AudioTrack.cpp revision 2148bf0e79c436b8764b9edc4c8f2730cce98a32
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18//#define LOG_NDEBUG 0 19#define LOG_TAG "AudioTrack" 20 21#include <inttypes.h> 22#include <math.h> 23#include <sys/resource.h> 24 25#include <audio_utils/primitives.h> 26#include <binder/IPCThreadState.h> 27#include <media/AudioTrack.h> 28#include <utils/Log.h> 29#include <private/media/AudioTrackShared.h> 30#include <media/IAudioFlinger.h> 31#include <media/AudioPolicyHelper.h> 32#include <media/AudioResamplerPublic.h> 33 34#define WAIT_PERIOD_MS 10 35#define WAIT_STREAM_END_TIMEOUT_SEC 120 36static const int kMaxLoopCountNotifications = 32; 37 38namespace android { 39// --------------------------------------------------------------------------- 40 41// TODO: Move to a separate .h 42 43template <typename T> 44static inline const T &min(const T &x, const T &y) { 45 return x < y ? x : y; 46} 47 48template <typename T> 49static inline const T &max(const T &x, const T &y) { 50 return x > y ? x : y; 51} 52 53static const int32_t NANOS_PER_SECOND = 1000000000; 54 55static inline nsecs_t framesToNanoseconds(ssize_t frames, uint32_t sampleRate, float speed) 56{ 57 return ((double)frames * 1000000000) / ((double)sampleRate * speed); 58} 59 60static int64_t convertTimespecToUs(const struct timespec &tv) 61{ 62 return tv.tv_sec * 1000000ll + tv.tv_nsec / 1000; 63} 64 65static inline nsecs_t convertTimespecToNs(const struct timespec &tv) 66{ 67 return tv.tv_sec * (long long)NANOS_PER_SECOND + tv.tv_nsec; 68} 69 70// current monotonic time in microseconds. 71static int64_t getNowUs() 72{ 73 struct timespec tv; 74 (void) clock_gettime(CLOCK_MONOTONIC, &tv); 75 return convertTimespecToUs(tv); 76} 77 78// FIXME: we don't use the pitch setting in the time stretcher (not working); 79// instead we emulate it using our sample rate converter. 80static const bool kFixPitch = true; // enable pitch fix 81static inline uint32_t adjustSampleRate(uint32_t sampleRate, float pitch) 82{ 83 return kFixPitch ? (sampleRate * pitch + 0.5) : sampleRate; 84} 85 86static inline float adjustSpeed(float speed, float pitch) 87{ 88 return kFixPitch ? speed / max(pitch, AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) : speed; 89} 90 91static inline float adjustPitch(float pitch) 92{ 93 return kFixPitch ? AUDIO_TIMESTRETCH_PITCH_NORMAL : pitch; 94} 95 96// Must match similar computation in createTrack_l in Threads.cpp. 97// TODO: Move to a common library 98static size_t calculateMinFrameCount( 99 uint32_t afLatencyMs, uint32_t afFrameCount, uint32_t afSampleRate, 100 uint32_t sampleRate, float speed /*, uint32_t notificationsPerBufferReq*/) 101{ 102 // Ensure that buffer depth covers at least audio hardware latency 103 uint32_t minBufCount = afLatencyMs / ((1000 * afFrameCount) / afSampleRate); 104 if (minBufCount < 2) { 105 minBufCount = 2; 106 } 107#if 0 108 // The notificationsPerBufferReq parameter is not yet used for non-fast tracks, 109 // but keeping the code here to make it easier to add later. 110 if (minBufCount < notificationsPerBufferReq) { 111 minBufCount = notificationsPerBufferReq; 112 } 113#endif 114 ALOGV("calculateMinFrameCount afLatency %u afFrameCount %u afSampleRate %u " 115 "sampleRate %u speed %f minBufCount: %u" /*" notificationsPerBufferReq %u"*/, 116 afLatencyMs, afFrameCount, afSampleRate, sampleRate, speed, minBufCount 117 /*, notificationsPerBufferReq*/); 118 return minBufCount * sourceFramesNeededWithTimestretch( 119 sampleRate, afFrameCount, afSampleRate, speed); 120} 121 122// static 123status_t AudioTrack::getMinFrameCount( 124 size_t* frameCount, 125 audio_stream_type_t streamType, 126 uint32_t sampleRate) 127{ 128 if (frameCount == NULL) { 129 return BAD_VALUE; 130 } 131 132 // FIXME handle in server, like createTrack_l(), possible missing info: 133 // audio_io_handle_t output 134 // audio_format_t format 135 // audio_channel_mask_t channelMask 136 // audio_output_flags_t flags (FAST) 137 uint32_t afSampleRate; 138 status_t status; 139 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 140 if (status != NO_ERROR) { 141 ALOGE("Unable to query output sample rate for stream type %d; status %d", 142 streamType, status); 143 return status; 144 } 145 size_t afFrameCount; 146 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 147 if (status != NO_ERROR) { 148 ALOGE("Unable to query output frame count for stream type %d; status %d", 149 streamType, status); 150 return status; 151 } 152 uint32_t afLatency; 153 status = AudioSystem::getOutputLatency(&afLatency, streamType); 154 if (status != NO_ERROR) { 155 ALOGE("Unable to query output latency for stream type %d; status %d", 156 streamType, status); 157 return status; 158 } 159 160 // When called from createTrack, speed is 1.0f (normal speed). 161 // This is rechecked again on setting playback rate (TODO: on setting sample rate, too). 162 *frameCount = calculateMinFrameCount(afLatency, afFrameCount, afSampleRate, sampleRate, 1.0f 163 /*, 0 notificationsPerBufferReq*/); 164 165 // The formula above should always produce a non-zero value under normal circumstances: 166 // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX. 167 // Return error in the unlikely event that it does not, as that's part of the API contract. 168 if (*frameCount == 0) { 169 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %u", 170 streamType, sampleRate); 171 return BAD_VALUE; 172 } 173 ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, afSampleRate=%u, afLatency=%u", 174 *frameCount, afFrameCount, afSampleRate, afLatency); 175 return NO_ERROR; 176} 177 178// --------------------------------------------------------------------------- 179 180AudioTrack::AudioTrack() 181 : mStatus(NO_INIT), 182 mState(STATE_STOPPED), 183 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 184 mPreviousSchedulingGroup(SP_DEFAULT), 185 mPausedPosition(0), 186 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE) 187{ 188 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN; 189 mAttributes.usage = AUDIO_USAGE_UNKNOWN; 190 mAttributes.flags = 0x0; 191 strcpy(mAttributes.tags, ""); 192} 193 194AudioTrack::AudioTrack( 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 int32_t notificationFrames, 204 audio_session_t sessionId, 205 transfer_type transferType, 206 const audio_offload_info_t *offloadInfo, 207 uid_t uid, 208 pid_t pid, 209 const audio_attributes_t* pAttributes, 210 bool doNotReconnect, 211 float maxRequiredSpeed) 212 : mStatus(NO_INIT), 213 mState(STATE_STOPPED), 214 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 215 mPreviousSchedulingGroup(SP_DEFAULT), 216 mPausedPosition(0), 217 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE) 218{ 219 mStatus = set(streamType, sampleRate, format, channelMask, 220 frameCount, flags, cbf, user, notificationFrames, 221 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 222 offloadInfo, uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed); 223} 224 225AudioTrack::AudioTrack( 226 audio_stream_type_t streamType, 227 uint32_t sampleRate, 228 audio_format_t format, 229 audio_channel_mask_t channelMask, 230 const sp<IMemory>& sharedBuffer, 231 audio_output_flags_t flags, 232 callback_t cbf, 233 void* user, 234 int32_t notificationFrames, 235 audio_session_t sessionId, 236 transfer_type transferType, 237 const audio_offload_info_t *offloadInfo, 238 uid_t uid, 239 pid_t pid, 240 const audio_attributes_t* pAttributes, 241 bool doNotReconnect, 242 float maxRequiredSpeed) 243 : mStatus(NO_INIT), 244 mState(STATE_STOPPED), 245 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 246 mPreviousSchedulingGroup(SP_DEFAULT), 247 mPausedPosition(0), 248 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE) 249{ 250 mStatus = set(streamType, sampleRate, format, channelMask, 251 0 /*frameCount*/, flags, cbf, user, notificationFrames, 252 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, 253 uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed); 254} 255 256AudioTrack::~AudioTrack() 257{ 258 if (mStatus == NO_ERROR) { 259 // Make sure that callback function exits in the case where 260 // it is looping on buffer full condition in obtainBuffer(). 261 // Otherwise the callback thread will never exit. 262 stop(); 263 if (mAudioTrackThread != 0) { 264 mProxy->interrupt(); 265 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 266 mAudioTrackThread->requestExitAndWait(); 267 mAudioTrackThread.clear(); 268 } 269 // No lock here: worst case we remove a NULL callback which will be a nop 270 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) { 271 AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); 272 } 273 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this); 274 mAudioTrack.clear(); 275 mCblkMemory.clear(); 276 mSharedBuffer.clear(); 277 IPCThreadState::self()->flushCommands(); 278 ALOGV("~AudioTrack, releasing session id %d from %d on behalf of %d", 279 mSessionId, IPCThreadState::self()->getCallingPid(), mClientPid); 280 AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); 281 } 282} 283 284status_t AudioTrack::set( 285 audio_stream_type_t streamType, 286 uint32_t sampleRate, 287 audio_format_t format, 288 audio_channel_mask_t channelMask, 289 size_t frameCount, 290 audio_output_flags_t flags, 291 callback_t cbf, 292 void* user, 293 int32_t notificationFrames, 294 const sp<IMemory>& sharedBuffer, 295 bool threadCanCallJava, 296 audio_session_t sessionId, 297 transfer_type transferType, 298 const audio_offload_info_t *offloadInfo, 299 uid_t uid, 300 pid_t pid, 301 const audio_attributes_t* pAttributes, 302 bool doNotReconnect, 303 float maxRequiredSpeed) 304{ 305 ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 306 "flags #%x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d", 307 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, 308 sessionId, transferType, uid, pid); 309 310 mThreadCanCallJava = threadCanCallJava; 311 312 switch (transferType) { 313 case TRANSFER_DEFAULT: 314 if (sharedBuffer != 0) { 315 transferType = TRANSFER_SHARED; 316 } else if (cbf == NULL || threadCanCallJava) { 317 transferType = TRANSFER_SYNC; 318 } else { 319 transferType = TRANSFER_CALLBACK; 320 } 321 break; 322 case TRANSFER_CALLBACK: 323 if (cbf == NULL || sharedBuffer != 0) { 324 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 325 return BAD_VALUE; 326 } 327 break; 328 case TRANSFER_OBTAIN: 329 case TRANSFER_SYNC: 330 if (sharedBuffer != 0) { 331 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 332 return BAD_VALUE; 333 } 334 break; 335 case TRANSFER_SHARED: 336 if (sharedBuffer == 0) { 337 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 338 return BAD_VALUE; 339 } 340 break; 341 default: 342 ALOGE("Invalid transfer type %d", transferType); 343 return BAD_VALUE; 344 } 345 mSharedBuffer = sharedBuffer; 346 mTransfer = transferType; 347 mDoNotReconnect = doNotReconnect; 348 349 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %zu", sharedBuffer->pointer(), 350 sharedBuffer->size()); 351 352 ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags); 353 354 // invariant that mAudioTrack != 0 is true only after set() returns successfully 355 if (mAudioTrack != 0) { 356 ALOGE("Track already in use"); 357 return INVALID_OPERATION; 358 } 359 360 // handle default values first. 361 if (streamType == AUDIO_STREAM_DEFAULT) { 362 streamType = AUDIO_STREAM_MUSIC; 363 } 364 if (pAttributes == NULL) { 365 if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) { 366 ALOGE("Invalid stream type %d", streamType); 367 return BAD_VALUE; 368 } 369 mStreamType = streamType; 370 371 } else { 372 // stream type shouldn't be looked at, this track has audio attributes 373 memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t)); 374 ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]", 375 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags); 376 mStreamType = AUDIO_STREAM_DEFAULT; 377 if ((mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) { 378 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC); 379 } 380 if ((mAttributes.flags & AUDIO_FLAG_LOW_LATENCY) != 0) { 381 flags = (audio_output_flags_t) (flags | AUDIO_OUTPUT_FLAG_FAST); 382 } 383 } 384 385 // these below should probably come from the audioFlinger too... 386 if (format == AUDIO_FORMAT_DEFAULT) { 387 format = AUDIO_FORMAT_PCM_16_BIT; 388 } else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through? 389 mAttributes.flags |= AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO; 390 } 391 392 // validate parameters 393 if (!audio_is_valid_format(format)) { 394 ALOGE("Invalid format %#x", format); 395 return BAD_VALUE; 396 } 397 mFormat = format; 398 399 if (!audio_is_output_channel(channelMask)) { 400 ALOGE("Invalid channel mask %#x", channelMask); 401 return BAD_VALUE; 402 } 403 mChannelMask = channelMask; 404 uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 405 mChannelCount = channelCount; 406 407 // force direct flag if format is not linear PCM 408 // or offload was requested 409 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 410 || !audio_is_linear_pcm(format)) { 411 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 412 ? "Offload request, forcing to Direct Output" 413 : "Not linear PCM, forcing to Direct Output"); 414 flags = (audio_output_flags_t) 415 // FIXME why can't we allow direct AND fast? 416 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 417 } 418 419 // force direct flag if HW A/V sync requested 420 if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) { 421 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT); 422 } 423 424 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { 425 if (audio_has_proportional_frames(format)) { 426 mFrameSize = channelCount * audio_bytes_per_sample(format); 427 } else { 428 mFrameSize = sizeof(uint8_t); 429 } 430 } else { 431 ALOG_ASSERT(audio_has_proportional_frames(format)); 432 mFrameSize = channelCount * audio_bytes_per_sample(format); 433 // createTrack will return an error if PCM format is not supported by server, 434 // so no need to check for specific PCM formats here 435 } 436 437 // sampling rate must be specified for direct outputs 438 if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) { 439 return BAD_VALUE; 440 } 441 mSampleRate = sampleRate; 442 mOriginalSampleRate = sampleRate; 443 mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT; 444 // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX 445 mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX); 446 447 // Make copy of input parameter offloadInfo so that in the future: 448 // (a) createTrack_l doesn't need it as an input parameter 449 // (b) we can support re-creation of offloaded tracks 450 if (offloadInfo != NULL) { 451 mOffloadInfoCopy = *offloadInfo; 452 mOffloadInfo = &mOffloadInfoCopy; 453 } else { 454 mOffloadInfo = NULL; 455 } 456 457 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f; 458 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f; 459 mSendLevel = 0.0f; 460 // mFrameCount is initialized in createTrack_l 461 mReqFrameCount = frameCount; 462 if (notificationFrames >= 0) { 463 mNotificationFramesReq = notificationFrames; 464 mNotificationsPerBufferReq = 0; 465 } else { 466 if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 467 ALOGE("notificationFrames=%d not permitted for non-fast track", 468 notificationFrames); 469 return BAD_VALUE; 470 } 471 if (frameCount > 0) { 472 ALOGE("notificationFrames=%d not permitted with non-zero frameCount=%zu", 473 notificationFrames, frameCount); 474 return BAD_VALUE; 475 } 476 mNotificationFramesReq = 0; 477 const uint32_t minNotificationsPerBuffer = 1; 478 const uint32_t maxNotificationsPerBuffer = 8; 479 mNotificationsPerBufferReq = min(maxNotificationsPerBuffer, 480 max((uint32_t) -notificationFrames, minNotificationsPerBuffer)); 481 ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames, 482 "notificationFrames=%d clamped to the range -%u to -%u", 483 notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer); 484 } 485 mNotificationFramesAct = 0; 486 if (sessionId == AUDIO_SESSION_ALLOCATE) { 487 mSessionId = (audio_session_t) AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 488 } else { 489 mSessionId = sessionId; 490 } 491 int callingpid = IPCThreadState::self()->getCallingPid(); 492 int mypid = getpid(); 493 if (uid == AUDIO_UID_INVALID || (callingpid != mypid)) { 494 mClientUid = IPCThreadState::self()->getCallingUid(); 495 } else { 496 mClientUid = uid; 497 } 498 if (pid == -1 || (callingpid != mypid)) { 499 mClientPid = callingpid; 500 } else { 501 mClientPid = pid; 502 } 503 mAuxEffectId = 0; 504 mOrigFlags = mFlags = flags; 505 mCbf = cbf; 506 507 if (cbf != NULL) { 508 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 509 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 510 // thread begins in paused state, and will not reference us until start() 511 } 512 513 // create the IAudioTrack 514 status_t status = createTrack_l(); 515 516 if (status != NO_ERROR) { 517 if (mAudioTrackThread != 0) { 518 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 519 mAudioTrackThread->requestExitAndWait(); 520 mAudioTrackThread.clear(); 521 } 522 return status; 523 } 524 525 mStatus = NO_ERROR; 526 mUserData = user; 527 mLoopCount = 0; 528 mLoopStart = 0; 529 mLoopEnd = 0; 530 mLoopCountNotified = 0; 531 mMarkerPosition = 0; 532 mMarkerReached = false; 533 mNewPosition = 0; 534 mUpdatePeriod = 0; 535 mPosition = 0; 536 mReleased = 0; 537 mStartUs = 0; 538 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 539 mSequence = 1; 540 mObservedSequence = mSequence; 541 mInUnderrun = false; 542 mPreviousTimestampValid = false; 543 mTimestampStartupGlitchReported = false; 544 mRetrogradeMotionReported = false; 545 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID; 546 mStartTs.mPosition = 0; 547 mUnderrunCountOffset = 0; 548 mFramesWritten = 0; 549 mFramesWrittenServerOffset = 0; 550 mFramesWrittenAtRestore = -1; // -1 is a unique initializer. 551 552 return NO_ERROR; 553} 554 555// ------------------------------------------------------------------------- 556 557status_t AudioTrack::start() 558{ 559 AutoMutex lock(mLock); 560 561 if (mState == STATE_ACTIVE) { 562 return INVALID_OPERATION; 563 } 564 565 mInUnderrun = true; 566 567 State previousState = mState; 568 if (previousState == STATE_PAUSED_STOPPING) { 569 mState = STATE_STOPPING; 570 } else { 571 mState = STATE_ACTIVE; 572 } 573 (void) updateAndGetPosition_l(); 574 575 // save start timestamp 576 if (isOffloadedOrDirect_l()) { 577 if (getTimestamp_l(mStartTs) != OK) { 578 mStartTs.mPosition = 0; 579 } 580 } else { 581 if (getTimestamp_l(&mStartEts) != OK) { 582 mStartEts.clear(); 583 } 584 } 585 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 586 // reset current position as seen by client to 0 587 mPosition = 0; 588 mPreviousTimestampValid = false; 589 mTimestampStartupGlitchReported = false; 590 mRetrogradeMotionReported = false; 591 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID; 592 593 if (!isOffloadedOrDirect_l() 594 && mStartEts.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) { 595 // Server side has consumed something, but is it finished consuming? 596 // It is possible since flush and stop are asynchronous that the server 597 // is still active at this point. 598 ALOGV("start: server read:%lld cumulative flushed:%lld client written:%lld", 599 (long long)(mFramesWrittenServerOffset 600 + mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER]), 601 (long long)mStartEts.mFlushed, 602 (long long)mFramesWritten); 603 mFramesWrittenServerOffset = -mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER]; 604 } 605 mFramesWritten = 0; 606 mProxy->clearTimestamp(); // need new server push for valid timestamp 607 mMarkerReached = false; 608 609 // For offloaded tracks, we don't know if the hardware counters are really zero here, 610 // since the flush is asynchronous and stop may not fully drain. 611 // We save the time when the track is started to later verify whether 612 // the counters are realistic (i.e. start from zero after this time). 613 mStartUs = getNowUs(); 614 615 // force refresh of remaining frames by processAudioBuffer() as last 616 // write before stop could be partial. 617 mRefreshRemaining = true; 618 } 619 mNewPosition = mPosition + mUpdatePeriod; 620 int32_t flags = android_atomic_and(~(CBLK_STREAM_END_DONE | CBLK_DISABLED), &mCblk->mFlags); 621 622 status_t status = NO_ERROR; 623 if (!(flags & CBLK_INVALID)) { 624 status = mAudioTrack->start(); 625 if (status == DEAD_OBJECT) { 626 flags |= CBLK_INVALID; 627 } 628 } 629 if (flags & CBLK_INVALID) { 630 status = restoreTrack_l("start"); 631 } 632 633 // resume or pause the callback thread as needed. 634 sp<AudioTrackThread> t = mAudioTrackThread; 635 if (status == NO_ERROR) { 636 if (t != 0) { 637 if (previousState == STATE_STOPPING) { 638 mProxy->interrupt(); 639 } else { 640 t->resume(); 641 } 642 } else { 643 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 644 get_sched_policy(0, &mPreviousSchedulingGroup); 645 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 646 } 647 } else { 648 ALOGE("start() status %d", status); 649 mState = previousState; 650 if (t != 0) { 651 if (previousState != STATE_STOPPING) { 652 t->pause(); 653 } 654 } else { 655 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 656 set_sched_policy(0, mPreviousSchedulingGroup); 657 } 658 } 659 660 return status; 661} 662 663void AudioTrack::stop() 664{ 665 AutoMutex lock(mLock); 666 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 667 return; 668 } 669 670 if (isOffloaded_l()) { 671 mState = STATE_STOPPING; 672 } else { 673 mState = STATE_STOPPED; 674 ALOGD_IF(mSharedBuffer == nullptr, 675 "stop() called with %u frames delivered", mReleased.value()); 676 mReleased = 0; 677 } 678 679 mProxy->interrupt(); 680 mAudioTrack->stop(); 681 682 // Note: legacy handling - stop does not clear playback marker 683 // and periodic update counter, but flush does for streaming tracks. 684 685 if (mSharedBuffer != 0) { 686 // clear buffer position and loop count. 687 mStaticProxy->setBufferPositionAndLoop(0 /* position */, 688 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */); 689 } 690 691 sp<AudioTrackThread> t = mAudioTrackThread; 692 if (t != 0) { 693 if (!isOffloaded_l()) { 694 t->pause(); 695 } 696 } else { 697 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 698 set_sched_policy(0, mPreviousSchedulingGroup); 699 } 700} 701 702bool AudioTrack::stopped() const 703{ 704 AutoMutex lock(mLock); 705 return mState != STATE_ACTIVE; 706} 707 708void AudioTrack::flush() 709{ 710 if (mSharedBuffer != 0) { 711 return; 712 } 713 AutoMutex lock(mLock); 714 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 715 return; 716 } 717 flush_l(); 718} 719 720void AudioTrack::flush_l() 721{ 722 ALOG_ASSERT(mState != STATE_ACTIVE); 723 724 // clear playback marker and periodic update counter 725 mMarkerPosition = 0; 726 mMarkerReached = false; 727 mUpdatePeriod = 0; 728 mRefreshRemaining = true; 729 730 mState = STATE_FLUSHED; 731 mReleased = 0; 732 if (isOffloaded_l()) { 733 mProxy->interrupt(); 734 } 735 mProxy->flush(); 736 mAudioTrack->flush(); 737} 738 739void AudioTrack::pause() 740{ 741 AutoMutex lock(mLock); 742 if (mState == STATE_ACTIVE) { 743 mState = STATE_PAUSED; 744 } else if (mState == STATE_STOPPING) { 745 mState = STATE_PAUSED_STOPPING; 746 } else { 747 return; 748 } 749 mProxy->interrupt(); 750 mAudioTrack->pause(); 751 752 if (isOffloaded_l()) { 753 if (mOutput != AUDIO_IO_HANDLE_NONE) { 754 // An offload output can be re-used between two audio tracks having 755 // the same configuration. A timestamp query for a paused track 756 // while the other is running would return an incorrect time. 757 // To fix this, cache the playback position on a pause() and return 758 // this time when requested until the track is resumed. 759 760 // OffloadThread sends HAL pause in its threadLoop. Time saved 761 // here can be slightly off. 762 763 // TODO: check return code for getRenderPosition. 764 765 uint32_t halFrames; 766 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); 767 ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); 768 } 769 } 770} 771 772status_t AudioTrack::setVolume(float left, float right) 773{ 774 // This duplicates a test by AudioTrack JNI, but that is not the only caller 775 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY || 776 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) { 777 return BAD_VALUE; 778 } 779 780 AutoMutex lock(mLock); 781 mVolume[AUDIO_INTERLEAVE_LEFT] = left; 782 mVolume[AUDIO_INTERLEAVE_RIGHT] = right; 783 784 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right))); 785 786 if (isOffloaded_l()) { 787 mAudioTrack->signal(); 788 } 789 return NO_ERROR; 790} 791 792status_t AudioTrack::setVolume(float volume) 793{ 794 return setVolume(volume, volume); 795} 796 797status_t AudioTrack::setAuxEffectSendLevel(float level) 798{ 799 // This duplicates a test by AudioTrack JNI, but that is not the only caller 800 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) { 801 return BAD_VALUE; 802 } 803 804 AutoMutex lock(mLock); 805 mSendLevel = level; 806 mProxy->setSendLevel(level); 807 808 return NO_ERROR; 809} 810 811void AudioTrack::getAuxEffectSendLevel(float* level) const 812{ 813 if (level != NULL) { 814 *level = mSendLevel; 815 } 816} 817 818status_t AudioTrack::setSampleRate(uint32_t rate) 819{ 820 AutoMutex lock(mLock); 821 if (rate == mSampleRate) { 822 return NO_ERROR; 823 } 824 if (isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 825 return INVALID_OPERATION; 826 } 827 if (mOutput == AUDIO_IO_HANDLE_NONE) { 828 return NO_INIT; 829 } 830 // NOTE: it is theoretically possible, but highly unlikely, that a device change 831 // could mean a previously allowed sampling rate is no longer allowed. 832 uint32_t afSamplingRate; 833 if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) { 834 return NO_INIT; 835 } 836 // pitch is emulated by adjusting speed and sampleRate 837 const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch); 838 if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 839 return BAD_VALUE; 840 } 841 // TODO: Should we also check if the buffer size is compatible? 842 843 mSampleRate = rate; 844 mProxy->setSampleRate(effectiveSampleRate); 845 846 return NO_ERROR; 847} 848 849uint32_t AudioTrack::getSampleRate() const 850{ 851 AutoMutex lock(mLock); 852 853 // sample rate can be updated during playback by the offloaded decoder so we need to 854 // query the HAL and update if needed. 855// FIXME use Proxy return channel to update the rate from server and avoid polling here 856 if (isOffloadedOrDirect_l()) { 857 if (mOutput != AUDIO_IO_HANDLE_NONE) { 858 uint32_t sampleRate = 0; 859 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate); 860 if (status == NO_ERROR) { 861 mSampleRate = sampleRate; 862 } 863 } 864 } 865 return mSampleRate; 866} 867 868uint32_t AudioTrack::getOriginalSampleRate() const 869{ 870 return mOriginalSampleRate; 871} 872 873status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate) 874{ 875 AutoMutex lock(mLock); 876 if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) { 877 return NO_ERROR; 878 } 879 if (isOffloadedOrDirect_l()) { 880 return INVALID_OPERATION; 881 } 882 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 883 return INVALID_OPERATION; 884 } 885 886 ALOGV("setPlaybackRate (input): mSampleRate:%u mSpeed:%f mPitch:%f", 887 mSampleRate, playbackRate.mSpeed, playbackRate.mPitch); 888 // pitch is emulated by adjusting speed and sampleRate 889 const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch); 890 const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch); 891 const float effectivePitch = adjustPitch(playbackRate.mPitch); 892 AudioPlaybackRate playbackRateTemp = playbackRate; 893 playbackRateTemp.mSpeed = effectiveSpeed; 894 playbackRateTemp.mPitch = effectivePitch; 895 896 ALOGV("setPlaybackRate (effective): mSampleRate:%u mSpeed:%f mPitch:%f", 897 effectiveRate, effectiveSpeed, effectivePitch); 898 899 if (!isAudioPlaybackRateValid(playbackRateTemp)) { 900 ALOGV("setPlaybackRate(%f, %f) failed (effective rate out of bounds)", 901 playbackRate.mSpeed, playbackRate.mPitch); 902 return BAD_VALUE; 903 } 904 // Check if the buffer size is compatible. 905 if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) { 906 ALOGV("setPlaybackRate(%f, %f) failed (buffer size)", 907 playbackRate.mSpeed, playbackRate.mPitch); 908 return BAD_VALUE; 909 } 910 911 // Check resampler ratios are within bounds 912 if ((uint64_t)effectiveRate > (uint64_t)mSampleRate * (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 913 ALOGV("setPlaybackRate(%f, %f) failed. Resample rate exceeds max accepted value", 914 playbackRate.mSpeed, playbackRate.mPitch); 915 return BAD_VALUE; 916 } 917 918 if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) { 919 ALOGV("setPlaybackRate(%f, %f) failed. Resample rate below min accepted value", 920 playbackRate.mSpeed, playbackRate.mPitch); 921 return BAD_VALUE; 922 } 923 mPlaybackRate = playbackRate; 924 //set effective rates 925 mProxy->setPlaybackRate(playbackRateTemp); 926 mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate 927 return NO_ERROR; 928} 929 930const AudioPlaybackRate& AudioTrack::getPlaybackRate() const 931{ 932 AutoMutex lock(mLock); 933 return mPlaybackRate; 934} 935 936ssize_t AudioTrack::getBufferSizeInFrames() 937{ 938 AutoMutex lock(mLock); 939 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) { 940 return NO_INIT; 941 } 942 return (ssize_t) mProxy->getBufferSizeInFrames(); 943} 944 945status_t AudioTrack::getBufferDurationInUs(int64_t *duration) 946{ 947 if (duration == nullptr) { 948 return BAD_VALUE; 949 } 950 AutoMutex lock(mLock); 951 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) { 952 return NO_INIT; 953 } 954 ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames(); 955 if (bufferSizeInFrames < 0) { 956 return (status_t)bufferSizeInFrames; 957 } 958 *duration = (int64_t)((double)bufferSizeInFrames * 1000000 959 / ((double)mSampleRate * mPlaybackRate.mSpeed)); 960 return NO_ERROR; 961} 962 963ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames) 964{ 965 AutoMutex lock(mLock); 966 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) { 967 return NO_INIT; 968 } 969 // Reject if timed track or compressed audio. 970 if (!audio_is_linear_pcm(mFormat)) { 971 return INVALID_OPERATION; 972 } 973 return (ssize_t) mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames); 974} 975 976status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 977{ 978 if (mSharedBuffer == 0 || isOffloadedOrDirect()) { 979 return INVALID_OPERATION; 980 } 981 982 if (loopCount == 0) { 983 ; 984 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 985 loopEnd - loopStart >= MIN_LOOP) { 986 ; 987 } else { 988 return BAD_VALUE; 989 } 990 991 AutoMutex lock(mLock); 992 // See setPosition() regarding setting parameters such as loop points or position while active 993 if (mState == STATE_ACTIVE) { 994 return INVALID_OPERATION; 995 } 996 setLoop_l(loopStart, loopEnd, loopCount); 997 return NO_ERROR; 998} 999 1000void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 1001{ 1002 // We do not update the periodic notification point. 1003 // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 1004 mLoopCount = loopCount; 1005 mLoopEnd = loopEnd; 1006 mLoopStart = loopStart; 1007 mLoopCountNotified = loopCount; 1008 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 1009 1010 // Waking the AudioTrackThread is not needed as this cannot be called when active. 1011} 1012 1013status_t AudioTrack::setMarkerPosition(uint32_t marker) 1014{ 1015 // The only purpose of setting marker position is to get a callback 1016 if (mCbf == NULL || isOffloadedOrDirect()) { 1017 return INVALID_OPERATION; 1018 } 1019 1020 AutoMutex lock(mLock); 1021 mMarkerPosition = marker; 1022 mMarkerReached = false; 1023 1024 sp<AudioTrackThread> t = mAudioTrackThread; 1025 if (t != 0) { 1026 t->wake(); 1027 } 1028 return NO_ERROR; 1029} 1030 1031status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 1032{ 1033 if (isOffloadedOrDirect()) { 1034 return INVALID_OPERATION; 1035 } 1036 if (marker == NULL) { 1037 return BAD_VALUE; 1038 } 1039 1040 AutoMutex lock(mLock); 1041 mMarkerPosition.getValue(marker); 1042 1043 return NO_ERROR; 1044} 1045 1046status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 1047{ 1048 // The only purpose of setting position update period is to get a callback 1049 if (mCbf == NULL || isOffloadedOrDirect()) { 1050 return INVALID_OPERATION; 1051 } 1052 1053 AutoMutex lock(mLock); 1054 mNewPosition = updateAndGetPosition_l() + updatePeriod; 1055 mUpdatePeriod = updatePeriod; 1056 1057 sp<AudioTrackThread> t = mAudioTrackThread; 1058 if (t != 0) { 1059 t->wake(); 1060 } 1061 return NO_ERROR; 1062} 1063 1064status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 1065{ 1066 if (isOffloadedOrDirect()) { 1067 return INVALID_OPERATION; 1068 } 1069 if (updatePeriod == NULL) { 1070 return BAD_VALUE; 1071 } 1072 1073 AutoMutex lock(mLock); 1074 *updatePeriod = mUpdatePeriod; 1075 1076 return NO_ERROR; 1077} 1078 1079status_t AudioTrack::setPosition(uint32_t position) 1080{ 1081 if (mSharedBuffer == 0 || isOffloadedOrDirect()) { 1082 return INVALID_OPERATION; 1083 } 1084 if (position > mFrameCount) { 1085 return BAD_VALUE; 1086 } 1087 1088 AutoMutex lock(mLock); 1089 // Currently we require that the player is inactive before setting parameters such as position 1090 // or loop points. Otherwise, there could be a race condition: the application could read the 1091 // current position, compute a new position or loop parameters, and then set that position or 1092 // loop parameters but it would do the "wrong" thing since the position has continued to advance 1093 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 1094 // to specify how it wants to handle such scenarios. 1095 if (mState == STATE_ACTIVE) { 1096 return INVALID_OPERATION; 1097 } 1098 // After setting the position, use full update period before notification. 1099 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 1100 mStaticProxy->setBufferPosition(position); 1101 1102 // Waking the AudioTrackThread is not needed as this cannot be called when active. 1103 return NO_ERROR; 1104} 1105 1106status_t AudioTrack::getPosition(uint32_t *position) 1107{ 1108 if (position == NULL) { 1109 return BAD_VALUE; 1110 } 1111 1112 AutoMutex lock(mLock); 1113 // FIXME: offloaded and direct tracks call into the HAL for render positions 1114 // for compressed/synced data; however, we use proxy position for pure linear pcm data 1115 // as we do not know the capability of the HAL for pcm position support and standby. 1116 // There may be some latency differences between the HAL position and the proxy position. 1117 if (isOffloadedOrDirect_l() && !isPurePcmData_l()) { 1118 uint32_t dspFrames = 0; 1119 1120 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) { 1121 ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); 1122 *position = mPausedPosition; 1123 return NO_ERROR; 1124 } 1125 1126 if (mOutput != AUDIO_IO_HANDLE_NONE) { 1127 uint32_t halFrames; // actually unused 1128 (void) AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 1129 // FIXME: on getRenderPosition() error, we return OK with frame position 0. 1130 } 1131 // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED) 1132 // due to hardware latency. We leave this behavior for now. 1133 *position = dspFrames; 1134 } else { 1135 if (mCblk->mFlags & CBLK_INVALID) { 1136 (void) restoreTrack_l("getPosition"); 1137 // FIXME: for compatibility with the Java API we ignore the restoreTrack_l() 1138 // error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position. 1139 } 1140 1141 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 1142 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 1143 0 : updateAndGetPosition_l().value(); 1144 } 1145 return NO_ERROR; 1146} 1147 1148status_t AudioTrack::getBufferPosition(uint32_t *position) 1149{ 1150 if (mSharedBuffer == 0) { 1151 return INVALID_OPERATION; 1152 } 1153 if (position == NULL) { 1154 return BAD_VALUE; 1155 } 1156 1157 AutoMutex lock(mLock); 1158 *position = mStaticProxy->getBufferPosition(); 1159 return NO_ERROR; 1160} 1161 1162status_t AudioTrack::reload() 1163{ 1164 if (mSharedBuffer == 0 || isOffloadedOrDirect()) { 1165 return INVALID_OPERATION; 1166 } 1167 1168 AutoMutex lock(mLock); 1169 // See setPosition() regarding setting parameters such as loop points or position while active 1170 if (mState == STATE_ACTIVE) { 1171 return INVALID_OPERATION; 1172 } 1173 mNewPosition = mUpdatePeriod; 1174 (void) updateAndGetPosition_l(); 1175 mPosition = 0; 1176 mPreviousTimestampValid = false; 1177#if 0 1178 // The documentation is not clear on the behavior of reload() and the restoration 1179 // of loop count. Historically we have not restored loop count, start, end, 1180 // but it makes sense if one desires to repeat playing a particular sound. 1181 if (mLoopCount != 0) { 1182 mLoopCountNotified = mLoopCount; 1183 mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount); 1184 } 1185#endif 1186 mStaticProxy->setBufferPosition(0); 1187 return NO_ERROR; 1188} 1189 1190audio_io_handle_t AudioTrack::getOutput() const 1191{ 1192 AutoMutex lock(mLock); 1193 return mOutput; 1194} 1195 1196status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) { 1197 AutoMutex lock(mLock); 1198 if (mSelectedDeviceId != deviceId) { 1199 mSelectedDeviceId = deviceId; 1200 android_atomic_or(CBLK_INVALID, &mCblk->mFlags); 1201 } 1202 return NO_ERROR; 1203} 1204 1205audio_port_handle_t AudioTrack::getOutputDevice() { 1206 AutoMutex lock(mLock); 1207 return mSelectedDeviceId; 1208} 1209 1210audio_port_handle_t AudioTrack::getRoutedDeviceId() { 1211 AutoMutex lock(mLock); 1212 if (mOutput == AUDIO_IO_HANDLE_NONE) { 1213 return AUDIO_PORT_HANDLE_NONE; 1214 } 1215 return AudioSystem::getDeviceIdForIo(mOutput); 1216} 1217 1218status_t AudioTrack::attachAuxEffect(int effectId) 1219{ 1220 AutoMutex lock(mLock); 1221 status_t status = mAudioTrack->attachAuxEffect(effectId); 1222 if (status == NO_ERROR) { 1223 mAuxEffectId = effectId; 1224 } 1225 return status; 1226} 1227 1228audio_stream_type_t AudioTrack::streamType() const 1229{ 1230 if (mStreamType == AUDIO_STREAM_DEFAULT) { 1231 return audio_attributes_to_stream_type(&mAttributes); 1232 } 1233 return mStreamType; 1234} 1235 1236// ------------------------------------------------------------------------- 1237 1238// must be called with mLock held 1239status_t AudioTrack::createTrack_l() 1240{ 1241 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 1242 if (audioFlinger == 0) { 1243 ALOGE("Could not get audioflinger"); 1244 return NO_INIT; 1245 } 1246 1247 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) { 1248 AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); 1249 } 1250 audio_io_handle_t output; 1251 audio_stream_type_t streamType = mStreamType; 1252 audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL; 1253 1254 // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted. 1255 // After fast request is denied, we will request again if IAudioTrack is re-created. 1256 1257 status_t status; 1258 status = AudioSystem::getOutputForAttr(attr, &output, 1259 mSessionId, &streamType, mClientUid, 1260 mSampleRate, mFormat, mChannelMask, 1261 mFlags, mSelectedDeviceId, mOffloadInfo); 1262 1263 if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) { 1264 ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u, format %#x," 1265 " channel mask %#x, flags %#x", 1266 mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags); 1267 return BAD_VALUE; 1268 } 1269 { 1270 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 1271 // we must release it ourselves if anything goes wrong. 1272 1273 // Not all of these values are needed under all conditions, but it is easier to get them all 1274 status = AudioSystem::getLatency(output, &mAfLatency); 1275 if (status != NO_ERROR) { 1276 ALOGE("getLatency(%d) failed status %d", output, status); 1277 goto release; 1278 } 1279 ALOGV("createTrack_l() output %d afLatency %u", output, mAfLatency); 1280 1281 status = AudioSystem::getFrameCount(output, &mAfFrameCount); 1282 if (status != NO_ERROR) { 1283 ALOGE("getFrameCount(output=%d) status %d", output, status); 1284 goto release; 1285 } 1286 1287 // TODO consider making this a member variable if there are other uses for it later 1288 size_t afFrameCountHAL; 1289 status = AudioSystem::getFrameCountHAL(output, &afFrameCountHAL); 1290 if (status != NO_ERROR) { 1291 ALOGE("getFrameCountHAL(output=%d) status %d", output, status); 1292 goto release; 1293 } 1294 ALOG_ASSERT(afFrameCountHAL > 0); 1295 1296 status = AudioSystem::getSamplingRate(output, &mAfSampleRate); 1297 if (status != NO_ERROR) { 1298 ALOGE("getSamplingRate(output=%d) status %d", output, status); 1299 goto release; 1300 } 1301 if (mSampleRate == 0) { 1302 mSampleRate = mAfSampleRate; 1303 mOriginalSampleRate = mAfSampleRate; 1304 } 1305 1306 // Client can only express a preference for FAST. Server will perform additional tests. 1307 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1308 bool useCaseAllowed = 1309 // either of these use cases: 1310 // use case 1: shared buffer 1311 (mSharedBuffer != 0) || 1312 // use case 2: callback transfer mode 1313 (mTransfer == TRANSFER_CALLBACK) || 1314 // use case 3: obtain/release mode 1315 (mTransfer == TRANSFER_OBTAIN) || 1316 // use case 4: synchronous write 1317 ((mTransfer == TRANSFER_SYNC) && mThreadCanCallJava); 1318 // sample rates must also match 1319 bool fastAllowed = useCaseAllowed && (mSampleRate == mAfSampleRate); 1320 if (!fastAllowed) { 1321 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client; transfer %d, " 1322 "track %u Hz, output %u Hz", 1323 mTransfer, mSampleRate, mAfSampleRate); 1324 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1325 } 1326 } 1327 1328 mNotificationFramesAct = mNotificationFramesReq; 1329 1330 size_t frameCount = mReqFrameCount; 1331 if (!audio_has_proportional_frames(mFormat)) { 1332 1333 if (mSharedBuffer != 0) { 1334 // Same comment as below about ignoring frameCount parameter for set() 1335 frameCount = mSharedBuffer->size(); 1336 } else if (frameCount == 0) { 1337 frameCount = mAfFrameCount; 1338 } 1339 if (mNotificationFramesAct != frameCount) { 1340 mNotificationFramesAct = frameCount; 1341 } 1342 } else if (mSharedBuffer != 0) { 1343 // FIXME: Ensure client side memory buffers need 1344 // not have additional alignment beyond sample 1345 // (e.g. 16 bit stereo accessed as 32 bit frame). 1346 size_t alignment = audio_bytes_per_sample(mFormat); 1347 if (alignment & 1) { 1348 // for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java). 1349 alignment = 1; 1350 } 1351 if (mChannelCount > 1) { 1352 // More than 2 channels does not require stronger alignment than stereo 1353 alignment <<= 1; 1354 } 1355 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 1356 ALOGE("Invalid buffer alignment: address %p, channel count %u", 1357 mSharedBuffer->pointer(), mChannelCount); 1358 status = BAD_VALUE; 1359 goto release; 1360 } 1361 1362 // When initializing a shared buffer AudioTrack via constructors, 1363 // there's no frameCount parameter. 1364 // But when initializing a shared buffer AudioTrack via set(), 1365 // there _is_ a frameCount parameter. We silently ignore it. 1366 frameCount = mSharedBuffer->size() / mFrameSize; 1367 } else { 1368 size_t minFrameCount = 0; 1369 // For fast tracks the frame count calculations and checks are mostly done by server, 1370 // but we try to respect the application's request for notifications per buffer. 1371 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1372 if (mNotificationsPerBufferReq > 0) { 1373 // Avoid possible arithmetic overflow during multiplication. 1374 // mNotificationsPerBuffer is clamped to a small integer earlier, so it is unlikely. 1375 if (mNotificationsPerBufferReq > SIZE_MAX / afFrameCountHAL) { 1376 ALOGE("Requested notificationPerBuffer=%u ignored for HAL frameCount=%zu", 1377 mNotificationsPerBufferReq, afFrameCountHAL); 1378 } else { 1379 minFrameCount = afFrameCountHAL * mNotificationsPerBufferReq; 1380 } 1381 } 1382 } else { 1383 // for normal tracks precompute the frame count based on speed. 1384 const float speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f : 1385 max(mMaxRequiredSpeed, mPlaybackRate.mSpeed); 1386 minFrameCount = calculateMinFrameCount( 1387 mAfLatency, mAfFrameCount, mAfSampleRate, mSampleRate, 1388 speed /*, 0 mNotificationsPerBufferReq*/); 1389 } 1390 if (frameCount < minFrameCount) { 1391 frameCount = minFrameCount; 1392 } 1393 } 1394 1395 audio_output_flags_t flags = mFlags; 1396 1397 pid_t tid = -1; 1398 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1399 if (mAudioTrackThread != 0 && !mThreadCanCallJava) { 1400 tid = mAudioTrackThread->getTid(); 1401 } 1402 } 1403 1404 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1405 // but we will still need the original value also 1406 audio_session_t originalSessionId = mSessionId; 1407 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 1408 mSampleRate, 1409 mFormat, 1410 mChannelMask, 1411 &temp, 1412 &flags, 1413 mSharedBuffer, 1414 output, 1415 mClientPid, 1416 tid, 1417 &mSessionId, 1418 mClientUid, 1419 &status); 1420 ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId, 1421 "session ID changed from %d to %d", originalSessionId, mSessionId); 1422 1423 if (status != NO_ERROR) { 1424 ALOGE("AudioFlinger could not create track, status: %d", status); 1425 goto release; 1426 } 1427 ALOG_ASSERT(track != 0); 1428 1429 // AudioFlinger now owns the reference to the I/O handle, 1430 // so we are no longer responsible for releasing it. 1431 1432 // FIXME compare to AudioRecord 1433 sp<IMemory> iMem = track->getCblk(); 1434 if (iMem == 0) { 1435 ALOGE("Could not get control block"); 1436 return NO_INIT; 1437 } 1438 void *iMemPointer = iMem->pointer(); 1439 if (iMemPointer == NULL) { 1440 ALOGE("Could not get control block pointer"); 1441 return NO_INIT; 1442 } 1443 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1444 if (mAudioTrack != 0) { 1445 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this); 1446 mDeathNotifier.clear(); 1447 } 1448 mAudioTrack = track; 1449 mCblkMemory = iMem; 1450 IPCThreadState::self()->flushCommands(); 1451 1452 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1453 mCblk = cblk; 1454 // note that temp is the (possibly revised) value of frameCount 1455 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1456 // In current design, AudioTrack client checks and ensures frame count validity before 1457 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1458 // for fast track as it uses a special method of assigning frame count. 1459 ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp); 1460 } 1461 frameCount = temp; 1462 1463 mAwaitBoost = false; 1464 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1465 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 1466 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount); 1467 if (!mThreadCanCallJava) { 1468 mAwaitBoost = true; 1469 } 1470 } else { 1471 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount); 1472 } 1473 } 1474 mFlags = flags; 1475 1476 // Make sure that application is notified with sufficient margin before underrun. 1477 // The client can divide the AudioTrack buffer into sub-buffers, 1478 // and expresses its desire to server as the notification frame count. 1479 if (mSharedBuffer == 0 && audio_is_linear_pcm(mFormat)) { 1480 size_t maxNotificationFrames; 1481 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1482 // notify every HAL buffer, regardless of the size of the track buffer 1483 maxNotificationFrames = afFrameCountHAL; 1484 } else { 1485 // For normal tracks, use at least double-buffering if no sample rate conversion, 1486 // or at least triple-buffering if there is sample rate conversion 1487 const int nBuffering = mOriginalSampleRate == mAfSampleRate ? 2 : 3; 1488 maxNotificationFrames = frameCount / nBuffering; 1489 } 1490 if (mNotificationFramesAct == 0 || mNotificationFramesAct > maxNotificationFrames) { 1491 if (mNotificationFramesAct == 0) { 1492 ALOGD("Client defaulted notificationFrames to %zu for frameCount %zu", 1493 maxNotificationFrames, frameCount); 1494 } else { 1495 ALOGW("Client adjusted notificationFrames from %u to %zu for frameCount %zu", 1496 mNotificationFramesAct, maxNotificationFrames, frameCount); 1497 } 1498 mNotificationFramesAct = (uint32_t) maxNotificationFrames; 1499 } 1500 } 1501 1502 // We retain a copy of the I/O handle, but don't own the reference 1503 mOutput = output; 1504 mRefreshRemaining = true; 1505 1506 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1507 // is the value of pointer() for the shared buffer, otherwise buffers points 1508 // immediately after the control block. This address is for the mapping within client 1509 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1510 void* buffers; 1511 if (mSharedBuffer == 0) { 1512 buffers = cblk + 1; 1513 } else { 1514 buffers = mSharedBuffer->pointer(); 1515 if (buffers == NULL) { 1516 ALOGE("Could not get buffer pointer"); 1517 return NO_INIT; 1518 } 1519 } 1520 1521 mAudioTrack->attachAuxEffect(mAuxEffectId); 1522 // FIXME doesn't take into account speed or future sample rate changes (until restoreTrack) 1523 // FIXME don't believe this lie 1524 mLatency = mAfLatency + (1000*frameCount) / mSampleRate; 1525 1526 mFrameCount = frameCount; 1527 // If IAudioTrack is re-created, don't let the requested frameCount 1528 // decrease. This can confuse clients that cache frameCount(). 1529 if (frameCount > mReqFrameCount) { 1530 mReqFrameCount = frameCount; 1531 } 1532 1533 // reset server position to 0 as we have new cblk. 1534 mServer = 0; 1535 1536 // update proxy 1537 if (mSharedBuffer == 0) { 1538 mStaticProxy.clear(); 1539 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize); 1540 } else { 1541 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize); 1542 mProxy = mStaticProxy; 1543 } 1544 1545 mProxy->setVolumeLR(gain_minifloat_pack( 1546 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]), 1547 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT]))); 1548 1549 mProxy->setSendLevel(mSendLevel); 1550 const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch); 1551 const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch); 1552 const float effectivePitch = adjustPitch(mPlaybackRate.mPitch); 1553 mProxy->setSampleRate(effectiveSampleRate); 1554 1555 AudioPlaybackRate playbackRateTemp = mPlaybackRate; 1556 playbackRateTemp.mSpeed = effectiveSpeed; 1557 playbackRateTemp.mPitch = effectivePitch; 1558 mProxy->setPlaybackRate(playbackRateTemp); 1559 mProxy->setMinimum(mNotificationFramesAct); 1560 1561 mDeathNotifier = new DeathNotifier(this); 1562 IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this); 1563 1564 if (mDeviceCallback != 0) { 1565 AudioSystem::addAudioDeviceCallback(mDeviceCallback, mOutput); 1566 } 1567 1568 return NO_ERROR; 1569 } 1570 1571release: 1572 AudioSystem::releaseOutput(output, streamType, mSessionId); 1573 if (status == NO_ERROR) { 1574 status = NO_INIT; 1575 } 1576 return status; 1577} 1578 1579status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig) 1580{ 1581 if (audioBuffer == NULL) { 1582 if (nonContig != NULL) { 1583 *nonContig = 0; 1584 } 1585 return BAD_VALUE; 1586 } 1587 if (mTransfer != TRANSFER_OBTAIN) { 1588 audioBuffer->frameCount = 0; 1589 audioBuffer->size = 0; 1590 audioBuffer->raw = NULL; 1591 if (nonContig != NULL) { 1592 *nonContig = 0; 1593 } 1594 return INVALID_OPERATION; 1595 } 1596 1597 const struct timespec *requested; 1598 struct timespec timeout; 1599 if (waitCount == -1) { 1600 requested = &ClientProxy::kForever; 1601 } else if (waitCount == 0) { 1602 requested = &ClientProxy::kNonBlocking; 1603 } else if (waitCount > 0) { 1604 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1605 timeout.tv_sec = ms / 1000; 1606 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1607 requested = &timeout; 1608 } else { 1609 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1610 requested = NULL; 1611 } 1612 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig); 1613} 1614 1615status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1616 struct timespec *elapsed, size_t *nonContig) 1617{ 1618 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1619 uint32_t oldSequence = 0; 1620 uint32_t newSequence; 1621 1622 Proxy::Buffer buffer; 1623 status_t status = NO_ERROR; 1624 1625 static const int32_t kMaxTries = 5; 1626 int32_t tryCounter = kMaxTries; 1627 1628 do { 1629 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1630 // keep them from going away if another thread re-creates the track during obtainBuffer() 1631 sp<AudioTrackClientProxy> proxy; 1632 sp<IMemory> iMem; 1633 1634 { // start of lock scope 1635 AutoMutex lock(mLock); 1636 1637 newSequence = mSequence; 1638 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1639 if (status == DEAD_OBJECT) { 1640 // re-create track, unless someone else has already done so 1641 if (newSequence == oldSequence) { 1642 status = restoreTrack_l("obtainBuffer"); 1643 if (status != NO_ERROR) { 1644 buffer.mFrameCount = 0; 1645 buffer.mRaw = NULL; 1646 buffer.mNonContig = 0; 1647 break; 1648 } 1649 } 1650 } 1651 oldSequence = newSequence; 1652 1653 if (status == NOT_ENOUGH_DATA) { 1654 restartIfDisabled(); 1655 } 1656 1657 // Keep the extra references 1658 proxy = mProxy; 1659 iMem = mCblkMemory; 1660 1661 if (mState == STATE_STOPPING) { 1662 status = -EINTR; 1663 buffer.mFrameCount = 0; 1664 buffer.mRaw = NULL; 1665 buffer.mNonContig = 0; 1666 break; 1667 } 1668 1669 // Non-blocking if track is stopped or paused 1670 if (mState != STATE_ACTIVE) { 1671 requested = &ClientProxy::kNonBlocking; 1672 } 1673 1674 } // end of lock scope 1675 1676 buffer.mFrameCount = audioBuffer->frameCount; 1677 // FIXME starts the requested timeout and elapsed over from scratch 1678 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1679 } while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0)); 1680 1681 audioBuffer->frameCount = buffer.mFrameCount; 1682 audioBuffer->size = buffer.mFrameCount * mFrameSize; 1683 audioBuffer->raw = buffer.mRaw; 1684 if (nonContig != NULL) { 1685 *nonContig = buffer.mNonContig; 1686 } 1687 return status; 1688} 1689 1690void AudioTrack::releaseBuffer(const Buffer* audioBuffer) 1691{ 1692 // FIXME add error checking on mode, by adding an internal version 1693 if (mTransfer == TRANSFER_SHARED) { 1694 return; 1695 } 1696 1697 size_t stepCount = audioBuffer->size / mFrameSize; 1698 if (stepCount == 0) { 1699 return; 1700 } 1701 1702 Proxy::Buffer buffer; 1703 buffer.mFrameCount = stepCount; 1704 buffer.mRaw = audioBuffer->raw; 1705 1706 AutoMutex lock(mLock); 1707 mReleased += stepCount; 1708 mInUnderrun = false; 1709 mProxy->releaseBuffer(&buffer); 1710 1711 // restart track if it was disabled by audioflinger due to previous underrun 1712 restartIfDisabled(); 1713} 1714 1715void AudioTrack::restartIfDisabled() 1716{ 1717 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 1718 if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) { 1719 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1720 // FIXME ignoring status 1721 mAudioTrack->start(); 1722 } 1723} 1724 1725// ------------------------------------------------------------------------- 1726 1727ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1728{ 1729 if (mTransfer != TRANSFER_SYNC) { 1730 return INVALID_OPERATION; 1731 } 1732 1733 if (isDirect()) { 1734 AutoMutex lock(mLock); 1735 int32_t flags = android_atomic_and( 1736 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), 1737 &mCblk->mFlags); 1738 if (flags & CBLK_INVALID) { 1739 return DEAD_OBJECT; 1740 } 1741 } 1742 1743 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1744 // Sanity-check: user is most-likely passing an error code, and it would 1745 // make the return value ambiguous (actualSize vs error). 1746 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1747 return BAD_VALUE; 1748 } 1749 1750 size_t written = 0; 1751 Buffer audioBuffer; 1752 1753 while (userSize >= mFrameSize) { 1754 audioBuffer.frameCount = userSize / mFrameSize; 1755 1756 status_t err = obtainBuffer(&audioBuffer, 1757 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1758 if (err < 0) { 1759 if (written > 0) { 1760 break; 1761 } 1762 if (err == TIMED_OUT || err == -EINTR) { 1763 err = WOULD_BLOCK; 1764 } 1765 return ssize_t(err); 1766 } 1767 1768 size_t toWrite = audioBuffer.size; 1769 memcpy(audioBuffer.i8, buffer, toWrite); 1770 buffer = ((const char *) buffer) + toWrite; 1771 userSize -= toWrite; 1772 written += toWrite; 1773 1774 releaseBuffer(&audioBuffer); 1775 } 1776 1777 if (written > 0) { 1778 mFramesWritten += written / mFrameSize; 1779 } 1780 return written; 1781} 1782 1783// ------------------------------------------------------------------------- 1784 1785nsecs_t AudioTrack::processAudioBuffer() 1786{ 1787 // Currently the AudioTrack thread is not created if there are no callbacks. 1788 // Would it ever make sense to run the thread, even without callbacks? 1789 // If so, then replace this by checks at each use for mCbf != NULL. 1790 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1791 1792 mLock.lock(); 1793 if (mAwaitBoost) { 1794 mAwaitBoost = false; 1795 mLock.unlock(); 1796 static const int32_t kMaxTries = 5; 1797 int32_t tryCounter = kMaxTries; 1798 uint32_t pollUs = 10000; 1799 do { 1800 int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK; 1801 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1802 break; 1803 } 1804 usleep(pollUs); 1805 pollUs <<= 1; 1806 } while (tryCounter-- > 0); 1807 if (tryCounter < 0) { 1808 ALOGE("did not receive expected priority boost on time"); 1809 } 1810 // Run again immediately 1811 return 0; 1812 } 1813 1814 // Can only reference mCblk while locked 1815 int32_t flags = android_atomic_and( 1816 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1817 1818 // Check for track invalidation 1819 if (flags & CBLK_INVALID) { 1820 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1821 // AudioSystem cache. We should not exit here but after calling the callback so 1822 // that the upper layers can recreate the track 1823 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) { 1824 status_t status __unused = restoreTrack_l("processAudioBuffer"); 1825 // FIXME unused status 1826 // after restoration, continue below to make sure that the loop and buffer events 1827 // are notified because they have been cleared from mCblk->mFlags above. 1828 } 1829 } 1830 1831 bool waitStreamEnd = mState == STATE_STOPPING; 1832 bool active = mState == STATE_ACTIVE; 1833 1834 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1835 bool newUnderrun = false; 1836 if (flags & CBLK_UNDERRUN) { 1837#if 0 1838 // Currently in shared buffer mode, when the server reaches the end of buffer, 1839 // the track stays active in continuous underrun state. It's up to the application 1840 // to pause or stop the track, or set the position to a new offset within buffer. 1841 // This was some experimental code to auto-pause on underrun. Keeping it here 1842 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1843 if (mTransfer == TRANSFER_SHARED) { 1844 mState = STATE_PAUSED; 1845 active = false; 1846 } 1847#endif 1848 if (!mInUnderrun) { 1849 mInUnderrun = true; 1850 newUnderrun = true; 1851 } 1852 } 1853 1854 // Get current position of server 1855 Modulo<uint32_t> position(updateAndGetPosition_l()); 1856 1857 // Manage marker callback 1858 bool markerReached = false; 1859 Modulo<uint32_t> markerPosition(mMarkerPosition); 1860 // uses 32 bit wraparound for comparison with position. 1861 if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) { 1862 mMarkerReached = markerReached = true; 1863 } 1864 1865 // Determine number of new position callback(s) that will be needed, while locked 1866 size_t newPosCount = 0; 1867 Modulo<uint32_t> newPosition(mNewPosition); 1868 uint32_t updatePeriod = mUpdatePeriod; 1869 // FIXME fails for wraparound, need 64 bits 1870 if (updatePeriod > 0 && position >= newPosition) { 1871 newPosCount = ((position - newPosition).value() / updatePeriod) + 1; 1872 mNewPosition += updatePeriod * newPosCount; 1873 } 1874 1875 // Cache other fields that will be needed soon 1876 uint32_t sampleRate = mSampleRate; 1877 float speed = mPlaybackRate.mSpeed; 1878 const uint32_t notificationFrames = mNotificationFramesAct; 1879 if (mRefreshRemaining) { 1880 mRefreshRemaining = false; 1881 mRemainingFrames = notificationFrames; 1882 mRetryOnPartialBuffer = false; 1883 } 1884 size_t misalignment = mProxy->getMisalignment(); 1885 uint32_t sequence = mSequence; 1886 sp<AudioTrackClientProxy> proxy = mProxy; 1887 1888 // Determine the number of new loop callback(s) that will be needed, while locked. 1889 int loopCountNotifications = 0; 1890 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END 1891 1892 if (mLoopCount > 0) { 1893 int loopCount; 1894 size_t bufferPosition; 1895 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount); 1896 loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition; 1897 loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications); 1898 mLoopCountNotified = loopCount; // discard any excess notifications 1899 } else if (mLoopCount < 0) { 1900 // FIXME: We're not accurate with notification count and position with infinite looping 1901 // since loopCount from server side will always return -1 (we could decrement it). 1902 size_t bufferPosition = mStaticProxy->getBufferPosition(); 1903 loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0); 1904 loopPeriod = mLoopEnd - bufferPosition; 1905 } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) { 1906 size_t bufferPosition = mStaticProxy->getBufferPosition(); 1907 loopPeriod = mFrameCount - bufferPosition; 1908 } 1909 1910 // These fields don't need to be cached, because they are assigned only by set(): 1911 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFlags 1912 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1913 1914 mLock.unlock(); 1915 1916 // get anchor time to account for callbacks. 1917 const nsecs_t timeBeforeCallbacks = systemTime(); 1918 1919 if (waitStreamEnd) { 1920 // FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread 1921 // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function 1922 // (and make sure we don't callback for more data while we're stopping). 1923 // This helps with position, marker notifications, and track invalidation. 1924 struct timespec timeout; 1925 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1926 timeout.tv_nsec = 0; 1927 1928 status_t status = proxy->waitStreamEndDone(&timeout); 1929 switch (status) { 1930 case NO_ERROR: 1931 case DEAD_OBJECT: 1932 case TIMED_OUT: 1933 if (status != DEAD_OBJECT) { 1934 // for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop(); 1935 // instead, the application should handle the EVENT_NEW_IAUDIOTRACK. 1936 mCbf(EVENT_STREAM_END, mUserData, NULL); 1937 } 1938 { 1939 AutoMutex lock(mLock); 1940 // The previously assigned value of waitStreamEnd is no longer valid, 1941 // since the mutex has been unlocked and either the callback handler 1942 // or another thread could have re-started the AudioTrack during that time. 1943 waitStreamEnd = mState == STATE_STOPPING; 1944 if (waitStreamEnd) { 1945 mState = STATE_STOPPED; 1946 mReleased = 0; 1947 } 1948 } 1949 if (waitStreamEnd && status != DEAD_OBJECT) { 1950 return NS_INACTIVE; 1951 } 1952 break; 1953 } 1954 return 0; 1955 } 1956 1957 // perform callbacks while unlocked 1958 if (newUnderrun) { 1959 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1960 } 1961 while (loopCountNotifications > 0) { 1962 mCbf(EVENT_LOOP_END, mUserData, NULL); 1963 --loopCountNotifications; 1964 } 1965 if (flags & CBLK_BUFFER_END) { 1966 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1967 } 1968 if (markerReached) { 1969 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1970 } 1971 while (newPosCount > 0) { 1972 size_t temp = newPosition.value(); // FIXME size_t != uint32_t 1973 mCbf(EVENT_NEW_POS, mUserData, &temp); 1974 newPosition += updatePeriod; 1975 newPosCount--; 1976 } 1977 1978 if (mObservedSequence != sequence) { 1979 mObservedSequence = sequence; 1980 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1981 // for offloaded tracks, just wait for the upper layers to recreate the track 1982 if (isOffloadedOrDirect()) { 1983 return NS_INACTIVE; 1984 } 1985 } 1986 1987 // if inactive, then don't run me again until re-started 1988 if (!active) { 1989 return NS_INACTIVE; 1990 } 1991 1992 // Compute the estimated time until the next timed event (position, markers, loops) 1993 // FIXME only for non-compressed audio 1994 uint32_t minFrames = ~0; 1995 if (!markerReached && position < markerPosition) { 1996 minFrames = (markerPosition - position).value(); 1997 } 1998 if (loopPeriod > 0 && loopPeriod < minFrames) { 1999 // loopPeriod is already adjusted for actual position. 2000 minFrames = loopPeriod; 2001 } 2002 if (updatePeriod > 0) { 2003 minFrames = min(minFrames, (newPosition - position).value()); 2004 } 2005 2006 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 2007 static const uint32_t kPoll = 0; 2008 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 2009 minFrames = kPoll * notificationFrames; 2010 } 2011 2012 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 2013 static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL; 2014 const nsecs_t timeAfterCallbacks = systemTime(); 2015 2016 // Convert frame units to time units 2017 nsecs_t ns = NS_WHENEVER; 2018 if (minFrames != (uint32_t) ~0) { 2019 ns = framesToNanoseconds(minFrames, sampleRate, speed) + kWaitPeriodNs; 2020 ns -= (timeAfterCallbacks - timeBeforeCallbacks); // account for callback time 2021 // TODO: Should we warn if the callback time is too long? 2022 if (ns < 0) ns = 0; 2023 } 2024 2025 // If not supplying data by EVENT_MORE_DATA, then we're done 2026 if (mTransfer != TRANSFER_CALLBACK) { 2027 return ns; 2028 } 2029 2030 // EVENT_MORE_DATA callback handling. 2031 // Timing for linear pcm audio data formats can be derived directly from the 2032 // buffer fill level. 2033 // Timing for compressed data is not directly available from the buffer fill level, 2034 // rather indirectly from waiting for blocking mode callbacks or waiting for obtain() 2035 // to return a certain fill level. 2036 2037 struct timespec timeout; 2038 const struct timespec *requested = &ClientProxy::kForever; 2039 if (ns != NS_WHENEVER) { 2040 timeout.tv_sec = ns / 1000000000LL; 2041 timeout.tv_nsec = ns % 1000000000LL; 2042 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 2043 requested = &timeout; 2044 } 2045 2046 size_t writtenFrames = 0; 2047 while (mRemainingFrames > 0) { 2048 2049 Buffer audioBuffer; 2050 audioBuffer.frameCount = mRemainingFrames; 2051 size_t nonContig; 2052 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 2053 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 2054 "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount); 2055 requested = &ClientProxy::kNonBlocking; 2056 size_t avail = audioBuffer.frameCount + nonContig; 2057 ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d", 2058 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 2059 if (err != NO_ERROR) { 2060 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 2061 (isOffloaded() && (err == DEAD_OBJECT))) { 2062 // FIXME bug 25195759 2063 return 1000000; 2064 } 2065 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 2066 return NS_NEVER; 2067 } 2068 2069 if (mRetryOnPartialBuffer && audio_has_proportional_frames(mFormat)) { 2070 mRetryOnPartialBuffer = false; 2071 if (avail < mRemainingFrames) { 2072 if (ns > 0) { // account for obtain time 2073 const nsecs_t timeNow = systemTime(); 2074 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks)); 2075 } 2076 nsecs_t myns = framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed); 2077 if (ns < 0 /* NS_WHENEVER */ || myns < ns) { 2078 ns = myns; 2079 } 2080 return ns; 2081 } 2082 } 2083 2084 size_t reqSize = audioBuffer.size; 2085 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 2086 size_t writtenSize = audioBuffer.size; 2087 2088 // Sanity check on returned size 2089 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 2090 ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes", 2091 reqSize, ssize_t(writtenSize)); 2092 return NS_NEVER; 2093 } 2094 2095 if (writtenSize == 0) { 2096 // The callback is done filling buffers 2097 // Keep this thread going to handle timed events and 2098 // still try to get more data in intervals of WAIT_PERIOD_MS 2099 // but don't just loop and block the CPU, so wait 2100 2101 // mCbf(EVENT_MORE_DATA, ...) might either 2102 // (1) Block until it can fill the buffer, returning 0 size on EOS. 2103 // (2) Block until it can fill the buffer, returning 0 data (silence) on EOS. 2104 // (3) Return 0 size when no data is available, does not wait for more data. 2105 // 2106 // (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer. 2107 // We try to compute the wait time to avoid a tight sleep-wait cycle, 2108 // especially for case (3). 2109 // 2110 // The decision to support (1) and (2) affect the sizing of mRemainingFrames 2111 // and this loop; whereas for case (3) we could simply check once with the full 2112 // buffer size and skip the loop entirely. 2113 2114 nsecs_t myns; 2115 if (audio_has_proportional_frames(mFormat)) { 2116 // time to wait based on buffer occupancy 2117 const nsecs_t datans = mRemainingFrames <= avail ? 0 : 2118 framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed); 2119 // audio flinger thread buffer size (TODO: adjust for fast tracks) 2120 // FIXME: use mAfFrameCountHAL instead of mAfFrameCount below for fast tracks. 2121 const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed); 2122 // add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0. 2123 myns = datans + (afns / 2); 2124 } else { 2125 // FIXME: This could ping quite a bit if the buffer isn't full. 2126 // Note that when mState is stopping we waitStreamEnd, so it never gets here. 2127 myns = kWaitPeriodNs; 2128 } 2129 if (ns > 0) { // account for obtain and callback time 2130 const nsecs_t timeNow = systemTime(); 2131 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks)); 2132 } 2133 if (ns < 0 /* NS_WHENEVER */ || myns < ns) { 2134 ns = myns; 2135 } 2136 return ns; 2137 } 2138 2139 size_t releasedFrames = writtenSize / mFrameSize; 2140 audioBuffer.frameCount = releasedFrames; 2141 mRemainingFrames -= releasedFrames; 2142 if (misalignment >= releasedFrames) { 2143 misalignment -= releasedFrames; 2144 } else { 2145 misalignment = 0; 2146 } 2147 2148 releaseBuffer(&audioBuffer); 2149 writtenFrames += releasedFrames; 2150 2151 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 2152 // if callback doesn't like to accept the full chunk 2153 if (writtenSize < reqSize) { 2154 continue; 2155 } 2156 2157 // There could be enough non-contiguous frames available to satisfy the remaining request 2158 if (mRemainingFrames <= nonContig) { 2159 continue; 2160 } 2161 2162#if 0 2163 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 2164 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 2165 // that total to a sum == notificationFrames. 2166 if (0 < misalignment && misalignment <= mRemainingFrames) { 2167 mRemainingFrames = misalignment; 2168 return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed); 2169 } 2170#endif 2171 2172 } 2173 if (writtenFrames > 0) { 2174 AutoMutex lock(mLock); 2175 mFramesWritten += writtenFrames; 2176 } 2177 mRemainingFrames = notificationFrames; 2178 mRetryOnPartialBuffer = true; 2179 2180 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 2181 return 0; 2182} 2183 2184status_t AudioTrack::restoreTrack_l(const char *from) 2185{ 2186 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 2187 isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from); 2188 ++mSequence; 2189 2190 // refresh the audio configuration cache in this process to make sure we get new 2191 // output parameters and new IAudioFlinger in createTrack_l() 2192 AudioSystem::clearAudioConfigCache(); 2193 2194 if (isOffloadedOrDirect_l() || mDoNotReconnect) { 2195 // FIXME re-creation of offloaded and direct tracks is not yet implemented; 2196 // reconsider enabling for linear PCM encodings when position can be preserved. 2197 return DEAD_OBJECT; 2198 } 2199 2200 // Save so we can return count since creation. 2201 mUnderrunCountOffset = getUnderrunCount_l(); 2202 2203 // save the old static buffer position 2204 uint32_t staticPosition = 0; 2205 size_t bufferPosition = 0; 2206 int loopCount = 0; 2207 if (mStaticProxy != 0) { 2208 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount); 2209 staticPosition = mStaticProxy->getPosition().unsignedValue(); 2210 } 2211 2212 mFlags = mOrigFlags; 2213 2214 // If a new IAudioTrack is successfully created, createTrack_l() will modify the 2215 // following member variables: mAudioTrack, mCblkMemory and mCblk. 2216 // It will also delete the strong references on previous IAudioTrack and IMemory. 2217 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact. 2218 status_t result = createTrack_l(); 2219 2220 if (result == NO_ERROR) { 2221 // take the frames that will be lost by track recreation into account in saved position 2222 // For streaming tracks, this is the amount we obtained from the user/client 2223 // (not the number actually consumed at the server - those are already lost). 2224 if (mStaticProxy == 0) { 2225 mPosition = mReleased; 2226 } 2227 // Continue playback from last known position and restore loop. 2228 if (mStaticProxy != 0) { 2229 if (loopCount != 0) { 2230 mStaticProxy->setBufferPositionAndLoop(bufferPosition, 2231 mLoopStart, mLoopEnd, loopCount); 2232 } else { 2233 mStaticProxy->setBufferPosition(bufferPosition); 2234 if (bufferPosition == mFrameCount) { 2235 ALOGD("restoring track at end of static buffer"); 2236 } 2237 } 2238 } 2239 if (mState == STATE_ACTIVE) { 2240 result = mAudioTrack->start(); 2241 } 2242 // server resets to zero so we offset 2243 mFramesWrittenServerOffset = 2244 mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten; 2245 mFramesWrittenAtRestore = mFramesWrittenServerOffset; 2246 } 2247 if (result != NO_ERROR) { 2248 ALOGW("restoreTrack_l() failed status %d", result); 2249 mState = STATE_STOPPED; 2250 mReleased = 0; 2251 } 2252 2253 return result; 2254} 2255 2256Modulo<uint32_t> AudioTrack::updateAndGetPosition_l() 2257{ 2258 // This is the sole place to read server consumed frames 2259 Modulo<uint32_t> newServer(mProxy->getPosition()); 2260 const int32_t delta = (newServer - mServer).signedValue(); 2261 // TODO There is controversy about whether there can be "negative jitter" in server position. 2262 // This should be investigated further, and if possible, it should be addressed. 2263 // A more definite failure mode is infrequent polling by client. 2264 // One could call (void)getPosition_l() in releaseBuffer(), 2265 // so mReleased and mPosition are always lock-step as best possible. 2266 // That should ensure delta never goes negative for infrequent polling 2267 // unless the server has more than 2^31 frames in its buffer, 2268 // in which case the use of uint32_t for these counters has bigger issues. 2269 ALOGE_IF(delta < 0, 2270 "detected illegal retrograde motion by the server: mServer advanced by %d", 2271 delta); 2272 mServer = newServer; 2273 if (delta > 0) { // avoid retrograde 2274 mPosition += delta; 2275 } 2276 return mPosition; 2277} 2278 2279bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const 2280{ 2281 // applicable for mixing tracks only (not offloaded or direct) 2282 if (mStaticProxy != 0) { 2283 return true; // static tracks do not have issues with buffer sizing. 2284 } 2285 const size_t minFrameCount = 2286 calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed 2287 /*, 0 mNotificationsPerBufferReq*/); 2288 ALOGV("isSampleRateSpeedAllowed_l mFrameCount %zu minFrameCount %zu", 2289 mFrameCount, minFrameCount); 2290 return mFrameCount >= minFrameCount; 2291} 2292 2293status_t AudioTrack::setParameters(const String8& keyValuePairs) 2294{ 2295 AutoMutex lock(mLock); 2296 return mAudioTrack->setParameters(keyValuePairs); 2297} 2298 2299status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp) 2300{ 2301 if (timestamp == nullptr) { 2302 return BAD_VALUE; 2303 } 2304 AutoMutex lock(mLock); 2305 return getTimestamp_l(timestamp); 2306} 2307 2308status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp) 2309{ 2310 if (mCblk->mFlags & CBLK_INVALID) { 2311 const status_t status = restoreTrack_l("getTimestampExtended"); 2312 if (status != OK) { 2313 // per getTimestamp() API doc in header, we return DEAD_OBJECT here, 2314 // recommending that the track be recreated. 2315 return DEAD_OBJECT; 2316 } 2317 } 2318 // check for offloaded/direct here in case restoring somehow changed those flags. 2319 if (isOffloadedOrDirect_l()) { 2320 return INVALID_OPERATION; // not supported 2321 } 2322 status_t status = mProxy->getTimestamp(timestamp); 2323 LOG_ALWAYS_FATAL_IF(status != OK, "status %d not allowed from proxy getTimestamp", status); 2324 bool found = false; 2325 timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten; 2326 timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0; 2327 // server side frame offset in case AudioTrack has been restored. 2328 for (int i = ExtendedTimestamp::LOCATION_SERVER; 2329 i < ExtendedTimestamp::LOCATION_MAX; ++i) { 2330 if (timestamp->mTimeNs[i] >= 0) { 2331 // apply server offset (frames flushed is ignored 2332 // so we don't report the jump when the flush occurs). 2333 timestamp->mPosition[i] += mFramesWrittenServerOffset; 2334 found = true; 2335 } 2336 } 2337 return found ? OK : WOULD_BLOCK; 2338} 2339 2340status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 2341{ 2342 AutoMutex lock(mLock); 2343 return getTimestamp_l(timestamp); 2344} 2345 2346status_t AudioTrack::getTimestamp_l(AudioTimestamp& timestamp) 2347{ 2348 bool previousTimestampValid = mPreviousTimestampValid; 2349 // Set false here to cover all the error return cases. 2350 mPreviousTimestampValid = false; 2351 2352 switch (mState) { 2353 case STATE_ACTIVE: 2354 case STATE_PAUSED: 2355 break; // handle below 2356 case STATE_FLUSHED: 2357 case STATE_STOPPED: 2358 return WOULD_BLOCK; 2359 case STATE_STOPPING: 2360 case STATE_PAUSED_STOPPING: 2361 if (!isOffloaded_l()) { 2362 return INVALID_OPERATION; 2363 } 2364 break; // offloaded tracks handled below 2365 default: 2366 LOG_ALWAYS_FATAL("Invalid mState in getTimestamp(): %d", mState); 2367 break; 2368 } 2369 2370 if (mCblk->mFlags & CBLK_INVALID) { 2371 const status_t status = restoreTrack_l("getTimestamp"); 2372 if (status != OK) { 2373 // per getTimestamp() API doc in header, we return DEAD_OBJECT here, 2374 // recommending that the track be recreated. 2375 return DEAD_OBJECT; 2376 } 2377 } 2378 2379 // The presented frame count must always lag behind the consumed frame count. 2380 // To avoid a race, read the presented frames first. This ensures that presented <= consumed. 2381 2382 status_t status; 2383 if (isOffloadedOrDirect_l()) { 2384 // use Binder to get timestamp 2385 status = mAudioTrack->getTimestamp(timestamp); 2386 } else { 2387 // read timestamp from shared memory 2388 ExtendedTimestamp ets; 2389 status = mProxy->getTimestamp(&ets); 2390 if (status == OK) { 2391 ExtendedTimestamp::Location location; 2392 status = ets.getBestTimestamp(×tamp, &location); 2393 2394 if (status == OK) { 2395 // It is possible that the best location has moved from the kernel to the server. 2396 // In this case we adjust the position from the previous computed latency. 2397 if (location == ExtendedTimestamp::LOCATION_SERVER) { 2398 ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL, 2399 "getTimestamp() location moved from kernel to server"); 2400 // check that the last kernel OK time info exists and the positions 2401 // are valid (if they predate the current track, the positions may 2402 // be zero or negative). 2403 const int64_t frames = 2404 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 || 2405 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 || 2406 ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 || 2407 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0) 2408 ? 2409 int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed 2410 / 1000) 2411 : 2412 (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] 2413 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]); 2414 ALOGV("frame adjustment:%lld timestamp:%s", 2415 (long long)frames, ets.toString().c_str()); 2416 if (frames >= ets.mPosition[location]) { 2417 timestamp.mPosition = 0; 2418 } else { 2419 timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames); 2420 } 2421 } else if (location == ExtendedTimestamp::LOCATION_KERNEL) { 2422 ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER, 2423 "getTimestamp() location moved from server to kernel"); 2424 } 2425 2426 // We update the timestamp time even when paused. 2427 if (mState == STATE_PAUSED /* not needed: STATE_PAUSED_STOPPING */) { 2428 const int64_t now = systemTime(); 2429 const int64_t at = convertTimespecToNs(timestamp.mTime); 2430 const int64_t lag = 2431 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 || 2432 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0) 2433 ? int64_t(mAfLatency * 1000000LL) 2434 : (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] 2435 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]) 2436 * NANOS_PER_SECOND / mSampleRate; 2437 const int64_t limit = now - lag; // no earlier than this limit 2438 if (at < limit) { 2439 ALOGV("timestamp pause lag:%lld adjusting from %lld to %lld", 2440 (long long)lag, (long long)at, (long long)limit); 2441 timestamp.mTime.tv_sec = limit / NANOS_PER_SECOND; 2442 timestamp.mTime.tv_nsec = limit % NANOS_PER_SECOND; // compiler opt. 2443 } 2444 } 2445 mPreviousLocation = location; 2446 } else { 2447 // right after AudioTrack is started, one may not find a timestamp 2448 ALOGV("getBestTimestamp did not find timestamp"); 2449 } 2450 } 2451 if (status == INVALID_OPERATION) { 2452 // INVALID_OPERATION occurs when no timestamp has been issued by the server; 2453 // other failures are signaled by a negative time. 2454 // If we come out of FLUSHED or STOPPED where the position is known 2455 // to be zero we convert this to WOULD_BLOCK (with the implicit meaning of 2456 // "zero" for NuPlayer). We don't convert for track restoration as position 2457 // does not reset. 2458 ALOGV("timestamp server offset:%lld restore frames:%lld", 2459 (long long)mFramesWrittenServerOffset, (long long)mFramesWrittenAtRestore); 2460 if (mFramesWrittenServerOffset != mFramesWrittenAtRestore) { 2461 status = WOULD_BLOCK; 2462 } 2463 } 2464 } 2465 if (status != NO_ERROR) { 2466 ALOGV_IF(status != WOULD_BLOCK, "getTimestamp error:%#x", status); 2467 return status; 2468 } 2469 if (isOffloadedOrDirect_l()) { 2470 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) { 2471 // use cached paused position in case another offloaded track is running. 2472 timestamp.mPosition = mPausedPosition; 2473 clock_gettime(CLOCK_MONOTONIC, ×tamp.mTime); 2474 // TODO: adjust for delay 2475 return NO_ERROR; 2476 } 2477 2478 // Check whether a pending flush or stop has completed, as those commands may 2479 // be asynchronous or return near finish or exhibit glitchy behavior. 2480 // 2481 // Originally this showed up as the first timestamp being a continuation of 2482 // the previous song under gapless playback. 2483 // However, we sometimes see zero timestamps, then a glitch of 2484 // the previous song's position, and then correct timestamps afterwards. 2485 if (mStartUs != 0 && mSampleRate != 0) { 2486 static const int kTimeJitterUs = 100000; // 100 ms 2487 static const int k1SecUs = 1000000; 2488 2489 const int64_t timeNow = getNowUs(); 2490 2491 if (timeNow < mStartUs + k1SecUs) { // within first second of starting 2492 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime); 2493 if (timestampTimeUs < mStartUs) { 2494 return WOULD_BLOCK; // stale timestamp time, occurs before start. 2495 } 2496 const int64_t deltaTimeUs = timestampTimeUs - mStartUs; 2497 const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000 2498 / ((double)mSampleRate * mPlaybackRate.mSpeed); 2499 2500 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) { 2501 // Verify that the counter can't count faster than the sample rate 2502 // since the start time. If greater, then that means we may have failed 2503 // to completely flush or stop the previous playing track. 2504 ALOGW_IF(!mTimestampStartupGlitchReported, 2505 "getTimestamp startup glitch detected" 2506 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)", 2507 (long long)deltaTimeUs, (long long)deltaPositionByUs, 2508 timestamp.mPosition); 2509 mTimestampStartupGlitchReported = true; 2510 if (previousTimestampValid 2511 && mPreviousTimestamp.mPosition == 0 /* should be true if valid */) { 2512 timestamp = mPreviousTimestamp; 2513 mPreviousTimestampValid = true; 2514 return NO_ERROR; 2515 } 2516 return WOULD_BLOCK; 2517 } 2518 if (deltaPositionByUs != 0) { 2519 mStartUs = 0; // don't check again, we got valid nonzero position. 2520 } 2521 } else { 2522 mStartUs = 0; // don't check again, start time expired. 2523 } 2524 mTimestampStartupGlitchReported = false; 2525 } 2526 } else { 2527 // Update the mapping between local consumed (mPosition) and server consumed (mServer) 2528 (void) updateAndGetPosition_l(); 2529 // Server consumed (mServer) and presented both use the same server time base, 2530 // and server consumed is always >= presented. 2531 // The delta between these represents the number of frames in the buffer pipeline. 2532 // If this delta between these is greater than the client position, it means that 2533 // actually presented is still stuck at the starting line (figuratively speaking), 2534 // waiting for the first frame to go by. So we can't report a valid timestamp yet. 2535 // Note: We explicitly use non-Modulo comparison here - potential wrap issue when 2536 // mPosition exceeds 32 bits. 2537 // TODO Remove when timestamp is updated to contain pipeline status info. 2538 const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue(); 2539 if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */ 2540 && (uint32_t)pipelineDepthInFrames > mPosition.value()) { 2541 return INVALID_OPERATION; 2542 } 2543 // Convert timestamp position from server time base to client time base. 2544 // TODO The following code should work OK now because timestamp.mPosition is 32-bit. 2545 // But if we change it to 64-bit then this could fail. 2546 // Use Modulo computation here. 2547 timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value(); 2548 // Immediately after a call to getPosition_l(), mPosition and 2549 // mServer both represent the same frame position. mPosition is 2550 // in client's point of view, and mServer is in server's point of 2551 // view. So the difference between them is the "fudge factor" 2552 // between client and server views due to stop() and/or new 2553 // IAudioTrack. And timestamp.mPosition is initially in server's 2554 // point of view, so we need to apply the same fudge factor to it. 2555 } 2556 2557 // Prevent retrograde motion in timestamp. 2558 // This is sometimes caused by erratic reports of the available space in the ALSA drivers. 2559 if (status == NO_ERROR) { 2560 if (previousTimestampValid) { 2561 const int64_t previousTimeNanos = convertTimespecToNs(mPreviousTimestamp.mTime); 2562 const int64_t currentTimeNanos = convertTimespecToNs(timestamp.mTime); 2563 if (currentTimeNanos < previousTimeNanos) { 2564 ALOGW("retrograde timestamp time corrected, %lld < %lld", 2565 (long long)currentTimeNanos, (long long)previousTimeNanos); 2566 timestamp.mTime = mPreviousTimestamp.mTime; 2567 } 2568 2569 // Looking at signed delta will work even when the timestamps 2570 // are wrapping around. 2571 int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition) 2572 - mPreviousTimestamp.mPosition).signedValue(); 2573 if (deltaPosition < 0) { 2574 // Only report once per position instead of spamming the log. 2575 if (!mRetrogradeMotionReported) { 2576 ALOGW("retrograde timestamp position corrected, %d = %u - %u", 2577 deltaPosition, 2578 timestamp.mPosition, 2579 mPreviousTimestamp.mPosition); 2580 mRetrogradeMotionReported = true; 2581 } 2582 } else { 2583 mRetrogradeMotionReported = false; 2584 } 2585 if (deltaPosition < 0) { 2586 timestamp.mPosition = mPreviousTimestamp.mPosition; 2587 deltaPosition = 0; 2588 } 2589#if 0 2590 // Uncomment this to verify audio timestamp rate. 2591 const int64_t deltaTime = 2592 convertTimespecToNs(timestamp.mTime) - previousTimeNanos; 2593 if (deltaTime != 0) { 2594 const int64_t computedSampleRate = 2595 deltaPosition * (long long)NANOS_PER_SECOND / deltaTime; 2596 ALOGD("computedSampleRate:%u sampleRate:%u", 2597 (unsigned)computedSampleRate, mSampleRate); 2598 } 2599#endif 2600 } 2601 mPreviousTimestamp = timestamp; 2602 mPreviousTimestampValid = true; 2603 } 2604 2605 return status; 2606} 2607 2608String8 AudioTrack::getParameters(const String8& keys) 2609{ 2610 audio_io_handle_t output = getOutput(); 2611 if (output != AUDIO_IO_HANDLE_NONE) { 2612 return AudioSystem::getParameters(output, keys); 2613 } else { 2614 return String8::empty(); 2615 } 2616} 2617 2618bool AudioTrack::isOffloaded() const 2619{ 2620 AutoMutex lock(mLock); 2621 return isOffloaded_l(); 2622} 2623 2624bool AudioTrack::isDirect() const 2625{ 2626 AutoMutex lock(mLock); 2627 return isDirect_l(); 2628} 2629 2630bool AudioTrack::isOffloadedOrDirect() const 2631{ 2632 AutoMutex lock(mLock); 2633 return isOffloadedOrDirect_l(); 2634} 2635 2636 2637status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 2638{ 2639 2640 const size_t SIZE = 256; 2641 char buffer[SIZE]; 2642 String8 result; 2643 2644 result.append(" AudioTrack::dump\n"); 2645 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 2646 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 2647 result.append(buffer); 2648 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 2649 mChannelCount, mFrameCount); 2650 result.append(buffer); 2651 snprintf(buffer, 255, " sample rate(%u), speed(%f), status(%d)\n", 2652 mSampleRate, mPlaybackRate.mSpeed, mStatus); 2653 result.append(buffer); 2654 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 2655 result.append(buffer); 2656 ::write(fd, result.string(), result.size()); 2657 return NO_ERROR; 2658} 2659 2660uint32_t AudioTrack::getUnderrunCount() const 2661{ 2662 AutoMutex lock(mLock); 2663 return getUnderrunCount_l(); 2664} 2665 2666uint32_t AudioTrack::getUnderrunCount_l() const 2667{ 2668 return mProxy->getUnderrunCount() + mUnderrunCountOffset; 2669} 2670 2671uint32_t AudioTrack::getUnderrunFrames() const 2672{ 2673 AutoMutex lock(mLock); 2674 return mProxy->getUnderrunFrames(); 2675} 2676 2677status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback) 2678{ 2679 if (callback == 0) { 2680 ALOGW("%s adding NULL callback!", __FUNCTION__); 2681 return BAD_VALUE; 2682 } 2683 AutoMutex lock(mLock); 2684 if (mDeviceCallback == callback) { 2685 ALOGW("%s adding same callback!", __FUNCTION__); 2686 return INVALID_OPERATION; 2687 } 2688 status_t status = NO_ERROR; 2689 if (mOutput != AUDIO_IO_HANDLE_NONE) { 2690 if (mDeviceCallback != 0) { 2691 ALOGW("%s callback already present!", __FUNCTION__); 2692 AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); 2693 } 2694 status = AudioSystem::addAudioDeviceCallback(callback, mOutput); 2695 } 2696 mDeviceCallback = callback; 2697 return status; 2698} 2699 2700status_t AudioTrack::removeAudioDeviceCallback( 2701 const sp<AudioSystem::AudioDeviceCallback>& callback) 2702{ 2703 if (callback == 0) { 2704 ALOGW("%s removing NULL callback!", __FUNCTION__); 2705 return BAD_VALUE; 2706 } 2707 AutoMutex lock(mLock); 2708 if (mDeviceCallback != callback) { 2709 ALOGW("%s removing different callback!", __FUNCTION__); 2710 return INVALID_OPERATION; 2711 } 2712 if (mOutput != AUDIO_IO_HANDLE_NONE) { 2713 AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); 2714 } 2715 mDeviceCallback = 0; 2716 return NO_ERROR; 2717} 2718 2719status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location) 2720{ 2721 if (msec == nullptr || 2722 (location != ExtendedTimestamp::LOCATION_SERVER 2723 && location != ExtendedTimestamp::LOCATION_KERNEL)) { 2724 return BAD_VALUE; 2725 } 2726 AutoMutex lock(mLock); 2727 // inclusive of offloaded and direct tracks. 2728 // 2729 // It is possible, but not enabled, to allow duration computation for non-pcm 2730 // audio_has_proportional_frames() formats because currently they have 2731 // the drain rate equivalent to the pcm sample rate * framesize. 2732 if (!isPurePcmData_l()) { 2733 return INVALID_OPERATION; 2734 } 2735 ExtendedTimestamp ets; 2736 if (getTimestamp_l(&ets) == OK 2737 && ets.mTimeNs[location] > 0) { 2738 int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT] 2739 - ets.mPosition[location]; 2740 if (diff < 0) { 2741 *msec = 0; 2742 } else { 2743 // ms is the playback time by frames 2744 int64_t ms = (int64_t)((double)diff * 1000 / 2745 ((double)mSampleRate * mPlaybackRate.mSpeed)); 2746 // clockdiff is the timestamp age (negative) 2747 int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 : 2748 ets.mTimeNs[location] 2749 + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC] 2750 - systemTime(SYSTEM_TIME_MONOTONIC); 2751 2752 //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff); 2753 static const int NANOS_PER_MILLIS = 1000000; 2754 *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS); 2755 } 2756 return NO_ERROR; 2757 } 2758 if (location != ExtendedTimestamp::LOCATION_SERVER) { 2759 return INVALID_OPERATION; // LOCATION_KERNEL is not available 2760 } 2761 // use server position directly (offloaded and direct arrive here) 2762 updateAndGetPosition_l(); 2763 int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue(); 2764 *msec = (diff <= 0) ? 0 2765 : (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed)); 2766 return NO_ERROR; 2767} 2768 2769bool AudioTrack::hasStarted() 2770{ 2771 AutoMutex lock(mLock); 2772 switch (mState) { 2773 case STATE_STOPPED: 2774 if (isOffloadedOrDirect_l()) { 2775 // check if we have started in the past to return true. 2776 return mStartUs > 0; 2777 } 2778 // A normal audio track may still be draining, so 2779 // check if stream has ended. This covers fasttrack position 2780 // instability and start/stop without any data written. 2781 if (mProxy->getStreamEndDone()) { 2782 return true; 2783 } 2784 // fall through 2785 case STATE_ACTIVE: 2786 case STATE_STOPPING: 2787 break; 2788 case STATE_PAUSED: 2789 case STATE_PAUSED_STOPPING: 2790 case STATE_FLUSHED: 2791 return false; // we're not active 2792 default: 2793 LOG_ALWAYS_FATAL("Invalid mState in hasStarted(): %d", mState); 2794 break; 2795 } 2796 2797 // wait indicates whether we need to wait for a timestamp. 2798 // This is conservatively figured - if we encounter an unexpected error 2799 // then we will not wait. 2800 bool wait = false; 2801 if (isOffloadedOrDirect_l()) { 2802 AudioTimestamp ts; 2803 status_t status = getTimestamp_l(ts); 2804 if (status == WOULD_BLOCK) { 2805 wait = true; 2806 } else if (status == OK) { 2807 wait = (ts.mPosition == 0 || ts.mPosition == mStartTs.mPosition); 2808 } 2809 ALOGV("hasStarted wait:%d ts:%u start position:%lld", 2810 (int)wait, 2811 ts.mPosition, 2812 (long long)mStartTs.mPosition); 2813 } else { 2814 int location = ExtendedTimestamp::LOCATION_SERVER; // for ALOG 2815 ExtendedTimestamp ets; 2816 status_t status = getTimestamp_l(&ets); 2817 if (status == WOULD_BLOCK) { // no SERVER or KERNEL frame info in ets 2818 wait = true; 2819 } else if (status == OK) { 2820 for (location = ExtendedTimestamp::LOCATION_KERNEL; 2821 location >= ExtendedTimestamp::LOCATION_SERVER; --location) { 2822 if (ets.mTimeNs[location] < 0 || mStartEts.mTimeNs[location] < 0) { 2823 continue; 2824 } 2825 wait = ets.mPosition[location] == 0 2826 || ets.mPosition[location] == mStartEts.mPosition[location]; 2827 break; 2828 } 2829 } 2830 ALOGV("hasStarted wait:%d ets:%lld start position:%lld", 2831 (int)wait, 2832 (long long)ets.mPosition[location], 2833 (long long)mStartEts.mPosition[location]); 2834 } 2835 return !wait; 2836} 2837 2838// ========================================================================= 2839 2840void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 2841{ 2842 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 2843 if (audioTrack != 0) { 2844 AutoMutex lock(audioTrack->mLock); 2845 audioTrack->mProxy->binderDied(); 2846 } 2847} 2848 2849// ========================================================================= 2850 2851AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 2852 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 2853 mIgnoreNextPausedInt(false) 2854{ 2855} 2856 2857AudioTrack::AudioTrackThread::~AudioTrackThread() 2858{ 2859} 2860 2861bool AudioTrack::AudioTrackThread::threadLoop() 2862{ 2863 { 2864 AutoMutex _l(mMyLock); 2865 if (mPaused) { 2866 mMyCond.wait(mMyLock); 2867 // caller will check for exitPending() 2868 return true; 2869 } 2870 if (mIgnoreNextPausedInt) { 2871 mIgnoreNextPausedInt = false; 2872 mPausedInt = false; 2873 } 2874 if (mPausedInt) { 2875 if (mPausedNs > 0) { 2876 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 2877 } else { 2878 mMyCond.wait(mMyLock); 2879 } 2880 mPausedInt = false; 2881 return true; 2882 } 2883 } 2884 if (exitPending()) { 2885 return false; 2886 } 2887 nsecs_t ns = mReceiver.processAudioBuffer(); 2888 switch (ns) { 2889 case 0: 2890 return true; 2891 case NS_INACTIVE: 2892 pauseInternal(); 2893 return true; 2894 case NS_NEVER: 2895 return false; 2896 case NS_WHENEVER: 2897 // Event driven: call wake() when callback notifications conditions change. 2898 ns = INT64_MAX; 2899 // fall through 2900 default: 2901 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns); 2902 pauseInternal(ns); 2903 return true; 2904 } 2905} 2906 2907void AudioTrack::AudioTrackThread::requestExit() 2908{ 2909 // must be in this order to avoid a race condition 2910 Thread::requestExit(); 2911 resume(); 2912} 2913 2914void AudioTrack::AudioTrackThread::pause() 2915{ 2916 AutoMutex _l(mMyLock); 2917 mPaused = true; 2918} 2919 2920void AudioTrack::AudioTrackThread::resume() 2921{ 2922 AutoMutex _l(mMyLock); 2923 mIgnoreNextPausedInt = true; 2924 if (mPaused || mPausedInt) { 2925 mPaused = false; 2926 mPausedInt = false; 2927 mMyCond.signal(); 2928 } 2929} 2930 2931void AudioTrack::AudioTrackThread::wake() 2932{ 2933 AutoMutex _l(mMyLock); 2934 if (!mPaused) { 2935 // wake() might be called while servicing a callback - ignore the next 2936 // pause time and call processAudioBuffer. 2937 mIgnoreNextPausedInt = true; 2938 if (mPausedInt && mPausedNs > 0) { 2939 // audio track is active and internally paused with timeout. 2940 mPausedInt = false; 2941 mMyCond.signal(); 2942 } 2943 } 2944} 2945 2946void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 2947{ 2948 AutoMutex _l(mMyLock); 2949 mPausedInt = true; 2950 mPausedNs = ns; 2951} 2952 2953} // namespace android 2954