Remove RTC_LOGGED_* macro.

webrtc/modules/audio_device/audio_device_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.
  */
 
@@ skipping 82 matching lines @@
     LOG(INFO) << "average: "
              << static_cast<float>(total_diff_time) / num_measurements;
   }
 
   // Add UMA histogram to keep track of the case when only zeros have been
   // recorded. Ensure that recording callbacks have started and that at least
   // one timer event has been able to update |num_rec_level_is_zero_|.
   // I am avoiding use of the task queue here since we are under destruction
   // and reading these members on the creating thread feels safe.
   if (rec_callbacks_ > 0 && num_stat_reports_ > 0) {
-    RTC_LOGGED_HISTOGRAM_BOOLEAN("WebRTC.Audio.RecordedOnlyZeros",
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.RecordedOnlyZeros",
     static_cast<int>(num_stat_reports_ == num_rec_level_is_zero_));
   }
 }
 
 int32_t AudioDeviceBuffer::RegisterAudioCallback(
     AudioTransport* audio_callback) {
   LOG(INFO) << __FUNCTION__;
   rtc::CritScope lock(&_critSectCb);
   audio_transport_cb_ = audio_callback;
   return 0;
@@ skipping 377 matching lines @@
     int16_t max_abs = WebRtcSpl_MaxAbsValueW16(
         static_cast<int16_t*>(const_cast<void*>(audio_buffer)),
         num_samples * play_channels_);
     if (max_abs > max_play_level_) {
       max_play_level_ = max_abs;
     }
   }
 }
 
 }  // namespace webrtc