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

webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.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.
  */
 
@@ skipping 199 matching lines @@
       const char payloadName[RTP_PAYLOAD_NAME_SIZE],
       const int frequency,
       const size_t channels,
       const uint32_t rate) = 0;
 
   virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0;
 
   virtual void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) = 0;
 };
 
+class RtpAudioFeedback {
+ public:
+  virtual void OnPlayTelephoneEvent(const uint8_t event,
+                                    const uint16_t lengthMs,
+                                    const uint8_t volume) = 0;
+
+ protected:
+  virtual ~RtpAudioFeedback() {}
+};
+
 class RtcpIntraFrameObserver {
  public:
   virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
 
   virtual void OnReceivedSLI(uint32_t ssrc,
                              uint8_t picture_id) = 0;
 
   virtual void OnReceivedRPSI(uint32_t ssrc,
                               uint64_t picture_id) = 0;
 
@@ skipping 110 matching lines @@
                                 const size_t payloadSize,
                                 const WebRtcRTPHeader* rtpHeader) override {
     return 0;
   }
 
   bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override {
     return true;
   }
 };
 
+// Null object version of RtpAudioFeedback.
+class NullRtpAudioFeedback : public RtpAudioFeedback {
+ public:
+  virtual ~NullRtpAudioFeedback() {}
+
+  void OnPlayTelephoneEvent(const uint8_t event,
+                            const uint16_t lengthMs,
+                            const uint8_t volume) override {}
+};
+
 // Statistics about packet loss for a single directional connection. All values
 // are totals since the connection initiated.
 struct RtpPacketLossStats {
   // The number of packets lost in events where no adjacent packets were also
   // lost.
   uint64_t single_packet_loss_count;
   // The number of events in which more than one adjacent packet was lost.
   uint64_t multiple_packet_loss_event_count;
   // The number of packets lost in events where more than one adjacent packet
   // was lost.
@@ skipping 26 matching lines @@
 class TransportSequenceNumberAllocator {
  public:
   TransportSequenceNumberAllocator() {}
   virtual ~TransportSequenceNumberAllocator() {}
 
   virtual uint16_t AllocateSequenceNumber() = 0;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_