Remove RTC_LOGGED_* macro.

webrtc/video/receive_statistics_proxy.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 22 matching lines @@
   stats_.ssrc = config_.rtp.remote_ssrc;
   for (auto it : config_.rtp.rtx)
     rtx_stats_[it.second.ssrc] = StreamDataCounters();
 }
 
 ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
   UpdateHistograms();
 }
 
 void ReceiveStatisticsProxy::UpdateHistograms() {
-  RTC_LOGGED_HISTOGRAM_COUNTS_100000(
+  RTC_HISTOGRAM_COUNTS_100000(
       "WebRTC.Video.ReceiveStreamLifetimeInSeconds",
       (clock_->TimeInMilliseconds() - start_ms_) / 1000);
 
   int fraction_lost = report_block_stats_.FractionLostInPercent();
   if (fraction_lost != -1) {
-    RTC_LOGGED_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
-                                    fraction_lost);
+    RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
+                             fraction_lost);
   }
   const int kMinRequiredSamples = 200;
   int samples = static_cast<int>(render_fps_tracker_.TotalSampleCount());
   if (samples > kMinRequiredSamples) {
-    RTC_LOGGED_HISTOGRAM_COUNTS_100(
-        "WebRTC.Video.RenderFramesPerSecond",
-        round(render_fps_tracker_.ComputeTotalRate()));
-    RTC_LOGGED_HISTOGRAM_COUNTS_100000(
+    RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.RenderFramesPerSecond",
+                             round(render_fps_tracker_.ComputeTotalRate()));
+    RTC_HISTOGRAM_COUNTS_100000(
         "WebRTC.Video.RenderSqrtPixelsPerSecond",
         round(render_pixel_tracker_.ComputeTotalRate()));
   }
   int width = render_width_counter_.Avg(kMinRequiredSamples);
   int height = render_height_counter_.Avg(kMinRequiredSamples);
   if (width != -1) {
-    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedWidthInPixels",
-                                      width);
-    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedHeightInPixels",
-                                      height);
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedWidthInPixels", width);
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedHeightInPixels", height);
   }
   int sync_offset_ms = sync_offset_counter_.Avg(kMinRequiredSamples);
   if (sync_offset_ms != -1) {
-    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.AVSyncOffsetInMs",
-                                      sync_offset_ms);
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.AVSyncOffsetInMs", sync_offset_ms);
   }
 
   int qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
   if (qp != -1)
-    RTC_LOGGED_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", qp);
+    RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", qp);
 
   // TODO(asapersson): DecoderTiming() is call periodically (each 1000ms) and
   // not per frame. Change decode time to include every frame.
   const int kMinRequiredDecodeSamples = 5;
   int decode_ms = decode_time_counter_.Avg(kMinRequiredDecodeSamples);
   if (decode_ms != -1)
-    RTC_LOGGED_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
+    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
 
   int jb_delay_ms = jitter_buffer_delay_counter_.Avg(kMinRequiredDecodeSamples);
   if (jb_delay_ms != -1) {
-    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
-                                      jb_delay_ms);
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
+                               jb_delay_ms);
   }
   int target_delay_ms = target_delay_counter_.Avg(kMinRequiredDecodeSamples);
   if (target_delay_ms != -1) {
-    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs",
-                                      target_delay_ms);
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs", target_delay_ms);
   }
   int current_delay_ms = current_delay_counter_.Avg(kMinRequiredDecodeSamples);
   if (current_delay_ms != -1) {
-    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
-                                      current_delay_ms);
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
+                               current_delay_ms);
   }
 
   int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples);
   if (delay_ms != -1)
-    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
 
   StreamDataCounters rtp = stats_.rtp_stats;
   StreamDataCounters rtx;
   for (auto it : rtx_stats_)
     rtx.Add(it.second);
   StreamDataCounters rtp_rtx = rtp;
   rtp_rtx.Add(rtx);
   int64_t elapsed_sec =
       rtp_rtx.TimeSinceFirstPacketInMs(clock_->TimeInMilliseconds()) / 1000;
   if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
-    RTC_LOGGED_HISTOGRAM_COUNTS_10000(
+    RTC_HISTOGRAM_COUNTS_10000(
         "WebRTC.Video.BitrateReceivedInKbps",
         static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
                          1000));
-    RTC_LOGGED_HISTOGRAM_COUNTS_10000(
+    RTC_HISTOGRAM_COUNTS_10000(
         "WebRTC.Video.MediaBitrateReceivedInKbps",
         static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
-    RTC_LOGGED_HISTOGRAM_COUNTS_10000(
+    RTC_HISTOGRAM_COUNTS_10000(
         "WebRTC.Video.PaddingBitrateReceivedInKbps",
         static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
                          1000));
-    RTC_LOGGED_HISTOGRAM_COUNTS_10000(
+    RTC_HISTOGRAM_COUNTS_10000(
         "WebRTC.Video.RetransmittedBitrateReceivedInKbps",
         static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec /
                          1000));
     if (!rtx_stats_.empty()) {
-      RTC_LOGGED_HISTOGRAM_COUNTS_10000(
-          "WebRTC.Video.RtxBitrateReceivedInKbps",
-          static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
-                           1000));
+      RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateReceivedInKbps",
+                                 static_cast<int>(rtx.transmitted.TotalBytes() *
+                                                  8 / elapsed_sec / 1000));
     }
     if (config_.rtp.fec.ulpfec_payload_type != -1) {
-      RTC_LOGGED_HISTOGRAM_COUNTS_10000(
+      RTC_HISTOGRAM_COUNTS_10000(
           "WebRTC.Video.FecBitrateReceivedInKbps",
           static_cast<int>(rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec / 1000));
     }
     const RtcpPacketTypeCounter& counters = stats_.rtcp_packet_type_counts;
-    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute",
-                                      counters.nack_packets * 60 / elapsed_sec);
-    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute",
-                                      counters.fir_packets * 60 / elapsed_sec);
-    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
-                                      counters.pli_packets * 60 / elapsed_sec);
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute",
+                               counters.nack_packets * 60 / elapsed_sec);
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute",
+                               counters.fir_packets * 60 / elapsed_sec);
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
+                               counters.pli_packets * 60 / elapsed_sec);
     if (counters.nack_requests > 0) {
-      RTC_LOGGED_HISTOGRAM_PERCENTAGE(
-          "WebRTC.Video.UniqueNackRequestsSentInPercent",
-          counters.UniqueNackRequestsInPercent());
+      RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent",
+                               counters.UniqueNackRequestsInPercent());
     }
   }
 }
 
 VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
   rtc::CritScope lock(&crit_);
   return stats_;
 }
 
 void ReceiveStatisticsProxy::OnIncomingPayloadType(int payload_type) {
@@ skipping 145 matching lines @@
   ++num_samples;
 }
 
 int ReceiveStatisticsProxy::SampleCounter::Avg(int min_required_samples) const {
   if (num_samples < min_required_samples || num_samples == 0)
     return -1;
   return sum / num_samples;
 }
 
 }  // namespace webrtc