content/renderer/media/webrtc_local_audio_renderer.h

   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 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_
   6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_
   7
   8 #include <string>
   9 #include <vector>
  10
  11 #include "base/callback.h"
  12 #include "base/memory/ref_counted.h"
  13 #include "base/single_thread_task_runner.h"
  14 #include "base/synchronization/lock.h"
  15 #include "base/threading/thread_checker.h"
  16 #include "content/common/content_export.h"
  17 #include "content/public/renderer/media_stream_audio_renderer.h"
  18 #include "content/public/renderer/media_stream_audio_sink.h"
  19 #include "content/renderer/media/webrtc_audio_device_impl.h"
  20 #include "content/renderer/media/webrtc_local_audio_track.h"
  21 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
  22
  23 namespace media {
  24 class AudioBus;
  25 class AudioShifter;
  26 class AudioOutputDevice;
  27 class AudioParameters;
  28 }
  29
  30 namespace content {
  31
  32 class WebRtcAudioCapturer;
  33
  34 // WebRtcLocalAudioRenderer is a MediaStreamAudioRenderer designed for rendering
  35 // local audio media stream tracks,
  36 // http://dev.w3.org/2011/webrtc/editor/getusermedia.html#mediastreamtrack
  37 // It also implements media::AudioRendererSink::RenderCallback to render audio
  38 // data provided from a WebRtcLocalAudioTrack source.
  39 // When the audio layer in the browser process asks for data to render, this
  40 // class provides the data by implementing the MediaStreamAudioSink
  41 // interface, i.e., we are a sink seen from the WebRtcAudioCapturer perspective.
  42 // TODO(henrika): improve by using similar principles as in
  43 // MediaStreamVideoRendererSink which register itself to the video track when
  44 // the provider is started and deregisters itself when it is stopped. Tracking
  45 // this at http://crbug.com/164813.
  46 class CONTENT_EXPORT WebRtcLocalAudioRenderer
  47     : NON_EXPORTED_BASE(public MediaStreamAudioRenderer),
  48       NON_EXPORTED_BASE(public MediaStreamAudioSink),
  49       NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback) {
  50  public:
  51   // Creates a local renderer and registers a capturing |source| object.
  52   // The |source| is owned by the WebRtcAudioDeviceImpl.
  53   // Called on the main thread.
  54   WebRtcLocalAudioRenderer(const blink::WebMediaStreamTrack& audio_track,
  55                            int source_render_frame_id,
  56                            int session_id,
  57                            int frames_per_buffer);
  58
  59   // MediaStreamAudioRenderer implementation.
  60   // Called on the main thread.
  61   void Start() override;
  62   void Stop() override;
  63   void Play() override;
  64   void Pause() override;
  65   void SetVolume(float volume) override;
  66   media::OutputDevice* GetOutputDevice() override;
  67   base::TimeDelta GetCurrentRenderTime() const override;
  68   bool IsLocalRenderer() const override;
  69
  70   const base::TimeDelta& total_render_time() const {
  71     return total_render_time_;
  72   }
  73
  74  protected:
  75   ~WebRtcLocalAudioRenderer() override;
  76
  77  private:
  78   // MediaStreamAudioSink implementation.
  79
  80   // Called on the AudioInputDevice worker thread.
  81   void OnData(const media::AudioBus& audio_bus,
  82               base::TimeTicks estimated_capture_time) override;
  83
  84   // Called on the AudioInputDevice worker thread.
  85   void OnSetFormat(const media::AudioParameters& params) override;
  86
  87   // media::AudioRendererSink::RenderCallback implementation.
  88   // Render() is called on the AudioOutputDevice thread and OnRenderError()
  89   // on the IO thread.
  90   int Render(media::AudioBus* audio_bus, int audio_delay_milliseconds) override;
  91   void OnRenderError() override;
  92
  93   // Initializes and starts the |sink_| if
  94   //  we have received valid |source_params_| &&
  95   //  |playing_| has been set to true &&
  96   //  |volume_| is not zero.
  97   void MaybeStartSink();
  98
  99   // Sets new |source_params_| and then re-initializes and restarts |sink_|.
 100   void ReconfigureSink(const media::AudioParameters& params);
 101
 102   // The audio track which provides data to render. Given that this class
 103   // implements local loopback, the audio track is getting data from a capture
 104   // instance like a selected microphone and forwards the recorded data to its
 105   // sinks. The recorded data is stored in a FIFO and consumed
 106   // by this class when the sink asks for new data.
 107   // This class is calling MediaStreamAudioSink::AddToAudioTrack() and
 108   // MediaStreamAudioSink::RemoveFromAudioTrack() to connect and disconnect
 109   // with the audio track.
 110   blink::WebMediaStreamTrack audio_track_;
 111
 112   // The render view and frame in which the audio is rendered into |sink_|.
 113   const int source_render_frame_id_;
 114   const int session_id_;
 115
 116   // MessageLoop associated with the single thread that performs all control
 117   // tasks.  Set to the MessageLoop that invoked the ctor.
 118   const scoped_refptr<base::SingleThreadTaskRunner> task_runner_;
 119
 120   // The sink (destination) for rendered audio.
 121   scoped_refptr<media::AudioOutputDevice> sink_;
 122
 123   // This does all the synchronization/resampling/smoothing.
 124   scoped_ptr<media::AudioShifter> audio_shifter_;
 125
 126   // Stores last time a render callback was received. The time difference
 127   // between a new time stamp and this value can be used to derive the
 128   // total render time.
 129   base::TimeTicks last_render_time_;
 130
 131   // Keeps track of total time audio has been rendered.
 132   base::TimeDelta total_render_time_;
 133
 134   // The audio parameters of the capture source.
 135   // Must only be touched on the main thread.
 136   media::AudioParameters source_params_;
 137
 138   // The audio parameters used by the sink.
 139   // Must only be touched on the main thread.
 140   media::AudioParameters sink_params_;
 141
 142   // Set when playing, cleared when paused.
 143   bool playing_;
 144
 145   // Protects |audio_shifter_|, |playing_| and |sink_|.
 146   mutable base::Lock thread_lock_;
 147
 148   // The preferred buffer size provided via the ctor.
 149   const int frames_per_buffer_;
 150
 151   // The preferred device id of the output device or empty for the default
 152   // output device.
 153   const std::string output_device_id_;
 154
 155   // Cache value for the volume.
 156   float volume_;
 157
 158   // Flag to indicate whether |sink_| has been started yet.
 159   bool sink_started_;
 160
 161   // Used to DCHECK that some methods are called on the capture audio thread.
 162   base::ThreadChecker capture_thread_checker_;
 163
 164   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer);
 165 };
 166
 167 }  // namespace content
 168
 169 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_