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webrtc_m130/modules
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Björn Terelius c0b0494860 Fix loss of precision in accumulation of RTT in GoogCC 
Bug: webrtc:14513
Change-Id: Iefa4cf906ded02b224b970cabeea5b8c4ed122de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277760
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38273}
2022-10-03 12:10:48 +00:00
..
async_audio_processing
Remove dependency on rtc_base_approved from most targets
2022-04-25 12:15:30 +00:00
audio_coding
Remove old checksums for older version of opus.
2022-09-27 18:33:52 +00:00
audio_device
Audio stack traces
2022-09-27 15:05:51 +00:00
audio_mixer
Audio stack traces
2022-09-27 15:05:51 +00:00
audio_processing
APM: Fix benign race in MaybeInitializeCapture()
2022-09-29 09:30:03 +00:00
congestion_controller
Fix loss of precision in accumulation of RTT in GoogCC
2022-10-03 12:10:48 +00:00
desktop_capture
Add WindowId to Source on ChromeOS
2022-09-28 21:05:22 +00:00
include
Delete ProcessThread and related Module interface
2022-07-04 10:20:35 +00:00
pacing
Revise video owners
2022-09-30 08:44:30 +00:00
remote_bitrate_estimator
Revise video owners
2022-09-30 08:44:30 +00:00
rtp_rtcp
Audio stack traces
2022-09-27 15:05:51 +00:00
third_party
Update portaudio to the latest
2022-05-13 09:01:34 +00:00
utility
Delete ProcessThread and related Module interface
2022-07-04 10:20:35 +00:00
video_capture
Update/delete old TODOs
2022-07-06 07:49:04 +00:00
video_coding
Remove the libaom av1 decoder.
2022-09-30 08:42:25 +00:00
BUILD.gn
Delete modules/video_processing
2022-09-30 13:50:49 +00:00
module_common_types_unittest.cc
…