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webrtc/modules/audio_coding
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Ivo Creusen 1a84b565ac Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 
This metric was recently added to the standard (see https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferminimumdelay). This CL implements it for audio streams.

Bug: webrtc:14141
Change-Id: I79d918639cd12361ebbc28c2be41549e33fa7e2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262770
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37567}
2022-07-20 09:14:03 +00:00
..
acm2 Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
audio_network_adaptor Add objc_class_prefix to the Audio Network Adaptor proto. 2022-02-14 21:04:20 +00:00
codecs Don't add libopus to public_deps, its headers are only used directly 2022-06-28 19:13:14 +00:00
g3doc Update links to point at main branch 2021-07-22 16:41:26 +00:00
include Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
neteq Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
test Remove ACMTestTimer in iSACTest 2022-06-09 12:14:02 +00:00
audio_coding.gni build: remove WEBRTC_CODEC_RED 2020-05-26 11:01:26 +00:00
BUILD.gn Don't add libopus to public_deps, its headers are only used directly 2022-06-28 19:13:14 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Add jakobi to modules/audio_coding OWNERS 2021-06-18 11:52:58 +00:00