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

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
index 1e96d17a67ddbfae7ab249a6b4d525ff05a7d1fb..25c5e4dd88a8dee24e6f7d18e0ceaa7a41c605b3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -21,9 +21,7 @@
 namespace webrtc {
 class RTPSenderAudio : public DTMFqueue {
  public:
-  RTPSenderAudio(Clock* clock,
-                 RTPSender* rtpSender,
-                 RtpAudioFeedback* audio_feedback);
+  RTPSenderAudio(Clock* clock, RTPSender* rtpSender);
   virtual ~RTPSenderAudio();
 
   int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE],
@@ -73,7 +71,6 @@ class RTPSenderAudio : public DTMFqueue {
  private:
   Clock* const _clock;
   RTPSender* const _rtpSender;
-  RtpAudioFeedback* const _audioFeedback;
 
   rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect;