Tracks.cpp revision 96f60d8f04432a1ed503b3e24d5736d28c63c9a2
1/* 2** 3** Copyright 2012, 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 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include "Configuration.h" 23#include <math.h> 24#include <utils/Log.h> 25 26#include <private/media/AudioTrackShared.h> 27 28#include <common_time/cc_helper.h> 29#include <common_time/local_clock.h> 30 31#include "AudioMixer.h" 32#include "AudioFlinger.h" 33#include "ServiceUtilities.h" 34 35#include <media/nbaio/Pipe.h> 36#include <media/nbaio/PipeReader.h> 37 38// ---------------------------------------------------------------------------- 39 40// Note: the following macro is used for extremely verbose logging message. In 41// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 42// 0; but one side effect of this is to turn all LOGV's as well. Some messages 43// are so verbose that we want to suppress them even when we have ALOG_ASSERT 44// turned on. Do not uncomment the #def below unless you really know what you 45// are doing and want to see all of the extremely verbose messages. 46//#define VERY_VERY_VERBOSE_LOGGING 47#ifdef VERY_VERY_VERBOSE_LOGGING 48#define ALOGVV ALOGV 49#else 50#define ALOGVV(a...) do { } while(0) 51#endif 52 53namespace android { 54 55// ---------------------------------------------------------------------------- 56// TrackBase 57// ---------------------------------------------------------------------------- 58 59static volatile int32_t nextTrackId = 55; 60 61// TrackBase constructor must be called with AudioFlinger::mLock held 62AudioFlinger::ThreadBase::TrackBase::TrackBase( 63 ThreadBase *thread, 64 const sp<Client>& client, 65 uint32_t sampleRate, 66 audio_format_t format, 67 audio_channel_mask_t channelMask, 68 size_t frameCount, 69 const sp<IMemory>& sharedBuffer, 70 int sessionId, 71 bool isOut) 72 : RefBase(), 73 mThread(thread), 74 mClient(client), 75 mCblk(NULL), 76 // mBuffer 77 mState(IDLE), 78 mSampleRate(sampleRate), 79 mFormat(format), 80 mChannelMask(channelMask), 81 mChannelCount(popcount(channelMask)), 82 mFrameSize(audio_is_linear_pcm(format) ? 83 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 84 mFrameCount(frameCount), 85 mSessionId(sessionId), 86 mIsOut(isOut), 87 mServerProxy(NULL), 88 mId(android_atomic_inc(&nextTrackId)), 89 mTerminated(false) 90{ 91 // client == 0 implies sharedBuffer == 0 92 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 93 94 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 95 sharedBuffer->size()); 96 97 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 98 size_t size = sizeof(audio_track_cblk_t); 99 size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize; 100 if (sharedBuffer == 0) { 101 size += bufferSize; 102 } 103 104 if (client != 0) { 105 mCblkMemory = client->heap()->allocate(size); 106 if (mCblkMemory != 0) { 107 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 108 // can't assume mCblk != NULL 109 } else { 110 ALOGE("not enough memory for AudioTrack size=%u", size); 111 client->heap()->dump("AudioTrack"); 112 return; 113 } 114 } else { 115 // this syntax avoids calling the audio_track_cblk_t constructor twice 116 mCblk = (audio_track_cblk_t *) new uint8_t[size]; 117 // assume mCblk != NULL 118 } 119 120 // construct the shared structure in-place. 121 if (mCblk != NULL) { 122 new(mCblk) audio_track_cblk_t(); 123 // clear all buffers 124 mCblk->frameCount_ = frameCount; 125 if (sharedBuffer == 0) { 126 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 127 memset(mBuffer, 0, bufferSize); 128 } else { 129 mBuffer = sharedBuffer->pointer(); 130#if 0 131 mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic 132#endif 133 } 134 135#ifdef TEE_SINK 136 if (mTeeSinkTrackEnabled) { 137 NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount); 138 if (pipeFormat != Format_Invalid) { 139 Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat); 140 size_t numCounterOffers = 0; 141 const NBAIO_Format offers[1] = {pipeFormat}; 142 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 143 ALOG_ASSERT(index == 0); 144 PipeReader *pipeReader = new PipeReader(*pipe); 145 numCounterOffers = 0; 146 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 147 ALOG_ASSERT(index == 0); 148 mTeeSink = pipe; 149 mTeeSource = pipeReader; 150 } 151 } 152#endif 153 154 } 155} 156 157AudioFlinger::ThreadBase::TrackBase::~TrackBase() 158{ 159#ifdef TEE_SINK 160 dumpTee(-1, mTeeSource, mId); 161#endif 162 // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference 163 delete mServerProxy; 164 if (mCblk != NULL) { 165 if (mClient == 0) { 166 delete mCblk; 167 } else { 168 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 169 } 170 } 171 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 172 if (mClient != 0) { 173 // Client destructor must run with AudioFlinger mutex locked 174 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 175 // If the client's reference count drops to zero, the associated destructor 176 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 177 // relying on the automatic clear() at end of scope. 178 mClient.clear(); 179 } 180} 181 182// AudioBufferProvider interface 183// getNextBuffer() = 0; 184// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 185void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 186{ 187#ifdef TEE_SINK 188 if (mTeeSink != 0) { 189 (void) mTeeSink->write(buffer->raw, buffer->frameCount); 190 } 191#endif 192 193 ServerProxy::Buffer buf; 194 buf.mFrameCount = buffer->frameCount; 195 buf.mRaw = buffer->raw; 196 buffer->frameCount = 0; 197 buffer->raw = NULL; 198 mServerProxy->releaseBuffer(&buf); 199} 200 201status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 202{ 203 mSyncEvents.add(event); 204 return NO_ERROR; 205} 206 207// ---------------------------------------------------------------------------- 208// Playback 209// ---------------------------------------------------------------------------- 210 211AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 212 : BnAudioTrack(), 213 mTrack(track) 214{ 215} 216 217AudioFlinger::TrackHandle::~TrackHandle() { 218 // just stop the track on deletion, associated resources 219 // will be freed from the main thread once all pending buffers have 220 // been played. Unless it's not in the active track list, in which 221 // case we free everything now... 222 mTrack->destroy(); 223} 224 225sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 226 return mTrack->getCblk(); 227} 228 229status_t AudioFlinger::TrackHandle::start() { 230 return mTrack->start(); 231} 232 233void AudioFlinger::TrackHandle::stop() { 234 mTrack->stop(); 235} 236 237void AudioFlinger::TrackHandle::flush() { 238 mTrack->flush(); 239} 240 241void AudioFlinger::TrackHandle::pause() { 242 mTrack->pause(); 243} 244 245status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 246{ 247 return mTrack->attachAuxEffect(EffectId); 248} 249 250status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 251 sp<IMemory>* buffer) { 252 if (!mTrack->isTimedTrack()) 253 return INVALID_OPERATION; 254 255 PlaybackThread::TimedTrack* tt = 256 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 257 return tt->allocateTimedBuffer(size, buffer); 258} 259 260status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 261 int64_t pts) { 262 if (!mTrack->isTimedTrack()) 263 return INVALID_OPERATION; 264 265 PlaybackThread::TimedTrack* tt = 266 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 267 return tt->queueTimedBuffer(buffer, pts); 268} 269 270status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 271 const LinearTransform& xform, int target) { 272 273 if (!mTrack->isTimedTrack()) 274 return INVALID_OPERATION; 275 276 PlaybackThread::TimedTrack* tt = 277 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 278 return tt->setMediaTimeTransform( 279 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 280} 281 282status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) { 283 return mTrack->setParameters(keyValuePairs); 284} 285 286status_t AudioFlinger::TrackHandle::onTransact( 287 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 288{ 289 return BnAudioTrack::onTransact(code, data, reply, flags); 290} 291 292// ---------------------------------------------------------------------------- 293 294// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 295AudioFlinger::PlaybackThread::Track::Track( 296 PlaybackThread *thread, 297 const sp<Client>& client, 298 audio_stream_type_t streamType, 299 uint32_t sampleRate, 300 audio_format_t format, 301 audio_channel_mask_t channelMask, 302 size_t frameCount, 303 const sp<IMemory>& sharedBuffer, 304 int sessionId, 305 IAudioFlinger::track_flags_t flags) 306 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 307 sessionId, true /*isOut*/), 308 mFillingUpStatus(FS_INVALID), 309 // mRetryCount initialized later when needed 310 mSharedBuffer(sharedBuffer), 311 mStreamType(streamType), 312 mName(-1), // see note below 313 mMainBuffer(thread->mixBuffer()), 314 mAuxBuffer(NULL), 315 mAuxEffectId(0), mHasVolumeController(false), 316 mPresentationCompleteFrames(0), 317 mFlags(flags), 318 mFastIndex(-1), 319 mUnderrunCount(0), 320 mCachedVolume(1.0), 321 mIsInvalid(false), 322 mAudioTrackServerProxy(NULL), 323 mResumeToStopping(false) 324{ 325 if (mCblk != NULL) { 326 if (sharedBuffer == 0) { 327 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 328 mFrameSize); 329 } else { 330 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, 331 mFrameSize); 332 } 333 mServerProxy = mAudioTrackServerProxy; 334 // to avoid leaking a track name, do not allocate one unless there is an mCblk 335 mName = thread->getTrackName_l(channelMask, sessionId); 336 mCblk->mName = mName; 337 if (mName < 0) { 338 ALOGE("no more track names available"); 339 return; 340 } 341 // only allocate a fast track index if we were able to allocate a normal track name 342 if (flags & IAudioFlinger::TRACK_FAST) { 343 mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); 344 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 345 int i = __builtin_ctz(thread->mFastTrackAvailMask); 346 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 347 // FIXME This is too eager. We allocate a fast track index before the 348 // fast track becomes active. Since fast tracks are a scarce resource, 349 // this means we are potentially denying other more important fast tracks from 350 // being created. It would be better to allocate the index dynamically. 351 mFastIndex = i; 352 mCblk->mName = i; 353 // Read the initial underruns because this field is never cleared by the fast mixer 354 mObservedUnderruns = thread->getFastTrackUnderruns(i); 355 thread->mFastTrackAvailMask &= ~(1 << i); 356 } 357 } 358 ALOGV("Track constructor name %d, calling pid %d", mName, 359 IPCThreadState::self()->getCallingPid()); 360} 361 362AudioFlinger::PlaybackThread::Track::~Track() 363{ 364 ALOGV("PlaybackThread::Track destructor"); 365} 366 367void AudioFlinger::PlaybackThread::Track::destroy() 368{ 369 // NOTE: destroyTrack_l() can remove a strong reference to this Track 370 // by removing it from mTracks vector, so there is a risk that this Tracks's 371 // destructor is called. As the destructor needs to lock mLock, 372 // we must acquire a strong reference on this Track before locking mLock 373 // here so that the destructor is called only when exiting this function. 374 // On the other hand, as long as Track::destroy() is only called by 375 // TrackHandle destructor, the TrackHandle still holds a strong ref on 376 // this Track with its member mTrack. 377 sp<Track> keep(this); 378 { // scope for mLock 379 sp<ThreadBase> thread = mThread.promote(); 380 if (thread != 0) { 381 Mutex::Autolock _l(thread->mLock); 382 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 383 bool wasActive = playbackThread->destroyTrack_l(this); 384 if (!isOutputTrack() && !wasActive) { 385 AudioSystem::releaseOutput(thread->id()); 386 } 387 } 388 } 389} 390 391/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 392{ 393 result.append(" Name Client Type Fmt Chn mask Session fCount S F SRate " 394 "L dB R dB Server Main buf Aux Buf Flags Underruns\n"); 395} 396 397void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 398{ 399 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 400 if (isFastTrack()) { 401 sprintf(buffer, " F %2d", mFastIndex); 402 } else { 403 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 404 } 405 track_state state = mState; 406 char stateChar; 407 if (isTerminated()) { 408 stateChar = 'T'; 409 } else { 410 switch (state) { 411 case IDLE: 412 stateChar = 'I'; 413 break; 414 case STOPPING_1: 415 stateChar = 's'; 416 break; 417 case STOPPING_2: 418 stateChar = '5'; 419 break; 420 case STOPPED: 421 stateChar = 'S'; 422 break; 423 case RESUMING: 424 stateChar = 'R'; 425 break; 426 case ACTIVE: 427 stateChar = 'A'; 428 break; 429 case PAUSING: 430 stateChar = 'p'; 431 break; 432 case PAUSED: 433 stateChar = 'P'; 434 break; 435 case FLUSHED: 436 stateChar = 'F'; 437 break; 438 default: 439 stateChar = '?'; 440 break; 441 } 442 } 443 char nowInUnderrun; 444 switch (mObservedUnderruns.mBitFields.mMostRecent) { 445 case UNDERRUN_FULL: 446 nowInUnderrun = ' '; 447 break; 448 case UNDERRUN_PARTIAL: 449 nowInUnderrun = '<'; 450 break; 451 case UNDERRUN_EMPTY: 452 nowInUnderrun = '*'; 453 break; 454 default: 455 nowInUnderrun = '?'; 456 break; 457 } 458 snprintf(&buffer[7], size-7, " %6u %4u %3u %08X %7u %6u %1c %1d %5u %5.2g %5.2g " 459 "%08X %08X %08X 0x%03X %9u%c\n", 460 (mClient == 0) ? getpid_cached : mClient->pid(), 461 mStreamType, 462 mFormat, 463 mChannelMask, 464 mSessionId, 465 mFrameCount, 466 stateChar, 467 mFillingUpStatus, 468 mAudioTrackServerProxy->getSampleRate(), 469 20.0 * log10((vlr & 0xFFFF) / 4096.0), 470 20.0 * log10((vlr >> 16) / 4096.0), 471 mCblk->mServer, 472 (int)mMainBuffer, 473 (int)mAuxBuffer, 474 mCblk->mFlags, 475 mUnderrunCount, 476 nowInUnderrun); 477} 478 479uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { 480 return mAudioTrackServerProxy->getSampleRate(); 481} 482 483// AudioBufferProvider interface 484status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 485 AudioBufferProvider::Buffer* buffer, int64_t pts) 486{ 487 ServerProxy::Buffer buf; 488 size_t desiredFrames = buffer->frameCount; 489 buf.mFrameCount = desiredFrames; 490 status_t status = mServerProxy->obtainBuffer(&buf); 491 buffer->frameCount = buf.mFrameCount; 492 buffer->raw = buf.mRaw; 493 if (buf.mFrameCount == 0) { 494 // only implemented so far for normal tracks, not fast tracks 495 mCblk->u.mStreaming.mUnderrunFrames += desiredFrames; 496 // FIXME also wake futex so that underrun is noticed more quickly 497 (void) android_atomic_or(CBLK_UNDERRUN, &mCblk->mFlags); 498 } 499 return status; 500} 501 502// Note that framesReady() takes a mutex on the control block using tryLock(). 503// This could result in priority inversion if framesReady() is called by the normal mixer, 504// as the normal mixer thread runs at lower 505// priority than the client's callback thread: there is a short window within framesReady() 506// during which the normal mixer could be preempted, and the client callback would block. 507// Another problem can occur if framesReady() is called by the fast mixer: 508// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 509// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 510size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 511 return mAudioTrackServerProxy->framesReady(); 512} 513 514// Don't call for fast tracks; the framesReady() could result in priority inversion 515bool AudioFlinger::PlaybackThread::Track::isReady() const { 516 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 517 return true; 518 } 519 520 if (framesReady() >= mFrameCount || 521 (mCblk->mFlags & CBLK_FORCEREADY)) { 522 mFillingUpStatus = FS_FILLED; 523 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 524 return true; 525 } 526 return false; 527} 528 529status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 530 int triggerSession) 531{ 532 status_t status = NO_ERROR; 533 ALOGV("start(%d), calling pid %d session %d", 534 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 535 536 sp<ThreadBase> thread = mThread.promote(); 537 if (thread != 0) { 538 Mutex::Autolock _l(thread->mLock); 539 track_state state = mState; 540 // here the track could be either new, or restarted 541 // in both cases "unstop" the track 542 543 if (state == PAUSED) { 544 if (mResumeToStopping) { 545 // happened we need to resume to STOPPING_1 546 mState = TrackBase::STOPPING_1; 547 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 548 } else { 549 mState = TrackBase::RESUMING; 550 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 551 } 552 } else { 553 mState = TrackBase::ACTIVE; 554 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 555 } 556 557 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 558 status = playbackThread->addTrack_l(this); 559 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 560 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 561 // restore previous state if start was rejected by policy manager 562 if (status == PERMISSION_DENIED) { 563 mState = state; 564 } 565 } 566 // track was already in the active list, not a problem 567 if (status == ALREADY_EXISTS) { 568 status = NO_ERROR; 569 } 570 } else { 571 status = BAD_VALUE; 572 } 573 return status; 574} 575 576void AudioFlinger::PlaybackThread::Track::stop() 577{ 578 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 579 sp<ThreadBase> thread = mThread.promote(); 580 if (thread != 0) { 581 Mutex::Autolock _l(thread->mLock); 582 track_state state = mState; 583 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 584 // If the track is not active (PAUSED and buffers full), flush buffers 585 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 586 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 587 reset(); 588 mState = STOPPED; 589 } else if (!isFastTrack() && !isOffloaded()) { 590 mState = STOPPED; 591 } else { 592 // For fast tracks prepareTracks_l() will set state to STOPPING_2 593 // presentation is complete 594 // For an offloaded track this starts a drain and state will 595 // move to STOPPING_2 when drain completes and then STOPPED 596 mState = STOPPING_1; 597 } 598 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 599 playbackThread); 600 } 601 } 602} 603 604void AudioFlinger::PlaybackThread::Track::pause() 605{ 606 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 607 sp<ThreadBase> thread = mThread.promote(); 608 if (thread != 0) { 609 Mutex::Autolock _l(thread->mLock); 610 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 611 switch (mState) { 612 case STOPPING_1: 613 case STOPPING_2: 614 if (!isOffloaded()) { 615 /* nothing to do if track is not offloaded */ 616 break; 617 } 618 619 // Offloaded track was draining, we need to carry on draining when resumed 620 mResumeToStopping = true; 621 // fall through... 622 case ACTIVE: 623 case RESUMING: 624 mState = PAUSING; 625 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 626 playbackThread->signal_l(); 627 break; 628 629 default: 630 break; 631 } 632 } 633} 634 635void AudioFlinger::PlaybackThread::Track::flush() 636{ 637 ALOGV("flush(%d)", mName); 638 sp<ThreadBase> thread = mThread.promote(); 639 if (thread != 0) { 640 Mutex::Autolock _l(thread->mLock); 641 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 642 643 if (isOffloaded()) { 644 // If offloaded we allow flush during any state except terminated 645 // and keep the track active to avoid problems if user is seeking 646 // rapidly and underlying hardware has a significant delay handling 647 // a pause 648 if (isTerminated()) { 649 return; 650 } 651 652 ALOGV("flush: offload flush"); 653 reset(); 654 655 if (mState == STOPPING_1 || mState == STOPPING_2) { 656 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 657 mState = ACTIVE; 658 } 659 660 if (mState == ACTIVE) { 661 ALOGV("flush called in active state, resetting buffer time out retry count"); 662 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 663 } 664 665 mResumeToStopping = false; 666 } else { 667 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 668 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 669 return; 670 } 671 // No point remaining in PAUSED state after a flush => go to 672 // FLUSHED state 673 mState = FLUSHED; 674 // do not reset the track if it is still in the process of being stopped or paused. 675 // this will be done by prepareTracks_l() when the track is stopped. 676 // prepareTracks_l() will see mState == FLUSHED, then 677 // remove from active track list, reset(), and trigger presentation complete 678 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 679 reset(); 680 } 681 } 682 // Prevent flush being lost if the track is flushed and then resumed 683 // before mixer thread can run. This is important when offloading 684 // because the hardware buffer could hold a large amount of audio 685 playbackThread->flushOutput_l(); 686 playbackThread->signal_l(); 687 } 688} 689 690void AudioFlinger::PlaybackThread::Track::reset() 691{ 692 // Do not reset twice to avoid discarding data written just after a flush and before 693 // the audioflinger thread detects the track is stopped. 694 if (!mResetDone) { 695 // Force underrun condition to avoid false underrun callback until first data is 696 // written to buffer 697 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 698 mFillingUpStatus = FS_FILLING; 699 mResetDone = true; 700 if (mState == FLUSHED) { 701 mState = IDLE; 702 } 703 } 704} 705 706status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 707{ 708 sp<ThreadBase> thread = mThread.promote(); 709 if (thread == 0) { 710 ALOGE("thread is dead"); 711 return FAILED_TRANSACTION; 712 } else if ((thread->type() == ThreadBase::DIRECT) || 713 (thread->type() == ThreadBase::OFFLOAD)) { 714 return thread->setParameters(keyValuePairs); 715 } else { 716 return PERMISSION_DENIED; 717 } 718} 719 720status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 721{ 722 status_t status = DEAD_OBJECT; 723 sp<ThreadBase> thread = mThread.promote(); 724 if (thread != 0) { 725 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 726 sp<AudioFlinger> af = mClient->audioFlinger(); 727 728 Mutex::Autolock _l(af->mLock); 729 730 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 731 732 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 733 Mutex::Autolock _dl(playbackThread->mLock); 734 Mutex::Autolock _sl(srcThread->mLock); 735 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 736 if (chain == 0) { 737 return INVALID_OPERATION; 738 } 739 740 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 741 if (effect == 0) { 742 return INVALID_OPERATION; 743 } 744 srcThread->removeEffect_l(effect); 745 playbackThread->addEffect_l(effect); 746 // removeEffect_l() has stopped the effect if it was active so it must be restarted 747 if (effect->state() == EffectModule::ACTIVE || 748 effect->state() == EffectModule::STOPPING) { 749 effect->start(); 750 } 751 752 sp<EffectChain> dstChain = effect->chain().promote(); 753 if (dstChain == 0) { 754 srcThread->addEffect_l(effect); 755 return INVALID_OPERATION; 756 } 757 AudioSystem::unregisterEffect(effect->id()); 758 AudioSystem::registerEffect(&effect->desc(), 759 srcThread->id(), 760 dstChain->strategy(), 761 AUDIO_SESSION_OUTPUT_MIX, 762 effect->id()); 763 } 764 status = playbackThread->attachAuxEffect(this, EffectId); 765 } 766 return status; 767} 768 769void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 770{ 771 mAuxEffectId = EffectId; 772 mAuxBuffer = buffer; 773} 774 775bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 776 size_t audioHalFrames) 777{ 778 // a track is considered presented when the total number of frames written to audio HAL 779 // corresponds to the number of frames written when presentationComplete() is called for the 780 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 781 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 782 // to detect when all frames have been played. In this case framesWritten isn't 783 // useful because it doesn't always reflect whether there is data in the h/w 784 // buffers, particularly if a track has been paused and resumed during draining 785 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 786 mPresentationCompleteFrames, framesWritten); 787 if (mPresentationCompleteFrames == 0) { 788 mPresentationCompleteFrames = framesWritten + audioHalFrames; 789 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 790 mPresentationCompleteFrames, audioHalFrames); 791 } 792 793 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 794 ALOGV("presentationComplete() session %d complete: framesWritten %d", 795 mSessionId, framesWritten); 796 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 797 mAudioTrackServerProxy->setStreamEndDone(); 798 return true; 799 } 800 return false; 801} 802 803void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 804{ 805 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 806 if (mSyncEvents[i]->type() == type) { 807 mSyncEvents[i]->trigger(); 808 mSyncEvents.removeAt(i); 809 i--; 810 } 811 } 812} 813 814// implement VolumeBufferProvider interface 815 816uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 817{ 818 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 819 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 820 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 821 uint32_t vl = vlr & 0xFFFF; 822 uint32_t vr = vlr >> 16; 823 // track volumes come from shared memory, so can't be trusted and must be clamped 824 if (vl > MAX_GAIN_INT) { 825 vl = MAX_GAIN_INT; 826 } 827 if (vr > MAX_GAIN_INT) { 828 vr = MAX_GAIN_INT; 829 } 830 // now apply the cached master volume and stream type volume; 831 // this is trusted but lacks any synchronization or barrier so may be stale 832 float v = mCachedVolume; 833 vl *= v; 834 vr *= v; 835 // re-combine into U4.16 836 vlr = (vr << 16) | (vl & 0xFFFF); 837 // FIXME look at mute, pause, and stop flags 838 return vlr; 839} 840 841status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 842{ 843 if (isTerminated() || mState == PAUSED || 844 ((framesReady() == 0) && ((mSharedBuffer != 0) || 845 (mState == STOPPED)))) { 846 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 847 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 848 event->cancel(); 849 return INVALID_OPERATION; 850 } 851 (void) TrackBase::setSyncEvent(event); 852 return NO_ERROR; 853} 854 855void AudioFlinger::PlaybackThread::Track::invalidate() 856{ 857 // FIXME should use proxy, and needs work 858 audio_track_cblk_t* cblk = mCblk; 859 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 860 android_atomic_release_store(0x40000000, &cblk->mFutex); 861 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 862 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 863 mIsInvalid = true; 864} 865 866// ---------------------------------------------------------------------------- 867 868sp<AudioFlinger::PlaybackThread::TimedTrack> 869AudioFlinger::PlaybackThread::TimedTrack::create( 870 PlaybackThread *thread, 871 const sp<Client>& client, 872 audio_stream_type_t streamType, 873 uint32_t sampleRate, 874 audio_format_t format, 875 audio_channel_mask_t channelMask, 876 size_t frameCount, 877 const sp<IMemory>& sharedBuffer, 878 int sessionId) { 879 if (!client->reserveTimedTrack()) 880 return 0; 881 882 return new TimedTrack( 883 thread, client, streamType, sampleRate, format, channelMask, frameCount, 884 sharedBuffer, sessionId); 885} 886 887AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 888 PlaybackThread *thread, 889 const sp<Client>& client, 890 audio_stream_type_t streamType, 891 uint32_t sampleRate, 892 audio_format_t format, 893 audio_channel_mask_t channelMask, 894 size_t frameCount, 895 const sp<IMemory>& sharedBuffer, 896 int sessionId) 897 : Track(thread, client, streamType, sampleRate, format, channelMask, 898 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 899 mQueueHeadInFlight(false), 900 mTrimQueueHeadOnRelease(false), 901 mFramesPendingInQueue(0), 902 mTimedSilenceBuffer(NULL), 903 mTimedSilenceBufferSize(0), 904 mTimedAudioOutputOnTime(false), 905 mMediaTimeTransformValid(false) 906{ 907 LocalClock lc; 908 mLocalTimeFreq = lc.getLocalFreq(); 909 910 mLocalTimeToSampleTransform.a_zero = 0; 911 mLocalTimeToSampleTransform.b_zero = 0; 912 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 913 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 914 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 915 &mLocalTimeToSampleTransform.a_to_b_denom); 916 917 mMediaTimeToSampleTransform.a_zero = 0; 918 mMediaTimeToSampleTransform.b_zero = 0; 919 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 920 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 921 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 922 &mMediaTimeToSampleTransform.a_to_b_denom); 923} 924 925AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 926 mClient->releaseTimedTrack(); 927 delete [] mTimedSilenceBuffer; 928} 929 930status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 931 size_t size, sp<IMemory>* buffer) { 932 933 Mutex::Autolock _l(mTimedBufferQueueLock); 934 935 trimTimedBufferQueue_l(); 936 937 // lazily initialize the shared memory heap for timed buffers 938 if (mTimedMemoryDealer == NULL) { 939 const int kTimedBufferHeapSize = 512 << 10; 940 941 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 942 "AudioFlingerTimed"); 943 if (mTimedMemoryDealer == NULL) 944 return NO_MEMORY; 945 } 946 947 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 948 if (newBuffer == NULL) { 949 newBuffer = mTimedMemoryDealer->allocate(size); 950 if (newBuffer == NULL) 951 return NO_MEMORY; 952 } 953 954 *buffer = newBuffer; 955 return NO_ERROR; 956} 957 958// caller must hold mTimedBufferQueueLock 959void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 960 int64_t mediaTimeNow; 961 { 962 Mutex::Autolock mttLock(mMediaTimeTransformLock); 963 if (!mMediaTimeTransformValid) 964 return; 965 966 int64_t targetTimeNow; 967 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 968 ? mCCHelper.getCommonTime(&targetTimeNow) 969 : mCCHelper.getLocalTime(&targetTimeNow); 970 971 if (OK != res) 972 return; 973 974 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 975 &mediaTimeNow)) { 976 return; 977 } 978 } 979 980 size_t trimEnd; 981 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 982 int64_t bufEnd; 983 984 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 985 // We have a next buffer. Just use its PTS as the PTS of the frame 986 // following the last frame in this buffer. If the stream is sparse 987 // (ie, there are deliberate gaps left in the stream which should be 988 // filled with silence by the TimedAudioTrack), then this can result 989 // in one extra buffer being left un-trimmed when it could have 990 // been. In general, this is not typical, and we would rather 991 // optimized away the TS calculation below for the more common case 992 // where PTSes are contiguous. 993 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 994 } else { 995 // We have no next buffer. Compute the PTS of the frame following 996 // the last frame in this buffer by computing the duration of of 997 // this frame in media time units and adding it to the PTS of the 998 // buffer. 999 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1000 / mFrameSize; 1001 1002 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1003 &bufEnd)) { 1004 ALOGE("Failed to convert frame count of %lld to media time" 1005 " duration" " (scale factor %d/%u) in %s", 1006 frameCount, 1007 mMediaTimeToSampleTransform.a_to_b_numer, 1008 mMediaTimeToSampleTransform.a_to_b_denom, 1009 __PRETTY_FUNCTION__); 1010 break; 1011 } 1012 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1013 } 1014 1015 if (bufEnd > mediaTimeNow) 1016 break; 1017 1018 // Is the buffer we want to use in the middle of a mix operation right 1019 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1020 // from the mixer which should be coming back shortly. 1021 if (!trimEnd && mQueueHeadInFlight) { 1022 mTrimQueueHeadOnRelease = true; 1023 } 1024 } 1025 1026 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1027 if (trimStart < trimEnd) { 1028 // Update the bookkeeping for framesReady() 1029 for (size_t i = trimStart; i < trimEnd; ++i) { 1030 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1031 } 1032 1033 // Now actually remove the buffers from the queue. 1034 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1035 } 1036} 1037 1038void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1039 const char* logTag) { 1040 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1041 "%s called (reason \"%s\"), but timed buffer queue has no" 1042 " elements to trim.", __FUNCTION__, logTag); 1043 1044 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1045 mTimedBufferQueue.removeAt(0); 1046} 1047 1048void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1049 const TimedBuffer& buf, 1050 const char* logTag) { 1051 uint32_t bufBytes = buf.buffer()->size(); 1052 uint32_t consumedAlready = buf.position(); 1053 1054 ALOG_ASSERT(consumedAlready <= bufBytes, 1055 "Bad bookkeeping while updating frames pending. Timed buffer is" 1056 " only %u bytes long, but claims to have consumed %u" 1057 " bytes. (update reason: \"%s\")", 1058 bufBytes, consumedAlready, logTag); 1059 1060 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1061 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1062 "Bad bookkeeping while updating frames pending. Should have at" 1063 " least %u queued frames, but we think we have only %u. (update" 1064 " reason: \"%s\")", 1065 bufFrames, mFramesPendingInQueue, logTag); 1066 1067 mFramesPendingInQueue -= bufFrames; 1068} 1069 1070status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1071 const sp<IMemory>& buffer, int64_t pts) { 1072 1073 { 1074 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1075 if (!mMediaTimeTransformValid) 1076 return INVALID_OPERATION; 1077 } 1078 1079 Mutex::Autolock _l(mTimedBufferQueueLock); 1080 1081 uint32_t bufFrames = buffer->size() / mFrameSize; 1082 mFramesPendingInQueue += bufFrames; 1083 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1084 1085 return NO_ERROR; 1086} 1087 1088status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1089 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1090 1091 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1092 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1093 target); 1094 1095 if (!(target == TimedAudioTrack::LOCAL_TIME || 1096 target == TimedAudioTrack::COMMON_TIME)) { 1097 return BAD_VALUE; 1098 } 1099 1100 Mutex::Autolock lock(mMediaTimeTransformLock); 1101 mMediaTimeTransform = xform; 1102 mMediaTimeTransformTarget = target; 1103 mMediaTimeTransformValid = true; 1104 1105 return NO_ERROR; 1106} 1107 1108#define min(a, b) ((a) < (b) ? (a) : (b)) 1109 1110// implementation of getNextBuffer for tracks whose buffers have timestamps 1111status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1112 AudioBufferProvider::Buffer* buffer, int64_t pts) 1113{ 1114 if (pts == AudioBufferProvider::kInvalidPTS) { 1115 buffer->raw = NULL; 1116 buffer->frameCount = 0; 1117 mTimedAudioOutputOnTime = false; 1118 return INVALID_OPERATION; 1119 } 1120 1121 Mutex::Autolock _l(mTimedBufferQueueLock); 1122 1123 ALOG_ASSERT(!mQueueHeadInFlight, 1124 "getNextBuffer called without releaseBuffer!"); 1125 1126 while (true) { 1127 1128 // if we have no timed buffers, then fail 1129 if (mTimedBufferQueue.isEmpty()) { 1130 buffer->raw = NULL; 1131 buffer->frameCount = 0; 1132 return NOT_ENOUGH_DATA; 1133 } 1134 1135 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1136 1137 // calculate the PTS of the head of the timed buffer queue expressed in 1138 // local time 1139 int64_t headLocalPTS; 1140 { 1141 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1142 1143 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1144 1145 if (mMediaTimeTransform.a_to_b_denom == 0) { 1146 // the transform represents a pause, so yield silence 1147 timedYieldSilence_l(buffer->frameCount, buffer); 1148 return NO_ERROR; 1149 } 1150 1151 int64_t transformedPTS; 1152 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1153 &transformedPTS)) { 1154 // the transform failed. this shouldn't happen, but if it does 1155 // then just drop this buffer 1156 ALOGW("timedGetNextBuffer transform failed"); 1157 buffer->raw = NULL; 1158 buffer->frameCount = 0; 1159 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1160 return NO_ERROR; 1161 } 1162 1163 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1164 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1165 &headLocalPTS)) { 1166 buffer->raw = NULL; 1167 buffer->frameCount = 0; 1168 return INVALID_OPERATION; 1169 } 1170 } else { 1171 headLocalPTS = transformedPTS; 1172 } 1173 } 1174 1175 uint32_t sr = sampleRate(); 1176 1177 // adjust the head buffer's PTS to reflect the portion of the head buffer 1178 // that has already been consumed 1179 int64_t effectivePTS = headLocalPTS + 1180 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1181 1182 // Calculate the delta in samples between the head of the input buffer 1183 // queue and the start of the next output buffer that will be written. 1184 // If the transformation fails because of over or underflow, it means 1185 // that the sample's position in the output stream is so far out of 1186 // whack that it should just be dropped. 1187 int64_t sampleDelta; 1188 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1189 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1190 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1191 " mix"); 1192 continue; 1193 } 1194 if (!mLocalTimeToSampleTransform.doForwardTransform( 1195 (effectivePTS - pts) << 32, &sampleDelta)) { 1196 ALOGV("*** too late during sample rate transform: dropped buffer"); 1197 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1198 continue; 1199 } 1200 1201 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1202 " sampleDelta=[%d.%08x]", 1203 head.pts(), head.position(), pts, 1204 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1205 + (sampleDelta >> 32)), 1206 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1207 1208 // if the delta between the ideal placement for the next input sample and 1209 // the current output position is within this threshold, then we will 1210 // concatenate the next input samples to the previous output 1211 const int64_t kSampleContinuityThreshold = 1212 (static_cast<int64_t>(sr) << 32) / 250; 1213 1214 // if this is the first buffer of audio that we're emitting from this track 1215 // then it should be almost exactly on time. 1216 const int64_t kSampleStartupThreshold = 1LL << 32; 1217 1218 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1219 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1220 // the next input is close enough to being on time, so concatenate it 1221 // with the last output 1222 timedYieldSamples_l(buffer); 1223 1224 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1225 head.position(), buffer->frameCount); 1226 return NO_ERROR; 1227 } 1228 1229 // Looks like our output is not on time. Reset our on timed status. 1230 // Next time we mix samples from our input queue, then should be within 1231 // the StartupThreshold. 1232 mTimedAudioOutputOnTime = false; 1233 if (sampleDelta > 0) { 1234 // the gap between the current output position and the proper start of 1235 // the next input sample is too big, so fill it with silence 1236 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1237 1238 timedYieldSilence_l(framesUntilNextInput, buffer); 1239 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1240 return NO_ERROR; 1241 } else { 1242 // the next input sample is late 1243 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1244 size_t onTimeSamplePosition = 1245 head.position() + lateFrames * mFrameSize; 1246 1247 if (onTimeSamplePosition > head.buffer()->size()) { 1248 // all the remaining samples in the head are too late, so 1249 // drop it and move on 1250 ALOGV("*** too late: dropped buffer"); 1251 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1252 continue; 1253 } else { 1254 // skip over the late samples 1255 head.setPosition(onTimeSamplePosition); 1256 1257 // yield the available samples 1258 timedYieldSamples_l(buffer); 1259 1260 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1261 return NO_ERROR; 1262 } 1263 } 1264 } 1265} 1266 1267// Yield samples from the timed buffer queue head up to the given output 1268// buffer's capacity. 1269// 1270// Caller must hold mTimedBufferQueueLock 1271void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1272 AudioBufferProvider::Buffer* buffer) { 1273 1274 const TimedBuffer& head = mTimedBufferQueue[0]; 1275 1276 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1277 head.position()); 1278 1279 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1280 mFrameSize); 1281 size_t framesRequested = buffer->frameCount; 1282 buffer->frameCount = min(framesLeftInHead, framesRequested); 1283 1284 mQueueHeadInFlight = true; 1285 mTimedAudioOutputOnTime = true; 1286} 1287 1288// Yield samples of silence up to the given output buffer's capacity 1289// 1290// Caller must hold mTimedBufferQueueLock 1291void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1292 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1293 1294 // lazily allocate a buffer filled with silence 1295 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1296 delete [] mTimedSilenceBuffer; 1297 mTimedSilenceBufferSize = numFrames * mFrameSize; 1298 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1299 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1300 } 1301 1302 buffer->raw = mTimedSilenceBuffer; 1303 size_t framesRequested = buffer->frameCount; 1304 buffer->frameCount = min(numFrames, framesRequested); 1305 1306 mTimedAudioOutputOnTime = false; 1307} 1308 1309// AudioBufferProvider interface 1310void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1311 AudioBufferProvider::Buffer* buffer) { 1312 1313 Mutex::Autolock _l(mTimedBufferQueueLock); 1314 1315 // If the buffer which was just released is part of the buffer at the head 1316 // of the queue, be sure to update the amt of the buffer which has been 1317 // consumed. If the buffer being returned is not part of the head of the 1318 // queue, its either because the buffer is part of the silence buffer, or 1319 // because the head of the timed queue was trimmed after the mixer called 1320 // getNextBuffer but before the mixer called releaseBuffer. 1321 if (buffer->raw == mTimedSilenceBuffer) { 1322 ALOG_ASSERT(!mQueueHeadInFlight, 1323 "Queue head in flight during release of silence buffer!"); 1324 goto done; 1325 } 1326 1327 ALOG_ASSERT(mQueueHeadInFlight, 1328 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1329 " head in flight."); 1330 1331 if (mTimedBufferQueue.size()) { 1332 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1333 1334 void* start = head.buffer()->pointer(); 1335 void* end = reinterpret_cast<void*>( 1336 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1337 + head.buffer()->size()); 1338 1339 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1340 "released buffer not within the head of the timed buffer" 1341 " queue; qHead = [%p, %p], released buffer = %p", 1342 start, end, buffer->raw); 1343 1344 head.setPosition(head.position() + 1345 (buffer->frameCount * mFrameSize)); 1346 mQueueHeadInFlight = false; 1347 1348 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1349 "Bad bookkeeping during releaseBuffer! Should have at" 1350 " least %u queued frames, but we think we have only %u", 1351 buffer->frameCount, mFramesPendingInQueue); 1352 1353 mFramesPendingInQueue -= buffer->frameCount; 1354 1355 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1356 || mTrimQueueHeadOnRelease) { 1357 trimTimedBufferQueueHead_l("releaseBuffer"); 1358 mTrimQueueHeadOnRelease = false; 1359 } 1360 } else { 1361 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1362 " buffers in the timed buffer queue"); 1363 } 1364 1365done: 1366 buffer->raw = 0; 1367 buffer->frameCount = 0; 1368} 1369 1370size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1371 Mutex::Autolock _l(mTimedBufferQueueLock); 1372 return mFramesPendingInQueue; 1373} 1374 1375AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1376 : mPTS(0), mPosition(0) {} 1377 1378AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1379 const sp<IMemory>& buffer, int64_t pts) 1380 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1381 1382 1383// ---------------------------------------------------------------------------- 1384 1385AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1386 PlaybackThread *playbackThread, 1387 DuplicatingThread *sourceThread, 1388 uint32_t sampleRate, 1389 audio_format_t format, 1390 audio_channel_mask_t channelMask, 1391 size_t frameCount) 1392 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1393 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 1394 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1395{ 1396 1397 if (mCblk != NULL) { 1398 mOutBuffer.frameCount = 0; 1399 playbackThread->mTracks.add(this); 1400 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1401 "mCblk->frameCount_ %u, mChannelMask 0x%08x", 1402 mCblk, mBuffer, 1403 mCblk->frameCount_, mChannelMask); 1404 // since client and server are in the same process, 1405 // the buffer has the same virtual address on both sides 1406 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize); 1407 mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000)); 1408 mClientProxy->setSendLevel(0.0); 1409 mClientProxy->setSampleRate(sampleRate); 1410 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1411 true /*clientInServer*/); 1412 } else { 1413 ALOGW("Error creating output track on thread %p", playbackThread); 1414 } 1415} 1416 1417AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1418{ 1419 clearBufferQueue(); 1420 delete mClientProxy; 1421 // superclass destructor will now delete the server proxy and shared memory both refer to 1422} 1423 1424status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1425 int triggerSession) 1426{ 1427 status_t status = Track::start(event, triggerSession); 1428 if (status != NO_ERROR) { 1429 return status; 1430 } 1431 1432 mActive = true; 1433 mRetryCount = 127; 1434 return status; 1435} 1436 1437void AudioFlinger::PlaybackThread::OutputTrack::stop() 1438{ 1439 Track::stop(); 1440 clearBufferQueue(); 1441 mOutBuffer.frameCount = 0; 1442 mActive = false; 1443} 1444 1445bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1446{ 1447 Buffer *pInBuffer; 1448 Buffer inBuffer; 1449 uint32_t channelCount = mChannelCount; 1450 bool outputBufferFull = false; 1451 inBuffer.frameCount = frames; 1452 inBuffer.i16 = data; 1453 1454 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1455 1456 if (!mActive && frames != 0) { 1457 start(); 1458 sp<ThreadBase> thread = mThread.promote(); 1459 if (thread != 0) { 1460 MixerThread *mixerThread = (MixerThread *)thread.get(); 1461 if (mFrameCount > frames) { 1462 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1463 uint32_t startFrames = (mFrameCount - frames); 1464 pInBuffer = new Buffer; 1465 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1466 pInBuffer->frameCount = startFrames; 1467 pInBuffer->i16 = pInBuffer->mBuffer; 1468 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1469 mBufferQueue.add(pInBuffer); 1470 } else { 1471 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1472 } 1473 } 1474 } 1475 } 1476 1477 while (waitTimeLeftMs) { 1478 // First write pending buffers, then new data 1479 if (mBufferQueue.size()) { 1480 pInBuffer = mBufferQueue.itemAt(0); 1481 } else { 1482 pInBuffer = &inBuffer; 1483 } 1484 1485 if (pInBuffer->frameCount == 0) { 1486 break; 1487 } 1488 1489 if (mOutBuffer.frameCount == 0) { 1490 mOutBuffer.frameCount = pInBuffer->frameCount; 1491 nsecs_t startTime = systemTime(); 1492 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1493 if (status != NO_ERROR) { 1494 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1495 mThread.unsafe_get(), status); 1496 outputBufferFull = true; 1497 break; 1498 } 1499 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1500 if (waitTimeLeftMs >= waitTimeMs) { 1501 waitTimeLeftMs -= waitTimeMs; 1502 } else { 1503 waitTimeLeftMs = 0; 1504 } 1505 } 1506 1507 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1508 pInBuffer->frameCount; 1509 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1510 Proxy::Buffer buf; 1511 buf.mFrameCount = outFrames; 1512 buf.mRaw = NULL; 1513 mClientProxy->releaseBuffer(&buf); 1514 pInBuffer->frameCount -= outFrames; 1515 pInBuffer->i16 += outFrames * channelCount; 1516 mOutBuffer.frameCount -= outFrames; 1517 mOutBuffer.i16 += outFrames * channelCount; 1518 1519 if (pInBuffer->frameCount == 0) { 1520 if (mBufferQueue.size()) { 1521 mBufferQueue.removeAt(0); 1522 delete [] pInBuffer->mBuffer; 1523 delete pInBuffer; 1524 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1525 mThread.unsafe_get(), mBufferQueue.size()); 1526 } else { 1527 break; 1528 } 1529 } 1530 } 1531 1532 // If we could not write all frames, allocate a buffer and queue it for next time. 1533 if (inBuffer.frameCount) { 1534 sp<ThreadBase> thread = mThread.promote(); 1535 if (thread != 0 && !thread->standby()) { 1536 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1537 pInBuffer = new Buffer; 1538 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1539 pInBuffer->frameCount = inBuffer.frameCount; 1540 pInBuffer->i16 = pInBuffer->mBuffer; 1541 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1542 sizeof(int16_t)); 1543 mBufferQueue.add(pInBuffer); 1544 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1545 mThread.unsafe_get(), mBufferQueue.size()); 1546 } else { 1547 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1548 mThread.unsafe_get(), this); 1549 } 1550 } 1551 } 1552 1553 // Calling write() with a 0 length buffer, means that no more data will be written: 1554 // If no more buffers are pending, fill output track buffer to make sure it is started 1555 // by output mixer. 1556 if (frames == 0 && mBufferQueue.size() == 0) { 1557 // FIXME borken, replace by getting framesReady() from proxy 1558 size_t user = 0; // was mCblk->user 1559 if (user < mFrameCount) { 1560 frames = mFrameCount - user; 1561 pInBuffer = new Buffer; 1562 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1563 pInBuffer->frameCount = frames; 1564 pInBuffer->i16 = pInBuffer->mBuffer; 1565 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1566 mBufferQueue.add(pInBuffer); 1567 } else if (mActive) { 1568 stop(); 1569 } 1570 } 1571 1572 return outputBufferFull; 1573} 1574 1575status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1576 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1577{ 1578 ClientProxy::Buffer buf; 1579 buf.mFrameCount = buffer->frameCount; 1580 struct timespec timeout; 1581 timeout.tv_sec = waitTimeMs / 1000; 1582 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1583 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1584 buffer->frameCount = buf.mFrameCount; 1585 buffer->raw = buf.mRaw; 1586 return status; 1587} 1588 1589void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1590{ 1591 size_t size = mBufferQueue.size(); 1592 1593 for (size_t i = 0; i < size; i++) { 1594 Buffer *pBuffer = mBufferQueue.itemAt(i); 1595 delete [] pBuffer->mBuffer; 1596 delete pBuffer; 1597 } 1598 mBufferQueue.clear(); 1599} 1600 1601 1602// ---------------------------------------------------------------------------- 1603// Record 1604// ---------------------------------------------------------------------------- 1605 1606AudioFlinger::RecordHandle::RecordHandle( 1607 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1608 : BnAudioRecord(), 1609 mRecordTrack(recordTrack) 1610{ 1611} 1612 1613AudioFlinger::RecordHandle::~RecordHandle() { 1614 stop_nonvirtual(); 1615 mRecordTrack->destroy(); 1616} 1617 1618sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 1619 return mRecordTrack->getCblk(); 1620} 1621 1622status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1623 int triggerSession) { 1624 ALOGV("RecordHandle::start()"); 1625 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1626} 1627 1628void AudioFlinger::RecordHandle::stop() { 1629 stop_nonvirtual(); 1630} 1631 1632void AudioFlinger::RecordHandle::stop_nonvirtual() { 1633 ALOGV("RecordHandle::stop()"); 1634 mRecordTrack->stop(); 1635} 1636 1637status_t AudioFlinger::RecordHandle::onTransact( 1638 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1639{ 1640 return BnAudioRecord::onTransact(code, data, reply, flags); 1641} 1642 1643// ---------------------------------------------------------------------------- 1644 1645// RecordTrack constructor must be called with AudioFlinger::mLock held 1646AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1647 RecordThread *thread, 1648 const sp<Client>& client, 1649 uint32_t sampleRate, 1650 audio_format_t format, 1651 audio_channel_mask_t channelMask, 1652 size_t frameCount, 1653 int sessionId) 1654 : TrackBase(thread, client, sampleRate, format, 1655 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/), 1656 mOverflow(false) 1657{ 1658 ALOGV("RecordTrack constructor"); 1659 if (mCblk != NULL) { 1660 mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, 1661 mFrameSize); 1662 mServerProxy = mAudioRecordServerProxy; 1663 } 1664} 1665 1666AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 1667{ 1668 ALOGV("%s", __func__); 1669} 1670 1671// AudioBufferProvider interface 1672status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 1673 int64_t pts) 1674{ 1675 ServerProxy::Buffer buf; 1676 buf.mFrameCount = buffer->frameCount; 1677 status_t status = mServerProxy->obtainBuffer(&buf); 1678 buffer->frameCount = buf.mFrameCount; 1679 buffer->raw = buf.mRaw; 1680 if (buf.mFrameCount == 0) { 1681 // FIXME also wake futex so that overrun is noticed more quickly 1682 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 1683 } 1684 return status; 1685} 1686 1687status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 1688 int triggerSession) 1689{ 1690 sp<ThreadBase> thread = mThread.promote(); 1691 if (thread != 0) { 1692 RecordThread *recordThread = (RecordThread *)thread.get(); 1693 return recordThread->start(this, event, triggerSession); 1694 } else { 1695 return BAD_VALUE; 1696 } 1697} 1698 1699void AudioFlinger::RecordThread::RecordTrack::stop() 1700{ 1701 sp<ThreadBase> thread = mThread.promote(); 1702 if (thread != 0) { 1703 RecordThread *recordThread = (RecordThread *)thread.get(); 1704 if (recordThread->stop(this)) { 1705 AudioSystem::stopInput(recordThread->id()); 1706 } 1707 } 1708} 1709 1710void AudioFlinger::RecordThread::RecordTrack::destroy() 1711{ 1712 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 1713 sp<RecordTrack> keep(this); 1714 { 1715 sp<ThreadBase> thread = mThread.promote(); 1716 if (thread != 0) { 1717 if (mState == ACTIVE || mState == RESUMING) { 1718 AudioSystem::stopInput(thread->id()); 1719 } 1720 AudioSystem::releaseInput(thread->id()); 1721 Mutex::Autolock _l(thread->mLock); 1722 RecordThread *recordThread = (RecordThread *) thread.get(); 1723 recordThread->destroyTrack_l(this); 1724 } 1725 } 1726} 1727 1728 1729/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 1730{ 1731 result.append("Client Fmt Chn mask Session S Server fCount\n"); 1732} 1733 1734void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 1735{ 1736 snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6u\n", 1737 (mClient == 0) ? getpid_cached : mClient->pid(), 1738 mFormat, 1739 mChannelMask, 1740 mSessionId, 1741 mState, 1742 mCblk->mServer, 1743 mFrameCount); 1744} 1745 1746}; // namespace android 1747