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webrtc_m130/audio/voip
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Evan Shrubsole 8267cf31ce [Unwrap] Use RtpTimestampUnwrapper in audio_ingress 
Bug: webrtc:13982
Change-Id: I748f8e9d5497eac3335b8a9397199ddf24ecc8a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288746
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39035}
2023-01-09 14:47:22 +00:00
..
test
Update rtc::Event::Wait call sites to use TimeDelta.
2022-08-19 10:07:28 +00:00
audio_channel.cc
Replace Thread::Invoke with Thread::BlockingCall
2022-09-09 10:44:17 +00:00
audio_channel.h
ModuleRtcRtcpImpl2: remove Module inheritance.
2021-06-22 14:51:04 +00:00
audio_egress.cc
VoipVolumeControl subAPI for VoIP API
2020-11-13 19:27:12 +00:00
audio_egress.h
Use backticks not vertical bars to denote variables in comments for /audio
2021-07-27 15:36:40 +00:00
audio_ingress.cc
[Unwrap] Use RtpTimestampUnwrapper in audio_ingress
2023-01-09 14:47:22 +00:00
audio_ingress.h
[Unwrap] Use RtpTimestampUnwrapper in audio_ingress
2023-01-09 14:47:22 +00:00
BUILD.gn
[Unwrap] Use RtpTimestampUnwrapper in audio_ingress
2023-01-09 14:47:22 +00:00
voip_core.cc
Revert "Use ADM internal state for init state check."
2022-05-16 14:08:04 +00:00
voip_core.h
Revert "Use ADM internal state for init state check."
2022-05-16 14:08:04 +00:00