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

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index c4c7dbb4cd3eafaf811573115f302af8bafb84b4..7a3b6fd829002b8a1d30891a02ae5e9fd71a5461 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -20,13 +20,11 @@
 
 namespace webrtc {
 RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
-    RtpData* data_callback,
-    RtpAudioFeedback* incoming_messages_callback) {
-  return new RTPReceiverAudio(data_callback, incoming_messages_callback);
+    RtpData* data_callback) {
+  return new RTPReceiverAudio(data_callback);
 }
 
-RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback,
-                                   RtpAudioFeedback* incoming_messages_callback)
+RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback)
     : RTPReceiverStrategy(data_callback),
       TelephoneEventHandler(),
       last_received_frequency_(8000),
@@ -40,8 +38,7 @@ RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback,
       g722_payload_type_(-1),
       last_received_g722_(false),
       num_energy_(0),
-      current_remote_energy_(),
-      cb_audio_feedback_(incoming_messages_callback) {
+      current_remote_energy_() {
   last_payload_.Audio.channels = 1;
   memset(current_remote_energy_, 0, sizeof(current_remote_energy_));
 }