media/filters/audio_renderer_impl.cc

   1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 #include "media/filters/audio_renderer_impl.h"
   6
   7 #include <math.h>
   8
   9 #include <algorithm>
  10
  11 #include "base/bind.h"
  12 #include "base/callback.h"
  13 #include "base/callback_helpers.h"
  14 #include "base/logging.h"
  15 #include "base/metrics/histogram.h"
  16 #include "base/single_thread_task_runner.h"
  17 #include "media/base/audio_buffer.h"
  18 #include "media/base/audio_buffer_converter.h"
  19 #include "media/base/audio_hardware_config.h"
  20 #include "media/base/audio_splicer.h"
  21 #include "media/base/bind_to_current_loop.h"
  22 #include "media/base/demuxer_stream.h"
  23 #include "media/filters/audio_clock.h"
  24 #include "media/filters/decrypting_demuxer_stream.h"
  25
  26 namespace media {
  27
  28 namespace {
  29
  30 enum AudioRendererEvent {
  31   INITIALIZED,
  32   RENDER_ERROR,
  33   RENDER_EVENT_MAX = RENDER_ERROR,
  34 };
  35
  36 void HistogramRendererEvent(AudioRendererEvent event) {
  37   UMA_HISTOGRAM_ENUMERATION(
  38       "Media.AudioRendererEvents", event, RENDER_EVENT_MAX + 1);
  39 }
  40
  41 }  // namespace
  42
  43 AudioRendererImpl::AudioRendererImpl(
  44     const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
  45     media::AudioRendererSink* sink,
  46     ScopedVector<AudioDecoder> decoders,
  47     const SetDecryptorReadyCB& set_decryptor_ready_cb,
  48     const AudioHardwareConfig& hardware_config,
  49     const scoped_refptr<MediaLog>& media_log)
  50     : task_runner_(task_runner),
  51       expecting_config_changes_(false),
  52       sink_(sink),
  53       audio_buffer_stream_(new AudioBufferStream(task_runner,
  54                                                  decoders.Pass(),
  55                                                  set_decryptor_ready_cb,
  56                                                  media_log)),
  57       hardware_config_(hardware_config),
  58       playback_rate_(0),
  59       state_(kUninitialized),
  60       buffering_state_(BUFFERING_HAVE_NOTHING),
  61       rendering_(false),
  62       sink_playing_(false),
  63       pending_read_(false),
  64       received_end_of_stream_(false),
  65       rendered_end_of_stream_(false),
  66       weak_factory_(this) {
  67   audio_buffer_stream_->set_splice_observer(base::Bind(
  68       &AudioRendererImpl::OnNewSpliceBuffer, weak_factory_.GetWeakPtr()));
  69   audio_buffer_stream_->set_config_change_observer(base::Bind(
  70       &AudioRendererImpl::OnConfigChange, weak_factory_.GetWeakPtr()));
  71 }
  72
  73 AudioRendererImpl::~AudioRendererImpl() {
  74   DVLOG(1) << __FUNCTION__;
  75   DCHECK(task_runner_->BelongsToCurrentThread());
  76
  77   // If Render() is in progress, this call will wait for Render() to finish.
  78   // After this call, the |sink_| will not call back into |this| anymore.
  79   sink_->Stop();
  80
  81   if (!init_cb_.is_null())
  82     base::ResetAndReturn(&init_cb_).Run(PIPELINE_ERROR_ABORT);
  83 }
  84
  85 void AudioRendererImpl::StartTicking() {
  86   DVLOG(1) << __FUNCTION__;
  87   DCHECK(task_runner_->BelongsToCurrentThread());
  88   DCHECK(!rendering_);
  89   rendering_ = true;
  90
  91   base::AutoLock auto_lock(lock_);
  92   // Wait for an eventual call to SetPlaybackRate() to start rendering.
  93   if (playback_rate_ == 0) {
  94     DCHECK(!sink_playing_);
  95     return;
  96   }
  97
  98   StartRendering_Locked();
  99 }
 100
 101 void AudioRendererImpl::StartRendering_Locked() {
 102   DVLOG(1) << __FUNCTION__;
 103   DCHECK(task_runner_->BelongsToCurrentThread());
 104   DCHECK_EQ(state_, kPlaying);
 105   DCHECK(!sink_playing_);
 106   DCHECK_NE(playback_rate_, 0);
 107   lock_.AssertAcquired();
 108
 109   sink_playing_ = true;
 110
 111   base::AutoUnlock auto_unlock(lock_);
 112   sink_->Play();
 113 }
 114
 115 void AudioRendererImpl::StopTicking() {
 116   DVLOG(1) << __FUNCTION__;
 117   DCHECK(task_runner_->BelongsToCurrentThread());
 118   DCHECK(rendering_);
 119   rendering_ = false;
 120
 121   base::AutoLock auto_lock(lock_);
 122   // Rendering should have already been stopped with a zero playback rate.
 123   if (playback_rate_ == 0) {
 124     DCHECK(!sink_playing_);
 125     return;
 126   }
 127
 128   StopRendering_Locked();
 129 }
 130
 131 void AudioRendererImpl::StopRendering_Locked() {
 132   DCHECK(task_runner_->BelongsToCurrentThread());
 133   DCHECK_EQ(state_, kPlaying);
 134   DCHECK(sink_playing_);
 135   lock_.AssertAcquired();
 136
 137   sink_playing_ = false;
 138
 139   base::AutoUnlock auto_unlock(lock_);
 140   sink_->Pause();
 141 }
 142
 143 void AudioRendererImpl::SetMediaTime(base::TimeDelta time) {
 144   DVLOG(1) << __FUNCTION__ << "(" << time.InMicroseconds() << ")";
 145   DCHECK(task_runner_->BelongsToCurrentThread());
 146
 147   base::AutoLock auto_lock(lock_);
 148   DCHECK(!rendering_);
 149   DCHECK_EQ(state_, kFlushed);
 150
 151   start_timestamp_ = time;
 152   ended_timestamp_ = kInfiniteDuration();
 153   last_render_ticks_ = base::TimeTicks();
 154   audio_clock_.reset(new AudioClock(time, audio_parameters_.sample_rate()));
 155 }
 156
 157 base::TimeDelta AudioRendererImpl::CurrentMediaTime() {
 158   DVLOG(2) << __FUNCTION__;
 159
 160   // In practice the Render() method is called with a high enough frequency
 161   // that returning only the front timestamp is good enough and also prevents
 162   // returning values that go backwards in time.
 163   base::AutoLock auto_lock(lock_);
 164   return audio_clock_->front_timestamp();
 165 }
 166
 167 base::TimeDelta AudioRendererImpl::CurrentMediaTimeForSyncingVideo() {
 168   DVLOG(2) << __FUNCTION__;
 169
 170   base::AutoLock auto_lock(lock_);
 171   if (last_render_ticks_.is_null())
 172     return audio_clock_->front_timestamp();
 173
 174   return audio_clock_->TimestampSinceWriting(base::TimeTicks::Now() -
 175                                              last_render_ticks_);
 176 }
 177
 178 TimeSource* AudioRendererImpl::GetTimeSource() {
 179   return this;
 180 }
 181
 182 void AudioRendererImpl::Flush(const base::Closure& callback) {
 183   DVLOG(1) << __FUNCTION__;
 184   DCHECK(task_runner_->BelongsToCurrentThread());
 185
 186   base::AutoLock auto_lock(lock_);
 187   DCHECK_EQ(state_, kPlaying);
 188   DCHECK(flush_cb_.is_null());
 189
 190   flush_cb_ = callback;
 191   ChangeState_Locked(kFlushing);
 192
 193   if (pending_read_)
 194     return;
 195
 196   ChangeState_Locked(kFlushed);
 197   DoFlush_Locked();
 198 }
 199
 200 void AudioRendererImpl::DoFlush_Locked() {
 201   DCHECK(task_runner_->BelongsToCurrentThread());
 202   lock_.AssertAcquired();
 203
 204   DCHECK(!pending_read_);
 205   DCHECK_EQ(state_, kFlushed);
 206
 207   audio_buffer_stream_->Reset(base::Bind(&AudioRendererImpl::ResetDecoderDone,
 208                                          weak_factory_.GetWeakPtr()));
 209 }
 210
 211 void AudioRendererImpl::ResetDecoderDone() {
 212   DCHECK(task_runner_->BelongsToCurrentThread());
 213   {
 214     base::AutoLock auto_lock(lock_);
 215
 216     DCHECK_EQ(state_, kFlushed);
 217     DCHECK(!flush_cb_.is_null());
 218
 219     received_end_of_stream_ = false;
 220     rendered_end_of_stream_ = false;
 221
 222     // Flush() may have been called while underflowed/not fully buffered.
 223     if (buffering_state_ != BUFFERING_HAVE_NOTHING)
 224       SetBufferingState_Locked(BUFFERING_HAVE_NOTHING);
 225
 226     splicer_->Reset();
 227     if (buffer_converter_)
 228       buffer_converter_->Reset();
 229     algorithm_->FlushBuffers();
 230   }
 231
 232   // Changes in buffering state are always posted. Flush callback must only be
 233   // run after buffering state has been set back to nothing.
 234   task_runner_->PostTask(FROM_HERE, base::ResetAndReturn(&flush_cb_));
 235 }
 236
 237 void AudioRendererImpl::StartPlaying() {
 238   DVLOG(1) << __FUNCTION__;
 239   DCHECK(task_runner_->BelongsToCurrentThread());
 240
 241   base::AutoLock auto_lock(lock_);
 242   DCHECK(!sink_playing_);
 243   DCHECK_EQ(state_, kFlushed);
 244   DCHECK_EQ(buffering_state_, BUFFERING_HAVE_NOTHING);
 245   DCHECK(!pending_read_) << "Pending read must complete before seeking";
 246
 247   ChangeState_Locked(kPlaying);
 248   AttemptRead_Locked();
 249 }
 250
 251 void AudioRendererImpl::Initialize(DemuxerStream* stream,
 252                                    const PipelineStatusCB& init_cb,
 253                                    const StatisticsCB& statistics_cb,
 254                                    const BufferingStateCB& buffering_state_cb,
 255                                    const base::Closure& ended_cb,
 256                                    const PipelineStatusCB& error_cb) {
 257   DCHECK(task_runner_->BelongsToCurrentThread());
 258   DCHECK(stream);
 259   DCHECK_EQ(stream->type(), DemuxerStream::AUDIO);
 260   DCHECK(!init_cb.is_null());
 261   DCHECK(!statistics_cb.is_null());
 262   DCHECK(!buffering_state_cb.is_null());
 263   DCHECK(!ended_cb.is_null());
 264   DCHECK(!error_cb.is_null());
 265   DCHECK_EQ(kUninitialized, state_);
 266   DCHECK(sink_.get());
 267
 268   state_ = kInitializing;
 269
 270   // Always post |init_cb_| because |this| could be destroyed if initialization
 271   // failed.
 272   init_cb_ = BindToCurrentLoop(init_cb);
 273
 274   buffering_state_cb_ = buffering_state_cb;
 275   ended_cb_ = ended_cb;
 276   error_cb_ = error_cb;
 277
 278   expecting_config_changes_ = stream->SupportsConfigChanges();
 279   if (!expecting_config_changes_) {
 280     // The actual buffer size is controlled via the size of the AudioBus
 281     // provided to Render(), so just choose something reasonable here for looks.
 282     int buffer_size = stream->audio_decoder_config().samples_per_second() / 100;
 283     audio_parameters_.Reset(
 284         AudioParameters::AUDIO_PCM_LOW_LATENCY,
 285         stream->audio_decoder_config().channel_layout(),
 286         ChannelLayoutToChannelCount(
 287             stream->audio_decoder_config().channel_layout()),
 288         stream->audio_decoder_config().samples_per_second(),
 289         stream->audio_decoder_config().bits_per_channel(),
 290         buffer_size);
 291     buffer_converter_.reset();
 292   } else {
 293     // TODO(rileya): Support hardware config changes
 294     const AudioParameters& hw_params = hardware_config_.GetOutputConfig();
 295     audio_parameters_.Reset(
 296         hw_params.format(),
 297         // Always use the source's channel layout and channel count to avoid
 298         // premature downmixing (http://crbug.com/379288), platform specific
 299         // issues around channel layouts (http://crbug.com/266674), and
 300         // unnecessary upmixing overhead.
 301         stream->audio_decoder_config().channel_layout(),
 302         ChannelLayoutToChannelCount(
 303             stream->audio_decoder_config().channel_layout()),
 304         hw_params.sample_rate(),
 305         hw_params.bits_per_sample(),
 306         hardware_config_.GetHighLatencyBufferSize());
 307   }
 308
 309   audio_clock_.reset(
 310       new AudioClock(base::TimeDelta(), audio_parameters_.sample_rate()));
 311
 312   audio_buffer_stream_->Initialize(
 313       stream,
 314       false,
 315       statistics_cb,
 316       base::Bind(&AudioRendererImpl::OnAudioBufferStreamInitialized,
 317                  weak_factory_.GetWeakPtr()));
 318 }
 319
 320 void AudioRendererImpl::OnAudioBufferStreamInitialized(bool success) {
 321   DCHECK(task_runner_->BelongsToCurrentThread());
 322
 323   base::AutoLock auto_lock(lock_);
 324
 325   if (!success) {
 326     state_ = kUninitialized;
 327     base::ResetAndReturn(&init_cb_).Run(DECODER_ERROR_NOT_SUPPORTED);
 328     return;
 329   }
 330
 331   if (!audio_parameters_.IsValid()) {
 332     ChangeState_Locked(kUninitialized);
 333     base::ResetAndReturn(&init_cb_).Run(PIPELINE_ERROR_INITIALIZATION_FAILED);
 334     return;
 335   }
 336
 337   if (expecting_config_changes_)
 338     buffer_converter_.reset(new AudioBufferConverter(audio_parameters_));
 339   splicer_.reset(new AudioSplicer(audio_parameters_.sample_rate()));
 340
 341   // We're all good! Continue initializing the rest of the audio renderer
 342   // based on the decoder format.
 343   algorithm_.reset(new AudioRendererAlgorithm());
 344   algorithm_->Initialize(audio_parameters_);
 345
 346   ChangeState_Locked(kFlushed);
 347
 348   HistogramRendererEvent(INITIALIZED);
 349
 350   {
 351     base::AutoUnlock auto_unlock(lock_);
 352     sink_->Initialize(audio_parameters_, this);
 353     sink_->Start();
 354
 355     // Some sinks play on start...
 356     sink_->Pause();
 357   }
 358
 359   DCHECK(!sink_playing_);
 360   base::ResetAndReturn(&init_cb_).Run(PIPELINE_OK);
 361 }
 362
 363 void AudioRendererImpl::SetVolume(float volume) {
 364   DCHECK(task_runner_->BelongsToCurrentThread());
 365   DCHECK(sink_.get());
 366   sink_->SetVolume(volume);
 367 }
 368
 369 void AudioRendererImpl::DecodedAudioReady(
 370     AudioBufferStream::Status status,
 371     const scoped_refptr<AudioBuffer>& buffer) {
 372   DVLOG(2) << __FUNCTION__ << "(" << status << ")";
 373   DCHECK(task_runner_->BelongsToCurrentThread());
 374
 375   base::AutoLock auto_lock(lock_);
 376   DCHECK(state_ != kUninitialized);
 377
 378   CHECK(pending_read_);
 379   pending_read_ = false;
 380
 381   if (status == AudioBufferStream::ABORTED ||
 382       status == AudioBufferStream::DEMUXER_READ_ABORTED) {
 383     HandleAbortedReadOrDecodeError(false);
 384     return;
 385   }
 386
 387   if (status == AudioBufferStream::DECODE_ERROR) {
 388     HandleAbortedReadOrDecodeError(true);
 389     return;
 390   }
 391
 392   DCHECK_EQ(status, AudioBufferStream::OK);
 393   DCHECK(buffer.get());
 394
 395   if (state_ == kFlushing) {
 396     ChangeState_Locked(kFlushed);
 397     DoFlush_Locked();
 398     return;
 399   }
 400
 401   if (expecting_config_changes_) {
 402     DCHECK(buffer_converter_);
 403     buffer_converter_->AddInput(buffer);
 404     while (buffer_converter_->HasNextBuffer()) {
 405       if (!splicer_->AddInput(buffer_converter_->GetNextBuffer())) {
 406         HandleAbortedReadOrDecodeError(true);
 407         return;
 408       }
 409     }
 410   } else {
 411     if (!splicer_->AddInput(buffer)) {
 412       HandleAbortedReadOrDecodeError(true);
 413       return;
 414     }
 415   }
 416
 417   if (!splicer_->HasNextBuffer()) {
 418     AttemptRead_Locked();
 419     return;
 420   }
 421
 422   bool need_another_buffer = false;
 423   while (splicer_->HasNextBuffer())
 424     need_another_buffer = HandleSplicerBuffer_Locked(splicer_->GetNextBuffer());
 425
 426   if (!need_another_buffer && !CanRead_Locked())
 427     return;
 428
 429   AttemptRead_Locked();
 430 }
 431
 432 bool AudioRendererImpl::HandleSplicerBuffer_Locked(
 433     const scoped_refptr<AudioBuffer>& buffer) {
 434   lock_.AssertAcquired();
 435   if (buffer->end_of_stream()) {
 436     received_end_of_stream_ = true;
 437   } else {
 438     if (state_ == kPlaying) {
 439       if (IsBeforeStartTime(buffer))
 440         return true;
 441
 442       // Trim off any additional time before the start timestamp.
 443       const base::TimeDelta trim_time = start_timestamp_ - buffer->timestamp();
 444       if (trim_time > base::TimeDelta()) {
 445         buffer->TrimStart(buffer->frame_count() *
 446                           (static_cast<double>(trim_time.InMicroseconds()) /
 447                            buffer->duration().InMicroseconds()));
 448       }
 449       // If the entire buffer was trimmed, request a new one.
 450       if (!buffer->frame_count())
 451         return true;
 452     }
 453
 454     if (state_ != kUninitialized)
 455       algorithm_->EnqueueBuffer(buffer);
 456   }
 457
 458   switch (state_) {
 459     case kUninitialized:
 460     case kInitializing:
 461     case kFlushing:
 462       NOTREACHED();
 463       return false;
 464
 465     case kFlushed:
 466       DCHECK(!pending_read_);
 467       return false;
 468
 469     case kPlaying:
 470       if (buffer->end_of_stream() || algorithm_->IsQueueFull()) {
 471         if (buffering_state_ == BUFFERING_HAVE_NOTHING)
 472           SetBufferingState_Locked(BUFFERING_HAVE_ENOUGH);
 473         return false;
 474       }
 475       return true;
 476   }
 477   return false;
 478 }
 479
 480 void AudioRendererImpl::AttemptRead() {
 481   base::AutoLock auto_lock(lock_);
 482   AttemptRead_Locked();
 483 }
 484
 485 void AudioRendererImpl::AttemptRead_Locked() {
 486   DCHECK(task_runner_->BelongsToCurrentThread());
 487   lock_.AssertAcquired();
 488
 489   if (!CanRead_Locked())
 490     return;
 491
 492   pending_read_ = true;
 493   audio_buffer_stream_->Read(base::Bind(&AudioRendererImpl::DecodedAudioReady,
 494                                         weak_factory_.GetWeakPtr()));
 495 }
 496
 497 bool AudioRendererImpl::CanRead_Locked() {
 498   lock_.AssertAcquired();
 499
 500   switch (state_) {
 501     case kUninitialized:
 502     case kInitializing:
 503     case kFlushing:
 504     case kFlushed:
 505       return false;
 506
 507     case kPlaying:
 508       break;
 509   }
 510
 511   return !pending_read_ && !received_end_of_stream_ &&
 512       !algorithm_->IsQueueFull();
 513 }
 514
 515 void AudioRendererImpl::SetPlaybackRate(float playback_rate) {
 516   DVLOG(1) << __FUNCTION__ << "(" << playback_rate << ")";
 517   DCHECK(task_runner_->BelongsToCurrentThread());
 518   DCHECK_GE(playback_rate, 0);
 519   DCHECK(sink_.get());
 520
 521   base::AutoLock auto_lock(lock_);
 522
 523   // We have two cases here:
 524   // Play: current_playback_rate == 0 && playback_rate != 0
 525   // Pause: current_playback_rate != 0 && playback_rate == 0
 526   float current_playback_rate = playback_rate_;
 527   playback_rate_ = playback_rate;
 528
 529   if (!rendering_)
 530     return;
 531
 532   if (current_playback_rate == 0 && playback_rate != 0) {
 533     StartRendering_Locked();
 534     return;
 535   }
 536
 537   if (current_playback_rate != 0 && playback_rate == 0) {
 538     StopRendering_Locked();
 539     return;
 540   }
 541 }
 542
 543 bool AudioRendererImpl::IsBeforeStartTime(
 544     const scoped_refptr<AudioBuffer>& buffer) {
 545   DCHECK_EQ(state_, kPlaying);
 546   return buffer.get() && !buffer->end_of_stream() &&
 547          (buffer->timestamp() + buffer->duration()) < start_timestamp_;
 548 }
 549
 550 int AudioRendererImpl::Render(AudioBus* audio_bus,
 551                               int audio_delay_milliseconds) {
 552   const int requested_frames = audio_bus->frames();
 553   base::TimeDelta playback_delay = base::TimeDelta::FromMilliseconds(
 554       audio_delay_milliseconds);
 555   const int delay_frames = static_cast<int>(playback_delay.InSecondsF() *
 556                                             audio_parameters_.sample_rate());
 557   int frames_written = 0;
 558   {
 559     base::AutoLock auto_lock(lock_);
 560     last_render_ticks_ = base::TimeTicks::Now();
 561
 562     // Ensure Stop() hasn't destroyed our |algorithm_| on the pipeline thread.
 563     if (!algorithm_) {
 564       audio_clock_->WroteAudio(
 565           0, requested_frames, delay_frames, playback_rate_);
 566       return 0;
 567     }
 568
 569     if (playback_rate_ == 0) {
 570       audio_clock_->WroteAudio(
 571           0, requested_frames, delay_frames, playback_rate_);
 572       return 0;
 573     }
 574
 575     // Mute audio by returning 0 when not playing.
 576     if (state_ != kPlaying) {
 577       audio_clock_->WroteAudio(
 578           0, requested_frames, delay_frames, playback_rate_);
 579       return 0;
 580     }
 581
 582     // We use the following conditions to determine end of playback:
 583     //   1) Algorithm can not fill the audio callback buffer
 584     //   2) We received an end of stream buffer
 585     //   3) We haven't already signalled that we've ended
 586     //   4) We've played all known audio data sent to hardware
 587     //
 588     // We use the following conditions to determine underflow:
 589     //   1) Algorithm can not fill the audio callback buffer
 590     //   2) We have NOT received an end of stream buffer
 591     //   3) We are in the kPlaying state
 592     //
 593     // Otherwise the buffer has data we can send to the device.
 594     if (algorithm_->frames_buffered() > 0) {
 595       frames_written =
 596           algorithm_->FillBuffer(audio_bus, requested_frames, playback_rate_);
 597     }
 598
 599     // Per the TimeSource API the media time should always increase even after
 600     // we've rendered all known audio data. Doing so simplifies scenarios where
 601     // we have other sources of media data that need to be scheduled after audio
 602     // data has ended.
 603     //
 604     // That being said, we don't want to advance time when underflowed as we
 605     // know more decoded frames will eventually arrive. If we did, we would
 606     // throw things out of sync when said decoded frames arrive.
 607     int frames_after_end_of_stream = 0;
 608     if (frames_written == 0) {
 609       if (received_end_of_stream_) {
 610         if (ended_timestamp_ == kInfiniteDuration())
 611           ended_timestamp_ = audio_clock_->back_timestamp();
 612         frames_after_end_of_stream = requested_frames;
 613       } else if (state_ == kPlaying &&
 614                  buffering_state_ != BUFFERING_HAVE_NOTHING) {
 615         algorithm_->IncreaseQueueCapacity();
 616         SetBufferingState_Locked(BUFFERING_HAVE_NOTHING);
 617       }
 618     }
 619
 620     audio_clock_->WroteAudio(frames_written + frames_after_end_of_stream,
 621                              requested_frames,
 622                              delay_frames,
 623                              playback_rate_);
 624
 625     if (CanRead_Locked()) {
 626       task_runner_->PostTask(FROM_HERE,
 627                              base::Bind(&AudioRendererImpl::AttemptRead,
 628                                         weak_factory_.GetWeakPtr()));
 629     }
 630
 631     if (audio_clock_->front_timestamp() >= ended_timestamp_ &&
 632         !rendered_end_of_stream_) {
 633       rendered_end_of_stream_ = true;
 634       task_runner_->PostTask(FROM_HERE, ended_cb_);
 635     }
 636   }
 637
 638   DCHECK_LE(frames_written, requested_frames);
 639   return frames_written;
 640 }
 641
 642 void AudioRendererImpl::OnRenderError() {
 643   // UMA data tells us this happens ~0.01% of the time. Trigger an error instead
 644   // of trying to gracefully fall back to a fake sink. It's very likely
 645   // OnRenderError() should be removed and the audio stack handle errors without
 646   // notifying clients. See http://crbug.com/234708 for details.
 647   HistogramRendererEvent(RENDER_ERROR);
 648   // Post to |task_runner_| as this is called on the audio callback thread.
 649   task_runner_->PostTask(FROM_HERE,
 650                          base::Bind(error_cb_, PIPELINE_ERROR_DECODE));
 651 }
 652
 653 void AudioRendererImpl::HandleAbortedReadOrDecodeError(bool is_decode_error) {
 654   DCHECK(task_runner_->BelongsToCurrentThread());
 655   lock_.AssertAcquired();
 656
 657   PipelineStatus status = is_decode_error ? PIPELINE_ERROR_DECODE : PIPELINE_OK;
 658   switch (state_) {
 659     case kUninitialized:
 660     case kInitializing:
 661       NOTREACHED();
 662       return;
 663     case kFlushing:
 664       ChangeState_Locked(kFlushed);
 665       if (status == PIPELINE_OK) {
 666         DoFlush_Locked();
 667         return;
 668       }
 669
 670       error_cb_.Run(status);
 671       base::ResetAndReturn(&flush_cb_).Run();
 672       return;
 673
 674     case kFlushed:
 675     case kPlaying:
 676       if (status != PIPELINE_OK)
 677         error_cb_.Run(status);
 678       return;
 679   }
 680 }
 681
 682 void AudioRendererImpl::ChangeState_Locked(State new_state) {
 683   DVLOG(1) << __FUNCTION__ << " : " << state_ << " -> " << new_state;
 684   lock_.AssertAcquired();
 685   state_ = new_state;
 686 }
 687
 688 void AudioRendererImpl::OnNewSpliceBuffer(base::TimeDelta splice_timestamp) {
 689   DCHECK(task_runner_->BelongsToCurrentThread());
 690   splicer_->SetSpliceTimestamp(splice_timestamp);
 691 }
 692
 693 void AudioRendererImpl::OnConfigChange() {
 694   DCHECK(task_runner_->BelongsToCurrentThread());
 695   DCHECK(expecting_config_changes_);
 696   buffer_converter_->ResetTimestampState();
 697   // Drain flushed buffers from the converter so the AudioSplicer receives all
 698   // data ahead of any OnNewSpliceBuffer() calls.  Since discontinuities should
 699   // only appear after config changes, AddInput() should never fail here.
 700   while (buffer_converter_->HasNextBuffer())
 701     CHECK(splicer_->AddInput(buffer_converter_->GetNextBuffer()));
 702 }
 703
 704 void AudioRendererImpl::SetBufferingState_Locked(
 705     BufferingState buffering_state) {
 706   DVLOG(1) << __FUNCTION__ << " : " << buffering_state_ << " -> "
 707            << buffering_state;
 708   DCHECK_NE(buffering_state_, buffering_state);
 709   lock_.AssertAcquired();
 710   buffering_state_ = buffering_state;
 711
 712   task_runner_->PostTask(FROM_HERE,
 713                          base::Bind(buffering_state_cb_, buffering_state_));
 714 }
 715
 716 }  // namespace media