admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/modules/audio_coding
History
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