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

webrtc/logging/rtc_event_log/rtc_event_log2text.cc

 /*
  *  Copyright (c) 2017 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 364 matching lines @@
                                   webrtc::kOutgoingPacket);
 
       if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_noincoming) {
         std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
                 << "\tssrc=" << config.remote_ssrc
                 << "\tfeedback_ssrc=" << config.local_ssrc << std::endl;
       }
     }
     if (parsed_stream.GetEventType(i) ==
         webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
-      webrtc::VideoSendStream::Config config(nullptr);
+      webrtc::rtclog::StreamConfig config;
       parsed_stream.GetVideoSendConfig(i, &config);
+      global_streams.emplace_back(config.local_ssrc, webrtc::MediaType::VIDEO,
+                                  webrtc::kOutgoingPacket);
 
-      for (uint32_t ssrc : config.rtp.ssrcs) {
-        global_streams.emplace_back(ssrc, webrtc::MediaType::VIDEO,
-                                    webrtc::kOutgoingPacket);
-      }
-      for (uint32_t ssrc : config.rtp.rtx.ssrcs) {
-        global_streams.emplace_back(ssrc, webrtc::MediaType::VIDEO,
-                                    webrtc::kOutgoingPacket);
-      }
+      global_streams.emplace_back(config.rtx_ssrc, webrtc::MediaType::VIDEO,
+                                  webrtc::kOutgoingPacket);
 
       if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing) {
         std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
-        std::cout << "\tssrcs=";
-        for (const auto& ssrc : config.rtp.ssrcs)
-          std::cout << ssrc << ',';
-        std::cout << "\trtx_ssrcs=";
-        for (const auto& ssrc : config.rtp.rtx.ssrcs)
-          std::cout << ssrc << ',';
+        std::cout << "\tssrcs=" << config.local_ssrc;
+        std::cout << "\trtx_ssrcs=" << config.rtx_ssrc;
         std::cout << std::endl;
       }
     }
     if (parsed_stream.GetEventType(i) ==
         webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
       webrtc::AudioReceiveStream::Config config;
       parsed_stream.GetAudioReceiveConfig(i, &config);
       global_streams.emplace_back(config.rtp.remote_ssrc,
                                   webrtc::MediaType::AUDIO,
                                   webrtc::kIncomingPacket);
@@ skipping 84 matching lines @@
             PrintPsFeedback(rtcp_block, log_timestamp, direction);
             break;
           default:
             break;
         }
       }
     }
   }
   return 0;
 }