Tracks.cpp revision 6ae6b811666865815ebb1f670aacb1a0f2edaa73
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 mCachedVolume(1.0), 320 mIsInvalid(false), 321 mAudioTrackServerProxy(NULL), 322 mResumeToStopping(false) 323{ 324 if (mCblk != NULL) { 325 if (sharedBuffer == 0) { 326 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 327 mFrameSize); 328 } else { 329 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, 330 mFrameSize); 331 } 332 mServerProxy = mAudioTrackServerProxy; 333 // to avoid leaking a track name, do not allocate one unless there is an mCblk 334 mName = thread->getTrackName_l(channelMask, sessionId); 335 if (mName < 0) { 336 ALOGE("no more track names available"); 337 return; 338 } 339 // only allocate a fast track index if we were able to allocate a normal track name 340 if (flags & IAudioFlinger::TRACK_FAST) { 341 mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); 342 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 343 int i = __builtin_ctz(thread->mFastTrackAvailMask); 344 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 345 // FIXME This is too eager. We allocate a fast track index before the 346 // fast track becomes active. Since fast tracks are a scarce resource, 347 // this means we are potentially denying other more important fast tracks from 348 // being created. It would be better to allocate the index dynamically. 349 mFastIndex = i; 350 // Read the initial underruns because this field is never cleared by the fast mixer 351 mObservedUnderruns = thread->getFastTrackUnderruns(i); 352 thread->mFastTrackAvailMask &= ~(1 << i); 353 } 354 } 355 ALOGV("Track constructor name %d, calling pid %d", mName, 356 IPCThreadState::self()->getCallingPid()); 357} 358 359AudioFlinger::PlaybackThread::Track::~Track() 360{ 361 ALOGV("PlaybackThread::Track destructor"); 362} 363 364void AudioFlinger::PlaybackThread::Track::destroy() 365{ 366 // NOTE: destroyTrack_l() can remove a strong reference to this Track 367 // by removing it from mTracks vector, so there is a risk that this Tracks's 368 // destructor is called. As the destructor needs to lock mLock, 369 // we must acquire a strong reference on this Track before locking mLock 370 // here so that the destructor is called only when exiting this function. 371 // On the other hand, as long as Track::destroy() is only called by 372 // TrackHandle destructor, the TrackHandle still holds a strong ref on 373 // this Track with its member mTrack. 374 sp<Track> keep(this); 375 { // scope for mLock 376 sp<ThreadBase> thread = mThread.promote(); 377 if (thread != 0) { 378 Mutex::Autolock _l(thread->mLock); 379 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 380 bool wasActive = playbackThread->destroyTrack_l(this); 381 if (!isOutputTrack() && !wasActive) { 382 AudioSystem::releaseOutput(thread->id()); 383 } 384 } 385 } 386} 387 388/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 389{ 390 result.append(" Name Client Type Fmt Chn mask Session fCount S F SRate " 391 "L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n"); 392} 393 394void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 395{ 396 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 397 if (isFastTrack()) { 398 sprintf(buffer, " F %2d", mFastIndex); 399 } else { 400 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 401 } 402 track_state state = mState; 403 char stateChar; 404 if (isTerminated()) { 405 stateChar = 'T'; 406 } else { 407 switch (state) { 408 case IDLE: 409 stateChar = 'I'; 410 break; 411 case STOPPING_1: 412 stateChar = 's'; 413 break; 414 case STOPPING_2: 415 stateChar = '5'; 416 break; 417 case STOPPED: 418 stateChar = 'S'; 419 break; 420 case RESUMING: 421 stateChar = 'R'; 422 break; 423 case ACTIVE: 424 stateChar = 'A'; 425 break; 426 case PAUSING: 427 stateChar = 'p'; 428 break; 429 case PAUSED: 430 stateChar = 'P'; 431 break; 432 case FLUSHED: 433 stateChar = 'F'; 434 break; 435 default: 436 stateChar = '?'; 437 break; 438 } 439 } 440 char nowInUnderrun; 441 switch (mObservedUnderruns.mBitFields.mMostRecent) { 442 case UNDERRUN_FULL: 443 nowInUnderrun = ' '; 444 break; 445 case UNDERRUN_PARTIAL: 446 nowInUnderrun = '<'; 447 break; 448 case UNDERRUN_EMPTY: 449 nowInUnderrun = '*'; 450 break; 451 default: 452 nowInUnderrun = '?'; 453 break; 454 } 455 snprintf(&buffer[7], size-7, " %6u %4u %3u %08X %7u %6u %1c %1d %5u %5.2g %5.2g " 456 "%08X %08X %08X 0x%03X %9u%c\n", 457 (mClient == 0) ? getpid_cached : mClient->pid(), 458 mStreamType, 459 mFormat, 460 mChannelMask, 461 mSessionId, 462 mFrameCount, 463 stateChar, 464 mFillingUpStatus, 465 mAudioTrackServerProxy->getSampleRate(), 466 20.0 * log10((vlr & 0xFFFF) / 4096.0), 467 20.0 * log10((vlr >> 16) / 4096.0), 468 mCblk->mServer, 469 (int)mMainBuffer, 470 (int)mAuxBuffer, 471 mCblk->mFlags, 472 mAudioTrackServerProxy->getUnderrunFrames(), 473 nowInUnderrun); 474} 475 476uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { 477 return mAudioTrackServerProxy->getSampleRate(); 478} 479 480// AudioBufferProvider interface 481status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 482 AudioBufferProvider::Buffer* buffer, int64_t pts) 483{ 484 ServerProxy::Buffer buf; 485 size_t desiredFrames = buffer->frameCount; 486 buf.mFrameCount = desiredFrames; 487 status_t status = mServerProxy->obtainBuffer(&buf); 488 buffer->frameCount = buf.mFrameCount; 489 buffer->raw = buf.mRaw; 490 if (buf.mFrameCount == 0) { 491 mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 492 } 493 return status; 494} 495 496// Note that framesReady() takes a mutex on the control block using tryLock(). 497// This could result in priority inversion if framesReady() is called by the normal mixer, 498// as the normal mixer thread runs at lower 499// priority than the client's callback thread: there is a short window within framesReady() 500// during which the normal mixer could be preempted, and the client callback would block. 501// Another problem can occur if framesReady() is called by the fast mixer: 502// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 503// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 504size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 505 return mAudioTrackServerProxy->framesReady(); 506} 507 508// Don't call for fast tracks; the framesReady() could result in priority inversion 509bool AudioFlinger::PlaybackThread::Track::isReady() const { 510 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 511 return true; 512 } 513 514 if (framesReady() >= mFrameCount || 515 (mCblk->mFlags & CBLK_FORCEREADY)) { 516 mFillingUpStatus = FS_FILLED; 517 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 518 return true; 519 } 520 return false; 521} 522 523status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 524 int triggerSession) 525{ 526 status_t status = NO_ERROR; 527 ALOGV("start(%d), calling pid %d session %d", 528 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 529 530 sp<ThreadBase> thread = mThread.promote(); 531 if (thread != 0) { 532 Mutex::Autolock _l(thread->mLock); 533 track_state state = mState; 534 // here the track could be either new, or restarted 535 // in both cases "unstop" the track 536 537 if (state == PAUSED) { 538 if (mResumeToStopping) { 539 // happened we need to resume to STOPPING_1 540 mState = TrackBase::STOPPING_1; 541 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 542 } else { 543 mState = TrackBase::RESUMING; 544 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 545 } 546 } else { 547 mState = TrackBase::ACTIVE; 548 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 549 } 550 551 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 552 status = playbackThread->addTrack_l(this); 553 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 554 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 555 // restore previous state if start was rejected by policy manager 556 if (status == PERMISSION_DENIED) { 557 mState = state; 558 } 559 } 560 // track was already in the active list, not a problem 561 if (status == ALREADY_EXISTS) { 562 status = NO_ERROR; 563 } 564 } else { 565 status = BAD_VALUE; 566 } 567 return status; 568} 569 570void AudioFlinger::PlaybackThread::Track::stop() 571{ 572 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 573 sp<ThreadBase> thread = mThread.promote(); 574 if (thread != 0) { 575 Mutex::Autolock _l(thread->mLock); 576 track_state state = mState; 577 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 578 // If the track is not active (PAUSED and buffers full), flush buffers 579 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 580 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 581 reset(); 582 mState = STOPPED; 583 } else if (!isFastTrack() && !isOffloaded()) { 584 mState = STOPPED; 585 } else { 586 // For fast tracks prepareTracks_l() will set state to STOPPING_2 587 // presentation is complete 588 // For an offloaded track this starts a drain and state will 589 // move to STOPPING_2 when drain completes and then STOPPED 590 mState = STOPPING_1; 591 } 592 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 593 playbackThread); 594 } 595 } 596} 597 598void AudioFlinger::PlaybackThread::Track::pause() 599{ 600 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 601 sp<ThreadBase> thread = mThread.promote(); 602 if (thread != 0) { 603 Mutex::Autolock _l(thread->mLock); 604 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 605 switch (mState) { 606 case STOPPING_1: 607 case STOPPING_2: 608 if (!isOffloaded()) { 609 /* nothing to do if track is not offloaded */ 610 break; 611 } 612 613 // Offloaded track was draining, we need to carry on draining when resumed 614 mResumeToStopping = true; 615 // fall through... 616 case ACTIVE: 617 case RESUMING: 618 mState = PAUSING; 619 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 620 playbackThread->signal_l(); 621 break; 622 623 default: 624 break; 625 } 626 } 627} 628 629void AudioFlinger::PlaybackThread::Track::flush() 630{ 631 ALOGV("flush(%d)", mName); 632 sp<ThreadBase> thread = mThread.promote(); 633 if (thread != 0) { 634 Mutex::Autolock _l(thread->mLock); 635 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 636 637 if (isOffloaded()) { 638 // If offloaded we allow flush during any state except terminated 639 // and keep the track active to avoid problems if user is seeking 640 // rapidly and underlying hardware has a significant delay handling 641 // a pause 642 if (isTerminated()) { 643 return; 644 } 645 646 ALOGV("flush: offload flush"); 647 reset(); 648 649 if (mState == STOPPING_1 || mState == STOPPING_2) { 650 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 651 mState = ACTIVE; 652 } 653 654 if (mState == ACTIVE) { 655 ALOGV("flush called in active state, resetting buffer time out retry count"); 656 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 657 } 658 659 mResumeToStopping = false; 660 } else { 661 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 662 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 663 return; 664 } 665 // No point remaining in PAUSED state after a flush => go to 666 // FLUSHED state 667 mState = FLUSHED; 668 // do not reset the track if it is still in the process of being stopped or paused. 669 // this will be done by prepareTracks_l() when the track is stopped. 670 // prepareTracks_l() will see mState == FLUSHED, then 671 // remove from active track list, reset(), and trigger presentation complete 672 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 673 reset(); 674 } 675 } 676 // Prevent flush being lost if the track is flushed and then resumed 677 // before mixer thread can run. This is important when offloading 678 // because the hardware buffer could hold a large amount of audio 679 playbackThread->flushOutput_l(); 680 playbackThread->signal_l(); 681 } 682} 683 684void AudioFlinger::PlaybackThread::Track::reset() 685{ 686 // Do not reset twice to avoid discarding data written just after a flush and before 687 // the audioflinger thread detects the track is stopped. 688 if (!mResetDone) { 689 // Force underrun condition to avoid false underrun callback until first data is 690 // written to buffer 691 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 692 mFillingUpStatus = FS_FILLING; 693 mResetDone = true; 694 if (mState == FLUSHED) { 695 mState = IDLE; 696 } 697 } 698} 699 700status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 701{ 702 sp<ThreadBase> thread = mThread.promote(); 703 if (thread == 0) { 704 ALOGE("thread is dead"); 705 return FAILED_TRANSACTION; 706 } else if ((thread->type() == ThreadBase::DIRECT) || 707 (thread->type() == ThreadBase::OFFLOAD)) { 708 return thread->setParameters(keyValuePairs); 709 } else { 710 return PERMISSION_DENIED; 711 } 712} 713 714status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 715{ 716 status_t status = DEAD_OBJECT; 717 sp<ThreadBase> thread = mThread.promote(); 718 if (thread != 0) { 719 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 720 sp<AudioFlinger> af = mClient->audioFlinger(); 721 722 Mutex::Autolock _l(af->mLock); 723 724 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 725 726 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 727 Mutex::Autolock _dl(playbackThread->mLock); 728 Mutex::Autolock _sl(srcThread->mLock); 729 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 730 if (chain == 0) { 731 return INVALID_OPERATION; 732 } 733 734 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 735 if (effect == 0) { 736 return INVALID_OPERATION; 737 } 738 srcThread->removeEffect_l(effect); 739 playbackThread->addEffect_l(effect); 740 // removeEffect_l() has stopped the effect if it was active so it must be restarted 741 if (effect->state() == EffectModule::ACTIVE || 742 effect->state() == EffectModule::STOPPING) { 743 effect->start(); 744 } 745 746 sp<EffectChain> dstChain = effect->chain().promote(); 747 if (dstChain == 0) { 748 srcThread->addEffect_l(effect); 749 return INVALID_OPERATION; 750 } 751 AudioSystem::unregisterEffect(effect->id()); 752 AudioSystem::registerEffect(&effect->desc(), 753 srcThread->id(), 754 dstChain->strategy(), 755 AUDIO_SESSION_OUTPUT_MIX, 756 effect->id()); 757 } 758 status = playbackThread->attachAuxEffect(this, EffectId); 759 } 760 return status; 761} 762 763void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 764{ 765 mAuxEffectId = EffectId; 766 mAuxBuffer = buffer; 767} 768 769bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 770 size_t audioHalFrames) 771{ 772 // a track is considered presented when the total number of frames written to audio HAL 773 // corresponds to the number of frames written when presentationComplete() is called for the 774 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 775 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 776 // to detect when all frames have been played. In this case framesWritten isn't 777 // useful because it doesn't always reflect whether there is data in the h/w 778 // buffers, particularly if a track has been paused and resumed during draining 779 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 780 mPresentationCompleteFrames, framesWritten); 781 if (mPresentationCompleteFrames == 0) { 782 mPresentationCompleteFrames = framesWritten + audioHalFrames; 783 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 784 mPresentationCompleteFrames, audioHalFrames); 785 } 786 787 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 788 ALOGV("presentationComplete() session %d complete: framesWritten %d", 789 mSessionId, framesWritten); 790 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 791 mAudioTrackServerProxy->setStreamEndDone(); 792 return true; 793 } 794 return false; 795} 796 797void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 798{ 799 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 800 if (mSyncEvents[i]->type() == type) { 801 mSyncEvents[i]->trigger(); 802 mSyncEvents.removeAt(i); 803 i--; 804 } 805 } 806} 807 808// implement VolumeBufferProvider interface 809 810uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 811{ 812 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 813 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 814 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 815 uint32_t vl = vlr & 0xFFFF; 816 uint32_t vr = vlr >> 16; 817 // track volumes come from shared memory, so can't be trusted and must be clamped 818 if (vl > MAX_GAIN_INT) { 819 vl = MAX_GAIN_INT; 820 } 821 if (vr > MAX_GAIN_INT) { 822 vr = MAX_GAIN_INT; 823 } 824 // now apply the cached master volume and stream type volume; 825 // this is trusted but lacks any synchronization or barrier so may be stale 826 float v = mCachedVolume; 827 vl *= v; 828 vr *= v; 829 // re-combine into U4.16 830 vlr = (vr << 16) | (vl & 0xFFFF); 831 // FIXME look at mute, pause, and stop flags 832 return vlr; 833} 834 835status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 836{ 837 if (isTerminated() || mState == PAUSED || 838 ((framesReady() == 0) && ((mSharedBuffer != 0) || 839 (mState == STOPPED)))) { 840 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 841 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 842 event->cancel(); 843 return INVALID_OPERATION; 844 } 845 (void) TrackBase::setSyncEvent(event); 846 return NO_ERROR; 847} 848 849void AudioFlinger::PlaybackThread::Track::invalidate() 850{ 851 // FIXME should use proxy, and needs work 852 audio_track_cblk_t* cblk = mCblk; 853 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 854 android_atomic_release_store(0x40000000, &cblk->mFutex); 855 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 856 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 857 mIsInvalid = true; 858} 859 860// ---------------------------------------------------------------------------- 861 862sp<AudioFlinger::PlaybackThread::TimedTrack> 863AudioFlinger::PlaybackThread::TimedTrack::create( 864 PlaybackThread *thread, 865 const sp<Client>& client, 866 audio_stream_type_t streamType, 867 uint32_t sampleRate, 868 audio_format_t format, 869 audio_channel_mask_t channelMask, 870 size_t frameCount, 871 const sp<IMemory>& sharedBuffer, 872 int sessionId) { 873 if (!client->reserveTimedTrack()) 874 return 0; 875 876 return new TimedTrack( 877 thread, client, streamType, sampleRate, format, channelMask, frameCount, 878 sharedBuffer, sessionId); 879} 880 881AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 882 PlaybackThread *thread, 883 const sp<Client>& client, 884 audio_stream_type_t streamType, 885 uint32_t sampleRate, 886 audio_format_t format, 887 audio_channel_mask_t channelMask, 888 size_t frameCount, 889 const sp<IMemory>& sharedBuffer, 890 int sessionId) 891 : Track(thread, client, streamType, sampleRate, format, channelMask, 892 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 893 mQueueHeadInFlight(false), 894 mTrimQueueHeadOnRelease(false), 895 mFramesPendingInQueue(0), 896 mTimedSilenceBuffer(NULL), 897 mTimedSilenceBufferSize(0), 898 mTimedAudioOutputOnTime(false), 899 mMediaTimeTransformValid(false) 900{ 901 LocalClock lc; 902 mLocalTimeFreq = lc.getLocalFreq(); 903 904 mLocalTimeToSampleTransform.a_zero = 0; 905 mLocalTimeToSampleTransform.b_zero = 0; 906 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 907 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 908 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 909 &mLocalTimeToSampleTransform.a_to_b_denom); 910 911 mMediaTimeToSampleTransform.a_zero = 0; 912 mMediaTimeToSampleTransform.b_zero = 0; 913 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 914 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 915 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 916 &mMediaTimeToSampleTransform.a_to_b_denom); 917} 918 919AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 920 mClient->releaseTimedTrack(); 921 delete [] mTimedSilenceBuffer; 922} 923 924status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 925 size_t size, sp<IMemory>* buffer) { 926 927 Mutex::Autolock _l(mTimedBufferQueueLock); 928 929 trimTimedBufferQueue_l(); 930 931 // lazily initialize the shared memory heap for timed buffers 932 if (mTimedMemoryDealer == NULL) { 933 const int kTimedBufferHeapSize = 512 << 10; 934 935 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 936 "AudioFlingerTimed"); 937 if (mTimedMemoryDealer == NULL) 938 return NO_MEMORY; 939 } 940 941 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 942 if (newBuffer == NULL) { 943 newBuffer = mTimedMemoryDealer->allocate(size); 944 if (newBuffer == NULL) 945 return NO_MEMORY; 946 } 947 948 *buffer = newBuffer; 949 return NO_ERROR; 950} 951 952// caller must hold mTimedBufferQueueLock 953void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 954 int64_t mediaTimeNow; 955 { 956 Mutex::Autolock mttLock(mMediaTimeTransformLock); 957 if (!mMediaTimeTransformValid) 958 return; 959 960 int64_t targetTimeNow; 961 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 962 ? mCCHelper.getCommonTime(&targetTimeNow) 963 : mCCHelper.getLocalTime(&targetTimeNow); 964 965 if (OK != res) 966 return; 967 968 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 969 &mediaTimeNow)) { 970 return; 971 } 972 } 973 974 size_t trimEnd; 975 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 976 int64_t bufEnd; 977 978 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 979 // We have a next buffer. Just use its PTS as the PTS of the frame 980 // following the last frame in this buffer. If the stream is sparse 981 // (ie, there are deliberate gaps left in the stream which should be 982 // filled with silence by the TimedAudioTrack), then this can result 983 // in one extra buffer being left un-trimmed when it could have 984 // been. In general, this is not typical, and we would rather 985 // optimized away the TS calculation below for the more common case 986 // where PTSes are contiguous. 987 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 988 } else { 989 // We have no next buffer. Compute the PTS of the frame following 990 // the last frame in this buffer by computing the duration of of 991 // this frame in media time units and adding it to the PTS of the 992 // buffer. 993 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 994 / mFrameSize; 995 996 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 997 &bufEnd)) { 998 ALOGE("Failed to convert frame count of %lld to media time" 999 " duration" " (scale factor %d/%u) in %s", 1000 frameCount, 1001 mMediaTimeToSampleTransform.a_to_b_numer, 1002 mMediaTimeToSampleTransform.a_to_b_denom, 1003 __PRETTY_FUNCTION__); 1004 break; 1005 } 1006 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1007 } 1008 1009 if (bufEnd > mediaTimeNow) 1010 break; 1011 1012 // Is the buffer we want to use in the middle of a mix operation right 1013 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1014 // from the mixer which should be coming back shortly. 1015 if (!trimEnd && mQueueHeadInFlight) { 1016 mTrimQueueHeadOnRelease = true; 1017 } 1018 } 1019 1020 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1021 if (trimStart < trimEnd) { 1022 // Update the bookkeeping for framesReady() 1023 for (size_t i = trimStart; i < trimEnd; ++i) { 1024 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1025 } 1026 1027 // Now actually remove the buffers from the queue. 1028 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1029 } 1030} 1031 1032void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1033 const char* logTag) { 1034 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1035 "%s called (reason \"%s\"), but timed buffer queue has no" 1036 " elements to trim.", __FUNCTION__, logTag); 1037 1038 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1039 mTimedBufferQueue.removeAt(0); 1040} 1041 1042void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1043 const TimedBuffer& buf, 1044 const char* logTag) { 1045 uint32_t bufBytes = buf.buffer()->size(); 1046 uint32_t consumedAlready = buf.position(); 1047 1048 ALOG_ASSERT(consumedAlready <= bufBytes, 1049 "Bad bookkeeping while updating frames pending. Timed buffer is" 1050 " only %u bytes long, but claims to have consumed %u" 1051 " bytes. (update reason: \"%s\")", 1052 bufBytes, consumedAlready, logTag); 1053 1054 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1055 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1056 "Bad bookkeeping while updating frames pending. Should have at" 1057 " least %u queued frames, but we think we have only %u. (update" 1058 " reason: \"%s\")", 1059 bufFrames, mFramesPendingInQueue, logTag); 1060 1061 mFramesPendingInQueue -= bufFrames; 1062} 1063 1064status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1065 const sp<IMemory>& buffer, int64_t pts) { 1066 1067 { 1068 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1069 if (!mMediaTimeTransformValid) 1070 return INVALID_OPERATION; 1071 } 1072 1073 Mutex::Autolock _l(mTimedBufferQueueLock); 1074 1075 uint32_t bufFrames = buffer->size() / mFrameSize; 1076 mFramesPendingInQueue += bufFrames; 1077 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1078 1079 return NO_ERROR; 1080} 1081 1082status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1083 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1084 1085 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1086 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1087 target); 1088 1089 if (!(target == TimedAudioTrack::LOCAL_TIME || 1090 target == TimedAudioTrack::COMMON_TIME)) { 1091 return BAD_VALUE; 1092 } 1093 1094 Mutex::Autolock lock(mMediaTimeTransformLock); 1095 mMediaTimeTransform = xform; 1096 mMediaTimeTransformTarget = target; 1097 mMediaTimeTransformValid = true; 1098 1099 return NO_ERROR; 1100} 1101 1102#define min(a, b) ((a) < (b) ? (a) : (b)) 1103 1104// implementation of getNextBuffer for tracks whose buffers have timestamps 1105status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1106 AudioBufferProvider::Buffer* buffer, int64_t pts) 1107{ 1108 if (pts == AudioBufferProvider::kInvalidPTS) { 1109 buffer->raw = NULL; 1110 buffer->frameCount = 0; 1111 mTimedAudioOutputOnTime = false; 1112 return INVALID_OPERATION; 1113 } 1114 1115 Mutex::Autolock _l(mTimedBufferQueueLock); 1116 1117 ALOG_ASSERT(!mQueueHeadInFlight, 1118 "getNextBuffer called without releaseBuffer!"); 1119 1120 while (true) { 1121 1122 // if we have no timed buffers, then fail 1123 if (mTimedBufferQueue.isEmpty()) { 1124 buffer->raw = NULL; 1125 buffer->frameCount = 0; 1126 return NOT_ENOUGH_DATA; 1127 } 1128 1129 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1130 1131 // calculate the PTS of the head of the timed buffer queue expressed in 1132 // local time 1133 int64_t headLocalPTS; 1134 { 1135 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1136 1137 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1138 1139 if (mMediaTimeTransform.a_to_b_denom == 0) { 1140 // the transform represents a pause, so yield silence 1141 timedYieldSilence_l(buffer->frameCount, buffer); 1142 return NO_ERROR; 1143 } 1144 1145 int64_t transformedPTS; 1146 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1147 &transformedPTS)) { 1148 // the transform failed. this shouldn't happen, but if it does 1149 // then just drop this buffer 1150 ALOGW("timedGetNextBuffer transform failed"); 1151 buffer->raw = NULL; 1152 buffer->frameCount = 0; 1153 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1154 return NO_ERROR; 1155 } 1156 1157 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1158 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1159 &headLocalPTS)) { 1160 buffer->raw = NULL; 1161 buffer->frameCount = 0; 1162 return INVALID_OPERATION; 1163 } 1164 } else { 1165 headLocalPTS = transformedPTS; 1166 } 1167 } 1168 1169 uint32_t sr = sampleRate(); 1170 1171 // adjust the head buffer's PTS to reflect the portion of the head buffer 1172 // that has already been consumed 1173 int64_t effectivePTS = headLocalPTS + 1174 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1175 1176 // Calculate the delta in samples between the head of the input buffer 1177 // queue and the start of the next output buffer that will be written. 1178 // If the transformation fails because of over or underflow, it means 1179 // that the sample's position in the output stream is so far out of 1180 // whack that it should just be dropped. 1181 int64_t sampleDelta; 1182 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1183 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1184 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1185 " mix"); 1186 continue; 1187 } 1188 if (!mLocalTimeToSampleTransform.doForwardTransform( 1189 (effectivePTS - pts) << 32, &sampleDelta)) { 1190 ALOGV("*** too late during sample rate transform: dropped buffer"); 1191 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1192 continue; 1193 } 1194 1195 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1196 " sampleDelta=[%d.%08x]", 1197 head.pts(), head.position(), pts, 1198 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1199 + (sampleDelta >> 32)), 1200 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1201 1202 // if the delta between the ideal placement for the next input sample and 1203 // the current output position is within this threshold, then we will 1204 // concatenate the next input samples to the previous output 1205 const int64_t kSampleContinuityThreshold = 1206 (static_cast<int64_t>(sr) << 32) / 250; 1207 1208 // if this is the first buffer of audio that we're emitting from this track 1209 // then it should be almost exactly on time. 1210 const int64_t kSampleStartupThreshold = 1LL << 32; 1211 1212 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1213 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1214 // the next input is close enough to being on time, so concatenate it 1215 // with the last output 1216 timedYieldSamples_l(buffer); 1217 1218 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1219 head.position(), buffer->frameCount); 1220 return NO_ERROR; 1221 } 1222 1223 // Looks like our output is not on time. Reset our on timed status. 1224 // Next time we mix samples from our input queue, then should be within 1225 // the StartupThreshold. 1226 mTimedAudioOutputOnTime = false; 1227 if (sampleDelta > 0) { 1228 // the gap between the current output position and the proper start of 1229 // the next input sample is too big, so fill it with silence 1230 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1231 1232 timedYieldSilence_l(framesUntilNextInput, buffer); 1233 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1234 return NO_ERROR; 1235 } else { 1236 // the next input sample is late 1237 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1238 size_t onTimeSamplePosition = 1239 head.position() + lateFrames * mFrameSize; 1240 1241 if (onTimeSamplePosition > head.buffer()->size()) { 1242 // all the remaining samples in the head are too late, so 1243 // drop it and move on 1244 ALOGV("*** too late: dropped buffer"); 1245 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1246 continue; 1247 } else { 1248 // skip over the late samples 1249 head.setPosition(onTimeSamplePosition); 1250 1251 // yield the available samples 1252 timedYieldSamples_l(buffer); 1253 1254 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1255 return NO_ERROR; 1256 } 1257 } 1258 } 1259} 1260 1261// Yield samples from the timed buffer queue head up to the given output 1262// buffer's capacity. 1263// 1264// Caller must hold mTimedBufferQueueLock 1265void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1266 AudioBufferProvider::Buffer* buffer) { 1267 1268 const TimedBuffer& head = mTimedBufferQueue[0]; 1269 1270 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1271 head.position()); 1272 1273 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1274 mFrameSize); 1275 size_t framesRequested = buffer->frameCount; 1276 buffer->frameCount = min(framesLeftInHead, framesRequested); 1277 1278 mQueueHeadInFlight = true; 1279 mTimedAudioOutputOnTime = true; 1280} 1281 1282// Yield samples of silence up to the given output buffer's capacity 1283// 1284// Caller must hold mTimedBufferQueueLock 1285void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1286 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1287 1288 // lazily allocate a buffer filled with silence 1289 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1290 delete [] mTimedSilenceBuffer; 1291 mTimedSilenceBufferSize = numFrames * mFrameSize; 1292 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1293 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1294 } 1295 1296 buffer->raw = mTimedSilenceBuffer; 1297 size_t framesRequested = buffer->frameCount; 1298 buffer->frameCount = min(numFrames, framesRequested); 1299 1300 mTimedAudioOutputOnTime = false; 1301} 1302 1303// AudioBufferProvider interface 1304void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1305 AudioBufferProvider::Buffer* buffer) { 1306 1307 Mutex::Autolock _l(mTimedBufferQueueLock); 1308 1309 // If the buffer which was just released is part of the buffer at the head 1310 // of the queue, be sure to update the amt of the buffer which has been 1311 // consumed. If the buffer being returned is not part of the head of the 1312 // queue, its either because the buffer is part of the silence buffer, or 1313 // because the head of the timed queue was trimmed after the mixer called 1314 // getNextBuffer but before the mixer called releaseBuffer. 1315 if (buffer->raw == mTimedSilenceBuffer) { 1316 ALOG_ASSERT(!mQueueHeadInFlight, 1317 "Queue head in flight during release of silence buffer!"); 1318 goto done; 1319 } 1320 1321 ALOG_ASSERT(mQueueHeadInFlight, 1322 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1323 " head in flight."); 1324 1325 if (mTimedBufferQueue.size()) { 1326 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1327 1328 void* start = head.buffer()->pointer(); 1329 void* end = reinterpret_cast<void*>( 1330 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1331 + head.buffer()->size()); 1332 1333 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1334 "released buffer not within the head of the timed buffer" 1335 " queue; qHead = [%p, %p], released buffer = %p", 1336 start, end, buffer->raw); 1337 1338 head.setPosition(head.position() + 1339 (buffer->frameCount * mFrameSize)); 1340 mQueueHeadInFlight = false; 1341 1342 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1343 "Bad bookkeeping during releaseBuffer! Should have at" 1344 " least %u queued frames, but we think we have only %u", 1345 buffer->frameCount, mFramesPendingInQueue); 1346 1347 mFramesPendingInQueue -= buffer->frameCount; 1348 1349 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1350 || mTrimQueueHeadOnRelease) { 1351 trimTimedBufferQueueHead_l("releaseBuffer"); 1352 mTrimQueueHeadOnRelease = false; 1353 } 1354 } else { 1355 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1356 " buffers in the timed buffer queue"); 1357 } 1358 1359done: 1360 buffer->raw = 0; 1361 buffer->frameCount = 0; 1362} 1363 1364size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1365 Mutex::Autolock _l(mTimedBufferQueueLock); 1366 return mFramesPendingInQueue; 1367} 1368 1369AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1370 : mPTS(0), mPosition(0) {} 1371 1372AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1373 const sp<IMemory>& buffer, int64_t pts) 1374 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1375 1376 1377// ---------------------------------------------------------------------------- 1378 1379AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1380 PlaybackThread *playbackThread, 1381 DuplicatingThread *sourceThread, 1382 uint32_t sampleRate, 1383 audio_format_t format, 1384 audio_channel_mask_t channelMask, 1385 size_t frameCount) 1386 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1387 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 1388 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1389{ 1390 1391 if (mCblk != NULL) { 1392 mOutBuffer.frameCount = 0; 1393 playbackThread->mTracks.add(this); 1394 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1395 "mCblk->frameCount_ %u, mChannelMask 0x%08x", 1396 mCblk, mBuffer, 1397 mCblk->frameCount_, mChannelMask); 1398 // since client and server are in the same process, 1399 // the buffer has the same virtual address on both sides 1400 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize); 1401 mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000)); 1402 mClientProxy->setSendLevel(0.0); 1403 mClientProxy->setSampleRate(sampleRate); 1404 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1405 true /*clientInServer*/); 1406 } else { 1407 ALOGW("Error creating output track on thread %p", playbackThread); 1408 } 1409} 1410 1411AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1412{ 1413 clearBufferQueue(); 1414 delete mClientProxy; 1415 // superclass destructor will now delete the server proxy and shared memory both refer to 1416} 1417 1418status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1419 int triggerSession) 1420{ 1421 status_t status = Track::start(event, triggerSession); 1422 if (status != NO_ERROR) { 1423 return status; 1424 } 1425 1426 mActive = true; 1427 mRetryCount = 127; 1428 return status; 1429} 1430 1431void AudioFlinger::PlaybackThread::OutputTrack::stop() 1432{ 1433 Track::stop(); 1434 clearBufferQueue(); 1435 mOutBuffer.frameCount = 0; 1436 mActive = false; 1437} 1438 1439bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1440{ 1441 Buffer *pInBuffer; 1442 Buffer inBuffer; 1443 uint32_t channelCount = mChannelCount; 1444 bool outputBufferFull = false; 1445 inBuffer.frameCount = frames; 1446 inBuffer.i16 = data; 1447 1448 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1449 1450 if (!mActive && frames != 0) { 1451 start(); 1452 sp<ThreadBase> thread = mThread.promote(); 1453 if (thread != 0) { 1454 MixerThread *mixerThread = (MixerThread *)thread.get(); 1455 if (mFrameCount > frames) { 1456 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1457 uint32_t startFrames = (mFrameCount - frames); 1458 pInBuffer = new Buffer; 1459 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1460 pInBuffer->frameCount = startFrames; 1461 pInBuffer->i16 = pInBuffer->mBuffer; 1462 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1463 mBufferQueue.add(pInBuffer); 1464 } else { 1465 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1466 } 1467 } 1468 } 1469 } 1470 1471 while (waitTimeLeftMs) { 1472 // First write pending buffers, then new data 1473 if (mBufferQueue.size()) { 1474 pInBuffer = mBufferQueue.itemAt(0); 1475 } else { 1476 pInBuffer = &inBuffer; 1477 } 1478 1479 if (pInBuffer->frameCount == 0) { 1480 break; 1481 } 1482 1483 if (mOutBuffer.frameCount == 0) { 1484 mOutBuffer.frameCount = pInBuffer->frameCount; 1485 nsecs_t startTime = systemTime(); 1486 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1487 if (status != NO_ERROR) { 1488 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1489 mThread.unsafe_get(), status); 1490 outputBufferFull = true; 1491 break; 1492 } 1493 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1494 if (waitTimeLeftMs >= waitTimeMs) { 1495 waitTimeLeftMs -= waitTimeMs; 1496 } else { 1497 waitTimeLeftMs = 0; 1498 } 1499 } 1500 1501 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1502 pInBuffer->frameCount; 1503 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1504 Proxy::Buffer buf; 1505 buf.mFrameCount = outFrames; 1506 buf.mRaw = NULL; 1507 mClientProxy->releaseBuffer(&buf); 1508 pInBuffer->frameCount -= outFrames; 1509 pInBuffer->i16 += outFrames * channelCount; 1510 mOutBuffer.frameCount -= outFrames; 1511 mOutBuffer.i16 += outFrames * channelCount; 1512 1513 if (pInBuffer->frameCount == 0) { 1514 if (mBufferQueue.size()) { 1515 mBufferQueue.removeAt(0); 1516 delete [] pInBuffer->mBuffer; 1517 delete pInBuffer; 1518 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1519 mThread.unsafe_get(), mBufferQueue.size()); 1520 } else { 1521 break; 1522 } 1523 } 1524 } 1525 1526 // If we could not write all frames, allocate a buffer and queue it for next time. 1527 if (inBuffer.frameCount) { 1528 sp<ThreadBase> thread = mThread.promote(); 1529 if (thread != 0 && !thread->standby()) { 1530 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1531 pInBuffer = new Buffer; 1532 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1533 pInBuffer->frameCount = inBuffer.frameCount; 1534 pInBuffer->i16 = pInBuffer->mBuffer; 1535 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1536 sizeof(int16_t)); 1537 mBufferQueue.add(pInBuffer); 1538 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1539 mThread.unsafe_get(), mBufferQueue.size()); 1540 } else { 1541 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1542 mThread.unsafe_get(), this); 1543 } 1544 } 1545 } 1546 1547 // Calling write() with a 0 length buffer, means that no more data will be written: 1548 // If no more buffers are pending, fill output track buffer to make sure it is started 1549 // by output mixer. 1550 if (frames == 0 && mBufferQueue.size() == 0) { 1551 // FIXME borken, replace by getting framesReady() from proxy 1552 size_t user = 0; // was mCblk->user 1553 if (user < mFrameCount) { 1554 frames = mFrameCount - user; 1555 pInBuffer = new Buffer; 1556 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1557 pInBuffer->frameCount = frames; 1558 pInBuffer->i16 = pInBuffer->mBuffer; 1559 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1560 mBufferQueue.add(pInBuffer); 1561 } else if (mActive) { 1562 stop(); 1563 } 1564 } 1565 1566 return outputBufferFull; 1567} 1568 1569status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1570 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1571{ 1572 ClientProxy::Buffer buf; 1573 buf.mFrameCount = buffer->frameCount; 1574 struct timespec timeout; 1575 timeout.tv_sec = waitTimeMs / 1000; 1576 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1577 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1578 buffer->frameCount = buf.mFrameCount; 1579 buffer->raw = buf.mRaw; 1580 return status; 1581} 1582 1583void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1584{ 1585 size_t size = mBufferQueue.size(); 1586 1587 for (size_t i = 0; i < size; i++) { 1588 Buffer *pBuffer = mBufferQueue.itemAt(i); 1589 delete [] pBuffer->mBuffer; 1590 delete pBuffer; 1591 } 1592 mBufferQueue.clear(); 1593} 1594 1595 1596// ---------------------------------------------------------------------------- 1597// Record 1598// ---------------------------------------------------------------------------- 1599 1600AudioFlinger::RecordHandle::RecordHandle( 1601 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1602 : BnAudioRecord(), 1603 mRecordTrack(recordTrack) 1604{ 1605} 1606 1607AudioFlinger::RecordHandle::~RecordHandle() { 1608 stop_nonvirtual(); 1609 mRecordTrack->destroy(); 1610} 1611 1612sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 1613 return mRecordTrack->getCblk(); 1614} 1615 1616status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1617 int triggerSession) { 1618 ALOGV("RecordHandle::start()"); 1619 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1620} 1621 1622void AudioFlinger::RecordHandle::stop() { 1623 stop_nonvirtual(); 1624} 1625 1626void AudioFlinger::RecordHandle::stop_nonvirtual() { 1627 ALOGV("RecordHandle::stop()"); 1628 mRecordTrack->stop(); 1629} 1630 1631status_t AudioFlinger::RecordHandle::onTransact( 1632 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1633{ 1634 return BnAudioRecord::onTransact(code, data, reply, flags); 1635} 1636 1637// ---------------------------------------------------------------------------- 1638 1639// RecordTrack constructor must be called with AudioFlinger::mLock held 1640AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1641 RecordThread *thread, 1642 const sp<Client>& client, 1643 uint32_t sampleRate, 1644 audio_format_t format, 1645 audio_channel_mask_t channelMask, 1646 size_t frameCount, 1647 int sessionId) 1648 : TrackBase(thread, client, sampleRate, format, 1649 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/), 1650 mOverflow(false) 1651{ 1652 ALOGV("RecordTrack constructor"); 1653 if (mCblk != NULL) { 1654 mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize); 1655 } 1656} 1657 1658AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 1659{ 1660 ALOGV("%s", __func__); 1661} 1662 1663// AudioBufferProvider interface 1664status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 1665 int64_t pts) 1666{ 1667 ServerProxy::Buffer buf; 1668 buf.mFrameCount = buffer->frameCount; 1669 status_t status = mServerProxy->obtainBuffer(&buf); 1670 buffer->frameCount = buf.mFrameCount; 1671 buffer->raw = buf.mRaw; 1672 if (buf.mFrameCount == 0) { 1673 // FIXME also wake futex so that overrun is noticed more quickly 1674 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 1675 } 1676 return status; 1677} 1678 1679status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 1680 int triggerSession) 1681{ 1682 sp<ThreadBase> thread = mThread.promote(); 1683 if (thread != 0) { 1684 RecordThread *recordThread = (RecordThread *)thread.get(); 1685 return recordThread->start(this, event, triggerSession); 1686 } else { 1687 return BAD_VALUE; 1688 } 1689} 1690 1691void AudioFlinger::RecordThread::RecordTrack::stop() 1692{ 1693 sp<ThreadBase> thread = mThread.promote(); 1694 if (thread != 0) { 1695 RecordThread *recordThread = (RecordThread *)thread.get(); 1696 if (recordThread->stop(this)) { 1697 AudioSystem::stopInput(recordThread->id()); 1698 } 1699 } 1700} 1701 1702void AudioFlinger::RecordThread::RecordTrack::destroy() 1703{ 1704 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 1705 sp<RecordTrack> keep(this); 1706 { 1707 sp<ThreadBase> thread = mThread.promote(); 1708 if (thread != 0) { 1709 if (mState == ACTIVE || mState == RESUMING) { 1710 AudioSystem::stopInput(thread->id()); 1711 } 1712 AudioSystem::releaseInput(thread->id()); 1713 Mutex::Autolock _l(thread->mLock); 1714 RecordThread *recordThread = (RecordThread *) thread.get(); 1715 recordThread->destroyTrack_l(this); 1716 } 1717 } 1718} 1719 1720 1721/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 1722{ 1723 result.append("Client Fmt Chn mask Session S Server fCount\n"); 1724} 1725 1726void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 1727{ 1728 snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6u\n", 1729 (mClient == 0) ? getpid_cached : mClient->pid(), 1730 mFormat, 1731 mChannelMask, 1732 mSessionId, 1733 mState, 1734 mCblk->mServer, 1735 mFrameCount); 1736} 1737 1738}; // namespace android 1739