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webrtc/modules/audio_coding
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Artem Titov 3eceaf4669 Migrate WebRTC documentation to new renderer 
Bug: b/258408932
Change-Id: Ib96f39fe0c3912f9746bcc09d079097a145d6115
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290987
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39205}
2023-01-26 14:58:00 +00:00
..
acm2 Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
audio_network_adaptor Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead 2022-11-30 20:19:36 +00:00
codecs Stop setting OPUS_SIGNAL_VOICE when DTX is enabled. 2022-12-20 11:06:48 +00:00
g3doc Migrate WebRTC documentation to new renderer 2023-01-26 14:58:00 +00:00
include Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
neteq Migrate WebRTC documentation to new renderer 2023-01-26 14:58:00 +00:00
test Stop setting OPUS_SIGNAL_VOICE when DTX is enabled. 2022-12-20 11:06:48 +00:00
audio_coding.gni Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
BUILD.gn Remove all usage of //rtc_base target 2023-01-16 14:36:06 +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