AudioTrack.cpp revision cd04484f4837b8ca0041d118286ab6a98e84fc75
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18//#define LOG_NDEBUG 0 19#define LOG_TAG "AudioTrack" 20 21#include <inttypes.h> 22#include <math.h> 23#include <sys/resource.h> 24 25#include <audio_utils/primitives.h> 26#include <binder/IPCThreadState.h> 27#include <media/AudioTrack.h> 28#include <utils/Log.h> 29#include <private/media/AudioTrackShared.h> 30#include <media/IAudioFlinger.h> 31#include <media/AudioResamplerPublic.h> 32 33#define WAIT_PERIOD_MS 10 34#define WAIT_STREAM_END_TIMEOUT_SEC 120 35 36 37namespace android { 38// --------------------------------------------------------------------------- 39 40// static 41status_t AudioTrack::getMinFrameCount( 42 size_t* frameCount, 43 audio_stream_type_t streamType, 44 uint32_t sampleRate) 45{ 46 if (frameCount == NULL) { 47 return BAD_VALUE; 48 } 49 50 // FIXME merge with similar code in createTrack_l(), except we're missing 51 // some information here that is available in createTrack_l(): 52 // audio_io_handle_t output 53 // audio_format_t format 54 // audio_channel_mask_t channelMask 55 // audio_output_flags_t flags 56 uint32_t afSampleRate; 57 status_t status; 58 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 59 if (status != NO_ERROR) { 60 ALOGE("Unable to query output sample rate for stream type %d; status %d", 61 streamType, status); 62 return status; 63 } 64 size_t afFrameCount; 65 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 66 if (status != NO_ERROR) { 67 ALOGE("Unable to query output frame count for stream type %d; status %d", 68 streamType, status); 69 return status; 70 } 71 uint32_t afLatency; 72 status = AudioSystem::getOutputLatency(&afLatency, streamType); 73 if (status != NO_ERROR) { 74 ALOGE("Unable to query output latency for stream type %d; status %d", 75 streamType, status); 76 return status; 77 } 78 79 // Ensure that buffer depth covers at least audio hardware latency 80 uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate); 81 if (minBufCount < 2) { 82 minBufCount = 2; 83 } 84 85 *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : 86 afFrameCount * minBufCount * uint64_t(sampleRate) / afSampleRate; 87 // The formula above should always produce a non-zero value, but return an error 88 // in the unlikely event that it does not, as that's part of the API contract. 89 if (*frameCount == 0) { 90 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", 91 streamType, sampleRate); 92 return BAD_VALUE; 93 } 94 ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%d, afSampleRate=%d, afLatency=%d", 95 *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); 96 return NO_ERROR; 97} 98 99// --------------------------------------------------------------------------- 100 101AudioTrack::AudioTrack() 102 : mStatus(NO_INIT), 103 mIsTimed(false), 104 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 105 mPreviousSchedulingGroup(SP_DEFAULT), 106 mPausedPosition(0) 107{ 108 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN; 109 mAttributes.usage = AUDIO_USAGE_UNKNOWN; 110 mAttributes.flags = 0x0; 111 strcpy(mAttributes.tags, ""); 112} 113 114AudioTrack::AudioTrack( 115 audio_stream_type_t streamType, 116 uint32_t sampleRate, 117 audio_format_t format, 118 audio_channel_mask_t channelMask, 119 size_t frameCount, 120 audio_output_flags_t flags, 121 callback_t cbf, 122 void* user, 123 uint32_t notificationFrames, 124 int sessionId, 125 transfer_type transferType, 126 const audio_offload_info_t *offloadInfo, 127 int uid, 128 pid_t pid, 129 const audio_attributes_t* pAttributes) 130 : mStatus(NO_INIT), 131 mIsTimed(false), 132 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 133 mPreviousSchedulingGroup(SP_DEFAULT), 134 mPausedPosition(0) 135{ 136 mStatus = set(streamType, sampleRate, format, channelMask, 137 frameCount, flags, cbf, user, notificationFrames, 138 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 139 offloadInfo, uid, pid, pAttributes); 140} 141 142AudioTrack::AudioTrack( 143 audio_stream_type_t streamType, 144 uint32_t sampleRate, 145 audio_format_t format, 146 audio_channel_mask_t channelMask, 147 const sp<IMemory>& sharedBuffer, 148 audio_output_flags_t flags, 149 callback_t cbf, 150 void* user, 151 uint32_t notificationFrames, 152 int sessionId, 153 transfer_type transferType, 154 const audio_offload_info_t *offloadInfo, 155 int uid, 156 pid_t pid, 157 const audio_attributes_t* pAttributes) 158 : mStatus(NO_INIT), 159 mIsTimed(false), 160 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 161 mPreviousSchedulingGroup(SP_DEFAULT), 162 mPausedPosition(0) 163{ 164 mStatus = set(streamType, sampleRate, format, channelMask, 165 0 /*frameCount*/, flags, cbf, user, notificationFrames, 166 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, 167 uid, pid, pAttributes); 168} 169 170AudioTrack::~AudioTrack() 171{ 172 if (mStatus == NO_ERROR) { 173 // Make sure that callback function exits in the case where 174 // it is looping on buffer full condition in obtainBuffer(). 175 // Otherwise the callback thread will never exit. 176 stop(); 177 if (mAudioTrackThread != 0) { 178 mProxy->interrupt(); 179 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 180 mAudioTrackThread->requestExitAndWait(); 181 mAudioTrackThread.clear(); 182 } 183 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 184 mAudioTrack.clear(); 185 mCblkMemory.clear(); 186 mSharedBuffer.clear(); 187 IPCThreadState::self()->flushCommands(); 188 ALOGV("~AudioTrack, releasing session id from %d on behalf of %d", 189 IPCThreadState::self()->getCallingPid(), mClientPid); 190 AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); 191 } 192} 193 194status_t AudioTrack::set( 195 audio_stream_type_t streamType, 196 uint32_t sampleRate, 197 audio_format_t format, 198 audio_channel_mask_t channelMask, 199 size_t frameCount, 200 audio_output_flags_t flags, 201 callback_t cbf, 202 void* user, 203 uint32_t notificationFrames, 204 const sp<IMemory>& sharedBuffer, 205 bool threadCanCallJava, 206 int sessionId, 207 transfer_type transferType, 208 const audio_offload_info_t *offloadInfo, 209 int uid, 210 pid_t pid, 211 const audio_attributes_t* pAttributes) 212{ 213 ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 214 "flags #%x, notificationFrames %u, sessionId %d, transferType %d", 215 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, 216 sessionId, transferType); 217 218 switch (transferType) { 219 case TRANSFER_DEFAULT: 220 if (sharedBuffer != 0) { 221 transferType = TRANSFER_SHARED; 222 } else if (cbf == NULL || threadCanCallJava) { 223 transferType = TRANSFER_SYNC; 224 } else { 225 transferType = TRANSFER_CALLBACK; 226 } 227 break; 228 case TRANSFER_CALLBACK: 229 if (cbf == NULL || sharedBuffer != 0) { 230 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 231 return BAD_VALUE; 232 } 233 break; 234 case TRANSFER_OBTAIN: 235 case TRANSFER_SYNC: 236 if (sharedBuffer != 0) { 237 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 238 return BAD_VALUE; 239 } 240 break; 241 case TRANSFER_SHARED: 242 if (sharedBuffer == 0) { 243 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 244 return BAD_VALUE; 245 } 246 break; 247 default: 248 ALOGE("Invalid transfer type %d", transferType); 249 return BAD_VALUE; 250 } 251 mSharedBuffer = sharedBuffer; 252 mTransfer = transferType; 253 254 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 255 sharedBuffer->size()); 256 257 ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags); 258 259 AutoMutex lock(mLock); 260 261 // invariant that mAudioTrack != 0 is true only after set() returns successfully 262 if (mAudioTrack != 0) { 263 ALOGE("Track already in use"); 264 return INVALID_OPERATION; 265 } 266 267 // handle default values first. 268 if (streamType == AUDIO_STREAM_DEFAULT) { 269 streamType = AUDIO_STREAM_MUSIC; 270 } 271 272 if (pAttributes == NULL) { 273 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 274 ALOGE("Invalid stream type %d", streamType); 275 return BAD_VALUE; 276 } 277 setAttributesFromStreamType(streamType); 278 mStreamType = streamType; 279 } else { 280 if (!isValidAttributes(pAttributes)) { 281 ALOGE("Invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]", 282 pAttributes->usage, pAttributes->content_type, pAttributes->flags, 283 pAttributes->tags); 284 } 285 // stream type shouldn't be looked at, this track has audio attributes 286 memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t)); 287 setStreamTypeFromAttributes(mAttributes); 288 ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]", 289 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags); 290 } 291 292 status_t status; 293 if (sampleRate == 0) { 294 status = AudioSystem::getOutputSamplingRateForAttr(&sampleRate, &mAttributes); 295 if (status != NO_ERROR) { 296 ALOGE("Could not get output sample rate for stream type %d; status %d", 297 mStreamType, status); 298 return status; 299 } 300 } 301 mSampleRate = sampleRate; 302 303 // these below should probably come from the audioFlinger too... 304 if (format == AUDIO_FORMAT_DEFAULT) { 305 format = AUDIO_FORMAT_PCM_16_BIT; 306 } 307 308 // validate parameters 309 if (!audio_is_valid_format(format)) { 310 ALOGE("Invalid format %#x", format); 311 return BAD_VALUE; 312 } 313 mFormat = format; 314 315 if (!audio_is_output_channel(channelMask)) { 316 ALOGE("Invalid channel mask %#x", channelMask); 317 return BAD_VALUE; 318 } 319 mChannelMask = channelMask; 320 uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 321 mChannelCount = channelCount; 322 323 // AudioFlinger does not currently support 8-bit data in shared memory 324 if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) { 325 ALOGE("8-bit data in shared memory is not supported"); 326 return BAD_VALUE; 327 } 328 329 // force direct flag if format is not linear PCM 330 // or offload was requested 331 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 332 || !audio_is_linear_pcm(format)) { 333 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 334 ? "Offload request, forcing to Direct Output" 335 : "Not linear PCM, forcing to Direct Output"); 336 flags = (audio_output_flags_t) 337 // FIXME why can't we allow direct AND fast? 338 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 339 } 340 // only allow deep buffering for music stream type 341 if (mStreamType != AUDIO_STREAM_MUSIC) { 342 flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER); 343 } 344 345 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { 346 if (audio_is_linear_pcm(format)) { 347 mFrameSize = channelCount * audio_bytes_per_sample(format); 348 } else { 349 mFrameSize = sizeof(uint8_t); 350 } 351 mFrameSizeAF = mFrameSize; 352 } else { 353 ALOG_ASSERT(audio_is_linear_pcm(format)); 354 mFrameSize = channelCount * audio_bytes_per_sample(format); 355 mFrameSizeAF = channelCount * audio_bytes_per_sample( 356 format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format); 357 // createTrack will return an error if PCM format is not supported by server, 358 // so no need to check for specific PCM formats here 359 } 360 361 // Make copy of input parameter offloadInfo so that in the future: 362 // (a) createTrack_l doesn't need it as an input parameter 363 // (b) we can support re-creation of offloaded tracks 364 if (offloadInfo != NULL) { 365 mOffloadInfoCopy = *offloadInfo; 366 mOffloadInfo = &mOffloadInfoCopy; 367 } else { 368 mOffloadInfo = NULL; 369 } 370 371 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f; 372 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f; 373 mSendLevel = 0.0f; 374 // mFrameCount is initialized in createTrack_l 375 mReqFrameCount = frameCount; 376 mNotificationFramesReq = notificationFrames; 377 mNotificationFramesAct = 0; 378 mSessionId = sessionId; 379 int callingpid = IPCThreadState::self()->getCallingPid(); 380 int mypid = getpid(); 381 if (uid == -1 || (callingpid != mypid)) { 382 mClientUid = IPCThreadState::self()->getCallingUid(); 383 } else { 384 mClientUid = uid; 385 } 386 if (pid == -1 || (callingpid != mypid)) { 387 mClientPid = callingpid; 388 } else { 389 mClientPid = pid; 390 } 391 mAuxEffectId = 0; 392 mFlags = flags; 393 mCbf = cbf; 394 395 if (cbf != NULL) { 396 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 397 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 398 } 399 400 // create the IAudioTrack 401 status = createTrack_l(0 /*epoch*/); 402 403 if (status != NO_ERROR) { 404 if (mAudioTrackThread != 0) { 405 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 406 mAudioTrackThread->requestExitAndWait(); 407 mAudioTrackThread.clear(); 408 } 409 return status; 410 } 411 412 mStatus = NO_ERROR; 413 mState = STATE_STOPPED; 414 mUserData = user; 415 mLoopPeriod = 0; 416 mMarkerPosition = 0; 417 mMarkerReached = false; 418 mNewPosition = 0; 419 mUpdatePeriod = 0; 420 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 421 mSequence = 1; 422 mObservedSequence = mSequence; 423 mInUnderrun = false; 424 425 return NO_ERROR; 426} 427 428// ------------------------------------------------------------------------- 429 430status_t AudioTrack::start() 431{ 432 AutoMutex lock(mLock); 433 434 if (mState == STATE_ACTIVE) { 435 return INVALID_OPERATION; 436 } 437 438 mInUnderrun = true; 439 440 State previousState = mState; 441 if (previousState == STATE_PAUSED_STOPPING) { 442 mState = STATE_STOPPING; 443 } else { 444 mState = STATE_ACTIVE; 445 } 446 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 447 // reset current position as seen by client to 0 448 mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition()); 449 // force refresh of remaining frames by processAudioBuffer() as last 450 // write before stop could be partial. 451 mRefreshRemaining = true; 452 } 453 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 454 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 455 456 sp<AudioTrackThread> t = mAudioTrackThread; 457 if (t != 0) { 458 if (previousState == STATE_STOPPING) { 459 mProxy->interrupt(); 460 } else { 461 t->resume(); 462 } 463 } else { 464 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 465 get_sched_policy(0, &mPreviousSchedulingGroup); 466 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 467 } 468 469 status_t status = NO_ERROR; 470 if (!(flags & CBLK_INVALID)) { 471 status = mAudioTrack->start(); 472 if (status == DEAD_OBJECT) { 473 flags |= CBLK_INVALID; 474 } 475 } 476 if (flags & CBLK_INVALID) { 477 status = restoreTrack_l("start"); 478 } 479 480 if (status != NO_ERROR) { 481 ALOGE("start() status %d", status); 482 mState = previousState; 483 if (t != 0) { 484 if (previousState != STATE_STOPPING) { 485 t->pause(); 486 } 487 } else { 488 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 489 set_sched_policy(0, mPreviousSchedulingGroup); 490 } 491 } 492 493 return status; 494} 495 496void AudioTrack::stop() 497{ 498 AutoMutex lock(mLock); 499 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 500 return; 501 } 502 503 if (isOffloaded_l()) { 504 mState = STATE_STOPPING; 505 } else { 506 mState = STATE_STOPPED; 507 } 508 509 mProxy->interrupt(); 510 mAudioTrack->stop(); 511 // the playback head position will reset to 0, so if a marker is set, we need 512 // to activate it again 513 mMarkerReached = false; 514#if 0 515 // Force flush if a shared buffer is used otherwise audioflinger 516 // will not stop before end of buffer is reached. 517 // It may be needed to make sure that we stop playback, likely in case looping is on. 518 if (mSharedBuffer != 0) { 519 flush_l(); 520 } 521#endif 522 523 sp<AudioTrackThread> t = mAudioTrackThread; 524 if (t != 0) { 525 if (!isOffloaded_l()) { 526 t->pause(); 527 } 528 } else { 529 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 530 set_sched_policy(0, mPreviousSchedulingGroup); 531 } 532} 533 534bool AudioTrack::stopped() const 535{ 536 AutoMutex lock(mLock); 537 return mState != STATE_ACTIVE; 538} 539 540void AudioTrack::flush() 541{ 542 if (mSharedBuffer != 0) { 543 return; 544 } 545 AutoMutex lock(mLock); 546 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 547 return; 548 } 549 flush_l(); 550} 551 552void AudioTrack::flush_l() 553{ 554 ALOG_ASSERT(mState != STATE_ACTIVE); 555 556 // clear playback marker and periodic update counter 557 mMarkerPosition = 0; 558 mMarkerReached = false; 559 mUpdatePeriod = 0; 560 mRefreshRemaining = true; 561 562 mState = STATE_FLUSHED; 563 if (isOffloaded_l()) { 564 mProxy->interrupt(); 565 } 566 mProxy->flush(); 567 mAudioTrack->flush(); 568} 569 570void AudioTrack::pause() 571{ 572 AutoMutex lock(mLock); 573 if (mState == STATE_ACTIVE) { 574 mState = STATE_PAUSED; 575 } else if (mState == STATE_STOPPING) { 576 mState = STATE_PAUSED_STOPPING; 577 } else { 578 return; 579 } 580 mProxy->interrupt(); 581 mAudioTrack->pause(); 582 583 if (isOffloaded_l()) { 584 if (mOutput != AUDIO_IO_HANDLE_NONE) { 585 uint32_t halFrames; 586 // OffloadThread sends HAL pause in its threadLoop.. time saved 587 // here can be slightly off 588 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); 589 ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); 590 } 591 } 592} 593 594status_t AudioTrack::setVolume(float left, float right) 595{ 596 // This duplicates a test by AudioTrack JNI, but that is not the only caller 597 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY || 598 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) { 599 return BAD_VALUE; 600 } 601 602 AutoMutex lock(mLock); 603 mVolume[AUDIO_INTERLEAVE_LEFT] = left; 604 mVolume[AUDIO_INTERLEAVE_RIGHT] = right; 605 606 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right))); 607 608 if (isOffloaded_l()) { 609 mAudioTrack->signal(); 610 } 611 return NO_ERROR; 612} 613 614status_t AudioTrack::setVolume(float volume) 615{ 616 return setVolume(volume, volume); 617} 618 619status_t AudioTrack::setAuxEffectSendLevel(float level) 620{ 621 // This duplicates a test by AudioTrack JNI, but that is not the only caller 622 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) { 623 return BAD_VALUE; 624 } 625 626 AutoMutex lock(mLock); 627 mSendLevel = level; 628 mProxy->setSendLevel(level); 629 630 return NO_ERROR; 631} 632 633void AudioTrack::getAuxEffectSendLevel(float* level) const 634{ 635 if (level != NULL) { 636 *level = mSendLevel; 637 } 638} 639 640status_t AudioTrack::setSampleRate(uint32_t rate) 641{ 642 if (mIsTimed || isOffloadedOrDirect()) { 643 return INVALID_OPERATION; 644 } 645 646 uint32_t afSamplingRate; 647 if (AudioSystem::getOutputSamplingRateForAttr(&afSamplingRate, &mAttributes) != NO_ERROR) { 648 return NO_INIT; 649 } 650 if (rate == 0 || rate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 651 return BAD_VALUE; 652 } 653 654 AutoMutex lock(mLock); 655 mSampleRate = rate; 656 mProxy->setSampleRate(rate); 657 658 return NO_ERROR; 659} 660 661uint32_t AudioTrack::getSampleRate() const 662{ 663 if (mIsTimed) { 664 return 0; 665 } 666 667 AutoMutex lock(mLock); 668 669 // sample rate can be updated during playback by the offloaded decoder so we need to 670 // query the HAL and update if needed. 671// FIXME use Proxy return channel to update the rate from server and avoid polling here 672 if (isOffloadedOrDirect_l()) { 673 if (mOutput != AUDIO_IO_HANDLE_NONE) { 674 uint32_t sampleRate = 0; 675 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate); 676 if (status == NO_ERROR) { 677 mSampleRate = sampleRate; 678 } 679 } 680 } 681 return mSampleRate; 682} 683 684status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 685{ 686 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 687 return INVALID_OPERATION; 688 } 689 690 if (loopCount == 0) { 691 ; 692 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 693 loopEnd - loopStart >= MIN_LOOP) { 694 ; 695 } else { 696 return BAD_VALUE; 697 } 698 699 AutoMutex lock(mLock); 700 // See setPosition() regarding setting parameters such as loop points or position while active 701 if (mState == STATE_ACTIVE) { 702 return INVALID_OPERATION; 703 } 704 setLoop_l(loopStart, loopEnd, loopCount); 705 return NO_ERROR; 706} 707 708void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 709{ 710 // FIXME If setting a loop also sets position to start of loop, then 711 // this is correct. Otherwise it should be removed. 712 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 713 mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0; 714 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 715} 716 717status_t AudioTrack::setMarkerPosition(uint32_t marker) 718{ 719 // The only purpose of setting marker position is to get a callback 720 if (mCbf == NULL || isOffloadedOrDirect()) { 721 return INVALID_OPERATION; 722 } 723 724 AutoMutex lock(mLock); 725 mMarkerPosition = marker; 726 mMarkerReached = false; 727 728 return NO_ERROR; 729} 730 731status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 732{ 733 if (isOffloadedOrDirect()) { 734 return INVALID_OPERATION; 735 } 736 if (marker == NULL) { 737 return BAD_VALUE; 738 } 739 740 AutoMutex lock(mLock); 741 *marker = mMarkerPosition; 742 743 return NO_ERROR; 744} 745 746status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 747{ 748 // The only purpose of setting position update period is to get a callback 749 if (mCbf == NULL || isOffloadedOrDirect()) { 750 return INVALID_OPERATION; 751 } 752 753 AutoMutex lock(mLock); 754 mNewPosition = mProxy->getPosition() + updatePeriod; 755 mUpdatePeriod = updatePeriod; 756 757 return NO_ERROR; 758} 759 760status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 761{ 762 if (isOffloadedOrDirect()) { 763 return INVALID_OPERATION; 764 } 765 if (updatePeriod == NULL) { 766 return BAD_VALUE; 767 } 768 769 AutoMutex lock(mLock); 770 *updatePeriod = mUpdatePeriod; 771 772 return NO_ERROR; 773} 774 775status_t AudioTrack::setPosition(uint32_t position) 776{ 777 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 778 return INVALID_OPERATION; 779 } 780 if (position > mFrameCount) { 781 return BAD_VALUE; 782 } 783 784 AutoMutex lock(mLock); 785 // Currently we require that the player is inactive before setting parameters such as position 786 // or loop points. Otherwise, there could be a race condition: the application could read the 787 // current position, compute a new position or loop parameters, and then set that position or 788 // loop parameters but it would do the "wrong" thing since the position has continued to advance 789 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 790 // to specify how it wants to handle such scenarios. 791 if (mState == STATE_ACTIVE) { 792 return INVALID_OPERATION; 793 } 794 mNewPosition = mProxy->getPosition() + mUpdatePeriod; 795 mLoopPeriod = 0; 796 // FIXME Check whether loops and setting position are incompatible in old code. 797 // If we use setLoop for both purposes we lose the capability to set the position while looping. 798 mStaticProxy->setLoop(position, mFrameCount, 0); 799 800 return NO_ERROR; 801} 802 803status_t AudioTrack::getPosition(uint32_t *position) const 804{ 805 if (position == NULL) { 806 return BAD_VALUE; 807 } 808 809 AutoMutex lock(mLock); 810 if (isOffloadedOrDirect_l()) { 811 uint32_t dspFrames = 0; 812 813 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) { 814 ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); 815 *position = mPausedPosition; 816 return NO_ERROR; 817 } 818 819 if (mOutput != AUDIO_IO_HANDLE_NONE) { 820 uint32_t halFrames; 821 AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 822 } 823 *position = dspFrames; 824 } else { 825 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 826 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 0 : 827 mProxy->getPosition(); 828 } 829 return NO_ERROR; 830} 831 832status_t AudioTrack::getBufferPosition(uint32_t *position) 833{ 834 if (mSharedBuffer == 0 || mIsTimed) { 835 return INVALID_OPERATION; 836 } 837 if (position == NULL) { 838 return BAD_VALUE; 839 } 840 841 AutoMutex lock(mLock); 842 *position = mStaticProxy->getBufferPosition(); 843 return NO_ERROR; 844} 845 846status_t AudioTrack::reload() 847{ 848 if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) { 849 return INVALID_OPERATION; 850 } 851 852 AutoMutex lock(mLock); 853 // See setPosition() regarding setting parameters such as loop points or position while active 854 if (mState == STATE_ACTIVE) { 855 return INVALID_OPERATION; 856 } 857 mNewPosition = mUpdatePeriod; 858 mLoopPeriod = 0; 859 // FIXME The new code cannot reload while keeping a loop specified. 860 // Need to check how the old code handled this, and whether it's a significant change. 861 mStaticProxy->setLoop(0, mFrameCount, 0); 862 return NO_ERROR; 863} 864 865audio_io_handle_t AudioTrack::getOutput() const 866{ 867 AutoMutex lock(mLock); 868 return mOutput; 869} 870 871status_t AudioTrack::attachAuxEffect(int effectId) 872{ 873 AutoMutex lock(mLock); 874 status_t status = mAudioTrack->attachAuxEffect(effectId); 875 if (status == NO_ERROR) { 876 mAuxEffectId = effectId; 877 } 878 return status; 879} 880 881// ------------------------------------------------------------------------- 882 883// must be called with mLock held 884status_t AudioTrack::createTrack_l(size_t epoch) 885{ 886 status_t status; 887 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 888 if (audioFlinger == 0) { 889 ALOGE("Could not get audioflinger"); 890 return NO_INIT; 891 } 892 893 audio_io_handle_t output = AudioSystem::getOutputForAttr(&mAttributes, mSampleRate, mFormat, 894 mChannelMask, mFlags, mOffloadInfo); 895 if (output == AUDIO_IO_HANDLE_NONE) { 896 ALOGE("Could not get audio output for stream type %d, usage %d, sample rate %u, format %#x," 897 " channel mask %#x, flags %#x", 898 mStreamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags); 899 return BAD_VALUE; 900 } 901 { 902 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 903 // we must release it ourselves if anything goes wrong. 904 905 // Not all of these values are needed under all conditions, but it is easier to get them all 906 907 uint32_t afLatency; 908 status = AudioSystem::getLatency(output, &afLatency); 909 if (status != NO_ERROR) { 910 ALOGE("getLatency(%d) failed status %d", output, status); 911 goto release; 912 } 913 914 size_t afFrameCount; 915 status = AudioSystem::getFrameCount(output, &afFrameCount); 916 if (status != NO_ERROR) { 917 ALOGE("getFrameCount(output=%d) status %d", output, status); 918 goto release; 919 } 920 921 uint32_t afSampleRate; 922 status = AudioSystem::getSamplingRate(output, &afSampleRate); 923 if (status != NO_ERROR) { 924 ALOGE("getSamplingRate(output=%d) status %d", output, status); 925 goto release; 926 } 927 928 // Client decides whether the track is TIMED (see below), but can only express a preference 929 // for FAST. Server will perform additional tests. 930 if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) && !(( 931 // either of these use cases: 932 // use case 1: shared buffer 933 (mSharedBuffer != 0) || 934 // use case 2: callback transfer mode 935 (mTransfer == TRANSFER_CALLBACK)) && 936 // matching sample rate 937 (mSampleRate == afSampleRate))) { 938 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client"); 939 // once denied, do not request again if IAudioTrack is re-created 940 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 941 } 942 ALOGV("createTrack_l() output %d afLatency %d", output, afLatency); 943 944 // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where 945 // n = 1 fast track with single buffering; nBuffering is ignored 946 // n = 2 fast track with double buffering 947 // n = 2 normal track, no sample rate conversion 948 // n = 3 normal track, with sample rate conversion 949 // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) 950 // n > 3 very high latency or very small notification interval; nBuffering is ignored 951 const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; 952 953 mNotificationFramesAct = mNotificationFramesReq; 954 955 size_t frameCount = mReqFrameCount; 956 if (!audio_is_linear_pcm(mFormat)) { 957 958 if (mSharedBuffer != 0) { 959 // Same comment as below about ignoring frameCount parameter for set() 960 frameCount = mSharedBuffer->size(); 961 } else if (frameCount == 0) { 962 frameCount = afFrameCount; 963 } 964 if (mNotificationFramesAct != frameCount) { 965 mNotificationFramesAct = frameCount; 966 } 967 } else if (mSharedBuffer != 0) { 968 969 // Ensure that buffer alignment matches channel count 970 // 8-bit data in shared memory is not currently supported by AudioFlinger 971 size_t alignment = audio_bytes_per_sample( 972 mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat); 973 if (alignment & 1) { 974 alignment = 1; 975 } 976 if (mChannelCount > 1) { 977 // More than 2 channels does not require stronger alignment than stereo 978 alignment <<= 1; 979 } 980 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 981 ALOGE("Invalid buffer alignment: address %p, channel count %u", 982 mSharedBuffer->pointer(), mChannelCount); 983 status = BAD_VALUE; 984 goto release; 985 } 986 987 // When initializing a shared buffer AudioTrack via constructors, 988 // there's no frameCount parameter. 989 // But when initializing a shared buffer AudioTrack via set(), 990 // there _is_ a frameCount parameter. We silently ignore it. 991 frameCount = mSharedBuffer->size() / mFrameSizeAF; 992 993 } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 994 995 // FIXME move these calculations and associated checks to server 996 997 // Ensure that buffer depth covers at least audio hardware latency 998 uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); 999 ALOGV("afFrameCount=%zu, minBufCount=%d, afSampleRate=%u, afLatency=%d", 1000 afFrameCount, minBufCount, afSampleRate, afLatency); 1001 if (minBufCount <= nBuffering) { 1002 minBufCount = nBuffering; 1003 } 1004 1005 size_t minFrameCount = afFrameCount * minBufCount * uint64_t(mSampleRate) / afSampleRate; 1006 ALOGV("minFrameCount: %zu, afFrameCount=%zu, minBufCount=%d, sampleRate=%u, afSampleRate=%u" 1007 ", afLatency=%d", 1008 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); 1009 1010 if (frameCount == 0) { 1011 frameCount = minFrameCount; 1012 } else if (frameCount < minFrameCount) { 1013 // not ALOGW because it happens all the time when playing key clicks over A2DP 1014 ALOGV("Minimum buffer size corrected from %zu to %zu", 1015 frameCount, minFrameCount); 1016 frameCount = minFrameCount; 1017 } 1018 // Make sure that application is notified with sufficient margin before underrun 1019 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1020 mNotificationFramesAct = frameCount/nBuffering; 1021 } 1022 1023 } else { 1024 // For fast tracks, the frame count calculations and checks are done by server 1025 } 1026 1027 IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; 1028 if (mIsTimed) { 1029 trackFlags |= IAudioFlinger::TRACK_TIMED; 1030 } 1031 1032 pid_t tid = -1; 1033 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1034 trackFlags |= IAudioFlinger::TRACK_FAST; 1035 if (mAudioTrackThread != 0) { 1036 tid = mAudioTrackThread->getTid(); 1037 } 1038 } 1039 1040 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1041 trackFlags |= IAudioFlinger::TRACK_OFFLOAD; 1042 } 1043 1044 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1045 trackFlags |= IAudioFlinger::TRACK_DIRECT; 1046 } 1047 1048 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1049 // but we will still need the original value also 1050 sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType, 1051 mSampleRate, 1052 // AudioFlinger only sees 16-bit PCM 1053 mFormat == AUDIO_FORMAT_PCM_8_BIT && 1054 !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT) ? 1055 AUDIO_FORMAT_PCM_16_BIT : mFormat, 1056 mChannelMask, 1057 &temp, 1058 &trackFlags, 1059 mSharedBuffer, 1060 output, 1061 tid, 1062 &mSessionId, 1063 mClientUid, 1064 &status); 1065 1066 if (status != NO_ERROR) { 1067 ALOGE("AudioFlinger could not create track, status: %d", status); 1068 goto release; 1069 } 1070 ALOG_ASSERT(track != 0); 1071 1072 // AudioFlinger now owns the reference to the I/O handle, 1073 // so we are no longer responsible for releasing it. 1074 1075 sp<IMemory> iMem = track->getCblk(); 1076 if (iMem == 0) { 1077 ALOGE("Could not get control block"); 1078 return NO_INIT; 1079 } 1080 void *iMemPointer = iMem->pointer(); 1081 if (iMemPointer == NULL) { 1082 ALOGE("Could not get control block pointer"); 1083 return NO_INIT; 1084 } 1085 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1086 if (mAudioTrack != 0) { 1087 mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this); 1088 mDeathNotifier.clear(); 1089 } 1090 mAudioTrack = track; 1091 mCblkMemory = iMem; 1092 IPCThreadState::self()->flushCommands(); 1093 1094 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1095 mCblk = cblk; 1096 // note that temp is the (possibly revised) value of frameCount 1097 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1098 // In current design, AudioTrack client checks and ensures frame count validity before 1099 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1100 // for fast track as it uses a special method of assigning frame count. 1101 ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp); 1102 } 1103 frameCount = temp; 1104 1105 mAwaitBoost = false; 1106 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1107 if (trackFlags & IAudioFlinger::TRACK_FAST) { 1108 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount); 1109 mAwaitBoost = true; 1110 if (mSharedBuffer == 0) { 1111 // Theoretically double-buffering is not required for fast tracks, 1112 // due to tighter scheduling. But in practice, to accommodate kernels with 1113 // scheduling jitter, and apps with computation jitter, we use double-buffering. 1114 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1115 mNotificationFramesAct = frameCount/nBuffering; 1116 } 1117 } 1118 } else { 1119 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount); 1120 // once denied, do not request again if IAudioTrack is re-created 1121 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1122 if (mSharedBuffer == 0) { 1123 if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { 1124 mNotificationFramesAct = frameCount/nBuffering; 1125 } 1126 } 1127 } 1128 } 1129 if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1130 if (trackFlags & IAudioFlinger::TRACK_OFFLOAD) { 1131 ALOGV("AUDIO_OUTPUT_FLAG_OFFLOAD successful"); 1132 } else { 1133 ALOGW("AUDIO_OUTPUT_FLAG_OFFLOAD denied by server"); 1134 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD); 1135 // FIXME This is a warning, not an error, so don't return error status 1136 //return NO_INIT; 1137 } 1138 } 1139 if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { 1140 if (trackFlags & IAudioFlinger::TRACK_DIRECT) { 1141 ALOGV("AUDIO_OUTPUT_FLAG_DIRECT successful"); 1142 } else { 1143 ALOGW("AUDIO_OUTPUT_FLAG_DIRECT denied by server"); 1144 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT); 1145 // FIXME This is a warning, not an error, so don't return error status 1146 //return NO_INIT; 1147 } 1148 } 1149 1150 // We retain a copy of the I/O handle, but don't own the reference 1151 mOutput = output; 1152 mRefreshRemaining = true; 1153 1154 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1155 // is the value of pointer() for the shared buffer, otherwise buffers points 1156 // immediately after the control block. This address is for the mapping within client 1157 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1158 void* buffers; 1159 if (mSharedBuffer == 0) { 1160 buffers = (char*)cblk + sizeof(audio_track_cblk_t); 1161 } else { 1162 buffers = mSharedBuffer->pointer(); 1163 } 1164 1165 mAudioTrack->attachAuxEffect(mAuxEffectId); 1166 // FIXME don't believe this lie 1167 mLatency = afLatency + (1000*frameCount) / mSampleRate; 1168 1169 mFrameCount = frameCount; 1170 // If IAudioTrack is re-created, don't let the requested frameCount 1171 // decrease. This can confuse clients that cache frameCount(). 1172 if (frameCount > mReqFrameCount) { 1173 mReqFrameCount = frameCount; 1174 } 1175 1176 // update proxy 1177 if (mSharedBuffer == 0) { 1178 mStaticProxy.clear(); 1179 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1180 } else { 1181 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF); 1182 mProxy = mStaticProxy; 1183 } 1184 mProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); 1185 mProxy->setSendLevel(mSendLevel); 1186 mProxy->setSampleRate(mSampleRate); 1187 mProxy->setEpoch(epoch); 1188 mProxy->setMinimum(mNotificationFramesAct); 1189 1190 mDeathNotifier = new DeathNotifier(this); 1191 mAudioTrack->asBinder()->linkToDeath(mDeathNotifier, this); 1192 1193 return NO_ERROR; 1194 } 1195 1196release: 1197 AudioSystem::releaseOutput(output); 1198 if (status == NO_ERROR) { 1199 status = NO_INIT; 1200 } 1201 return status; 1202} 1203 1204status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount) 1205{ 1206 if (audioBuffer == NULL) { 1207 return BAD_VALUE; 1208 } 1209 if (mTransfer != TRANSFER_OBTAIN) { 1210 audioBuffer->frameCount = 0; 1211 audioBuffer->size = 0; 1212 audioBuffer->raw = NULL; 1213 return INVALID_OPERATION; 1214 } 1215 1216 const struct timespec *requested; 1217 struct timespec timeout; 1218 if (waitCount == -1) { 1219 requested = &ClientProxy::kForever; 1220 } else if (waitCount == 0) { 1221 requested = &ClientProxy::kNonBlocking; 1222 } else if (waitCount > 0) { 1223 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1224 timeout.tv_sec = ms / 1000; 1225 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1226 requested = &timeout; 1227 } else { 1228 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1229 requested = NULL; 1230 } 1231 return obtainBuffer(audioBuffer, requested); 1232} 1233 1234status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1235 struct timespec *elapsed, size_t *nonContig) 1236{ 1237 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1238 uint32_t oldSequence = 0; 1239 uint32_t newSequence; 1240 1241 Proxy::Buffer buffer; 1242 status_t status = NO_ERROR; 1243 1244 static const int32_t kMaxTries = 5; 1245 int32_t tryCounter = kMaxTries; 1246 1247 do { 1248 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1249 // keep them from going away if another thread re-creates the track during obtainBuffer() 1250 sp<AudioTrackClientProxy> proxy; 1251 sp<IMemory> iMem; 1252 1253 { // start of lock scope 1254 AutoMutex lock(mLock); 1255 1256 newSequence = mSequence; 1257 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1258 if (status == DEAD_OBJECT) { 1259 // re-create track, unless someone else has already done so 1260 if (newSequence == oldSequence) { 1261 status = restoreTrack_l("obtainBuffer"); 1262 if (status != NO_ERROR) { 1263 buffer.mFrameCount = 0; 1264 buffer.mRaw = NULL; 1265 buffer.mNonContig = 0; 1266 break; 1267 } 1268 } 1269 } 1270 oldSequence = newSequence; 1271 1272 // Keep the extra references 1273 proxy = mProxy; 1274 iMem = mCblkMemory; 1275 1276 if (mState == STATE_STOPPING) { 1277 status = -EINTR; 1278 buffer.mFrameCount = 0; 1279 buffer.mRaw = NULL; 1280 buffer.mNonContig = 0; 1281 break; 1282 } 1283 1284 // Non-blocking if track is stopped or paused 1285 if (mState != STATE_ACTIVE) { 1286 requested = &ClientProxy::kNonBlocking; 1287 } 1288 1289 } // end of lock scope 1290 1291 buffer.mFrameCount = audioBuffer->frameCount; 1292 // FIXME starts the requested timeout and elapsed over from scratch 1293 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1294 1295 } while ((status == DEAD_OBJECT) && (tryCounter-- > 0)); 1296 1297 audioBuffer->frameCount = buffer.mFrameCount; 1298 audioBuffer->size = buffer.mFrameCount * mFrameSizeAF; 1299 audioBuffer->raw = buffer.mRaw; 1300 if (nonContig != NULL) { 1301 *nonContig = buffer.mNonContig; 1302 } 1303 return status; 1304} 1305 1306void AudioTrack::releaseBuffer(Buffer* audioBuffer) 1307{ 1308 if (mTransfer == TRANSFER_SHARED) { 1309 return; 1310 } 1311 1312 size_t stepCount = audioBuffer->size / mFrameSizeAF; 1313 if (stepCount == 0) { 1314 return; 1315 } 1316 1317 Proxy::Buffer buffer; 1318 buffer.mFrameCount = stepCount; 1319 buffer.mRaw = audioBuffer->raw; 1320 1321 AutoMutex lock(mLock); 1322 mInUnderrun = false; 1323 mProxy->releaseBuffer(&buffer); 1324 1325 // restart track if it was disabled by audioflinger due to previous underrun 1326 if (mState == STATE_ACTIVE) { 1327 audio_track_cblk_t* cblk = mCblk; 1328 if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) { 1329 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1330 // FIXME ignoring status 1331 mAudioTrack->start(); 1332 } 1333 } 1334} 1335 1336// ------------------------------------------------------------------------- 1337 1338ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1339{ 1340 if (mTransfer != TRANSFER_SYNC || mIsTimed) { 1341 return INVALID_OPERATION; 1342 } 1343 1344 if (isDirect()) { 1345 AutoMutex lock(mLock); 1346 int32_t flags = android_atomic_and( 1347 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), 1348 &mCblk->mFlags); 1349 if (flags & CBLK_INVALID) { 1350 return DEAD_OBJECT; 1351 } 1352 } 1353 1354 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1355 // Sanity-check: user is most-likely passing an error code, and it would 1356 // make the return value ambiguous (actualSize vs error). 1357 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1358 return BAD_VALUE; 1359 } 1360 1361 size_t written = 0; 1362 Buffer audioBuffer; 1363 1364 while (userSize >= mFrameSize) { 1365 audioBuffer.frameCount = userSize / mFrameSize; 1366 1367 status_t err = obtainBuffer(&audioBuffer, 1368 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1369 if (err < 0) { 1370 if (written > 0) { 1371 break; 1372 } 1373 return ssize_t(err); 1374 } 1375 1376 size_t toWrite; 1377 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1378 // Divide capacity by 2 to take expansion into account 1379 toWrite = audioBuffer.size >> 1; 1380 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite); 1381 } else { 1382 toWrite = audioBuffer.size; 1383 memcpy(audioBuffer.i8, buffer, toWrite); 1384 } 1385 buffer = ((const char *) buffer) + toWrite; 1386 userSize -= toWrite; 1387 written += toWrite; 1388 1389 releaseBuffer(&audioBuffer); 1390 } 1391 1392 return written; 1393} 1394 1395// ------------------------------------------------------------------------- 1396 1397TimedAudioTrack::TimedAudioTrack() { 1398 mIsTimed = true; 1399} 1400 1401status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer) 1402{ 1403 AutoMutex lock(mLock); 1404 status_t result = UNKNOWN_ERROR; 1405 1406#if 1 1407 // acquire a strong reference on the IMemory and IAudioTrack so that they cannot be destroyed 1408 // while we are accessing the cblk 1409 sp<IAudioTrack> audioTrack = mAudioTrack; 1410 sp<IMemory> iMem = mCblkMemory; 1411#endif 1412 1413 // If the track is not invalid already, try to allocate a buffer. alloc 1414 // fails indicating that the server is dead, flag the track as invalid so 1415 // we can attempt to restore in just a bit. 1416 audio_track_cblk_t* cblk = mCblk; 1417 if (!(cblk->mFlags & CBLK_INVALID)) { 1418 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1419 if (result == DEAD_OBJECT) { 1420 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1421 } 1422 } 1423 1424 // If the track is invalid at this point, attempt to restore it. and try the 1425 // allocation one more time. 1426 if (cblk->mFlags & CBLK_INVALID) { 1427 result = restoreTrack_l("allocateTimedBuffer"); 1428 1429 if (result == NO_ERROR) { 1430 result = mAudioTrack->allocateTimedBuffer(size, buffer); 1431 } 1432 } 1433 1434 return result; 1435} 1436 1437status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer, 1438 int64_t pts) 1439{ 1440 status_t status = mAudioTrack->queueTimedBuffer(buffer, pts); 1441 { 1442 AutoMutex lock(mLock); 1443 audio_track_cblk_t* cblk = mCblk; 1444 // restart track if it was disabled by audioflinger due to previous underrun 1445 if (buffer->size() != 0 && status == NO_ERROR && 1446 (mState == STATE_ACTIVE) && (cblk->mFlags & CBLK_DISABLED)) { 1447 android_atomic_and(~CBLK_DISABLED, &cblk->mFlags); 1448 ALOGW("queueTimedBuffer() track %p disabled, restarting", this); 1449 // FIXME ignoring status 1450 mAudioTrack->start(); 1451 } 1452 } 1453 return status; 1454} 1455 1456status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform, 1457 TargetTimeline target) 1458{ 1459 return mAudioTrack->setMediaTimeTransform(xform, target); 1460} 1461 1462// ------------------------------------------------------------------------- 1463 1464nsecs_t AudioTrack::processAudioBuffer() 1465{ 1466 // Currently the AudioTrack thread is not created if there are no callbacks. 1467 // Would it ever make sense to run the thread, even without callbacks? 1468 // If so, then replace this by checks at each use for mCbf != NULL. 1469 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1470 1471 mLock.lock(); 1472 if (mAwaitBoost) { 1473 mAwaitBoost = false; 1474 mLock.unlock(); 1475 static const int32_t kMaxTries = 5; 1476 int32_t tryCounter = kMaxTries; 1477 uint32_t pollUs = 10000; 1478 do { 1479 int policy = sched_getscheduler(0); 1480 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1481 break; 1482 } 1483 usleep(pollUs); 1484 pollUs <<= 1; 1485 } while (tryCounter-- > 0); 1486 if (tryCounter < 0) { 1487 ALOGE("did not receive expected priority boost on time"); 1488 } 1489 // Run again immediately 1490 return 0; 1491 } 1492 1493 // Can only reference mCblk while locked 1494 int32_t flags = android_atomic_and( 1495 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1496 1497 // Check for track invalidation 1498 if (flags & CBLK_INVALID) { 1499 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1500 // AudioSystem cache. We should not exit here but after calling the callback so 1501 // that the upper layers can recreate the track 1502 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) { 1503 status_t status = restoreTrack_l("processAudioBuffer"); 1504 mLock.unlock(); 1505 // Run again immediately, but with a new IAudioTrack 1506 return 0; 1507 } 1508 } 1509 1510 bool waitStreamEnd = mState == STATE_STOPPING; 1511 bool active = mState == STATE_ACTIVE; 1512 1513 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1514 bool newUnderrun = false; 1515 if (flags & CBLK_UNDERRUN) { 1516#if 0 1517 // Currently in shared buffer mode, when the server reaches the end of buffer, 1518 // the track stays active in continuous underrun state. It's up to the application 1519 // to pause or stop the track, or set the position to a new offset within buffer. 1520 // This was some experimental code to auto-pause on underrun. Keeping it here 1521 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1522 if (mTransfer == TRANSFER_SHARED) { 1523 mState = STATE_PAUSED; 1524 active = false; 1525 } 1526#endif 1527 if (!mInUnderrun) { 1528 mInUnderrun = true; 1529 newUnderrun = true; 1530 } 1531 } 1532 1533 // Get current position of server 1534 size_t position = mProxy->getPosition(); 1535 1536 // Manage marker callback 1537 bool markerReached = false; 1538 size_t markerPosition = mMarkerPosition; 1539 // FIXME fails for wraparound, need 64 bits 1540 if (!mMarkerReached && (markerPosition > 0) && (position >= markerPosition)) { 1541 mMarkerReached = markerReached = true; 1542 } 1543 1544 // Determine number of new position callback(s) that will be needed, while locked 1545 size_t newPosCount = 0; 1546 size_t newPosition = mNewPosition; 1547 size_t updatePeriod = mUpdatePeriod; 1548 // FIXME fails for wraparound, need 64 bits 1549 if (updatePeriod > 0 && position >= newPosition) { 1550 newPosCount = ((position - newPosition) / updatePeriod) + 1; 1551 mNewPosition += updatePeriod * newPosCount; 1552 } 1553 1554 // Cache other fields that will be needed soon 1555 uint32_t loopPeriod = mLoopPeriod; 1556 uint32_t sampleRate = mSampleRate; 1557 uint32_t notificationFrames = mNotificationFramesAct; 1558 if (mRefreshRemaining) { 1559 mRefreshRemaining = false; 1560 mRemainingFrames = notificationFrames; 1561 mRetryOnPartialBuffer = false; 1562 } 1563 size_t misalignment = mProxy->getMisalignment(); 1564 uint32_t sequence = mSequence; 1565 sp<AudioTrackClientProxy> proxy = mProxy; 1566 1567 // These fields don't need to be cached, because they are assigned only by set(): 1568 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags 1569 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1570 1571 mLock.unlock(); 1572 1573 if (waitStreamEnd) { 1574 struct timespec timeout; 1575 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1576 timeout.tv_nsec = 0; 1577 1578 status_t status = proxy->waitStreamEndDone(&timeout); 1579 switch (status) { 1580 case NO_ERROR: 1581 case DEAD_OBJECT: 1582 case TIMED_OUT: 1583 mCbf(EVENT_STREAM_END, mUserData, NULL); 1584 { 1585 AutoMutex lock(mLock); 1586 // The previously assigned value of waitStreamEnd is no longer valid, 1587 // since the mutex has been unlocked and either the callback handler 1588 // or another thread could have re-started the AudioTrack during that time. 1589 waitStreamEnd = mState == STATE_STOPPING; 1590 if (waitStreamEnd) { 1591 mState = STATE_STOPPED; 1592 } 1593 } 1594 if (waitStreamEnd && status != DEAD_OBJECT) { 1595 return NS_INACTIVE; 1596 } 1597 break; 1598 } 1599 return 0; 1600 } 1601 1602 // perform callbacks while unlocked 1603 if (newUnderrun) { 1604 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1605 } 1606 // FIXME we will miss loops if loop cycle was signaled several times since last call 1607 // to processAudioBuffer() 1608 if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) { 1609 mCbf(EVENT_LOOP_END, mUserData, NULL); 1610 } 1611 if (flags & CBLK_BUFFER_END) { 1612 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1613 } 1614 if (markerReached) { 1615 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1616 } 1617 while (newPosCount > 0) { 1618 size_t temp = newPosition; 1619 mCbf(EVENT_NEW_POS, mUserData, &temp); 1620 newPosition += updatePeriod; 1621 newPosCount--; 1622 } 1623 1624 if (mObservedSequence != sequence) { 1625 mObservedSequence = sequence; 1626 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1627 // for offloaded tracks, just wait for the upper layers to recreate the track 1628 if (isOffloadedOrDirect()) { 1629 return NS_INACTIVE; 1630 } 1631 } 1632 1633 // if inactive, then don't run me again until re-started 1634 if (!active) { 1635 return NS_INACTIVE; 1636 } 1637 1638 // Compute the estimated time until the next timed event (position, markers, loops) 1639 // FIXME only for non-compressed audio 1640 uint32_t minFrames = ~0; 1641 if (!markerReached && position < markerPosition) { 1642 minFrames = markerPosition - position; 1643 } 1644 if (loopPeriod > 0 && loopPeriod < minFrames) { 1645 minFrames = loopPeriod; 1646 } 1647 if (updatePeriod > 0 && updatePeriod < minFrames) { 1648 minFrames = updatePeriod; 1649 } 1650 1651 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1652 static const uint32_t kPoll = 0; 1653 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1654 minFrames = kPoll * notificationFrames; 1655 } 1656 1657 // Convert frame units to time units 1658 nsecs_t ns = NS_WHENEVER; 1659 if (minFrames != (uint32_t) ~0) { 1660 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1661 static const nsecs_t kFudgeNs = 10000000LL; // 10 ms 1662 ns = ((minFrames * 1000000000LL) / sampleRate) + kFudgeNs; 1663 } 1664 1665 // If not supplying data by EVENT_MORE_DATA, then we're done 1666 if (mTransfer != TRANSFER_CALLBACK) { 1667 return ns; 1668 } 1669 1670 struct timespec timeout; 1671 const struct timespec *requested = &ClientProxy::kForever; 1672 if (ns != NS_WHENEVER) { 1673 timeout.tv_sec = ns / 1000000000LL; 1674 timeout.tv_nsec = ns % 1000000000LL; 1675 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 1676 requested = &timeout; 1677 } 1678 1679 while (mRemainingFrames > 0) { 1680 1681 Buffer audioBuffer; 1682 audioBuffer.frameCount = mRemainingFrames; 1683 size_t nonContig; 1684 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 1685 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 1686 "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount); 1687 requested = &ClientProxy::kNonBlocking; 1688 size_t avail = audioBuffer.frameCount + nonContig; 1689 ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d", 1690 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 1691 if (err != NO_ERROR) { 1692 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 1693 (isOffloaded() && (err == DEAD_OBJECT))) { 1694 return 0; 1695 } 1696 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 1697 return NS_NEVER; 1698 } 1699 1700 if (mRetryOnPartialBuffer && !isOffloaded()) { 1701 mRetryOnPartialBuffer = false; 1702 if (avail < mRemainingFrames) { 1703 int64_t myns = ((mRemainingFrames - avail) * 1100000000LL) / sampleRate; 1704 if (ns < 0 || myns < ns) { 1705 ns = myns; 1706 } 1707 return ns; 1708 } 1709 } 1710 1711 // Divide buffer size by 2 to take into account the expansion 1712 // due to 8 to 16 bit conversion: the callback must fill only half 1713 // of the destination buffer 1714 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1715 audioBuffer.size >>= 1; 1716 } 1717 1718 size_t reqSize = audioBuffer.size; 1719 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 1720 size_t writtenSize = audioBuffer.size; 1721 1722 // Sanity check on returned size 1723 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 1724 ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes", 1725 reqSize, ssize_t(writtenSize)); 1726 return NS_NEVER; 1727 } 1728 1729 if (writtenSize == 0) { 1730 // The callback is done filling buffers 1731 // Keep this thread going to handle timed events and 1732 // still try to get more data in intervals of WAIT_PERIOD_MS 1733 // but don't just loop and block the CPU, so wait 1734 return WAIT_PERIOD_MS * 1000000LL; 1735 } 1736 1737 if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) { 1738 // 8 to 16 bit conversion, note that source and destination are the same address 1739 memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize); 1740 audioBuffer.size <<= 1; 1741 } 1742 1743 size_t releasedFrames = audioBuffer.size / mFrameSizeAF; 1744 audioBuffer.frameCount = releasedFrames; 1745 mRemainingFrames -= releasedFrames; 1746 if (misalignment >= releasedFrames) { 1747 misalignment -= releasedFrames; 1748 } else { 1749 misalignment = 0; 1750 } 1751 1752 releaseBuffer(&audioBuffer); 1753 1754 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 1755 // if callback doesn't like to accept the full chunk 1756 if (writtenSize < reqSize) { 1757 continue; 1758 } 1759 1760 // There could be enough non-contiguous frames available to satisfy the remaining request 1761 if (mRemainingFrames <= nonContig) { 1762 continue; 1763 } 1764 1765#if 0 1766 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 1767 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 1768 // that total to a sum == notificationFrames. 1769 if (0 < misalignment && misalignment <= mRemainingFrames) { 1770 mRemainingFrames = misalignment; 1771 return (mRemainingFrames * 1100000000LL) / sampleRate; 1772 } 1773#endif 1774 1775 } 1776 mRemainingFrames = notificationFrames; 1777 mRetryOnPartialBuffer = true; 1778 1779 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 1780 return 0; 1781} 1782 1783status_t AudioTrack::restoreTrack_l(const char *from) 1784{ 1785 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 1786 isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from); 1787 ++mSequence; 1788 status_t result; 1789 1790 // refresh the audio configuration cache in this process to make sure we get new 1791 // output parameters in createTrack_l() 1792 AudioSystem::clearAudioConfigCache(); 1793 1794 if (isOffloadedOrDirect_l()) { 1795 // FIXME re-creation of offloaded tracks is not yet implemented 1796 return DEAD_OBJECT; 1797 } 1798 1799 // if the new IAudioTrack is created, createTrack_l() will modify the 1800 // following member variables: mAudioTrack, mCblkMemory and mCblk. 1801 // It will also delete the strong references on previous IAudioTrack and IMemory 1802 1803 // take the frames that will be lost by track recreation into account in saved position 1804 size_t position = mProxy->getPosition() + mProxy->getFramesFilled(); 1805 size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0; 1806 result = createTrack_l(position /*epoch*/); 1807 1808 if (result == NO_ERROR) { 1809 // continue playback from last known position, but 1810 // don't attempt to restore loop after invalidation; it's difficult and not worthwhile 1811 if (mStaticProxy != NULL) { 1812 mLoopPeriod = 0; 1813 mStaticProxy->setLoop(bufferPosition, mFrameCount, 0); 1814 } 1815 // FIXME How do we simulate the fact that all frames present in the buffer at the time of 1816 // track destruction have been played? This is critical for SoundPool implementation 1817 // This must be broken, and needs to be tested/debugged. 1818#if 0 1819 // restore write index and set other indexes to reflect empty buffer status 1820 if (!strcmp(from, "start")) { 1821 // Make sure that a client relying on callback events indicating underrun or 1822 // the actual amount of audio frames played (e.g SoundPool) receives them. 1823 if (mSharedBuffer == 0) { 1824 // restart playback even if buffer is not completely filled. 1825 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1826 } 1827 } 1828#endif 1829 if (mState == STATE_ACTIVE) { 1830 result = mAudioTrack->start(); 1831 } 1832 } 1833 if (result != NO_ERROR) { 1834 ALOGW("restoreTrack_l() failed status %d", result); 1835 mState = STATE_STOPPED; 1836 } 1837 1838 return result; 1839} 1840 1841status_t AudioTrack::setParameters(const String8& keyValuePairs) 1842{ 1843 AutoMutex lock(mLock); 1844 return mAudioTrack->setParameters(keyValuePairs); 1845} 1846 1847status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 1848{ 1849 AutoMutex lock(mLock); 1850 // FIXME not implemented for fast tracks; should use proxy and SSQ 1851 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1852 return INVALID_OPERATION; 1853 } 1854 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 1855 return INVALID_OPERATION; 1856 } 1857 status_t status = mAudioTrack->getTimestamp(timestamp); 1858 if (status == NO_ERROR) { 1859 timestamp.mPosition += mProxy->getEpoch(); 1860 } 1861 return status; 1862} 1863 1864String8 AudioTrack::getParameters(const String8& keys) 1865{ 1866 audio_io_handle_t output = getOutput(); 1867 if (output != AUDIO_IO_HANDLE_NONE) { 1868 return AudioSystem::getParameters(output, keys); 1869 } else { 1870 return String8::empty(); 1871 } 1872} 1873 1874bool AudioTrack::isOffloaded() const 1875{ 1876 AutoMutex lock(mLock); 1877 return isOffloaded_l(); 1878} 1879 1880bool AudioTrack::isDirect() const 1881{ 1882 AutoMutex lock(mLock); 1883 return isDirect_l(); 1884} 1885 1886bool AudioTrack::isOffloadedOrDirect() const 1887{ 1888 AutoMutex lock(mLock); 1889 return isOffloadedOrDirect_l(); 1890} 1891 1892 1893status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 1894{ 1895 1896 const size_t SIZE = 256; 1897 char buffer[SIZE]; 1898 String8 result; 1899 1900 result.append(" AudioTrack::dump\n"); 1901 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 1902 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 1903 result.append(buffer); 1904 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 1905 mChannelCount, mFrameCount); 1906 result.append(buffer); 1907 snprintf(buffer, 255, " sample rate(%u), status(%d)\n", mSampleRate, mStatus); 1908 result.append(buffer); 1909 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 1910 result.append(buffer); 1911 ::write(fd, result.string(), result.size()); 1912 return NO_ERROR; 1913} 1914 1915uint32_t AudioTrack::getUnderrunFrames() const 1916{ 1917 AutoMutex lock(mLock); 1918 return mProxy->getUnderrunFrames(); 1919} 1920 1921void AudioTrack::setAttributesFromStreamType(audio_stream_type_t streamType) { 1922 mAttributes.flags = 0x0; 1923 1924 switch(streamType) { 1925 case AUDIO_STREAM_DEFAULT: 1926 case AUDIO_STREAM_MUSIC: 1927 mAttributes.content_type = AUDIO_CONTENT_TYPE_MUSIC; 1928 mAttributes.usage = AUDIO_USAGE_MEDIA; 1929 break; 1930 case AUDIO_STREAM_VOICE_CALL: 1931 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 1932 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION; 1933 break; 1934 case AUDIO_STREAM_ENFORCED_AUDIBLE: 1935 mAttributes.flags |= AUDIO_FLAG_AUDIBILITY_ENFORCED; 1936 // intended fall through, attributes in common with STREAM_SYSTEM 1937 case AUDIO_STREAM_SYSTEM: 1938 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 1939 mAttributes.usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION; 1940 break; 1941 case AUDIO_STREAM_RING: 1942 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 1943 mAttributes.usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE; 1944 break; 1945 case AUDIO_STREAM_ALARM: 1946 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 1947 mAttributes.usage = AUDIO_USAGE_ALARM; 1948 break; 1949 case AUDIO_STREAM_NOTIFICATION: 1950 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 1951 mAttributes.usage = AUDIO_USAGE_NOTIFICATION; 1952 break; 1953 case AUDIO_STREAM_BLUETOOTH_SCO: 1954 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 1955 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION; 1956 mAttributes.flags |= AUDIO_FLAG_SCO; 1957 break; 1958 case AUDIO_STREAM_DTMF: 1959 mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION; 1960 mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING; 1961 break; 1962 case AUDIO_STREAM_TTS: 1963 mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH; 1964 mAttributes.usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY; 1965 break; 1966 default: 1967 ALOGE("invalid stream type %d when converting to attributes", streamType); 1968 } 1969} 1970 1971void AudioTrack::setStreamTypeFromAttributes(audio_attributes_t& aa) { 1972 // flags to stream type mapping 1973 if ((aa.flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { 1974 mStreamType = AUDIO_STREAM_ENFORCED_AUDIBLE; 1975 return; 1976 } 1977 if ((aa.flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) { 1978 mStreamType = AUDIO_STREAM_BLUETOOTH_SCO; 1979 return; 1980 } 1981 1982 // usage to stream type mapping 1983 switch (aa.usage) { 1984 case AUDIO_USAGE_MEDIA: 1985 case AUDIO_USAGE_GAME: 1986 case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: 1987 case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: 1988 mStreamType = AUDIO_STREAM_MUSIC; 1989 return; 1990 case AUDIO_USAGE_ASSISTANCE_SONIFICATION: 1991 mStreamType = AUDIO_STREAM_SYSTEM; 1992 return; 1993 case AUDIO_USAGE_VOICE_COMMUNICATION: 1994 mStreamType = AUDIO_STREAM_VOICE_CALL; 1995 return; 1996 1997 case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: 1998 mStreamType = AUDIO_STREAM_DTMF; 1999 return; 2000 2001 case AUDIO_USAGE_ALARM: 2002 mStreamType = AUDIO_STREAM_ALARM; 2003 return; 2004 case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: 2005 mStreamType = AUDIO_STREAM_RING; 2006 return; 2007 2008 case AUDIO_USAGE_NOTIFICATION: 2009 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: 2010 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: 2011 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: 2012 case AUDIO_USAGE_NOTIFICATION_EVENT: 2013 mStreamType = AUDIO_STREAM_NOTIFICATION; 2014 return; 2015 2016 case AUDIO_USAGE_UNKNOWN: 2017 default: 2018 mStreamType = AUDIO_STREAM_MUSIC; 2019 } 2020} 2021 2022bool AudioTrack::isValidAttributes(const audio_attributes_t *paa) { 2023 // has flags that map to a strategy? 2024 if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO)) != 0) { 2025 return true; 2026 } 2027 2028 // has known usage? 2029 switch (paa->usage) { 2030 case AUDIO_USAGE_UNKNOWN: 2031 case AUDIO_USAGE_MEDIA: 2032 case AUDIO_USAGE_VOICE_COMMUNICATION: 2033 case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: 2034 case AUDIO_USAGE_ALARM: 2035 case AUDIO_USAGE_NOTIFICATION: 2036 case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: 2037 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: 2038 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: 2039 case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: 2040 case AUDIO_USAGE_NOTIFICATION_EVENT: 2041 case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: 2042 case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: 2043 case AUDIO_USAGE_ASSISTANCE_SONIFICATION: 2044 case AUDIO_USAGE_GAME: 2045 break; 2046 default: 2047 return false; 2048 } 2049 return true; 2050} 2051// ========================================================================= 2052 2053void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 2054{ 2055 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 2056 if (audioTrack != 0) { 2057 AutoMutex lock(audioTrack->mLock); 2058 audioTrack->mProxy->binderDied(); 2059 } 2060} 2061 2062// ========================================================================= 2063 2064AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 2065 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 2066 mIgnoreNextPausedInt(false) 2067{ 2068} 2069 2070AudioTrack::AudioTrackThread::~AudioTrackThread() 2071{ 2072} 2073 2074bool AudioTrack::AudioTrackThread::threadLoop() 2075{ 2076 { 2077 AutoMutex _l(mMyLock); 2078 if (mPaused) { 2079 mMyCond.wait(mMyLock); 2080 // caller will check for exitPending() 2081 return true; 2082 } 2083 if (mIgnoreNextPausedInt) { 2084 mIgnoreNextPausedInt = false; 2085 mPausedInt = false; 2086 } 2087 if (mPausedInt) { 2088 if (mPausedNs > 0) { 2089 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 2090 } else { 2091 mMyCond.wait(mMyLock); 2092 } 2093 mPausedInt = false; 2094 return true; 2095 } 2096 } 2097 nsecs_t ns = mReceiver.processAudioBuffer(); 2098 switch (ns) { 2099 case 0: 2100 return true; 2101 case NS_INACTIVE: 2102 pauseInternal(); 2103 return true; 2104 case NS_NEVER: 2105 return false; 2106 case NS_WHENEVER: 2107 // FIXME increase poll interval, or make event-driven 2108 ns = 1000000000LL; 2109 // fall through 2110 default: 2111 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns); 2112 pauseInternal(ns); 2113 return true; 2114 } 2115} 2116 2117void AudioTrack::AudioTrackThread::requestExit() 2118{ 2119 // must be in this order to avoid a race condition 2120 Thread::requestExit(); 2121 resume(); 2122} 2123 2124void AudioTrack::AudioTrackThread::pause() 2125{ 2126 AutoMutex _l(mMyLock); 2127 mPaused = true; 2128} 2129 2130void AudioTrack::AudioTrackThread::resume() 2131{ 2132 AutoMutex _l(mMyLock); 2133 mIgnoreNextPausedInt = true; 2134 if (mPaused || mPausedInt) { 2135 mPaused = false; 2136 mPausedInt = false; 2137 mMyCond.signal(); 2138 } 2139} 2140 2141void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 2142{ 2143 AutoMutex _l(mMyLock); 2144 mPausedInt = true; 2145 mPausedNs = ns; 2146} 2147 2148}; // namespace android 2149