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

webrtc/logging/rtc_event_log/rtc_event_log_parser.cc

 /*
  *  Copyright (c) 2016 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 360 matching lines @@
       } else {
         config->rtx_ssrc = rtx_it->second.rtx_ssrc();
       }
     }
     config->codecs.emplace_back(receiver_config.decoders(i).name(),
                                 receiver_config.decoders(i).payload_type(),
                                 rtx_payload_type);
   }
 }
 
-void ParsedRtcEventLog::GetVideoSendConfig(
-    size_t index,
-    VideoSendStream::Config* config) const {
+void ParsedRtcEventLog::GetVideoSendConfig(size_t index,
+                                           rtclog::StreamConfig* config) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(config != nullptr);
   RTC_CHECK(event.has_type());
   RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
   RTC_CHECK(event.has_video_sender_config());
   const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
   // Get SSRCs.
-  config->rtp.ssrcs.clear();
-  for (int i = 0; i < sender_config.ssrcs_size(); i++) {
-    config->rtp.ssrcs.push_back(sender_config.ssrcs(i));
-  }
-  // Get header extensions.
-  GetHeaderExtensions(&config->rtp.extensions,
-                      sender_config.header_extensions());
-  // Get RTX settings.
-  config->rtp.rtx.ssrcs.clear();
-  for (int i = 0; i < sender_config.rtx_ssrcs_size(); i++) {
-    config->rtp.rtx.ssrcs.push_back(sender_config.rtx_ssrcs(i));
+  if (sender_config.ssrcs_size() > 0) {
+    config->local_ssrc = sender_config.ssrcs(0);
+    if (sender_config.ssrcs().size() > 1) {
+      LOG(WARNING) << "VideoSendConfig contains multiple ssrcs.";
+    }
   }
   if (sender_config.rtx_ssrcs_size() > 0) {
     RTC_CHECK(sender_config.has_rtx_payload_type());
-    config->rtp.rtx.payload_type = sender_config.rtx_payload_type();
-  } else {
-    // Reset RTX payload type default value if no RTX SSRCs are used.
-    config->rtp.rtx.payload_type = -1;
+    config->rtx_ssrc = sender_config.rtx_ssrcs(0);
+    if (sender_config.rtx_ssrcs_size() > 1) {
+      LOG(WARNING) << "VideoSendConfig contains multiple rtx ssrcs.";
+    }
   }
-  // Get encoder.
+  // Get header extensions.
+  GetHeaderExtensions(&config->rtp_extensions,
+                      sender_config.header_extensions());
+
+  // Get the codec.
   RTC_CHECK(sender_config.has_encoder());
   RTC_CHECK(sender_config.encoder().has_name());
   RTC_CHECK(sender_config.encoder().has_payload_type());
-  config->encoder_settings.payload_name = sender_config.encoder().name();
-  config->encoder_settings.payload_type =
-      sender_config.encoder().payload_type();
+  config->codecs.emplace_back(
+      sender_config.encoder().name(), sender_config.encoder().payload_type(),
+      sender_config.has_rtx_payload_type() ? sender_config.rtx_payload_type()
+                                           : 0);
 }
 
 void ParsedRtcEventLog::GetAudioReceiveConfig(
     size_t index,
     AudioReceiveStream::Config* config) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(config != nullptr);
   RTC_CHECK(event.has_type());
   RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT);
@@ skipping 157 matching lines @@
         rtc::Optional<ProbeFailureReason>(kInvalidSendReceiveRatio);
   } else if (pr_event.result() == rtclog::BweProbeResult::TIMEOUT) {
     res.failure_reason = rtc::Optional<ProbeFailureReason>(kTimeout);
   } else {
     RTC_NOTREACHED();
   }
 
   return res;
 }
 }  // namespace webrtc