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webrtc/modules/audio_coding
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Jakob Ivarsson 10403ae87c Add PeerConnection option to configure minimum audio jitter buffer delay. 
Note that this value will override the minimum delay that is used for audio/video sync.

Bug: webrtc:10053
Change-Id: Ia129f6c9ee9da5d00a3d955afaaa6e8f0c2bee33
Reviewed-on: https://webrtc-review.googlesource.com/c/112121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25805}
2018-11-27 19:49:48 +00:00
..
acm2 Expose delayed packet outage as a cumulative metric of samples in the new getStats API. 2018-11-27 15:10:09 +00:00
audio_network_adaptor Removing ANA enabling field trials. 2018-11-22 22:26:28 +00:00
codecs Bump variable sizes in response to fuzzer bug 2018-11-26 16:16:50 +00:00
include Expose delayed packet outage as a cumulative metric of samples in the new getStats API. 2018-11-27 15:10:09 +00:00
neteq Add PeerConnection option to configure minimum audio jitter buffer delay. 2018-11-27 19:49:48 +00:00
test Hide the AudioEncoderCng class behind a create function 2018-11-02 13:00:05 +00:00
audio_coding.gni Don't select audio codecs depending on GN vars build_with_{chromium|mozilla} 2017-11-01 18:59:27 +00:00
BUILD.gn Move RtcpStatistics from common_types.h to a new header file 2018-11-27 13:46:42 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Make ivoc owner of audio_coding. 2018-10-15 15:08:28 +00:00