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      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/media_stream_audio_renderer.h"
     12 #include "content/renderer/media/webrtc_audio_device_impl.h"
     13 #include "media/base/audio_decoder.h"
     14 #include "media/base/audio_pull_fifo.h"
     15 #include "media/base/audio_renderer_sink.h"
     16 #include "media/base/channel_layout.h"
     17 
     18 namespace media {
     19 class AudioOutputDevice;
     20 }
     21 
     22 namespace content {
     23 
     24 class WebRtcAudioRendererSource;
     25 
     26 // This renderer handles calls from the pipeline and WebRtc ADM. It is used
     27 // for connecting WebRtc MediaStream with the audio pipeline.
     28 class CONTENT_EXPORT WebRtcAudioRenderer
     29     : NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
     30       NON_EXPORTED_BASE(public MediaStreamAudioRenderer) {
     31  public:
     32   WebRtcAudioRenderer(int source_render_view_id,
     33                       int session_id,
     34                       int sample_rate,
     35                       int frames_per_buffer);
     36 
     37   // Initialize function called by clients like WebRtcAudioDeviceImpl.
     38   // Stop() has to be called before |source| is deleted.
     39   bool Initialize(WebRtcAudioRendererSource* source);
     40 
     41   // When sharing a single instance of WebRtcAudioRenderer between multiple
     42   // users (e.g. WebMediaPlayerMS), call this method to create a proxy object
     43   // that maintains the Play and Stop states per caller.
     44   // The wrapper ensures that Play() won't be called when the caller's state
     45   // is "playing", Pause() won't be called when the state already is "paused"
     46   // etc and similarly maintains the same state for Stop().
     47   // When Stop() is called or when the proxy goes out of scope, the proxy
     48   // will ensure that Pause() is called followed by a call to Stop(), which
     49   // is the usage pattern that WebRtcAudioRenderer requires.
     50   scoped_refptr<MediaStreamAudioRenderer> CreateSharedAudioRendererProxy();
     51 
     52   // Used to DCHECK on the expected state.
     53   bool IsStarted() const;
     54 
     55  private:
     56   // MediaStreamAudioRenderer implementation.  This is private since we want
     57   // callers to use proxy objects.
     58   // TODO(tommi): Make the MediaStreamAudioRenderer implementation a pimpl?
     59   virtual void Start() OVERRIDE;
     60   virtual void Play() OVERRIDE;
     61   virtual void Pause() OVERRIDE;
     62   virtual void Stop() OVERRIDE;
     63   virtual void SetVolume(float volume) OVERRIDE;
     64   virtual base::TimeDelta GetCurrentRenderTime() const OVERRIDE;
     65   virtual bool IsLocalRenderer() const OVERRIDE;
     66 
     67  protected:
     68   virtual ~WebRtcAudioRenderer();
     69 
     70  private:
     71   enum State {
     72     UNINITIALIZED,
     73     PLAYING,
     74     PAUSED,
     75   };
     76 
     77   // Used to DCHECK that we are called on the correct thread.
     78   base::ThreadChecker thread_checker_;
     79 
     80   // Flag to keep track the state of the renderer.
     81   State state_;
     82 
     83   // media::AudioRendererSink::RenderCallback implementation.
     84   // These two methods are called on the AudioOutputDevice worker thread.
     85   virtual int Render(media::AudioBus* audio_bus,
     86                      int audio_delay_milliseconds) OVERRIDE;
     87   virtual void OnRenderError() OVERRIDE;
     88 
     89   // Called by AudioPullFifo when more data is necessary.
     90   // This method is called on the AudioOutputDevice worker thread.
     91   void SourceCallback(int fifo_frame_delay, media::AudioBus* audio_bus);
     92 
     93   // The render view in which the audio is rendered into |sink_|.
     94   const int source_render_view_id_;
     95   const int session_id_;
     96 
     97   // The sink (destination) for rendered audio.
     98   scoped_refptr<media::AudioOutputDevice> sink_;
     99 
    100   // Audio data source from the browser process.
    101   WebRtcAudioRendererSource* source_;
    102 
    103   // Buffers used for temporary storage during render callbacks.
    104   // Allocated during initialization.
    105   scoped_ptr<int16[]> buffer_;
    106 
    107   // Protects access to |state_|, |source_| and |sink_|.
    108   base::Lock lock_;
    109 
    110   // Ref count for the MediaPlayers which are playing audio.
    111   int play_ref_count_;
    112 
    113   // Ref count for the MediaPlayers which have called Start() but not Stop().
    114   int start_ref_count_;
    115 
    116   // Used to buffer data between the client and the output device in cases where
    117   // the client buffer size is not the same as the output device buffer size.
    118   scoped_ptr<media::AudioPullFifo> audio_fifo_;
    119 
    120   // Contains the accumulated delay estimate which is provided to the WebRTC
    121   // AEC.
    122   int audio_delay_milliseconds_;
    123 
    124   // Delay due to the FIFO in milliseconds.
    125   int fifo_delay_milliseconds_;
    126 
    127   // The preferred sample rate and buffer sizes provided via the ctor.
    128   const int sample_rate_;
    129   const int frames_per_buffer_;
    130 
    131   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
    132 };
    133 
    134 }  // namespace content
    135 
    136 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
    137