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/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 41756fb..12c98ee 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -656,6 +656,12 @@
virtual int FilteredCurrentDelayMs() const = 0;
///////////////////////////////////////////////////////////////////////////
+ // int FilteredCurrentDelayMs()
+ // Returns the current target delay for NetEq in ms.
+ //
+ virtual int TargetDelayMs() const = 0;
+
+ ///////////////////////////////////////////////////////////////////////////
// int32_t PlayoutData10Ms(
// Get 10 milliseconds of raw audio data for playout, at the given sampling
// frequency. ACM will perform a resampling if required.