Add a parameter to set a maximum filesize when starting an RTC event log on the PeerConnectionFacto…

webrtc/media/engine/webrtcvoiceengine.h

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 67 matching lines @@
 
   // Starts AEC dump using an existing file. A maximum file size in bytes can be
   // specified. When the maximum file size is reached, logging is stopped and
   // the file is closed. If max_size_bytes is set to <= 0, no limit will be
   // used.
   bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
 
   // Stops AEC dump.
   void StopAecDump();
 
-  // Starts recording an RtcEventLog using an existing file until 10 minutes
-  // pass or the StopRtcEventLog function is called.
-  bool StartRtcEventLog(rtc::PlatformFile file);
+  // Starts recording an RtcEventLog using an existing file until the log file
+  // reaches the maximum filesize or the StopRtcEventLog function is called.
+  // If the value of max_size_bytes is <= 0, no limit is used.
+  bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes);
 
   // Stops recording the RtcEventLog.
   void StopRtcEventLog();
 
  private:
   // Every option that is "set" will be applied. Every option not "set" will be
   // ignored. This allows us to selectively turn on and off different options
   // easily at any time.
   bool ApplyOptions(const AudioOptions& options);
   void SetDefaultDevices();
@@ skipping 185 matching lines @@
     int cng_payload_type = -1;
     int cng_plfreq = -1;
     webrtc::CodecInst codec_inst;
   } send_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_