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 55 namespace android { 56 57 // ---------------------------------------------------------------------------- 58 // TrackBase 59 // ---------------------------------------------------------------------------- 60 61 static volatile int32_t nextTrackId = 55; 62 63 // TrackBase constructor must be called with AudioFlinger::mLock held 64 AudioFlinger::ThreadBase::TrackBase::TrackBase( 65 ThreadBase *thread, 66 const sp<Client>& client, 67 uint32_t sampleRate, 68 audio_format_t format, 69 audio_channel_mask_t channelMask, 70 size_t frameCount, 71 void *buffer, 72 int sessionId, 73 int clientUid, 74 IAudioFlinger::track_flags_t flags, 75 bool isOut, 76 alloc_type alloc, 77 track_type type) 78 : RefBase(), 79 mThread(thread), 80 mClient(client), 81 mCblk(NULL), 82 // mBuffer 83 mState(IDLE), 84 mSampleRate(sampleRate), 85 mFormat(format), 86 mChannelMask(channelMask), 87 mChannelCount(isOut ? 88 audio_channel_count_from_out_mask(channelMask) : 89 audio_channel_count_from_in_mask(channelMask)), 90 mFrameSize(audio_is_linear_pcm(format) ? 91 mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)), 92 mFrameCount(frameCount), 93 mSessionId(sessionId), 94 mFlags(flags), 95 mIsOut(isOut), 96 mServerProxy(NULL), 97 mId(android_atomic_inc(&nextTrackId)), 98 mTerminated(false), 99 mType(type), 100 mThreadIoHandle(thread->id()) 101 { 102 // if the caller is us, trust the specified uid 103 if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) { 104 int newclientUid = IPCThreadState::self()->getCallingUid(); 105 if (clientUid != -1 && clientUid != newclientUid) { 106 ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid); 107 } 108 clientUid = newclientUid; 109 } 110 // clientUid contains the uid of the app that is responsible for this track, so we can blame 111 // battery usage on it. 112 mUid = clientUid; 113 114 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 115 size_t size = sizeof(audio_track_cblk_t); 116 size_t bufferSize = (buffer == NULL ? roundup(frameCount) : frameCount) * mFrameSize; 117 if (buffer == NULL && alloc == ALLOC_CBLK) { 118 size += bufferSize; 119 } 120 121 if (client != 0) { 122 mCblkMemory = client->heap()->allocate(size); 123 if (mCblkMemory == 0 || 124 (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer())) == NULL) { 125 ALOGE("not enough memory for AudioTrack size=%u", size); 126 client->heap()->dump("AudioTrack"); 127 mCblkMemory.clear(); 128 return; 129 } 130 } else { 131 // this syntax avoids calling the audio_track_cblk_t constructor twice 132 mCblk = (audio_track_cblk_t *) new uint8_t[size]; 133 // assume mCblk != NULL 134 } 135 136 // construct the shared structure in-place. 137 if (mCblk != NULL) { 138 new(mCblk) audio_track_cblk_t(); 139 switch (alloc) { 140 case ALLOC_READONLY: { 141 const sp<MemoryDealer> roHeap(thread->readOnlyHeap()); 142 if (roHeap == 0 || 143 (mBufferMemory = roHeap->allocate(bufferSize)) == 0 || 144 (mBuffer = mBufferMemory->pointer()) == NULL) { 145 ALOGE("not enough memory for read-only buffer size=%zu", bufferSize); 146 if (roHeap != 0) { 147 roHeap->dump("buffer"); 148 } 149 mCblkMemory.clear(); 150 mBufferMemory.clear(); 151 return; 152 } 153 memset(mBuffer, 0, bufferSize); 154 } break; 155 case ALLOC_PIPE: 156 mBufferMemory = thread->pipeMemory(); 157 // mBuffer is the virtual address as seen from current process (mediaserver), 158 // and should normally be coming from mBufferMemory->pointer(). 159 // However in this case the TrackBase does not reference the buffer directly. 160 // It should references the buffer via the pipe. 161 // Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL. 162 mBuffer = NULL; 163 break; 164 case ALLOC_CBLK: 165 // clear all buffers 166 if (buffer == NULL) { 167 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 168 memset(mBuffer, 0, bufferSize); 169 } else { 170 mBuffer = buffer; 171 #if 0 172 mCblk->mFlags = CBLK_FORCEREADY; // FIXME hack, need to fix the track ready logic 173 #endif 174 } 175 break; 176 case ALLOC_LOCAL: 177 mBuffer = calloc(1, bufferSize); 178 break; 179 case ALLOC_NONE: 180 mBuffer = buffer; 181 break; 182 } 183 184 #ifdef TEE_SINK 185 if (mTeeSinkTrackEnabled) { 186 NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount, mFormat); 187 if (Format_isValid(pipeFormat)) { 188 Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat); 189 size_t numCounterOffers = 0; 190 const NBAIO_Format offers[1] = {pipeFormat}; 191 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 192 ALOG_ASSERT(index == 0); 193 PipeReader *pipeReader = new PipeReader(*pipe); 194 numCounterOffers = 0; 195 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 196 ALOG_ASSERT(index == 0); 197 mTeeSink = pipe; 198 mTeeSource = pipeReader; 199 } 200 } 201 #endif 202 203 } 204 } 205 206 status_t AudioFlinger::ThreadBase::TrackBase::initCheck() const 207 { 208 status_t status; 209 if (mType == TYPE_OUTPUT || mType == TYPE_PATCH) { 210 status = cblk() != NULL ? NO_ERROR : NO_MEMORY; 211 } else { 212 status = getCblk() != 0 ? NO_ERROR : NO_MEMORY; 213 } 214 return status; 215 } 216 217 AudioFlinger::ThreadBase::TrackBase::~TrackBase() 218 { 219 #ifdef TEE_SINK 220 dumpTee(-1, mTeeSource, mId); 221 #endif 222 // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference 223 delete mServerProxy; 224 if (mCblk != NULL) { 225 if (mClient == 0) { 226 delete mCblk; 227 } else { 228 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 229 } 230 } 231 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 232 if (mClient != 0) { 233 // Client destructor must run with AudioFlinger client mutex locked 234 Mutex::Autolock _l(mClient->audioFlinger()->mClientLock); 235 // If the client's reference count drops to zero, the associated destructor 236 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 237 // relying on the automatic clear() at end of scope. 238 mClient.clear(); 239 } 240 // flush the binder command buffer 241 IPCThreadState::self()->flushCommands(); 242 } 243 244 // AudioBufferProvider interface 245 // getNextBuffer() = 0; 246 // This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 247 void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 248 { 249 #ifdef TEE_SINK 250 if (mTeeSink != 0) { 251 (void) mTeeSink->write(buffer->raw, buffer->frameCount); 252 } 253 #endif 254 255 ServerProxy::Buffer buf; 256 buf.mFrameCount = buffer->frameCount; 257 buf.mRaw = buffer->raw; 258 buffer->frameCount = 0; 259 buffer->raw = NULL; 260 mServerProxy->releaseBuffer(&buf); 261 } 262 263 status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 264 { 265 mSyncEvents.add(event); 266 return NO_ERROR; 267 } 268 269 // ---------------------------------------------------------------------------- 270 // Playback 271 // ---------------------------------------------------------------------------- 272 273 AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 274 : BnAudioTrack(), 275 mTrack(track) 276 { 277 } 278 279 AudioFlinger::TrackHandle::~TrackHandle() { 280 // just stop the track on deletion, associated resources 281 // will be freed from the main thread once all pending buffers have 282 // been played. Unless it's not in the active track list, in which 283 // case we free everything now... 284 mTrack->destroy(); 285 } 286 287 sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 288 return mTrack->getCblk(); 289 } 290 291 status_t AudioFlinger::TrackHandle::start() { 292 return mTrack->start(); 293 } 294 295 void AudioFlinger::TrackHandle::stop() { 296 mTrack->stop(); 297 } 298 299 void AudioFlinger::TrackHandle::flush() { 300 mTrack->flush(); 301 } 302 303 void AudioFlinger::TrackHandle::pause() { 304 mTrack->pause(); 305 } 306 307 status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 308 { 309 return mTrack->attachAuxEffect(EffectId); 310 } 311 312 status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 313 sp<IMemory>* buffer) { 314 if (!mTrack->isTimedTrack()) 315 return INVALID_OPERATION; 316 317 PlaybackThread::TimedTrack* tt = 318 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 319 return tt->allocateTimedBuffer(size, buffer); 320 } 321 322 status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 323 int64_t pts) { 324 if (!mTrack->isTimedTrack()) 325 return INVALID_OPERATION; 326 327 if (buffer == 0 || buffer->pointer() == NULL) { 328 ALOGE("queueTimedBuffer() buffer is 0 or has NULL pointer()"); 329 return BAD_VALUE; 330 } 331 332 PlaybackThread::TimedTrack* tt = 333 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 334 return tt->queueTimedBuffer(buffer, pts); 335 } 336 337 status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 338 const LinearTransform& xform, int target) { 339 340 if (!mTrack->isTimedTrack()) 341 return INVALID_OPERATION; 342 343 PlaybackThread::TimedTrack* tt = 344 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 345 return tt->setMediaTimeTransform( 346 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 347 } 348 349 status_t AudioFlinger::TrackHandle::setParameters(const String8& keyValuePairs) { 350 return mTrack->setParameters(keyValuePairs); 351 } 352 353 status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp) 354 { 355 return mTrack->getTimestamp(timestamp); 356 } 357 358 359 void AudioFlinger::TrackHandle::signal() 360 { 361 return mTrack->signal(); 362 } 363 364 status_t AudioFlinger::TrackHandle::onTransact( 365 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 366 { 367 return BnAudioTrack::onTransact(code, data, reply, flags); 368 } 369 370 // ---------------------------------------------------------------------------- 371 372 // Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 373 AudioFlinger::PlaybackThread::Track::Track( 374 PlaybackThread *thread, 375 const sp<Client>& client, 376 audio_stream_type_t streamType, 377 uint32_t sampleRate, 378 audio_format_t format, 379 audio_channel_mask_t channelMask, 380 size_t frameCount, 381 void *buffer, 382 const sp<IMemory>& sharedBuffer, 383 int sessionId, 384 int uid, 385 IAudioFlinger::track_flags_t flags, 386 track_type type) 387 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 388 (sharedBuffer != 0) ? sharedBuffer->pointer() : buffer, 389 sessionId, uid, flags, true /*isOut*/, 390 (type == TYPE_PATCH) ? ( buffer == NULL ? ALLOC_LOCAL : ALLOC_NONE) : ALLOC_CBLK, 391 type), 392 mFillingUpStatus(FS_INVALID), 393 // mRetryCount initialized later when needed 394 mSharedBuffer(sharedBuffer), 395 mStreamType(streamType), 396 mName(-1), // see note below 397 mMainBuffer(thread->mixBuffer()), 398 mAuxBuffer(NULL), 399 mAuxEffectId(0), mHasVolumeController(false), 400 mPresentationCompleteFrames(0), 401 mFastIndex(-1), 402 mCachedVolume(1.0), 403 mIsInvalid(false), 404 mAudioTrackServerProxy(NULL), 405 mResumeToStopping(false), 406 mFlushHwPending(false), 407 mPreviousValid(false), 408 mPreviousFramesWritten(0) 409 // mPreviousTimestamp 410 { 411 // client == 0 implies sharedBuffer == 0 412 ALOG_ASSERT(!(client == 0 && sharedBuffer != 0)); 413 414 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), 415 sharedBuffer->size()); 416 417 if (mCblk == NULL) { 418 return; 419 } 420 421 if (sharedBuffer == 0) { 422 mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, 423 mFrameSize, !isExternalTrack(), sampleRate); 424 } else { 425 mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, 426 mFrameSize); 427 } 428 mServerProxy = mAudioTrackServerProxy; 429 430 mName = thread->getTrackName_l(channelMask, format, sessionId); 431 if (mName < 0) { 432 ALOGE("no more track names available"); 433 return; 434 } 435 // only allocate a fast track index if we were able to allocate a normal track name 436 if (flags & IAudioFlinger::TRACK_FAST) { 437 mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads(); 438 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 439 int i = __builtin_ctz(thread->mFastTrackAvailMask); 440 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 441 // FIXME This is too eager. We allocate a fast track index before the 442 // fast track becomes active. Since fast tracks are a scarce resource, 443 // this means we are potentially denying other more important fast tracks from 444 // being created. It would be better to allocate the index dynamically. 445 mFastIndex = i; 446 // Read the initial underruns because this field is never cleared by the fast mixer 447 mObservedUnderruns = thread->getFastTrackUnderruns(i); 448 thread->mFastTrackAvailMask &= ~(1 << i); 449 } 450 } 451 452 AudioFlinger::PlaybackThread::Track::~Track() 453 { 454 ALOGV("PlaybackThread::Track destructor"); 455 456 // The destructor would clear mSharedBuffer, 457 // but it will not push the decremented reference count, 458 // leaving the client's IMemory dangling indefinitely. 459 // This prevents that leak. 460 if (mSharedBuffer != 0) { 461 mSharedBuffer.clear(); 462 } 463 } 464 465 status_t AudioFlinger::PlaybackThread::Track::initCheck() const 466 { 467 status_t status = TrackBase::initCheck(); 468 if (status == NO_ERROR && mName < 0) { 469 status = NO_MEMORY; 470 } 471 return status; 472 } 473 474 void AudioFlinger::PlaybackThread::Track::destroy() 475 { 476 // NOTE: destroyTrack_l() can remove a strong reference to this Track 477 // by removing it from mTracks vector, so there is a risk that this Tracks's 478 // destructor is called. As the destructor needs to lock mLock, 479 // we must acquire a strong reference on this Track before locking mLock 480 // here so that the destructor is called only when exiting this function. 481 // On the other hand, as long as Track::destroy() is only called by 482 // TrackHandle destructor, the TrackHandle still holds a strong ref on 483 // this Track with its member mTrack. 484 sp<Track> keep(this); 485 { // scope for mLock 486 bool wasActive = false; 487 sp<ThreadBase> thread = mThread.promote(); 488 if (thread != 0) { 489 Mutex::Autolock _l(thread->mLock); 490 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 491 wasActive = playbackThread->destroyTrack_l(this); 492 } 493 if (isExternalTrack() && !wasActive) { 494 AudioSystem::releaseOutput(mThreadIoHandle); 495 } 496 } 497 } 498 499 /*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 500 { 501 result.append(" Name Active Client Type Fmt Chn mask Session fCount S F SRate " 502 "L dB R dB Server Main buf Aux Buf Flags UndFrmCnt\n"); 503 } 504 505 void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size, bool active) 506 { 507 gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR(); 508 if (isFastTrack()) { 509 sprintf(buffer, " F %2d", mFastIndex); 510 } else if (mName >= AudioMixer::TRACK0) { 511 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 512 } else { 513 sprintf(buffer, " none"); 514 } 515 track_state state = mState; 516 char stateChar; 517 if (isTerminated()) { 518 stateChar = 'T'; 519 } else { 520 switch (state) { 521 case IDLE: 522 stateChar = 'I'; 523 break; 524 case STOPPING_1: 525 stateChar = 's'; 526 break; 527 case STOPPING_2: 528 stateChar = '5'; 529 break; 530 case STOPPED: 531 stateChar = 'S'; 532 break; 533 case RESUMING: 534 stateChar = 'R'; 535 break; 536 case ACTIVE: 537 stateChar = 'A'; 538 break; 539 case PAUSING: 540 stateChar = 'p'; 541 break; 542 case PAUSED: 543 stateChar = 'P'; 544 break; 545 case FLUSHED: 546 stateChar = 'F'; 547 break; 548 default: 549 stateChar = '?'; 550 break; 551 } 552 } 553 char nowInUnderrun; 554 switch (mObservedUnderruns.mBitFields.mMostRecent) { 555 case UNDERRUN_FULL: 556 nowInUnderrun = ' '; 557 break; 558 case UNDERRUN_PARTIAL: 559 nowInUnderrun = '<'; 560 break; 561 case UNDERRUN_EMPTY: 562 nowInUnderrun = '*'; 563 break; 564 default: 565 nowInUnderrun = '?'; 566 break; 567 } 568 snprintf(&buffer[8], size-8, " %6s %6u %4u %08X %08X %7u %6zu %1c %1d %5u %5.2g %5.2g " 569 "%08X %p %p 0x%03X %9u%c\n", 570 active ? "yes" : "no", 571 (mClient == 0) ? getpid_cached : mClient->pid(), 572 mStreamType, 573 mFormat, 574 mChannelMask, 575 mSessionId, 576 mFrameCount, 577 stateChar, 578 mFillingUpStatus, 579 mAudioTrackServerProxy->getSampleRate(), 580 20.0 * log10(float_from_gain(gain_minifloat_unpack_left(vlr))), 581 20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))), 582 mCblk->mServer, 583 mMainBuffer, 584 mAuxBuffer, 585 mCblk->mFlags, 586 mAudioTrackServerProxy->getUnderrunFrames(), 587 nowInUnderrun); 588 } 589 590 uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const { 591 return mAudioTrackServerProxy->getSampleRate(); 592 } 593 594 // AudioBufferProvider interface 595 status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 596 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 597 { 598 ServerProxy::Buffer buf; 599 size_t desiredFrames = buffer->frameCount; 600 buf.mFrameCount = desiredFrames; 601 status_t status = mServerProxy->obtainBuffer(&buf); 602 buffer->frameCount = buf.mFrameCount; 603 buffer->raw = buf.mRaw; 604 if (buf.mFrameCount == 0) { 605 mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 606 } 607 return status; 608 } 609 610 // releaseBuffer() is not overridden 611 612 // ExtendedAudioBufferProvider interface 613 614 // Note that framesReady() takes a mutex on the control block using tryLock(). 615 // This could result in priority inversion if framesReady() is called by the normal mixer, 616 // as the normal mixer thread runs at lower 617 // priority than the client's callback thread: there is a short window within framesReady() 618 // during which the normal mixer could be preempted, and the client callback would block. 619 // Another problem can occur if framesReady() is called by the fast mixer: 620 // the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 621 // FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 622 size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 623 return mAudioTrackServerProxy->framesReady(); 624 } 625 626 size_t AudioFlinger::PlaybackThread::Track::framesReleased() const 627 { 628 return mAudioTrackServerProxy->framesReleased(); 629 } 630 631 // Don't call for fast tracks; the framesReady() could result in priority inversion 632 bool AudioFlinger::PlaybackThread::Track::isReady() const { 633 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) { 634 return true; 635 } 636 637 if (isStopping()) { 638 if (framesReady() > 0) { 639 mFillingUpStatus = FS_FILLED; 640 } 641 return true; 642 } 643 644 if (framesReady() >= mFrameCount || 645 (mCblk->mFlags & CBLK_FORCEREADY)) { 646 mFillingUpStatus = FS_FILLED; 647 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 648 return true; 649 } 650 return false; 651 } 652 653 status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event __unused, 654 int triggerSession __unused) 655 { 656 status_t status = NO_ERROR; 657 ALOGV("start(%d), calling pid %d session %d", 658 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 659 660 sp<ThreadBase> thread = mThread.promote(); 661 if (thread != 0) { 662 if (isOffloaded()) { 663 Mutex::Autolock _laf(thread->mAudioFlinger->mLock); 664 Mutex::Autolock _lth(thread->mLock); 665 sp<EffectChain> ec = thread->getEffectChain_l(mSessionId); 666 if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() || 667 (ec != 0 && ec->isNonOffloadableEnabled())) { 668 invalidate(); 669 return PERMISSION_DENIED; 670 } 671 } 672 Mutex::Autolock _lth(thread->mLock); 673 track_state state = mState; 674 // here the track could be either new, or restarted 675 // in both cases "unstop" the track 676 677 // initial state-stopping. next state-pausing. 678 // What if resume is called ? 679 680 if (state == PAUSED || state == PAUSING) { 681 if (mResumeToStopping) { 682 // happened we need to resume to STOPPING_1 683 mState = TrackBase::STOPPING_1; 684 ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this); 685 } else { 686 mState = TrackBase::RESUMING; 687 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 688 } 689 } else { 690 mState = TrackBase::ACTIVE; 691 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 692 } 693 694 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 695 status = playbackThread->addTrack_l(this); 696 if (status == INVALID_OPERATION || status == PERMISSION_DENIED) { 697 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 698 // restore previous state if start was rejected by policy manager 699 if (status == PERMISSION_DENIED) { 700 mState = state; 701 } 702 } 703 // track was already in the active list, not a problem 704 if (status == ALREADY_EXISTS) { 705 status = NO_ERROR; 706 } else { 707 // Acknowledge any pending flush(), so that subsequent new data isn't discarded. 708 // It is usually unsafe to access the server proxy from a binder thread. 709 // But in this case we know the mixer thread (whether normal mixer or fast mixer) 710 // isn't looking at this track yet: we still hold the normal mixer thread lock, 711 // and for fast tracks the track is not yet in the fast mixer thread's active set. 712 ServerProxy::Buffer buffer; 713 buffer.mFrameCount = 1; 714 (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/); 715 } 716 } else { 717 status = BAD_VALUE; 718 } 719 return status; 720 } 721 722 void AudioFlinger::PlaybackThread::Track::stop() 723 { 724 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 725 sp<ThreadBase> thread = mThread.promote(); 726 if (thread != 0) { 727 Mutex::Autolock _l(thread->mLock); 728 track_state state = mState; 729 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 730 // If the track is not active (PAUSED and buffers full), flush buffers 731 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 732 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 733 reset(); 734 mState = STOPPED; 735 } else if (!isFastTrack() && !isOffloaded() && !isDirect()) { 736 mState = STOPPED; 737 } else { 738 // For fast tracks prepareTracks_l() will set state to STOPPING_2 739 // presentation is complete 740 // For an offloaded track this starts a drain and state will 741 // move to STOPPING_2 when drain completes and then STOPPED 742 mState = STOPPING_1; 743 } 744 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, 745 playbackThread); 746 } 747 } 748 } 749 750 void AudioFlinger::PlaybackThread::Track::pause() 751 { 752 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 753 sp<ThreadBase> thread = mThread.promote(); 754 if (thread != 0) { 755 Mutex::Autolock _l(thread->mLock); 756 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 757 switch (mState) { 758 case STOPPING_1: 759 case STOPPING_2: 760 if (!isOffloaded()) { 761 /* nothing to do if track is not offloaded */ 762 break; 763 } 764 765 // Offloaded track was draining, we need to carry on draining when resumed 766 mResumeToStopping = true; 767 // fall through... 768 case ACTIVE: 769 case RESUMING: 770 mState = PAUSING; 771 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 772 playbackThread->broadcast_l(); 773 break; 774 775 default: 776 break; 777 } 778 } 779 } 780 781 void AudioFlinger::PlaybackThread::Track::flush() 782 { 783 ALOGV("flush(%d)", mName); 784 sp<ThreadBase> thread = mThread.promote(); 785 if (thread != 0) { 786 Mutex::Autolock _l(thread->mLock); 787 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 788 789 if (isOffloaded()) { 790 // If offloaded we allow flush during any state except terminated 791 // and keep the track active to avoid problems if user is seeking 792 // rapidly and underlying hardware has a significant delay handling 793 // a pause 794 if (isTerminated()) { 795 return; 796 } 797 798 ALOGV("flush: offload flush"); 799 reset(); 800 801 if (mState == STOPPING_1 || mState == STOPPING_2) { 802 ALOGV("flushed in STOPPING_1 or 2 state, change state to ACTIVE"); 803 mState = ACTIVE; 804 } 805 806 if (mState == ACTIVE) { 807 ALOGV("flush called in active state, resetting buffer time out retry count"); 808 mRetryCount = PlaybackThread::kMaxTrackRetriesOffload; 809 } 810 811 mFlushHwPending = true; 812 mResumeToStopping = false; 813 } else { 814 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && 815 mState != PAUSED && mState != PAUSING && mState != IDLE && mState != FLUSHED) { 816 return; 817 } 818 // No point remaining in PAUSED state after a flush => go to 819 // FLUSHED state 820 mState = FLUSHED; 821 // do not reset the track if it is still in the process of being stopped or paused. 822 // this will be done by prepareTracks_l() when the track is stopped. 823 // prepareTracks_l() will see mState == FLUSHED, then 824 // remove from active track list, reset(), and trigger presentation complete 825 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 826 reset(); 827 if (thread->type() == ThreadBase::DIRECT) { 828 DirectOutputThread *t = (DirectOutputThread *)playbackThread; 829 t->flushHw_l(); 830 } 831 } 832 } 833 // Prevent flush being lost if the track is flushed and then resumed 834 // before mixer thread can run. This is important when offloading 835 // because the hardware buffer could hold a large amount of audio 836 playbackThread->broadcast_l(); 837 } 838 } 839 840 // must be called with thread lock held 841 void AudioFlinger::PlaybackThread::Track::flushAck() 842 { 843 if (!isOffloaded()) 844 return; 845 846 mFlushHwPending = false; 847 } 848 849 void AudioFlinger::PlaybackThread::Track::reset() 850 { 851 // Do not reset twice to avoid discarding data written just after a flush and before 852 // the audioflinger thread detects the track is stopped. 853 if (!mResetDone) { 854 // Force underrun condition to avoid false underrun callback until first data is 855 // written to buffer 856 android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags); 857 mFillingUpStatus = FS_FILLING; 858 mResetDone = true; 859 if (mState == FLUSHED) { 860 mState = IDLE; 861 } 862 } 863 } 864 865 status_t AudioFlinger::PlaybackThread::Track::setParameters(const String8& keyValuePairs) 866 { 867 sp<ThreadBase> thread = mThread.promote(); 868 if (thread == 0) { 869 ALOGE("thread is dead"); 870 return FAILED_TRANSACTION; 871 } else if ((thread->type() == ThreadBase::DIRECT) || 872 (thread->type() == ThreadBase::OFFLOAD)) { 873 return thread->setParameters(keyValuePairs); 874 } else { 875 return PERMISSION_DENIED; 876 } 877 } 878 879 status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp) 880 { 881 // Client should implement this using SSQ; the unpresented frame count in latch is irrelevant 882 if (isFastTrack()) { 883 // FIXME no lock held to set mPreviousValid = false 884 return INVALID_OPERATION; 885 } 886 sp<ThreadBase> thread = mThread.promote(); 887 if (thread == 0) { 888 // FIXME no lock held to set mPreviousValid = false 889 return INVALID_OPERATION; 890 } 891 Mutex::Autolock _l(thread->mLock); 892 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 893 if (!isOffloaded() && !isDirect()) { 894 if (!playbackThread->mLatchQValid) { 895 mPreviousValid = false; 896 return INVALID_OPERATION; 897 } 898 uint32_t unpresentedFrames = 899 ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) / 900 playbackThread->mSampleRate; 901 // FIXME Since we're using a raw pointer as the key, it is theoretically possible 902 // for a brand new track to share the same address as a recently destroyed 903 // track, and thus for us to get the frames released of the wrong track. 904 // It is unlikely that we would be able to call getTimestamp() so quickly 905 // right after creating a new track. Nevertheless, the index here should 906 // be changed to something that is unique. Or use a completely different strategy. 907 ssize_t i = playbackThread->mLatchQ.mFramesReleased.indexOfKey(this); 908 uint32_t framesWritten = i >= 0 ? 909 playbackThread->mLatchQ.mFramesReleased[i] : 910 mAudioTrackServerProxy->framesReleased(); 911 bool checkPreviousTimestamp = mPreviousValid && framesWritten >= mPreviousFramesWritten; 912 if (framesWritten < unpresentedFrames) { 913 mPreviousValid = false; 914 return INVALID_OPERATION; 915 } 916 mPreviousFramesWritten = framesWritten; 917 uint32_t position = framesWritten - unpresentedFrames; 918 struct timespec time = playbackThread->mLatchQ.mTimestamp.mTime; 919 if (checkPreviousTimestamp) { 920 if (time.tv_sec < mPreviousTimestamp.mTime.tv_sec || 921 (time.tv_sec == mPreviousTimestamp.mTime.tv_sec && 922 time.tv_nsec < mPreviousTimestamp.mTime.tv_nsec)) { 923 ALOGW("Time is going backwards"); 924 } 925 // position can bobble slightly as an artifact; this hides the bobble 926 static const uint32_t MINIMUM_POSITION_DELTA = 8u; 927 if ((position <= mPreviousTimestamp.mPosition) || 928 (position - mPreviousTimestamp.mPosition) < MINIMUM_POSITION_DELTA) { 929 position = mPreviousTimestamp.mPosition; 930 time = mPreviousTimestamp.mTime; 931 } 932 } 933 timestamp.mPosition = position; 934 timestamp.mTime = time; 935 mPreviousTimestamp = timestamp; 936 mPreviousValid = true; 937 return NO_ERROR; 938 } 939 940 return playbackThread->getTimestamp_l(timestamp); 941 } 942 943 status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 944 { 945 status_t status = DEAD_OBJECT; 946 sp<ThreadBase> thread = mThread.promote(); 947 if (thread != 0) { 948 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 949 sp<AudioFlinger> af = mClient->audioFlinger(); 950 951 Mutex::Autolock _l(af->mLock); 952 953 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 954 955 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 956 Mutex::Autolock _dl(playbackThread->mLock); 957 Mutex::Autolock _sl(srcThread->mLock); 958 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 959 if (chain == 0) { 960 return INVALID_OPERATION; 961 } 962 963 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 964 if (effect == 0) { 965 return INVALID_OPERATION; 966 } 967 srcThread->removeEffect_l(effect); 968 status = playbackThread->addEffect_l(effect); 969 if (status != NO_ERROR) { 970 srcThread->addEffect_l(effect); 971 return INVALID_OPERATION; 972 } 973 // removeEffect_l() has stopped the effect if it was active so it must be restarted 974 if (effect->state() == EffectModule::ACTIVE || 975 effect->state() == EffectModule::STOPPING) { 976 effect->start(); 977 } 978 979 sp<EffectChain> dstChain = effect->chain().promote(); 980 if (dstChain == 0) { 981 srcThread->addEffect_l(effect); 982 return INVALID_OPERATION; 983 } 984 AudioSystem::unregisterEffect(effect->id()); 985 AudioSystem::registerEffect(&effect->desc(), 986 srcThread->id(), 987 dstChain->strategy(), 988 AUDIO_SESSION_OUTPUT_MIX, 989 effect->id()); 990 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 991 } 992 status = playbackThread->attachAuxEffect(this, EffectId); 993 } 994 return status; 995 } 996 997 void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 998 { 999 mAuxEffectId = EffectId; 1000 mAuxBuffer = buffer; 1001 } 1002 1003 bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 1004 size_t audioHalFrames) 1005 { 1006 // a track is considered presented when the total number of frames written to audio HAL 1007 // corresponds to the number of frames written when presentationComplete() is called for the 1008 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 1009 // For an offloaded track the HAL+h/w delay is variable so a HAL drain() is used 1010 // to detect when all frames have been played. In this case framesWritten isn't 1011 // useful because it doesn't always reflect whether there is data in the h/w 1012 // buffers, particularly if a track has been paused and resumed during draining 1013 ALOGV("presentationComplete() mPresentationCompleteFrames %d framesWritten %d", 1014 mPresentationCompleteFrames, framesWritten); 1015 if (mPresentationCompleteFrames == 0) { 1016 mPresentationCompleteFrames = framesWritten + audioHalFrames; 1017 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 1018 mPresentationCompleteFrames, audioHalFrames); 1019 } 1020 1021 if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) { 1022 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1023 mAudioTrackServerProxy->setStreamEndDone(); 1024 return true; 1025 } 1026 return false; 1027 } 1028 1029 void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 1030 { 1031 for (size_t i = 0; i < mSyncEvents.size(); i++) { 1032 if (mSyncEvents[i]->type() == type) { 1033 mSyncEvents[i]->trigger(); 1034 mSyncEvents.removeAt(i); 1035 i--; 1036 } 1037 } 1038 } 1039 1040 // implement VolumeBufferProvider interface 1041 1042 gain_minifloat_packed_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 1043 { 1044 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 1045 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 1046 gain_minifloat_packed_t vlr = mAudioTrackServerProxy->getVolumeLR(); 1047 float vl = float_from_gain(gain_minifloat_unpack_left(vlr)); 1048 float vr = float_from_gain(gain_minifloat_unpack_right(vlr)); 1049 // track volumes come from shared memory, so can't be trusted and must be clamped 1050 if (vl > GAIN_FLOAT_UNITY) { 1051 vl = GAIN_FLOAT_UNITY; 1052 } 1053 if (vr > GAIN_FLOAT_UNITY) { 1054 vr = GAIN_FLOAT_UNITY; 1055 } 1056 // now apply the cached master volume and stream type volume; 1057 // this is trusted but lacks any synchronization or barrier so may be stale 1058 float v = mCachedVolume; 1059 vl *= v; 1060 vr *= v; 1061 // re-combine into packed minifloat 1062 vlr = gain_minifloat_pack(gain_from_float(vl), gain_from_float(vr)); 1063 // FIXME look at mute, pause, and stop flags 1064 return vlr; 1065 } 1066 1067 status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 1068 { 1069 if (isTerminated() || mState == PAUSED || 1070 ((framesReady() == 0) && ((mSharedBuffer != 0) || 1071 (mState == STOPPED)))) { 1072 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 1073 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 1074 event->cancel(); 1075 return INVALID_OPERATION; 1076 } 1077 (void) TrackBase::setSyncEvent(event); 1078 return NO_ERROR; 1079 } 1080 1081 void AudioFlinger::PlaybackThread::Track::invalidate() 1082 { 1083 // FIXME should use proxy, and needs work 1084 audio_track_cblk_t* cblk = mCblk; 1085 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 1086 android_atomic_release_store(0x40000000, &cblk->mFutex); 1087 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 1088 (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); 1089 mIsInvalid = true; 1090 } 1091 1092 void AudioFlinger::PlaybackThread::Track::signal() 1093 { 1094 sp<ThreadBase> thread = mThread.promote(); 1095 if (thread != 0) { 1096 PlaybackThread *t = (PlaybackThread *)thread.get(); 1097 Mutex::Autolock _l(t->mLock); 1098 t->broadcast_l(); 1099 } 1100 } 1101 1102 //To be called with thread lock held 1103 bool AudioFlinger::PlaybackThread::Track::isResumePending() { 1104 1105 if (mState == RESUMING) 1106 return true; 1107 /* Resume is pending if track was stopping before pause was called */ 1108 if (mState == STOPPING_1 && 1109 mResumeToStopping) 1110 return true; 1111 1112 return false; 1113 } 1114 1115 //To be called with thread lock held 1116 void AudioFlinger::PlaybackThread::Track::resumeAck() { 1117 1118 1119 if (mState == RESUMING) 1120 mState = ACTIVE; 1121 1122 // Other possibility of pending resume is stopping_1 state 1123 // Do not update the state from stopping as this prevents 1124 // drain being called. 1125 if (mState == STOPPING_1) { 1126 mResumeToStopping = false; 1127 } 1128 } 1129 // ---------------------------------------------------------------------------- 1130 1131 sp<AudioFlinger::PlaybackThread::TimedTrack> 1132 AudioFlinger::PlaybackThread::TimedTrack::create( 1133 PlaybackThread *thread, 1134 const sp<Client>& client, 1135 audio_stream_type_t streamType, 1136 uint32_t sampleRate, 1137 audio_format_t format, 1138 audio_channel_mask_t channelMask, 1139 size_t frameCount, 1140 const sp<IMemory>& sharedBuffer, 1141 int sessionId, 1142 int uid) 1143 { 1144 if (!client->reserveTimedTrack()) 1145 return 0; 1146 1147 return new TimedTrack( 1148 thread, client, streamType, sampleRate, format, channelMask, frameCount, 1149 sharedBuffer, sessionId, uid); 1150 } 1151 1152 AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 1153 PlaybackThread *thread, 1154 const sp<Client>& client, 1155 audio_stream_type_t streamType, 1156 uint32_t sampleRate, 1157 audio_format_t format, 1158 audio_channel_mask_t channelMask, 1159 size_t frameCount, 1160 const sp<IMemory>& sharedBuffer, 1161 int sessionId, 1162 int uid) 1163 : Track(thread, client, streamType, sampleRate, format, channelMask, 1164 frameCount, (sharedBuffer != 0) ? sharedBuffer->pointer() : NULL, sharedBuffer, 1165 sessionId, uid, IAudioFlinger::TRACK_TIMED, TYPE_TIMED), 1166 mQueueHeadInFlight(false), 1167 mTrimQueueHeadOnRelease(false), 1168 mFramesPendingInQueue(0), 1169 mTimedSilenceBuffer(NULL), 1170 mTimedSilenceBufferSize(0), 1171 mTimedAudioOutputOnTime(false), 1172 mMediaTimeTransformValid(false) 1173 { 1174 LocalClock lc; 1175 mLocalTimeFreq = lc.getLocalFreq(); 1176 1177 mLocalTimeToSampleTransform.a_zero = 0; 1178 mLocalTimeToSampleTransform.b_zero = 0; 1179 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 1180 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 1181 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 1182 &mLocalTimeToSampleTransform.a_to_b_denom); 1183 1184 mMediaTimeToSampleTransform.a_zero = 0; 1185 mMediaTimeToSampleTransform.b_zero = 0; 1186 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 1187 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 1188 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 1189 &mMediaTimeToSampleTransform.a_to_b_denom); 1190 } 1191 1192 AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 1193 mClient->releaseTimedTrack(); 1194 delete [] mTimedSilenceBuffer; 1195 } 1196 1197 status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 1198 size_t size, sp<IMemory>* buffer) { 1199 1200 Mutex::Autolock _l(mTimedBufferQueueLock); 1201 1202 trimTimedBufferQueue_l(); 1203 1204 // lazily initialize the shared memory heap for timed buffers 1205 if (mTimedMemoryDealer == NULL) { 1206 const int kTimedBufferHeapSize = 512 << 10; 1207 1208 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 1209 "AudioFlingerTimed"); 1210 if (mTimedMemoryDealer == NULL) { 1211 return NO_MEMORY; 1212 } 1213 } 1214 1215 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 1216 if (newBuffer == 0 || newBuffer->pointer() == NULL) { 1217 return NO_MEMORY; 1218 } 1219 1220 *buffer = newBuffer; 1221 return NO_ERROR; 1222 } 1223 1224 // caller must hold mTimedBufferQueueLock 1225 void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 1226 int64_t mediaTimeNow; 1227 { 1228 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1229 if (!mMediaTimeTransformValid) 1230 return; 1231 1232 int64_t targetTimeNow; 1233 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 1234 ? mCCHelper.getCommonTime(&targetTimeNow) 1235 : mCCHelper.getLocalTime(&targetTimeNow); 1236 1237 if (OK != res) 1238 return; 1239 1240 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 1241 &mediaTimeNow)) { 1242 return; 1243 } 1244 } 1245 1246 size_t trimEnd; 1247 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 1248 int64_t bufEnd; 1249 1250 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 1251 // We have a next buffer. Just use its PTS as the PTS of the frame 1252 // following the last frame in this buffer. If the stream is sparse 1253 // (ie, there are deliberate gaps left in the stream which should be 1254 // filled with silence by the TimedAudioTrack), then this can result 1255 // in one extra buffer being left un-trimmed when it could have 1256 // been. In general, this is not typical, and we would rather 1257 // optimized away the TS calculation below for the more common case 1258 // where PTSes are contiguous. 1259 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 1260 } else { 1261 // We have no next buffer. Compute the PTS of the frame following 1262 // the last frame in this buffer by computing the duration of of 1263 // this frame in media time units and adding it to the PTS of the 1264 // buffer. 1265 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 1266 / mFrameSize; 1267 1268 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 1269 &bufEnd)) { 1270 ALOGE("Failed to convert frame count of %lld to media time" 1271 " duration" " (scale factor %d/%u) in %s", 1272 frameCount, 1273 mMediaTimeToSampleTransform.a_to_b_numer, 1274 mMediaTimeToSampleTransform.a_to_b_denom, 1275 __PRETTY_FUNCTION__); 1276 break; 1277 } 1278 bufEnd += mTimedBufferQueue[trimEnd].pts(); 1279 } 1280 1281 if (bufEnd > mediaTimeNow) 1282 break; 1283 1284 // Is the buffer we want to use in the middle of a mix operation right 1285 // now? If so, don't actually trim it. Just wait for the releaseBuffer 1286 // from the mixer which should be coming back shortly. 1287 if (!trimEnd && mQueueHeadInFlight) { 1288 mTrimQueueHeadOnRelease = true; 1289 } 1290 } 1291 1292 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 1293 if (trimStart < trimEnd) { 1294 // Update the bookkeeping for framesReady() 1295 for (size_t i = trimStart; i < trimEnd; ++i) { 1296 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 1297 } 1298 1299 // Now actually remove the buffers from the queue. 1300 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 1301 } 1302 } 1303 1304 void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 1305 const char* logTag) { 1306 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 1307 "%s called (reason \"%s\"), but timed buffer queue has no" 1308 " elements to trim.", __FUNCTION__, logTag); 1309 1310 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 1311 mTimedBufferQueue.removeAt(0); 1312 } 1313 1314 void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 1315 const TimedBuffer& buf, 1316 const char* logTag __unused) { 1317 uint32_t bufBytes = buf.buffer()->size(); 1318 uint32_t consumedAlready = buf.position(); 1319 1320 ALOG_ASSERT(consumedAlready <= bufBytes, 1321 "Bad bookkeeping while updating frames pending. Timed buffer is" 1322 " only %u bytes long, but claims to have consumed %u" 1323 " bytes. (update reason: \"%s\")", 1324 bufBytes, consumedAlready, logTag); 1325 1326 uint32_t bufFrames = (bufBytes - consumedAlready) / mFrameSize; 1327 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 1328 "Bad bookkeeping while updating frames pending. Should have at" 1329 " least %u queued frames, but we think we have only %u. (update" 1330 " reason: \"%s\")", 1331 bufFrames, mFramesPendingInQueue, logTag); 1332 1333 mFramesPendingInQueue -= bufFrames; 1334 } 1335 1336 status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 1337 const sp<IMemory>& buffer, int64_t pts) { 1338 1339 { 1340 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1341 if (!mMediaTimeTransformValid) 1342 return INVALID_OPERATION; 1343 } 1344 1345 Mutex::Autolock _l(mTimedBufferQueueLock); 1346 1347 uint32_t bufFrames = buffer->size() / mFrameSize; 1348 mFramesPendingInQueue += bufFrames; 1349 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 1350 1351 return NO_ERROR; 1352 } 1353 1354 status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 1355 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 1356 1357 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 1358 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 1359 target); 1360 1361 if (!(target == TimedAudioTrack::LOCAL_TIME || 1362 target == TimedAudioTrack::COMMON_TIME)) { 1363 return BAD_VALUE; 1364 } 1365 1366 Mutex::Autolock lock(mMediaTimeTransformLock); 1367 mMediaTimeTransform = xform; 1368 mMediaTimeTransformTarget = target; 1369 mMediaTimeTransformValid = true; 1370 1371 return NO_ERROR; 1372 } 1373 1374 #define min(a, b) ((a) < (b) ? (a) : (b)) 1375 1376 // implementation of getNextBuffer for tracks whose buffers have timestamps 1377 status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 1378 AudioBufferProvider::Buffer* buffer, int64_t pts) 1379 { 1380 if (pts == AudioBufferProvider::kInvalidPTS) { 1381 buffer->raw = NULL; 1382 buffer->frameCount = 0; 1383 mTimedAudioOutputOnTime = false; 1384 return INVALID_OPERATION; 1385 } 1386 1387 Mutex::Autolock _l(mTimedBufferQueueLock); 1388 1389 ALOG_ASSERT(!mQueueHeadInFlight, 1390 "getNextBuffer called without releaseBuffer!"); 1391 1392 while (true) { 1393 1394 // if we have no timed buffers, then fail 1395 if (mTimedBufferQueue.isEmpty()) { 1396 buffer->raw = NULL; 1397 buffer->frameCount = 0; 1398 return NOT_ENOUGH_DATA; 1399 } 1400 1401 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1402 1403 // calculate the PTS of the head of the timed buffer queue expressed in 1404 // local time 1405 int64_t headLocalPTS; 1406 { 1407 Mutex::Autolock mttLock(mMediaTimeTransformLock); 1408 1409 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 1410 1411 if (mMediaTimeTransform.a_to_b_denom == 0) { 1412 // the transform represents a pause, so yield silence 1413 timedYieldSilence_l(buffer->frameCount, buffer); 1414 return NO_ERROR; 1415 } 1416 1417 int64_t transformedPTS; 1418 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 1419 &transformedPTS)) { 1420 // the transform failed. this shouldn't happen, but if it does 1421 // then just drop this buffer 1422 ALOGW("timedGetNextBuffer transform failed"); 1423 buffer->raw = NULL; 1424 buffer->frameCount = 0; 1425 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 1426 return NO_ERROR; 1427 } 1428 1429 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 1430 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 1431 &headLocalPTS)) { 1432 buffer->raw = NULL; 1433 buffer->frameCount = 0; 1434 return INVALID_OPERATION; 1435 } 1436 } else { 1437 headLocalPTS = transformedPTS; 1438 } 1439 } 1440 1441 uint32_t sr = sampleRate(); 1442 1443 // adjust the head buffer's PTS to reflect the portion of the head buffer 1444 // that has already been consumed 1445 int64_t effectivePTS = headLocalPTS + 1446 ((head.position() / mFrameSize) * mLocalTimeFreq / sr); 1447 1448 // Calculate the delta in samples between the head of the input buffer 1449 // queue and the start of the next output buffer that will be written. 1450 // If the transformation fails because of over or underflow, it means 1451 // that the sample's position in the output stream is so far out of 1452 // whack that it should just be dropped. 1453 int64_t sampleDelta; 1454 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 1455 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 1456 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 1457 " mix"); 1458 continue; 1459 } 1460 if (!mLocalTimeToSampleTransform.doForwardTransform( 1461 (effectivePTS - pts) << 32, &sampleDelta)) { 1462 ALOGV("*** too late during sample rate transform: dropped buffer"); 1463 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 1464 continue; 1465 } 1466 1467 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 1468 " sampleDelta=[%d.%08x]", 1469 head.pts(), head.position(), pts, 1470 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 1471 + (sampleDelta >> 32)), 1472 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 1473 1474 // if the delta between the ideal placement for the next input sample and 1475 // the current output position is within this threshold, then we will 1476 // concatenate the next input samples to the previous output 1477 const int64_t kSampleContinuityThreshold = 1478 (static_cast<int64_t>(sr) << 32) / 250; 1479 1480 // if this is the first buffer of audio that we're emitting from this track 1481 // then it should be almost exactly on time. 1482 const int64_t kSampleStartupThreshold = 1LL << 32; 1483 1484 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 1485 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 1486 // the next input is close enough to being on time, so concatenate it 1487 // with the last output 1488 timedYieldSamples_l(buffer); 1489 1490 ALOGVV("*** on time: head.pos=%d frameCount=%u", 1491 head.position(), buffer->frameCount); 1492 return NO_ERROR; 1493 } 1494 1495 // Looks like our output is not on time. Reset our on timed status. 1496 // Next time we mix samples from our input queue, then should be within 1497 // the StartupThreshold. 1498 mTimedAudioOutputOnTime = false; 1499 if (sampleDelta > 0) { 1500 // the gap between the current output position and the proper start of 1501 // the next input sample is too big, so fill it with silence 1502 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 1503 1504 timedYieldSilence_l(framesUntilNextInput, buffer); 1505 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 1506 return NO_ERROR; 1507 } else { 1508 // the next input sample is late 1509 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 1510 size_t onTimeSamplePosition = 1511 head.position() + lateFrames * mFrameSize; 1512 1513 if (onTimeSamplePosition > head.buffer()->size()) { 1514 // all the remaining samples in the head are too late, so 1515 // drop it and move on 1516 ALOGV("*** too late: dropped buffer"); 1517 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 1518 continue; 1519 } else { 1520 // skip over the late samples 1521 head.setPosition(onTimeSamplePosition); 1522 1523 // yield the available samples 1524 timedYieldSamples_l(buffer); 1525 1526 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 1527 return NO_ERROR; 1528 } 1529 } 1530 } 1531 } 1532 1533 // Yield samples from the timed buffer queue head up to the given output 1534 // buffer's capacity. 1535 // 1536 // Caller must hold mTimedBufferQueueLock 1537 void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 1538 AudioBufferProvider::Buffer* buffer) { 1539 1540 const TimedBuffer& head = mTimedBufferQueue[0]; 1541 1542 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 1543 head.position()); 1544 1545 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 1546 mFrameSize); 1547 size_t framesRequested = buffer->frameCount; 1548 buffer->frameCount = min(framesLeftInHead, framesRequested); 1549 1550 mQueueHeadInFlight = true; 1551 mTimedAudioOutputOnTime = true; 1552 } 1553 1554 // Yield samples of silence up to the given output buffer's capacity 1555 // 1556 // Caller must hold mTimedBufferQueueLock 1557 void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 1558 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 1559 1560 // lazily allocate a buffer filled with silence 1561 if (mTimedSilenceBufferSize < numFrames * mFrameSize) { 1562 delete [] mTimedSilenceBuffer; 1563 mTimedSilenceBufferSize = numFrames * mFrameSize; 1564 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 1565 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 1566 } 1567 1568 buffer->raw = mTimedSilenceBuffer; 1569 size_t framesRequested = buffer->frameCount; 1570 buffer->frameCount = min(numFrames, framesRequested); 1571 1572 mTimedAudioOutputOnTime = false; 1573 } 1574 1575 // AudioBufferProvider interface 1576 void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 1577 AudioBufferProvider::Buffer* buffer) { 1578 1579 Mutex::Autolock _l(mTimedBufferQueueLock); 1580 1581 // If the buffer which was just released is part of the buffer at the head 1582 // of the queue, be sure to update the amt of the buffer which has been 1583 // consumed. If the buffer being returned is not part of the head of the 1584 // queue, its either because the buffer is part of the silence buffer, or 1585 // because the head of the timed queue was trimmed after the mixer called 1586 // getNextBuffer but before the mixer called releaseBuffer. 1587 if (buffer->raw == mTimedSilenceBuffer) { 1588 ALOG_ASSERT(!mQueueHeadInFlight, 1589 "Queue head in flight during release of silence buffer!"); 1590 goto done; 1591 } 1592 1593 ALOG_ASSERT(mQueueHeadInFlight, 1594 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 1595 " head in flight."); 1596 1597 if (mTimedBufferQueue.size()) { 1598 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 1599 1600 void* start = head.buffer()->pointer(); 1601 void* end = reinterpret_cast<void*>( 1602 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 1603 + head.buffer()->size()); 1604 1605 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 1606 "released buffer not within the head of the timed buffer" 1607 " queue; qHead = [%p, %p], released buffer = %p", 1608 start, end, buffer->raw); 1609 1610 head.setPosition(head.position() + 1611 (buffer->frameCount * mFrameSize)); 1612 mQueueHeadInFlight = false; 1613 1614 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 1615 "Bad bookkeeping during releaseBuffer! Should have at" 1616 " least %u queued frames, but we think we have only %u", 1617 buffer->frameCount, mFramesPendingInQueue); 1618 1619 mFramesPendingInQueue -= buffer->frameCount; 1620 1621 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 1622 || mTrimQueueHeadOnRelease) { 1623 trimTimedBufferQueueHead_l("releaseBuffer"); 1624 mTrimQueueHeadOnRelease = false; 1625 } 1626 } else { 1627 LOG_ALWAYS_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 1628 " buffers in the timed buffer queue"); 1629 } 1630 1631 done: 1632 buffer->raw = 0; 1633 buffer->frameCount = 0; 1634 } 1635 1636 size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 1637 Mutex::Autolock _l(mTimedBufferQueueLock); 1638 return mFramesPendingInQueue; 1639 } 1640 1641 AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 1642 : mPTS(0), mPosition(0) {} 1643 1644 AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 1645 const sp<IMemory>& buffer, int64_t pts) 1646 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 1647 1648 1649 // ---------------------------------------------------------------------------- 1650 1651 AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 1652 PlaybackThread *playbackThread, 1653 DuplicatingThread *sourceThread, 1654 uint32_t sampleRate, 1655 audio_format_t format, 1656 audio_channel_mask_t channelMask, 1657 size_t frameCount, 1658 int uid) 1659 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1660 NULL, 0, 0, uid, IAudioFlinger::TRACK_DEFAULT, TYPE_OUTPUT), 1661 mActive(false), mSourceThread(sourceThread), mClientProxy(NULL) 1662 { 1663 1664 if (mCblk != NULL) { 1665 mOutBuffer.frameCount = 0; 1666 playbackThread->mTracks.add(this); 1667 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, " 1668 "frameCount %u, mChannelMask 0x%08x", 1669 mCblk, mBuffer, 1670 frameCount, mChannelMask); 1671 // since client and server are in the same process, 1672 // the buffer has the same virtual address on both sides 1673 mClientProxy = new AudioTrackClientProxy(mCblk, mBuffer, mFrameCount, mFrameSize, 1674 true /*clientInServer*/); 1675 mClientProxy->setVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY); 1676 mClientProxy->setSendLevel(0.0); 1677 mClientProxy->setSampleRate(sampleRate); 1678 } else { 1679 ALOGW("Error creating output track on thread %p", playbackThread); 1680 } 1681 } 1682 1683 AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 1684 { 1685 clearBufferQueue(); 1686 delete mClientProxy; 1687 // superclass destructor will now delete the server proxy and shared memory both refer to 1688 } 1689 1690 status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 1691 int triggerSession) 1692 { 1693 status_t status = Track::start(event, triggerSession); 1694 if (status != NO_ERROR) { 1695 return status; 1696 } 1697 1698 mActive = true; 1699 mRetryCount = 127; 1700 return status; 1701 } 1702 1703 void AudioFlinger::PlaybackThread::OutputTrack::stop() 1704 { 1705 Track::stop(); 1706 clearBufferQueue(); 1707 mOutBuffer.frameCount = 0; 1708 mActive = false; 1709 } 1710 1711 bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 1712 { 1713 Buffer *pInBuffer; 1714 Buffer inBuffer; 1715 uint32_t channelCount = mChannelCount; 1716 bool outputBufferFull = false; 1717 inBuffer.frameCount = frames; 1718 inBuffer.i16 = data; 1719 1720 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 1721 1722 if (!mActive && frames != 0) { 1723 start(); 1724 sp<ThreadBase> thread = mThread.promote(); 1725 if (thread != 0) { 1726 MixerThread *mixerThread = (MixerThread *)thread.get(); 1727 if (mFrameCount > frames) { 1728 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1729 uint32_t startFrames = (mFrameCount - frames); 1730 pInBuffer = new Buffer; 1731 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 1732 pInBuffer->frameCount = startFrames; 1733 pInBuffer->i16 = pInBuffer->mBuffer; 1734 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 1735 mBufferQueue.add(pInBuffer); 1736 } else { 1737 ALOGW("OutputTrack::write() %p no more buffers in queue", this); 1738 } 1739 } 1740 } 1741 } 1742 1743 while (waitTimeLeftMs) { 1744 // First write pending buffers, then new data 1745 if (mBufferQueue.size()) { 1746 pInBuffer = mBufferQueue.itemAt(0); 1747 } else { 1748 pInBuffer = &inBuffer; 1749 } 1750 1751 if (pInBuffer->frameCount == 0) { 1752 break; 1753 } 1754 1755 if (mOutBuffer.frameCount == 0) { 1756 mOutBuffer.frameCount = pInBuffer->frameCount; 1757 nsecs_t startTime = systemTime(); 1758 status_t status = obtainBuffer(&mOutBuffer, waitTimeLeftMs); 1759 if (status != NO_ERROR) { 1760 ALOGV("OutputTrack::write() %p thread %p no more output buffers; status %d", this, 1761 mThread.unsafe_get(), status); 1762 outputBufferFull = true; 1763 break; 1764 } 1765 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 1766 if (waitTimeLeftMs >= waitTimeMs) { 1767 waitTimeLeftMs -= waitTimeMs; 1768 } else { 1769 waitTimeLeftMs = 0; 1770 } 1771 } 1772 1773 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : 1774 pInBuffer->frameCount; 1775 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 1776 Proxy::Buffer buf; 1777 buf.mFrameCount = outFrames; 1778 buf.mRaw = NULL; 1779 mClientProxy->releaseBuffer(&buf); 1780 pInBuffer->frameCount -= outFrames; 1781 pInBuffer->i16 += outFrames * channelCount; 1782 mOutBuffer.frameCount -= outFrames; 1783 mOutBuffer.i16 += outFrames * channelCount; 1784 1785 if (pInBuffer->frameCount == 0) { 1786 if (mBufferQueue.size()) { 1787 mBufferQueue.removeAt(0); 1788 delete [] pInBuffer->mBuffer; 1789 delete pInBuffer; 1790 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, 1791 mThread.unsafe_get(), mBufferQueue.size()); 1792 } else { 1793 break; 1794 } 1795 } 1796 } 1797 1798 // If we could not write all frames, allocate a buffer and queue it for next time. 1799 if (inBuffer.frameCount) { 1800 sp<ThreadBase> thread = mThread.promote(); 1801 if (thread != 0 && !thread->standby()) { 1802 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 1803 pInBuffer = new Buffer; 1804 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 1805 pInBuffer->frameCount = inBuffer.frameCount; 1806 pInBuffer->i16 = pInBuffer->mBuffer; 1807 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * 1808 sizeof(int16_t)); 1809 mBufferQueue.add(pInBuffer); 1810 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, 1811 mThread.unsafe_get(), mBufferQueue.size()); 1812 } else { 1813 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", 1814 mThread.unsafe_get(), this); 1815 } 1816 } 1817 } 1818 1819 // Calling write() with a 0 length buffer, means that no more data will be written: 1820 // If no more buffers are pending, fill output track buffer to make sure it is started 1821 // by output mixer. 1822 if (frames == 0 && mBufferQueue.size() == 0) { 1823 // FIXME borken, replace by getting framesReady() from proxy 1824 size_t user = 0; // was mCblk->user 1825 if (user < mFrameCount) { 1826 frames = mFrameCount - user; 1827 pInBuffer = new Buffer; 1828 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 1829 pInBuffer->frameCount = frames; 1830 pInBuffer->i16 = pInBuffer->mBuffer; 1831 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 1832 mBufferQueue.add(pInBuffer); 1833 } else if (mActive) { 1834 stop(); 1835 } 1836 } 1837 1838 return outputBufferFull; 1839 } 1840 1841 status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer( 1842 AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 1843 { 1844 ClientProxy::Buffer buf; 1845 buf.mFrameCount = buffer->frameCount; 1846 struct timespec timeout; 1847 timeout.tv_sec = waitTimeMs / 1000; 1848 timeout.tv_nsec = (int) (waitTimeMs % 1000) * 1000000; 1849 status_t status = mClientProxy->obtainBuffer(&buf, &timeout); 1850 buffer->frameCount = buf.mFrameCount; 1851 buffer->raw = buf.mRaw; 1852 return status; 1853 } 1854 1855 void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 1856 { 1857 size_t size = mBufferQueue.size(); 1858 1859 for (size_t i = 0; i < size; i++) { 1860 Buffer *pBuffer = mBufferQueue.itemAt(i); 1861 delete [] pBuffer->mBuffer; 1862 delete pBuffer; 1863 } 1864 mBufferQueue.clear(); 1865 } 1866 1867 1868 AudioFlinger::PlaybackThread::PatchTrack::PatchTrack(PlaybackThread *playbackThread, 1869 uint32_t sampleRate, 1870 audio_channel_mask_t channelMask, 1871 audio_format_t format, 1872 size_t frameCount, 1873 void *buffer, 1874 IAudioFlinger::track_flags_t flags) 1875 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 1876 buffer, 0, 0, getuid(), flags, TYPE_PATCH), 1877 mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true)) 1878 { 1879 uint64_t mixBufferNs = ((uint64_t)2 * playbackThread->frameCount() * 1000000000) / 1880 playbackThread->sampleRate(); 1881 mPeerTimeout.tv_sec = mixBufferNs / 1000000000; 1882 mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000); 1883 1884 ALOGV("PatchTrack %p sampleRate %d mPeerTimeout %d.%03d sec", 1885 this, sampleRate, 1886 (int)mPeerTimeout.tv_sec, 1887 (int)(mPeerTimeout.tv_nsec / 1000000)); 1888 } 1889 1890 AudioFlinger::PlaybackThread::PatchTrack::~PatchTrack() 1891 { 1892 } 1893 1894 // AudioBufferProvider interface 1895 status_t AudioFlinger::PlaybackThread::PatchTrack::getNextBuffer( 1896 AudioBufferProvider::Buffer* buffer, int64_t pts) 1897 { 1898 ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::getNextBuffer() called without peer proxy"); 1899 Proxy::Buffer buf; 1900 buf.mFrameCount = buffer->frameCount; 1901 status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout); 1902 ALOGV_IF(status != NO_ERROR, "PatchTrack() %p getNextBuffer status %d", this, status); 1903 buffer->frameCount = buf.mFrameCount; 1904 if (buf.mFrameCount == 0) { 1905 return WOULD_BLOCK; 1906 } 1907 status = Track::getNextBuffer(buffer, pts); 1908 return status; 1909 } 1910 1911 void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(AudioBufferProvider::Buffer* buffer) 1912 { 1913 ALOG_ASSERT(mPeerProxy != 0, "PatchTrack::releaseBuffer() called without peer proxy"); 1914 Proxy::Buffer buf; 1915 buf.mFrameCount = buffer->frameCount; 1916 buf.mRaw = buffer->raw; 1917 mPeerProxy->releaseBuffer(&buf); 1918 TrackBase::releaseBuffer(buffer); 1919 } 1920 1921 status_t AudioFlinger::PlaybackThread::PatchTrack::obtainBuffer(Proxy::Buffer* buffer, 1922 const struct timespec *timeOut) 1923 { 1924 return mProxy->obtainBuffer(buffer, timeOut); 1925 } 1926 1927 void AudioFlinger::PlaybackThread::PatchTrack::releaseBuffer(Proxy::Buffer* buffer) 1928 { 1929 mProxy->releaseBuffer(buffer); 1930 if (android_atomic_and(~CBLK_DISABLED, &mCblk->mFlags) & CBLK_DISABLED) { 1931 ALOGW("PatchTrack::releaseBuffer() disabled due to previous underrun, restarting"); 1932 start(); 1933 } 1934 android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags); 1935 } 1936 1937 // ---------------------------------------------------------------------------- 1938 // Record 1939 // ---------------------------------------------------------------------------- 1940 1941 AudioFlinger::RecordHandle::RecordHandle( 1942 const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 1943 : BnAudioRecord(), 1944 mRecordTrack(recordTrack) 1945 { 1946 } 1947 1948 AudioFlinger::RecordHandle::~RecordHandle() { 1949 stop_nonvirtual(); 1950 mRecordTrack->destroy(); 1951 } 1952 1953 status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, 1954 int triggerSession) { 1955 ALOGV("RecordHandle::start()"); 1956 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 1957 } 1958 1959 void AudioFlinger::RecordHandle::stop() { 1960 stop_nonvirtual(); 1961 } 1962 1963 void AudioFlinger::RecordHandle::stop_nonvirtual() { 1964 ALOGV("RecordHandle::stop()"); 1965 mRecordTrack->stop(); 1966 } 1967 1968 status_t AudioFlinger::RecordHandle::onTransact( 1969 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 1970 { 1971 return BnAudioRecord::onTransact(code, data, reply, flags); 1972 } 1973 1974 // ---------------------------------------------------------------------------- 1975 1976 // RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 1977 AudioFlinger::RecordThread::RecordTrack::RecordTrack( 1978 RecordThread *thread, 1979 const sp<Client>& client, 1980 uint32_t sampleRate, 1981 audio_format_t format, 1982 audio_channel_mask_t channelMask, 1983 size_t frameCount, 1984 void *buffer, 1985 int sessionId, 1986 int uid, 1987 IAudioFlinger::track_flags_t flags, 1988 track_type type) 1989 : TrackBase(thread, client, sampleRate, format, 1990 channelMask, frameCount, buffer, sessionId, uid, 1991 flags, false /*isOut*/, 1992 (type == TYPE_DEFAULT) ? 1993 ((flags & IAudioFlinger::TRACK_FAST) ? ALLOC_PIPE : ALLOC_CBLK) : 1994 ((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE), 1995 type), 1996 mOverflow(false), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0), 1997 // See real initialization of mRsmpInFront at RecordThread::start() 1998 mRsmpInUnrel(0), mRsmpInFront(0), mFramesToDrop(0), mResamplerBufferProvider(NULL) 1999 { 2000 if (mCblk == NULL) { 2001 return; 2002 } 2003 2004 mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, 2005 mFrameSize, !isExternalTrack()); 2006 2007 uint32_t channelCount = audio_channel_count_from_in_mask(channelMask); 2008 // FIXME I don't understand either of the channel count checks 2009 if (thread->mSampleRate != sampleRate && thread->mChannelCount <= FCC_2 && 2010 channelCount <= FCC_2) { 2011 // sink SR 2012 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT, 2013 thread->mChannelCount, sampleRate); 2014 // source SR 2015 mResampler->setSampleRate(thread->mSampleRate); 2016 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 2017 mResamplerBufferProvider = new ResamplerBufferProvider(this); 2018 } 2019 2020 if (flags & IAudioFlinger::TRACK_FAST) { 2021 ALOG_ASSERT(thread->mFastTrackAvail); 2022 thread->mFastTrackAvail = false; 2023 } 2024 } 2025 2026 AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 2027 { 2028 ALOGV("%s", __func__); 2029 delete mResampler; 2030 delete[] mRsmpOutBuffer; 2031 delete mResamplerBufferProvider; 2032 } 2033 2034 // AudioBufferProvider interface 2035 status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, 2036 int64_t pts __unused) 2037 { 2038 ServerProxy::Buffer buf; 2039 buf.mFrameCount = buffer->frameCount; 2040 status_t status = mServerProxy->obtainBuffer(&buf); 2041 buffer->frameCount = buf.mFrameCount; 2042 buffer->raw = buf.mRaw; 2043 if (buf.mFrameCount == 0) { 2044 // FIXME also wake futex so that overrun is noticed more quickly 2045 (void) android_atomic_or(CBLK_OVERRUN, &mCblk->mFlags); 2046 } 2047 return status; 2048 } 2049 2050 status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 2051 int triggerSession) 2052 { 2053 sp<ThreadBase> thread = mThread.promote(); 2054 if (thread != 0) { 2055 RecordThread *recordThread = (RecordThread *)thread.get(); 2056 return recordThread->start(this, event, triggerSession); 2057 } else { 2058 return BAD_VALUE; 2059 } 2060 } 2061 2062 void AudioFlinger::RecordThread::RecordTrack::stop() 2063 { 2064 sp<ThreadBase> thread = mThread.promote(); 2065 if (thread != 0) { 2066 RecordThread *recordThread = (RecordThread *)thread.get(); 2067 if (recordThread->stop(this) && isExternalTrack()) { 2068 AudioSystem::stopInput(mThreadIoHandle, (audio_session_t)mSessionId); 2069 } 2070 } 2071 } 2072 2073 void AudioFlinger::RecordThread::RecordTrack::destroy() 2074 { 2075 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 2076 sp<RecordTrack> keep(this); 2077 { 2078 if (isExternalTrack()) { 2079 if (mState == ACTIVE || mState == RESUMING) { 2080 AudioSystem::stopInput(mThreadIoHandle, (audio_session_t)mSessionId); 2081 } 2082 AudioSystem::releaseInput(mThreadIoHandle, (audio_session_t)mSessionId); 2083 } 2084 sp<ThreadBase> thread = mThread.promote(); 2085 if (thread != 0) { 2086 Mutex::Autolock _l(thread->mLock); 2087 RecordThread *recordThread = (RecordThread *) thread.get(); 2088 recordThread->destroyTrack_l(this); 2089 } 2090 } 2091 } 2092 2093 void AudioFlinger::RecordThread::RecordTrack::invalidate() 2094 { 2095 // FIXME should use proxy, and needs work 2096 audio_track_cblk_t* cblk = mCblk; 2097 android_atomic_or(CBLK_INVALID, &cblk->mFlags); 2098 android_atomic_release_store(0x40000000, &cblk->mFutex); 2099 // client is not in server, so FUTEX_WAKE is needed instead of FUTEX_WAKE_PRIVATE 2100 (void) syscall(__NR_futex, &cblk->mFutex, FUTEX_WAKE, INT_MAX); 2101 } 2102 2103 2104 /*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 2105 { 2106 result.append(" Active Client Fmt Chn mask Session S Server fCount SRate\n"); 2107 } 2108 2109 void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size, bool active) 2110 { 2111 snprintf(buffer, size, " %6s %6u %3u %08X %7u %1d %08X %6zu %5u\n", 2112 active ? "yes" : "no", 2113 (mClient == 0) ? getpid_cached : mClient->pid(), 2114 mFormat, 2115 mChannelMask, 2116 mSessionId, 2117 mState, 2118 mCblk->mServer, 2119 mFrameCount, 2120 mSampleRate); 2121 2122 } 2123 2124 void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event) 2125 { 2126 if (event == mSyncStartEvent) { 2127 ssize_t framesToDrop = 0; 2128 sp<ThreadBase> threadBase = mThread.promote(); 2129 if (threadBase != 0) { 2130 // TODO: use actual buffer filling status instead of 2 buffers when info is available 2131 // from audio HAL 2132 framesToDrop = threadBase->mFrameCount * 2; 2133 } 2134 mFramesToDrop = framesToDrop; 2135 } 2136 } 2137 2138 void AudioFlinger::RecordThread::RecordTrack::clearSyncStartEvent() 2139 { 2140 if (mSyncStartEvent != 0) { 2141 mSyncStartEvent->cancel(); 2142 mSyncStartEvent.clear(); 2143 } 2144 mFramesToDrop = 0; 2145 } 2146 2147 2148 AudioFlinger::RecordThread::PatchRecord::PatchRecord(RecordThread *recordThread, 2149 uint32_t sampleRate, 2150 audio_channel_mask_t channelMask, 2151 audio_format_t format, 2152 size_t frameCount, 2153 void *buffer, 2154 IAudioFlinger::track_flags_t flags) 2155 : RecordTrack(recordThread, NULL, sampleRate, format, channelMask, frameCount, 2156 buffer, 0, getuid(), flags, TYPE_PATCH), 2157 mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true)) 2158 { 2159 uint64_t mixBufferNs = ((uint64_t)2 * recordThread->frameCount() * 1000000000) / 2160 recordThread->sampleRate(); 2161 mPeerTimeout.tv_sec = mixBufferNs / 1000000000; 2162 mPeerTimeout.tv_nsec = (int) (mixBufferNs % 1000000000); 2163 2164 ALOGV("PatchRecord %p sampleRate %d mPeerTimeout %d.%03d sec", 2165 this, sampleRate, 2166 (int)mPeerTimeout.tv_sec, 2167 (int)(mPeerTimeout.tv_nsec / 1000000)); 2168 } 2169 2170 AudioFlinger::RecordThread::PatchRecord::~PatchRecord() 2171 { 2172 } 2173 2174 // AudioBufferProvider interface 2175 status_t AudioFlinger::RecordThread::PatchRecord::getNextBuffer( 2176 AudioBufferProvider::Buffer* buffer, int64_t pts) 2177 { 2178 ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::getNextBuffer() called without peer proxy"); 2179 Proxy::Buffer buf; 2180 buf.mFrameCount = buffer->frameCount; 2181 status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout); 2182 ALOGV_IF(status != NO_ERROR, 2183 "PatchRecord() %p mPeerProxy->obtainBuffer status %d", this, status); 2184 buffer->frameCount = buf.mFrameCount; 2185 if (buf.mFrameCount == 0) { 2186 return WOULD_BLOCK; 2187 } 2188 status = RecordTrack::getNextBuffer(buffer, pts); 2189 return status; 2190 } 2191 2192 void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(AudioBufferProvider::Buffer* buffer) 2193 { 2194 ALOG_ASSERT(mPeerProxy != 0, "PatchRecord::releaseBuffer() called without peer proxy"); 2195 Proxy::Buffer buf; 2196 buf.mFrameCount = buffer->frameCount; 2197 buf.mRaw = buffer->raw; 2198 mPeerProxy->releaseBuffer(&buf); 2199 TrackBase::releaseBuffer(buffer); 2200 } 2201 2202 status_t AudioFlinger::RecordThread::PatchRecord::obtainBuffer(Proxy::Buffer* buffer, 2203 const struct timespec *timeOut) 2204 { 2205 return mProxy->obtainBuffer(buffer, timeOut); 2206 } 2207 2208 void AudioFlinger::RecordThread::PatchRecord::releaseBuffer(Proxy::Buffer* buffer) 2209 { 2210 mProxy->releaseBuffer(buffer); 2211 } 2212 2213 }; // namespace android 2214
