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

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 70 matching lines @@
 }
 
 bool UseSendSideBwe(const AudioReceiveStream::Config& config) {
   return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc);
 }
 
 bool UseSendSideBwe(const FlexfecReceiveStream::Config& config) {
   return UseSendSideBwe(config.rtp_header_extensions, config.transport_cc);
 }
 
+rtclog::StreamConfig CreateRtcLogStreamConfig(
+    const VideoReceiveStream::Config& config) {
+  rtclog::StreamConfig rtclog_config;
+  rtclog_config.remote_ssrc = config.rtp.remote_ssrc;
+  rtclog_config.local_ssrc = config.rtp.local_ssrc;
+  rtclog_config.rtx_ssrc = config.rtp.rtx_ssrc;
+  rtclog_config.rtcp_mode = config.rtp.rtcp_mode;
+  rtclog_config.remb = config.rtp.remb;
+  rtclog_config.rtp_extensions = config.rtp.extensions;
+
+  for (const auto& d : config.decoders) {
+    auto search = config.rtp.rtx_payload_types.find(d.payload_type);
+    rtclog_config.codecs.emplace_back(
+        d.payload_name, d.payload_type,
+        search != config.rtp.rtx_payload_types.end() ? search->second : 0);
+  }
+  return rtclog_config;
+}
+
 }  // namespace
 
 namespace internal {
 
 class Call : public webrtc::Call,
              public PacketReceiver,
              public RecoveredPacketReceiver,
              public SendSideCongestionController::Observer,
              public BitrateAllocator::LimitObserver {
  public:
@@ skipping 602 matching lines @@
       // type, we may get an incorrect value for the rtx stream, but
       // that is unlikely to matter in practice.
       receive_rtp_config_[config.rtp.rtx_ssrc] = receive_config;
     }
     receive_rtp_config_[config.rtp.remote_ssrc] = receive_config;
     video_receive_streams_.insert(receive_stream);
     ConfigureSync(config.sync_group);
   }
   receive_stream->SignalNetworkState(video_network_state_);
   UpdateAggregateNetworkState();
-  event_log_->LogVideoReceiveStreamConfig(config);
+  event_log_->LogVideoReceiveStreamConfig(CreateRtcLogStreamConfig(config));
   return receive_stream;
 }
 
 void Call::DestroyVideoReceiveStream(
     webrtc::VideoReceiveStream* receive_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(receive_stream != nullptr);
   VideoReceiveStream* receive_stream_impl =
       static_cast<VideoReceiveStream*>(receive_stream);
@@ skipping 508 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc