Support multiple files for AEC dump.

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 99 matching lines @@
   // This function is triggered by provideInput() on the WebAudio audio thread,
   // TODO(xians): Remove after moving APM from WebRtc to Chrome.
   void GetAudioProcessingParams(base::TimeDelta* delay, int* volume,
                                 bool* key_pressed);
 
   // Used by the unittests to inject their own source to the capturer.
   void SetCapturerSourceForTesting(
       const scoped_refptr<media::AudioCapturerSource>& source,
       media::AudioParameters params);
 
-  void StartAecDump(base::File aec_dump_file);
-  void StopAecDump();
-
  protected:
   friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>;
   virtual ~WebRtcAudioCapturer();
 
  private:
   class TrackOwner;
   typedef TaggedList<TrackOwner> TrackList;
 
   WebRtcAudioCapturer(int render_view_id,
                       const StreamDeviceInfo& device_info,
@@ skipping 92 matching lines @@
 
   // Records when the last time audio power level is logged.
   base::TimeTicks last_audio_level_log_time_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_