Enable platform echo cancellation through the AudioRecord path.

content/renderer/media/webrtc_audio_capturer.h

 // Copyright (c) 2012 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_CAPTURER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
 
 #include <list>
 #include <string>
 
@@ skipping 37 matching lines @@
   // consuming audio for capture.  |session_id| is passed to the browser to
   // decide which device to use.  |device_id| is used to identify which device
   // the capturer is created for.  Called on the main render thread.
   bool Initialize(int render_view_id,
                   media::ChannelLayout channel_layout,
                   int sample_rate,
                   int buffer_size,
                   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);

[Ami GONE FROM CHROMIUM, 2013/12/06 19:22:53]

I'm going to pretty sad if each platform effect means touching all the files
that this CL touches, and adding new bool use_platform_fooblart params each
time.  Time for a
struct PlatformEffects {
  bool echo_canceller;
};
and pass that around instead of the bool in all the new call-sites?

[ajm, 2013/12/06 19:31:24]

Agreed, I was thinking the same thing after reading Tommi's comments.

 
   // Add a audio track to the sinks of the capturer.
   // WebRtcAudioDeviceImpl calls this method on the main render thread but
   // other clients may call it from other threads. The current implementation
   // does not support multi-thread calling.
   // The first AddTrack will implicitly trigger the Start() of this object.
   // Called on the main render thread or libjingle working thread.
   void AddTrack(WebRtcLocalAudioTrack* track);
 
   // Remove a audio track from the sinks of the capturer.
   // If the track has been added to the capturer, it  must call RemoveTrack()
   // before it goes away.
   // Called on the main render thread or libjingle working thread.
   void RemoveTrack(WebRtcLocalAudioTrack* track);
 
   // SetCapturerSource() is called if the client on the source side desires to
   // provide their own captured audio data. Client is responsible for calling
   // Start() on its own source to have the ball rolling.
   // Called on the main render thread.
   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
   // performance for peer connection.
   void EnablePeerConnectionMode();
 
   // Volume APIs used by WebRtcAudioDeviceImpl.
   // Called on the AudioInputDevice audio thread.
   void SetVolume(int volume);
   int Volume() const;
@@ skipping 42 matching lines @@
   // AudioCapturerSource::CaptureCallback implementation.
   // Called on the AudioInputDevice audio thread.
   virtual void Capture(media::AudioBus* audio_source,
                        int audio_delay_milliseconds,
                        double volume,
                        bool key_pressed) OVERRIDE;
   virtual void OnCaptureError() OVERRIDE;
 
   // 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
   // thread. It should NOT be called under |lock_|.
   void Start();
 
   // Helper function to get the buffer size based on |peer_connection_mode_|
   // and sample rate;
   int GetBufferSize(int sample_rate) const;
 
@@ skipping 45 matching lines @@
   // Cache value for the audio processing params.
   base::TimeDelta audio_delay_;
   bool key_pressed_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_