Replace AudioReceiveStream::Config with rtclog::StreamConfig.

webrtc/logging/rtc_event_log/rtc_event_log.h

 /*
  *  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 101 matching lines @@
   // Stops logging to file and waits until the thread has finished.
   virtual void StopLogging() = 0;
 
   // Logs configuration information for a video receive stream.
   virtual void LogVideoReceiveStreamConfig(
       const rtclog::StreamConfig& config) = 0;
 
   // Logs configuration information for a video send stream.
   virtual void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) = 0;
 
-  // Logs configuration information for webrtc::AudioReceiveStream.
+  // Logs configuration information for an audio receive stream.
   virtual void LogAudioReceiveStreamConfig(
-      const webrtc::AudioReceiveStream::Config& config) = 0;
+      const rtclog::StreamConfig& config) = 0;
 
   // Logs configuration information for webrtc::AudioSendStream.
   virtual void LogAudioSendStreamConfig(
       const webrtc::AudioSendStream::Config& config) = 0;
 
   // Logs the header of an incoming or outgoing RTP packet. packet_length
   // is the total length of the packet, including both header and payload.
   virtual void LogRtpHeader(PacketDirection direction,
                             MediaType media_type,
                             const uint8_t* header,
@@ skipping 60 matching lines @@
                     int64_t max_size_bytes) override {
     return false;
   }
   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 AudioReceiveStream::Config& config) override {}
+      const rtclog::StreamConfig& config) override {}
   void LogAudioSendStreamConfig(
       const AudioSendStream::Config& 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,
@@ skipping 15 matching lines @@
                               int min_probes,
                               int min_bytes) override{};
   void LogProbeResultSuccess(int id, int bitrate_bps) override{};
   void LogProbeResultFailure(int id,
                              ProbeFailureReason failure_reason) override{};
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_