Tracks.cpp revision d72b7c0180ee83fc3754629ed68fc5887a125c4c
1f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org/* 2f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org** 3f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org** Copyright 2012, The Android Open Source Project 4f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org** 5f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org** Licensed under the Apache License, Version 2.0 (the "License"); 6f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org** you may not use this file except in compliance with the License. 7f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org** You may obtain a copy of the License at 8f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org** 9f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org** http://www.apache.org/licenses/LICENSE-2.0 10f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org** 11f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org** Unless required by applicable law or agreed to in writing, software 12f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org** distributed under the License is distributed on an "AS IS" BASIS, 13f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org** See the License for the specific language governing permissions and 15f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org** limitations under the License. 16f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org*/ 17f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org 18f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org 19f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#define LOG_TAG "AudioFlinger" 20f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org//#define LOG_NDEBUG 0 21f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org 22f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#include "Configuration.h" 23f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#include <math.h> 24f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#include <utils/Log.h> 25f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org 26f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#include <private/media/AudioTrackShared.h> 27f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org 28f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#include <common_time/cc_helper.h> 29f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#include <common_time/local_clock.h> 30f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org 31f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#include "AudioMixer.h" 32f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#include "AudioFlinger.h" 33f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#include "ServiceUtilities.h" 34f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org 35f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#include <media/nbaio/Pipe.h> 36f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#include <media/nbaio/PipeReader.h> 37f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org 38f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org// ---------------------------------------------------------------------------- 39f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org 40f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org// Note: the following macro is used for extremely verbose logging message. In 41f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 42f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org// 0; but one side effect of this is to turn all LOGV's as well. Some messages 43f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org// are so verbose that we want to suppress them even when we have ALOG_ASSERT 44f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org// turned on. Do not uncomment the #def below unless you really know what you 45f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org// are doing and want to see all of the extremely verbose messages. 46f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org//#define VERY_VERY_VERBOSE_LOGGING 47f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#ifdef VERY_VERY_VERBOSE_LOGGING 48f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#define ALOGVV ALOGV 49f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#else 50f2ba7591b1407a7ee9209f842c50696914dc2dedkbr@chromium.org#define ALOGVV(a...) do { } while(0) 51#endif 52 53namespace android { 54 55// ---------------------------------------------------------------------------- 56// TrackBase 57// ---------------------------------------------------------------------------- 58 59static 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} 290 291 292void AudioFlinger::TrackHandle::signal() 293{ 294 return mTrack->signal(); 295} 296 297status_t AudioFlinger::TrackHandle::onTransact( 298 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 299{ 300 return BnAudioTrack::onTransact(code, data, reply, flags); 301} 302 303// ---------------------------------------------------------------------------- 304 305// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 306AudioFlinger::PlaybackThread::Track::Track( 307 PlaybackThread *thread, 308 const sp<Client>& client, 309 audio_stream_type_t streamType, 310 uint32_t sampleRate, 311 audio_format_t format, 312 audio_channel_mask_t channelMask, 313 size_t frameCount, 314 const sp<IMemory>& sharedBuffer, 315 int sessionId, 316 IAudioFlinger::track_flags_t flags) 317 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, 318 sessionId, true /*isOut*/), 319 mFillingUpStatus(FS_INVALID), 320 // mRetryCount initialized later when needed 321 mSharedBuffer(sharedBuffer), 322 mStreamType(streamType), 323 mName(-1), // see note below 324 mMainBuffer(thread->mixBuffer()), 325 mAuxBuffer(NULL), 326 mAuxEffectId(0), mHasVolumeController(false), 327 mPresentationCompleteFrames(0), 328 mFlags(flags), 329 mFastIndex(-1), 330 mCachedVolume(1.0), 331 mIsInvalid(false), 332 mAudioTrackServerProxy(NULL), 333 mResumeToStopping(false) 334{ 335 if (mCblk != NULL) { 336 if (sharedBuffer == 0) { 337 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 338 mFrameSize); 339 } else { 340 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, 341 mFrameSize); 342 } 343 mServerProxy = mAudioTrackServerProxy; 344 // to avoid leaking a track name, do not allocate one unless there is an mCblk 345 mName = thread->getTrackName_l(channelMask, sessionId); 346 if (mName < 0) { 347 ALOGE("no more track names available"); 348 return; 349 } 350 // only allocate a fast track index if we were able to allocate a normal track name 351 if (flags & IAudioFlinger::TRACK_FAST) { 352 mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); 353 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 354 int i = __builtin_ctz(thread->mFastTrackAvailMask); 355 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 356 // FIXME This is too eager. We allocate a fast track index before the 357 // fast track becomes active. Since fast tracks are a scarce resource, 358 // this means we are potentially denying other more important fast tracks from 359 // being created. It would be better to allocate the index dynamically. 360 mFastIndex = i; 361 // Read the initial underruns because this field is never cleared by the fast mixer 362 mObservedUnderruns = thread->getFastTrackUnderruns(i); 363 thread->mFastTrackAvailMask &= ~(1 << i); 364 } 365 } 366 ALOGV("Track constructor name %d, calling pid %d", mName, 367 IPCThreadState::self()->getCallingPid()); 368} 369 370AudioFlinger::PlaybackThread::Track::~Track() 371{ 372 ALOGV("PlaybackThread::Track destructor"); 373 374 // The destructor would clear mSharedBuffer, 375 // but it will not push the decremented reference count, 376 // leaving the client's IMemory dangling indefinitely. 377 // This prevents that leak. 378 if (mSharedBuffer != 0) { 379 mSharedBuffer.clear(); 380 // flush the binder command buffer 381 IPCThreadState::self()->flushCommands(); 382 } 383} 384 385void AudioFlinger::PlaybackThread::Track::destroy() 386{ 387 // NOTE: destroyTrack_l() can remove a strong reference to this Track 388 // by removing it from mTracks vector, so there is a risk that this Tracks's 389 // destructor is called. As the destructor needs to lock mLock, 390 // we must acquire a strong reference on this Track before locking mLock 391 // here so that the destructor is called only when exiting this function. 392 // On the other hand, as long as Track::destroy() is only called by 393 // TrackHandle destructor, the TrackHandle still holds a strong ref on 394 // this Track with its member mTrack. 395 sp<Track> keep(this); 396 { // scope for mLock 397 sp<ThreadBase> thread = mThread.promote(); 398 if (thread != 0) { 399 Mutex::Autolock _l(thread->mLock); 400 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 401 bool wasActive = playbackThread->destroyTrack_l(this); 402 if (!isOutputTrack() && !wasActive) { 403 AudioSystem::releaseOutput(thread->id()); 404 } 405 } 406 } 407} 408 409/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 410{ 411 result.append(" Name Client Type Fmt Chn mask Session fCount S F SRate " 412 "L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n"); 413} 414 415void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 416{ 417 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 418 if (isFastTrack()) { 419 sprintf(buffer, " F %2d", mFastIndex); 420 } else { 421 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 422 } 423 track_state state = mState; 424 char stateChar; 425 if (isTerminated()) { 426 stateChar = 'T'; 427 } else { 428 switch (state) { 429 case IDLE: 430 stateChar = 'I'; 431 break; 432 case STOPPING_1: 433 stateChar = 's'; 434 break; 435 case STOPPING_2: 436 stateChar = '5'; 437 break; 438 case STOPPED: 439 stateChar = 'S'; 440 break; 441 case RESUMING: 442 stateChar = 'R'; 443 break; 444 case ACTIVE: 445 stateChar = 'A'; 446 break; 447 case PAUSING: 448 stateChar = 'p'; 449 break; 450 case PAUSED: 451 stateChar = 'P'; 452 break; 453 case FLUSHED: 454 stateChar = 'F'; 455 break; 456 default: 457 stateChar = '?'; 458 break; 459 } 460 } 461 char nowInUnderrun; 462 switch (mObservedUnderruns.mBitFields.mMostRecent) { 463 case UNDERRUN_FULL: 464 nowInUnderrun = ' '; 465 break; 466 case UNDERRUN_PARTIAL: 467 nowInUnderrun = '<'; 468 break; 469 case UNDERRUN_EMPTY: 470 nowInUnderrun = '*'; 471 break; 472 default: 473 nowInUnderrun = '?'; 474 break; 475 } 476 snprintf(&buffer[7], size-7, " %6u %4u %08X %08X %7u %6u %1c %1d %5u %5.2g %5.2g " 477 "%08X %08X %08X 0x%03X %9u%c\n", 478 (mClient == 0) ? getpid_cached : mClient->pid(), 479 mStreamType, 480 mFormat, 481 mChannelMask, 482 mSessionId, 483 mFrameCount, 484 stateChar, 485 mFillingUpStatus, 486 mAudioTrackServerProxy->getSampleRate(), 487 20.0 * log10((vlr & 0xFFFF) / 4096.0), 488 20.0 * log10((vlr >> 16) / 4096.0), 489 mCblk->mServer, 490 (int)mMainBuffer, 491 (int)mAuxBuffer, 492 mCblk->mFlags, 493 mAudioTrackServerProxy->getUnderrunFrames(), 494 nowInUnderrun); 495} 496 497uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { 498 return mAudioTrackServerProxy->getSampleRate(); 499} 500 501// AudioBufferProvider interface 502status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 503 AudioBufferProvider::Buffer* buffer, int64_t pts) 504{ 505 ServerProxy::Buffer buf; 506 size_t desiredFrames = buffer->frameCount; 507 buf.mFrameCount = desiredFrames; 508 status_t status = mServerProxy->obtainBuffer(&buf); 509 buffer->frameCount = buf.mFrameCount; 510 buffer->raw = buf.mRaw; 511 if (buf.mFrameCount == 0) { 512 mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 513 } 514 return status; 515} 516 517// releaseBuffer() is not overridden 518 519// ExtendedAudioBufferProvider interface 520 521// Note that framesReady() takes a mutex on the control block using tryLock(). 522// This could result in priority inversion if framesReady() is called by the normal mixer, 523// as the normal mixer thread runs at lower 524// priority than the client's callback thread: there is a short window within framesReady() 525// during which the normal mixer could be preempted, and the client callback would block. 526// Another problem can occur if framesReady() is called by the fast mixer: 527// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 528// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 529size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 530 return mAudioTrackServerProxy->framesReady(); 531} 532 533size_t AudioFlinger::PlaybackThread::Track::framesReleased() const 534{ 535 return mAudioTrackServerProxy->framesReleased(); 536} 537 538// Don't call for fast tracks; the framesReady() could result in priority inversion 539bool AudioFlinger::PlaybackThread::Track::isReady() const { 540 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 541 return true; 542 } 543 544 if (framesReady() >= mFrameCount || 545 (mCblk->mFlags & CBLK_FORCEREADY)) { 546 mFillingUpStatus = FS_FILLED; 547 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 548 return true; 549 } 550 return false; 551} 552 553status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 554 int triggerSession) 555{ 556 status_t status = NO_ERROR; 557 ALOGV("start(%d), calling pid %d session %d", 558 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 559 560 sp<ThreadBase> thread = mThread.promote(); 561 if (thread != 0) { 562 if (isOffloaded()) { 563 Mutex::Autolock _laf(thread->mAudioFlinger->mLock); 564 Mutex::Autolock _lth(thread->mLock); 565 sp<EffectChain> ec = thread->getEffectChain_l(mSessionId); 566 if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() || 567 (ec != 0 && ec->isNonOffloadableEnabled())) { 568 invalidate(); 569 return PERMISSION_DENIED; 570 } 571 } 572 Mutex::Autolock _lth(thread->mLock); 573 track_state state = mState; 574 // here the track could be either new, or restarted 575 // in both cases "unstop" the track 576 577 if (state == PAUSED) { 578 if (mResumeToStopping) { 579 // happened we need to resume to STOPPING_1 580 mState = TrackBase::STOPPING_1; 581 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 582 } else { 583 mState = TrackBase::RESUMING; 584 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 585 } 586 } else { 587 mState = TrackBase::ACTIVE; 588 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 589 } 590 591 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 592 status = playbackThread->addTrack_l(this); 593 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 594 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 595 // restore previous state if start was rejected by policy manager 596 if (status == PERMISSION_DENIED) { 597 mState = state; 598 } 599 } 600 // track was already in the active list, not a problem 601 if (status == ALREADY_EXISTS) { 602 status = NO_ERROR; 603 } 604 } else { 605 status = BAD_VALUE; 606 } 607 return status; 608} 609 610void AudioFlinger::PlaybackThread::Track::stop() 611{ 612 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 613 sp<ThreadBase> thread = mThread.promote(); 614 if (thread != 0) { 615 Mutex::Autolock _l(thread->mLock); 616 track_state state = mState; 617 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 618 // If the track is not active (PAUSED and buffers full), flush buffers 619 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 620 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 621 reset(); 622 mState = STOPPED; 623 } else if (!isFastTrack() && !isOffloaded()) { 624 mState = STOPPED; 625 } else { 626 // For fast tracks prepareTracks_l() will set state to STOPPING_2 627 // presentation is complete 628 // For an offloaded track this starts a drain and state will 629 // move to STOPPING_2 when drain completes and then STOPPED 630 mState = STOPPING_1; 631 } 632 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 633 playbackThread); 634 } 635 } 636} 637 638void AudioFlinger::PlaybackThread::Track::pause() 639{ 640 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 641 sp<ThreadBase> thread = mThread.promote(); 642 if (thread != 0) { 643 Mutex::Autolock _l(thread->mLock); 644 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 645 switch (mState) { 646 case STOPPING_1: 647 case STOPPING_2: 648 if (!isOffloaded()) { 649 /* nothing to do if track is not offloaded */ 650 break; 651 } 652 653 // Offloaded track was draining, we need to carry on draining when resumed 654 mResumeToStopping = true; 655 // fall through... 656 case ACTIVE: 657 case RESUMING: 658 mState = PAUSING; 659 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 660 playbackThread->broadcast_l(); 661 break; 662 663 default: 664 break; 665 } 666 } 667} 668 669void AudioFlinger::PlaybackThread::Track::flush() 670{ 671 ALOGV("flush(%d)", mName); 672 sp<ThreadBase> thread = mThread.promote(); 673 if (thread != 0) { 674 Mutex::Autolock _l(thread->mLock); 675 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 676 677 if (isOffloaded()) { 678 // If offloaded we allow flush during any state except terminated 679 // and keep the track active to avoid problems if user is seeking 680 // rapidly and underlying hardware has a significant delay handling 681 // a pause 682 if (isTerminated()) { 683 return; 684 } 685 686 ALOGV("flush: offload flush"); 687 reset(); 688 689 if (mState == STOPPING_1 || mState == STOPPING_2) { 690 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 691 mState = ACTIVE; 692 } 693 694 if (mState == ACTIVE) { 695 ALOGV("flush called in active state, resetting buffer time out retry count"); 696 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 697 } 698 699 mResumeToStopping = false; 700 } else { 701 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 702 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 703 return; 704 } 705 // No point remaining in PAUSED state after a flush => go to 706 // FLUSHED state 707 mState = FLUSHED; 708 // do not reset the track if it is still in the process of being stopped or paused. 709 // this will be done by prepareTracks_l() when the track is stopped. 710 // prepareTracks_l() will see mState == FLUSHED, then 711 // remove from active track list, reset(), and trigger presentation complete 712 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 713 reset(); 714 } 715 } 716 // Prevent flush being lost if the track is flushed and then resumed 717 // before mixer thread can run. This is important when offloading 718 // because the hardware buffer could hold a large amount of audio 719 playbackThread->flushOutput_l(); 720 playbackThread->broadcast_l(); 721 } 722} 723 724void AudioFlinger::PlaybackThread::Track::reset() 725{ 726 // Do not reset twice to avoid discarding data written just after a flush and before 727 // the audioflinger thread detects the track is stopped. 728 if (!mResetDone) { 729 // Force underrun condition to avoid false underrun callback until first data is 730 // written to buffer 731 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 732 mFillingUpStatus = FS_FILLING; 733 mResetDone = true; 734 if (mState == FLUSHED) { 735 mState = IDLE; 736 } 737 } 738} 739 740status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 741{ 742 sp<ThreadBase> thread = mThread.promote(); 743 if (thread == 0) { 744 ALOGE("thread is dead"); 745 return FAILED_TRANSACTION; 746 } else if ((thread->type() == ThreadBase::DIRECT) || 747 (thread->type() == ThreadBase::OFFLOAD)) { 748 return thread->setParameters(keyValuePairs); 749 } else { 750 return PERMISSION_DENIED; 751 } 752} 753 754status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp) 755{ 756 // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant 757 if (isFastTrack()) { 758 return INVALID_OPERATION; 759 } 760 sp<ThreadBase> thread = mThread.promote(); 761 if (thread == 0) { 762 return INVALID_OPERATION; 763 } 764 Mutex::Autolock _l(thread->mLock); 765 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 766 if (!isOffloaded()) { 767 if (!playbackThread->mLatchQValid) { 768 return INVALID_OPERATION; 769 } 770 uint32_t unpresentedFrames = 771 ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) / 772 playbackThread->mSampleRate; 773 uint32_t framesWritten = mAudioTrackServerProxy->framesReleased(); 774 if (framesWritten < unpresentedFrames) { 775 return INVALID_OPERATION; 776 } 777 timestamp.mPosition = framesWritten - unpresentedFrames; 778 timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime; 779 return NO_ERROR; 780 } 781 782 return playbackThread->getTimestamp_l(timestamp); 783} 784 785status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 786{ 787 status_t status = DEAD_OBJECT; 788 sp<ThreadBase> thread = mThread.promote(); 789 if (thread != 0) { 790 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 791 sp<AudioFlinger> af = mClient->audioFlinger(); 792 793 Mutex::Autolock _l(af->mLock); 794 795 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 796 797 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 798 Mutex::Autolock _dl(playbackThread->mLock); 799 Mutex::Autolock _sl(srcThread->mLock); 800 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 801 if (chain == 0) { 802 return INVALID_OPERATION; 803 } 804 805 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 806 if (effect == 0) { 807 return INVALID_OPERATION; 808 } 809 srcThread->removeEffect_l(effect); 810 status = playbackThread->addEffect_l(effect); 811 if (status != NO_ERROR) { 812 srcThread->addEffect_l(effect); 813 return INVALID_OPERATION; 814 } 815 // removeEffect_l() has stopped the effect if it was active so it must be restarted 816 if (effect->state() == EffectModule::ACTIVE || 817 effect->state() == EffectModule::STOPPING) { 818 effect->start(); 819 } 820 821 sp<EffectChain> dstChain = effect->chain().promote(); 822 if (dstChain == 0) { 823 srcThread->addEffect_l(effect); 824 return INVALID_OPERATION; 825 } 826 AudioSystem::unregisterEffect(effect->id()); 827 AudioSystem::registerEffect(&effect->desc(), 828 srcThread->id(), 829 dstChain->strategy(), 830 AUDIO_SESSION_OUTPUT_MIX, 831 effect->id()); 832 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 833 } 834 status = playbackThread->attachAuxEffect(this, EffectId); 835 } 836 return status; 837} 838 839void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 840{ 841 mAuxEffectId = EffectId; 842 mAuxBuffer = buffer; 843} 844 845bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 846 size_t audioHalFrames) 847{ 848 // a track is considered presented when the total number of frames written to audio HAL 849 // corresponds to the number of frames written when presentationComplete() is called for the 850 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 851 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 852 // to detect when all frames have been played. In this case framesWritten isn't 853 // useful because it doesn't always reflect whether there is data in the h/w 854 // buffers, particularly if a track has been paused and resumed during draining 855 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 856 mPresentationCompleteFrames, framesWritten); 857 if (mPresentationCompleteFrames == 0) { 858 mPresentationCompleteFrames = framesWritten + audioHalFrames; 859 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 860 mPresentationCompleteFrames, audioHalFrames); 861 } 862 863 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 864 ALOGV("presentationComplete() session %d complete: framesWritten %d", 865 mSessionId, framesWritten); 866 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 867 mAudioTrackServerProxy->setStreamEndDone(); 868 return true; 869 } 870 return false; 871} 872 873void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 874{ 875 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 876 if (mSyncEvents[i]->type() == type) { 877 mSyncEvents[i]->trigger(); 878 mSyncEvents.removeAt(i); 879 i--; 880 } 881 } 882} 883 884// implement VolumeBufferProvider interface 885 886uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 887{ 888 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 889 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 890 uint32_t vlr = mAudioTrackServerProxy->getVolumeLR(); 891 uint32_t vl = vlr & 0xFFFF; 892 uint32_t vr = vlr >> 16; 893 // track volumes come from shared memory, so can't be trusted and must be clamped 894 if (vl > MAX_GAIN_INT) { 895 vl = MAX_GAIN_INT; 896 } 897 if (vr > MAX_GAIN_INT) { 898 vr = MAX_GAIN_INT; 899 } 900 // now apply the cached master volume and stream type volume; 901 // this is trusted but lacks any synchronization or barrier so may be stale 902 float v = mCachedVolume; 903 vl *= v; 904 vr *= v; 905 // re-combine into U4.16 906 vlr = (vr << 16) | (vl & 0xFFFF); 907 // FIXME look at mute, pause, and stop flags 908 return vlr; 909} 910 911status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 912{ 913 if (isTerminated() || mState == PAUSED || 914 ((framesReady() == 0) && ((mSharedBuffer != 0) || 915 (mState == STOPPED)))) { 916 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 917 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 918 event->cancel(); 919 return INVALID_OPERATION; 920 } 921 (void) TrackBase::setSyncEvent(event); 922 return NO_ERROR; 923} 924 925void AudioFlinger::PlaybackThread::Track::invalidate() 926{ 927 // FIXME should use proxy, and needs work 928 audio_track_cblk_t* cblk = mCblk; 929 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 930 android_atomic_release_store(0x40000000, &cblk->mFutex); 931 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 932 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 933 mIsInvalid = true; 934} 935 936void AudioFlinger::PlaybackThread::Track::signal() 937{ 938 sp<ThreadBase> thread = mThread.promote(); 939 if (thread != 0) { 940 PlaybackThread *t = (PlaybackThread *)thread.get(); 941 Mutex::Autolock _l(t->mLock); 942 t->broadcast_l(); 943 } 944} 945 946// ---------------------------------------------------------------------------- 947 948sp<AudioFlinger::PlaybackThread::TimedTrack> 949AudioFlinger::PlaybackThread::TimedTrack::create( 950 PlaybackThread *thread, 951 const sp<Client>& client, 952 audio_stream_type_t streamType, 953 uint32_t sampleRate, 954 audio_format_t format, 955 audio_channel_mask_t channelMask, 956 size_t frameCount, 957 const sp<IMemory>& sharedBuffer, 958 int sessionId) { 959 if (!client->reserveTimedTrack()) 960 return 0; 961 962 return new TimedTrack( 963 thread, client, streamType, sampleRate, format, channelMask, frameCount, 964 sharedBuffer, sessionId); 965} 966 967AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 968 PlaybackThread *thread, 969 const sp<Client>& client, 970 audio_stream_type_t streamType, 971 uint32_t sampleRate, 972 audio_format_t format, 973 audio_channel_mask_t channelMask, 974 size_t frameCount, 975 const sp<IMemory>& sharedBuffer, 976 int sessionId) 977 : Track(thread, client, streamType, sampleRate, format, channelMask, 978 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 979 mQueueHeadInFlight(false), 980 mTrimQueueHeadOnRelease(false), 981 mFramesPendingInQueue(0), 982 mTimedSilenceBuffer(NULL), 983 mTimedSilenceBufferSize(0), 984 mTimedAudioOutputOnTime(false), 985 mMediaTimeTransformValid(false) 986{ 987 LocalClock lc; 988 mLocalTimeFreq = lc.getLocalFreq(); 989 990 mLocalTimeToSampleTransform.a_zero = 0; 991 mLocalTimeToSampleTransform.b_zero = 0; 992 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 993 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 994 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 995 &mLocalTimeToSampleTransform.a_to_b_denom); 996 997 mMediaTimeToSampleTransform.a_zero = 0; 998 mMediaTimeToSampleTransform.b_zero = 0; 999 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 1000 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 1001 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 1002 &mMediaTimeToSampleTransform.a_to_b_denom); 1003} 1004 1005AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 1006 mClient->releaseTimedTrack(); 1007 delete [] mTimedSilenceBuffer; 1008} 1009 1010status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 1011 size_t size, sp<IMemory>* buffer) { 1012 1013 Mutex::Autolock _l(mTimedBufferQueueLock); 1014 1015 trimTimedBufferQueue_l(); 1016 1017 // lazily initialize the shared memory heap for timed buffers 1018 if (mTimedMemoryDealer == NULL) { 1019 const int kTimedBufferHeapSize = 512 << 10; 1020 1021 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 1022 "AudioFlingerTimed"); 1023 if (mTimedMemoryDealer == NULL) 1024 return NO_MEMORY; 1025 } 1026 1027 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 1028 if (newBuffer == NULL) { 1029 newBuffer = mTimedMemoryDealer->allocate(size); 1030 if (newBuffer == NULL) 1031 return NO_MEMORY; 1032 } 1033 1034 *buffer = newBuffer; 1035 return NO_ERROR; 1036} 1037 1038// caller must hold mTimedBufferQueueLock 1039void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 1040 int64_t mediaTimeNow; 1041 { 1042 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1043 if (!mMediaTimeTransformValid) 1044 return; 1045 1046 int64_t targetTimeNow; 1047 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 1048 ? mCCHelper.getCommonTime(&targetTimeNow) 1049 : mCCHelper.getLocalTime(&targetTimeNow); 1050 1051 if (OK != res) 1052 return; 1053 1054 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 1055 &mediaTimeNow)) { 1056 return; 1057 } 1058 } 1059 1060 size_t trimEnd; 1061 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 1062 int64_t bufEnd; 1063 1064 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 1065 // We have a next buffer. Just use its PTS as the PTS of the frame 1066 // following the last frame in this buffer. If the stream is sparse 1067 // (ie, there are deliberate gaps left in the stream which should be 1068 // filled with silence by the TimedAudioTrack), then this can result 1069 // in one extra buffer being left un-trimmed when it could have 1070 // been. In general, this is not typical, and we would rather 1071 // optimized away the TS calculation below for the more common case 1072 // where PTSes are contiguous. 1073 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 1074 } else { 1075 // We have no next buffer. Compute the PTS of the frame following 1076 // the last frame in this buffer by computing the duration of of 1077 // this frame in media time units and adding it to the PTS of the 1078 // buffer. 1079 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1080 / mFrameSize; 1081 1082 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1083 &bufEnd)) { 1084 ALOGE("Failed to convert frame count of %lld to media time" 1085 " duration" " (scale factor %d/%u) in %s", 1086 frameCount, 1087 mMediaTimeToSampleTransform.a_to_b_numer, 1088 mMediaTimeToSampleTransform.a_to_b_denom, 1089 __PRETTY_FUNCTION__); 1090 break; 1091 } 1092 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1093 } 1094 1095 if (bufEnd > mediaTimeNow) 1096 break; 1097 1098 // Is the buffer we want to use in the middle of a mix operation right 1099 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1100 // from the mixer which should be coming back shortly. 1101 if (!trimEnd && mQueueHeadInFlight) { 1102 mTrimQueueHeadOnRelease = true; 1103 } 1104 } 1105 1106 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1107 if (trimStart < trimEnd) { 1108 // Update the bookkeeping for framesReady() 1109 for (size_t i = trimStart; i < trimEnd; ++i) { 1110 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1111 } 1112 1113 // Now actually remove the buffers from the queue. 1114 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1115 } 1116} 1117 1118void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1119 const char* logTag) { 1120 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1121 "%s called (reason \"%s\"), but timed buffer queue has no" 1122 " elements to trim.", __FUNCTION__, logTag); 1123 1124 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1125 mTimedBufferQueue.removeAt(0); 1126} 1127 1128void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1129 const TimedBuffer& buf, 1130 const char* logTag) { 1131 uint32_t bufBytes = buf.buffer()->size(); 1132 uint32_t consumedAlready = buf.position(); 1133 1134 ALOG_ASSERT(consumedAlready <= bufBytes, 1135 "Bad bookkeeping while updating frames pending. Timed buffer is" 1136 " only %u bytes long, but claims to have consumed %u" 1137 " bytes. (update reason: \"%s\")", 1138 bufBytes, consumedAlready, logTag); 1139 1140 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1141 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1142 "Bad bookkeeping while updating frames pending. Should have at" 1143 " least %u queued frames, but we think we have only %u. (update" 1144 " reason: \"%s\")", 1145 bufFrames, mFramesPendingInQueue, logTag); 1146 1147 mFramesPendingInQueue -= bufFrames; 1148} 1149 1150status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1151 const sp<IMemory>& buffer, int64_t pts) { 1152 1153 { 1154 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1155 if (!mMediaTimeTransformValid) 1156 return INVALID_OPERATION; 1157 } 1158 1159 Mutex::Autolock _l(mTimedBufferQueueLock); 1160 1161 uint32_t bufFrames = buffer->size() / mFrameSize; 1162 mFramesPendingInQueue += bufFrames; 1163 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1164 1165 return NO_ERROR; 1166} 1167 1168status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1169 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1170 1171 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1172 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1173 target); 1174 1175 if (!(target == TimedAudioTrack::LOCAL_TIME || 1176 target == TimedAudioTrack::COMMON_TIME)) { 1177 return BAD_VALUE; 1178 } 1179 1180 Mutex::Autolock lock(mMediaTimeTransformLock); 1181 mMediaTimeTransform = xform; 1182 mMediaTimeTransformTarget = target; 1183 mMediaTimeTransformValid = true; 1184 1185 return NO_ERROR; 1186} 1187 1188#define min(a, b) ((a) < (b) ? (a) : (b)) 1189 1190// implementation of getNextBuffer for tracks whose buffers have timestamps 1191status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1192 AudioBufferProvider::Buffer* buffer, int64_t pts) 1193{ 1194 if (pts == AudioBufferProvider::kInvalidPTS) { 1195 buffer->raw = NULL; 1196 buffer->frameCount = 0; 1197 mTimedAudioOutputOnTime = false; 1198 return INVALID_OPERATION; 1199 } 1200 1201 Mutex::Autolock _l(mTimedBufferQueueLock); 1202 1203 ALOG_ASSERT(!mQueueHeadInFlight, 1204 "getNextBuffer called without releaseBuffer!"); 1205 1206 while (true) { 1207 1208 // if we have no timed buffers, then fail 1209 if (mTimedBufferQueue.isEmpty()) { 1210 buffer->raw = NULL; 1211 buffer->frameCount = 0; 1212 return NOT_ENOUGH_DATA; 1213 } 1214 1215 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1216 1217 // calculate the PTS of the head of the timed buffer queue expressed in 1218 // local time 1219 int64_t headLocalPTS; 1220 { 1221 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1222 1223 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1224 1225 if (mMediaTimeTransform.a_to_b_denom == 0) { 1226 // the transform represents a pause, so yield silence 1227 timedYieldSilence_l(buffer->frameCount, buffer); 1228 return NO_ERROR; 1229 } 1230 1231 int64_t transformedPTS; 1232 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1233 &transformedPTS)) { 1234 // the transform failed. this shouldn't happen, but if it does 1235 // then just drop this buffer 1236 ALOGW("timedGetNextBuffer transform failed"); 1237 buffer->raw = NULL; 1238 buffer->frameCount = 0; 1239 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1240 return NO_ERROR; 1241 } 1242 1243 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1244 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1245 &headLocalPTS)) { 1246 buffer->raw = NULL; 1247 buffer->frameCount = 0; 1248 return INVALID_OPERATION; 1249 } 1250 } else { 1251 headLocalPTS = transformedPTS; 1252 } 1253 } 1254 1255 uint32_t sr = sampleRate(); 1256 1257 // adjust the head buffer's PTS to reflect the portion of the head buffer 1258 // that has already been consumed 1259 int64_t effectivePTS = headLocalPTS + 1260 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1261 1262 // Calculate the delta in samples between the head of the input buffer 1263 // queue and the start of the next output buffer that will be written. 1264 // If the transformation fails because of over or underflow, it means 1265 // that the sample's position in the output stream is so far out of 1266 // whack that it should just be dropped. 1267 int64_t sampleDelta; 1268 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1269 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1270 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1271 " mix"); 1272 continue; 1273 } 1274 if (!mLocalTimeToSampleTransform.doForwardTransform( 1275 (effectivePTS - pts) << 32, &sampleDelta)) { 1276 ALOGV("*** too late during sample rate transform: dropped buffer"); 1277 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1278 continue; 1279 } 1280 1281 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1282 " sampleDelta=[%d.%08x]", 1283 head.pts(), head.position(), pts, 1284 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1285 + (sampleDelta >> 32)), 1286 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1287 1288 // if the delta between the ideal placement for the next input sample and 1289 // the current output position is within this threshold, then we will 1290 // concatenate the next input samples to the previous output 1291 const int64_t kSampleContinuityThreshold = 1292 (static_cast<int64_t>(sr) << 32) / 250; 1293 1294 // if this is the first buffer of audio that we're emitting from this track 1295 // then it should be almost exactly on time. 1296 const int64_t kSampleStartupThreshold = 1LL << 32; 1297 1298 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1299 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1300 // the next input is close enough to being on time, so concatenate it 1301 // with the last output 1302 timedYieldSamples_l(buffer); 1303 1304 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1305 head.position(), buffer->frameCount); 1306 return NO_ERROR; 1307 } 1308 1309 // Looks like our output is not on time. Reset our on timed status. 1310 // Next time we mix samples from our input queue, then should be within 1311 // the StartupThreshold. 1312 mTimedAudioOutputOnTime = false; 1313 if (sampleDelta > 0) { 1314 // the gap between the current output position and the proper start of 1315 // the next input sample is too big, so fill it with silence 1316 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1317 1318 timedYieldSilence_l(framesUntilNextInput, buffer); 1319 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1320 return NO_ERROR; 1321 } else { 1322 // the next input sample is late 1323 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1324 size_t onTimeSamplePosition = 1325 head.position() + lateFrames * mFrameSize; 1326 1327 if (onTimeSamplePosition > head.buffer()->size()) { 1328 // all the remaining samples in the head are too late, so 1329 // drop it and move on 1330 ALOGV("*** too late: dropped buffer"); 1331 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1332 continue; 1333 } else { 1334 // skip over the late samples 1335 head.setPosition(onTimeSamplePosition); 1336 1337 // yield the available samples 1338 timedYieldSamples_l(buffer); 1339 1340 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1341 return NO_ERROR; 1342 } 1343 } 1344 } 1345} 1346 1347// Yield samples from the timed buffer queue head up to the given output 1348// buffer's capacity. 1349// 1350// Caller must hold mTimedBufferQueueLock 1351void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1352 AudioBufferProvider::Buffer* buffer) { 1353 1354 const TimedBuffer& head = mTimedBufferQueue[0]; 1355 1356 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1357 head.position()); 1358 1359 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1360 mFrameSize); 1361 size_t framesRequested = buffer->frameCount; 1362 buffer->frameCount = min(framesLeftInHead, framesRequested); 1363 1364 mQueueHeadInFlight = true; 1365 mTimedAudioOutputOnTime = true; 1366} 1367 1368// Yield samples of silence up to the given output buffer's capacity 1369// 1370// Caller must hold mTimedBufferQueueLock 1371void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1372 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1373 1374 // lazily allocate a buffer filled with silence 1375 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1376 delete [] mTimedSilenceBuffer; 1377 mTimedSilenceBufferSize = numFrames * mFrameSize; 1378 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1379 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1380 } 1381 1382 buffer->raw = mTimedSilenceBuffer; 1383 size_t framesRequested = buffer->frameCount; 1384 buffer->frameCount = min(numFrames, framesRequested); 1385 1386 mTimedAudioOutputOnTime = false; 1387} 1388 1389// AudioBufferProvider interface 1390void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1391 AudioBufferProvider::Buffer* buffer) { 1392 1393 Mutex::Autolock _l(mTimedBufferQueueLock); 1394 1395 // If the buffer which was just released is part of the buffer at the head 1396 // of the queue, be sure to update the amt of the buffer which has been 1397 // consumed. If the buffer being returned is not part of the head of the 1398 // queue, its either because the buffer is part of the silence buffer, or 1399 // because the head of the timed queue was trimmed after the mixer called 1400 // getNextBuffer but before the mixer called releaseBuffer. 1401 if (buffer->raw == mTimedSilenceBuffer) { 1402 ALOG_ASSERT(!mQueueHeadInFlight, 1403 "Queue head in flight during release of silence buffer!"); 1404 goto done; 1405 } 1406 1407 ALOG_ASSERT(mQueueHeadInFlight, 1408 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1409 " head in flight."); 1410 1411 if (mTimedBufferQueue.size()) { 1412 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1413 1414 void* start = head.buffer()->pointer(); 1415 void* end = reinterpret_cast<void*>( 1416 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1417 + head.buffer()->size()); 1418 1419 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1420 "released buffer not within the head of the timed buffer" 1421 " queue; qHead = [%p, %p], released buffer = %p", 1422 start, end, buffer->raw); 1423 1424 head.setPosition(head.position() + 1425 (buffer->frameCount * mFrameSize)); 1426 mQueueHeadInFlight = false; 1427 1428 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1429 "Bad bookkeeping during releaseBuffer! Should have at" 1430 " least %u queued frames, but we think we have only %u", 1431 buffer->frameCount, mFramesPendingInQueue); 1432 1433 mFramesPendingInQueue -= buffer->frameCount; 1434 1435 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1436 || mTrimQueueHeadOnRelease) { 1437 trimTimedBufferQueueHead_l("releaseBuffer"); 1438 mTrimQueueHeadOnRelease = false; 1439 } 1440 } else { 1441 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1442 " buffers in the timed buffer queue"); 1443 } 1444 1445done: 1446 buffer->raw = 0; 1447 buffer->frameCount = 0; 1448} 1449 1450size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1451 Mutex::Autolock _l(mTimedBufferQueueLock); 1452 return mFramesPendingInQueue; 1453} 1454 1455AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1456 : mPTS(0), mPosition(0) {} 1457 1458AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1459 const sp<IMemory>& buffer, int64_t pts) 1460 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1461 1462 1463// ---------------------------------------------------------------------------- 1464 1465AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1466 PlaybackThread *playbackThread, 1467 DuplicatingThread *sourceThread, 1468 uint32_t sampleRate, 1469 audio_format_t format, 1470 audio_channel_mask_t channelMask, 1471 size_t frameCount) 1472 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1473 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 1474 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1475{ 1476 1477 if (mCblk != NULL) { 1478 mOutBuffer.frameCount = 0; 1479 playbackThread->mTracks.add(this); 1480 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1481 "mCblk->frameCount_ %u, mChannelMask 0x%08x", 1482 mCblk, mBuffer, 1483 mCblk->frameCount_, mChannelMask); 1484 // since client and server are in the same process, 1485 // the buffer has the same virtual address on both sides 1486 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize); 1487 mClientProxy->setVolumeLR((uint32_t(uint16_t(0x1000)) << 16) | uint16_t(0x1000)); 1488 mClientProxy->setSendLevel(0.0); 1489 mClientProxy->setSampleRate(sampleRate); 1490 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1491 true /*clientInServer*/); 1492 } else { 1493 ALOGW("Error creating output track on thread %p", playbackThread); 1494 } 1495} 1496 1497AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1498{ 1499 clearBufferQueue(); 1500 delete mClientProxy; 1501 // superclass destructor will now delete the server proxy and shared memory both refer to 1502} 1503 1504status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1505 int triggerSession) 1506{ 1507 status_t status = Track::start(event, triggerSession); 1508 if (status != NO_ERROR) { 1509 return status; 1510 } 1511 1512 mActive = true; 1513 mRetryCount = 127; 1514 return status; 1515} 1516 1517void AudioFlinger::PlaybackThread::OutputTrack::stop() 1518{ 1519 Track::stop(); 1520 clearBufferQueue(); 1521 mOutBuffer.frameCount = 0; 1522 mActive = false; 1523} 1524 1525bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1526{ 1527 Buffer *pInBuffer; 1528 Buffer inBuffer; 1529 uint32_t channelCount = mChannelCount; 1530 bool outputBufferFull = false; 1531 inBuffer.frameCount = frames; 1532 inBuffer.i16 = data; 1533 1534 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1535 1536 if (!mActive && frames != 0) { 1537 start(); 1538 sp<ThreadBase> thread = mThread.promote(); 1539 if (thread != 0) { 1540 MixerThread *mixerThread = (MixerThread *)thread.get(); 1541 if (mFrameCount > frames) { 1542 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1543 uint32_t startFrames = (mFrameCount - frames); 1544 pInBuffer = new Buffer; 1545 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1546 pInBuffer->frameCount = startFrames; 1547 pInBuffer->i16 = pInBuffer->mBuffer; 1548 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1549 mBufferQueue.add(pInBuffer); 1550 } else { 1551 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1552 } 1553 } 1554 } 1555 } 1556 1557 while (waitTimeLeftMs) { 1558 // First write pending buffers, then new data 1559 if (mBufferQueue.size()) { 1560 pInBuffer = mBufferQueue.itemAt(0); 1561 } else { 1562 pInBuffer = &inBuffer; 1563 } 1564 1565 if (pInBuffer->frameCount == 0) { 1566 break; 1567 } 1568 1569 if (mOutBuffer.frameCount == 0) { 1570 mOutBuffer.frameCount = pInBuffer->frameCount; 1571 nsecs_t startTime = systemTime(); 1572 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1573 if (status != NO_ERROR) { 1574 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1575 mThread.unsafe_get(), status); 1576 outputBufferFull = true; 1577 break; 1578 } 1579 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1580 if (waitTimeLeftMs >= waitTimeMs) { 1581 waitTimeLeftMs -= waitTimeMs; 1582 } else { 1583 waitTimeLeftMs = 0; 1584 } 1585 } 1586 1587 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1588 pInBuffer->frameCount; 1589 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1590 Proxy::Buffer buf; 1591 buf.mFrameCount = outFrames; 1592 buf.mRaw = NULL; 1593 mClientProxy->releaseBuffer(&buf); 1594 pInBuffer->frameCount -= outFrames; 1595 pInBuffer->i16 += outFrames * channelCount; 1596 mOutBuffer.frameCount -= outFrames; 1597 mOutBuffer.i16 += outFrames * channelCount; 1598 1599 if (pInBuffer->frameCount == 0) { 1600 if (mBufferQueue.size()) { 1601 mBufferQueue.removeAt(0); 1602 delete [] pInBuffer->mBuffer; 1603 delete pInBuffer; 1604 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1605 mThread.unsafe_get(), mBufferQueue.size()); 1606 } else { 1607 break; 1608 } 1609 } 1610 } 1611 1612 // If we could not write all frames, allocate a buffer and queue it for next time. 1613 if (inBuffer.frameCount) { 1614 sp<ThreadBase> thread = mThread.promote(); 1615 if (thread != 0 && !thread->standby()) { 1616 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1617 pInBuffer = new Buffer; 1618 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1619 pInBuffer->frameCount = inBuffer.frameCount; 1620 pInBuffer->i16 = pInBuffer->mBuffer; 1621 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1622 sizeof(int16_t)); 1623 mBufferQueue.add(pInBuffer); 1624 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1625 mThread.unsafe_get(), mBufferQueue.size()); 1626 } else { 1627 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1628 mThread.unsafe_get(), this); 1629 } 1630 } 1631 } 1632 1633 // Calling write() with a 0 length buffer, means that no more data will be written: 1634 // If no more buffers are pending, fill output track buffer to make sure it is started 1635 // by output mixer. 1636 if (frames == 0 && mBufferQueue.size() == 0) { 1637 // FIXME borken, replace by getting framesReady() from proxy 1638 size_t user = 0; // was mCblk->user 1639 if (user < mFrameCount) { 1640 frames = mFrameCount - user; 1641 pInBuffer = new Buffer; 1642 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1643 pInBuffer->frameCount = frames; 1644 pInBuffer->i16 = pInBuffer->mBuffer; 1645 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1646 mBufferQueue.add(pInBuffer); 1647 } else if (mActive) { 1648 stop(); 1649 } 1650 } 1651 1652 return outputBufferFull; 1653} 1654 1655status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1656 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1657{ 1658 ClientProxy::Buffer buf; 1659 buf.mFrameCount = buffer->frameCount; 1660 struct timespec timeout; 1661 timeout.tv_sec = waitTimeMs / 1000; 1662 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1663 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1664 buffer->frameCount = buf.mFrameCount; 1665 buffer->raw = buf.mRaw; 1666 return status; 1667} 1668 1669void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1670{ 1671 size_t size = mBufferQueue.size(); 1672 1673 for (size_t i = 0; i < size; i++) { 1674 Buffer *pBuffer = mBufferQueue.itemAt(i); 1675 delete [] pBuffer->mBuffer; 1676 delete pBuffer; 1677 } 1678 mBufferQueue.clear(); 1679} 1680 1681 1682// ---------------------------------------------------------------------------- 1683// Record 1684// ---------------------------------------------------------------------------- 1685 1686AudioFlinger::RecordHandle::RecordHandle( 1687 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1688 : BnAudioRecord(), 1689 mRecordTrack(recordTrack) 1690{ 1691} 1692 1693AudioFlinger::RecordHandle::~RecordHandle() { 1694 stop_nonvirtual(); 1695 mRecordTrack->destroy(); 1696} 1697 1698sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 1699 return mRecordTrack->getCblk(); 1700} 1701 1702status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1703 int triggerSession) { 1704 ALOGV("RecordHandle::start()"); 1705 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1706} 1707 1708void AudioFlinger::RecordHandle::stop() { 1709 stop_nonvirtual(); 1710} 1711 1712void AudioFlinger::RecordHandle::stop_nonvirtual() { 1713 ALOGV("RecordHandle::stop()"); 1714 mRecordTrack->stop(); 1715} 1716 1717status_t AudioFlinger::RecordHandle::onTransact( 1718 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1719{ 1720 return BnAudioRecord::onTransact(code, data, reply, flags); 1721} 1722 1723// ---------------------------------------------------------------------------- 1724 1725// RecordTrack constructor must be called with AudioFlinger::mLock held 1726AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1727 RecordThread *thread, 1728 const sp<Client>& client, 1729 uint32_t sampleRate, 1730 audio_format_t format, 1731 audio_channel_mask_t channelMask, 1732 size_t frameCount, 1733 int sessionId) 1734 : TrackBase(thread, client, sampleRate, format, 1735 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/), 1736 mOverflow(false) 1737{ 1738 ALOGV("RecordTrack constructor"); 1739 if (mCblk != NULL) { 1740 mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, 1741 mFrameSize); 1742 mServerProxy = mAudioRecordServerProxy; 1743 } 1744} 1745 1746AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 1747{ 1748 ALOGV("%s", __func__); 1749} 1750 1751// AudioBufferProvider interface 1752status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 1753 int64_t pts) 1754{ 1755 ServerProxy::Buffer buf; 1756 buf.mFrameCount = buffer->frameCount; 1757 status_t status = mServerProxy->obtainBuffer(&buf); 1758 buffer->frameCount = buf.mFrameCount; 1759 buffer->raw = buf.mRaw; 1760 if (buf.mFrameCount == 0) { 1761 // FIXME also wake futex so that overrun is noticed more quickly 1762 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 1763 } 1764 return status; 1765} 1766 1767status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 1768 int triggerSession) 1769{ 1770 sp<ThreadBase> thread = mThread.promote(); 1771 if (thread != 0) { 1772 RecordThread *recordThread = (RecordThread *)thread.get(); 1773 return recordThread->start(this, event, triggerSession); 1774 } else { 1775 return BAD_VALUE; 1776 } 1777} 1778 1779void AudioFlinger::RecordThread::RecordTrack::stop() 1780{ 1781 sp<ThreadBase> thread = mThread.promote(); 1782 if (thread != 0) { 1783 RecordThread *recordThread = (RecordThread *)thread.get(); 1784 if (recordThread->stop(this)) { 1785 AudioSystem::stopInput(recordThread->id()); 1786 } 1787 } 1788} 1789 1790void AudioFlinger::RecordThread::RecordTrack::destroy() 1791{ 1792 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 1793 sp<RecordTrack> keep(this); 1794 { 1795 sp<ThreadBase> thread = mThread.promote(); 1796 if (thread != 0) { 1797 if (mState == ACTIVE || mState == RESUMING) { 1798 AudioSystem::stopInput(thread->id()); 1799 } 1800 AudioSystem::releaseInput(thread->id()); 1801 Mutex::Autolock _l(thread->mLock); 1802 RecordThread *recordThread = (RecordThread *) thread.get(); 1803 recordThread->destroyTrack_l(this); 1804 } 1805 } 1806} 1807 1808void AudioFlinger::RecordThread::RecordTrack::invalidate() 1809{ 1810 // FIXME should use proxy, and needs work 1811 audio_track_cblk_t* cblk = mCblk; 1812 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1813 android_atomic_release_store(0x40000000, &cblk->mFutex); 1814 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 1815 (void) __futex_syscall3(&cblk->mFutex, FUTEX_WAKE, INT_MAX); 1816} 1817 1818 1819/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 1820{ 1821 result.append("Client Fmt Chn mask Session S Server fCount\n"); 1822} 1823 1824void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 1825{ 1826 snprintf(buffer, size, "%6u %3u %08X %7u %1d %08X %6u\n", 1827 (mClient == 0) ? getpid_cached : mClient->pid(), 1828 mFormat, 1829 mChannelMask, 1830 mSessionId, 1831 mState, 1832 mCblk->mServer, 1833 mFrameCount); 1834} 1835 1836}; // namespace android 1837