Set audio format before adding pepper sink to audio track.

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 87 matching lines @@
 
   const std::string& device_id() const { return device_info_.device.id; }
   int session_id() const { return device_info_.session_id; }
 
   // Stops recording audio. This method will empty its track lists since
   // stopping the capturer will implicitly invalidate all its tracks.
   // This method is exposed to the public because the MediaStreamAudioSource can
   // call Stop()
   void Stop();
 
+  // Returns the output format.
+  // Called on the main render thread.
+  const media::AudioParameters& GetOutputFormat() const;
+
   // Used by the unittests to inject their own source to the capturer.
   void SetCapturerSourceForTesting(
       const scoped_refptr<media::AudioCapturerSource>& source,
       media::AudioParameters params);
 
  protected:
   friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>;
   ~WebRtcAudioCapturer() override;
 
  private:
@@ skipping 84 matching lines @@
   // guaranteed to exist as long as a WebRtcLocalAudioTrack is connected to this
   // WebRtcAudioCapturer.
   MediaStreamAudioSource* const audio_source_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_