From 9bde6b76980f1233a4c252cb05b10a48c350cb6d Mon Sep 17 00:00:00 2001
From: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Thu, 2 Nov 2017 15:01:56 +0100
Subject: [PATCH] Add new UMA metric for the audio receiver delay

The UMA metric will log the same information that goes into the
googCurrentDelayMs stat.

Bug: webrtc:8488
Change-Id: I26abb3d86a07e8c0ddb4168540a8e2458115f004
Reviewed-on: https://webrtc-review.googlesource.com/18201
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20557}
---
 voice_engine/channel.cc | 11 +++++++++++
 1 file changed, 11 insertions(+)

diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 01aa96525c..d978c457c9 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -548,6 +548,17 @@ AudioMixer::Source::AudioFrameInfo Channel::GetAudioFrameWithInfo(
     }
   }
 
+  {
+    const int jitter_buffer_delay = audio_coding_->FilteredCurrentDelayMs();
+    rtc::CritScope lock(&video_sync_lock_);
+    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDelayEstimateMs",
+                              jitter_buffer_delay + playout_delay_ms_);
+    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverJitterBufferDelayMs",
+                              jitter_buffer_delay);
+    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDeviceDelayMs",
+                              playout_delay_ms_);
+  }
+
   return muted ? AudioMixer::Source::AudioFrameInfo::kMuted
                : AudioMixer::Source::AudioFrameInfo::kNormal;
 }