Lionel Koenig a656b9d781 Use absolute capture timestamp from the beginning of payload ... This ensure the absolute capture timestamp from the first audio sample encoded in the payload is used for the corresponding rtp header. Bug: webrtc:42226041 Change-Id: Ib8f2e3a5df5c82c5806171bd5b36a26d92fbea72 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349265 Commit-Queue: Lionel Koenig <lionelk@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42281}		2024-05-13 08:10:56 +00:00
..
acm2	Use absolute capture timestamp from the beginning of payload	2024-05-13 08:10:56 +00:00
audio_network_adaptor	Remove expired field trial UseTwccPlrForAna	2024-04-15 14:26:33 +00:00
codecs	Remove expired WebRTC-Audio-OpusSetSignalVoiceWithDtx	2024-04-05 07:49:33 +00:00
g3doc	Reland "Migrate WebRTC documentation to new renderer"	2023-01-31 09:30:04 +00:00
include	Format /modules	2023-04-20 02:02:45 +00:00
neteq	Make muted param in GetAudio optional.	2024-05-06 18:07:34 +00:00
test	Reland "Unify access to SDP codec parameters"	2024-01-03 12:03:11 +00:00
audio_coding.gni	Reland "[ACM] iSAC audio codec removed"	2022-11-17 12:52:35 +00:00
BUILD.gn	Revert "Split digest methods from ssl target into digest target"	2024-05-08 06:42:32 +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