Replace AudioSendStream::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 115 matching lines @@
 
 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;
 }
 
+rtclog::StreamConfig CreateRtcLogStreamConfig(
+    const AudioSendStream::Config& config) {
+  rtclog::StreamConfig rtclog_config;
+  rtclog_config.local_ssrc = config.rtp.ssrc;
+  rtclog_config.rtp_extensions = config.rtp.extensions;
+  if (config.send_codec_spec) {
+    rtclog_config.codecs.emplace_back(config.send_codec_spec->format.name,
+                                      config.send_codec_spec->payload_type, 0);
+  }
+  return rtclog_config;
+}
+
 }  // namespace
 
 namespace internal {
 
 class Call : public webrtc::Call,
              public PacketReceiver,
              public RecoveredPacketReceiver,
              public SendSideCongestionController::Observer,
              public BitrateAllocator::LimitObserver {
  public:
@@ skipping 396 matching lines @@
   // TODO(solenberg): Some test cases in EndToEndTest use this from a different
   // thread. Re-enable once that is fixed.
   // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   return this;
 }
 
 webrtc::AudioSendStream* Call::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
-  event_log_->LogAudioSendStreamConfig(config);
+  event_log_->LogAudioSendStreamConfig(CreateRtcLogStreamConfig(config));
   AudioSendStream* send_stream = new AudioSendStream(
       config, config_.audio_state, &worker_queue_, transport_send_.get(),
       bitrate_allocator_.get(), event_log_, call_stats_->rtcp_rtt_stats());
   {
     WriteLockScoped write_lock(*send_crit_);
     RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
                audio_send_ssrcs_.end());
     audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
   }
   {
@@ skipping 718 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