Remove ResetStatistics from RTP feedback.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  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 1083 matching lines @@
   return rtp_header_length;
 }
 
 uint16_t RTPSender::AllocateSequenceNumber(uint16_t packets_to_send) {
   CriticalSectionScoped cs(send_critsect_.get());
   uint16_t first_allocated_sequence_number = sequence_number_;
   sequence_number_ += packets_to_send;
   return first_allocated_sequence_number;
 }
 
-void RTPSender::ResetDataCounters() {
-  uint32_t ssrc;
-  uint32_t ssrc_rtx;
-  bool report_rtx;
-  {
-    CriticalSectionScoped ssrc_lock(send_critsect_.get());
-    ssrc = ssrc_;
-    ssrc_rtx = ssrc_rtx_;
-    report_rtx = rtx_ != kRtxOff;
-  }
-  CriticalSectionScoped lock(statistics_crit_.get());
-  rtp_stats_ = StreamDataCounters();
-  rtx_rtp_stats_ = StreamDataCounters();
-  if (rtp_stats_callback_) {
-    rtp_stats_callback_->DataCountersUpdated(rtp_stats_, ssrc);
-    if (report_rtx)
-      rtp_stats_callback_->DataCountersUpdated(rtx_rtp_stats_, ssrc_rtx);
-  }
-}
-
 void RTPSender::GetDataCounters(StreamDataCounters* rtp_stats,
                                 StreamDataCounters* rtx_stats) const {
   CriticalSectionScoped lock(statistics_crit_.get());
   *rtp_stats = rtp_stats_;
   *rtx_stats = rtx_rtp_stats_;
 }
 
 size_t RTPSender::CreateRtpHeader(uint8_t* header,
                                   int8_t payload_type,
                                   uint32_t ssrc,
@@ skipping 743 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc