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

webrtc/tools/event_log_visualizer/analyzer.cc

 /*
  *  Copyright (c) 2016 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 313 matching lines @@
         event_type != ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT &&
         event_type != ParsedRtcEventLog::LOG_START &&
         event_type != ParsedRtcEventLog::LOG_END) {
       uint64_t timestamp = parsed_log_.GetTimestamp(i);
       first_timestamp = std::min(first_timestamp, timestamp);
       last_timestamp = std::max(last_timestamp, timestamp);
     }
 
     switch (parsed_log_.GetEventType(i)) {
       case ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT: {
-        VideoReceiveStream::Config config(nullptr);
+        rtclog::StreamConfig config;
         parsed_log_.GetVideoReceiveConfig(i, &config);
-        StreamId stream(config.rtp.remote_ssrc, kIncomingPacket);
-        extension_maps[stream] = RtpHeaderExtensionMap(config.rtp.extensions);
+        StreamId stream(config.remote_ssrc, kIncomingPacket);
+        extension_maps[stream] = RtpHeaderExtensionMap(config.rtp_extensions);
         video_ssrcs_.insert(stream);
-        StreamId rtx_stream(config.rtp.rtx_ssrc, kIncomingPacket);
+        StreamId rtx_stream(config.rtx_ssrc, kIncomingPacket);
         extension_maps[rtx_stream] =
-            RtpHeaderExtensionMap(config.rtp.extensions);
+            RtpHeaderExtensionMap(config.rtp_extensions);
         video_ssrcs_.insert(rtx_stream);
         rtx_ssrcs_.insert(rtx_stream);
         break;
       }
       case ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT: {
         VideoSendStream::Config config(nullptr);
         parsed_log_.GetVideoSendConfig(i, &config);
         for (auto ssrc : config.rtp.ssrcs) {
           StreamId stream(ssrc, kOutgoingPacket);
           extension_maps[stream] = RtpHeaderExtensionMap(config.rtp.extensions);
@@ skipping 1047 matching lines @@
       },
       audio_network_adaptation_events_, begin_time_, &time_series);
   plot->AppendTimeSeries(std::move(time_series));
   plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
   plot->SetSuggestedYAxis(0, 1, "Number of channels (1 (mono)/2 (stereo))",
                           kBottomMargin, kTopMargin);
   plot->SetTitle("Reported audio encoder number of channels");
 }
 }  // namespace plotting
 }  // namespace webrtc