Switch to use new implementation in metrics.h.

webrtc/modules/video_coding/jitter_buffer.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.
  */
 #include "webrtc/modules/video_coding/jitter_buffer.h"
@@ skipping 263 matching lines @@
 void VCMJitterBuffer::UpdateHistograms() {
   if (num_packets_ <= 0 || !running_) {
     return;
   }
   int64_t elapsed_sec =
       (clock_->TimeInMilliseconds() - time_first_packet_ms_) / 1000;
   if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
     return;
   }
 
-  RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.DiscardedPacketsInPercent",
-                                  num_discarded_packets_ * 100 / num_packets_);
-  RTC_HISTOGRAM_PERCENTAGE_SPARSE("WebRTC.Video.DuplicatedPacketsInPercent",
-                                  num_duplicated_packets_ * 100 / num_packets_);
+  RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DiscardedPacketsInPercent",
+                           num_discarded_packets_ * 100 / num_packets_);
+  RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DuplicatedPacketsInPercent",
+                           num_duplicated_packets_ * 100 / num_packets_);
 
   int total_frames =
       receive_statistics_.key_frames + receive_statistics_.delta_frames;
   if (total_frames > 0) {
-    RTC_HISTOGRAM_COUNTS_SPARSE_100(
+    RTC_HISTOGRAM_COUNTS_100(
         "WebRTC.Video.CompleteFramesReceivedPerSecond",
         static_cast<int>((total_frames / elapsed_sec) + 0.5f));
-    RTC_HISTOGRAM_COUNTS_SPARSE_1000(
+    RTC_HISTOGRAM_COUNTS_1000(
         "WebRTC.Video.KeyFramesReceivedInPermille",
         static_cast<int>(
             (receive_statistics_.key_frames * 1000.0f / total_frames) + 0.5f));
   }
 }
 
 void VCMJitterBuffer::Start() {
   CriticalSectionScoped cs(crit_sect_);
   running_ = true;
   incoming_frame_count_ = 0;
@@ skipping 1031 matching lines @@
   }
   // Evaluate if the RTT is higher than |high_rtt_nack_threshold_ms_|, and in
   // that case we don't wait for retransmissions.
   if (high_rtt_nack_threshold_ms_ >= 0 &&
       rtt_ms_ >= high_rtt_nack_threshold_ms_) {
     return false;
   }
   return true;
 }
 }  // namespace webrtc