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webrtc_m130/modules/rtp_rtcp
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Danil Chapovalov 920abcc9bc In RtpSenderVideo::UpdateConditionalRetransmit use typed time and framerate instead of plain ints
Bug: webrtc:13757
Change-Id: If2df5418dacd2b95387fa74a9bc226426b207aee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313041
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40483}
2023-07-27 14:35:42 +00:00
..
include
In RtpRtcp use BitrateTracker instead of RateStatistics to measure bitrate
2023-07-24 14:57:29 +00:00
mocks
Update RtpSenderVideo::SendVideo/SendEncodedImage to take Timestamp/TimeDelta types
2023-07-21 10:36:49 +00:00
source
In RtpSenderVideo::UpdateConditionalRetransmit use typed time and framerate instead of plain ints
2023-07-27 14:35:42 +00:00
test/testFec
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
2021-11-15 21:44:59 +00:00
BUILD.gn
In RtpSenderVideo::UpdateConditionalRetransmit use typed time and framerate instead of plain ints
2023-07-27 14:35:42 +00:00
DEPS
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
OWNERS
Remove wildcard ownership for build files.
2020-02-19 14:05:46 +00:00
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