Added logging for audio send/receive stream configs.

webrtc/call/rtc_event_log_parser.h

 /*
  *  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.
  */
 #ifndef WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_
@@ skipping 78 matching lines @@
 
   // Reads a config event to a (non-NULL) VideoReceiveStream::Config struct.
   // Only the fields that are stored in the protobuf will be written.
   void GetVideoReceiveConfig(size_t index,
                              VideoReceiveStream::Config* config) const;
 
   // Reads a config event to a (non-NULL) VideoSendStream::Config struct.
   // Only the fields that are stored in the protobuf will be written.
   void GetVideoSendConfig(size_t index, VideoSendStream::Config* config) const;
 
+  // Reads a config event to a (non-NULL) AudioReceiveStream::Config struct.
+  // Only the fields that are stored in the protobuf will be written.
+  void GetAudioReceiveConfig(size_t index,
+                             AudioReceiveStream::Config* config) const;
+
+  // Reads a config event to a (non-NULL) AudioSendStream::Config struct.
+  // Only the fields that are stored in the protobuf will be written.
+  void GetAudioSendConfig(size_t index, AudioSendStream::Config* config) const;
+
   // Reads the SSRC from the audio playout event at |index|. The SSRC is stored
   // in the output parameter ssrc. The output parameter can be set to nullptr
   // and in that case the function only asserts that the event is well formed.
   void GetAudioPlayout(size_t index, uint32_t* ssrc) const;
 
   // Reads bitrate, fraction loss (as defined in RFC 1889) and total number of
   // expected packets from the BWE event at |index| and stores the values in
   // the corresponding output parameters. The output parameters can be set to
   // nullptr if those values aren't needed.
   // NB: The packet must have space for at least IP_PACKET_SIZE bytes.
   void GetBwePacketLossEvent(size_t index,
                              int32_t* bitrate,
                              uint8_t* fraction_loss,
                              int32_t* total_packets) const;
 
  private:
   std::vector<rtclog::Event> events_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_