AudioTrack.cpp revision 6d8018f0b7be9deec6b0acab10a0dca6e91d0fb8
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 mPortId(AUDIO_PORT_HANDLE_NONE) 188{ 189 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN; 190 mAttributes.usage = AUDIO_USAGE_UNKNOWN; 191 mAttributes.flags = 0x0; 192 strcpy(mAttributes.tags, ""); 193} 194 195AudioTrack::AudioTrack( 196 audio_stream_type_t streamType, 197 uint32_t sampleRate, 198 audio_format_t format, 199 audio_channel_mask_t channelMask, 200 size_t frameCount, 201 audio_output_flags_t flags, 202 callback_t cbf, 203 void* user, 204 int32_t notificationFrames, 205 audio_session_t sessionId, 206 transfer_type transferType, 207 const audio_offload_info_t *offloadInfo, 208 uid_t uid, 209 pid_t pid, 210 const audio_attributes_t* pAttributes, 211 bool doNotReconnect, 212 float maxRequiredSpeed) 213 : mStatus(NO_INIT), 214 mState(STATE_STOPPED), 215 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 216 mPreviousSchedulingGroup(SP_DEFAULT), 217 mPausedPosition(0), 218 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE), 219 mPortId(AUDIO_PORT_HANDLE_NONE) 220{ 221 mStatus = set(streamType, sampleRate, format, channelMask, 222 frameCount, flags, cbf, user, notificationFrames, 223 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 224 offloadInfo, uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed); 225} 226 227AudioTrack::AudioTrack( 228 audio_stream_type_t streamType, 229 uint32_t sampleRate, 230 audio_format_t format, 231 audio_channel_mask_t channelMask, 232 const sp<IMemory>& sharedBuffer, 233 audio_output_flags_t flags, 234 callback_t cbf, 235 void* user, 236 int32_t notificationFrames, 237 audio_session_t sessionId, 238 transfer_type transferType, 239 const audio_offload_info_t *offloadInfo, 240 uid_t uid, 241 pid_t pid, 242 const audio_attributes_t* pAttributes, 243 bool doNotReconnect, 244 float maxRequiredSpeed) 245 : mStatus(NO_INIT), 246 mState(STATE_STOPPED), 247 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 248 mPreviousSchedulingGroup(SP_DEFAULT), 249 mPausedPosition(0), 250 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE), 251 mPortId(AUDIO_PORT_HANDLE_NONE) 252{ 253 mStatus = set(streamType, sampleRate, format, channelMask, 254 0 /*frameCount*/, flags, cbf, user, notificationFrames, 255 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, 256 uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed); 257} 258 259AudioTrack::~AudioTrack() 260{ 261 if (mStatus == NO_ERROR) { 262 // Make sure that callback function exits in the case where 263 // it is looping on buffer full condition in obtainBuffer(). 264 // Otherwise the callback thread will never exit. 265 stop(); 266 if (mAudioTrackThread != 0) { 267 mProxy->interrupt(); 268 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 269 mAudioTrackThread->requestExitAndWait(); 270 mAudioTrackThread.clear(); 271 } 272 // No lock here: worst case we remove a NULL callback which will be a nop 273 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) { 274 AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); 275 } 276 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this); 277 mAudioTrack.clear(); 278 mCblkMemory.clear(); 279 mSharedBuffer.clear(); 280 IPCThreadState::self()->flushCommands(); 281 ALOGV("~AudioTrack, releasing session id %d from %d on behalf of %d", 282 mSessionId, IPCThreadState::self()->getCallingPid(), mClientPid); 283 AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); 284 } 285} 286 287status_t AudioTrack::set( 288 audio_stream_type_t streamType, 289 uint32_t sampleRate, 290 audio_format_t format, 291 audio_channel_mask_t channelMask, 292 size_t frameCount, 293 audio_output_flags_t flags, 294 callback_t cbf, 295 void* user, 296 int32_t notificationFrames, 297 const sp<IMemory>& sharedBuffer, 298 bool threadCanCallJava, 299 audio_session_t sessionId, 300 transfer_type transferType, 301 const audio_offload_info_t *offloadInfo, 302 uid_t uid, 303 pid_t pid, 304 const audio_attributes_t* pAttributes, 305 bool doNotReconnect, 306 float maxRequiredSpeed) 307{ 308 ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 309 "flags #%x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d", 310 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, 311 sessionId, transferType, uid, pid); 312 313 mThreadCanCallJava = threadCanCallJava; 314 315 switch (transferType) { 316 case TRANSFER_DEFAULT: 317 if (sharedBuffer != 0) { 318 transferType = TRANSFER_SHARED; 319 } else if (cbf == NULL || threadCanCallJava) { 320 transferType = TRANSFER_SYNC; 321 } else { 322 transferType = TRANSFER_CALLBACK; 323 } 324 break; 325 case TRANSFER_CALLBACK: 326 if (cbf == NULL || sharedBuffer != 0) { 327 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 328 return BAD_VALUE; 329 } 330 break; 331 case TRANSFER_OBTAIN: 332 case TRANSFER_SYNC: 333 if (sharedBuffer != 0) { 334 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 335 return BAD_VALUE; 336 } 337 break; 338 case TRANSFER_SHARED: 339 if (sharedBuffer == 0) { 340 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 341 return BAD_VALUE; 342 } 343 break; 344 default: 345 ALOGE("Invalid transfer type %d", transferType); 346 return BAD_VALUE; 347 } 348 mSharedBuffer = sharedBuffer; 349 mTransfer = transferType; 350 mDoNotReconnect = doNotReconnect; 351 352 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %zu", sharedBuffer->pointer(), 353 sharedBuffer->size()); 354 355 ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags); 356 357 // invariant that mAudioTrack != 0 is true only after set() returns successfully 358 if (mAudioTrack != 0) { 359 ALOGE("Track already in use"); 360 return INVALID_OPERATION; 361 } 362 363 // handle default values first. 364 if (streamType == AUDIO_STREAM_DEFAULT) { 365 streamType = AUDIO_STREAM_MUSIC; 366 } 367 if (pAttributes == NULL) { 368 if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) { 369 ALOGE("Invalid stream type %d", streamType); 370 return BAD_VALUE; 371 } 372 mStreamType = streamType; 373 374 } else { 375 // stream type shouldn't be looked at, this track has audio attributes 376 memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t)); 377 ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]", 378 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags); 379 mStreamType = AUDIO_STREAM_DEFAULT; 380 if ((mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) { 381 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC); 382 } 383 if ((mAttributes.flags & AUDIO_FLAG_LOW_LATENCY) != 0) { 384 flags = (audio_output_flags_t) (flags | AUDIO_OUTPUT_FLAG_FAST); 385 } 386 // check deep buffer after flags have been modified above 387 if (flags == AUDIO_OUTPUT_FLAG_NONE && (mAttributes.flags & AUDIO_FLAG_DEEP_BUFFER) != 0) { 388 flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER; 389 } 390 } 391 392 // these below should probably come from the audioFlinger too... 393 if (format == AUDIO_FORMAT_DEFAULT) { 394 format = AUDIO_FORMAT_PCM_16_BIT; 395 } else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through? 396 mAttributes.flags |= AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO; 397 } 398 399 // validate parameters 400 if (!audio_is_valid_format(format)) { 401 ALOGE("Invalid format %#x", format); 402 return BAD_VALUE; 403 } 404 mFormat = format; 405 406 if (!audio_is_output_channel(channelMask)) { 407 ALOGE("Invalid channel mask %#x", channelMask); 408 return BAD_VALUE; 409 } 410 mChannelMask = channelMask; 411 uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 412 mChannelCount = channelCount; 413 414 // force direct flag if format is not linear PCM 415 // or offload was requested 416 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 417 || !audio_is_linear_pcm(format)) { 418 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 419 ? "Offload request, forcing to Direct Output" 420 : "Not linear PCM, forcing to Direct Output"); 421 flags = (audio_output_flags_t) 422 // FIXME why can't we allow direct AND fast? 423 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 424 } 425 426 // force direct flag if HW A/V sync requested 427 if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) { 428 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT); 429 } 430 431 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { 432 if (audio_has_proportional_frames(format)) { 433 mFrameSize = channelCount * audio_bytes_per_sample(format); 434 } else { 435 mFrameSize = sizeof(uint8_t); 436 } 437 } else { 438 ALOG_ASSERT(audio_has_proportional_frames(format)); 439 mFrameSize = channelCount * audio_bytes_per_sample(format); 440 // createTrack will return an error if PCM format is not supported by server, 441 // so no need to check for specific PCM formats here 442 } 443 444 // sampling rate must be specified for direct outputs 445 if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) { 446 return BAD_VALUE; 447 } 448 mSampleRate = sampleRate; 449 mOriginalSampleRate = sampleRate; 450 mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT; 451 // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX 452 mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX); 453 454 // Make copy of input parameter offloadInfo so that in the future: 455 // (a) createTrack_l doesn't need it as an input parameter 456 // (b) we can support re-creation of offloaded tracks 457 if (offloadInfo != NULL) { 458 mOffloadInfoCopy = *offloadInfo; 459 mOffloadInfo = &mOffloadInfoCopy; 460 } else { 461 mOffloadInfo = NULL; 462 memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t)); 463 } 464 465 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f; 466 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f; 467 mSendLevel = 0.0f; 468 // mFrameCount is initialized in createTrack_l 469 mReqFrameCount = frameCount; 470 if (notificationFrames >= 0) { 471 mNotificationFramesReq = notificationFrames; 472 mNotificationsPerBufferReq = 0; 473 } else { 474 if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 475 ALOGE("notificationFrames=%d not permitted for non-fast track", 476 notificationFrames); 477 return BAD_VALUE; 478 } 479 if (frameCount > 0) { 480 ALOGE("notificationFrames=%d not permitted with non-zero frameCount=%zu", 481 notificationFrames, frameCount); 482 return BAD_VALUE; 483 } 484 mNotificationFramesReq = 0; 485 const uint32_t minNotificationsPerBuffer = 1; 486 const uint32_t maxNotificationsPerBuffer = 8; 487 mNotificationsPerBufferReq = min(maxNotificationsPerBuffer, 488 max((uint32_t) -notificationFrames, minNotificationsPerBuffer)); 489 ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames, 490 "notificationFrames=%d clamped to the range -%u to -%u", 491 notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer); 492 } 493 mNotificationFramesAct = 0; 494 if (sessionId == AUDIO_SESSION_ALLOCATE) { 495 mSessionId = (audio_session_t) AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 496 } else { 497 mSessionId = sessionId; 498 } 499 int callingpid = IPCThreadState::self()->getCallingPid(); 500 int mypid = getpid(); 501 if (uid == AUDIO_UID_INVALID || (callingpid != mypid)) { 502 mClientUid = IPCThreadState::self()->getCallingUid(); 503 } else { 504 mClientUid = uid; 505 } 506 if (pid == -1 || (callingpid != mypid)) { 507 mClientPid = callingpid; 508 } else { 509 mClientPid = pid; 510 } 511 mAuxEffectId = 0; 512 mOrigFlags = mFlags = flags; 513 mCbf = cbf; 514 515 if (cbf != NULL) { 516 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 517 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 518 // thread begins in paused state, and will not reference us until start() 519 } 520 521 // create the IAudioTrack 522 status_t status = createTrack_l(); 523 524 if (status != NO_ERROR) { 525 if (mAudioTrackThread != 0) { 526 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 527 mAudioTrackThread->requestExitAndWait(); 528 mAudioTrackThread.clear(); 529 } 530 return status; 531 } 532 533 mStatus = NO_ERROR; 534 mUserData = user; 535 mLoopCount = 0; 536 mLoopStart = 0; 537 mLoopEnd = 0; 538 mLoopCountNotified = 0; 539 mMarkerPosition = 0; 540 mMarkerReached = false; 541 mNewPosition = 0; 542 mUpdatePeriod = 0; 543 mPosition = 0; 544 mReleased = 0; 545 mStartUs = 0; 546 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 547 mSequence = 1; 548 mObservedSequence = mSequence; 549 mInUnderrun = false; 550 mPreviousTimestampValid = false; 551 mTimestampStartupGlitchReported = false; 552 mRetrogradeMotionReported = false; 553 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID; 554 mStartTs.mPosition = 0; 555 mUnderrunCountOffset = 0; 556 mFramesWritten = 0; 557 mFramesWrittenServerOffset = 0; 558 mFramesWrittenAtRestore = -1; // -1 is a unique initializer. 559 mVolumeHandler = new VolumeHandler(); 560 return NO_ERROR; 561} 562 563// ------------------------------------------------------------------------- 564 565status_t AudioTrack::start() 566{ 567 AutoMutex lock(mLock); 568 569 if (mState == STATE_ACTIVE) { 570 return INVALID_OPERATION; 571 } 572 573 mInUnderrun = true; 574 575 State previousState = mState; 576 if (previousState == STATE_PAUSED_STOPPING) { 577 mState = STATE_STOPPING; 578 } else { 579 mState = STATE_ACTIVE; 580 } 581 (void) updateAndGetPosition_l(); 582 583 // save start timestamp 584 if (isOffloadedOrDirect_l()) { 585 if (getTimestamp_l(mStartTs) != OK) { 586 mStartTs.mPosition = 0; 587 } 588 } else { 589 if (getTimestamp_l(&mStartEts) != OK) { 590 mStartEts.clear(); 591 } 592 } 593 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 594 // reset current position as seen by client to 0 595 mPosition = 0; 596 mPreviousTimestampValid = false; 597 mTimestampStartupGlitchReported = false; 598 mRetrogradeMotionReported = false; 599 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID; 600 601 if (!isOffloadedOrDirect_l() 602 && mStartEts.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) { 603 // Server side has consumed something, but is it finished consuming? 604 // It is possible since flush and stop are asynchronous that the server 605 // is still active at this point. 606 ALOGV("start: server read:%lld cumulative flushed:%lld client written:%lld", 607 (long long)(mFramesWrittenServerOffset 608 + mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER]), 609 (long long)mStartEts.mFlushed, 610 (long long)mFramesWritten); 611 mFramesWrittenServerOffset = -mStartEts.mPosition[ExtendedTimestamp::LOCATION_SERVER]; 612 } 613 mFramesWritten = 0; 614 mProxy->clearTimestamp(); // need new server push for valid timestamp 615 mMarkerReached = false; 616 617 // For offloaded tracks, we don't know if the hardware counters are really zero here, 618 // since the flush is asynchronous and stop may not fully drain. 619 // We save the time when the track is started to later verify whether 620 // the counters are realistic (i.e. start from zero after this time). 621 mStartUs = getNowUs(); 622 623 // force refresh of remaining frames by processAudioBuffer() as last 624 // write before stop could be partial. 625 mRefreshRemaining = true; 626 } 627 mNewPosition = mPosition + mUpdatePeriod; 628 int32_t flags = android_atomic_and(~(CBLK_STREAM_END_DONE | CBLK_DISABLED), &mCblk->mFlags); 629 630 status_t status = NO_ERROR; 631 if (!(flags & CBLK_INVALID)) { 632 status = mAudioTrack->start(); 633 if (status == DEAD_OBJECT) { 634 flags |= CBLK_INVALID; 635 } 636 } 637 if (flags & CBLK_INVALID) { 638 status = restoreTrack_l("start"); 639 } 640 641 // resume or pause the callback thread as needed. 642 sp<AudioTrackThread> t = mAudioTrackThread; 643 if (status == NO_ERROR) { 644 if (t != 0) { 645 if (previousState == STATE_STOPPING) { 646 mProxy->interrupt(); 647 } else { 648 t->resume(); 649 } 650 } else { 651 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 652 get_sched_policy(0, &mPreviousSchedulingGroup); 653 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 654 } 655 } else { 656 ALOGE("start() status %d", status); 657 mState = previousState; 658 if (t != 0) { 659 if (previousState != STATE_STOPPING) { 660 t->pause(); 661 } 662 } else { 663 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 664 set_sched_policy(0, mPreviousSchedulingGroup); 665 } 666 } 667 668 return status; 669} 670 671void AudioTrack::stop() 672{ 673 AutoMutex lock(mLock); 674 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 675 return; 676 } 677 678 if (isOffloaded_l()) { 679 mState = STATE_STOPPING; 680 } else { 681 mState = STATE_STOPPED; 682 ALOGD_IF(mSharedBuffer == nullptr, 683 "stop() called with %u frames delivered", mReleased.value()); 684 mReleased = 0; 685 } 686 687 mProxy->interrupt(); 688 mAudioTrack->stop(); 689 690 // Note: legacy handling - stop does not clear playback marker 691 // and periodic update counter, but flush does for streaming tracks. 692 693 if (mSharedBuffer != 0) { 694 // clear buffer position and loop count. 695 mStaticProxy->setBufferPositionAndLoop(0 /* position */, 696 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */); 697 } 698 699 sp<AudioTrackThread> t = mAudioTrackThread; 700 if (t != 0) { 701 if (!isOffloaded_l()) { 702 t->pause(); 703 } 704 } else { 705 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 706 set_sched_policy(0, mPreviousSchedulingGroup); 707 } 708} 709 710bool AudioTrack::stopped() const 711{ 712 AutoMutex lock(mLock); 713 return mState != STATE_ACTIVE; 714} 715 716void AudioTrack::flush() 717{ 718 if (mSharedBuffer != 0) { 719 return; 720 } 721 AutoMutex lock(mLock); 722 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 723 return; 724 } 725 flush_l(); 726} 727 728void AudioTrack::flush_l() 729{ 730 ALOG_ASSERT(mState != STATE_ACTIVE); 731 732 // clear playback marker and periodic update counter 733 mMarkerPosition = 0; 734 mMarkerReached = false; 735 mUpdatePeriod = 0; 736 mRefreshRemaining = true; 737 738 mState = STATE_FLUSHED; 739 mReleased = 0; 740 if (isOffloaded_l()) { 741 mProxy->interrupt(); 742 } 743 mProxy->flush(); 744 mAudioTrack->flush(); 745} 746 747void AudioTrack::pause() 748{ 749 AutoMutex lock(mLock); 750 if (mState == STATE_ACTIVE) { 751 mState = STATE_PAUSED; 752 } else if (mState == STATE_STOPPING) { 753 mState = STATE_PAUSED_STOPPING; 754 } else { 755 return; 756 } 757 mProxy->interrupt(); 758 mAudioTrack->pause(); 759 760 if (isOffloaded_l()) { 761 if (mOutput != AUDIO_IO_HANDLE_NONE) { 762 // An offload output can be re-used between two audio tracks having 763 // the same configuration. A timestamp query for a paused track 764 // while the other is running would return an incorrect time. 765 // To fix this, cache the playback position on a pause() and return 766 // this time when requested until the track is resumed. 767 768 // OffloadThread sends HAL pause in its threadLoop. Time saved 769 // here can be slightly off. 770 771 // TODO: check return code for getRenderPosition. 772 773 uint32_t halFrames; 774 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); 775 ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); 776 } 777 } 778} 779 780status_t AudioTrack::setVolume(float left, float right) 781{ 782 // This duplicates a test by AudioTrack JNI, but that is not the only caller 783 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY || 784 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) { 785 return BAD_VALUE; 786 } 787 788 AutoMutex lock(mLock); 789 mVolume[AUDIO_INTERLEAVE_LEFT] = left; 790 mVolume[AUDIO_INTERLEAVE_RIGHT] = right; 791 792 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right))); 793 794 if (isOffloaded_l()) { 795 mAudioTrack->signal(); 796 } 797 return NO_ERROR; 798} 799 800status_t AudioTrack::setVolume(float volume) 801{ 802 return setVolume(volume, volume); 803} 804 805status_t AudioTrack::setAuxEffectSendLevel(float level) 806{ 807 // This duplicates a test by AudioTrack JNI, but that is not the only caller 808 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) { 809 return BAD_VALUE; 810 } 811 812 AutoMutex lock(mLock); 813 mSendLevel = level; 814 mProxy->setSendLevel(level); 815 816 return NO_ERROR; 817} 818 819void AudioTrack::getAuxEffectSendLevel(float* level) const 820{ 821 if (level != NULL) { 822 *level = mSendLevel; 823 } 824} 825 826status_t AudioTrack::setSampleRate(uint32_t rate) 827{ 828 AutoMutex lock(mLock); 829 if (rate == mSampleRate) { 830 return NO_ERROR; 831 } 832 if (isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 833 return INVALID_OPERATION; 834 } 835 if (mOutput == AUDIO_IO_HANDLE_NONE) { 836 return NO_INIT; 837 } 838 // NOTE: it is theoretically possible, but highly unlikely, that a device change 839 // could mean a previously allowed sampling rate is no longer allowed. 840 uint32_t afSamplingRate; 841 if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) { 842 return NO_INIT; 843 } 844 // pitch is emulated by adjusting speed and sampleRate 845 const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch); 846 if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 847 return BAD_VALUE; 848 } 849 // TODO: Should we also check if the buffer size is compatible? 850 851 mSampleRate = rate; 852 mProxy->setSampleRate(effectiveSampleRate); 853 854 return NO_ERROR; 855} 856 857uint32_t AudioTrack::getSampleRate() const 858{ 859 AutoMutex lock(mLock); 860 861 // sample rate can be updated during playback by the offloaded decoder so we need to 862 // query the HAL and update if needed. 863// FIXME use Proxy return channel to update the rate from server and avoid polling here 864 if (isOffloadedOrDirect_l()) { 865 if (mOutput != AUDIO_IO_HANDLE_NONE) { 866 uint32_t sampleRate = 0; 867 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate); 868 if (status == NO_ERROR) { 869 mSampleRate = sampleRate; 870 } 871 } 872 } 873 return mSampleRate; 874} 875 876uint32_t AudioTrack::getOriginalSampleRate() const 877{ 878 return mOriginalSampleRate; 879} 880 881status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate) 882{ 883 AutoMutex lock(mLock); 884 if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) { 885 return NO_ERROR; 886 } 887 if (isOffloadedOrDirect_l()) { 888 return INVALID_OPERATION; 889 } 890 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 891 return INVALID_OPERATION; 892 } 893 894 ALOGV("setPlaybackRate (input): mSampleRate:%u mSpeed:%f mPitch:%f", 895 mSampleRate, playbackRate.mSpeed, playbackRate.mPitch); 896 // pitch is emulated by adjusting speed and sampleRate 897 const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch); 898 const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch); 899 const float effectivePitch = adjustPitch(playbackRate.mPitch); 900 AudioPlaybackRate playbackRateTemp = playbackRate; 901 playbackRateTemp.mSpeed = effectiveSpeed; 902 playbackRateTemp.mPitch = effectivePitch; 903 904 ALOGV("setPlaybackRate (effective): mSampleRate:%u mSpeed:%f mPitch:%f", 905 effectiveRate, effectiveSpeed, effectivePitch); 906 907 if (!isAudioPlaybackRateValid(playbackRateTemp)) { 908 ALOGV("setPlaybackRate(%f, %f) failed (effective rate out of bounds)", 909 playbackRate.mSpeed, playbackRate.mPitch); 910 return BAD_VALUE; 911 } 912 // Check if the buffer size is compatible. 913 if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) { 914 ALOGV("setPlaybackRate(%f, %f) failed (buffer size)", 915 playbackRate.mSpeed, playbackRate.mPitch); 916 return BAD_VALUE; 917 } 918 919 // Check resampler ratios are within bounds 920 if ((uint64_t)effectiveRate > (uint64_t)mSampleRate * 921 (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 922 ALOGV("setPlaybackRate(%f, %f) failed. Resample rate exceeds max accepted value", 923 playbackRate.mSpeed, playbackRate.mPitch); 924 return BAD_VALUE; 925 } 926 927 if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) { 928 ALOGV("setPlaybackRate(%f, %f) failed. Resample rate below min accepted value", 929 playbackRate.mSpeed, playbackRate.mPitch); 930 return BAD_VALUE; 931 } 932 mPlaybackRate = playbackRate; 933 //set effective rates 934 mProxy->setPlaybackRate(playbackRateTemp); 935 mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate 936 return NO_ERROR; 937} 938 939const AudioPlaybackRate& AudioTrack::getPlaybackRate() const 940{ 941 AutoMutex lock(mLock); 942 return mPlaybackRate; 943} 944 945ssize_t AudioTrack::getBufferSizeInFrames() 946{ 947 AutoMutex lock(mLock); 948 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) { 949 return NO_INIT; 950 } 951 return (ssize_t) mProxy->getBufferSizeInFrames(); 952} 953 954status_t AudioTrack::getBufferDurationInUs(int64_t *duration) 955{ 956 if (duration == nullptr) { 957 return BAD_VALUE; 958 } 959 AutoMutex lock(mLock); 960 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) { 961 return NO_INIT; 962 } 963 ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames(); 964 if (bufferSizeInFrames < 0) { 965 return (status_t)bufferSizeInFrames; 966 } 967 *duration = (int64_t)((double)bufferSizeInFrames * 1000000 968 / ((double)mSampleRate * mPlaybackRate.mSpeed)); 969 return NO_ERROR; 970} 971 972ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames) 973{ 974 AutoMutex lock(mLock); 975 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) { 976 return NO_INIT; 977 } 978 // Reject if timed track or compressed audio. 979 if (!audio_is_linear_pcm(mFormat)) { 980 return INVALID_OPERATION; 981 } 982 return (ssize_t) mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames); 983} 984 985status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 986{ 987 if (mSharedBuffer == 0 || isOffloadedOrDirect()) { 988 return INVALID_OPERATION; 989 } 990 991 if (loopCount == 0) { 992 ; 993 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 994 loopEnd - loopStart >= MIN_LOOP) { 995 ; 996 } else { 997 return BAD_VALUE; 998 } 999 1000 AutoMutex lock(mLock); 1001 // See setPosition() regarding setting parameters such as loop points or position while active 1002 if (mState == STATE_ACTIVE) { 1003 return INVALID_OPERATION; 1004 } 1005 setLoop_l(loopStart, loopEnd, loopCount); 1006 return NO_ERROR; 1007} 1008 1009void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 1010{ 1011 // We do not update the periodic notification point. 1012 // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 1013 mLoopCount = loopCount; 1014 mLoopEnd = loopEnd; 1015 mLoopStart = loopStart; 1016 mLoopCountNotified = loopCount; 1017 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 1018 1019 // Waking the AudioTrackThread is not needed as this cannot be called when active. 1020} 1021 1022status_t AudioTrack::setMarkerPosition(uint32_t marker) 1023{ 1024 // The only purpose of setting marker position is to get a callback 1025 if (mCbf == NULL || isOffloadedOrDirect()) { 1026 return INVALID_OPERATION; 1027 } 1028 1029 AutoMutex lock(mLock); 1030 mMarkerPosition = marker; 1031 mMarkerReached = false; 1032 1033 sp<AudioTrackThread> t = mAudioTrackThread; 1034 if (t != 0) { 1035 t->wake(); 1036 } 1037 return NO_ERROR; 1038} 1039 1040status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 1041{ 1042 if (isOffloadedOrDirect()) { 1043 return INVALID_OPERATION; 1044 } 1045 if (marker == NULL) { 1046 return BAD_VALUE; 1047 } 1048 1049 AutoMutex lock(mLock); 1050 mMarkerPosition.getValue(marker); 1051 1052 return NO_ERROR; 1053} 1054 1055status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 1056{ 1057 // The only purpose of setting position update period is to get a callback 1058 if (mCbf == NULL || isOffloadedOrDirect()) { 1059 return INVALID_OPERATION; 1060 } 1061 1062 AutoMutex lock(mLock); 1063 mNewPosition = updateAndGetPosition_l() + updatePeriod; 1064 mUpdatePeriod = updatePeriod; 1065 1066 sp<AudioTrackThread> t = mAudioTrackThread; 1067 if (t != 0) { 1068 t->wake(); 1069 } 1070 return NO_ERROR; 1071} 1072 1073status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 1074{ 1075 if (isOffloadedOrDirect()) { 1076 return INVALID_OPERATION; 1077 } 1078 if (updatePeriod == NULL) { 1079 return BAD_VALUE; 1080 } 1081 1082 AutoMutex lock(mLock); 1083 *updatePeriod = mUpdatePeriod; 1084 1085 return NO_ERROR; 1086} 1087 1088status_t AudioTrack::setPosition(uint32_t position) 1089{ 1090 if (mSharedBuffer == 0 || isOffloadedOrDirect()) { 1091 return INVALID_OPERATION; 1092 } 1093 if (position > mFrameCount) { 1094 return BAD_VALUE; 1095 } 1096 1097 AutoMutex lock(mLock); 1098 // Currently we require that the player is inactive before setting parameters such as position 1099 // or loop points. Otherwise, there could be a race condition: the application could read the 1100 // current position, compute a new position or loop parameters, and then set that position or 1101 // loop parameters but it would do the "wrong" thing since the position has continued to advance 1102 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 1103 // to specify how it wants to handle such scenarios. 1104 if (mState == STATE_ACTIVE) { 1105 return INVALID_OPERATION; 1106 } 1107 // After setting the position, use full update period before notification. 1108 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 1109 mStaticProxy->setBufferPosition(position); 1110 1111 // Waking the AudioTrackThread is not needed as this cannot be called when active. 1112 return NO_ERROR; 1113} 1114 1115status_t AudioTrack::getPosition(uint32_t *position) 1116{ 1117 if (position == NULL) { 1118 return BAD_VALUE; 1119 } 1120 1121 AutoMutex lock(mLock); 1122 // FIXME: offloaded and direct tracks call into the HAL for render positions 1123 // for compressed/synced data; however, we use proxy position for pure linear pcm data 1124 // as we do not know the capability of the HAL for pcm position support and standby. 1125 // There may be some latency differences between the HAL position and the proxy position. 1126 if (isOffloadedOrDirect_l() && !isPurePcmData_l()) { 1127 uint32_t dspFrames = 0; 1128 1129 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) { 1130 ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); 1131 *position = mPausedPosition; 1132 return NO_ERROR; 1133 } 1134 1135 if (mOutput != AUDIO_IO_HANDLE_NONE) { 1136 uint32_t halFrames; // actually unused 1137 (void) AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 1138 // FIXME: on getRenderPosition() error, we return OK with frame position 0. 1139 } 1140 // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED) 1141 // due to hardware latency. We leave this behavior for now. 1142 *position = dspFrames; 1143 } else { 1144 if (mCblk->mFlags & CBLK_INVALID) { 1145 (void) restoreTrack_l("getPosition"); 1146 // FIXME: for compatibility with the Java API we ignore the restoreTrack_l() 1147 // error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position. 1148 } 1149 1150 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 1151 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 1152 0 : updateAndGetPosition_l().value(); 1153 } 1154 return NO_ERROR; 1155} 1156 1157status_t AudioTrack::getBufferPosition(uint32_t *position) 1158{ 1159 if (mSharedBuffer == 0) { 1160 return INVALID_OPERATION; 1161 } 1162 if (position == NULL) { 1163 return BAD_VALUE; 1164 } 1165 1166 AutoMutex lock(mLock); 1167 *position = mStaticProxy->getBufferPosition(); 1168 return NO_ERROR; 1169} 1170 1171status_t AudioTrack::reload() 1172{ 1173 if (mSharedBuffer == 0 || isOffloadedOrDirect()) { 1174 return INVALID_OPERATION; 1175 } 1176 1177 AutoMutex lock(mLock); 1178 // See setPosition() regarding setting parameters such as loop points or position while active 1179 if (mState == STATE_ACTIVE) { 1180 return INVALID_OPERATION; 1181 } 1182 mNewPosition = mUpdatePeriod; 1183 (void) updateAndGetPosition_l(); 1184 mPosition = 0; 1185 mPreviousTimestampValid = false; 1186#if 0 1187 // The documentation is not clear on the behavior of reload() and the restoration 1188 // of loop count. Historically we have not restored loop count, start, end, 1189 // but it makes sense if one desires to repeat playing a particular sound. 1190 if (mLoopCount != 0) { 1191 mLoopCountNotified = mLoopCount; 1192 mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount); 1193 } 1194#endif 1195 mStaticProxy->setBufferPosition(0); 1196 return NO_ERROR; 1197} 1198 1199audio_io_handle_t AudioTrack::getOutput() const 1200{ 1201 AutoMutex lock(mLock); 1202 return mOutput; 1203} 1204 1205status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) { 1206 AutoMutex lock(mLock); 1207 if (mSelectedDeviceId != deviceId) { 1208 mSelectedDeviceId = deviceId; 1209 android_atomic_or(CBLK_INVALID, &mCblk->mFlags); 1210 } 1211 return NO_ERROR; 1212} 1213 1214audio_port_handle_t AudioTrack::getOutputDevice() { 1215 AutoMutex lock(mLock); 1216 return mSelectedDeviceId; 1217} 1218 1219audio_port_handle_t AudioTrack::getRoutedDeviceId() { 1220 AutoMutex lock(mLock); 1221 if (mOutput == AUDIO_IO_HANDLE_NONE) { 1222 return AUDIO_PORT_HANDLE_NONE; 1223 } 1224 return AudioSystem::getDeviceIdForIo(mOutput); 1225} 1226 1227status_t AudioTrack::attachAuxEffect(int effectId) 1228{ 1229 AutoMutex lock(mLock); 1230 status_t status = mAudioTrack->attachAuxEffect(effectId); 1231 if (status == NO_ERROR) { 1232 mAuxEffectId = effectId; 1233 } 1234 return status; 1235} 1236 1237audio_stream_type_t AudioTrack::streamType() const 1238{ 1239 if (mStreamType == AUDIO_STREAM_DEFAULT) { 1240 return audio_attributes_to_stream_type(&mAttributes); 1241 } 1242 return mStreamType; 1243} 1244 1245// ------------------------------------------------------------------------- 1246 1247// must be called with mLock held 1248status_t AudioTrack::createTrack_l() 1249{ 1250 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 1251 if (audioFlinger == 0) { 1252 ALOGE("Could not get audioflinger"); 1253 return NO_INIT; 1254 } 1255 1256 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) { 1257 AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); 1258 } 1259 audio_io_handle_t output; 1260 audio_stream_type_t streamType = mStreamType; 1261 audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL; 1262 1263 // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted. 1264 // After fast request is denied, we will request again if IAudioTrack is re-created. 1265 1266 status_t status; 1267 audio_config_t config = AUDIO_CONFIG_INITIALIZER; 1268 config.sample_rate = mSampleRate; 1269 config.channel_mask = mChannelMask; 1270 config.format = mFormat; 1271 config.offload_info = mOffloadInfoCopy; 1272 status = AudioSystem::getOutputForAttr(attr, &output, 1273 mSessionId, &streamType, mClientUid, 1274 &config, 1275 mFlags, mSelectedDeviceId, &mPortId); 1276 1277 if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) { 1278 ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u," 1279 " format %#x, channel mask %#x, flags %#x", 1280 mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, 1281 mFlags); 1282 return BAD_VALUE; 1283 } 1284 { 1285 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 1286 // we must release it ourselves if anything goes wrong. 1287 1288 // Not all of these values are needed under all conditions, but it is easier to get them all 1289 status = AudioSystem::getLatency(output, &mAfLatency); 1290 if (status != NO_ERROR) { 1291 ALOGE("getLatency(%d) failed status %d", output, status); 1292 goto release; 1293 } 1294 ALOGV("createTrack_l() output %d afLatency %u", output, mAfLatency); 1295 1296 status = AudioSystem::getFrameCount(output, &mAfFrameCount); 1297 if (status != NO_ERROR) { 1298 ALOGE("getFrameCount(output=%d) status %d", output, status); 1299 goto release; 1300 } 1301 1302 // TODO consider making this a member variable if there are other uses for it later 1303 size_t afFrameCountHAL; 1304 status = AudioSystem::getFrameCountHAL(output, &afFrameCountHAL); 1305 if (status != NO_ERROR) { 1306 ALOGE("getFrameCountHAL(output=%d) status %d", output, status); 1307 goto release; 1308 } 1309 ALOG_ASSERT(afFrameCountHAL > 0); 1310 1311 status = AudioSystem::getSamplingRate(output, &mAfSampleRate); 1312 if (status != NO_ERROR) { 1313 ALOGE("getSamplingRate(output=%d) status %d", output, status); 1314 goto release; 1315 } 1316 if (mSampleRate == 0) { 1317 mSampleRate = mAfSampleRate; 1318 mOriginalSampleRate = mAfSampleRate; 1319 } 1320 1321 // Client can only express a preference for FAST. Server will perform additional tests. 1322 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1323 bool useCaseAllowed = 1324 // either of these use cases: 1325 // use case 1: shared buffer 1326 (mSharedBuffer != 0) || 1327 // use case 2: callback transfer mode 1328 (mTransfer == TRANSFER_CALLBACK) || 1329 // use case 3: obtain/release mode 1330 (mTransfer == TRANSFER_OBTAIN) || 1331 // use case 4: synchronous write 1332 ((mTransfer == TRANSFER_SYNC) && mThreadCanCallJava); 1333 // sample rates must also match 1334 bool fastAllowed = useCaseAllowed && (mSampleRate == mAfSampleRate); 1335 if (!fastAllowed) { 1336 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client; transfer %d, " 1337 "track %u Hz, output %u Hz", 1338 mTransfer, mSampleRate, mAfSampleRate); 1339 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1340 } 1341 } 1342 1343 mNotificationFramesAct = mNotificationFramesReq; 1344 1345 size_t frameCount = mReqFrameCount; 1346 if (!audio_has_proportional_frames(mFormat)) { 1347 1348 if (mSharedBuffer != 0) { 1349 // Same comment as below about ignoring frameCount parameter for set() 1350 frameCount = mSharedBuffer->size(); 1351 } else if (frameCount == 0) { 1352 frameCount = mAfFrameCount; 1353 } 1354 if (mNotificationFramesAct != frameCount) { 1355 mNotificationFramesAct = frameCount; 1356 } 1357 } else if (mSharedBuffer != 0) { 1358 // FIXME: Ensure client side memory buffers need 1359 // not have additional alignment beyond sample 1360 // (e.g. 16 bit stereo accessed as 32 bit frame). 1361 size_t alignment = audio_bytes_per_sample(mFormat); 1362 if (alignment & 1) { 1363 // for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java). 1364 alignment = 1; 1365 } 1366 if (mChannelCount > 1) { 1367 // More than 2 channels does not require stronger alignment than stereo 1368 alignment <<= 1; 1369 } 1370 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 1371 ALOGE("Invalid buffer alignment: address %p, channel count %u", 1372 mSharedBuffer->pointer(), mChannelCount); 1373 status = BAD_VALUE; 1374 goto release; 1375 } 1376 1377 // When initializing a shared buffer AudioTrack via constructors, 1378 // there's no frameCount parameter. 1379 // But when initializing a shared buffer AudioTrack via set(), 1380 // there _is_ a frameCount parameter. We silently ignore it. 1381 frameCount = mSharedBuffer->size() / mFrameSize; 1382 } else { 1383 size_t minFrameCount = 0; 1384 // For fast tracks the frame count calculations and checks are mostly done by server, 1385 // but we try to respect the application's request for notifications per buffer. 1386 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1387 if (mNotificationsPerBufferReq > 0) { 1388 // Avoid possible arithmetic overflow during multiplication. 1389 // mNotificationsPerBuffer is clamped to a small integer earlier, so it is unlikely. 1390 if (mNotificationsPerBufferReq > SIZE_MAX / afFrameCountHAL) { 1391 ALOGE("Requested notificationPerBuffer=%u ignored for HAL frameCount=%zu", 1392 mNotificationsPerBufferReq, afFrameCountHAL); 1393 } else { 1394 minFrameCount = afFrameCountHAL * mNotificationsPerBufferReq; 1395 } 1396 } 1397 } else { 1398 // for normal tracks precompute the frame count based on speed. 1399 const float speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f : 1400 max(mMaxRequiredSpeed, mPlaybackRate.mSpeed); 1401 minFrameCount = calculateMinFrameCount( 1402 mAfLatency, mAfFrameCount, mAfSampleRate, mSampleRate, 1403 speed /*, 0 mNotificationsPerBufferReq*/); 1404 } 1405 if (frameCount < minFrameCount) { 1406 frameCount = minFrameCount; 1407 } 1408 } 1409 1410 audio_output_flags_t flags = mFlags; 1411 1412 pid_t tid = -1; 1413 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1414 if (mAudioTrackThread != 0 && !mThreadCanCallJava) { 1415 tid = mAudioTrackThread->getTid(); 1416 } 1417 } 1418 1419 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1420 // but we will still need the original value also 1421 audio_session_t originalSessionId = mSessionId; 1422 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 1423 mSampleRate, 1424 mFormat, 1425 mChannelMask, 1426 &temp, 1427 &flags, 1428 mSharedBuffer, 1429 output, 1430 mClientPid, 1431 tid, 1432 &mSessionId, 1433 mClientUid, 1434 &status, 1435 mPortId); 1436 ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId, 1437 "session ID changed from %d to %d", originalSessionId, mSessionId); 1438 1439 if (status != NO_ERROR) { 1440 ALOGE("AudioFlinger could not create track, status: %d", status); 1441 goto release; 1442 } 1443 ALOG_ASSERT(track != 0); 1444 1445 // AudioFlinger now owns the reference to the I/O handle, 1446 // so we are no longer responsible for releasing it. 1447 1448 // FIXME compare to AudioRecord 1449 sp<IMemory> iMem = track->getCblk(); 1450 if (iMem == 0) { 1451 ALOGE("Could not get control block"); 1452 return NO_INIT; 1453 } 1454 void *iMemPointer = iMem->pointer(); 1455 if (iMemPointer == NULL) { 1456 ALOGE("Could not get control block pointer"); 1457 return NO_INIT; 1458 } 1459 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1460 if (mAudioTrack != 0) { 1461 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this); 1462 mDeathNotifier.clear(); 1463 } 1464 mAudioTrack = track; 1465 mCblkMemory = iMem; 1466 IPCThreadState::self()->flushCommands(); 1467 1468 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1469 mCblk = cblk; 1470 // note that temp is the (possibly revised) value of frameCount 1471 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1472 // In current design, AudioTrack client checks and ensures frame count validity before 1473 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1474 // for fast track as it uses a special method of assigning frame count. 1475 ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp); 1476 } 1477 frameCount = temp; 1478 1479 mAwaitBoost = false; 1480 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1481 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 1482 ALOGI("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu", frameCount, temp); 1483 if (!mThreadCanCallJava) { 1484 mAwaitBoost = true; 1485 } 1486 } else { 1487 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu", frameCount, 1488 temp); 1489 } 1490 } 1491 mFlags = flags; 1492 1493 // Make sure that application is notified with sufficient margin before underrun. 1494 // The client can divide the AudioTrack buffer into sub-buffers, 1495 // and expresses its desire to server as the notification frame count. 1496 if (mSharedBuffer == 0 && audio_is_linear_pcm(mFormat)) { 1497 size_t maxNotificationFrames; 1498 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1499 // notify every HAL buffer, regardless of the size of the track buffer 1500 maxNotificationFrames = afFrameCountHAL; 1501 } else { 1502 // For normal tracks, use at least double-buffering if no sample rate conversion, 1503 // or at least triple-buffering if there is sample rate conversion 1504 const int nBuffering = mOriginalSampleRate == mAfSampleRate ? 2 : 3; 1505 maxNotificationFrames = frameCount / nBuffering; 1506 } 1507 if (mNotificationFramesAct == 0 || mNotificationFramesAct > maxNotificationFrames) { 1508 if (mNotificationFramesAct == 0) { 1509 ALOGD("Client defaulted notificationFrames to %zu for frameCount %zu", 1510 maxNotificationFrames, frameCount); 1511 } else { 1512 ALOGW("Client adjusted notificationFrames from %u to %zu for frameCount %zu", 1513 mNotificationFramesAct, maxNotificationFrames, frameCount); 1514 } 1515 mNotificationFramesAct = (uint32_t) maxNotificationFrames; 1516 } 1517 } 1518 1519 // We retain a copy of the I/O handle, but don't own the reference 1520 mOutput = output; 1521 mRefreshRemaining = true; 1522 1523 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1524 // is the value of pointer() for the shared buffer, otherwise buffers points 1525 // immediately after the control block. This address is for the mapping within client 1526 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1527 void* buffers; 1528 if (mSharedBuffer == 0) { 1529 buffers = cblk + 1; 1530 } else { 1531 buffers = mSharedBuffer->pointer(); 1532 if (buffers == NULL) { 1533 ALOGE("Could not get buffer pointer"); 1534 return NO_INIT; 1535 } 1536 } 1537 1538 mAudioTrack->attachAuxEffect(mAuxEffectId); 1539 // FIXME doesn't take into account speed or future sample rate changes (until restoreTrack) 1540 // FIXME don't believe this lie 1541 mLatency = mAfLatency + (1000*frameCount) / mSampleRate; 1542 1543 mFrameCount = frameCount; 1544 // If IAudioTrack is re-created, don't let the requested frameCount 1545 // decrease. This can confuse clients that cache frameCount(). 1546 if (frameCount > mReqFrameCount) { 1547 mReqFrameCount = frameCount; 1548 } 1549 1550 // reset server position to 0 as we have new cblk. 1551 mServer = 0; 1552 1553 // update proxy 1554 if (mSharedBuffer == 0) { 1555 mStaticProxy.clear(); 1556 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize); 1557 } else { 1558 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize); 1559 mProxy = mStaticProxy; 1560 } 1561 1562 mProxy->setVolumeLR(gain_minifloat_pack( 1563 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]), 1564 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT]))); 1565 1566 mProxy->setSendLevel(mSendLevel); 1567 const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch); 1568 const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch); 1569 const float effectivePitch = adjustPitch(mPlaybackRate.mPitch); 1570 mProxy->setSampleRate(effectiveSampleRate); 1571 1572 AudioPlaybackRate playbackRateTemp = mPlaybackRate; 1573 playbackRateTemp.mSpeed = effectiveSpeed; 1574 playbackRateTemp.mPitch = effectivePitch; 1575 mProxy->setPlaybackRate(playbackRateTemp); 1576 mProxy->setMinimum(mNotificationFramesAct); 1577 1578 mDeathNotifier = new DeathNotifier(this); 1579 IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this); 1580 1581 if (mDeviceCallback != 0) { 1582 AudioSystem::addAudioDeviceCallback(mDeviceCallback, mOutput); 1583 } 1584 1585 return NO_ERROR; 1586 } 1587 1588release: 1589 AudioSystem::releaseOutput(output, streamType, mSessionId); 1590 if (status == NO_ERROR) { 1591 status = NO_INIT; 1592 } 1593 return status; 1594} 1595 1596status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig) 1597{ 1598 if (audioBuffer == NULL) { 1599 if (nonContig != NULL) { 1600 *nonContig = 0; 1601 } 1602 return BAD_VALUE; 1603 } 1604 if (mTransfer != TRANSFER_OBTAIN) { 1605 audioBuffer->frameCount = 0; 1606 audioBuffer->size = 0; 1607 audioBuffer->raw = NULL; 1608 if (nonContig != NULL) { 1609 *nonContig = 0; 1610 } 1611 return INVALID_OPERATION; 1612 } 1613 1614 const struct timespec *requested; 1615 struct timespec timeout; 1616 if (waitCount == -1) { 1617 requested = &ClientProxy::kForever; 1618 } else if (waitCount == 0) { 1619 requested = &ClientProxy::kNonBlocking; 1620 } else if (waitCount > 0) { 1621 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1622 timeout.tv_sec = ms / 1000; 1623 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1624 requested = &timeout; 1625 } else { 1626 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1627 requested = NULL; 1628 } 1629 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig); 1630} 1631 1632status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1633 struct timespec *elapsed, size_t *nonContig) 1634{ 1635 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1636 uint32_t oldSequence = 0; 1637 uint32_t newSequence; 1638 1639 Proxy::Buffer buffer; 1640 status_t status = NO_ERROR; 1641 1642 static const int32_t kMaxTries = 5; 1643 int32_t tryCounter = kMaxTries; 1644 1645 do { 1646 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1647 // keep them from going away if another thread re-creates the track during obtainBuffer() 1648 sp<AudioTrackClientProxy> proxy; 1649 sp<IMemory> iMem; 1650 1651 { // start of lock scope 1652 AutoMutex lock(mLock); 1653 1654 newSequence = mSequence; 1655 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1656 if (status == DEAD_OBJECT) { 1657 // re-create track, unless someone else has already done so 1658 if (newSequence == oldSequence) { 1659 status = restoreTrack_l("obtainBuffer"); 1660 if (status != NO_ERROR) { 1661 buffer.mFrameCount = 0; 1662 buffer.mRaw = NULL; 1663 buffer.mNonContig = 0; 1664 break; 1665 } 1666 } 1667 } 1668 oldSequence = newSequence; 1669 1670 if (status == NOT_ENOUGH_DATA) { 1671 restartIfDisabled(); 1672 } 1673 1674 // Keep the extra references 1675 proxy = mProxy; 1676 iMem = mCblkMemory; 1677 1678 if (mState == STATE_STOPPING) { 1679 status = -EINTR; 1680 buffer.mFrameCount = 0; 1681 buffer.mRaw = NULL; 1682 buffer.mNonContig = 0; 1683 break; 1684 } 1685 1686 // Non-blocking if track is stopped or paused 1687 if (mState != STATE_ACTIVE) { 1688 requested = &ClientProxy::kNonBlocking; 1689 } 1690 1691 } // end of lock scope 1692 1693 buffer.mFrameCount = audioBuffer->frameCount; 1694 // FIXME starts the requested timeout and elapsed over from scratch 1695 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1696 } while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0)); 1697 1698 audioBuffer->frameCount = buffer.mFrameCount; 1699 audioBuffer->size = buffer.mFrameCount * mFrameSize; 1700 audioBuffer->raw = buffer.mRaw; 1701 if (nonContig != NULL) { 1702 *nonContig = buffer.mNonContig; 1703 } 1704 return status; 1705} 1706 1707void AudioTrack::releaseBuffer(const Buffer* audioBuffer) 1708{ 1709 // FIXME add error checking on mode, by adding an internal version 1710 if (mTransfer == TRANSFER_SHARED) { 1711 return; 1712 } 1713 1714 size_t stepCount = audioBuffer->size / mFrameSize; 1715 if (stepCount == 0) { 1716 return; 1717 } 1718 1719 Proxy::Buffer buffer; 1720 buffer.mFrameCount = stepCount; 1721 buffer.mRaw = audioBuffer->raw; 1722 1723 AutoMutex lock(mLock); 1724 mReleased += stepCount; 1725 mInUnderrun = false; 1726 mProxy->releaseBuffer(&buffer); 1727 1728 // restart track if it was disabled by audioflinger due to previous underrun 1729 restartIfDisabled(); 1730} 1731 1732void AudioTrack::restartIfDisabled() 1733{ 1734 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 1735 if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) { 1736 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1737 // FIXME ignoring status 1738 mAudioTrack->start(); 1739 } 1740} 1741 1742// ------------------------------------------------------------------------- 1743 1744ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1745{ 1746 if (mTransfer != TRANSFER_SYNC) { 1747 return INVALID_OPERATION; 1748 } 1749 1750 if (isDirect()) { 1751 AutoMutex lock(mLock); 1752 int32_t flags = android_atomic_and( 1753 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), 1754 &mCblk->mFlags); 1755 if (flags & CBLK_INVALID) { 1756 return DEAD_OBJECT; 1757 } 1758 } 1759 1760 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1761 // Sanity-check: user is most-likely passing an error code, and it would 1762 // make the return value ambiguous (actualSize vs error). 1763 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1764 return BAD_VALUE; 1765 } 1766 1767 size_t written = 0; 1768 Buffer audioBuffer; 1769 1770 while (userSize >= mFrameSize) { 1771 audioBuffer.frameCount = userSize / mFrameSize; 1772 1773 status_t err = obtainBuffer(&audioBuffer, 1774 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1775 if (err < 0) { 1776 if (written > 0) { 1777 break; 1778 } 1779 if (err == TIMED_OUT || err == -EINTR) { 1780 err = WOULD_BLOCK; 1781 } 1782 return ssize_t(err); 1783 } 1784 1785 size_t toWrite = audioBuffer.size; 1786 memcpy(audioBuffer.i8, buffer, toWrite); 1787 buffer = ((const char *) buffer) + toWrite; 1788 userSize -= toWrite; 1789 written += toWrite; 1790 1791 releaseBuffer(&audioBuffer); 1792 } 1793 1794 if (written > 0) { 1795 mFramesWritten += written / mFrameSize; 1796 } 1797 return written; 1798} 1799 1800// ------------------------------------------------------------------------- 1801 1802nsecs_t AudioTrack::processAudioBuffer() 1803{ 1804 // Currently the AudioTrack thread is not created if there are no callbacks. 1805 // Would it ever make sense to run the thread, even without callbacks? 1806 // If so, then replace this by checks at each use for mCbf != NULL. 1807 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1808 1809 mLock.lock(); 1810 if (mAwaitBoost) { 1811 mAwaitBoost = false; 1812 mLock.unlock(); 1813 static const int32_t kMaxTries = 5; 1814 int32_t tryCounter = kMaxTries; 1815 uint32_t pollUs = 10000; 1816 do { 1817 int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK; 1818 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1819 break; 1820 } 1821 usleep(pollUs); 1822 pollUs <<= 1; 1823 } while (tryCounter-- > 0); 1824 if (tryCounter < 0) { 1825 ALOGE("did not receive expected priority boost on time"); 1826 } 1827 // Run again immediately 1828 return 0; 1829 } 1830 1831 // Can only reference mCblk while locked 1832 int32_t flags = android_atomic_and( 1833 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1834 1835 // Check for track invalidation 1836 if (flags & CBLK_INVALID) { 1837 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1838 // AudioSystem cache. We should not exit here but after calling the callback so 1839 // that the upper layers can recreate the track 1840 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) { 1841 status_t status __unused = restoreTrack_l("processAudioBuffer"); 1842 // FIXME unused status 1843 // after restoration, continue below to make sure that the loop and buffer events 1844 // are notified because they have been cleared from mCblk->mFlags above. 1845 } 1846 } 1847 1848 bool waitStreamEnd = mState == STATE_STOPPING; 1849 bool active = mState == STATE_ACTIVE; 1850 1851 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1852 bool newUnderrun = false; 1853 if (flags & CBLK_UNDERRUN) { 1854#if 0 1855 // Currently in shared buffer mode, when the server reaches the end of buffer, 1856 // the track stays active in continuous underrun state. It's up to the application 1857 // to pause or stop the track, or set the position to a new offset within buffer. 1858 // This was some experimental code to auto-pause on underrun. Keeping it here 1859 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1860 if (mTransfer == TRANSFER_SHARED) { 1861 mState = STATE_PAUSED; 1862 active = false; 1863 } 1864#endif 1865 if (!mInUnderrun) { 1866 mInUnderrun = true; 1867 newUnderrun = true; 1868 } 1869 } 1870 1871 // Get current position of server 1872 Modulo<uint32_t> position(updateAndGetPosition_l()); 1873 1874 // Manage marker callback 1875 bool markerReached = false; 1876 Modulo<uint32_t> markerPosition(mMarkerPosition); 1877 // uses 32 bit wraparound for comparison with position. 1878 if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) { 1879 mMarkerReached = markerReached = true; 1880 } 1881 1882 // Determine number of new position callback(s) that will be needed, while locked 1883 size_t newPosCount = 0; 1884 Modulo<uint32_t> newPosition(mNewPosition); 1885 uint32_t updatePeriod = mUpdatePeriod; 1886 // FIXME fails for wraparound, need 64 bits 1887 if (updatePeriod > 0 && position >= newPosition) { 1888 newPosCount = ((position - newPosition).value() / updatePeriod) + 1; 1889 mNewPosition += updatePeriod * newPosCount; 1890 } 1891 1892 // Cache other fields that will be needed soon 1893 uint32_t sampleRate = mSampleRate; 1894 float speed = mPlaybackRate.mSpeed; 1895 const uint32_t notificationFrames = mNotificationFramesAct; 1896 if (mRefreshRemaining) { 1897 mRefreshRemaining = false; 1898 mRemainingFrames = notificationFrames; 1899 mRetryOnPartialBuffer = false; 1900 } 1901 size_t misalignment = mProxy->getMisalignment(); 1902 uint32_t sequence = mSequence; 1903 sp<AudioTrackClientProxy> proxy = mProxy; 1904 1905 // Determine the number of new loop callback(s) that will be needed, while locked. 1906 int loopCountNotifications = 0; 1907 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END 1908 1909 if (mLoopCount > 0) { 1910 int loopCount; 1911 size_t bufferPosition; 1912 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount); 1913 loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition; 1914 loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications); 1915 mLoopCountNotified = loopCount; // discard any excess notifications 1916 } else if (mLoopCount < 0) { 1917 // FIXME: We're not accurate with notification count and position with infinite looping 1918 // since loopCount from server side will always return -1 (we could decrement it). 1919 size_t bufferPosition = mStaticProxy->getBufferPosition(); 1920 loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0); 1921 loopPeriod = mLoopEnd - bufferPosition; 1922 } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) { 1923 size_t bufferPosition = mStaticProxy->getBufferPosition(); 1924 loopPeriod = mFrameCount - bufferPosition; 1925 } 1926 1927 // These fields don't need to be cached, because they are assigned only by set(): 1928 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFlags 1929 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1930 1931 mLock.unlock(); 1932 1933 // get anchor time to account for callbacks. 1934 const nsecs_t timeBeforeCallbacks = systemTime(); 1935 1936 if (waitStreamEnd) { 1937 // FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread 1938 // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function 1939 // (and make sure we don't callback for more data while we're stopping). 1940 // This helps with position, marker notifications, and track invalidation. 1941 struct timespec timeout; 1942 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1943 timeout.tv_nsec = 0; 1944 1945 status_t status = proxy->waitStreamEndDone(&timeout); 1946 switch (status) { 1947 case NO_ERROR: 1948 case DEAD_OBJECT: 1949 case TIMED_OUT: 1950 if (status != DEAD_OBJECT) { 1951 // for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop(); 1952 // instead, the application should handle the EVENT_NEW_IAUDIOTRACK. 1953 mCbf(EVENT_STREAM_END, mUserData, NULL); 1954 } 1955 { 1956 AutoMutex lock(mLock); 1957 // The previously assigned value of waitStreamEnd is no longer valid, 1958 // since the mutex has been unlocked and either the callback handler 1959 // or another thread could have re-started the AudioTrack during that time. 1960 waitStreamEnd = mState == STATE_STOPPING; 1961 if (waitStreamEnd) { 1962 mState = STATE_STOPPED; 1963 mReleased = 0; 1964 } 1965 } 1966 if (waitStreamEnd && status != DEAD_OBJECT) { 1967 return NS_INACTIVE; 1968 } 1969 break; 1970 } 1971 return 0; 1972 } 1973 1974 // perform callbacks while unlocked 1975 if (newUnderrun) { 1976 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1977 } 1978 while (loopCountNotifications > 0) { 1979 mCbf(EVENT_LOOP_END, mUserData, NULL); 1980 --loopCountNotifications; 1981 } 1982 if (flags & CBLK_BUFFER_END) { 1983 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1984 } 1985 if (markerReached) { 1986 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1987 } 1988 while (newPosCount > 0) { 1989 size_t temp = newPosition.value(); // FIXME size_t != uint32_t 1990 mCbf(EVENT_NEW_POS, mUserData, &temp); 1991 newPosition += updatePeriod; 1992 newPosCount--; 1993 } 1994 1995 if (mObservedSequence != sequence) { 1996 mObservedSequence = sequence; 1997 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1998 // for offloaded tracks, just wait for the upper layers to recreate the track 1999 if (isOffloadedOrDirect()) { 2000 return NS_INACTIVE; 2001 } 2002 } 2003 2004 // if inactive, then don't run me again until re-started 2005 if (!active) { 2006 return NS_INACTIVE; 2007 } 2008 2009 // Compute the estimated time until the next timed event (position, markers, loops) 2010 // FIXME only for non-compressed audio 2011 uint32_t minFrames = ~0; 2012 if (!markerReached && position < markerPosition) { 2013 minFrames = (markerPosition - position).value(); 2014 } 2015 if (loopPeriod > 0 && loopPeriod < minFrames) { 2016 // loopPeriod is already adjusted for actual position. 2017 minFrames = loopPeriod; 2018 } 2019 if (updatePeriod > 0) { 2020 minFrames = min(minFrames, (newPosition - position).value()); 2021 } 2022 2023 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 2024 static const uint32_t kPoll = 0; 2025 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 2026 minFrames = kPoll * notificationFrames; 2027 } 2028 2029 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 2030 static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL; 2031 const nsecs_t timeAfterCallbacks = systemTime(); 2032 2033 // Convert frame units to time units 2034 nsecs_t ns = NS_WHENEVER; 2035 if (minFrames != (uint32_t) ~0) { 2036 ns = framesToNanoseconds(minFrames, sampleRate, speed) + kWaitPeriodNs; 2037 ns -= (timeAfterCallbacks - timeBeforeCallbacks); // account for callback time 2038 // TODO: Should we warn if the callback time is too long? 2039 if (ns < 0) ns = 0; 2040 } 2041 2042 // If not supplying data by EVENT_MORE_DATA, then we're done 2043 if (mTransfer != TRANSFER_CALLBACK) { 2044 return ns; 2045 } 2046 2047 // EVENT_MORE_DATA callback handling. 2048 // Timing for linear pcm audio data formats can be derived directly from the 2049 // buffer fill level. 2050 // Timing for compressed data is not directly available from the buffer fill level, 2051 // rather indirectly from waiting for blocking mode callbacks or waiting for obtain() 2052 // to return a certain fill level. 2053 2054 struct timespec timeout; 2055 const struct timespec *requested = &ClientProxy::kForever; 2056 if (ns != NS_WHENEVER) { 2057 timeout.tv_sec = ns / 1000000000LL; 2058 timeout.tv_nsec = ns % 1000000000LL; 2059 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 2060 requested = &timeout; 2061 } 2062 2063 size_t writtenFrames = 0; 2064 while (mRemainingFrames > 0) { 2065 2066 Buffer audioBuffer; 2067 audioBuffer.frameCount = mRemainingFrames; 2068 size_t nonContig; 2069 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 2070 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 2071 "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount); 2072 requested = &ClientProxy::kNonBlocking; 2073 size_t avail = audioBuffer.frameCount + nonContig; 2074 ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d", 2075 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 2076 if (err != NO_ERROR) { 2077 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 2078 (isOffloaded() && (err == DEAD_OBJECT))) { 2079 // FIXME bug 25195759 2080 return 1000000; 2081 } 2082 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 2083 return NS_NEVER; 2084 } 2085 2086 if (mRetryOnPartialBuffer && audio_has_proportional_frames(mFormat)) { 2087 mRetryOnPartialBuffer = false; 2088 if (avail < mRemainingFrames) { 2089 if (ns > 0) { // account for obtain time 2090 const nsecs_t timeNow = systemTime(); 2091 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks)); 2092 } 2093 nsecs_t myns = framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed); 2094 if (ns < 0 /* NS_WHENEVER */ || myns < ns) { 2095 ns = myns; 2096 } 2097 return ns; 2098 } 2099 } 2100 2101 size_t reqSize = audioBuffer.size; 2102 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 2103 size_t writtenSize = audioBuffer.size; 2104 2105 // Sanity check on returned size 2106 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 2107 ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes", 2108 reqSize, ssize_t(writtenSize)); 2109 return NS_NEVER; 2110 } 2111 2112 if (writtenSize == 0) { 2113 // The callback is done filling buffers 2114 // Keep this thread going to handle timed events and 2115 // still try to get more data in intervals of WAIT_PERIOD_MS 2116 // but don't just loop and block the CPU, so wait 2117 2118 // mCbf(EVENT_MORE_DATA, ...) might either 2119 // (1) Block until it can fill the buffer, returning 0 size on EOS. 2120 // (2) Block until it can fill the buffer, returning 0 data (silence) on EOS. 2121 // (3) Return 0 size when no data is available, does not wait for more data. 2122 // 2123 // (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer. 2124 // We try to compute the wait time to avoid a tight sleep-wait cycle, 2125 // especially for case (3). 2126 // 2127 // The decision to support (1) and (2) affect the sizing of mRemainingFrames 2128 // and this loop; whereas for case (3) we could simply check once with the full 2129 // buffer size and skip the loop entirely. 2130 2131 nsecs_t myns; 2132 if (audio_has_proportional_frames(mFormat)) { 2133 // time to wait based on buffer occupancy 2134 const nsecs_t datans = mRemainingFrames <= avail ? 0 : 2135 framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed); 2136 // audio flinger thread buffer size (TODO: adjust for fast tracks) 2137 // FIXME: use mAfFrameCountHAL instead of mAfFrameCount below for fast tracks. 2138 const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed); 2139 // add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0. 2140 myns = datans + (afns / 2); 2141 } else { 2142 // FIXME: This could ping quite a bit if the buffer isn't full. 2143 // Note that when mState is stopping we waitStreamEnd, so it never gets here. 2144 myns = kWaitPeriodNs; 2145 } 2146 if (ns > 0) { // account for obtain and callback time 2147 const nsecs_t timeNow = systemTime(); 2148 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks)); 2149 } 2150 if (ns < 0 /* NS_WHENEVER */ || myns < ns) { 2151 ns = myns; 2152 } 2153 return ns; 2154 } 2155 2156 size_t releasedFrames = writtenSize / mFrameSize; 2157 audioBuffer.frameCount = releasedFrames; 2158 mRemainingFrames -= releasedFrames; 2159 if (misalignment >= releasedFrames) { 2160 misalignment -= releasedFrames; 2161 } else { 2162 misalignment = 0; 2163 } 2164 2165 releaseBuffer(&audioBuffer); 2166 writtenFrames += releasedFrames; 2167 2168 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 2169 // if callback doesn't like to accept the full chunk 2170 if (writtenSize < reqSize) { 2171 continue; 2172 } 2173 2174 // There could be enough non-contiguous frames available to satisfy the remaining request 2175 if (mRemainingFrames <= nonContig) { 2176 continue; 2177 } 2178 2179#if 0 2180 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 2181 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 2182 // that total to a sum == notificationFrames. 2183 if (0 < misalignment && misalignment <= mRemainingFrames) { 2184 mRemainingFrames = misalignment; 2185 return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed); 2186 } 2187#endif 2188 2189 } 2190 if (writtenFrames > 0) { 2191 AutoMutex lock(mLock); 2192 mFramesWritten += writtenFrames; 2193 } 2194 mRemainingFrames = notificationFrames; 2195 mRetryOnPartialBuffer = true; 2196 2197 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 2198 return 0; 2199} 2200 2201status_t AudioTrack::restoreTrack_l(const char *from) 2202{ 2203 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 2204 isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from); 2205 ++mSequence; 2206 2207 // refresh the audio configuration cache in this process to make sure we get new 2208 // output parameters and new IAudioFlinger in createTrack_l() 2209 AudioSystem::clearAudioConfigCache(); 2210 2211 if (isOffloadedOrDirect_l() || mDoNotReconnect) { 2212 // FIXME re-creation of offloaded and direct tracks is not yet implemented; 2213 // reconsider enabling for linear PCM encodings when position can be preserved. 2214 return DEAD_OBJECT; 2215 } 2216 2217 // Save so we can return count since creation. 2218 mUnderrunCountOffset = getUnderrunCount_l(); 2219 2220 // save the old static buffer position 2221 uint32_t staticPosition = 0; 2222 size_t bufferPosition = 0; 2223 int loopCount = 0; 2224 if (mStaticProxy != 0) { 2225 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount); 2226 staticPosition = mStaticProxy->getPosition().unsignedValue(); 2227 } 2228 2229 mFlags = mOrigFlags; 2230 2231 // If a new IAudioTrack is successfully created, createTrack_l() will modify the 2232 // following member variables: mAudioTrack, mCblkMemory and mCblk. 2233 // It will also delete the strong references on previous IAudioTrack and IMemory. 2234 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact. 2235 status_t result = createTrack_l(); 2236 2237 if (result == NO_ERROR) { 2238 // take the frames that will be lost by track recreation into account in saved position 2239 // For streaming tracks, this is the amount we obtained from the user/client 2240 // (not the number actually consumed at the server - those are already lost). 2241 if (mStaticProxy == 0) { 2242 mPosition = mReleased; 2243 } 2244 // Continue playback from last known position and restore loop. 2245 if (mStaticProxy != 0) { 2246 if (loopCount != 0) { 2247 mStaticProxy->setBufferPositionAndLoop(bufferPosition, 2248 mLoopStart, mLoopEnd, loopCount); 2249 } else { 2250 mStaticProxy->setBufferPosition(bufferPosition); 2251 if (bufferPosition == mFrameCount) { 2252 ALOGD("restoring track at end of static buffer"); 2253 } 2254 } 2255 } 2256 // restore volume handler 2257 mVolumeHandler->forall([this](const sp<VolumeShaper::Configuration> &configuration, 2258 const sp<VolumeShaper::Operation> &operation) -> VolumeShaper::Status { 2259 sp<VolumeShaper::Operation> operationToEnd = new VolumeShaper::Operation(*operation); 2260 // TODO: Ideally we would restore to the exact xOffset position 2261 // as returned by getVolumeShaperState(), but we don't have that 2262 // information when restoring at the client unless we periodically poll 2263 // the server or create shared memory state. 2264 // 2265 // For now, we simply advance to the end of the VolumeShaper effect. 2266 operationToEnd->setXOffset(1.f); 2267 return mAudioTrack->applyVolumeShaper(configuration, operationToEnd); 2268 }); 2269 2270 if (mState == STATE_ACTIVE) { 2271 result = mAudioTrack->start(); 2272 } 2273 // server resets to zero so we offset 2274 mFramesWrittenServerOffset = 2275 mStaticProxy.get() != nullptr ? staticPosition : mFramesWritten; 2276 mFramesWrittenAtRestore = mFramesWrittenServerOffset; 2277 } 2278 if (result != NO_ERROR) { 2279 ALOGW("restoreTrack_l() failed status %d", result); 2280 mState = STATE_STOPPED; 2281 mReleased = 0; 2282 } 2283 2284 return result; 2285} 2286 2287Modulo<uint32_t> AudioTrack::updateAndGetPosition_l() 2288{ 2289 // This is the sole place to read server consumed frames 2290 Modulo<uint32_t> newServer(mProxy->getPosition()); 2291 const int32_t delta = (newServer - mServer).signedValue(); 2292 // TODO There is controversy about whether there can be "negative jitter" in server position. 2293 // This should be investigated further, and if possible, it should be addressed. 2294 // A more definite failure mode is infrequent polling by client. 2295 // One could call (void)getPosition_l() in releaseBuffer(), 2296 // so mReleased and mPosition are always lock-step as best possible. 2297 // That should ensure delta never goes negative for infrequent polling 2298 // unless the server has more than 2^31 frames in its buffer, 2299 // in which case the use of uint32_t for these counters has bigger issues. 2300 ALOGE_IF(delta < 0, 2301 "detected illegal retrograde motion by the server: mServer advanced by %d", 2302 delta); 2303 mServer = newServer; 2304 if (delta > 0) { // avoid retrograde 2305 mPosition += delta; 2306 } 2307 return mPosition; 2308} 2309 2310bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const 2311{ 2312 // applicable for mixing tracks only (not offloaded or direct) 2313 if (mStaticProxy != 0) { 2314 return true; // static tracks do not have issues with buffer sizing. 2315 } 2316 const size_t minFrameCount = 2317 calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed 2318 /*, 0 mNotificationsPerBufferReq*/); 2319 ALOGV("isSampleRateSpeedAllowed_l mFrameCount %zu minFrameCount %zu", 2320 mFrameCount, minFrameCount); 2321 return mFrameCount >= minFrameCount; 2322} 2323 2324status_t AudioTrack::setParameters(const String8& keyValuePairs) 2325{ 2326 AutoMutex lock(mLock); 2327 return mAudioTrack->setParameters(keyValuePairs); 2328} 2329 2330VolumeShaper::Status AudioTrack::applyVolumeShaper( 2331 const sp<VolumeShaper::Configuration>& configuration, 2332 const sp<VolumeShaper::Operation>& operation) 2333{ 2334 AutoMutex lock(mLock); 2335 mVolumeHandler->setIdIfNecessary(configuration); 2336 VolumeShaper::Status status = mAudioTrack->applyVolumeShaper(configuration, operation); 2337 if (status >= 0) { 2338 // save VolumeShaper for restore 2339 mVolumeHandler->applyVolumeShaper(configuration, operation); 2340 } 2341 return status; 2342} 2343 2344sp<VolumeShaper::State> AudioTrack::getVolumeShaperState(int id) 2345{ 2346 // TODO: To properly restore the AudioTrack 2347 // we will need to save the last state in AudioTrackShared. 2348 AutoMutex lock(mLock); 2349 return mAudioTrack->getVolumeShaperState(id); 2350} 2351 2352status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp) 2353{ 2354 if (timestamp == nullptr) { 2355 return BAD_VALUE; 2356 } 2357 AutoMutex lock(mLock); 2358 return getTimestamp_l(timestamp); 2359} 2360 2361status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp) 2362{ 2363 if (mCblk->mFlags & CBLK_INVALID) { 2364 const status_t status = restoreTrack_l("getTimestampExtended"); 2365 if (status != OK) { 2366 // per getTimestamp() API doc in header, we return DEAD_OBJECT here, 2367 // recommending that the track be recreated. 2368 return DEAD_OBJECT; 2369 } 2370 } 2371 // check for offloaded/direct here in case restoring somehow changed those flags. 2372 if (isOffloadedOrDirect_l()) { 2373 return INVALID_OPERATION; // not supported 2374 } 2375 status_t status = mProxy->getTimestamp(timestamp); 2376 LOG_ALWAYS_FATAL_IF(status != OK, "status %d not allowed from proxy getTimestamp", status); 2377 bool found = false; 2378 timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten; 2379 timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0; 2380 // server side frame offset in case AudioTrack has been restored. 2381 for (int i = ExtendedTimestamp::LOCATION_SERVER; 2382 i < ExtendedTimestamp::LOCATION_MAX; ++i) { 2383 if (timestamp->mTimeNs[i] >= 0) { 2384 // apply server offset (frames flushed is ignored 2385 // so we don't report the jump when the flush occurs). 2386 timestamp->mPosition[i] += mFramesWrittenServerOffset; 2387 found = true; 2388 } 2389 } 2390 return found ? OK : WOULD_BLOCK; 2391} 2392 2393status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 2394{ 2395 AutoMutex lock(mLock); 2396 return getTimestamp_l(timestamp); 2397} 2398 2399status_t AudioTrack::getTimestamp_l(AudioTimestamp& timestamp) 2400{ 2401 bool previousTimestampValid = mPreviousTimestampValid; 2402 // Set false here to cover all the error return cases. 2403 mPreviousTimestampValid = false; 2404 2405 switch (mState) { 2406 case STATE_ACTIVE: 2407 case STATE_PAUSED: 2408 break; // handle below 2409 case STATE_FLUSHED: 2410 case STATE_STOPPED: 2411 return WOULD_BLOCK; 2412 case STATE_STOPPING: 2413 case STATE_PAUSED_STOPPING: 2414 if (!isOffloaded_l()) { 2415 return INVALID_OPERATION; 2416 } 2417 break; // offloaded tracks handled below 2418 default: 2419 LOG_ALWAYS_FATAL("Invalid mState in getTimestamp(): %d", mState); 2420 break; 2421 } 2422 2423 if (mCblk->mFlags & CBLK_INVALID) { 2424 const status_t status = restoreTrack_l("getTimestamp"); 2425 if (status != OK) { 2426 // per getTimestamp() API doc in header, we return DEAD_OBJECT here, 2427 // recommending that the track be recreated. 2428 return DEAD_OBJECT; 2429 } 2430 } 2431 2432 // The presented frame count must always lag behind the consumed frame count. 2433 // To avoid a race, read the presented frames first. This ensures that presented <= consumed. 2434 2435 status_t status; 2436 if (isOffloadedOrDirect_l()) { 2437 // use Binder to get timestamp 2438 status = mAudioTrack->getTimestamp(timestamp); 2439 } else { 2440 // read timestamp from shared memory 2441 ExtendedTimestamp ets; 2442 status = mProxy->getTimestamp(&ets); 2443 if (status == OK) { 2444 ExtendedTimestamp::Location location; 2445 status = ets.getBestTimestamp(×tamp, &location); 2446 2447 if (status == OK) { 2448 // It is possible that the best location has moved from the kernel to the server. 2449 // In this case we adjust the position from the previous computed latency. 2450 if (location == ExtendedTimestamp::LOCATION_SERVER) { 2451 ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL, 2452 "getTimestamp() location moved from kernel to server"); 2453 // check that the last kernel OK time info exists and the positions 2454 // are valid (if they predate the current track, the positions may 2455 // be zero or negative). 2456 const int64_t frames = 2457 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 || 2458 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 || 2459 ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 || 2460 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0) 2461 ? 2462 int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed 2463 / 1000) 2464 : 2465 (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] 2466 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]); 2467 ALOGV("frame adjustment:%lld timestamp:%s", 2468 (long long)frames, ets.toString().c_str()); 2469 if (frames >= ets.mPosition[location]) { 2470 timestamp.mPosition = 0; 2471 } else { 2472 timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames); 2473 } 2474 } else if (location == ExtendedTimestamp::LOCATION_KERNEL) { 2475 ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER, 2476 "getTimestamp() location moved from server to kernel"); 2477 } 2478 2479 // We update the timestamp time even when paused. 2480 if (mState == STATE_PAUSED /* not needed: STATE_PAUSED_STOPPING */) { 2481 const int64_t now = systemTime(); 2482 const int64_t at = convertTimespecToNs(timestamp.mTime); 2483 const int64_t lag = 2484 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 || 2485 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0) 2486 ? int64_t(mAfLatency * 1000000LL) 2487 : (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] 2488 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]) 2489 * NANOS_PER_SECOND / mSampleRate; 2490 const int64_t limit = now - lag; // no earlier than this limit 2491 if (at < limit) { 2492 ALOGV("timestamp pause lag:%lld adjusting from %lld to %lld", 2493 (long long)lag, (long long)at, (long long)limit); 2494 timestamp.mTime.tv_sec = limit / NANOS_PER_SECOND; 2495 timestamp.mTime.tv_nsec = limit % NANOS_PER_SECOND; // compiler opt. 2496 } 2497 } 2498 mPreviousLocation = location; 2499 } else { 2500 // right after AudioTrack is started, one may not find a timestamp 2501 ALOGV("getBestTimestamp did not find timestamp"); 2502 } 2503 } 2504 if (status == INVALID_OPERATION) { 2505 // INVALID_OPERATION occurs when no timestamp has been issued by the server; 2506 // other failures are signaled by a negative time. 2507 // If we come out of FLUSHED or STOPPED where the position is known 2508 // to be zero we convert this to WOULD_BLOCK (with the implicit meaning of 2509 // "zero" for NuPlayer). We don't convert for track restoration as position 2510 // does not reset. 2511 ALOGV("timestamp server offset:%lld restore frames:%lld", 2512 (long long)mFramesWrittenServerOffset, (long long)mFramesWrittenAtRestore); 2513 if (mFramesWrittenServerOffset != mFramesWrittenAtRestore) { 2514 status = WOULD_BLOCK; 2515 } 2516 } 2517 } 2518 if (status != NO_ERROR) { 2519 ALOGV_IF(status != WOULD_BLOCK, "getTimestamp error:%#x", status); 2520 return status; 2521 } 2522 if (isOffloadedOrDirect_l()) { 2523 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) { 2524 // use cached paused position in case another offloaded track is running. 2525 timestamp.mPosition = mPausedPosition; 2526 clock_gettime(CLOCK_MONOTONIC, ×tamp.mTime); 2527 // TODO: adjust for delay 2528 return NO_ERROR; 2529 } 2530 2531 // Check whether a pending flush or stop has completed, as those commands may 2532 // be asynchronous or return near finish or exhibit glitchy behavior. 2533 // 2534 // Originally this showed up as the first timestamp being a continuation of 2535 // the previous song under gapless playback. 2536 // However, we sometimes see zero timestamps, then a glitch of 2537 // the previous song's position, and then correct timestamps afterwards. 2538 if (mStartUs != 0 && mSampleRate != 0) { 2539 static const int kTimeJitterUs = 100000; // 100 ms 2540 static const int k1SecUs = 1000000; 2541 2542 const int64_t timeNow = getNowUs(); 2543 2544 if (timeNow < mStartUs + k1SecUs) { // within first second of starting 2545 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime); 2546 if (timestampTimeUs < mStartUs) { 2547 return WOULD_BLOCK; // stale timestamp time, occurs before start. 2548 } 2549 const int64_t deltaTimeUs = timestampTimeUs - mStartUs; 2550 const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000 2551 / ((double)mSampleRate * mPlaybackRate.mSpeed); 2552 2553 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) { 2554 // Verify that the counter can't count faster than the sample rate 2555 // since the start time. If greater, then that means we may have failed 2556 // to completely flush or stop the previous playing track. 2557 ALOGW_IF(!mTimestampStartupGlitchReported, 2558 "getTimestamp startup glitch detected" 2559 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)", 2560 (long long)deltaTimeUs, (long long)deltaPositionByUs, 2561 timestamp.mPosition); 2562 mTimestampStartupGlitchReported = true; 2563 if (previousTimestampValid 2564 && mPreviousTimestamp.mPosition == 0 /* should be true if valid */) { 2565 timestamp = mPreviousTimestamp; 2566 mPreviousTimestampValid = true; 2567 return NO_ERROR; 2568 } 2569 return WOULD_BLOCK; 2570 } 2571 if (deltaPositionByUs != 0) { 2572 mStartUs = 0; // don't check again, we got valid nonzero position. 2573 } 2574 } else { 2575 mStartUs = 0; // don't check again, start time expired. 2576 } 2577 mTimestampStartupGlitchReported = false; 2578 } 2579 } else { 2580 // Update the mapping between local consumed (mPosition) and server consumed (mServer) 2581 (void) updateAndGetPosition_l(); 2582 // Server consumed (mServer) and presented both use the same server time base, 2583 // and server consumed is always >= presented. 2584 // The delta between these represents the number of frames in the buffer pipeline. 2585 // If this delta between these is greater than the client position, it means that 2586 // actually presented is still stuck at the starting line (figuratively speaking), 2587 // waiting for the first frame to go by. So we can't report a valid timestamp yet. 2588 // Note: We explicitly use non-Modulo comparison here - potential wrap issue when 2589 // mPosition exceeds 32 bits. 2590 // TODO Remove when timestamp is updated to contain pipeline status info. 2591 const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue(); 2592 if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */ 2593 && (uint32_t)pipelineDepthInFrames > mPosition.value()) { 2594 return INVALID_OPERATION; 2595 } 2596 // Convert timestamp position from server time base to client time base. 2597 // TODO The following code should work OK now because timestamp.mPosition is 32-bit. 2598 // But if we change it to 64-bit then this could fail. 2599 // Use Modulo computation here. 2600 timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value(); 2601 // Immediately after a call to getPosition_l(), mPosition and 2602 // mServer both represent the same frame position. mPosition is 2603 // in client's point of view, and mServer is in server's point of 2604 // view. So the difference between them is the "fudge factor" 2605 // between client and server views due to stop() and/or new 2606 // IAudioTrack. And timestamp.mPosition is initially in server's 2607 // point of view, so we need to apply the same fudge factor to it. 2608 } 2609 2610 // Prevent retrograde motion in timestamp. 2611 // This is sometimes caused by erratic reports of the available space in the ALSA drivers. 2612 if (status == NO_ERROR) { 2613 if (previousTimestampValid) { 2614 const int64_t previousTimeNanos = convertTimespecToNs(mPreviousTimestamp.mTime); 2615 const int64_t currentTimeNanos = convertTimespecToNs(timestamp.mTime); 2616 if (currentTimeNanos < previousTimeNanos) { 2617 ALOGW("retrograde timestamp time corrected, %lld < %lld", 2618 (long long)currentTimeNanos, (long long)previousTimeNanos); 2619 timestamp.mTime = mPreviousTimestamp.mTime; 2620 } 2621 2622 // Looking at signed delta will work even when the timestamps 2623 // are wrapping around. 2624 int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition) 2625 - mPreviousTimestamp.mPosition).signedValue(); 2626 if (deltaPosition < 0) { 2627 // Only report once per position instead of spamming the log. 2628 if (!mRetrogradeMotionReported) { 2629 ALOGW("retrograde timestamp position corrected, %d = %u - %u", 2630 deltaPosition, 2631 timestamp.mPosition, 2632 mPreviousTimestamp.mPosition); 2633 mRetrogradeMotionReported = true; 2634 } 2635 } else { 2636 mRetrogradeMotionReported = false; 2637 } 2638 if (deltaPosition < 0) { 2639 timestamp.mPosition = mPreviousTimestamp.mPosition; 2640 deltaPosition = 0; 2641 } 2642#if 0 2643 // Uncomment this to verify audio timestamp rate. 2644 const int64_t deltaTime = 2645 convertTimespecToNs(timestamp.mTime) - previousTimeNanos; 2646 if (deltaTime != 0) { 2647 const int64_t computedSampleRate = 2648 deltaPosition * (long long)NANOS_PER_SECOND / deltaTime; 2649 ALOGD("computedSampleRate:%u sampleRate:%u", 2650 (unsigned)computedSampleRate, mSampleRate); 2651 } 2652#endif 2653 } 2654 mPreviousTimestamp = timestamp; 2655 mPreviousTimestampValid = true; 2656 } 2657 2658 return status; 2659} 2660 2661String8 AudioTrack::getParameters(const String8& keys) 2662{ 2663 audio_io_handle_t output = getOutput(); 2664 if (output != AUDIO_IO_HANDLE_NONE) { 2665 return AudioSystem::getParameters(output, keys); 2666 } else { 2667 return String8::empty(); 2668 } 2669} 2670 2671bool AudioTrack::isOffloaded() const 2672{ 2673 AutoMutex lock(mLock); 2674 return isOffloaded_l(); 2675} 2676 2677bool AudioTrack::isDirect() const 2678{ 2679 AutoMutex lock(mLock); 2680 return isDirect_l(); 2681} 2682 2683bool AudioTrack::isOffloadedOrDirect() const 2684{ 2685 AutoMutex lock(mLock); 2686 return isOffloadedOrDirect_l(); 2687} 2688 2689 2690status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 2691{ 2692 2693 const size_t SIZE = 256; 2694 char buffer[SIZE]; 2695 String8 result; 2696 2697 result.append(" AudioTrack::dump\n"); 2698 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 2699 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 2700 result.append(buffer); 2701 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 2702 mChannelCount, mFrameCount); 2703 result.append(buffer); 2704 snprintf(buffer, 255, " sample rate(%u), speed(%f), status(%d)\n", 2705 mSampleRate, mPlaybackRate.mSpeed, mStatus); 2706 result.append(buffer); 2707 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 2708 result.append(buffer); 2709 ::write(fd, result.string(), result.size()); 2710 return NO_ERROR; 2711} 2712 2713uint32_t AudioTrack::getUnderrunCount() const 2714{ 2715 AutoMutex lock(mLock); 2716 return getUnderrunCount_l(); 2717} 2718 2719uint32_t AudioTrack::getUnderrunCount_l() const 2720{ 2721 return mProxy->getUnderrunCount() + mUnderrunCountOffset; 2722} 2723 2724uint32_t AudioTrack::getUnderrunFrames() const 2725{ 2726 AutoMutex lock(mLock); 2727 return mProxy->getUnderrunFrames(); 2728} 2729 2730status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback) 2731{ 2732 if (callback == 0) { 2733 ALOGW("%s adding NULL callback!", __FUNCTION__); 2734 return BAD_VALUE; 2735 } 2736 AutoMutex lock(mLock); 2737 if (mDeviceCallback == callback) { 2738 ALOGW("%s adding same callback!", __FUNCTION__); 2739 return INVALID_OPERATION; 2740 } 2741 status_t status = NO_ERROR; 2742 if (mOutput != AUDIO_IO_HANDLE_NONE) { 2743 if (mDeviceCallback != 0) { 2744 ALOGW("%s callback already present!", __FUNCTION__); 2745 AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); 2746 } 2747 status = AudioSystem::addAudioDeviceCallback(callback, mOutput); 2748 } 2749 mDeviceCallback = callback; 2750 return status; 2751} 2752 2753status_t AudioTrack::removeAudioDeviceCallback( 2754 const sp<AudioSystem::AudioDeviceCallback>& callback) 2755{ 2756 if (callback == 0) { 2757 ALOGW("%s removing NULL callback!", __FUNCTION__); 2758 return BAD_VALUE; 2759 } 2760 AutoMutex lock(mLock); 2761 if (mDeviceCallback != callback) { 2762 ALOGW("%s removing different callback!", __FUNCTION__); 2763 return INVALID_OPERATION; 2764 } 2765 if (mOutput != AUDIO_IO_HANDLE_NONE) { 2766 AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); 2767 } 2768 mDeviceCallback = 0; 2769 return NO_ERROR; 2770} 2771 2772status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location) 2773{ 2774 if (msec == nullptr || 2775 (location != ExtendedTimestamp::LOCATION_SERVER 2776 && location != ExtendedTimestamp::LOCATION_KERNEL)) { 2777 return BAD_VALUE; 2778 } 2779 AutoMutex lock(mLock); 2780 // inclusive of offloaded and direct tracks. 2781 // 2782 // It is possible, but not enabled, to allow duration computation for non-pcm 2783 // audio_has_proportional_frames() formats because currently they have 2784 // the drain rate equivalent to the pcm sample rate * framesize. 2785 if (!isPurePcmData_l()) { 2786 return INVALID_OPERATION; 2787 } 2788 ExtendedTimestamp ets; 2789 if (getTimestamp_l(&ets) == OK 2790 && ets.mTimeNs[location] > 0) { 2791 int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT] 2792 - ets.mPosition[location]; 2793 if (diff < 0) { 2794 *msec = 0; 2795 } else { 2796 // ms is the playback time by frames 2797 int64_t ms = (int64_t)((double)diff * 1000 / 2798 ((double)mSampleRate * mPlaybackRate.mSpeed)); 2799 // clockdiff is the timestamp age (negative) 2800 int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 : 2801 ets.mTimeNs[location] 2802 + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC] 2803 - systemTime(SYSTEM_TIME_MONOTONIC); 2804 2805 //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff); 2806 static const int NANOS_PER_MILLIS = 1000000; 2807 *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS); 2808 } 2809 return NO_ERROR; 2810 } 2811 if (location != ExtendedTimestamp::LOCATION_SERVER) { 2812 return INVALID_OPERATION; // LOCATION_KERNEL is not available 2813 } 2814 // use server position directly (offloaded and direct arrive here) 2815 updateAndGetPosition_l(); 2816 int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue(); 2817 *msec = (diff <= 0) ? 0 2818 : (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed)); 2819 return NO_ERROR; 2820} 2821 2822bool AudioTrack::hasStarted() 2823{ 2824 AutoMutex lock(mLock); 2825 switch (mState) { 2826 case STATE_STOPPED: 2827 if (isOffloadedOrDirect_l()) { 2828 // check if we have started in the past to return true. 2829 return mStartUs > 0; 2830 } 2831 // A normal audio track may still be draining, so 2832 // check if stream has ended. This covers fasttrack position 2833 // instability and start/stop without any data written. 2834 if (mProxy->getStreamEndDone()) { 2835 return true; 2836 } 2837 // fall through 2838 case STATE_ACTIVE: 2839 case STATE_STOPPING: 2840 break; 2841 case STATE_PAUSED: 2842 case STATE_PAUSED_STOPPING: 2843 case STATE_FLUSHED: 2844 return false; // we're not active 2845 default: 2846 LOG_ALWAYS_FATAL("Invalid mState in hasStarted(): %d", mState); 2847 break; 2848 } 2849 2850 // wait indicates whether we need to wait for a timestamp. 2851 // This is conservatively figured - if we encounter an unexpected error 2852 // then we will not wait. 2853 bool wait = false; 2854 if (isOffloadedOrDirect_l()) { 2855 AudioTimestamp ts; 2856 status_t status = getTimestamp_l(ts); 2857 if (status == WOULD_BLOCK) { 2858 wait = true; 2859 } else if (status == OK) { 2860 wait = (ts.mPosition == 0 || ts.mPosition == mStartTs.mPosition); 2861 } 2862 ALOGV("hasStarted wait:%d ts:%u start position:%lld", 2863 (int)wait, 2864 ts.mPosition, 2865 (long long)mStartTs.mPosition); 2866 } else { 2867 int location = ExtendedTimestamp::LOCATION_SERVER; // for ALOG 2868 ExtendedTimestamp ets; 2869 status_t status = getTimestamp_l(&ets); 2870 if (status == WOULD_BLOCK) { // no SERVER or KERNEL frame info in ets 2871 wait = true; 2872 } else if (status == OK) { 2873 for (location = ExtendedTimestamp::LOCATION_KERNEL; 2874 location >= ExtendedTimestamp::LOCATION_SERVER; --location) { 2875 if (ets.mTimeNs[location] < 0 || mStartEts.mTimeNs[location] < 0) { 2876 continue; 2877 } 2878 wait = ets.mPosition[location] == 0 2879 || ets.mPosition[location] == mStartEts.mPosition[location]; 2880 break; 2881 } 2882 } 2883 ALOGV("hasStarted wait:%d ets:%lld start position:%lld", 2884 (int)wait, 2885 (long long)ets.mPosition[location], 2886 (long long)mStartEts.mPosition[location]); 2887 } 2888 return !wait; 2889} 2890 2891// ========================================================================= 2892 2893void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 2894{ 2895 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 2896 if (audioTrack != 0) { 2897 AutoMutex lock(audioTrack->mLock); 2898 audioTrack->mProxy->binderDied(); 2899 } 2900} 2901 2902// ========================================================================= 2903 2904AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 2905 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 2906 mIgnoreNextPausedInt(false) 2907{ 2908} 2909 2910AudioTrack::AudioTrackThread::~AudioTrackThread() 2911{ 2912} 2913 2914bool AudioTrack::AudioTrackThread::threadLoop() 2915{ 2916 { 2917 AutoMutex _l(mMyLock); 2918 if (mPaused) { 2919 mMyCond.wait(mMyLock); 2920 // caller will check for exitPending() 2921 return true; 2922 } 2923 if (mIgnoreNextPausedInt) { 2924 mIgnoreNextPausedInt = false; 2925 mPausedInt = false; 2926 } 2927 if (mPausedInt) { 2928 if (mPausedNs > 0) { 2929 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 2930 } else { 2931 mMyCond.wait(mMyLock); 2932 } 2933 mPausedInt = false; 2934 return true; 2935 } 2936 } 2937 if (exitPending()) { 2938 return false; 2939 } 2940 nsecs_t ns = mReceiver.processAudioBuffer(); 2941 switch (ns) { 2942 case 0: 2943 return true; 2944 case NS_INACTIVE: 2945 pauseInternal(); 2946 return true; 2947 case NS_NEVER: 2948 return false; 2949 case NS_WHENEVER: 2950 // Event driven: call wake() when callback notifications conditions change. 2951 ns = INT64_MAX; 2952 // fall through 2953 default: 2954 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns); 2955 pauseInternal(ns); 2956 return true; 2957 } 2958} 2959 2960void AudioTrack::AudioTrackThread::requestExit() 2961{ 2962 // must be in this order to avoid a race condition 2963 Thread::requestExit(); 2964 resume(); 2965} 2966 2967void AudioTrack::AudioTrackThread::pause() 2968{ 2969 AutoMutex _l(mMyLock); 2970 mPaused = true; 2971} 2972 2973void AudioTrack::AudioTrackThread::resume() 2974{ 2975 AutoMutex _l(mMyLock); 2976 mIgnoreNextPausedInt = true; 2977 if (mPaused || mPausedInt) { 2978 mPaused = false; 2979 mPausedInt = false; 2980 mMyCond.signal(); 2981 } 2982} 2983 2984void AudioTrack::AudioTrackThread::wake() 2985{ 2986 AutoMutex _l(mMyLock); 2987 if (!mPaused) { 2988 // wake() might be called while servicing a callback - ignore the next 2989 // pause time and call processAudioBuffer. 2990 mIgnoreNextPausedInt = true; 2991 if (mPausedInt && mPausedNs > 0) { 2992 // audio track is active and internally paused with timeout. 2993 mPausedInt = false; 2994 mMyCond.signal(); 2995 } 2996 } 2997} 2998 2999void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 3000{ 3001 AutoMutex _l(mMyLock); 3002 mPausedInt = true; 3003 mPausedNs = ns; 3004} 3005 3006} // namespace android 3007