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