AudioTrack.cpp revision d2d089fc86c62843992e7d5b371ee9227189a1e6
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 40static int64_t convertTimespecToUs(const struct timespec &tv) 41{ 42 return tv.tv_sec * 1000000ll + tv.tv_nsec / 1000; 43} 44 45// current monotonic time in microseconds. 46static int64_t getNowUs() 47{ 48 struct timespec tv; 49 (void) clock_gettime(CLOCK_MONOTONIC, &tv); 50 return convertTimespecToUs(tv); 51} 52 53// static 54status_t AudioTrack::getMinFrameCount( 55 size_t* frameCount, 56 audio_stream_type_t streamType, 57 uint32_t sampleRate) 58{ 59 if (frameCount == NULL) { 60 return BAD_VALUE; 61 } 62 63 // FIXME merge with similar code in createTrack_l(), except we're missing 64 // some information here that is available in createTrack_l(): 65 // audio_io_handle_t output 66 // audio_format_t format 67 // audio_channel_mask_t channelMask 68 // audio_output_flags_t flags 69 uint32_t afSampleRate; 70 status_t status; 71 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 72 if (status != NO_ERROR) { 73 ALOGE("Unable to query output sample rate for stream type %d; status %d", 74 streamType, status); 75 return status; 76 } 77 size_t afFrameCount; 78 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 79 if (status != NO_ERROR) { 80 ALOGE("Unable to query output frame count for stream type %d; status %d", 81 streamType, status); 82 return status; 83 } 84 uint32_t afLatency; 85 status = AudioSystem::getOutputLatency(&afLatency, streamType); 86 if (status != NO_ERROR) { 87 ALOGE("Unable to query output latency for stream type %d; status %d", 88 streamType, status); 89 return status; 90 } 91 92 // Ensure that buffer depth covers at least audio hardware latency 93 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 94 if (minBufCount < 2) { 95 minBufCount = 2; 96 } 97 98 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 99 afFrameCount * minBufCount * uint64_t(sampleRate) / afSampleRate; 100 // The formula above should always produce a non-zero value, but return an error 101 // in the unlikely event that it does not, as that's part of the API contract. 102 if (*frameCount == 0) { 103 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", 104 streamType, sampleRate); 105 return BAD_VALUE; 106 } 107 ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%d, afSampleRate=%d, afLatency=%d", 108 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 109 return NO_ERROR; 110} 111 112// --------------------------------------------------------------------------- 113 114AudioTrack::AudioTrack() 115 : mStatus(NO_INIT), 116 mIsTimed(false), 117 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 118 mPreviousSchedulingGroup(SP_DEFAULT), 119 mPausedPosition(0) 120{ 121 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN; 122 mAttributes.usage = AUDIO_USAGE_UNKNOWN; 123 mAttributes.flags = 0x0; 124 strcpy(mAttributes.tags, ""); 125} 126 127AudioTrack::AudioTrack( 128 audio_stream_type_t streamType, 129 uint32_t sampleRate, 130 audio_format_t format, 131 audio_channel_mask_t channelMask, 132 size_t frameCount, 133 audio_output_flags_t flags, 134 callback_t cbf, 135 void* user, 136 uint32_t notificationFrames, 137 int sessionId, 138 transfer_type transferType, 139 const audio_offload_info_t *offloadInfo, 140 int uid, 141 pid_t pid, 142 const audio_attributes_t* pAttributes) 143 : mStatus(NO_INIT), 144 mIsTimed(false), 145 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 146 mPreviousSchedulingGroup(SP_DEFAULT), 147 mPausedPosition(0) 148{ 149 mStatus = set(streamType, sampleRate, format, channelMask, 150 frameCount, flags, cbf, user, notificationFrames, 151 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 152 offloadInfo, uid, pid, pAttributes); 153} 154 155AudioTrack::AudioTrack( 156 audio_stream_type_t streamType, 157 uint32_t sampleRate, 158 audio_format_t format, 159 audio_channel_mask_t channelMask, 160 const sp<IMemory>& sharedBuffer, 161 audio_output_flags_t flags, 162 callback_t cbf, 163 void* user, 164 uint32_t notificationFrames, 165 int sessionId, 166 transfer_type transferType, 167 const audio_offload_info_t *offloadInfo, 168 int uid, 169 pid_t pid, 170 const audio_attributes_t* pAttributes) 171 : mStatus(NO_INIT), 172 mIsTimed(false), 173 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 174 mPreviousSchedulingGroup(SP_DEFAULT), 175 mPausedPosition(0) 176{ 177 mStatus = set(streamType, sampleRate, format, channelMask, 178 0 /*frameCount*/, flags, cbf, user, notificationFrames, 179 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, 180 uid, pid, pAttributes); 181} 182 183AudioTrack::~AudioTrack() 184{ 185 if (mStatus == NO_ERROR) { 186 // Make sure that callback function exits in the case where 187 // it is looping on buffer full condition in obtainBuffer(). 188 // Otherwise the callback thread will never exit. 189 stop(); 190 if (mAudioTrackThread != 0) { 191 mProxy->interrupt(); 192 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 193 mAudioTrackThread->requestExitAndWait(); 194 mAudioTrackThread.clear(); 195 } 196 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 197 mAudioTrack.clear(); 198 mCblkMemory.clear(); 199 mSharedBuffer.clear(); 200 IPCThreadState::self()->flushCommands(); 201 ALOGV("~AudioTrack, releasing session id from %d on behalf of %d", 202 IPCThreadState::self()->getCallingPid(), mClientPid); 203 AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); 204 } 205} 206 207status_t AudioTrack::set( 208 audio_stream_type_t streamType, 209 uint32_t sampleRate, 210 audio_format_t format, 211 audio_channel_mask_t channelMask, 212 size_t frameCount, 213 audio_output_flags_t flags, 214 callback_t cbf, 215 void* user, 216 uint32_t notificationFrames, 217 const sp<IMemory>& sharedBuffer, 218 bool threadCanCallJava, 219 int sessionId, 220 transfer_type transferType, 221 const audio_offload_info_t *offloadInfo, 222 int uid, 223 pid_t pid, 224 const audio_attributes_t* pAttributes) 225{ 226 ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 227 "flags #%x, notificationFrames %u, sessionId %d, transferType %d", 228 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, 229 sessionId, transferType); 230 231 switch (transferType) { 232 case TRANSFER_DEFAULT: 233 if (sharedBuffer != 0) { 234 transferType = TRANSFER_SHARED; 235 } else if (cbf == NULL || threadCanCallJava) { 236 transferType = TRANSFER_SYNC; 237 } else { 238 transferType = TRANSFER_CALLBACK; 239 } 240 break; 241 case TRANSFER_CALLBACK: 242 if (cbf == NULL || sharedBuffer != 0) { 243 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 244 return BAD_VALUE; 245 } 246 break; 247 case TRANSFER_OBTAIN: 248 case TRANSFER_SYNC: 249 if (sharedBuffer != 0) { 250 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 251 return BAD_VALUE; 252 } 253 break; 254 case TRANSFER_SHARED: 255 if (sharedBuffer == 0) { 256 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 257 return BAD_VALUE; 258 } 259 break; 260 default: 261 ALOGE("Invalid transfer type %d", transferType); 262 return BAD_VALUE; 263 } 264 mSharedBuffer = sharedBuffer; 265 mTransfer = transferType; 266 267 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 268 sharedBuffer->size()); 269 270 ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags); 271 272 AutoMutex lock(mLock); 273 274 // invariant that mAudioTrack != 0 is true only after set() returns successfully 275 if (mAudioTrack != 0) { 276 ALOGE("Track already in use"); 277 return INVALID_OPERATION; 278 } 279 280 // handle default values first. 281 // TODO once AudioPolicyManager fully supports audio_attributes_t, 282 // remove stream "text-to-speech" redirect 283 if ((streamType == AUDIO_STREAM_DEFAULT) || (streamType == AUDIO_STREAM_TTS)) { 284 streamType = AUDIO_STREAM_MUSIC; 285 } 286 287 if (pAttributes == NULL) { 288 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 289 ALOGE("Invalid stream type %d", streamType); 290 return BAD_VALUE; 291 } 292 setAttributesFromStreamType(streamType); 293 mStreamType = streamType; 294 } else { 295 if (!isValidAttributes(pAttributes)) { 296 ALOGE("Invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]", 297 pAttributes->usage, pAttributes->content_type, pAttributes->flags, 298 pAttributes->tags); 299 } 300 // stream type shouldn't be looked at, this track has audio attributes 301 memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t)); 302 setStreamTypeFromAttributes(mAttributes); 303 ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]", 304 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags); 305 } 306 307 status_t status; 308 if (sampleRate == 0) { 309 status = AudioSystem::getOutputSamplingRateForAttr(&sampleRate, &mAttributes); 310 if (status != NO_ERROR) { 311 ALOGE("Could not get output sample rate for stream type %d; status %d", 312 mStreamType, status); 313 return status; 314 } 315 } 316 mSampleRate = sampleRate; 317 318 // these below should probably come from the audioFlinger too... 319 if (format == AUDIO_FORMAT_DEFAULT) { 320 format = AUDIO_FORMAT_PCM_16_BIT; 321 } 322 323 // validate parameters 324 if (!audio_is_valid_format(format)) { 325 ALOGE("Invalid format %#x", format); 326 return BAD_VALUE; 327 } 328 mFormat = format; 329 330 if (!audio_is_output_channel(channelMask)) { 331 ALOGE("Invalid channel mask %#x", channelMask); 332 return BAD_VALUE; 333 } 334 mChannelMask = channelMask; 335 uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 336 mChannelCount = channelCount; 337 338 // AudioFlinger does not currently support 8-bit data in shared memory 339 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 340 ALOGE("8-bit data in shared memory is not supported"); 341 return BAD_VALUE; 342 } 343 344 // force direct flag if format is not linear PCM 345 // or offload was requested 346 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 347 || !audio_is_linear_pcm(format)) { 348 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 349 ? "Offload request, forcing to Direct Output" 350 : "Not linear PCM, forcing to Direct Output"); 351 flags = (audio_output_flags_t) 352 // FIXME why can't we allow direct AND fast? 353 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 354 } 355 // only allow deep buffering for music stream type 356 if (mStreamType != AUDIO_STREAM_MUSIC) { 357 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 358 } 359 360 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { 361 if (audio_is_linear_pcm(format)) { 362 mFrameSize = channelCount * audio_bytes_per_sample(format); 363 } else { 364 mFrameSize = sizeof(uint8_t); 365 } 366 mFrameSizeAF = mFrameSize; 367 } else { 368 ALOG_ASSERT(audio_is_linear_pcm(format)); 369 mFrameSize = channelCount * audio_bytes_per_sample(format); 370 mFrameSizeAF = channelCount * audio_bytes_per_sample( 371 format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format); 372 // createTrack will return an error if PCM format is not supported by server, 373 // so no need to check for specific PCM formats here 374 } 375 376 // Make copy of input parameter offloadInfo so that in the future: 377 // (a) createTrack_l doesn't need it as an input parameter 378 // (b) we can support re-creation of offloaded tracks 379 if (offloadInfo != NULL) { 380 mOffloadInfoCopy = *offloadInfo; 381 mOffloadInfo = &mOffloadInfoCopy; 382 } else { 383 mOffloadInfo = NULL; 384 } 385 386 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f; 387 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f; 388 mSendLevel = 0.0f; 389 // mFrameCount is initialized in createTrack_l 390 mReqFrameCount = frameCount; 391 mNotificationFramesReq = notificationFrames; 392 mNotificationFramesAct = 0; 393 mSessionId = sessionId; 394 int callingpid = IPCThreadState::self()->getCallingPid(); 395 int mypid = getpid(); 396 if (uid == -1 || (callingpid != mypid)) { 397 mClientUid = IPCThreadState::self()->getCallingUid(); 398 } else { 399 mClientUid = uid; 400 } 401 if (pid == -1 || (callingpid != mypid)) { 402 mClientPid = callingpid; 403 } else { 404 mClientPid = pid; 405 } 406 mAuxEffectId = 0; 407 mFlags = flags; 408 mCbf = cbf; 409 410 if (cbf != NULL) { 411 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 412 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 413 } 414 415 // create the IAudioTrack 416 status = createTrack_l(); 417 418 if (status != NO_ERROR) { 419 if (mAudioTrackThread != 0) { 420 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 421 mAudioTrackThread->requestExitAndWait(); 422 mAudioTrackThread.clear(); 423 } 424 return status; 425 } 426 427 mStatus = NO_ERROR; 428 mState = STATE_STOPPED; 429 mUserData = user; 430 mLoopPeriod = 0; 431 mMarkerPosition = 0; 432 mMarkerReached = false; 433 mNewPosition = 0; 434 mUpdatePeriod = 0; 435 mServer = 0; 436 mPosition = 0; 437 mReleased = 0; 438 mStartUs = 0; 439 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 440 mSequence = 1; 441 mObservedSequence = mSequence; 442 mInUnderrun = false; 443 444 return NO_ERROR; 445} 446 447// ------------------------------------------------------------------------- 448 449status_t AudioTrack::start() 450{ 451 AutoMutex lock(mLock); 452 453 if (mState == STATE_ACTIVE) { 454 return INVALID_OPERATION; 455 } 456 457 mInUnderrun = true; 458 459 State previousState = mState; 460 if (previousState == STATE_PAUSED_STOPPING) { 461 mState = STATE_STOPPING; 462 } else { 463 mState = STATE_ACTIVE; 464 } 465 (void) updateAndGetPosition_l(); 466 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 467 // reset current position as seen by client to 0 468 mPosition = 0; 469 // For offloaded tracks, we don't know if the hardware counters are really zero here, 470 // since the flush is asynchronous and stop may not fully drain. 471 // We save the time when the track is started to later verify whether 472 // the counters are realistic (i.e. start from zero after this time). 473 mStartUs = getNowUs(); 474 475 // force refresh of remaining frames by processAudioBuffer() as last 476 // write before stop could be partial. 477 mRefreshRemaining = true; 478 } 479 mNewPosition = mPosition + mUpdatePeriod; 480 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 481 482 sp<AudioTrackThread> t = mAudioTrackThread; 483 if (t != 0) { 484 if (previousState == STATE_STOPPING) { 485 mProxy->interrupt(); 486 } else { 487 t->resume(); 488 } 489 } else { 490 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 491 get_sched_policy(0, &mPreviousSchedulingGroup); 492 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 493 } 494 495 status_t status = NO_ERROR; 496 if (!(flags & CBLK_INVALID)) { 497 status = mAudioTrack->start(); 498 if (status == DEAD_OBJECT) { 499 flags |= CBLK_INVALID; 500 } 501 } 502 if (flags & CBLK_INVALID) { 503 status = restoreTrack_l("start"); 504 } 505 506 if (status != NO_ERROR) { 507 ALOGE("start() status %d", status); 508 mState = previousState; 509 if (t != 0) { 510 if (previousState != STATE_STOPPING) { 511 t->pause(); 512 } 513 } else { 514 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 515 set_sched_policy(0, mPreviousSchedulingGroup); 516 } 517 } 518 519 return status; 520} 521 522void AudioTrack::stop() 523{ 524 AutoMutex lock(mLock); 525 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 526 return; 527 } 528 529 if (isOffloaded_l()) { 530 mState = STATE_STOPPING; 531 } else { 532 mState = STATE_STOPPED; 533 mReleased = 0; 534 } 535 536 mProxy->interrupt(); 537 mAudioTrack->stop(); 538 // the playback head position will reset to 0, so if a marker is set, we need 539 // to activate it again 540 mMarkerReached = false; 541#if 0 542 // Force flush if a shared buffer is used otherwise audioflinger 543 // will not stop before end of buffer is reached. 544 // It may be needed to make sure that we stop playback, likely in case looping is on. 545 if (mSharedBuffer != 0) { 546 flush_l(); 547 } 548#endif 549 550 sp<AudioTrackThread> t = mAudioTrackThread; 551 if (t != 0) { 552 if (!isOffloaded_l()) { 553 t->pause(); 554 } 555 } else { 556 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 557 set_sched_policy(0, mPreviousSchedulingGroup); 558 } 559} 560 561bool AudioTrack::stopped() const 562{ 563 AutoMutex lock(mLock); 564 return mState != STATE_ACTIVE; 565} 566 567void AudioTrack::flush() 568{ 569 if (mSharedBuffer != 0) { 570 return; 571 } 572 AutoMutex lock(mLock); 573 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 574 return; 575 } 576 flush_l(); 577} 578 579void AudioTrack::flush_l() 580{ 581 ALOG_ASSERT(mState != STATE_ACTIVE); 582 583 // clear playback marker and periodic update counter 584 mMarkerPosition = 0; 585 mMarkerReached = false; 586 mUpdatePeriod = 0; 587 mRefreshRemaining = true; 588 589 mState = STATE_FLUSHED; 590 mReleased = 0; 591 if (isOffloaded_l()) { 592 mProxy->interrupt(); 593 } 594 mProxy->flush(); 595 mAudioTrack->flush(); 596} 597 598void AudioTrack::pause() 599{ 600 AutoMutex lock(mLock); 601 if (mState == STATE_ACTIVE) { 602 mState = STATE_PAUSED; 603 } else if (mState == STATE_STOPPING) { 604 mState = STATE_PAUSED_STOPPING; 605 } else { 606 return; 607 } 608 mProxy->interrupt(); 609 mAudioTrack->pause(); 610 611 if (isOffloaded_l()) { 612 if (mOutput != AUDIO_IO_HANDLE_NONE) { 613 // An offload output can be re-used between two audio tracks having 614 // the same configuration. A timestamp query for a paused track 615 // while the other is running would return an incorrect time. 616 // To fix this, cache the playback position on a pause() and return 617 // this time when requested until the track is resumed. 618 619 // OffloadThread sends HAL pause in its threadLoop. Time saved 620 // here can be slightly off. 621 622 // TODO: check return code for getRenderPosition. 623 624 uint32_t halFrames; 625 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); 626 ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); 627 } 628 } 629} 630 631status_t AudioTrack::setVolume(float left, float right) 632{ 633 // This duplicates a test by AudioTrack JNI, but that is not the only caller 634 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY || 635 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) { 636 return BAD_VALUE; 637 } 638 639 AutoMutex lock(mLock); 640 mVolume[AUDIO_INTERLEAVE_LEFT] = left; 641 mVolume[AUDIO_INTERLEAVE_RIGHT] = right; 642 643 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right))); 644 645 if (isOffloaded_l()) { 646 mAudioTrack->signal(); 647 } 648 return NO_ERROR; 649} 650 651status_t AudioTrack::setVolume(float volume) 652{ 653 return setVolume(volume, volume); 654} 655 656status_t AudioTrack::setAuxEffectSendLevel(float level) 657{ 658 // This duplicates a test by AudioTrack JNI, but that is not the only caller 659 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) { 660 return BAD_VALUE; 661 } 662 663 AutoMutex lock(mLock); 664 mSendLevel = level; 665 mProxy->setSendLevel(level); 666 667 return NO_ERROR; 668} 669 670void AudioTrack::getAuxEffectSendLevel(float* level) const 671{ 672 if (level != NULL) { 673 *level = mSendLevel; 674 } 675} 676 677status_t AudioTrack::setSampleRate(uint32_t rate) 678{ 679 if (mIsTimed || isOffloadedOrDirect()) { 680 return INVALID_OPERATION; 681 } 682 683 uint32_t afSamplingRate; 684 if (AudioSystem::getOutputSamplingRateForAttr(&afSamplingRate, &mAttributes) != NO_ERROR) { 685 return NO_INIT; 686 } 687 if (rate == 0 || rate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 688 return BAD_VALUE; 689 } 690 691 AutoMutex lock(mLock); 692 mSampleRate = rate; 693 mProxy->setSampleRate(rate); 694 695 return NO_ERROR; 696} 697 698uint32_t AudioTrack::getSampleRate() const 699{ 700 if (mIsTimed) { 701 return 0; 702 } 703 704 AutoMutex lock(mLock); 705 706 // sample rate can be updated during playback by the offloaded decoder so we need to 707 // query the HAL and update if needed. 708// FIXME use Proxy return channel to update the rate from server and avoid polling here 709 if (isOffloadedOrDirect_l()) { 710 if (mOutput != AUDIO_IO_HANDLE_NONE) { 711 uint32_t sampleRate = 0; 712 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate); 713 if (status == NO_ERROR) { 714 mSampleRate = sampleRate; 715 } 716 } 717 } 718 return mSampleRate; 719} 720 721status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 722{ 723 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 724 return INVALID_OPERATION; 725 } 726 727 if (loopCount == 0) { 728 ; 729 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 730 loopEnd - loopStart >= MIN_LOOP) { 731 ; 732 } else { 733 return BAD_VALUE; 734 } 735 736 AutoMutex lock(mLock); 737 // See setPosition() regarding setting parameters such as loop points or position while active 738 if (mState == STATE_ACTIVE) { 739 return INVALID_OPERATION; 740 } 741 setLoop_l(loopStart, loopEnd, loopCount); 742 return NO_ERROR; 743} 744 745void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 746{ 747 // FIXME If setting a loop also sets position to start of loop, then 748 // this is correct. Otherwise it should be removed. 749 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 750 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 751 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 752} 753 754status_t AudioTrack::setMarkerPosition(uint32_t marker) 755{ 756 // The only purpose of setting marker position is to get a callback 757 if (mCbf == NULL || isOffloadedOrDirect()) { 758 return INVALID_OPERATION; 759 } 760 761 AutoMutex lock(mLock); 762 mMarkerPosition = marker; 763 mMarkerReached = false; 764 765 return NO_ERROR; 766} 767 768status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 769{ 770 if (isOffloadedOrDirect()) { 771 return INVALID_OPERATION; 772 } 773 if (marker == NULL) { 774 return BAD_VALUE; 775 } 776 777 AutoMutex lock(mLock); 778 *marker = mMarkerPosition; 779 780 return NO_ERROR; 781} 782 783status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 784{ 785 // The only purpose of setting position update period is to get a callback 786 if (mCbf == NULL || isOffloadedOrDirect()) { 787 return INVALID_OPERATION; 788 } 789 790 AutoMutex lock(mLock); 791 mNewPosition = updateAndGetPosition_l() + updatePeriod; 792 mUpdatePeriod = updatePeriod; 793 794 return NO_ERROR; 795} 796 797status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 798{ 799 if (isOffloadedOrDirect()) { 800 return INVALID_OPERATION; 801 } 802 if (updatePeriod == NULL) { 803 return BAD_VALUE; 804 } 805 806 AutoMutex lock(mLock); 807 *updatePeriod = mUpdatePeriod; 808 809 return NO_ERROR; 810} 811 812status_t AudioTrack::setPosition(uint32_t position) 813{ 814 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 815 return INVALID_OPERATION; 816 } 817 if (position > mFrameCount) { 818 return BAD_VALUE; 819 } 820 821 AutoMutex lock(mLock); 822 // Currently we require that the player is inactive before setting parameters such as position 823 // or loop points. Otherwise, there could be a race condition: the application could read the 824 // current position, compute a new position or loop parameters, and then set that position or 825 // loop parameters but it would do the "wrong" thing since the position has continued to advance 826 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 827 // to specify how it wants to handle such scenarios. 828 if (mState == STATE_ACTIVE) { 829 return INVALID_OPERATION; 830 } 831 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 832 mLoopPeriod = 0; 833 // FIXME Check whether loops and setting position are incompatible in old code. 834 // If we use setLoop for both purposes we lose the capability to set the position while looping. 835 mStaticProxy->setLoop(position, mFrameCount, 0); 836 837 return NO_ERROR; 838} 839 840status_t AudioTrack::getPosition(uint32_t *position) 841{ 842 if (position == NULL) { 843 return BAD_VALUE; 844 } 845 846 AutoMutex lock(mLock); 847 if (isOffloadedOrDirect_l()) { 848 uint32_t dspFrames = 0; 849 850 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) { 851 ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); 852 *position = mPausedPosition; 853 return NO_ERROR; 854 } 855 856 if (mOutput != AUDIO_IO_HANDLE_NONE) { 857 uint32_t halFrames; 858 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 859 } 860 // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED) 861 // due to hardware latency. We leave this behavior for now. 862 *position = dspFrames; 863 } else { 864 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 865 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 866 0 : updateAndGetPosition_l(); 867 } 868 return NO_ERROR; 869} 870 871status_t AudioTrack::getBufferPosition(uint32_t *position) 872{ 873 if (mSharedBuffer == 0 || mIsTimed) { 874 return INVALID_OPERATION; 875 } 876 if (position == NULL) { 877 return BAD_VALUE; 878 } 879 880 AutoMutex lock(mLock); 881 *position = mStaticProxy->getBufferPosition(); 882 return NO_ERROR; 883} 884 885status_t AudioTrack::reload() 886{ 887 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 888 return INVALID_OPERATION; 889 } 890 891 AutoMutex lock(mLock); 892 // See setPosition() regarding setting parameters such as loop points or position while active 893 if (mState == STATE_ACTIVE) { 894 return INVALID_OPERATION; 895 } 896 mNewPosition = mUpdatePeriod; 897 mLoopPeriod = 0; 898 // FIXME The new code cannot reload while keeping a loop specified. 899 // Need to check how the old code handled this, and whether it's a significant change. 900 mStaticProxy->setLoop(0, mFrameCount, 0); 901 return NO_ERROR; 902} 903 904audio_io_handle_t AudioTrack::getOutput() const 905{ 906 AutoMutex lock(mLock); 907 return mOutput; 908} 909 910status_t AudioTrack::attachAuxEffect(int effectId) 911{ 912 AutoMutex lock(mLock); 913 status_t status = mAudioTrack->attachAuxEffect(effectId); 914 if (status == NO_ERROR) { 915 mAuxEffectId = effectId; 916 } 917 return status; 918} 919 920// ------------------------------------------------------------------------- 921 922// must be called with mLock held 923status_t AudioTrack::createTrack_l() 924{ 925 status_t status; 926 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 927 if (audioFlinger == 0) { 928 ALOGE("Could not get audioflinger"); 929 return NO_INIT; 930 } 931 932 audio_io_handle_t output = AudioSystem::getOutputForAttr(&mAttributes, mSampleRate, mFormat, 933 mChannelMask, mFlags, mOffloadInfo); 934 if (output == AUDIO_IO_HANDLE_NONE) { 935 ALOGE("Could not get audio output for stream type %d, usage %d, sample rate %u, format %#x," 936 " channel mask %#x, flags %#x", 937 mStreamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags); 938 return BAD_VALUE; 939 } 940 { 941 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 942 // we must release it ourselves if anything goes wrong. 943 944 // Not all of these values are needed under all conditions, but it is easier to get them all 945 946 uint32_t afLatency; 947 status = AudioSystem::getLatency(output, &afLatency); 948 if (status != NO_ERROR) { 949 ALOGE("getLatency(%d) failed status %d", output, status); 950 goto release; 951 } 952 953 size_t afFrameCount; 954 status = AudioSystem::getFrameCount(output, &afFrameCount); 955 if (status != NO_ERROR) { 956 ALOGE("getFrameCount(output=%d) status %d", output, status); 957 goto release; 958 } 959 960 uint32_t afSampleRate; 961 status = AudioSystem::getSamplingRate(output, &afSampleRate); 962 if (status != NO_ERROR) { 963 ALOGE("getSamplingRate(output=%d) status %d", output, status); 964 goto release; 965 } 966 967 // Client decides whether the track is TIMED (see below), but can only express a preference 968 // for FAST. Server will perform additional tests. 969 if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( 970 // either of these use cases: 971 // use case 1: shared buffer 972 (mSharedBuffer != 0) || 973 // use case 2: callback transfer mode 974 (mTransfer == TRANSFER_CALLBACK)) && 975 // matching sample rate 976 (mSampleRate == afSampleRate))) { 977 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 978 // once denied, do not request again if IAudioTrack is re-created 979 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 980 } 981 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 982 983 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 984 // n = 1 fast track with single buffering; nBuffering is ignored 985 // n = 2 fast track with double buffering 986 // n = 2 normal track, no sample rate conversion 987 // n = 3 normal track, with sample rate conversion 988 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 989 // n > 3 very high latency or very small notification interval; nBuffering is ignored 990 const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; 991 992 mNotificationFramesAct = mNotificationFramesReq; 993 994 size_t frameCount = mReqFrameCount; 995 if (!audio_is_linear_pcm(mFormat)) { 996 997 if (mSharedBuffer != 0) { 998 // Same comment as below about ignoring frameCount parameter for set() 999 frameCount = mSharedBuffer->size(); 1000 } else if (frameCount == 0) { 1001 frameCount = afFrameCount; 1002 } 1003 if (mNotificationFramesAct != frameCount) { 1004 mNotificationFramesAct = frameCount; 1005 } 1006 } else if (mSharedBuffer != 0) { 1007 1008 // Ensure that buffer alignment matches channel count 1009 // 8-bit data in shared memory is not currently supported by AudioFlinger 1010 size_t alignment = audio_bytes_per_sample( 1011 mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat); 1012 if (alignment & 1) { 1013 alignment = 1; 1014 } 1015 if (mChannelCount > 1) { 1016 // More than 2 channels does not require stronger alignment than stereo 1017 alignment <<= 1; 1018 } 1019 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 1020 ALOGE("Invalid buffer alignment: address %p, channel count %u", 1021 mSharedBuffer->pointer(), mChannelCount); 1022 status = BAD_VALUE; 1023 goto release; 1024 } 1025 1026 // When initializing a shared buffer AudioTrack via constructors, 1027 // there's no frameCount parameter. 1028 // But when initializing a shared buffer AudioTrack via set(), 1029 // there _is_ a frameCount parameter. We silently ignore it. 1030 frameCount = mSharedBuffer->size() / mFrameSizeAF; 1031 1032 } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 1033 1034 // FIXME move these calculations and associated checks to server 1035 1036 // Ensure that buffer depth covers at least audio hardware latency 1037 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 1038 ALOGV("afFrameCount=%zu, minBufCount=%d, afSampleRate=%u, afLatency=%d", 1039 afFrameCount, minBufCount, afSampleRate, afLatency); 1040 if (minBufCount <= nBuffering) { 1041 minBufCount = nBuffering; 1042 } 1043 1044 size_t minFrameCount = afFrameCount * minBufCount * uint64_t(mSampleRate) / afSampleRate; 1045 ALOGV("minFrameCount: %zu, afFrameCount=%zu, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 1046 ", afLatency=%d", 1047 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); 1048 1049 if (frameCount == 0) { 1050 frameCount = minFrameCount; 1051 } else if (frameCount < minFrameCount) { 1052 // not ALOGW because it happens all the time when playing key clicks over A2DP 1053 ALOGV("Minimum buffer size corrected from %zu to %zu", 1054 frameCount, minFrameCount); 1055 frameCount = minFrameCount; 1056 } 1057 // Make sure that application is notified with sufficient margin before underrun 1058 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1059 mNotificationFramesAct = frameCount/nBuffering; 1060 } 1061 1062 } else { 1063 // For fast tracks, the frame count calculations and checks are done by server 1064 } 1065 1066 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 1067 if (mIsTimed) { 1068 trackFlags |= IAudioFlinger::TRACK_TIMED; 1069 } 1070 1071 pid_t tid = -1; 1072 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1073 trackFlags |= IAudioFlinger::TRACK_FAST; 1074 if (mAudioTrackThread != 0) { 1075 tid = mAudioTrackThread->getTid(); 1076 } 1077 } 1078 1079 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1080 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 1081 } 1082 1083 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1084 trackFlags |= IAudioFlinger::TRACK_DIRECT; 1085 } 1086 1087 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1088 // but we will still need the original value also 1089 sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType, 1090 mSampleRate, 1091 // AudioFlinger only sees 16-bit PCM 1092 mFormat == AUDIO_FORMAT_PCM_8_BIT && 1093 !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT) ? 1094 AUDIO_FORMAT_PCM_16_BIT : mFormat, 1095 mChannelMask, 1096 &temp, 1097 &trackFlags, 1098 mSharedBuffer, 1099 output, 1100 tid, 1101 &mSessionId, 1102 mClientUid, 1103 &status); 1104 1105 if (status != NO_ERROR) { 1106 ALOGE("AudioFlinger could not create track, status: %d", status); 1107 goto release; 1108 } 1109 ALOG_ASSERT(track != 0); 1110 1111 // AudioFlinger now owns the reference to the I/O handle, 1112 // so we are no longer responsible for releasing it. 1113 1114 sp<IMemory> iMem = track->getCblk(); 1115 if (iMem == 0) { 1116 ALOGE("Could not get control block"); 1117 return NO_INIT; 1118 } 1119 void *iMemPointer = iMem->pointer(); 1120 if (iMemPointer == NULL) { 1121 ALOGE("Could not get control block pointer"); 1122 return NO_INIT; 1123 } 1124 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1125 if (mAudioTrack != 0) { 1126 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1127 mDeathNotifier.clear(); 1128 } 1129 mAudioTrack = track; 1130 mCblkMemory = iMem; 1131 IPCThreadState::self()->flushCommands(); 1132 1133 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1134 mCblk = cblk; 1135 // note that temp is the (possibly revised) value of frameCount 1136 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1137 // In current design, AudioTrack client checks and ensures frame count validity before 1138 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1139 // for fast track as it uses a special method of assigning frame count. 1140 ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp); 1141 } 1142 frameCount = temp; 1143 1144 mAwaitBoost = false; 1145 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1146 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1147 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount); 1148 mAwaitBoost = true; 1149 if (mSharedBuffer == 0) { 1150 // Theoretically double-buffering is not required for fast tracks, 1151 // due to tighter scheduling. But in practice, to accommodate kernels with 1152 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1153 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1154 mNotificationFramesAct = frameCount/nBuffering; 1155 } 1156 } 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 if (mSharedBuffer == 0) { 1162 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1163 mNotificationFramesAct = frameCount/nBuffering; 1164 } 1165 } 1166 } 1167 } 1168 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1169 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1170 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1171 } else { 1172 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1173 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1174 // FIXME This is a warning, not an error, so don't return error status 1175 //return NO_INIT; 1176 } 1177 } 1178 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1179 if (trackFlags & IAudioFlinger::TRACK_DIRECT) { 1180 ALOGV("AUDIO_OUTPUT_FLAG_DIRECT successful"); 1181 } else { 1182 ALOGW("AUDIO_OUTPUT_FLAG_DIRECT denied by server"); 1183 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT); 1184 // FIXME This is a warning, not an error, so don't return error status 1185 //return NO_INIT; 1186 } 1187 } 1188 1189 // We retain a copy of the I/O handle, but don't own the reference 1190 mOutput = output; 1191 mRefreshRemaining = true; 1192 1193 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1194 // is the value of pointer() for the shared buffer, otherwise buffers points 1195 // immediately after the control block. This address is for the mapping within client 1196 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1197 void* buffers; 1198 if (mSharedBuffer == 0) { 1199 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1200 } else { 1201 buffers = mSharedBuffer->pointer(); 1202 } 1203 1204 mAudioTrack->attachAuxEffect(mAuxEffectId); 1205 // FIXME don't believe this lie 1206 mLatency = afLatency + (1000*frameCount) / mSampleRate; 1207 1208 mFrameCount = frameCount; 1209 // If IAudioTrack is re-created, don't let the requested frameCount 1210 // decrease. This can confuse clients that cache frameCount(). 1211 if (frameCount > mReqFrameCount) { 1212 mReqFrameCount = frameCount; 1213 } 1214 1215 // update proxy 1216 if (mSharedBuffer == 0) { 1217 mStaticProxy.clear(); 1218 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1219 } else { 1220 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1221 mProxy = mStaticProxy; 1222 } 1223 mProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); 1224 mProxy->setSendLevel(mSendLevel); 1225 mProxy->setSampleRate(mSampleRate); 1226 mProxy->setMinimum(mNotificationFramesAct); 1227 1228 mDeathNotifier = new DeathNotifier(this); 1229 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1230 1231 return NO_ERROR; 1232 } 1233 1234release: 1235 AudioSystem::releaseOutput(output); 1236 if (status == NO_ERROR) { 1237 status = NO_INIT; 1238 } 1239 return status; 1240} 1241 1242status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1243{ 1244 if (audioBuffer == NULL) { 1245 return BAD_VALUE; 1246 } 1247 if (mTransfer != TRANSFER_OBTAIN) { 1248 audioBuffer->frameCount = 0; 1249 audioBuffer->size = 0; 1250 audioBuffer->raw = NULL; 1251 return INVALID_OPERATION; 1252 } 1253 1254 const struct timespec *requested; 1255 struct timespec timeout; 1256 if (waitCount == -1) { 1257 requested = &ClientProxy::kForever; 1258 } else if (waitCount == 0) { 1259 requested = &ClientProxy::kNonBlocking; 1260 } else if (waitCount > 0) { 1261 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1262 timeout.tv_sec = ms / 1000; 1263 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1264 requested = &timeout; 1265 } else { 1266 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1267 requested = NULL; 1268 } 1269 return obtainBuffer(audioBuffer, requested); 1270} 1271 1272status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1273 struct timespec *elapsed, size_t *nonContig) 1274{ 1275 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1276 uint32_t oldSequence = 0; 1277 uint32_t newSequence; 1278 1279 Proxy::Buffer buffer; 1280 status_t status = NO_ERROR; 1281 1282 static const int32_t kMaxTries = 5; 1283 int32_t tryCounter = kMaxTries; 1284 1285 do { 1286 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1287 // keep them from going away if another thread re-creates the track during obtainBuffer() 1288 sp<AudioTrackClientProxy> proxy; 1289 sp<IMemory> iMem; 1290 1291 { // start of lock scope 1292 AutoMutex lock(mLock); 1293 1294 newSequence = mSequence; 1295 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1296 if (status == DEAD_OBJECT) { 1297 // re-create track, unless someone else has already done so 1298 if (newSequence == oldSequence) { 1299 status = restoreTrack_l("obtainBuffer"); 1300 if (status != NO_ERROR) { 1301 buffer.mFrameCount = 0; 1302 buffer.mRaw = NULL; 1303 buffer.mNonContig = 0; 1304 break; 1305 } 1306 } 1307 } 1308 oldSequence = newSequence; 1309 1310 // Keep the extra references 1311 proxy = mProxy; 1312 iMem = mCblkMemory; 1313 1314 if (mState == STATE_STOPPING) { 1315 status = -EINTR; 1316 buffer.mFrameCount = 0; 1317 buffer.mRaw = NULL; 1318 buffer.mNonContig = 0; 1319 break; 1320 } 1321 1322 // Non-blocking if track is stopped or paused 1323 if (mState != STATE_ACTIVE) { 1324 requested = &ClientProxy::kNonBlocking; 1325 } 1326 1327 } // end of lock scope 1328 1329 buffer.mFrameCount = audioBuffer->frameCount; 1330 // FIXME starts the requested timeout and elapsed over from scratch 1331 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1332 1333 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1334 1335 audioBuffer->frameCount = buffer.mFrameCount; 1336 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1337 audioBuffer->raw = buffer.mRaw; 1338 if (nonContig != NULL) { 1339 *nonContig = buffer.mNonContig; 1340 } 1341 return status; 1342} 1343 1344void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1345{ 1346 if (mTransfer == TRANSFER_SHARED) { 1347 return; 1348 } 1349 1350 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1351 if (stepCount == 0) { 1352 return; 1353 } 1354 1355 Proxy::Buffer buffer; 1356 buffer.mFrameCount = stepCount; 1357 buffer.mRaw = audioBuffer->raw; 1358 1359 AutoMutex lock(mLock); 1360 mReleased += stepCount; 1361 mInUnderrun = false; 1362 mProxy->releaseBuffer(&buffer); 1363 1364 // restart track if it was disabled by audioflinger due to previous underrun 1365 if (mState == STATE_ACTIVE) { 1366 audio_track_cblk_t* cblk = mCblk; 1367 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1368 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1369 // FIXME ignoring status 1370 mAudioTrack->start(); 1371 } 1372 } 1373} 1374 1375// ------------------------------------------------------------------------- 1376 1377ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1378{ 1379 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1380 return INVALID_OPERATION; 1381 } 1382 1383 if (isDirect()) { 1384 AutoMutex lock(mLock); 1385 int32_t flags = android_atomic_and( 1386 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), 1387 &mCblk->mFlags); 1388 if (flags & CBLK_INVALID) { 1389 return DEAD_OBJECT; 1390 } 1391 } 1392 1393 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1394 // Sanity-check: user is most-likely passing an error code, and it would 1395 // make the return value ambiguous (actualSize vs error). 1396 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1397 return BAD_VALUE; 1398 } 1399 1400 size_t written = 0; 1401 Buffer audioBuffer; 1402 1403 while (userSize >= mFrameSize) { 1404 audioBuffer.frameCount = userSize / mFrameSize; 1405 1406 status_t err = obtainBuffer(&audioBuffer, 1407 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1408 if (err < 0) { 1409 if (written > 0) { 1410 break; 1411 } 1412 return ssize_t(err); 1413 } 1414 1415 size_t toWrite; 1416 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1417 // Divide capacity by 2 to take expansion into account 1418 toWrite = audioBuffer.size >> 1; 1419 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1420 } else { 1421 toWrite = audioBuffer.size; 1422 memcpy(audioBuffer.i8, buffer, toWrite); 1423 } 1424 buffer = ((const char *) buffer) + toWrite; 1425 userSize -= toWrite; 1426 written += toWrite; 1427 1428 releaseBuffer(&audioBuffer); 1429 } 1430 1431 return written; 1432} 1433 1434// ------------------------------------------------------------------------- 1435 1436TimedAudioTrack::TimedAudioTrack() { 1437 mIsTimed = true; 1438} 1439 1440status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1441{ 1442 AutoMutex lock(mLock); 1443 status_t result = UNKNOWN_ERROR; 1444 1445#if 1 1446 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1447 // while we are accessing the cblk 1448 sp<IAudioTrack> audioTrack = mAudioTrack; 1449 sp<IMemory> iMem = mCblkMemory; 1450#endif 1451 1452 // If the track is not invalid already, try to allocate a buffer. alloc 1453 // fails indicating that the server is dead, flag the track as invalid so 1454 // we can attempt to restore in just a bit. 1455 audio_track_cblk_t* cblk = mCblk; 1456 if (!(cblk->mFlags & CBLK_INVALID)) { 1457 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1458 if (result == DEAD_OBJECT) { 1459 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1460 } 1461 } 1462 1463 // If the track is invalid at this point, attempt to restore it. and try the 1464 // allocation one more time. 1465 if (cblk->mFlags & CBLK_INVALID) { 1466 result = restoreTrack_l("allocateTimedBuffer"); 1467 1468 if (result == NO_ERROR) { 1469 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1470 } 1471 } 1472 1473 return result; 1474} 1475 1476status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1477 int64_t pts) 1478{ 1479 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1480 { 1481 AutoMutex lock(mLock); 1482 audio_track_cblk_t* cblk = mCblk; 1483 // restart track if it was disabled by audioflinger due to previous underrun 1484 if (buffer->size() != 0 && status == NO_ERROR && 1485 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1486 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1487 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1488 // FIXME ignoring status 1489 mAudioTrack->start(); 1490 } 1491 } 1492 return status; 1493} 1494 1495status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1496 TargetTimeline target) 1497{ 1498 return mAudioTrack->setMediaTimeTransform(xform, target); 1499} 1500 1501// ------------------------------------------------------------------------- 1502 1503nsecs_t AudioTrack::processAudioBuffer() 1504{ 1505 // Currently the AudioTrack thread is not created if there are no callbacks. 1506 // Would it ever make sense to run the thread, even without callbacks? 1507 // If so, then replace this by checks at each use for mCbf != NULL. 1508 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1509 1510 mLock.lock(); 1511 if (mAwaitBoost) { 1512 mAwaitBoost = false; 1513 mLock.unlock(); 1514 static const int32_t kMaxTries = 5; 1515 int32_t tryCounter = kMaxTries; 1516 uint32_t pollUs = 10000; 1517 do { 1518 int policy = sched_getscheduler(0); 1519 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1520 break; 1521 } 1522 usleep(pollUs); 1523 pollUs <<= 1; 1524 } while (tryCounter-- > 0); 1525 if (tryCounter < 0) { 1526 ALOGE("did not receive expected priority boost on time"); 1527 } 1528 // Run again immediately 1529 return 0; 1530 } 1531 1532 // Can only reference mCblk while locked 1533 int32_t flags = android_atomic_and( 1534 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1535 1536 // Check for track invalidation 1537 if (flags & CBLK_INVALID) { 1538 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1539 // AudioSystem cache. We should not exit here but after calling the callback so 1540 // that the upper layers can recreate the track 1541 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) { 1542 status_t status = restoreTrack_l("processAudioBuffer"); 1543 mLock.unlock(); 1544 // Run again immediately, but with a new IAudioTrack 1545 return 0; 1546 } 1547 } 1548 1549 bool waitStreamEnd = mState == STATE_STOPPING; 1550 bool active = mState == STATE_ACTIVE; 1551 1552 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1553 bool newUnderrun = false; 1554 if (flags & CBLK_UNDERRUN) { 1555#if 0 1556 // Currently in shared buffer mode, when the server reaches the end of buffer, 1557 // the track stays active in continuous underrun state. It's up to the application 1558 // to pause or stop the track, or set the position to a new offset within buffer. 1559 // This was some experimental code to auto-pause on underrun. Keeping it here 1560 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1561 if (mTransfer == TRANSFER_SHARED) { 1562 mState = STATE_PAUSED; 1563 active = false; 1564 } 1565#endif 1566 if (!mInUnderrun) { 1567 mInUnderrun = true; 1568 newUnderrun = true; 1569 } 1570 } 1571 1572 // Get current position of server 1573 size_t position = updateAndGetPosition_l(); 1574 1575 // Manage marker callback 1576 bool markerReached = false; 1577 size_t markerPosition = mMarkerPosition; 1578 // FIXME fails for wraparound, need 64 bits 1579 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1580 mMarkerReached = markerReached = true; 1581 } 1582 1583 // Determine number of new position callback(s) that will be needed, while locked 1584 size_t newPosCount = 0; 1585 size_t newPosition = mNewPosition; 1586 size_t updatePeriod = mUpdatePeriod; 1587 // FIXME fails for wraparound, need 64 bits 1588 if (updatePeriod > 0 && position >= newPosition) { 1589 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1590 mNewPosition += updatePeriod * newPosCount; 1591 } 1592 1593 // Cache other fields that will be needed soon 1594 uint32_t loopPeriod = mLoopPeriod; 1595 uint32_t sampleRate = mSampleRate; 1596 uint32_t notificationFrames = mNotificationFramesAct; 1597 if (mRefreshRemaining) { 1598 mRefreshRemaining = false; 1599 mRemainingFrames = notificationFrames; 1600 mRetryOnPartialBuffer = false; 1601 } 1602 size_t misalignment = mProxy->getMisalignment(); 1603 uint32_t sequence = mSequence; 1604 sp<AudioTrackClientProxy> proxy = mProxy; 1605 1606 // These fields don't need to be cached, because they are assigned only by set(): 1607 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1608 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1609 1610 mLock.unlock(); 1611 1612 if (waitStreamEnd) { 1613 struct timespec timeout; 1614 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1615 timeout.tv_nsec = 0; 1616 1617 status_t status = proxy->waitStreamEndDone(&timeout); 1618 switch (status) { 1619 case NO_ERROR: 1620 case DEAD_OBJECT: 1621 case TIMED_OUT: 1622 mCbf(EVENT_STREAM_END, mUserData, NULL); 1623 { 1624 AutoMutex lock(mLock); 1625 // The previously assigned value of waitStreamEnd is no longer valid, 1626 // since the mutex has been unlocked and either the callback handler 1627 // or another thread could have re-started the AudioTrack during that time. 1628 waitStreamEnd = mState == STATE_STOPPING; 1629 if (waitStreamEnd) { 1630 mState = STATE_STOPPED; 1631 mReleased = 0; 1632 } 1633 } 1634 if (waitStreamEnd && status != DEAD_OBJECT) { 1635 return NS_INACTIVE; 1636 } 1637 break; 1638 } 1639 return 0; 1640 } 1641 1642 // perform callbacks while unlocked 1643 if (newUnderrun) { 1644 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1645 } 1646 // FIXME we will miss loops if loop cycle was signaled several times since last call 1647 // to processAudioBuffer() 1648 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1649 mCbf(EVENT_LOOP_END, mUserData, NULL); 1650 } 1651 if (flags & CBLK_BUFFER_END) { 1652 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1653 } 1654 if (markerReached) { 1655 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1656 } 1657 while (newPosCount > 0) { 1658 size_t temp = newPosition; 1659 mCbf(EVENT_NEW_POS, mUserData, &temp); 1660 newPosition += updatePeriod; 1661 newPosCount--; 1662 } 1663 1664 if (mObservedSequence != sequence) { 1665 mObservedSequence = sequence; 1666 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1667 // for offloaded tracks, just wait for the upper layers to recreate the track 1668 if (isOffloadedOrDirect()) { 1669 return NS_INACTIVE; 1670 } 1671 } 1672 1673 // if inactive, then don't run me again until re-started 1674 if (!active) { 1675 return NS_INACTIVE; 1676 } 1677 1678 // Compute the estimated time until the next timed event (position, markers, loops) 1679 // FIXME only for non-compressed audio 1680 uint32_t minFrames = ~0; 1681 if (!markerReached && position < markerPosition) { 1682 minFrames = markerPosition - position; 1683 } 1684 if (loopPeriod > 0 && loopPeriod < minFrames) { 1685 minFrames = loopPeriod; 1686 } 1687 if (updatePeriod > 0 && updatePeriod < minFrames) { 1688 minFrames = updatePeriod; 1689 } 1690 1691 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1692 static const uint32_t kPoll = 0; 1693 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1694 minFrames = kPoll * notificationFrames; 1695 } 1696 1697 // Convert frame units to time units 1698 nsecs_t ns = NS_WHENEVER; 1699 if (minFrames != (uint32_t) ~0) { 1700 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1701 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1702 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1703 } 1704 1705 // If not supplying data by EVENT_MORE_DATA, then we're done 1706 if (mTransfer != TRANSFER_CALLBACK) { 1707 return ns; 1708 } 1709 1710 struct timespec timeout; 1711 const struct timespec *requested = &ClientProxy::kForever; 1712 if (ns != NS_WHENEVER) { 1713 timeout.tv_sec = ns / 1000000000LL; 1714 timeout.tv_nsec = ns % 1000000000LL; 1715 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1716 requested = &timeout; 1717 } 1718 1719 while (mRemainingFrames > 0) { 1720 1721 Buffer audioBuffer; 1722 audioBuffer.frameCount = mRemainingFrames; 1723 size_t nonContig; 1724 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1725 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1726 "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount); 1727 requested = &ClientProxy::kNonBlocking; 1728 size_t avail = audioBuffer.frameCount + nonContig; 1729 ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d", 1730 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1731 if (err != NO_ERROR) { 1732 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1733 (isOffloaded() && (err == DEAD_OBJECT))) { 1734 return 0; 1735 } 1736 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1737 return NS_NEVER; 1738 } 1739 1740 if (mRetryOnPartialBuffer && !isOffloaded()) { 1741 mRetryOnPartialBuffer = false; 1742 if (avail < mRemainingFrames) { 1743 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1744 if (ns < 0 || myns < ns) { 1745 ns = myns; 1746 } 1747 return ns; 1748 } 1749 } 1750 1751 // Divide buffer size by 2 to take into account the expansion 1752 // due to 8 to 16 bit conversion: the callback must fill only half 1753 // of the destination buffer 1754 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1755 audioBuffer.size >>= 1; 1756 } 1757 1758 size_t reqSize = audioBuffer.size; 1759 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1760 size_t writtenSize = audioBuffer.size; 1761 1762 // Sanity check on returned size 1763 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1764 ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes", 1765 reqSize, ssize_t(writtenSize)); 1766 return NS_NEVER; 1767 } 1768 1769 if (writtenSize == 0) { 1770 // The callback is done filling buffers 1771 // Keep this thread going to handle timed events and 1772 // still try to get more data in intervals of WAIT_PERIOD_MS 1773 // but don't just loop and block the CPU, so wait 1774 return WAIT_PERIOD_MS * 1000000LL; 1775 } 1776 1777 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1778 // 8 to 16 bit conversion, note that source and destination are the same address 1779 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1780 audioBuffer.size <<= 1; 1781 } 1782 1783 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1784 audioBuffer.frameCount = releasedFrames; 1785 mRemainingFrames -= releasedFrames; 1786 if (misalignment >= releasedFrames) { 1787 misalignment -= releasedFrames; 1788 } else { 1789 misalignment = 0; 1790 } 1791 1792 releaseBuffer(&audioBuffer); 1793 1794 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1795 // if callback doesn't like to accept the full chunk 1796 if (writtenSize < reqSize) { 1797 continue; 1798 } 1799 1800 // There could be enough non-contiguous frames available to satisfy the remaining request 1801 if (mRemainingFrames <= nonContig) { 1802 continue; 1803 } 1804 1805#if 0 1806 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1807 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1808 // that total to a sum == notificationFrames. 1809 if (0 < misalignment && misalignment <= mRemainingFrames) { 1810 mRemainingFrames = misalignment; 1811 return (mRemainingFrames * 1100000000LL) / sampleRate; 1812 } 1813#endif 1814 1815 } 1816 mRemainingFrames = notificationFrames; 1817 mRetryOnPartialBuffer = true; 1818 1819 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1820 return 0; 1821} 1822 1823status_t AudioTrack::restoreTrack_l(const char *from) 1824{ 1825 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1826 isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from); 1827 ++mSequence; 1828 status_t result; 1829 1830 // refresh the audio configuration cache in this process to make sure we get new 1831 // output parameters and new IAudioFlinger in createTrack_l() 1832 AudioSystem::clearAudioConfigCache(); 1833 1834 if (isOffloadedOrDirect_l()) { 1835 // FIXME re-creation of offloaded tracks is not yet implemented 1836 return DEAD_OBJECT; 1837 } 1838 1839 // save the old static buffer position 1840 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1841 1842 // If a new IAudioTrack is successfully created, createTrack_l() will modify the 1843 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1844 // It will also delete the strong references on previous IAudioTrack and IMemory. 1845 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact. 1846 result = createTrack_l(); 1847 1848 // take the frames that will be lost by track recreation into account in saved position 1849 (void) updateAndGetPosition_l(); 1850 mPosition = mReleased; 1851 1852 if (result == NO_ERROR) { 1853 // continue playback from last known position, but 1854 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1855 if (mStaticProxy != NULL) { 1856 mLoopPeriod = 0; 1857 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1858 } 1859 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1860 // track destruction have been played? This is critical for SoundPool implementation 1861 // This must be broken, and needs to be tested/debugged. 1862#if 0 1863 // restore write index and set other indexes to reflect empty buffer status 1864 if (!strcmp(from, "start")) { 1865 // Make sure that a client relying on callback events indicating underrun or 1866 // the actual amount of audio frames played (e.g SoundPool) receives them. 1867 if (mSharedBuffer == 0) { 1868 // restart playback even if buffer is not completely filled. 1869 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1870 } 1871 } 1872#endif 1873 if (mState == STATE_ACTIVE) { 1874 result = mAudioTrack->start(); 1875 } 1876 } 1877 if (result != NO_ERROR) { 1878 ALOGW("restoreTrack_l() failed status %d", result); 1879 mState = STATE_STOPPED; 1880 mReleased = 0; 1881 } 1882 1883 return result; 1884} 1885 1886uint32_t AudioTrack::updateAndGetPosition_l() 1887{ 1888 // This is the sole place to read server consumed frames 1889 uint32_t newServer = mProxy->getPosition(); 1890 int32_t delta = newServer - mServer; 1891 mServer = newServer; 1892 // TODO There is controversy about whether there can be "negative jitter" in server position. 1893 // This should be investigated further, and if possible, it should be addressed. 1894 // A more definite failure mode is infrequent polling by client. 1895 // One could call (void)getPosition_l() in releaseBuffer(), 1896 // so mReleased and mPosition are always lock-step as best possible. 1897 // That should ensure delta never goes negative for infrequent polling 1898 // unless the server has more than 2^31 frames in its buffer, 1899 // in which case the use of uint32_t for these counters has bigger issues. 1900 if (delta < 0) { 1901 ALOGE("detected illegal retrograde motion by the server: mServer advanced by %d", delta); 1902 delta = 0; 1903 } 1904 return mPosition += (uint32_t) delta; 1905} 1906 1907status_t AudioTrack::setParameters(const String8& keyValuePairs) 1908{ 1909 AutoMutex lock(mLock); 1910 return mAudioTrack->setParameters(keyValuePairs); 1911} 1912 1913status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1914{ 1915 AutoMutex lock(mLock); 1916 // FIXME not implemented for fast tracks; should use proxy and SSQ 1917 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1918 return INVALID_OPERATION; 1919 } 1920 1921 switch (mState) { 1922 case STATE_ACTIVE: 1923 case STATE_PAUSED: 1924 break; // handle below 1925 case STATE_FLUSHED: 1926 case STATE_STOPPED: 1927 return WOULD_BLOCK; 1928 case STATE_STOPPING: 1929 case STATE_PAUSED_STOPPING: 1930 if (!isOffloaded_l()) { 1931 return INVALID_OPERATION; 1932 } 1933 break; // offloaded tracks handled below 1934 default: 1935 LOG_ALWAYS_FATAL("Invalid mState in getTimestamp(): %d", mState); 1936 break; 1937 } 1938 1939 // The presented frame count must always lag behind the consumed frame count. 1940 // To avoid a race, read the presented frames first. This ensures that presented <= consumed. 1941 status_t status = mAudioTrack->getTimestamp(timestamp); 1942 if (status != NO_ERROR) { 1943 ALOGV_IF(status != WOULD_BLOCK, "getTimestamp error:%#x", status); 1944 return status; 1945 } 1946 if (isOffloadedOrDirect_l()) { 1947 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) { 1948 // use cached paused position in case another offloaded track is running. 1949 timestamp.mPosition = mPausedPosition; 1950 clock_gettime(CLOCK_MONOTONIC, ×tamp.mTime); 1951 return NO_ERROR; 1952 } 1953 1954 // Check whether a pending flush or stop has completed, as those commands may 1955 // be asynchronous or return near finish. 1956 if (mStartUs != 0 && mSampleRate != 0) { 1957 static const int kTimeJitterUs = 100000; // 100 ms 1958 static const int k1SecUs = 1000000; 1959 1960 const int64_t timeNow = getNowUs(); 1961 1962 if (timeNow < mStartUs + k1SecUs) { // within first second of starting 1963 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime); 1964 if (timestampTimeUs < mStartUs) { 1965 return WOULD_BLOCK; // stale timestamp time, occurs before start. 1966 } 1967 const int64_t deltaTimeUs = timestampTimeUs - mStartUs; 1968 const int64_t deltaPositionByUs = timestamp.mPosition * 1000000LL / mSampleRate; 1969 1970 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) { 1971 // Verify that the counter can't count faster than the sample rate 1972 // since the start time. If greater, then that means we have failed 1973 // to completely flush or stop the previous playing track. 1974 ALOGW("incomplete flush or stop:" 1975 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)", 1976 (long long)deltaTimeUs, (long long)deltaPositionByUs, 1977 timestamp.mPosition); 1978 return WOULD_BLOCK; 1979 } 1980 } 1981 mStartUs = 0; // no need to check again, start timestamp has either expired or unneeded. 1982 } 1983 } else { 1984 // Update the mapping between local consumed (mPosition) and server consumed (mServer) 1985 (void) updateAndGetPosition_l(); 1986 // Server consumed (mServer) and presented both use the same server time base, 1987 // and server consumed is always >= presented. 1988 // The delta between these represents the number of frames in the buffer pipeline. 1989 // If this delta between these is greater than the client position, it means that 1990 // actually presented is still stuck at the starting line (figuratively speaking), 1991 // waiting for the first frame to go by. So we can't report a valid timestamp yet. 1992 if ((uint32_t) (mServer - timestamp.mPosition) > mPosition) { 1993 return INVALID_OPERATION; 1994 } 1995 // Convert timestamp position from server time base to client time base. 1996 // TODO The following code should work OK now because timestamp.mPosition is 32-bit. 1997 // But if we change it to 64-bit then this could fail. 1998 // If (mPosition - mServer) can be negative then should use: 1999 // (int32_t)(mPosition - mServer) 2000 timestamp.mPosition += mPosition - mServer; 2001 // Immediately after a call to getPosition_l(), mPosition and 2002 // mServer both represent the same frame position. mPosition is 2003 // in client's point of view, and mServer is in server's point of 2004 // view. So the difference between them is the "fudge factor" 2005 // between client and server views due to stop() and/or new 2006 // IAudioTrack. And timestamp.mPosition is initially in server's 2007 // point of view, so we need to apply the same fudge factor to it. 2008 } 2009 return status; 2010} 2011 2012String8 AudioTrack::getParameters(const String8& keys) 2013{ 2014 audio_io_handle_t output = getOutput(); 2015 if (output != AUDIO_IO_HANDLE_NONE) { 2016 return AudioSystem::getParameters(output, keys); 2017 } else { 2018 return String8::empty(); 2019 } 2020} 2021 2022bool AudioTrack::isOffloaded() const 2023{ 2024 AutoMutex lock(mLock); 2025 return isOffloaded_l(); 2026} 2027 2028bool AudioTrack::isDirect() const 2029{ 2030 AutoMutex lock(mLock); 2031 return isDirect_l(); 2032} 2033 2034bool AudioTrack::isOffloadedOrDirect() const 2035{ 2036 AutoMutex lock(mLock); 2037 return isOffloadedOrDirect_l(); 2038} 2039 2040 2041status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 2042{ 2043 2044 const size_t SIZE = 256; 2045 char buffer[SIZE]; 2046 String8 result; 2047 2048 result.append(" AudioTrack::dump\n"); 2049 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 2050 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 2051 result.append(buffer); 2052 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 2053 mChannelCount, mFrameCount); 2054 result.append(buffer); 2055 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 2056 result.append(buffer); 2057 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 2058 result.append(buffer); 2059 ::write(fd, result.string(), result.size()); 2060 return NO_ERROR; 2061} 2062 2063uint32_t AudioTrack::getUnderrunFrames() const 2064{ 2065 AutoMutex lock(mLock); 2066 return mProxy->getUnderrunFrames(); 2067} 2068 2069void AudioTrack::setAttributesFromStreamType(audio_stream_type_t streamType) { 2070 mAttributes.flags = 0x0; 2071 2072 switch(streamType) { 2073 case AUDIO_STREAM_DEFAULT: 2074 case AUDIO_STREAM_MUSIC: 2075 mAttributes.content_type = AUDIO_CONTENT_TYPE_MUSIC; 2076 mAttributes.usage = AUDIO_USAGE_MEDIA; 2077 break; 2078 case AUDIO_STREAM_VOICE_CALL: 2079 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 2080 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION; 2081 break; 2082 case AUDIO_STREAM_ENFORCED_AUDIBLE: 2083 mAttributes.flags |= AUDIO_FLAG_AUDIBILITY_ENFORCED; 2084 // intended fall through, attributes in common with STREAM_SYSTEM 2085 case AUDIO_STREAM_SYSTEM: 2086 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2087 mAttributes.usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION; 2088 break; 2089 case AUDIO_STREAM_RING: 2090 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2091 mAttributes.usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE; 2092 break; 2093 case AUDIO_STREAM_ALARM: 2094 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2095 mAttributes.usage = AUDIO_USAGE_ALARM; 2096 break; 2097 case AUDIO_STREAM_NOTIFICATION: 2098 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2099 mAttributes.usage = AUDIO_USAGE_NOTIFICATION; 2100 break; 2101 case AUDIO_STREAM_BLUETOOTH_SCO: 2102 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 2103 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION; 2104 mAttributes.flags |= AUDIO_FLAG_SCO; 2105 break; 2106 case AUDIO_STREAM_DTMF: 2107 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 2108 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING; 2109 break; 2110 case AUDIO_STREAM_TTS: 2111 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 2112 mAttributes.usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY; 2113 break; 2114 default: 2115 ALOGE("invalid stream type %d when converting to attributes", streamType); 2116 } 2117} 2118 2119void AudioTrack::setStreamTypeFromAttributes(audio_attributes_t& aa) { 2120 // flags to stream type mapping 2121 if ((aa.flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { 2122 mStreamType = AUDIO_STREAM_ENFORCED_AUDIBLE; 2123 return; 2124 } 2125 if ((aa.flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) { 2126 mStreamType = AUDIO_STREAM_BLUETOOTH_SCO; 2127 return; 2128 } 2129 // TODO once AudioPolicyManager fully supports audio_attributes_t, 2130 // remove stream remap, the flag will be enough 2131 if ((aa.flags & AUDIO_FLAG_BEACON) == AUDIO_FLAG_BEACON) { 2132 mStreamType = AUDIO_STREAM_TTS; 2133 return; 2134 } 2135 2136 // usage to stream type mapping 2137 switch (aa.usage) { 2138 case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: { 2139 // TODO once AudioPolicyManager fully supports audio_attributes_t, 2140 // remove stream change based on stream activity 2141 bool active; 2142 status_t status = AudioSystem::isStreamActive(AUDIO_STREAM_RING, &active, 0); 2143 if (status == NO_ERROR && active == true) { 2144 mStreamType = AUDIO_STREAM_RING; 2145 break; 2146 } 2147 status = AudioSystem::isStreamActive(AUDIO_STREAM_ALARM, &active, 0); 2148 if (status == NO_ERROR && active == true) { 2149 mStreamType = AUDIO_STREAM_ALARM; 2150 break; 2151 } 2152 } /// FALL THROUGH 2153 case AUDIO_USAGE_MEDIA: 2154 case AUDIO_USAGE_GAME: 2155 case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: 2156 mStreamType = AUDIO_STREAM_MUSIC; 2157 return; 2158 case AUDIO_USAGE_ASSISTANCE_SONIFICATION: 2159 mStreamType = AUDIO_STREAM_SYSTEM; 2160 return; 2161 case AUDIO_USAGE_VOICE_COMMUNICATION: 2162 mStreamType = AUDIO_STREAM_VOICE_CALL; 2163 return; 2164 2165 case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: 2166 mStreamType = AUDIO_STREAM_DTMF; 2167 return; 2168 2169 case AUDIO_USAGE_ALARM: 2170 mStreamType = AUDIO_STREAM_ALARM; 2171 return; 2172 case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: 2173 mStreamType = AUDIO_STREAM_RING; 2174 return; 2175 2176 case AUDIO_USAGE_NOTIFICATION: 2177 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: 2178 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: 2179 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: 2180 case AUDIO_USAGE_NOTIFICATION_EVENT: 2181 mStreamType = AUDIO_STREAM_NOTIFICATION; 2182 return; 2183 2184 case AUDIO_USAGE_UNKNOWN: 2185 default: 2186 mStreamType = AUDIO_STREAM_MUSIC; 2187 } 2188} 2189 2190bool AudioTrack::isValidAttributes(const audio_attributes_t *paa) { 2191 // has flags that map to a strategy? 2192 if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO | AUDIO_FLAG_BEACON)) != 0) { 2193 return true; 2194 } 2195 2196 // has known usage? 2197 switch (paa->usage) { 2198 case AUDIO_USAGE_UNKNOWN: 2199 case AUDIO_USAGE_MEDIA: 2200 case AUDIO_USAGE_VOICE_COMMUNICATION: 2201 case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: 2202 case AUDIO_USAGE_ALARM: 2203 case AUDIO_USAGE_NOTIFICATION: 2204 case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: 2205 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: 2206 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: 2207 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: 2208 case AUDIO_USAGE_NOTIFICATION_EVENT: 2209 case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: 2210 case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: 2211 case AUDIO_USAGE_ASSISTANCE_SONIFICATION: 2212 case AUDIO_USAGE_GAME: 2213 break; 2214 default: 2215 return false; 2216 } 2217 return true; 2218} 2219// ========================================================================= 2220 2221void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 2222{ 2223 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 2224 if (audioTrack != 0) { 2225 AutoMutex lock(audioTrack->mLock); 2226 audioTrack->mProxy->binderDied(); 2227 } 2228} 2229 2230// ========================================================================= 2231 2232AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 2233 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 2234 mIgnoreNextPausedInt(false) 2235{ 2236} 2237 2238AudioTrack::AudioTrackThread::~AudioTrackThread() 2239{ 2240} 2241 2242bool AudioTrack::AudioTrackThread::threadLoop() 2243{ 2244 { 2245 AutoMutex _l(mMyLock); 2246 if (mPaused) { 2247 mMyCond.wait(mMyLock); 2248 // caller will check for exitPending() 2249 return true; 2250 } 2251 if (mIgnoreNextPausedInt) { 2252 mIgnoreNextPausedInt = false; 2253 mPausedInt = false; 2254 } 2255 if (mPausedInt) { 2256 if (mPausedNs > 0) { 2257 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 2258 } else { 2259 mMyCond.wait(mMyLock); 2260 } 2261 mPausedInt = false; 2262 return true; 2263 } 2264 } 2265 if (exitPending()) { 2266 return false; 2267 } 2268 nsecs_t ns = mReceiver.processAudioBuffer(); 2269 switch (ns) { 2270 case 0: 2271 return true; 2272 case NS_INACTIVE: 2273 pauseInternal(); 2274 return true; 2275 case NS_NEVER: 2276 return false; 2277 case NS_WHENEVER: 2278 // FIXME increase poll interval, or make event-driven 2279 ns = 1000000000LL; 2280 // fall through 2281 default: 2282 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns); 2283 pauseInternal(ns); 2284 return true; 2285 } 2286} 2287 2288void AudioTrack::AudioTrackThread::requestExit() 2289{ 2290 // must be in this order to avoid a race condition 2291 Thread::requestExit(); 2292 resume(); 2293} 2294 2295void AudioTrack::AudioTrackThread::pause() 2296{ 2297 AutoMutex _l(mMyLock); 2298 mPaused = true; 2299} 2300 2301void AudioTrack::AudioTrackThread::resume() 2302{ 2303 AutoMutex _l(mMyLock); 2304 mIgnoreNextPausedInt = true; 2305 if (mPaused || mPausedInt) { 2306 mPaused = false; 2307 mPausedInt = false; 2308 mMyCond.signal(); 2309 } 2310} 2311 2312void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 2313{ 2314 AutoMutex _l(mMyLock); 2315 mPausedInt = true; 2316 mPausedNs = ns; 2317} 2318 2319}; // namespace android 2320