Tracks.cpp revision 3bcffa136909c1fb6e88ee4efd12ccac18360a85
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 <sys/syscall.h> 25#include <utils/Log.h> 26 27#include <private/media/AudioTrackShared.h> 28 29#include <common_time/cc_helper.h> 30#include <common_time/local_clock.h> 31 32#include "AudioMixer.h" 33#include "AudioFlinger.h" 34#include "ServiceUtilities.h" 35 36#include <media/nbaio/Pipe.h> 37#include <media/nbaio/PipeReader.h> 38#include <audio_utils/minifloat.h> 39 40// ---------------------------------------------------------------------------- 41 42// Note: the following macro is used for extremely verbose logging message. In 43// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 44// 0; but one side effect of this is to turn all LOGV's as well. Some messages 45// are so verbose that we want to suppress them even when we have ALOG_ASSERT 46// turned on. Do not uncomment the #def below unless you really know what you 47// are doing and want to see all of the extremely verbose messages. 48//#define VERY_VERY_VERBOSE_LOGGING 49#ifdef VERY_VERY_VERBOSE_LOGGING 50#define ALOGVV ALOGV 51#else 52#define ALOGVV(a...) do { } while(0) 53#endif 54 55namespace android { 56 57// ---------------------------------------------------------------------------- 58// TrackBase 59// ---------------------------------------------------------------------------- 60 61static volatile int32_t nextTrackId = 55; 62 63// TrackBase constructor must be called with AudioFlinger::mLock held 64AudioFlinger::ThreadBase::TrackBase::TrackBase( 65 ThreadBase *thread, 66 const sp<Client>& client, 67 uint32_t sampleRate, 68 audio_format_t format, 69 audio_channel_mask_t channelMask, 70 size_t frameCount, 71 const sp<IMemory>& sharedBuffer, 72 int sessionId, 73 int clientUid, 74 IAudioFlinger::track_flags_t flags, 75 bool isOut, 76 alloc_type alloc) 77 : RefBase(), 78 mThread(thread), 79 mClient(client), 80 mCblk(NULL), 81 // mBuffer 82 mState(IDLE), 83 mSampleRate(sampleRate), 84 mFormat(format), 85 mChannelMask(channelMask), 86 mChannelCount(isOut ? 87 audio_channel_count_from_out_mask(channelMask) : 88 audio_channel_count_from_in_mask(channelMask)), 89 mFrameSize(audio_is_linear_pcm(format) ? 90 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 91 mFrameCount(frameCount), 92 mSessionId(sessionId), 93 mFlags(flags), 94 mIsOut(isOut), 95 mServerProxy(NULL), 96 mId(android_atomic_inc(&nextTrackId)), 97 mTerminated(false) 98{ 99 // if the caller is us, trust the specified uid 100 if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) { 101 int newclientUid = IPCThreadState::self()->getCallingUid(); 102 if (clientUid != -1 && clientUid != newclientUid) { 103 ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid); 104 } 105 clientUid = newclientUid; 106 } 107 // clientUid contains the uid of the app that is responsible for this track, so we can blame 108 // battery usage on it. 109 mUid = clientUid; 110 111 // client == 0 implies sharedBuffer == 0 112 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 113 114 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 115 sharedBuffer->size()); 116 117 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 118 size_t size = sizeof(audio_track_cblk_t); 119 size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize; 120 if (sharedBuffer == 0 && alloc == ALLOC_CBLK) { 121 size += bufferSize; 122 } 123 124 if (client != 0) { 125 mCblkMemory = client->heap()->allocate(size); 126 if (mCblkMemory == 0 || 127 (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) { 128 ALOGE("not enough memory for AudioTrack size=%u", size); 129 client->heap()->dump("AudioTrack"); 130 mCblkMemory.clear(); 131 return; 132 } 133 } else { 134 // this syntax avoids calling the audio_track_cblk_t constructor twice 135 mCblk = (audio_track_cblk_t *) new uint8_t[size]; 136 // assume mCblk != NULL 137 } 138 139 // construct the shared structure in-place. 140 if (mCblk != NULL) { 141 new(mCblk) audio_track_cblk_t(); 142 switch (alloc) { 143 case ALLOC_READONLY: { 144 const sp<MemoryDealer> roHeap(thread->readOnlyHeap()); 145 if (roHeap == 0 || 146 (mBufferMemory = roHeap->allocate(bufferSize)) == 0 || 147 (mBuffer = mBufferMemory->pointer()) == NULL) { 148 ALOGE("not enough memory for read-only buffer size=%zu", bufferSize); 149 if (roHeap != 0) { 150 roHeap->dump("buffer"); 151 } 152 mCblkMemory.clear(); 153 mBufferMemory.clear(); 154 return; 155 } 156 memset(mBuffer, 0, bufferSize); 157 } break; 158 case ALLOC_PIPE: 159 mBufferMemory = thread->pipeMemory(); 160 // mBuffer is the virtual address as seen from current process (mediaserver), 161 // and should normally be coming from mBufferMemory->pointer(). 162 // However in this case the TrackBase does not reference the buffer directly. 163 // It should references the buffer via the pipe. 164 // Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL. 165 mBuffer = NULL; 166 break; 167 case ALLOC_CBLK: 168 // clear all buffers 169 if (sharedBuffer == 0) { 170 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 171 memset(mBuffer, 0, bufferSize); 172 } else { 173 mBuffer = sharedBuffer->pointer(); 174#if 0 175 mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic 176#endif 177 } 178 break; 179 } 180 181#ifdef TEE_SINK 182 if (mTeeSinkTrackEnabled) { 183 NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount); 184 if (Format_isValid(pipeFormat)) { 185 Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat); 186 size_t numCounterOffers = 0; 187 const NBAIO_Format offers[1] = {pipeFormat}; 188 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 189 ALOG_ASSERT(index == 0); 190 PipeReader *pipeReader = new PipeReader(*pipe); 191 numCounterOffers = 0; 192 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 193 ALOG_ASSERT(index == 0); 194 mTeeSink = pipe; 195 mTeeSource = pipeReader; 196 } 197 } 198#endif 199 200 } 201} 202 203AudioFlinger::ThreadBase::TrackBase::~TrackBase() 204{ 205#ifdef TEE_SINK 206 dumpTee(-1, mTeeSource, mId); 207#endif 208 // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference 209 delete mServerProxy; 210 if (mCblk != NULL) { 211 if (mClient == 0) { 212 delete mCblk; 213 } else { 214 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 215 } 216 } 217 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 218 if (mClient != 0) { 219 // Client destructor must run with AudioFlinger client mutex locked 220 Mutex::Autolock _l(mClient->audioFlinger()->mClientLock); 221 // If the client's reference count drops to zero, the associated destructor 222 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 223 // relying on the automatic clear() at end of scope. 224 mClient.clear(); 225 } 226 // flush the binder command buffer 227 IPCThreadState::self()->flushCommands(); 228} 229 230// AudioBufferProvider interface 231// getNextBuffer() = 0; 232// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 233void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 234{ 235#ifdef TEE_SINK 236 if (mTeeSink != 0) { 237 (void) mTeeSink->write(buffer->raw, buffer->frameCount); 238 } 239#endif 240 241 ServerProxy::Buffer buf; 242 buf.mFrameCount = buffer->frameCount; 243 buf.mRaw = buffer->raw; 244 buffer->frameCount = 0; 245 buffer->raw = NULL; 246 mServerProxy->releaseBuffer(&buf); 247} 248 249status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 250{ 251 mSyncEvents.add(event); 252 return NO_ERROR; 253} 254 255// ---------------------------------------------------------------------------- 256// Playback 257// ---------------------------------------------------------------------------- 258 259AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 260 : BnAudioTrack(), 261 mTrack(track) 262{ 263} 264 265AudioFlinger::TrackHandle::~TrackHandle() { 266 // just stop the track on deletion, associated resources 267 // will be freed from the main thread once all pending buffers have 268 // been played. Unless it's not in the active track list, in which 269 // case we free everything now... 270 mTrack->destroy(); 271} 272 273sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 274 return mTrack->getCblk(); 275} 276 277status_t AudioFlinger::TrackHandle::start() { 278 return mTrack->start(); 279} 280 281void AudioFlinger::TrackHandle::stop() { 282 mTrack->stop(); 283} 284 285void AudioFlinger::TrackHandle::flush() { 286 mTrack->flush(); 287} 288 289void AudioFlinger::TrackHandle::pause() { 290 mTrack->pause(); 291} 292 293status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 294{ 295 return mTrack->attachAuxEffect(EffectId); 296} 297 298status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 299 sp<IMemory>* buffer) { 300 if (!mTrack->isTimedTrack()) 301 return INVALID_OPERATION; 302 303 PlaybackThread::TimedTrack* tt = 304 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 305 return tt->allocateTimedBuffer(size, buffer); 306} 307 308status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 309 int64_t pts) { 310 if (!mTrack->isTimedTrack()) 311 return INVALID_OPERATION; 312 313 if (buffer == 0 || buffer->pointer() == NULL) { 314 ALOGE("queueTimedBuffer() buffer is 0 or has NULL pointer()"); 315 return BAD_VALUE; 316 } 317 318 PlaybackThread::TimedTrack* tt = 319 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 320 return tt->queueTimedBuffer(buffer, pts); 321} 322 323status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 324 const LinearTransform& xform, int target) { 325 326 if (!mTrack->isTimedTrack()) 327 return INVALID_OPERATION; 328 329 PlaybackThread::TimedTrack* tt = 330 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 331 return tt->setMediaTimeTransform( 332 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 333} 334 335status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) { 336 return mTrack->setParameters(keyValuePairs); 337} 338 339status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp) 340{ 341 return mTrack->getTimestamp(timestamp); 342} 343 344 345void AudioFlinger::TrackHandle::signal() 346{ 347 return mTrack->signal(); 348} 349 350status_t AudioFlinger::TrackHandle::onTransact( 351 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 352{ 353 return BnAudioTrack::onTransact(code, data, reply, flags); 354} 355 356// ---------------------------------------------------------------------------- 357 358// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 359AudioFlinger::PlaybackThread::Track::Track( 360 PlaybackThread *thread, 361 const sp<Client>& client, 362 audio_stream_type_t streamType, 363 uint32_t sampleRate, 364 audio_format_t format, 365 audio_channel_mask_t channelMask, 366 size_t frameCount, 367 const sp<IMemory>& sharedBuffer, 368 int sessionId, 369 int uid, 370 IAudioFlinger::track_flags_t flags) 371 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 372 sessionId, uid, flags, true /*isOut*/), 373 mFillingUpStatus(FS_INVALID), 374 // mRetryCount initialized later when needed 375 mSharedBuffer(sharedBuffer), 376 mStreamType(streamType), 377 mName(-1), // see note below 378 mMainBuffer(thread->mixBuffer()), 379 mAuxBuffer(NULL), 380 mAuxEffectId(0), mHasVolumeController(false), 381 mPresentationCompleteFrames(0), 382 mFastIndex(-1), 383 mCachedVolume(1.0), 384 mIsInvalid(false), 385 mAudioTrackServerProxy(NULL), 386 mResumeToStopping(false), 387 mFlushHwPending(false), 388 mPreviousValid(false), 389 mPreviousFramesWritten(0) 390 // mPreviousTimestamp 391{ 392 if (mCblk == NULL) { 393 return; 394 } 395 396 if (sharedBuffer == 0) { 397 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 398 mFrameSize); 399 } else { 400 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, 401 mFrameSize); 402 } 403 mServerProxy = mAudioTrackServerProxy; 404 405 mName = thread->getTrackName_l(channelMask, format, sessionId); 406 if (mName < 0) { 407 ALOGE("no more track names available"); 408 return; 409 } 410 // only allocate a fast track index if we were able to allocate a normal track name 411 if (flags & IAudioFlinger::TRACK_FAST) { 412 mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); 413 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 414 int i = __builtin_ctz(thread->mFastTrackAvailMask); 415 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 416 // FIXME This is too eager. We allocate a fast track index before the 417 // fast track becomes active. Since fast tracks are a scarce resource, 418 // this means we are potentially denying other more important fast tracks from 419 // being created. It would be better to allocate the index dynamically. 420 mFastIndex = i; 421 // Read the initial underruns because this field is never cleared by the fast mixer 422 mObservedUnderruns = thread->getFastTrackUnderruns(i); 423 thread->mFastTrackAvailMask &= ~(1 << i); 424 } 425} 426 427AudioFlinger::PlaybackThread::Track::~Track() 428{ 429 ALOGV("PlaybackThread::Track destructor"); 430 431 // The destructor would clear mSharedBuffer, 432 // but it will not push the decremented reference count, 433 // leaving the client's IMemory dangling indefinitely. 434 // This prevents that leak. 435 if (mSharedBuffer != 0) { 436 mSharedBuffer.clear(); 437 } 438} 439 440status_t AudioFlinger::PlaybackThread::Track::initCheck() const 441{ 442 status_t status = TrackBase::initCheck(); 443 if (status == NO_ERROR && mName < 0) { 444 status = NO_MEMORY; 445 } 446 return status; 447} 448 449void AudioFlinger::PlaybackThread::Track::destroy() 450{ 451 // NOTE: destroyTrack_l() can remove a strong reference to this Track 452 // by removing it from mTracks vector, so there is a risk that this Tracks's 453 // destructor is called. As the destructor needs to lock mLock, 454 // we must acquire a strong reference on this Track before locking mLock 455 // here so that the destructor is called only when exiting this function. 456 // On the other hand, as long as Track::destroy() is only called by 457 // TrackHandle destructor, the TrackHandle still holds a strong ref on 458 // this Track with its member mTrack. 459 sp<Track> keep(this); 460 { // scope for mLock 461 sp<ThreadBase> thread = mThread.promote(); 462 if (thread != 0) { 463 Mutex::Autolock _l(thread->mLock); 464 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 465 bool wasActive = playbackThread->destroyTrack_l(this); 466 if (!isOutputTrack() && !wasActive) { 467 AudioSystem::releaseOutput(thread->id()); 468 } 469 } 470 } 471} 472 473/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 474{ 475 result.append(" Name Active Client Type Fmt Chn mask Session fCount S F SRate " 476 "L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n"); 477} 478 479void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active) 480{ 481 gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR(); 482 if (isFastTrack()) { 483 sprintf(buffer, " F %2d", mFastIndex); 484 } else if (mName >= AudioMixer::TRACK0) { 485 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 486 } else { 487 sprintf(buffer, " none"); 488 } 489 track_state state = mState; 490 char stateChar; 491 if (isTerminated()) { 492 stateChar = 'T'; 493 } else { 494 switch (state) { 495 case IDLE: 496 stateChar = 'I'; 497 break; 498 case STOPPING_1: 499 stateChar = 's'; 500 break; 501 case STOPPING_2: 502 stateChar = '5'; 503 break; 504 case STOPPED: 505 stateChar = 'S'; 506 break; 507 case RESUMING: 508 stateChar = 'R'; 509 break; 510 case ACTIVE: 511 stateChar = 'A'; 512 break; 513 case PAUSING: 514 stateChar = 'p'; 515 break; 516 case PAUSED: 517 stateChar = 'P'; 518 break; 519 case FLUSHED: 520 stateChar = 'F'; 521 break; 522 default: 523 stateChar = '?'; 524 break; 525 } 526 } 527 char nowInUnderrun; 528 switch (mObservedUnderruns.mBitFields.mMostRecent) { 529 case UNDERRUN_FULL: 530 nowInUnderrun = ' '; 531 break; 532 case UNDERRUN_PARTIAL: 533 nowInUnderrun = '<'; 534 break; 535 case UNDERRUN_EMPTY: 536 nowInUnderrun = '*'; 537 break; 538 default: 539 nowInUnderrun = '?'; 540 break; 541 } 542 snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g " 543 "%08X %p %p 0x%03X %9u%c\n", 544 active ? "yes" : "no", 545 (mClient == 0) ? getpid_cached : mClient->pid(), 546 mStreamType, 547 mFormat, 548 mChannelMask, 549 mSessionId, 550 mFrameCount, 551 stateChar, 552 mFillingUpStatus, 553 mAudioTrackServerProxy->getSampleRate(), 554 20.0 * log10(float_from_gain(gain_minifloat_unpack_left(vlr))), 555 20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))), 556 mCblk->mServer, 557 mMainBuffer, 558 mAuxBuffer, 559 mCblk->mFlags, 560 mAudioTrackServerProxy->getUnderrunFrames(), 561 nowInUnderrun); 562} 563 564uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { 565 return mAudioTrackServerProxy->getSampleRate(); 566} 567 568// AudioBufferProvider interface 569status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 570 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 571{ 572 ServerProxy::Buffer buf; 573 size_t desiredFrames = buffer->frameCount; 574 buf.mFrameCount = desiredFrames; 575 status_t status = mServerProxy->obtainBuffer(&buf); 576 buffer->frameCount = buf.mFrameCount; 577 buffer->raw = buf.mRaw; 578 if (buf.mFrameCount == 0) { 579 mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 580 } 581 return status; 582} 583 584// releaseBuffer() is not overridden 585 586// ExtendedAudioBufferProvider interface 587 588// Note that framesReady() takes a mutex on the control block using tryLock(). 589// This could result in priority inversion if framesReady() is called by the normal mixer, 590// as the normal mixer thread runs at lower 591// priority than the client's callback thread: there is a short window within framesReady() 592// during which the normal mixer could be preempted, and the client callback would block. 593// Another problem can occur if framesReady() is called by the fast mixer: 594// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 595// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 596size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 597 return mAudioTrackServerProxy->framesReady(); 598} 599 600size_t AudioFlinger::PlaybackThread::Track::framesReleased() const 601{ 602 return mAudioTrackServerProxy->framesReleased(); 603} 604 605// Don't call for fast tracks; the framesReady() could result in priority inversion 606bool AudioFlinger::PlaybackThread::Track::isReady() const { 607 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 608 return true; 609 } 610 611 if (isStopping()) { 612 if (framesReady() > 0) { 613 mFillingUpStatus = FS_FILLED; 614 } 615 return true; 616 } 617 618 if (framesReady() >= mFrameCount || 619 (mCblk->mFlags & CBLK_FORCEREADY)) { 620 mFillingUpStatus = FS_FILLED; 621 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 622 return true; 623 } 624 return false; 625} 626 627status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused, 628 int triggerSession __unused) 629{ 630 status_t status = NO_ERROR; 631 ALOGV("start(%d), calling pid %d session %d", 632 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 633 634 sp<ThreadBase> thread = mThread.promote(); 635 if (thread != 0) { 636 if (isOffloaded()) { 637 Mutex::Autolock _laf(thread->mAudioFlinger->mLock); 638 Mutex::Autolock _lth(thread->mLock); 639 sp<EffectChain> ec = thread->getEffectChain_l(mSessionId); 640 if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() || 641 (ec != 0 && ec->isNonOffloadableEnabled())) { 642 invalidate(); 643 return PERMISSION_DENIED; 644 } 645 } 646 Mutex::Autolock _lth(thread->mLock); 647 track_state state = mState; 648 // here the track could be either new, or restarted 649 // in both cases "unstop" the track 650 651 // initial state-stopping. next state-pausing. 652 // What if resume is called ? 653 654 if (state == PAUSED || state == PAUSING) { 655 if (mResumeToStopping) { 656 // happened we need to resume to STOPPING_1 657 mState = TrackBase::STOPPING_1; 658 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 659 } else { 660 mState = TrackBase::RESUMING; 661 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 662 } 663 } else { 664 mState = TrackBase::ACTIVE; 665 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 666 } 667 668 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 669 status = playbackThread->addTrack_l(this); 670 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 671 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 672 // restore previous state if start was rejected by policy manager 673 if (status == PERMISSION_DENIED) { 674 mState = state; 675 } 676 } 677 // track was already in the active list, not a problem 678 if (status == ALREADY_EXISTS) { 679 status = NO_ERROR; 680 } else { 681 // Acknowledge any pending flush(), so that subsequent new data isn't discarded. 682 // It is usually unsafe to access the server proxy from a binder thread. 683 // But in this case we know the mixer thread (whether normal mixer or fast mixer) 684 // isn't looking at this track yet: we still hold the normal mixer thread lock, 685 // and for fast tracks the track is not yet in the fast mixer thread's active set. 686 ServerProxy::Buffer buffer; 687 buffer.mFrameCount = 1; 688 (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/); 689 } 690 } else { 691 status = BAD_VALUE; 692 } 693 return status; 694} 695 696void AudioFlinger::PlaybackThread::Track::stop() 697{ 698 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 699 sp<ThreadBase> thread = mThread.promote(); 700 if (thread != 0) { 701 Mutex::Autolock _l(thread->mLock); 702 track_state state = mState; 703 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 704 // If the track is not active (PAUSED and buffers full), flush buffers 705 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 706 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 707 reset(); 708 mState = STOPPED; 709 } else if (!isFastTrack() && !isOffloaded()) { 710 mState = STOPPED; 711 } else { 712 // For fast tracks prepareTracks_l() will set state to STOPPING_2 713 // presentation is complete 714 // For an offloaded track this starts a drain and state will 715 // move to STOPPING_2 when drain completes and then STOPPED 716 mState = STOPPING_1; 717 } 718 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 719 playbackThread); 720 } 721 } 722} 723 724void AudioFlinger::PlaybackThread::Track::pause() 725{ 726 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 727 sp<ThreadBase> thread = mThread.promote(); 728 if (thread != 0) { 729 Mutex::Autolock _l(thread->mLock); 730 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 731 switch (mState) { 732 case STOPPING_1: 733 case STOPPING_2: 734 if (!isOffloaded()) { 735 /* nothing to do if track is not offloaded */ 736 break; 737 } 738 739 // Offloaded track was draining, we need to carry on draining when resumed 740 mResumeToStopping = true; 741 // fall through... 742 case ACTIVE: 743 case RESUMING: 744 mState = PAUSING; 745 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 746 playbackThread->broadcast_l(); 747 break; 748 749 default: 750 break; 751 } 752 } 753} 754 755void AudioFlinger::PlaybackThread::Track::flush() 756{ 757 ALOGV("flush(%d)", mName); 758 sp<ThreadBase> thread = mThread.promote(); 759 if (thread != 0) { 760 Mutex::Autolock _l(thread->mLock); 761 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 762 763 if (isOffloaded()) { 764 // If offloaded we allow flush during any state except terminated 765 // and keep the track active to avoid problems if user is seeking 766 // rapidly and underlying hardware has a significant delay handling 767 // a pause 768 if (isTerminated()) { 769 return; 770 } 771 772 ALOGV("flush: offload flush"); 773 reset(); 774 775 if (mState == STOPPING_1 || mState == STOPPING_2) { 776 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 777 mState = ACTIVE; 778 } 779 780 if (mState == ACTIVE) { 781 ALOGV("flush called in active state, resetting buffer time out retry count"); 782 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 783 } 784 785 mFlushHwPending = true; 786 mResumeToStopping = false; 787 } else { 788 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 789 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 790 return; 791 } 792 // No point remaining in PAUSED state after a flush => go to 793 // FLUSHED state 794 mState = FLUSHED; 795 // do not reset the track if it is still in the process of being stopped or paused. 796 // this will be done by prepareTracks_l() when the track is stopped. 797 // prepareTracks_l() will see mState == FLUSHED, then 798 // remove from active track list, reset(), and trigger presentation complete 799 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 800 reset(); 801 } 802 } 803 // Prevent flush being lost if the track is flushed and then resumed 804 // before mixer thread can run. This is important when offloading 805 // because the hardware buffer could hold a large amount of audio 806 playbackThread->broadcast_l(); 807 } 808} 809 810// must be called with thread lock held 811void AudioFlinger::PlaybackThread::Track::flushAck() 812{ 813 if (!isOffloaded()) 814 return; 815 816 mFlushHwPending = false; 817} 818 819void AudioFlinger::PlaybackThread::Track::reset() 820{ 821 // Do not reset twice to avoid discarding data written just after a flush and before 822 // the audioflinger thread detects the track is stopped. 823 if (!mResetDone) { 824 // Force underrun condition to avoid false underrun callback until first data is 825 // written to buffer 826 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 827 mFillingUpStatus = FS_FILLING; 828 mResetDone = true; 829 if (mState == FLUSHED) { 830 mState = IDLE; 831 } 832 } 833} 834 835status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 836{ 837 sp<ThreadBase> thread = mThread.promote(); 838 if (thread == 0) { 839 ALOGE("thread is dead"); 840 return FAILED_TRANSACTION; 841 } else if ((thread->type() == ThreadBase::DIRECT) || 842 (thread->type() == ThreadBase::OFFLOAD)) { 843 return thread->setParameters(keyValuePairs); 844 } else { 845 return PERMISSION_DENIED; 846 } 847} 848 849status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp) 850{ 851 // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant 852 if (isFastTrack()) { 853 // FIXME no lock held to set mPreviousValid = false 854 return INVALID_OPERATION; 855 } 856 sp<ThreadBase> thread = mThread.promote(); 857 if (thread == 0) { 858 // FIXME no lock held to set mPreviousValid = false 859 return INVALID_OPERATION; 860 } 861 Mutex::Autolock _l(thread->mLock); 862 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 863 if (!isOffloaded()) { 864 if (!playbackThread->mLatchQValid) { 865 mPreviousValid = false; 866 return INVALID_OPERATION; 867 } 868 uint32_t unpresentedFrames = 869 ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) / 870 playbackThread->mSampleRate; 871 uint32_t framesWritten = mAudioTrackServerProxy->framesReleased(); 872 bool checkPreviousTimestamp = mPreviousValid && framesWritten >= mPreviousFramesWritten; 873 if (framesWritten < unpresentedFrames) { 874 mPreviousValid = false; 875 return INVALID_OPERATION; 876 } 877 mPreviousFramesWritten = framesWritten; 878 uint32_t position = framesWritten - unpresentedFrames; 879 struct timespec time = playbackThread->mLatchQ.mTimestamp.mTime; 880 if (checkPreviousTimestamp) { 881 if (time.tv_sec < mPreviousTimestamp.mTime.tv_sec || 882 (time.tv_sec == mPreviousTimestamp.mTime.tv_sec && 883 time.tv_nsec < mPreviousTimestamp.mTime.tv_nsec)) { 884 ALOGW("Time is going backwards"); 885 } 886 // position can bobble slightly as an artifact; this hides the bobble 887 static const uint32_t MINIMUM_POSITION_DELTA = 8u; 888 if ((position <= mPreviousTimestamp.mPosition) || 889 (position - mPreviousTimestamp.mPosition) < MINIMUM_POSITION_DELTA) { 890 position = mPreviousTimestamp.mPosition; 891 time = mPreviousTimestamp.mTime; 892 } 893 } 894 timestamp.mPosition = position; 895 timestamp.mTime = time; 896 mPreviousTimestamp = timestamp; 897 mPreviousValid = true; 898 return NO_ERROR; 899 } 900 901 return playbackThread->getTimestamp_l(timestamp); 902} 903 904status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 905{ 906 status_t status = DEAD_OBJECT; 907 sp<ThreadBase> thread = mThread.promote(); 908 if (thread != 0) { 909 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 910 sp<AudioFlinger> af = mClient->audioFlinger(); 911 912 Mutex::Autolock _l(af->mLock); 913 914 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 915 916 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 917 Mutex::Autolock _dl(playbackThread->mLock); 918 Mutex::Autolock _sl(srcThread->mLock); 919 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 920 if (chain == 0) { 921 return INVALID_OPERATION; 922 } 923 924 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 925 if (effect == 0) { 926 return INVALID_OPERATION; 927 } 928 srcThread->removeEffect_l(effect); 929 status = playbackThread->addEffect_l(effect); 930 if (status != NO_ERROR) { 931 srcThread->addEffect_l(effect); 932 return INVALID_OPERATION; 933 } 934 // removeEffect_l() has stopped the effect if it was active so it must be restarted 935 if (effect->state() == EffectModule::ACTIVE || 936 effect->state() == EffectModule::STOPPING) { 937 effect->start(); 938 } 939 940 sp<EffectChain> dstChain = effect->chain().promote(); 941 if (dstChain == 0) { 942 srcThread->addEffect_l(effect); 943 return INVALID_OPERATION; 944 } 945 AudioSystem::unregisterEffect(effect->id()); 946 AudioSystem::registerEffect(&effect->desc(), 947 srcThread->id(), 948 dstChain->strategy(), 949 AUDIO_SESSION_OUTPUT_MIX, 950 effect->id()); 951 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 952 } 953 status = playbackThread->attachAuxEffect(this, EffectId); 954 } 955 return status; 956} 957 958void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 959{ 960 mAuxEffectId = EffectId; 961 mAuxBuffer = buffer; 962} 963 964bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 965 size_t audioHalFrames) 966{ 967 // a track is considered presented when the total number of frames written to audio HAL 968 // corresponds to the number of frames written when presentationComplete() is called for the 969 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 970 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 971 // to detect when all frames have been played. In this case framesWritten isn't 972 // useful because it doesn't always reflect whether there is data in the h/w 973 // buffers, particularly if a track has been paused and resumed during draining 974 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 975 mPresentationCompleteFrames, framesWritten); 976 if (mPresentationCompleteFrames == 0) { 977 mPresentationCompleteFrames = framesWritten + audioHalFrames; 978 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 979 mPresentationCompleteFrames, audioHalFrames); 980 } 981 982 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 983 ALOGV("presentationComplete() session %d complete: framesWritten %d", 984 mSessionId, framesWritten); 985 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 986 mAudioTrackServerProxy->setStreamEndDone(); 987 return true; 988 } 989 return false; 990} 991 992void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 993{ 994 for (size_t i = 0; i < mSyncEvents.size(); i++) { 995 if (mSyncEvents[i]->type() == type) { 996 mSyncEvents[i]->trigger(); 997 mSyncEvents.removeAt(i); 998 i--; 999 } 1000 } 1001} 1002 1003// implement VolumeBufferProvider interface 1004 1005gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 1006{ 1007 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 1008 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 1009 gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR(); 1010 float vl = float_from_gain(gain_minifloat_unpack_left(vlr)); 1011 float vr = float_from_gain(gain_minifloat_unpack_right(vlr)); 1012 // track volumes come from shared memory, so can't be trusted and must be clamped 1013 if (vl > GAIN_FLOAT_UNITY) { 1014 vl = GAIN_FLOAT_UNITY; 1015 } 1016 if (vr > GAIN_FLOAT_UNITY) { 1017 vr = GAIN_FLOAT_UNITY; 1018 } 1019 // now apply the cached master volume and stream type volume; 1020 // this is trusted but lacks any synchronization or barrier so may be stale 1021 float v = mCachedVolume; 1022 vl *= v; 1023 vr *= v; 1024 // re-combine into packed minifloat 1025 vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr)); 1026 // FIXME look at mute, pause, and stop flags 1027 return vlr; 1028} 1029 1030status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 1031{ 1032 if (isTerminated() || mState == PAUSED || 1033 ((framesReady() == 0) && ((mSharedBuffer != 0) || 1034 (mState == STOPPED)))) { 1035 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 1036 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 1037 event->cancel(); 1038 return INVALID_OPERATION; 1039 } 1040 (void) TrackBase::setSyncEvent(event); 1041 return NO_ERROR; 1042} 1043 1044void AudioFlinger::PlaybackThread::Track::invalidate() 1045{ 1046 // FIXME should use proxy, and needs work 1047 audio_track_cblk_t* cblk = mCblk; 1048 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1049 android_atomic_release_store(0x40000000, &cblk->mFutex); 1050 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 1051 (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); 1052 mIsInvalid = true; 1053} 1054 1055void AudioFlinger::PlaybackThread::Track::signal() 1056{ 1057 sp<ThreadBase> thread = mThread.promote(); 1058 if (thread != 0) { 1059 PlaybackThread *t = (PlaybackThread *)thread.get(); 1060 Mutex::Autolock _l(t->mLock); 1061 t->broadcast_l(); 1062 } 1063} 1064 1065//To be called with thread lock held 1066bool AudioFlinger::PlaybackThread::Track::isResumePending() { 1067 1068 if (mState == RESUMING) 1069 return true; 1070 /* Resume is pending if track was stopping before pause was called */ 1071 if (mState == STOPPING_1 && 1072 mResumeToStopping) 1073 return true; 1074 1075 return false; 1076} 1077 1078//To be called with thread lock held 1079void AudioFlinger::PlaybackThread::Track::resumeAck() { 1080 1081 1082 if (mState == RESUMING) 1083 mState = ACTIVE; 1084 1085 // Other possibility of pending resume is stopping_1 state 1086 // Do not update the state from stopping as this prevents 1087 // drain being called. 1088 if (mState == STOPPING_1) { 1089 mResumeToStopping = false; 1090 } 1091} 1092// ---------------------------------------------------------------------------- 1093 1094sp<AudioFlinger::PlaybackThread::TimedTrack> 1095AudioFlinger::PlaybackThread::TimedTrack::create( 1096 PlaybackThread *thread, 1097 const sp<Client>& client, 1098 audio_stream_type_t streamType, 1099 uint32_t sampleRate, 1100 audio_format_t format, 1101 audio_channel_mask_t channelMask, 1102 size_t frameCount, 1103 const sp<IMemory>& sharedBuffer, 1104 int sessionId, 1105 int uid) 1106{ 1107 if (!client->reserveTimedTrack()) 1108 return 0; 1109 1110 return new TimedTrack( 1111 thread, client, streamType, sampleRate, format, channelMask, frameCount, 1112 sharedBuffer, sessionId, uid); 1113} 1114 1115AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 1116 PlaybackThread *thread, 1117 const sp<Client>& client, 1118 audio_stream_type_t streamType, 1119 uint32_t sampleRate, 1120 audio_format_t format, 1121 audio_channel_mask_t channelMask, 1122 size_t frameCount, 1123 const sp<IMemory>& sharedBuffer, 1124 int sessionId, 1125 int uid) 1126 : Track(thread, client, streamType, sampleRate, format, channelMask, 1127 frameCount, sharedBuffer, sessionId, uid, IAudioFlinger::TRACK_TIMED), 1128 mQueueHeadInFlight(false), 1129 mTrimQueueHeadOnRelease(false), 1130 mFramesPendingInQueue(0), 1131 mTimedSilenceBuffer(NULL), 1132 mTimedSilenceBufferSize(0), 1133 mTimedAudioOutputOnTime(false), 1134 mMediaTimeTransformValid(false) 1135{ 1136 LocalClock lc; 1137 mLocalTimeFreq = lc.getLocalFreq(); 1138 1139 mLocalTimeToSampleTransform.a_zero = 0; 1140 mLocalTimeToSampleTransform.b_zero = 0; 1141 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 1142 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 1143 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 1144 &mLocalTimeToSampleTransform.a_to_b_denom); 1145 1146 mMediaTimeToSampleTransform.a_zero = 0; 1147 mMediaTimeToSampleTransform.b_zero = 0; 1148 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 1149 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 1150 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 1151 &mMediaTimeToSampleTransform.a_to_b_denom); 1152} 1153 1154AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 1155 mClient->releaseTimedTrack(); 1156 delete [] mTimedSilenceBuffer; 1157} 1158 1159status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 1160 size_t size, sp<IMemory>* buffer) { 1161 1162 Mutex::Autolock _l(mTimedBufferQueueLock); 1163 1164 trimTimedBufferQueue_l(); 1165 1166 // lazily initialize the shared memory heap for timed buffers 1167 if (mTimedMemoryDealer == NULL) { 1168 const int kTimedBufferHeapSize = 512 << 10; 1169 1170 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 1171 "AudioFlingerTimed"); 1172 if (mTimedMemoryDealer == NULL) { 1173 return NO_MEMORY; 1174 } 1175 } 1176 1177 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 1178 if (newBuffer == 0 || newBuffer->pointer() == NULL) { 1179 return NO_MEMORY; 1180 } 1181 1182 *buffer = newBuffer; 1183 return NO_ERROR; 1184} 1185 1186// caller must hold mTimedBufferQueueLock 1187void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 1188 int64_t mediaTimeNow; 1189 { 1190 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1191 if (!mMediaTimeTransformValid) 1192 return; 1193 1194 int64_t targetTimeNow; 1195 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 1196 ? mCCHelper.getCommonTime(&targetTimeNow) 1197 : mCCHelper.getLocalTime(&targetTimeNow); 1198 1199 if (OK != res) 1200 return; 1201 1202 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 1203 &mediaTimeNow)) { 1204 return; 1205 } 1206 } 1207 1208 size_t trimEnd; 1209 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 1210 int64_t bufEnd; 1211 1212 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 1213 // We have a next buffer. Just use its PTS as the PTS of the frame 1214 // following the last frame in this buffer. If the stream is sparse 1215 // (ie, there are deliberate gaps left in the stream which should be 1216 // filled with silence by the TimedAudioTrack), then this can result 1217 // in one extra buffer being left un-trimmed when it could have 1218 // been. In general, this is not typical, and we would rather 1219 // optimized away the TS calculation below for the more common case 1220 // where PTSes are contiguous. 1221 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 1222 } else { 1223 // We have no next buffer. Compute the PTS of the frame following 1224 // the last frame in this buffer by computing the duration of of 1225 // this frame in media time units and adding it to the PTS of the 1226 // buffer. 1227 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1228 / mFrameSize; 1229 1230 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1231 &bufEnd)) { 1232 ALOGE("Failed to convert frame count of %lld to media time" 1233 " duration" " (scale factor %d/%u) in %s", 1234 frameCount, 1235 mMediaTimeToSampleTransform.a_to_b_numer, 1236 mMediaTimeToSampleTransform.a_to_b_denom, 1237 __PRETTY_FUNCTION__); 1238 break; 1239 } 1240 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1241 } 1242 1243 if (bufEnd > mediaTimeNow) 1244 break; 1245 1246 // Is the buffer we want to use in the middle of a mix operation right 1247 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1248 // from the mixer which should be coming back shortly. 1249 if (!trimEnd && mQueueHeadInFlight) { 1250 mTrimQueueHeadOnRelease = true; 1251 } 1252 } 1253 1254 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1255 if (trimStart < trimEnd) { 1256 // Update the bookkeeping for framesReady() 1257 for (size_t i = trimStart; i < trimEnd; ++i) { 1258 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1259 } 1260 1261 // Now actually remove the buffers from the queue. 1262 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1263 } 1264} 1265 1266void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1267 const char* logTag) { 1268 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1269 "%s called (reason \"%s\"), but timed buffer queue has no" 1270 " elements to trim.", __FUNCTION__, logTag); 1271 1272 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1273 mTimedBufferQueue.removeAt(0); 1274} 1275 1276void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1277 const TimedBuffer& buf, 1278 const char* logTag __unused) { 1279 uint32_t bufBytes = buf.buffer()->size(); 1280 uint32_t consumedAlready = buf.position(); 1281 1282 ALOG_ASSERT(consumedAlready <= bufBytes, 1283 "Bad bookkeeping while updating frames pending. Timed buffer is" 1284 " only %u bytes long, but claims to have consumed %u" 1285 " bytes. (update reason: \"%s\")", 1286 bufBytes, consumedAlready, logTag); 1287 1288 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1289 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1290 "Bad bookkeeping while updating frames pending. Should have at" 1291 " least %u queued frames, but we think we have only %u. (update" 1292 " reason: \"%s\")", 1293 bufFrames, mFramesPendingInQueue, logTag); 1294 1295 mFramesPendingInQueue -= bufFrames; 1296} 1297 1298status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1299 const sp<IMemory>& buffer, int64_t pts) { 1300 1301 { 1302 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1303 if (!mMediaTimeTransformValid) 1304 return INVALID_OPERATION; 1305 } 1306 1307 Mutex::Autolock _l(mTimedBufferQueueLock); 1308 1309 uint32_t bufFrames = buffer->size() / mFrameSize; 1310 mFramesPendingInQueue += bufFrames; 1311 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1312 1313 return NO_ERROR; 1314} 1315 1316status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1317 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1318 1319 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1320 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1321 target); 1322 1323 if (!(target == TimedAudioTrack::LOCAL_TIME || 1324 target == TimedAudioTrack::COMMON_TIME)) { 1325 return BAD_VALUE; 1326 } 1327 1328 Mutex::Autolock lock(mMediaTimeTransformLock); 1329 mMediaTimeTransform = xform; 1330 mMediaTimeTransformTarget = target; 1331 mMediaTimeTransformValid = true; 1332 1333 return NO_ERROR; 1334} 1335 1336#define min(a, b) ((a) < (b) ? (a) : (b)) 1337 1338// implementation of getNextBuffer for tracks whose buffers have timestamps 1339status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1340 AudioBufferProvider::Buffer* buffer, int64_t pts) 1341{ 1342 if (pts == AudioBufferProvider::kInvalidPTS) { 1343 buffer->raw = NULL; 1344 buffer->frameCount = 0; 1345 mTimedAudioOutputOnTime = false; 1346 return INVALID_OPERATION; 1347 } 1348 1349 Mutex::Autolock _l(mTimedBufferQueueLock); 1350 1351 ALOG_ASSERT(!mQueueHeadInFlight, 1352 "getNextBuffer called without releaseBuffer!"); 1353 1354 while (true) { 1355 1356 // if we have no timed buffers, then fail 1357 if (mTimedBufferQueue.isEmpty()) { 1358 buffer->raw = NULL; 1359 buffer->frameCount = 0; 1360 return NOT_ENOUGH_DATA; 1361 } 1362 1363 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1364 1365 // calculate the PTS of the head of the timed buffer queue expressed in 1366 // local time 1367 int64_t headLocalPTS; 1368 { 1369 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1370 1371 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1372 1373 if (mMediaTimeTransform.a_to_b_denom == 0) { 1374 // the transform represents a pause, so yield silence 1375 timedYieldSilence_l(buffer->frameCount, buffer); 1376 return NO_ERROR; 1377 } 1378 1379 int64_t transformedPTS; 1380 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1381 &transformedPTS)) { 1382 // the transform failed. this shouldn't happen, but if it does 1383 // then just drop this buffer 1384 ALOGW("timedGetNextBuffer transform failed"); 1385 buffer->raw = NULL; 1386 buffer->frameCount = 0; 1387 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1388 return NO_ERROR; 1389 } 1390 1391 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1392 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1393 &headLocalPTS)) { 1394 buffer->raw = NULL; 1395 buffer->frameCount = 0; 1396 return INVALID_OPERATION; 1397 } 1398 } else { 1399 headLocalPTS = transformedPTS; 1400 } 1401 } 1402 1403 uint32_t sr = sampleRate(); 1404 1405 // adjust the head buffer's PTS to reflect the portion of the head buffer 1406 // that has already been consumed 1407 int64_t effectivePTS = headLocalPTS + 1408 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1409 1410 // Calculate the delta in samples between the head of the input buffer 1411 // queue and the start of the next output buffer that will be written. 1412 // If the transformation fails because of over or underflow, it means 1413 // that the sample's position in the output stream is so far out of 1414 // whack that it should just be dropped. 1415 int64_t sampleDelta; 1416 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1417 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1418 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1419 " mix"); 1420 continue; 1421 } 1422 if (!mLocalTimeToSampleTransform.doForwardTransform( 1423 (effectivePTS - pts) << 32, &sampleDelta)) { 1424 ALOGV("*** too late during sample rate transform: dropped buffer"); 1425 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1426 continue; 1427 } 1428 1429 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1430 " sampleDelta=[%d.%08x]", 1431 head.pts(), head.position(), pts, 1432 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1433 + (sampleDelta >> 32)), 1434 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1435 1436 // if the delta between the ideal placement for the next input sample and 1437 // the current output position is within this threshold, then we will 1438 // concatenate the next input samples to the previous output 1439 const int64_t kSampleContinuityThreshold = 1440 (static_cast<int64_t>(sr) << 32) / 250; 1441 1442 // if this is the first buffer of audio that we're emitting from this track 1443 // then it should be almost exactly on time. 1444 const int64_t kSampleStartupThreshold = 1LL << 32; 1445 1446 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1447 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1448 // the next input is close enough to being on time, so concatenate it 1449 // with the last output 1450 timedYieldSamples_l(buffer); 1451 1452 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1453 head.position(), buffer->frameCount); 1454 return NO_ERROR; 1455 } 1456 1457 // Looks like our output is not on time. Reset our on timed status. 1458 // Next time we mix samples from our input queue, then should be within 1459 // the StartupThreshold. 1460 mTimedAudioOutputOnTime = false; 1461 if (sampleDelta > 0) { 1462 // the gap between the current output position and the proper start of 1463 // the next input sample is too big, so fill it with silence 1464 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1465 1466 timedYieldSilence_l(framesUntilNextInput, buffer); 1467 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1468 return NO_ERROR; 1469 } else { 1470 // the next input sample is late 1471 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1472 size_t onTimeSamplePosition = 1473 head.position() + lateFrames * mFrameSize; 1474 1475 if (onTimeSamplePosition > head.buffer()->size()) { 1476 // all the remaining samples in the head are too late, so 1477 // drop it and move on 1478 ALOGV("*** too late: dropped buffer"); 1479 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1480 continue; 1481 } else { 1482 // skip over the late samples 1483 head.setPosition(onTimeSamplePosition); 1484 1485 // yield the available samples 1486 timedYieldSamples_l(buffer); 1487 1488 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1489 return NO_ERROR; 1490 } 1491 } 1492 } 1493} 1494 1495// Yield samples from the timed buffer queue head up to the given output 1496// buffer's capacity. 1497// 1498// Caller must hold mTimedBufferQueueLock 1499void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1500 AudioBufferProvider::Buffer* buffer) { 1501 1502 const TimedBuffer& head = mTimedBufferQueue[0]; 1503 1504 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1505 head.position()); 1506 1507 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1508 mFrameSize); 1509 size_t framesRequested = buffer->frameCount; 1510 buffer->frameCount = min(framesLeftInHead, framesRequested); 1511 1512 mQueueHeadInFlight = true; 1513 mTimedAudioOutputOnTime = true; 1514} 1515 1516// Yield samples of silence up to the given output buffer's capacity 1517// 1518// Caller must hold mTimedBufferQueueLock 1519void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1520 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1521 1522 // lazily allocate a buffer filled with silence 1523 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1524 delete [] mTimedSilenceBuffer; 1525 mTimedSilenceBufferSize = numFrames * mFrameSize; 1526 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1527 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1528 } 1529 1530 buffer->raw = mTimedSilenceBuffer; 1531 size_t framesRequested = buffer->frameCount; 1532 buffer->frameCount = min(numFrames, framesRequested); 1533 1534 mTimedAudioOutputOnTime = false; 1535} 1536 1537// AudioBufferProvider interface 1538void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1539 AudioBufferProvider::Buffer* buffer) { 1540 1541 Mutex::Autolock _l(mTimedBufferQueueLock); 1542 1543 // If the buffer which was just released is part of the buffer at the head 1544 // of the queue, be sure to update the amt of the buffer which has been 1545 // consumed. If the buffer being returned is not part of the head of the 1546 // queue, its either because the buffer is part of the silence buffer, or 1547 // because the head of the timed queue was trimmed after the mixer called 1548 // getNextBuffer but before the mixer called releaseBuffer. 1549 if (buffer->raw == mTimedSilenceBuffer) { 1550 ALOG_ASSERT(!mQueueHeadInFlight, 1551 "Queue head in flight during release of silence buffer!"); 1552 goto done; 1553 } 1554 1555 ALOG_ASSERT(mQueueHeadInFlight, 1556 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1557 " head in flight."); 1558 1559 if (mTimedBufferQueue.size()) { 1560 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1561 1562 void* start = head.buffer()->pointer(); 1563 void* end = reinterpret_cast<void*>( 1564 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1565 + head.buffer()->size()); 1566 1567 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1568 "released buffer not within the head of the timed buffer" 1569 " queue; qHead = [%p, %p], released buffer = %p", 1570 start, end, buffer->raw); 1571 1572 head.setPosition(head.position() + 1573 (buffer->frameCount * mFrameSize)); 1574 mQueueHeadInFlight = false; 1575 1576 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1577 "Bad bookkeeping during releaseBuffer! Should have at" 1578 " least %u queued frames, but we think we have only %u", 1579 buffer->frameCount, mFramesPendingInQueue); 1580 1581 mFramesPendingInQueue -= buffer->frameCount; 1582 1583 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1584 || mTrimQueueHeadOnRelease) { 1585 trimTimedBufferQueueHead_l("releaseBuffer"); 1586 mTrimQueueHeadOnRelease = false; 1587 } 1588 } else { 1589 LOG_ALWAYS_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1590 " buffers in the timed buffer queue"); 1591 } 1592 1593done: 1594 buffer->raw = 0; 1595 buffer->frameCount = 0; 1596} 1597 1598size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1599 Mutex::Autolock _l(mTimedBufferQueueLock); 1600 return mFramesPendingInQueue; 1601} 1602 1603AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1604 : mPTS(0), mPosition(0) {} 1605 1606AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1607 const sp<IMemory>& buffer, int64_t pts) 1608 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1609 1610 1611// ---------------------------------------------------------------------------- 1612 1613AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1614 PlaybackThread *playbackThread, 1615 DuplicatingThread *sourceThread, 1616 uint32_t sampleRate, 1617 audio_format_t format, 1618 audio_channel_mask_t channelMask, 1619 size_t frameCount, 1620 int uid) 1621 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1622 NULL, 0, uid, IAudioFlinger::TRACK_DEFAULT), 1623 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1624{ 1625 1626 if (mCblk != NULL) { 1627 mOutBuffer.frameCount = 0; 1628 playbackThread->mTracks.add(this); 1629 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1630 "frameCount %u, mChannelMask 0x%08x", 1631 mCblk, mBuffer, 1632 frameCount, mChannelMask); 1633 // since client and server are in the same process, 1634 // the buffer has the same virtual address on both sides 1635 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize); 1636 mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); 1637 mClientProxy->setSendLevel(0.0); 1638 mClientProxy->setSampleRate(sampleRate); 1639 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1640 true /*clientInServer*/); 1641 } else { 1642 ALOGW("Error creating output track on thread %p", playbackThread); 1643 } 1644} 1645 1646AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1647{ 1648 clearBufferQueue(); 1649 delete mClientProxy; 1650 // superclass destructor will now delete the server proxy and shared memory both refer to 1651} 1652 1653status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1654 int triggerSession) 1655{ 1656 status_t status = Track::start(event, triggerSession); 1657 if (status != NO_ERROR) { 1658 return status; 1659 } 1660 1661 mActive = true; 1662 mRetryCount = 127; 1663 return status; 1664} 1665 1666void AudioFlinger::PlaybackThread::OutputTrack::stop() 1667{ 1668 Track::stop(); 1669 clearBufferQueue(); 1670 mOutBuffer.frameCount = 0; 1671 mActive = false; 1672} 1673 1674bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1675{ 1676 Buffer *pInBuffer; 1677 Buffer inBuffer; 1678 uint32_t channelCount = mChannelCount; 1679 bool outputBufferFull = false; 1680 inBuffer.frameCount = frames; 1681 inBuffer.i16 = data; 1682 1683 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1684 1685 if (!mActive && frames != 0) { 1686 start(); 1687 sp<ThreadBase> thread = mThread.promote(); 1688 if (thread != 0) { 1689 MixerThread *mixerThread = (MixerThread *)thread.get(); 1690 if (mFrameCount > frames) { 1691 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1692 uint32_t startFrames = (mFrameCount - frames); 1693 pInBuffer = new Buffer; 1694 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1695 pInBuffer->frameCount = startFrames; 1696 pInBuffer->i16 = pInBuffer->mBuffer; 1697 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1698 mBufferQueue.add(pInBuffer); 1699 } else { 1700 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1701 } 1702 } 1703 } 1704 } 1705 1706 while (waitTimeLeftMs) { 1707 // First write pending buffers, then new data 1708 if (mBufferQueue.size()) { 1709 pInBuffer = mBufferQueue.itemAt(0); 1710 } else { 1711 pInBuffer = &inBuffer; 1712 } 1713 1714 if (pInBuffer->frameCount == 0) { 1715 break; 1716 } 1717 1718 if (mOutBuffer.frameCount == 0) { 1719 mOutBuffer.frameCount = pInBuffer->frameCount; 1720 nsecs_t startTime = systemTime(); 1721 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1722 if (status != NO_ERROR) { 1723 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1724 mThread.unsafe_get(), status); 1725 outputBufferFull = true; 1726 break; 1727 } 1728 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1729 if (waitTimeLeftMs >= waitTimeMs) { 1730 waitTimeLeftMs -= waitTimeMs; 1731 } else { 1732 waitTimeLeftMs = 0; 1733 } 1734 } 1735 1736 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1737 pInBuffer->frameCount; 1738 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1739 Proxy::Buffer buf; 1740 buf.mFrameCount = outFrames; 1741 buf.mRaw = NULL; 1742 mClientProxy->releaseBuffer(&buf); 1743 pInBuffer->frameCount -= outFrames; 1744 pInBuffer->i16 += outFrames * channelCount; 1745 mOutBuffer.frameCount -= outFrames; 1746 mOutBuffer.i16 += outFrames * channelCount; 1747 1748 if (pInBuffer->frameCount == 0) { 1749 if (mBufferQueue.size()) { 1750 mBufferQueue.removeAt(0); 1751 delete [] pInBuffer->mBuffer; 1752 delete pInBuffer; 1753 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1754 mThread.unsafe_get(), mBufferQueue.size()); 1755 } else { 1756 break; 1757 } 1758 } 1759 } 1760 1761 // If we could not write all frames, allocate a buffer and queue it for next time. 1762 if (inBuffer.frameCount) { 1763 sp<ThreadBase> thread = mThread.promote(); 1764 if (thread != 0 && !thread->standby()) { 1765 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1766 pInBuffer = new Buffer; 1767 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1768 pInBuffer->frameCount = inBuffer.frameCount; 1769 pInBuffer->i16 = pInBuffer->mBuffer; 1770 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1771 sizeof(int16_t)); 1772 mBufferQueue.add(pInBuffer); 1773 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1774 mThread.unsafe_get(), mBufferQueue.size()); 1775 } else { 1776 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1777 mThread.unsafe_get(), this); 1778 } 1779 } 1780 } 1781 1782 // Calling write() with a 0 length buffer, means that no more data will be written: 1783 // If no more buffers are pending, fill output track buffer to make sure it is started 1784 // by output mixer. 1785 if (frames == 0 && mBufferQueue.size() == 0) { 1786 // FIXME borken, replace by getting framesReady() from proxy 1787 size_t user = 0; // was mCblk->user 1788 if (user < mFrameCount) { 1789 frames = mFrameCount - user; 1790 pInBuffer = new Buffer; 1791 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1792 pInBuffer->frameCount = frames; 1793 pInBuffer->i16 = pInBuffer->mBuffer; 1794 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1795 mBufferQueue.add(pInBuffer); 1796 } else if (mActive) { 1797 stop(); 1798 } 1799 } 1800 1801 return outputBufferFull; 1802} 1803 1804status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1805 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1806{ 1807 ClientProxy::Buffer buf; 1808 buf.mFrameCount = buffer->frameCount; 1809 struct timespec timeout; 1810 timeout.tv_sec = waitTimeMs / 1000; 1811 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1812 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1813 buffer->frameCount = buf.mFrameCount; 1814 buffer->raw = buf.mRaw; 1815 return status; 1816} 1817 1818void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1819{ 1820 size_t size = mBufferQueue.size(); 1821 1822 for (size_t i = 0; i < size; i++) { 1823 Buffer *pBuffer = mBufferQueue.itemAt(i); 1824 delete [] pBuffer->mBuffer; 1825 delete pBuffer; 1826 } 1827 mBufferQueue.clear(); 1828} 1829 1830 1831// ---------------------------------------------------------------------------- 1832// Record 1833// ---------------------------------------------------------------------------- 1834 1835AudioFlinger::RecordHandle::RecordHandle( 1836 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1837 : BnAudioRecord(), 1838 mRecordTrack(recordTrack) 1839{ 1840} 1841 1842AudioFlinger::RecordHandle::~RecordHandle() { 1843 stop_nonvirtual(); 1844 mRecordTrack->destroy(); 1845} 1846 1847status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1848 int triggerSession) { 1849 ALOGV("RecordHandle::start()"); 1850 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1851} 1852 1853void AudioFlinger::RecordHandle::stop() { 1854 stop_nonvirtual(); 1855} 1856 1857void AudioFlinger::RecordHandle::stop_nonvirtual() { 1858 ALOGV("RecordHandle::stop()"); 1859 mRecordTrack->stop(); 1860} 1861 1862status_t AudioFlinger::RecordHandle::onTransact( 1863 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1864{ 1865 return BnAudioRecord::onTransact(code, data, reply, flags); 1866} 1867 1868// ---------------------------------------------------------------------------- 1869 1870// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 1871AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1872 RecordThread *thread, 1873 const sp<Client>& client, 1874 uint32_t sampleRate, 1875 audio_format_t format, 1876 audio_channel_mask_t channelMask, 1877 size_t frameCount, 1878 int sessionId, 1879 int uid, 1880 IAudioFlinger::track_flags_t flags) 1881 : TrackBase(thread, client, sampleRate, format, 1882 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid, 1883 flags, false /*isOut*/, 1884 flags & IAudioFlinger::TRACK_FAST ? ALLOC_PIPE : ALLOC_CBLK), 1885 mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0), 1886 // See real initialization of mRsmpInFront at RecordThread::start() 1887 mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL) 1888{ 1889 if (mCblk == NULL) { 1890 return; 1891 } 1892 1893 mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize); 1894 1895 uint32_t channelCount = audio_channel_count_from_in_mask(channelMask); 1896 // FIXME I don't understand either of the channel count checks 1897 if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 && 1898 channelCount <= FCC_2) { 1899 // sink SR 1900 mResampler = AudioResampler::create(16, thread->mChannelCount, sampleRate); 1901 // source SR 1902 mResampler->setSampleRate(thread->mSampleRate); 1903 mResampler->setVolume(AudioMixer::UNITY_GAIN_INT, AudioMixer::UNITY_GAIN_INT); 1904 mResamplerBufferProvider = new ResamplerBufferProvider(this); 1905 } 1906 1907 if (flags & IAudioFlinger::TRACK_FAST) { 1908 ALOG_ASSERT(thread->mFastTrackAvail); 1909 thread->mFastTrackAvail = false; 1910 } 1911} 1912 1913AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 1914{ 1915 ALOGV("%s", __func__); 1916 delete mResampler; 1917 delete[] mRsmpOutBuffer; 1918 delete mResamplerBufferProvider; 1919} 1920 1921// AudioBufferProvider interface 1922status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 1923 int64_t pts __unused) 1924{ 1925 ServerProxy::Buffer buf; 1926 buf.mFrameCount = buffer->frameCount; 1927 status_t status = mServerProxy->obtainBuffer(&buf); 1928 buffer->frameCount = buf.mFrameCount; 1929 buffer->raw = buf.mRaw; 1930 if (buf.mFrameCount == 0) { 1931 // FIXME also wake futex so that overrun is noticed more quickly 1932 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 1933 } 1934 return status; 1935} 1936 1937status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 1938 int triggerSession) 1939{ 1940 sp<ThreadBase> thread = mThread.promote(); 1941 if (thread != 0) { 1942 RecordThread *recordThread = (RecordThread *)thread.get(); 1943 return recordThread->start(this, event, triggerSession); 1944 } else { 1945 return BAD_VALUE; 1946 } 1947} 1948 1949void AudioFlinger::RecordThread::RecordTrack::stop() 1950{ 1951 sp<ThreadBase> thread = mThread.promote(); 1952 if (thread != 0) { 1953 RecordThread *recordThread = (RecordThread *)thread.get(); 1954 if (recordThread->stop(this)) { 1955 AudioSystem::stopInput(recordThread->id()); 1956 } 1957 } 1958} 1959 1960void AudioFlinger::RecordThread::RecordTrack::destroy() 1961{ 1962 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 1963 sp<RecordTrack> keep(this); 1964 { 1965 sp<ThreadBase> thread = mThread.promote(); 1966 if (thread != 0) { 1967 if (mState == ACTIVE || mState == RESUMING) { 1968 AudioSystem::stopInput(thread->id()); 1969 } 1970 AudioSystem::releaseInput(thread->id()); 1971 Mutex::Autolock _l(thread->mLock); 1972 RecordThread *recordThread = (RecordThread *) thread.get(); 1973 recordThread->destroyTrack_l(this); 1974 } 1975 } 1976} 1977 1978void AudioFlinger::RecordThread::RecordTrack::invalidate() 1979{ 1980 // FIXME should use proxy, and needs work 1981 audio_track_cblk_t* cblk = mCblk; 1982 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1983 android_atomic_release_store(0x40000000, &cblk->mFutex); 1984 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 1985 (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); 1986} 1987 1988 1989/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 1990{ 1991 result.append(" Active Client Fmt Chn mask Session S Server fCount Resampling\n"); 1992} 1993 1994void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active) 1995{ 1996 snprintf(buffer, size, " %6s %6u %3u %08X %7u %1d %08X %6zu %10d\n", 1997 active ? "yes" : "no", 1998 (mClient == 0) ? getpid_cached : mClient->pid(), 1999 mFormat, 2000 mChannelMask, 2001 mSessionId, 2002 mState, 2003 mCblk->mServer, 2004 mFrameCount, 2005 mResampler != NULL); 2006 2007} 2008 2009void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event) 2010{ 2011 if (event == mSyncStartEvent) { 2012 ssize_t framesToDrop = 0; 2013 sp<ThreadBase> threadBase = mThread.promote(); 2014 if (threadBase != 0) { 2015 // TODO: use actual buffer filling status instead of 2 buffers when info is available 2016 // from audio HAL 2017 framesToDrop = threadBase->mFrameCount * 2; 2018 } 2019 mFramesToDrop = framesToDrop; 2020 } 2021} 2022 2023void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent() 2024{ 2025 if (mSyncStartEvent != 0) { 2026 mSyncStartEvent->cancel(); 2027 mSyncStartEvent.clear(); 2028 } 2029 mFramesToDrop = 0; 2030} 2031 2032}; // namespace android 2033