commit a2c58e21986410041fa72a4fd6c2f05bfd76b81b
author asapersson <asapersson@webrtc.org> Mon Mar 07 01:52:59 2016 -0800
committer Commit bot <commit-bot@chromium.org> Mon Mar 07 09:53:08 2016 +0000
tree 38e1c0db62cc866b3786a974226d6f92c04a2101
parent 7620be8492fc96ac8e3ca9fb28fd0d64b99c8e4a

Switch to use new implementation in metrics.h for gathering statistics.
Sparse macro replaced for all audio histograms that have a constant name.

BUG=webrtc:5283

Review URL: https://codereview.webrtc.org/1762863003

Cr-Commit-Position: refs/heads/master@{#11885}

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 9c587c2..a7c120f 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -1282,9 +1282,8 @@
         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.
       }
@@ -1303,9 +1302,9 @@
         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.
       }
@@ -1321,7 +1320,7 @@
   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);
   }
@@ -1329,8 +1328,8 @@
   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;