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webrtc/modules/audio_coding
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Jakob Ivarsson‎ 8f7ad88d0e Revert "Change default NetEq sample rate to 48k." 
This reverts commit 38fcd58429.

Reason for revert: Breaks downstream test

Original change's description:
> Change default NetEq sample rate to 48k.
>
> This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k).
>
> Bug: none
> Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662
> Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38536}

Bug: none
Change-Id: I03181168ab14d2d99320767c3a25ba3cfb726b2c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281441
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Auto-Submit: Jakob Ivarsson‎ <jakobi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38538}
2022-11-02 16:00:16 +00:00
..
acm2 Revert "Change default NetEq sample rate to 48k." 2022-11-02 16:00:16 +00:00
audio_network_adaptor Adopt absl::string_view in modules/audio_coding/ 2022-07-20 13:34:23 +00:00
codecs Make header files self contained. 2022-10-08 08:38:36 +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 Make it easier to specify in/out files for neteq_quality_test. 2022-10-11 21:10:11 +00:00
test Adopt absl::string_view in modules/audio_coding/ 2022-07-20 13:34:23 +00:00
audio_coding.gni build: remove WEBRTC_CODEC_RED 2020-05-26 11:01:26 +00:00
BUILD.gn Make it easier to specify in/out files for neteq_quality_test. 2022-10-11 21:10:11 +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