Logo
Explore Help
Register Sign In
admin/webrtc_m130
1
0
Fork 0
You've already forked webrtc_m130
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc/modules/rtp_rtcp
History
sprang e2d83d6560 Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()
Also move some stats reporting from vie_channel to send stats proxy

BUG=

Review URL: https://codereview.webrtc.org/1669623004

Cr-Commit-Position: refs/heads/master@{#11688}
2016-02-19 17:03:34 +00:00
..
include
Remove video-codec max bitrate from TMMBN.
2016-02-16 16:59:36 +00:00
mocks
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
2015-12-15 10:54:50 +00:00
source
Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()
2016-02-19 17:03:34 +00:00
test
iOS: Add resource files for tests and implement OutputPath
2016-02-17 06:06:17 +00:00
BUILD.gn
[rtp_rtcp] Added Sender Report Request rtcp packet.
2016-02-09 13:57:56 +00:00
OWNERS
OWNERS: Add * to .gyp{i,} everywhere.
2015-12-16 19:44:39 +00:00
rtp_rtcp.gypi
[rtp_rtcp] Added Sender Report Request rtcp packet.
2016-02-09 13:57:56 +00:00
Powered by Gitea Version: 1.23.5 Page: 177ms Template: 3ms
English
Bahasa Indonesia Deutsch English Español Français Gaeilge Italiano Latviešu Magyar nyelv Nederlands Polski Português de Portugal Português do Brasil Suomi Svenska Türkçe Čeština Ελληνικά Български Русский Українська فارسی മലയാളം 日本語 简体中文 繁體中文(台灣) 繁體中文(香港) 한국어
Licenses API