Remove ResetStatistics from RTP feedback.

webrtc/voice_engine/channel.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 56 matching lines @@
 
     CriticalSectionScoped cs(stats_lock_.get());
     stats_.rtcp = statistics;
     if (statistics.jitter > stats_.max_jitter) {
       stats_.max_jitter = statistics.jitter;
     }
   }
 
   void CNameChanged(const char* cname, uint32_t ssrc) override {}
 
-  void ResetStatistics() {
-    CriticalSectionScoped cs(stats_lock_.get());
-    stats_ = ChannelStatistics();
-  }
-
   ChannelStatistics GetStats() {
     CriticalSectionScoped cs(stats_lock_.get());
     return stats_;
   }
 
  private:
   // StatisticsUpdated calls are triggered from threads in the RTP module,
   // while GetStats calls can be triggered from the public voice engine API,
   // hence synchronization is needed.
   rtc::scoped_ptr<CriticalSectionWrapper> stats_lock_;
@@ skipping 241 matching lines @@
 
 void Channel::OnIncomingCSRCChanged(int32_t id,
                                     uint32_t CSRC,
                                     bool added)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::OnIncomingCSRCChanged(id=%d, CSRC=%d, added=%d)",
                  id, CSRC, added);
 }
 
-void Channel::ResetStatistics(uint32_t ssrc) {
-  StreamStatistician* statistician =
-      rtp_receive_statistics_->GetStatistician(ssrc);
-  if (statistician) {
-    statistician->ResetStatistics();
-  }
-  statistics_proxy_->ResetStatistics();
-}
-
 int32_t
 Channel::OnInitializeDecoder(
     int32_t id,
     int8_t payloadType,
     const char payloadName[RTP_PAYLOAD_NAME_SIZE],
     int frequency,
     uint8_t channels,
     uint32_t rate)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
@@ skipping 837 matching lines @@
     // Store the sequence number to be able to pick up the same sequence for
     // the next StartSend(). This is needed for restarting device, otherwise
     // it might cause libSRTP to complain about packets being replayed.
     // TODO(xians): Remove this workaround after RtpRtcpModule's refactoring
     // CL is landed. See issue
     // https://code.google.com/p/webrtc/issues/detail?id=2111 .
     send_sequence_number_ = _rtpRtcpModule->SequenceNumber();
 
     // Reset sending SSRC and sequence number and triggers direct transmission
     // of RTCP BYE
-    if (_rtpRtcpModule->SetSendingStatus(false) == -1 ||
-        _rtpRtcpModule->ResetSendDataCountersRTP() == -1)
+    if (_rtpRtcpModule->SetSendingStatus(false) == -1)
     {
         _engineStatisticsPtr->SetLastError(
             VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
             "StartSend() RTP/RTCP failed to stop sending");
     }
 
     return 0;
 }
 
 int32_t
@@ skipping 2925 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt)
       != 0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc