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 36 static const int kMaxLoopCountNotifications = 32; 37 38 namespace android { 39 // --------------------------------------------------------------------------- 40 41 // TODO: Move to a separate .h 42 43 template <typename T> 44 static inline const T &min(const T &x, const T &y) { 45 return x < y ? x : y; 46 } 47 48 template <typename T> 49 static inline const T &max(const T &x, const T &y) { 50 return x > y ? x : y; 51 } 52 53 static inline nsecs_t framesToNanoseconds(ssize_t frames, uint32_t sampleRate, float speed) 54 { 55 return ((double)frames * 1000000000) / ((double)sampleRate * speed); 56 } 57 58 static int64_t convertTimespecToUs(const struct timespec &tv) 59 { 60 return tv.tv_sec * 1000000ll + tv.tv_nsec / 1000; 61 } 62 63 // current monotonic time in microseconds. 64 static int64_t getNowUs() 65 { 66 struct timespec tv; 67 (void) clock_gettime(CLOCK_MONOTONIC, &tv); 68 return convertTimespecToUs(tv); 69 } 70 71 // FIXME: we don't use the pitch setting in the time stretcher (not working); 72 // instead we emulate it using our sample rate converter. 73 static const bool kFixPitch = true; // enable pitch fix 74 static inline uint32_t adjustSampleRate(uint32_t sampleRate, float pitch) 75 { 76 return kFixPitch ? (sampleRate * pitch + 0.5) : sampleRate; 77 } 78 79 static inline float adjustSpeed(float speed, float pitch) 80 { 81 return kFixPitch ? speed / max(pitch, AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) : speed; 82 } 83 84 static inline float adjustPitch(float pitch) 85 { 86 return kFixPitch ? AUDIO_TIMESTRETCH_PITCH_NORMAL : pitch; 87 } 88 89 // Must match similar computation in createTrack_l in Threads.cpp. 90 // TODO: Move to a common library 91 static size_t calculateMinFrameCount( 92 uint32_t afLatencyMs, uint32_t afFrameCount, uint32_t afSampleRate, 93 uint32_t sampleRate, float speed /*, uint32_t notificationsPerBufferReq*/) 94 { 95 // Ensure that buffer depth covers at least audio hardware latency 96 uint32_t minBufCount = afLatencyMs / ((1000 * afFrameCount) / afSampleRate); 97 if (minBufCount < 2) { 98 minBufCount = 2; 99 } 100 #if 0 101 // The notificationsPerBufferReq parameter is not yet used for non-fast tracks, 102 // but keeping the code here to make it easier to add later. 103 if (minBufCount < notificationsPerBufferReq) { 104 minBufCount = notificationsPerBufferReq; 105 } 106 #endif 107 ALOGV("calculateMinFrameCount afLatency %u afFrameCount %u afSampleRate %u " 108 "sampleRate %u speed %f minBufCount: %u" /*" notificationsPerBufferReq %u"*/, 109 afLatencyMs, afFrameCount, afSampleRate, sampleRate, speed, minBufCount 110 /*, notificationsPerBufferReq*/); 111 return minBufCount * sourceFramesNeededWithTimestretch( 112 sampleRate, afFrameCount, afSampleRate, speed); 113 } 114 115 // static 116 status_t AudioTrack::getMinFrameCount( 117 size_t* frameCount, 118 audio_stream_type_t streamType, 119 uint32_t sampleRate) 120 { 121 if (frameCount == NULL) { 122 return BAD_VALUE; 123 } 124 125 // FIXME handle in server, like createTrack_l(), possible missing info: 126 // audio_io_handle_t output 127 // audio_format_t format 128 // audio_channel_mask_t channelMask 129 // audio_output_flags_t flags (FAST) 130 uint32_t afSampleRate; 131 status_t status; 132 status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); 133 if (status != NO_ERROR) { 134 ALOGE("Unable to query output sample rate for stream type %d; status %d", 135 streamType, status); 136 return status; 137 } 138 size_t afFrameCount; 139 status = AudioSystem::getOutputFrameCount(&afFrameCount, streamType); 140 if (status != NO_ERROR) { 141 ALOGE("Unable to query output frame count for stream type %d; status %d", 142 streamType, status); 143 return status; 144 } 145 uint32_t afLatency; 146 status = AudioSystem::getOutputLatency(&afLatency, streamType); 147 if (status != NO_ERROR) { 148 ALOGE("Unable to query output latency for stream type %d; status %d", 149 streamType, status); 150 return status; 151 } 152 153 // When called from createTrack, speed is 1.0f (normal speed). 154 // This is rechecked again on setting playback rate (TODO: on setting sample rate, too). 155 *frameCount = calculateMinFrameCount(afLatency, afFrameCount, afSampleRate, sampleRate, 1.0f 156 /*, 0 notificationsPerBufferReq*/); 157 158 // The formula above should always produce a non-zero value under normal circumstances: 159 // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX. 160 // Return error in the unlikely event that it does not, as that's part of the API contract. 161 if (*frameCount == 0) { 162 ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %u", 163 streamType, sampleRate); 164 return BAD_VALUE; 165 } 166 ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, afSampleRate=%u, afLatency=%u", 167 *frameCount, afFrameCount, afSampleRate, afLatency); 168 return NO_ERROR; 169 } 170 171 // --------------------------------------------------------------------------- 172 173 AudioTrack::AudioTrack() 174 : mStatus(NO_INIT), 175 mState(STATE_STOPPED), 176 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 177 mPreviousSchedulingGroup(SP_DEFAULT), 178 mPausedPosition(0), 179 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE) 180 { 181 mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN; 182 mAttributes.usage = AUDIO_USAGE_UNKNOWN; 183 mAttributes.flags = 0x0; 184 strcpy(mAttributes.tags, ""); 185 } 186 187 AudioTrack::AudioTrack( 188 audio_stream_type_t streamType, 189 uint32_t sampleRate, 190 audio_format_t format, 191 audio_channel_mask_t channelMask, 192 size_t frameCount, 193 audio_output_flags_t flags, 194 callback_t cbf, 195 void* user, 196 int32_t notificationFrames, 197 audio_session_t sessionId, 198 transfer_type transferType, 199 const audio_offload_info_t *offloadInfo, 200 int uid, 201 pid_t pid, 202 const audio_attributes_t* pAttributes, 203 bool doNotReconnect, 204 float maxRequiredSpeed) 205 : mStatus(NO_INIT), 206 mState(STATE_STOPPED), 207 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 208 mPreviousSchedulingGroup(SP_DEFAULT), 209 mPausedPosition(0), 210 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE) 211 { 212 mStatus = set(streamType, sampleRate, format, channelMask, 213 frameCount, flags, cbf, user, notificationFrames, 214 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, 215 offloadInfo, uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed); 216 } 217 218 AudioTrack::AudioTrack( 219 audio_stream_type_t streamType, 220 uint32_t sampleRate, 221 audio_format_t format, 222 audio_channel_mask_t channelMask, 223 const sp<IMemory>& sharedBuffer, 224 audio_output_flags_t flags, 225 callback_t cbf, 226 void* user, 227 int32_t notificationFrames, 228 audio_session_t sessionId, 229 transfer_type transferType, 230 const audio_offload_info_t *offloadInfo, 231 int uid, 232 pid_t pid, 233 const audio_attributes_t* pAttributes, 234 bool doNotReconnect, 235 float maxRequiredSpeed) 236 : mStatus(NO_INIT), 237 mState(STATE_STOPPED), 238 mPreviousPriority(ANDROID_PRIORITY_NORMAL), 239 mPreviousSchedulingGroup(SP_DEFAULT), 240 mPausedPosition(0), 241 mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE) 242 { 243 mStatus = set(streamType, sampleRate, format, channelMask, 244 0 /*frameCount*/, flags, cbf, user, notificationFrames, 245 sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo, 246 uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed); 247 } 248 249 AudioTrack::~AudioTrack() 250 { 251 if (mStatus == NO_ERROR) { 252 // Make sure that callback function exits in the case where 253 // it is looping on buffer full condition in obtainBuffer(). 254 // Otherwise the callback thread will never exit. 255 stop(); 256 if (mAudioTrackThread != 0) { 257 mProxy->interrupt(); 258 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 259 mAudioTrackThread->requestExitAndWait(); 260 mAudioTrackThread.clear(); 261 } 262 // No lock here: worst case we remove a NULL callback which will be a nop 263 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) { 264 AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); 265 } 266 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this); 267 mAudioTrack.clear(); 268 mCblkMemory.clear(); 269 mSharedBuffer.clear(); 270 IPCThreadState::self()->flushCommands(); 271 ALOGV("~AudioTrack, releasing session id %d from %d on behalf of %d", 272 mSessionId, IPCThreadState::self()->getCallingPid(), mClientPid); 273 AudioSystem::releaseAudioSessionId(mSessionId, mClientPid); 274 } 275 } 276 277 status_t AudioTrack::set( 278 audio_stream_type_t streamType, 279 uint32_t sampleRate, 280 audio_format_t format, 281 audio_channel_mask_t channelMask, 282 size_t frameCount, 283 audio_output_flags_t flags, 284 callback_t cbf, 285 void* user, 286 int32_t notificationFrames, 287 const sp<IMemory>& sharedBuffer, 288 bool threadCanCallJava, 289 audio_session_t sessionId, 290 transfer_type transferType, 291 const audio_offload_info_t *offloadInfo, 292 int uid, 293 pid_t pid, 294 const audio_attributes_t* pAttributes, 295 bool doNotReconnect, 296 float maxRequiredSpeed) 297 { 298 ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, " 299 "flags #%x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d", 300 streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames, 301 sessionId, transferType, uid, pid); 302 303 mThreadCanCallJava = threadCanCallJava; 304 305 switch (transferType) { 306 case TRANSFER_DEFAULT: 307 if (sharedBuffer != 0) { 308 transferType = TRANSFER_SHARED; 309 } else if (cbf == NULL || threadCanCallJava) { 310 transferType = TRANSFER_SYNC; 311 } else { 312 transferType = TRANSFER_CALLBACK; 313 } 314 break; 315 case TRANSFER_CALLBACK: 316 if (cbf == NULL || sharedBuffer != 0) { 317 ALOGE("Transfer type TRANSFER_CALLBACK but cbf == NULL || sharedBuffer != 0"); 318 return BAD_VALUE; 319 } 320 break; 321 case TRANSFER_OBTAIN: 322 case TRANSFER_SYNC: 323 if (sharedBuffer != 0) { 324 ALOGE("Transfer type TRANSFER_OBTAIN but sharedBuffer != 0"); 325 return BAD_VALUE; 326 } 327 break; 328 case TRANSFER_SHARED: 329 if (sharedBuffer == 0) { 330 ALOGE("Transfer type TRANSFER_SHARED but sharedBuffer == 0"); 331 return BAD_VALUE; 332 } 333 break; 334 default: 335 ALOGE("Invalid transfer type %d", transferType); 336 return BAD_VALUE; 337 } 338 mSharedBuffer = sharedBuffer; 339 mTransfer = transferType; 340 mDoNotReconnect = doNotReconnect; 341 342 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %zu", sharedBuffer->pointer(), 343 sharedBuffer->size()); 344 345 ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags); 346 347 // invariant that mAudioTrack != 0 is true only after set() returns successfully 348 if (mAudioTrack != 0) { 349 ALOGE("Track already in use"); 350 return INVALID_OPERATION; 351 } 352 353 // handle default values first. 354 if (streamType == AUDIO_STREAM_DEFAULT) { 355 streamType = AUDIO_STREAM_MUSIC; 356 } 357 if (pAttributes == NULL) { 358 if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) { 359 ALOGE("Invalid stream type %d", streamType); 360 return BAD_VALUE; 361 } 362 mStreamType = streamType; 363 364 } else { 365 // stream type shouldn't be looked at, this track has audio attributes 366 memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t)); 367 ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]", 368 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags); 369 mStreamType = AUDIO_STREAM_DEFAULT; 370 if ((mAttributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) { 371 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC); 372 } 373 if ((mAttributes.flags & AUDIO_FLAG_LOW_LATENCY) != 0) { 374 flags = (audio_output_flags_t) (flags | AUDIO_OUTPUT_FLAG_FAST); 375 } 376 } 377 378 // these below should probably come from the audioFlinger too... 379 if (format == AUDIO_FORMAT_DEFAULT) { 380 format = AUDIO_FORMAT_PCM_16_BIT; 381 } else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through? 382 mAttributes.flags |= AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO; 383 } 384 385 // validate parameters 386 if (!audio_is_valid_format(format)) { 387 ALOGE("Invalid format %#x", format); 388 return BAD_VALUE; 389 } 390 mFormat = format; 391 392 if (!audio_is_output_channel(channelMask)) { 393 ALOGE("Invalid channel mask %#x", channelMask); 394 return BAD_VALUE; 395 } 396 mChannelMask = channelMask; 397 uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 398 mChannelCount = channelCount; 399 400 // force direct flag if format is not linear PCM 401 // or offload was requested 402 if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 403 || !audio_is_linear_pcm(format)) { 404 ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) 405 ? "Offload request, forcing to Direct Output" 406 : "Not linear PCM, forcing to Direct Output"); 407 flags = (audio_output_flags_t) 408 // FIXME why can't we allow direct AND fast? 409 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST); 410 } 411 412 // force direct flag if HW A/V sync requested 413 if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) { 414 flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT); 415 } 416 417 if (flags & AUDIO_OUTPUT_FLAG_DIRECT) { 418 if (audio_has_proportional_frames(format)) { 419 mFrameSize = channelCount * audio_bytes_per_sample(format); 420 } else { 421 mFrameSize = sizeof(uint8_t); 422 } 423 } else { 424 ALOG_ASSERT(audio_has_proportional_frames(format)); 425 mFrameSize = channelCount * audio_bytes_per_sample(format); 426 // createTrack will return an error if PCM format is not supported by server, 427 // so no need to check for specific PCM formats here 428 } 429 430 // sampling rate must be specified for direct outputs 431 if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) { 432 return BAD_VALUE; 433 } 434 mSampleRate = sampleRate; 435 mOriginalSampleRate = sampleRate; 436 mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT; 437 // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX 438 mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX); 439 440 // Make copy of input parameter offloadInfo so that in the future: 441 // (a) createTrack_l doesn't need it as an input parameter 442 // (b) we can support re-creation of offloaded tracks 443 if (offloadInfo != NULL) { 444 mOffloadInfoCopy = *offloadInfo; 445 mOffloadInfo = &mOffloadInfoCopy; 446 } else { 447 mOffloadInfo = NULL; 448 } 449 450 mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f; 451 mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f; 452 mSendLevel = 0.0f; 453 // mFrameCount is initialized in createTrack_l 454 mReqFrameCount = frameCount; 455 if (notificationFrames >= 0) { 456 mNotificationFramesReq = notificationFrames; 457 mNotificationsPerBufferReq = 0; 458 } else { 459 if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) { 460 ALOGE("notificationFrames=%d not permitted for non-fast track", 461 notificationFrames); 462 return BAD_VALUE; 463 } 464 if (frameCount > 0) { 465 ALOGE("notificationFrames=%d not permitted with non-zero frameCount=%zu", 466 notificationFrames, frameCount); 467 return BAD_VALUE; 468 } 469 mNotificationFramesReq = 0; 470 const uint32_t minNotificationsPerBuffer = 1; 471 const uint32_t maxNotificationsPerBuffer = 8; 472 mNotificationsPerBufferReq = min(maxNotificationsPerBuffer, 473 max((uint32_t) -notificationFrames, minNotificationsPerBuffer)); 474 ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames, 475 "notificationFrames=%d clamped to the range -%u to -%u", 476 notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer); 477 } 478 mNotificationFramesAct = 0; 479 if (sessionId == AUDIO_SESSION_ALLOCATE) { 480 mSessionId = (audio_session_t) AudioSystem::newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 481 } else { 482 mSessionId = sessionId; 483 } 484 int callingpid = IPCThreadState::self()->getCallingPid(); 485 int mypid = getpid(); 486 if (uid == -1 || (callingpid != mypid)) { 487 mClientUid = IPCThreadState::self()->getCallingUid(); 488 } else { 489 mClientUid = uid; 490 } 491 if (pid == -1 || (callingpid != mypid)) { 492 mClientPid = callingpid; 493 } else { 494 mClientPid = pid; 495 } 496 mAuxEffectId = 0; 497 mOrigFlags = mFlags = flags; 498 mCbf = cbf; 499 500 if (cbf != NULL) { 501 mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava); 502 mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/); 503 // thread begins in paused state, and will not reference us until start() 504 } 505 506 // create the IAudioTrack 507 status_t status = createTrack_l(); 508 509 if (status != NO_ERROR) { 510 if (mAudioTrackThread != 0) { 511 mAudioTrackThread->requestExit(); // see comment in AudioTrack.h 512 mAudioTrackThread->requestExitAndWait(); 513 mAudioTrackThread.clear(); 514 } 515 return status; 516 } 517 518 mStatus = NO_ERROR; 519 mUserData = user; 520 mLoopCount = 0; 521 mLoopStart = 0; 522 mLoopEnd = 0; 523 mLoopCountNotified = 0; 524 mMarkerPosition = 0; 525 mMarkerReached = false; 526 mNewPosition = 0; 527 mUpdatePeriod = 0; 528 mPosition = 0; 529 mReleased = 0; 530 mStartUs = 0; 531 AudioSystem::acquireAudioSessionId(mSessionId, mClientPid); 532 mSequence = 1; 533 mObservedSequence = mSequence; 534 mInUnderrun = false; 535 mPreviousTimestampValid = false; 536 mTimestampStartupGlitchReported = false; 537 mRetrogradeMotionReported = false; 538 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID; 539 mUnderrunCountOffset = 0; 540 mFramesWritten = 0; 541 mFramesWrittenServerOffset = 0; 542 543 return NO_ERROR; 544 } 545 546 // ------------------------------------------------------------------------- 547 548 status_t AudioTrack::start() 549 { 550 AutoMutex lock(mLock); 551 552 if (mState == STATE_ACTIVE) { 553 return INVALID_OPERATION; 554 } 555 556 mInUnderrun = true; 557 558 State previousState = mState; 559 if (previousState == STATE_PAUSED_STOPPING) { 560 mState = STATE_STOPPING; 561 } else { 562 mState = STATE_ACTIVE; 563 } 564 (void) updateAndGetPosition_l(); 565 if (previousState == STATE_STOPPED || previousState == STATE_FLUSHED) { 566 // reset current position as seen by client to 0 567 mPosition = 0; 568 mPreviousTimestampValid = false; 569 mTimestampStartupGlitchReported = false; 570 mRetrogradeMotionReported = false; 571 mPreviousLocation = ExtendedTimestamp::LOCATION_INVALID; 572 573 // read last server side position change via timestamp. 574 ExtendedTimestamp ets; 575 if (mProxy->getTimestamp(&ets) == OK && 576 ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) { 577 // Server side has consumed something, but is it finished consuming? 578 // It is possible since flush and stop are asynchronous that the server 579 // is still active at this point. 580 ALOGV("start: server read:%lld cumulative flushed:%lld client written:%lld", 581 (long long)(mFramesWrittenServerOffset 582 + ets.mPosition[ExtendedTimestamp::LOCATION_SERVER]), 583 (long long)ets.mFlushed, 584 (long long)mFramesWritten); 585 mFramesWrittenServerOffset = -ets.mPosition[ExtendedTimestamp::LOCATION_SERVER]; 586 } 587 mFramesWritten = 0; 588 mProxy->clearTimestamp(); // need new server push for valid timestamp 589 mMarkerReached = false; 590 591 // For offloaded tracks, we don't know if the hardware counters are really zero here, 592 // since the flush is asynchronous and stop may not fully drain. 593 // We save the time when the track is started to later verify whether 594 // the counters are realistic (i.e. start from zero after this time). 595 mStartUs = getNowUs(); 596 597 // force refresh of remaining frames by processAudioBuffer() as last 598 // write before stop could be partial. 599 mRefreshRemaining = true; 600 } 601 mNewPosition = mPosition + mUpdatePeriod; 602 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 603 604 status_t status = NO_ERROR; 605 if (!(flags & CBLK_INVALID)) { 606 status = mAudioTrack->start(); 607 if (status == DEAD_OBJECT) { 608 flags |= CBLK_INVALID; 609 } 610 } 611 if (flags & CBLK_INVALID) { 612 status = restoreTrack_l("start"); 613 } 614 615 // resume or pause the callback thread as needed. 616 sp<AudioTrackThread> t = mAudioTrackThread; 617 if (status == NO_ERROR) { 618 if (t != 0) { 619 if (previousState == STATE_STOPPING) { 620 mProxy->interrupt(); 621 } else { 622 t->resume(); 623 } 624 } else { 625 mPreviousPriority = getpriority(PRIO_PROCESS, 0); 626 get_sched_policy(0, &mPreviousSchedulingGroup); 627 androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO); 628 } 629 } else { 630 ALOGE("start() status %d", status); 631 mState = previousState; 632 if (t != 0) { 633 if (previousState != STATE_STOPPING) { 634 t->pause(); 635 } 636 } else { 637 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 638 set_sched_policy(0, mPreviousSchedulingGroup); 639 } 640 } 641 642 return status; 643 } 644 645 void AudioTrack::stop() 646 { 647 AutoMutex lock(mLock); 648 if (mState != STATE_ACTIVE && mState != STATE_PAUSED) { 649 return; 650 } 651 652 if (isOffloaded_l()) { 653 mState = STATE_STOPPING; 654 } else { 655 mState = STATE_STOPPED; 656 mReleased = 0; 657 } 658 659 mProxy->interrupt(); 660 mAudioTrack->stop(); 661 662 // Note: legacy handling - stop does not clear playback marker 663 // and periodic update counter, but flush does for streaming tracks. 664 665 if (mSharedBuffer != 0) { 666 // clear buffer position and loop count. 667 mStaticProxy->setBufferPositionAndLoop(0 /* position */, 668 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */); 669 } 670 671 sp<AudioTrackThread> t = mAudioTrackThread; 672 if (t != 0) { 673 if (!isOffloaded_l()) { 674 t->pause(); 675 } 676 } else { 677 setpriority(PRIO_PROCESS, 0, mPreviousPriority); 678 set_sched_policy(0, mPreviousSchedulingGroup); 679 } 680 } 681 682 bool AudioTrack::stopped() const 683 { 684 AutoMutex lock(mLock); 685 return mState != STATE_ACTIVE; 686 } 687 688 void AudioTrack::flush() 689 { 690 if (mSharedBuffer != 0) { 691 return; 692 } 693 AutoMutex lock(mLock); 694 if (mState == STATE_ACTIVE || mState == STATE_FLUSHED) { 695 return; 696 } 697 flush_l(); 698 } 699 700 void AudioTrack::flush_l() 701 { 702 ALOG_ASSERT(mState != STATE_ACTIVE); 703 704 // clear playback marker and periodic update counter 705 mMarkerPosition = 0; 706 mMarkerReached = false; 707 mUpdatePeriod = 0; 708 mRefreshRemaining = true; 709 710 mState = STATE_FLUSHED; 711 mReleased = 0; 712 if (isOffloaded_l()) { 713 mProxy->interrupt(); 714 } 715 mProxy->flush(); 716 mAudioTrack->flush(); 717 } 718 719 void AudioTrack::pause() 720 { 721 AutoMutex lock(mLock); 722 if (mState == STATE_ACTIVE) { 723 mState = STATE_PAUSED; 724 } else if (mState == STATE_STOPPING) { 725 mState = STATE_PAUSED_STOPPING; 726 } else { 727 return; 728 } 729 mProxy->interrupt(); 730 mAudioTrack->pause(); 731 732 if (isOffloaded_l()) { 733 if (mOutput != AUDIO_IO_HANDLE_NONE) { 734 // An offload output can be re-used between two audio tracks having 735 // the same configuration. A timestamp query for a paused track 736 // while the other is running would return an incorrect time. 737 // To fix this, cache the playback position on a pause() and return 738 // this time when requested until the track is resumed. 739 740 // OffloadThread sends HAL pause in its threadLoop. Time saved 741 // here can be slightly off. 742 743 // TODO: check return code for getRenderPosition. 744 745 uint32_t halFrames; 746 AudioSystem::getRenderPosition(mOutput, &halFrames, &mPausedPosition); 747 ALOGV("AudioTrack::pause for offload, cache current position %u", mPausedPosition); 748 } 749 } 750 } 751 752 status_t AudioTrack::setVolume(float left, float right) 753 { 754 // This duplicates a test by AudioTrack JNI, but that is not the only caller 755 if (isnanf(left) || left < GAIN_FLOAT_ZERO || left > GAIN_FLOAT_UNITY || 756 isnanf(right) || right < GAIN_FLOAT_ZERO || right > GAIN_FLOAT_UNITY) { 757 return BAD_VALUE; 758 } 759 760 AutoMutex lock(mLock); 761 mVolume[AUDIO_INTERLEAVE_LEFT] = left; 762 mVolume[AUDIO_INTERLEAVE_RIGHT] = right; 763 764 mProxy->setVolumeLR(gain_minifloat_pack(gain_from_float(left), gain_from_float(right))); 765 766 if (isOffloaded_l()) { 767 mAudioTrack->signal(); 768 } 769 return NO_ERROR; 770 } 771 772 status_t AudioTrack::setVolume(float volume) 773 { 774 return setVolume(volume, volume); 775 } 776 777 status_t AudioTrack::setAuxEffectSendLevel(float level) 778 { 779 // This duplicates a test by AudioTrack JNI, but that is not the only caller 780 if (isnanf(level) || level < GAIN_FLOAT_ZERO || level > GAIN_FLOAT_UNITY) { 781 return BAD_VALUE; 782 } 783 784 AutoMutex lock(mLock); 785 mSendLevel = level; 786 mProxy->setSendLevel(level); 787 788 return NO_ERROR; 789 } 790 791 void AudioTrack::getAuxEffectSendLevel(float* level) const 792 { 793 if (level != NULL) { 794 *level = mSendLevel; 795 } 796 } 797 798 status_t AudioTrack::setSampleRate(uint32_t rate) 799 { 800 AutoMutex lock(mLock); 801 if (rate == mSampleRate) { 802 return NO_ERROR; 803 } 804 if (isOffloadedOrDirect_l() || (mFlags & AUDIO_OUTPUT_FLAG_FAST)) { 805 return INVALID_OPERATION; 806 } 807 if (mOutput == AUDIO_IO_HANDLE_NONE) { 808 return NO_INIT; 809 } 810 // NOTE: it is theoretically possible, but highly unlikely, that a device change 811 // could mean a previously allowed sampling rate is no longer allowed. 812 uint32_t afSamplingRate; 813 if (AudioSystem::getSamplingRate(mOutput, &afSamplingRate) != NO_ERROR) { 814 return NO_INIT; 815 } 816 // pitch is emulated by adjusting speed and sampleRate 817 const uint32_t effectiveSampleRate = adjustSampleRate(rate, mPlaybackRate.mPitch); 818 if (rate == 0 || effectiveSampleRate > afSamplingRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 819 return BAD_VALUE; 820 } 821 // TODO: Should we also check if the buffer size is compatible? 822 823 mSampleRate = rate; 824 mProxy->setSampleRate(effectiveSampleRate); 825 826 return NO_ERROR; 827 } 828 829 uint32_t AudioTrack::getSampleRate() const 830 { 831 AutoMutex lock(mLock); 832 833 // sample rate can be updated during playback by the offloaded decoder so we need to 834 // query the HAL and update if needed. 835 // FIXME use Proxy return channel to update the rate from server and avoid polling here 836 if (isOffloadedOrDirect_l()) { 837 if (mOutput != AUDIO_IO_HANDLE_NONE) { 838 uint32_t sampleRate = 0; 839 status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate); 840 if (status == NO_ERROR) { 841 mSampleRate = sampleRate; 842 } 843 } 844 } 845 return mSampleRate; 846 } 847 848 uint32_t AudioTrack::getOriginalSampleRate() const 849 { 850 return mOriginalSampleRate; 851 } 852 853 status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate) 854 { 855 AutoMutex lock(mLock); 856 if (isAudioPlaybackRateEqual(playbackRate, mPlaybackRate)) { 857 return NO_ERROR; 858 } 859 if (isOffloadedOrDirect_l()) { 860 return INVALID_OPERATION; 861 } 862 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 863 return INVALID_OPERATION; 864 } 865 866 ALOGV("setPlaybackRate (input): mSampleRate:%u mSpeed:%f mPitch:%f", 867 mSampleRate, playbackRate.mSpeed, playbackRate.mPitch); 868 // pitch is emulated by adjusting speed and sampleRate 869 const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch); 870 const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch); 871 const float effectivePitch = adjustPitch(playbackRate.mPitch); 872 AudioPlaybackRate playbackRateTemp = playbackRate; 873 playbackRateTemp.mSpeed = effectiveSpeed; 874 playbackRateTemp.mPitch = effectivePitch; 875 876 ALOGV("setPlaybackRate (effective): mSampleRate:%u mSpeed:%f mPitch:%f", 877 effectiveRate, effectiveSpeed, effectivePitch); 878 879 if (!isAudioPlaybackRateValid(playbackRateTemp)) { 880 ALOGV("setPlaybackRate(%f, %f) failed (effective rate out of bounds)", 881 playbackRate.mSpeed, playbackRate.mPitch); 882 return BAD_VALUE; 883 } 884 // Check if the buffer size is compatible. 885 if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) { 886 ALOGV("setPlaybackRate(%f, %f) failed (buffer size)", 887 playbackRate.mSpeed, playbackRate.mPitch); 888 return BAD_VALUE; 889 } 890 891 // Check resampler ratios are within bounds 892 if ((uint64_t)effectiveRate > (uint64_t)mSampleRate * (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 893 ALOGV("setPlaybackRate(%f, %f) failed. Resample rate exceeds max accepted value", 894 playbackRate.mSpeed, playbackRate.mPitch); 895 return BAD_VALUE; 896 } 897 898 if ((uint64_t)effectiveRate * (uint64_t)AUDIO_RESAMPLER_UP_RATIO_MAX < (uint64_t)mSampleRate) { 899 ALOGV("setPlaybackRate(%f, %f) failed. Resample rate below min accepted value", 900 playbackRate.mSpeed, playbackRate.mPitch); 901 return BAD_VALUE; 902 } 903 mPlaybackRate = playbackRate; 904 //set effective rates 905 mProxy->setPlaybackRate(playbackRateTemp); 906 mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate 907 return NO_ERROR; 908 } 909 910 const AudioPlaybackRate& AudioTrack::getPlaybackRate() const 911 { 912 AutoMutex lock(mLock); 913 return mPlaybackRate; 914 } 915 916 ssize_t AudioTrack::getBufferSizeInFrames() 917 { 918 AutoMutex lock(mLock); 919 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) { 920 return NO_INIT; 921 } 922 return (ssize_t) mProxy->getBufferSizeInFrames(); 923 } 924 925 status_t AudioTrack::getBufferDurationInUs(int64_t *duration) 926 { 927 if (duration == nullptr) { 928 return BAD_VALUE; 929 } 930 AutoMutex lock(mLock); 931 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) { 932 return NO_INIT; 933 } 934 ssize_t bufferSizeInFrames = (ssize_t) mProxy->getBufferSizeInFrames(); 935 if (bufferSizeInFrames < 0) { 936 return (status_t)bufferSizeInFrames; 937 } 938 *duration = (int64_t)((double)bufferSizeInFrames * 1000000 939 / ((double)mSampleRate * mPlaybackRate.mSpeed)); 940 return NO_ERROR; 941 } 942 943 ssize_t AudioTrack::setBufferSizeInFrames(size_t bufferSizeInFrames) 944 { 945 AutoMutex lock(mLock); 946 if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) { 947 return NO_INIT; 948 } 949 // Reject if timed track or compressed audio. 950 if (!audio_is_linear_pcm(mFormat)) { 951 return INVALID_OPERATION; 952 } 953 return (ssize_t) mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames); 954 } 955 956 status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount) 957 { 958 if (mSharedBuffer == 0 || isOffloadedOrDirect()) { 959 return INVALID_OPERATION; 960 } 961 962 if (loopCount == 0) { 963 ; 964 } else if (loopCount >= -1 && loopStart < loopEnd && loopEnd <= mFrameCount && 965 loopEnd - loopStart >= MIN_LOOP) { 966 ; 967 } else { 968 return BAD_VALUE; 969 } 970 971 AutoMutex lock(mLock); 972 // See setPosition() regarding setting parameters such as loop points or position while active 973 if (mState == STATE_ACTIVE) { 974 return INVALID_OPERATION; 975 } 976 setLoop_l(loopStart, loopEnd, loopCount); 977 return NO_ERROR; 978 } 979 980 void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount) 981 { 982 // We do not update the periodic notification point. 983 // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 984 mLoopCount = loopCount; 985 mLoopEnd = loopEnd; 986 mLoopStart = loopStart; 987 mLoopCountNotified = loopCount; 988 mStaticProxy->setLoop(loopStart, loopEnd, loopCount); 989 990 // Waking the AudioTrackThread is not needed as this cannot be called when active. 991 } 992 993 status_t AudioTrack::setMarkerPosition(uint32_t marker) 994 { 995 // The only purpose of setting marker position is to get a callback 996 if (mCbf == NULL || isOffloadedOrDirect()) { 997 return INVALID_OPERATION; 998 } 999 1000 AutoMutex lock(mLock); 1001 mMarkerPosition = marker; 1002 mMarkerReached = false; 1003 1004 sp<AudioTrackThread> t = mAudioTrackThread; 1005 if (t != 0) { 1006 t->wake(); 1007 } 1008 return NO_ERROR; 1009 } 1010 1011 status_t AudioTrack::getMarkerPosition(uint32_t *marker) const 1012 { 1013 if (isOffloadedOrDirect()) { 1014 return INVALID_OPERATION; 1015 } 1016 if (marker == NULL) { 1017 return BAD_VALUE; 1018 } 1019 1020 AutoMutex lock(mLock); 1021 mMarkerPosition.getValue(marker); 1022 1023 return NO_ERROR; 1024 } 1025 1026 status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod) 1027 { 1028 // The only purpose of setting position update period is to get a callback 1029 if (mCbf == NULL || isOffloadedOrDirect()) { 1030 return INVALID_OPERATION; 1031 } 1032 1033 AutoMutex lock(mLock); 1034 mNewPosition = updateAndGetPosition_l() + updatePeriod; 1035 mUpdatePeriod = updatePeriod; 1036 1037 sp<AudioTrackThread> t = mAudioTrackThread; 1038 if (t != 0) { 1039 t->wake(); 1040 } 1041 return NO_ERROR; 1042 } 1043 1044 status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const 1045 { 1046 if (isOffloadedOrDirect()) { 1047 return INVALID_OPERATION; 1048 } 1049 if (updatePeriod == NULL) { 1050 return BAD_VALUE; 1051 } 1052 1053 AutoMutex lock(mLock); 1054 *updatePeriod = mUpdatePeriod; 1055 1056 return NO_ERROR; 1057 } 1058 1059 status_t AudioTrack::setPosition(uint32_t position) 1060 { 1061 if (mSharedBuffer == 0 || isOffloadedOrDirect()) { 1062 return INVALID_OPERATION; 1063 } 1064 if (position > mFrameCount) { 1065 return BAD_VALUE; 1066 } 1067 1068 AutoMutex lock(mLock); 1069 // Currently we require that the player is inactive before setting parameters such as position 1070 // or loop points. Otherwise, there could be a race condition: the application could read the 1071 // current position, compute a new position or loop parameters, and then set that position or 1072 // loop parameters but it would do the "wrong" thing since the position has continued to advance 1073 // in the mean time. If we ever provide a sequencer in server, we could allow a way for the app 1074 // to specify how it wants to handle such scenarios. 1075 if (mState == STATE_ACTIVE) { 1076 return INVALID_OPERATION; 1077 } 1078 // After setting the position, use full update period before notification. 1079 mNewPosition = updateAndGetPosition_l() + mUpdatePeriod; 1080 mStaticProxy->setBufferPosition(position); 1081 1082 // Waking the AudioTrackThread is not needed as this cannot be called when active. 1083 return NO_ERROR; 1084 } 1085 1086 status_t AudioTrack::getPosition(uint32_t *position) 1087 { 1088 if (position == NULL) { 1089 return BAD_VALUE; 1090 } 1091 1092 AutoMutex lock(mLock); 1093 // FIXME: offloaded and direct tracks call into the HAL for render positions 1094 // for compressed/synced data; however, we use proxy position for pure linear pcm data 1095 // as we do not know the capability of the HAL for pcm position support and standby. 1096 // There may be some latency differences between the HAL position and the proxy position. 1097 if (isOffloadedOrDirect_l() && !isPurePcmData_l()) { 1098 uint32_t dspFrames = 0; 1099 1100 if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) { 1101 ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition); 1102 *position = mPausedPosition; 1103 return NO_ERROR; 1104 } 1105 1106 if (mOutput != AUDIO_IO_HANDLE_NONE) { 1107 uint32_t halFrames; // actually unused 1108 (void) AudioSystem::getRenderPosition(mOutput, &halFrames, &dspFrames); 1109 // FIXME: on getRenderPosition() error, we return OK with frame position 0. 1110 } 1111 // FIXME: dspFrames may not be zero in (mState == STATE_STOPPED || mState == STATE_FLUSHED) 1112 // due to hardware latency. We leave this behavior for now. 1113 *position = dspFrames; 1114 } else { 1115 if (mCblk->mFlags & CBLK_INVALID) { 1116 (void) restoreTrack_l("getPosition"); 1117 // FIXME: for compatibility with the Java API we ignore the restoreTrack_l() 1118 // error here (e.g. DEAD_OBJECT) and return OK with the last recorded server position. 1119 } 1120 1121 // IAudioTrack::stop() isn't synchronous; we don't know when presentation completes 1122 *position = (mState == STATE_STOPPED || mState == STATE_FLUSHED) ? 1123 0 : updateAndGetPosition_l().value(); 1124 } 1125 return NO_ERROR; 1126 } 1127 1128 status_t AudioTrack::getBufferPosition(uint32_t *position) 1129 { 1130 if (mSharedBuffer == 0) { 1131 return INVALID_OPERATION; 1132 } 1133 if (position == NULL) { 1134 return BAD_VALUE; 1135 } 1136 1137 AutoMutex lock(mLock); 1138 *position = mStaticProxy->getBufferPosition(); 1139 return NO_ERROR; 1140 } 1141 1142 status_t AudioTrack::reload() 1143 { 1144 if (mSharedBuffer == 0 || isOffloadedOrDirect()) { 1145 return INVALID_OPERATION; 1146 } 1147 1148 AutoMutex lock(mLock); 1149 // See setPosition() regarding setting parameters such as loop points or position while active 1150 if (mState == STATE_ACTIVE) { 1151 return INVALID_OPERATION; 1152 } 1153 mNewPosition = mUpdatePeriod; 1154 (void) updateAndGetPosition_l(); 1155 mPosition = 0; 1156 mPreviousTimestampValid = false; 1157 #if 0 1158 // The documentation is not clear on the behavior of reload() and the restoration 1159 // of loop count. Historically we have not restored loop count, start, end, 1160 // but it makes sense if one desires to repeat playing a particular sound. 1161 if (mLoopCount != 0) { 1162 mLoopCountNotified = mLoopCount; 1163 mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount); 1164 } 1165 #endif 1166 mStaticProxy->setBufferPosition(0); 1167 return NO_ERROR; 1168 } 1169 1170 audio_io_handle_t AudioTrack::getOutput() const 1171 { 1172 AutoMutex lock(mLock); 1173 return mOutput; 1174 } 1175 1176 status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) { 1177 AutoMutex lock(mLock); 1178 if (mSelectedDeviceId != deviceId) { 1179 mSelectedDeviceId = deviceId; 1180 android_atomic_or(CBLK_INVALID, &mCblk->mFlags); 1181 } 1182 return NO_ERROR; 1183 } 1184 1185 audio_port_handle_t AudioTrack::getOutputDevice() { 1186 AutoMutex lock(mLock); 1187 return mSelectedDeviceId; 1188 } 1189 1190 audio_port_handle_t AudioTrack::getRoutedDeviceId() { 1191 AutoMutex lock(mLock); 1192 if (mOutput == AUDIO_IO_HANDLE_NONE) { 1193 return AUDIO_PORT_HANDLE_NONE; 1194 } 1195 return AudioSystem::getDeviceIdForIo(mOutput); 1196 } 1197 1198 status_t AudioTrack::attachAuxEffect(int effectId) 1199 { 1200 AutoMutex lock(mLock); 1201 status_t status = mAudioTrack->attachAuxEffect(effectId); 1202 if (status == NO_ERROR) { 1203 mAuxEffectId = effectId; 1204 } 1205 return status; 1206 } 1207 1208 audio_stream_type_t AudioTrack::streamType() const 1209 { 1210 if (mStreamType == AUDIO_STREAM_DEFAULT) { 1211 return audio_attributes_to_stream_type(&mAttributes); 1212 } 1213 return mStreamType; 1214 } 1215 1216 // ------------------------------------------------------------------------- 1217 1218 // must be called with mLock held 1219 status_t AudioTrack::createTrack_l() 1220 { 1221 const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger(); 1222 if (audioFlinger == 0) { 1223 ALOGE("Could not get audioflinger"); 1224 return NO_INIT; 1225 } 1226 1227 if (mDeviceCallback != 0 && mOutput != AUDIO_IO_HANDLE_NONE) { 1228 AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); 1229 } 1230 audio_io_handle_t output; 1231 audio_stream_type_t streamType = mStreamType; 1232 audio_attributes_t *attr = (mStreamType == AUDIO_STREAM_DEFAULT) ? &mAttributes : NULL; 1233 1234 // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted. 1235 // After fast request is denied, we will request again if IAudioTrack is re-created. 1236 1237 status_t status; 1238 status = AudioSystem::getOutputForAttr(attr, &output, 1239 mSessionId, &streamType, mClientUid, 1240 mSampleRate, mFormat, mChannelMask, 1241 mFlags, mSelectedDeviceId, mOffloadInfo); 1242 1243 if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) { 1244 ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u, format %#x," 1245 " channel mask %#x, flags %#x", 1246 mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags); 1247 return BAD_VALUE; 1248 } 1249 { 1250 // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger, 1251 // we must release it ourselves if anything goes wrong. 1252 1253 // Not all of these values are needed under all conditions, but it is easier to get them all 1254 status = AudioSystem::getLatency(output, &mAfLatency); 1255 if (status != NO_ERROR) { 1256 ALOGE("getLatency(%d) failed status %d", output, status); 1257 goto release; 1258 } 1259 ALOGV("createTrack_l() output %d afLatency %u", output, mAfLatency); 1260 1261 status = AudioSystem::getFrameCount(output, &mAfFrameCount); 1262 if (status != NO_ERROR) { 1263 ALOGE("getFrameCount(output=%d) status %d", output, status); 1264 goto release; 1265 } 1266 1267 // TODO consider making this a member variable if there are other uses for it later 1268 size_t afFrameCountHAL; 1269 status = AudioSystem::getFrameCountHAL(output, &afFrameCountHAL); 1270 if (status != NO_ERROR) { 1271 ALOGE("getFrameCountHAL(output=%d) status %d", output, status); 1272 goto release; 1273 } 1274 ALOG_ASSERT(afFrameCountHAL > 0); 1275 1276 status = AudioSystem::getSamplingRate(output, &mAfSampleRate); 1277 if (status != NO_ERROR) { 1278 ALOGE("getSamplingRate(output=%d) status %d", output, status); 1279 goto release; 1280 } 1281 if (mSampleRate == 0) { 1282 mSampleRate = mAfSampleRate; 1283 mOriginalSampleRate = mAfSampleRate; 1284 } 1285 1286 // Client can only express a preference for FAST. Server will perform additional tests. 1287 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1288 bool useCaseAllowed = 1289 // either of these use cases: 1290 // use case 1: shared buffer 1291 (mSharedBuffer != 0) || 1292 // use case 2: callback transfer mode 1293 (mTransfer == TRANSFER_CALLBACK) || 1294 // use case 3: obtain/release mode 1295 (mTransfer == TRANSFER_OBTAIN) || 1296 // use case 4: synchronous write 1297 ((mTransfer == TRANSFER_SYNC) && mThreadCanCallJava); 1298 // sample rates must also match 1299 bool fastAllowed = useCaseAllowed && (mSampleRate == mAfSampleRate); 1300 if (!fastAllowed) { 1301 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client; transfer %d, " 1302 "track %u Hz, output %u Hz", 1303 mTransfer, mSampleRate, mAfSampleRate); 1304 mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); 1305 } 1306 } 1307 1308 mNotificationFramesAct = mNotificationFramesReq; 1309 1310 size_t frameCount = mReqFrameCount; 1311 if (!audio_has_proportional_frames(mFormat)) { 1312 1313 if (mSharedBuffer != 0) { 1314 // Same comment as below about ignoring frameCount parameter for set() 1315 frameCount = mSharedBuffer->size(); 1316 } else if (frameCount == 0) { 1317 frameCount = mAfFrameCount; 1318 } 1319 if (mNotificationFramesAct != frameCount) { 1320 mNotificationFramesAct = frameCount; 1321 } 1322 } else if (mSharedBuffer != 0) { 1323 // FIXME: Ensure client side memory buffers need 1324 // not have additional alignment beyond sample 1325 // (e.g. 16 bit stereo accessed as 32 bit frame). 1326 size_t alignment = audio_bytes_per_sample(mFormat); 1327 if (alignment & 1) { 1328 // for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java). 1329 alignment = 1; 1330 } 1331 if (mChannelCount > 1) { 1332 // More than 2 channels does not require stronger alignment than stereo 1333 alignment <<= 1; 1334 } 1335 if (((uintptr_t)mSharedBuffer->pointer() & (alignment - 1)) != 0) { 1336 ALOGE("Invalid buffer alignment: address %p, channel count %u", 1337 mSharedBuffer->pointer(), mChannelCount); 1338 status = BAD_VALUE; 1339 goto release; 1340 } 1341 1342 // When initializing a shared buffer AudioTrack via constructors, 1343 // there's no frameCount parameter. 1344 // But when initializing a shared buffer AudioTrack via set(), 1345 // there _is_ a frameCount parameter. We silently ignore it. 1346 frameCount = mSharedBuffer->size() / mFrameSize; 1347 } else { 1348 size_t minFrameCount = 0; 1349 // For fast tracks the frame count calculations and checks are mostly done by server, 1350 // but we try to respect the application's request for notifications per buffer. 1351 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1352 if (mNotificationsPerBufferReq > 0) { 1353 // Avoid possible arithmetic overflow during multiplication. 1354 // mNotificationsPerBuffer is clamped to a small integer earlier, so it is unlikely. 1355 if (mNotificationsPerBufferReq > SIZE_MAX / afFrameCountHAL) { 1356 ALOGE("Requested notificationPerBuffer=%u ignored for HAL frameCount=%zu", 1357 mNotificationsPerBufferReq, afFrameCountHAL); 1358 } else { 1359 minFrameCount = afFrameCountHAL * mNotificationsPerBufferReq; 1360 } 1361 } 1362 } else { 1363 // for normal tracks precompute the frame count based on speed. 1364 const float speed = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f : 1365 max(mMaxRequiredSpeed, mPlaybackRate.mSpeed); 1366 minFrameCount = calculateMinFrameCount( 1367 mAfLatency, mAfFrameCount, mAfSampleRate, mSampleRate, 1368 speed /*, 0 mNotificationsPerBufferReq*/); 1369 } 1370 if (frameCount < minFrameCount) { 1371 frameCount = minFrameCount; 1372 } 1373 } 1374 1375 audio_output_flags_t flags = mFlags; 1376 1377 pid_t tid = -1; 1378 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1379 if (mAudioTrackThread != 0 && !mThreadCanCallJava) { 1380 tid = mAudioTrackThread->getTid(); 1381 } 1382 } 1383 1384 size_t temp = frameCount; // temp may be replaced by a revised value of frameCount, 1385 // but we will still need the original value also 1386 audio_session_t originalSessionId = mSessionId; 1387 sp<IAudioTrack> track = audioFlinger->createTrack(streamType, 1388 mSampleRate, 1389 mFormat, 1390 mChannelMask, 1391 &temp, 1392 &flags, 1393 mSharedBuffer, 1394 output, 1395 mClientPid, 1396 tid, 1397 &mSessionId, 1398 mClientUid, 1399 &status); 1400 ALOGE_IF(originalSessionId != AUDIO_SESSION_ALLOCATE && mSessionId != originalSessionId, 1401 "session ID changed from %d to %d", originalSessionId, mSessionId); 1402 1403 if (status != NO_ERROR) { 1404 ALOGE("AudioFlinger could not create track, status: %d", status); 1405 goto release; 1406 } 1407 ALOG_ASSERT(track != 0); 1408 1409 // AudioFlinger now owns the reference to the I/O handle, 1410 // so we are no longer responsible for releasing it. 1411 1412 // FIXME compare to AudioRecord 1413 sp<IMemory> iMem = track->getCblk(); 1414 if (iMem == 0) { 1415 ALOGE("Could not get control block"); 1416 return NO_INIT; 1417 } 1418 void *iMemPointer = iMem->pointer(); 1419 if (iMemPointer == NULL) { 1420 ALOGE("Could not get control block pointer"); 1421 return NO_INIT; 1422 } 1423 // invariant that mAudioTrack != 0 is true only after set() returns successfully 1424 if (mAudioTrack != 0) { 1425 IInterface::asBinder(mAudioTrack)->unlinkToDeath(mDeathNotifier, this); 1426 mDeathNotifier.clear(); 1427 } 1428 mAudioTrack = track; 1429 mCblkMemory = iMem; 1430 IPCThreadState::self()->flushCommands(); 1431 1432 audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer); 1433 mCblk = cblk; 1434 // note that temp is the (possibly revised) value of frameCount 1435 if (temp < frameCount || (frameCount == 0 && temp == 0)) { 1436 // In current design, AudioTrack client checks and ensures frame count validity before 1437 // passing it to AudioFlinger so AudioFlinger should not return a different value except 1438 // for fast track as it uses a special method of assigning frame count. 1439 ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp); 1440 } 1441 frameCount = temp; 1442 1443 mAwaitBoost = false; 1444 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1445 if (flags & AUDIO_OUTPUT_FLAG_FAST) { 1446 ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount); 1447 if (!mThreadCanCallJava) { 1448 mAwaitBoost = true; 1449 } 1450 } else { 1451 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount); 1452 } 1453 } 1454 mFlags = flags; 1455 1456 // Make sure that application is notified with sufficient margin before underrun. 1457 // The client can divide the AudioTrack buffer into sub-buffers, 1458 // and expresses its desire to server as the notification frame count. 1459 if (mSharedBuffer == 0 && audio_is_linear_pcm(mFormat)) { 1460 size_t maxNotificationFrames; 1461 if (mFlags & AUDIO_OUTPUT_FLAG_FAST) { 1462 // notify every HAL buffer, regardless of the size of the track buffer 1463 maxNotificationFrames = afFrameCountHAL; 1464 } else { 1465 // For normal tracks, use at least double-buffering if no sample rate conversion, 1466 // or at least triple-buffering if there is sample rate conversion 1467 const int nBuffering = mOriginalSampleRate == mAfSampleRate ? 2 : 3; 1468 maxNotificationFrames = frameCount / nBuffering; 1469 } 1470 if (mNotificationFramesAct == 0 || mNotificationFramesAct > maxNotificationFrames) { 1471 if (mNotificationFramesAct == 0) { 1472 ALOGD("Client defaulted notificationFrames to %zu for frameCount %zu", 1473 maxNotificationFrames, frameCount); 1474 } else { 1475 ALOGW("Client adjusted notificationFrames from %u to %zu for frameCount %zu", 1476 mNotificationFramesAct, maxNotificationFrames, frameCount); 1477 } 1478 mNotificationFramesAct = (uint32_t) maxNotificationFrames; 1479 } 1480 } 1481 1482 // We retain a copy of the I/O handle, but don't own the reference 1483 mOutput = output; 1484 mRefreshRemaining = true; 1485 1486 // Starting address of buffers in shared memory. If there is a shared buffer, buffers 1487 // is the value of pointer() for the shared buffer, otherwise buffers points 1488 // immediately after the control block. This address is for the mapping within client 1489 // address space. AudioFlinger::TrackBase::mBuffer is for the server address space. 1490 void* buffers; 1491 if (mSharedBuffer == 0) { 1492 buffers = cblk + 1; 1493 } else { 1494 buffers = mSharedBuffer->pointer(); 1495 if (buffers == NULL) { 1496 ALOGE("Could not get buffer pointer"); 1497 return NO_INIT; 1498 } 1499 } 1500 1501 mAudioTrack->attachAuxEffect(mAuxEffectId); 1502 // FIXME doesn't take into account speed or future sample rate changes (until restoreTrack) 1503 // FIXME don't believe this lie 1504 mLatency = mAfLatency + (1000*frameCount) / mSampleRate; 1505 1506 mFrameCount = frameCount; 1507 // If IAudioTrack is re-created, don't let the requested frameCount 1508 // decrease. This can confuse clients that cache frameCount(). 1509 if (frameCount > mReqFrameCount) { 1510 mReqFrameCount = frameCount; 1511 } 1512 1513 // reset server position to 0 as we have new cblk. 1514 mServer = 0; 1515 1516 // update proxy 1517 if (mSharedBuffer == 0) { 1518 mStaticProxy.clear(); 1519 mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize); 1520 } else { 1521 mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize); 1522 mProxy = mStaticProxy; 1523 } 1524 1525 mProxy->setVolumeLR(gain_minifloat_pack( 1526 gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]), 1527 gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT]))); 1528 1529 mProxy->setSendLevel(mSendLevel); 1530 const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch); 1531 const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch); 1532 const float effectivePitch = adjustPitch(mPlaybackRate.mPitch); 1533 mProxy->setSampleRate(effectiveSampleRate); 1534 1535 AudioPlaybackRate playbackRateTemp = mPlaybackRate; 1536 playbackRateTemp.mSpeed = effectiveSpeed; 1537 playbackRateTemp.mPitch = effectivePitch; 1538 mProxy->setPlaybackRate(playbackRateTemp); 1539 mProxy->setMinimum(mNotificationFramesAct); 1540 1541 mDeathNotifier = new DeathNotifier(this); 1542 IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this); 1543 1544 if (mDeviceCallback != 0) { 1545 AudioSystem::addAudioDeviceCallback(mDeviceCallback, mOutput); 1546 } 1547 1548 return NO_ERROR; 1549 } 1550 1551 release: 1552 AudioSystem::releaseOutput(output, streamType, mSessionId); 1553 if (status == NO_ERROR) { 1554 status = NO_INIT; 1555 } 1556 return status; 1557 } 1558 1559 status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig) 1560 { 1561 if (audioBuffer == NULL) { 1562 if (nonContig != NULL) { 1563 *nonContig = 0; 1564 } 1565 return BAD_VALUE; 1566 } 1567 if (mTransfer != TRANSFER_OBTAIN) { 1568 audioBuffer->frameCount = 0; 1569 audioBuffer->size = 0; 1570 audioBuffer->raw = NULL; 1571 if (nonContig != NULL) { 1572 *nonContig = 0; 1573 } 1574 return INVALID_OPERATION; 1575 } 1576 1577 const struct timespec *requested; 1578 struct timespec timeout; 1579 if (waitCount == -1) { 1580 requested = &ClientProxy::kForever; 1581 } else if (waitCount == 0) { 1582 requested = &ClientProxy::kNonBlocking; 1583 } else if (waitCount > 0) { 1584 long long ms = WAIT_PERIOD_MS * (long long) waitCount; 1585 timeout.tv_sec = ms / 1000; 1586 timeout.tv_nsec = (int) (ms % 1000) * 1000000; 1587 requested = &timeout; 1588 } else { 1589 ALOGE("%s invalid waitCount %d", __func__, waitCount); 1590 requested = NULL; 1591 } 1592 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig); 1593 } 1594 1595 status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 1596 struct timespec *elapsed, size_t *nonContig) 1597 { 1598 // previous and new IAudioTrack sequence numbers are used to detect track re-creation 1599 uint32_t oldSequence = 0; 1600 uint32_t newSequence; 1601 1602 Proxy::Buffer buffer; 1603 status_t status = NO_ERROR; 1604 1605 static const int32_t kMaxTries = 5; 1606 int32_t tryCounter = kMaxTries; 1607 1608 do { 1609 // obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to 1610 // keep them from going away if another thread re-creates the track during obtainBuffer() 1611 sp<AudioTrackClientProxy> proxy; 1612 sp<IMemory> iMem; 1613 1614 { // start of lock scope 1615 AutoMutex lock(mLock); 1616 1617 newSequence = mSequence; 1618 // did previous obtainBuffer() fail due to media server death or voluntary invalidation? 1619 if (status == DEAD_OBJECT) { 1620 // re-create track, unless someone else has already done so 1621 if (newSequence == oldSequence) { 1622 status = restoreTrack_l("obtainBuffer"); 1623 if (status != NO_ERROR) { 1624 buffer.mFrameCount = 0; 1625 buffer.mRaw = NULL; 1626 buffer.mNonContig = 0; 1627 break; 1628 } 1629 } 1630 } 1631 oldSequence = newSequence; 1632 1633 if (status == NOT_ENOUGH_DATA) { 1634 restartIfDisabled(); 1635 } 1636 1637 // Keep the extra references 1638 proxy = mProxy; 1639 iMem = mCblkMemory; 1640 1641 if (mState == STATE_STOPPING) { 1642 status = -EINTR; 1643 buffer.mFrameCount = 0; 1644 buffer.mRaw = NULL; 1645 buffer.mNonContig = 0; 1646 break; 1647 } 1648 1649 // Non-blocking if track is stopped or paused 1650 if (mState != STATE_ACTIVE) { 1651 requested = &ClientProxy::kNonBlocking; 1652 } 1653 1654 } // end of lock scope 1655 1656 buffer.mFrameCount = audioBuffer->frameCount; 1657 // FIXME starts the requested timeout and elapsed over from scratch 1658 status = proxy->obtainBuffer(&buffer, requested, elapsed); 1659 } while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0)); 1660 1661 audioBuffer->frameCount = buffer.mFrameCount; 1662 audioBuffer->size = buffer.mFrameCount * mFrameSize; 1663 audioBuffer->raw = buffer.mRaw; 1664 if (nonContig != NULL) { 1665 *nonContig = buffer.mNonContig; 1666 } 1667 return status; 1668 } 1669 1670 void AudioTrack::releaseBuffer(const Buffer* audioBuffer) 1671 { 1672 // FIXME add error checking on mode, by adding an internal version 1673 if (mTransfer == TRANSFER_SHARED) { 1674 return; 1675 } 1676 1677 size_t stepCount = audioBuffer->size / mFrameSize; 1678 if (stepCount == 0) { 1679 return; 1680 } 1681 1682 Proxy::Buffer buffer; 1683 buffer.mFrameCount = stepCount; 1684 buffer.mRaw = audioBuffer->raw; 1685 1686 AutoMutex lock(mLock); 1687 mReleased += stepCount; 1688 mInUnderrun = false; 1689 mProxy->releaseBuffer(&buffer); 1690 1691 // restart track if it was disabled by audioflinger due to previous underrun 1692 restartIfDisabled(); 1693 } 1694 1695 void AudioTrack::restartIfDisabled() 1696 { 1697 int32_t flags = android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags); 1698 if ((mState == STATE_ACTIVE) && (flags & CBLK_DISABLED)) { 1699 ALOGW("releaseBuffer() track %p disabled due to previous underrun, restarting", this); 1700 // FIXME ignoring status 1701 mAudioTrack->start(); 1702 } 1703 } 1704 1705 // ------------------------------------------------------------------------- 1706 1707 ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking) 1708 { 1709 if (mTransfer != TRANSFER_SYNC) { 1710 return INVALID_OPERATION; 1711 } 1712 1713 if (isDirect()) { 1714 AutoMutex lock(mLock); 1715 int32_t flags = android_atomic_and( 1716 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), 1717 &mCblk->mFlags); 1718 if (flags & CBLK_INVALID) { 1719 return DEAD_OBJECT; 1720 } 1721 } 1722 1723 if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) { 1724 // Sanity-check: user is most-likely passing an error code, and it would 1725 // make the return value ambiguous (actualSize vs error). 1726 ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize); 1727 return BAD_VALUE; 1728 } 1729 1730 size_t written = 0; 1731 Buffer audioBuffer; 1732 1733 while (userSize >= mFrameSize) { 1734 audioBuffer.frameCount = userSize / mFrameSize; 1735 1736 status_t err = obtainBuffer(&audioBuffer, 1737 blocking ? &ClientProxy::kForever : &ClientProxy::kNonBlocking); 1738 if (err < 0) { 1739 if (written > 0) { 1740 break; 1741 } 1742 if (err == TIMED_OUT || err == -EINTR) { 1743 err = WOULD_BLOCK; 1744 } 1745 return ssize_t(err); 1746 } 1747 1748 size_t toWrite = audioBuffer.size; 1749 memcpy(audioBuffer.i8, buffer, toWrite); 1750 buffer = ((const char *) buffer) + toWrite; 1751 userSize -= toWrite; 1752 written += toWrite; 1753 1754 releaseBuffer(&audioBuffer); 1755 } 1756 1757 if (written > 0) { 1758 mFramesWritten += written / mFrameSize; 1759 } 1760 return written; 1761 } 1762 1763 // ------------------------------------------------------------------------- 1764 1765 nsecs_t AudioTrack::processAudioBuffer() 1766 { 1767 // Currently the AudioTrack thread is not created if there are no callbacks. 1768 // Would it ever make sense to run the thread, even without callbacks? 1769 // If so, then replace this by checks at each use for mCbf != NULL. 1770 LOG_ALWAYS_FATAL_IF(mCblk == NULL); 1771 1772 mLock.lock(); 1773 if (mAwaitBoost) { 1774 mAwaitBoost = false; 1775 mLock.unlock(); 1776 static const int32_t kMaxTries = 5; 1777 int32_t tryCounter = kMaxTries; 1778 uint32_t pollUs = 10000; 1779 do { 1780 int policy = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK; 1781 if (policy == SCHED_FIFO || policy == SCHED_RR) { 1782 break; 1783 } 1784 usleep(pollUs); 1785 pollUs <<= 1; 1786 } while (tryCounter-- > 0); 1787 if (tryCounter < 0) { 1788 ALOGE("did not receive expected priority boost on time"); 1789 } 1790 // Run again immediately 1791 return 0; 1792 } 1793 1794 // Can only reference mCblk while locked 1795 int32_t flags = android_atomic_and( 1796 ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END), &mCblk->mFlags); 1797 1798 // Check for track invalidation 1799 if (flags & CBLK_INVALID) { 1800 // for offloaded tracks restoreTrack_l() will just update the sequence and clear 1801 // AudioSystem cache. We should not exit here but after calling the callback so 1802 // that the upper layers can recreate the track 1803 if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) { 1804 status_t status __unused = restoreTrack_l("processAudioBuffer"); 1805 // FIXME unused status 1806 // after restoration, continue below to make sure that the loop and buffer events 1807 // are notified because they have been cleared from mCblk->mFlags above. 1808 } 1809 } 1810 1811 bool waitStreamEnd = mState == STATE_STOPPING; 1812 bool active = mState == STATE_ACTIVE; 1813 1814 // Manage underrun callback, must be done under lock to avoid race with releaseBuffer() 1815 bool newUnderrun = false; 1816 if (flags & CBLK_UNDERRUN) { 1817 #if 0 1818 // Currently in shared buffer mode, when the server reaches the end of buffer, 1819 // the track stays active in continuous underrun state. It's up to the application 1820 // to pause or stop the track, or set the position to a new offset within buffer. 1821 // This was some experimental code to auto-pause on underrun. Keeping it here 1822 // in "if 0" so we can re-visit this if we add a real sequencer for shared memory content. 1823 if (mTransfer == TRANSFER_SHARED) { 1824 mState = STATE_PAUSED; 1825 active = false; 1826 } 1827 #endif 1828 if (!mInUnderrun) { 1829 mInUnderrun = true; 1830 newUnderrun = true; 1831 } 1832 } 1833 1834 // Get current position of server 1835 Modulo<uint32_t> position(updateAndGetPosition_l()); 1836 1837 // Manage marker callback 1838 bool markerReached = false; 1839 Modulo<uint32_t> markerPosition(mMarkerPosition); 1840 // uses 32 bit wraparound for comparison with position. 1841 if (!mMarkerReached && markerPosition.value() > 0 && position >= markerPosition) { 1842 mMarkerReached = markerReached = true; 1843 } 1844 1845 // Determine number of new position callback(s) that will be needed, while locked 1846 size_t newPosCount = 0; 1847 Modulo<uint32_t> newPosition(mNewPosition); 1848 uint32_t updatePeriod = mUpdatePeriod; 1849 // FIXME fails for wraparound, need 64 bits 1850 if (updatePeriod > 0 && position >= newPosition) { 1851 newPosCount = ((position - newPosition).value() / updatePeriod) + 1; 1852 mNewPosition += updatePeriod * newPosCount; 1853 } 1854 1855 // Cache other fields that will be needed soon 1856 uint32_t sampleRate = mSampleRate; 1857 float speed = mPlaybackRate.mSpeed; 1858 const uint32_t notificationFrames = mNotificationFramesAct; 1859 if (mRefreshRemaining) { 1860 mRefreshRemaining = false; 1861 mRemainingFrames = notificationFrames; 1862 mRetryOnPartialBuffer = false; 1863 } 1864 size_t misalignment = mProxy->getMisalignment(); 1865 uint32_t sequence = mSequence; 1866 sp<AudioTrackClientProxy> proxy = mProxy; 1867 1868 // Determine the number of new loop callback(s) that will be needed, while locked. 1869 int loopCountNotifications = 0; 1870 uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END 1871 1872 if (mLoopCount > 0) { 1873 int loopCount; 1874 size_t bufferPosition; 1875 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount); 1876 loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition; 1877 loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications); 1878 mLoopCountNotified = loopCount; // discard any excess notifications 1879 } else if (mLoopCount < 0) { 1880 // FIXME: We're not accurate with notification count and position with infinite looping 1881 // since loopCount from server side will always return -1 (we could decrement it). 1882 size_t bufferPosition = mStaticProxy->getBufferPosition(); 1883 loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0); 1884 loopPeriod = mLoopEnd - bufferPosition; 1885 } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) { 1886 size_t bufferPosition = mStaticProxy->getBufferPosition(); 1887 loopPeriod = mFrameCount - bufferPosition; 1888 } 1889 1890 // These fields don't need to be cached, because they are assigned only by set(): 1891 // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFlags 1892 // mFlags is also assigned by createTrack_l(), but not the bit we care about. 1893 1894 mLock.unlock(); 1895 1896 // get anchor time to account for callbacks. 1897 const nsecs_t timeBeforeCallbacks = systemTime(); 1898 1899 if (waitStreamEnd) { 1900 // FIXME: Instead of blocking in proxy->waitStreamEndDone(), Callback thread 1901 // should wait on proxy futex and handle CBLK_STREAM_END_DONE within this function 1902 // (and make sure we don't callback for more data while we're stopping). 1903 // This helps with position, marker notifications, and track invalidation. 1904 struct timespec timeout; 1905 timeout.tv_sec = WAIT_STREAM_END_TIMEOUT_SEC; 1906 timeout.tv_nsec = 0; 1907 1908 status_t status = proxy->waitStreamEndDone(&timeout); 1909 switch (status) { 1910 case NO_ERROR: 1911 case DEAD_OBJECT: 1912 case TIMED_OUT: 1913 if (status != DEAD_OBJECT) { 1914 // for DEAD_OBJECT, we do not send a EVENT_STREAM_END after stop(); 1915 // instead, the application should handle the EVENT_NEW_IAUDIOTRACK. 1916 mCbf(EVENT_STREAM_END, mUserData, NULL); 1917 } 1918 { 1919 AutoMutex lock(mLock); 1920 // The previously assigned value of waitStreamEnd is no longer valid, 1921 // since the mutex has been unlocked and either the callback handler 1922 // or another thread could have re-started the AudioTrack during that time. 1923 waitStreamEnd = mState == STATE_STOPPING; 1924 if (waitStreamEnd) { 1925 mState = STATE_STOPPED; 1926 mReleased = 0; 1927 } 1928 } 1929 if (waitStreamEnd && status != DEAD_OBJECT) { 1930 return NS_INACTIVE; 1931 } 1932 break; 1933 } 1934 return 0; 1935 } 1936 1937 // perform callbacks while unlocked 1938 if (newUnderrun) { 1939 mCbf(EVENT_UNDERRUN, mUserData, NULL); 1940 } 1941 while (loopCountNotifications > 0) { 1942 mCbf(EVENT_LOOP_END, mUserData, NULL); 1943 --loopCountNotifications; 1944 } 1945 if (flags & CBLK_BUFFER_END) { 1946 mCbf(EVENT_BUFFER_END, mUserData, NULL); 1947 } 1948 if (markerReached) { 1949 mCbf(EVENT_MARKER, mUserData, &markerPosition); 1950 } 1951 while (newPosCount > 0) { 1952 size_t temp = newPosition.value(); // FIXME size_t != uint32_t 1953 mCbf(EVENT_NEW_POS, mUserData, &temp); 1954 newPosition += updatePeriod; 1955 newPosCount--; 1956 } 1957 1958 if (mObservedSequence != sequence) { 1959 mObservedSequence = sequence; 1960 mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL); 1961 // for offloaded tracks, just wait for the upper layers to recreate the track 1962 if (isOffloadedOrDirect()) { 1963 return NS_INACTIVE; 1964 } 1965 } 1966 1967 // if inactive, then don't run me again until re-started 1968 if (!active) { 1969 return NS_INACTIVE; 1970 } 1971 1972 // Compute the estimated time until the next timed event (position, markers, loops) 1973 // FIXME only for non-compressed audio 1974 uint32_t minFrames = ~0; 1975 if (!markerReached && position < markerPosition) { 1976 minFrames = (markerPosition - position).value(); 1977 } 1978 if (loopPeriod > 0 && loopPeriod < minFrames) { 1979 // loopPeriod is already adjusted for actual position. 1980 minFrames = loopPeriod; 1981 } 1982 if (updatePeriod > 0) { 1983 minFrames = min(minFrames, (newPosition - position).value()); 1984 } 1985 1986 // If > 0, poll periodically to recover from a stuck server. A good value is 2. 1987 static const uint32_t kPoll = 0; 1988 if (kPoll > 0 && mTransfer == TRANSFER_CALLBACK && kPoll * notificationFrames < minFrames) { 1989 minFrames = kPoll * notificationFrames; 1990 } 1991 1992 // This "fudge factor" avoids soaking CPU, and compensates for late progress by server 1993 static const nsecs_t kWaitPeriodNs = WAIT_PERIOD_MS * 1000000LL; 1994 const nsecs_t timeAfterCallbacks = systemTime(); 1995 1996 // Convert frame units to time units 1997 nsecs_t ns = NS_WHENEVER; 1998 if (minFrames != (uint32_t) ~0) { 1999 ns = framesToNanoseconds(minFrames, sampleRate, speed) + kWaitPeriodNs; 2000 ns -= (timeAfterCallbacks - timeBeforeCallbacks); // account for callback time 2001 // TODO: Should we warn if the callback time is too long? 2002 if (ns < 0) ns = 0; 2003 } 2004 2005 // If not supplying data by EVENT_MORE_DATA, then we're done 2006 if (mTransfer != TRANSFER_CALLBACK) { 2007 return ns; 2008 } 2009 2010 // EVENT_MORE_DATA callback handling. 2011 // Timing for linear pcm audio data formats can be derived directly from the 2012 // buffer fill level. 2013 // Timing for compressed data is not directly available from the buffer fill level, 2014 // rather indirectly from waiting for blocking mode callbacks or waiting for obtain() 2015 // to return a certain fill level. 2016 2017 struct timespec timeout; 2018 const struct timespec *requested = &ClientProxy::kForever; 2019 if (ns != NS_WHENEVER) { 2020 timeout.tv_sec = ns / 1000000000LL; 2021 timeout.tv_nsec = ns % 1000000000LL; 2022 ALOGV("timeout %ld.%03d", timeout.tv_sec, (int) timeout.tv_nsec / 1000000); 2023 requested = &timeout; 2024 } 2025 2026 size_t writtenFrames = 0; 2027 while (mRemainingFrames > 0) { 2028 2029 Buffer audioBuffer; 2030 audioBuffer.frameCount = mRemainingFrames; 2031 size_t nonContig; 2032 status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig); 2033 LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0), 2034 "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount); 2035 requested = &ClientProxy::kNonBlocking; 2036 size_t avail = audioBuffer.frameCount + nonContig; 2037 ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d", 2038 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err); 2039 if (err != NO_ERROR) { 2040 if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR || 2041 (isOffloaded() && (err == DEAD_OBJECT))) { 2042 // FIXME bug 25195759 2043 return 1000000; 2044 } 2045 ALOGE("Error %d obtaining an audio buffer, giving up.", err); 2046 return NS_NEVER; 2047 } 2048 2049 if (mRetryOnPartialBuffer && audio_has_proportional_frames(mFormat)) { 2050 mRetryOnPartialBuffer = false; 2051 if (avail < mRemainingFrames) { 2052 if (ns > 0) { // account for obtain time 2053 const nsecs_t timeNow = systemTime(); 2054 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks)); 2055 } 2056 nsecs_t myns = framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed); 2057 if (ns < 0 /* NS_WHENEVER */ || myns < ns) { 2058 ns = myns; 2059 } 2060 return ns; 2061 } 2062 } 2063 2064 size_t reqSize = audioBuffer.size; 2065 mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer); 2066 size_t writtenSize = audioBuffer.size; 2067 2068 // Sanity check on returned size 2069 if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) { 2070 ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes", 2071 reqSize, ssize_t(writtenSize)); 2072 return NS_NEVER; 2073 } 2074 2075 if (writtenSize == 0) { 2076 // The callback is done filling buffers 2077 // Keep this thread going to handle timed events and 2078 // still try to get more data in intervals of WAIT_PERIOD_MS 2079 // but don't just loop and block the CPU, so wait 2080 2081 // mCbf(EVENT_MORE_DATA, ...) might either 2082 // (1) Block until it can fill the buffer, returning 0 size on EOS. 2083 // (2) Block until it can fill the buffer, returning 0 data (silence) on EOS. 2084 // (3) Return 0 size when no data is available, does not wait for more data. 2085 // 2086 // (1) and (2) occurs with AudioPlayer/AwesomePlayer; (3) occurs with NuPlayer. 2087 // We try to compute the wait time to avoid a tight sleep-wait cycle, 2088 // especially for case (3). 2089 // 2090 // The decision to support (1) and (2) affect the sizing of mRemainingFrames 2091 // and this loop; whereas for case (3) we could simply check once with the full 2092 // buffer size and skip the loop entirely. 2093 2094 nsecs_t myns; 2095 if (audio_has_proportional_frames(mFormat)) { 2096 // time to wait based on buffer occupancy 2097 const nsecs_t datans = mRemainingFrames <= avail ? 0 : 2098 framesToNanoseconds(mRemainingFrames - avail, sampleRate, speed); 2099 // audio flinger thread buffer size (TODO: adjust for fast tracks) 2100 // FIXME: use mAfFrameCountHAL instead of mAfFrameCount below for fast tracks. 2101 const nsecs_t afns = framesToNanoseconds(mAfFrameCount, mAfSampleRate, speed); 2102 // add a half the AudioFlinger buffer time to avoid soaking CPU if datans is 0. 2103 myns = datans + (afns / 2); 2104 } else { 2105 // FIXME: This could ping quite a bit if the buffer isn't full. 2106 // Note that when mState is stopping we waitStreamEnd, so it never gets here. 2107 myns = kWaitPeriodNs; 2108 } 2109 if (ns > 0) { // account for obtain and callback time 2110 const nsecs_t timeNow = systemTime(); 2111 ns = max((nsecs_t)0, ns - (timeNow - timeAfterCallbacks)); 2112 } 2113 if (ns < 0 /* NS_WHENEVER */ || myns < ns) { 2114 ns = myns; 2115 } 2116 return ns; 2117 } 2118 2119 size_t releasedFrames = writtenSize / mFrameSize; 2120 audioBuffer.frameCount = releasedFrames; 2121 mRemainingFrames -= releasedFrames; 2122 if (misalignment >= releasedFrames) { 2123 misalignment -= releasedFrames; 2124 } else { 2125 misalignment = 0; 2126 } 2127 2128 releaseBuffer(&audioBuffer); 2129 writtenFrames += releasedFrames; 2130 2131 // FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer 2132 // if callback doesn't like to accept the full chunk 2133 if (writtenSize < reqSize) { 2134 continue; 2135 } 2136 2137 // There could be enough non-contiguous frames available to satisfy the remaining request 2138 if (mRemainingFrames <= nonContig) { 2139 continue; 2140 } 2141 2142 #if 0 2143 // This heuristic tries to collapse a series of EVENT_MORE_DATA that would total to a 2144 // sum <= notificationFrames. It replaces that series by at most two EVENT_MORE_DATA 2145 // that total to a sum == notificationFrames. 2146 if (0 < misalignment && misalignment <= mRemainingFrames) { 2147 mRemainingFrames = misalignment; 2148 return ((double)mRemainingFrames * 1100000000) / ((double)sampleRate * speed); 2149 } 2150 #endif 2151 2152 } 2153 if (writtenFrames > 0) { 2154 AutoMutex lock(mLock); 2155 mFramesWritten += writtenFrames; 2156 } 2157 mRemainingFrames = notificationFrames; 2158 mRetryOnPartialBuffer = true; 2159 2160 // A lot has transpired since ns was calculated, so run again immediately and re-calculate 2161 return 0; 2162 } 2163 2164 status_t AudioTrack::restoreTrack_l(const char *from) 2165 { 2166 ALOGW("dead IAudioTrack, %s, creating a new one from %s()", 2167 isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from); 2168 ++mSequence; 2169 2170 // refresh the audio configuration cache in this process to make sure we get new 2171 // output parameters and new IAudioFlinger in createTrack_l() 2172 AudioSystem::clearAudioConfigCache(); 2173 2174 if (isOffloadedOrDirect_l() || mDoNotReconnect) { 2175 // FIXME re-creation of offloaded and direct tracks is not yet implemented; 2176 // reconsider enabling for linear PCM encodings when position can be preserved. 2177 return DEAD_OBJECT; 2178 } 2179 2180 // Save so we can return count since creation. 2181 mUnderrunCountOffset = getUnderrunCount_l(); 2182 2183 // save the old static buffer position 2184 size_t bufferPosition = 0; 2185 int loopCount = 0; 2186 if (mStaticProxy != 0) { 2187 mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount); 2188 } 2189 2190 mFlags = mOrigFlags; 2191 2192 // If a new IAudioTrack is successfully created, createTrack_l() will modify the 2193 // following member variables: mAudioTrack, mCblkMemory and mCblk. 2194 // It will also delete the strong references on previous IAudioTrack and IMemory. 2195 // If a new IAudioTrack cannot be created, the previous (dead) instance will be left intact. 2196 status_t result = createTrack_l(); 2197 2198 if (result == NO_ERROR) { 2199 // take the frames that will be lost by track recreation into account in saved position 2200 // For streaming tracks, this is the amount we obtained from the user/client 2201 // (not the number actually consumed at the server - those are already lost). 2202 if (mStaticProxy == 0) { 2203 mPosition = mReleased; 2204 } 2205 // Continue playback from last known position and restore loop. 2206 if (mStaticProxy != 0) { 2207 if (loopCount != 0) { 2208 mStaticProxy->setBufferPositionAndLoop(bufferPosition, 2209 mLoopStart, mLoopEnd, loopCount); 2210 } else { 2211 mStaticProxy->setBufferPosition(bufferPosition); 2212 if (bufferPosition == mFrameCount) { 2213 ALOGD("restoring track at end of static buffer"); 2214 } 2215 } 2216 } 2217 if (mState == STATE_ACTIVE) { 2218 result = mAudioTrack->start(); 2219 mFramesWrittenServerOffset = mFramesWritten; // server resets to zero so we offset 2220 } 2221 } 2222 if (result != NO_ERROR) { 2223 ALOGW("restoreTrack_l() failed status %d", result); 2224 mState = STATE_STOPPED; 2225 mReleased = 0; 2226 } 2227 2228 return result; 2229 } 2230 2231 Modulo<uint32_t> AudioTrack::updateAndGetPosition_l() 2232 { 2233 // This is the sole place to read server consumed frames 2234 Modulo<uint32_t> newServer(mProxy->getPosition()); 2235 const int32_t delta = (newServer - mServer).signedValue(); 2236 // TODO There is controversy about whether there can be "negative jitter" in server position. 2237 // This should be investigated further, and if possible, it should be addressed. 2238 // A more definite failure mode is infrequent polling by client. 2239 // One could call (void)getPosition_l() in releaseBuffer(), 2240 // so mReleased and mPosition are always lock-step as best possible. 2241 // That should ensure delta never goes negative for infrequent polling 2242 // unless the server has more than 2^31 frames in its buffer, 2243 // in which case the use of uint32_t for these counters has bigger issues. 2244 ALOGE_IF(delta < 0, 2245 "detected illegal retrograde motion by the server: mServer advanced by %d", 2246 delta); 2247 mServer = newServer; 2248 if (delta > 0) { // avoid retrograde 2249 mPosition += delta; 2250 } 2251 return mPosition; 2252 } 2253 2254 bool AudioTrack::isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const 2255 { 2256 // applicable for mixing tracks only (not offloaded or direct) 2257 if (mStaticProxy != 0) { 2258 return true; // static tracks do not have issues with buffer sizing. 2259 } 2260 const size_t minFrameCount = 2261 calculateMinFrameCount(mAfLatency, mAfFrameCount, mAfSampleRate, sampleRate, speed 2262 /*, 0 mNotificationsPerBufferReq*/); 2263 ALOGV("isSampleRateSpeedAllowed_l mFrameCount %zu minFrameCount %zu", 2264 mFrameCount, minFrameCount); 2265 return mFrameCount >= minFrameCount; 2266 } 2267 2268 status_t AudioTrack::setParameters(const String8& keyValuePairs) 2269 { 2270 AutoMutex lock(mLock); 2271 return mAudioTrack->setParameters(keyValuePairs); 2272 } 2273 2274 status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp) 2275 { 2276 if (timestamp == nullptr) { 2277 return BAD_VALUE; 2278 } 2279 AutoMutex lock(mLock); 2280 return getTimestamp_l(timestamp); 2281 } 2282 2283 status_t AudioTrack::getTimestamp_l(ExtendedTimestamp *timestamp) 2284 { 2285 if (mCblk->mFlags & CBLK_INVALID) { 2286 const status_t status = restoreTrack_l("getTimestampExtended"); 2287 if (status != OK) { 2288 // per getTimestamp() API doc in header, we return DEAD_OBJECT here, 2289 // recommending that the track be recreated. 2290 return DEAD_OBJECT; 2291 } 2292 } 2293 // check for offloaded/direct here in case restoring somehow changed those flags. 2294 if (isOffloadedOrDirect_l()) { 2295 return INVALID_OPERATION; // not supported 2296 } 2297 status_t status = mProxy->getTimestamp(timestamp); 2298 LOG_ALWAYS_FATAL_IF(status != OK, "status %d not allowed from proxy getTimestamp", status); 2299 bool found = false; 2300 timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten; 2301 timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0; 2302 // server side frame offset in case AudioTrack has been restored. 2303 for (int i = ExtendedTimestamp::LOCATION_SERVER; 2304 i < ExtendedTimestamp::LOCATION_MAX; ++i) { 2305 if (timestamp->mTimeNs[i] >= 0) { 2306 // apply server offset (frames flushed is ignored 2307 // so we don't report the jump when the flush occurs). 2308 timestamp->mPosition[i] += mFramesWrittenServerOffset; 2309 found = true; 2310 } 2311 } 2312 return found ? OK : WOULD_BLOCK; 2313 } 2314 2315 status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp) 2316 { 2317 AutoMutex lock(mLock); 2318 2319 bool previousTimestampValid = mPreviousTimestampValid; 2320 // Set false here to cover all the error return cases. 2321 mPreviousTimestampValid = false; 2322 2323 switch (mState) { 2324 case STATE_ACTIVE: 2325 case STATE_PAUSED: 2326 break; // handle below 2327 case STATE_FLUSHED: 2328 case STATE_STOPPED: 2329 return WOULD_BLOCK; 2330 case STATE_STOPPING: 2331 case STATE_PAUSED_STOPPING: 2332 if (!isOffloaded_l()) { 2333 return INVALID_OPERATION; 2334 } 2335 break; // offloaded tracks handled below 2336 default: 2337 LOG_ALWAYS_FATAL("Invalid mState in getTimestamp(): %d", mState); 2338 break; 2339 } 2340 2341 if (mCblk->mFlags & CBLK_INVALID) { 2342 const status_t status = restoreTrack_l("getTimestamp"); 2343 if (status != OK) { 2344 // per getTimestamp() API doc in header, we return DEAD_OBJECT here, 2345 // recommending that the track be recreated. 2346 return DEAD_OBJECT; 2347 } 2348 } 2349 2350 // The presented frame count must always lag behind the consumed frame count. 2351 // To avoid a race, read the presented frames first. This ensures that presented <= consumed. 2352 2353 status_t status; 2354 if (isOffloadedOrDirect_l()) { 2355 // use Binder to get timestamp 2356 status = mAudioTrack->getTimestamp(timestamp); 2357 } else { 2358 // read timestamp from shared memory 2359 ExtendedTimestamp ets; 2360 status = mProxy->getTimestamp(&ets); 2361 if (status == OK) { 2362 ExtendedTimestamp::Location location; 2363 status = ets.getBestTimestamp(×tamp, &location); 2364 2365 if (status == OK) { 2366 // It is possible that the best location has moved from the kernel to the server. 2367 // In this case we adjust the position from the previous computed latency. 2368 if (location == ExtendedTimestamp::LOCATION_SERVER) { 2369 ALOGW_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_KERNEL, 2370 "getTimestamp() location moved from kernel to server"); 2371 // check that the last kernel OK time info exists and the positions 2372 // are valid (if they predate the current track, the positions may 2373 // be zero or negative). 2374 const int64_t frames = 2375 (ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] < 0 || 2376 ets.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] < 0 || 2377 ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] <= 0 || 2378 ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] <= 0) 2379 ? 2380 int64_t((double)mAfLatency * mSampleRate * mPlaybackRate.mSpeed 2381 / 1000) 2382 : 2383 (ets.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] 2384 - ets.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK]); 2385 ALOGV("frame adjustment:%lld timestamp:%s", 2386 (long long)frames, ets.toString().c_str()); 2387 if (frames >= ets.mPosition[location]) { 2388 timestamp.mPosition = 0; 2389 } else { 2390 timestamp.mPosition = (uint32_t)(ets.mPosition[location] - frames); 2391 } 2392 } else if (location == ExtendedTimestamp::LOCATION_KERNEL) { 2393 ALOGV_IF(mPreviousLocation == ExtendedTimestamp::LOCATION_SERVER, 2394 "getTimestamp() location moved from server to kernel"); 2395 } 2396 mPreviousLocation = location; 2397 } else { 2398 // right after AudioTrack is started, one may not find a timestamp 2399 ALOGV("getBestTimestamp did not find timestamp"); 2400 } 2401 } 2402 if (status == INVALID_OPERATION) { 2403 status = WOULD_BLOCK; 2404 } 2405 } 2406 if (status != NO_ERROR) { 2407 ALOGV_IF(status != WOULD_BLOCK, "getTimestamp error:%#x", status); 2408 return status; 2409 } 2410 if (isOffloadedOrDirect_l()) { 2411 if (isOffloaded_l() && (mState == STATE_PAUSED || mState == STATE_PAUSED_STOPPING)) { 2412 // use cached paused position in case another offloaded track is running. 2413 timestamp.mPosition = mPausedPosition; 2414 clock_gettime(CLOCK_MONOTONIC, ×tamp.mTime); 2415 return NO_ERROR; 2416 } 2417 2418 // Check whether a pending flush or stop has completed, as those commands may 2419 // be asynchronous or return near finish or exhibit glitchy behavior. 2420 // 2421 // Originally this showed up as the first timestamp being a continuation of 2422 // the previous song under gapless playback. 2423 // However, we sometimes see zero timestamps, then a glitch of 2424 // the previous song's position, and then correct timestamps afterwards. 2425 if (mStartUs != 0 && mSampleRate != 0) { 2426 static const int kTimeJitterUs = 100000; // 100 ms 2427 static const int k1SecUs = 1000000; 2428 2429 const int64_t timeNow = getNowUs(); 2430 2431 if (timeNow < mStartUs + k1SecUs) { // within first second of starting 2432 const int64_t timestampTimeUs = convertTimespecToUs(timestamp.mTime); 2433 if (timestampTimeUs < mStartUs) { 2434 return WOULD_BLOCK; // stale timestamp time, occurs before start. 2435 } 2436 const int64_t deltaTimeUs = timestampTimeUs - mStartUs; 2437 const int64_t deltaPositionByUs = (double)timestamp.mPosition * 1000000 2438 / ((double)mSampleRate * mPlaybackRate.mSpeed); 2439 2440 if (deltaPositionByUs > deltaTimeUs + kTimeJitterUs) { 2441 // Verify that the counter can't count faster than the sample rate 2442 // since the start time. If greater, then that means we may have failed 2443 // to completely flush or stop the previous playing track. 2444 ALOGW_IF(!mTimestampStartupGlitchReported, 2445 "getTimestamp startup glitch detected" 2446 " deltaTimeUs(%lld) deltaPositionUs(%lld) tsmPosition(%u)", 2447 (long long)deltaTimeUs, (long long)deltaPositionByUs, 2448 timestamp.mPosition); 2449 mTimestampStartupGlitchReported = true; 2450 if (previousTimestampValid 2451 && mPreviousTimestamp.mPosition == 0 /* should be true if valid */) { 2452 timestamp = mPreviousTimestamp; 2453 mPreviousTimestampValid = true; 2454 return NO_ERROR; 2455 } 2456 return WOULD_BLOCK; 2457 } 2458 if (deltaPositionByUs != 0) { 2459 mStartUs = 0; // don't check again, we got valid nonzero position. 2460 } 2461 } else { 2462 mStartUs = 0; // don't check again, start time expired. 2463 } 2464 mTimestampStartupGlitchReported = false; 2465 } 2466 } else { 2467 // Update the mapping between local consumed (mPosition) and server consumed (mServer) 2468 (void) updateAndGetPosition_l(); 2469 // Server consumed (mServer) and presented both use the same server time base, 2470 // and server consumed is always >= presented. 2471 // The delta between these represents the number of frames in the buffer pipeline. 2472 // If this delta between these is greater than the client position, it means that 2473 // actually presented is still stuck at the starting line (figuratively speaking), 2474 // waiting for the first frame to go by. So we can't report a valid timestamp yet. 2475 // Note: We explicitly use non-Modulo comparison here - potential wrap issue when 2476 // mPosition exceeds 32 bits. 2477 // TODO Remove when timestamp is updated to contain pipeline status info. 2478 const int32_t pipelineDepthInFrames = (mServer - timestamp.mPosition).signedValue(); 2479 if (pipelineDepthInFrames > 0 /* should be true, but we check anyways */ 2480 && (uint32_t)pipelineDepthInFrames > mPosition.value()) { 2481 return INVALID_OPERATION; 2482 } 2483 // Convert timestamp position from server time base to client time base. 2484 // TODO The following code should work OK now because timestamp.mPosition is 32-bit. 2485 // But if we change it to 64-bit then this could fail. 2486 // Use Modulo computation here. 2487 timestamp.mPosition = (mPosition - mServer + timestamp.mPosition).value(); 2488 // Immediately after a call to getPosition_l(), mPosition and 2489 // mServer both represent the same frame position. mPosition is 2490 // in client's point of view, and mServer is in server's point of 2491 // view. So the difference between them is the "fudge factor" 2492 // between client and server views due to stop() and/or new 2493 // IAudioTrack. And timestamp.mPosition is initially in server's 2494 // point of view, so we need to apply the same fudge factor to it. 2495 } 2496 2497 // Prevent retrograde motion in timestamp. 2498 // This is sometimes caused by erratic reports of the available space in the ALSA drivers. 2499 if (status == NO_ERROR) { 2500 if (previousTimestampValid) { 2501 #define TIME_TO_NANOS(time) ((int64_t)time.tv_sec * 1000000000 + time.tv_nsec) 2502 const int64_t previousTimeNanos = TIME_TO_NANOS(mPreviousTimestamp.mTime); 2503 const int64_t currentTimeNanos = TIME_TO_NANOS(timestamp.mTime); 2504 #undef TIME_TO_NANOS 2505 if (currentTimeNanos < previousTimeNanos) { 2506 ALOGW("retrograde timestamp time"); 2507 // FIXME Consider blocking this from propagating upwards. 2508 } 2509 2510 // Looking at signed delta will work even when the timestamps 2511 // are wrapping around. 2512 int32_t deltaPosition = (Modulo<uint32_t>(timestamp.mPosition) 2513 - mPreviousTimestamp.mPosition).signedValue(); 2514 // position can bobble slightly as an artifact; this hides the bobble 2515 static const int32_t MINIMUM_POSITION_DELTA = 8; 2516 if (deltaPosition < 0) { 2517 // Only report once per position instead of spamming the log. 2518 if (!mRetrogradeMotionReported) { 2519 ALOGW("retrograde timestamp position corrected, %d = %u - %u", 2520 deltaPosition, 2521 timestamp.mPosition, 2522 mPreviousTimestamp.mPosition); 2523 mRetrogradeMotionReported = true; 2524 } 2525 } else { 2526 mRetrogradeMotionReported = false; 2527 } 2528 if (deltaPosition < MINIMUM_POSITION_DELTA) { 2529 timestamp = mPreviousTimestamp; // Use last valid timestamp. 2530 } 2531 } 2532 mPreviousTimestamp = timestamp; 2533 mPreviousTimestampValid = true; 2534 } 2535 2536 return status; 2537 } 2538 2539 String8 AudioTrack::getParameters(const String8& keys) 2540 { 2541 audio_io_handle_t output = getOutput(); 2542 if (output != AUDIO_IO_HANDLE_NONE) { 2543 return AudioSystem::getParameters(output, keys); 2544 } else { 2545 return String8::empty(); 2546 } 2547 } 2548 2549 bool AudioTrack::isOffloaded() const 2550 { 2551 AutoMutex lock(mLock); 2552 return isOffloaded_l(); 2553 } 2554 2555 bool AudioTrack::isDirect() const 2556 { 2557 AutoMutex lock(mLock); 2558 return isDirect_l(); 2559 } 2560 2561 bool AudioTrack::isOffloadedOrDirect() const 2562 { 2563 AutoMutex lock(mLock); 2564 return isOffloadedOrDirect_l(); 2565 } 2566 2567 2568 status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const 2569 { 2570 2571 const size_t SIZE = 256; 2572 char buffer[SIZE]; 2573 String8 result; 2574 2575 result.append(" AudioTrack::dump\n"); 2576 snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, 2577 mVolume[AUDIO_INTERLEAVE_LEFT], mVolume[AUDIO_INTERLEAVE_RIGHT]); 2578 result.append(buffer); 2579 snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%zu)\n", mFormat, 2580 mChannelCount, mFrameCount); 2581 result.append(buffer); 2582 snprintf(buffer, 255, " sample rate(%u), speed(%f), status(%d)\n", 2583 mSampleRate, mPlaybackRate.mSpeed, mStatus); 2584 result.append(buffer); 2585 snprintf(buffer, 255, " state(%d), latency (%d)\n", mState, mLatency); 2586 result.append(buffer); 2587 ::write(fd, result.string(), result.size()); 2588 return NO_ERROR; 2589 } 2590 2591 uint32_t AudioTrack::getUnderrunCount() const 2592 { 2593 AutoMutex lock(mLock); 2594 return getUnderrunCount_l(); 2595 } 2596 2597 uint32_t AudioTrack::getUnderrunCount_l() const 2598 { 2599 return mProxy->getUnderrunCount() + mUnderrunCountOffset; 2600 } 2601 2602 uint32_t AudioTrack::getUnderrunFrames() const 2603 { 2604 AutoMutex lock(mLock); 2605 return mProxy->getUnderrunFrames(); 2606 } 2607 2608 status_t AudioTrack::addAudioDeviceCallback(const sp<AudioSystem::AudioDeviceCallback>& callback) 2609 { 2610 if (callback == 0) { 2611 ALOGW("%s adding NULL callback!", __FUNCTION__); 2612 return BAD_VALUE; 2613 } 2614 AutoMutex lock(mLock); 2615 if (mDeviceCallback == callback) { 2616 ALOGW("%s adding same callback!", __FUNCTION__); 2617 return INVALID_OPERATION; 2618 } 2619 status_t status = NO_ERROR; 2620 if (mOutput != AUDIO_IO_HANDLE_NONE) { 2621 if (mDeviceCallback != 0) { 2622 ALOGW("%s callback already present!", __FUNCTION__); 2623 AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); 2624 } 2625 status = AudioSystem::addAudioDeviceCallback(callback, mOutput); 2626 } 2627 mDeviceCallback = callback; 2628 return status; 2629 } 2630 2631 status_t AudioTrack::removeAudioDeviceCallback( 2632 const sp<AudioSystem::AudioDeviceCallback>& callback) 2633 { 2634 if (callback == 0) { 2635 ALOGW("%s removing NULL callback!", __FUNCTION__); 2636 return BAD_VALUE; 2637 } 2638 AutoMutex lock(mLock); 2639 if (mDeviceCallback != callback) { 2640 ALOGW("%s removing different callback!", __FUNCTION__); 2641 return INVALID_OPERATION; 2642 } 2643 if (mOutput != AUDIO_IO_HANDLE_NONE) { 2644 AudioSystem::removeAudioDeviceCallback(mDeviceCallback, mOutput); 2645 } 2646 mDeviceCallback = 0; 2647 return NO_ERROR; 2648 } 2649 2650 status_t AudioTrack::pendingDuration(int32_t *msec, ExtendedTimestamp::Location location) 2651 { 2652 if (msec == nullptr || 2653 (location != ExtendedTimestamp::LOCATION_SERVER 2654 && location != ExtendedTimestamp::LOCATION_KERNEL)) { 2655 return BAD_VALUE; 2656 } 2657 AutoMutex lock(mLock); 2658 // inclusive of offloaded and direct tracks. 2659 // 2660 // It is possible, but not enabled, to allow duration computation for non-pcm 2661 // audio_has_proportional_frames() formats because currently they have 2662 // the drain rate equivalent to the pcm sample rate * framesize. 2663 if (!isPurePcmData_l()) { 2664 return INVALID_OPERATION; 2665 } 2666 ExtendedTimestamp ets; 2667 if (getTimestamp_l(&ets) == OK 2668 && ets.mTimeNs[location] > 0) { 2669 int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT] 2670 - ets.mPosition[location]; 2671 if (diff < 0) { 2672 *msec = 0; 2673 } else { 2674 // ms is the playback time by frames 2675 int64_t ms = (int64_t)((double)diff * 1000 / 2676 ((double)mSampleRate * mPlaybackRate.mSpeed)); 2677 // clockdiff is the timestamp age (negative) 2678 int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 : 2679 ets.mTimeNs[location] 2680 + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC] 2681 - systemTime(SYSTEM_TIME_MONOTONIC); 2682 2683 //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff); 2684 static const int NANOS_PER_MILLIS = 1000000; 2685 *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS); 2686 } 2687 return NO_ERROR; 2688 } 2689 if (location != ExtendedTimestamp::LOCATION_SERVER) { 2690 return INVALID_OPERATION; // LOCATION_KERNEL is not available 2691 } 2692 // use server position directly (offloaded and direct arrive here) 2693 updateAndGetPosition_l(); 2694 int32_t diff = (Modulo<uint32_t>(mFramesWritten) - mPosition).signedValue(); 2695 *msec = (diff <= 0) ? 0 2696 : (int32_t)((double)diff * 1000 / ((double)mSampleRate * mPlaybackRate.mSpeed)); 2697 return NO_ERROR; 2698 } 2699 2700 // ========================================================================= 2701 2702 void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused) 2703 { 2704 sp<AudioTrack> audioTrack = mAudioTrack.promote(); 2705 if (audioTrack != 0) { 2706 AutoMutex lock(audioTrack->mLock); 2707 audioTrack->mProxy->binderDied(); 2708 } 2709 } 2710 2711 // ========================================================================= 2712 2713 AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava) 2714 : Thread(bCanCallJava), mReceiver(receiver), mPaused(true), mPausedInt(false), mPausedNs(0LL), 2715 mIgnoreNextPausedInt(false) 2716 { 2717 } 2718 2719 AudioTrack::AudioTrackThread::~AudioTrackThread() 2720 { 2721 } 2722 2723 bool AudioTrack::AudioTrackThread::threadLoop() 2724 { 2725 { 2726 AutoMutex _l(mMyLock); 2727 if (mPaused) { 2728 mMyCond.wait(mMyLock); 2729 // caller will check for exitPending() 2730 return true; 2731 } 2732 if (mIgnoreNextPausedInt) { 2733 mIgnoreNextPausedInt = false; 2734 mPausedInt = false; 2735 } 2736 if (mPausedInt) { 2737 if (mPausedNs > 0) { 2738 (void) mMyCond.waitRelative(mMyLock, mPausedNs); 2739 } else { 2740 mMyCond.wait(mMyLock); 2741 } 2742 mPausedInt = false; 2743 return true; 2744 } 2745 } 2746 if (exitPending()) { 2747 return false; 2748 } 2749 nsecs_t ns = mReceiver.processAudioBuffer(); 2750 switch (ns) { 2751 case 0: 2752 return true; 2753 case NS_INACTIVE: 2754 pauseInternal(); 2755 return true; 2756 case NS_NEVER: 2757 return false; 2758 case NS_WHENEVER: 2759 // Event driven: call wake() when callback notifications conditions change. 2760 ns = INT64_MAX; 2761 // fall through 2762 default: 2763 LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns); 2764 pauseInternal(ns); 2765 return true; 2766 } 2767 } 2768 2769 void AudioTrack::AudioTrackThread::requestExit() 2770 { 2771 // must be in this order to avoid a race condition 2772 Thread::requestExit(); 2773 resume(); 2774 } 2775 2776 void AudioTrack::AudioTrackThread::pause() 2777 { 2778 AutoMutex _l(mMyLock); 2779 mPaused = true; 2780 } 2781 2782 void AudioTrack::AudioTrackThread::resume() 2783 { 2784 AutoMutex _l(mMyLock); 2785 mIgnoreNextPausedInt = true; 2786 if (mPaused || mPausedInt) { 2787 mPaused = false; 2788 mPausedInt = false; 2789 mMyCond.signal(); 2790 } 2791 } 2792 2793 void AudioTrack::AudioTrackThread::wake() 2794 { 2795 AutoMutex _l(mMyLock); 2796 if (!mPaused) { 2797 // wake() might be called while servicing a callback - ignore the next 2798 // pause time and call processAudioBuffer. 2799 mIgnoreNextPausedInt = true; 2800 if (mPausedInt && mPausedNs > 0) { 2801 // audio track is active and internally paused with timeout. 2802 mPausedInt = false; 2803 mMyCond.signal(); 2804 } 2805 } 2806 } 2807 2808 void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns) 2809 { 2810 AutoMutex _l(mMyLock); 2811 mPausedInt = true; 2812 mPausedNs = ns; 2813 } 2814 2815 } // namespace android 2816
