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Add new UMA metric for the audio receiver delay

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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}
This commit is contained in:
Henrik Lundin
2017-11-02 15:01:56 +01:00
committed by Commit Bot
parent 969d4a9386
commit 9bde6b7698
1 changed files with 11 additions and 0 deletions
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11
voice_engine/channel.cc
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@ -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;
}