Switch to use new implementation in metrics.h for gathering statistics.

webrtc/modules/audio_coding/neteq/statistics_calculator.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 169 matching lines @@
     timestamps_since_last_report_ = 0;
     discarded_packets_ = 0;
   }
 }
 
 void StatisticsCalculator::SecondaryDecodedSamples(int num_samples) {
   secondary_decoded_samples_ += num_samples;
 }
 
 void StatisticsCalculator::LogDelayedPacketOutageEvent(int outage_duration_ms) {
-  RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.Audio.DelayedPacketOutageEventMs",
-                              outage_duration_ms, 1 /* min */, 2000 /* max */,
-                              100 /* bucket count */);
+  RTC_HISTOGRAM_COUNTS("WebRTC.Audio.DelayedPacketOutageEventMs",
+                       outage_duration_ms, 1 /* min */, 2000 /* max */,
+                       100 /* bucket count */);
   delayed_packet_outage_counter_.RegisterSample();
 }
 
 void StatisticsCalculator::StoreWaitingTime(int waiting_time_ms) {
   excess_buffer_delay_.RegisterSample(waiting_time_ms);
   RTC_DCHECK_LE(waiting_times_.size(), kLenWaitingTimes);
   if (waiting_times_.size() == kLenWaitingTimes) {
     // Erase first value.
     waiting_times_.pop_front();
   }
@@ skipping 84 matching lines @@
     // Ratio must be smaller than 1 in Q14.
     assert((numerator << 14) / denominator < (1 << 14));
     return static_cast<uint16_t>((numerator << 14) / denominator);
   } else {
     // Will not produce a ratio larger than 1, since this is probably an error.
     return 1 << 14;
   }
 }
 
 }  // namespace webrtc