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webrtc/modules/audio_coding
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Jakob Ivarsson 2237eb07c3 Reland "Change default NetEq sample rate to 48k." 
This is a reland of commit 38fcd58429

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: Id634799286f6d1f1eaf315ebe8e70de669d589db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281900
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38572}
2022-11-07 18:14:33 +00:00
..
acm2 Reland "Change default NetEq sample rate to 48k." 2022-11-07 18:14:33 +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
OWNERS Add jakobi to modules/audio_coding OWNERS 2021-06-18 11:52:58 +00:00