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

webrtc/modules/audio_processing/audio_processing_impl.cc

Index: webrtc/modules/audio_processing/audio_processing_impl.cc
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index dbc04b7cf34f5fedf29b613139ba93a5b8c281fe..49e8210b72509988daa126bce8034a1297103982 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -1277,9 +1277,8 @@ void AudioProcessingImpl::MaybeUpdateHistograms() {
         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.
       }
@@ -1296,9 +1295,9 @@ void AudioProcessingImpl::MaybeUpdateHistograms() {
         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.
       }
@@ -1314,7 +1313,7 @@ void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
   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);
   }
@@ -1322,8 +1321,8 @@ void AudioProcessingImpl::UpdateHistogramsOnCallEnd() {
   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;