Tracks.cpp revision accc147666bfd37fc8b4ef745f18a8c751555ec2
15d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)/* 25d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)** 35d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)** Copyright 2012, The Android Open Source Project 45d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)** 55d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)** Licensed under the Apache License, Version 2.0 (the "License"); 65d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)** you may not use this file except in compliance with the License. 75d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)** You may obtain a copy of the License at 85d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)** 95d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)** http://www.apache.org/licenses/LICENSE-2.0 105d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)** 115d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)** Unless required by applicable law or agreed to in writing, software 125d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)** distributed under the License is distributed on an "AS IS" BASIS, 135d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 145d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)** See the License for the specific language governing permissions and 155d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)** limitations under the License. 165d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)*/ 175d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles) 185d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles) 195d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#define LOG_TAG "AudioFlinger" 205d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)//#define LOG_NDEBUG 0 215d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles) 225d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#include "Configuration.h" 235d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#include <math.h> 245d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#include <utils/Log.h> 255d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles) 265d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#include <private/media/AudioTrackShared.h> 275d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles) 285d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#include <common_time/cc_helper.h> 295d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#include <common_time/local_clock.h> 305d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles) 315d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#include "AudioMixer.h" 325d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#include "AudioFlinger.h" 335d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#include "ServiceUtilities.h" 345d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles) 355d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#include <media/nbaio/Pipe.h> 365d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#include <media/nbaio/PipeReader.h> 375d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles) 385d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)// ---------------------------------------------------------------------------- 395d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles) 405d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)// Note: the following macro is used for extremely verbose logging message. In 415d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 425d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)// 0; but one side effect of this is to turn all LOGV's as well. Some messages 435d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)// are so verbose that we want to suppress them even when we have ALOG_ASSERT 445d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)// turned on. Do not uncomment the #def below unless you really know what you 455d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)// are doing and want to see all of the extremely verbose messages. 465d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)//#define VERY_VERY_VERBOSE_LOGGING 475d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#ifdef VERY_VERY_VERBOSE_LOGGING 485d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#define ALOGVV ALOGV 495d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#else 505d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#define ALOGVV(a...) do { } while(0) 515d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)#endif 525d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles) 535d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)namespace android { 545d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles) 555d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)// ---------------------------------------------------------------------------- 565d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)// TrackBase 575d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)// ---------------------------------------------------------------------------- 585d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles) 595d1f7b1de12d16ceb2c938c56701a3e8bfa558f7Torne (Richard Coles)static volatile int32_t nextTrackId = 55; 60 61// TrackBase constructor must be called with AudioFlinger::mLock held 62AudioFlinger::ThreadBase::TrackBase::TrackBase( 63 ThreadBase *thread, 64 const sp<Client>& client, 65 uint32_t sampleRate, 66 audio_format_t format, 67 audio_channel_mask_t channelMask, 68 size_t frameCount, 69 const sp<IMemory>& sharedBuffer, 70 int sessionId, 71 bool isOut) 72 : RefBase(), 73 mThread(thread), 74 mClient(client), 75 mCblk(NULL), 76 // mBuffer 77 mState(IDLE), 78 mSampleRate(sampleRate), 79 mFormat(format), 80 mChannelMask(channelMask), 81 mChannelCount(popcount(channelMask)), 82 mFrameSize(audio_is_linear_pcm(format) ? 83 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 84 mFrameCount(frameCount), 85 mSessionId(sessionId), 86 mIsOut(isOut), 87 mServerProxy(NULL), 88 mId(android_atomic_inc(&nextTrackId)), 89 mTerminated(false) 90{ 91 // client == 0 implies sharedBuffer == 0 92 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 93 94 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 95 sharedBuffer->size()); 96 97 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 98 size_t size = sizeof(audio_track_cblk_t); 99 size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize; 100 if (sharedBuffer == 0) { 101 size += bufferSize; 102 } 103 104 if (client != 0) { 105 mCblkMemory = client->heap()->allocate(size); 106 if (mCblkMemory != 0) { 107 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 108 // can't assume mCblk != NULL 109 } else { 110 ALOGE("not enough memory for AudioTrack size=%u", size); 111 client->heap()->dump("AudioTrack"); 112 return; 113 } 114 } else { 115 // this syntax avoids calling the audio_track_cblk_t constructor twice 116 mCblk = (audio_track_cblk_t *) new uint8_t[size]; 117 // assume mCblk != NULL 118 } 119 120 // construct the shared structure in-place. 121 if (mCblk != NULL) { 122 new(mCblk) audio_track_cblk_t(); 123 // clear all buffers 124 mCblk->frameCount_ = frameCount; 125 if (sharedBuffer == 0) { 126 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 127 memset(mBuffer, 0, bufferSize); 128 } else { 129 mBuffer = sharedBuffer->pointer(); 130#if 0 131 mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic 132#endif 133 } 134 135#ifdef TEE_SINK 136 if (mTeeSinkTrackEnabled) { 137 NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount); 138 if (pipeFormat != Format_Invalid) { 139 Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat); 140 size_t numCounterOffers = 0; 141 const NBAIO_Format offers[1] = {pipeFormat}; 142 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 143 ALOG_ASSERT(index == 0); 144 PipeReader *pipeReader = new PipeReader(*pipe); 145 numCounterOffers = 0; 146 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 147 ALOG_ASSERT(index == 0); 148 mTeeSink = pipe; 149 mTeeSource = pipeReader; 150 } 151 } 152#endif 153 154 } 155} 156 157AudioFlinger::ThreadBase::TrackBase::~TrackBase() 158{ 159#ifdef TEE_SINK 160 dumpTee(-1, mTeeSource, mId); 161#endif 162 // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference 163 delete mServerProxy; 164 if (mCblk != NULL) { 165 if (mClient == 0) { 166 delete mCblk; 167 } else { 168 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 169 } 170 } 171 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 172 if (mClient != 0) { 173 // Client destructor must run with AudioFlinger mutex locked 174 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 175 // If the client's reference count drops to zero, the associated destructor 176 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 177 // relying on the automatic clear() at end of scope. 178 mClient.clear(); 179 } 180} 181 182// AudioBufferProvider interface 183// getNextBuffer() = 0; 184// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 185void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 186{ 187#ifdef TEE_SINK 188 if (mTeeSink != 0) { 189 (void) mTeeSink->write(buffer->raw, buffer->frameCount); 190 } 191#endif 192 193 ServerProxy::Buffer buf; 194 buf.mFrameCount = buffer->frameCount; 195 buf.mRaw = buffer->raw; 196 buffer->frameCount = 0; 197 buffer->raw = NULL; 198 mServerProxy->releaseBuffer(&buf); 199} 200 201status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 202{ 203 mSyncEvents.add(event); 204 return NO_ERROR; 205} 206 207// ---------------------------------------------------------------------------- 208// Playback 209// ---------------------------------------------------------------------------- 210 211AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 212 : BnAudioTrack(), 213 mTrack(track) 214{ 215} 216 217AudioFlinger::TrackHandle::~TrackHandle() { 218 // just stop the track on deletion, associated resources 219 // will be freed from the main thread once all pending buffers have 220 // been played. Unless it's not in the active track list, in which 221 // case we free everything now... 222 mTrack->destroy(); 223} 224 225sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 226 return mTrack->getCblk(); 227} 228 229status_t AudioFlinger::TrackHandle::start() { 230 return mTrack->start(); 231} 232 233void AudioFlinger::TrackHandle::stop() { 234 mTrack->stop(); 235} 236 237void AudioFlinger::TrackHandle::flush() { 238 mTrack->flush(); 239} 240 241void AudioFlinger::TrackHandle::pause() { 242 mTrack->pause(); 243} 244 245status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 246{ 247 return mTrack->attachAuxEffect(EffectId); 248} 249 250status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 251 sp<IMemory>* buffer) { 252 if (!mTrack->isTimedTrack()) 253 return INVALID_OPERATION; 254 255 PlaybackThread::TimedTrack* tt = 256 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 257 return tt->allocateTimedBuffer(size, buffer); 258} 259 260status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 261 int64_t pts) { 262 if (!mTrack->isTimedTrack()) 263 return INVALID_OPERATION; 264 265 PlaybackThread::TimedTrack* tt = 266 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 267 return tt->queueTimedBuffer(buffer, pts); 268} 269 270status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 271 const LinearTransform& xform, int target) { 272 273 if (!mTrack->isTimedTrack()) 274 return INVALID_OPERATION; 275 276 PlaybackThread::TimedTrack* tt = 277 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 278 return tt->setMediaTimeTransform( 279 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 280} 281 282status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) { 283 return mTrack->setParameters(keyValuePairs); 284} 285 286status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp) 287{ 288 return mTrack->getTimestamp(timestamp); 289} 290 291status_t AudioFlinger::TrackHandle::onTransact( 292 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 293{ 294 return BnAudioTrack::onTransact(code, data, reply, flags); 295} 296 297// ---------------------------------------------------------------------------- 298 299// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 300AudioFlinger::PlaybackThread::Track::Track( 301 PlaybackThread *thread, 302 const sp<Client>& client, 303 audio_stream_type_t streamType, 304 uint32_t sampleRate, 305 audio_format_t format, 306 audio_channel_mask_t channelMask, 307 size_t frameCount, 308 const sp<IMemory>& sharedBuffer, 309 int sessionId, 310 IAudioFlinger::track_flags_t flags) 311 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 312 sessionId, true /*isOut*/), 313 mFillingUpStatus(FS_INVALID), 314 // mRetryCount initialized later when needed 315 mSharedBuffer(sharedBuffer), 316 mStreamType(streamType), 317 mName(-1), // see note below 318 mMainBuffer(thread->mixBuffer()), 319 mAuxBuffer(NULL), 320 mAuxEffectId(0), mHasVolumeController(false), 321 mPresentationCompleteFrames(0), 322 mFlags(flags), 323 mFastIndex(-1), 324 mCachedVolume(1.0), 325 mIsInvalid(false), 326 mAudioTrackServerProxy(NULL), 327 mResumeToStopping(false) 328{ 329 if (mCblk != NULL) { 330 if (sharedBuffer == 0) { 331 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 332 mFrameSize); 333 } else { 334 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, 335 mFrameSize); 336 } 337 mServerProxy = mAudioTrackServerProxy; 338 // to avoid leaking a track name, do not allocate one unless there is an mCblk 339 mName = thread->getTrackName_l(channelMask, sessionId); 340 if (mName < 0) { 341 ALOGE("no more track names available"); 342 return; 343 } 344 // only allocate a fast track index if we were able to allocate a normal track name 345 if (flags & IAudioFlinger::TRACK_FAST) { 346 mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); 347 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 348 int i = __builtin_ctz(thread->mFastTrackAvailMask); 349 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 350 // FIXME This is too eager. We allocate a fast track index before the 351 // fast track becomes active. Since fast tracks are a scarce resource, 352 // this means we are potentially denying other more important fast tracks from 353 // being created. It would be better to allocate the index dynamically. 354 mFastIndex = i; 355 // Read the initial underruns because this field is never cleared by the fast mixer 356 mObservedUnderruns = thread->getFastTrackUnderruns(i); 357 thread->mFastTrackAvailMask &= ~(1 << i); 358 } 359 } 360 ALOGV("Track constructor name %d, calling pid %d", mName, 361 IPCThreadState::self()->getCallingPid()); 362} 363 364AudioFlinger::PlaybackThread::Track::~Track() 365{ 366 ALOGV("PlaybackThread::Track destructor"); 367 368 // The destructor would clear mSharedBuffer, 369 // but it will not push the decremented reference count, 370 // leaving the client's IMemory dangling indefinitely. 371 // This prevents that leak. 372 if (mSharedBuffer != 0) { 373 mSharedBuffer.clear(); 374 // flush the binder command buffer 375 IPCThreadState::self()->flushCommands(); 376 } 377} 378 379void AudioFlinger::PlaybackThread::Track::destroy() 380{ 381 // NOTE: destroyTrack_l() can remove a strong reference to this Track 382 // by removing it from mTracks vector, so there is a risk that this Tracks's 383 // destructor is called. As the destructor needs to lock mLock, 384 // we must acquire a strong reference on this Track before locking mLock 385 // here so that the destructor is called only when exiting this function. 386 // On the other hand, as long as Track::destroy() is only called by 387 // TrackHandle destructor, the TrackHandle still holds a strong ref on 388 // this Track with its member mTrack. 389 sp<Track> keep(this); 390 { // scope for mLock 391 sp<ThreadBase> thread = mThread.promote(); 392 if (thread != 0) { 393 Mutex::Autolock _l(thread->mLock); 394 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 395 bool wasActive = playbackThread->destroyTrack_l(this); 396 if (!isOutputTrack() && !wasActive) { 397 AudioSystem::releaseOutput(thread->id()); 398 } 399 } 400 } 401} 402 403/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 404{ 405 result.append(" Name Client Type Fmt Chn mask Session fCount S F SRate " 406 "L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n"); 407} 408 409void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 410{ 411 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 412 if (isFastTrack()) { 413 sprintf(buffer, " F %2d", mFastIndex); 414 } else { 415 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 416 } 417 track_state state = mState; 418 char stateChar; 419 if (isTerminated()) { 420 stateChar = 'T'; 421 } else { 422 switch (state) { 423 case IDLE: 424 stateChar = 'I'; 425 break; 426 case STOPPING_1: 427 stateChar = 's'; 428 break; 429 case STOPPING_2: 430 stateChar = '5'; 431 break; 432 case STOPPED: 433 stateChar = 'S'; 434 break; 435 case RESUMING: 436 stateChar = 'R'; 437 break; 438 case ACTIVE: 439 stateChar = 'A'; 440 break; 441 case PAUSING: 442 stateChar = 'p'; 443 break; 444 case PAUSED: 445 stateChar = 'P'; 446 break; 447 case FLUSHED: 448 stateChar = 'F'; 449 break; 450 default: 451 stateChar = '?'; 452 break; 453 } 454 } 455 char nowInUnderrun; 456 switch (mObservedUnderruns.mBitFields.mMostRecent) { 457 case UNDERRUN_FULL: 458 nowInUnderrun = ' '; 459 break; 460 case UNDERRUN_PARTIAL: 461 nowInUnderrun = '<'; 462 break; 463 case UNDERRUN_EMPTY: 464 nowInUnderrun = '*'; 465 break; 466 default: 467 nowInUnderrun = '?'; 468 break; 469 } 470 snprintf(&buffer[7], size-7, " %6u %4u %08X %08X %7u %6u %1c %1d %5u %5.2g %5.2g " 471 "%08X %08X %08X 0x%03X %9u%c\n", 472 (mClient == 0) ? getpid_cached : mClient->pid(), 473 mStreamType, 474 mFormat, 475 mChannelMask, 476 mSessionId, 477 mFrameCount, 478 stateChar, 479 mFillingUpStatus, 480 mAudioTrackServerProxy->getSampleRate(), 481 20.0 * log10((vlr & 0xFFFF) / 4096.0), 482 20.0 * log10((vlr >> 16) / 4096.0), 483 mCblk->mServer, 484 (int)mMainBuffer, 485 (int)mAuxBuffer, 486 mCblk->mFlags, 487 mAudioTrackServerProxy->getUnderrunFrames(), 488 nowInUnderrun); 489} 490 491uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { 492 return mAudioTrackServerProxy->getSampleRate(); 493} 494 495// AudioBufferProvider interface 496status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 497 AudioBufferProvider::Buffer* buffer, int64_t pts) 498{ 499 ServerProxy::Buffer buf; 500 size_t desiredFrames = buffer->frameCount; 501 buf.mFrameCount = desiredFrames; 502 status_t status = mServerProxy->obtainBuffer(&buf); 503 buffer->frameCount = buf.mFrameCount; 504 buffer->raw = buf.mRaw; 505 if (buf.mFrameCount == 0) { 506 mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 507 } 508 return status; 509} 510 511// releaseBuffer() is not overridden 512 513// ExtendedAudioBufferProvider interface 514 515// Note that framesReady() takes a mutex on the control block using tryLock(). 516// This could result in priority inversion if framesReady() is called by the normal mixer, 517// as the normal mixer thread runs at lower 518// priority than the client's callback thread: there is a short window within framesReady() 519// during which the normal mixer could be preempted, and the client callback would block. 520// Another problem can occur if framesReady() is called by the fast mixer: 521// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 522// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 523size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 524 return mAudioTrackServerProxy->framesReady(); 525} 526 527size_t AudioFlinger::PlaybackThread::Track::framesReleased() const 528{ 529 return mAudioTrackServerProxy->framesReleased(); 530} 531 532// Don't call for fast tracks; the framesReady() could result in priority inversion 533bool AudioFlinger::PlaybackThread::Track::isReady() const { 534 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 535 return true; 536 } 537 538 if (framesReady() >= mFrameCount || 539 (mCblk->mFlags & CBLK_FORCEREADY)) { 540 mFillingUpStatus = FS_FILLED; 541 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 542 return true; 543 } 544 return false; 545} 546 547status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 548 int triggerSession) 549{ 550 status_t status = NO_ERROR; 551 ALOGV("start(%d), calling pid %d session %d", 552 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 553 554 sp<ThreadBase> thread = mThread.promote(); 555 if (thread != 0) { 556 if (isOffloaded()) { 557 Mutex::Autolock _laf(thread->mAudioFlinger->mLock); 558 Mutex::Autolock _lth(thread->mLock); 559 sp<EffectChain> ec = thread->getEffectChain_l(mSessionId); 560 if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() || 561 (ec != 0 && ec->isNonOffloadableEnabled())) { 562 invalidate(); 563 return PERMISSION_DENIED; 564 } 565 } 566 Mutex::Autolock _lth(thread->mLock); 567 track_state state = mState; 568 // here the track could be either new, or restarted 569 // in both cases "unstop" the track 570 571 if (state == PAUSED) { 572 if (mResumeToStopping) { 573 // happened we need to resume to STOPPING_1 574 mState = TrackBase::STOPPING_1; 575 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 576 } else { 577 mState = TrackBase::RESUMING; 578 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 579 } 580 } else { 581 mState = TrackBase::ACTIVE; 582 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 583 } 584 585 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 586 status = playbackThread->addTrack_l(this); 587 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 588 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 589 // restore previous state if start was rejected by policy manager 590 if (status == PERMISSION_DENIED) { 591 mState = state; 592 } 593 } 594 // track was already in the active list, not a problem 595 if (status == ALREADY_EXISTS) { 596 status = NO_ERROR; 597 } 598 } else { 599 status = BAD_VALUE; 600 } 601 return status; 602} 603 604void AudioFlinger::PlaybackThread::Track::stop() 605{ 606 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 607 sp<ThreadBase> thread = mThread.promote(); 608 if (thread != 0) { 609 Mutex::Autolock _l(thread->mLock); 610 track_state state = mState; 611 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 612 // If the track is not active (PAUSED and buffers full), flush buffers 613 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 614 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 615 reset(); 616 mState = STOPPED; 617 } else if (!isFastTrack() && !isOffloaded()) { 618 mState = STOPPED; 619 } else { 620 // For fast tracks prepareTracks_l() will set state to STOPPING_2 621 // presentation is complete 622 // For an offloaded track this starts a drain and state will 623 // move to STOPPING_2 when drain completes and then STOPPED 624 mState = STOPPING_1; 625 } 626 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 627 playbackThread); 628 } 629 } 630} 631 632void AudioFlinger::PlaybackThread::Track::pause() 633{ 634 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 635 sp<ThreadBase> thread = mThread.promote(); 636 if (thread != 0) { 637 Mutex::Autolock _l(thread->mLock); 638 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 639 switch (mState) { 640 case STOPPING_1: 641 case STOPPING_2: 642 if (!isOffloaded()) { 643 /* nothing to do if track is not offloaded */ 644 break; 645 } 646 647 // Offloaded track was draining, we need to carry on draining when resumed 648 mResumeToStopping = true; 649 // fall through... 650 case ACTIVE: 651 case RESUMING: 652 mState = PAUSING; 653 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 654 playbackThread->signal_l(); 655 break; 656 657 default: 658 break; 659 } 660 } 661} 662 663void AudioFlinger::PlaybackThread::Track::flush() 664{ 665 ALOGV("flush(%d)", mName); 666 sp<ThreadBase> thread = mThread.promote(); 667 if (thread != 0) { 668 Mutex::Autolock _l(thread->mLock); 669 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 670 671 if (isOffloaded()) { 672 // If offloaded we allow flush during any state except terminated 673 // and keep the track active to avoid problems if user is seeking 674 // rapidly and underlying hardware has a significant delay handling 675 // a pause 676 if (isTerminated()) { 677 return; 678 } 679 680 ALOGV("flush: offload flush"); 681 reset(); 682 683 if (mState == STOPPING_1 || mState == STOPPING_2) { 684 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 685 mState = ACTIVE; 686 } 687 688 if (mState == ACTIVE) { 689 ALOGV("flush called in active state, resetting buffer time out retry count"); 690 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 691 } 692 693 mResumeToStopping = false; 694 } else { 695 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 696 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 697 return; 698 } 699 // No point remaining in PAUSED state after a flush => go to 700 // FLUSHED state 701 mState = FLUSHED; 702 // do not reset the track if it is still in the process of being stopped or paused. 703 // this will be done by prepareTracks_l() when the track is stopped. 704 // prepareTracks_l() will see mState == FLUSHED, then 705 // remove from active track list, reset(), and trigger presentation complete 706 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 707 reset(); 708 } 709 } 710 // Prevent flush being lost if the track is flushed and then resumed 711 // before mixer thread can run. This is important when offloading 712 // because the hardware buffer could hold a large amount of audio 713 playbackThread->flushOutput_l(); 714 playbackThread->signal_l(); 715 } 716} 717 718void AudioFlinger::PlaybackThread::Track::reset() 719{ 720 // Do not reset twice to avoid discarding data written just after a flush and before 721 // the audioflinger thread detects the track is stopped. 722 if (!mResetDone) { 723 // Force underrun condition to avoid false underrun callback until first data is 724 // written to buffer 725 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 726 mFillingUpStatus = FS_FILLING; 727 mResetDone = true; 728 if (mState == FLUSHED) { 729 mState = IDLE; 730 } 731 } 732} 733 734status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 735{ 736 sp<ThreadBase> thread = mThread.promote(); 737 if (thread == 0) { 738 ALOGE("thread is dead"); 739 return FAILED_TRANSACTION; 740 } else if ((thread->type() == ThreadBase::DIRECT) || 741 (thread->type() == ThreadBase::OFFLOAD)) { 742 return thread->setParameters(keyValuePairs); 743 } else { 744 return PERMISSION_DENIED; 745 } 746} 747 748status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp) 749{ 750 // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant 751 if (isFastTrack()) { 752 return INVALID_OPERATION; 753 } 754 sp<ThreadBase> thread = mThread.promote(); 755 if (thread == 0) { 756 return INVALID_OPERATION; 757 } 758 Mutex::Autolock _l(thread->mLock); 759 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 760 if (!isOffloaded()) { 761 if (!playbackThread->mLatchQValid) { 762 return INVALID_OPERATION; 763 } 764 uint32_t unpresentedFrames = 765 ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) / 766 playbackThread->mSampleRate; 767 uint32_t framesWritten = mAudioTrackServerProxy->framesReleased(); 768 if (framesWritten < unpresentedFrames) { 769 return INVALID_OPERATION; 770 } 771 timestamp.mPosition = framesWritten - unpresentedFrames; 772 timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime; 773 return NO_ERROR; 774 } 775 776 return playbackThread->getTimestamp_l(timestamp); 777} 778 779status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 780{ 781 status_t status = DEAD_OBJECT; 782 sp<ThreadBase> thread = mThread.promote(); 783 if (thread != 0) { 784 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 785 sp<AudioFlinger> af = mClient->audioFlinger(); 786 787 Mutex::Autolock _l(af->mLock); 788 789 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 790 791 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 792 Mutex::Autolock _dl(playbackThread->mLock); 793 Mutex::Autolock _sl(srcThread->mLock); 794 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 795 if (chain == 0) { 796 return INVALID_OPERATION; 797 } 798 799 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 800 if (effect == 0) { 801 return INVALID_OPERATION; 802 } 803 srcThread->removeEffect_l(effect); 804 status = playbackThread->addEffect_l(effect); 805 if (status != NO_ERROR) { 806 srcThread->addEffect_l(effect); 807 return INVALID_OPERATION; 808 } 809 // removeEffect_l() has stopped the effect if it was active so it must be restarted 810 if (effect->state() == EffectModule::ACTIVE || 811 effect->state() == EffectModule::STOPPING) { 812 effect->start(); 813 } 814 815 sp<EffectChain> dstChain = effect->chain().promote(); 816 if (dstChain == 0) { 817 srcThread->addEffect_l(effect); 818 return INVALID_OPERATION; 819 } 820 AudioSystem::unregisterEffect(effect->id()); 821 AudioSystem::registerEffect(&effect->desc(), 822 srcThread->id(), 823 dstChain->strategy(), 824 AUDIO_SESSION_OUTPUT_MIX, 825 effect->id()); 826 } 827 status = playbackThread->attachAuxEffect(this, EffectId); 828 } 829 return status; 830} 831 832void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 833{ 834 mAuxEffectId = EffectId; 835 mAuxBuffer = buffer; 836} 837 838bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 839 size_t audioHalFrames) 840{ 841 // a track is considered presented when the total number of frames written to audio HAL 842 // corresponds to the number of frames written when presentationComplete() is called for the 843 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 844 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 845 // to detect when all frames have been played. In this case framesWritten isn't 846 // useful because it doesn't always reflect whether there is data in the h/w 847 // buffers, particularly if a track has been paused and resumed during draining 848 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 849 mPresentationCompleteFrames, framesWritten); 850 if (mPresentationCompleteFrames == 0) { 851 mPresentationCompleteFrames = framesWritten + audioHalFrames; 852 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 853 mPresentationCompleteFrames, audioHalFrames); 854 } 855 856 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 857 ALOGV("presentationComplete() session %d complete: framesWritten %d", 858 mSessionId, framesWritten); 859 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 860 mAudioTrackServerProxy->setStreamEndDone(); 861 return true; 862 } 863 return false; 864} 865 866void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 867{ 868 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 869 if (mSyncEvents[i]->type() == type) { 870 mSyncEvents[i]->trigger(); 871 mSyncEvents.removeAt(i); 872 i--; 873 } 874 } 875} 876 877// implement VolumeBufferProvider interface 878 879uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 880{ 881 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 882 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 883 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 884 uint32_t vl = vlr & 0xFFFF; 885 uint32_t vr = vlr >> 16; 886 // track volumes come from shared memory, so can't be trusted and must be clamped 887 if (vl > MAX_GAIN_INT) { 888 vl = MAX_GAIN_INT; 889 } 890 if (vr > MAX_GAIN_INT) { 891 vr = MAX_GAIN_INT; 892 } 893 // now apply the cached master volume and stream type volume; 894 // this is trusted but lacks any synchronization or barrier so may be stale 895 float v = mCachedVolume; 896 vl *= v; 897 vr *= v; 898 // re-combine into U4.16 899 vlr = (vr << 16) | (vl & 0xFFFF); 900 // FIXME look at mute, pause, and stop flags 901 return vlr; 902} 903 904status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 905{ 906 if (isTerminated() || mState == PAUSED || 907 ((framesReady() == 0) && ((mSharedBuffer != 0) || 908 (mState == STOPPED)))) { 909 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 910 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 911 event->cancel(); 912 return INVALID_OPERATION; 913 } 914 (void) TrackBase::setSyncEvent(event); 915 return NO_ERROR; 916} 917 918void AudioFlinger::PlaybackThread::Track::invalidate() 919{ 920 // FIXME should use proxy, and needs work 921 audio_track_cblk_t* cblk = mCblk; 922 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 923 android_atomic_release_store(0x40000000, &cblk->mFutex); 924 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 925 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 926 mIsInvalid = true; 927} 928 929// ---------------------------------------------------------------------------- 930 931sp<AudioFlinger::PlaybackThread::TimedTrack> 932AudioFlinger::PlaybackThread::TimedTrack::create( 933 PlaybackThread *thread, 934 const sp<Client>& client, 935 audio_stream_type_t streamType, 936 uint32_t sampleRate, 937 audio_format_t format, 938 audio_channel_mask_t channelMask, 939 size_t frameCount, 940 const sp<IMemory>& sharedBuffer, 941 int sessionId) { 942 if (!client->reserveTimedTrack()) 943 return 0; 944 945 return new TimedTrack( 946 thread, client, streamType, sampleRate, format, channelMask, frameCount, 947 sharedBuffer, sessionId); 948} 949 950AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 951 PlaybackThread *thread, 952 const sp<Client>& client, 953 audio_stream_type_t streamType, 954 uint32_t sampleRate, 955 audio_format_t format, 956 audio_channel_mask_t channelMask, 957 size_t frameCount, 958 const sp<IMemory>& sharedBuffer, 959 int sessionId) 960 : Track(thread, client, streamType, sampleRate, format, channelMask, 961 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 962 mQueueHeadInFlight(false), 963 mTrimQueueHeadOnRelease(false), 964 mFramesPendingInQueue(0), 965 mTimedSilenceBuffer(NULL), 966 mTimedSilenceBufferSize(0), 967 mTimedAudioOutputOnTime(false), 968 mMediaTimeTransformValid(false) 969{ 970 LocalClock lc; 971 mLocalTimeFreq = lc.getLocalFreq(); 972 973 mLocalTimeToSampleTransform.a_zero = 0; 974 mLocalTimeToSampleTransform.b_zero = 0; 975 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 976 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 977 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 978 &mLocalTimeToSampleTransform.a_to_b_denom); 979 980 mMediaTimeToSampleTransform.a_zero = 0; 981 mMediaTimeToSampleTransform.b_zero = 0; 982 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 983 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 984 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 985 &mMediaTimeToSampleTransform.a_to_b_denom); 986} 987 988AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 989 mClient->releaseTimedTrack(); 990 delete [] mTimedSilenceBuffer; 991} 992 993status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 994 size_t size, sp<IMemory>* buffer) { 995 996 Mutex::Autolock _l(mTimedBufferQueueLock); 997 998 trimTimedBufferQueue_l(); 999 1000 // lazily initialize the shared memory heap for timed buffers 1001 if (mTimedMemoryDealer == NULL) { 1002 const int kTimedBufferHeapSize = 512 << 10; 1003 1004 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 1005 "AudioFlingerTimed"); 1006 if (mTimedMemoryDealer == NULL) 1007 return NO_MEMORY; 1008 } 1009 1010 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 1011 if (newBuffer == NULL) { 1012 newBuffer = mTimedMemoryDealer->allocate(size); 1013 if (newBuffer == NULL) 1014 return NO_MEMORY; 1015 } 1016 1017 *buffer = newBuffer; 1018 return NO_ERROR; 1019} 1020 1021// caller must hold mTimedBufferQueueLock 1022void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 1023 int64_t mediaTimeNow; 1024 { 1025 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1026 if (!mMediaTimeTransformValid) 1027 return; 1028 1029 int64_t targetTimeNow; 1030 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 1031 ? mCCHelper.getCommonTime(&targetTimeNow) 1032 : mCCHelper.getLocalTime(&targetTimeNow); 1033 1034 if (OK != res) 1035 return; 1036 1037 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 1038 &mediaTimeNow)) { 1039 return; 1040 } 1041 } 1042 1043 size_t trimEnd; 1044 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 1045 int64_t bufEnd; 1046 1047 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 1048 // We have a next buffer. Just use its PTS as the PTS of the frame 1049 // following the last frame in this buffer. If the stream is sparse 1050 // (ie, there are deliberate gaps left in the stream which should be 1051 // filled with silence by the TimedAudioTrack), then this can result 1052 // in one extra buffer being left un-trimmed when it could have 1053 // been. In general, this is not typical, and we would rather 1054 // optimized away the TS calculation below for the more common case 1055 // where PTSes are contiguous. 1056 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 1057 } else { 1058 // We have no next buffer. Compute the PTS of the frame following 1059 // the last frame in this buffer by computing the duration of of 1060 // this frame in media time units and adding it to the PTS of the 1061 // buffer. 1062 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1063 / mFrameSize; 1064 1065 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1066 &bufEnd)) { 1067 ALOGE("Failed to convert frame count of %lld to media time" 1068 " duration" " (scale factor %d/%u) in %s", 1069 frameCount, 1070 mMediaTimeToSampleTransform.a_to_b_numer, 1071 mMediaTimeToSampleTransform.a_to_b_denom, 1072 __PRETTY_FUNCTION__); 1073 break; 1074 } 1075 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1076 } 1077 1078 if (bufEnd > mediaTimeNow) 1079 break; 1080 1081 // Is the buffer we want to use in the middle of a mix operation right 1082 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1083 // from the mixer which should be coming back shortly. 1084 if (!trimEnd && mQueueHeadInFlight) { 1085 mTrimQueueHeadOnRelease = true; 1086 } 1087 } 1088 1089 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1090 if (trimStart < trimEnd) { 1091 // Update the bookkeeping for framesReady() 1092 for (size_t i = trimStart; i < trimEnd; ++i) { 1093 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1094 } 1095 1096 // Now actually remove the buffers from the queue. 1097 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1098 } 1099} 1100 1101void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1102 const char* logTag) { 1103 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1104 "%s called (reason \"%s\"), but timed buffer queue has no" 1105 " elements to trim.", __FUNCTION__, logTag); 1106 1107 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1108 mTimedBufferQueue.removeAt(0); 1109} 1110 1111void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1112 const TimedBuffer& buf, 1113 const char* logTag) { 1114 uint32_t bufBytes = buf.buffer()->size(); 1115 uint32_t consumedAlready = buf.position(); 1116 1117 ALOG_ASSERT(consumedAlready <= bufBytes, 1118 "Bad bookkeeping while updating frames pending. Timed buffer is" 1119 " only %u bytes long, but claims to have consumed %u" 1120 " bytes. (update reason: \"%s\")", 1121 bufBytes, consumedAlready, logTag); 1122 1123 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1124 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1125 "Bad bookkeeping while updating frames pending. Should have at" 1126 " least %u queued frames, but we think we have only %u. (update" 1127 " reason: \"%s\")", 1128 bufFrames, mFramesPendingInQueue, logTag); 1129 1130 mFramesPendingInQueue -= bufFrames; 1131} 1132 1133status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1134 const sp<IMemory>& buffer, int64_t pts) { 1135 1136 { 1137 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1138 if (!mMediaTimeTransformValid) 1139 return INVALID_OPERATION; 1140 } 1141 1142 Mutex::Autolock _l(mTimedBufferQueueLock); 1143 1144 uint32_t bufFrames = buffer->size() / mFrameSize; 1145 mFramesPendingInQueue += bufFrames; 1146 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1147 1148 return NO_ERROR; 1149} 1150 1151status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1152 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1153 1154 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1155 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1156 target); 1157 1158 if (!(target == TimedAudioTrack::LOCAL_TIME || 1159 target == TimedAudioTrack::COMMON_TIME)) { 1160 return BAD_VALUE; 1161 } 1162 1163 Mutex::Autolock lock(mMediaTimeTransformLock); 1164 mMediaTimeTransform = xform; 1165 mMediaTimeTransformTarget = target; 1166 mMediaTimeTransformValid = true; 1167 1168 return NO_ERROR; 1169} 1170 1171#define min(a, b) ((a) < (b) ? (a) : (b)) 1172 1173// implementation of getNextBuffer for tracks whose buffers have timestamps 1174status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1175 AudioBufferProvider::Buffer* buffer, int64_t pts) 1176{ 1177 if (pts == AudioBufferProvider::kInvalidPTS) { 1178 buffer->raw = NULL; 1179 buffer->frameCount = 0; 1180 mTimedAudioOutputOnTime = false; 1181 return INVALID_OPERATION; 1182 } 1183 1184 Mutex::Autolock _l(mTimedBufferQueueLock); 1185 1186 ALOG_ASSERT(!mQueueHeadInFlight, 1187 "getNextBuffer called without releaseBuffer!"); 1188 1189 while (true) { 1190 1191 // if we have no timed buffers, then fail 1192 if (mTimedBufferQueue.isEmpty()) { 1193 buffer->raw = NULL; 1194 buffer->frameCount = 0; 1195 return NOT_ENOUGH_DATA; 1196 } 1197 1198 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1199 1200 // calculate the PTS of the head of the timed buffer queue expressed in 1201 // local time 1202 int64_t headLocalPTS; 1203 { 1204 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1205 1206 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1207 1208 if (mMediaTimeTransform.a_to_b_denom == 0) { 1209 // the transform represents a pause, so yield silence 1210 timedYieldSilence_l(buffer->frameCount, buffer); 1211 return NO_ERROR; 1212 } 1213 1214 int64_t transformedPTS; 1215 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1216 &transformedPTS)) { 1217 // the transform failed. this shouldn't happen, but if it does 1218 // then just drop this buffer 1219 ALOGW("timedGetNextBuffer transform failed"); 1220 buffer->raw = NULL; 1221 buffer->frameCount = 0; 1222 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1223 return NO_ERROR; 1224 } 1225 1226 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1227 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1228 &headLocalPTS)) { 1229 buffer->raw = NULL; 1230 buffer->frameCount = 0; 1231 return INVALID_OPERATION; 1232 } 1233 } else { 1234 headLocalPTS = transformedPTS; 1235 } 1236 } 1237 1238 uint32_t sr = sampleRate(); 1239 1240 // adjust the head buffer's PTS to reflect the portion of the head buffer 1241 // that has already been consumed 1242 int64_t effectivePTS = headLocalPTS + 1243 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1244 1245 // Calculate the delta in samples between the head of the input buffer 1246 // queue and the start of the next output buffer that will be written. 1247 // If the transformation fails because of over or underflow, it means 1248 // that the sample's position in the output stream is so far out of 1249 // whack that it should just be dropped. 1250 int64_t sampleDelta; 1251 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1252 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1253 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1254 " mix"); 1255 continue; 1256 } 1257 if (!mLocalTimeToSampleTransform.doForwardTransform( 1258 (effectivePTS - pts) << 32, &sampleDelta)) { 1259 ALOGV("*** too late during sample rate transform: dropped buffer"); 1260 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1261 continue; 1262 } 1263 1264 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1265 " sampleDelta=[%d.%08x]", 1266 head.pts(), head.position(), pts, 1267 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1268 + (sampleDelta >> 32)), 1269 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1270 1271 // if the delta between the ideal placement for the next input sample and 1272 // the current output position is within this threshold, then we will 1273 // concatenate the next input samples to the previous output 1274 const int64_t kSampleContinuityThreshold = 1275 (static_cast<int64_t>(sr) << 32) / 250; 1276 1277 // if this is the first buffer of audio that we're emitting from this track 1278 // then it should be almost exactly on time. 1279 const int64_t kSampleStartupThreshold = 1LL << 32; 1280 1281 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1282 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1283 // the next input is close enough to being on time, so concatenate it 1284 // with the last output 1285 timedYieldSamples_l(buffer); 1286 1287 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1288 head.position(), buffer->frameCount); 1289 return NO_ERROR; 1290 } 1291 1292 // Looks like our output is not on time. Reset our on timed status. 1293 // Next time we mix samples from our input queue, then should be within 1294 // the StartupThreshold. 1295 mTimedAudioOutputOnTime = false; 1296 if (sampleDelta > 0) { 1297 // the gap between the current output position and the proper start of 1298 // the next input sample is too big, so fill it with silence 1299 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1300 1301 timedYieldSilence_l(framesUntilNextInput, buffer); 1302 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1303 return NO_ERROR; 1304 } else { 1305 // the next input sample is late 1306 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1307 size_t onTimeSamplePosition = 1308 head.position() + lateFrames * mFrameSize; 1309 1310 if (onTimeSamplePosition > head.buffer()->size()) { 1311 // all the remaining samples in the head are too late, so 1312 // drop it and move on 1313 ALOGV("*** too late: dropped buffer"); 1314 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1315 continue; 1316 } else { 1317 // skip over the late samples 1318 head.setPosition(onTimeSamplePosition); 1319 1320 // yield the available samples 1321 timedYieldSamples_l(buffer); 1322 1323 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1324 return NO_ERROR; 1325 } 1326 } 1327 } 1328} 1329 1330// Yield samples from the timed buffer queue head up to the given output 1331// buffer's capacity. 1332// 1333// Caller must hold mTimedBufferQueueLock 1334void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1335 AudioBufferProvider::Buffer* buffer) { 1336 1337 const TimedBuffer& head = mTimedBufferQueue[0]; 1338 1339 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1340 head.position()); 1341 1342 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1343 mFrameSize); 1344 size_t framesRequested = buffer->frameCount; 1345 buffer->frameCount = min(framesLeftInHead, framesRequested); 1346 1347 mQueueHeadInFlight = true; 1348 mTimedAudioOutputOnTime = true; 1349} 1350 1351// Yield samples of silence up to the given output buffer's capacity 1352// 1353// Caller must hold mTimedBufferQueueLock 1354void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1355 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1356 1357 // lazily allocate a buffer filled with silence 1358 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1359 delete [] mTimedSilenceBuffer; 1360 mTimedSilenceBufferSize = numFrames * mFrameSize; 1361 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1362 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1363 } 1364 1365 buffer->raw = mTimedSilenceBuffer; 1366 size_t framesRequested = buffer->frameCount; 1367 buffer->frameCount = min(numFrames, framesRequested); 1368 1369 mTimedAudioOutputOnTime = false; 1370} 1371 1372// AudioBufferProvider interface 1373void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1374 AudioBufferProvider::Buffer* buffer) { 1375 1376 Mutex::Autolock _l(mTimedBufferQueueLock); 1377 1378 // If the buffer which was just released is part of the buffer at the head 1379 // of the queue, be sure to update the amt of the buffer which has been 1380 // consumed. If the buffer being returned is not part of the head of the 1381 // queue, its either because the buffer is part of the silence buffer, or 1382 // because the head of the timed queue was trimmed after the mixer called 1383 // getNextBuffer but before the mixer called releaseBuffer. 1384 if (buffer->raw == mTimedSilenceBuffer) { 1385 ALOG_ASSERT(!mQueueHeadInFlight, 1386 "Queue head in flight during release of silence buffer!"); 1387 goto done; 1388 } 1389 1390 ALOG_ASSERT(mQueueHeadInFlight, 1391 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1392 " head in flight."); 1393 1394 if (mTimedBufferQueue.size()) { 1395 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1396 1397 void* start = head.buffer()->pointer(); 1398 void* end = reinterpret_cast<void*>( 1399 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1400 + head.buffer()->size()); 1401 1402 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1403 "released buffer not within the head of the timed buffer" 1404 " queue; qHead = [%p, %p], released buffer = %p", 1405 start, end, buffer->raw); 1406 1407 head.setPosition(head.position() + 1408 (buffer->frameCount * mFrameSize)); 1409 mQueueHeadInFlight = false; 1410 1411 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1412 "Bad bookkeeping during releaseBuffer! Should have at" 1413 " least %u queued frames, but we think we have only %u", 1414 buffer->frameCount, mFramesPendingInQueue); 1415 1416 mFramesPendingInQueue -= buffer->frameCount; 1417 1418 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1419 || mTrimQueueHeadOnRelease) { 1420 trimTimedBufferQueueHead_l("releaseBuffer"); 1421 mTrimQueueHeadOnRelease = false; 1422 } 1423 } else { 1424 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1425 " buffers in the timed buffer queue"); 1426 } 1427 1428done: 1429 buffer->raw = 0; 1430 buffer->frameCount = 0; 1431} 1432 1433size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1434 Mutex::Autolock _l(mTimedBufferQueueLock); 1435 return mFramesPendingInQueue; 1436} 1437 1438AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1439 : mPTS(0), mPosition(0) {} 1440 1441AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1442 const sp<IMemory>& buffer, int64_t pts) 1443 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1444 1445 1446// ---------------------------------------------------------------------------- 1447 1448AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1449 PlaybackThread *playbackThread, 1450 DuplicatingThread *sourceThread, 1451 uint32_t sampleRate, 1452 audio_format_t format, 1453 audio_channel_mask_t channelMask, 1454 size_t frameCount) 1455 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1456 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 1457 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1458{ 1459 1460 if (mCblk != NULL) { 1461 mOutBuffer.frameCount = 0; 1462 playbackThread->mTracks.add(this); 1463 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1464 "mCblk->frameCount_ %u, mChannelMask 0x%08x", 1465 mCblk, mBuffer, 1466 mCblk->frameCount_, mChannelMask); 1467 // since client and server are in the same process, 1468 // the buffer has the same virtual address on both sides 1469 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize); 1470 mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000)); 1471 mClientProxy->setSendLevel(0.0); 1472 mClientProxy->setSampleRate(sampleRate); 1473 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1474 true /*clientInServer*/); 1475 } else { 1476 ALOGW("Error creating output track on thread %p", playbackThread); 1477 } 1478} 1479 1480AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1481{ 1482 clearBufferQueue(); 1483 delete mClientProxy; 1484 // superclass destructor will now delete the server proxy and shared memory both refer to 1485} 1486 1487status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1488 int triggerSession) 1489{ 1490 status_t status = Track::start(event, triggerSession); 1491 if (status != NO_ERROR) { 1492 return status; 1493 } 1494 1495 mActive = true; 1496 mRetryCount = 127; 1497 return status; 1498} 1499 1500void AudioFlinger::PlaybackThread::OutputTrack::stop() 1501{ 1502 Track::stop(); 1503 clearBufferQueue(); 1504 mOutBuffer.frameCount = 0; 1505 mActive = false; 1506} 1507 1508bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1509{ 1510 Buffer *pInBuffer; 1511 Buffer inBuffer; 1512 uint32_t channelCount = mChannelCount; 1513 bool outputBufferFull = false; 1514 inBuffer.frameCount = frames; 1515 inBuffer.i16 = data; 1516 1517 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1518 1519 if (!mActive && frames != 0) { 1520 start(); 1521 sp<ThreadBase> thread = mThread.promote(); 1522 if (thread != 0) { 1523 MixerThread *mixerThread = (MixerThread *)thread.get(); 1524 if (mFrameCount > frames) { 1525 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1526 uint32_t startFrames = (mFrameCount - frames); 1527 pInBuffer = new Buffer; 1528 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1529 pInBuffer->frameCount = startFrames; 1530 pInBuffer->i16 = pInBuffer->mBuffer; 1531 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1532 mBufferQueue.add(pInBuffer); 1533 } else { 1534 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1535 } 1536 } 1537 } 1538 } 1539 1540 while (waitTimeLeftMs) { 1541 // First write pending buffers, then new data 1542 if (mBufferQueue.size()) { 1543 pInBuffer = mBufferQueue.itemAt(0); 1544 } else { 1545 pInBuffer = &inBuffer; 1546 } 1547 1548 if (pInBuffer->frameCount == 0) { 1549 break; 1550 } 1551 1552 if (mOutBuffer.frameCount == 0) { 1553 mOutBuffer.frameCount = pInBuffer->frameCount; 1554 nsecs_t startTime = systemTime(); 1555 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1556 if (status != NO_ERROR) { 1557 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1558 mThread.unsafe_get(), status); 1559 outputBufferFull = true; 1560 break; 1561 } 1562 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1563 if (waitTimeLeftMs >= waitTimeMs) { 1564 waitTimeLeftMs -= waitTimeMs; 1565 } else { 1566 waitTimeLeftMs = 0; 1567 } 1568 } 1569 1570 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1571 pInBuffer->frameCount; 1572 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1573 Proxy::Buffer buf; 1574 buf.mFrameCount = outFrames; 1575 buf.mRaw = NULL; 1576 mClientProxy->releaseBuffer(&buf); 1577 pInBuffer->frameCount -= outFrames; 1578 pInBuffer->i16 += outFrames * channelCount; 1579 mOutBuffer.frameCount -= outFrames; 1580 mOutBuffer.i16 += outFrames * channelCount; 1581 1582 if (pInBuffer->frameCount == 0) { 1583 if (mBufferQueue.size()) { 1584 mBufferQueue.removeAt(0); 1585 delete [] pInBuffer->mBuffer; 1586 delete pInBuffer; 1587 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1588 mThread.unsafe_get(), mBufferQueue.size()); 1589 } else { 1590 break; 1591 } 1592 } 1593 } 1594 1595 // If we could not write all frames, allocate a buffer and queue it for next time. 1596 if (inBuffer.frameCount) { 1597 sp<ThreadBase> thread = mThread.promote(); 1598 if (thread != 0 && !thread->standby()) { 1599 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1600 pInBuffer = new Buffer; 1601 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1602 pInBuffer->frameCount = inBuffer.frameCount; 1603 pInBuffer->i16 = pInBuffer->mBuffer; 1604 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1605 sizeof(int16_t)); 1606 mBufferQueue.add(pInBuffer); 1607 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1608 mThread.unsafe_get(), mBufferQueue.size()); 1609 } else { 1610 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1611 mThread.unsafe_get(), this); 1612 } 1613 } 1614 } 1615 1616 // Calling write() with a 0 length buffer, means that no more data will be written: 1617 // If no more buffers are pending, fill output track buffer to make sure it is started 1618 // by output mixer. 1619 if (frames == 0 && mBufferQueue.size() == 0) { 1620 // FIXME borken, replace by getting framesReady() from proxy 1621 size_t user = 0; // was mCblk->user 1622 if (user < mFrameCount) { 1623 frames = mFrameCount - user; 1624 pInBuffer = new Buffer; 1625 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1626 pInBuffer->frameCount = frames; 1627 pInBuffer->i16 = pInBuffer->mBuffer; 1628 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1629 mBufferQueue.add(pInBuffer); 1630 } else if (mActive) { 1631 stop(); 1632 } 1633 } 1634 1635 return outputBufferFull; 1636} 1637 1638status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1639 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1640{ 1641 ClientProxy::Buffer buf; 1642 buf.mFrameCount = buffer->frameCount; 1643 struct timespec timeout; 1644 timeout.tv_sec = waitTimeMs / 1000; 1645 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1646 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1647 buffer->frameCount = buf.mFrameCount; 1648 buffer->raw = buf.mRaw; 1649 return status; 1650} 1651 1652void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1653{ 1654 size_t size = mBufferQueue.size(); 1655 1656 for (size_t i = 0; i < size; i++) { 1657 Buffer *pBuffer = mBufferQueue.itemAt(i); 1658 delete [] pBuffer->mBuffer; 1659 delete pBuffer; 1660 } 1661 mBufferQueue.clear(); 1662} 1663 1664 1665// ---------------------------------------------------------------------------- 1666// Record 1667// ---------------------------------------------------------------------------- 1668 1669AudioFlinger::RecordHandle::RecordHandle( 1670 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1671 : BnAudioRecord(), 1672 mRecordTrack(recordTrack) 1673{ 1674} 1675 1676AudioFlinger::RecordHandle::~RecordHandle() { 1677 stop_nonvirtual(); 1678 mRecordTrack->destroy(); 1679} 1680 1681sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 1682 return mRecordTrack->getCblk(); 1683} 1684 1685status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1686 int triggerSession) { 1687 ALOGV("RecordHandle::start()"); 1688 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1689} 1690 1691void AudioFlinger::RecordHandle::stop() { 1692 stop_nonvirtual(); 1693} 1694 1695void AudioFlinger::RecordHandle::stop_nonvirtual() { 1696 ALOGV("RecordHandle::stop()"); 1697 mRecordTrack->stop(); 1698} 1699 1700status_t AudioFlinger::RecordHandle::onTransact( 1701 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1702{ 1703 return BnAudioRecord::onTransact(code, data, reply, flags); 1704} 1705 1706// ---------------------------------------------------------------------------- 1707 1708// RecordTrack constructor must be called with AudioFlinger::mLock held 1709AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1710 RecordThread *thread, 1711 const sp<Client>& client, 1712 uint32_t sampleRate, 1713 audio_format_t format, 1714 audio_channel_mask_t channelMask, 1715 size_t frameCount, 1716 int sessionId) 1717 : TrackBase(thread, client, sampleRate, format, 1718 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/), 1719 mOverflow(false) 1720{ 1721 ALOGV("RecordTrack constructor"); 1722 if (mCblk != NULL) { 1723 mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, 1724 mFrameSize); 1725 mServerProxy = mAudioRecordServerProxy; 1726 } 1727} 1728 1729AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 1730{ 1731 ALOGV("%s", __func__); 1732} 1733 1734// AudioBufferProvider interface 1735status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 1736 int64_t pts) 1737{ 1738 ServerProxy::Buffer buf; 1739 buf.mFrameCount = buffer->frameCount; 1740 status_t status = mServerProxy->obtainBuffer(&buf); 1741 buffer->frameCount = buf.mFrameCount; 1742 buffer->raw = buf.mRaw; 1743 if (buf.mFrameCount == 0) { 1744 // FIXME also wake futex so that overrun is noticed more quickly 1745 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 1746 } 1747 return status; 1748} 1749 1750status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 1751 int triggerSession) 1752{ 1753 sp<ThreadBase> thread = mThread.promote(); 1754 if (thread != 0) { 1755 RecordThread *recordThread = (RecordThread *)thread.get(); 1756 return recordThread->start(this, event, triggerSession); 1757 } else { 1758 return BAD_VALUE; 1759 } 1760} 1761 1762void AudioFlinger::RecordThread::RecordTrack::stop() 1763{ 1764 sp<ThreadBase> thread = mThread.promote(); 1765 if (thread != 0) { 1766 RecordThread *recordThread = (RecordThread *)thread.get(); 1767 if (recordThread->stop(this)) { 1768 AudioSystem::stopInput(recordThread->id()); 1769 } 1770 } 1771} 1772 1773void AudioFlinger::RecordThread::RecordTrack::destroy() 1774{ 1775 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 1776 sp<RecordTrack> keep(this); 1777 { 1778 sp<ThreadBase> thread = mThread.promote(); 1779 if (thread != 0) { 1780 if (mState == ACTIVE || mState == RESUMING) { 1781 AudioSystem::stopInput(thread->id()); 1782 } 1783 AudioSystem::releaseInput(thread->id()); 1784 Mutex::Autolock _l(thread->mLock); 1785 RecordThread *recordThread = (RecordThread *) thread.get(); 1786 recordThread->destroyTrack_l(this); 1787 } 1788 } 1789} 1790 1791void AudioFlinger::RecordThread::RecordTrack::invalidate() 1792{ 1793 // FIXME should use proxy, and needs work 1794 audio_track_cblk_t* cblk = mCblk; 1795 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1796 android_atomic_release_store(0x40000000, &cblk->mFutex); 1797 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 1798 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 1799} 1800 1801 1802/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 1803{ 1804 result.append("Client Fmt Chn mask Session S Server fCount\n"); 1805} 1806 1807void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 1808{ 1809 snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6u\n", 1810 (mClient == 0) ? getpid_cached : mClient->pid(), 1811 mFormat, 1812 mChannelMask, 1813 mSessionId, 1814 mState, 1815 mCblk->mServer, 1816 mFrameCount); 1817} 1818 1819}; // namespace android 1820