Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
74ba99062c48b278675cfe52643719202296fddc
/
call
/
rtp_bitrate_configurator_unittest.cc
b9b146c
Replace rtc::Optional with absl::optional in audio, call and video
by Danil Chapovalov
· 6 years ago
0c4f7be
New api struct BitrateSettings.
by Niels Möller
· 6 years ago
df023aa
Extracted bitrate configuration from call class.
by Sebastian Jansson
· 7 years ago