Remove RTC_LOGGED_* macro.

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 294 matching lines @@
     rtc::CritScope lock(&bitrate_crit_);
     UpdateSendHistograms();
   }
   UpdateReceiveHistograms();
   UpdateHistograms();
 
   Trace::ReturnTrace();
 }
 
 void Call::UpdateHistograms() {
-  RTC_LOGGED_HISTOGRAM_COUNTS_100000(
+  RTC_HISTOGRAM_COUNTS_100000(
       "WebRTC.Call.LifetimeInSeconds",
       (clock_->TimeInMilliseconds() - start_ms_) / 1000);
 }
 
 void Call::UpdateSendHistograms() {
   if (first_packet_sent_ms_ == -1)
     return;
   int64_t elapsed_sec =
       (clock_->TimeInMilliseconds() - first_packet_sent_ms_) / 1000;
   if (elapsed_sec < metrics::kMinRunTimeInSeconds)
     return;
   const int kMinRequiredPeriodicSamples = 5;
   AggregatedStats send_bitrate_stats =
       estimated_send_bitrate_kbps_counter_.ProcessAndGetStats();
   if (send_bitrate_stats.num_samples > kMinRequiredPeriodicSamples) {
-    RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
-                                       send_bitrate_stats.average);
+    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
+                                send_bitrate_stats.average);
   }
   AggregatedStats pacer_bitrate_stats =
       pacer_bitrate_kbps_counter_.ProcessAndGetStats();
   if (pacer_bitrate_stats.num_samples > kMinRequiredPeriodicSamples) {
-    RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
-                                       pacer_bitrate_stats.average);
+    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
+                                pacer_bitrate_stats.average);
   }
 }
 
 void Call::UpdateReceiveHistograms() {
   const int kMinRequiredPeriodicSamples = 5;
   AggregatedStats video_bytes_per_sec =
       received_video_bytes_per_second_counter_.GetStats();
   if (video_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
-    RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
-                                       video_bytes_per_sec.average * 8 / 1000);
+    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
+                                video_bytes_per_sec.average * 8 / 1000);
   }
   AggregatedStats audio_bytes_per_sec =
       received_audio_bytes_per_second_counter_.GetStats();
   if (audio_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
-    RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
-                                       audio_bytes_per_sec.average * 8 / 1000);
+    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
+                                audio_bytes_per_sec.average * 8 / 1000);
   }
   AggregatedStats rtcp_bytes_per_sec =
       received_rtcp_bytes_per_second_counter_.GetStats();
   if (rtcp_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
-    RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
-                                       rtcp_bytes_per_sec.average * 8);
+    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
+                                rtcp_bytes_per_sec.average * 8);
   }
   AggregatedStats recv_bytes_per_sec =
       received_bytes_per_second_counter_.GetStats();
   if (recv_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
-    RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps",
-                                       recv_bytes_per_sec.average * 8 / 1000);
+    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps",
+                                recv_bytes_per_sec.average * 8 / 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 563 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