Add new UMA metric for NetEq target buffer delay
The UMA metric will log the same information that goes into the
googPreferredJitterBufferMs stat.
Bug: webrtc:8488
Change-Id: I4e4e1e362dd42377105d52d2c4cd49c1ecb1a90d
Reviewed-on: https://webrtc-review.googlesource.com/26740
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20923}diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index e9a14be..0d5dcae 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -300,6 +300,10 @@
return neteq_->FilteredCurrentDelayMs();
}
+int AcmReceiver::TargetDelayMs() const {
+ return neteq_->TargetDelayMs();
+}
+
int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
rtc::CritScope lock(&crit_sect_);
if (!last_audio_decoder_) {
