Add generic interfaces for the sinks of the media stream audio track

content/renderer/media/webrtc_audio_device_impl.h

 // Copyright 2013 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 
 #include <string>
 #include <vector>
 
@@ skipping 184 matching lines @@
   // Set the format for the capture audio parameters.
   virtual void SetRenderFormat(const media::AudioParameters& params) = 0;
 
   // Callback to notify the client that the renderer is going away.
   virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) = 0;
 
  protected:
   virtual ~WebRtcAudioRendererSource() {}
 };
 
-class WebRtcAudioCapturerSink {
+class PeerConnectionAudioSink {
  public:
   // Callback to deliver the captured interleaved data.
   // |channels| contains a vector of WebRtc VoE channels.
   // |audio_data| is the pointer to the audio data.
   // |sample_rate| is the sample frequency of audio data.
   // |number_of_channels| is the number of channels reflecting the order of
   // surround sound channels.
   // |audio_delay_milliseconds| is recording delay value.
   // |current_volume| is current microphone volume, in range of |0, 255].
   // |need_audio_processing| indicates if the audio needs WebRtc AEC/NS/AGC
   // audio processing.
   // The return value is the new microphone volume, in the range of |0, 255].
   // When the volume does not need to be updated, it returns 0.
-  virtual int CaptureData(const std::vector<int>& channels,
-                          const int16* audio_data,
-                          int sample_rate,
-                          int number_of_channels,
-                          int number_of_frames,
-                          int audio_delay_milliseconds,
-                          int current_volume,
-                          bool need_audio_processing,
-                          bool key_pressed) = 0;
+  virtual int OnData(const int16* audio_data,
+                     int sample_rate,
+                     int number_of_channels,
+                     int number_of_frames,
+                     const std::vector<int>& channels,
+                     int audio_delay_milliseconds,
+                     int current_volume,
+                     bool need_audio_processing,
+                     bool key_pressed) = 0;
 
   // Set the format for the capture audio parameters.
   // This is called when the capture format has changed, and it must be called
   // on the same thread as calling CaptureData().
-  virtual void SetCaptureFormat(const media::AudioParameters& params) = 0;
+  virtual void OnSetFormat(const media::AudioParameters& params) = 0;
 
  protected:
-  virtual ~WebRtcAudioCapturerSink() {}
+ virtual ~PeerConnectionAudioSink() {}
 };
 
 // Note that this class inherits from webrtc::AudioDeviceModule but due to
 // the high number of non-implemented methods, we move the cruft over to the
 // WebRtcAudioDeviceNotImpl.
 class CONTENT_EXPORT WebRtcAudioDeviceImpl
-    : NON_EXPORTED_BASE(public WebRtcAudioDeviceNotImpl),
-      NON_EXPORTED_BASE(public WebRtcAudioCapturerSink),
+    : NON_EXPORTED_BASE(public PeerConnectionAudioSink),
+      NON_EXPORTED_BASE(public WebRtcAudioDeviceNotImpl),
       NON_EXPORTED_BASE(public WebRtcAudioRendererSource) {
  public:
   // The maximum volume value WebRtc uses.
   static const int kMaxVolumeLevel = 255;
 
   // Instances of this object are created on the main render thread.
   WebRtcAudioDeviceImpl();
 
   // webrtc::RefCountedModule implementation.
   // The creator must call AddRef() after construction and use Release()
@@ skipping 66 matching lines @@
   int output_sample_rate() const {
     return output_audio_parameters_.sample_rate();
   }
 
  private:
   typedef std::list<scoped_refptr<WebRtcAudioCapturer> > CapturerList;
 
   // Make destructor private to ensure that we can only be deleted by Release().
   virtual ~WebRtcAudioDeviceImpl();
 
-  // WebRtcAudioCapturerSink implementation.
+  // PeerConnectionAudioSink implementation.
 
   // Called on the AudioInputDevice worker thread.
-  virtual int CaptureData(const std::vector<int>& channels,
-                          const int16* audio_data,
-                          int sample_rate,
-                          int number_of_channels,
-                          int number_of_frames,
-                          int audio_delay_milliseconds,
-                          int current_volume,
-                          bool need_audio_processing,
-                          bool key_pressed) OVERRIDE;
+  virtual int OnData(const int16* audio_data,
+                     int sample_rate,
+                     int number_of_channels,
+                     int number_of_frames,
+                     const std::vector<int>& channels,
+                     int audio_delay_milliseconds,
+                     int current_volume,
+                     bool need_audio_processing,
+                     bool key_pressed) OVERRIDE;
 
-  // Called on the main render thread.
-  virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE;
+  // Called on the AudioInputDevice worker thread.
+  virtual void OnSetFormat(const media::AudioParameters& params) OVERRIDE;
 
   // WebRtcAudioRendererSource implementation.
 
   // Called on the AudioInputDevice worker thread.
   virtual void RenderData(uint8* audio_data,
                           int number_of_channels,
                           int number_of_frames,
                           int audio_delay_milliseconds) OVERRIDE;
 
   // Called on the main render thread.
@@ skipping 41 matching lines @@
   // Stores latest microphone volume received in a CaptureData() callback.
   // Range is [0, 255].
   uint32_t microphone_volume_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_