Remove ResetStatistics from RTP feedback.

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.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 75 matching lines @@
       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
       size_t payload_name_length,
       uint32_t frequency,
       uint8_t channels,
       uint32_t rate) const;
 
   // We need to look out for special payload types here and sometimes reset
   // statistics. In addition we sometimes need to tweak the frequency.
   void CheckPayloadChanged(int8_t payload_type,
                            PayloadUnion* specific_payload,
-                           bool* should_reset_statistics,
                            bool* should_discard_changes) override;
 
   int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
 
  private:
 
   int32_t ParseAudioCodecSpecific(
       WebRtcRTPHeader* rtp_header,
       const uint8_t* payload_data,
       size_t payload_length,
@@ skipping 20 matching lines @@
   bool last_received_g722_;
 
   uint8_t num_energy_;
   uint8_t current_remote_energy_[kRtpCsrcSize];
 
   RtpAudioFeedback* cb_audio_feedback_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_