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webrtc/modules/audio_coding
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Artem Titov 0f2ce5cc1c Revert "Migrate WebRTC documentation to new renderer" 
This reverts commit 3eceaf4669.

Reason for revert: 

Original change's description:
> 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}

Bug: b/258408932
Change-Id: I16cb4088bee3fc15c2bb88bd692c592b3a7db9fe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291560
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39209}
2023-01-26 20:19:12 +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 Revert "Migrate WebRTC documentation to new renderer" 2023-01-26 20:19:12 +00:00
include Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
neteq Revert "Migrate WebRTC documentation to new renderer" 2023-01-26 20:19:12 +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