Move the call to enable the WebRTC event log from PeerConnectionFactory to PeerConnection.

content/public/browser/render_process_host.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_PUBLIC_BROWSER_RENDER_PROCESS_HOST_H_
 #define CONTENT_PUBLIC_BROWSER_RENDER_PROCESS_HOST_H_
 
 #include <stddef.h>
 #include <stdint.h>
 
@@ skipping 201 matching lines @@
 
   // Checks that the given renderer can request |url|, if not it sets it to
   // about:blank.
   // |empty_allowed| must be set to false for navigations for security reasons.
   virtual void FilterURL(bool empty_allowed, GURL* url) = 0;
 
 #if defined(ENABLE_WEBRTC)
   virtual void EnableAudioDebugRecordings(const base::FilePath& file) = 0;
   virtual void DisableAudioDebugRecordings() = 0;
 
-  virtual void EnableEventLogRecordings(const base::FilePath& file) = 0;
-  virtual void DisableEventLogRecordings() = 0;
-
   // When set, |callback| receives log messages regarding, for example, media
   // devices (webcams, mics, etc) that were initially requested in the render
   // process associated with this RenderProcessHost.
   virtual void SetWebRtcLogMessageCallback(
       base::Callback<void(const std::string&)> callback) = 0;
   virtual void ClearWebRtcLogMessageCallback() = 0;
 
   typedef base::Callback<void(std::unique_ptr<uint8_t[]> packet_header,
                               size_t header_length,
                               size_t packet_length,
                               bool incoming)>
       WebRtcRtpPacketCallback;
 
   typedef base::Callback<void(bool incoming, bool outgoing)>
       WebRtcStopRtpDumpCallback;
 
   // Starts passing RTP packets to |packet_callback| and returns the callback
   // used to stop dumping.
   virtual WebRtcStopRtpDumpCallback StartRtpDump(
       bool incoming,
       bool outgoing,
       const WebRtcRtpPacketCallback& packet_callback) = 0;
+
+  // Starts a WebRTC event log for each peerconnection on the render process.
+  // A base file_path can be supplied, which will be extended to include several
+  // identifiers to ensure uniqueness. If a recording was already in progress,
+  // this call will return false and have no other effect.
+  virtual bool StartWebRTCEventLog(const base::FilePath& file_path) = 0;

[ncarter (slow), 2016/05/20 18:27:21]

This looks basically like a rename of the old functions (I like the new names
and documentation much better). I'd hoist these up to line 221, where
EnableEventLogRecordings() used to be.

[Ivo-OOO until feb 6, 2016/05/25 14:57:00]

Good idea, done.

+
+  // Stops recording a WebRTC event log for each peerconnection on the render
+  // process. If no recording was in progress, this call will return false.
+  virtual bool StopWebRTCEventLog() = 0;
 #endif
 
   // Tells the ResourceDispatcherHost to resume a deferred navigation without
   // transferring it to a new renderer process.
   virtual void ResumeDeferredNavigation(const GlobalRequestID& request_id) = 0;
 
   // Notifies the renderer that the timezone configuration of the system might
   // have changed.
   virtual void NotifyTimezoneChange(const std::string& zone_id) = 0;
 
@@ skipping 98 matching lines @@
   static void SetMaxRendererProcessCount(size_t count);
 
   // Returns the current maximum number of renderer process hosts kept by the
   // content module.
   static size_t GetMaxRendererProcessCount();
 };
 
 }  // namespace content.
 
 #endif  // CONTENT_PUBLIC_BROWSER_RENDER_PROCESS_HOST_H_