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

webrtc/modules/audio_processing/audio_processing_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 1259 matching lines @@
     if (capture_.aec_system_delay_jumps == -1 &&
         echo_cancellation()->stream_has_echo()) {
       capture_.aec_system_delay_jumps = 0;
     }
 
     // Detect a jump in platform reported system delay and log the difference.
     const int diff_stream_delay_ms =
         capture_nonlocked_.stream_delay_ms - capture_.last_stream_delay_ms;
     if (diff_stream_delay_ms > kMinDiffDelayMs &&
         capture_.last_stream_delay_ms != 0) {
-      RTC_HISTOGRAM_COUNTS_SPARSE(
-          "WebRTC.Audio.PlatformReportedStreamDelayJump", diff_stream_delay_ms,
-          kMinDiffDelayMs, 1000, 100);
+      RTC_HISTOGRAM_COUNTS("WebRTC.Audio.PlatformReportedStreamDelayJump",
+                           diff_stream_delay_ms, kMinDiffDelayMs, 1000, 100);
       if (capture_.stream_delay_jumps == -1) {
         capture_.stream_delay_jumps = 0;  // Activate counter if needed.
       }
       capture_.stream_delay_jumps++;
     }
     capture_.last_stream_delay_ms = capture_nonlocked_.stream_delay_ms;
 
     // Detect a jump in AEC system delay and log the difference.
     const int frames_per_ms =
         rtc::CheckedDivExact(capture_nonlocked_.split_rate, 1000);
     const int aec_system_delay_ms =
         WebRtcAec_system_delay(echo_cancellation()->aec_core()) / frames_per_ms;
     const int diff_aec_system_delay_ms =
         aec_system_delay_ms - capture_.last_aec_system_delay_ms;
     if (diff_aec_system_delay_ms > kMinDiffDelayMs &&
         capture_.last_aec_system_delay_ms != 0) {
-      RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.Audio.AecSystemDelayJump",
-                                  diff_aec_system_delay_ms, kMinDiffDelayMs,
-                                  1000, 100);
+      RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecSystemDelayJump",
+                           diff_aec_system_delay_ms, kMinDiffDelayMs, 1000,
+                           100);
       if (capture_.aec_system_delay_jumps == -1) {
         capture_.aec_system_delay_jumps = 0;  // Activate counter if needed.
       }
       capture_.aec_system_delay_jumps++;
     }
     capture_.last_aec_system_delay_ms = aec_system_delay_ms;
   }
 }
 
 void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
   // Run in a single-threaded manner.
   rtc::CritScope cs_render(&crit_render_);
   rtc::CritScope cs_capture(&crit_capture_);
 
   if (capture_.stream_delay_jumps > -1) {
-    RTC_HISTOGRAM_ENUMERATION_SPARSE(
+    RTC_HISTOGRAM_ENUMERATION(
         "WebRTC.Audio.NumOfPlatformReportedStreamDelayJumps",
         capture_.stream_delay_jumps, 51);
   }
   capture_.stream_delay_jumps = -1;
   capture_.last_stream_delay_ms = 0;
 
   if (capture_.aec_system_delay_jumps > -1) {
-    RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.Audio.NumOfAecSystemDelayJumps",
-                                     capture_.aec_system_delay_jumps, 51);
+    RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.NumOfAecSystemDelayJumps",
+                              capture_.aec_system_delay_jumps, 51);
   }
   capture_.aec_system_delay_jumps = -1;
   capture_.last_aec_system_delay_ms = 0;
 }
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 int AudioProcessingImpl::WriteMessageToDebugFile(
     FileWrapper* debug_file,
     int64_t* filesize_limit_bytes,
     rtc::CriticalSection* crit_debug,
@@ skipping 113 matching lines @@
   debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
 
   RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                         &debug_dump_.num_bytes_left_for_log_,
                                         &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc