Tracks.cpp revision 4dc680607181e6a76f4e91a39366c4f5dfb7b03e
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include "Configuration.h" 23#include <math.h> 24#include <sys/syscall.h> 25#include <utils/Log.h> 26 27#include <private/media/AudioTrackShared.h> 28 29#include <common_time/cc_helper.h> 30#include <common_time/local_clock.h> 31 32#include "AudioMixer.h" 33#include "AudioFlinger.h" 34#include "ServiceUtilities.h" 35 36#include <media/nbaio/Pipe.h> 37#include <media/nbaio/PipeReader.h> 38#include <audio_utils/minifloat.h> 39 40// ---------------------------------------------------------------------------- 41 42// Note: the following macro is used for extremely verbose logging message. In 43// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 44// 0; but one side effect of this is to turn all LOGV's as well. Some messages 45// are so verbose that we want to suppress them even when we have ALOG_ASSERT 46// turned on. Do not uncomment the #def below unless you really know what you 47// are doing and want to see all of the extremely verbose messages. 48//#define VERY_VERY_VERBOSE_LOGGING 49#ifdef VERY_VERY_VERBOSE_LOGGING 50#define ALOGVV ALOGV 51#else 52#define ALOGVV(a...) do { } while(0) 53#endif 54 55namespace android { 56 57// ---------------------------------------------------------------------------- 58// TrackBase 59// ---------------------------------------------------------------------------- 60 61static volatile int32_t nextTrackId = 55; 62 63// TrackBase constructor must be called with AudioFlinger::mLock held 64AudioFlinger::ThreadBase::TrackBase::TrackBase( 65 ThreadBase *thread, 66 const sp<Client>& client, 67 uint32_t sampleRate, 68 audio_format_t format, 69 audio_channel_mask_t channelMask, 70 size_t frameCount, 71 void *buffer, 72 int sessionId, 73 int clientUid, 74 IAudioFlinger::track_flags_t flags, 75 bool isOut, 76 alloc_type alloc, 77 track_type type) 78 : RefBase(), 79 mThread(thread), 80 mClient(client), 81 mCblk(NULL), 82 // mBuffer 83 mState(IDLE), 84 mSampleRate(sampleRate), 85 mFormat(format), 86 mChannelMask(channelMask), 87 mChannelCount(isOut ? 88 audio_channel_count_from_out_mask(channelMask) : 89 audio_channel_count_from_in_mask(channelMask)), 90 mFrameSize(audio_is_linear_pcm(format) ? 91 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 92 mFrameCount(frameCount), 93 mSessionId(sessionId), 94 mFlags(flags), 95 mIsOut(isOut), 96 mServerProxy(NULL), 97 mId(android_atomic_inc(&nextTrackId)), 98 mTerminated(false), 99 mType(type) 100{ 101 // if the caller is us, trust the specified uid 102 if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) { 103 int newclientUid = IPCThreadState::self()->getCallingUid(); 104 if (clientUid != -1 && clientUid != newclientUid) { 105 ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid); 106 } 107 clientUid = newclientUid; 108 } 109 // clientUid contains the uid of the app that is responsible for this track, so we can blame 110 // battery usage on it. 111 mUid = clientUid; 112 113 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 114 size_t size = sizeof(audio_track_cblk_t); 115 size_t bufferSize = (buffer == NULL ? roundup(frameCount) : frameCount) * mFrameSize; 116 if (buffer == NULL && alloc == ALLOC_CBLK) { 117 size += bufferSize; 118 } 119 120 if (client != 0) { 121 mCblkMemory = client->heap()->allocate(size); 122 if (mCblkMemory == 0 || 123 (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) { 124 ALOGE("not enough memory for AudioTrack size=%u", size); 125 client->heap()->dump("AudioTrack"); 126 mCblkMemory.clear(); 127 return; 128 } 129 } else { 130 // this syntax avoids calling the audio_track_cblk_t constructor twice 131 mCblk = (audio_track_cblk_t *) new uint8_t[size]; 132 // assume mCblk != NULL 133 } 134 135 // construct the shared structure in-place. 136 if (mCblk != NULL) { 137 new(mCblk) audio_track_cblk_t(); 138 switch (alloc) { 139 case ALLOC_READONLY: { 140 const sp<MemoryDealer> roHeap(thread->readOnlyHeap()); 141 if (roHeap == 0 || 142 (mBufferMemory = roHeap->allocate(bufferSize)) == 0 || 143 (mBuffer = mBufferMemory->pointer()) == NULL) { 144 ALOGE("not enough memory for read-only buffer size=%zu", bufferSize); 145 if (roHeap != 0) { 146 roHeap->dump("buffer"); 147 } 148 mCblkMemory.clear(); 149 mBufferMemory.clear(); 150 return; 151 } 152 memset(mBuffer, 0, bufferSize); 153 } break; 154 case ALLOC_PIPE: 155 mBufferMemory = thread->pipeMemory(); 156 // mBuffer is the virtual address as seen from current process (mediaserver), 157 // and should normally be coming from mBufferMemory->pointer(). 158 // However in this case the TrackBase does not reference the buffer directly. 159 // It should references the buffer via the pipe. 160 // Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL. 161 mBuffer = NULL; 162 break; 163 case ALLOC_CBLK: 164 // clear all buffers 165 if (buffer == NULL) { 166 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 167 memset(mBuffer, 0, bufferSize); 168 } else { 169 mBuffer = buffer; 170#if 0 171 mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic 172#endif 173 } 174 break; 175 case ALLOC_LOCAL: 176 mBuffer = calloc(1, bufferSize); 177 break; 178 case ALLOC_NONE: 179 mBuffer = buffer; 180 break; 181 } 182 183#ifdef TEE_SINK 184 if (mTeeSinkTrackEnabled) { 185 NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount); 186 if (Format_isValid(pipeFormat)) { 187 Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat); 188 size_t numCounterOffers = 0; 189 const NBAIO_Format offers[1] = {pipeFormat}; 190 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 191 ALOG_ASSERT(index == 0); 192 PipeReader *pipeReader = new PipeReader(*pipe); 193 numCounterOffers = 0; 194 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 195 ALOG_ASSERT(index == 0); 196 mTeeSink = pipe; 197 mTeeSource = pipeReader; 198 } 199 } 200#endif 201 202 } 203} 204 205status_t AudioFlinger::ThreadBase::TrackBase::initCheck() const 206{ 207 status_t status; 208 if (mType == TYPE_OUTPUT || mType == TYPE_PATCH) { 209 status = cblk() != NULL ? NO_ERROR : NO_MEMORY; 210 } else { 211 status = getCblk() != 0 ? NO_ERROR : NO_MEMORY; 212 } 213 return status; 214} 215 216AudioFlinger::ThreadBase::TrackBase::~TrackBase() 217{ 218#ifdef TEE_SINK 219 dumpTee(-1, mTeeSource, mId); 220#endif 221 // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference 222 delete mServerProxy; 223 if (mCblk != NULL) { 224 if (mClient == 0) { 225 delete mCblk; 226 } else { 227 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 228 } 229 } 230 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 231 if (mClient != 0) { 232 // Client destructor must run with AudioFlinger client mutex locked 233 Mutex::Autolock _l(mClient->audioFlinger()->mClientLock); 234 // If the client's reference count drops to zero, the associated destructor 235 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 236 // relying on the automatic clear() at end of scope. 237 mClient.clear(); 238 } 239 // flush the binder command buffer 240 IPCThreadState::self()->flushCommands(); 241} 242 243// AudioBufferProvider interface 244// getNextBuffer() = 0; 245// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 246void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 247{ 248#ifdef TEE_SINK 249 if (mTeeSink != 0) { 250 (void) mTeeSink->write(buffer->raw, buffer->frameCount); 251 } 252#endif 253 254 ServerProxy::Buffer buf; 255 buf.mFrameCount = buffer->frameCount; 256 buf.mRaw = buffer->raw; 257 buffer->frameCount = 0; 258 buffer->raw = NULL; 259 mServerProxy->releaseBuffer(&buf); 260} 261 262status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 263{ 264 mSyncEvents.add(event); 265 return NO_ERROR; 266} 267 268// ---------------------------------------------------------------------------- 269// Playback 270// ---------------------------------------------------------------------------- 271 272AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 273 : BnAudioTrack(), 274 mTrack(track) 275{ 276} 277 278AudioFlinger::TrackHandle::~TrackHandle() { 279 // just stop the track on deletion, associated resources 280 // will be freed from the main thread once all pending buffers have 281 // been played. Unless it's not in the active track list, in which 282 // case we free everything now... 283 mTrack->destroy(); 284} 285 286sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 287 return mTrack->getCblk(); 288} 289 290status_t AudioFlinger::TrackHandle::start() { 291 return mTrack->start(); 292} 293 294void AudioFlinger::TrackHandle::stop() { 295 mTrack->stop(); 296} 297 298void AudioFlinger::TrackHandle::flush() { 299 mTrack->flush(); 300} 301 302void AudioFlinger::TrackHandle::pause() { 303 mTrack->pause(); 304} 305 306status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 307{ 308 return mTrack->attachAuxEffect(EffectId); 309} 310 311status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 312 sp<IMemory>* buffer) { 313 if (!mTrack->isTimedTrack()) 314 return INVALID_OPERATION; 315 316 PlaybackThread::TimedTrack* tt = 317 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 318 return tt->allocateTimedBuffer(size, buffer); 319} 320 321status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 322 int64_t pts) { 323 if (!mTrack->isTimedTrack()) 324 return INVALID_OPERATION; 325 326 if (buffer == 0 || buffer->pointer() == NULL) { 327 ALOGE("queueTimedBuffer() buffer is 0 or has NULL pointer()"); 328 return BAD_VALUE; 329 } 330 331 PlaybackThread::TimedTrack* tt = 332 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 333 return tt->queueTimedBuffer(buffer, pts); 334} 335 336status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 337 const LinearTransform& xform, int target) { 338 339 if (!mTrack->isTimedTrack()) 340 return INVALID_OPERATION; 341 342 PlaybackThread::TimedTrack* tt = 343 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 344 return tt->setMediaTimeTransform( 345 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 346} 347 348status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) { 349 return mTrack->setParameters(keyValuePairs); 350} 351 352status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp) 353{ 354 return mTrack->getTimestamp(timestamp); 355} 356 357 358void AudioFlinger::TrackHandle::signal() 359{ 360 return mTrack->signal(); 361} 362 363status_t AudioFlinger::TrackHandle::onTransact( 364 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 365{ 366 return BnAudioTrack::onTransact(code, data, reply, flags); 367} 368 369// ---------------------------------------------------------------------------- 370 371// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 372AudioFlinger::PlaybackThread::Track::Track( 373 PlaybackThread *thread, 374 const sp<Client>& client, 375 audio_stream_type_t streamType, 376 uint32_t sampleRate, 377 audio_format_t format, 378 audio_channel_mask_t channelMask, 379 size_t frameCount, 380 void *buffer, 381 const sp<IMemory>& sharedBuffer, 382 int sessionId, 383 int uid, 384 IAudioFlinger::track_flags_t flags, 385 track_type type) 386 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 387 (sharedBuffer != 0) ? sharedBuffer->pointer() : buffer, 388 sessionId, uid, flags, true /*isOut*/, 389 (type == TYPE_PATCH) ? ( buffer == NULL ? ALLOC_LOCAL : ALLOC_NONE) : ALLOC_CBLK, 390 type), 391 mFillingUpStatus(FS_INVALID), 392 // mRetryCount initialized later when needed 393 mSharedBuffer(sharedBuffer), 394 mStreamType(streamType), 395 mName(-1), // see note below 396 mMainBuffer(thread->mixBuffer()), 397 mAuxBuffer(NULL), 398 mAuxEffectId(0), mHasVolumeController(false), 399 mPresentationCompleteFrames(0), 400 mFastIndex(-1), 401 mCachedVolume(1.0), 402 mIsInvalid(false), 403 mAudioTrackServerProxy(NULL), 404 mResumeToStopping(false), 405 mFlushHwPending(false), 406 mPreviousValid(false), 407 mPreviousFramesWritten(0) 408 // mPreviousTimestamp 409{ 410 // client == 0 implies sharedBuffer == 0 411 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 412 413 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 414 sharedBuffer->size()); 415 416 if (mCblk == NULL) { 417 return; 418 } 419 420 if (sharedBuffer == 0) { 421 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 422 mFrameSize, !isExternalTrack(), sampleRate); 423 } else { 424 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, 425 mFrameSize); 426 } 427 mServerProxy = mAudioTrackServerProxy; 428 429 mName = thread->getTrackName_l(channelMask, format, sessionId); 430 if (mName < 0) { 431 ALOGE("no more track names available"); 432 return; 433 } 434 // only allocate a fast track index if we were able to allocate a normal track name 435 if (flags & IAudioFlinger::TRACK_FAST) { 436 mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); 437 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 438 int i = __builtin_ctz(thread->mFastTrackAvailMask); 439 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 440 // FIXME This is too eager. We allocate a fast track index before the 441 // fast track becomes active. Since fast tracks are a scarce resource, 442 // this means we are potentially denying other more important fast tracks from 443 // being created. It would be better to allocate the index dynamically. 444 mFastIndex = i; 445 // Read the initial underruns because this field is never cleared by the fast mixer 446 mObservedUnderruns = thread->getFastTrackUnderruns(i); 447 thread->mFastTrackAvailMask &= ~(1 << i); 448 } 449} 450 451AudioFlinger::PlaybackThread::Track::~Track() 452{ 453 ALOGV("PlaybackThread::Track destructor"); 454 455 // The destructor would clear mSharedBuffer, 456 // but it will not push the decremented reference count, 457 // leaving the client's IMemory dangling indefinitely. 458 // This prevents that leak. 459 if (mSharedBuffer != 0) { 460 mSharedBuffer.clear(); 461 } 462} 463 464status_t AudioFlinger::PlaybackThread::Track::initCheck() const 465{ 466 status_t status = TrackBase::initCheck(); 467 if (status == NO_ERROR && mName < 0) { 468 status = NO_MEMORY; 469 } 470 return status; 471} 472 473void AudioFlinger::PlaybackThread::Track::destroy() 474{ 475 // NOTE: destroyTrack_l() can remove a strong reference to this Track 476 // by removing it from mTracks vector, so there is a risk that this Tracks's 477 // destructor is called. As the destructor needs to lock mLock, 478 // we must acquire a strong reference on this Track before locking mLock 479 // here so that the destructor is called only when exiting this function. 480 // On the other hand, as long as Track::destroy() is only called by 481 // TrackHandle destructor, the TrackHandle still holds a strong ref on 482 // this Track with its member mTrack. 483 sp<Track> keep(this); 484 { // scope for mLock 485 sp<ThreadBase> thread = mThread.promote(); 486 if (thread != 0) { 487 Mutex::Autolock _l(thread->mLock); 488 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 489 bool wasActive = playbackThread->destroyTrack_l(this); 490 if (isExternalTrack() && !wasActive) { 491 AudioSystem::releaseOutput(thread->id()); 492 } 493 } 494 } 495} 496 497/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 498{ 499 result.append(" Name Active Client Type Fmt Chn mask Session fCount S F SRate " 500 "L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n"); 501} 502 503void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active) 504{ 505 gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR(); 506 if (isFastTrack()) { 507 sprintf(buffer, " F %2d", mFastIndex); 508 } else if (mName >= AudioMixer::TRACK0) { 509 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 510 } else { 511 sprintf(buffer, " none"); 512 } 513 track_state state = mState; 514 char stateChar; 515 if (isTerminated()) { 516 stateChar = 'T'; 517 } else { 518 switch (state) { 519 case IDLE: 520 stateChar = 'I'; 521 break; 522 case STOPPING_1: 523 stateChar = 's'; 524 break; 525 case STOPPING_2: 526 stateChar = '5'; 527 break; 528 case STOPPED: 529 stateChar = 'S'; 530 break; 531 case RESUMING: 532 stateChar = 'R'; 533 break; 534 case ACTIVE: 535 stateChar = 'A'; 536 break; 537 case PAUSING: 538 stateChar = 'p'; 539 break; 540 case PAUSED: 541 stateChar = 'P'; 542 break; 543 case FLUSHED: 544 stateChar = 'F'; 545 break; 546 default: 547 stateChar = '?'; 548 break; 549 } 550 } 551 char nowInUnderrun; 552 switch (mObservedUnderruns.mBitFields.mMostRecent) { 553 case UNDERRUN_FULL: 554 nowInUnderrun = ' '; 555 break; 556 case UNDERRUN_PARTIAL: 557 nowInUnderrun = '<'; 558 break; 559 case UNDERRUN_EMPTY: 560 nowInUnderrun = '*'; 561 break; 562 default: 563 nowInUnderrun = '?'; 564 break; 565 } 566 snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g " 567 "%08X %p %p 0x%03X %9u%c\n", 568 active ? "yes" : "no", 569 (mClient == 0) ? getpid_cached : mClient->pid(), 570 mStreamType, 571 mFormat, 572 mChannelMask, 573 mSessionId, 574 mFrameCount, 575 stateChar, 576 mFillingUpStatus, 577 mAudioTrackServerProxy->getSampleRate(), 578 20.0 * log10(float_from_gain(gain_minifloat_unpack_left(vlr))), 579 20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))), 580 mCblk->mServer, 581 mMainBuffer, 582 mAuxBuffer, 583 mCblk->mFlags, 584 mAudioTrackServerProxy->getUnderrunFrames(), 585 nowInUnderrun); 586} 587 588uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { 589 return mAudioTrackServerProxy->getSampleRate(); 590} 591 592// AudioBufferProvider interface 593status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 594 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 595{ 596 ServerProxy::Buffer buf; 597 size_t desiredFrames = buffer->frameCount; 598 buf.mFrameCount = desiredFrames; 599 status_t status = mServerProxy->obtainBuffer(&buf); 600 buffer->frameCount = buf.mFrameCount; 601 buffer->raw = buf.mRaw; 602 if (buf.mFrameCount == 0) { 603 mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 604 } 605 return status; 606} 607 608// releaseBuffer() is not overridden 609 610// ExtendedAudioBufferProvider interface 611 612// Note that framesReady() takes a mutex on the control block using tryLock(). 613// This could result in priority inversion if framesReady() is called by the normal mixer, 614// as the normal mixer thread runs at lower 615// priority than the client's callback thread: there is a short window within framesReady() 616// during which the normal mixer could be preempted, and the client callback would block. 617// Another problem can occur if framesReady() is called by the fast mixer: 618// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 619// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 620size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 621 return mAudioTrackServerProxy->framesReady(); 622} 623 624size_t AudioFlinger::PlaybackThread::Track::framesReleased() const 625{ 626 return mAudioTrackServerProxy->framesReleased(); 627} 628 629// Don't call for fast tracks; the framesReady() could result in priority inversion 630bool AudioFlinger::PlaybackThread::Track::isReady() const { 631 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 632 return true; 633 } 634 635 if (isStopping()) { 636 if (framesReady() > 0) { 637 mFillingUpStatus = FS_FILLED; 638 } 639 return true; 640 } 641 642 if (framesReady() >= mFrameCount || 643 (mCblk->mFlags & CBLK_FORCEREADY)) { 644 mFillingUpStatus = FS_FILLED; 645 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 646 return true; 647 } 648 return false; 649} 650 651status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused, 652 int triggerSession __unused) 653{ 654 status_t status = NO_ERROR; 655 ALOGV("start(%d), calling pid %d session %d", 656 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 657 658 sp<ThreadBase> thread = mThread.promote(); 659 if (thread != 0) { 660 if (isOffloaded()) { 661 Mutex::Autolock _laf(thread->mAudioFlinger->mLock); 662 Mutex::Autolock _lth(thread->mLock); 663 sp<EffectChain> ec = thread->getEffectChain_l(mSessionId); 664 if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() || 665 (ec != 0 && ec->isNonOffloadableEnabled())) { 666 invalidate(); 667 return PERMISSION_DENIED; 668 } 669 } 670 Mutex::Autolock _lth(thread->mLock); 671 track_state state = mState; 672 // here the track could be either new, or restarted 673 // in both cases "unstop" the track 674 675 // initial state-stopping. next state-pausing. 676 // What if resume is called ? 677 678 if (state == PAUSED || state == PAUSING) { 679 if (mResumeToStopping) { 680 // happened we need to resume to STOPPING_1 681 mState = TrackBase::STOPPING_1; 682 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 683 } else { 684 mState = TrackBase::RESUMING; 685 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 686 } 687 } else { 688 mState = TrackBase::ACTIVE; 689 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 690 } 691 692 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 693 status = playbackThread->addTrack_l(this); 694 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 695 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 696 // restore previous state if start was rejected by policy manager 697 if (status == PERMISSION_DENIED) { 698 mState = state; 699 } 700 } 701 // track was already in the active list, not a problem 702 if (status == ALREADY_EXISTS) { 703 status = NO_ERROR; 704 } else { 705 // Acknowledge any pending flush(), so that subsequent new data isn't discarded. 706 // It is usually unsafe to access the server proxy from a binder thread. 707 // But in this case we know the mixer thread (whether normal mixer or fast mixer) 708 // isn't looking at this track yet: we still hold the normal mixer thread lock, 709 // and for fast tracks the track is not yet in the fast mixer thread's active set. 710 ServerProxy::Buffer buffer; 711 buffer.mFrameCount = 1; 712 (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/); 713 } 714 } else { 715 status = BAD_VALUE; 716 } 717 return status; 718} 719 720void AudioFlinger::PlaybackThread::Track::stop() 721{ 722 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 723 sp<ThreadBase> thread = mThread.promote(); 724 if (thread != 0) { 725 Mutex::Autolock _l(thread->mLock); 726 track_state state = mState; 727 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 728 // If the track is not active (PAUSED and buffers full), flush buffers 729 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 730 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 731 reset(); 732 mState = STOPPED; 733 } else if (!isFastTrack() && !isOffloaded() && !isDirect()) { 734 mState = STOPPED; 735 } else { 736 // For fast tracks prepareTracks_l() will set state to STOPPING_2 737 // presentation is complete 738 // For an offloaded track this starts a drain and state will 739 // move to STOPPING_2 when drain completes and then STOPPED 740 mState = STOPPING_1; 741 } 742 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 743 playbackThread); 744 } 745 } 746} 747 748void AudioFlinger::PlaybackThread::Track::pause() 749{ 750 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 751 sp<ThreadBase> thread = mThread.promote(); 752 if (thread != 0) { 753 Mutex::Autolock _l(thread->mLock); 754 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 755 switch (mState) { 756 case STOPPING_1: 757 case STOPPING_2: 758 if (!isOffloaded()) { 759 /* nothing to do if track is not offloaded */ 760 break; 761 } 762 763 // Offloaded track was draining, we need to carry on draining when resumed 764 mResumeToStopping = true; 765 // fall through... 766 case ACTIVE: 767 case RESUMING: 768 mState = PAUSING; 769 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 770 playbackThread->broadcast_l(); 771 break; 772 773 default: 774 break; 775 } 776 } 777} 778 779void AudioFlinger::PlaybackThread::Track::flush() 780{ 781 ALOGV("flush(%d)", mName); 782 sp<ThreadBase> thread = mThread.promote(); 783 if (thread != 0) { 784 Mutex::Autolock _l(thread->mLock); 785 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 786 787 if (isOffloaded()) { 788 // If offloaded we allow flush during any state except terminated 789 // and keep the track active to avoid problems if user is seeking 790 // rapidly and underlying hardware has a significant delay handling 791 // a pause 792 if (isTerminated()) { 793 return; 794 } 795 796 ALOGV("flush: offload flush"); 797 reset(); 798 799 if (mState == STOPPING_1 || mState == STOPPING_2) { 800 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 801 mState = ACTIVE; 802 } 803 804 if (mState == ACTIVE) { 805 ALOGV("flush called in active state, resetting buffer time out retry count"); 806 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 807 } 808 809 mFlushHwPending = true; 810 mResumeToStopping = false; 811 } else { 812 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 813 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 814 return; 815 } 816 // No point remaining in PAUSED state after a flush => go to 817 // FLUSHED state 818 mState = FLUSHED; 819 // do not reset the track if it is still in the process of being stopped or paused. 820 // this will be done by prepareTracks_l() when the track is stopped. 821 // prepareTracks_l() will see mState == FLUSHED, then 822 // remove from active track list, reset(), and trigger presentation complete 823 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 824 reset(); 825 } 826 } 827 // Prevent flush being lost if the track is flushed and then resumed 828 // before mixer thread can run. This is important when offloading 829 // because the hardware buffer could hold a large amount of audio 830 playbackThread->broadcast_l(); 831 } 832} 833 834// must be called with thread lock held 835void AudioFlinger::PlaybackThread::Track::flushAck() 836{ 837 if (!isOffloaded()) 838 return; 839 840 mFlushHwPending = false; 841} 842 843void AudioFlinger::PlaybackThread::Track::reset() 844{ 845 // Do not reset twice to avoid discarding data written just after a flush and before 846 // the audioflinger thread detects the track is stopped. 847 if (!mResetDone) { 848 // Force underrun condition to avoid false underrun callback until first data is 849 // written to buffer 850 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 851 mFillingUpStatus = FS_FILLING; 852 mResetDone = true; 853 if (mState == FLUSHED) { 854 mState = IDLE; 855 } 856 } 857} 858 859status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 860{ 861 sp<ThreadBase> thread = mThread.promote(); 862 if (thread == 0) { 863 ALOGE("thread is dead"); 864 return FAILED_TRANSACTION; 865 } else if ((thread->type() == ThreadBase::DIRECT) || 866 (thread->type() == ThreadBase::OFFLOAD)) { 867 return thread->setParameters(keyValuePairs); 868 } else { 869 return PERMISSION_DENIED; 870 } 871} 872 873status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp) 874{ 875 // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant 876 if (isFastTrack()) { 877 // FIXME no lock held to set mPreviousValid = false 878 return INVALID_OPERATION; 879 } 880 sp<ThreadBase> thread = mThread.promote(); 881 if (thread == 0) { 882 // FIXME no lock held to set mPreviousValid = false 883 return INVALID_OPERATION; 884 } 885 Mutex::Autolock _l(thread->mLock); 886 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 887 if (!isOffloaded() && !isDirect()) { 888 if (!playbackThread->mLatchQValid) { 889 mPreviousValid = false; 890 return INVALID_OPERATION; 891 } 892 uint32_t unpresentedFrames = 893 ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) / 894 playbackThread->mSampleRate; 895 uint32_t framesWritten = mAudioTrackServerProxy->framesReleased(); 896 bool checkPreviousTimestamp = mPreviousValid && framesWritten >= mPreviousFramesWritten; 897 if (framesWritten < unpresentedFrames) { 898 mPreviousValid = false; 899 return INVALID_OPERATION; 900 } 901 mPreviousFramesWritten = framesWritten; 902 uint32_t position = framesWritten - unpresentedFrames; 903 struct timespec time = playbackThread->mLatchQ.mTimestamp.mTime; 904 if (checkPreviousTimestamp) { 905 if (time.tv_sec < mPreviousTimestamp.mTime.tv_sec || 906 (time.tv_sec == mPreviousTimestamp.mTime.tv_sec && 907 time.tv_nsec < mPreviousTimestamp.mTime.tv_nsec)) { 908 ALOGW("Time is going backwards"); 909 } 910 // position can bobble slightly as an artifact; this hides the bobble 911 static const uint32_t MINIMUM_POSITION_DELTA = 8u; 912 if ((position <= mPreviousTimestamp.mPosition) || 913 (position - mPreviousTimestamp.mPosition) < MINIMUM_POSITION_DELTA) { 914 position = mPreviousTimestamp.mPosition; 915 time = mPreviousTimestamp.mTime; 916 } 917 } 918 timestamp.mPosition = position; 919 timestamp.mTime = time; 920 mPreviousTimestamp = timestamp; 921 mPreviousValid = true; 922 return NO_ERROR; 923 } 924 925 return playbackThread->getTimestamp_l(timestamp); 926} 927 928status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 929{ 930 status_t status = DEAD_OBJECT; 931 sp<ThreadBase> thread = mThread.promote(); 932 if (thread != 0) { 933 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 934 sp<AudioFlinger> af = mClient->audioFlinger(); 935 936 Mutex::Autolock _l(af->mLock); 937 938 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 939 940 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 941 Mutex::Autolock _dl(playbackThread->mLock); 942 Mutex::Autolock _sl(srcThread->mLock); 943 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 944 if (chain == 0) { 945 return INVALID_OPERATION; 946 } 947 948 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 949 if (effect == 0) { 950 return INVALID_OPERATION; 951 } 952 srcThread->removeEffect_l(effect); 953 status = playbackThread->addEffect_l(effect); 954 if (status != NO_ERROR) { 955 srcThread->addEffect_l(effect); 956 return INVALID_OPERATION; 957 } 958 // removeEffect_l() has stopped the effect if it was active so it must be restarted 959 if (effect->state() == EffectModule::ACTIVE || 960 effect->state() == EffectModule::STOPPING) { 961 effect->start(); 962 } 963 964 sp<EffectChain> dstChain = effect->chain().promote(); 965 if (dstChain == 0) { 966 srcThread->addEffect_l(effect); 967 return INVALID_OPERATION; 968 } 969 AudioSystem::unregisterEffect(effect->id()); 970 AudioSystem::registerEffect(&effect->desc(), 971 srcThread->id(), 972 dstChain->strategy(), 973 AUDIO_SESSION_OUTPUT_MIX, 974 effect->id()); 975 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 976 } 977 status = playbackThread->attachAuxEffect(this, EffectId); 978 } 979 return status; 980} 981 982void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 983{ 984 mAuxEffectId = EffectId; 985 mAuxBuffer = buffer; 986} 987 988bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 989 size_t audioHalFrames) 990{ 991 // a track is considered presented when the total number of frames written to audio HAL 992 // corresponds to the number of frames written when presentationComplete() is called for the 993 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 994 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 995 // to detect when all frames have been played. In this case framesWritten isn't 996 // useful because it doesn't always reflect whether there is data in the h/w 997 // buffers, particularly if a track has been paused and resumed during draining 998 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 999 mPresentationCompleteFrames, framesWritten); 1000 if (mPresentationCompleteFrames == 0) { 1001 mPresentationCompleteFrames = framesWritten + audioHalFrames; 1002 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 1003 mPresentationCompleteFrames, audioHalFrames); 1004 } 1005 1006 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 1007 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1008 mAudioTrackServerProxy->setStreamEndDone(); 1009 return true; 1010 } 1011 return false; 1012} 1013 1014void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 1015{ 1016 for (size_t i = 0; i < mSyncEvents.size(); i++) { 1017 if (mSyncEvents[i]->type() == type) { 1018 mSyncEvents[i]->trigger(); 1019 mSyncEvents.removeAt(i); 1020 i--; 1021 } 1022 } 1023} 1024 1025// implement VolumeBufferProvider interface 1026 1027gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 1028{ 1029 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 1030 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 1031 gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR(); 1032 float vl = float_from_gain(gain_minifloat_unpack_left(vlr)); 1033 float vr = float_from_gain(gain_minifloat_unpack_right(vlr)); 1034 // track volumes come from shared memory, so can't be trusted and must be clamped 1035 if (vl > GAIN_FLOAT_UNITY) { 1036 vl = GAIN_FLOAT_UNITY; 1037 } 1038 if (vr > GAIN_FLOAT_UNITY) { 1039 vr = GAIN_FLOAT_UNITY; 1040 } 1041 // now apply the cached master volume and stream type volume; 1042 // this is trusted but lacks any synchronization or barrier so may be stale 1043 float v = mCachedVolume; 1044 vl *= v; 1045 vr *= v; 1046 // re-combine into packed minifloat 1047 vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr)); 1048 // FIXME look at mute, pause, and stop flags 1049 return vlr; 1050} 1051 1052status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 1053{ 1054 if (isTerminated() || mState == PAUSED || 1055 ((framesReady() == 0) && ((mSharedBuffer != 0) || 1056 (mState == STOPPED)))) { 1057 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 1058 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 1059 event->cancel(); 1060 return INVALID_OPERATION; 1061 } 1062 (void) TrackBase::setSyncEvent(event); 1063 return NO_ERROR; 1064} 1065 1066void AudioFlinger::PlaybackThread::Track::invalidate() 1067{ 1068 // FIXME should use proxy, and needs work 1069 audio_track_cblk_t* cblk = mCblk; 1070 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1071 android_atomic_release_store(0x40000000, &cblk->mFutex); 1072 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 1073 (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); 1074 mIsInvalid = true; 1075} 1076 1077void AudioFlinger::PlaybackThread::Track::signal() 1078{ 1079 sp<ThreadBase> thread = mThread.promote(); 1080 if (thread != 0) { 1081 PlaybackThread *t = (PlaybackThread *)thread.get(); 1082 Mutex::Autolock _l(t->mLock); 1083 t->broadcast_l(); 1084 } 1085} 1086 1087//To be called with thread lock held 1088bool AudioFlinger::PlaybackThread::Track::isResumePending() { 1089 1090 if (mState == RESUMING) 1091 return true; 1092 /* Resume is pending if track was stopping before pause was called */ 1093 if (mState == STOPPING_1 && 1094 mResumeToStopping) 1095 return true; 1096 1097 return false; 1098} 1099 1100//To be called with thread lock held 1101void AudioFlinger::PlaybackThread::Track::resumeAck() { 1102 1103 1104 if (mState == RESUMING) 1105 mState = ACTIVE; 1106 1107 // Other possibility of pending resume is stopping_1 state 1108 // Do not update the state from stopping as this prevents 1109 // drain being called. 1110 if (mState == STOPPING_1) { 1111 mResumeToStopping = false; 1112 } 1113} 1114// ---------------------------------------------------------------------------- 1115 1116sp<AudioFlinger::PlaybackThread::TimedTrack> 1117AudioFlinger::PlaybackThread::TimedTrack::create( 1118 PlaybackThread *thread, 1119 const sp<Client>& client, 1120 audio_stream_type_t streamType, 1121 uint32_t sampleRate, 1122 audio_format_t format, 1123 audio_channel_mask_t channelMask, 1124 size_t frameCount, 1125 const sp<IMemory>& sharedBuffer, 1126 int sessionId, 1127 int uid) 1128{ 1129 if (!client->reserveTimedTrack()) 1130 return 0; 1131 1132 return new TimedTrack( 1133 thread, client, streamType, sampleRate, format, channelMask, frameCount, 1134 sharedBuffer, sessionId, uid); 1135} 1136 1137AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 1138 PlaybackThread *thread, 1139 const sp<Client>& client, 1140 audio_stream_type_t streamType, 1141 uint32_t sampleRate, 1142 audio_format_t format, 1143 audio_channel_mask_t channelMask, 1144 size_t frameCount, 1145 const sp<IMemory>& sharedBuffer, 1146 int sessionId, 1147 int uid) 1148 : Track(thread, client, streamType, sampleRate, format, channelMask, 1149 frameCount, (sharedBuffer != 0) ? sharedBuffer->pointer() : NULL, sharedBuffer, 1150 sessionId, uid, IAudioFlinger::TRACK_TIMED, TYPE_TIMED), 1151 mQueueHeadInFlight(false), 1152 mTrimQueueHeadOnRelease(false), 1153 mFramesPendingInQueue(0), 1154 mTimedSilenceBuffer(NULL), 1155 mTimedSilenceBufferSize(0), 1156 mTimedAudioOutputOnTime(false), 1157 mMediaTimeTransformValid(false) 1158{ 1159 LocalClock lc; 1160 mLocalTimeFreq = lc.getLocalFreq(); 1161 1162 mLocalTimeToSampleTransform.a_zero = 0; 1163 mLocalTimeToSampleTransform.b_zero = 0; 1164 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 1165 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 1166 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 1167 &mLocalTimeToSampleTransform.a_to_b_denom); 1168 1169 mMediaTimeToSampleTransform.a_zero = 0; 1170 mMediaTimeToSampleTransform.b_zero = 0; 1171 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 1172 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 1173 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 1174 &mMediaTimeToSampleTransform.a_to_b_denom); 1175} 1176 1177AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 1178 mClient->releaseTimedTrack(); 1179 delete [] mTimedSilenceBuffer; 1180} 1181 1182status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 1183 size_t size, sp<IMemory>* buffer) { 1184 1185 Mutex::Autolock _l(mTimedBufferQueueLock); 1186 1187 trimTimedBufferQueue_l(); 1188 1189 // lazily initialize the shared memory heap for timed buffers 1190 if (mTimedMemoryDealer == NULL) { 1191 const int kTimedBufferHeapSize = 512 << 10; 1192 1193 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 1194 "AudioFlingerTimed"); 1195 if (mTimedMemoryDealer == NULL) { 1196 return NO_MEMORY; 1197 } 1198 } 1199 1200 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 1201 if (newBuffer == 0 || newBuffer->pointer() == NULL) { 1202 return NO_MEMORY; 1203 } 1204 1205 *buffer = newBuffer; 1206 return NO_ERROR; 1207} 1208 1209// caller must hold mTimedBufferQueueLock 1210void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 1211 int64_t mediaTimeNow; 1212 { 1213 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1214 if (!mMediaTimeTransformValid) 1215 return; 1216 1217 int64_t targetTimeNow; 1218 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 1219 ? mCCHelper.getCommonTime(&targetTimeNow) 1220 : mCCHelper.getLocalTime(&targetTimeNow); 1221 1222 if (OK != res) 1223 return; 1224 1225 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 1226 &mediaTimeNow)) { 1227 return; 1228 } 1229 } 1230 1231 size_t trimEnd; 1232 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 1233 int64_t bufEnd; 1234 1235 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 1236 // We have a next buffer. Just use its PTS as the PTS of the frame 1237 // following the last frame in this buffer. If the stream is sparse 1238 // (ie, there are deliberate gaps left in the stream which should be 1239 // filled with silence by the TimedAudioTrack), then this can result 1240 // in one extra buffer being left un-trimmed when it could have 1241 // been. In general, this is not typical, and we would rather 1242 // optimized away the TS calculation below for the more common case 1243 // where PTSes are contiguous. 1244 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 1245 } else { 1246 // We have no next buffer. Compute the PTS of the frame following 1247 // the last frame in this buffer by computing the duration of of 1248 // this frame in media time units and adding it to the PTS of the 1249 // buffer. 1250 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1251 / mFrameSize; 1252 1253 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1254 &bufEnd)) { 1255 ALOGE("Failed to convert frame count of %lld to media time" 1256 " duration" " (scale factor %d/%u) in %s", 1257 frameCount, 1258 mMediaTimeToSampleTransform.a_to_b_numer, 1259 mMediaTimeToSampleTransform.a_to_b_denom, 1260 __PRETTY_FUNCTION__); 1261 break; 1262 } 1263 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1264 } 1265 1266 if (bufEnd > mediaTimeNow) 1267 break; 1268 1269 // Is the buffer we want to use in the middle of a mix operation right 1270 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1271 // from the mixer which should be coming back shortly. 1272 if (!trimEnd && mQueueHeadInFlight) { 1273 mTrimQueueHeadOnRelease = true; 1274 } 1275 } 1276 1277 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1278 if (trimStart < trimEnd) { 1279 // Update the bookkeeping for framesReady() 1280 for (size_t i = trimStart; i < trimEnd; ++i) { 1281 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1282 } 1283 1284 // Now actually remove the buffers from the queue. 1285 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1286 } 1287} 1288 1289void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1290 const char* logTag) { 1291 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1292 "%s called (reason \"%s\"), but timed buffer queue has no" 1293 " elements to trim.", __FUNCTION__, logTag); 1294 1295 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1296 mTimedBufferQueue.removeAt(0); 1297} 1298 1299void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1300 const TimedBuffer& buf, 1301 const char* logTag __unused) { 1302 uint32_t bufBytes = buf.buffer()->size(); 1303 uint32_t consumedAlready = buf.position(); 1304 1305 ALOG_ASSERT(consumedAlready <= bufBytes, 1306 "Bad bookkeeping while updating frames pending. Timed buffer is" 1307 " only %u bytes long, but claims to have consumed %u" 1308 " bytes. (update reason: \"%s\")", 1309 bufBytes, consumedAlready, logTag); 1310 1311 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1312 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1313 "Bad bookkeeping while updating frames pending. Should have at" 1314 " least %u queued frames, but we think we have only %u. (update" 1315 " reason: \"%s\")", 1316 bufFrames, mFramesPendingInQueue, logTag); 1317 1318 mFramesPendingInQueue -= bufFrames; 1319} 1320 1321status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1322 const sp<IMemory>& buffer, int64_t pts) { 1323 1324 { 1325 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1326 if (!mMediaTimeTransformValid) 1327 return INVALID_OPERATION; 1328 } 1329 1330 Mutex::Autolock _l(mTimedBufferQueueLock); 1331 1332 uint32_t bufFrames = buffer->size() / mFrameSize; 1333 mFramesPendingInQueue += bufFrames; 1334 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1335 1336 return NO_ERROR; 1337} 1338 1339status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1340 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1341 1342 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1343 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1344 target); 1345 1346 if (!(target == TimedAudioTrack::LOCAL_TIME || 1347 target == TimedAudioTrack::COMMON_TIME)) { 1348 return BAD_VALUE; 1349 } 1350 1351 Mutex::Autolock lock(mMediaTimeTransformLock); 1352 mMediaTimeTransform = xform; 1353 mMediaTimeTransformTarget = target; 1354 mMediaTimeTransformValid = true; 1355 1356 return NO_ERROR; 1357} 1358 1359#define min(a, b) ((a) < (b) ? (a) : (b)) 1360 1361// implementation of getNextBuffer for tracks whose buffers have timestamps 1362status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1363 AudioBufferProvider::Buffer* buffer, int64_t pts) 1364{ 1365 if (pts == AudioBufferProvider::kInvalidPTS) { 1366 buffer->raw = NULL; 1367 buffer->frameCount = 0; 1368 mTimedAudioOutputOnTime = false; 1369 return INVALID_OPERATION; 1370 } 1371 1372 Mutex::Autolock _l(mTimedBufferQueueLock); 1373 1374 ALOG_ASSERT(!mQueueHeadInFlight, 1375 "getNextBuffer called without releaseBuffer!"); 1376 1377 while (true) { 1378 1379 // if we have no timed buffers, then fail 1380 if (mTimedBufferQueue.isEmpty()) { 1381 buffer->raw = NULL; 1382 buffer->frameCount = 0; 1383 return NOT_ENOUGH_DATA; 1384 } 1385 1386 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1387 1388 // calculate the PTS of the head of the timed buffer queue expressed in 1389 // local time 1390 int64_t headLocalPTS; 1391 { 1392 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1393 1394 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1395 1396 if (mMediaTimeTransform.a_to_b_denom == 0) { 1397 // the transform represents a pause, so yield silence 1398 timedYieldSilence_l(buffer->frameCount, buffer); 1399 return NO_ERROR; 1400 } 1401 1402 int64_t transformedPTS; 1403 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1404 &transformedPTS)) { 1405 // the transform failed. this shouldn't happen, but if it does 1406 // then just drop this buffer 1407 ALOGW("timedGetNextBuffer transform failed"); 1408 buffer->raw = NULL; 1409 buffer->frameCount = 0; 1410 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1411 return NO_ERROR; 1412 } 1413 1414 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1415 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1416 &headLocalPTS)) { 1417 buffer->raw = NULL; 1418 buffer->frameCount = 0; 1419 return INVALID_OPERATION; 1420 } 1421 } else { 1422 headLocalPTS = transformedPTS; 1423 } 1424 } 1425 1426 uint32_t sr = sampleRate(); 1427 1428 // adjust the head buffer's PTS to reflect the portion of the head buffer 1429 // that has already been consumed 1430 int64_t effectivePTS = headLocalPTS + 1431 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1432 1433 // Calculate the delta in samples between the head of the input buffer 1434 // queue and the start of the next output buffer that will be written. 1435 // If the transformation fails because of over or underflow, it means 1436 // that the sample's position in the output stream is so far out of 1437 // whack that it should just be dropped. 1438 int64_t sampleDelta; 1439 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1440 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1441 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1442 " mix"); 1443 continue; 1444 } 1445 if (!mLocalTimeToSampleTransform.doForwardTransform( 1446 (effectivePTS - pts) << 32, &sampleDelta)) { 1447 ALOGV("*** too late during sample rate transform: dropped buffer"); 1448 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1449 continue; 1450 } 1451 1452 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1453 " sampleDelta=[%d.%08x]", 1454 head.pts(), head.position(), pts, 1455 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1456 + (sampleDelta >> 32)), 1457 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1458 1459 // if the delta between the ideal placement for the next input sample and 1460 // the current output position is within this threshold, then we will 1461 // concatenate the next input samples to the previous output 1462 const int64_t kSampleContinuityThreshold = 1463 (static_cast<int64_t>(sr) << 32) / 250; 1464 1465 // if this is the first buffer of audio that we're emitting from this track 1466 // then it should be almost exactly on time. 1467 const int64_t kSampleStartupThreshold = 1LL << 32; 1468 1469 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1470 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1471 // the next input is close enough to being on time, so concatenate it 1472 // with the last output 1473 timedYieldSamples_l(buffer); 1474 1475 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1476 head.position(), buffer->frameCount); 1477 return NO_ERROR; 1478 } 1479 1480 // Looks like our output is not on time. Reset our on timed status. 1481 // Next time we mix samples from our input queue, then should be within 1482 // the StartupThreshold. 1483 mTimedAudioOutputOnTime = false; 1484 if (sampleDelta > 0) { 1485 // the gap between the current output position and the proper start of 1486 // the next input sample is too big, so fill it with silence 1487 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1488 1489 timedYieldSilence_l(framesUntilNextInput, buffer); 1490 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1491 return NO_ERROR; 1492 } else { 1493 // the next input sample is late 1494 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1495 size_t onTimeSamplePosition = 1496 head.position() + lateFrames * mFrameSize; 1497 1498 if (onTimeSamplePosition > head.buffer()->size()) { 1499 // all the remaining samples in the head are too late, so 1500 // drop it and move on 1501 ALOGV("*** too late: dropped buffer"); 1502 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1503 continue; 1504 } else { 1505 // skip over the late samples 1506 head.setPosition(onTimeSamplePosition); 1507 1508 // yield the available samples 1509 timedYieldSamples_l(buffer); 1510 1511 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1512 return NO_ERROR; 1513 } 1514 } 1515 } 1516} 1517 1518// Yield samples from the timed buffer queue head up to the given output 1519// buffer's capacity. 1520// 1521// Caller must hold mTimedBufferQueueLock 1522void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1523 AudioBufferProvider::Buffer* buffer) { 1524 1525 const TimedBuffer& head = mTimedBufferQueue[0]; 1526 1527 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1528 head.position()); 1529 1530 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1531 mFrameSize); 1532 size_t framesRequested = buffer->frameCount; 1533 buffer->frameCount = min(framesLeftInHead, framesRequested); 1534 1535 mQueueHeadInFlight = true; 1536 mTimedAudioOutputOnTime = true; 1537} 1538 1539// Yield samples of silence up to the given output buffer's capacity 1540// 1541// Caller must hold mTimedBufferQueueLock 1542void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1543 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1544 1545 // lazily allocate a buffer filled with silence 1546 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1547 delete [] mTimedSilenceBuffer; 1548 mTimedSilenceBufferSize = numFrames * mFrameSize; 1549 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1550 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1551 } 1552 1553 buffer->raw = mTimedSilenceBuffer; 1554 size_t framesRequested = buffer->frameCount; 1555 buffer->frameCount = min(numFrames, framesRequested); 1556 1557 mTimedAudioOutputOnTime = false; 1558} 1559 1560// AudioBufferProvider interface 1561void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1562 AudioBufferProvider::Buffer* buffer) { 1563 1564 Mutex::Autolock _l(mTimedBufferQueueLock); 1565 1566 // If the buffer which was just released is part of the buffer at the head 1567 // of the queue, be sure to update the amt of the buffer which has been 1568 // consumed. If the buffer being returned is not part of the head of the 1569 // queue, its either because the buffer is part of the silence buffer, or 1570 // because the head of the timed queue was trimmed after the mixer called 1571 // getNextBuffer but before the mixer called releaseBuffer. 1572 if (buffer->raw == mTimedSilenceBuffer) { 1573 ALOG_ASSERT(!mQueueHeadInFlight, 1574 "Queue head in flight during release of silence buffer!"); 1575 goto done; 1576 } 1577 1578 ALOG_ASSERT(mQueueHeadInFlight, 1579 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1580 " head in flight."); 1581 1582 if (mTimedBufferQueue.size()) { 1583 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1584 1585 void* start = head.buffer()->pointer(); 1586 void* end = reinterpret_cast<void*>( 1587 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1588 + head.buffer()->size()); 1589 1590 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1591 "released buffer not within the head of the timed buffer" 1592 " queue; qHead = [%p, %p], released buffer = %p", 1593 start, end, buffer->raw); 1594 1595 head.setPosition(head.position() + 1596 (buffer->frameCount * mFrameSize)); 1597 mQueueHeadInFlight = false; 1598 1599 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1600 "Bad bookkeeping during releaseBuffer! Should have at" 1601 " least %u queued frames, but we think we have only %u", 1602 buffer->frameCount, mFramesPendingInQueue); 1603 1604 mFramesPendingInQueue -= buffer->frameCount; 1605 1606 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1607 || mTrimQueueHeadOnRelease) { 1608 trimTimedBufferQueueHead_l("releaseBuffer"); 1609 mTrimQueueHeadOnRelease = false; 1610 } 1611 } else { 1612 LOG_ALWAYS_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1613 " buffers in the timed buffer queue"); 1614 } 1615 1616done: 1617 buffer->raw = 0; 1618 buffer->frameCount = 0; 1619} 1620 1621size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1622 Mutex::Autolock _l(mTimedBufferQueueLock); 1623 return mFramesPendingInQueue; 1624} 1625 1626AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1627 : mPTS(0), mPosition(0) {} 1628 1629AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1630 const sp<IMemory>& buffer, int64_t pts) 1631 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1632 1633 1634// ---------------------------------------------------------------------------- 1635 1636AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1637 PlaybackThread *playbackThread, 1638 DuplicatingThread *sourceThread, 1639 uint32_t sampleRate, 1640 audio_format_t format, 1641 audio_channel_mask_t channelMask, 1642 size_t frameCount, 1643 int uid) 1644 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1645 NULL, 0, 0, uid, IAudioFlinger::TRACK_DEFAULT, TYPE_OUTPUT), 1646 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1647{ 1648 1649 if (mCblk != NULL) { 1650 mOutBuffer.frameCount = 0; 1651 playbackThread->mTracks.add(this); 1652 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1653 "frameCount %u, mChannelMask 0x%08x", 1654 mCblk, mBuffer, 1655 frameCount, mChannelMask); 1656 // since client and server are in the same process, 1657 // the buffer has the same virtual address on both sides 1658 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1659 true /*clientInServer*/); 1660 mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); 1661 mClientProxy->setSendLevel(0.0); 1662 mClientProxy->setSampleRate(sampleRate); 1663 } else { 1664 ALOGW("Error creating output track on thread %p", playbackThread); 1665 } 1666} 1667 1668AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1669{ 1670 clearBufferQueue(); 1671 delete mClientProxy; 1672 // superclass destructor will now delete the server proxy and shared memory both refer to 1673} 1674 1675status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1676 int triggerSession) 1677{ 1678 status_t status = Track::start(event, triggerSession); 1679 if (status != NO_ERROR) { 1680 return status; 1681 } 1682 1683 mActive = true; 1684 mRetryCount = 127; 1685 return status; 1686} 1687 1688void AudioFlinger::PlaybackThread::OutputTrack::stop() 1689{ 1690 Track::stop(); 1691 clearBufferQueue(); 1692 mOutBuffer.frameCount = 0; 1693 mActive = false; 1694} 1695 1696bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1697{ 1698 Buffer *pInBuffer; 1699 Buffer inBuffer; 1700 uint32_t channelCount = mChannelCount; 1701 bool outputBufferFull = false; 1702 inBuffer.frameCount = frames; 1703 inBuffer.i16 = data; 1704 1705 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1706 1707 if (!mActive && frames != 0) { 1708 start(); 1709 sp<ThreadBase> thread = mThread.promote(); 1710 if (thread != 0) { 1711 MixerThread *mixerThread = (MixerThread *)thread.get(); 1712 if (mFrameCount > frames) { 1713 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1714 uint32_t startFrames = (mFrameCount - frames); 1715 pInBuffer = new Buffer; 1716 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1717 pInBuffer->frameCount = startFrames; 1718 pInBuffer->i16 = pInBuffer->mBuffer; 1719 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1720 mBufferQueue.add(pInBuffer); 1721 } else { 1722 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1723 } 1724 } 1725 } 1726 } 1727 1728 while (waitTimeLeftMs) { 1729 // First write pending buffers, then new data 1730 if (mBufferQueue.size()) { 1731 pInBuffer = mBufferQueue.itemAt(0); 1732 } else { 1733 pInBuffer = &inBuffer; 1734 } 1735 1736 if (pInBuffer->frameCount == 0) { 1737 break; 1738 } 1739 1740 if (mOutBuffer.frameCount == 0) { 1741 mOutBuffer.frameCount = pInBuffer->frameCount; 1742 nsecs_t startTime = systemTime(); 1743 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1744 if (status != NO_ERROR) { 1745 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1746 mThread.unsafe_get(), status); 1747 outputBufferFull = true; 1748 break; 1749 } 1750 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1751 if (waitTimeLeftMs >= waitTimeMs) { 1752 waitTimeLeftMs -= waitTimeMs; 1753 } else { 1754 waitTimeLeftMs = 0; 1755 } 1756 } 1757 1758 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1759 pInBuffer->frameCount; 1760 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1761 Proxy::Buffer buf; 1762 buf.mFrameCount = outFrames; 1763 buf.mRaw = NULL; 1764 mClientProxy->releaseBuffer(&buf); 1765 pInBuffer->frameCount -= outFrames; 1766 pInBuffer->i16 += outFrames * channelCount; 1767 mOutBuffer.frameCount -= outFrames; 1768 mOutBuffer.i16 += outFrames * channelCount; 1769 1770 if (pInBuffer->frameCount == 0) { 1771 if (mBufferQueue.size()) { 1772 mBufferQueue.removeAt(0); 1773 delete [] pInBuffer->mBuffer; 1774 delete pInBuffer; 1775 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1776 mThread.unsafe_get(), mBufferQueue.size()); 1777 } else { 1778 break; 1779 } 1780 } 1781 } 1782 1783 // If we could not write all frames, allocate a buffer and queue it for next time. 1784 if (inBuffer.frameCount) { 1785 sp<ThreadBase> thread = mThread.promote(); 1786 if (thread != 0 && !thread->standby()) { 1787 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1788 pInBuffer = new Buffer; 1789 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1790 pInBuffer->frameCount = inBuffer.frameCount; 1791 pInBuffer->i16 = pInBuffer->mBuffer; 1792 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1793 sizeof(int16_t)); 1794 mBufferQueue.add(pInBuffer); 1795 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1796 mThread.unsafe_get(), mBufferQueue.size()); 1797 } else { 1798 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1799 mThread.unsafe_get(), this); 1800 } 1801 } 1802 } 1803 1804 // Calling write() with a 0 length buffer, means that no more data will be written: 1805 // If no more buffers are pending, fill output track buffer to make sure it is started 1806 // by output mixer. 1807 if (frames == 0 && mBufferQueue.size() == 0) { 1808 // FIXME borken, replace by getting framesReady() from proxy 1809 size_t user = 0; // was mCblk->user 1810 if (user < mFrameCount) { 1811 frames = mFrameCount - user; 1812 pInBuffer = new Buffer; 1813 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1814 pInBuffer->frameCount = frames; 1815 pInBuffer->i16 = pInBuffer->mBuffer; 1816 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1817 mBufferQueue.add(pInBuffer); 1818 } else if (mActive) { 1819 stop(); 1820 } 1821 } 1822 1823 return outputBufferFull; 1824} 1825 1826status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1827 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1828{ 1829 ClientProxy::Buffer buf; 1830 buf.mFrameCount = buffer->frameCount; 1831 struct timespec timeout; 1832 timeout.tv_sec = waitTimeMs / 1000; 1833 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1834 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1835 buffer->frameCount = buf.mFrameCount; 1836 buffer->raw = buf.mRaw; 1837 return status; 1838} 1839 1840void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1841{ 1842 size_t size = mBufferQueue.size(); 1843 1844 for (size_t i = 0; i < size; i++) { 1845 Buffer *pBuffer = mBufferQueue.itemAt(i); 1846 delete [] pBuffer->mBuffer; 1847 delete pBuffer; 1848 } 1849 mBufferQueue.clear(); 1850} 1851 1852 1853AudioFlinger::PlaybackThread::PatchTrack::PatchTrack(PlaybackThread *playbackThread, 1854 uint32_t sampleRate, 1855 audio_channel_mask_t channelMask, 1856 audio_format_t format, 1857 size_t frameCount, 1858 void *buffer, 1859 IAudioFlinger::track_flags_t flags) 1860 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1861 buffer, 0, 0, getuid(), flags, TYPE_PATCH), 1862 mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true)) 1863{ 1864 uint64_t mixBufferNs = ((uint64_t)2 * playbackThread->frameCount() * 1000000000) / 1865 playbackThread->sampleRate(); 1866 mPeerTimeout.tv_sec = mixBufferNs / 1000000000; 1867 mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000); 1868 1869 ALOGV("PatchTrack %p sampleRate %d mPeerTimeout %d.%03d sec", 1870 this, sampleRate, 1871 (int)mPeerTimeout.tv_sec, 1872 (int)(mPeerTimeout.tv_nsec / 1000000)); 1873} 1874 1875AudioFlinger::PlaybackThread::PatchTrack::~PatchTrack() 1876{ 1877} 1878 1879// AudioBufferProvider interface 1880status_t AudioFlinger::PlaybackThread::PatchTrack::getNextBuffer( 1881 AudioBufferProvider::Buffer* buffer, int64_t pts) 1882{ 1883 ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::getNextBuffer() called without peer proxy"); 1884 Proxy::Buffer buf; 1885 buf.mFrameCount = buffer->frameCount; 1886 status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout); 1887 ALOGV_IF(status != NO_ERROR, "PatchTrack() %p getNextBuffer status %d", this, status); 1888 if (buf.mFrameCount == 0) { 1889 return WOULD_BLOCK; 1890 } 1891 buffer->frameCount = buf.mFrameCount; 1892 status = Track::getNextBuffer(buffer, pts); 1893 return status; 1894} 1895 1896void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(AudioBufferProvider::Buffer* buffer) 1897{ 1898 ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::releaseBuffer() called without peer proxy"); 1899 Proxy::Buffer buf; 1900 buf.mFrameCount = buffer->frameCount; 1901 buf.mRaw = buffer->raw; 1902 mPeerProxy->releaseBuffer(&buf); 1903 TrackBase::releaseBuffer(buffer); 1904} 1905 1906status_t AudioFlinger::PlaybackThread::PatchTrack::obtainBuffer(Proxy::Buffer* buffer, 1907 const struct timespec *timeOut) 1908{ 1909 return mProxy->obtainBuffer(buffer, timeOut); 1910} 1911 1912void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(Proxy::Buffer* buffer) 1913{ 1914 mProxy->releaseBuffer(buffer); 1915 if (android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags) & CBLK_DISABLED) { 1916 ALOGW("PatchTrack::releaseBuffer() disabled due to previous underrun, restarting"); 1917 start(); 1918 } 1919 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1920} 1921 1922// ---------------------------------------------------------------------------- 1923// Record 1924// ---------------------------------------------------------------------------- 1925 1926AudioFlinger::RecordHandle::RecordHandle( 1927 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1928 : BnAudioRecord(), 1929 mRecordTrack(recordTrack) 1930{ 1931} 1932 1933AudioFlinger::RecordHandle::~RecordHandle() { 1934 stop_nonvirtual(); 1935 mRecordTrack->destroy(); 1936} 1937 1938status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1939 int triggerSession) { 1940 ALOGV("RecordHandle::start()"); 1941 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1942} 1943 1944void AudioFlinger::RecordHandle::stop() { 1945 stop_nonvirtual(); 1946} 1947 1948void AudioFlinger::RecordHandle::stop_nonvirtual() { 1949 ALOGV("RecordHandle::stop()"); 1950 mRecordTrack->stop(); 1951} 1952 1953status_t AudioFlinger::RecordHandle::onTransact( 1954 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1955{ 1956 return BnAudioRecord::onTransact(code, data, reply, flags); 1957} 1958 1959// ---------------------------------------------------------------------------- 1960 1961// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 1962AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1963 RecordThread *thread, 1964 const sp<Client>& client, 1965 uint32_t sampleRate, 1966 audio_format_t format, 1967 audio_channel_mask_t channelMask, 1968 size_t frameCount, 1969 void *buffer, 1970 int sessionId, 1971 int uid, 1972 IAudioFlinger::track_flags_t flags, 1973 track_type type) 1974 : TrackBase(thread, client, sampleRate, format, 1975 channelMask, frameCount, buffer, sessionId, uid, 1976 flags, false /*isOut*/, 1977 (type == TYPE_DEFAULT) ? 1978 ((flags & IAudioFlinger::TRACK_FAST) ? ALLOC_PIPE : ALLOC_CBLK) : 1979 ((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE), 1980 type), 1981 mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0), 1982 // See real initialization of mRsmpInFront at RecordThread::start() 1983 mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL) 1984{ 1985 if (mCblk == NULL) { 1986 return; 1987 } 1988 1989 mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, 1990 mFrameSize, !isExternalTrack()); 1991 1992 uint32_t channelCount = audio_channel_count_from_in_mask(channelMask); 1993 // FIXME I don't understand either of the channel count checks 1994 if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 && 1995 channelCount <= FCC_2) { 1996 // sink SR 1997 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT, 1998 thread->mChannelCount, sampleRate); 1999 // source SR 2000 mResampler->setSampleRate(thread->mSampleRate); 2001 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 2002 mResamplerBufferProvider = new ResamplerBufferProvider(this); 2003 } 2004 2005 if (flags & IAudioFlinger::TRACK_FAST) { 2006 ALOG_ASSERT(thread->mFastTrackAvail); 2007 thread->mFastTrackAvail = false; 2008 } 2009} 2010 2011AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 2012{ 2013 ALOGV("%s", __func__); 2014 delete mResampler; 2015 delete[] mRsmpOutBuffer; 2016 delete mResamplerBufferProvider; 2017} 2018 2019// AudioBufferProvider interface 2020status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 2021 int64_t pts __unused) 2022{ 2023 ServerProxy::Buffer buf; 2024 buf.mFrameCount = buffer->frameCount; 2025 status_t status = mServerProxy->obtainBuffer(&buf); 2026 buffer->frameCount = buf.mFrameCount; 2027 buffer->raw = buf.mRaw; 2028 if (buf.mFrameCount == 0) { 2029 // FIXME also wake futex so that overrun is noticed more quickly 2030 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 2031 } 2032 return status; 2033} 2034 2035status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 2036 int triggerSession) 2037{ 2038 sp<ThreadBase> thread = mThread.promote(); 2039 if (thread != 0) { 2040 RecordThread *recordThread = (RecordThread *)thread.get(); 2041 return recordThread->start(this, event, triggerSession); 2042 } else { 2043 return BAD_VALUE; 2044 } 2045} 2046 2047void AudioFlinger::RecordThread::RecordTrack::stop() 2048{ 2049 sp<ThreadBase> thread = mThread.promote(); 2050 if (thread != 0) { 2051 RecordThread *recordThread = (RecordThread *)thread.get(); 2052 if (recordThread->stop(this) && isExternalTrack()) { 2053 AudioSystem::stopInput(recordThread->id(), (audio_session_t)mSessionId); 2054 } 2055 } 2056} 2057 2058void AudioFlinger::RecordThread::RecordTrack::destroy() 2059{ 2060 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 2061 sp<RecordTrack> keep(this); 2062 { 2063 sp<ThreadBase> thread = mThread.promote(); 2064 if (thread != 0) { 2065 if (isExternalTrack()) { 2066 if (mState == ACTIVE || mState == RESUMING) { 2067 AudioSystem::stopInput(thread->id(), (audio_session_t)mSessionId); 2068 } 2069 AudioSystem::releaseInput(thread->id(), (audio_session_t)mSessionId); 2070 } 2071 Mutex::Autolock _l(thread->mLock); 2072 RecordThread *recordThread = (RecordThread *) thread.get(); 2073 recordThread->destroyTrack_l(this); 2074 } 2075 } 2076} 2077 2078void AudioFlinger::RecordThread::RecordTrack::invalidate() 2079{ 2080 // FIXME should use proxy, and needs work 2081 audio_track_cblk_t* cblk = mCblk; 2082 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 2083 android_atomic_release_store(0x40000000, &cblk->mFutex); 2084 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 2085 (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); 2086} 2087 2088 2089/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 2090{ 2091 result.append(" Active Client Fmt Chn mask Session S Server fCount SRate\n"); 2092} 2093 2094void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active) 2095{ 2096 snprintf(buffer, size, " %6s %6u %3u %08X %7u %1d %08X %6zu %5u\n", 2097 active ? "yes" : "no", 2098 (mClient == 0) ? getpid_cached : mClient->pid(), 2099 mFormat, 2100 mChannelMask, 2101 mSessionId, 2102 mState, 2103 mCblk->mServer, 2104 mFrameCount, 2105 mSampleRate); 2106 2107} 2108 2109void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event) 2110{ 2111 if (event == mSyncStartEvent) { 2112 ssize_t framesToDrop = 0; 2113 sp<ThreadBase> threadBase = mThread.promote(); 2114 if (threadBase != 0) { 2115 // TODO: use actual buffer filling status instead of 2 buffers when info is available 2116 // from audio HAL 2117 framesToDrop = threadBase->mFrameCount * 2; 2118 } 2119 mFramesToDrop = framesToDrop; 2120 } 2121} 2122 2123void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent() 2124{ 2125 if (mSyncStartEvent != 0) { 2126 mSyncStartEvent->cancel(); 2127 mSyncStartEvent.clear(); 2128 } 2129 mFramesToDrop = 0; 2130} 2131 2132 2133AudioFlinger::RecordThread::PatchRecord::PatchRecord(RecordThread *recordThread, 2134 uint32_t sampleRate, 2135 audio_channel_mask_t channelMask, 2136 audio_format_t format, 2137 size_t frameCount, 2138 void *buffer, 2139 IAudioFlinger::track_flags_t flags) 2140 : RecordTrack(recordThread, NULL, sampleRate, format, channelMask, frameCount, 2141 buffer, 0, getuid(), flags, TYPE_PATCH), 2142 mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true)) 2143{ 2144 uint64_t mixBufferNs = ((uint64_t)2 * recordThread->frameCount() * 1000000000) / 2145 recordThread->sampleRate(); 2146 mPeerTimeout.tv_sec = mixBufferNs / 1000000000; 2147 mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000); 2148 2149 ALOGV("PatchRecord %p sampleRate %d mPeerTimeout %d.%03d sec", 2150 this, sampleRate, 2151 (int)mPeerTimeout.tv_sec, 2152 (int)(mPeerTimeout.tv_nsec / 1000000)); 2153} 2154 2155AudioFlinger::RecordThread::PatchRecord::~PatchRecord() 2156{ 2157} 2158 2159// AudioBufferProvider interface 2160status_t AudioFlinger::RecordThread::PatchRecord::getNextBuffer( 2161 AudioBufferProvider::Buffer* buffer, int64_t pts) 2162{ 2163 ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::getNextBuffer() called without peer proxy"); 2164 Proxy::Buffer buf; 2165 buf.mFrameCount = buffer->frameCount; 2166 status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout); 2167 ALOGV_IF(status != NO_ERROR, 2168 "PatchRecord() %p mPeerProxy->obtainBuffer status %d", this, status); 2169 if (buf.mFrameCount == 0) { 2170 return WOULD_BLOCK; 2171 } 2172 buffer->frameCount = buf.mFrameCount; 2173 status = RecordTrack::getNextBuffer(buffer, pts); 2174 return status; 2175} 2176 2177void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(AudioBufferProvider::Buffer* buffer) 2178{ 2179 ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::releaseBuffer() called without peer proxy"); 2180 Proxy::Buffer buf; 2181 buf.mFrameCount = buffer->frameCount; 2182 buf.mRaw = buffer->raw; 2183 mPeerProxy->releaseBuffer(&buf); 2184 TrackBase::releaseBuffer(buffer); 2185} 2186 2187status_t AudioFlinger::RecordThread::PatchRecord::obtainBuffer(Proxy::Buffer* buffer, 2188 const struct timespec *timeOut) 2189{ 2190 return mProxy->obtainBuffer(buffer, timeOut); 2191} 2192 2193void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(Proxy::Buffer* buffer) 2194{ 2195 mProxy->releaseBuffer(buffer); 2196} 2197 2198}; // namespace android 2199