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

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 89 matching lines @@
 
   for (const auto& d : config.decoders) {
     auto search = config.rtp.rtx_payload_types.find(d.payload_type);
     rtclog_config.codecs.emplace_back(
         d.payload_name, d.payload_type,
         search != config.rtp.rtx_payload_types.end() ? search->second : 0);
   }
   return rtclog_config;
 }
 
+rtclog::StreamConfig CreateRtcLogStreamConfig(
+    const VideoSendStream::Config& config,
+    size_t ssrc_index) {
+  rtclog::StreamConfig rtclog_config;
+  rtclog_config.local_ssrc = config.rtp.ssrcs[ssrc_index];
+  if (ssrc_index < config.rtp.rtx.ssrcs.size()) {
+    rtclog_config.rtx_ssrc = config.rtp.rtx.ssrcs[ssrc_index];
+  }
+  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;
+}
+
 }  // namespace
 
 namespace internal {
 
 class Call : public webrtc::Call,
              public PacketReceiver,
              public RecoveredPacketReceiver,
              public SendSideCongestionController::Observer,
              public BitrateAllocator::LimitObserver {
  public:
@@ skipping 511 matching lines @@
   delete audio_receive_stream;
 }
 
 webrtc::VideoSendStream* Call::CreateVideoSendStream(
     webrtc::VideoSendStream::Config config,
     VideoEncoderConfig encoder_config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   video_send_delay_stats_->AddSsrcs(config);
-  event_log_->LogVideoSendStreamConfig(config);
+  for (size_t ssrc_index = 0; ssrc_index < config.rtp.ssrcs.size();
+       ++ssrc_index) {
+    event_log_->LogVideoSendStreamConfig(
+        CreateRtcLogStreamConfig(config, ssrc_index));
+  }
 
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
   // Copy ssrcs from |config| since |config| is moved.
   std::vector<uint32_t> ssrcs = config.rtp.ssrcs;
   VideoSendStream* send_stream = new VideoSendStream(
       num_cpu_cores_, module_process_thread_.get(), &worker_queue_,
       call_stats_.get(), transport_send_.get(), bitrate_allocator_.get(),
       video_send_delay_stats_.get(), event_log_, std::move(config),
       std::move(encoder_config), suspended_video_send_ssrcs_);
@@ skipping 599 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