ACTIVE; 556} 557 558void AudioTrack::flush() 559{ 560 if (mSharedBuffer != 0) { 561 return; 562 } 563 AutoMutex lock(mLock); 564 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 565 return; 566 } 567 flush_l(); 568} 569 570void AudioTrack::flush_l() 571{ 572 ALOG_ASSERT(mState != STATE_ACTIVE); 573 574 // clear playback marker and periodic update counter 575 mMarkerPosition = 0; 576 mMarkerReached = false; 577 mUpdatePeriod = 0; 578 mRefreshRemaining = true; 579 580 mState = STATE_FLUSHED; 581 mReleased = 0; 582 if (isOffloaded_l()) { 583 mProxy->interrupt(); 584 } 585 mProxy->flush(); 586 mAudioTrack->flush(); 587} 588 589void AudioTrack::pause() 590{ 591 AutoMutex lock(mLock); 592 if (mState == STATE_ACTIVE) { 593 mState = STATE_PAUSED; 594 } else if (mState == STATE_STOPPING) { 595 mState = STATE_PAUSED_STOPPING; 596 } else { 597 return; 598 } 599 mProxy->interrupt(); 600 mAudioTrack->pause(); 601 602 if (isOffloaded_l()) { 603 if (mOutput != AUDIO_IO_HANDLE_NONE) { 604 // An offload output can be re-used between two audio tracks having 605 // the same configuration. A timestamp query for a paused track 606 // while the other is running would return an incorrect time. 607 // To fix this, cache the playback position on a pause() and return 608 // this time when requested until the track is resumed. 609 610 // OffloadThread sends HAL pause in its threadLoop. Time saved 611 // here can be slightly off. 612 613 // TODO: check return code for getRenderPosition. 614 615 uint32_t halFrames; 616 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); 617 ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); 618 } 619 } 620} 621 622status_t AudioTrack::setVolume(float left, float right) 623{ 624 // This duplicates a test by AudioTrack JNI, but that is not the only caller 625 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY || 626 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) { 627 return BAD_VALUE; 628 } 629 630 AutoMutex lock(mLock); 631 mVolume[AUDIO_INTERLEAVE_LEFT] = left; 632 mVolume[AUDIO_INTERLEAVE_RIGHT] = right; 633 634 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right))); 635 636 if (isOffloaded_l()) { 637 mAudioTrack->signal(); 638 } 639 return NO_ERROR; 640} 641 642status_t AudioTrack::setVolume(float volume) 643{ 644 return setVolume(volume, volume); 645} 646 647status_t AudioTrack::setAuxEffectSendLevel(float level) 648{ 649 // This duplicates a test by AudioTrack JNI, but that is not the only caller 650 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) { 651 return BAD_VALUE; 652 } 653 654 AutoMutex lock(mLock); 655 mSendLevel = level; 656 mProxy->setSendLevel(level); 657 658 return NO_ERROR; 659} 660 661void AudioTrack::getAuxEffectSendLevel(float* level) const 662{ 663 if (level != NULL) { 664 *level = mSendLevel; 665 } 666} 667 668status_t AudioTrack::setSampleRate(uint32_t rate) 669{ 670 AutoMutex lock(mLock); 671 if (rate == mSampleRate) { 672 return NO_ERROR; 673 } 674 if (mIsTimed || isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 675 return INVALID_OPERATION; 676 } 677 if (mOutput == AUDIO_IO_HANDLE_NONE) { 678 return NO_INIT; 679 } 680 // NOTE: it is theoretically possible, but highly unlikely, that a device change 681 // could mean a previously allowed sampling rate is no longer allowed. 682 uint32_t afSamplingRate; 683 if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) { 684 return NO_INIT; 685 } 686 if (rate == 0 || rate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 687 return BAD_VALUE; 688 } 689 690 mSampleRate = rate; 691 mProxy->setSampleRate(rate); 692 693 return NO_ERROR; 694} 695 696uint32_t AudioTrack::getSampleRate() const 697{ 698 if (mIsTimed) { 699 return 0; 700 } 701 702 AutoMutex lock(mLock); 703 704 // sample rate can be updated during playback by the offloaded decoder so we need to 705 // query the HAL and update if needed. 706// FIXME use Proxy return channel to update the rate from server and avoid polling here 707 if (isOffloadedOrDirect_l()) { 708 if (mOutput != AUDIO_IO_HANDLE_NONE) { 709 uint32_t sampleRate = 0; 710 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate); 711 if (status == NO_ERROR) { 712 mSampleRate = sampleRate; 713 } 714 } 715 } 716 return mSampleRate; 717} 718 719status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 720{ 721 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 722 return INVALID_OPERATION; 723 } 724 725 if (loopCount == 0) { 726 ; 727 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 728 loopEnd - loopStart >= MIN_LOOP) { 729 ; 730 } else { 731 return BAD_VALUE; 732 } 733 734 AutoMutex lock(mLock); 735 // See setPosition() regarding setting parameters such as loop points or position while active 736 if (mState == STATE_ACTIVE) { 737 return INVALID_OPERATION; 738 } 739 setLoop_l(loopStart, loopEnd, loopCount); 740 return NO_ERROR; 741} 742 743void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 744{ 745 // We do not update the periodic notification point. 746 // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 747 mLoopCount = loopCount; 748 mLoopEnd = loopEnd; 749 mLoopStart = loopStart; 750 mLoopCountNotified = loopCount; 751 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 752 753 // Waking the AudioTrackThread is not needed as this cannot be called when active. 754} 755 756status_t AudioTrack::setMarkerPosition(uint32_t marker) 757{ 758 // The only purpose of setting marker position is to get a callback 759 if (mCbf == NULL || isOffloadedOrDirect()) { 760 return INVALID_OPERATION; 761 } 762 763 AutoMutex lock(mLock); 764 mMarkerPosition = marker; 765 mMarkerReached = false; 766 767 sp<AudioTrackThread> t = mAudioTrackThread; 768 if (t != 0) { 769 t->wake(); 770 } 771 return NO_ERROR; 772} 773 774status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 775{ 776 if (isOffloadedOrDirect()) { 777 return INVALID_OPERATION; 778 } 779 if (marker == NULL) { 780 return BAD_VALUE; 781 } 782 783 AutoMutex lock(mLock); 784 *marker = mMarkerPosition; 785 786 return NO_ERROR; 787} 788 789status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 790{ 791 // The only purpose of setting position update period is to get a callback 792 if (mCbf == NULL || isOffloadedOrDirect()) { 793 return INVALID_OPERATION; 794 } 795 796 AutoMutex lock(mLock); 797 mNewPosition = updateAndGetPosition_l() + updatePeriod; 798 mUpdatePeriod = updatePeriod; 799 800 sp<AudioTrackThread> t = mAudioTrackThread; 801 if (t != 0) { 802 t->wake(); 803 } 804 return NO_ERROR; 805} 806 807status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 808{ 809 if (isOffloadedOrDirect()) { 810 return INVALID_OPERATION; 811 } 812 if (updatePeriod == NULL) { 813 return BAD_VALUE; 814 } 815 816 AutoMutex lock(mLock); 817 *updatePeriod = mUpdatePeriod; 818 819 return NO_ERROR; 820} 821 822status_t AudioTrack::setPosition(uint32_t position) 823{ 824 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 825 return INVALID_OPERATION; 826 } 827 if (position > mFrameCount) { 828 return BAD_VALUE; 829 } 830 831 AutoMutex lock(mLock); 832 // Currently we require that the player is inactive before setting parameters such as position 833 // or loop points. Otherwise, there could be a race condition: the application could read the 834 // current position, compute a new position or loop parameters, and then set that position or 835 // loop parameters but it would do the "wrong" thing since the position has continued to advance 836 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 837 // to specify how it wants to handle such scenarios. 838 if (mState == STATE_ACTIVE) { 839 return INVALID_OPERATION; 840 } 841 // After setting the position, use full update period before notification. 842 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 843 mStaticProxy->setBufferPosition(position); 844 845 // Waking the AudioTrackThread is not needed as this cannot be called when active. 846 return NO_ERROR; 847} 848 849status_t AudioTrack::getPosition(uint32_t *position) 850{ 851 if (position == NULL) { 852 return BAD_VALUE; 853 } 854 855 AutoMutex lock(mLock); 856 if (isOffloadedOrDirect_l()) { 857 uint32_t dspFrames = 0; 858 859 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) { 860 ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); 861 *position = mPausedPosition; 862 return NO_ERROR; 863 } 864 865 if (mOutput != AUDIO_IO_HANDLE_NONE) { 866 uint32_t halFrames; 867 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 868 } 869 // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED) 870 // due to hardware latency. We leave this behavior for now. 871 *position = dspFrames; 872 } else { 873 if (mCblk->mFlags & CBLK_INVALID) { 874 restoreTrack_l("getPosition"); 875 } 876 877 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 878 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 879 0 : updateAndGetPosition_l(); 880 } 881 return NO_ERROR; 882} 883 884status_t AudioTrack::getBufferPosition(uint32_t *position) 885{ 886 if (mSharedBuffer == 0 || mIsTimed) { 887 return INVALID_OPERATION; 888 } 889 if (position == NULL) { 890 return BAD_VALUE; 891 } 892 893 AutoMutex lock(mLock); 894 *position = mStaticProxy->getBufferPosition(); 895 return NO_ERROR; 896} 897 898status_t AudioTrack::reload() 899{ 900 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 901 return INVALID_OPERATION; 902 } 903 904 AutoMutex lock(mLock); 905 // See setPosition() regarding setting parameters such as loop points or position while active 906 if (mState == STATE_ACTIVE) { 907 return INVALID_OPERATION; 908 } 909 mNewPosition = mUpdatePeriod; 910 (void) updateAndGetPosition_l(); 911 mPosition = 0; 912#if 0 913 // The documentation is not clear on the behavior of reload() and the restoration 914 // of loop count. Historically we have not restored loop count, start, end, 915 // but it makes sense if one desires to repeat playing a particular sound. 916 if (mLoopCount != 0) { 917 mLoopCountNotified = mLoopCount; 918 mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount); 919 } 920#endif 921 mStaticProxy->setBufferPosition(0); 922 return NO_ERROR; 923} 924 925audio_io_handle_t AudioTrack::getOutput() const 926{ 927 AutoMutex lock(mLock); 928 return mOutput; 929} 930 931status_t AudioTrack::attachAuxEffect(int effectId) 932{ 933 AutoMutex lock(mLock); 934 status_t status = mAudioTrack->attachAuxEffect(effectId); 935 if (status == NO_ERROR) { 936 mAuxEffectId = effectId; 937 } 938 return status; 939} 940 941audio_stream_type_t AudioTrack::streamType() const 942{ 943 if (mStreamType == AUDIO_STREAM_DEFAULT) { 944 return audio_attributes_to_stream_type(&mAttributes); 945 } 946 return mStreamType; 947} 948 949// ------------------------------------------------------------------------- 950 951// must be called with mLock held 952status_t AudioTrack::createTrack_l() 953{ 954 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 955 if (audioFlinger == 0) { 956 ALOGE("Could not get audioflinger"); 957 return NO_INIT; 958 } 959 960 audio_io_handle_t output; 961 audio_stream_type_t streamType = mStreamType; 962 audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL; 963 status_t status = AudioSystem::getOutputForAttr(attr, &output, 964 (audio_session_t)mSessionId, &streamType, 965 mSampleRate, mFormat, mChannelMask, 966 mFlags, mOffloadInfo); 967 968 969 if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) { 970 ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u, format %#x," 971 " channel mask %#x, flags %#x", 972 mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags); 973 return BAD_VALUE; 974 } 975 { 976 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 977 // we must release it ourselves if anything goes wrong. 978 979 // Not all of these values are needed under all conditions, but it is easier to get them all 980 981 uint32_t afLatency; 982 status = AudioSystem::getLatency(output, &afLatency); 983 if (status != NO_ERROR) { 984 ALOGE("getLatency(%d) failed status %d", output, status); 985 goto release; 986 } 987 ALOGV("createTrack_l() output %d afLatency %u", output, afLatency); 988 989 size_t afFrameCount; 990 status = AudioSystem::getFrameCount(output, &afFrameCount); 991 if (status != NO_ERROR) { 992 ALOGE("getFrameCount(output=%d) status %d", output, status); 993 goto release; 994 } 995 996 uint32_t afSampleRate; 997 status = AudioSystem::getSamplingRate(output, &afSampleRate); 998 if (status != NO_ERROR) { 999 ALOGE("getSamplingRate(output=%d) status %d", output, status); 1000 goto release; 1001 } 1002 if (mSampleRate == 0) { 1003 mSampleRate = afSampleRate; 1004 } 1005 // Client decides whether the track is TIMED (see below), but can only express a preference 1006 // for FAST. Server will perform additional tests. 1007 if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( 1008 // either of these use cases: 1009 // use case 1: shared buffer 1010 (mSharedBuffer != 0) || 1011 // use case 2: callback transfer mode 1012 (mTransfer == TRANSFER_CALLBACK) || 1013 // use case 3: obtain/release mode 1014 (mTransfer == TRANSFER_OBTAIN)) && 1015 // matching sample rate 1016 (mSampleRate == afSampleRate))) { 1017 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client; transfer %d, track %u Hz, output %u Hz", 1018 mTransfer, mSampleRate, afSampleRate); 1019 // once denied, do not request again if IAudioTrack is re-created 1020 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1021 } 1022 1023 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 1024 // n = 1 fast track with single buffering; nBuffering is ignored 1025 // n = 2 fast track with double buffering 1026 // n = 2 normal track, (including those with sample rate conversion) 1027 // n >= 3 very high latency or very small notification interval (unused). 1028 const uint32_t nBuffering = 2; 1029 1030 mNotificationFramesAct = mNotificationFramesReq; 1031 1032 size_t frameCount = mReqFrameCount; 1033 if (!audio_is_linear_pcm(mFormat)) { 1034 1035 if (mSharedBuffer != 0) { 1036 // Same comment as below about ignoring frameCount parameter for set() 1037 frameCount = mSharedBuffer->size(); 1038 } else if (frameCount == 0) { 1039 frameCount = afFrameCount; 1040 } 1041 if (mNotificationFramesAct != frameCount) { 1042 mNotificationFramesAct = frameCount; 1043 } 1044 } else if (mSharedBuffer != 0) { 1045 // FIXME: Ensure client side memory buffers need 1046 // not have additional alignment beyond sample 1047 // (e.g. 16 bit stereo accessed as 32 bit frame). 1048 size_t alignment = audio_bytes_per_sample(mFormat); 1049 if (alignment & 1) { 1050 // for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java). 1051 alignment = 1; 1052 } 1053 if (mChannelCount > 1) { 1054 // More than 2 channels does not require stronger alignment than stereo 1055 alignment <<= 1; 1056 } 1057 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 1058 ALOGE("Invalid buffer alignment: address %p, channel count %u", 1059 mSharedBuffer->pointer(), mChannelCount); 1060 status = BAD_VALUE; 1061 goto release; 1062 } 1063 1064 // When initializing a shared buffer AudioTrack via constructors, 1065 // there's no frameCount parameter. 1066 // But when initializing a shared buffer AudioTrack via set(), 1067 // there _is_ a frameCount parameter. We silently ignore it. 1068 frameCount = mSharedBuffer->size() / mFrameSize; 1069 } else { 1070 // For fast and normal streaming tracks, 1071 // the frame count calculations and checks are done by server 1072 } 1073 1074 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 1075 if (mIsTimed) { 1076 trackFlags |= IAudioFlinger::TRACK_TIMED; 1077 } 1078 1079 pid_t tid = -1; 1080 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1081 trackFlags |= IAudioFlinger::TRACK_FAST; 1082 if (mAudioTrackThread != 0) { 1083 tid = mAudioTrackThread->getTid(); 1084 } 1085 } 1086 1087 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1088 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 1089 } 1090 1091 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1092 trackFlags |= IAudioFlinger::TRACK_DIRECT; 1093 } 1094 1095 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1096 // but we will still need the original value also 1097 int originalSessionId = mSessionId; 1098 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 1099 mSampleRate, 1100 mFormat, 1101 mChannelMask, 1102 &temp, 1103 &trackFlags, 1104 mSharedBuffer, 1105 output, 1106 tid, 1107 &mSessionId, 1108 mClientUid, 1109 &status); 1110 ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId, 1111 "session ID changed from %d to %d", originalSessionId, mSessionId); 1112 1113 if (status != NO_ERROR) { 1114 ALOGE("AudioFlinger could not create track, status: %d", status); 1115 goto release; 1116 } 1117 ALOG_ASSERT(track != 0); 1118 1119 // AudioFlinger now owns the reference to the I/O handle, 1120 // so we are no longer responsible for releasing it. 1121 1122 sp<IMemory> iMem = track->getCblk(); 1123 if (iMem == 0) { 1124 ALOGE("Could not get control block"); 1125 return NO_INIT; 1126 } 1127 void *iMemPointer = iMem->pointer(); 1128 if (iMemPointer == NULL) { 1129 ALOGE("Could not get control block pointer"); 1130 return NO_INIT; 1131 } 1132 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1133 if (mAudioTrack != 0) { 1134 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this); 1135 mDeathNotifier.clear(); 1136 } 1137 mAudioTrack = track; 1138 mCblkMemory = iMem; 1139 IPCThreadState::self()->flushCommands(); 1140 1141 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1142 mCblk = cblk; 1143 // note that temp is the (possibly revised) value of frameCount 1144 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1145 // In current design, AudioTrack client checks and ensures frame count validity before 1146 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1147 // for fast track as it uses a special method of assigning frame count. 1148 ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp); 1149 } 1150 frameCount = temp; 1151 1152 mAwaitBoost = false; 1153 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1154 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1155 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount); 1156 mAwaitBoost = true; 1157 } else { 1158 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount); 1159 // once denied, do not request again if IAudioTrack is re-created 1160 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1161 } 1162 } 1163 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1164 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1165 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1166 } else { 1167 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1168 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1169 // FIXME This is a warning, not an error, so don't return error status 1170 //return NO_INIT; 1171 } 1172 } 1173 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1174 if (trackFlags & IAudioFlinger::TRACK_DIRECT) { 1175 ALOGV("AUDIO_OUTPUT_FLAG_DIRECT successful"); 1176 } else { 1177 ALOGW("AUDIO_OUTPUT_FLAG_DIRECT denied by server"); 1178 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT); 1179 // FIXME This is a warning, not an error, so don't return error status 1180 //return NO_INIT; 1181 } 1182 } 1183 // Make sure that application is notified with sufficient margin before underrun 1184 if (mSharedBuffer == 0 && audio_is_linear_pcm(mFormat)) { 1185 // Theoretically double-buffering is not required for fast tracks, 1186 // due to tighter scheduling. But in practice, to accommodate kernels with 1187 // scheduling jitter, and apps with computation jitter, we use double-buffering 1188 // for fast tracks just like normal streaming tracks. 1189 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount / nBuffering) { 1190 mNotificationFramesAct = frameCount / nBuffering; 1191 } 1192 } 1193 1194 // We retain a copy of the I/O handle, but don't own the reference 1195 mOutput = output; 1196 mRefreshRemaining = true; 1197 1198 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1199 // is the value of pointer() for the shared buffer, otherwise buffers points 1200 // immediately after the control block. This address is for the mapping within client 1201 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1202 void* buffers; 1203 if (mSharedBuffer == 0) { 1204 buffers = cblk + 1; 1205 } else { 1206 buffers = mSharedBuffer->pointer(); 1207 if (buffers == NULL) { 1208 ALOGE("Could not get buffer pointer"); 1209 return NO_INIT; 1210 } 1211 } 1212 1213 mAudioTrack->attachAuxEffect(mAuxEffectId); 1214 // FIXME don't believe this lie 1215 mLatency = afLatency + (1000*frameCount) / mSampleRate; 1216 1217 mFrameCount = frameCount; 1218 // If IAudioTrack is re-created, don't let the requested frameCount 1219 // decrease. This can confuse clients that cache frameCount(). 1220 if (frameCount > mReqFrameCount) { 1221 mReqFrameCount = frameCount; 1222 } 1223 1224 // update proxy 1225 if (mSharedBuffer == 0) { 1226 mStaticProxy.clear(); 1227 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize); 1228 } else { 1229 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize); 1230 mProxy = mStaticProxy; 1231 } 1232 1233 mProxy->setVolumeLR(gain_minifloat_pack( 1234 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]), 1235 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT]))); 1236 1237 mProxy->setSendLevel(mSendLevel); 1238 mProxy->setSampleRate(mSampleRate); 1239 mProxy->setMinimum(mNotificationFramesAct); 1240 1241 mDeathNotifier = new DeathNotifier(this); 1242 IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this); 1243 1244 return NO_ERROR; 1245 } 1246 1247release: 1248 AudioSystem::releaseOutput(output, streamType, (audio_session_t)mSessionId); 1249 if (status == NO_ERROR) { 1250 status = NO_INIT; 1251 } 1252 return status; 1253} 1254 1255status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig) 1256{ 1257 if (audioBuffer == NULL) { 1258 if (nonContig != NULL) { 1259 *nonContig = 0; 1260 } 1261 return BAD_VALUE; 1262 } 1263 if (mTransfer != TRANSFER_OBTAIN) { 1264 audioBuffer->frameCount = 0; 1265 audioBuffer->size = 0; 1266 audioBuffer->raw = NULL; 1267 if (nonContig != NULL) { 1268 *nonContig = 0; 1269 } 1270 return INVALID_OPERATION; 1271 } 1272 1273 const struct timespec *requested; 1274 struct timespec timeout; 1275 if (waitCount == -1) { 1276 requested = &ClientProxy::kForever; 1277 } else if (waitCount == 0) { 1278 requested = &ClientProxy::kNonBlocking; 1279 } else if (waitCount > 0) { 1280 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1281 timeout.tv_sec = ms / 1000; 1282 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1283 requested = &timeout; 1284 } else { 1285 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1286 requested = NULL; 1287 } 1288 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig); 1289} 1290 1291status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1292 struct timespec *elapsed, size_t *nonContig) 1293{ 1294 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1295 uint32_t oldSequence = 0; 1296 uint32_t newSequence; 1297 1298 Proxy::Buffer buffer; 1299 status_t status = NO_ERROR; 1300 1301 static const int32_t kMaxTries = 5; 1302 int32_t tryCounter = kMaxTries; 1303 1304 do { 1305 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1306 // keep them from going away if another thread re-creates the track during obtainBuffer() 1307 sp<AudioTrackClientProxy> proxy; 1308 sp<IMemory> iMem; 1309 1310 { // start of lock scope 1311 AutoMutex lock(mLock); 1312 1313 newSequence = mSequence; 1314 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1315 if (status == DEAD_OBJECT) { 1316 // re-create track, unless someone else has already done so 1317 if (newSequence == oldSequence) { 1318 status = restoreTrack_l("obtainBuffer"); 1319 if (status != NO_ERROR) { 1320 buffer.mFrameCount = 0; 1321 buffer.mRaw = NULL; 1322 buffer.mNonContig = 0; 1323 break; 1324 } 1325 } 1326 } 1327 oldSequence = newSequence; 1328 1329 // Keep the extra references 1330 proxy = mProxy; 1331 iMem = mCblkMemory; 1332 1333 if (mState == STATE_STOPPING) { 1334 status = -EINTR; 1335 buffer.mFrameCount = 0; 1336 buffer.mRaw = NULL; 1337 buffer.mNonContig = 0; 1338 break; 1339 } 1340 1341 // Non-blocking if track is stopped or paused 1342 if (mState != STATE_ACTIVE) { 1343 requested = &ClientProxy::kNonBlocking; 1344 } 1345 1346 } // end of lock scope 1347 1348 buffer.mFrameCount = audioBuffer->frameCount; 1349 // FIXME starts the requested timeout and elapsed over from scratch 1350 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1351 1352 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1353 1354 audioBuffer->frameCount = buffer.mFrameCount; 1355 audioBuffer->size = buffer.mFrameCount * mFrameSize; 1356 audioBuffer->raw = buffer.mRaw; 1357 if (nonContig != NULL) { 1358 *nonContig = buffer.mNonContig; 1359 } 1360 return status; 1361} 1362 1363void AudioTrack::releaseBuffer(const Buffer* audioBuffer) 1364{ 1365 // FIXME add error checking on mode, by adding an internal version 1366 if (mTransfer == TRANSFER_SHARED) { 1367 return; 1368 } 1369 1370 size_t stepCount = audioBuffer->size / mFrameSize; 1371 if (stepCount == 0) { 1372 return; 1373 } 1374 1375 Proxy::Buffer buffer; 1376 buffer.mFrameCount = stepCount; 1377 buffer.mRaw = audioBuffer->raw; 1378 1379 AutoMutex lock(mLock); 1380 mReleased += stepCount; 1381 mInUnderrun = false; 1382 mProxy->releaseBuffer(&buffer); 1383 1384 // restart track if it was disabled by audioflinger due to previous underrun 1385 if (mState == STATE_ACTIVE) { 1386 audio_track_cblk_t* cblk = mCblk; 1387 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1388 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1389 // FIXME ignoring status 1390 mAudioTrack->start(); 1391 } 1392 } 1393} 1394 1395// ------------------------------------------------------------------------- 1396 1397ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1398{ 1399 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1400 return INVALID_OPERATION; 1401 } 1402 1403 if (isDirect()) { 1404 AutoMutex lock(mLock); 1405 int32_t flags = android_atomic_and( 1406 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), 1407 &mCblk->mFlags); 1408 if (flags & CBLK_INVALID) { 1409 return DEAD_OBJECT; 1410 } 1411 } 1412 1413 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1414 // Sanity-check: user is most-likely passing an error code, and it would 1415 // make the return value ambiguous (actualSize vs error). 1416 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1417 return BAD_VALUE; 1418 } 1419 1420 size_t written = 0; 1421 Buffer audioBuffer; 1422 1423 while (userSize >= mFrameSize) { 1424 audioBuffer.frameCount = userSize / mFrameSize; 1425 1426 status_t err = obtainBuffer(&audioBuffer, 1427 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1428 if (err < 0) { 1429 if (written > 0) { 1430 break; 1431 } 1432 return ssize_t(err); 1433 } 1434 1435 size_t toWrite = audioBuffer.size; 1436 memcpy(audioBuffer.i8, buffer, toWrite); 1437 buffer = ((const char *) buffer) + toWrite; 1438 userSize -= toWrite; 1439 written += toWrite; 1440 1441 releaseBuffer(&audioBuffer); 1442 } 1443 1444 return written; 1445} 1446 1447// ------------------------------------------------------------------------- 1448 1449TimedAudioTrack::TimedAudioTrack() { 1450 mIsTimed = true; 1451} 1452 1453status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1454{ 1455 AutoMutex lock(mLock); 1456 status_t result = UNKNOWN_ERROR; 1457 1458#if 1 1459 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1460 // while we are accessing the cblk 1461 sp<IAudioTrack> audioTrack = mAudioTrack; 1462 sp<IMemory> iMem = mCblkMemory; 1463#endif 1464 1465 // If the track is not invalid already, try to allocate a buffer. alloc 1466 // fails indicating that the server is dead, flag the track as invalid so 1467 // we can attempt to restore in just a bit. 1468 audio_track_cblk_t* cblk = mCblk; 1469 if (!(cblk->mFlags & CBLK_INVALID)) { 1470 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1471 if (result == DEAD_OBJECT) { 1472 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1473 } 1474 } 1475 1476 // If the track is invalid at this point, attempt to restore it. and try the 1477 // allocation one more time. 1478 if (cblk->mFlags & CBLK_INVALID) { 1479 result = restoreTrack_l("allocateTimedBuffer"); 1480 1481 if (result == NO_ERROR) { 1482 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1483 } 1484 } 1485 1486 return result; 1487} 1488 1489status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1490 int64_t pts) 1491{ 1492 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1493 { 1494 AutoMutex lock(mLock); 1495 audio_track_cblk_t* cblk = mCblk; 1496 // restart track if it was disabled by audioflinger due to previous underrun 1497 if (buffer->size() != 0 && status == NO_ERROR && 1498 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1499 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1500 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1501 // FIXME ignoring status 1502 mAudioTrack->start(); 1503 } 1504 } 1505 return status; 1506} 1507 1508status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1509 TargetTimeline target) 1510{ 1511 return mAudioTrack->setMediaTimeTransform(xform, target); 1512} 1513 1514// ------------------------------------------------------------------------- 1515 1516nsecs_t AudioTrack::processAudioBuffer() 1517{ 1518 // Currently the AudioTrack thread is not created if there are no callbacks. 1519 // Would it ever make sense to run the thread, even without callbacks? 1520 // If so, then replace this by checks at each use for mCbf != NULL. 1521 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1522 1523 mLock.lock(); 1524 if (mAwaitBoost) { 1525 mAwaitBoost = false; 1526 mLock.unlock(); 1527 static const int32_t kMaxTries = 5; 1528 int32_t tryCounter = kMaxTries; 1529 uint32_t pollUs = 10000; 1530 do { 1531 int policy = sched_getscheduler(0); 1532 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1533 break; 1534 } 1535 usleep(pollUs); 1536 pollUs <<= 1; 1537 } while (tryCounter-- > 0); 1538 if (tryCounter < 0) { 1539 ALOGE("did not receive expected priority boost on time"); 1540 } 1541 // Run again immediately 1542 return 0; 1543 } 1544 1545 // Can only reference mCblk while locked 1546 int32_t flags = android_atomic_and( 1547 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1548 1549 // Check for track invalidation 1550 if (flags & CBLK_INVALID) { 1551 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1552 // AudioSystem cache. We should not exit here but after calling the callback so 1553 // that the upper layers can recreate the track 1554 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) { 1555 status_t status = restoreTrack_l("processAudioBuffer"); 1556 // after restoration, continue below to make sure that the loop and buffer events 1557 // are notified because they have been cleared from mCblk->mFlags above. 1558 } 1559 } 1560 1561 bool waitStreamEnd = mState == STATE_STOPPING; 1562 bool active = mState == STATE_ACTIVE; 1563 1564 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1565 bool newUnderrun = false; 1566 if (flags & CBLK_UNDERRUN) { 1567#if 0 1568 // Currently in shared buffer mode, when the server reaches the end of buffer, 1569 // the track stays active in continuous underrun state. It's up to the application 1570 // to pause or stop the track, or set the position to a new offset within buffer. 1571 // This was some experimental code to auto-pause on underrun. Keeping it here 1572 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1573 if (mTransfer == TRANSFER_SHARED) { 1574 mState = STATE_PAUSED; 1575 active = false; 1576 } 1577#endif 1578 if (!mInUnderrun) { 1579 mInUnderrun = true; 1580 newUnderrun = true; 1581 } 1582 } 1583 1584 // Get current position of server 1585 size_t position = updateAndGetPosition_l(); 1586 1587 // Manage marker callback 1588 bool markerReached = false; 1589 size_t markerPosition = mMarkerPosition; 1590 // FIXME fails for wraparound, need 64 bits 1591 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1592 mMarkerReached = markerReached = true; 1593 } 1594 1595 // Determine number of new position callback(s) that will be needed, while locked 1596 size_t newPosCount = 0; 1597 size_t newPosition = mNewPosition; 1598 size_t updatePeriod = mUpdatePeriod; 1599 // FIXME fails for wraparound, need 64 bits 1600 if (updatePeriod > 0 && position >= newPosition) { 1601 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1602 mNewPosition += updatePeriod * newPosCount; 1603 } 1604 1605 // Cache other fields that will be needed soon 1606 uint32_t sampleRate = mSampleRate; 1607 uint32_t notificationFrames = mNotificationFramesAct; 1608 if (mRefreshRemaining) { 1609 mRefreshRemaining = false; 1610 mRemainingFrames = notificationFrames; 1611 mRetryOnPartialBuffer = false; 1612 } 1613 size_t misalignment = mProxy->getMisalignment(); 1614 uint32_t sequence = mSequence; 1615 sp<AudioTrackClientProxy> proxy = mProxy; 1616 1617 // Determine the number of new loop callback(s) that will be needed, while locked. 1618 int loopCountNotifications = 0; 1619 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END 1620 1621 if (mLoopCount > 0) { 1622 int loopCount; 1623 size_t bufferPosition; 1624 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount); 1625 loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition; 1626 loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications); 1627 mLoopCountNotified = loopCount; // discard any excess notifications 1628 } else if (mLoopCount < 0) { 1629 // FIXME: We're not accurate with notification count and position with infinite looping 1630 // since loopCount from server side will always return -1 (we could decrement it). 1631 size_t bufferPosition = mStaticProxy->getBufferPosition(); 1632 loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0); 1633 loopPeriod = mLoopEnd - bufferPosition; 1634 } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) { 1635 size_t bufferPosition = mStaticProxy->getBufferPosition(); 1636 loopPeriod = mFrameCount - bufferPosition; 1637 } 1638 1639 // These fields don't need to be cached, because they are assigned only by set(): 1640 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFlags 1641 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1642 1643 mLock.unlock(); 1644 1645 if (waitStreamEnd) { 1646 struct timespec timeout; 1647 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1648 timeout.tv_nsec = 0; 1649 1650 status_t status = proxy->waitStreamEndDone(&timeout); 1651 switch (status) { 1652 case NO_ERROR: 1653 case DEAD_OBJECT: 1654 case TIMED_OUT: 1655 mCbf(EVENT_STREAM_END, mUserData, NULL); 1656 { 1657 AutoMutex lock(mLock); 1658 // The previously assigned value of waitStreamEnd is no longer valid, 1659 // since the mutex has been unlocked and either the callback handler 1660 // or another thread could have re-started the AudioTrack during that time. 1661 waitStreamEnd = mState == STATE_STOPPING; 1662 if (waitStreamEnd) { 1663 mState = STATE_STOPPED; 1664 mReleased = 0; 1665 } 1666 } 1667 if (waitStreamEnd && status != DEAD_OBJECT) { 1668 return NS_INACTIVE; 1669 } 1670 break; 1671 } 1672 return 0; 1673 } 1674 1675 // perform callbacks while unlocked 1676 if (newUnderrun) { 1677 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1678 } 1679 while (loopCountNotifications > 0) { 1680 mCbf(EVENT_LOOP_END, mUserData, NULL); 1681 --loopCountNotifications; 1682 } 1683 if (flags & CBLK_BUFFER_END) { 1684 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1685 } 1686 if (markerReached) { 1687 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1688 } 1689 while (newPosCount > 0) { 1690 size_t temp = newPosition; 1691 mCbf(EVENT_NEW_POS, mUserData, &temp); 1692 newPosition += updatePeriod; 1693 newPosCount--; 1694 } 1695 1696 if (mObservedSequence != sequence) { 1697 mObservedSequence = sequence; 1698 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1699 // for offloaded tracks, just wait for the upper layers to recreate the track 1700 if (isOffloadedOrDirect()) { 1701 return NS_INACTIVE; 1702 } 1703 } 1704 1705 // if inactive, then don't run me again until re-started 1706 if (!active) { 1707 return NS_INACTIVE; 1708 } 1709 1710 // Compute the estimated time until the next timed event (position, markers, loops) 1711 // FIXME only for non-compressed audio 1712 uint32_t minFrames = ~0; 1713 if (!markerReached && position < markerPosition) { 1714 minFrames = markerPosition - position; 1715 } 1716 if (loopPeriod > 0 && loopPeriod < minFrames) { 1717 // loopPeriod is already adjusted for actual position. 1718 minFrames = loopPeriod; 1719 } 1720 if (updatePeriod > 0) { 1721 minFrames = min(minFrames, uint32_t(newPosition - position)); 1722 } 1723 1724 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1725 static const uint32_t kPoll = 0; 1726 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1727 minFrames = kPoll * notificationFrames; 1728 } 1729 1730 // Convert frame units to time units 1731 nsecs_t ns = NS_WHENEVER; 1732 if (minFrames != (uint32_t) ~0) { 1733 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1734 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1735 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1736 } 1737 1738 // If not supplying data by EVENT_MORE_DATA, then we're done 1739 if (mTransfer != TRANSFER_CALLBACK) { 1740 return ns; 1741 } 1742 1743 struct timespec timeout; 1744 const struct timespec *requested = &ClientProxy::kForever; 1745 if (ns != NS_WHENEVER) { 1746 timeout.tv_sec = ns / 1000000000LL; 1747 timeout.tv_nsec = ns % 1000000000LL; 1748 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1749 requested = &timeout; 1750 } 1751 1752 while (mRemainingFrames > 0) { 1753 1754 Buffer audioBuffer; 1755 audioBuffer.frameCount = mRemainingFrames; 1756 size_t nonContig; 1757 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1758 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1759 "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount); 1760 requested = &ClientProxy::kNonBlocking; 1761 size_t avail = audioBuffer.frameCount + nonContig; 1762 ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d", 1763 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1764 if (err != NO_ERROR) { 1765 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1766 (isOffloaded() && (err == DEAD_OBJECT))) { 1767 return 0; 1768 } 1769 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1770 return NS_NEVER; 1771 } 1772 1773 if (mRetryOnPartialBuffer && !isOffloaded()) { 1774 mRetryOnPartialBuffer = false; 1775 if (avail < mRemainingFrames) { 1776 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1777 if (ns < 0 || myns < ns) { 1778 ns = myns; 1779 } 1780 return ns; 1781 } 1782 } 1783 1784 size_t reqSize = audioBuffer.size; 1785 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1786 size_t writtenSize = audioBuffer.size; 1787 1788 // Sanity check on returned size 1789 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1790 ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes", 1791 reqSize, ssize_t(writtenSize)); 1792 return NS_NEVER; 1793 } 1794 1795 if (writtenSize == 0) { 1796 // The callback is done filling buffers 1797 // Keep this thread going to handle timed events and 1798 // still try to get more data in intervals of WAIT_PERIOD_MS 1799 // but don't just loop and block the CPU, so wait 1800 return WAIT_PERIOD_MS * 1000000LL; 1801 } 1802 1803 size_t releasedFrames = writtenSize / mFrameSize; 1804 audioBuffer.frameCount = releasedFrames; 1805 mRemainingFrames -= releasedFrames; 1806 if (misalignment >= releasedFrames) { 1807 misalignment -= releasedFrames; 1808 } else { 1809 misalignment = 0; 1810 } 1811 1812 releaseBuffer(&audioBuffer); 1813 1814 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1815 // if callback doesn't like to accept the full chunk 1816 if (writtenSize < reqSize) { 1817 continue; 1818 } 1819 1820 // There could be enough non-contiguous frames available to satisfy the remaining request 1821 if (mRemainingFrames <= nonContig) { 1822 continue; 1823 } 1824 1825#if 0 1826 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1827 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1828 // that total to a sum == notificationFrames. 1829 if (0 < misalignment && misalignment <= mRemainingFrames) { 1830 mRemainingFrames = misalignment; 1831 return (mRemainingFrames * 1100000000LL) / sampleRate; 1832 } 1833#endif 1834 1835 } 1836 mRemainingFrames = notificationFrames; 1837 mRetryOnPartialBuffer = true; 1838 1839 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1840 return 0; 1841} 1842 1843status_t AudioTrack::restoreTrack_l(const char *from) 1844{ 1845 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1846 isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from); 1847 ++mSequence; 1848 1849 // refresh the audio configuration cache in this process to make sure we get new 1850 // output parameters and new IAudioFlinger in createTrack_l() 1851 AudioSystem::clearAudioConfigCache(); 1852 1853 if (isOffloadedOrDirect_l()) { 1854 // FIXME re-creation of offloaded tracks is not yet implemented 1855 return DEAD_OBJECT; 1856 } 1857 1858 // save the old static buffer position 1859 size_t bufferPosition = 0; 1860 int loopCount = 0; 1861 if (mStaticProxy != 0) { 1862 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount); 1863 } 1864 1865 // If a new IAudioTrack is successfully created, createTrack_l() will modify the 1866 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1867 // It will also delete the strong references on previous IAudioTrack and IMemory. 1868 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact. 1869 status_t result = createTrack_l(); 1870 1871 // take the frames that will be lost by track recreation into account in saved position 1872 // For streaming tracks, this is the amount we obtained from the user/client 1873 // (not the number actually consumed at the server - those are already lost). 1874 (void) updateAndGetPosition_l(); 1875 if (mStaticProxy == 0) { 1876 mPosition = mReleased; 1877 } 1878 1879 if (result == NO_ERROR) { 1880 // Continue playback from last known position and restore loop. 1881 if (mStaticProxy != 0) { 1882 if (loopCount != 0) { 1883 mStaticProxy->setBufferPositionAndLoop(bufferPosition, 1884 mLoopStart, mLoopEnd, loopCount); 1885 } else { 1886 mStaticProxy->setBufferPosition(bufferPosition); 1887 if (bufferPosition == mFrameCount) { 1888 ALOGD("restoring track at end of static buffer"); 1889 } 1890 } 1891 } 1892 if (mState == STATE_ACTIVE) { 1893 result = mAudioTrack->start(); 1894 } 1895 } 1896 if (result != NO_ERROR) { 1897 ALOGW("restoreTrack_l() failed status %d", result); 1898 mState = STATE_STOPPED; 1899 mReleased = 0; 1900 } 1901 1902 return result; 1903} 1904 1905uint32_t AudioTrack::updateAndGetPosition_l() 1906{ 1907 // This is the sole place to read server consumed frames 1908 uint32_t newServer = mProxy->getPosition(); 1909 int32_t delta = newServer - mServer; 1910 mServer = newServer; 1911 // TODO There is controversy about whether there can be "negative jitter" in server position. 1912 // This should be investigated further, and if possible, it should be addressed. 1913 // A more definite failure mode is infrequent polling by client. 1914 // One could call (void)getPosition_l() in releaseBuffer(), 1915 // so mReleased and mPosition are always lock-step as best possible. 1916 // That should ensure delta never goes negative for infrequent polling 1917 // unless the server has more than 2^31 frames in its buffer, 1918 // in which case the use of uint32_t for these counters has bigger issues. 1919 if (delta < 0) { 1920 ALOGE("detected illegal retrograde motion by the server: mServer advanced by %d", delta); 1921 delta = 0; 1922 } 1923 return mPosition += (uint32_t) delta; 1924} 1925 1926status_t AudioTrack::setParameters(const String8& keyValuePairs) 1927{ 1928 AutoMutex lock(mLock); 1929 return mAudioTrack->setParameters(keyValuePairs); 1930} 1931 1932status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1933{ 1934 AutoMutex lock(mLock); 1935 // FIXME not implemented for fast tracks; should use proxy and SSQ 1936 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1937 return INVALID_OPERATION; 1938 } 1939 1940 switch (mState) { 1941 case STATE_ACTIVE: 1942 case STATE_PAUSED: 1943 break; // handle below 1944 case STATE_FLUSHED: 1945 case STATE_STOPPED: 1946 return WOULD_BLOCK; 1947 case STATE_STOPPING: 1948 case STATE_PAUSED_STOPPING: 1949 if (!isOffloaded_l()) { 1950 return INVALID_OPERATION; 1951 } 1952 break; // offloaded tracks handled below 1953 default: 1954 LOG_ALWAYS_FATAL("Invalid mState in getTimestamp(): %d", mState); 1955 break; 1956 } 1957 1958 if (mCblk->mFlags & CBLK_INVALID) { 1959 restoreTrack_l("getTimestamp"); 1960 } 1961 1962 // The presented frame count must always lag behind the consumed frame count. 1963 // To avoid a race, read the presented frames first. This ensures that presented <= consumed. 1964 status_t status = mAudioTrack->getTimestamp(timestamp); 1965 if (status != NO_ERROR) { 1966 ALOGV_IF(status != WOULD_BLOCK, "getTimestamp error:%#x", status); 1967 return status; 1968 } 1969 if (isOffloadedOrDirect_l()) { 1970 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) { 1971 // use cached paused position in case another offloaded track is running. 1972 timestamp.mPosition = mPausedPosition; 1973 clock_gettime(CLOCK_MONOTONIC, ×tamp.mTime); 1974 return NO_ERROR; 1975 } 1976 1977 // Check whether a pending flush or stop has completed, as those commands may 1978 // be asynchronous or return near finish. 1979 if (mStartUs != 0 && mSampleRate != 0) { 1980 static const int kTimeJitterUs = 100000; // 100 ms 1981 static const int k1SecUs = 1000000; 1982 1983 const int64_t timeNow = getNowUs(); 1984 1985 if (timeNow < mStartUs + k1SecUs) { // within first second of starting 1986 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime); 1987 if (timestampTimeUs < mStartUs) { 1988 return WOULD_BLOCK; // stale timestamp time, occurs before start. 1989 } 1990 const int64_t deltaTimeUs = timestampTimeUs - mStartUs; 1991 const int64_t deltaPositionByUs = timestamp.mPosition * 1000000LL / mSampleRate; 1992 1993 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) { 1994 // Verify that the counter can't count faster than the sample rate 1995 // since the start time. If greater, then that means we have failed 1996 // to completely flush or stop the previous playing track. 1997 ALOGW("incomplete flush or stop:" 1998 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)", 1999 (long long)deltaTimeUs, (long long)deltaPositionByUs, 2000 timestamp.mPosition); 2001 return WOULD_BLOCK; 2002 } 2003 } 2004 mStartUs = 0; // no need to check again, start timestamp has either expired or unneeded. 2005 } 2006 } else { 2007 // Update the mapping between local consumed (mPosition) and server consumed (mServer) 2008 (void) updateAndGetPosition_l(); 2009 // Server consumed (mServer) and presented both use the same server time base, 2010 // and server consumed is always >= presented. 2011 // The delta between these represents the number of frames in the buffer pipeline. 2012 // If this delta between these is greater than the client position, it means that 2013 // actually presented is still stuck at the starting line (figuratively speaking), 2014 // waiting for the first frame to go by. So we can't report a valid timestamp yet. 2015 if ((uint32_t) (mServer - timestamp.mPosition) > mPosition) { 2016 return INVALID_OPERATION; 2017 } 2018 // Convert timestamp position from server time base to client time base. 2019 // TODO The following code should work OK now because timestamp.mPosition is 32-bit. 2020 // But if we change it to 64-bit then this could fail. 2021 // If (mPosition - mServer) can be negative then should use: 2022 // (int32_t)(mPosition - mServer) 2023 timestamp.mPosition += mPosition - mServer; 2024 // Immediately after a call to getPosition_l(), mPosition and 2025 // mServer both represent the same frame position. mPosition is 2026 // in client's point of view, and mServer is in server's point of 2027 // view. So the difference between them is the "fudge factor" 2028 // between client and server views due to stop() and/or new 2029 // IAudioTrack. And timestamp.mPosition is initially in server's 2030 // point of view, so we need to apply the same fudge factor to it. 2031 } 2032 return status; 2033} 2034 2035String8 AudioTrack::getParameters(const String8& keys) 2036{ 2037 audio_io_handle_t output = getOutput(); 2038 if (output != AUDIO_IO_HANDLE_NONE) { 2039 return AudioSystem::getParameters(output, keys); 2040 } else { 2041 return String8::empty(); 2042 } 2043} 2044 2045bool AudioTrack::isOffloaded() const 2046{ 2047 AutoMutex lock(mLock); 2048 return isOffloaded_l(); 2049} 2050 2051bool AudioTrack::isDirect() const 2052{ 2053 AutoMutex lock(mLock); 2054 return isDirect_l(); 2055} 2056 2057bool AudioTrack::isOffloadedOrDirect() const 2058{ 2059 AutoMutex lock(mLock); 2060 return isOffloadedOrDirect_l(); 2061} 2062 2063 2064status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 2065{ 2066 2067 const size_t SIZE = 256; 2068 char buffer[SIZE]; 2069 String8 result; 2070 2071 result.append(" AudioTrack::dump\n"); 2072 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 2073 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 2074 result.append(buffer); 2075 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 2076 mChannelCount, mFrameCount); 2077 result.append(buffer); 2078 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 2079 result.append(buffer); 2080 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 2081 result.append(buffer); 2082 ::write(fd, result.string(), result.size()); 2083 return NO_ERROR; 2084} 2085 2086uint32_t AudioTrack::getUnderrunFrames() const 2087{ 2088 AutoMutex lock(mLock); 2089 return mProxy->getUnderrunFrames(); 2090} 2091 2092// ========================================================================= 2093 2094void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 2095{ 2096 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 2097 if (audioTrack != 0) { 2098 AutoMutex lock(audioTrack->mLock); 2099 audioTrack->mProxy->binderDied(); 2100 } 2101} 2102 2103// ========================================================================= 2104 2105AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 2106 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 2107 mIgnoreNextPausedInt(false) 2108{ 2109} 2110 2111AudioTrack::AudioTrackThread::~AudioTrackThread() 2112{ 2113} 2114 2115bool AudioTrack::AudioTrackThread::threadLoop() 2116{ 2117 { 2118 AutoMutex _l(mMyLock); 2119 if (mPaused) { 2120 mMyCond.wait(mMyLock); 2121 // caller will check for exitPending() 2122 return true; 2123 } 2124 if (mIgnoreNextPausedInt) { 2125 mIgnoreNextPausedInt = false; 2126 mPausedInt = false; 2127 } 2128 if (mPausedInt) { 2129 if (mPausedNs > 0) { 2130 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 2131 } else { 2132 mMyCond.wait(mMyLock); 2133 } 2134 mPausedInt = false; 2135 return true; 2136 } 2137 } 2138 if (exitPending()) { 2139 return false; 2140 } 2141 nsecs_t ns = mReceiver.processAudioBuffer(); 2142 switch (ns) { 2143 case 0: 2144 return true; 2145 case NS_INACTIVE: 2146 pauseInternal(); 2147 return true; 2148 case NS_NEVER: 2149 return false; 2150 case NS_WHENEVER: 2151 // Event driven: call wake() when callback notifications conditions change. 2152 ns = INT64_MAX; 2153 // fall through 2154 default: 2155 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns); 2156 pauseInternal(ns); 2157 return true; 2158 } 2159} 2160 2161void AudioTrack::AudioTrackThread::requestExit() 2162{ 2163 // must be in this order to avoid a race condition 2164 Thread::requestExit(); 2165 resume(); 2166} 2167 2168void AudioTrack::AudioTrackThread::pause() 2169{ 2170 AutoMutex _l(mMyLock); 2171 mPaused = true; 2172} 2173 2174void AudioTrack::AudioTrackThread::resume() 2175{ 2176 AutoMutex _l(mMyLock); 2177 mIgnoreNextPausedInt = true; 2178 if (mPaused || mPausedInt) { 2179 mPaused = false; 2180 mPausedInt = false; 2181 mMyCond.signal(); 2182 } 2183} 2184 2185void AudioTrack::AudioTrackThread::wake() 2186{ 2187 AutoMutex _l(mMyLock); 2188 if (!mPaused && mPausedInt && mPausedNs > 0) { 2189 // audio track is active and internally paused with timeout. 2190 mIgnoreNextPausedInt = true; 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