admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/modules/audio_coding
History
Evan Shrubsole c3891e3a4e [Unwrap] Migrate NetEqDelayAnalyzer to use RtpTimestampUnwrapper 
Bug: webrtc:13982
Change-Id: I35c08921c8c1be31f0de4bd81f918250bee25313
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288961
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39052}
2023-01-10 09:53:17 +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
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
[Unwrap] Migrate NetEqDelayAnalyzer to use RtpTimestampUnwrapper
2023-01-10 09:53:17 +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
[Unwrap] Migrate NetEqDelayAnalyzer to use RtpTimestampUnwrapper
2023-01-10 09:53:17 +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