Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*).

webrtc/video/vie_receiver.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 39 matching lines @@
       restored_packet_in_use_(false),
       last_packet_log_ms_(-1) {}
 
 ViEReceiver::~ViEReceiver() {
   UpdateHistograms();
 }
 
 void ViEReceiver::UpdateHistograms() {
   FecPacketCounter counter = fec_receiver_->GetPacketCounter();
   if (counter.num_packets > 0) {
-    RTC_HISTOGRAM_PERCENTAGE(
+    RTC_LOGGED_HISTOGRAM_PERCENTAGE(
         "WebRTC.Video.ReceivedFecPacketsInPercent",
         static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets));
   }
   if (counter.num_fec_packets > 0) {
-    RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
-                             static_cast<int>(counter.num_recovered_packets *
-                                              100 / counter.num_fec_packets));
+    RTC_LOGGED_HISTOGRAM_PERCENTAGE(
+        "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
+        static_cast<int>(counter.num_recovered_packets * 100 /
+                         counter.num_fec_packets));
   }
 }
 
 bool ViEReceiver::SetReceiveCodec(const VideoCodec& video_codec) {
   int8_t old_pltype = -1;
   if (rtp_payload_registry_.ReceivePayloadType(
           video_codec.plName, kVideoPayloadTypeFrequency, 0,
           video_codec.maxBitrate, &old_pltype) != -1) {
     rtp_payload_registry_.DeRegisterReceivePayload(old_pltype);
   }
@@ skipping 301 matching lines @@
       rtp_receive_statistics_->GetStatistician(header.ssrc);
   if (!statistician)
     return false;
   // Check if this is a retransmission.
   int64_t min_rtt = 0;
   rtp_rtcp_[0]->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
   return !in_order &&
       statistician->IsRetransmitOfOldPacket(header, min_rtt);
 }
 }  // namespace webrtc