Added support for logging the SSRC corresponding to AudioPlayout events.

webrtc/voice_engine/channel.h

 /*
  *  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 470 matching lines @@
     int64_t GetRTT(bool allow_associate_channel) const;
 
     CriticalSectionWrapper& _fileCritSect;
     CriticalSectionWrapper& _callbackCritSect;
     CriticalSectionWrapper& volume_settings_critsect_;
     uint32_t _instanceId;
     int32_t _channelId;
 
     ChannelState channel_state_;
 
+    RtcEventLog* const event_log_;
+
     rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
     rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
     rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
     rtc::scoped_ptr<StatisticsProxy> statistics_proxy_;
     rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
     TelephoneEventHandler* telephone_event_handler_;
     rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule;
     rtc::scoped_ptr<AudioCodingModule> audio_coding_;
     AudioLevel _outputAudioLevel;
     bool _externalTransport;
@@ skipping 82 matching lines @@
     rtc::scoped_ptr<NetworkPredictor> network_predictor_;
     // An associated send channel.
     rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
     ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_