content/renderer/media/webrtc_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_AUDIO_RENDERER_H_
   6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
   7
   8 #include "base/memory/ref_counted.h"
   9 #include "base/synchronization/lock.h"
  10 #include "base/threading/thread_checker.h"
  11 #include "content/renderer/media/webrtc_audio_device_impl.h"
  12 #include "media/base/audio_decoder.h"
  13 #include "media/base/audio_pull_fifo.h"
  14 #include "media/base/audio_renderer_sink.h"
  15 #include "webkit/media/media_stream_audio_renderer.h"
  16
  17 namespace content {
  18
  19 class RendererAudioOutputDevice;
  20 class WebRtcAudioRendererSource;
  21
  22 // This renderer handles calls from the pipeline and WebRtc ADM. It is used
  23 // for connecting WebRtc MediaStream with the audio pipeline.
  24 class CONTENT_EXPORT WebRtcAudioRenderer
  25     : NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
  26       NON_EXPORTED_BASE(public webkit_media::MediaStreamAudioRenderer) {
  27  public:
  28   explicit WebRtcAudioRenderer(int source_render_view_id);
  29
  30   // Initialize function called by clients like WebRtcAudioDeviceImpl.
  31   // Stop() has to be called before |source| is deleted.
  32   bool Initialize(WebRtcAudioRendererSource* source);
  33
  34   // Methods called by WebMediaPlayerMS and WebRtcAudioDeviceImpl.
  35   // MediaStreamAudioRenderer implementation.
  36   virtual void Start() OVERRIDE;
  37   virtual void Play() OVERRIDE;
  38   virtual void Pause() OVERRIDE;
  39   virtual void Stop() OVERRIDE;
  40   virtual void SetVolume(float volume) OVERRIDE;
  41   virtual base::TimeDelta GetCurrentRenderTime() const OVERRIDE;
  42   virtual bool IsLocalRenderer() const OVERRIDE;
  43
  44  protected:
  45   virtual ~WebRtcAudioRenderer();
  46
  47  private:
  48   enum State {
  49     UNINITIALIZED,
  50     PLAYING,
  51     PAUSED,
  52   };
  53
  54   // Used to DCHECK that we are called on the correct thread.
  55   base::ThreadChecker thread_checker_;
  56
  57   // Flag to keep track the state of the renderer.
  58   State state_;
  59
  60   // media::AudioRendererSink::RenderCallback implementation.
  61   // These two methods are called on the AudioOutputDevice worker thread.
  62   virtual int Render(media::AudioBus* audio_bus,
  63                      int audio_delay_milliseconds) OVERRIDE;
  64   virtual void OnRenderError() OVERRIDE;
  65
  66   // Called by AudioPullFifo when more data is necessary.
  67   // This method is called on the AudioOutputDevice worker thread.
  68   void SourceCallback(int fifo_frame_delay, media::AudioBus* audio_bus);
  69
  70   // The render view in which the audio is rendered into |sink_|.
  71   const int source_render_view_id_;
  72
  73   // The sink (destination) for rendered audio.
  74   scoped_refptr<RendererAudioOutputDevice> sink_;
  75
  76   // Audio data source from the browser process.
  77   WebRtcAudioRendererSource* source_;
  78
  79   // Buffers used for temporary storage during render callbacks.
  80   // Allocated during initialization.
  81   scoped_ptr<int16[]> buffer_;
  82
  83   // Protects access to |state_|, |source_| and |sink_|.
  84   base::Lock lock_;
  85
  86   // Ref count for the MediaPlayers which are playing audio.
  87   int play_ref_count_;
  88
  89   // Used to buffer data between the client and the output device in cases where
  90   // the client buffer size is not the same as the output device buffer size.
  91   scoped_ptr<media::AudioPullFifo> audio_fifo_;
  92
  93   // Contains the accumulated delay estimate which is provided to the WebRTC
  94   // AEC.
  95   int audio_delay_milliseconds_;
  96
  97   // Lengh of an audio frame in milliseconds.
  98   double frame_duration_milliseconds_;
  99
 100   double fifo_io_ratio_;
 101
 102   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 103 };
 104
 105 }  // namespace content
 106
 107 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_