root/content/renderer/media/webrtc_audio_renderer.cc

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DEFINITIONS

This source file includes following definitions.
  1. AsAudioFramesPerBuffer
  2. AddHistogramFramesPerBuffer
  3. on_play_state_changed_
  4. Start
  5. Play
  6. Pause
  7. Stop
  8. SetVolume
  9. GetCurrentRenderTime
  10. IsLocalRenderer
  11. sink_params_
  12. Initialize
  13. CreateSharedAudioRendererProxy
  14. IsStarted
  15. Start
  16. Play
  17. EnterPlayState
  18. Pause
  19. EnterPauseState
  20. Stop
  21. SetVolume
  22. GetCurrentRenderTime
  23. IsLocalRenderer
  24. OnRenderError
  25. SourceCallback
  26. UpdateSourceVolume
  27. AddPlayingState
  28. RemovePlayingState
  29. OnPlayStateChanged

// Copyright (c) 2012 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#include "content/renderer/media/webrtc_audio_renderer.h"

#include "base/logging.h"
#include "base/metrics/histogram.h"
#include "base/strings/string_util.h"
#include "base/strings/stringprintf.h"
#include "content/renderer/media/audio_device_factory.h"
#include "content/renderer/media/webrtc_audio_device_impl.h"
#include "content/renderer/media/webrtc_logging.h"
#include "media/audio/audio_output_device.h"
#include "media/audio/audio_parameters.h"
#include "media/audio/sample_rates.h"
#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
#include "third_party/libjingle/source/talk/media/base/audiorenderer.h"


#if defined(OS_WIN)
#include "base/win/windows_version.h"
#include "media/audio/win/core_audio_util_win.h"
#endif

namespace content {

namespace {

// Supported hardware sample rates for output sides.
#if defined(OS_WIN) || defined(OS_MACOSX)
// AudioHardwareConfig::GetOutputSampleRate() asks the audio layer for its
// current sample rate (set by the user) on Windows and Mac OS X.  The listed
// rates below adds restrictions and Initialize() will fail if the user selects
// any rate outside these ranges.
const int kValidOutputRates[] = {96000, 48000, 44100, 32000, 16000};
#elif defined(OS_LINUX) || defined(OS_OPENBSD)
const int kValidOutputRates[] = {48000, 44100};
#elif defined(OS_ANDROID)
// TODO(leozwang): We want to use native sampling rate on Android to achieve
// low latency, currently 16000 is used to work around audio problem on some
// Android devices.
const int kValidOutputRates[] = {48000, 44100, 16000};
#else
const int kValidOutputRates[] = {44100};
#endif

// TODO(xians): Merge the following code to WebRtcAudioCapturer, or remove.
enum AudioFramesPerBuffer {
  k160,
  k320,
  k440,
  k480,
  k640,
  k880,
  k960,
  k1440,
  k1920,
  kUnexpectedAudioBufferSize  // Must always be last!
};

// Helper method to convert integral values to their respective enum values
// above, or kUnexpectedAudioBufferSize if no match exists.
// We map 441 to k440 to avoid changes in the XML part for histograms.
// It is still possible to map the histogram result to the actual buffer size.
// See http://crbug.com/243450 for details.
AudioFramesPerBuffer AsAudioFramesPerBuffer(int frames_per_buffer) {
  switch (frames_per_buffer) {
    case 160: return k160;
    case 320: return k320;
    case 441: return k440;
    case 480: return k480;
    case 640: return k640;
    case 880: return k880;
    case 960: return k960;
    case 1440: return k1440;
    case 1920: return k1920;
  }
  return kUnexpectedAudioBufferSize;
}

void AddHistogramFramesPerBuffer(int param) {
  AudioFramesPerBuffer afpb = AsAudioFramesPerBuffer(param);
  if (afpb != kUnexpectedAudioBufferSize) {
    UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputFramesPerBuffer",
                              afpb, kUnexpectedAudioBufferSize);
  } else {
    // Report unexpected sample rates using a unique histogram name.
    UMA_HISTOGRAM_COUNTS("WebRTC.AudioOutputFramesPerBufferUnexpected", param);
  }
}

// This is a simple wrapper class that's handed out to users of a shared
// WebRtcAudioRenderer instance.  This class maintains the per-user 'playing'
// and 'started' states to avoid problems related to incorrect usage which
// might violate the implementation assumptions inside WebRtcAudioRenderer
// (see the play reference count).
class SharedAudioRenderer : public MediaStreamAudioRenderer {
 public:
  // Callback definition for a callback that is called when when Play(), Pause()
  // or SetVolume are called (whenever the internal |playing_state_| changes).
  typedef base::Callback<
      void(const scoped_refptr<webrtc::MediaStreamInterface>&,
           WebRtcAudioRenderer::PlayingState*)> OnPlayStateChanged;

  SharedAudioRenderer(
      const scoped_refptr<MediaStreamAudioRenderer>& delegate,
      const scoped_refptr<webrtc::MediaStreamInterface>& media_stream,
      const OnPlayStateChanged& on_play_state_changed)
      : delegate_(delegate), media_stream_(media_stream), started_(false),
        on_play_state_changed_(on_play_state_changed) {
    DCHECK(!on_play_state_changed_.is_null());
    DCHECK(media_stream_.get());
  }

 protected:
  virtual ~SharedAudioRenderer() {
    DCHECK(thread_checker_.CalledOnValidThread());
    DVLOG(1) << __FUNCTION__;
    Stop();
  }

  virtual void Start() OVERRIDE {
    DCHECK(thread_checker_.CalledOnValidThread());
    if (started_)
      return;
    started_ = true;
    delegate_->Start();
  }

  virtual void Play() OVERRIDE {
    DCHECK(thread_checker_.CalledOnValidThread());
    DCHECK(started_);
    if (playing_state_.playing())
      return;
    playing_state_.set_playing(true);
    on_play_state_changed_.Run(media_stream_, &playing_state_);
  }

  virtual void Pause() OVERRIDE {
    DCHECK(thread_checker_.CalledOnValidThread());
    DCHECK(started_);
    if (!playing_state_.playing())
      return;
    playing_state_.set_playing(false);
    on_play_state_changed_.Run(media_stream_, &playing_state_);
  }

  virtual void Stop() OVERRIDE {
    DCHECK(thread_checker_.CalledOnValidThread());
    if (!started_)
      return;
    Pause();
    started_ = false;
    delegate_->Stop();
  }

  virtual void SetVolume(float volume) OVERRIDE {
    DCHECK(thread_checker_.CalledOnValidThread());
    DCHECK(volume >= 0.0f && volume <= 1.0f);
    playing_state_.set_volume(volume);
    on_play_state_changed_.Run(media_stream_, &playing_state_);
  }

  virtual base::TimeDelta GetCurrentRenderTime() const OVERRIDE {
    DCHECK(thread_checker_.CalledOnValidThread());
    return delegate_->GetCurrentRenderTime();
  }

  virtual bool IsLocalRenderer() const OVERRIDE {
    DCHECK(thread_checker_.CalledOnValidThread());
    return delegate_->IsLocalRenderer();
  }

 private:
  base::ThreadChecker thread_checker_;
  const scoped_refptr<MediaStreamAudioRenderer> delegate_;
  const scoped_refptr<webrtc::MediaStreamInterface> media_stream_;
  bool started_;
  WebRtcAudioRenderer::PlayingState playing_state_;
  OnPlayStateChanged on_play_state_changed_;
};

}  // namespace

WebRtcAudioRenderer::WebRtcAudioRenderer(
    const scoped_refptr<webrtc::MediaStreamInterface>& media_stream,
    int source_render_view_id,
    int source_render_frame_id,
    int session_id,
    int sample_rate,
    int frames_per_buffer)
    : state_(UNINITIALIZED),
      source_render_view_id_(source_render_view_id),
      source_render_frame_id_(source_render_frame_id),
      session_id_(session_id),
      media_stream_(media_stream),
      source_(NULL),
      play_ref_count_(0),
      start_ref_count_(0),
      audio_delay_milliseconds_(0),
      fifo_delay_milliseconds_(0),
      sink_params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                   media::CHANNEL_LAYOUT_STEREO, 0, sample_rate, 16,
                   frames_per_buffer, media::AudioParameters::DUCKING) {
  WebRtcLogMessage(base::StringPrintf(
      "WAR::WAR. source_render_view_id=%d"
      ", session_id=%d, sample_rate=%d, frames_per_buffer=%d",
      source_render_view_id,
      session_id,
      sample_rate,
      frames_per_buffer));
}

WebRtcAudioRenderer::~WebRtcAudioRenderer() {
  DCHECK(thread_checker_.CalledOnValidThread());
  DCHECK_EQ(state_, UNINITIALIZED);
}

bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) {
  DVLOG(1) << "WebRtcAudioRenderer::Initialize()";
  DCHECK(thread_checker_.CalledOnValidThread());
  base::AutoLock auto_lock(lock_);
  DCHECK_EQ(state_, UNINITIALIZED);
  DCHECK(source);
  DCHECK(!sink_.get());
  DCHECK(!source_);

  // WebRTC does not yet support higher rates than 96000 on the client side
  // and 48000 is the preferred sample rate. Therefore, if 192000 is detected,
  // we change the rate to 48000 instead. The consequence is that the native
  // layer will be opened up at 192kHz but WebRTC will provide data at 48kHz
  // which will then be resampled by the audio converted on the browser side
  // to match the native audio layer.
  int sample_rate = sink_params_.sample_rate();
  DVLOG(1) << "Audio output hardware sample rate: " << sample_rate;
  if (sample_rate == 192000) {
    DVLOG(1) << "Resampling from 48000 to 192000 is required";
    sample_rate = 48000;
  }
  media::AudioSampleRate asr;
  if (media::ToAudioSampleRate(sample_rate, &asr)) {
    UMA_HISTOGRAM_ENUMERATION(
        "WebRTC.AudioOutputSampleRate", asr, media::kAudioSampleRateMax + 1);
  } else {
    UMA_HISTOGRAM_COUNTS("WebRTC.AudioOutputSampleRateUnexpected",
                         sample_rate);
  }

  // Verify that the reported output hardware sample rate is supported
  // on the current platform.
  if (std::find(&kValidOutputRates[0],
                &kValidOutputRates[0] + arraysize(kValidOutputRates),
                sample_rate) ==
                    &kValidOutputRates[arraysize(kValidOutputRates)]) {
    DLOG(ERROR) << sample_rate << " is not a supported output rate.";
    return false;
  }

  // Set up audio parameters for the source, i.e., the WebRTC client.

  // The WebRTC client only supports multiples of 10ms as buffer size where
  // 10ms is preferred for lowest possible delay.
  media::AudioParameters source_params;
  const int frames_per_10ms = (sample_rate / 100);
  DVLOG(1) << "Using WebRTC output buffer size: " << frames_per_10ms;

  source_params.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                      sink_params_.channel_layout(), sink_params_.channels(), 0,
                      sample_rate, 16, frames_per_10ms);

  // Update audio parameters for the sink, i.e., the native audio output stream.
  // We strive to open up using native parameters to achieve best possible
  // performance and to ensure that no FIFO is needed on the browser side to
  // match the client request. Any mismatch between the source and the sink is
  // taken care of in this class instead using a pull FIFO.

  // Use native output size as default.
  int frames_per_buffer = sink_params_.frames_per_buffer();
#if defined(OS_ANDROID)
  // TODO(henrika): Keep tuning this scheme and espcicially for low-latency
  // cases. Might not be possible to come up with the perfect solution using
  // the render side only.
  if (frames_per_buffer < 2 * frames_per_10ms) {
    // Examples of low-latency frame sizes and the resulting |buffer_size|:
    //  Nexus 7     : 240 audio frames => 2*480 = 960
    //  Nexus 10    : 256              => 2*441 = 882
    //  Galaxy Nexus: 144              => 2*441 = 882
    frames_per_buffer = 2 * frames_per_10ms;
    DVLOG(1) << "Low-latency output detected on Android";
  }
#endif
  DVLOG(1) << "Using sink output buffer size: " << frames_per_buffer;

  sink_params_.Reset(sink_params_.format(), sink_params_.channel_layout(),
                     sink_params_.channels(), 0, sample_rate, 16,
                     frames_per_buffer);

  // Create a FIFO if re-buffering is required to match the source input with
  // the sink request. The source acts as provider here and the sink as
  // consumer.
  fifo_delay_milliseconds_ = 0;
  if (source_params.frames_per_buffer() != sink_params_.frames_per_buffer()) {
    DVLOG(1) << "Rebuffering from " << source_params.frames_per_buffer()
             << " to " << sink_params_.frames_per_buffer();
    audio_fifo_.reset(new media::AudioPullFifo(
        source_params.channels(),
        source_params.frames_per_buffer(),
        base::Bind(
            &WebRtcAudioRenderer::SourceCallback,
            base::Unretained(this))));

    if (sink_params_.frames_per_buffer() > source_params.frames_per_buffer()) {
      int frame_duration_milliseconds = base::Time::kMillisecondsPerSecond /
          static_cast<double>(source_params.sample_rate());
      fifo_delay_milliseconds_ = (sink_params_.frames_per_buffer() -
        source_params.frames_per_buffer()) * frame_duration_milliseconds;
    }
  }

  source_ = source;

  // Configure the audio rendering client and start rendering.
  sink_ = AudioDeviceFactory::NewOutputDevice(
      source_render_view_id_, source_render_frame_id_);

  // TODO(tommi): Rename InitializeUnifiedStream to rather reflect association
  // with a session.
  DCHECK_GE(session_id_, 0);
  sink_->InitializeUnifiedStream(sink_params_, this, session_id_);

  sink_->Start();

  // User must call Play() before any audio can be heard.
  state_ = PAUSED;

  UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputFramesPerBuffer",
                            source_params.frames_per_buffer(),
                            kUnexpectedAudioBufferSize);
  AddHistogramFramesPerBuffer(source_params.frames_per_buffer());

  return true;
}

scoped_refptr<MediaStreamAudioRenderer>
WebRtcAudioRenderer::CreateSharedAudioRendererProxy(
    const scoped_refptr<webrtc::MediaStreamInterface>& media_stream) {
  content::SharedAudioRenderer::OnPlayStateChanged on_play_state_changed =
      base::Bind(&WebRtcAudioRenderer::OnPlayStateChanged, this);
  return new SharedAudioRenderer(this, media_stream, on_play_state_changed);
}

bool WebRtcAudioRenderer::IsStarted() const {
  DCHECK(thread_checker_.CalledOnValidThread());
  return start_ref_count_ != 0;
}

void WebRtcAudioRenderer::Start() {
  DVLOG(1) << "WebRtcAudioRenderer::Start()";
  DCHECK(thread_checker_.CalledOnValidThread());
  ++start_ref_count_;
}

void WebRtcAudioRenderer::Play() {
  DVLOG(1) << "WebRtcAudioRenderer::Play()";
  DCHECK(thread_checker_.CalledOnValidThread());

  if (playing_state_.playing())
    return;

  playing_state_.set_playing(true);

  OnPlayStateChanged(media_stream_, &playing_state_);
}

void WebRtcAudioRenderer::EnterPlayState() {
  DVLOG(1) << "WebRtcAudioRenderer::EnterPlayState()";
  DCHECK(thread_checker_.CalledOnValidThread());
  DCHECK_GT(start_ref_count_, 0) << "Did you forget to call Start()?";
  base::AutoLock auto_lock(lock_);
  if (state_ == UNINITIALIZED)
    return;

  DCHECK(play_ref_count_ == 0 || state_ == PLAYING);
  ++play_ref_count_;

  if (state_ != PLAYING) {
    state_ = PLAYING;

    if (audio_fifo_) {
      audio_delay_milliseconds_ = 0;
      audio_fifo_->Clear();
    }
  }
}

void WebRtcAudioRenderer::Pause() {
  DVLOG(1) << "WebRtcAudioRenderer::Pause()";
  DCHECK(thread_checker_.CalledOnValidThread());
  if (!playing_state_.playing())
    return;

  playing_state_.set_playing(false);

  OnPlayStateChanged(media_stream_, &playing_state_);
}

void WebRtcAudioRenderer::EnterPauseState() {
  DVLOG(1) << "WebRtcAudioRenderer::EnterPauseState()";
  DCHECK(thread_checker_.CalledOnValidThread());
  DCHECK_GT(start_ref_count_, 0) << "Did you forget to call Start()?";
  base::AutoLock auto_lock(lock_);
  if (state_ == UNINITIALIZED)
    return;

  DCHECK_EQ(state_, PLAYING);
  DCHECK_GT(play_ref_count_, 0);
  if (!--play_ref_count_)
    state_ = PAUSED;
}

void WebRtcAudioRenderer::Stop() {
  DVLOG(1) << "WebRtcAudioRenderer::Stop()";
  DCHECK(thread_checker_.CalledOnValidThread());
  {
    base::AutoLock auto_lock(lock_);
    if (state_ == UNINITIALIZED)
      return;

    if (--start_ref_count_)
      return;

    DVLOG(1) << "Calling RemoveAudioRenderer and Stop().";

    source_->RemoveAudioRenderer(this);
    source_ = NULL;
    state_ = UNINITIALIZED;
  }

  // Make sure to stop the sink while _not_ holding the lock since the Render()
  // callback may currently be executing and try to grab the lock while we're
  // stopping the thread on which it runs.
  sink_->Stop();
}

void WebRtcAudioRenderer::SetVolume(float volume) {
  DCHECK(thread_checker_.CalledOnValidThread());
  DCHECK(volume >= 0.0f && volume <= 1.0f);

  playing_state_.set_volume(volume);
  OnPlayStateChanged(media_stream_, &playing_state_);
}

base::TimeDelta WebRtcAudioRenderer::GetCurrentRenderTime() const {
  return base::TimeDelta();
}

bool WebRtcAudioRenderer::IsLocalRenderer() const {
  return false;
}

int WebRtcAudioRenderer::Render(media::AudioBus* audio_bus,
                                int audio_delay_milliseconds) {
  base::AutoLock auto_lock(lock_);
  if (!source_)
    return 0;

  DVLOG(2) << "WebRtcAudioRenderer::Render()";
  DVLOG(2) << "audio_delay_milliseconds: " << audio_delay_milliseconds;

  audio_delay_milliseconds_ = audio_delay_milliseconds;

  if (audio_fifo_)
    audio_fifo_->Consume(audio_bus, audio_bus->frames());
  else
    SourceCallback(0, audio_bus);

  return (state_ == PLAYING) ? audio_bus->frames() : 0;
}

void WebRtcAudioRenderer::OnRenderError() {
  NOTIMPLEMENTED();
  LOG(ERROR) << "OnRenderError()";
}

// Called by AudioPullFifo when more data is necessary.
void WebRtcAudioRenderer::SourceCallback(
    int fifo_frame_delay, media::AudioBus* audio_bus) {
  DVLOG(2) << "WebRtcAudioRenderer::SourceCallback("
           << fifo_frame_delay << ", "
           << audio_bus->frames() << ")";

  int output_delay_milliseconds = audio_delay_milliseconds_;
  output_delay_milliseconds += fifo_delay_milliseconds_;
  DVLOG(2) << "output_delay_milliseconds: " << output_delay_milliseconds;

  // We need to keep render data for the |source_| regardless of |state_|,
  // otherwise the data will be buffered up inside |source_|.
  source_->RenderData(audio_bus, sink_params_.sample_rate(),
                      output_delay_milliseconds);

  // Avoid filling up the audio bus if we are not playing; instead
  // return here and ensure that the returned value in Render() is 0.
  if (state_ != PLAYING)
    audio_bus->Zero();
}

void WebRtcAudioRenderer::UpdateSourceVolume(
    webrtc::AudioSourceInterface* source) {
  DCHECK(thread_checker_.CalledOnValidThread());

  // Note: If there are no playing audio renderers, then the volume will be
  // set to 0.0.
  float volume = 0.0f;

  SourcePlayingStates::iterator entry = source_playing_states_.find(source);
  if (entry != source_playing_states_.end()) {
    PlayingStates& states = entry->second;
    for (PlayingStates::const_iterator it = states.begin();
         it != states.end(); ++it) {
      if ((*it)->playing())
        volume += (*it)->volume();
    }
  }

  // The valid range for volume scaling of a remote webrtc source is
  // 0.0-10.0 where 1.0 is no attenuation/boost.
  DCHECK(volume >= 0.0f);
  if (volume > 10.0f)
    volume = 10.0f;

  DVLOG(1) << "Setting remote source volume: " << volume;
  source->SetVolume(volume);
}

bool WebRtcAudioRenderer::AddPlayingState(
    webrtc::AudioSourceInterface* source,
    PlayingState* state) {
  DCHECK(thread_checker_.CalledOnValidThread());
  DCHECK(state->playing());
  // Look up or add the |source| to the map.
  PlayingStates& array = source_playing_states_[source];
  if (std::find(array.begin(), array.end(), state) != array.end())
    return false;

  array.push_back(state);

  return true;
}

bool WebRtcAudioRenderer::RemovePlayingState(
    webrtc::AudioSourceInterface* source,
    PlayingState* state) {
  DCHECK(thread_checker_.CalledOnValidThread());
  DCHECK(!state->playing());
  SourcePlayingStates::iterator found = source_playing_states_.find(source);
  if (found == source_playing_states_.end())
    return false;

  PlayingStates& array = found->second;
  PlayingStates::iterator state_it =
      std::find(array.begin(), array.end(), state);
  if (state_it == array.end())
    return false;

  array.erase(state_it);

  if (array.empty())
    source_playing_states_.erase(found);

  return true;
}

void WebRtcAudioRenderer::OnPlayStateChanged(
    const scoped_refptr<webrtc::MediaStreamInterface>& media_stream,
    PlayingState* state) {
  webrtc::AudioTrackVector tracks(media_stream->GetAudioTracks());
  for (webrtc::AudioTrackVector::iterator it = tracks.begin();
       it != tracks.end(); ++it) {
    webrtc::AudioSourceInterface* source = (*it)->GetSource();
    DCHECK(source);
    if (!state->playing()) {
      if (RemovePlayingState(source, state))
        EnterPauseState();
    } else if (AddPlayingState(source, state)) {
      EnterPlayState();
    }
    UpdateSourceVolume(source);
  }
}

}  // namespace content

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