Replace AudioSendStream::Config with rtclog::StreamConfig.

webrtc/logging/rtc_event_log/rtc_event_log.cc

 /*
  *  Copyright (c) 2015 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 47 matching lines @@
   ~RtcEventLogImpl() override;
 
   bool StartLogging(const std::string& file_name,
                     int64_t max_size_bytes) override;
   bool StartLogging(rtc::PlatformFile platform_file,
                     int64_t max_size_bytes) override;
   void StopLogging() override;
   void LogVideoReceiveStreamConfig(const rtclog::StreamConfig& config) override;
   void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override;
   void LogAudioReceiveStreamConfig(const rtclog::StreamConfig& config) override;
-  void LogAudioSendStreamConfig(const AudioSendStream::Config& config) override;
+  void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override;
   void LogRtpHeader(PacketDirection direction,
                     MediaType media_type,
                     const uint8_t* header,
                     size_t packet_length) override;
   void LogRtpHeader(PacketDirection direction,
                     MediaType media_type,
                     const uint8_t* header,
                     size_t packet_length,
                     int probe_cluster_id) override;
   void LogRtcpPacket(PacketDirection direction,
@@ skipping 284 matching lines @@
   for (const auto& e : config.rtp_extensions) {
     rtclog::RtpHeaderExtension* extension =
         receiver_config->add_header_extensions();
     extension->set_name(e.uri);
     extension->set_id(e.id);
   }
   StoreEvent(&event);
 }
 
 void RtcEventLogImpl::LogAudioSendStreamConfig(
-    const AudioSendStream::Config& config) {
+    const rtclog::StreamConfig& config) {
   std::unique_ptr<rtclog::Event> event(new rtclog::Event());
   event->set_timestamp_us(rtc::TimeMicros());
   event->set_type(rtclog::Event::AUDIO_SENDER_CONFIG_EVENT);
 
   rtclog::AudioSendConfig* sender_config = event->mutable_audio_sender_config();
 
-  sender_config->set_ssrc(config.rtp.ssrc);
+  sender_config->set_ssrc(config.local_ssrc);
 
-  for (const auto& e : config.rtp.extensions) {
+  for (const auto& e : config.rtp_extensions) {
     rtclog::RtpHeaderExtension* extension =
         sender_config->add_header_extensions();
     extension->set_name(e.uri);
     extension->set_id(e.id);
   }
 
   StoreEvent(&event);
 }
 
 void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
@@ skipping 228 matching lines @@
 #else
   return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
 #endif  // ENABLE_RTC_EVENT_LOG
 }
 
 std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() {
   return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
 }
 
 }  // namespace webrtc