media/renderers/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/renderers/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 AudioHardwareConfig& hardware_config,
  48     const scoped_refptr<MediaLog>& media_log)
  49     : task_runner_(task_runner),
  50       expecting_config_changes_(false),
  51       sink_(sink),
  52       audio_buffer_stream_(
  53           new AudioBufferStream(task_runner, decoders.Pass(), media_log)),
  54       hardware_config_(hardware_config),
  55       playback_rate_(0),
  56       state_(kUninitialized),
  57       buffering_state_(BUFFERING_HAVE_NOTHING),
  58       rendering_(false),
  59       sink_playing_(false),
  60       pending_read_(false),
  61       received_end_of_stream_(false),
  62       rendered_end_of_stream_(false),
  63       weak_factory_(this) {
  64   audio_buffer_stream_->set_splice_observer(base::Bind(
  65       &AudioRendererImpl::OnNewSpliceBuffer, weak_factory_.GetWeakPtr()));
  66   audio_buffer_stream_->set_config_change_observer(base::Bind(
  67       &AudioRendererImpl::OnConfigChange, weak_factory_.GetWeakPtr()));
  68 }
  69
  70 AudioRendererImpl::~AudioRendererImpl() {
  71   DVLOG(1) << __FUNCTION__;
  72   DCHECK(task_runner_->BelongsToCurrentThread());
  73
  74   // If Render() is in progress, this call will wait for Render() to finish.
  75   // After this call, the |sink_| will not call back into |this| anymore.
  76   sink_->Stop();
  77
  78   if (!init_cb_.is_null())
  79     base::ResetAndReturn(&init_cb_).Run(PIPELINE_ERROR_ABORT);
  80 }
  81
  82 void AudioRendererImpl::StartTicking() {
  83   DVLOG(1) << __FUNCTION__;
  84   DCHECK(task_runner_->BelongsToCurrentThread());
  85   DCHECK(!rendering_);
  86   rendering_ = true;
  87
  88   base::AutoLock auto_lock(lock_);
  89   // Wait for an eventual call to SetPlaybackRate() to start rendering.
  90   if (playback_rate_ == 0) {
  91     DCHECK(!sink_playing_);
  92     return;
  93   }
  94
  95   StartRendering_Locked();
  96 }
  97
  98 void AudioRendererImpl::StartRendering_Locked() {
  99   DVLOG(1) << __FUNCTION__;
 100   DCHECK(task_runner_->BelongsToCurrentThread());
 101   DCHECK_EQ(state_, kPlaying);
 102   DCHECK(!sink_playing_);
 103   DCHECK_NE(playback_rate_, 0);
 104   lock_.AssertAcquired();
 105
 106   sink_playing_ = true;
 107
 108   base::AutoUnlock auto_unlock(lock_);
 109   sink_->Play();
 110 }
 111
 112 void AudioRendererImpl::StopTicking() {
 113   DVLOG(1) << __FUNCTION__;
 114   DCHECK(task_runner_->BelongsToCurrentThread());
 115   DCHECK(rendering_);
 116   rendering_ = false;
 117
 118   base::AutoLock auto_lock(lock_);
 119   // Rendering should have already been stopped with a zero playback rate.
 120   if (playback_rate_ == 0) {
 121     DCHECK(!sink_playing_);
 122     return;
 123   }
 124
 125   StopRendering_Locked();
 126 }
 127
 128 void AudioRendererImpl::StopRendering_Locked() {
 129   DCHECK(task_runner_->BelongsToCurrentThread());
 130   DCHECK_EQ(state_, kPlaying);
 131   DCHECK(sink_playing_);
 132   lock_.AssertAcquired();
 133
 134   sink_playing_ = false;
 135
 136   base::AutoUnlock auto_unlock(lock_);
 137   sink_->Pause();
 138 }
 139
 140 void AudioRendererImpl::SetMediaTime(base::TimeDelta time) {
 141   DVLOG(1) << __FUNCTION__ << "(" << time << ")";
 142   DCHECK(task_runner_->BelongsToCurrentThread());
 143
 144   base::AutoLock auto_lock(lock_);
 145   DCHECK(!rendering_);
 146   DCHECK_EQ(state_, kFlushed);
 147
 148   start_timestamp_ = time;
 149   ended_timestamp_ = kInfiniteDuration();
 150   last_render_ticks_ = base::TimeTicks();
 151   first_packet_timestamp_ = kNoTimestamp();
 152   audio_clock_.reset(new AudioClock(time, audio_parameters_.sample_rate()));
 153 }
 154
 155 base::TimeDelta AudioRendererImpl::CurrentMediaTime() {
 156   // In practice the Render() method is called with a high enough frequency
 157   // that returning only the front timestamp is good enough and also prevents
 158   // returning values that go backwards in time.
 159   base::TimeDelta current_media_time;
 160   {
 161     base::AutoLock auto_lock(lock_);
 162     current_media_time = audio_clock_->front_timestamp();
 163   }
 164
 165   DVLOG(2) << __FUNCTION__ << ": " << current_media_time;
 166   return current_media_time;
 167 }
 168
 169 base::TimeDelta AudioRendererImpl::CurrentMediaTimeForSyncingVideo() {
 170   DVLOG(3) << __FUNCTION__;
 171
 172   base::AutoLock auto_lock(lock_);
 173   if (last_render_ticks_.is_null())
 174     return audio_clock_->front_timestamp();
 175
 176   return audio_clock_->TimestampSinceWriting(base::TimeTicks::Now() -
 177                                              last_render_ticks_);
 178 }
 179
 180 TimeSource* AudioRendererImpl::GetTimeSource() {
 181   return this;
 182 }
 183
 184 void AudioRendererImpl::Flush(const base::Closure& callback) {
 185   DVLOG(1) << __FUNCTION__;
 186   DCHECK(task_runner_->BelongsToCurrentThread());
 187
 188   base::AutoLock auto_lock(lock_);
 189   DCHECK_EQ(state_, kPlaying);
 190   DCHECK(flush_cb_.is_null());
 191
 192   flush_cb_ = callback;
 193   ChangeState_Locked(kFlushing);
 194
 195   if (pending_read_)
 196     return;
 197
 198   ChangeState_Locked(kFlushed);
 199   DoFlush_Locked();
 200 }
 201
 202 void AudioRendererImpl::DoFlush_Locked() {
 203   DCHECK(task_runner_->BelongsToCurrentThread());
 204   lock_.AssertAcquired();
 205
 206   DCHECK(!pending_read_);
 207   DCHECK_EQ(state_, kFlushed);
 208
 209   audio_buffer_stream_->Reset(base::Bind(&AudioRendererImpl::ResetDecoderDone,
 210                                          weak_factory_.GetWeakPtr()));
 211 }
 212
 213 void AudioRendererImpl::ResetDecoderDone() {
 214   DCHECK(task_runner_->BelongsToCurrentThread());
 215   {
 216     base::AutoLock auto_lock(lock_);
 217
 218     DCHECK_EQ(state_, kFlushed);
 219     DCHECK(!flush_cb_.is_null());
 220
 221     received_end_of_stream_ = false;
 222     rendered_end_of_stream_ = false;
 223
 224     // Flush() may have been called while underflowed/not fully buffered.
 225     if (buffering_state_ != BUFFERING_HAVE_NOTHING)
 226       SetBufferingState_Locked(BUFFERING_HAVE_NOTHING);
 227
 228     splicer_->Reset();
 229     if (buffer_converter_)
 230       buffer_converter_->Reset();
 231     algorithm_->FlushBuffers();
 232   }
 233
 234   // Changes in buffering state are always posted. Flush callback must only be
 235   // run after buffering state has been set back to nothing.
 236   task_runner_->PostTask(FROM_HERE, base::ResetAndReturn(&flush_cb_));
 237 }
 238
 239 void AudioRendererImpl::StartPlaying() {
 240   DVLOG(1) << __FUNCTION__;
 241   DCHECK(task_runner_->BelongsToCurrentThread());
 242
 243   base::AutoLock auto_lock(lock_);
 244   DCHECK(!sink_playing_);
 245   DCHECK_EQ(state_, kFlushed);
 246   DCHECK_EQ(buffering_state_, BUFFERING_HAVE_NOTHING);
 247   DCHECK(!pending_read_) << "Pending read must complete before seeking";
 248
 249   ChangeState_Locked(kPlaying);
 250   AttemptRead_Locked();
 251 }
 252
 253 void AudioRendererImpl::Initialize(
 254     DemuxerStream* stream,
 255     const PipelineStatusCB& init_cb,
 256     const SetDecryptorReadyCB& set_decryptor_ready_cb,
 257     const StatisticsCB& statistics_cb,
 258     const BufferingStateCB& buffering_state_cb,
 259     const base::Closure& ended_cb,
 260     const PipelineStatusCB& error_cb,
 261     const base::Closure& waiting_for_decryption_key_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, base::Bind(&AudioRendererImpl::OnAudioBufferStreamInitialized,
 320                          weak_factory_.GetWeakPtr()),
 321       set_decryptor_ready_cb, statistics_cb, waiting_for_decryption_key_cb);
 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   // Store the timestamp of the first packet so we know when to start actual
 466   // audio playback.
 467   if (first_packet_timestamp_ == kNoTimestamp())
 468     first_packet_timestamp_ = buffer->timestamp();
 469
 470   switch (state_) {
 471     case kUninitialized:
 472     case kInitializing:
 473     case kFlushing:
 474       NOTREACHED();
 475       return false;
 476
 477     case kFlushed:
 478       DCHECK(!pending_read_);
 479       return false;
 480
 481     case kPlaying:
 482       if (buffer->end_of_stream() || algorithm_->IsQueueFull()) {
 483         if (buffering_state_ == BUFFERING_HAVE_NOTHING)
 484           SetBufferingState_Locked(BUFFERING_HAVE_ENOUGH);
 485         return false;
 486       }
 487       return true;
 488   }
 489   return false;
 490 }
 491
 492 void AudioRendererImpl::AttemptRead() {
 493   base::AutoLock auto_lock(lock_);
 494   AttemptRead_Locked();
 495 }
 496
 497 void AudioRendererImpl::AttemptRead_Locked() {
 498   DCHECK(task_runner_->BelongsToCurrentThread());
 499   lock_.AssertAcquired();
 500
 501   if (!CanRead_Locked())
 502     return;
 503
 504   pending_read_ = true;
 505   audio_buffer_stream_->Read(base::Bind(&AudioRendererImpl::DecodedAudioReady,
 506                                         weak_factory_.GetWeakPtr()));
 507 }
 508
 509 bool AudioRendererImpl::CanRead_Locked() {
 510   lock_.AssertAcquired();
 511
 512   switch (state_) {
 513     case kUninitialized:
 514     case kInitializing:
 515     case kFlushing:
 516     case kFlushed:
 517       return false;
 518
 519     case kPlaying:
 520       break;
 521   }
 522
 523   return !pending_read_ && !received_end_of_stream_ &&
 524       !algorithm_->IsQueueFull();
 525 }
 526
 527 void AudioRendererImpl::SetPlaybackRate(float playback_rate) {
 528   DVLOG(1) << __FUNCTION__ << "(" << playback_rate << ")";
 529   DCHECK(task_runner_->BelongsToCurrentThread());
 530   DCHECK_GE(playback_rate, 0);
 531   DCHECK(sink_.get());
 532
 533   base::AutoLock auto_lock(lock_);
 534
 535   // We have two cases here:
 536   // Play: current_playback_rate == 0 && playback_rate != 0
 537   // Pause: current_playback_rate != 0 && playback_rate == 0
 538   float current_playback_rate = playback_rate_;
 539   playback_rate_ = playback_rate;
 540
 541   if (!rendering_)
 542     return;
 543
 544   if (current_playback_rate == 0 && playback_rate != 0) {
 545     StartRendering_Locked();
 546     return;
 547   }
 548
 549   if (current_playback_rate != 0 && playback_rate == 0) {
 550     StopRendering_Locked();
 551     return;
 552   }
 553 }
 554
 555 bool AudioRendererImpl::IsBeforeStartTime(
 556     const scoped_refptr<AudioBuffer>& buffer) {
 557   DCHECK_EQ(state_, kPlaying);
 558   return buffer.get() && !buffer->end_of_stream() &&
 559          (buffer->timestamp() + buffer->duration()) < start_timestamp_;
 560 }
 561
 562 int AudioRendererImpl::Render(AudioBus* audio_bus,
 563                               int audio_delay_milliseconds) {
 564   const int requested_frames = audio_bus->frames();
 565   base::TimeDelta playback_delay = base::TimeDelta::FromMilliseconds(
 566       audio_delay_milliseconds);
 567   const int delay_frames = static_cast<int>(playback_delay.InSecondsF() *
 568                                             audio_parameters_.sample_rate());
 569   int frames_written = 0;
 570   {
 571     base::AutoLock auto_lock(lock_);
 572     last_render_ticks_ = base::TimeTicks::Now();
 573
 574     // Ensure Stop() hasn't destroyed our |algorithm_| on the pipeline thread.
 575     if (!algorithm_) {
 576       audio_clock_->WroteAudio(
 577           0, requested_frames, delay_frames, playback_rate_);
 578       return 0;
 579     }
 580
 581     if (playback_rate_ == 0) {
 582       audio_clock_->WroteAudio(
 583           0, requested_frames, delay_frames, playback_rate_);
 584       return 0;
 585     }
 586
 587     // Mute audio by returning 0 when not playing.
 588     if (state_ != kPlaying) {
 589       audio_clock_->WroteAudio(
 590           0, requested_frames, delay_frames, playback_rate_);
 591       return 0;
 592     }
 593
 594     // Delay playback by writing silence if we haven't reached the first
 595     // timestamp yet; this can occur if the video starts before the audio.
 596     if (algorithm_->frames_buffered() > 0) {
 597       DCHECK(first_packet_timestamp_ != kNoTimestamp());
 598       const base::TimeDelta play_delay =
 599           first_packet_timestamp_ - audio_clock_->back_timestamp();
 600       if (play_delay > base::TimeDelta()) {
 601         DCHECK_EQ(frames_written, 0);
 602         frames_written =
 603             std::min(static_cast<int>(play_delay.InSecondsF() *
 604                                       audio_parameters_.sample_rate()),
 605                      requested_frames);
 606         audio_bus->ZeroFramesPartial(0, frames_written);
 607       }
 608
 609       // If there's any space left, actually render the audio; this is where the
 610       // aural magic happens.
 611       if (frames_written < requested_frames) {
 612         frames_written += algorithm_->FillBuffer(
 613             audio_bus, frames_written, requested_frames - frames_written,
 614             playback_rate_);
 615       }
 616     }
 617
 618     // We use the following conditions to determine end of playback:
 619     //   1) Algorithm can not fill the audio callback buffer
 620     //   2) We received an end of stream buffer
 621     //   3) We haven't already signalled that we've ended
 622     //   4) We've played all known audio data sent to hardware
 623     //
 624     // We use the following conditions to determine underflow:
 625     //   1) Algorithm can not fill the audio callback buffer
 626     //   2) We have NOT received an end of stream buffer
 627     //   3) We are in the kPlaying state
 628     //
 629     // Otherwise the buffer has data we can send to the device.
 630     //
 631     // Per the TimeSource API the media time should always increase even after
 632     // we've rendered all known audio data. Doing so simplifies scenarios where
 633     // we have other sources of media data that need to be scheduled after audio
 634     // data has ended.
 635     //
 636     // That being said, we don't want to advance time when underflowed as we
 637     // know more decoded frames will eventually arrive. If we did, we would
 638     // throw things out of sync when said decoded frames arrive.
 639     int frames_after_end_of_stream = 0;
 640     if (frames_written == 0) {
 641       if (received_end_of_stream_) {
 642         if (ended_timestamp_ == kInfiniteDuration())
 643           ended_timestamp_ = audio_clock_->back_timestamp();
 644         frames_after_end_of_stream = requested_frames;
 645       } else if (state_ == kPlaying &&
 646                  buffering_state_ != BUFFERING_HAVE_NOTHING) {
 647         algorithm_->IncreaseQueueCapacity();
 648         SetBufferingState_Locked(BUFFERING_HAVE_NOTHING);
 649       }
 650     }
 651
 652     audio_clock_->WroteAudio(frames_written + frames_after_end_of_stream,
 653                              requested_frames,
 654                              delay_frames,
 655                              playback_rate_);
 656
 657     if (CanRead_Locked()) {
 658       task_runner_->PostTask(FROM_HERE,
 659                              base::Bind(&AudioRendererImpl::AttemptRead,
 660                                         weak_factory_.GetWeakPtr()));
 661     }
 662
 663     if (audio_clock_->front_timestamp() >= ended_timestamp_ &&
 664         !rendered_end_of_stream_) {
 665       rendered_end_of_stream_ = true;
 666       task_runner_->PostTask(FROM_HERE, ended_cb_);
 667     }
 668   }
 669
 670   DCHECK_LE(frames_written, requested_frames);
 671   return frames_written;
 672 }
 673
 674 void AudioRendererImpl::OnRenderError() {
 675   // UMA data tells us this happens ~0.01% of the time. Trigger an error instead
 676   // of trying to gracefully fall back to a fake sink. It's very likely
 677   // OnRenderError() should be removed and the audio stack handle errors without
 678   // notifying clients. See http://crbug.com/234708 for details.
 679   HistogramRendererEvent(RENDER_ERROR);
 680   // Post to |task_runner_| as this is called on the audio callback thread.
 681   task_runner_->PostTask(FROM_HERE,
 682                          base::Bind(error_cb_, PIPELINE_ERROR_DECODE));
 683 }
 684
 685 void AudioRendererImpl::HandleAbortedReadOrDecodeError(bool is_decode_error) {
 686   DCHECK(task_runner_->BelongsToCurrentThread());
 687   lock_.AssertAcquired();
 688
 689   PipelineStatus status = is_decode_error ? PIPELINE_ERROR_DECODE : PIPELINE_OK;
 690   switch (state_) {
 691     case kUninitialized:
 692     case kInitializing:
 693       NOTREACHED();
 694       return;
 695     case kFlushing:
 696       ChangeState_Locked(kFlushed);
 697       if (status == PIPELINE_OK) {
 698         DoFlush_Locked();
 699         return;
 700       }
 701
 702       error_cb_.Run(status);
 703       base::ResetAndReturn(&flush_cb_).Run();
 704       return;
 705
 706     case kFlushed:
 707     case kPlaying:
 708       if (status != PIPELINE_OK)
 709         error_cb_.Run(status);
 710       return;
 711   }
 712 }
 713
 714 void AudioRendererImpl::ChangeState_Locked(State new_state) {
 715   DVLOG(1) << __FUNCTION__ << " : " << state_ << " -> " << new_state;
 716   lock_.AssertAcquired();
 717   state_ = new_state;
 718 }
 719
 720 void AudioRendererImpl::OnNewSpliceBuffer(base::TimeDelta splice_timestamp) {
 721   DCHECK(task_runner_->BelongsToCurrentThread());
 722   splicer_->SetSpliceTimestamp(splice_timestamp);
 723 }
 724
 725 void AudioRendererImpl::OnConfigChange() {
 726   DCHECK(task_runner_->BelongsToCurrentThread());
 727   DCHECK(expecting_config_changes_);
 728   buffer_converter_->ResetTimestampState();
 729   // Drain flushed buffers from the converter so the AudioSplicer receives all
 730   // data ahead of any OnNewSpliceBuffer() calls.  Since discontinuities should
 731   // only appear after config changes, AddInput() should never fail here.
 732   while (buffer_converter_->HasNextBuffer())
 733     CHECK(splicer_->AddInput(buffer_converter_->GetNextBuffer()));
 734 }
 735
 736 void AudioRendererImpl::SetBufferingState_Locked(
 737     BufferingState buffering_state) {
 738   DVLOG(1) << __FUNCTION__ << " : " << buffering_state_ << " -> "
 739            << buffering_state;
 740   DCHECK_NE(buffering_state_, buffering_state);
 741   lock_.AssertAcquired();
 742   buffering_state_ = buffering_state;
 743
 744   task_runner_->PostTask(FROM_HERE,
 745                          base::Bind(buffering_state_cb_, buffering_state_));
 746 }
 747
 748 }  // namespace media