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webrtc/modules/audio_coding
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Jakob Ivarsson d5ebc33562 Refactor NetEq rtp dump input. 
NetEq packet source input doesn't have any other uses than rtp dump,
so remove that layer.

Bug: None
Change-Id: I667bb4aead9f0f2fe8a1c0d6d911a4670ded67e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300542
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39810}
2023-04-11 14:32:35 +00:00
..
acm2 Delete RTPHeader::payload_type_frequency as unused 2023-03-09 16:32:22 +00:00
audio_network_adaptor Remove mentions of already deleted field trials 2023-03-01 15:53:37 +00:00
codecs Stop setting OPUS_SIGNAL_VOICE when DTX is enabled. 2022-12-20 11:06:48 +00:00
g3doc Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
include Delete deprecated Create method and config from AudioCodingModule 2023-02-02 17:06:29 +00:00
neteq Refactor NetEq rtp dump input. 2023-04-11 14:32:35 +00:00
test Break apart AudioCodingModule and AcmReceiver 2023-02-01 16:09:26 +00:00
audio_coding.gni Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
BUILD.gn Refactor NetEq rtp dump input. 2023-04-11 14:32:35 +00:00
DEPS
OWNERS Add jakobi to modules/audio_coding OWNERS 2021-06-18 11:52:58 +00:00