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   // In practice the Render() method is called with a high enough frequency
 159   // that returning only the front timestamp is good enough and also prevents
 160   // returning values that go backwards in time.
 161   base::TimeDelta current_media_time;
 162   {
 163     base::AutoLock auto_lock(lock_);
 164     current_media_time = audio_clock_->front_timestamp();
 165   }
 166
 167   DVLOG(3) << __FUNCTION__ << ": " << current_media_time.InMilliseconds()
 168            << " ms";
 169   return current_media_time;
 170 }
 171
 172 base::TimeDelta AudioRendererImpl::CurrentMediaTimeForSyncingVideo() {
 173   DVLOG(2) << __FUNCTION__;
 174
 175   base::AutoLock auto_lock(lock_);
 176   if (last_render_ticks_.is_null())
 177     return audio_clock_->front_timestamp();
 178
 179   return audio_clock_->TimestampSinceWriting(base::TimeTicks::Now() -
 180                                              last_render_ticks_);
 181 }
 182
 183 TimeSource* AudioRendererImpl::GetTimeSource() {
 184   return this;
 185 }
 186
 187 void AudioRendererImpl::Flush(const base::Closure& callback) {
 188   DVLOG(1) << __FUNCTION__;
 189   DCHECK(task_runner_->BelongsToCurrentThread());
 190
 191   base::AutoLock auto_lock(lock_);
 192   DCHECK_EQ(state_, kPlaying);
 193   DCHECK(flush_cb_.is_null());
 194
 195   flush_cb_ = callback;
 196   ChangeState_Locked(kFlushing);
 197
 198   if (pending_read_)
 199     return;
 200
 201   ChangeState_Locked(kFlushed);
 202   DoFlush_Locked();
 203 }
 204
 205 void AudioRendererImpl::DoFlush_Locked() {
 206   DCHECK(task_runner_->BelongsToCurrentThread());
 207   lock_.AssertAcquired();
 208
 209   DCHECK(!pending_read_);
 210   DCHECK_EQ(state_, kFlushed);
 211
 212   audio_buffer_stream_->Reset(base::Bind(&AudioRendererImpl::ResetDecoderDone,
 213                                          weak_factory_.GetWeakPtr()));
 214 }
 215
 216 void AudioRendererImpl::ResetDecoderDone() {
 217   DCHECK(task_runner_->BelongsToCurrentThread());
 218   {
 219     base::AutoLock auto_lock(lock_);
 220
 221     DCHECK_EQ(state_, kFlushed);
 222     DCHECK(!flush_cb_.is_null());
 223
 224     received_end_of_stream_ = false;
 225     rendered_end_of_stream_ = false;
 226
 227     // Flush() may have been called while underflowed/not fully buffered.
 228     if (buffering_state_ != BUFFERING_HAVE_NOTHING)
 229       SetBufferingState_Locked(BUFFERING_HAVE_NOTHING);
 230
 231     splicer_->Reset();
 232     if (buffer_converter_)
 233       buffer_converter_->Reset();
 234     algorithm_->FlushBuffers();
 235   }
 236
 237   // Changes in buffering state are always posted. Flush callback must only be
 238   // run after buffering state has been set back to nothing.
 239   task_runner_->PostTask(FROM_HERE, base::ResetAndReturn(&flush_cb_));
 240 }
 241
 242 void AudioRendererImpl::StartPlaying() {
 243   DVLOG(1) << __FUNCTION__;
 244   DCHECK(task_runner_->BelongsToCurrentThread());
 245
 246   base::AutoLock auto_lock(lock_);
 247   DCHECK(!sink_playing_);
 248   DCHECK_EQ(state_, kFlushed);
 249   DCHECK_EQ(buffering_state_, BUFFERING_HAVE_NOTHING);
 250   DCHECK(!pending_read_) << "Pending read must complete before seeking";
 251
 252   ChangeState_Locked(kPlaying);
 253   AttemptRead_Locked();
 254 }
 255
 256 void AudioRendererImpl::Initialize(DemuxerStream* stream,
 257                                    const PipelineStatusCB& init_cb,
 258                                    const StatisticsCB& statistics_cb,
 259                                    const BufferingStateCB& buffering_state_cb,
 260                                    const base::Closure& ended_cb,
 261                                    const PipelineStatusCB& error_cb) {
 262   DVLOG(1) << __FUNCTION__;
 263   DCHECK(task_runner_->BelongsToCurrentThread());
 264   DCHECK(stream);
 265   DCHECK_EQ(stream->type(), DemuxerStream::AUDIO);
 266   DCHECK(!init_cb.is_null());
 267   DCHECK(!statistics_cb.is_null());
 268   DCHECK(!buffering_state_cb.is_null());
 269   DCHECK(!ended_cb.is_null());
 270   DCHECK(!error_cb.is_null());
 271   DCHECK_EQ(kUninitialized, state_);
 272   DCHECK(sink_.get());
 273
 274   state_ = kInitializing;
 275
 276   // Always post |init_cb_| because |this| could be destroyed if initialization
 277   // failed.
 278   init_cb_ = BindToCurrentLoop(init_cb);
 279
 280   buffering_state_cb_ = buffering_state_cb;
 281   ended_cb_ = ended_cb;
 282   error_cb_ = error_cb;
 283
 284   expecting_config_changes_ = stream->SupportsConfigChanges();
 285   if (!expecting_config_changes_) {
 286     // The actual buffer size is controlled via the size of the AudioBus
 287     // provided to Render(), so just choose something reasonable here for looks.
 288     int buffer_size = stream->audio_decoder_config().samples_per_second() / 100;
 289     audio_parameters_.Reset(
 290         AudioParameters::AUDIO_PCM_LOW_LATENCY,
 291         stream->audio_decoder_config().channel_layout(),
 292         ChannelLayoutToChannelCount(
 293             stream->audio_decoder_config().channel_layout()),
 294         stream->audio_decoder_config().samples_per_second(),
 295         stream->audio_decoder_config().bits_per_channel(),
 296         buffer_size);
 297     buffer_converter_.reset();
 298   } else {
 299     // TODO(rileya): Support hardware config changes
 300     const AudioParameters& hw_params = hardware_config_.GetOutputConfig();
 301     audio_parameters_.Reset(
 302         hw_params.format(),
 303         // Always use the source's channel layout and channel count to avoid
 304         // premature downmixing (http://crbug.com/379288), platform specific
 305         // issues around channel layouts (http://crbug.com/266674), and
 306         // unnecessary upmixing overhead.
 307         stream->audio_decoder_config().channel_layout(),
 308         ChannelLayoutToChannelCount(
 309             stream->audio_decoder_config().channel_layout()),
 310         hw_params.sample_rate(),
 311         hw_params.bits_per_sample(),
 312         hardware_config_.GetHighLatencyBufferSize());
 313   }
 314
 315   audio_clock_.reset(
 316       new AudioClock(base::TimeDelta(), audio_parameters_.sample_rate()));
 317
 318   audio_buffer_stream_->Initialize(
 319       stream, statistics_cb,
 320       base::Bind(&AudioRendererImpl::OnAudioBufferStreamInitialized,
 321                  weak_factory_.GetWeakPtr()));
 322 }
 323
 324 void AudioRendererImpl::OnAudioBufferStreamInitialized(bool success) {
 325   DVLOG(1) << __FUNCTION__ << ": " << success;
 326   DCHECK(task_runner_->BelongsToCurrentThread());
 327
 328   base::AutoLock auto_lock(lock_);
 329
 330   if (!success) {
 331     state_ = kUninitialized;
 332     base::ResetAndReturn(&init_cb_).Run(DECODER_ERROR_NOT_SUPPORTED);
 333     return;
 334   }
 335
 336   if (!audio_parameters_.IsValid()) {
 337     DVLOG(1) << __FUNCTION__ << ": Invalid audio parameters: "
 338              << audio_parameters_.AsHumanReadableString();
 339     ChangeState_Locked(kUninitialized);
 340     base::ResetAndReturn(&init_cb_).Run(PIPELINE_ERROR_INITIALIZATION_FAILED);
 341     return;
 342   }
 343
 344   if (expecting_config_changes_)
 345     buffer_converter_.reset(new AudioBufferConverter(audio_parameters_));
 346   splicer_.reset(new AudioSplicer(audio_parameters_.sample_rate()));
 347
 348   // We're all good! Continue initializing the rest of the audio renderer
 349   // based on the decoder format.
 350   algorithm_.reset(new AudioRendererAlgorithm());
 351   algorithm_->Initialize(audio_parameters_);
 352
 353   ChangeState_Locked(kFlushed);
 354
 355   HistogramRendererEvent(INITIALIZED);
 356
 357   {
 358     base::AutoUnlock auto_unlock(lock_);
 359     sink_->Initialize(audio_parameters_, this);
 360     sink_->Start();
 361
 362     // Some sinks play on start...
 363     sink_->Pause();
 364   }
 365
 366   DCHECK(!sink_playing_);
 367   base::ResetAndReturn(&init_cb_).Run(PIPELINE_OK);
 368 }
 369
 370 void AudioRendererImpl::SetVolume(float volume) {
 371   DCHECK(task_runner_->BelongsToCurrentThread());
 372   DCHECK(sink_.get());
 373   sink_->SetVolume(volume);
 374 }
 375
 376 void AudioRendererImpl::DecodedAudioReady(
 377     AudioBufferStream::Status status,
 378     const scoped_refptr<AudioBuffer>& buffer) {
 379   DVLOG(2) << __FUNCTION__ << "(" << status << ")";
 380   DCHECK(task_runner_->BelongsToCurrentThread());
 381
 382   base::AutoLock auto_lock(lock_);
 383   DCHECK(state_ != kUninitialized);
 384
 385   CHECK(pending_read_);
 386   pending_read_ = false;
 387
 388   if (status == AudioBufferStream::ABORTED ||
 389       status == AudioBufferStream::DEMUXER_READ_ABORTED) {
 390     HandleAbortedReadOrDecodeError(false);
 391     return;
 392   }
 393
 394   if (status == AudioBufferStream::DECODE_ERROR) {
 395     HandleAbortedReadOrDecodeError(true);
 396     return;
 397   }
 398
 399   DCHECK_EQ(status, AudioBufferStream::OK);
 400   DCHECK(buffer.get());
 401
 402   if (state_ == kFlushing) {
 403     ChangeState_Locked(kFlushed);
 404     DoFlush_Locked();
 405     return;
 406   }
 407
 408   if (expecting_config_changes_) {
 409     DCHECK(buffer_converter_);
 410     buffer_converter_->AddInput(buffer);
 411     while (buffer_converter_->HasNextBuffer()) {
 412       if (!splicer_->AddInput(buffer_converter_->GetNextBuffer())) {
 413         HandleAbortedReadOrDecodeError(true);
 414         return;
 415       }
 416     }
 417   } else {
 418     if (!splicer_->AddInput(buffer)) {
 419       HandleAbortedReadOrDecodeError(true);
 420       return;
 421     }
 422   }
 423
 424   if (!splicer_->HasNextBuffer()) {
 425     AttemptRead_Locked();
 426     return;
 427   }
 428
 429   bool need_another_buffer = false;
 430   while (splicer_->HasNextBuffer())
 431     need_another_buffer = HandleSplicerBuffer_Locked(splicer_->GetNextBuffer());
 432
 433   if (!need_another_buffer && !CanRead_Locked())
 434     return;
 435
 436   AttemptRead_Locked();
 437 }
 438
 439 bool AudioRendererImpl::HandleSplicerBuffer_Locked(
 440     const scoped_refptr<AudioBuffer>& buffer) {
 441   lock_.AssertAcquired();
 442   if (buffer->end_of_stream()) {
 443     received_end_of_stream_ = true;
 444   } else {
 445     if (state_ == kPlaying) {
 446       if (IsBeforeStartTime(buffer))
 447         return true;
 448
 449       // Trim off any additional time before the start timestamp.
 450       const base::TimeDelta trim_time = start_timestamp_ - buffer->timestamp();
 451       if (trim_time > base::TimeDelta()) {
 452         buffer->TrimStart(buffer->frame_count() *
 453                           (static_cast<double>(trim_time.InMicroseconds()) /
 454                            buffer->duration().InMicroseconds()));
 455       }
 456       // If the entire buffer was trimmed, request a new one.
 457       if (!buffer->frame_count())
 458         return true;
 459     }
 460
 461     if (state_ != kUninitialized)
 462       algorithm_->EnqueueBuffer(buffer);
 463   }
 464
 465   switch (state_) {
 466     case kUninitialized:
 467     case kInitializing:
 468     case kFlushing:
 469       NOTREACHED();
 470       return false;
 471
 472     case kFlushed:
 473       DCHECK(!pending_read_);
 474       return false;
 475
 476     case kPlaying:
 477       if (buffer->end_of_stream() || algorithm_->IsQueueFull()) {
 478         if (buffering_state_ == BUFFERING_HAVE_NOTHING)
 479           SetBufferingState_Locked(BUFFERING_HAVE_ENOUGH);
 480         return false;
 481       }
 482       return true;
 483   }
 484   return false;
 485 }
 486
 487 void AudioRendererImpl::AttemptRead() {
 488   base::AutoLock auto_lock(lock_);
 489   AttemptRead_Locked();
 490 }
 491
 492 void AudioRendererImpl::AttemptRead_Locked() {
 493   DCHECK(task_runner_->BelongsToCurrentThread());
 494   lock_.AssertAcquired();
 495
 496   if (!CanRead_Locked())
 497     return;
 498
 499   pending_read_ = true;
 500   audio_buffer_stream_->Read(base::Bind(&AudioRendererImpl::DecodedAudioReady,
 501                                         weak_factory_.GetWeakPtr()));
 502 }
 503
 504 bool AudioRendererImpl::CanRead_Locked() {
 505   lock_.AssertAcquired();
 506
 507   switch (state_) {
 508     case kUninitialized:
 509     case kInitializing:
 510     case kFlushing:
 511     case kFlushed:
 512       return false;
 513
 514     case kPlaying:
 515       break;
 516   }
 517
 518   return !pending_read_ && !received_end_of_stream_ &&
 519       !algorithm_->IsQueueFull();
 520 }
 521
 522 void AudioRendererImpl::SetPlaybackRate(float playback_rate) {
 523   DVLOG(1) << __FUNCTION__ << "(" << playback_rate << ")";
 524   DCHECK(task_runner_->BelongsToCurrentThread());
 525   DCHECK_GE(playback_rate, 0);
 526   DCHECK(sink_.get());
 527
 528   base::AutoLock auto_lock(lock_);
 529
 530   // We have two cases here:
 531   // Play: current_playback_rate == 0 && playback_rate != 0
 532   // Pause: current_playback_rate != 0 && playback_rate == 0
 533   float current_playback_rate = playback_rate_;
 534   playback_rate_ = playback_rate;
 535
 536   if (!rendering_)
 537     return;
 538
 539   if (current_playback_rate == 0 && playback_rate != 0) {
 540     StartRendering_Locked();
 541     return;
 542   }
 543
 544   if (current_playback_rate != 0 && playback_rate == 0) {
 545     StopRendering_Locked();
 546     return;
 547   }
 548 }
 549
 550 bool AudioRendererImpl::IsBeforeStartTime(
 551     const scoped_refptr<AudioBuffer>& buffer) {
 552   DCHECK_EQ(state_, kPlaying);
 553   return buffer.get() && !buffer->end_of_stream() &&
 554          (buffer->timestamp() + buffer->duration()) < start_timestamp_;
 555 }
 556
 557 int AudioRendererImpl::Render(AudioBus* audio_bus,
 558                               int audio_delay_milliseconds) {
 559   const int requested_frames = audio_bus->frames();
 560   base::TimeDelta playback_delay = base::TimeDelta::FromMilliseconds(
 561       audio_delay_milliseconds);
 562   const int delay_frames = static_cast<int>(playback_delay.InSecondsF() *
 563                                             audio_parameters_.sample_rate());
 564   int frames_written = 0;
 565   {
 566     base::AutoLock auto_lock(lock_);
 567     last_render_ticks_ = base::TimeTicks::Now();
 568
 569     // Ensure Stop() hasn't destroyed our |algorithm_| on the pipeline thread.
 570     if (!algorithm_) {
 571       audio_clock_->WroteAudio(
 572           0, requested_frames, delay_frames, playback_rate_);
 573       return 0;
 574     }
 575
 576     if (playback_rate_ == 0) {
 577       audio_clock_->WroteAudio(
 578           0, requested_frames, delay_frames, playback_rate_);
 579       return 0;
 580     }
 581
 582     // Mute audio by returning 0 when not playing.
 583     if (state_ != kPlaying) {
 584       audio_clock_->WroteAudio(
 585           0, requested_frames, delay_frames, playback_rate_);
 586       return 0;
 587     }
 588
 589     // We use the following conditions to determine end of playback:
 590     //   1) Algorithm can not fill the audio callback buffer
 591     //   2) We received an end of stream buffer
 592     //   3) We haven't already signalled that we've ended
 593     //   4) We've played all known audio data sent to hardware
 594     //
 595     // We use the following conditions to determine underflow:
 596     //   1) Algorithm can not fill the audio callback buffer
 597     //   2) We have NOT received an end of stream buffer
 598     //   3) We are in the kPlaying state
 599     //
 600     // Otherwise the buffer has data we can send to the device.
 601     if (algorithm_->frames_buffered() > 0) {
 602       frames_written =
 603           algorithm_->FillBuffer(audio_bus, requested_frames, playback_rate_);
 604     }
 605
 606     // Per the TimeSource API the media time should always increase even after
 607     // we've rendered all known audio data. Doing so simplifies scenarios where
 608     // we have other sources of media data that need to be scheduled after audio
 609     // data has ended.
 610     //
 611     // That being said, we don't want to advance time when underflowed as we
 612     // know more decoded frames will eventually arrive. If we did, we would
 613     // throw things out of sync when said decoded frames arrive.
 614     int frames_after_end_of_stream = 0;
 615     if (frames_written == 0) {
 616       if (received_end_of_stream_) {
 617         if (ended_timestamp_ == kInfiniteDuration())
 618           ended_timestamp_ = audio_clock_->back_timestamp();
 619         frames_after_end_of_stream = requested_frames;
 620       } else if (state_ == kPlaying &&
 621                  buffering_state_ != BUFFERING_HAVE_NOTHING) {
 622         algorithm_->IncreaseQueueCapacity();
 623         SetBufferingState_Locked(BUFFERING_HAVE_NOTHING);
 624       }
 625     }
 626
 627     audio_clock_->WroteAudio(frames_written + frames_after_end_of_stream,
 628                              requested_frames,
 629                              delay_frames,
 630                              playback_rate_);
 631
 632     if (CanRead_Locked()) {
 633       task_runner_->PostTask(FROM_HERE,
 634                              base::Bind(&AudioRendererImpl::AttemptRead,
 635                                         weak_factory_.GetWeakPtr()));
 636     }
 637
 638     if (audio_clock_->front_timestamp() >= ended_timestamp_ &&
 639         !rendered_end_of_stream_) {
 640       rendered_end_of_stream_ = true;
 641       task_runner_->PostTask(FROM_HERE, ended_cb_);
 642     }
 643   }
 644
 645   DCHECK_LE(frames_written, requested_frames);
 646   return frames_written;
 647 }
 648
 649 void AudioRendererImpl::OnRenderError() {
 650   // UMA data tells us this happens ~0.01% of the time. Trigger an error instead
 651   // of trying to gracefully fall back to a fake sink. It's very likely
 652   // OnRenderError() should be removed and the audio stack handle errors without
 653   // notifying clients. See http://crbug.com/234708 for details.
 654   HistogramRendererEvent(RENDER_ERROR);
 655   // Post to |task_runner_| as this is called on the audio callback thread.
 656   task_runner_->PostTask(FROM_HERE,
 657                          base::Bind(error_cb_, PIPELINE_ERROR_DECODE));
 658 }
 659
 660 void AudioRendererImpl::HandleAbortedReadOrDecodeError(bool is_decode_error) {
 661   DCHECK(task_runner_->BelongsToCurrentThread());
 662   lock_.AssertAcquired();
 663
 664   PipelineStatus status = is_decode_error ? PIPELINE_ERROR_DECODE : PIPELINE_OK;
 665   switch (state_) {
 666     case kUninitialized:
 667     case kInitializing:
 668       NOTREACHED();
 669       return;
 670     case kFlushing:
 671       ChangeState_Locked(kFlushed);
 672       if (status == PIPELINE_OK) {
 673         DoFlush_Locked();
 674         return;
 675       }
 676
 677       error_cb_.Run(status);
 678       base::ResetAndReturn(&flush_cb_).Run();
 679       return;
 680
 681     case kFlushed:
 682     case kPlaying:
 683       if (status != PIPELINE_OK)
 684         error_cb_.Run(status);
 685       return;
 686   }
 687 }
 688
 689 void AudioRendererImpl::ChangeState_Locked(State new_state) {
 690   DVLOG(1) << __FUNCTION__ << " : " << state_ << " -> " << new_state;
 691   lock_.AssertAcquired();
 692   state_ = new_state;
 693 }
 694
 695 void AudioRendererImpl::OnNewSpliceBuffer(base::TimeDelta splice_timestamp) {
 696   DCHECK(task_runner_->BelongsToCurrentThread());
 697   splicer_->SetSpliceTimestamp(splice_timestamp);
 698 }
 699
 700 void AudioRendererImpl::OnConfigChange() {
 701   DCHECK(task_runner_->BelongsToCurrentThread());
 702   DCHECK(expecting_config_changes_);
 703   buffer_converter_->ResetTimestampState();
 704   // Drain flushed buffers from the converter so the AudioSplicer receives all
 705   // data ahead of any OnNewSpliceBuffer() calls.  Since discontinuities should
 706   // only appear after config changes, AddInput() should never fail here.
 707   while (buffer_converter_->HasNextBuffer())
 708     CHECK(splicer_->AddInput(buffer_converter_->GetNextBuffer()));
 709 }
 710
 711 void AudioRendererImpl::SetBufferingState_Locked(
 712     BufferingState buffering_state) {
 713   DVLOG(1) << __FUNCTION__ << " : " << buffering_state_ << " -> "
 714            << buffering_state;
 715   DCHECK_NE(buffering_state_, buffering_state);
 716   lock_.AssertAcquired();
 717   buffering_state_ = buffering_state;
 718
 719   task_runner_->PostTask(FROM_HERE,
 720                          base::Bind(buffering_state_cb_, buffering_state_));
 721 }
 722
 723 }  // namespace media