Removed RtcEventLog deps to call:call_interfaces.

webrtc/voice_engine/channel.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.
  */
 
@@ skipping 83 matching lines @@
 
   void LogAudioSendStreamConfig(
       const webrtc::rtclog::StreamConfig& config) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
       event_log_->LogAudioSendStreamConfig(config);
     }
   }
 
   void LogRtpHeader(webrtc::PacketDirection direction,
-                    webrtc::MediaType media_type,
                     const uint8_t* header,
                     size_t packet_length) override {
-    LogRtpHeader(direction, media_type, header, packet_length,
-                 PacedPacketInfo::kNotAProbe);
+    LogRtpHeader(direction, header, packet_length, PacedPacketInfo::kNotAProbe);
   }
 
   void LogRtpHeader(webrtc::PacketDirection direction,
-                    webrtc::MediaType media_type,
                     const uint8_t* header,
                     size_t packet_length,
                     int probe_cluster_id) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogRtpHeader(direction, media_type, header, packet_length,
+      event_log_->LogRtpHeader(direction, header, packet_length,
                                probe_cluster_id);
     }
   }
 
   void LogRtcpPacket(webrtc::PacketDirection direction,
-                     webrtc::MediaType media_type,
                      const uint8_t* packet,
                      size_t length) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogRtcpPacket(direction, media_type, packet, length);
+      event_log_->LogRtcpPacket(direction, packet, length);
     }
   }
 
   void LogAudioPlayout(uint32_t ssrc) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
       event_log_->LogAudioPlayout(ssrc);
     }
   }
 
@@ skipping 2993 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc