Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 
 #include <stdlib.h>  // srand
 #include <algorithm>
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call.h"
-#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/call/rtc_event_log_proxy.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/time_util.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
@@ skipping 76 matching lines @@
     bool audio,
     Clock* clock,
     Transport* transport,
     RtpAudioFeedback* audio_feedback,
     RtpPacketSender* paced_sender,
     TransportSequenceNumberAllocator* sequence_number_allocator,
     TransportFeedbackObserver* transport_feedback_observer,
     BitrateStatisticsObserver* bitrate_callback,
     FrameCountObserver* frame_count_observer,
     SendSideDelayObserver* send_side_delay_observer,
-    RtcEventLog* event_log)
+    RtcEventLogProxy* event_log)
     : clock_(clock),
       // TODO(holmer): Remove this conversion when we remove the use of
       // TickTime.
       clock_delta_ms_(clock_->TimeInMilliseconds() -
                       TickTime::MillisecondTimestamp()),
       random_(clock_->TimeInMicroseconds()),
       bitrates_(bitrate_callback),
       total_bitrate_sent_(clock, bitrates_.total_bitrate_observer()),
       audio_configured_(audio),
       audio_(audio ? new RTPSenderAudio(clock, this, audio_feedback) : nullptr),
@@ skipping 1782 matching lines @@
   rtc::CritScope lock(&send_critsect_);
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc