AudioTrack.cpp revision 1f12a8ad958344c50733b948628ffa06db9c5bc6
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 mReleased = 0; 675 } 676 677 mProxy->interrupt(); 678 mAudioTrack->stop(); 679 680 // Note: legacy handling - stop does not clear playback marker 681 // and periodic update counter, but flush does for streaming tracks. 682 683 if (mSharedBuffer != 0) { 684 // clear buffer position and loop count. 685 mStaticProxy->setBufferPositionAndLoop(0 /* position */, 686 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */); 687 } 688 689 sp<AudioTrackThread> t = mAudioTrackThread; 690 if (t != 0) { 691 if (!isOffloaded_l()) { 692 t->pause(); 693 } 694 } else { 695 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 696 set_sched_policy(0, mPreviousSchedulingGroup); 697 } 698} 699 700bool AudioTrack::stopped() const 701{ 702 AutoMutex lock(mLock); 703 return mState != STATE_ACTIVE; 704} 705 706void AudioTrack::flush() 707{ 708 if (mSharedBuffer != 0) { 709 return; 710 } 711 AutoMutex lock(mLock); 712 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 713 return; 714 } 715 flush_l(); 716} 717 718void AudioTrack::flush_l() 719{ 720 ALOG_ASSERT(mState != STATE_ACTIVE); 721 722 // clear playback marker and periodic update counter 723 mMarkerPosition = 0; 724 mMarkerReached = false; 725 mUpdatePeriod = 0; 726 mRefreshRemaining = true; 727 728 mState = STATE_FLUSHED; 729 mReleased = 0; 730 if (isOffloaded_l()) { 731 mProxy->interrupt(); 732 } 733 mProxy->flush(); 734 mAudioTrack->flush(); 735} 736 737void AudioTrack::pause() 738{ 739 AutoMutex lock(mLock); 740 if (mState == STATE_ACTIVE) { 741 mState = STATE_PAUSED; 742 } else if (mState == STATE_STOPPING) { 743 mState = STATE_PAUSED_STOPPING; 744 } else { 745 return; 746 } 747 mProxy->interrupt(); 748 mAudioTrack->pause(); 749 750 if (isOffloaded_l()) { 751 if (mOutput != AUDIO_IO_HANDLE_NONE) { 752 // An offload output can be re-used between two audio tracks having 753 // the same configuration. A timestamp query for a paused track 754 // while the other is running would return an incorrect time. 755 // To fix this, cache the playback position on a pause() and return 756 // this time when requested until the track is resumed. 757 758 // OffloadThread sends HAL pause in its threadLoop. Time saved 759 // here can be slightly off. 760 761 // TODO: check return code for getRenderPosition. 762 763 uint32_t halFrames; 764 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); 765 ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); 766 } 767 } 768} 769 770status_t AudioTrack::setVolume(float left, float right) 771{ 772 // This duplicates a test by AudioTrack JNI, but that is not the only caller 773 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY || 774 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) { 775 return BAD_VALUE; 776 } 777 778 AutoMutex lock(mLock); 779 mVolume[AUDIO_INTERLEAVE_LEFT] = left; 780 mVolume[AUDIO_INTERLEAVE_RIGHT] = right; 781 782 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right))); 783 784 if (isOffloaded_l()) { 785 mAudioTrack->signal(); 786 } 787 return NO_ERROR; 788} 789 790status_t AudioTrack::setVolume(float volume) 791{ 792 return setVolume(volume, volume); 793} 794 795status_t AudioTrack::setAuxEffectSendLevel(float level) 796{ 797 // This duplicates a test by AudioTrack JNI, but that is not the only caller 798 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) { 799 return BAD_VALUE; 800 } 801 802 AutoMutex lock(mLock); 803 mSendLevel = level; 804 mProxy->setSendLevel(level); 805 806 return NO_ERROR; 807} 808 809void AudioTrack::getAuxEffectSendLevel(float* level) const 810{ 811 if (level != NULL) { 812 *level = mSendLevel; 813 } 814} 815 816status_t AudioTrack::setSampleRate(uint32_t rate) 817{ 818 AutoMutex lock(mLock); 819 if (rate == mSampleRate) { 820 return NO_ERROR; 821 } 822 if (isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 823 return INVALID_OPERATION; 824 } 825 if (mOutput == AUDIO_IO_HANDLE_NONE) { 826 return NO_INIT; 827 } 828 // NOTE: it is theoretically possible, but highly unlikely, that a device change 829 // could mean a previously allowed sampling rate is no longer allowed. 830 uint32_t afSamplingRate; 831 if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) { 832 return NO_INIT; 833 } 834 // pitch is emulated by adjusting speed and sampleRate 835 const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch); 836 if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 837 return BAD_VALUE; 838 } 839 // TODO: Should we also check if the buffer size is compatible? 840 841 mSampleRate = rate; 842 mProxy->setSampleRate(effectiveSampleRate); 843 844 return NO_ERROR; 845} 846 847uint32_t AudioTrack::getSampleRate() const 848{ 849 AutoMutex lock(mLock); 850 851 // sample rate can be updated during playback by the offloaded decoder so we need to 852 // query the HAL and update if needed. 853// FIXME use Proxy return channel to update the rate from server and avoid polling here 854 if (isOffloadedOrDirect_l()) { 855 if (mOutput != AUDIO_IO_HANDLE_NONE) { 856 uint32_t sampleRate = 0; 857 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate); 858 if (status == NO_ERROR) { 859 mSampleRate = sampleRate; 860 } 861 } 862 } 863 return mSampleRate; 864} 865 866uint32_t AudioTrack::getOriginalSampleRate() const 867{ 868 return mOriginalSampleRate; 869} 870 871status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate) 872{ 873 AutoMutex lock(mLock); 874 if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) { 875 return NO_ERROR; 876 } 877 if (isOffloadedOrDirect_l()) { 878 return INVALID_OPERATION; 879 } 880 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 881 return INVALID_OPERATION; 882 } 883 884 ALOGV("setPlaybackRate (input): mSampleRate:%u mSpeed:%f mPitch:%f", 885 mSampleRate, playbackRate.mSpeed, playbackRate.mPitch); 886 // pitch is emulated by adjusting speed and sampleRate 887 const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch); 888 const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch); 889 const float effectivePitch = adjustPitch(playbackRate.mPitch); 890 AudioPlaybackRate playbackRateTemp = playbackRate; 891 playbackRateTemp.mSpeed = effectiveSpeed; 892 playbackRateTemp.mPitch = effectivePitch; 893 894 ALOGV("setPlaybackRate (effective): mSampleRate:%u mSpeed:%f mPitch:%f", 895 effectiveRate, effectiveSpeed, effectivePitch); 896 897 if (!isAudioPlaybackRateValid(playbackRateTemp)) { 898 ALOGV("setPlaybackRate(%f, %f) failed (effective rate out of bounds)", 899 playbackRate.mSpeed, playbackRate.mPitch); 900 return BAD_VALUE; 901 } 902 // Check if the buffer size is compatible. 903 if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) { 904 ALOGV("setPlaybackRate(%f, %f) failed (buffer size)", 905 playbackRate.mSpeed, playbackRate.mPitch); 906 return BAD_VALUE; 907 } 908 909 // Check resampler ratios are within bounds 910 if ((uint64_t)effectiveRate > (uint64_t)mSampleRate * (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 911 ALOGV("setPlaybackRate(%f, %f) failed. Resample rate exceeds max accepted value", 912 playbackRate.mSpeed, playbackRate.mPitch); 913 return BAD_VALUE; 914 } 915 916 if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) { 917 ALOGV("setPlaybackRate(%f, %f) failed. Resample rate below min accepted value", 918 playbackRate.mSpeed, playbackRate.mPitch); 919 return BAD_VALUE; 920 } 921 mPlaybackRate = playbackRate; 922 //set effective rates 923 mProxy->setPlaybackRate(playbackRateTemp); 924 mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate 925 return NO_ERROR; 926} 927 928const AudioPlaybackRate& AudioTrack::getPlaybackRate() const 929{ 930 AutoMutex lock(mLock); 931 return mPlaybackRate; 932} 933 934ssize_t AudioTrack::getBufferSizeInFrames() 935{ 936 AutoMutex lock(mLock); 937 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) { 938 return NO_INIT; 939 } 940 return (ssize_t) mProxy->getBufferSizeInFrames(); 941} 942 943status_t AudioTrack::getBufferDurationInUs(int64_t *duration) 944{ 945 if (duration == nullptr) { 946 return BAD_VALUE; 947 } 948 AutoMutex lock(mLock); 949 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) { 950 return NO_INIT; 951 } 952 ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames(); 953 if (bufferSizeInFrames < 0) { 954 return (status_t)bufferSizeInFrames; 955 } 956 *duration = (int64_t)((double)bufferSizeInFrames * 1000000 957 / ((double)mSampleRate * mPlaybackRate.mSpeed)); 958 return NO_ERROR; 959} 960 961ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames) 962{ 963 AutoMutex lock(mLock); 964 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) { 965 return NO_INIT; 966 } 967 // Reject if timed track or compressed audio. 968 if (!audio_is_linear_pcm(mFormat)) { 969 return INVALID_OPERATION; 970 } 971 return (ssize_t) mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames); 972} 973 974status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 975{ 976 if (mSharedBuffer == 0 || isOffloadedOrDirect()) { 977 return INVALID_OPERATION; 978 } 979 980 if (loopCount == 0) { 981 ; 982 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 983 loopEnd - loopStart >= MIN_LOOP) { 984 ; 985 } else { 986 return BAD_VALUE; 987 } 988 989 AutoMutex lock(mLock); 990 // See setPosition() regarding setting parameters such as loop points or position while active 991 if (mState == STATE_ACTIVE) { 992 return INVALID_OPERATION; 993 } 994 setLoop_l(loopStart, loopEnd, loopCount); 995 return NO_ERROR; 996} 997 998void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 999{ 1000 // We do not update the periodic notification point. 1001 // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 1002 mLoopCount = loopCount; 1003 mLoopEnd = loopEnd; 1004 mLoopStart = loopStart; 1005 mLoopCountNotified = loopCount; 1006 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 1007 1008 // Waking the AudioTrackThread is not needed as this cannot be called when active. 1009} 1010 1011status_t AudioTrack::setMarkerPosition(uint32_t marker) 1012{ 1013 // The only purpose of setting marker position is to get a callback 1014 if (mCbf == NULL || isOffloadedOrDirect()) { 1015 return INVALID_OPERATION; 1016 } 1017 1018 AutoMutex lock(mLock); 1019 mMarkerPosition = marker; 1020 mMarkerReached = false; 1021 1022 sp<AudioTrackThread> t = mAudioTrackThread; 1023 if (t != 0) { 1024 t->wake(); 1025 } 1026 return NO_ERROR; 1027} 1028 1029status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 1030{ 1031 if (isOffloadedOrDirect()) { 1032 return INVALID_OPERATION; 1033 } 1034 if (marker == NULL) { 1035 return BAD_VALUE; 1036 } 1037 1038 AutoMutex lock(mLock); 1039 mMarkerPosition.getValue(marker); 1040 1041 return NO_ERROR; 1042} 1043 1044status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 1045{ 1046 // The only purpose of setting position update period is to get a callback 1047 if (mCbf == NULL || isOffloadedOrDirect()) { 1048 return INVALID_OPERATION; 1049 } 1050 1051 AutoMutex lock(mLock); 1052 mNewPosition = updateAndGetPosition_l() + updatePeriod; 1053 mUpdatePeriod = updatePeriod; 1054 1055 sp<AudioTrackThread> t = mAudioTrackThread; 1056 if (t != 0) { 1057 t->wake(); 1058 } 1059 return NO_ERROR; 1060} 1061 1062status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 1063{ 1064 if (isOffloadedOrDirect()) { 1065 return INVALID_OPERATION; 1066 } 1067 if (updatePeriod == NULL) { 1068 return BAD_VALUE; 1069 } 1070 1071 AutoMutex lock(mLock); 1072 *updatePeriod = mUpdatePeriod; 1073 1074 return NO_ERROR; 1075} 1076 1077status_t AudioTrack::setPosition(uint32_t position) 1078{ 1079 if (mSharedBuffer == 0 || isOffloadedOrDirect()) { 1080 return INVALID_OPERATION; 1081 } 1082 if (position > mFrameCount) { 1083 return BAD_VALUE; 1084 } 1085 1086 AutoMutex lock(mLock); 1087 // Currently we require that the player is inactive before setting parameters such as position 1088 // or loop points. Otherwise, there could be a race condition: the application could read the 1089 // current position, compute a new position or loop parameters, and then set that position or 1090 // loop parameters but it would do the "wrong" thing since the position has continued to advance 1091 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 1092 // to specify how it wants to handle such scenarios. 1093 if (mState == STATE_ACTIVE) { 1094 return INVALID_OPERATION; 1095 } 1096 // After setting the position, use full update period before notification. 1097 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 1098 mStaticProxy->setBufferPosition(position); 1099 1100 // Waking the AudioTrackThread is not needed as this cannot be called when active. 1101 return NO_ERROR; 1102} 1103 1104status_t AudioTrack::getPosition(uint32_t *position) 1105{ 1106 if (position == NULL) { 1107 return BAD_VALUE; 1108 } 1109 1110 AutoMutex lock(mLock); 1111 // FIXME: offloaded and direct tracks call into the HAL for render positions 1112 // for compressed/synced data; however, we use proxy position for pure linear pcm data 1113 // as we do not know the capability of the HAL for pcm position support and standby. 1114 // There may be some latency differences between the HAL position and the proxy position. 1115 if (isOffloadedOrDirect_l() && !isPurePcmData_l()) { 1116 uint32_t dspFrames = 0; 1117 1118 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) { 1119 ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); 1120 *position = mPausedPosition; 1121 return NO_ERROR; 1122 } 1123 1124 if (mOutput != AUDIO_IO_HANDLE_NONE) { 1125 uint32_t halFrames; // actually unused 1126 (void) AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 1127 // FIXME: on getRenderPosition() error, we return OK with frame position 0. 1128 } 1129 // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED) 1130 // due to hardware latency. We leave this behavior for now. 1131 *position = dspFrames; 1132 } else { 1133 if (mCblk->mFlags & CBLK_INVALID) { 1134 (void) restoreTrack_l("getPosition"); 1135 // FIXME: for compatibility with the Java API we ignore the restoreTrack_l() 1136 // error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position. 1137 } 1138 1139 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 1140 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 1141 0 : updateAndGetPosition_l().value(); 1142 } 1143 return NO_ERROR; 1144} 1145 1146status_t AudioTrack::getBufferPosition(uint32_t *position) 1147{ 1148 if (mSharedBuffer == 0) { 1149 return INVALID_OPERATION; 1150 } 1151 if (position == NULL) { 1152 return BAD_VALUE; 1153 } 1154 1155 AutoMutex lock(mLock); 1156 *position = mStaticProxy->getBufferPosition(); 1157 return NO_ERROR; 1158} 1159 1160status_t AudioTrack::reload() 1161{ 1162 if (mSharedBuffer == 0 || isOffloadedOrDirect()) { 1163 return INVALID_OPERATION; 1164 } 1165 1166 AutoMutex lock(mLock); 1167 // See setPosition() regarding setting parameters such as loop points or position while active 1168 if (mState == STATE_ACTIVE) { 1169 return INVALID_OPERATION; 1170 } 1171 mNewPosition = mUpdatePeriod; 1172 (void) updateAndGetPosition_l(); 1173 mPosition = 0; 1174 mPreviousTimestampValid = false; 1175#if 0 1176 // The documentation is not clear on the behavior of reload() and the restoration 1177 // of loop count. Historically we have not restored loop count, start, end, 1178 // but it makes sense if one desires to repeat playing a particular sound. 1179 if (mLoopCount != 0) { 1180 mLoopCountNotified = mLoopCount; 1181 mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount); 1182 } 1183#endif 1184 mStaticProxy->setBufferPosition(0); 1185 return NO_ERROR; 1186} 1187 1188audio_io_handle_t AudioTrack::getOutput() const 1189{ 1190 AutoMutex lock(mLock); 1191 return mOutput; 1192} 1193 1194status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) { 1195 AutoMutex lock(mLock); 1196 if (mSelectedDeviceId != deviceId) { 1197 mSelectedDeviceId = deviceId; 1198 android_atomic_or(CBLK_INVALID, &mCblk->mFlags); 1199 } 1200 return NO_ERROR; 1201} 1202 1203audio_port_handle_t AudioTrack::getOutputDevice() { 1204 AutoMutex lock(mLock); 1205 return mSelectedDeviceId; 1206} 1207 1208audio_port_handle_t AudioTrack::getRoutedDeviceId() { 1209 AutoMutex lock(mLock); 1210 if (mOutput == AUDIO_IO_HANDLE_NONE) { 1211 return AUDIO_PORT_HANDLE_NONE; 1212 } 1213 return AudioSystem::getDeviceIdForIo(mOutput); 1214} 1215 1216status_t AudioTrack::attachAuxEffect(int effectId) 1217{ 1218 AutoMutex lock(mLock); 1219 status_t status = mAudioTrack->attachAuxEffect(effectId); 1220 if (status == NO_ERROR) { 1221 mAuxEffectId = effectId; 1222 } 1223 return status; 1224} 1225 1226audio_stream_type_t AudioTrack::streamType() const 1227{ 1228 if (mStreamType == AUDIO_STREAM_DEFAULT) { 1229 return audio_attributes_to_stream_type(&mAttributes); 1230 } 1231 return mStreamType; 1232} 1233 1234// ------------------------------------------------------------------------- 1235 1236// must be called with mLock held 1237status_t AudioTrack::createTrack_l() 1238{ 1239 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 1240 if (audioFlinger == 0) { 1241 ALOGE("Could not get audioflinger"); 1242 return NO_INIT; 1243 } 1244 1245 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) { 1246 AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); 1247 } 1248 audio_io_handle_t output; 1249 audio_stream_type_t streamType = mStreamType; 1250 audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL; 1251 1252 // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted. 1253 // After fast request is denied, we will request again if IAudioTrack is re-created. 1254 1255 status_t status; 1256 status = AudioSystem::getOutputForAttr(attr, &output, 1257 mSessionId, &streamType, mClientUid, 1258 mSampleRate, mFormat, mChannelMask, 1259 mFlags, mSelectedDeviceId, mOffloadInfo); 1260 1261 if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) { 1262 ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u, format %#x," 1263 " channel mask %#x, flags %#x", 1264 mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags); 1265 return BAD_VALUE; 1266 } 1267 { 1268 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 1269 // we must release it ourselves if anything goes wrong. 1270 1271 // Not all of these values are needed under all conditions, but it is easier to get them all 1272 status = AudioSystem::getLatency(output, &mAfLatency); 1273 if (status != NO_ERROR) { 1274 ALOGE("getLatency(%d) failed status %d", output, status); 1275 goto release; 1276 } 1277 ALOGV("createTrack_l() output %d afLatency %u", output, mAfLatency); 1278 1279 status = AudioSystem::getFrameCount(output, &mAfFrameCount); 1280 if (status != NO_ERROR) { 1281 ALOGE("getFrameCount(output=%d) status %d", output, status); 1282 goto release; 1283 } 1284 1285 // TODO consider making this a member variable if there are other uses for it later 1286 size_t afFrameCountHAL; 1287 status = AudioSystem::getFrameCountHAL(output, &afFrameCountHAL); 1288 if (status != NO_ERROR) { 1289 ALOGE("getFrameCountHAL(output=%d) status %d", output, status); 1290 goto release; 1291 } 1292 ALOG_ASSERT(afFrameCountHAL > 0); 1293 1294 status = AudioSystem::getSamplingRate(output, &mAfSampleRate); 1295 if (status != NO_ERROR) { 1296 ALOGE("getSamplingRate(output=%d) status %d", output, status); 1297 goto release; 1298 } 1299 if (mSampleRate == 0) { 1300 mSampleRate = mAfSampleRate; 1301 mOriginalSampleRate = mAfSampleRate; 1302 } 1303 1304 // Client can only express a preference for FAST. Server will perform additional tests. 1305 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1306 bool useCaseAllowed = 1307 // either of these use cases: 1308 // use case 1: shared buffer 1309 (mSharedBuffer != 0) || 1310 // use case 2: callback transfer mode 1311 (mTransfer == TRANSFER_CALLBACK) || 1312 // use case 3: obtain/release mode 1313 (mTransfer == TRANSFER_OBTAIN) || 1314 // use case 4: synchronous write 1315 ((mTransfer == TRANSFER_SYNC) && mThreadCanCallJava); 1316 // sample rates must also match 1317 bool fastAllowed = useCaseAllowed && (mSampleRate == mAfSampleRate); 1318 if (!fastAllowed) { 1319 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client; transfer %d, " 1320 "track %u Hz, output %u Hz", 1321 mTransfer, mSampleRate, mAfSampleRate); 1322 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1323 } 1324 } 1325 1326 mNotificationFramesAct = mNotificationFramesReq; 1327 1328 size_t frameCount = mReqFrameCount; 1329 if (!audio_has_proportional_frames(mFormat)) { 1330 1331 if (mSharedBuffer != 0) { 1332 // Same comment as below about ignoring frameCount parameter for set() 1333 frameCount = mSharedBuffer->size(); 1334 } else if (frameCount == 0) { 1335 frameCount = mAfFrameCount; 1336 } 1337 if (mNotificationFramesAct != frameCount) { 1338 mNotificationFramesAct = frameCount; 1339 } 1340 } else if (mSharedBuffer != 0) { 1341 // FIXME: Ensure client side memory buffers need 1342 // not have additional alignment beyond sample 1343 // (e.g. 16 bit stereo accessed as 32 bit frame). 1344 size_t alignment = audio_bytes_per_sample(mFormat); 1345 if (alignment & 1) { 1346 // for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java). 1347 alignment = 1; 1348 } 1349 if (mChannelCount > 1) { 1350 // More than 2 channels does not require stronger alignment than stereo 1351 alignment <<= 1; 1352 } 1353 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 1354 ALOGE("Invalid buffer alignment: address %p, channel count %u", 1355 mSharedBuffer->pointer(), mChannelCount); 1356 status = BAD_VALUE; 1357 goto release; 1358 } 1359 1360 // When initializing a shared buffer AudioTrack via constructors, 1361 // there's no frameCount parameter. 1362 // But when initializing a shared buffer AudioTrack via set(), 1363 // there _is_ a frameCount parameter. We silently ignore it. 1364 frameCount = mSharedBuffer->size() / mFrameSize; 1365 } else { 1366 size_t minFrameCount = 0; 1367 // For fast tracks the frame count calculations and checks are mostly done by server, 1368 // but we try to respect the application's request for notifications per buffer. 1369 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1370 if (mNotificationsPerBufferReq > 0) { 1371 // Avoid possible arithmetic overflow during multiplication. 1372 // mNotificationsPerBuffer is clamped to a small integer earlier, so it is unlikely. 1373 if (mNotificationsPerBufferReq > SIZE_MAX / afFrameCountHAL) { 1374 ALOGE("Requested notificationPerBuffer=%u ignored for HAL frameCount=%zu", 1375 mNotificationsPerBufferReq, afFrameCountHAL); 1376 } else { 1377 minFrameCount = afFrameCountHAL * mNotificationsPerBufferReq; 1378 } 1379 } 1380 } else { 1381 // for normal tracks precompute the frame count based on speed. 1382 const float speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f : 1383 max(mMaxRequiredSpeed, mPlaybackRate.mSpeed); 1384 minFrameCount = calculateMinFrameCount( 1385 mAfLatency, mAfFrameCount, mAfSampleRate, mSampleRate, 1386 speed /*, 0 mNotificationsPerBufferReq*/); 1387 } 1388 if (frameCount < minFrameCount) { 1389 frameCount = minFrameCount; 1390 } 1391 } 1392 1393 audio_output_flags_t flags = mFlags; 1394 1395 pid_t tid = -1; 1396 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1397 if (mAudioTrackThread != 0 && !mThreadCanCallJava) { 1398 tid = mAudioTrackThread->getTid(); 1399 } 1400 } 1401 1402 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1403 // but we will still need the original value also 1404 audio_session_t originalSessionId = mSessionId; 1405 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 1406 mSampleRate, 1407 mFormat, 1408 mChannelMask, 1409 &temp, 1410 &flags, 1411 mSharedBuffer, 1412 output, 1413 mClientPid, 1414 tid, 1415 &mSessionId, 1416 mClientUid, 1417 &status); 1418 ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId, 1419 "session ID changed from %d to %d", originalSessionId, mSessionId); 1420 1421 if (status != NO_ERROR) { 1422 ALOGE("AudioFlinger could not create track, status: %d", status); 1423 goto release; 1424 } 1425 ALOG_ASSERT(track != 0); 1426 1427 // AudioFlinger now owns the reference to the I/O handle, 1428 // so we are no longer responsible for releasing it. 1429 1430 // FIXME compare to AudioRecord 1431 sp<IMemory> iMem = track->getCblk(); 1432 if (iMem == 0) { 1433 ALOGE("Could not get control block"); 1434 return NO_INIT; 1435 } 1436 void *iMemPointer = iMem->pointer(); 1437 if (iMemPointer == NULL) { 1438 ALOGE("Could not get control block pointer"); 1439 return NO_INIT; 1440 } 1441 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1442 if (mAudioTrack != 0) { 1443 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this); 1444 mDeathNotifier.clear(); 1445 } 1446 mAudioTrack = track; 1447 mCblkMemory = iMem; 1448 IPCThreadState::self()->flushCommands(); 1449 1450 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1451 mCblk = cblk; 1452 // note that temp is the (possibly revised) value of frameCount 1453 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1454 // In current design, AudioTrack client checks and ensures frame count validity before 1455 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1456 // for fast track as it uses a special method of assigning frame count. 1457 ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp); 1458 } 1459 frameCount = temp; 1460 1461 mAwaitBoost = false; 1462 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1463 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 1464 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount); 1465 if (!mThreadCanCallJava) { 1466 mAwaitBoost = true; 1467 } 1468 } else { 1469 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount); 1470 } 1471 } 1472 mFlags = flags; 1473 1474 // Make sure that application is notified with sufficient margin before underrun. 1475 // The client can divide the AudioTrack buffer into sub-buffers, 1476 // and expresses its desire to server as the notification frame count. 1477 if (mSharedBuffer == 0 && audio_is_linear_pcm(mFormat)) { 1478 size_t maxNotificationFrames; 1479 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1480 // notify every HAL buffer, regardless of the size of the track buffer 1481 maxNotificationFrames = afFrameCountHAL; 1482 } else { 1483 // For normal tracks, use at least double-buffering if no sample rate conversion, 1484 // or at least triple-buffering if there is sample rate conversion 1485 const int nBuffering = mOriginalSampleRate == mAfSampleRate ? 2 : 3; 1486 maxNotificationFrames = frameCount / nBuffering; 1487 } 1488 if (mNotificationFramesAct == 0 || mNotificationFramesAct > maxNotificationFrames) { 1489 if (mNotificationFramesAct == 0) { 1490 ALOGD("Client defaulted notificationFrames to %zu for frameCount %zu", 1491 maxNotificationFrames, frameCount); 1492 } else { 1493 ALOGW("Client adjusted notificationFrames from %u to %zu for frameCount %zu", 1494 mNotificationFramesAct, maxNotificationFrames, frameCount); 1495 } 1496 mNotificationFramesAct = (uint32_t) maxNotificationFrames; 1497 } 1498 } 1499 1500 // We retain a copy of the I/O handle, but don't own the reference 1501 mOutput = output; 1502 mRefreshRemaining = true; 1503 1504 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1505 // is the value of pointer() for the shared buffer, otherwise buffers points 1506 // immediately after the control block. This address is for the mapping within client 1507 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1508 void* buffers; 1509 if (mSharedBuffer == 0) { 1510 buffers = cblk + 1; 1511 } else { 1512 buffers = mSharedBuffer->pointer(); 1513 if (buffers == NULL) { 1514 ALOGE("Could not get buffer pointer"); 1515 return NO_INIT; 1516 } 1517 } 1518 1519 mAudioTrack->attachAuxEffect(mAuxEffectId); 1520 // FIXME doesn't take into account speed or future sample rate changes (until restoreTrack) 1521 // FIXME don't believe this lie 1522 mLatency = mAfLatency + (1000*frameCount) / mSampleRate; 1523 1524 mFrameCount = frameCount; 1525 // If IAudioTrack is re-created, don't let the requested frameCount 1526 // decrease. This can confuse clients that cache frameCount(). 1527 if (frameCount > mReqFrameCount) { 1528 mReqFrameCount = frameCount; 1529 } 1530 1531 // reset server position to 0 as we have new cblk. 1532 mServer = 0; 1533 1534 // update proxy 1535 if (mSharedBuffer == 0) { 1536 mStaticProxy.clear(); 1537 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize); 1538 } else { 1539 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize); 1540 mProxy = mStaticProxy; 1541 } 1542 1543 mProxy->setVolumeLR(gain_minifloat_pack( 1544 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]), 1545 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT]))); 1546 1547 mProxy->setSendLevel(mSendLevel); 1548 const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch); 1549 const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch); 1550 const float effectivePitch = adjustPitch(mPlaybackRate.mPitch); 1551 mProxy->setSampleRate(effectiveSampleRate); 1552 1553 AudioPlaybackRate playbackRateTemp = mPlaybackRate; 1554 playbackRateTemp.mSpeed = effectiveSpeed; 1555 playbackRateTemp.mPitch = effectivePitch; 1556 mProxy->setPlaybackRate(playbackRateTemp); 1557 mProxy->setMinimum(mNotificationFramesAct); 1558 1559 mDeathNotifier = new DeathNotifier(this); 1560 IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this); 1561 1562 if (mDeviceCallback != 0) { 1563 AudioSystem::addAudioDeviceCallback(mDeviceCallback, mOutput); 1564 } 1565 1566 return NO_ERROR; 1567 } 1568 1569release: 1570 AudioSystem::releaseOutput(output, streamType, mSessionId); 1571 if (status == NO_ERROR) { 1572 status = NO_INIT; 1573 } 1574 return status; 1575} 1576 1577status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig) 1578{ 1579 if (audioBuffer == NULL) { 1580 if (nonContig != NULL) { 1581 *nonContig = 0; 1582 } 1583 return BAD_VALUE; 1584 } 1585 if (mTransfer != TRANSFER_OBTAIN) { 1586 audioBuffer->frameCount = 0; 1587 audioBuffer->size = 0; 1588 audioBuffer->raw = NULL; 1589 if (nonContig != NULL) { 1590 *nonContig = 0; 1591 } 1592 return INVALID_OPERATION; 1593 } 1594 1595 const struct timespec *requested; 1596 struct timespec timeout; 1597 if (waitCount == -1) { 1598 requested = &ClientProxy::kForever; 1599 } else if (waitCount == 0) { 1600 requested = &ClientProxy::kNonBlocking; 1601 } else if (waitCount > 0) { 1602 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1603 timeout.tv_sec = ms / 1000; 1604 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1605 requested = &timeout; 1606 } else { 1607 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1608 requested = NULL; 1609 } 1610 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig); 1611} 1612 1613status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1614 struct timespec *elapsed, size_t *nonContig) 1615{ 1616 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1617 uint32_t oldSequence = 0; 1618 uint32_t newSequence; 1619 1620 Proxy::Buffer buffer; 1621 status_t status = NO_ERROR; 1622 1623 static const int32_t kMaxTries = 5; 1624 int32_t tryCounter = kMaxTries; 1625 1626 do { 1627 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1628 // keep them from going away if another thread re-creates the track during obtainBuffer() 1629 sp<AudioTrackClientProxy> proxy; 1630 sp<IMemory> iMem; 1631 1632 { // start of lock scope 1633 AutoMutex lock(mLock); 1634 1635 newSequence = mSequence; 1636 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1637 if (status == DEAD_OBJECT) { 1638 // re-create track, unless someone else has already done so 1639 if (newSequence == oldSequence) { 1640 status = restoreTrack_l("obtainBuffer"); 1641 if (status != NO_ERROR) { 1642 buffer.mFrameCount = 0; 1643 buffer.mRaw = NULL; 1644 buffer.mNonContig = 0; 1645 break; 1646 } 1647 } 1648 } 1649 oldSequence = newSequence; 1650 1651 if (status == NOT_ENOUGH_DATA) { 1652 restartIfDisabled(); 1653 } 1654 1655 // Keep the extra references 1656 proxy = mProxy; 1657 iMem = mCblkMemory; 1658 1659 if (mState == STATE_STOPPING) { 1660 status = -EINTR; 1661 buffer.mFrameCount = 0; 1662 buffer.mRaw = NULL; 1663 buffer.mNonContig = 0; 1664 break; 1665 } 1666 1667 // Non-blocking if track is stopped or paused 1668 if (mState != STATE_ACTIVE) { 1669 requested = &ClientProxy::kNonBlocking; 1670 } 1671 1672 } // end of lock scope 1673 1674 buffer.mFrameCount = audioBuffer->frameCount; 1675 // FIXME starts the requested timeout and elapsed over from scratch 1676 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1677 } while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0)); 1678 1679 audioBuffer->frameCount = buffer.mFrameCount; 1680 audioBuffer->size = buffer.mFrameCount * mFrameSize; 1681 audioBuffer->raw = buffer.mRaw; 1682 if (nonContig != NULL) { 1683 *nonContig = buffer.mNonContig; 1684 } 1685 return status; 1686} 1687 1688void AudioTrack::releaseBuffer(const Buffer* audioBuffer) 1689{ 1690 // FIXME add error checking on mode, by adding an internal version 1691 if (mTransfer == TRANSFER_SHARED) { 1692 return; 1693 } 1694 1695 size_t stepCount = audioBuffer->size / mFrameSize; 1696 if (stepCount == 0) { 1697 return; 1698 } 1699 1700 Proxy::Buffer buffer; 1701 buffer.mFrameCount = stepCount; 1702 buffer.mRaw = audioBuffer->raw; 1703 1704 AutoMutex lock(mLock); 1705 mReleased += stepCount; 1706 mInUnderrun = false; 1707 mProxy->releaseBuffer(&buffer); 1708 1709 // restart track if it was disabled by audioflinger due to previous underrun 1710 restartIfDisabled(); 1711} 1712 1713void AudioTrack::restartIfDisabled() 1714{ 1715 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 1716 if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) { 1717 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1718 // FIXME ignoring status 1719 mAudioTrack->start(); 1720 } 1721} 1722 1723// ------------------------------------------------------------------------- 1724 1725ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1726{ 1727 if (mTransfer != TRANSFER_SYNC) { 1728 return INVALID_OPERATION; 1729 } 1730 1731 if (isDirect()) { 1732 AutoMutex lock(mLock); 1733 int32_t flags = android_atomic_and( 1734 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), 1735 &mCblk->mFlags); 1736 if (flags & CBLK_INVALID) { 1737 return DEAD_OBJECT; 1738 } 1739 } 1740 1741 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1742 // Sanity-check: user is most-likely passing an error code, and it would 1743 // make the return value ambiguous (actualSize vs error). 1744 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1745 return BAD_VALUE; 1746 } 1747 1748 size_t written = 0; 1749 Buffer audioBuffer; 1750 1751 while (userSize >= mFrameSize) { 1752 audioBuffer.frameCount = userSize / mFrameSize; 1753 1754 status_t err = obtainBuffer(&audioBuffer, 1755 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1756 if (err < 0) { 1757 if (written > 0) { 1758 break; 1759 } 1760 if (err == TIMED_OUT || err == -EINTR) { 1761 err = WOULD_BLOCK; 1762 } 1763 return ssize_t(err); 1764 } 1765 1766 size_t toWrite = audioBuffer.size; 1767 memcpy(audioBuffer.i8, buffer, toWrite); 1768 buffer = ((const char *) buffer) + toWrite; 1769 userSize -= toWrite; 1770 written += toWrite; 1771 1772 releaseBuffer(&audioBuffer); 1773 } 1774 1775 if (written > 0) { 1776 mFramesWritten += written / mFrameSize; 1777 } 1778 return written; 1779} 1780 1781// ------------------------------------------------------------------------- 1782 1783nsecs_t AudioTrack::processAudioBuffer() 1784{ 1785 // Currently the AudioTrack thread is not created if there are no callbacks. 1786 // Would it ever make sense to run the thread, even without callbacks? 1787 // If so, then replace this by checks at each use for mCbf != NULL. 1788 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1789 1790 mLock.lock(); 1791 if (mAwaitBoost) { 1792 mAwaitBoost = false; 1793 mLock.unlock(); 1794 static const int32_t kMaxTries = 5; 1795 int32_t tryCounter = kMaxTries; 1796 uint32_t pollUs = 10000; 1797 do { 1798 int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK; 1799 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1800 break; 1801 } 1802 usleep(pollUs); 1803 pollUs <<= 1; 1804 } while (tryCounter-- > 0); 1805 if (tryCounter < 0) { 1806 ALOGE("did not receive expected priority boost on time"); 1807 } 1808 // Run again immediately 1809 return 0; 1810 } 1811 1812 // Can only reference mCblk while locked 1813 int32_t flags = android_atomic_and( 1814 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1815 1816 // Check for track invalidation 1817 if (flags & CBLK_INVALID) { 1818 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1819 // AudioSystem cache. We should not exit here but after calling the callback so 1820 // that the upper layers can recreate the track 1821 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) { 1822 status_t status __unused = restoreTrack_l("processAudioBuffer"); 1823 // FIXME unused status 1824 // after restoration, continue below to make sure that the loop and buffer events 1825 // are notified because they have been cleared from mCblk->mFlags above. 1826 } 1827 } 1828 1829 bool waitStreamEnd = mState == STATE_STOPPING; 1830 bool active = mState == STATE_ACTIVE; 1831 1832 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1833 bool newUnderrun = false; 1834 if (flags & CBLK_UNDERRUN) { 1835#if 0 1836 // Currently in shared buffer mode, when the server reaches the end of buffer, 1837 // the track stays active in continuous underrun state. It's up to the application 1838 // to pause or stop the track, or set the position to a new offset within buffer. 1839 // This was some experimental code to auto-pause on underrun. Keeping it here 1840 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1841 if (mTransfer == TRANSFER_SHARED) { 1842 mState = STATE_PAUSED; 1843 active = false; 1844 } 1845#endif 1846 if (!mInUnderrun) { 1847 mInUnderrun = true; 1848 newUnderrun = true; 1849 } 1850 } 1851 1852 // Get current position of server 1853 Modulo<uint32_t> position(updateAndGetPosition_l()); 1854 1855 // Manage marker callback 1856 bool markerReached = false; 1857 Modulo<uint32_t> markerPosition(mMarkerPosition); 1858 // uses 32 bit wraparound for comparison with position. 1859 if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) { 1860 mMarkerReached = markerReached = true; 1861 } 1862 1863 // Determine number of new position callback(s) that will be needed, while locked 1864 size_t newPosCount = 0; 1865 Modulo<uint32_t> newPosition(mNewPosition); 1866 uint32_t updatePeriod = mUpdatePeriod; 1867 // FIXME fails for wraparound, need 64 bits 1868 if (updatePeriod > 0 && position >= newPosition) { 1869 newPosCount = ((position - newPosition).value() / updatePeriod) + 1; 1870 mNewPosition += updatePeriod * newPosCount; 1871 } 1872 1873 // Cache other fields that will be needed soon 1874 uint32_t sampleRate = mSampleRate; 1875 float speed = mPlaybackRate.mSpeed; 1876 const uint32_t notificationFrames = mNotificationFramesAct; 1877 if (mRefreshRemaining) { 1878 mRefreshRemaining = false; 1879 mRemainingFrames = notificationFrames; 1880 mRetryOnPartialBuffer = false; 1881 } 1882 size_t misalignment = mProxy->getMisalignment(); 1883 uint32_t sequence = mSequence; 1884 sp<AudioTrackClientProxy> proxy = mProxy; 1885 1886 // Determine the number of new loop callback(s) that will be needed, while locked. 1887 int loopCountNotifications = 0; 1888 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END 1889 1890 if (mLoopCount > 0) { 1891 int loopCount; 1892 size_t bufferPosition; 1893 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount); 1894 loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition; 1895 loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications); 1896 mLoopCountNotified = loopCount; // discard any excess notifications 1897 } else if (mLoopCount < 0) { 1898 // FIXME: We're not accurate with notification count and position with infinite looping 1899 // since loopCount from server side will always return -1 (we could decrement it). 1900 size_t bufferPosition = mStaticProxy->getBufferPosition(); 1901 loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0); 1902 loopPeriod = mLoopEnd - bufferPosition; 1903 } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) { 1904 size_t bufferPosition = mStaticProxy->getBufferPosition(); 1905 loopPeriod = mFrameCount - bufferPosition; 1906 } 1907 1908 // These fields don't need to be cached, because they are assigned only by set(): 1909 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFlags 1910 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1911 1912 mLock.unlock(); 1913 1914 // get anchor time to account for callbacks. 1915 const nsecs_t timeBeforeCallbacks = systemTime(); 1916 1917 if (waitStreamEnd) { 1918 // FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread 1919 // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function 1920 // (and make sure we don't callback for more data while we're stopping). 1921 // This helps with position, marker notifications, and track invalidation. 1922 struct timespec timeout; 1923 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1924 timeout.tv_nsec = 0; 1925 1926 status_t status = proxy->waitStreamEndDone(&timeout); 1927 switch (status) { 1928 case NO_ERROR: 1929 case DEAD_OBJECT: 1930 case TIMED_OUT: 1931 if (status != DEAD_OBJECT) { 1932 // for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop(); 1933 // instead, the application should handle the EVENT_NEW_IAUDIOTRACK. 1934 mCbf(EVENT_STREAM_END, mUserData, NULL); 1935 } 1936 { 1937 AutoMutex lock(mLock); 1938 // The previously assigned value of waitStreamEnd is no longer valid, 1939 // since the mutex has been unlocked and either the callback handler 1940 // or another thread could have re-started the AudioTrack during that time. 1941 waitStreamEnd = mState == STATE_STOPPING; 1942 if (waitStreamEnd) { 1943 mState = STATE_STOPPED; 1944 mReleased = 0; 1945 } 1946 } 1947 if (waitStreamEnd && status != DEAD_OBJECT) { 1948 return NS_INACTIVE; 1949 } 1950 break; 1951 } 1952 return 0; 1953 } 1954 1955 // perform callbacks while unlocked 1956 if (newUnderrun) { 1957 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1958 } 1959 while (loopCountNotifications > 0) { 1960 mCbf(EVENT_LOOP_END, mUserData, NULL); 1961 --loopCountNotifications; 1962 } 1963 if (flags & CBLK_BUFFER_END) { 1964 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1965 } 1966 if (markerReached) { 1967 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1968 } 1969 while (newPosCount > 0) { 1970 size_t temp = newPosition.value(); // FIXME size_t != uint32_t 1971 mCbf(EVENT_NEW_POS, mUserData, &temp); 1972 newPosition += updatePeriod; 1973 newPosCount--; 1974 } 1975 1976 if (mObservedSequence != sequence) { 1977 mObservedSequence = sequence; 1978 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1979 // for offloaded tracks, just wait for the upper layers to recreate the track 1980 if (isOffloadedOrDirect()) { 1981 return NS_INACTIVE; 1982 } 1983 } 1984 1985 // if inactive, then don't run me again until re-started 1986 if (!active) { 1987 return NS_INACTIVE; 1988 } 1989 1990 // Compute the estimated time until the next timed event (position, markers, loops) 1991 // FIXME only for non-compressed audio 1992 uint32_t minFrames = ~0; 1993 if (!markerReached && position < markerPosition) { 1994 minFrames = (markerPosition - position).value(); 1995 } 1996 if (loopPeriod > 0 && loopPeriod < minFrames) { 1997 // loopPeriod is already adjusted for actual position. 1998 minFrames = loopPeriod; 1999 } 2000 if (updatePeriod > 0) { 2001 minFrames = min(minFrames, (newPosition - position).value()); 2002 } 2003 2004 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 2005 static const uint32_t kPoll = 0; 2006 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 2007 minFrames = kPoll * notificationFrames; 2008 } 2009 2010 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 2011 static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL; 2012 const nsecs_t timeAfterCallbacks = systemTime(); 2013 2014 // Convert frame units to time units 2015 nsecs_t ns = NS_WHENEVER; 2016 if (minFrames != (uint32_t) ~0) { 2017 ns = framesToNanoseconds(minFrames, sampleRate, speed) + kWaitPeriodNs; 2018 ns -= (timeAfterCallbacks - timeBeforeCallbacks); // account for callback time 2019 // TODO: Should we warn if the callback time is too long? 2020 if (ns < 0) ns = 0; 2021 } 2022 2023 // If not supplying data by EVENT_MORE_DATA, then we're done 2024 if (mTransfer != TRANSFER_CALLBACK) { 2025 return ns; 2026 } 2027 2028 // EVENT_MORE_DATA callback handling. 2029 // Timing for linear pcm audio data formats can be derived directly from the 2030 // buffer fill level. 2031 // Timing for compressed data is not directly available from the buffer fill level, 2032 // rather indirectly from waiting for blocking mode callbacks or waiting for obtain() 2033 // to return a certain fill level. 2034 2035 struct timespec timeout; 2036 const struct timespec *requested = &ClientProxy::kForever; 2037 if (ns != NS_WHENEVER) { 2038 timeout.tv_sec = ns / 1000000000LL; 2039 timeout.tv_nsec = ns % 1000000000LL; 2040 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 2041 requested = &timeout; 2042 } 2043 2044 size_t writtenFrames = 0; 2045 while (mRemainingFrames > 0) { 2046 2047 Buffer audioBuffer; 2048 audioBuffer.frameCount = mRemainingFrames; 2049 size_t nonContig; 2050 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 2051 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 2052 "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount); 2053 requested = &ClientProxy::kNonBlocking; 2054 size_t avail = audioBuffer.frameCount + nonContig; 2055 ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d", 2056 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 2057 if (err != NO_ERROR) { 2058 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 2059 (isOffloaded() && (err == DEAD_OBJECT))) { 2060 // FIXME bug 25195759 2061 return 1000000; 2062 } 2063 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 2064 return NS_NEVER; 2065 } 2066 2067 if (mRetryOnPartialBuffer && audio_has_proportional_frames(mFormat)) { 2068 mRetryOnPartialBuffer = false; 2069 if (avail < mRemainingFrames) { 2070 if (ns > 0) { // account for obtain time 2071 const nsecs_t timeNow = systemTime(); 2072 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks)); 2073 } 2074 nsecs_t myns = framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed); 2075 if (ns < 0 /* NS_WHENEVER */ || myns < ns) { 2076 ns = myns; 2077 } 2078 return ns; 2079 } 2080 } 2081 2082 size_t reqSize = audioBuffer.size; 2083 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 2084 size_t writtenSize = audioBuffer.size; 2085 2086 // Sanity check on returned size 2087 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 2088 ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes", 2089 reqSize, ssize_t(writtenSize)); 2090 return NS_NEVER; 2091 } 2092 2093 if (writtenSize == 0) { 2094 // The callback is done filling buffers 2095 // Keep this thread going to handle timed events and 2096 // still try to get more data in intervals of WAIT_PERIOD_MS 2097 // but don't just loop and block the CPU, so wait 2098 2099 // mCbf(EVENT_MORE_DATA, ...) might either 2100 // (1) Block until it can fill the buffer, returning 0 size on EOS. 2101 // (2) Block until it can fill the buffer, returning 0 data (silence) on EOS. 2102 // (3) Return 0 size when no data is available, does not wait for more data. 2103 // 2104 // (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer. 2105 // We try to compute the wait time to avoid a tight sleep-wait cycle, 2106 // especially for case (3). 2107 // 2108 // The decision to support (1) and (2) affect the sizing of mRemainingFrames 2109 // and this loop; whereas for case (3) we could simply check once with the full 2110 // buffer size and skip the loop entirely. 2111 2112 nsecs_t myns; 2113 if (audio_has_proportional_frames(mFormat)) { 2114 // time to wait based on buffer occupancy 2115 const nsecs_t datans = mRemainingFrames <= avail ? 0 : 2116 framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed); 2117 // audio flinger thread buffer size (TODO: adjust for fast tracks) 2118 // FIXME: use mAfFrameCountHAL instead of mAfFrameCount below for fast tracks. 2119 const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed); 2120 // add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0. 2121 myns = datans + (afns / 2); 2122 } else { 2123 // FIXME: This could ping quite a bit if the buffer isn't full. 2124 // Note that when mState is stopping we waitStreamEnd, so it never gets here. 2125 myns = kWaitPeriodNs; 2126 } 2127 if (ns > 0) { // account for obtain and callback time 2128 const nsecs_t timeNow = systemTime(); 2129 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks)); 2130 } 2131 if (ns < 0 /* NS_WHENEVER */ || myns < ns) { 2132 ns = myns; 2133 } 2134 return ns; 2135 } 2136 2137 size_t releasedFrames = writtenSize / mFrameSize; 2138 audioBuffer.frameCount = releasedFrames; 2139 mRemainingFrames -= releasedFrames; 2140 if (misalignment >= releasedFrames) { 2141 misalignment -= releasedFrames; 2142 } else { 2143 misalignment = 0; 2144 } 2145 2146 releaseBuffer(&audioBuffer); 2147 writtenFrames += releasedFrames; 2148 2149 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 2150 // if callback doesn't like to accept the full chunk 2151 if (writtenSize < reqSize) { 2152 continue; 2153 } 2154 2155 // There could be enough non-contiguous frames available to satisfy the remaining request 2156 if (mRemainingFrames <= nonContig) { 2157 continue; 2158 } 2159 2160#if 0 2161 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 2162 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 2163 // that total to a sum == notificationFrames. 2164 if (0 < misalignment && misalignment <= mRemainingFrames) { 2165 mRemainingFrames = misalignment; 2166 return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed); 2167 } 2168#endif 2169 2170 } 2171 if (writtenFrames > 0) { 2172 AutoMutex lock(mLock); 2173 mFramesWritten += writtenFrames; 2174 } 2175 mRemainingFrames = notificationFrames; 2176 mRetryOnPartialBuffer = true; 2177 2178 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 2179 return 0; 2180} 2181 2182status_t AudioTrack::restoreTrack_l(const char *from) 2183{ 2184 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 2185 isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from); 2186 ++mSequence; 2187 2188 // refresh the audio configuration cache in this process to make sure we get new 2189 // output parameters and new IAudioFlinger in createTrack_l() 2190 AudioSystem::clearAudioConfigCache(); 2191 2192 if (isOffloadedOrDirect_l() || mDoNotReconnect) { 2193 // FIXME re-creation of offloaded and direct tracks is not yet implemented; 2194 // reconsider enabling for linear PCM encodings when position can be preserved. 2195 return DEAD_OBJECT; 2196 } 2197 2198 // Save so we can return count since creation. 2199 mUnderrunCountOffset = getUnderrunCount_l(); 2200 2201 // save the old static buffer position 2202 uint32_t staticPosition = 0; 2203 size_t bufferPosition = 0; 2204 int loopCount = 0; 2205 if (mStaticProxy != 0) { 2206 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount); 2207 staticPosition = mStaticProxy->getPosition().unsignedValue(); 2208 } 2209 2210 mFlags = mOrigFlags; 2211 2212 // If a new IAudioTrack is successfully created, createTrack_l() will modify the 2213 // following member variables: mAudioTrack, mCblkMemory and mCblk. 2214 // It will also delete the strong references on previous IAudioTrack and IMemory. 2215 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact. 2216 status_t result = createTrack_l(); 2217 2218 if (result == NO_ERROR) { 2219 // take the frames that will be lost by track recreation into account in saved position 2220 // For streaming tracks, this is the amount we obtained from the user/client 2221 // (not the number actually consumed at the server - those are already lost). 2222 if (mStaticProxy == 0) { 2223 mPosition = mReleased; 2224 } 2225 // Continue playback from last known position and restore loop. 2226 if (mStaticProxy != 0) { 2227 if (loopCount != 0) { 2228 mStaticProxy->setBufferPositionAndLoop(bufferPosition, 2229 mLoopStart, mLoopEnd, loopCount); 2230 } else { 2231 mStaticProxy->setBufferPosition(bufferPosition); 2232 if (bufferPosition == mFrameCount) { 2233 ALOGD("restoring track at end of static buffer"); 2234 } 2235 } 2236 } 2237 if (mState == STATE_ACTIVE) { 2238 result = mAudioTrack->start(); 2239 } 2240 // server resets to zero so we offset 2241 mFramesWrittenServerOffset = 2242 mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten; 2243 mFramesWrittenAtRestore = mFramesWrittenServerOffset; 2244 } 2245 if (result != NO_ERROR) { 2246 ALOGW("restoreTrack_l() failed status %d", result); 2247 mState = STATE_STOPPED; 2248 mReleased = 0; 2249 } 2250 2251 return result; 2252} 2253 2254Modulo<uint32_t> AudioTrack::updateAndGetPosition_l() 2255{ 2256 // This is the sole place to read server consumed frames 2257 Modulo<uint32_t> newServer(mProxy->getPosition()); 2258 const int32_t delta = (newServer - mServer).signedValue(); 2259 // TODO There is controversy about whether there can be "negative jitter" in server position. 2260 // This should be investigated further, and if possible, it should be addressed. 2261 // A more definite failure mode is infrequent polling by client. 2262 // One could call (void)getPosition_l() in releaseBuffer(), 2263 // so mReleased and mPosition are always lock-step as best possible. 2264 // That should ensure delta never goes negative for infrequent polling 2265 // unless the server has more than 2^31 frames in its buffer, 2266 // in which case the use of uint32_t for these counters has bigger issues. 2267 ALOGE_IF(delta < 0, 2268 "detected illegal retrograde motion by the server: mServer advanced by %d", 2269 delta); 2270 mServer = newServer; 2271 if (delta > 0) { // avoid retrograde 2272 mPosition += delta; 2273 } 2274 return mPosition; 2275} 2276 2277bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const 2278{ 2279 // applicable for mixing tracks only (not offloaded or direct) 2280 if (mStaticProxy != 0) { 2281 return true; // static tracks do not have issues with buffer sizing. 2282 } 2283 const size_t minFrameCount = 2284 calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed 2285 /*, 0 mNotificationsPerBufferReq*/); 2286 ALOGV("isSampleRateSpeedAllowed_l mFrameCount %zu minFrameCount %zu", 2287 mFrameCount, minFrameCount); 2288 return mFrameCount >= minFrameCount; 2289} 2290 2291status_t AudioTrack::setParameters(const String8& keyValuePairs) 2292{ 2293 AutoMutex lock(mLock); 2294 return mAudioTrack->setParameters(keyValuePairs); 2295} 2296 2297status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp) 2298{ 2299 if (timestamp == nullptr) { 2300 return BAD_VALUE; 2301 } 2302 AutoMutex lock(mLock); 2303 return getTimestamp_l(timestamp); 2304} 2305 2306status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp) 2307{ 2308 if (mCblk->mFlags & CBLK_INVALID) { 2309 const status_t status = restoreTrack_l("getTimestampExtended"); 2310 if (status != OK) { 2311 // per getTimestamp() API doc in header, we return DEAD_OBJECT here, 2312 // recommending that the track be recreated. 2313 return DEAD_OBJECT; 2314 } 2315 } 2316 // check for offloaded/direct here in case restoring somehow changed those flags. 2317 if (isOffloadedOrDirect_l()) { 2318 return INVALID_OPERATION; // not supported 2319 } 2320 status_t status = mProxy->getTimestamp(timestamp); 2321 LOG_ALWAYS_FATAL_IF(status != OK, "status %d not allowed from proxy getTimestamp", status); 2322 bool found = false; 2323 timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten; 2324 timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0; 2325 // server side frame offset in case AudioTrack has been restored. 2326 for (int i = ExtendedTimestamp::LOCATION_SERVER; 2327 i < ExtendedTimestamp::LOCATION_MAX; ++i) { 2328 if (timestamp->mTimeNs[i] >= 0) { 2329 // apply server offset (frames flushed is ignored 2330 // so we don't report the jump when the flush occurs). 2331 timestamp->mPosition[i] += mFramesWrittenServerOffset; 2332 found = true; 2333 } 2334 } 2335 return found ? OK : WOULD_BLOCK; 2336} 2337 2338status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 2339{ 2340 AutoMutex lock(mLock); 2341 return getTimestamp_l(timestamp); 2342} 2343 2344status_t AudioTrack::getTimestamp_l(AudioTimestamp& timestamp) 2345{ 2346 bool previousTimestampValid = mPreviousTimestampValid; 2347 // Set false here to cover all the error return cases. 2348 mPreviousTimestampValid = false; 2349 2350 switch (mState) { 2351 case STATE_ACTIVE: 2352 case STATE_PAUSED: 2353 break; // handle below 2354 case STATE_FLUSHED: 2355 case STATE_STOPPED: 2356 return WOULD_BLOCK; 2357 case STATE_STOPPING: 2358 case STATE_PAUSED_STOPPING: 2359 if (!isOffloaded_l()) { 2360 return INVALID_OPERATION; 2361 } 2362 break; // offloaded tracks handled below 2363 default: 2364 LOG_ALWAYS_FATAL("Invalid mState in getTimestamp(): %d", mState); 2365 break; 2366 } 2367 2368 if (mCblk->mFlags & CBLK_INVALID) { 2369 const status_t status = restoreTrack_l("getTimestamp"); 2370 if (status != OK) { 2371 // per getTimestamp() API doc in header, we return DEAD_OBJECT here, 2372 // recommending that the track be recreated. 2373 return DEAD_OBJECT; 2374 } 2375 } 2376 2377 // The presented frame count must always lag behind the consumed frame count. 2378 // To avoid a race, read the presented frames first. This ensures that presented <= consumed. 2379 2380 status_t status; 2381 if (isOffloadedOrDirect_l()) { 2382 // use Binder to get timestamp 2383 status = mAudioTrack->getTimestamp(timestamp); 2384 } else { 2385 // read timestamp from shared memory 2386 ExtendedTimestamp ets; 2387 status = mProxy->getTimestamp(&ets); 2388 if (status == OK) { 2389 ExtendedTimestamp::Location location; 2390 status = ets.getBestTimestamp(×tamp, &location); 2391 2392 if (status == OK) { 2393 // It is possible that the best location has moved from the kernel to the server. 2394 // In this case we adjust the position from the previous computed latency. 2395 if (location == ExtendedTimestamp::LOCATION_SERVER) { 2396 ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL, 2397 "getTimestamp() location moved from kernel to server"); 2398 // check that the last kernel OK time info exists and the positions 2399 // are valid (if they predate the current track, the positions may 2400 // be zero or negative). 2401 const int64_t frames = 2402 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 || 2403 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 || 2404 ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 || 2405 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0) 2406 ? 2407 int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed 2408 / 1000) 2409 : 2410 (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] 2411 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]); 2412 ALOGV("frame adjustment:%lld timestamp:%s", 2413 (long long)frames, ets.toString().c_str()); 2414 if (frames >= ets.mPosition[location]) { 2415 timestamp.mPosition = 0; 2416 } else { 2417 timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames); 2418 } 2419 } else if (location == ExtendedTimestamp::LOCATION_KERNEL) { 2420 ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER, 2421 "getTimestamp() location moved from server to kernel"); 2422 } 2423 2424 // We update the timestamp time even when paused. 2425 if (mState == STATE_PAUSED /* not needed: STATE_PAUSED_STOPPING */) { 2426 const int64_t now = systemTime(); 2427 const int64_t at = convertTimespecToNs(timestamp.mTime); 2428 const int64_t lag = 2429 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 || 2430 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0) 2431 ? int64_t(mAfLatency * 1000000LL) 2432 : (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] 2433 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]) 2434 * NANOS_PER_SECOND / mSampleRate; 2435 const int64_t limit = now - lag; // no earlier than this limit 2436 if (at < limit) { 2437 ALOGV("timestamp pause lag:%lld adjusting from %lld to %lld", 2438 (long long)lag, (long long)at, (long long)limit); 2439 timestamp.mTime.tv_sec = limit / NANOS_PER_SECOND; 2440 timestamp.mTime.tv_nsec = limit % NANOS_PER_SECOND; // compiler opt. 2441 } 2442 } 2443 mPreviousLocation = location; 2444 } else { 2445 // right after AudioTrack is started, one may not find a timestamp 2446 ALOGV("getBestTimestamp did not find timestamp"); 2447 } 2448 } 2449 if (status == INVALID_OPERATION) { 2450 // INVALID_OPERATION occurs when no timestamp has been issued by the server; 2451 // other failures are signaled by a negative time. 2452 // If we come out of FLUSHED or STOPPED where the position is known 2453 // to be zero we convert this to WOULD_BLOCK (with the implicit meaning of 2454 // "zero" for NuPlayer). We don't convert for track restoration as position 2455 // does not reset. 2456 ALOGV("timestamp server offset:%lld restore frames:%lld", 2457 (long long)mFramesWrittenServerOffset, (long long)mFramesWrittenAtRestore); 2458 if (mFramesWrittenServerOffset != mFramesWrittenAtRestore) { 2459 status = WOULD_BLOCK; 2460 } 2461 } 2462 } 2463 if (status != NO_ERROR) { 2464 ALOGV_IF(status != WOULD_BLOCK, "getTimestamp error:%#x", status); 2465 return status; 2466 } 2467 if (isOffloadedOrDirect_l()) { 2468 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) { 2469 // use cached paused position in case another offloaded track is running. 2470 timestamp.mPosition = mPausedPosition; 2471 clock_gettime(CLOCK_MONOTONIC, ×tamp.mTime); 2472 // TODO: adjust for delay 2473 return NO_ERROR; 2474 } 2475 2476 // Check whether a pending flush or stop has completed, as those commands may 2477 // be asynchronous or return near finish or exhibit glitchy behavior. 2478 // 2479 // Originally this showed up as the first timestamp being a continuation of 2480 // the previous song under gapless playback. 2481 // However, we sometimes see zero timestamps, then a glitch of 2482 // the previous song's position, and then correct timestamps afterwards. 2483 if (mStartUs != 0 && mSampleRate != 0) { 2484 static const int kTimeJitterUs = 100000; // 100 ms 2485 static const int k1SecUs = 1000000; 2486 2487 const int64_t timeNow = getNowUs(); 2488 2489 if (timeNow < mStartUs + k1SecUs) { // within first second of starting 2490 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime); 2491 if (timestampTimeUs < mStartUs) { 2492 return WOULD_BLOCK; // stale timestamp time, occurs before start. 2493 } 2494 const int64_t deltaTimeUs = timestampTimeUs - mStartUs; 2495 const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000 2496 / ((double)mSampleRate * mPlaybackRate.mSpeed); 2497 2498 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) { 2499 // Verify that the counter can't count faster than the sample rate 2500 // since the start time. If greater, then that means we may have failed 2501 // to completely flush or stop the previous playing track. 2502 ALOGW_IF(!mTimestampStartupGlitchReported, 2503 "getTimestamp startup glitch detected" 2504 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)", 2505 (long long)deltaTimeUs, (long long)deltaPositionByUs, 2506 timestamp.mPosition); 2507 mTimestampStartupGlitchReported = true; 2508 if (previousTimestampValid 2509 && mPreviousTimestamp.mPosition == 0 /* should be true if valid */) { 2510 timestamp = mPreviousTimestamp; 2511 mPreviousTimestampValid = true; 2512 return NO_ERROR; 2513 } 2514 return WOULD_BLOCK; 2515 } 2516 if (deltaPositionByUs != 0) { 2517 mStartUs = 0; // don't check again, we got valid nonzero position. 2518 } 2519 } else { 2520 mStartUs = 0; // don't check again, start time expired. 2521 } 2522 mTimestampStartupGlitchReported = false; 2523 } 2524 } else { 2525 // Update the mapping between local consumed (mPosition) and server consumed (mServer) 2526 (void) updateAndGetPosition_l(); 2527 // Server consumed (mServer) and presented both use the same server time base, 2528 // and server consumed is always >= presented. 2529 // The delta between these represents the number of frames in the buffer pipeline. 2530 // If this delta between these is greater than the client position, it means that 2531 // actually presented is still stuck at the starting line (figuratively speaking), 2532 // waiting for the first frame to go by. So we can't report a valid timestamp yet. 2533 // Note: We explicitly use non-Modulo comparison here - potential wrap issue when 2534 // mPosition exceeds 32 bits. 2535 // TODO Remove when timestamp is updated to contain pipeline status info. 2536 const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue(); 2537 if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */ 2538 && (uint32_t)pipelineDepthInFrames > mPosition.value()) { 2539 return INVALID_OPERATION; 2540 } 2541 // Convert timestamp position from server time base to client time base. 2542 // TODO The following code should work OK now because timestamp.mPosition is 32-bit. 2543 // But if we change it to 64-bit then this could fail. 2544 // Use Modulo computation here. 2545 timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value(); 2546 // Immediately after a call to getPosition_l(), mPosition and 2547 // mServer both represent the same frame position. mPosition is 2548 // in client's point of view, and mServer is in server's point of 2549 // view. So the difference between them is the "fudge factor" 2550 // between client and server views due to stop() and/or new 2551 // IAudioTrack. And timestamp.mPosition is initially in server's 2552 // point of view, so we need to apply the same fudge factor to it. 2553 } 2554 2555 // Prevent retrograde motion in timestamp. 2556 // This is sometimes caused by erratic reports of the available space in the ALSA drivers. 2557 if (status == NO_ERROR) { 2558 if (previousTimestampValid) { 2559 const int64_t previousTimeNanos = convertTimespecToNs(mPreviousTimestamp.mTime); 2560 const int64_t currentTimeNanos = convertTimespecToNs(timestamp.mTime); 2561 if (currentTimeNanos < previousTimeNanos) { 2562 ALOGW("retrograde timestamp time corrected, %lld < %lld", 2563 (long long)currentTimeNanos, (long long)previousTimeNanos); 2564 timestamp.mTime = mPreviousTimestamp.mTime; 2565 } 2566 2567 // Looking at signed delta will work even when the timestamps 2568 // are wrapping around. 2569 int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition) 2570 - mPreviousTimestamp.mPosition).signedValue(); 2571 if (deltaPosition < 0) { 2572 // Only report once per position instead of spamming the log. 2573 if (!mRetrogradeMotionReported) { 2574 ALOGW("retrograde timestamp position corrected, %d = %u - %u", 2575 deltaPosition, 2576 timestamp.mPosition, 2577 mPreviousTimestamp.mPosition); 2578 mRetrogradeMotionReported = true; 2579 } 2580 } else { 2581 mRetrogradeMotionReported = false; 2582 } 2583 if (deltaPosition < 0) { 2584 timestamp.mPosition = mPreviousTimestamp.mPosition; 2585 deltaPosition = 0; 2586 } 2587#if 0 2588 // Uncomment this to verify audio timestamp rate. 2589 const int64_t deltaTime = 2590 convertTimespecToNs(timestamp.mTime) - previousTimeNanos; 2591 if (deltaTime != 0) { 2592 const int64_t computedSampleRate = 2593 deltaPosition * (long long)NANOS_PER_SECOND / deltaTime; 2594 ALOGD("computedSampleRate:%u sampleRate:%u", 2595 (unsigned)computedSampleRate, mSampleRate); 2596 } 2597#endif 2598 } 2599 mPreviousTimestamp = timestamp; 2600 mPreviousTimestampValid = true; 2601 } 2602 2603 return status; 2604} 2605 2606String8 AudioTrack::getParameters(const String8& keys) 2607{ 2608 audio_io_handle_t output = getOutput(); 2609 if (output != AUDIO_IO_HANDLE_NONE) { 2610 return AudioSystem::getParameters(output, keys); 2611 } else { 2612 return String8::empty(); 2613 } 2614} 2615 2616bool AudioTrack::isOffloaded() const 2617{ 2618 AutoMutex lock(mLock); 2619 return isOffloaded_l(); 2620} 2621 2622bool AudioTrack::isDirect() const 2623{ 2624 AutoMutex lock(mLock); 2625 return isDirect_l(); 2626} 2627 2628bool AudioTrack::isOffloadedOrDirect() const 2629{ 2630 AutoMutex lock(mLock); 2631 return isOffloadedOrDirect_l(); 2632} 2633 2634 2635status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 2636{ 2637 2638 const size_t SIZE = 256; 2639 char buffer[SIZE]; 2640 String8 result; 2641 2642 result.append(" AudioTrack::dump\n"); 2643 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 2644 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 2645 result.append(buffer); 2646 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 2647 mChannelCount, mFrameCount); 2648 result.append(buffer); 2649 snprintf(buffer, 255, " sample rate(%u), speed(%f), status(%d)\n", 2650 mSampleRate, mPlaybackRate.mSpeed, mStatus); 2651 result.append(buffer); 2652 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 2653 result.append(buffer); 2654 ::write(fd, result.string(), result.size()); 2655 return NO_ERROR; 2656} 2657 2658uint32_t AudioTrack::getUnderrunCount() const 2659{ 2660 AutoMutex lock(mLock); 2661 return getUnderrunCount_l(); 2662} 2663 2664uint32_t AudioTrack::getUnderrunCount_l() const 2665{ 2666 return mProxy->getUnderrunCount() + mUnderrunCountOffset; 2667} 2668 2669uint32_t AudioTrack::getUnderrunFrames() const 2670{ 2671 AutoMutex lock(mLock); 2672 return mProxy->getUnderrunFrames(); 2673} 2674 2675status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback) 2676{ 2677 if (callback == 0) { 2678 ALOGW("%s adding NULL callback!", __FUNCTION__); 2679 return BAD_VALUE; 2680 } 2681 AutoMutex lock(mLock); 2682 if (mDeviceCallback == callback) { 2683 ALOGW("%s adding same callback!", __FUNCTION__); 2684 return INVALID_OPERATION; 2685 } 2686 status_t status = NO_ERROR; 2687 if (mOutput != AUDIO_IO_HANDLE_NONE) { 2688 if (mDeviceCallback != 0) { 2689 ALOGW("%s callback already present!", __FUNCTION__); 2690 AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); 2691 } 2692 status = AudioSystem::addAudioDeviceCallback(callback, mOutput); 2693 } 2694 mDeviceCallback = callback; 2695 return status; 2696} 2697 2698status_t AudioTrack::removeAudioDeviceCallback( 2699 const sp<AudioSystem::AudioDeviceCallback>& callback) 2700{ 2701 if (callback == 0) { 2702 ALOGW("%s removing NULL callback!", __FUNCTION__); 2703 return BAD_VALUE; 2704 } 2705 AutoMutex lock(mLock); 2706 if (mDeviceCallback != callback) { 2707 ALOGW("%s removing different callback!", __FUNCTION__); 2708 return INVALID_OPERATION; 2709 } 2710 if (mOutput != AUDIO_IO_HANDLE_NONE) { 2711 AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); 2712 } 2713 mDeviceCallback = 0; 2714 return NO_ERROR; 2715} 2716 2717status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location) 2718{ 2719 if (msec == nullptr || 2720 (location != ExtendedTimestamp::LOCATION_SERVER 2721 && location != ExtendedTimestamp::LOCATION_KERNEL)) { 2722 return BAD_VALUE; 2723 } 2724 AutoMutex lock(mLock); 2725 // inclusive of offloaded and direct tracks. 2726 // 2727 // It is possible, but not enabled, to allow duration computation for non-pcm 2728 // audio_has_proportional_frames() formats because currently they have 2729 // the drain rate equivalent to the pcm sample rate * framesize. 2730 if (!isPurePcmData_l()) { 2731 return INVALID_OPERATION; 2732 } 2733 ExtendedTimestamp ets; 2734 if (getTimestamp_l(&ets) == OK 2735 && ets.mTimeNs[location] > 0) { 2736 int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT] 2737 - ets.mPosition[location]; 2738 if (diff < 0) { 2739 *msec = 0; 2740 } else { 2741 // ms is the playback time by frames 2742 int64_t ms = (int64_t)((double)diff * 1000 / 2743 ((double)mSampleRate * mPlaybackRate.mSpeed)); 2744 // clockdiff is the timestamp age (negative) 2745 int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 : 2746 ets.mTimeNs[location] 2747 + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC] 2748 - systemTime(SYSTEM_TIME_MONOTONIC); 2749 2750 //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff); 2751 static const int NANOS_PER_MILLIS = 1000000; 2752 *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS); 2753 } 2754 return NO_ERROR; 2755 } 2756 if (location != ExtendedTimestamp::LOCATION_SERVER) { 2757 return INVALID_OPERATION; // LOCATION_KERNEL is not available 2758 } 2759 // use server position directly (offloaded and direct arrive here) 2760 updateAndGetPosition_l(); 2761 int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue(); 2762 *msec = (diff <= 0) ? 0 2763 : (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed)); 2764 return NO_ERROR; 2765} 2766 2767bool AudioTrack::hasStarted() 2768{ 2769 AutoMutex lock(mLock); 2770 switch (mState) { 2771 case STATE_STOPPED: 2772 if (isOffloadedOrDirect_l()) { 2773 // check if we have started in the past to return true. 2774 return mStartUs > 0; 2775 } 2776 // A normal audio track may still be draining, so 2777 // check if stream has ended. This covers fasttrack position 2778 // instability and start/stop without any data written. 2779 if (mProxy->getStreamEndDone()) { 2780 return true; 2781 } 2782 // fall through 2783 case STATE_ACTIVE: 2784 case STATE_STOPPING: 2785 break; 2786 case STATE_PAUSED: 2787 case STATE_PAUSED_STOPPING: 2788 case STATE_FLUSHED: 2789 return false; // we're not active 2790 default: 2791 LOG_ALWAYS_FATAL("Invalid mState in hasStarted(): %d", mState); 2792 break; 2793 } 2794 2795 // wait indicates whether we need to wait for a timestamp. 2796 // This is conservatively figured - if we encounter an unexpected error 2797 // then we will not wait. 2798 bool wait = false; 2799 if (isOffloadedOrDirect_l()) { 2800 AudioTimestamp ts; 2801 status_t status = getTimestamp_l(ts); 2802 if (status == WOULD_BLOCK) { 2803 wait = true; 2804 } else if (status == OK) { 2805 wait = (ts.mPosition == 0 || ts.mPosition == mStartTs.mPosition); 2806 } 2807 ALOGV("hasStarted wait:%d ts:%u start position:%lld", 2808 (int)wait, 2809 ts.mPosition, 2810 (long long)mStartTs.mPosition); 2811 } else { 2812 int location = ExtendedTimestamp::LOCATION_SERVER; // for ALOG 2813 ExtendedTimestamp ets; 2814 status_t status = getTimestamp_l(&ets); 2815 if (status == WOULD_BLOCK) { // no SERVER or KERNEL frame info in ets 2816 wait = true; 2817 } else if (status == OK) { 2818 for (location = ExtendedTimestamp::LOCATION_KERNEL; 2819 location >= ExtendedTimestamp::LOCATION_SERVER; --location) { 2820 if (ets.mTimeNs[location] < 0 || mStartEts.mTimeNs[location] < 0) { 2821 continue; 2822 } 2823 wait = ets.mPosition[location] == 0 2824 || ets.mPosition[location] == mStartEts.mPosition[location]; 2825 break; 2826 } 2827 } 2828 ALOGV("hasStarted wait:%d ets:%lld start position:%lld", 2829 (int)wait, 2830 (long long)ets.mPosition[location], 2831 (long long)mStartEts.mPosition[location]); 2832 } 2833 return !wait; 2834} 2835 2836// ========================================================================= 2837 2838void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 2839{ 2840 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 2841 if (audioTrack != 0) { 2842 AutoMutex lock(audioTrack->mLock); 2843 audioTrack->mProxy->binderDied(); 2844 } 2845} 2846 2847// ========================================================================= 2848 2849AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 2850 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 2851 mIgnoreNextPausedInt(false) 2852{ 2853} 2854 2855AudioTrack::AudioTrackThread::~AudioTrackThread() 2856{ 2857} 2858 2859bool AudioTrack::AudioTrackThread::threadLoop() 2860{ 2861 { 2862 AutoMutex _l(mMyLock); 2863 if (mPaused) { 2864 mMyCond.wait(mMyLock); 2865 // caller will check for exitPending() 2866 return true; 2867 } 2868 if (mIgnoreNextPausedInt) { 2869 mIgnoreNextPausedInt = false; 2870 mPausedInt = false; 2871 } 2872 if (mPausedInt) { 2873 if (mPausedNs > 0) { 2874 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 2875 } else { 2876 mMyCond.wait(mMyLock); 2877 } 2878 mPausedInt = false; 2879 return true; 2880 } 2881 } 2882 if (exitPending()) { 2883 return false; 2884 } 2885 nsecs_t ns = mReceiver.processAudioBuffer(); 2886 switch (ns) { 2887 case 0: 2888 return true; 2889 case NS_INACTIVE: 2890 pauseInternal(); 2891 return true; 2892 case NS_NEVER: 2893 return false; 2894 case NS_WHENEVER: 2895 // Event driven: call wake() when callback notifications conditions change. 2896 ns = INT64_MAX; 2897 // fall through 2898 default: 2899 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns); 2900 pauseInternal(ns); 2901 return true; 2902 } 2903} 2904 2905void AudioTrack::AudioTrackThread::requestExit() 2906{ 2907 // must be in this order to avoid a race condition 2908 Thread::requestExit(); 2909 resume(); 2910} 2911 2912void AudioTrack::AudioTrackThread::pause() 2913{ 2914 AutoMutex _l(mMyLock); 2915 mPaused = true; 2916} 2917 2918void AudioTrack::AudioTrackThread::resume() 2919{ 2920 AutoMutex _l(mMyLock); 2921 mIgnoreNextPausedInt = true; 2922 if (mPaused || mPausedInt) { 2923 mPaused = false; 2924 mPausedInt = false; 2925 mMyCond.signal(); 2926 } 2927} 2928 2929void AudioTrack::AudioTrackThread::wake() 2930{ 2931 AutoMutex _l(mMyLock); 2932 if (!mPaused) { 2933 // wake() might be called while servicing a callback - ignore the next 2934 // pause time and call processAudioBuffer. 2935 mIgnoreNextPausedInt = true; 2936 if (mPausedInt && mPausedNs > 0) { 2937 // audio track is active and internally paused with timeout. 2938 mPausedInt = false; 2939 mMyCond.signal(); 2940 } 2941 } 2942} 2943 2944void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 2945{ 2946 AutoMutex _l(mMyLock); 2947 mPausedInt = true; 2948 mPausedNs = ns; 2949} 2950 2951} // namespace android 2952