Wire up AGC to the MediaStreamAudioProcessor

content/renderer/media/media_stream_audio_processor.h

Index: content/renderer/media/media_stream_audio_processor.h
diff --git a/content/renderer/media/media_stream_audio_processor.h b/content/renderer/media/media_stream_audio_processor.h
index be5b8b0409ea82d73e22cab068a0dbe8ce68282e..53adea23c1baecc3a4d48c123476d8cbb5f32ea6 100644
--- a/content/renderer/media/media_stream_audio_processor.h
+++ b/content/renderer/media/media_stream_audio_processor.h
@@ -63,6 +63,8 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
   // the post-processed data if the method is returning a true. The lifetime
   // of the data represeted by |out| is guaranteed to outlive the method call.
   // That also says *|out| won't change until this method is called again.
+  // |new_volume| contains the new microphone volume from the AGC, the value
+  // will be 0 if the microphone volume is not adjusted.

[tommi (sloooow) - chröme, 2014/01/21 15:06:20]

nit:
// |new_volume| receives the new microphone volume from the AGC.
// The value will be 0 if the microphone volume should not be
// adjusted.

do you want to add to that a note about the range?

[no longer working on chromium, 2014/01/23 12:46:08]

Done.

   // Returns true if the internal FIFO has at least 10 ms data for processing,
   // otherwise false.
   // |capture_delay|, |volume| and |key_pressed| will be passed to
@@ -71,6 +73,7 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
   bool ProcessAndConsumeData(base::TimeDelta capture_delay,
                              int volume,
                              bool key_pressed,
+                             int* new_volume,
                              int16** out);
 
 
@@ -88,6 +91,8 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
   virtual ~MediaStreamAudioProcessor();
 
  private:
+  friend class MediaStreamAudioProcessorTest;
+
   class MediaStreamAudioConverter;
 
   // Helper to initialize the WebRtc AudioProcessing.
@@ -103,10 +108,12 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
                                          int frames_per_buffer);
 
   // Called by ProcessAndConsumeData().
-  void ProcessData(webrtc::AudioFrame* audio_frame,
-                   base::TimeDelta capture_delay,
-                   int volume,
-                   bool key_pressed);
+  // Returns the new microphone volume in the range of |0, 255].
+  // When the volume does not need to be updated, it returns 0.
+  int ProcessData(webrtc::AudioFrame* audio_frame,
+                  base::TimeDelta capture_delay,
+                  int volume,
+                  bool key_pressed);
 
   // Called when the processor is going away.
   void StopAudioProcessing();
@@ -143,6 +150,9 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
 
   // Used to DCHECK that PushRenderData() is called on the render audio thread.
   base::ThreadChecker render_thread_checker_;
+
+  // Flag to enable the stereo channels swapping.
+  bool stereo_channels_swapping_;

[tommi (sloooow) - chröme, 2014/01/21 15:06:20]

nit: audio_mirroring_

[no longer working on chromium, 2014/01/23 12:46:08]

Done.

 };
 
 }  // namespace content