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/audio_coding
History
henrik.lundin@webrtc.org e5be877476 Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets
BUG=2935
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5725 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19 13:36:58 +00:00
..
codecs
adding FEC support to WebRTC Opus wrapper and tests.
2014-03-07 11:49:11 +00:00
main
Disable TestOpusNewACM on Android.
2014-03-11 20:40:59 +00:00
neteq
Including algorithm header to avoid VS2013 breakage
2014-03-04 15:10:03 +00:00
neteq4
Changing the buffer size (in packets) to 1.5 seconds @ 30 ms packets
2014-03-19 13:36:58 +00:00
Powered by Gitea Version: 1.23.5 Page: 140ms 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