Replace AudioReceiveStream::Config with rtclog::StreamConfig.

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 381 matching lines @@
 
       if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing) {
         std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
         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;
+      webrtc::rtclog::StreamConfig config;
       parsed_stream.GetAudioReceiveConfig(i, &config);
-      global_streams.emplace_back(config.rtp.remote_ssrc,
+      global_streams.emplace_back(config.remote_ssrc,
                                   webrtc::MediaType::AUDIO,
                                   webrtc::kIncomingPacket);
-      global_streams.emplace_back(config.rtp.local_ssrc,
+      global_streams.emplace_back(config.local_ssrc,
                                   webrtc::MediaType::AUDIO,
                                   webrtc::kOutgoingPacket);
       if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming) {
         std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
-                  << "\tssrc=" << config.rtp.remote_ssrc
-                  << "\tfeedback_ssrc=" << config.rtp.local_ssrc << std::endl;
+                  << "\tssrc=" << config.remote_ssrc
+                  << "\tfeedback_ssrc=" << config.local_ssrc << std::endl;
       }
     }
     if (parsed_stream.GetEventType(i) ==
         webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
       webrtc::AudioSendStream::Config config(nullptr);
       parsed_stream.GetAudioSendConfig(i, &config);
       global_streams.emplace_back(config.rtp.ssrc, webrtc::MediaType::AUDIO,
                                   webrtc::kOutgoingPacket);
       if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_nooutgoing) {
         std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
@@ skipping 67 matching lines @@
             PrintPsFeedback(rtcp_block, log_timestamp, direction);
             break;
           default:
             break;
         }
       }
     }
   }
   return 0;
 }