Replace AudioReceiveStream::Config with rtclog::StreamConfig.

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 106 matching lines @@
   }
   rtclog_config.rtcp_mode = config.rtp.rtcp_mode;
   rtclog_config.rtp_extensions = config.rtp.extensions;
 
   rtclog_config.codecs.emplace_back(config.encoder_settings.payload_name,
                                     config.encoder_settings.payload_type,
                                     config.rtp.rtx.payload_type);
   return rtclog_config;
 }
 
+rtclog::StreamConfig CreateRtcLogStreamConfig(
+    const AudioReceiveStream::Config& config) {
+  rtclog::StreamConfig rtclog_config;
+  rtclog_config.remote_ssrc = config.rtp.remote_ssrc;
+  rtclog_config.local_ssrc = config.rtp.local_ssrc;
+  rtclog_config.rtp_extensions = config.rtp.extensions;
+  return rtclog_config;
+}
+
 }  // namespace
 
 namespace internal {
 
 class Call : public webrtc::Call,
              public PacketReceiver,
              public RecoveredPacketReceiver,
              public SendSideCongestionController::Observer,
              public BitrateAllocator::LimitObserver {
  public:
@@ skipping 450 matching lines @@
     }
   }
   UpdateAggregateNetworkState();
   delete audio_send_stream;
 }
 
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
-  event_log_->LogAudioReceiveStreamConfig(config);
+  event_log_->LogAudioReceiveStreamConfig(CreateRtcLogStreamConfig(config));
   AudioReceiveStream* receive_stream =
       new AudioReceiveStream(transport_send_->packet_router(), config,
                              config_.audio_state, event_log_);
   {
     WriteLockScoped write_lock(*receive_crit_);
     audio_rtp_demuxer_.AddSink(config.rtp.remote_ssrc, receive_stream);
     receive_rtp_config_[config.rtp.remote_ssrc] =
         ReceiveRtpConfig(config.rtp.extensions, UseSendSideBwe(config));
     audio_receive_streams_.insert(receive_stream);
 
@@ skipping 664 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