Added logging for audio send/receive stream configs.

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.
  */
 
 #ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
 #define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
 
 #include <memory>
 #include <string>
 
+#include "webrtc/api/call/audio_receive_stream.h"
+#include "webrtc/api/call/audio_send_stream.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
 // Forward declaration of storage class that is automatically generated from
 // the protobuf file.
 namespace rtclog {
 class EventStream;
@@ skipping 43 matching lines @@
   virtual void StopLogging() = 0;
 
   // Logs configuration information for webrtc::VideoReceiveStream.
   virtual void LogVideoReceiveStreamConfig(
       const webrtc::VideoReceiveStream::Config& config) = 0;
 
   // Logs configuration information for webrtc::VideoSendStream.
   virtual void LogVideoSendStreamConfig(
       const webrtc::VideoSendStream::Config& config) = 0;
 
+  // Logs configuration information for webrtc::AudioReceiveStream.
+  virtual void LogAudioReceiveStreamConfig(
+      const webrtc::AudioReceiveStream::Config& 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,
                             size_t packet_length) = 0;
 
   // Logs an incoming or outgoing RTCP packet.
   virtual void LogRtcpPacket(PacketDirection direction,
                              MediaType media_type,
@@ skipping 26 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 VideoReceiveStream::Config& config) override {}
   void LogVideoSendStreamConfig(
       const VideoSendStream::Config& config) override {}
+  void LogAudioReceiveStreamConfig(
+      const AudioReceiveStream::Config& 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 LogRtcpPacket(PacketDirection direction,
                      MediaType media_type,
                      const uint8_t* packet,
                      size_t length) override {}
   void LogAudioPlayout(uint32_t ssrc) override {}
   void LogBwePacketLossEvent(int32_t bitrate,
                              uint8_t fraction_loss,
                              int32_t total_packets) override {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_