Tracks.cpp revision 6954127b7ace022677ac407ff943c2793f8a11be
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 void *buffer, 72 int sessionId, 73 int clientUid, 74 IAudioFlinger::track_flags_t flags, 75 bool isOut, 76 alloc_type alloc, 77 track_type type) 78 : RefBase(), 79 mThread(thread), 80 mClient(client), 81 mCblk(NULL), 82 // mBuffer 83 mState(IDLE), 84 mSampleRate(sampleRate), 85 mFormat(format), 86 mChannelMask(channelMask), 87 mChannelCount(isOut ? 88 audio_channel_count_from_out_mask(channelMask) : 89 audio_channel_count_from_in_mask(channelMask)), 90 mFrameSize(audio_is_linear_pcm(format) ? 91 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 92 mFrameCount(frameCount), 93 mSessionId(sessionId), 94 mFlags(flags), 95 mIsOut(isOut), 96 mServerProxy(NULL), 97 mId(android_atomic_inc(&nextTrackId)), 98 mTerminated(false), 99 mType(type), 100 mThreadIoHandle(thread->id()) 101{ 102 // if the caller is us, trust the specified uid 103 if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) { 104 int newclientUid = IPCThreadState::self()->getCallingUid(); 105 if (clientUid != -1 && clientUid != newclientUid) { 106 ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid); 107 } 108 clientUid = newclientUid; 109 } 110 // clientUid contains the uid of the app that is responsible for this track, so we can blame 111 // battery usage on it. 112 mUid = clientUid; 113 114 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 115 size_t size = sizeof(audio_track_cblk_t); 116 size_t bufferSize = (buffer == NULL ? roundup(frameCount) : frameCount) * mFrameSize; 117 if (buffer == NULL && alloc == ALLOC_CBLK) { 118 size += bufferSize; 119 } 120 121 if (client != 0) { 122 mCblkMemory = client->heap()->allocate(size); 123 if (mCblkMemory == 0 || 124 (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) { 125 ALOGE("not enough memory for AudioTrack size=%u", size); 126 client->heap()->dump("AudioTrack"); 127 mCblkMemory.clear(); 128 return; 129 } 130 } else { 131 // this syntax avoids calling the audio_track_cblk_t constructor twice 132 mCblk = (audio_track_cblk_t *) new uint8_t[size]; 133 // assume mCblk != NULL 134 } 135 136 // construct the shared structure in-place. 137 if (mCblk != NULL) { 138 new(mCblk) audio_track_cblk_t(); 139 switch (alloc) { 140 case ALLOC_READONLY: { 141 const sp<MemoryDealer> roHeap(thread->readOnlyHeap()); 142 if (roHeap == 0 || 143 (mBufferMemory = roHeap->allocate(bufferSize)) == 0 || 144 (mBuffer = mBufferMemory->pointer()) == NULL) { 145 ALOGE("not enough memory for read-only buffer size=%zu", bufferSize); 146 if (roHeap != 0) { 147 roHeap->dump("buffer"); 148 } 149 mCblkMemory.clear(); 150 mBufferMemory.clear(); 151 return; 152 } 153 memset(mBuffer, 0, bufferSize); 154 } break; 155 case ALLOC_PIPE: 156 mBufferMemory = thread->pipeMemory(); 157 // mBuffer is the virtual address as seen from current process (mediaserver), 158 // and should normally be coming from mBufferMemory->pointer(). 159 // However in this case the TrackBase does not reference the buffer directly. 160 // It should references the buffer via the pipe. 161 // Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL. 162 mBuffer = NULL; 163 break; 164 case ALLOC_CBLK: 165 // clear all buffers 166 if (buffer == NULL) { 167 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 168 memset(mBuffer, 0, bufferSize); 169 } else { 170 mBuffer = buffer; 171#if 0 172 mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic 173#endif 174 } 175 break; 176 case ALLOC_LOCAL: 177 mBuffer = calloc(1, bufferSize); 178 break; 179 case ALLOC_NONE: 180 mBuffer = buffer; 181 break; 182 } 183 184#ifdef TEE_SINK 185 if (mTeeSinkTrackEnabled) { 186 NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount, mFormat); 187 if (Format_isValid(pipeFormat)) { 188 Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat); 189 size_t numCounterOffers = 0; 190 const NBAIO_Format offers[1] = {pipeFormat}; 191 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 192 ALOG_ASSERT(index == 0); 193 PipeReader *pipeReader = new PipeReader(*pipe); 194 numCounterOffers = 0; 195 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 196 ALOG_ASSERT(index == 0); 197 mTeeSink = pipe; 198 mTeeSource = pipeReader; 199 } 200 } 201#endif 202 203 } 204} 205 206status_t AudioFlinger::ThreadBase::TrackBase::initCheck() const 207{ 208 status_t status; 209 if (mType == TYPE_OUTPUT || mType == TYPE_PATCH) { 210 status = cblk() != NULL ? NO_ERROR : NO_MEMORY; 211 } else { 212 status = getCblk() != 0 ? NO_ERROR : NO_MEMORY; 213 } 214 return status; 215} 216 217AudioFlinger::ThreadBase::TrackBase::~TrackBase() 218{ 219#ifdef TEE_SINK 220 dumpTee(-1, mTeeSource, mId); 221#endif 222 // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference 223 delete mServerProxy; 224 if (mCblk != NULL) { 225 if (mClient == 0) { 226 delete mCblk; 227 } else { 228 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 229 } 230 } 231 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 232 if (mClient != 0) { 233 // Client destructor must run with AudioFlinger client mutex locked 234 Mutex::Autolock _l(mClient->audioFlinger()->mClientLock); 235 // If the client's reference count drops to zero, the associated destructor 236 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 237 // relying on the automatic clear() at end of scope. 238 mClient.clear(); 239 } 240 // flush the binder command buffer 241 IPCThreadState::self()->flushCommands(); 242} 243 244// AudioBufferProvider interface 245// getNextBuffer() = 0; 246// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 247void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 248{ 249#ifdef TEE_SINK 250 if (mTeeSink != 0) { 251 (void) mTeeSink->write(buffer->raw, buffer->frameCount); 252 } 253#endif 254 255 ServerProxy::Buffer buf; 256 buf.mFrameCount = buffer->frameCount; 257 buf.mRaw = buffer->raw; 258 buffer->frameCount = 0; 259 buffer->raw = NULL; 260 mServerProxy->releaseBuffer(&buf); 261} 262 263status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 264{ 265 mSyncEvents.add(event); 266 return NO_ERROR; 267} 268 269// ---------------------------------------------------------------------------- 270// Playback 271// ---------------------------------------------------------------------------- 272 273AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 274 : BnAudioTrack(), 275 mTrack(track) 276{ 277} 278 279AudioFlinger::TrackHandle::~TrackHandle() { 280 // just stop the track on deletion, associated resources 281 // will be freed from the main thread once all pending buffers have 282 // been played. Unless it's not in the active track list, in which 283 // case we free everything now... 284 mTrack->destroy(); 285} 286 287sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 288 return mTrack->getCblk(); 289} 290 291status_t AudioFlinger::TrackHandle::start() { 292 return mTrack->start(); 293} 294 295void AudioFlinger::TrackHandle::stop() { 296 mTrack->stop(); 297} 298 299void AudioFlinger::TrackHandle::flush() { 300 mTrack->flush(); 301} 302 303void AudioFlinger::TrackHandle::pause() { 304 mTrack->pause(); 305} 306 307status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 308{ 309 return mTrack->attachAuxEffect(EffectId); 310} 311 312status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 313 sp<IMemory>* buffer) { 314 if (!mTrack->isTimedTrack()) 315 return INVALID_OPERATION; 316 317 PlaybackThread::TimedTrack* tt = 318 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 319 return tt->allocateTimedBuffer(size, buffer); 320} 321 322status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 323 int64_t pts) { 324 if (!mTrack->isTimedTrack()) 325 return INVALID_OPERATION; 326 327 if (buffer == 0 || buffer->pointer() == NULL) { 328 ALOGE("queueTimedBuffer() buffer is 0 or has NULL pointer()"); 329 return BAD_VALUE; 330 } 331 332 PlaybackThread::TimedTrack* tt = 333 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 334 return tt->queueTimedBuffer(buffer, pts); 335} 336 337status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 338 const LinearTransform& xform, int target) { 339 340 if (!mTrack->isTimedTrack()) 341 return INVALID_OPERATION; 342 343 PlaybackThread::TimedTrack* tt = 344 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 345 return tt->setMediaTimeTransform( 346 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 347} 348 349status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) { 350 return mTrack->setParameters(keyValuePairs); 351} 352 353status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp) 354{ 355 return mTrack->getTimestamp(timestamp); 356} 357 358 359void AudioFlinger::TrackHandle::signal() 360{ 361 return mTrack->signal(); 362} 363 364status_t AudioFlinger::TrackHandle::onTransact( 365 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 366{ 367 return BnAudioTrack::onTransact(code, data, reply, flags); 368} 369 370// ---------------------------------------------------------------------------- 371 372// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 373AudioFlinger::PlaybackThread::Track::Track( 374 PlaybackThread *thread, 375 const sp<Client>& client, 376 audio_stream_type_t streamType, 377 uint32_t sampleRate, 378 audio_format_t format, 379 audio_channel_mask_t channelMask, 380 size_t frameCount, 381 void *buffer, 382 const sp<IMemory>& sharedBuffer, 383 int sessionId, 384 int uid, 385 IAudioFlinger::track_flags_t flags, 386 track_type type) 387 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 388 (sharedBuffer != 0) ? sharedBuffer->pointer() : buffer, 389 sessionId, uid, flags, true /*isOut*/, 390 (type == TYPE_PATCH) ? ( buffer == NULL ? ALLOC_LOCAL : ALLOC_NONE) : ALLOC_CBLK, 391 type), 392 mFillingUpStatus(FS_INVALID), 393 // mRetryCount initialized later when needed 394 mSharedBuffer(sharedBuffer), 395 mStreamType(streamType), 396 mName(-1), // see note below 397 mMainBuffer(thread->mixBuffer()), 398 mAuxBuffer(NULL), 399 mAuxEffectId(0), mHasVolumeController(false), 400 mPresentationCompleteFrames(0), 401 mFastIndex(-1), 402 mCachedVolume(1.0), 403 mIsInvalid(false), 404 mAudioTrackServerProxy(NULL), 405 mResumeToStopping(false), 406 mFlushHwPending(false), 407 mPreviousValid(false), 408 mPreviousFramesWritten(0) 409 // mPreviousTimestamp 410{ 411 // client == 0 implies sharedBuffer == 0 412 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 413 414 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 415 sharedBuffer->size()); 416 417 if (mCblk == NULL) { 418 return; 419 } 420 421 if (sharedBuffer == 0) { 422 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 423 mFrameSize, !isExternalTrack(), sampleRate); 424 } else { 425 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, 426 mFrameSize); 427 } 428 mServerProxy = mAudioTrackServerProxy; 429 430 mName = thread->getTrackName_l(channelMask, format, sessionId); 431 if (mName < 0) { 432 ALOGE("no more track names available"); 433 return; 434 } 435 // only allocate a fast track index if we were able to allocate a normal track name 436 if (flags & IAudioFlinger::TRACK_FAST) { 437 mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); 438 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 439 int i = __builtin_ctz(thread->mFastTrackAvailMask); 440 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 441 // FIXME This is too eager. We allocate a fast track index before the 442 // fast track becomes active. Since fast tracks are a scarce resource, 443 // this means we are potentially denying other more important fast tracks from 444 // being created. It would be better to allocate the index dynamically. 445 mFastIndex = i; 446 // Read the initial underruns because this field is never cleared by the fast mixer 447 mObservedUnderruns = thread->getFastTrackUnderruns(i); 448 thread->mFastTrackAvailMask &= ~(1 << i); 449 } 450} 451 452AudioFlinger::PlaybackThread::Track::~Track() 453{ 454 ALOGV("PlaybackThread::Track destructor"); 455 456 // The destructor would clear mSharedBuffer, 457 // but it will not push the decremented reference count, 458 // leaving the client's IMemory dangling indefinitely. 459 // This prevents that leak. 460 if (mSharedBuffer != 0) { 461 mSharedBuffer.clear(); 462 } 463} 464 465status_t AudioFlinger::PlaybackThread::Track::initCheck() const 466{ 467 status_t status = TrackBase::initCheck(); 468 if (status == NO_ERROR && mName < 0) { 469 status = NO_MEMORY; 470 } 471 return status; 472} 473 474void AudioFlinger::PlaybackThread::Track::destroy() 475{ 476 // NOTE: destroyTrack_l() can remove a strong reference to this Track 477 // by removing it from mTracks vector, so there is a risk that this Tracks's 478 // destructor is called. As the destructor needs to lock mLock, 479 // we must acquire a strong reference on this Track before locking mLock 480 // here so that the destructor is called only when exiting this function. 481 // On the other hand, as long as Track::destroy() is only called by 482 // TrackHandle destructor, the TrackHandle still holds a strong ref on 483 // this Track with its member mTrack. 484 sp<Track> keep(this); 485 { // scope for mLock 486 bool wasActive = false; 487 sp<ThreadBase> thread = mThread.promote(); 488 if (thread != 0) { 489 Mutex::Autolock _l(thread->mLock); 490 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 491 wasActive = playbackThread->destroyTrack_l(this); 492 } 493 if (isExternalTrack() && !wasActive) { 494 AudioSystem::releaseOutput(mThreadIoHandle); 495 } 496 } 497} 498 499/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 500{ 501 result.append(" Name Active Client Type Fmt Chn mask Session fCount S F SRate " 502 "L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n"); 503} 504 505void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active) 506{ 507 gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR(); 508 if (isFastTrack()) { 509 sprintf(buffer, " F %2d", mFastIndex); 510 } else if (mName >= AudioMixer::TRACK0) { 511 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 512 } else { 513 sprintf(buffer, " none"); 514 } 515 track_state state = mState; 516 char stateChar; 517 if (isTerminated()) { 518 stateChar = 'T'; 519 } else { 520 switch (state) { 521 case IDLE: 522 stateChar = 'I'; 523 break; 524 case STOPPING_1: 525 stateChar = 's'; 526 break; 527 case STOPPING_2: 528 stateChar = '5'; 529 break; 530 case STOPPED: 531 stateChar = 'S'; 532 break; 533 case RESUMING: 534 stateChar = 'R'; 535 break; 536 case ACTIVE: 537 stateChar = 'A'; 538 break; 539 case PAUSING: 540 stateChar = 'p'; 541 break; 542 case PAUSED: 543 stateChar = 'P'; 544 break; 545 case FLUSHED: 546 stateChar = 'F'; 547 break; 548 default: 549 stateChar = '?'; 550 break; 551 } 552 } 553 char nowInUnderrun; 554 switch (mObservedUnderruns.mBitFields.mMostRecent) { 555 case UNDERRUN_FULL: 556 nowInUnderrun = ' '; 557 break; 558 case UNDERRUN_PARTIAL: 559 nowInUnderrun = '<'; 560 break; 561 case UNDERRUN_EMPTY: 562 nowInUnderrun = '*'; 563 break; 564 default: 565 nowInUnderrun = '?'; 566 break; 567 } 568 snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g " 569 "%08X %p %p 0x%03X %9u%c\n", 570 active ? "yes" : "no", 571 (mClient == 0) ? getpid_cached : mClient->pid(), 572 mStreamType, 573 mFormat, 574 mChannelMask, 575 mSessionId, 576 mFrameCount, 577 stateChar, 578 mFillingUpStatus, 579 mAudioTrackServerProxy->getSampleRate(), 580 20.0 * log10(float_from_gain(gain_minifloat_unpack_left(vlr))), 581 20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))), 582 mCblk->mServer, 583 mMainBuffer, 584 mAuxBuffer, 585 mCblk->mFlags, 586 mAudioTrackServerProxy->getUnderrunFrames(), 587 nowInUnderrun); 588} 589 590uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { 591 return mAudioTrackServerProxy->getSampleRate(); 592} 593 594// AudioBufferProvider interface 595status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 596 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 597{ 598 ServerProxy::Buffer buf; 599 size_t desiredFrames = buffer->frameCount; 600 buf.mFrameCount = desiredFrames; 601 status_t status = mServerProxy->obtainBuffer(&buf); 602 buffer->frameCount = buf.mFrameCount; 603 buffer->raw = buf.mRaw; 604 if (buf.mFrameCount == 0) { 605 mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 606 } 607 return status; 608} 609 610// releaseBuffer() is not overridden 611 612// ExtendedAudioBufferProvider interface 613 614// Note that framesReady() takes a mutex on the control block using tryLock(). 615// This could result in priority inversion if framesReady() is called by the normal mixer, 616// as the normal mixer thread runs at lower 617// priority than the client's callback thread: there is a short window within framesReady() 618// during which the normal mixer could be preempted, and the client callback would block. 619// Another problem can occur if framesReady() is called by the fast mixer: 620// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 621// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 622size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 623 return mAudioTrackServerProxy->framesReady(); 624} 625 626size_t AudioFlinger::PlaybackThread::Track::framesReleased() const 627{ 628 return mAudioTrackServerProxy->framesReleased(); 629} 630 631// Don't call for fast tracks; the framesReady() could result in priority inversion 632bool AudioFlinger::PlaybackThread::Track::isReady() const { 633 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 634 return true; 635 } 636 637 if (isStopping()) { 638 if (framesReady() > 0) { 639 mFillingUpStatus = FS_FILLED; 640 } 641 return true; 642 } 643 644 if (framesReady() >= mFrameCount || 645 (mCblk->mFlags & CBLK_FORCEREADY)) { 646 mFillingUpStatus = FS_FILLED; 647 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 648 return true; 649 } 650 return false; 651} 652 653status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused, 654 int triggerSession __unused) 655{ 656 status_t status = NO_ERROR; 657 ALOGV("start(%d), calling pid %d session %d", 658 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 659 660 sp<ThreadBase> thread = mThread.promote(); 661 if (thread != 0) { 662 if (isOffloaded()) { 663 Mutex::Autolock _laf(thread->mAudioFlinger->mLock); 664 Mutex::Autolock _lth(thread->mLock); 665 sp<EffectChain> ec = thread->getEffectChain_l(mSessionId); 666 if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() || 667 (ec != 0 && ec->isNonOffloadableEnabled())) { 668 invalidate(); 669 return PERMISSION_DENIED; 670 } 671 } 672 Mutex::Autolock _lth(thread->mLock); 673 track_state state = mState; 674 // here the track could be either new, or restarted 675 // in both cases "unstop" the track 676 677 // initial state-stopping. next state-pausing. 678 // What if resume is called ? 679 680 if (state == PAUSED || state == PAUSING) { 681 if (mResumeToStopping) { 682 // happened we need to resume to STOPPING_1 683 mState = TrackBase::STOPPING_1; 684 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 685 } else { 686 mState = TrackBase::RESUMING; 687 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 688 } 689 } else { 690 mState = TrackBase::ACTIVE; 691 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 692 } 693 694 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 695 status = playbackThread->addTrack_l(this); 696 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 697 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 698 // restore previous state if start was rejected by policy manager 699 if (status == PERMISSION_DENIED) { 700 mState = state; 701 } 702 } 703 // track was already in the active list, not a problem 704 if (status == ALREADY_EXISTS) { 705 status = NO_ERROR; 706 } else { 707 // Acknowledge any pending flush(), so that subsequent new data isn't discarded. 708 // It is usually unsafe to access the server proxy from a binder thread. 709 // But in this case we know the mixer thread (whether normal mixer or fast mixer) 710 // isn't looking at this track yet: we still hold the normal mixer thread lock, 711 // and for fast tracks the track is not yet in the fast mixer thread's active set. 712 ServerProxy::Buffer buffer; 713 buffer.mFrameCount = 1; 714 (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/); 715 } 716 } else { 717 status = BAD_VALUE; 718 } 719 return status; 720} 721 722void AudioFlinger::PlaybackThread::Track::stop() 723{ 724 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 725 sp<ThreadBase> thread = mThread.promote(); 726 if (thread != 0) { 727 Mutex::Autolock _l(thread->mLock); 728 track_state state = mState; 729 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 730 // If the track is not active (PAUSED and buffers full), flush buffers 731 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 732 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 733 reset(); 734 mState = STOPPED; 735 } else if (!isFastTrack() && !isOffloaded() && !isDirect()) { 736 mState = STOPPED; 737 } else { 738 // For fast tracks prepareTracks_l() will set state to STOPPING_2 739 // presentation is complete 740 // For an offloaded track this starts a drain and state will 741 // move to STOPPING_2 when drain completes and then STOPPED 742 mState = STOPPING_1; 743 } 744 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 745 playbackThread); 746 } 747 } 748} 749 750void AudioFlinger::PlaybackThread::Track::pause() 751{ 752 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 753 sp<ThreadBase> thread = mThread.promote(); 754 if (thread != 0) { 755 Mutex::Autolock _l(thread->mLock); 756 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 757 switch (mState) { 758 case STOPPING_1: 759 case STOPPING_2: 760 if (!isOffloaded()) { 761 /* nothing to do if track is not offloaded */ 762 break; 763 } 764 765 // Offloaded track was draining, we need to carry on draining when resumed 766 mResumeToStopping = true; 767 // fall through... 768 case ACTIVE: 769 case RESUMING: 770 mState = PAUSING; 771 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 772 playbackThread->broadcast_l(); 773 break; 774 775 default: 776 break; 777 } 778 } 779} 780 781void AudioFlinger::PlaybackThread::Track::flush() 782{ 783 ALOGV("flush(%d)", mName); 784 sp<ThreadBase> thread = mThread.promote(); 785 if (thread != 0) { 786 Mutex::Autolock _l(thread->mLock); 787 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 788 789 if (isOffloaded()) { 790 // If offloaded we allow flush during any state except terminated 791 // and keep the track active to avoid problems if user is seeking 792 // rapidly and underlying hardware has a significant delay handling 793 // a pause 794 if (isTerminated()) { 795 return; 796 } 797 798 ALOGV("flush: offload flush"); 799 reset(); 800 801 if (mState == STOPPING_1 || mState == STOPPING_2) { 802 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 803 mState = ACTIVE; 804 } 805 806 if (mState == ACTIVE) { 807 ALOGV("flush called in active state, resetting buffer time out retry count"); 808 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 809 } 810 811 mFlushHwPending = true; 812 mResumeToStopping = false; 813 } else { 814 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 815 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 816 return; 817 } 818 // No point remaining in PAUSED state after a flush => go to 819 // FLUSHED state 820 mState = FLUSHED; 821 // do not reset the track if it is still in the process of being stopped or paused. 822 // this will be done by prepareTracks_l() when the track is stopped. 823 // prepareTracks_l() will see mState == FLUSHED, then 824 // remove from active track list, reset(), and trigger presentation complete 825 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 826 reset(); 827 } 828 } 829 // Prevent flush being lost if the track is flushed and then resumed 830 // before mixer thread can run. This is important when offloading 831 // because the hardware buffer could hold a large amount of audio 832 playbackThread->broadcast_l(); 833 } 834} 835 836// must be called with thread lock held 837void AudioFlinger::PlaybackThread::Track::flushAck() 838{ 839 if (!isOffloaded()) 840 return; 841 842 mFlushHwPending = false; 843} 844 845void AudioFlinger::PlaybackThread::Track::reset() 846{ 847 // Do not reset twice to avoid discarding data written just after a flush and before 848 // the audioflinger thread detects the track is stopped. 849 if (!mResetDone) { 850 // Force underrun condition to avoid false underrun callback until first data is 851 // written to buffer 852 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 853 mFillingUpStatus = FS_FILLING; 854 mResetDone = true; 855 if (mState == FLUSHED) { 856 mState = IDLE; 857 } 858 } 859} 860 861status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 862{ 863 sp<ThreadBase> thread = mThread.promote(); 864 if (thread == 0) { 865 ALOGE("thread is dead"); 866 return FAILED_TRANSACTION; 867 } else if ((thread->type() == ThreadBase::DIRECT) || 868 (thread->type() == ThreadBase::OFFLOAD)) { 869 return thread->setParameters(keyValuePairs); 870 } else { 871 return PERMISSION_DENIED; 872 } 873} 874 875status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp) 876{ 877 // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant 878 if (isFastTrack()) { 879 // FIXME no lock held to set mPreviousValid = false 880 return INVALID_OPERATION; 881 } 882 sp<ThreadBase> thread = mThread.promote(); 883 if (thread == 0) { 884 // FIXME no lock held to set mPreviousValid = false 885 return INVALID_OPERATION; 886 } 887 Mutex::Autolock _l(thread->mLock); 888 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 889 if (!isOffloaded() && !isDirect()) { 890 if (!playbackThread->mLatchQValid) { 891 mPreviousValid = false; 892 return INVALID_OPERATION; 893 } 894 uint32_t unpresentedFrames = 895 ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) / 896 playbackThread->mSampleRate; 897 uint32_t framesWritten = mAudioTrackServerProxy->framesReleased(); 898 bool checkPreviousTimestamp = mPreviousValid && framesWritten >= mPreviousFramesWritten; 899 if (framesWritten < unpresentedFrames) { 900 mPreviousValid = false; 901 return INVALID_OPERATION; 902 } 903 mPreviousFramesWritten = framesWritten; 904 uint32_t position = framesWritten - unpresentedFrames; 905 struct timespec time = playbackThread->mLatchQ.mTimestamp.mTime; 906 if (checkPreviousTimestamp) { 907 if (time.tv_sec < mPreviousTimestamp.mTime.tv_sec || 908 (time.tv_sec == mPreviousTimestamp.mTime.tv_sec && 909 time.tv_nsec < mPreviousTimestamp.mTime.tv_nsec)) { 910 ALOGW("Time is going backwards"); 911 } 912 // position can bobble slightly as an artifact; this hides the bobble 913 static const uint32_t MINIMUM_POSITION_DELTA = 8u; 914 if ((position <= mPreviousTimestamp.mPosition) || 915 (position - mPreviousTimestamp.mPosition) < MINIMUM_POSITION_DELTA) { 916 position = mPreviousTimestamp.mPosition; 917 time = mPreviousTimestamp.mTime; 918 } 919 } 920 timestamp.mPosition = position; 921 timestamp.mTime = time; 922 mPreviousTimestamp = timestamp; 923 mPreviousValid = true; 924 return NO_ERROR; 925 } 926 927 return playbackThread->getTimestamp_l(timestamp); 928} 929 930status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 931{ 932 status_t status = DEAD_OBJECT; 933 sp<ThreadBase> thread = mThread.promote(); 934 if (thread != 0) { 935 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 936 sp<AudioFlinger> af = mClient->audioFlinger(); 937 938 Mutex::Autolock _l(af->mLock); 939 940 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 941 942 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 943 Mutex::Autolock _dl(playbackThread->mLock); 944 Mutex::Autolock _sl(srcThread->mLock); 945 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 946 if (chain == 0) { 947 return INVALID_OPERATION; 948 } 949 950 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 951 if (effect == 0) { 952 return INVALID_OPERATION; 953 } 954 srcThread->removeEffect_l(effect); 955 status = playbackThread->addEffect_l(effect); 956 if (status != NO_ERROR) { 957 srcThread->addEffect_l(effect); 958 return INVALID_OPERATION; 959 } 960 // removeEffect_l() has stopped the effect if it was active so it must be restarted 961 if (effect->state() == EffectModule::ACTIVE || 962 effect->state() == EffectModule::STOPPING) { 963 effect->start(); 964 } 965 966 sp<EffectChain> dstChain = effect->chain().promote(); 967 if (dstChain == 0) { 968 srcThread->addEffect_l(effect); 969 return INVALID_OPERATION; 970 } 971 AudioSystem::unregisterEffect(effect->id()); 972 AudioSystem::registerEffect(&effect->desc(), 973 srcThread->id(), 974 dstChain->strategy(), 975 AUDIO_SESSION_OUTPUT_MIX, 976 effect->id()); 977 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 978 } 979 status = playbackThread->attachAuxEffect(this, EffectId); 980 } 981 return status; 982} 983 984void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 985{ 986 mAuxEffectId = EffectId; 987 mAuxBuffer = buffer; 988} 989 990bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 991 size_t audioHalFrames) 992{ 993 // a track is considered presented when the total number of frames written to audio HAL 994 // corresponds to the number of frames written when presentationComplete() is called for the 995 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 996 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 997 // to detect when all frames have been played. In this case framesWritten isn't 998 // useful because it doesn't always reflect whether there is data in the h/w 999 // buffers, particularly if a track has been paused and resumed during draining 1000 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 1001 mPresentationCompleteFrames, framesWritten); 1002 if (mPresentationCompleteFrames == 0) { 1003 mPresentationCompleteFrames = framesWritten + audioHalFrames; 1004 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 1005 mPresentationCompleteFrames, audioHalFrames); 1006 } 1007 1008 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 1009 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1010 mAudioTrackServerProxy->setStreamEndDone(); 1011 return true; 1012 } 1013 return false; 1014} 1015 1016void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 1017{ 1018 for (size_t i = 0; i < mSyncEvents.size(); i++) { 1019 if (mSyncEvents[i]->type() == type) { 1020 mSyncEvents[i]->trigger(); 1021 mSyncEvents.removeAt(i); 1022 i--; 1023 } 1024 } 1025} 1026 1027// implement VolumeBufferProvider interface 1028 1029gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 1030{ 1031 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 1032 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 1033 gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR(); 1034 float vl = float_from_gain(gain_minifloat_unpack_left(vlr)); 1035 float vr = float_from_gain(gain_minifloat_unpack_right(vlr)); 1036 // track volumes come from shared memory, so can't be trusted and must be clamped 1037 if (vl > GAIN_FLOAT_UNITY) { 1038 vl = GAIN_FLOAT_UNITY; 1039 } 1040 if (vr > GAIN_FLOAT_UNITY) { 1041 vr = GAIN_FLOAT_UNITY; 1042 } 1043 // now apply the cached master volume and stream type volume; 1044 // this is trusted but lacks any synchronization or barrier so may be stale 1045 float v = mCachedVolume; 1046 vl *= v; 1047 vr *= v; 1048 // re-combine into packed minifloat 1049 vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr)); 1050 // FIXME look at mute, pause, and stop flags 1051 return vlr; 1052} 1053 1054status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 1055{ 1056 if (isTerminated() || mState == PAUSED || 1057 ((framesReady() == 0) && ((mSharedBuffer != 0) || 1058 (mState == STOPPED)))) { 1059 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 1060 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 1061 event->cancel(); 1062 return INVALID_OPERATION; 1063 } 1064 (void) TrackBase::setSyncEvent(event); 1065 return NO_ERROR; 1066} 1067 1068void AudioFlinger::PlaybackThread::Track::invalidate() 1069{ 1070 // FIXME should use proxy, and needs work 1071 audio_track_cblk_t* cblk = mCblk; 1072 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1073 android_atomic_release_store(0x40000000, &cblk->mFutex); 1074 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 1075 (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); 1076 mIsInvalid = true; 1077} 1078 1079void AudioFlinger::PlaybackThread::Track::signal() 1080{ 1081 sp<ThreadBase> thread = mThread.promote(); 1082 if (thread != 0) { 1083 PlaybackThread *t = (PlaybackThread *)thread.get(); 1084 Mutex::Autolock _l(t->mLock); 1085 t->broadcast_l(); 1086 } 1087} 1088 1089//To be called with thread lock held 1090bool AudioFlinger::PlaybackThread::Track::isResumePending() { 1091 1092 if (mState == RESUMING) 1093 return true; 1094 /* Resume is pending if track was stopping before pause was called */ 1095 if (mState == STOPPING_1 && 1096 mResumeToStopping) 1097 return true; 1098 1099 return false; 1100} 1101 1102//To be called with thread lock held 1103void AudioFlinger::PlaybackThread::Track::resumeAck() { 1104 1105 1106 if (mState == RESUMING) 1107 mState = ACTIVE; 1108 1109 // Other possibility of pending resume is stopping_1 state 1110 // Do not update the state from stopping as this prevents 1111 // drain being called. 1112 if (mState == STOPPING_1) { 1113 mResumeToStopping = false; 1114 } 1115} 1116// ---------------------------------------------------------------------------- 1117 1118sp<AudioFlinger::PlaybackThread::TimedTrack> 1119AudioFlinger::PlaybackThread::TimedTrack::create( 1120 PlaybackThread *thread, 1121 const sp<Client>& client, 1122 audio_stream_type_t streamType, 1123 uint32_t sampleRate, 1124 audio_format_t format, 1125 audio_channel_mask_t channelMask, 1126 size_t frameCount, 1127 const sp<IMemory>& sharedBuffer, 1128 int sessionId, 1129 int uid) 1130{ 1131 if (!client->reserveTimedTrack()) 1132 return 0; 1133 1134 return new TimedTrack( 1135 thread, client, streamType, sampleRate, format, channelMask, frameCount, 1136 sharedBuffer, sessionId, uid); 1137} 1138 1139AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 1140 PlaybackThread *thread, 1141 const sp<Client>& client, 1142 audio_stream_type_t streamType, 1143 uint32_t sampleRate, 1144 audio_format_t format, 1145 audio_channel_mask_t channelMask, 1146 size_t frameCount, 1147 const sp<IMemory>& sharedBuffer, 1148 int sessionId, 1149 int uid) 1150 : Track(thread, client, streamType, sampleRate, format, channelMask, 1151 frameCount, (sharedBuffer != 0) ? sharedBuffer->pointer() : NULL, sharedBuffer, 1152 sessionId, uid, IAudioFlinger::TRACK_TIMED, TYPE_TIMED), 1153 mQueueHeadInFlight(false), 1154 mTrimQueueHeadOnRelease(false), 1155 mFramesPendingInQueue(0), 1156 mTimedSilenceBuffer(NULL), 1157 mTimedSilenceBufferSize(0), 1158 mTimedAudioOutputOnTime(false), 1159 mMediaTimeTransformValid(false) 1160{ 1161 LocalClock lc; 1162 mLocalTimeFreq = lc.getLocalFreq(); 1163 1164 mLocalTimeToSampleTransform.a_zero = 0; 1165 mLocalTimeToSampleTransform.b_zero = 0; 1166 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 1167 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 1168 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 1169 &mLocalTimeToSampleTransform.a_to_b_denom); 1170 1171 mMediaTimeToSampleTransform.a_zero = 0; 1172 mMediaTimeToSampleTransform.b_zero = 0; 1173 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 1174 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 1175 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 1176 &mMediaTimeToSampleTransform.a_to_b_denom); 1177} 1178 1179AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 1180 mClient->releaseTimedTrack(); 1181 delete [] mTimedSilenceBuffer; 1182} 1183 1184status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 1185 size_t size, sp<IMemory>* buffer) { 1186 1187 Mutex::Autolock _l(mTimedBufferQueueLock); 1188 1189 trimTimedBufferQueue_l(); 1190 1191 // lazily initialize the shared memory heap for timed buffers 1192 if (mTimedMemoryDealer == NULL) { 1193 const int kTimedBufferHeapSize = 512 << 10; 1194 1195 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 1196 "AudioFlingerTimed"); 1197 if (mTimedMemoryDealer == NULL) { 1198 return NO_MEMORY; 1199 } 1200 } 1201 1202 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 1203 if (newBuffer == 0 || newBuffer->pointer() == NULL) { 1204 return NO_MEMORY; 1205 } 1206 1207 *buffer = newBuffer; 1208 return NO_ERROR; 1209} 1210 1211// caller must hold mTimedBufferQueueLock 1212void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 1213 int64_t mediaTimeNow; 1214 { 1215 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1216 if (!mMediaTimeTransformValid) 1217 return; 1218 1219 int64_t targetTimeNow; 1220 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 1221 ? mCCHelper.getCommonTime(&targetTimeNow) 1222 : mCCHelper.getLocalTime(&targetTimeNow); 1223 1224 if (OK != res) 1225 return; 1226 1227 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 1228 &mediaTimeNow)) { 1229 return; 1230 } 1231 } 1232 1233 size_t trimEnd; 1234 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 1235 int64_t bufEnd; 1236 1237 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 1238 // We have a next buffer. Just use its PTS as the PTS of the frame 1239 // following the last frame in this buffer. If the stream is sparse 1240 // (ie, there are deliberate gaps left in the stream which should be 1241 // filled with silence by the TimedAudioTrack), then this can result 1242 // in one extra buffer being left un-trimmed when it could have 1243 // been. In general, this is not typical, and we would rather 1244 // optimized away the TS calculation below for the more common case 1245 // where PTSes are contiguous. 1246 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 1247 } else { 1248 // We have no next buffer. Compute the PTS of the frame following 1249 // the last frame in this buffer by computing the duration of of 1250 // this frame in media time units and adding it to the PTS of the 1251 // buffer. 1252 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1253 / mFrameSize; 1254 1255 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1256 &bufEnd)) { 1257 ALOGE("Failed to convert frame count of %lld to media time" 1258 " duration" " (scale factor %d/%u) in %s", 1259 frameCount, 1260 mMediaTimeToSampleTransform.a_to_b_numer, 1261 mMediaTimeToSampleTransform.a_to_b_denom, 1262 __PRETTY_FUNCTION__); 1263 break; 1264 } 1265 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1266 } 1267 1268 if (bufEnd > mediaTimeNow) 1269 break; 1270 1271 // Is the buffer we want to use in the middle of a mix operation right 1272 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1273 // from the mixer which should be coming back shortly. 1274 if (!trimEnd && mQueueHeadInFlight) { 1275 mTrimQueueHeadOnRelease = true; 1276 } 1277 } 1278 1279 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1280 if (trimStart < trimEnd) { 1281 // Update the bookkeeping for framesReady() 1282 for (size_t i = trimStart; i < trimEnd; ++i) { 1283 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1284 } 1285 1286 // Now actually remove the buffers from the queue. 1287 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1288 } 1289} 1290 1291void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1292 const char* logTag) { 1293 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1294 "%s called (reason \"%s\"), but timed buffer queue has no" 1295 " elements to trim.", __FUNCTION__, logTag); 1296 1297 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1298 mTimedBufferQueue.removeAt(0); 1299} 1300 1301void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1302 const TimedBuffer& buf, 1303 const char* logTag __unused) { 1304 uint32_t bufBytes = buf.buffer()->size(); 1305 uint32_t consumedAlready = buf.position(); 1306 1307 ALOG_ASSERT(consumedAlready <= bufBytes, 1308 "Bad bookkeeping while updating frames pending. Timed buffer is" 1309 " only %u bytes long, but claims to have consumed %u" 1310 " bytes. (update reason: \"%s\")", 1311 bufBytes, consumedAlready, logTag); 1312 1313 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1314 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1315 "Bad bookkeeping while updating frames pending. Should have at" 1316 " least %u queued frames, but we think we have only %u. (update" 1317 " reason: \"%s\")", 1318 bufFrames, mFramesPendingInQueue, logTag); 1319 1320 mFramesPendingInQueue -= bufFrames; 1321} 1322 1323status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1324 const sp<IMemory>& buffer, int64_t pts) { 1325 1326 { 1327 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1328 if (!mMediaTimeTransformValid) 1329 return INVALID_OPERATION; 1330 } 1331 1332 Mutex::Autolock _l(mTimedBufferQueueLock); 1333 1334 uint32_t bufFrames = buffer->size() / mFrameSize; 1335 mFramesPendingInQueue += bufFrames; 1336 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1337 1338 return NO_ERROR; 1339} 1340 1341status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1342 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1343 1344 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1345 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1346 target); 1347 1348 if (!(target == TimedAudioTrack::LOCAL_TIME || 1349 target == TimedAudioTrack::COMMON_TIME)) { 1350 return BAD_VALUE; 1351 } 1352 1353 Mutex::Autolock lock(mMediaTimeTransformLock); 1354 mMediaTimeTransform = xform; 1355 mMediaTimeTransformTarget = target; 1356 mMediaTimeTransformValid = true; 1357 1358 return NO_ERROR; 1359} 1360 1361#define min(a, b) ((a) < (b) ? (a) : (b)) 1362 1363// implementation of getNextBuffer for tracks whose buffers have timestamps 1364status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1365 AudioBufferProvider::Buffer* buffer, int64_t pts) 1366{ 1367 if (pts == AudioBufferProvider::kInvalidPTS) { 1368 buffer->raw = NULL; 1369 buffer->frameCount = 0; 1370 mTimedAudioOutputOnTime = false; 1371 return INVALID_OPERATION; 1372 } 1373 1374 Mutex::Autolock _l(mTimedBufferQueueLock); 1375 1376 ALOG_ASSERT(!mQueueHeadInFlight, 1377 "getNextBuffer called without releaseBuffer!"); 1378 1379 while (true) { 1380 1381 // if we have no timed buffers, then fail 1382 if (mTimedBufferQueue.isEmpty()) { 1383 buffer->raw = NULL; 1384 buffer->frameCount = 0; 1385 return NOT_ENOUGH_DATA; 1386 } 1387 1388 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1389 1390 // calculate the PTS of the head of the timed buffer queue expressed in 1391 // local time 1392 int64_t headLocalPTS; 1393 { 1394 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1395 1396 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1397 1398 if (mMediaTimeTransform.a_to_b_denom == 0) { 1399 // the transform represents a pause, so yield silence 1400 timedYieldSilence_l(buffer->frameCount, buffer); 1401 return NO_ERROR; 1402 } 1403 1404 int64_t transformedPTS; 1405 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1406 &transformedPTS)) { 1407 // the transform failed. this shouldn't happen, but if it does 1408 // then just drop this buffer 1409 ALOGW("timedGetNextBuffer transform failed"); 1410 buffer->raw = NULL; 1411 buffer->frameCount = 0; 1412 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1413 return NO_ERROR; 1414 } 1415 1416 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1417 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1418 &headLocalPTS)) { 1419 buffer->raw = NULL; 1420 buffer->frameCount = 0; 1421 return INVALID_OPERATION; 1422 } 1423 } else { 1424 headLocalPTS = transformedPTS; 1425 } 1426 } 1427 1428 uint32_t sr = sampleRate(); 1429 1430 // adjust the head buffer's PTS to reflect the portion of the head buffer 1431 // that has already been consumed 1432 int64_t effectivePTS = headLocalPTS + 1433 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1434 1435 // Calculate the delta in samples between the head of the input buffer 1436 // queue and the start of the next output buffer that will be written. 1437 // If the transformation fails because of over or underflow, it means 1438 // that the sample's position in the output stream is so far out of 1439 // whack that it should just be dropped. 1440 int64_t sampleDelta; 1441 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1442 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1443 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1444 " mix"); 1445 continue; 1446 } 1447 if (!mLocalTimeToSampleTransform.doForwardTransform( 1448 (effectivePTS - pts) << 32, &sampleDelta)) { 1449 ALOGV("*** too late during sample rate transform: dropped buffer"); 1450 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1451 continue; 1452 } 1453 1454 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1455 " sampleDelta=[%d.%08x]", 1456 head.pts(), head.position(), pts, 1457 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1458 + (sampleDelta >> 32)), 1459 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1460 1461 // if the delta between the ideal placement for the next input sample and 1462 // the current output position is within this threshold, then we will 1463 // concatenate the next input samples to the previous output 1464 const int64_t kSampleContinuityThreshold = 1465 (static_cast<int64_t>(sr) << 32) / 250; 1466 1467 // if this is the first buffer of audio that we're emitting from this track 1468 // then it should be almost exactly on time. 1469 const int64_t kSampleStartupThreshold = 1LL << 32; 1470 1471 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1472 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1473 // the next input is close enough to being on time, so concatenate it 1474 // with the last output 1475 timedYieldSamples_l(buffer); 1476 1477 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1478 head.position(), buffer->frameCount); 1479 return NO_ERROR; 1480 } 1481 1482 // Looks like our output is not on time. Reset our on timed status. 1483 // Next time we mix samples from our input queue, then should be within 1484 // the StartupThreshold. 1485 mTimedAudioOutputOnTime = false; 1486 if (sampleDelta > 0) { 1487 // the gap between the current output position and the proper start of 1488 // the next input sample is too big, so fill it with silence 1489 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1490 1491 timedYieldSilence_l(framesUntilNextInput, buffer); 1492 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1493 return NO_ERROR; 1494 } else { 1495 // the next input sample is late 1496 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1497 size_t onTimeSamplePosition = 1498 head.position() + lateFrames * mFrameSize; 1499 1500 if (onTimeSamplePosition > head.buffer()->size()) { 1501 // all the remaining samples in the head are too late, so 1502 // drop it and move on 1503 ALOGV("*** too late: dropped buffer"); 1504 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1505 continue; 1506 } else { 1507 // skip over the late samples 1508 head.setPosition(onTimeSamplePosition); 1509 1510 // yield the available samples 1511 timedYieldSamples_l(buffer); 1512 1513 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1514 return NO_ERROR; 1515 } 1516 } 1517 } 1518} 1519 1520// Yield samples from the timed buffer queue head up to the given output 1521// buffer's capacity. 1522// 1523// Caller must hold mTimedBufferQueueLock 1524void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1525 AudioBufferProvider::Buffer* buffer) { 1526 1527 const TimedBuffer& head = mTimedBufferQueue[0]; 1528 1529 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1530 head.position()); 1531 1532 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1533 mFrameSize); 1534 size_t framesRequested = buffer->frameCount; 1535 buffer->frameCount = min(framesLeftInHead, framesRequested); 1536 1537 mQueueHeadInFlight = true; 1538 mTimedAudioOutputOnTime = true; 1539} 1540 1541// Yield samples of silence up to the given output buffer's capacity 1542// 1543// Caller must hold mTimedBufferQueueLock 1544void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1545 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1546 1547 // lazily allocate a buffer filled with silence 1548 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1549 delete [] mTimedSilenceBuffer; 1550 mTimedSilenceBufferSize = numFrames * mFrameSize; 1551 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1552 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1553 } 1554 1555 buffer->raw = mTimedSilenceBuffer; 1556 size_t framesRequested = buffer->frameCount; 1557 buffer->frameCount = min(numFrames, framesRequested); 1558 1559 mTimedAudioOutputOnTime = false; 1560} 1561 1562// AudioBufferProvider interface 1563void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1564 AudioBufferProvider::Buffer* buffer) { 1565 1566 Mutex::Autolock _l(mTimedBufferQueueLock); 1567 1568 // If the buffer which was just released is part of the buffer at the head 1569 // of the queue, be sure to update the amt of the buffer which has been 1570 // consumed. If the buffer being returned is not part of the head of the 1571 // queue, its either because the buffer is part of the silence buffer, or 1572 // because the head of the timed queue was trimmed after the mixer called 1573 // getNextBuffer but before the mixer called releaseBuffer. 1574 if (buffer->raw == mTimedSilenceBuffer) { 1575 ALOG_ASSERT(!mQueueHeadInFlight, 1576 "Queue head in flight during release of silence buffer!"); 1577 goto done; 1578 } 1579 1580 ALOG_ASSERT(mQueueHeadInFlight, 1581 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1582 " head in flight."); 1583 1584 if (mTimedBufferQueue.size()) { 1585 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1586 1587 void* start = head.buffer()->pointer(); 1588 void* end = reinterpret_cast<void*>( 1589 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1590 + head.buffer()->size()); 1591 1592 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1593 "released buffer not within the head of the timed buffer" 1594 " queue; qHead = [%p, %p], released buffer = %p", 1595 start, end, buffer->raw); 1596 1597 head.setPosition(head.position() + 1598 (buffer->frameCount * mFrameSize)); 1599 mQueueHeadInFlight = false; 1600 1601 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1602 "Bad bookkeeping during releaseBuffer! Should have at" 1603 " least %u queued frames, but we think we have only %u", 1604 buffer->frameCount, mFramesPendingInQueue); 1605 1606 mFramesPendingInQueue -= buffer->frameCount; 1607 1608 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1609 || mTrimQueueHeadOnRelease) { 1610 trimTimedBufferQueueHead_l("releaseBuffer"); 1611 mTrimQueueHeadOnRelease = false; 1612 } 1613 } else { 1614 LOG_ALWAYS_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1615 " buffers in the timed buffer queue"); 1616 } 1617 1618done: 1619 buffer->raw = 0; 1620 buffer->frameCount = 0; 1621} 1622 1623size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1624 Mutex::Autolock _l(mTimedBufferQueueLock); 1625 return mFramesPendingInQueue; 1626} 1627 1628AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1629 : mPTS(0), mPosition(0) {} 1630 1631AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1632 const sp<IMemory>& buffer, int64_t pts) 1633 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1634 1635 1636// ---------------------------------------------------------------------------- 1637 1638AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1639 PlaybackThread *playbackThread, 1640 DuplicatingThread *sourceThread, 1641 uint32_t sampleRate, 1642 audio_format_t format, 1643 audio_channel_mask_t channelMask, 1644 size_t frameCount, 1645 int uid) 1646 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1647 NULL, 0, 0, uid, IAudioFlinger::TRACK_DEFAULT, TYPE_OUTPUT), 1648 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1649{ 1650 1651 if (mCblk != NULL) { 1652 mOutBuffer.frameCount = 0; 1653 playbackThread->mTracks.add(this); 1654 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1655 "frameCount %u, mChannelMask 0x%08x", 1656 mCblk, mBuffer, 1657 frameCount, mChannelMask); 1658 // since client and server are in the same process, 1659 // the buffer has the same virtual address on both sides 1660 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1661 true /*clientInServer*/); 1662 mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); 1663 mClientProxy->setSendLevel(0.0); 1664 mClientProxy->setSampleRate(sampleRate); 1665 } else { 1666 ALOGW("Error creating output track on thread %p", playbackThread); 1667 } 1668} 1669 1670AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1671{ 1672 clearBufferQueue(); 1673 delete mClientProxy; 1674 // superclass destructor will now delete the server proxy and shared memory both refer to 1675} 1676 1677status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1678 int triggerSession) 1679{ 1680 status_t status = Track::start(event, triggerSession); 1681 if (status != NO_ERROR) { 1682 return status; 1683 } 1684 1685 mActive = true; 1686 mRetryCount = 127; 1687 return status; 1688} 1689 1690void AudioFlinger::PlaybackThread::OutputTrack::stop() 1691{ 1692 Track::stop(); 1693 clearBufferQueue(); 1694 mOutBuffer.frameCount = 0; 1695 mActive = false; 1696} 1697 1698bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1699{ 1700 Buffer *pInBuffer; 1701 Buffer inBuffer; 1702 uint32_t channelCount = mChannelCount; 1703 bool outputBufferFull = false; 1704 inBuffer.frameCount = frames; 1705 inBuffer.i16 = data; 1706 1707 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1708 1709 if (!mActive && frames != 0) { 1710 start(); 1711 sp<ThreadBase> thread = mThread.promote(); 1712 if (thread != 0) { 1713 MixerThread *mixerThread = (MixerThread *)thread.get(); 1714 if (mFrameCount > frames) { 1715 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1716 uint32_t startFrames = (mFrameCount - frames); 1717 pInBuffer = new Buffer; 1718 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1719 pInBuffer->frameCount = startFrames; 1720 pInBuffer->i16 = pInBuffer->mBuffer; 1721 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1722 mBufferQueue.add(pInBuffer); 1723 } else { 1724 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1725 } 1726 } 1727 } 1728 } 1729 1730 while (waitTimeLeftMs) { 1731 // First write pending buffers, then new data 1732 if (mBufferQueue.size()) { 1733 pInBuffer = mBufferQueue.itemAt(0); 1734 } else { 1735 pInBuffer = &inBuffer; 1736 } 1737 1738 if (pInBuffer->frameCount == 0) { 1739 break; 1740 } 1741 1742 if (mOutBuffer.frameCount == 0) { 1743 mOutBuffer.frameCount = pInBuffer->frameCount; 1744 nsecs_t startTime = systemTime(); 1745 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1746 if (status != NO_ERROR) { 1747 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1748 mThread.unsafe_get(), status); 1749 outputBufferFull = true; 1750 break; 1751 } 1752 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1753 if (waitTimeLeftMs >= waitTimeMs) { 1754 waitTimeLeftMs -= waitTimeMs; 1755 } else { 1756 waitTimeLeftMs = 0; 1757 } 1758 } 1759 1760 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1761 pInBuffer->frameCount; 1762 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1763 Proxy::Buffer buf; 1764 buf.mFrameCount = outFrames; 1765 buf.mRaw = NULL; 1766 mClientProxy->releaseBuffer(&buf); 1767 pInBuffer->frameCount -= outFrames; 1768 pInBuffer->i16 += outFrames * channelCount; 1769 mOutBuffer.frameCount -= outFrames; 1770 mOutBuffer.i16 += outFrames * channelCount; 1771 1772 if (pInBuffer->frameCount == 0) { 1773 if (mBufferQueue.size()) { 1774 mBufferQueue.removeAt(0); 1775 delete [] pInBuffer->mBuffer; 1776 delete pInBuffer; 1777 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1778 mThread.unsafe_get(), mBufferQueue.size()); 1779 } else { 1780 break; 1781 } 1782 } 1783 } 1784 1785 // If we could not write all frames, allocate a buffer and queue it for next time. 1786 if (inBuffer.frameCount) { 1787 sp<ThreadBase> thread = mThread.promote(); 1788 if (thread != 0 && !thread->standby()) { 1789 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1790 pInBuffer = new Buffer; 1791 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1792 pInBuffer->frameCount = inBuffer.frameCount; 1793 pInBuffer->i16 = pInBuffer->mBuffer; 1794 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1795 sizeof(int16_t)); 1796 mBufferQueue.add(pInBuffer); 1797 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1798 mThread.unsafe_get(), mBufferQueue.size()); 1799 } else { 1800 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1801 mThread.unsafe_get(), this); 1802 } 1803 } 1804 } 1805 1806 // Calling write() with a 0 length buffer, means that no more data will be written: 1807 // If no more buffers are pending, fill output track buffer to make sure it is started 1808 // by output mixer. 1809 if (frames == 0 && mBufferQueue.size() == 0) { 1810 // FIXME borken, replace by getting framesReady() from proxy 1811 size_t user = 0; // was mCblk->user 1812 if (user < mFrameCount) { 1813 frames = mFrameCount - user; 1814 pInBuffer = new Buffer; 1815 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1816 pInBuffer->frameCount = frames; 1817 pInBuffer->i16 = pInBuffer->mBuffer; 1818 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1819 mBufferQueue.add(pInBuffer); 1820 } else if (mActive) { 1821 stop(); 1822 } 1823 } 1824 1825 return outputBufferFull; 1826} 1827 1828status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1829 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1830{ 1831 ClientProxy::Buffer buf; 1832 buf.mFrameCount = buffer->frameCount; 1833 struct timespec timeout; 1834 timeout.tv_sec = waitTimeMs / 1000; 1835 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1836 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1837 buffer->frameCount = buf.mFrameCount; 1838 buffer->raw = buf.mRaw; 1839 return status; 1840} 1841 1842void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1843{ 1844 size_t size = mBufferQueue.size(); 1845 1846 for (size_t i = 0; i < size; i++) { 1847 Buffer *pBuffer = mBufferQueue.itemAt(i); 1848 delete [] pBuffer->mBuffer; 1849 delete pBuffer; 1850 } 1851 mBufferQueue.clear(); 1852} 1853 1854 1855AudioFlinger::PlaybackThread::PatchTrack::PatchTrack(PlaybackThread *playbackThread, 1856 uint32_t sampleRate, 1857 audio_channel_mask_t channelMask, 1858 audio_format_t format, 1859 size_t frameCount, 1860 void *buffer, 1861 IAudioFlinger::track_flags_t flags) 1862 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1863 buffer, 0, 0, getuid(), flags, TYPE_PATCH), 1864 mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true)) 1865{ 1866 uint64_t mixBufferNs = ((uint64_t)2 * playbackThread->frameCount() * 1000000000) / 1867 playbackThread->sampleRate(); 1868 mPeerTimeout.tv_sec = mixBufferNs / 1000000000; 1869 mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000); 1870 1871 ALOGV("PatchTrack %p sampleRate %d mPeerTimeout %d.%03d sec", 1872 this, sampleRate, 1873 (int)mPeerTimeout.tv_sec, 1874 (int)(mPeerTimeout.tv_nsec / 1000000)); 1875} 1876 1877AudioFlinger::PlaybackThread::PatchTrack::~PatchTrack() 1878{ 1879} 1880 1881// AudioBufferProvider interface 1882status_t AudioFlinger::PlaybackThread::PatchTrack::getNextBuffer( 1883 AudioBufferProvider::Buffer* buffer, int64_t pts) 1884{ 1885 ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::getNextBuffer() called without peer proxy"); 1886 Proxy::Buffer buf; 1887 buf.mFrameCount = buffer->frameCount; 1888 status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout); 1889 ALOGV_IF(status != NO_ERROR, "PatchTrack() %p getNextBuffer status %d", this, status); 1890 buffer->frameCount = buf.mFrameCount; 1891 if (buf.mFrameCount == 0) { 1892 return WOULD_BLOCK; 1893 } 1894 status = Track::getNextBuffer(buffer, pts); 1895 return status; 1896} 1897 1898void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(AudioBufferProvider::Buffer* buffer) 1899{ 1900 ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::releaseBuffer() called without peer proxy"); 1901 Proxy::Buffer buf; 1902 buf.mFrameCount = buffer->frameCount; 1903 buf.mRaw = buffer->raw; 1904 mPeerProxy->releaseBuffer(&buf); 1905 TrackBase::releaseBuffer(buffer); 1906} 1907 1908status_t AudioFlinger::PlaybackThread::PatchTrack::obtainBuffer(Proxy::Buffer* buffer, 1909 const struct timespec *timeOut) 1910{ 1911 return mProxy->obtainBuffer(buffer, timeOut); 1912} 1913 1914void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(Proxy::Buffer* buffer) 1915{ 1916 mProxy->releaseBuffer(buffer); 1917 if (android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags) & CBLK_DISABLED) { 1918 ALOGW("PatchTrack::releaseBuffer() disabled due to previous underrun, restarting"); 1919 start(); 1920 } 1921 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1922} 1923 1924// ---------------------------------------------------------------------------- 1925// Record 1926// ---------------------------------------------------------------------------- 1927 1928AudioFlinger::RecordHandle::RecordHandle( 1929 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1930 : BnAudioRecord(), 1931 mRecordTrack(recordTrack) 1932{ 1933} 1934 1935AudioFlinger::RecordHandle::~RecordHandle() { 1936 stop_nonvirtual(); 1937 mRecordTrack->destroy(); 1938} 1939 1940status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1941 int triggerSession) { 1942 ALOGV("RecordHandle::start()"); 1943 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1944} 1945 1946void AudioFlinger::RecordHandle::stop() { 1947 stop_nonvirtual(); 1948} 1949 1950void AudioFlinger::RecordHandle::stop_nonvirtual() { 1951 ALOGV("RecordHandle::stop()"); 1952 mRecordTrack->stop(); 1953} 1954 1955status_t AudioFlinger::RecordHandle::onTransact( 1956 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1957{ 1958 return BnAudioRecord::onTransact(code, data, reply, flags); 1959} 1960 1961// ---------------------------------------------------------------------------- 1962 1963// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 1964AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1965 RecordThread *thread, 1966 const sp<Client>& client, 1967 uint32_t sampleRate, 1968 audio_format_t format, 1969 audio_channel_mask_t channelMask, 1970 size_t frameCount, 1971 void *buffer, 1972 int sessionId, 1973 int uid, 1974 IAudioFlinger::track_flags_t flags, 1975 track_type type) 1976 : TrackBase(thread, client, sampleRate, format, 1977 channelMask, frameCount, buffer, sessionId, uid, 1978 flags, false /*isOut*/, 1979 (type == TYPE_DEFAULT) ? 1980 ((flags & IAudioFlinger::TRACK_FAST) ? ALLOC_PIPE : ALLOC_CBLK) : 1981 ((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE), 1982 type), 1983 mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0), 1984 // See real initialization of mRsmpInFront at RecordThread::start() 1985 mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL) 1986{ 1987 if (mCblk == NULL) { 1988 return; 1989 } 1990 1991 mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, 1992 mFrameSize, !isExternalTrack()); 1993 1994 uint32_t channelCount = audio_channel_count_from_in_mask(channelMask); 1995 // FIXME I don't understand either of the channel count checks 1996 if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 && 1997 channelCount <= FCC_2) { 1998 // sink SR 1999 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT, 2000 thread->mChannelCount, sampleRate); 2001 // source SR 2002 mResampler->setSampleRate(thread->mSampleRate); 2003 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 2004 mResamplerBufferProvider = new ResamplerBufferProvider(this); 2005 } 2006 2007 if (flags & IAudioFlinger::TRACK_FAST) { 2008 ALOG_ASSERT(thread->mFastTrackAvail); 2009 thread->mFastTrackAvail = false; 2010 } 2011} 2012 2013AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 2014{ 2015 ALOGV("%s", __func__); 2016 delete mResampler; 2017 delete[] mRsmpOutBuffer; 2018 delete mResamplerBufferProvider; 2019} 2020 2021// AudioBufferProvider interface 2022status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 2023 int64_t pts __unused) 2024{ 2025 ServerProxy::Buffer buf; 2026 buf.mFrameCount = buffer->frameCount; 2027 status_t status = mServerProxy->obtainBuffer(&buf); 2028 buffer->frameCount = buf.mFrameCount; 2029 buffer->raw = buf.mRaw; 2030 if (buf.mFrameCount == 0) { 2031 // FIXME also wake futex so that overrun is noticed more quickly 2032 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 2033 } 2034 return status; 2035} 2036 2037status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 2038 int triggerSession) 2039{ 2040 sp<ThreadBase> thread = mThread.promote(); 2041 if (thread != 0) { 2042 RecordThread *recordThread = (RecordThread *)thread.get(); 2043 return recordThread->start(this, event, triggerSession); 2044 } else { 2045 return BAD_VALUE; 2046 } 2047} 2048 2049void AudioFlinger::RecordThread::RecordTrack::stop() 2050{ 2051 sp<ThreadBase> thread = mThread.promote(); 2052 if (thread != 0) { 2053 RecordThread *recordThread = (RecordThread *)thread.get(); 2054 if (recordThread->stop(this) && isExternalTrack()) { 2055 AudioSystem::stopInput(mThreadIoHandle, (audio_session_t)mSessionId); 2056 } 2057 } 2058} 2059 2060void AudioFlinger::RecordThread::RecordTrack::destroy() 2061{ 2062 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 2063 sp<RecordTrack> keep(this); 2064 { 2065 if (isExternalTrack()) { 2066 if (mState == ACTIVE || mState == RESUMING) { 2067 AudioSystem::stopInput(mThreadIoHandle, (audio_session_t)mSessionId); 2068 } 2069 AudioSystem::releaseInput(mThreadIoHandle, (audio_session_t)mSessionId); 2070 } 2071 sp<ThreadBase> thread = mThread.promote(); 2072 if (thread != 0) { 2073 Mutex::Autolock _l(thread->mLock); 2074 RecordThread *recordThread = (RecordThread *) thread.get(); 2075 recordThread->destroyTrack_l(this); 2076 } 2077 } 2078} 2079 2080void AudioFlinger::RecordThread::RecordTrack::invalidate() 2081{ 2082 // FIXME should use proxy, and needs work 2083 audio_track_cblk_t* cblk = mCblk; 2084 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 2085 android_atomic_release_store(0x40000000, &cblk->mFutex); 2086 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 2087 (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); 2088} 2089 2090 2091/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 2092{ 2093 result.append(" Active Client Fmt Chn mask Session S Server fCount SRate\n"); 2094} 2095 2096void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active) 2097{ 2098 snprintf(buffer, size, " %6s %6u %3u %08X %7u %1d %08X %6zu %5u\n", 2099 active ? "yes" : "no", 2100 (mClient == 0) ? getpid_cached : mClient->pid(), 2101 mFormat, 2102 mChannelMask, 2103 mSessionId, 2104 mState, 2105 mCblk->mServer, 2106 mFrameCount, 2107 mSampleRate); 2108 2109} 2110 2111void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event) 2112{ 2113 if (event == mSyncStartEvent) { 2114 ssize_t framesToDrop = 0; 2115 sp<ThreadBase> threadBase = mThread.promote(); 2116 if (threadBase != 0) { 2117 // TODO: use actual buffer filling status instead of 2 buffers when info is available 2118 // from audio HAL 2119 framesToDrop = threadBase->mFrameCount * 2; 2120 } 2121 mFramesToDrop = framesToDrop; 2122 } 2123} 2124 2125void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent() 2126{ 2127 if (mSyncStartEvent != 0) { 2128 mSyncStartEvent->cancel(); 2129 mSyncStartEvent.clear(); 2130 } 2131 mFramesToDrop = 0; 2132} 2133 2134 2135AudioFlinger::RecordThread::PatchRecord::PatchRecord(RecordThread *recordThread, 2136 uint32_t sampleRate, 2137 audio_channel_mask_t channelMask, 2138 audio_format_t format, 2139 size_t frameCount, 2140 void *buffer, 2141 IAudioFlinger::track_flags_t flags) 2142 : RecordTrack(recordThread, NULL, sampleRate, format, channelMask, frameCount, 2143 buffer, 0, getuid(), flags, TYPE_PATCH), 2144 mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true)) 2145{ 2146 uint64_t mixBufferNs = ((uint64_t)2 * recordThread->frameCount() * 1000000000) / 2147 recordThread->sampleRate(); 2148 mPeerTimeout.tv_sec = mixBufferNs / 1000000000; 2149 mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000); 2150 2151 ALOGV("PatchRecord %p sampleRate %d mPeerTimeout %d.%03d sec", 2152 this, sampleRate, 2153 (int)mPeerTimeout.tv_sec, 2154 (int)(mPeerTimeout.tv_nsec / 1000000)); 2155} 2156 2157AudioFlinger::RecordThread::PatchRecord::~PatchRecord() 2158{ 2159} 2160 2161// AudioBufferProvider interface 2162status_t AudioFlinger::RecordThread::PatchRecord::getNextBuffer( 2163 AudioBufferProvider::Buffer* buffer, int64_t pts) 2164{ 2165 ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::getNextBuffer() called without peer proxy"); 2166 Proxy::Buffer buf; 2167 buf.mFrameCount = buffer->frameCount; 2168 status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout); 2169 ALOGV_IF(status != NO_ERROR, 2170 "PatchRecord() %p mPeerProxy->obtainBuffer status %d", this, status); 2171 buffer->frameCount = buf.mFrameCount; 2172 if (buf.mFrameCount == 0) { 2173 return WOULD_BLOCK; 2174 } 2175 status = RecordTrack::getNextBuffer(buffer, pts); 2176 return status; 2177} 2178 2179void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(AudioBufferProvider::Buffer* buffer) 2180{ 2181 ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::releaseBuffer() called without peer proxy"); 2182 Proxy::Buffer buf; 2183 buf.mFrameCount = buffer->frameCount; 2184 buf.mRaw = buffer->raw; 2185 mPeerProxy->releaseBuffer(&buf); 2186 TrackBase::releaseBuffer(buffer); 2187} 2188 2189status_t AudioFlinger::RecordThread::PatchRecord::obtainBuffer(Proxy::Buffer* buffer, 2190 const struct timespec *timeOut) 2191{ 2192 return mProxy->obtainBuffer(buffer, timeOut); 2193} 2194 2195void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(Proxy::Buffer* buffer) 2196{ 2197 mProxy->releaseBuffer(buffer); 2198} 2199 2200}; // namespace android 2201