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

webrtc/modules/rtp_rtcp/include/rtp_receiver.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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
 
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
+class RtpAudioFeedback;
 class RTPPayloadRegistry;
 
 class TelephoneEventHandler {
  public:
   virtual ~TelephoneEventHandler() {}
 
   // The following three methods implement the TelephoneEventHandler interface.
   // Forward DTMFs to decoder for playout.
   virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0;
 
   // Is forwarding of outband telephone events turned on/off?
   virtual bool TelephoneEventForwardToDecoder() const = 0;
 
   // Is TelephoneEvent configured with payload type payload_type
   virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0;
 };
 
 class RtpReceiver {
  public:
   // Creates a video-enabled RTP receiver.
   static RtpReceiver* CreateVideoReceiver(
       Clock* clock,
       RtpData* incoming_payload_callback,
       RtpFeedback* incoming_messages_callback,
       RTPPayloadRegistry* rtp_payload_registry);
 
   // Creates an audio-enabled RTP receiver.
   static RtpReceiver* CreateAudioReceiver(
       Clock* clock,
+      RtpData* incoming_payload_callback,
+      RtpFeedback* incoming_messages_callback,
+      RTPPayloadRegistry* rtp_payload_registry);
+
+  // DEPRECATED: Creates an audio-enabled RTP receiver.
+  // TODO(solenberg): Remove, after updating downstream code.
+  static RtpReceiver* CreateAudioReceiver(
+      Clock* clock,
       RtpAudioFeedback* incoming_audio_feedback,
       RtpData* incoming_payload_callback,
       RtpFeedback* incoming_messages_callback,
       RTPPayloadRegistry* rtp_payload_registry);
 
   virtual ~RtpReceiver() {}
 
   // Returns a TelephoneEventHandler if available.
   virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
 
@@ skipping 36 matching lines @@
 
   // Returns the current remote CSRCs.
   virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0;
 
   // Returns the current energy of the RTP stream received.
   virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_