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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
191e38fb47bc1b8f77888e261bc034149f401c65
/
api
/
rtp_parameters.cc
46bbdec
Allow AbsSendTime extension to be used for audio streams.
by Sebastian Jansson
· 5 years ago
cd8a6e2
Add writing and parsing of the `abs-capture-time` RTP header extension.
by Chen Xing
· 5 years ago
a59dcc3
Use Abseil container algorithms in api/
by Steve Anton
· 6 years ago
8cc711a
Update URI of TransportSequenceNumberV2
by Johannes Kron
· 6 years ago
48e7065
Remove default IDs for RTP extensions from rtp_parameters.h
by Elad Alon
· 6 years ago
ce8e867
Add support for TransportSequenceNumberV2 in SDP negotiation
by Johannes Kron
· 6 years ago
ccb9b75
Create version 01 of Generic Frame Descriptor - with discardability flag
by Elad Alon
· 6 years ago
7ff164e
Plumbing of feedback on request setting
by Johannes Kron
· 6 years ago
10542f2
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
by Steve Anton
· 6 years ago
1c05765
(3) Rename files to snake_case: move the files
by Steve Anton
· 6 years ago
[Renamed from api/rtpparameters.cc]
77938e6
Simulcast work to enable RID mux.
by Amit Hilbuch
· 6 years ago
d0b69a8
Send and receive color space information if available
by Johannes Kron
· 6 years ago
988cc08
[Cleanup] Add missing #include. Remove useless ones.
by Yves Gerey
· 6 years ago
2e00abc
Reland "[cleanup] Remove useless includes."
by Yves Gerey
· 6 years ago
96a0f61
Revert "[cleanup] Remove useless includes."
by Oleh Prypin
· 6 years ago
be8b534
[cleanup] Remove useless includes.
by Yves Gerey
· 6 years ago
78cdde3
Add support for sending RTP two-byte header extensions.
by Johannes Kron
· 6 years ago
07ba2b9
Parse two-byte header extensions.
by Johannes Kron
· 6 years ago
f3119ef
Add generic frame descriptor to list of supported video extensions
by Danil Chapovalov
· 6 years ago
e0c8b23
Frame marking RTP header extension (PART 1: implement extension)
by Johnny Lee
· 6 years ago
2ffed6d
Enable clang::find_bad_constructs for sdk/android (part 1/2).
by Mirko Bonadei
· 6 years ago
dacec71
Add Rtcp parameters for PeerConnection senders
by Florent Castelli
· 7 years ago
866d6dc
Remove the remaining non-test stringstreams from api/
by Jonas Olsson
· 7 years ago
bb50ce5
Wire up MID send value to the PeerConnection API
by Steve Anton
· 7 years ago
f32795e
Updates to video config to allow changes in google3 tests, in order to not break anything.
by Seth Hampson
· 7 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago
[Renamed from webrtc/api/rtpparameters.cc]
1acbd68
Move RtpExtension to api/ directory and config.h/.cc to call/.
by Stefan Holmer
· 7 years ago