Removed RtcEventLog deps to call:call_interfaces.

webrtc/logging/rtc_event_log/rtc_event_log.h

 /*
  *  Copyright (c) 2015 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.
  */
 
 #ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
 #define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/platform_file.h"
-#include "webrtc/call/audio_receive_stream.h"
-#include "webrtc/call/audio_send_stream.h"
-#include "webrtc/video_receive_stream.h"
-#include "webrtc/video_send_stream.h"
+#include "webrtc/config.h"
 
 namespace webrtc {
 
 // Forward declaration of storage class that is automatically generated from
 // the protobuf file.
 namespace rtclog {
 class EventStream;
 
 struct StreamConfig {
   uint32_t local_ssrc = 0;
@@ skipping 89 matching lines @@
   // Logs configuration information for an audio receive stream.
   virtual void LogAudioReceiveStreamConfig(
       const rtclog::StreamConfig& config) = 0;
 
   // Logs configuration information for an audio send stream.
   virtual void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) = 0;
 
   // Logs the header of an incoming or outgoing RTP packet. packet_length
   // is the total length of the packet, including both header and payload.
   virtual void LogRtpHeader(PacketDirection direction,
-                            MediaType media_type,
                             const uint8_t* header,
                             size_t packet_length) = 0;
 
   // Same as above but used on the sender side to log packets that are part of
   // a probe cluster.
   virtual void LogRtpHeader(PacketDirection direction,
-                            MediaType media_type,
                             const uint8_t* header,
                             size_t packet_length,
                             int probe_cluster_id) = 0;
 
   // Logs an incoming or outgoing RTCP packet.
   virtual void LogRtcpPacket(PacketDirection direction,
-                             MediaType media_type,
                              const uint8_t* packet,
                              size_t length) = 0;
 
   // Logs an audio playout event.
   virtual void LogAudioPlayout(uint32_t ssrc) = 0;
 
   // Logs a bitrate update from the bandwidth estimator based on packet loss.
   virtual void LogLossBasedBweUpdate(int32_t bitrate_bps,
                                      uint8_t fraction_loss,
                                      int32_t total_packets) = 0;
@@ skipping 40 matching lines @@
   bool StartLogging(rtc::PlatformFile platform_file,
                     int64_t max_size_bytes) override;
   void StopLogging() override {}
   void LogVideoReceiveStreamConfig(
       const rtclog::StreamConfig& config) override {}
   void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override {}
   void LogAudioReceiveStreamConfig(
       const rtclog::StreamConfig& config) override {}
   void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override {}
   void LogRtpHeader(PacketDirection direction,
-                    MediaType media_type,
                     const uint8_t* header,
                     size_t packet_length) override {}
   void LogRtpHeader(PacketDirection direction,
-                    MediaType media_type,
                     const uint8_t* header,
                     size_t packet_length,
                     int probe_cluster_id) override {}
   void LogRtcpPacket(PacketDirection direction,
-                     MediaType media_type,
                      const uint8_t* packet,
                      size_t length) override {}
   void LogAudioPlayout(uint32_t ssrc) override {}
   void LogLossBasedBweUpdate(int32_t bitrate_bps,
                              uint8_t fraction_loss,
                              int32_t total_packets) override {}
   void LogDelayBasedBweUpdate(int32_t bitrate_bps,
                               BandwidthUsage detector_state) override {}
   void LogAudioNetworkAdaptation(
       const AudioEncoderRuntimeConfig& config) override {}
   void LogProbeClusterCreated(int id,
                               int bitrate_bps,
                               int min_probes,
                               int min_bytes) override{};
   void LogProbeResultSuccess(int id, int bitrate_bps) override{};
   void LogProbeResultFailure(int id,
                              ProbeFailureReason failure_reason) override{};
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_