Set audio format before adding pepper sink to audio track.

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>
 
@@ skipping 49 matching lines @@
   // Overrides for MediaStreamTrack.
 
   void SetEnabled(bool enabled) override;
 
   // Stops the local audio track. Called on the main render thread and
   // should be called only once when audio track going away.
   void Stop() override;
 
   webrtc::AudioTrackInterface* GetAudioAdapter() override;
 
+  // Returns the output format of the capture source. May return an invalid
+  // AudioParameters if the format is not yet available.
+  // Called on the main render thread.
+  media::AudioParameters GetOutputFormat() const;
+
   // Method called by the capturer to deliver the capture data.
   // Called on the capture audio thread.
   void Capture(const media::AudioBus& audio_bus,
                base::TimeTicks estimated_capture_time,
                bool force_report_nonzero_energy);
 
   // Method called by the capturer to set the audio parameters used by source
   // of the capture data..
   // Called on the capture audio thread.
   void OnSetFormat(const media::AudioParameters& params);
@@ skipping 41 matching lines @@
   // Used to calculate the signal level that shows in the UI.
   // Accessed on only the audio thread.
   scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_