Replace AudioSendStream::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 397 matching lines @@
                                   webrtc::MediaType::AUDIO,
                                   webrtc::kOutgoingPacket);
       if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming) {
         std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
                   << "\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);
+      webrtc::rtclog::StreamConfig config;
       parsed_stream.GetAudioSendConfig(i, &config);
-      global_streams.emplace_back(config.rtp.ssrc, webrtc::MediaType::AUDIO,
+      global_streams.emplace_back(config.local_ssrc, webrtc::MediaType::AUDIO,
                                   webrtc::kOutgoingPacket);
       if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_nooutgoing) {
         std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
-                  << "\tssrc=" << config.rtp.ssrc << std::endl;
+                  << "\tssrc=" << config.local_ssrc << std::endl;
       }
     }
     if (!FLAGS_nortp &&
         parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
       size_t header_length;
       size_t total_length;
       uint8_t header[IP_PACKET_SIZE];
       webrtc::PacketDirection direction;
       webrtc::MediaType media_type;
       parsed_stream.GetRtpHeader(i, &direction, &media_type, header,
@@ skipping 56 matching lines @@
             PrintPsFeedback(rtcp_block, log_timestamp, direction);
             break;
           default:
             break;
         }
       }
     }
   }
   return 0;
 }