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

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 250 matching lines @@
   if (num_bitrate_updates_ == 0 || first_packet_sent_ms_ == -1)
     return;
   int64_t elapsed_sec =
       (clock_->TimeInMilliseconds() - first_packet_sent_ms_) / 1000;
   if (elapsed_sec < metrics::kMinRunTimeInSeconds)
     return;
   int send_bitrate_kbps =
       estimated_send_bitrate_sum_kbits_ / num_bitrate_updates_;
   int pacer_bitrate_kbps = pacer_bitrate_sum_kbits_ / num_bitrate_updates_;
   if (send_bitrate_kbps > 0) {
-    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
-                                send_bitrate_kbps);
+    RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
+                                       send_bitrate_kbps);
   }
   if (pacer_bitrate_kbps > 0) {
-    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
-                                pacer_bitrate_kbps);
+    RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
+                                       pacer_bitrate_kbps);
   }
 }
 
 void Call::UpdateReceiveHistograms() {
   if (first_rtp_packet_received_ms_ == -1)
     return;
   int64_t elapsed_sec =
       (last_rtp_packet_received_ms_ - first_rtp_packet_received_ms_) / 1000;
   if (elapsed_sec < metrics::kMinRunTimeInSeconds)
     return;
   int audio_bitrate_kbps = received_audio_bytes_ * 8 / elapsed_sec / 1000;
   int video_bitrate_kbps = received_video_bytes_ * 8 / elapsed_sec / 1000;
   int rtcp_bitrate_bps = received_rtcp_bytes_ * 8 / elapsed_sec;
   if (video_bitrate_kbps > 0) {
-    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
-                                video_bitrate_kbps);
+    RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
+                                       video_bitrate_kbps);
   }
   if (audio_bitrate_kbps > 0) {
-    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
-                                audio_bitrate_kbps);
+    RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
+                                       audio_bitrate_kbps);
   }
   if (rtcp_bitrate_bps > 0) {
-    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
-                                rtcp_bitrate_bps);
+    RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
+                                       rtcp_bitrate_bps);
   }
-  RTC_HISTOGRAM_COUNTS_100000(
+  RTC_LOGGED_HISTOGRAM_COUNTS_100000(
       "WebRTC.Call.BitrateReceivedInKbps",
       audio_bitrate_kbps + video_bitrate_kbps + rtcp_bitrate_bps / 1000);
 }
 
 PacketReceiver* Call::Receiver() {
   // TODO(solenberg): Some test cases in EndToEndTest use this from a different
   // thread. Re-enable once that is fixed.
   // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   return this;
 }
@@ skipping 432 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc