Tracks.cpp revision e659ef420dae0caae84ab78f9df8952acb9ad3a0
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 if (thread->type() == ThreadBase::DIRECT) { 828 DirectOutputThread *t = (DirectOutputThread *)playbackThread; 829 t->flushHw_l(); 830 } 831 } 832 } 833 // Prevent flush being lost if the track is flushed and then resumed 834 // before mixer thread can run. This is important when offloading 835 // because the hardware buffer could hold a large amount of audio 836 playbackThread->broadcast_l(); 837 } 838} 839 840// must be called with thread lock held 841void AudioFlinger::PlaybackThread::Track::flushAck() 842{ 843 if (!isOffloaded()) 844 return; 845 846 mFlushHwPending = false; 847} 848 849void AudioFlinger::PlaybackThread::Track::reset() 850{ 851 // Do not reset twice to avoid discarding data written just after a flush and before 852 // the audioflinger thread detects the track is stopped. 853 if (!mResetDone) { 854 // Force underrun condition to avoid false underrun callback until first data is 855 // written to buffer 856 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 857 mFillingUpStatus = FS_FILLING; 858 mResetDone = true; 859 if (mState == FLUSHED) { 860 mState = IDLE; 861 } 862 } 863} 864 865status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 866{ 867 sp<ThreadBase> thread = mThread.promote(); 868 if (thread == 0) { 869 ALOGE("thread is dead"); 870 return FAILED_TRANSACTION; 871 } else if ((thread->type() == ThreadBase::DIRECT) || 872 (thread->type() == ThreadBase::OFFLOAD)) { 873 return thread->setParameters(keyValuePairs); 874 } else { 875 return PERMISSION_DENIED; 876 } 877} 878 879status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp) 880{ 881 // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant 882 if (isFastTrack()) { 883 // FIXME no lock held to set mPreviousValid = false 884 return INVALID_OPERATION; 885 } 886 sp<ThreadBase> thread = mThread.promote(); 887 if (thread == 0) { 888 // FIXME no lock held to set mPreviousValid = false 889 return INVALID_OPERATION; 890 } 891 Mutex::Autolock _l(thread->mLock); 892 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 893 if (!isOffloaded() && !isDirect()) { 894 if (!playbackThread->mLatchQValid) { 895 mPreviousValid = false; 896 return INVALID_OPERATION; 897 } 898 uint32_t unpresentedFrames = 899 ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) / 900 playbackThread->mSampleRate; 901 uint32_t framesWritten = mAudioTrackServerProxy->framesReleased(); 902 bool checkPreviousTimestamp = mPreviousValid && framesWritten >= mPreviousFramesWritten; 903 if (framesWritten < unpresentedFrames) { 904 mPreviousValid = false; 905 return INVALID_OPERATION; 906 } 907 mPreviousFramesWritten = framesWritten; 908 uint32_t position = framesWritten - unpresentedFrames; 909 struct timespec time = playbackThread->mLatchQ.mTimestamp.mTime; 910 if (checkPreviousTimestamp) { 911 if (time.tv_sec < mPreviousTimestamp.mTime.tv_sec || 912 (time.tv_sec == mPreviousTimestamp.mTime.tv_sec && 913 time.tv_nsec < mPreviousTimestamp.mTime.tv_nsec)) { 914 ALOGW("Time is going backwards"); 915 } 916 // position can bobble slightly as an artifact; this hides the bobble 917 static const uint32_t MINIMUM_POSITION_DELTA = 8u; 918 if ((position <= mPreviousTimestamp.mPosition) || 919 (position - mPreviousTimestamp.mPosition) < MINIMUM_POSITION_DELTA) { 920 position = mPreviousTimestamp.mPosition; 921 time = mPreviousTimestamp.mTime; 922 } 923 } 924 timestamp.mPosition = position; 925 timestamp.mTime = time; 926 mPreviousTimestamp = timestamp; 927 mPreviousValid = true; 928 return NO_ERROR; 929 } 930 931 return playbackThread->getTimestamp_l(timestamp); 932} 933 934status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 935{ 936 status_t status = DEAD_OBJECT; 937 sp<ThreadBase> thread = mThread.promote(); 938 if (thread != 0) { 939 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 940 sp<AudioFlinger> af = mClient->audioFlinger(); 941 942 Mutex::Autolock _l(af->mLock); 943 944 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 945 946 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 947 Mutex::Autolock _dl(playbackThread->mLock); 948 Mutex::Autolock _sl(srcThread->mLock); 949 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 950 if (chain == 0) { 951 return INVALID_OPERATION; 952 } 953 954 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 955 if (effect == 0) { 956 return INVALID_OPERATION; 957 } 958 srcThread->removeEffect_l(effect); 959 status = playbackThread->addEffect_l(effect); 960 if (status != NO_ERROR) { 961 srcThread->addEffect_l(effect); 962 return INVALID_OPERATION; 963 } 964 // removeEffect_l() has stopped the effect if it was active so it must be restarted 965 if (effect->state() == EffectModule::ACTIVE || 966 effect->state() == EffectModule::STOPPING) { 967 effect->start(); 968 } 969 970 sp<EffectChain> dstChain = effect->chain().promote(); 971 if (dstChain == 0) { 972 srcThread->addEffect_l(effect); 973 return INVALID_OPERATION; 974 } 975 AudioSystem::unregisterEffect(effect->id()); 976 AudioSystem::registerEffect(&effect->desc(), 977 srcThread->id(), 978 dstChain->strategy(), 979 AUDIO_SESSION_OUTPUT_MIX, 980 effect->id()); 981 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 982 } 983 status = playbackThread->attachAuxEffect(this, EffectId); 984 } 985 return status; 986} 987 988void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 989{ 990 mAuxEffectId = EffectId; 991 mAuxBuffer = buffer; 992} 993 994bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 995 size_t audioHalFrames) 996{ 997 // a track is considered presented when the total number of frames written to audio HAL 998 // corresponds to the number of frames written when presentationComplete() is called for the 999 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 1000 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 1001 // to detect when all frames have been played. In this case framesWritten isn't 1002 // useful because it doesn't always reflect whether there is data in the h/w 1003 // buffers, particularly if a track has been paused and resumed during draining 1004 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 1005 mPresentationCompleteFrames, framesWritten); 1006 if (mPresentationCompleteFrames == 0) { 1007 mPresentationCompleteFrames = framesWritten + audioHalFrames; 1008 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 1009 mPresentationCompleteFrames, audioHalFrames); 1010 } 1011 1012 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 1013 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1014 mAudioTrackServerProxy->setStreamEndDone(); 1015 return true; 1016 } 1017 return false; 1018} 1019 1020void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 1021{ 1022 for (size_t i = 0; i < mSyncEvents.size(); i++) { 1023 if (mSyncEvents[i]->type() == type) { 1024 mSyncEvents[i]->trigger(); 1025 mSyncEvents.removeAt(i); 1026 i--; 1027 } 1028 } 1029} 1030 1031// implement VolumeBufferProvider interface 1032 1033gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 1034{ 1035 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 1036 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 1037 gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR(); 1038 float vl = float_from_gain(gain_minifloat_unpack_left(vlr)); 1039 float vr = float_from_gain(gain_minifloat_unpack_right(vlr)); 1040 // track volumes come from shared memory, so can't be trusted and must be clamped 1041 if (vl > GAIN_FLOAT_UNITY) { 1042 vl = GAIN_FLOAT_UNITY; 1043 } 1044 if (vr > GAIN_FLOAT_UNITY) { 1045 vr = GAIN_FLOAT_UNITY; 1046 } 1047 // now apply the cached master volume and stream type volume; 1048 // this is trusted but lacks any synchronization or barrier so may be stale 1049 float v = mCachedVolume; 1050 vl *= v; 1051 vr *= v; 1052 // re-combine into packed minifloat 1053 vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr)); 1054 // FIXME look at mute, pause, and stop flags 1055 return vlr; 1056} 1057 1058status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 1059{ 1060 if (isTerminated() || mState == PAUSED || 1061 ((framesReady() == 0) && ((mSharedBuffer != 0) || 1062 (mState == STOPPED)))) { 1063 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 1064 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 1065 event->cancel(); 1066 return INVALID_OPERATION; 1067 } 1068 (void) TrackBase::setSyncEvent(event); 1069 return NO_ERROR; 1070} 1071 1072void AudioFlinger::PlaybackThread::Track::invalidate() 1073{ 1074 // FIXME should use proxy, and needs work 1075 audio_track_cblk_t* cblk = mCblk; 1076 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1077 android_atomic_release_store(0x40000000, &cblk->mFutex); 1078 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 1079 (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); 1080 mIsInvalid = true; 1081} 1082 1083void AudioFlinger::PlaybackThread::Track::signal() 1084{ 1085 sp<ThreadBase> thread = mThread.promote(); 1086 if (thread != 0) { 1087 PlaybackThread *t = (PlaybackThread *)thread.get(); 1088 Mutex::Autolock _l(t->mLock); 1089 t->broadcast_l(); 1090 } 1091} 1092 1093//To be called with thread lock held 1094bool AudioFlinger::PlaybackThread::Track::isResumePending() { 1095 1096 if (mState == RESUMING) 1097 return true; 1098 /* Resume is pending if track was stopping before pause was called */ 1099 if (mState == STOPPING_1 && 1100 mResumeToStopping) 1101 return true; 1102 1103 return false; 1104} 1105 1106//To be called with thread lock held 1107void AudioFlinger::PlaybackThread::Track::resumeAck() { 1108 1109 1110 if (mState == RESUMING) 1111 mState = ACTIVE; 1112 1113 // Other possibility of pending resume is stopping_1 state 1114 // Do not update the state from stopping as this prevents 1115 // drain being called. 1116 if (mState == STOPPING_1) { 1117 mResumeToStopping = false; 1118 } 1119} 1120// ---------------------------------------------------------------------------- 1121 1122sp<AudioFlinger::PlaybackThread::TimedTrack> 1123AudioFlinger::PlaybackThread::TimedTrack::create( 1124 PlaybackThread *thread, 1125 const sp<Client>& client, 1126 audio_stream_type_t streamType, 1127 uint32_t sampleRate, 1128 audio_format_t format, 1129 audio_channel_mask_t channelMask, 1130 size_t frameCount, 1131 const sp<IMemory>& sharedBuffer, 1132 int sessionId, 1133 int uid) 1134{ 1135 if (!client->reserveTimedTrack()) 1136 return 0; 1137 1138 return new TimedTrack( 1139 thread, client, streamType, sampleRate, format, channelMask, frameCount, 1140 sharedBuffer, sessionId, uid); 1141} 1142 1143AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 1144 PlaybackThread *thread, 1145 const sp<Client>& client, 1146 audio_stream_type_t streamType, 1147 uint32_t sampleRate, 1148 audio_format_t format, 1149 audio_channel_mask_t channelMask, 1150 size_t frameCount, 1151 const sp<IMemory>& sharedBuffer, 1152 int sessionId, 1153 int uid) 1154 : Track(thread, client, streamType, sampleRate, format, channelMask, 1155 frameCount, (sharedBuffer != 0) ? sharedBuffer->pointer() : NULL, sharedBuffer, 1156 sessionId, uid, IAudioFlinger::TRACK_TIMED, TYPE_TIMED), 1157 mQueueHeadInFlight(false), 1158 mTrimQueueHeadOnRelease(false), 1159 mFramesPendingInQueue(0), 1160 mTimedSilenceBuffer(NULL), 1161 mTimedSilenceBufferSize(0), 1162 mTimedAudioOutputOnTime(false), 1163 mMediaTimeTransformValid(false) 1164{ 1165 LocalClock lc; 1166 mLocalTimeFreq = lc.getLocalFreq(); 1167 1168 mLocalTimeToSampleTransform.a_zero = 0; 1169 mLocalTimeToSampleTransform.b_zero = 0; 1170 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 1171 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 1172 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 1173 &mLocalTimeToSampleTransform.a_to_b_denom); 1174 1175 mMediaTimeToSampleTransform.a_zero = 0; 1176 mMediaTimeToSampleTransform.b_zero = 0; 1177 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 1178 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 1179 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 1180 &mMediaTimeToSampleTransform.a_to_b_denom); 1181} 1182 1183AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 1184 mClient->releaseTimedTrack(); 1185 delete [] mTimedSilenceBuffer; 1186} 1187 1188status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 1189 size_t size, sp<IMemory>* buffer) { 1190 1191 Mutex::Autolock _l(mTimedBufferQueueLock); 1192 1193 trimTimedBufferQueue_l(); 1194 1195 // lazily initialize the shared memory heap for timed buffers 1196 if (mTimedMemoryDealer == NULL) { 1197 const int kTimedBufferHeapSize = 512 << 10; 1198 1199 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 1200 "AudioFlingerTimed"); 1201 if (mTimedMemoryDealer == NULL) { 1202 return NO_MEMORY; 1203 } 1204 } 1205 1206 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 1207 if (newBuffer == 0 || newBuffer->pointer() == NULL) { 1208 return NO_MEMORY; 1209 } 1210 1211 *buffer = newBuffer; 1212 return NO_ERROR; 1213} 1214 1215// caller must hold mTimedBufferQueueLock 1216void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 1217 int64_t mediaTimeNow; 1218 { 1219 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1220 if (!mMediaTimeTransformValid) 1221 return; 1222 1223 int64_t targetTimeNow; 1224 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 1225 ? mCCHelper.getCommonTime(&targetTimeNow) 1226 : mCCHelper.getLocalTime(&targetTimeNow); 1227 1228 if (OK != res) 1229 return; 1230 1231 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 1232 &mediaTimeNow)) { 1233 return; 1234 } 1235 } 1236 1237 size_t trimEnd; 1238 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 1239 int64_t bufEnd; 1240 1241 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 1242 // We have a next buffer. Just use its PTS as the PTS of the frame 1243 // following the last frame in this buffer. If the stream is sparse 1244 // (ie, there are deliberate gaps left in the stream which should be 1245 // filled with silence by the TimedAudioTrack), then this can result 1246 // in one extra buffer being left un-trimmed when it could have 1247 // been. In general, this is not typical, and we would rather 1248 // optimized away the TS calculation below for the more common case 1249 // where PTSes are contiguous. 1250 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 1251 } else { 1252 // We have no next buffer. Compute the PTS of the frame following 1253 // the last frame in this buffer by computing the duration of of 1254 // this frame in media time units and adding it to the PTS of the 1255 // buffer. 1256 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1257 / mFrameSize; 1258 1259 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1260 &bufEnd)) { 1261 ALOGE("Failed to convert frame count of %lld to media time" 1262 " duration" " (scale factor %d/%u) in %s", 1263 frameCount, 1264 mMediaTimeToSampleTransform.a_to_b_numer, 1265 mMediaTimeToSampleTransform.a_to_b_denom, 1266 __PRETTY_FUNCTION__); 1267 break; 1268 } 1269 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1270 } 1271 1272 if (bufEnd > mediaTimeNow) 1273 break; 1274 1275 // Is the buffer we want to use in the middle of a mix operation right 1276 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1277 // from the mixer which should be coming back shortly. 1278 if (!trimEnd && mQueueHeadInFlight) { 1279 mTrimQueueHeadOnRelease = true; 1280 } 1281 } 1282 1283 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1284 if (trimStart < trimEnd) { 1285 // Update the bookkeeping for framesReady() 1286 for (size_t i = trimStart; i < trimEnd; ++i) { 1287 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1288 } 1289 1290 // Now actually remove the buffers from the queue. 1291 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1292 } 1293} 1294 1295void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1296 const char* logTag) { 1297 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1298 "%s called (reason \"%s\"), but timed buffer queue has no" 1299 " elements to trim.", __FUNCTION__, logTag); 1300 1301 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1302 mTimedBufferQueue.removeAt(0); 1303} 1304 1305void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1306 const TimedBuffer& buf, 1307 const char* logTag __unused) { 1308 uint32_t bufBytes = buf.buffer()->size(); 1309 uint32_t consumedAlready = buf.position(); 1310 1311 ALOG_ASSERT(consumedAlready <= bufBytes, 1312 "Bad bookkeeping while updating frames pending. Timed buffer is" 1313 " only %u bytes long, but claims to have consumed %u" 1314 " bytes. (update reason: \"%s\")", 1315 bufBytes, consumedAlready, logTag); 1316 1317 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1318 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1319 "Bad bookkeeping while updating frames pending. Should have at" 1320 " least %u queued frames, but we think we have only %u. (update" 1321 " reason: \"%s\")", 1322 bufFrames, mFramesPendingInQueue, logTag); 1323 1324 mFramesPendingInQueue -= bufFrames; 1325} 1326 1327status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1328 const sp<IMemory>& buffer, int64_t pts) { 1329 1330 { 1331 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1332 if (!mMediaTimeTransformValid) 1333 return INVALID_OPERATION; 1334 } 1335 1336 Mutex::Autolock _l(mTimedBufferQueueLock); 1337 1338 uint32_t bufFrames = buffer->size() / mFrameSize; 1339 mFramesPendingInQueue += bufFrames; 1340 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1341 1342 return NO_ERROR; 1343} 1344 1345status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1346 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1347 1348 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1349 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1350 target); 1351 1352 if (!(target == TimedAudioTrack::LOCAL_TIME || 1353 target == TimedAudioTrack::COMMON_TIME)) { 1354 return BAD_VALUE; 1355 } 1356 1357 Mutex::Autolock lock(mMediaTimeTransformLock); 1358 mMediaTimeTransform = xform; 1359 mMediaTimeTransformTarget = target; 1360 mMediaTimeTransformValid = true; 1361 1362 return NO_ERROR; 1363} 1364 1365#define min(a, b) ((a) < (b) ? (a) : (b)) 1366 1367// implementation of getNextBuffer for tracks whose buffers have timestamps 1368status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1369 AudioBufferProvider::Buffer* buffer, int64_t pts) 1370{ 1371 if (pts == AudioBufferProvider::kInvalidPTS) { 1372 buffer->raw = NULL; 1373 buffer->frameCount = 0; 1374 mTimedAudioOutputOnTime = false; 1375 return INVALID_OPERATION; 1376 } 1377 1378 Mutex::Autolock _l(mTimedBufferQueueLock); 1379 1380 ALOG_ASSERT(!mQueueHeadInFlight, 1381 "getNextBuffer called without releaseBuffer!"); 1382 1383 while (true) { 1384 1385 // if we have no timed buffers, then fail 1386 if (mTimedBufferQueue.isEmpty()) { 1387 buffer->raw = NULL; 1388 buffer->frameCount = 0; 1389 return NOT_ENOUGH_DATA; 1390 } 1391 1392 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1393 1394 // calculate the PTS of the head of the timed buffer queue expressed in 1395 // local time 1396 int64_t headLocalPTS; 1397 { 1398 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1399 1400 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1401 1402 if (mMediaTimeTransform.a_to_b_denom == 0) { 1403 // the transform represents a pause, so yield silence 1404 timedYieldSilence_l(buffer->frameCount, buffer); 1405 return NO_ERROR; 1406 } 1407 1408 int64_t transformedPTS; 1409 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1410 &transformedPTS)) { 1411 // the transform failed. this shouldn't happen, but if it does 1412 // then just drop this buffer 1413 ALOGW("timedGetNextBuffer transform failed"); 1414 buffer->raw = NULL; 1415 buffer->frameCount = 0; 1416 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1417 return NO_ERROR; 1418 } 1419 1420 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1421 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1422 &headLocalPTS)) { 1423 buffer->raw = NULL; 1424 buffer->frameCount = 0; 1425 return INVALID_OPERATION; 1426 } 1427 } else { 1428 headLocalPTS = transformedPTS; 1429 } 1430 } 1431 1432 uint32_t sr = sampleRate(); 1433 1434 // adjust the head buffer's PTS to reflect the portion of the head buffer 1435 // that has already been consumed 1436 int64_t effectivePTS = headLocalPTS + 1437 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1438 1439 // Calculate the delta in samples between the head of the input buffer 1440 // queue and the start of the next output buffer that will be written. 1441 // If the transformation fails because of over or underflow, it means 1442 // that the sample's position in the output stream is so far out of 1443 // whack that it should just be dropped. 1444 int64_t sampleDelta; 1445 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1446 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1447 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1448 " mix"); 1449 continue; 1450 } 1451 if (!mLocalTimeToSampleTransform.doForwardTransform( 1452 (effectivePTS - pts) << 32, &sampleDelta)) { 1453 ALOGV("*** too late during sample rate transform: dropped buffer"); 1454 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1455 continue; 1456 } 1457 1458 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1459 " sampleDelta=[%d.%08x]", 1460 head.pts(), head.position(), pts, 1461 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1462 + (sampleDelta >> 32)), 1463 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1464 1465 // if the delta between the ideal placement for the next input sample and 1466 // the current output position is within this threshold, then we will 1467 // concatenate the next input samples to the previous output 1468 const int64_t kSampleContinuityThreshold = 1469 (static_cast<int64_t>(sr) << 32) / 250; 1470 1471 // if this is the first buffer of audio that we're emitting from this track 1472 // then it should be almost exactly on time. 1473 const int64_t kSampleStartupThreshold = 1LL << 32; 1474 1475 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1476 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1477 // the next input is close enough to being on time, so concatenate it 1478 // with the last output 1479 timedYieldSamples_l(buffer); 1480 1481 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1482 head.position(), buffer->frameCount); 1483 return NO_ERROR; 1484 } 1485 1486 // Looks like our output is not on time. Reset our on timed status. 1487 // Next time we mix samples from our input queue, then should be within 1488 // the StartupThreshold. 1489 mTimedAudioOutputOnTime = false; 1490 if (sampleDelta > 0) { 1491 // the gap between the current output position and the proper start of 1492 // the next input sample is too big, so fill it with silence 1493 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1494 1495 timedYieldSilence_l(framesUntilNextInput, buffer); 1496 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1497 return NO_ERROR; 1498 } else { 1499 // the next input sample is late 1500 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1501 size_t onTimeSamplePosition = 1502 head.position() + lateFrames * mFrameSize; 1503 1504 if (onTimeSamplePosition > head.buffer()->size()) { 1505 // all the remaining samples in the head are too late, so 1506 // drop it and move on 1507 ALOGV("*** too late: dropped buffer"); 1508 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1509 continue; 1510 } else { 1511 // skip over the late samples 1512 head.setPosition(onTimeSamplePosition); 1513 1514 // yield the available samples 1515 timedYieldSamples_l(buffer); 1516 1517 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1518 return NO_ERROR; 1519 } 1520 } 1521 } 1522} 1523 1524// Yield samples from the timed buffer queue head up to the given output 1525// buffer's capacity. 1526// 1527// Caller must hold mTimedBufferQueueLock 1528void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1529 AudioBufferProvider::Buffer* buffer) { 1530 1531 const TimedBuffer& head = mTimedBufferQueue[0]; 1532 1533 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1534 head.position()); 1535 1536 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1537 mFrameSize); 1538 size_t framesRequested = buffer->frameCount; 1539 buffer->frameCount = min(framesLeftInHead, framesRequested); 1540 1541 mQueueHeadInFlight = true; 1542 mTimedAudioOutputOnTime = true; 1543} 1544 1545// Yield samples of silence up to the given output buffer's capacity 1546// 1547// Caller must hold mTimedBufferQueueLock 1548void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1549 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1550 1551 // lazily allocate a buffer filled with silence 1552 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1553 delete [] mTimedSilenceBuffer; 1554 mTimedSilenceBufferSize = numFrames * mFrameSize; 1555 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1556 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1557 } 1558 1559 buffer->raw = mTimedSilenceBuffer; 1560 size_t framesRequested = buffer->frameCount; 1561 buffer->frameCount = min(numFrames, framesRequested); 1562 1563 mTimedAudioOutputOnTime = false; 1564} 1565 1566// AudioBufferProvider interface 1567void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1568 AudioBufferProvider::Buffer* buffer) { 1569 1570 Mutex::Autolock _l(mTimedBufferQueueLock); 1571 1572 // If the buffer which was just released is part of the buffer at the head 1573 // of the queue, be sure to update the amt of the buffer which has been 1574 // consumed. If the buffer being returned is not part of the head of the 1575 // queue, its either because the buffer is part of the silence buffer, or 1576 // because the head of the timed queue was trimmed after the mixer called 1577 // getNextBuffer but before the mixer called releaseBuffer. 1578 if (buffer->raw == mTimedSilenceBuffer) { 1579 ALOG_ASSERT(!mQueueHeadInFlight, 1580 "Queue head in flight during release of silence buffer!"); 1581 goto done; 1582 } 1583 1584 ALOG_ASSERT(mQueueHeadInFlight, 1585 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1586 " head in flight."); 1587 1588 if (mTimedBufferQueue.size()) { 1589 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1590 1591 void* start = head.buffer()->pointer(); 1592 void* end = reinterpret_cast<void*>( 1593 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1594 + head.buffer()->size()); 1595 1596 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1597 "released buffer not within the head of the timed buffer" 1598 " queue; qHead = [%p, %p], released buffer = %p", 1599 start, end, buffer->raw); 1600 1601 head.setPosition(head.position() + 1602 (buffer->frameCount * mFrameSize)); 1603 mQueueHeadInFlight = false; 1604 1605 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1606 "Bad bookkeeping during releaseBuffer! Should have at" 1607 " least %u queued frames, but we think we have only %u", 1608 buffer->frameCount, mFramesPendingInQueue); 1609 1610 mFramesPendingInQueue -= buffer->frameCount; 1611 1612 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1613 || mTrimQueueHeadOnRelease) { 1614 trimTimedBufferQueueHead_l("releaseBuffer"); 1615 mTrimQueueHeadOnRelease = false; 1616 } 1617 } else { 1618 LOG_ALWAYS_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1619 " buffers in the timed buffer queue"); 1620 } 1621 1622done: 1623 buffer->raw = 0; 1624 buffer->frameCount = 0; 1625} 1626 1627size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1628 Mutex::Autolock _l(mTimedBufferQueueLock); 1629 return mFramesPendingInQueue; 1630} 1631 1632AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1633 : mPTS(0), mPosition(0) {} 1634 1635AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1636 const sp<IMemory>& buffer, int64_t pts) 1637 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1638 1639 1640// ---------------------------------------------------------------------------- 1641 1642AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1643 PlaybackThread *playbackThread, 1644 DuplicatingThread *sourceThread, 1645 uint32_t sampleRate, 1646 audio_format_t format, 1647 audio_channel_mask_t channelMask, 1648 size_t frameCount, 1649 int uid) 1650 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1651 NULL, 0, 0, uid, IAudioFlinger::TRACK_DEFAULT, TYPE_OUTPUT), 1652 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1653{ 1654 1655 if (mCblk != NULL) { 1656 mOutBuffer.frameCount = 0; 1657 playbackThread->mTracks.add(this); 1658 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1659 "frameCount %u, mChannelMask 0x%08x", 1660 mCblk, mBuffer, 1661 frameCount, mChannelMask); 1662 // since client and server are in the same process, 1663 // the buffer has the same virtual address on both sides 1664 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1665 true /*clientInServer*/); 1666 mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); 1667 mClientProxy->setSendLevel(0.0); 1668 mClientProxy->setSampleRate(sampleRate); 1669 } else { 1670 ALOGW("Error creating output track on thread %p", playbackThread); 1671 } 1672} 1673 1674AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1675{ 1676 clearBufferQueue(); 1677 delete mClientProxy; 1678 // superclass destructor will now delete the server proxy and shared memory both refer to 1679} 1680 1681status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1682 int triggerSession) 1683{ 1684 status_t status = Track::start(event, triggerSession); 1685 if (status != NO_ERROR) { 1686 return status; 1687 } 1688 1689 mActive = true; 1690 mRetryCount = 127; 1691 return status; 1692} 1693 1694void AudioFlinger::PlaybackThread::OutputTrack::stop() 1695{ 1696 Track::stop(); 1697 clearBufferQueue(); 1698 mOutBuffer.frameCount = 0; 1699 mActive = false; 1700} 1701 1702bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1703{ 1704 Buffer *pInBuffer; 1705 Buffer inBuffer; 1706 uint32_t channelCount = mChannelCount; 1707 bool outputBufferFull = false; 1708 inBuffer.frameCount = frames; 1709 inBuffer.i16 = data; 1710 1711 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1712 1713 if (!mActive && frames != 0) { 1714 start(); 1715 sp<ThreadBase> thread = mThread.promote(); 1716 if (thread != 0) { 1717 MixerThread *mixerThread = (MixerThread *)thread.get(); 1718 if (mFrameCount > frames) { 1719 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1720 uint32_t startFrames = (mFrameCount - frames); 1721 pInBuffer = new Buffer; 1722 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1723 pInBuffer->frameCount = startFrames; 1724 pInBuffer->i16 = pInBuffer->mBuffer; 1725 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1726 mBufferQueue.add(pInBuffer); 1727 } else { 1728 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1729 } 1730 } 1731 } 1732 } 1733 1734 while (waitTimeLeftMs) { 1735 // First write pending buffers, then new data 1736 if (mBufferQueue.size()) { 1737 pInBuffer = mBufferQueue.itemAt(0); 1738 } else { 1739 pInBuffer = &inBuffer; 1740 } 1741 1742 if (pInBuffer->frameCount == 0) { 1743 break; 1744 } 1745 1746 if (mOutBuffer.frameCount == 0) { 1747 mOutBuffer.frameCount = pInBuffer->frameCount; 1748 nsecs_t startTime = systemTime(); 1749 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1750 if (status != NO_ERROR) { 1751 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1752 mThread.unsafe_get(), status); 1753 outputBufferFull = true; 1754 break; 1755 } 1756 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1757 if (waitTimeLeftMs >= waitTimeMs) { 1758 waitTimeLeftMs -= waitTimeMs; 1759 } else { 1760 waitTimeLeftMs = 0; 1761 } 1762 } 1763 1764 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1765 pInBuffer->frameCount; 1766 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1767 Proxy::Buffer buf; 1768 buf.mFrameCount = outFrames; 1769 buf.mRaw = NULL; 1770 mClientProxy->releaseBuffer(&buf); 1771 pInBuffer->frameCount -= outFrames; 1772 pInBuffer->i16 += outFrames * channelCount; 1773 mOutBuffer.frameCount -= outFrames; 1774 mOutBuffer.i16 += outFrames * channelCount; 1775 1776 if (pInBuffer->frameCount == 0) { 1777 if (mBufferQueue.size()) { 1778 mBufferQueue.removeAt(0); 1779 delete [] pInBuffer->mBuffer; 1780 delete pInBuffer; 1781 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1782 mThread.unsafe_get(), mBufferQueue.size()); 1783 } else { 1784 break; 1785 } 1786 } 1787 } 1788 1789 // If we could not write all frames, allocate a buffer and queue it for next time. 1790 if (inBuffer.frameCount) { 1791 sp<ThreadBase> thread = mThread.promote(); 1792 if (thread != 0 && !thread->standby()) { 1793 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1794 pInBuffer = new Buffer; 1795 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1796 pInBuffer->frameCount = inBuffer.frameCount; 1797 pInBuffer->i16 = pInBuffer->mBuffer; 1798 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1799 sizeof(int16_t)); 1800 mBufferQueue.add(pInBuffer); 1801 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1802 mThread.unsafe_get(), mBufferQueue.size()); 1803 } else { 1804 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1805 mThread.unsafe_get(), this); 1806 } 1807 } 1808 } 1809 1810 // Calling write() with a 0 length buffer, means that no more data will be written: 1811 // If no more buffers are pending, fill output track buffer to make sure it is started 1812 // by output mixer. 1813 if (frames == 0 && mBufferQueue.size() == 0) { 1814 // FIXME borken, replace by getting framesReady() from proxy 1815 size_t user = 0; // was mCblk->user 1816 if (user < mFrameCount) { 1817 frames = mFrameCount - user; 1818 pInBuffer = new Buffer; 1819 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1820 pInBuffer->frameCount = frames; 1821 pInBuffer->i16 = pInBuffer->mBuffer; 1822 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1823 mBufferQueue.add(pInBuffer); 1824 } else if (mActive) { 1825 stop(); 1826 } 1827 } 1828 1829 return outputBufferFull; 1830} 1831 1832status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1833 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1834{ 1835 ClientProxy::Buffer buf; 1836 buf.mFrameCount = buffer->frameCount; 1837 struct timespec timeout; 1838 timeout.tv_sec = waitTimeMs / 1000; 1839 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1840 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1841 buffer->frameCount = buf.mFrameCount; 1842 buffer->raw = buf.mRaw; 1843 return status; 1844} 1845 1846void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1847{ 1848 size_t size = mBufferQueue.size(); 1849 1850 for (size_t i = 0; i < size; i++) { 1851 Buffer *pBuffer = mBufferQueue.itemAt(i); 1852 delete [] pBuffer->mBuffer; 1853 delete pBuffer; 1854 } 1855 mBufferQueue.clear(); 1856} 1857 1858 1859AudioFlinger::PlaybackThread::PatchTrack::PatchTrack(PlaybackThread *playbackThread, 1860 uint32_t sampleRate, 1861 audio_channel_mask_t channelMask, 1862 audio_format_t format, 1863 size_t frameCount, 1864 void *buffer, 1865 IAudioFlinger::track_flags_t flags) 1866 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1867 buffer, 0, 0, getuid(), flags, TYPE_PATCH), 1868 mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true)) 1869{ 1870 uint64_t mixBufferNs = ((uint64_t)2 * playbackThread->frameCount() * 1000000000) / 1871 playbackThread->sampleRate(); 1872 mPeerTimeout.tv_sec = mixBufferNs / 1000000000; 1873 mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000); 1874 1875 ALOGV("PatchTrack %p sampleRate %d mPeerTimeout %d.%03d sec", 1876 this, sampleRate, 1877 (int)mPeerTimeout.tv_sec, 1878 (int)(mPeerTimeout.tv_nsec / 1000000)); 1879} 1880 1881AudioFlinger::PlaybackThread::PatchTrack::~PatchTrack() 1882{ 1883} 1884 1885// AudioBufferProvider interface 1886status_t AudioFlinger::PlaybackThread::PatchTrack::getNextBuffer( 1887 AudioBufferProvider::Buffer* buffer, int64_t pts) 1888{ 1889 ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::getNextBuffer() called without peer proxy"); 1890 Proxy::Buffer buf; 1891 buf.mFrameCount = buffer->frameCount; 1892 status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout); 1893 ALOGV_IF(status != NO_ERROR, "PatchTrack() %p getNextBuffer status %d", this, status); 1894 buffer->frameCount = buf.mFrameCount; 1895 if (buf.mFrameCount == 0) { 1896 return WOULD_BLOCK; 1897 } 1898 status = Track::getNextBuffer(buffer, pts); 1899 return status; 1900} 1901 1902void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(AudioBufferProvider::Buffer* buffer) 1903{ 1904 ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::releaseBuffer() called without peer proxy"); 1905 Proxy::Buffer buf; 1906 buf.mFrameCount = buffer->frameCount; 1907 buf.mRaw = buffer->raw; 1908 mPeerProxy->releaseBuffer(&buf); 1909 TrackBase::releaseBuffer(buffer); 1910} 1911 1912status_t AudioFlinger::PlaybackThread::PatchTrack::obtainBuffer(Proxy::Buffer* buffer, 1913 const struct timespec *timeOut) 1914{ 1915 return mProxy->obtainBuffer(buffer, timeOut); 1916} 1917 1918void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(Proxy::Buffer* buffer) 1919{ 1920 mProxy->releaseBuffer(buffer); 1921 if (android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags) & CBLK_DISABLED) { 1922 ALOGW("PatchTrack::releaseBuffer() disabled due to previous underrun, restarting"); 1923 start(); 1924 } 1925 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1926} 1927 1928// ---------------------------------------------------------------------------- 1929// Record 1930// ---------------------------------------------------------------------------- 1931 1932AudioFlinger::RecordHandle::RecordHandle( 1933 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1934 : BnAudioRecord(), 1935 mRecordTrack(recordTrack) 1936{ 1937} 1938 1939AudioFlinger::RecordHandle::~RecordHandle() { 1940 stop_nonvirtual(); 1941 mRecordTrack->destroy(); 1942} 1943 1944status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1945 int triggerSession) { 1946 ALOGV("RecordHandle::start()"); 1947 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1948} 1949 1950void AudioFlinger::RecordHandle::stop() { 1951 stop_nonvirtual(); 1952} 1953 1954void AudioFlinger::RecordHandle::stop_nonvirtual() { 1955 ALOGV("RecordHandle::stop()"); 1956 mRecordTrack->stop(); 1957} 1958 1959status_t AudioFlinger::RecordHandle::onTransact( 1960 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1961{ 1962 return BnAudioRecord::onTransact(code, data, reply, flags); 1963} 1964 1965// ---------------------------------------------------------------------------- 1966 1967// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 1968AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1969 RecordThread *thread, 1970 const sp<Client>& client, 1971 uint32_t sampleRate, 1972 audio_format_t format, 1973 audio_channel_mask_t channelMask, 1974 size_t frameCount, 1975 void *buffer, 1976 int sessionId, 1977 int uid, 1978 IAudioFlinger::track_flags_t flags, 1979 track_type type) 1980 : TrackBase(thread, client, sampleRate, format, 1981 channelMask, frameCount, buffer, sessionId, uid, 1982 flags, false /*isOut*/, 1983 (type == TYPE_DEFAULT) ? 1984 ((flags & IAudioFlinger::TRACK_FAST) ? ALLOC_PIPE : ALLOC_CBLK) : 1985 ((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE), 1986 type), 1987 mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0), 1988 // See real initialization of mRsmpInFront at RecordThread::start() 1989 mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL) 1990{ 1991 if (mCblk == NULL) { 1992 return; 1993 } 1994 1995 mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, 1996 mFrameSize, !isExternalTrack()); 1997 1998 uint32_t channelCount = audio_channel_count_from_in_mask(channelMask); 1999 // FIXME I don't understand either of the channel count checks 2000 if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 && 2001 channelCount <= FCC_2) { 2002 // sink SR 2003 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT, 2004 thread->mChannelCount, sampleRate); 2005 // source SR 2006 mResampler->setSampleRate(thread->mSampleRate); 2007 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 2008 mResamplerBufferProvider = new ResamplerBufferProvider(this); 2009 } 2010 2011 if (flags & IAudioFlinger::TRACK_FAST) { 2012 ALOG_ASSERT(thread->mFastTrackAvail); 2013 thread->mFastTrackAvail = false; 2014 } 2015} 2016 2017AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 2018{ 2019 ALOGV("%s", __func__); 2020 delete mResampler; 2021 delete[] mRsmpOutBuffer; 2022 delete mResamplerBufferProvider; 2023} 2024 2025// AudioBufferProvider interface 2026status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 2027 int64_t pts __unused) 2028{ 2029 ServerProxy::Buffer buf; 2030 buf.mFrameCount = buffer->frameCount; 2031 status_t status = mServerProxy->obtainBuffer(&buf); 2032 buffer->frameCount = buf.mFrameCount; 2033 buffer->raw = buf.mRaw; 2034 if (buf.mFrameCount == 0) { 2035 // FIXME also wake futex so that overrun is noticed more quickly 2036 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 2037 } 2038 return status; 2039} 2040 2041status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 2042 int triggerSession) 2043{ 2044 sp<ThreadBase> thread = mThread.promote(); 2045 if (thread != 0) { 2046 RecordThread *recordThread = (RecordThread *)thread.get(); 2047 return recordThread->start(this, event, triggerSession); 2048 } else { 2049 return BAD_VALUE; 2050 } 2051} 2052 2053void AudioFlinger::RecordThread::RecordTrack::stop() 2054{ 2055 sp<ThreadBase> thread = mThread.promote(); 2056 if (thread != 0) { 2057 RecordThread *recordThread = (RecordThread *)thread.get(); 2058 if (recordThread->stop(this) && isExternalTrack()) { 2059 AudioSystem::stopInput(mThreadIoHandle, (audio_session_t)mSessionId); 2060 } 2061 } 2062} 2063 2064void AudioFlinger::RecordThread::RecordTrack::destroy() 2065{ 2066 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 2067 sp<RecordTrack> keep(this); 2068 { 2069 if (isExternalTrack()) { 2070 if (mState == ACTIVE || mState == RESUMING) { 2071 AudioSystem::stopInput(mThreadIoHandle, (audio_session_t)mSessionId); 2072 } 2073 AudioSystem::releaseInput(mThreadIoHandle, (audio_session_t)mSessionId); 2074 } 2075 sp<ThreadBase> thread = mThread.promote(); 2076 if (thread != 0) { 2077 Mutex::Autolock _l(thread->mLock); 2078 RecordThread *recordThread = (RecordThread *) thread.get(); 2079 recordThread->destroyTrack_l(this); 2080 } 2081 } 2082} 2083 2084void AudioFlinger::RecordThread::RecordTrack::invalidate() 2085{ 2086 // FIXME should use proxy, and needs work 2087 audio_track_cblk_t* cblk = mCblk; 2088 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 2089 android_atomic_release_store(0x40000000, &cblk->mFutex); 2090 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 2091 (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); 2092} 2093 2094 2095/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 2096{ 2097 result.append(" Active Client Fmt Chn mask Session S Server fCount SRate\n"); 2098} 2099 2100void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active) 2101{ 2102 snprintf(buffer, size, " %6s %6u %3u %08X %7u %1d %08X %6zu %5u\n", 2103 active ? "yes" : "no", 2104 (mClient == 0) ? getpid_cached : mClient->pid(), 2105 mFormat, 2106 mChannelMask, 2107 mSessionId, 2108 mState, 2109 mCblk->mServer, 2110 mFrameCount, 2111 mSampleRate); 2112 2113} 2114 2115void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event) 2116{ 2117 if (event == mSyncStartEvent) { 2118 ssize_t framesToDrop = 0; 2119 sp<ThreadBase> threadBase = mThread.promote(); 2120 if (threadBase != 0) { 2121 // TODO: use actual buffer filling status instead of 2 buffers when info is available 2122 // from audio HAL 2123 framesToDrop = threadBase->mFrameCount * 2; 2124 } 2125 mFramesToDrop = framesToDrop; 2126 } 2127} 2128 2129void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent() 2130{ 2131 if (mSyncStartEvent != 0) { 2132 mSyncStartEvent->cancel(); 2133 mSyncStartEvent.clear(); 2134 } 2135 mFramesToDrop = 0; 2136} 2137 2138 2139AudioFlinger::RecordThread::PatchRecord::PatchRecord(RecordThread *recordThread, 2140 uint32_t sampleRate, 2141 audio_channel_mask_t channelMask, 2142 audio_format_t format, 2143 size_t frameCount, 2144 void *buffer, 2145 IAudioFlinger::track_flags_t flags) 2146 : RecordTrack(recordThread, NULL, sampleRate, format, channelMask, frameCount, 2147 buffer, 0, getuid(), flags, TYPE_PATCH), 2148 mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true)) 2149{ 2150 uint64_t mixBufferNs = ((uint64_t)2 * recordThread->frameCount() * 1000000000) / 2151 recordThread->sampleRate(); 2152 mPeerTimeout.tv_sec = mixBufferNs / 1000000000; 2153 mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000); 2154 2155 ALOGV("PatchRecord %p sampleRate %d mPeerTimeout %d.%03d sec", 2156 this, sampleRate, 2157 (int)mPeerTimeout.tv_sec, 2158 (int)(mPeerTimeout.tv_nsec / 1000000)); 2159} 2160 2161AudioFlinger::RecordThread::PatchRecord::~PatchRecord() 2162{ 2163} 2164 2165// AudioBufferProvider interface 2166status_t AudioFlinger::RecordThread::PatchRecord::getNextBuffer( 2167 AudioBufferProvider::Buffer* buffer, int64_t pts) 2168{ 2169 ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::getNextBuffer() called without peer proxy"); 2170 Proxy::Buffer buf; 2171 buf.mFrameCount = buffer->frameCount; 2172 status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout); 2173 ALOGV_IF(status != NO_ERROR, 2174 "PatchRecord() %p mPeerProxy->obtainBuffer status %d", this, status); 2175 buffer->frameCount = buf.mFrameCount; 2176 if (buf.mFrameCount == 0) { 2177 return WOULD_BLOCK; 2178 } 2179 status = RecordTrack::getNextBuffer(buffer, pts); 2180 return status; 2181} 2182 2183void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(AudioBufferProvider::Buffer* buffer) 2184{ 2185 ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::releaseBuffer() called without peer proxy"); 2186 Proxy::Buffer buf; 2187 buf.mFrameCount = buffer->frameCount; 2188 buf.mRaw = buffer->raw; 2189 mPeerProxy->releaseBuffer(&buf); 2190 TrackBase::releaseBuffer(buffer); 2191} 2192 2193status_t AudioFlinger::RecordThread::PatchRecord::obtainBuffer(Proxy::Buffer* buffer, 2194 const struct timespec *timeOut) 2195{ 2196 return mProxy->obtainBuffer(buffer, timeOut); 2197} 2198 2199void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(Proxy::Buffer* buffer) 2200{ 2201 mProxy->releaseBuffer(buffer); 2202} 2203 2204}; // namespace android 2205