Remove ResetStatistics from RTP feedback.

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index e19378aa2e022804f534ec0c4453d1e035d7d9f8..c9a1adf19676a5776bd75e5fb684f301509fb386 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -228,10 +228,8 @@ RTPAliveType RTPReceiverAudio::ProcessDeadOrAlive(
 
 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.
@@ -244,8 +242,6 @@ void RTPReceiverAudio::CheckPayloadChanged(int8_t 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;