Tracks.cpp revision 81784c37c61b09289654b979567a42bf73cd2b12
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 <math.h> 23#include <cutils/compiler.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// ---------------------------------------------------------------------------- 36 37// Note: the following macro is used for extremely verbose logging message. In 38// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 39// 0; but one side effect of this is to turn all LOGV's as well. Some messages 40// are so verbose that we want to suppress them even when we have ALOG_ASSERT 41// turned on. Do not uncomment the #def below unless you really know what you 42// are doing and want to see all of the extremely verbose messages. 43//#define VERY_VERY_VERBOSE_LOGGING 44#ifdef VERY_VERY_VERBOSE_LOGGING 45#define ALOGVV ALOGV 46#else 47#define ALOGVV(a...) do { } while(0) 48#endif 49 50namespace android { 51 52// ---------------------------------------------------------------------------- 53// TrackBase 54// ---------------------------------------------------------------------------- 55 56// TrackBase constructor must be called with AudioFlinger::mLock held 57AudioFlinger::ThreadBase::TrackBase::TrackBase( 58 ThreadBase *thread, 59 const sp<Client>& client, 60 uint32_t sampleRate, 61 audio_format_t format, 62 audio_channel_mask_t channelMask, 63 size_t frameCount, 64 const sp<IMemory>& sharedBuffer, 65 int sessionId) 66 : RefBase(), 67 mThread(thread), 68 mClient(client), 69 mCblk(NULL), 70 // mBuffer 71 // mBufferEnd 72 mStepCount(0), 73 mState(IDLE), 74 mSampleRate(sampleRate), 75 mFormat(format), 76 mChannelMask(channelMask), 77 mChannelCount(popcount(channelMask)), 78 mFrameSize(audio_is_linear_pcm(format) ? 79 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 80 mFrameCount(frameCount), 81 mStepServerFailed(false), 82 mSessionId(sessionId) 83{ 84 // client == 0 implies sharedBuffer == 0 85 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 86 87 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 88 sharedBuffer->size()); 89 90 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 91 size_t size = sizeof(audio_track_cblk_t); 92 size_t bufferSize = frameCount * mFrameSize; 93 if (sharedBuffer == 0) { 94 size += bufferSize; 95 } 96 97 if (client != 0) { 98 mCblkMemory = client->heap()->allocate(size); 99 if (mCblkMemory != 0) { 100 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 101 // can't assume mCblk != NULL 102 } else { 103 ALOGE("not enough memory for AudioTrack size=%u", size); 104 client->heap()->dump("AudioTrack"); 105 return; 106 } 107 } else { 108 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 109 // assume mCblk != NULL 110 } 111 112 // construct the shared structure in-place. 113 if (mCblk != NULL) { 114 new(mCblk) audio_track_cblk_t(); 115 // clear all buffers 116 mCblk->frameCount_ = frameCount; 117 mCblk->sampleRate = sampleRate; 118// uncomment the following lines to quickly test 32-bit wraparound 119// mCblk->user = 0xffff0000; 120// mCblk->server = 0xffff0000; 121// mCblk->userBase = 0xffff0000; 122// mCblk->serverBase = 0xffff0000; 123 if (sharedBuffer == 0) { 124 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 125 memset(mBuffer, 0, bufferSize); 126 // Force underrun condition to avoid false underrun callback until first data is 127 // written to buffer (other flags are cleared) 128 mCblk->flags = CBLK_UNDERRUN; 129 } else { 130 mBuffer = sharedBuffer->pointer(); 131 } 132 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 133 } 134} 135 136AudioFlinger::ThreadBase::TrackBase::~TrackBase() 137{ 138 if (mCblk != NULL) { 139 if (mClient == 0) { 140 delete mCblk; 141 } else { 142 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 143 } 144 } 145 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 146 if (mClient != 0) { 147 // Client destructor must run with AudioFlinger mutex locked 148 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 149 // If the client's reference count drops to zero, the associated destructor 150 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 151 // relying on the automatic clear() at end of scope. 152 mClient.clear(); 153 } 154} 155 156// AudioBufferProvider interface 157// getNextBuffer() = 0; 158// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 159void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 160{ 161 buffer->raw = NULL; 162 mStepCount = buffer->frameCount; 163 // FIXME See note at getNextBuffer() 164 (void) step(); // ignore return value of step() 165 buffer->frameCount = 0; 166} 167 168bool AudioFlinger::ThreadBase::TrackBase::step() { 169 bool result; 170 audio_track_cblk_t* cblk = this->cblk(); 171 172 result = cblk->stepServer(mStepCount, mFrameCount, isOut()); 173 if (!result) { 174 ALOGV("stepServer failed acquiring cblk mutex"); 175 mStepServerFailed = true; 176 } 177 return result; 178} 179 180void AudioFlinger::ThreadBase::TrackBase::reset() { 181 audio_track_cblk_t* cblk = this->cblk(); 182 183 cblk->user = 0; 184 cblk->server = 0; 185 cblk->userBase = 0; 186 cblk->serverBase = 0; 187 mStepServerFailed = false; 188 ALOGV("TrackBase::reset"); 189} 190 191uint32_t AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 192 return mCblk->sampleRate; 193} 194 195void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 196 audio_track_cblk_t* cblk = this->cblk(); 197 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase) * mFrameSize; 198 int8_t *bufferEnd = bufferStart + frames * mFrameSize; 199 200 // Check validity of returned pointer in case the track control block would have been corrupted. 201 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 202 "TrackBase::getBuffer buffer out of range:\n" 203 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 204 " server %u, serverBase %u, user %u, userBase %u, frameSize %u", 205 bufferStart, bufferEnd, mBuffer, mBufferEnd, 206 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, mFrameSize); 207 208 return bufferStart; 209} 210 211status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 212{ 213 mSyncEvents.add(event); 214 return NO_ERROR; 215} 216 217// ---------------------------------------------------------------------------- 218// Playback 219// ---------------------------------------------------------------------------- 220 221AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 222 : BnAudioTrack(), 223 mTrack(track) 224{ 225} 226 227AudioFlinger::TrackHandle::~TrackHandle() { 228 // just stop the track on deletion, associated resources 229 // will be freed from the main thread once all pending buffers have 230 // been played. Unless it's not in the active track list, in which 231 // case we free everything now... 232 mTrack->destroy(); 233} 234 235sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 236 return mTrack->getCblk(); 237} 238 239status_t AudioFlinger::TrackHandle::start() { 240 return mTrack->start(); 241} 242 243void AudioFlinger::TrackHandle::stop() { 244 mTrack->stop(); 245} 246 247void AudioFlinger::TrackHandle::flush() { 248 mTrack->flush(); 249} 250 251void AudioFlinger::TrackHandle::mute(bool e) { 252 mTrack->mute(e); 253} 254 255void AudioFlinger::TrackHandle::pause() { 256 mTrack->pause(); 257} 258 259status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 260{ 261 return mTrack->attachAuxEffect(EffectId); 262} 263 264status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 265 sp<IMemory>* buffer) { 266 if (!mTrack->isTimedTrack()) 267 return INVALID_OPERATION; 268 269 PlaybackThread::TimedTrack* tt = 270 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 271 return tt->allocateTimedBuffer(size, buffer); 272} 273 274status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 275 int64_t pts) { 276 if (!mTrack->isTimedTrack()) 277 return INVALID_OPERATION; 278 279 PlaybackThread::TimedTrack* tt = 280 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 281 return tt->queueTimedBuffer(buffer, pts); 282} 283 284status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 285 const LinearTransform& xform, int target) { 286 287 if (!mTrack->isTimedTrack()) 288 return INVALID_OPERATION; 289 290 PlaybackThread::TimedTrack* tt = 291 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 292 return tt->setMediaTimeTransform( 293 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 294} 295 296status_t AudioFlinger::TrackHandle::onTransact( 297 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 298{ 299 return BnAudioTrack::onTransact(code, data, reply, flags); 300} 301 302// ---------------------------------------------------------------------------- 303 304// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 305AudioFlinger::PlaybackThread::Track::Track( 306 PlaybackThread *thread, 307 const sp<Client>& client, 308 audio_stream_type_t streamType, 309 uint32_t sampleRate, 310 audio_format_t format, 311 audio_channel_mask_t channelMask, 312 size_t frameCount, 313 const sp<IMemory>& sharedBuffer, 314 int sessionId, 315 IAudioFlinger::track_flags_t flags) 316 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 317 sessionId), 318 mMute(false), 319 mFillingUpStatus(FS_INVALID), 320 // mRetryCount initialized later when needed 321 mSharedBuffer(sharedBuffer), 322 mStreamType(streamType), 323 mName(-1), // see note below 324 mMainBuffer(thread->mixBuffer()), 325 mAuxBuffer(NULL), 326 mAuxEffectId(0), mHasVolumeController(false), 327 mPresentationCompleteFrames(0), 328 mFlags(flags), 329 mFastIndex(-1), 330 mUnderrunCount(0), 331 mCachedVolume(1.0) 332{ 333 if (mCblk != NULL) { 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 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 344 int i = __builtin_ctz(thread->mFastTrackAvailMask); 345 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 346 // FIXME This is too eager. We allocate a fast track index before the 347 // fast track becomes active. Since fast tracks are a scarce resource, 348 // this means we are potentially denying other more important fast tracks from 349 // being created. It would be better to allocate the index dynamically. 350 mFastIndex = i; 351 mCblk->mName = i; 352 // Read the initial underruns because this field is never cleared by the fast mixer 353 mObservedUnderruns = thread->getFastTrackUnderruns(i); 354 thread->mFastTrackAvailMask &= ~(1 << i); 355 } 356 } 357 ALOGV("Track constructor name %d, calling pid %d", mName, 358 IPCThreadState::self()->getCallingPid()); 359} 360 361AudioFlinger::PlaybackThread::Track::~Track() 362{ 363 ALOGV("PlaybackThread::Track destructor"); 364} 365 366void AudioFlinger::PlaybackThread::Track::destroy() 367{ 368 // NOTE: destroyTrack_l() can remove a strong reference to this Track 369 // by removing it from mTracks vector, so there is a risk that this Tracks's 370 // destructor is called. As the destructor needs to lock mLock, 371 // we must acquire a strong reference on this Track before locking mLock 372 // here so that the destructor is called only when exiting this function. 373 // On the other hand, as long as Track::destroy() is only called by 374 // TrackHandle destructor, the TrackHandle still holds a strong ref on 375 // this Track with its member mTrack. 376 sp<Track> keep(this); 377 { // scope for mLock 378 sp<ThreadBase> thread = mThread.promote(); 379 if (thread != 0) { 380 if (!isOutputTrack()) { 381 if (mState == ACTIVE || mState == RESUMING) { 382 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 383 384#ifdef ADD_BATTERY_DATA 385 // to track the speaker usage 386 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 387#endif 388 } 389 AudioSystem::releaseOutput(thread->id()); 390 } 391 Mutex::Autolock _l(thread->mLock); 392 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 393 playbackThread->destroyTrack_l(this); 394 } 395 } 396} 397 398/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 399{ 400 result.append(" Name Client Type Fmt Chn mask Session StpCnt fCount S M F SRate " 401 "L dB R dB Server User Main buf Aux Buf Flags Underruns\n"); 402} 403 404void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 405{ 406 uint32_t vlr = mCblk->getVolumeLR(); 407 if (isFastTrack()) { 408 sprintf(buffer, " F %2d", mFastIndex); 409 } else { 410 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 411 } 412 track_state state = mState; 413 char stateChar; 414 switch (state) { 415 case IDLE: 416 stateChar = 'I'; 417 break; 418 case TERMINATED: 419 stateChar = 'T'; 420 break; 421 case STOPPING_1: 422 stateChar = 's'; 423 break; 424 case STOPPING_2: 425 stateChar = '5'; 426 break; 427 case STOPPED: 428 stateChar = 'S'; 429 break; 430 case RESUMING: 431 stateChar = 'R'; 432 break; 433 case ACTIVE: 434 stateChar = 'A'; 435 break; 436 case PAUSING: 437 stateChar = 'p'; 438 break; 439 case PAUSED: 440 stateChar = 'P'; 441 break; 442 case FLUSHED: 443 stateChar = 'F'; 444 break; 445 default: 446 stateChar = '?'; 447 break; 448 } 449 char nowInUnderrun; 450 switch (mObservedUnderruns.mBitFields.mMostRecent) { 451 case UNDERRUN_FULL: 452 nowInUnderrun = ' '; 453 break; 454 case UNDERRUN_PARTIAL: 455 nowInUnderrun = '<'; 456 break; 457 case UNDERRUN_EMPTY: 458 nowInUnderrun = '*'; 459 break; 460 default: 461 nowInUnderrun = '?'; 462 break; 463 } 464 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 465 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 466 (mClient == 0) ? getpid_cached : mClient->pid(), 467 mStreamType, 468 mFormat, 469 mChannelMask, 470 mSessionId, 471 mStepCount, 472 mFrameCount, 473 stateChar, 474 mMute, 475 mFillingUpStatus, 476 mCblk->sampleRate, 477 20.0 * log10((vlr & 0xFFFF) / 4096.0), 478 20.0 * log10((vlr >> 16) / 4096.0), 479 mCblk->server, 480 mCblk->user, 481 (int)mMainBuffer, 482 (int)mAuxBuffer, 483 mCblk->flags, 484 mUnderrunCount, 485 nowInUnderrun); 486} 487 488// AudioBufferProvider interface 489status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 490 AudioBufferProvider::Buffer* buffer, int64_t pts) 491{ 492 audio_track_cblk_t* cblk = this->cblk(); 493 uint32_t framesReady; 494 uint32_t framesReq = buffer->frameCount; 495 496 // Check if last stepServer failed, try to step now 497 if (mStepServerFailed) { 498 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 499 // Since the fast mixer is higher priority than client callback thread, 500 // it does not result in priority inversion for client. 501 // But a non-blocking solution would be preferable to avoid 502 // fast mixer being unable to tryLock(), and 503 // to avoid the extra context switches if the client wakes up, 504 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 505 if (!step()) goto getNextBuffer_exit; 506 ALOGV("stepServer recovered"); 507 mStepServerFailed = false; 508 } 509 510 // FIXME Same as above 511 framesReady = cblk->framesReadyOut(); 512 513 if (CC_LIKELY(framesReady)) { 514 uint32_t s = cblk->server; 515 uint32_t bufferEnd = cblk->serverBase + mFrameCount; 516 517 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 518 if (framesReq > framesReady) { 519 framesReq = framesReady; 520 } 521 if (framesReq > bufferEnd - s) { 522 framesReq = bufferEnd - s; 523 } 524 525 buffer->raw = getBuffer(s, framesReq); 526 buffer->frameCount = framesReq; 527 return NO_ERROR; 528 } 529 530getNextBuffer_exit: 531 buffer->raw = NULL; 532 buffer->frameCount = 0; 533 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 534 return NOT_ENOUGH_DATA; 535} 536 537// Note that framesReady() takes a mutex on the control block using tryLock(). 538// This could result in priority inversion if framesReady() is called by the normal mixer, 539// as the normal mixer thread runs at lower 540// priority than the client's callback thread: there is a short window within framesReady() 541// during which the normal mixer could be preempted, and the client callback would block. 542// Another problem can occur if framesReady() is called by the fast mixer: 543// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 544// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 545size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 546 return mCblk->framesReadyOut(); 547} 548 549// Don't call for fast tracks; the framesReady() could result in priority inversion 550bool AudioFlinger::PlaybackThread::Track::isReady() const { 551 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 552 return true; 553 } 554 555 if (framesReady() >= mFrameCount || 556 (mCblk->flags & CBLK_FORCEREADY)) { 557 mFillingUpStatus = FS_FILLED; 558 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 559 return true; 560 } 561 return false; 562} 563 564status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 565 int triggerSession) 566{ 567 status_t status = NO_ERROR; 568 ALOGV("start(%d), calling pid %d session %d", 569 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 570 571 sp<ThreadBase> thread = mThread.promote(); 572 if (thread != 0) { 573 Mutex::Autolock _l(thread->mLock); 574 track_state state = mState; 575 // here the track could be either new, or restarted 576 // in both cases "unstop" the track 577 if (mState == PAUSED) { 578 mState = TrackBase::RESUMING; 579 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 580 } else { 581 mState = TrackBase::ACTIVE; 582 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 583 } 584 585 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 586 thread->mLock.unlock(); 587 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 588 thread->mLock.lock(); 589 590#ifdef ADD_BATTERY_DATA 591 // to track the speaker usage 592 if (status == NO_ERROR) { 593 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 594 } 595#endif 596 } 597 if (status == NO_ERROR) { 598 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 599 playbackThread->addTrack_l(this); 600 } else { 601 mState = state; 602 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 603 } 604 } else { 605 status = BAD_VALUE; 606 } 607 return status; 608} 609 610void AudioFlinger::PlaybackThread::Track::stop() 611{ 612 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 613 sp<ThreadBase> thread = mThread.promote(); 614 if (thread != 0) { 615 Mutex::Autolock _l(thread->mLock); 616 track_state state = mState; 617 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 618 // If the track is not active (PAUSED and buffers full), flush buffers 619 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 620 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 621 reset(); 622 mState = STOPPED; 623 } else if (!isFastTrack()) { 624 mState = STOPPED; 625 } else { 626 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 627 // and then to STOPPED and reset() when presentation is complete 628 mState = STOPPING_1; 629 } 630 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 631 playbackThread); 632 } 633 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 634 thread->mLock.unlock(); 635 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 636 thread->mLock.lock(); 637 638#ifdef ADD_BATTERY_DATA 639 // to track the speaker usage 640 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 641#endif 642 } 643 } 644} 645 646void AudioFlinger::PlaybackThread::Track::pause() 647{ 648 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 649 sp<ThreadBase> thread = mThread.promote(); 650 if (thread != 0) { 651 Mutex::Autolock _l(thread->mLock); 652 if (mState == ACTIVE || mState == RESUMING) { 653 mState = PAUSING; 654 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 655 if (!isOutputTrack()) { 656 thread->mLock.unlock(); 657 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 658 thread->mLock.lock(); 659 660#ifdef ADD_BATTERY_DATA 661 // to track the speaker usage 662 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 663#endif 664 } 665 } 666 } 667} 668 669void AudioFlinger::PlaybackThread::Track::flush() 670{ 671 ALOGV("flush(%d)", mName); 672 sp<ThreadBase> thread = mThread.promote(); 673 if (thread != 0) { 674 Mutex::Autolock _l(thread->mLock); 675 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 676 mState != PAUSING && mState != IDLE && mState != FLUSHED) { 677 return; 678 } 679 // No point remaining in PAUSED state after a flush => go to 680 // FLUSHED state 681 mState = FLUSHED; 682 // do not reset the track if it is still in the process of being stopped or paused. 683 // this will be done by prepareTracks_l() when the track is stopped. 684 // prepareTracks_l() will see mState == FLUSHED, then 685 // remove from active track list, reset(), and trigger presentation complete 686 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 687 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 688 reset(); 689 } 690 } 691} 692 693void AudioFlinger::PlaybackThread::Track::reset() 694{ 695 // Do not reset twice to avoid discarding data written just after a flush and before 696 // the audioflinger thread detects the track is stopped. 697 if (!mResetDone) { 698 TrackBase::reset(); 699 // Force underrun condition to avoid false underrun callback until first data is 700 // written to buffer 701 android_atomic_and(~CBLK_FORCEREADY, &mCblk->flags); 702 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 703 mFillingUpStatus = FS_FILLING; 704 mResetDone = true; 705 if (mState == FLUSHED) { 706 mState = IDLE; 707 } 708 } 709} 710 711void AudioFlinger::PlaybackThread::Track::mute(bool muted) 712{ 713 mMute = muted; 714} 715 716status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 717{ 718 status_t status = DEAD_OBJECT; 719 sp<ThreadBase> thread = mThread.promote(); 720 if (thread != 0) { 721 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 722 sp<AudioFlinger> af = mClient->audioFlinger(); 723 724 Mutex::Autolock _l(af->mLock); 725 726 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 727 728 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 729 Mutex::Autolock _dl(playbackThread->mLock); 730 Mutex::Autolock _sl(srcThread->mLock); 731 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 732 if (chain == 0) { 733 return INVALID_OPERATION; 734 } 735 736 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 737 if (effect == 0) { 738 return INVALID_OPERATION; 739 } 740 srcThread->removeEffect_l(effect); 741 playbackThread->addEffect_l(effect); 742 // removeEffect_l() has stopped the effect if it was active so it must be restarted 743 if (effect->state() == EffectModule::ACTIVE || 744 effect->state() == EffectModule::STOPPING) { 745 effect->start(); 746 } 747 748 sp<EffectChain> dstChain = effect->chain().promote(); 749 if (dstChain == 0) { 750 srcThread->addEffect_l(effect); 751 return INVALID_OPERATION; 752 } 753 AudioSystem::unregisterEffect(effect->id()); 754 AudioSystem::registerEffect(&effect->desc(), 755 srcThread->id(), 756 dstChain->strategy(), 757 AUDIO_SESSION_OUTPUT_MIX, 758 effect->id()); 759 } 760 status = playbackThread->attachAuxEffect(this, EffectId); 761 } 762 return status; 763} 764 765void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 766{ 767 mAuxEffectId = EffectId; 768 mAuxBuffer = buffer; 769} 770 771bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 772 size_t audioHalFrames) 773{ 774 // a track is considered presented when the total number of frames written to audio HAL 775 // corresponds to the number of frames written when presentationComplete() is called for the 776 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 777 if (mPresentationCompleteFrames == 0) { 778 mPresentationCompleteFrames = framesWritten + audioHalFrames; 779 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 780 mPresentationCompleteFrames, audioHalFrames); 781 } 782 if (framesWritten >= mPresentationCompleteFrames) { 783 ALOGV("presentationComplete() session %d complete: framesWritten %d", 784 mSessionId, framesWritten); 785 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 786 return true; 787 } 788 return false; 789} 790 791void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 792{ 793 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 794 if (mSyncEvents[i]->type() == type) { 795 mSyncEvents[i]->trigger(); 796 mSyncEvents.removeAt(i); 797 i--; 798 } 799 } 800} 801 802// implement VolumeBufferProvider interface 803 804uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 805{ 806 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 807 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 808 uint32_t vlr = mCblk->getVolumeLR(); 809 uint32_t vl = vlr & 0xFFFF; 810 uint32_t vr = vlr >> 16; 811 // track volumes come from shared memory, so can't be trusted and must be clamped 812 if (vl > MAX_GAIN_INT) { 813 vl = MAX_GAIN_INT; 814 } 815 if (vr > MAX_GAIN_INT) { 816 vr = MAX_GAIN_INT; 817 } 818 // now apply the cached master volume and stream type volume; 819 // this is trusted but lacks any synchronization or barrier so may be stale 820 float v = mCachedVolume; 821 vl *= v; 822 vr *= v; 823 // re-combine into U4.16 824 vlr = (vr << 16) | (vl & 0xFFFF); 825 // FIXME look at mute, pause, and stop flags 826 return vlr; 827} 828 829status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 830{ 831 if (mState == TERMINATED || mState == PAUSED || 832 ((framesReady() == 0) && ((mSharedBuffer != 0) || 833 (mState == STOPPED)))) { 834 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 835 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 836 event->cancel(); 837 return INVALID_OPERATION; 838 } 839 (void) TrackBase::setSyncEvent(event); 840 return NO_ERROR; 841} 842 843bool AudioFlinger::PlaybackThread::Track::isOut() const 844{ 845 return true; 846} 847 848// ---------------------------------------------------------------------------- 849 850sp<AudioFlinger::PlaybackThread::TimedTrack> 851AudioFlinger::PlaybackThread::TimedTrack::create( 852 PlaybackThread *thread, 853 const sp<Client>& client, 854 audio_stream_type_t streamType, 855 uint32_t sampleRate, 856 audio_format_t format, 857 audio_channel_mask_t channelMask, 858 size_t frameCount, 859 const sp<IMemory>& sharedBuffer, 860 int sessionId) { 861 if (!client->reserveTimedTrack()) 862 return 0; 863 864 return new TimedTrack( 865 thread, client, streamType, sampleRate, format, channelMask, frameCount, 866 sharedBuffer, sessionId); 867} 868 869AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 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 : Track(thread, client, streamType, sampleRate, format, channelMask, 880 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 881 mQueueHeadInFlight(false), 882 mTrimQueueHeadOnRelease(false), 883 mFramesPendingInQueue(0), 884 mTimedSilenceBuffer(NULL), 885 mTimedSilenceBufferSize(0), 886 mTimedAudioOutputOnTime(false), 887 mMediaTimeTransformValid(false) 888{ 889 LocalClock lc; 890 mLocalTimeFreq = lc.getLocalFreq(); 891 892 mLocalTimeToSampleTransform.a_zero = 0; 893 mLocalTimeToSampleTransform.b_zero = 0; 894 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 895 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 896 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 897 &mLocalTimeToSampleTransform.a_to_b_denom); 898 899 mMediaTimeToSampleTransform.a_zero = 0; 900 mMediaTimeToSampleTransform.b_zero = 0; 901 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 902 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 903 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 904 &mMediaTimeToSampleTransform.a_to_b_denom); 905} 906 907AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 908 mClient->releaseTimedTrack(); 909 delete [] mTimedSilenceBuffer; 910} 911 912status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 913 size_t size, sp<IMemory>* buffer) { 914 915 Mutex::Autolock _l(mTimedBufferQueueLock); 916 917 trimTimedBufferQueue_l(); 918 919 // lazily initialize the shared memory heap for timed buffers 920 if (mTimedMemoryDealer == NULL) { 921 const int kTimedBufferHeapSize = 512 << 10; 922 923 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 924 "AudioFlingerTimed"); 925 if (mTimedMemoryDealer == NULL) 926 return NO_MEMORY; 927 } 928 929 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 930 if (newBuffer == NULL) { 931 newBuffer = mTimedMemoryDealer->allocate(size); 932 if (newBuffer == NULL) 933 return NO_MEMORY; 934 } 935 936 *buffer = newBuffer; 937 return NO_ERROR; 938} 939 940// caller must hold mTimedBufferQueueLock 941void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 942 int64_t mediaTimeNow; 943 { 944 Mutex::Autolock mttLock(mMediaTimeTransformLock); 945 if (!mMediaTimeTransformValid) 946 return; 947 948 int64_t targetTimeNow; 949 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 950 ? mCCHelper.getCommonTime(&targetTimeNow) 951 : mCCHelper.getLocalTime(&targetTimeNow); 952 953 if (OK != res) 954 return; 955 956 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 957 &mediaTimeNow)) { 958 return; 959 } 960 } 961 962 size_t trimEnd; 963 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 964 int64_t bufEnd; 965 966 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 967 // We have a next buffer. Just use its PTS as the PTS of the frame 968 // following the last frame in this buffer. If the stream is sparse 969 // (ie, there are deliberate gaps left in the stream which should be 970 // filled with silence by the TimedAudioTrack), then this can result 971 // in one extra buffer being left un-trimmed when it could have 972 // been. In general, this is not typical, and we would rather 973 // optimized away the TS calculation below for the more common case 974 // where PTSes are contiguous. 975 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 976 } else { 977 // We have no next buffer. Compute the PTS of the frame following 978 // the last frame in this buffer by computing the duration of of 979 // this frame in media time units and adding it to the PTS of the 980 // buffer. 981 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 982 / mFrameSize; 983 984 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 985 &bufEnd)) { 986 ALOGE("Failed to convert frame count of %lld to media time" 987 " duration" " (scale factor %d/%u) in %s", 988 frameCount, 989 mMediaTimeToSampleTransform.a_to_b_numer, 990 mMediaTimeToSampleTransform.a_to_b_denom, 991 __PRETTY_FUNCTION__); 992 break; 993 } 994 bufEnd += mTimedBufferQueue[trimEnd].pts(); 995 } 996 997 if (bufEnd > mediaTimeNow) 998 break; 999 1000 // Is the buffer we want to use in the middle of a mix operation right 1001 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1002 // from the mixer which should be coming back shortly. 1003 if (!trimEnd && mQueueHeadInFlight) { 1004 mTrimQueueHeadOnRelease = true; 1005 } 1006 } 1007 1008 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1009 if (trimStart < trimEnd) { 1010 // Update the bookkeeping for framesReady() 1011 for (size_t i = trimStart; i < trimEnd; ++i) { 1012 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1013 } 1014 1015 // Now actually remove the buffers from the queue. 1016 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1017 } 1018} 1019 1020void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1021 const char* logTag) { 1022 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1023 "%s called (reason \"%s\"), but timed buffer queue has no" 1024 " elements to trim.", __FUNCTION__, logTag); 1025 1026 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1027 mTimedBufferQueue.removeAt(0); 1028} 1029 1030void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1031 const TimedBuffer& buf, 1032 const char* logTag) { 1033 uint32_t bufBytes = buf.buffer()->size(); 1034 uint32_t consumedAlready = buf.position(); 1035 1036 ALOG_ASSERT(consumedAlready <= bufBytes, 1037 "Bad bookkeeping while updating frames pending. Timed buffer is" 1038 " only %u bytes long, but claims to have consumed %u" 1039 " bytes. (update reason: \"%s\")", 1040 bufBytes, consumedAlready, logTag); 1041 1042 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1043 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1044 "Bad bookkeeping while updating frames pending. Should have at" 1045 " least %u queued frames, but we think we have only %u. (update" 1046 " reason: \"%s\")", 1047 bufFrames, mFramesPendingInQueue, logTag); 1048 1049 mFramesPendingInQueue -= bufFrames; 1050} 1051 1052status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1053 const sp<IMemory>& buffer, int64_t pts) { 1054 1055 { 1056 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1057 if (!mMediaTimeTransformValid) 1058 return INVALID_OPERATION; 1059 } 1060 1061 Mutex::Autolock _l(mTimedBufferQueueLock); 1062 1063 uint32_t bufFrames = buffer->size() / mFrameSize; 1064 mFramesPendingInQueue += bufFrames; 1065 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1066 1067 return NO_ERROR; 1068} 1069 1070status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1071 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1072 1073 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1074 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1075 target); 1076 1077 if (!(target == TimedAudioTrack::LOCAL_TIME || 1078 target == TimedAudioTrack::COMMON_TIME)) { 1079 return BAD_VALUE; 1080 } 1081 1082 Mutex::Autolock lock(mMediaTimeTransformLock); 1083 mMediaTimeTransform = xform; 1084 mMediaTimeTransformTarget = target; 1085 mMediaTimeTransformValid = true; 1086 1087 return NO_ERROR; 1088} 1089 1090#define min(a, b) ((a) < (b) ? (a) : (b)) 1091 1092// implementation of getNextBuffer for tracks whose buffers have timestamps 1093status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1094 AudioBufferProvider::Buffer* buffer, int64_t pts) 1095{ 1096 if (pts == AudioBufferProvider::kInvalidPTS) { 1097 buffer->raw = NULL; 1098 buffer->frameCount = 0; 1099 mTimedAudioOutputOnTime = false; 1100 return INVALID_OPERATION; 1101 } 1102 1103 Mutex::Autolock _l(mTimedBufferQueueLock); 1104 1105 ALOG_ASSERT(!mQueueHeadInFlight, 1106 "getNextBuffer called without releaseBuffer!"); 1107 1108 while (true) { 1109 1110 // if we have no timed buffers, then fail 1111 if (mTimedBufferQueue.isEmpty()) { 1112 buffer->raw = NULL; 1113 buffer->frameCount = 0; 1114 return NOT_ENOUGH_DATA; 1115 } 1116 1117 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1118 1119 // calculate the PTS of the head of the timed buffer queue expressed in 1120 // local time 1121 int64_t headLocalPTS; 1122 { 1123 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1124 1125 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1126 1127 if (mMediaTimeTransform.a_to_b_denom == 0) { 1128 // the transform represents a pause, so yield silence 1129 timedYieldSilence_l(buffer->frameCount, buffer); 1130 return NO_ERROR; 1131 } 1132 1133 int64_t transformedPTS; 1134 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1135 &transformedPTS)) { 1136 // the transform failed. this shouldn't happen, but if it does 1137 // then just drop this buffer 1138 ALOGW("timedGetNextBuffer transform failed"); 1139 buffer->raw = NULL; 1140 buffer->frameCount = 0; 1141 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1142 return NO_ERROR; 1143 } 1144 1145 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1146 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1147 &headLocalPTS)) { 1148 buffer->raw = NULL; 1149 buffer->frameCount = 0; 1150 return INVALID_OPERATION; 1151 } 1152 } else { 1153 headLocalPTS = transformedPTS; 1154 } 1155 } 1156 1157 // adjust the head buffer's PTS to reflect the portion of the head buffer 1158 // that has already been consumed 1159 int64_t effectivePTS = headLocalPTS + 1160 ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate()); 1161 1162 // Calculate the delta in samples between the head of the input buffer 1163 // queue and the start of the next output buffer that will be written. 1164 // If the transformation fails because of over or underflow, it means 1165 // that the sample's position in the output stream is so far out of 1166 // whack that it should just be dropped. 1167 int64_t sampleDelta; 1168 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1169 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1170 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1171 " mix"); 1172 continue; 1173 } 1174 if (!mLocalTimeToSampleTransform.doForwardTransform( 1175 (effectivePTS - pts) << 32, &sampleDelta)) { 1176 ALOGV("*** too late during sample rate transform: dropped buffer"); 1177 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1178 continue; 1179 } 1180 1181 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1182 " sampleDelta=[%d.%08x]", 1183 head.pts(), head.position(), pts, 1184 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1185 + (sampleDelta >> 32)), 1186 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1187 1188 // if the delta between the ideal placement for the next input sample and 1189 // the current output position is within this threshold, then we will 1190 // concatenate the next input samples to the previous output 1191 const int64_t kSampleContinuityThreshold = 1192 (static_cast<int64_t>(sampleRate()) << 32) / 250; 1193 1194 // if this is the first buffer of audio that we're emitting from this track 1195 // then it should be almost exactly on time. 1196 const int64_t kSampleStartupThreshold = 1LL << 32; 1197 1198 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1199 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1200 // the next input is close enough to being on time, so concatenate it 1201 // with the last output 1202 timedYieldSamples_l(buffer); 1203 1204 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1205 head.position(), buffer->frameCount); 1206 return NO_ERROR; 1207 } 1208 1209 // Looks like our output is not on time. Reset our on timed status. 1210 // Next time we mix samples from our input queue, then should be within 1211 // the StartupThreshold. 1212 mTimedAudioOutputOnTime = false; 1213 if (sampleDelta > 0) { 1214 // the gap between the current output position and the proper start of 1215 // the next input sample is too big, so fill it with silence 1216 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1217 1218 timedYieldSilence_l(framesUntilNextInput, buffer); 1219 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1220 return NO_ERROR; 1221 } else { 1222 // the next input sample is late 1223 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1224 size_t onTimeSamplePosition = 1225 head.position() + lateFrames * mFrameSize; 1226 1227 if (onTimeSamplePosition > head.buffer()->size()) { 1228 // all the remaining samples in the head are too late, so 1229 // drop it and move on 1230 ALOGV("*** too late: dropped buffer"); 1231 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1232 continue; 1233 } else { 1234 // skip over the late samples 1235 head.setPosition(onTimeSamplePosition); 1236 1237 // yield the available samples 1238 timedYieldSamples_l(buffer); 1239 1240 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1241 return NO_ERROR; 1242 } 1243 } 1244 } 1245} 1246 1247// Yield samples from the timed buffer queue head up to the given output 1248// buffer's capacity. 1249// 1250// Caller must hold mTimedBufferQueueLock 1251void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1252 AudioBufferProvider::Buffer* buffer) { 1253 1254 const TimedBuffer& head = mTimedBufferQueue[0]; 1255 1256 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1257 head.position()); 1258 1259 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1260 mFrameSize); 1261 size_t framesRequested = buffer->frameCount; 1262 buffer->frameCount = min(framesLeftInHead, framesRequested); 1263 1264 mQueueHeadInFlight = true; 1265 mTimedAudioOutputOnTime = true; 1266} 1267 1268// Yield samples of silence up to the given output buffer's capacity 1269// 1270// Caller must hold mTimedBufferQueueLock 1271void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1272 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1273 1274 // lazily allocate a buffer filled with silence 1275 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1276 delete [] mTimedSilenceBuffer; 1277 mTimedSilenceBufferSize = numFrames * mFrameSize; 1278 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1279 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1280 } 1281 1282 buffer->raw = mTimedSilenceBuffer; 1283 size_t framesRequested = buffer->frameCount; 1284 buffer->frameCount = min(numFrames, framesRequested); 1285 1286 mTimedAudioOutputOnTime = false; 1287} 1288 1289// AudioBufferProvider interface 1290void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1291 AudioBufferProvider::Buffer* buffer) { 1292 1293 Mutex::Autolock _l(mTimedBufferQueueLock); 1294 1295 // If the buffer which was just released is part of the buffer at the head 1296 // of the queue, be sure to update the amt of the buffer which has been 1297 // consumed. If the buffer being returned is not part of the head of the 1298 // queue, its either because the buffer is part of the silence buffer, or 1299 // because the head of the timed queue was trimmed after the mixer called 1300 // getNextBuffer but before the mixer called releaseBuffer. 1301 if (buffer->raw == mTimedSilenceBuffer) { 1302 ALOG_ASSERT(!mQueueHeadInFlight, 1303 "Queue head in flight during release of silence buffer!"); 1304 goto done; 1305 } 1306 1307 ALOG_ASSERT(mQueueHeadInFlight, 1308 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1309 " head in flight."); 1310 1311 if (mTimedBufferQueue.size()) { 1312 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1313 1314 void* start = head.buffer()->pointer(); 1315 void* end = reinterpret_cast<void*>( 1316 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1317 + head.buffer()->size()); 1318 1319 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1320 "released buffer not within the head of the timed buffer" 1321 " queue; qHead = [%p, %p], released buffer = %p", 1322 start, end, buffer->raw); 1323 1324 head.setPosition(head.position() + 1325 (buffer->frameCount * mFrameSize)); 1326 mQueueHeadInFlight = false; 1327 1328 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1329 "Bad bookkeeping during releaseBuffer! Should have at" 1330 " least %u queued frames, but we think we have only %u", 1331 buffer->frameCount, mFramesPendingInQueue); 1332 1333 mFramesPendingInQueue -= buffer->frameCount; 1334 1335 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1336 || mTrimQueueHeadOnRelease) { 1337 trimTimedBufferQueueHead_l("releaseBuffer"); 1338 mTrimQueueHeadOnRelease = false; 1339 } 1340 } else { 1341 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1342 " buffers in the timed buffer queue"); 1343 } 1344 1345done: 1346 buffer->raw = 0; 1347 buffer->frameCount = 0; 1348} 1349 1350size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1351 Mutex::Autolock _l(mTimedBufferQueueLock); 1352 return mFramesPendingInQueue; 1353} 1354 1355AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1356 : mPTS(0), mPosition(0) {} 1357 1358AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1359 const sp<IMemory>& buffer, int64_t pts) 1360 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1361 1362 1363// ---------------------------------------------------------------------------- 1364 1365AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1366 PlaybackThread *playbackThread, 1367 DuplicatingThread *sourceThread, 1368 uint32_t sampleRate, 1369 audio_format_t format, 1370 audio_channel_mask_t channelMask, 1371 size_t frameCount) 1372 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1373 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 1374 mActive(false), mSourceThread(sourceThread), mBuffers(NULL) 1375{ 1376 1377 if (mCblk != NULL) { 1378 mBuffers = (char*)mCblk + sizeof(audio_track_cblk_t); 1379 mOutBuffer.frameCount = 0; 1380 playbackThread->mTracks.add(this); 1381 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mBuffers %p, " \ 1382 "mCblk->frameCount %d, mCblk->sampleRate %u, mChannelMask 0x%08x mBufferEnd %p", 1383 mCblk, mBuffer, mBuffers, 1384 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 1385 } else { 1386 ALOGW("Error creating output track on thread %p", playbackThread); 1387 } 1388} 1389 1390AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1391{ 1392 clearBufferQueue(); 1393} 1394 1395status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1396 int triggerSession) 1397{ 1398 status_t status = Track::start(event, triggerSession); 1399 if (status != NO_ERROR) { 1400 return status; 1401 } 1402 1403 mActive = true; 1404 mRetryCount = 127; 1405 return status; 1406} 1407 1408void AudioFlinger::PlaybackThread::OutputTrack::stop() 1409{ 1410 Track::stop(); 1411 clearBufferQueue(); 1412 mOutBuffer.frameCount = 0; 1413 mActive = false; 1414} 1415 1416bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1417{ 1418 Buffer *pInBuffer; 1419 Buffer inBuffer; 1420 uint32_t channelCount = mChannelCount; 1421 bool outputBufferFull = false; 1422 inBuffer.frameCount = frames; 1423 inBuffer.i16 = data; 1424 1425 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1426 1427 if (!mActive && frames != 0) { 1428 start(); 1429 sp<ThreadBase> thread = mThread.promote(); 1430 if (thread != 0) { 1431 MixerThread *mixerThread = (MixerThread *)thread.get(); 1432 if (mFrameCount > frames) { 1433 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1434 uint32_t startFrames = (mFrameCount - frames); 1435 pInBuffer = new Buffer; 1436 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1437 pInBuffer->frameCount = startFrames; 1438 pInBuffer->i16 = pInBuffer->mBuffer; 1439 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1440 mBufferQueue.add(pInBuffer); 1441 } else { 1442 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 1443 } 1444 } 1445 } 1446 } 1447 1448 while (waitTimeLeftMs) { 1449 // First write pending buffers, then new data 1450 if (mBufferQueue.size()) { 1451 pInBuffer = mBufferQueue.itemAt(0); 1452 } else { 1453 pInBuffer = &inBuffer; 1454 } 1455 1456 if (pInBuffer->frameCount == 0) { 1457 break; 1458 } 1459 1460 if (mOutBuffer.frameCount == 0) { 1461 mOutBuffer.frameCount = pInBuffer->frameCount; 1462 nsecs_t startTime = systemTime(); 1463 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 1464 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, 1465 mThread.unsafe_get()); 1466 outputBufferFull = true; 1467 break; 1468 } 1469 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1470 if (waitTimeLeftMs >= waitTimeMs) { 1471 waitTimeLeftMs -= waitTimeMs; 1472 } else { 1473 waitTimeLeftMs = 0; 1474 } 1475 } 1476 1477 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1478 pInBuffer->frameCount; 1479 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1480 mCblk->stepUserOut(outFrames, mFrameCount); 1481 pInBuffer->frameCount -= outFrames; 1482 pInBuffer->i16 += outFrames * channelCount; 1483 mOutBuffer.frameCount -= outFrames; 1484 mOutBuffer.i16 += outFrames * channelCount; 1485 1486 if (pInBuffer->frameCount == 0) { 1487 if (mBufferQueue.size()) { 1488 mBufferQueue.removeAt(0); 1489 delete [] pInBuffer->mBuffer; 1490 delete pInBuffer; 1491 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1492 mThread.unsafe_get(), mBufferQueue.size()); 1493 } else { 1494 break; 1495 } 1496 } 1497 } 1498 1499 // If we could not write all frames, allocate a buffer and queue it for next time. 1500 if (inBuffer.frameCount) { 1501 sp<ThreadBase> thread = mThread.promote(); 1502 if (thread != 0 && !thread->standby()) { 1503 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1504 pInBuffer = new Buffer; 1505 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1506 pInBuffer->frameCount = inBuffer.frameCount; 1507 pInBuffer->i16 = pInBuffer->mBuffer; 1508 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1509 sizeof(int16_t)); 1510 mBufferQueue.add(pInBuffer); 1511 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1512 mThread.unsafe_get(), mBufferQueue.size()); 1513 } else { 1514 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1515 mThread.unsafe_get(), this); 1516 } 1517 } 1518 } 1519 1520 // Calling write() with a 0 length buffer, means that no more data will be written: 1521 // If no more buffers are pending, fill output track buffer to make sure it is started 1522 // by output mixer. 1523 if (frames == 0 && mBufferQueue.size() == 0) { 1524 if (mCblk->user < mFrameCount) { 1525 frames = mFrameCount - mCblk->user; 1526 pInBuffer = new Buffer; 1527 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1528 pInBuffer->frameCount = frames; 1529 pInBuffer->i16 = pInBuffer->mBuffer; 1530 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1531 mBufferQueue.add(pInBuffer); 1532 } else if (mActive) { 1533 stop(); 1534 } 1535 } 1536 1537 return outputBufferFull; 1538} 1539 1540status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1541 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1542{ 1543 int active; 1544 status_t result; 1545 audio_track_cblk_t* cblk = mCblk; 1546 uint32_t framesReq = buffer->frameCount; 1547 1548 ALOGVV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 1549 buffer->frameCount = 0; 1550 1551 uint32_t framesAvail = cblk->framesAvailableOut(mFrameCount); 1552 1553 1554 if (framesAvail == 0) { 1555 Mutex::Autolock _l(cblk->lock); 1556 goto start_loop_here; 1557 while (framesAvail == 0) { 1558 active = mActive; 1559 if (CC_UNLIKELY(!active)) { 1560 ALOGV("Not active and NO_MORE_BUFFERS"); 1561 return NO_MORE_BUFFERS; 1562 } 1563 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 1564 if (result != NO_ERROR) { 1565 return NO_MORE_BUFFERS; 1566 } 1567 // read the server count again 1568 start_loop_here: 1569 framesAvail = cblk->framesAvailableOut_l(mFrameCount); 1570 } 1571 } 1572 1573// if (framesAvail < framesReq) { 1574// return NO_MORE_BUFFERS; 1575// } 1576 1577 if (framesReq > framesAvail) { 1578 framesReq = framesAvail; 1579 } 1580 1581 uint32_t u = cblk->user; 1582 uint32_t bufferEnd = cblk->userBase + mFrameCount; 1583 1584 if (framesReq > bufferEnd - u) { 1585 framesReq = bufferEnd - u; 1586 } 1587 1588 buffer->frameCount = framesReq; 1589 buffer->raw = cblk->buffer(mBuffers, mFrameSize, u); 1590 return NO_ERROR; 1591} 1592 1593 1594void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1595{ 1596 size_t size = mBufferQueue.size(); 1597 1598 for (size_t i = 0; i < size; i++) { 1599 Buffer *pBuffer = mBufferQueue.itemAt(i); 1600 delete [] pBuffer->mBuffer; 1601 delete pBuffer; 1602 } 1603 mBufferQueue.clear(); 1604} 1605 1606 1607// ---------------------------------------------------------------------------- 1608// Record 1609// ---------------------------------------------------------------------------- 1610 1611AudioFlinger::RecordHandle::RecordHandle( 1612 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1613 : BnAudioRecord(), 1614 mRecordTrack(recordTrack) 1615{ 1616} 1617 1618AudioFlinger::RecordHandle::~RecordHandle() { 1619 stop_nonvirtual(); 1620 mRecordTrack->destroy(); 1621} 1622 1623sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 1624 return mRecordTrack->getCblk(); 1625} 1626 1627status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1628 int triggerSession) { 1629 ALOGV("RecordHandle::start()"); 1630 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1631} 1632 1633void AudioFlinger::RecordHandle::stop() { 1634 stop_nonvirtual(); 1635} 1636 1637void AudioFlinger::RecordHandle::stop_nonvirtual() { 1638 ALOGV("RecordHandle::stop()"); 1639 mRecordTrack->stop(); 1640} 1641 1642status_t AudioFlinger::RecordHandle::onTransact( 1643 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1644{ 1645 return BnAudioRecord::onTransact(code, data, reply, flags); 1646} 1647 1648// ---------------------------------------------------------------------------- 1649 1650// RecordTrack constructor must be called with AudioFlinger::mLock held 1651AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1652 RecordThread *thread, 1653 const sp<Client>& client, 1654 uint32_t sampleRate, 1655 audio_format_t format, 1656 audio_channel_mask_t channelMask, 1657 size_t frameCount, 1658 int sessionId) 1659 : TrackBase(thread, client, sampleRate, format, 1660 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 1661 mOverflow(false) 1662{ 1663 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 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 audio_track_cblk_t* cblk = this->cblk(); 1676 uint32_t framesAvail; 1677 uint32_t framesReq = buffer->frameCount; 1678 1679 // Check if last stepServer failed, try to step now 1680 if (mStepServerFailed) { 1681 if (!step()) { 1682 goto getNextBuffer_exit; 1683 } 1684 ALOGV("stepServer recovered"); 1685 mStepServerFailed = false; 1686 } 1687 1688 // FIXME lock is not actually held, so overrun is possible 1689 framesAvail = cblk->framesAvailableIn_l(mFrameCount); 1690 1691 if (CC_LIKELY(framesAvail)) { 1692 uint32_t s = cblk->server; 1693 uint32_t bufferEnd = cblk->serverBase + mFrameCount; 1694 1695 if (framesReq > framesAvail) { 1696 framesReq = framesAvail; 1697 } 1698 if (framesReq > bufferEnd - s) { 1699 framesReq = bufferEnd - s; 1700 } 1701 1702 buffer->raw = getBuffer(s, framesReq); 1703 buffer->frameCount = framesReq; 1704 return NO_ERROR; 1705 } 1706 1707getNextBuffer_exit: 1708 buffer->raw = NULL; 1709 buffer->frameCount = 0; 1710 return NOT_ENOUGH_DATA; 1711} 1712 1713status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 1714 int triggerSession) 1715{ 1716 sp<ThreadBase> thread = mThread.promote(); 1717 if (thread != 0) { 1718 RecordThread *recordThread = (RecordThread *)thread.get(); 1719 return recordThread->start(this, event, triggerSession); 1720 } else { 1721 return BAD_VALUE; 1722 } 1723} 1724 1725void AudioFlinger::RecordThread::RecordTrack::stop() 1726{ 1727 sp<ThreadBase> thread = mThread.promote(); 1728 if (thread != 0) { 1729 RecordThread *recordThread = (RecordThread *)thread.get(); 1730 recordThread->mLock.lock(); 1731 bool doStop = recordThread->stop_l(this); 1732 if (doStop) { 1733 TrackBase::reset(); 1734 // Force overrun condition to avoid false overrun callback until first data is 1735 // read from buffer 1736 android_atomic_or(CBLK_UNDERRUN, &mCblk->flags); 1737 } 1738 recordThread->mLock.unlock(); 1739 if (doStop) { 1740 AudioSystem::stopInput(recordThread->id()); 1741 } 1742 } 1743} 1744 1745void AudioFlinger::RecordThread::RecordTrack::destroy() 1746{ 1747 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 1748 sp<RecordTrack> keep(this); 1749 { 1750 sp<ThreadBase> thread = mThread.promote(); 1751 if (thread != 0) { 1752 if (mState == ACTIVE || mState == RESUMING) { 1753 AudioSystem::stopInput(thread->id()); 1754 } 1755 AudioSystem::releaseInput(thread->id()); 1756 Mutex::Autolock _l(thread->mLock); 1757 RecordThread *recordThread = (RecordThread *) thread.get(); 1758 recordThread->destroyTrack_l(this); 1759 } 1760 } 1761} 1762 1763 1764/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 1765{ 1766 result.append(" Clien Fmt Chn mask Session Step S SRate Serv User FrameCount\n"); 1767} 1768 1769void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 1770{ 1771 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x %05d\n", 1772 (mClient == 0) ? getpid_cached : mClient->pid(), 1773 mFormat, 1774 mChannelMask, 1775 mSessionId, 1776 mStepCount, 1777 mState, 1778 mCblk->sampleRate, 1779 mCblk->server, 1780 mCblk->user, 1781 mFrameCount); 1782} 1783 1784bool AudioFlinger::RecordThread::RecordTrack::isOut() const 1785{ 1786 return false; 1787} 1788 1789}; // namespace android 1790