Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that these are for infrequent

webrtc/modules/video_coding/timing.cc

 /*
  *  Copyright (c) 2011 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 44 matching lines @@
 void VCMTiming::UpdateHistograms() const {
   CriticalSectionScoped cs(crit_sect_);
   if (num_decoded_frames_ == 0) {
     return;
   }
   int64_t elapsed_sec =
       (clock_->TimeInMilliseconds() - first_decoded_frame_ms_) / 1000;
   if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
     return;
   }
-  RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.DecodedFramesPerSecond",
+  RTC_HISTOGRAM_COUNTS_SPARSE_100(
+      "WebRTC.Video.DecodedFramesPerSecond",
       static_cast<int>((num_decoded_frames_ / elapsed_sec) + 0.5f));
-  RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DelayedFramesToRenderer",
+  RTC_HISTOGRAM_PERCENTAGE_SPARSE(
+      "WebRTC.Video.DelayedFramesToRenderer",
       num_delayed_decoded_frames_ * 100 / num_decoded_frames_);
   if (num_delayed_decoded_frames_ > 0) {
-    RTC_HISTOGRAM_COUNTS_1000(
+    RTC_HISTOGRAM_COUNTS_SPARSE_1000(
         "WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs",
-            sum_missed_render_deadline_ms_ / num_delayed_decoded_frames_);
+        sum_missed_render_deadline_ms_ / num_delayed_decoded_frames_);
   }
 }
 
 void VCMTiming::Reset() {
   CriticalSectionScoped cs(crit_sect_);
   ts_extrapolator_->Reset(clock_->TimeInMilliseconds());
   codec_timer_.Reset();
   render_delay_ms_ = kDefaultRenderDelayMs;
   min_playout_delay_ms_ = 0;
   jitter_delay_ms_ = 0;
@@ skipping 187 matching lines @@
   *decode_ms = last_decode_ms_;
   *max_decode_ms = MaxDecodeTimeMs();
   *current_delay_ms = current_delay_ms_;
   *target_delay_ms = TargetDelayInternal();
   *jitter_buffer_ms = jitter_delay_ms_;
   *min_playout_delay_ms = min_playout_delay_ms_;
   *render_delay_ms = render_delay_ms_;
 }
 
 }  // namespace webrtc