Replace AudioReceiveStream::Config with rtclog::StreamConfig.

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 399 matching lines @@
   RTC_CHECK(sender_config.encoder().has_name());
   RTC_CHECK(sender_config.encoder().has_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 {
+    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::AUDIO_RECEIVER_CONFIG_EVENT);
   RTC_CHECK(event.has_audio_receiver_config());
   const rtclog::AudioReceiveConfig& receiver_config =
       event.audio_receiver_config();
   // Get SSRCs.
   RTC_CHECK(receiver_config.has_remote_ssrc());
-  config->rtp.remote_ssrc = receiver_config.remote_ssrc();
+  config->remote_ssrc = receiver_config.remote_ssrc();
   RTC_CHECK(receiver_config.has_local_ssrc());
-  config->rtp.local_ssrc = receiver_config.local_ssrc();
+  config->local_ssrc = receiver_config.local_ssrc();
   // Get header extensions.
-  GetHeaderExtensions(&config->rtp.extensions,
+  GetHeaderExtensions(&config->rtp_extensions,
                       receiver_config.header_extensions());
 }
 
 void ParsedRtcEventLog::GetAudioSendConfig(
     size_t index,
     AudioSendStream::Config* config) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(config != nullptr);
   RTC_CHECK(event.has_type());
@@ skipping 137 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