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

webrtc/modules/audio_coding/acm2/audio_coding_module_impl.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.
  */
 
@@ skipping 79 matching lines @@
         info.encoded_timestamp - info.redundant[i].encoded_timestamp);
     frag->fragmentationPlType[i] = info.redundant[i].payload_type;
   }
 }
 }  // namespace
 
 void AudioCodingModuleImpl::ChangeLogger::MaybeLog(int value) {
   if (value != last_value_ || first_time_) {
     first_time_ = false;
     last_value_ = value;
-    RTC_HISTOGRAM_COUNTS_100(histogram_name_, value);
+    RTC_HISTOGRAM_COUNTS_SPARSE_100(histogram_name_, value);
   }
 }
 
 AudioCodingModuleImpl::AudioCodingModuleImpl(
     const AudioCodingModule::Config& config)
     : acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       id_(config.id),
       expected_codec_ts_(0xD87F3F9F),
       expected_in_ts_(0xD87F3F9F),
       receiver_(config),
@@ skipping 667 matching lines @@
   return receiver_.LeastRequiredDelayMs();
 }
 
 void AudioCodingModuleImpl::GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const {
   receiver_.GetDecodingCallStatistics(call_stats);
 }
 
 }  // namespace acm2
 }  // namespace webrtc