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

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  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 1038 matching lines @@
   if (is_dumping_aec_) {
     // Stop dumping AEC when we are dumping.
     if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() !=
         webrtc::AudioProcessing::kNoError) {
       LOG_RTCERR0(StopDebugRecording);
     }
     is_dumping_aec_ = false;
   }
 }
 
-bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) {
+bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file,
+                                         int64_t max_size_bytes) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog();
   if (event_log) {
-    return event_log->StartLogging(file);
+    return event_log->StartLogging(file, max_size_bytes);
   }
   LOG_RTCERR0(StartRtcEventLog);
   return false;
 }
 
 void WebRtcVoiceEngine::StopRtcEventLog() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog();
   if (event_log) {
     event_log->StopLogging();
@@ skipping 1485 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE