Remove ResetStatistics from RTP feedback.

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.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 210 matching lines @@
   // kRtpNoRtp against NetEq speech_type kOutputPLCtoCNG.
   if (last_payload_length < 10) {  // our CNG is 9 bytes
     return kRtpNoRtp;
   } else {
     return kRtpDead;
   }
 }
 
 void RTPReceiverAudio::CheckPayloadChanged(int8_t payload_type,
                                            PayloadUnion* specific_payload,
-                                           bool* should_reset_statistics,
                                            bool* should_discard_changes) {
   *should_discard_changes = false;
-  *should_reset_statistics = false;
 
   if (TelephoneEventPayloadType(payload_type)) {
     // Don't do callbacks for DTMF packets.
     *should_discard_changes = true;
     return;
   }
   // frequency is updated for CNG
   bool cng_payload_type_has_changed = false;
   bool is_cng_payload_type = CNGPayloadType(payload_type,
                                             &specific_payload->Audio.frequency,
                                             &cng_payload_type_has_changed);
 
-  *should_reset_statistics = cng_payload_type_has_changed;
-
   if (is_cng_payload_type) {
     // Don't do callbacks for DTMF packets.
     *should_discard_changes = true;
     return;
   }
 }
 
 int RTPReceiverAudio::Energy(uint8_t array_of_energy[kRtpCsrcSize]) const {
   CriticalSectionScoped cs(crit_sect_.get());
 
@@ skipping 126 matching lines @@
     // only one frame in the RED strip the one byte to help NetEq
     return data_callback_->OnReceivedPayloadData(
         payload_data + 1, payload_length - 1, rtp_header);
   }
 
   rtp_header->type.Audio.channel = audio_specific.channels;
   return data_callback_->OnReceivedPayloadData(
       payload_data, payload_length, rtp_header);
 }
 }  // namespace webrtc