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