Remove the RTP recording related code from webrtc-internals.

content/browser/media/webrtc_internals.h

 // Copyright (c) 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_BROWSER_MEDIA_WEBRTC_INTERNALS_H_
 #define CONTENT_BROWSER_MEDIA_WEBRTC_INTERNALS_H_
 
 #include "base/gtest_prod_util.h"
 #include "base/memory/singleton.h"
 #include "base/observer_list.h"
@@ skipping 67 matching lines @@
                       const std::string& audio_constraints,
                       const std::string& video_constraints);
 
   // Methods for adding or removing WebRTCInternalsUIObserver.
   void AddObserver(WebRTCInternalsUIObserver *observer);
   void RemoveObserver(WebRTCInternalsUIObserver *observer);
 
   // Sends all update data to |observer|.
   void UpdateObserver(WebRTCInternalsUIObserver* observer);
 
-  // Tells the renderer processes to start or stop recording RTP packets.
-  void StartRtpRecording();
-  void StopRtpRecording();
-
   // Enables or disables AEC dump (diagnostic echo canceller recording).
   void EnableAecDump(content::WebContents* web_contents);
   void DisableAecDump();
 
   bool aec_dump_enabled() {
     return aec_dump_enabled_;
   }
 
   base::FilePath aec_dump_file_path() {
     return aec_dump_file_path_;
@@ skipping 22 matching lines @@
                        const NotificationDetails& details) OVERRIDE;
 
   // ui::SelectFileDialog::Listener implementation.
   virtual void FileSelected(const base::FilePath& path,
                             int index,
                             void* unused_params) OVERRIDE;
 
   // Called when a renderer exits (including crashes).
   void OnRendererExit(int render_process_id);
 
-  void SendRtpRecordingUpdate();
-
   ObserverList<WebRTCInternalsUIObserver> observers_;
 
   // |peer_connection_data_| is a list containing all the PeerConnection
   // updates.
   // Each item of the list represents the data for one PeerConnection, which
   // contains these fields:
   // "rid" -- the renderer id.
   // "pid" -- OS process id of the renderer that creates the PeerConnection.
   // "lid" -- local Id assigned to the PeerConnection.
   // "url" -- url of the web page that created the PeerConnection.
   // "servers" and "constraints" -- server configuration and media constraints
   // used to initialize the PeerConnection respectively.
   // "log" -- a ListValue contains all the updates for the PeerConnection. Each
   // list item is a DictionaryValue containing "type" and "value", both of which
   // are strings.
   base::ListValue peer_connection_data_;
 
   // A list of getUserMedia requests. Each item is a DictionaryValue that
   // contains these fields:
   // "rid" -- the renderer id.
   // "pid" -- proceddId of the renderer.
   // "origin" -- the security origin of the request.
   // "audio" -- the serialized audio constraints if audio is requested.
   // "video" -- the serialized video constraints if video is requested.
   base::ListValue get_user_media_requests_;
 
   NotificationRegistrar registrar_;
 
-  bool is_recording_rtp_;
-
   // For managing select file dialog.
   scoped_refptr<ui::SelectFileDialog> select_file_dialog_;
 
   // AEC dump (diagnostic echo canceller recording) state.
   bool aec_dump_enabled_;
   base::FilePath aec_dump_file_path_;
 };
 
 }  // namespace content
 
 #endif  // CONTENT_BROWSER_MEDIA_WEBRTC_INTERNALS_H_