Adds Analog Gain Control (AGC) to the WebRTC client.

content/browser/renderer_host/media/audio_input_renderer_host.h

Index: content/browser/renderer_host/media/audio_input_renderer_host.h
diff --git a/content/browser/renderer_host/media/audio_input_renderer_host.h b/content/browser/renderer_host/media/audio_input_renderer_host.h
index a7645514a2ba9e1dfcf501ddc102eb323a70c07e..a0c9913ce460363aa2d538d2b9359689d9504842 100644
--- a/content/browser/renderer_host/media/audio_input_renderer_host.h
+++ b/content/browser/renderer_host/media/audio_input_renderer_host.h
@@ -140,7 +140,8 @@ class CONTENT_EXPORT AudioInputRendererHost
   // required properties.
   void OnCreateStream(int stream_id,
                       const AudioParameters& params,
-                      const std::string& device_id);
+                      const std::string& device_id,
+                      bool automatic_gain_control);
 
   // Record the audio input stream referenced by |stream_id|.
   void OnRecordStream(int stream_id);
@@ -154,6 +155,10 @@ class CONTENT_EXPORT AudioInputRendererHost
   // Get the volume of the audio stream referenced by |stream_id|.
   void OnGetVolume(int stream_id);
 
+  // Set the automatic gain control state for the audio stream referenced
+  // by |stream_id|.
+  void OnSetAutomaticGainControl(int stream_id, bool enabled);
+
   // Complete the process of creating an audio input stream. This will set up
   // the shared memory or shared socket in low latency mode.
   void DoCompleteCreation(media::AudioInputController* controller);