Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.

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 c89b0e0c8bb9a36efbc4dda32aa6e211cc5c1f22..7d9a17fcf39c86b79fd33b8a9289cf6d9b00cc2a 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -156,7 +156,6 @@ class Channel
       public FileCallback,  // receiving notification from file player &
                             // recorder
       public Transport,
-      public RtpAudioFeedback,
       public AudioPacketizationCallback,  // receive encoded packets from the
                                           // ACM
       public ACMVADCallback,              // receive voice activity from the ACM
@@ -385,11 +384,6 @@ class Channel
   void OnIncomingSSRCChanged(uint32_t ssrc) override;
   void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override;
 
-  // From RtpAudioFeedback in the RTP/RTCP module
-  void OnPlayTelephoneEvent(uint8_t event,
-                            uint16_t lengthMs,
-                            uint8_t volume) override;
-
   // From Transport (called by the RTP/RTCP module)
   bool SendRtp(const uint8_t* data,
                size_t len,