Ignore this CL - here as a baseline only (originally Bjorn's CL)

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.
  */
 
 #include "webrtc/voice_engine/channel.h"
 
 #include <algorithm>
+#include <map>
+#include <string>
 #include <utility>
+#include <vector>
 
 #include "webrtc/api/array_view.h"
 #include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/pacing/packet_router.h"
@@ skipping 68 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,
-                    const uint8_t* header,
-                    size_t packet_length) override {
-    LogRtpHeader(direction, header, packet_length, PacedPacketInfo::kNotAProbe);
-  }
-
-  void LogRtpHeader(webrtc::PacketDirection direction,
-                    const uint8_t* header,
-                    size_t packet_length,
-                    int probe_cluster_id) override {
+  void LogIncomingRtpHeader(const RtpPacketReceived& packet) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogRtpHeader(direction, header, packet_length,
-                               probe_cluster_id);
+      event_log_->LogIncomingRtpHeader(packet);
     }
   }
 
-  void LogRtcpPacket(webrtc::PacketDirection direction,
-                     const uint8_t* packet,
-                     size_t length) override {
+  void LogOutgoingRtpHeader(const RtpPacketToSend& packet,
+                            int probe_cluster_id) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogRtcpPacket(direction, packet, length);
+      event_log_->LogOutgoingRtpHeader(packet, probe_cluster_id);
     }
   }
 
+  void LogIncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override {
+    rtc::CritScope lock(&crit_);
+    if (event_log_) {
+      event_log_->LogIncomingRtcpPacket(packet);
+    }
+  }
+
+  void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override {
+    rtc::CritScope lock(&crit_);
+    if (event_log_) {
+      event_log_->LogOutgoingRtcpPacket(packet);
+    }
+  }
+
   void LogAudioPlayout(uint32_t ssrc) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
       event_log_->LogAudioPlayout(ssrc);
     }
   }
 
   void LogLossBasedBweUpdate(int32_t bitrate_bps,
                              uint8_t fraction_loss,
                              int32_t total_packets) override {
@@ 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