content/renderer/media/webrtc_audio_renderer.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 "content/renderer/media/webrtc_audio_renderer.h"
   6
   7 #include "base/logging.h"
   8 #include "base/metrics/histogram.h"
   9 #include "base/strings/string_util.h"
  10 #include "base/strings/stringprintf.h"
  11 #include "content/renderer/media/audio_device_factory.h"
  12 #include "content/renderer/media/media_stream_dispatcher.h"
  13 #include "content/renderer/media/webrtc_audio_device_impl.h"
  14 #include "content/renderer/media/webrtc_logging.h"
  15 #include "content/renderer/render_frame_impl.h"
  16 #include "media/audio/audio_output_device.h"
  17 #include "media/audio/audio_parameters.h"
  18 #include "media/audio/sample_rates.h"
  19 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
  20 #include "third_party/libjingle/source/talk/media/base/audiorenderer.h"
  21
  22
  23 #if defined(OS_WIN)
  24 #include "base/win/windows_version.h"
  25 #include "media/audio/win/core_audio_util_win.h"
  26 #endif
  27
  28 namespace content {
  29
  30 namespace {
  31
  32 // We add a UMA histogram measuring the execution time of the Render() method
  33 // every |kNumCallbacksBetweenRenderTimeHistograms| callback. Assuming 10ms
  34 // between each callback leads to one UMA update each 100ms.
  35 const int kNumCallbacksBetweenRenderTimeHistograms = 10;
  36
  37 // This is a simple wrapper class that's handed out to users of a shared
  38 // WebRtcAudioRenderer instance.  This class maintains the per-user 'playing'
  39 // and 'started' states to avoid problems related to incorrect usage which
  40 // might violate the implementation assumptions inside WebRtcAudioRenderer
  41 // (see the play reference count).
  42 class SharedAudioRenderer : public MediaStreamAudioRenderer {
  43  public:
  44   // Callback definition for a callback that is called when when Play(), Pause()
  45   // or SetVolume are called (whenever the internal |playing_state_| changes).
  46   typedef base::Callback<
  47       void(const scoped_refptr<webrtc::MediaStreamInterface>&,
  48            WebRtcAudioRenderer::PlayingState*)> OnPlayStateChanged;
  49
  50   SharedAudioRenderer(
  51       const scoped_refptr<MediaStreamAudioRenderer>& delegate,
  52       const scoped_refptr<webrtc::MediaStreamInterface>& media_stream,
  53       const OnPlayStateChanged& on_play_state_changed)
  54       : delegate_(delegate), media_stream_(media_stream), started_(false),
  55         on_play_state_changed_(on_play_state_changed) {
  56     DCHECK(!on_play_state_changed_.is_null());
  57     DCHECK(media_stream_.get());
  58   }
  59
  60  protected:
  61   ~SharedAudioRenderer() override {
  62     DCHECK(thread_checker_.CalledOnValidThread());
  63     DVLOG(1) << __FUNCTION__;
  64     Stop();
  65   }
  66
  67   void Start() override {
  68     DCHECK(thread_checker_.CalledOnValidThread());
  69     if (started_)
  70       return;
  71     started_ = true;
  72     delegate_->Start();
  73   }
  74
  75   void Play() override {
  76     DCHECK(thread_checker_.CalledOnValidThread());
  77     DCHECK(started_);
  78     if (playing_state_.playing())
  79       return;
  80     playing_state_.set_playing(true);
  81     on_play_state_changed_.Run(media_stream_, &playing_state_);
  82   }
  83
  84   void Pause() override {
  85     DCHECK(thread_checker_.CalledOnValidThread());
  86     DCHECK(started_);
  87     if (!playing_state_.playing())
  88       return;
  89     playing_state_.set_playing(false);
  90     on_play_state_changed_.Run(media_stream_, &playing_state_);
  91   }
  92
  93   void Stop() override {
  94     DCHECK(thread_checker_.CalledOnValidThread());
  95     if (!started_)
  96       return;
  97     Pause();
  98     started_ = false;
  99     delegate_->Stop();
 100   }
 101
 102   void SetVolume(float volume) override {
 103     DCHECK(thread_checker_.CalledOnValidThread());
 104     DCHECK(volume >= 0.0f && volume <= 1.0f);
 105     playing_state_.set_volume(volume);
 106     on_play_state_changed_.Run(media_stream_, &playing_state_);
 107   }
 108
 109   void SwitchOutputDevice(
 110       const std::string& device_id,
 111       const GURL& security_origin,
 112       const media::SwitchOutputDeviceCB& callback) override {
 113     DCHECK(thread_checker_.CalledOnValidThread());
 114     DVLOG(1) << __FUNCTION__
 115              << "(" << device_id << ", " << security_origin << ")";
 116     delegate_->SwitchOutputDevice(device_id, security_origin, callback);
 117   }
 118
 119   base::TimeDelta GetCurrentRenderTime() const override {
 120     DCHECK(thread_checker_.CalledOnValidThread());
 121     return delegate_->GetCurrentRenderTime();
 122   }
 123
 124   bool IsLocalRenderer() const override {
 125     DCHECK(thread_checker_.CalledOnValidThread());
 126     return delegate_->IsLocalRenderer();
 127   }
 128
 129  private:
 130   base::ThreadChecker thread_checker_;
 131   const scoped_refptr<MediaStreamAudioRenderer> delegate_;
 132   const scoped_refptr<webrtc::MediaStreamInterface> media_stream_;
 133   bool started_;
 134   WebRtcAudioRenderer::PlayingState playing_state_;
 135   OnPlayStateChanged on_play_state_changed_;
 136 };
 137
 138 // Returns either AudioParameters::NO_EFFECTS or AudioParameters::DUCKING
 139 // depending on whether or not an input element is currently open with
 140 // ducking enabled.
 141 int GetCurrentDuckingFlag(int render_frame_id) {
 142   RenderFrameImpl* const frame =
 143       RenderFrameImpl::FromRoutingID(render_frame_id);
 144   MediaStreamDispatcher* const dispatcher = frame ?
 145       frame->GetMediaStreamDispatcher() : NULL;
 146   if (dispatcher && dispatcher->IsAudioDuckingActive()) {
 147     return media::AudioParameters::DUCKING;
 148   }
 149
 150   return media::AudioParameters::NO_EFFECTS;
 151 }
 152
 153 }  // namespace
 154
 155 int WebRtcAudioRenderer::GetOptimalBufferSize(int sample_rate,
 156                                               int hardware_buffer_size) {
 157   // Use native hardware buffer size as default. On Windows, we strive to open
 158   // up using this native hardware buffer size to achieve best
 159   // possible performance and to ensure that no FIFO is needed on the browser
 160   // side to match the client request. That is why there is no #if case for
 161   // Windows below.
 162   int frames_per_buffer = hardware_buffer_size;
 163
 164 #if defined(OS_LINUX) || defined(OS_MACOSX)
 165   // On Linux and MacOS, the low level IO implementations on the browser side
 166   // supports all buffer size the clients want. We use the native peer
 167   // connection buffer size (10ms) to achieve best possible performance.
 168   frames_per_buffer = sample_rate / 100;
 169 #elif defined(OS_ANDROID)
 170   // TODO(henrika): Keep tuning this scheme and espcicially for low-latency
 171   // cases. Might not be possible to come up with the perfect solution using
 172   // the render side only.
 173   int frames_per_10ms = sample_rate / 100;
 174   if (frames_per_buffer < 2 * frames_per_10ms) {
 175     // Examples of low-latency frame sizes and the resulting |buffer_size|:
 176     //  Nexus 7     : 240 audio frames => 2*480 = 960
 177     //  Nexus 10    : 256              => 2*441 = 882
 178     //  Galaxy Nexus: 144              => 2*441 = 882
 179     frames_per_buffer = 2 * frames_per_10ms;
 180     DVLOG(1) << "Low-latency output detected on Android";
 181   }
 182 #endif
 183
 184   DVLOG(1) << "Using sink output buffer size: " << frames_per_buffer;
 185   return frames_per_buffer;
 186 }
 187
 188 WebRtcAudioRenderer::WebRtcAudioRenderer(
 189     const scoped_refptr<base::SingleThreadTaskRunner>& signaling_thread,
 190     const scoped_refptr<webrtc::MediaStreamInterface>& media_stream,
 191     int source_render_frame_id,
 192     int session_id,
 193     int sample_rate,
 194     int frames_per_buffer)
 195     : state_(UNINITIALIZED),
 196       source_render_frame_id_(source_render_frame_id),
 197       session_id_(session_id),
 198       signaling_thread_(signaling_thread),
 199       media_stream_(media_stream),
 200       source_(NULL),
 201       play_ref_count_(0),
 202       start_ref_count_(0),
 203       audio_delay_milliseconds_(0),
 204       fifo_delay_milliseconds_(0),
 205       sink_params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
 206                    media::CHANNEL_LAYOUT_STEREO, sample_rate, 16,
 207                    frames_per_buffer,
 208                    GetCurrentDuckingFlag(source_render_frame_id)),
 209       render_callback_count_(0) {
 210   WebRtcLogMessage(base::StringPrintf(
 211       "WAR::WAR. source_render_frame_id=%d"
 212       ", session_id=%d, sample_rate=%d, frames_per_buffer=%d, effects=%i",
 213       source_render_frame_id, session_id, sample_rate, frames_per_buffer,
 214       sink_params_.effects()));
 215 }
 216
 217 WebRtcAudioRenderer::~WebRtcAudioRenderer() {
 218   DCHECK(thread_checker_.CalledOnValidThread());
 219   DCHECK_EQ(state_, UNINITIALIZED);
 220 }
 221
 222 bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) {
 223   DVLOG(1) << "WebRtcAudioRenderer::Initialize()";
 224   DCHECK(thread_checker_.CalledOnValidThread());
 225   base::AutoLock auto_lock(lock_);
 226   DCHECK_EQ(state_, UNINITIALIZED);
 227   DCHECK(source);
 228   DCHECK(!sink_.get());
 229   DCHECK(!source_);
 230
 231   // WebRTC does not yet support higher rates than 96000 on the client side
 232   // and 48000 is the preferred sample rate. Therefore, if 192000 is detected,
 233   // we change the rate to 48000 instead. The consequence is that the native
 234   // layer will be opened up at 192kHz but WebRTC will provide data at 48kHz
 235   // which will then be resampled by the audio converted on the browser side
 236   // to match the native audio layer.
 237   int sample_rate = sink_params_.sample_rate();
 238   DVLOG(1) << "Audio output hardware sample rate: " << sample_rate;
 239   if (sample_rate == 192000) {
 240     DVLOG(1) << "Resampling from 48000 to 192000 is required";
 241     sample_rate = 48000;
 242   }
 243   media::AudioSampleRate asr;
 244   if (media::ToAudioSampleRate(sample_rate, &asr)) {
 245     UMA_HISTOGRAM_ENUMERATION(
 246         "WebRTC.AudioOutputSampleRate", asr, media::kAudioSampleRateMax + 1);
 247   } else {
 248     UMA_HISTOGRAM_COUNTS("WebRTC.AudioOutputSampleRateUnexpected",
 249                          sample_rate);
 250   }
 251
 252   // Set up audio parameters for the source, i.e., the WebRTC client.
 253
 254   // The WebRTC client only supports multiples of 10ms as buffer size where
 255   // 10ms is preferred for lowest possible delay.
 256   media::AudioParameters source_params;
 257   const int frames_per_10ms = (sample_rate / 100);
 258   DVLOG(1) << "Using WebRTC output buffer size: " << frames_per_10ms;
 259
 260   source_params.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
 261                       sink_params_.channel_layout(), sink_params_.channels(),
 262                       sample_rate, 16, frames_per_10ms);
 263
 264   const int frames_per_buffer =
 265       GetOptimalBufferSize(sample_rate, sink_params_.frames_per_buffer());
 266
 267   sink_params_.Reset(sink_params_.format(), sink_params_.channel_layout(),
 268                      sink_params_.channels(), sample_rate, 16,
 269                      frames_per_buffer);
 270
 271   // Create a FIFO if re-buffering is required to match the source input with
 272   // the sink request. The source acts as provider here and the sink as
 273   // consumer.
 274   fifo_delay_milliseconds_ = 0;
 275   if (source_params.frames_per_buffer() != sink_params_.frames_per_buffer()) {
 276     DVLOG(1) << "Rebuffering from " << source_params.frames_per_buffer()
 277              << " to " << sink_params_.frames_per_buffer();
 278     audio_fifo_.reset(new media::AudioPullFifo(
 279         source_params.channels(),
 280         source_params.frames_per_buffer(),
 281         base::Bind(
 282             &WebRtcAudioRenderer::SourceCallback,
 283             base::Unretained(this))));
 284
 285     if (sink_params_.frames_per_buffer() > source_params.frames_per_buffer()) {
 286       int frame_duration_milliseconds = base::Time::kMillisecondsPerSecond /
 287           static_cast<double>(source_params.sample_rate());
 288       fifo_delay_milliseconds_ = (sink_params_.frames_per_buffer() -
 289         source_params.frames_per_buffer()) * frame_duration_milliseconds;
 290     }
 291   }
 292
 293   source_ = source;
 294
 295   // Configure the audio rendering client and start rendering.
 296   sink_ = AudioDeviceFactory::NewOutputDevice(source_render_frame_id_);
 297
 298   DCHECK_GE(session_id_, 0);
 299   sink_->InitializeWithSessionId(sink_params_, this, session_id_);
 300
 301   sink_->Start();
 302
 303   // User must call Play() before any audio can be heard.
 304   state_ = PAUSED;
 305
 306   return true;
 307 }
 308
 309 scoped_refptr<MediaStreamAudioRenderer>
 310 WebRtcAudioRenderer::CreateSharedAudioRendererProxy(
 311     const scoped_refptr<webrtc::MediaStreamInterface>& media_stream) {
 312   content::SharedAudioRenderer::OnPlayStateChanged on_play_state_changed =
 313       base::Bind(&WebRtcAudioRenderer::OnPlayStateChanged, this);
 314   return new SharedAudioRenderer(this, media_stream, on_play_state_changed);
 315 }
 316
 317 bool WebRtcAudioRenderer::IsStarted() const {
 318   DCHECK(thread_checker_.CalledOnValidThread());
 319   return start_ref_count_ != 0;
 320 }
 321
 322 void WebRtcAudioRenderer::Start() {
 323   DVLOG(1) << "WebRtcAudioRenderer::Start()";
 324   DCHECK(thread_checker_.CalledOnValidThread());
 325   ++start_ref_count_;
 326 }
 327
 328 void WebRtcAudioRenderer::Play() {
 329   DVLOG(1) << "WebRtcAudioRenderer::Play()";
 330   DCHECK(thread_checker_.CalledOnValidThread());
 331
 332   if (playing_state_.playing())
 333     return;
 334
 335   playing_state_.set_playing(true);
 336   render_callback_count_ = 0;
 337
 338   OnPlayStateChanged(media_stream_, &playing_state_);
 339 }
 340
 341 void WebRtcAudioRenderer::EnterPlayState() {
 342   DVLOG(1) << "WebRtcAudioRenderer::EnterPlayState()";
 343   DCHECK(thread_checker_.CalledOnValidThread());
 344   DCHECK_GT(start_ref_count_, 0) << "Did you forget to call Start()?";
 345   base::AutoLock auto_lock(lock_);
 346   if (state_ == UNINITIALIZED)
 347     return;
 348
 349   DCHECK(play_ref_count_ == 0 || state_ == PLAYING);
 350   ++play_ref_count_;
 351
 352   if (state_ != PLAYING) {
 353     state_ = PLAYING;
 354
 355     if (audio_fifo_) {
 356       audio_delay_milliseconds_ = 0;
 357       audio_fifo_->Clear();
 358     }
 359   }
 360 }
 361
 362 void WebRtcAudioRenderer::Pause() {
 363   DVLOG(1) << "WebRtcAudioRenderer::Pause()";
 364   DCHECK(thread_checker_.CalledOnValidThread());
 365   if (!playing_state_.playing())
 366     return;
 367
 368   playing_state_.set_playing(false);
 369
 370   OnPlayStateChanged(media_stream_, &playing_state_);
 371 }
 372
 373 void WebRtcAudioRenderer::EnterPauseState() {
 374   DVLOG(1) << "WebRtcAudioRenderer::EnterPauseState()";
 375   DCHECK(thread_checker_.CalledOnValidThread());
 376   DCHECK_GT(start_ref_count_, 0) << "Did you forget to call Start()?";
 377   base::AutoLock auto_lock(lock_);
 378   if (state_ == UNINITIALIZED)
 379     return;
 380
 381   DCHECK_EQ(state_, PLAYING);
 382   DCHECK_GT(play_ref_count_, 0);
 383   if (!--play_ref_count_)
 384     state_ = PAUSED;
 385 }
 386
 387 void WebRtcAudioRenderer::Stop() {
 388   DVLOG(1) << "WebRtcAudioRenderer::Stop()";
 389   DCHECK(thread_checker_.CalledOnValidThread());
 390   {
 391     base::AutoLock auto_lock(lock_);
 392     if (state_ == UNINITIALIZED)
 393       return;
 394
 395     if (--start_ref_count_)
 396       return;
 397
 398     DVLOG(1) << "Calling RemoveAudioRenderer and Stop().";
 399
 400     source_->RemoveAudioRenderer(this);
 401     source_ = NULL;
 402     state_ = UNINITIALIZED;
 403   }
 404
 405   // Make sure to stop the sink while _not_ holding the lock since the Render()
 406   // callback may currently be executing and try to grab the lock while we're
 407   // stopping the thread on which it runs.
 408   sink_->Stop();
 409 }
 410
 411 void WebRtcAudioRenderer::SetVolume(float volume) {
 412   DCHECK(thread_checker_.CalledOnValidThread());
 413   DCHECK(volume >= 0.0f && volume <= 1.0f);
 414
 415   playing_state_.set_volume(volume);
 416   OnPlayStateChanged(media_stream_, &playing_state_);
 417 }
 418
 419 void WebRtcAudioRenderer::SwitchOutputDevice(
 420     const std::string& device_id,
 421     const GURL& security_origin,
 422     const media::SwitchOutputDeviceCB& callback) {
 423   DCHECK(thread_checker_.CalledOnValidThread());
 424   DCHECK(sink_);
 425   DVLOG(1) << __FUNCTION__
 426            << "(" << device_id << ", " << security_origin << ")";
 427   sink_->SwitchOutputDevice(device_id, security_origin, callback);
 428 }
 429
 430 base::TimeDelta WebRtcAudioRenderer::GetCurrentRenderTime() const {
 431   DCHECK(thread_checker_.CalledOnValidThread());
 432   base::AutoLock auto_lock(lock_);
 433   return current_time_;
 434 }
 435
 436 bool WebRtcAudioRenderer::IsLocalRenderer() const {
 437   return false;
 438 }
 439
 440 int WebRtcAudioRenderer::Render(media::AudioBus* audio_bus,
 441                                 int audio_delay_milliseconds) {
 442   base::AutoLock auto_lock(lock_);
 443   if (!source_)
 444     return 0;
 445
 446   DVLOG(2) << "WebRtcAudioRenderer::Render()";
 447   DVLOG(2) << "audio_delay_milliseconds: " << audio_delay_milliseconds;
 448
 449   audio_delay_milliseconds_ = audio_delay_milliseconds;
 450
 451   if (audio_fifo_)
 452     audio_fifo_->Consume(audio_bus, audio_bus->frames());
 453   else
 454     SourceCallback(0, audio_bus);
 455
 456   return (state_ == PLAYING) ? audio_bus->frames() : 0;
 457 }
 458
 459 void WebRtcAudioRenderer::OnRenderError() {
 460   NOTIMPLEMENTED();
 461   LOG(ERROR) << "OnRenderError()";
 462 }
 463
 464 // Called by AudioPullFifo when more data is necessary.
 465 void WebRtcAudioRenderer::SourceCallback(
 466     int fifo_frame_delay, media::AudioBus* audio_bus) {
 467   base::TimeTicks start_time = base::TimeTicks::Now() ;
 468   DVLOG(2) << "WebRtcAudioRenderer::SourceCallback("
 469            << fifo_frame_delay << ", "
 470            << audio_bus->frames() << ")";
 471
 472   int output_delay_milliseconds = audio_delay_milliseconds_;
 473   output_delay_milliseconds += fifo_delay_milliseconds_;
 474   DVLOG(2) << "output_delay_milliseconds: " << output_delay_milliseconds;
 475
 476   // We need to keep render data for the |source_| regardless of |state_|,
 477   // otherwise the data will be buffered up inside |source_|.
 478   source_->RenderData(audio_bus, sink_params_.sample_rate(),
 479                       output_delay_milliseconds,
 480                       &current_time_);
 481
 482   // Avoid filling up the audio bus if we are not playing; instead
 483   // return here and ensure that the returned value in Render() is 0.
 484   if (state_ != PLAYING)
 485     audio_bus->Zero();
 486
 487   if (++render_callback_count_ == kNumCallbacksBetweenRenderTimeHistograms) {
 488     base::TimeDelta elapsed = base::TimeTicks::Now() - start_time;
 489     render_callback_count_ = 0;
 490     UMA_HISTOGRAM_TIMES("WebRTC.AudioRenderTimes", elapsed);
 491   }
 492 }
 493
 494 void WebRtcAudioRenderer::UpdateSourceVolume(
 495     webrtc::AudioSourceInterface* source) {
 496   DCHECK(thread_checker_.CalledOnValidThread());
 497
 498   // Note: If there are no playing audio renderers, then the volume will be
 499   // set to 0.0.
 500   float volume = 0.0f;
 501
 502   SourcePlayingStates::iterator entry = source_playing_states_.find(source);
 503   if (entry != source_playing_states_.end()) {
 504     PlayingStates& states = entry->second;
 505     for (PlayingStates::const_iterator it = states.begin();
 506          it != states.end(); ++it) {
 507       if ((*it)->playing())
 508         volume += (*it)->volume();
 509     }
 510   }
 511
 512   // The valid range for volume scaling of a remote webrtc source is
 513   // 0.0-10.0 where 1.0 is no attenuation/boost.
 514   DCHECK(volume >= 0.0f);
 515   if (volume > 10.0f)
 516     volume = 10.0f;
 517
 518   DVLOG(1) << "Setting remote source volume: " << volume;
 519   if (!signaling_thread_->BelongsToCurrentThread()) {
 520     // Libjingle hands out proxy objects in most cases, but the audio source
 521     // object is an exception (bug?).  So, to work around that, we need to make
 522     // sure we call SetVolume on the signaling thread.
 523     signaling_thread_->PostTask(FROM_HERE,
 524         base::Bind(&webrtc::AudioSourceInterface::SetVolume, source, volume));
 525   } else {
 526     source->SetVolume(volume);
 527   }
 528 }
 529
 530 bool WebRtcAudioRenderer::AddPlayingState(
 531     webrtc::AudioSourceInterface* source,
 532     PlayingState* state) {
 533   DCHECK(thread_checker_.CalledOnValidThread());
 534   DCHECK(state->playing());
 535   // Look up or add the |source| to the map.
 536   PlayingStates& array = source_playing_states_[source];
 537   if (std::find(array.begin(), array.end(), state) != array.end())
 538     return false;
 539
 540   array.push_back(state);
 541
 542   return true;
 543 }
 544
 545 bool WebRtcAudioRenderer::RemovePlayingState(
 546     webrtc::AudioSourceInterface* source,
 547     PlayingState* state) {
 548   DCHECK(thread_checker_.CalledOnValidThread());
 549   DCHECK(!state->playing());
 550   SourcePlayingStates::iterator found = source_playing_states_.find(source);
 551   if (found == source_playing_states_.end())
 552     return false;
 553
 554   PlayingStates& array = found->second;
 555   PlayingStates::iterator state_it =
 556       std::find(array.begin(), array.end(), state);
 557   if (state_it == array.end())
 558     return false;
 559
 560   array.erase(state_it);
 561
 562   if (array.empty())
 563     source_playing_states_.erase(found);
 564
 565   return true;
 566 }
 567
 568 void WebRtcAudioRenderer::OnPlayStateChanged(
 569     const scoped_refptr<webrtc::MediaStreamInterface>& media_stream,
 570     PlayingState* state) {
 571   webrtc::AudioTrackVector tracks(media_stream->GetAudioTracks());
 572   for (webrtc::AudioTrackVector::iterator it = tracks.begin();
 573        it != tracks.end(); ++it) {
 574     webrtc::AudioSourceInterface* source = (*it)->GetSource();
 575     DCHECK(source);
 576     if (!state->playing()) {
 577       if (RemovePlayingState(source, state))
 578         EnterPauseState();
 579     } else if (AddPlayingState(source, state)) {
 580       EnterPlayState();
 581     }
 582     UpdateSourceVolume(source);
 583   }
 584 }
 585
 586 }  // namespace content