Added logging for audio send/receive stream configs.

webrtc/call/mock/mock_rtc_event_log.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.
  */
 
@@ skipping 16 matching lines @@
                bool(rtc::PlatformFile log_file, int64_t max_size_bytes));
 
   MOCK_METHOD0(StopLogging, void());
 
   MOCK_METHOD1(LogVideoReceiveStreamConfig,
                void(const webrtc::VideoReceiveStream::Config& config));
 
   MOCK_METHOD1(LogVideoSendStreamConfig,
                void(const webrtc::VideoSendStream::Config& config));
 
+  MOCK_METHOD1(LogAudioReceiveStreamConfig,
+               void(const webrtc::AudioReceiveStream::Config& config));
+
+  MOCK_METHOD1(LogAudioSendStreamConfig,
+               void(const webrtc::AudioSendStream::Config& config));
+
   MOCK_METHOD4(LogRtpHeader,
                void(PacketDirection direction,
                     MediaType media_type,
                     const uint8_t* header,
                     size_t packet_length));
 
   MOCK_METHOD4(LogRtcpPacket,
                void(PacketDirection direction,
                     MediaType media_type,
                     const uint8_t* packet,
                     size_t length));
 
   MOCK_METHOD1(LogAudioPlayout, void(uint32_t ssrc));
 
   MOCK_METHOD3(LogBwePacketLossEvent,
                void(int32_t bitrate,
                     uint8_t fraction_loss,
                     int32_t total_packets));
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_MOCK_MOCK_RTC_EVENT_LOG_H_