Feed audio constraints over to WebRtcLocalAudioTrack

content/renderer/media/webrtc_local_audio_track.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_LOCAL_AUDIO_TRACK_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
 
 #include <list>
 #include <string>
 
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
+#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreamtrack.h"
 #include "third_party/libjingle/source/talk/media/base/audiorenderer.h"
 
 namespace cricket {
 class AudioRenderer;
 }
 
 namespace content {
 
 class WebRtcAudioCapturer;
 class WebRtcAudioCapturerSinkOwner;
 
 // A WebRtcLocalAudioTrack instance contains the implementations of
 // MediaStreamTrack and WebRtcAudioCapturerSink.
 // When an instance is created, it will register itself as a track to the
 // WebRtcAudioCapturer to get the captured data, and forward the data to
 // its |sinks_|. The data flow can be stopped by disabling the audio track.
 class CONTENT_EXPORT WebRtcLocalAudioTrack
     : NON_EXPORTED_BASE(public cricket::AudioRenderer),
       NON_EXPORTED_BASE(
           public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) {
  public:
   static scoped_refptr<WebRtcLocalAudioTrack> Create(
       const std::string& id,
       const scoped_refptr<WebRtcAudioCapturer>& capturer,
-      webrtc::AudioSourceInterface* stream_source);
+      webrtc::AudioSourceInterface* stream_source,
+      const webrtc::MediaConstraintsInterface* constraints);
 
   // Add a sink to the track. This function will trigger a SetCaptureFormat()
   // call on the |sink|.
   // Called on the main render thread.
   void AddSink(WebRtcAudioCapturerSink* sink);
 
   // Remove a sink from the track.
   // Called on the main render thread.
   void RemoveSink(WebRtcAudioCapturerSink* sink);
 
@@ skipping 14 matching lines @@
                    bool key_pressed);
 
   // Method called by the capturer to set the audio parameters used by source
   // of the capture data..
   // Can be called on different user threads.
   void SetCaptureFormat(const media::AudioParameters& params);
 
  protected:
   WebRtcLocalAudioTrack(const std::string& label,
                         const scoped_refptr<WebRtcAudioCapturer>& capturer,
-                        webrtc::AudioSourceInterface* track_source);
+                        webrtc::AudioSourceInterface* track_source,
+                        const webrtc::MediaConstraintsInterface* constraints);
   virtual ~WebRtcLocalAudioTrack();
 
  private:
   typedef std::list<scoped_refptr<WebRtcAudioCapturerSinkOwner> > SinkList;
 
   // cricket::AudioCapturer implementation.
   virtual void AddChannel(int channel_id) OVERRIDE;
   virtual void RemoveChannel(int channel_id) OVERRIDE;
 
   // webrtc::AudioTrackInterface implementation.
@@ skipping 27 matching lines @@
   std::vector<int> voe_channels_;
 
   bool need_audio_processing_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_