Replace VideoReceiveStream::Config with new rtclog::StreamConfig in RtcEventLog.

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"
 
 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;
+  uint32_t remote_ssrc = 0;
+  uint32_t rtx_ssrc = 0;
+  std::string rsid;
+
+  bool remb = false;
+  std::vector<RtpExtension> rtp_extensions;
+
+  RtcpMode rtcp_mode = RtcpMode::kReducedSize;
+
+  struct Codec {
+    Codec(const std::string& payload_name,
+          int payload_type,
+          int rtx_payload_type)
+        : payload_name(payload_name),
+          payload_type(payload_type),
+          rtx_payload_type(rtx_payload_type) {}
+
+    std::string payload_name;
+    int payload_type;
+    int rtx_payload_type;
+  };
+  std::vector<Codec> codecs;
+};
+
 }  // namespace rtclog
 
 class Clock;
 class RtcEventLogImpl;
 struct AudioEncoderRuntimeConfig;
 
 enum class MediaType;
 enum class BandwidthUsage;
 
 enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket };
@@ skipping 38 matching lines @@
   }
 
   // Deprecated. Pass an explicit file size limit.
   bool StartLogging(rtc::PlatformFile platform_file) {
     return StartLogging(platform_file, 10000000);
   }
 
   // Stops logging to file and waits until the thread has finished.
   virtual void StopLogging() = 0;
 
-  // Logs configuration information for webrtc::VideoReceiveStream.
+  // Logs configuration information for video receive stream.
   virtual void LogVideoReceiveStreamConfig(
-      const webrtc::VideoReceiveStream::Config& config) = 0;
+      const rtclog::StreamConfig& config) = 0;
 
   // Logs configuration information for webrtc::VideoSendStream.
   virtual void LogVideoSendStreamConfig(
       const webrtc::VideoSendStream::Config& config) = 0;
 
   // Logs configuration information for webrtc::AudioReceiveStream.
   virtual void LogAudioReceiveStreamConfig(
       const webrtc::AudioReceiveStream::Config& config) = 0;
 
   // Logs configuration information for webrtc::AudioSendStream.
@@ skipping 65 matching lines @@
 class RtcEventLogNullImpl final : public RtcEventLog {
  public:
   bool StartLogging(const std::string& file_name,
                     int64_t max_size_bytes) override {
     return false;
   }
   bool StartLogging(rtc::PlatformFile platform_file,
                     int64_t max_size_bytes) override;
   void StopLogging() override {}
   void LogVideoReceiveStreamConfig(
-      const VideoReceiveStream::Config& config) override {}
+      const rtclog::StreamConfig& config) override {}
   void LogVideoSendStreamConfig(
       const VideoSendStream::Config& config) override {}
   void LogAudioReceiveStreamConfig(
       const AudioReceiveStream::Config& config) override {}
   void LogAudioSendStreamConfig(
       const AudioSendStream::Config& config) override {}
   void LogRtpHeader(PacketDirection direction,
                     MediaType media_type,
                     const uint8_t* header,
                     size_t packet_length) override {}
@@ skipping 19 matching lines @@
                               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_