Replace AudioSendStream::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 418 matching lines @@
   // Get SSRCs.
   RTC_CHECK(receiver_config.has_remote_ssrc());
   config->remote_ssrc = receiver_config.remote_ssrc();
   RTC_CHECK(receiver_config.has_local_ssrc());
   config->local_ssrc = receiver_config.local_ssrc();
   // Get header extensions.
   GetHeaderExtensions(&config->rtp_extensions,
                       receiver_config.header_extensions());
 }
 
-void ParsedRtcEventLog::GetAudioSendConfig(
-    size_t index,
-    AudioSendStream::Config* config) const {
+void ParsedRtcEventLog::GetAudioSendConfig(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::AUDIO_SENDER_CONFIG_EVENT);
   RTC_CHECK(event.has_audio_sender_config());
   const rtclog::AudioSendConfig& sender_config = event.audio_sender_config();
   // Get SSRCs.
   RTC_CHECK(sender_config.has_ssrc());
-  config->rtp.ssrc = sender_config.ssrc();
+  config->local_ssrc = sender_config.ssrc();
   // Get header extensions.
-  GetHeaderExtensions(&config->rtp.extensions,
+  GetHeaderExtensions(&config->rtp_extensions,
                       sender_config.header_extensions());
 }
 
 void ParsedRtcEventLog::GetAudioPlayout(size_t index, uint32_t* ssrc) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(event.has_type());
   RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_PLAYOUT_EVENT);
   RTC_CHECK(event.has_audio_playout_event());
   const rtclog::AudioPlayoutEvent& loss_event = event.audio_playout_event();
@@ skipping 119 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