Enable platform echo cancellation through the AudioRecord path.

content/renderer/media/webrtc_audio_capturer.h

Index: content/renderer/media/webrtc_audio_capturer.h
diff --git a/content/renderer/media/webrtc_audio_capturer.h b/content/renderer/media/webrtc_audio_capturer.h
index 580e9fdcf6ed2ea905ea36682fae0502d6eaecb7..663f63d3e5740e1e21fadf887c80fa1af526b54a 100644
--- a/content/renderer/media/webrtc_audio_capturer.h
+++ b/content/renderer/media/webrtc_audio_capturer.h
@@ -55,7 +55,8 @@ class CONTENT_EXPORT WebRtcAudioCapturer
                   int session_id,
                   const std::string& device_id,
                   int paired_output_sample_rate,
-                  int paired_output_frames_per_buffer);
+                  int paired_output_frames_per_buffer,
+                  bool use_platform_aec);
 
   // Add a audio track to the sinks of the capturer.
   // WebRtcAudioDeviceImpl calls this method on the main render thread but
@@ -78,7 +79,8 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   void SetCapturerSource(
       const scoped_refptr<media::AudioCapturerSource>& source,
       media::ChannelLayout channel_layout,
-      float sample_rate);
+      float sample_rate,
+      bool use_platform_aec);
 
   // Called when a stream is connecting to a peer connection. This will set
   // up the native buffer size for the stream in order to optimize the
@@ -141,7 +143,8 @@ class CONTENT_EXPORT WebRtcAudioCapturer
 
   // Reconfigures the capturer with a new capture parameters.
   // Must be called without holding the lock.
-  void Reconfigure(int sample_rate, media::ChannelLayout channel_layout);
+  void Reconfigure(int sample_rate, media::ChannelLayout channel_layout,
+                   bool use_platform_aec);
 
   // Starts recording audio.
   // Triggered by AddSink() on the main render thread or a Libjingle working