Added logging for audio send/receive stream configs.

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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 76 matching lines @@
   }
 
   void LogVideoSendStreamConfig(
       const webrtc::VideoSendStream::Config& config) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
       event_log_->LogVideoSendStreamConfig(config);
     }
   }
 
+  void LogAudioReceiveStreamConfig(
+      const webrtc::AudioReceiveStream::Config& config) override {
+    rtc::CritScope lock(&crit_);
+    if (event_log_) {
+      event_log_->LogAudioReceiveStreamConfig(config);
+    }
+  }
+
+  void LogAudioSendStreamConfig(
+      const webrtc::AudioSendStream::Config& config) override {
+    rtc::CritScope lock(&crit_);
+    if (event_log_) {
+      event_log_->LogAudioSendStreamConfig(config);
+    }
+  }
+
   void LogRtpHeader(webrtc::PacketDirection direction,
                     webrtc::MediaType media_type,
                     const uint8_t* header,
                     size_t packet_length) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
       event_log_->LogRtpHeader(direction, media_type, header, packet_length);
     }
   }
 
@@ skipping 3098 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc