Remove Channel::UpdatePacketDelay and some member variables

webrtc/voice_engine/channel.h

Index: webrtc/voice_engine/channel.h
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 6fc4f3404009af5c2211164bfd6f27934a28dc88..65f3258a4f8d94d371141992734d23bcdbb75127 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -466,14 +466,13 @@ class Channel
   int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
   int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
   void UpdatePlayoutTimestamp(bool rtcp);
-  void UpdatePacketDelay(uint32_t timestamp, uint16_t sequenceNumber);
   void RegisterReceiveCodecsToRTPModule();
 
   int SetSendRtpHeaderExtension(bool enable,
                                 RTPExtensionType type,
                                 unsigned char id);
 
-  int32_t GetPlayoutFrequency();
+  int32_t GetPlayoutFrequency() const;

[hlundin-webrtc, 2016/08/30 12:19:49]

This change is not exactly needed, but makes it clear that we can safely omit
the call go GetPlayoutFrequency.

   int64_t GetRTT(bool allow_associate_channel) const;
 
   rtc::CriticalSection _fileCritSect;
@@ -565,9 +564,6 @@ class Channel
   AudioFrame::SpeechType _outputSpeechType;
   // VoEVideoSync
   rtc::CriticalSection video_sync_lock_;
-  uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_);
-  uint32_t _previousTimestamp;
-  uint16_t _recPacketDelayMs GUARDED_BY(video_sync_lock_);
   // VoEAudioProcessing
   bool _RxVadDetection;
   bool _rxAgcIsEnabled;