Remove ResetStatistics from RTP feedback.

webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.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 61 matching lines @@
       RtpFeedback* callback,
       int32_t id,
       int8_t payload_type,
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       const PayloadUnion& specific_payload) const = 0;
 
   // Checks if the payload type has changed, and returns whether we should
   // reset statistics and/or discard this packet.
   virtual void CheckPayloadChanged(int8_t payload_type,
                                    PayloadUnion* specific_payload,
-                                   bool* should_reset_statistics,
                                    bool* should_discard_changes);
 
   virtual int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const;
 
   // Stores / retrieves the last media specific payload for later reference.
   void GetLastMediaSpecificPayload(PayloadUnion* payload) const;
   void SetLastMediaSpecificPayload(const PayloadUnion& payload);
 
  protected:
   // The data callback is where we should send received payload data.
   // See ParseRtpPacket. This class does not claim ownership of the callback.
   // Implementations must NOT hold any critical sections while calling the
   // callback.
   //
   // Note: Implementations may call the callback for other reasons than calls
   // to ParseRtpPacket, for instance if the implementation somehow recovers a
   // packet.
   RTPReceiverStrategy(RtpData* data_callback);
 
   rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
   PayloadUnion last_payload_;
   RtpData* data_callback_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_