Remove ResetStatistics from RTP feedback.

webrtc/modules/rtp_rtcp/interface/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 234 matching lines @@
         const int frequency,
         const uint8_t channels,
         const uint32_t rate) = 0;
 
     virtual void OnIncomingSSRCChanged( const int32_t id,
                                         const uint32_t ssrc) = 0;
 
     virtual void OnIncomingCSRCChanged( const int32_t id,
                                         const uint32_t CSRC,
                                         const bool added) = 0;
-
-    virtual void ResetStatistics(uint32_t ssrc) = 0;
 };
 
 class RtpAudioFeedback {
  public:
 
   virtual void OnPlayTelephoneEvent(const int32_t id,
                                     const uint8_t event,
                                     const uint16_t lengthMs,
                                     const uint8_t volume) = 0;
  protected:
@@ skipping 49 matching lines @@
                               const uint8_t channels,
                               const uint32_t rate) override {
     return 0;
   }
 
   void OnIncomingSSRCChanged(const int32_t id, const uint32_t ssrc) override {}
 
   void OnIncomingCSRCChanged(const int32_t id,
                              const uint32_t CSRC,
                              const bool added) override {}
-
-  void ResetStatistics(uint32_t ssrc) override {}
 };
 
 // Null object version of RtpData.
 class NullRtpData : public RtpData {
  public:
   virtual ~NullRtpData() {}
 
   int32_t OnReceivedPayloadData(const uint8_t* payloadData,
                                 const size_t payloadSize,
                                 const WebRtcRTPHeader* rtpHeader) override {
@@ skipping 11 matching lines @@
   virtual ~NullRtpAudioFeedback() {}
 
   void OnPlayTelephoneEvent(const int32_t id,
                             const uint8_t event,
                             const uint16_t lengthMs,
                             const uint8_t volume) override {}
 };
 
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_