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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
c6ce9c59385102e766913dcfc5eb69cb5cf135ba
/
call
/
video_receive_stream.h
c6ce9c5
New file api/video/BUILD.gn
by Niels Möller
· 6 years ago
be214a2
Move videosinkinterface.h to common_video to solve a circular dep.
by Patrik Höglund
· 7 years ago
3e11343
Fix circular dependencies in webrtc_common.
by Patrik Höglund
· 7 years ago
3b3622f
Delete member VideoReceiveStream::Config::Rtp::ulpfec.
by nisse
· 7 years ago
7120742
Adding NOLINT for typedefs.h and common_types.h
by Mirko Bonadei
· 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/call/video_receive_stream.h]
ca5706d
Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ )
by nisse
· 7 years ago
8e7eee0
Revert of Use RtxReceiveStream. (patchset #5 id:320001 of https://codereview.webrtc.org/3006063002/ )
by nisse
· 7 years ago
35713ea
Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3010983002/ )
by nisse
· 7 years ago
2e1b40b
Implement googContentType GetStats metric reported on receive side.
by ilnik
· 7 years ago
3c39c01
Revert of Use RtxReceiveStream. (patchset #5 id:80001 of https://codereview.webrtc.org/3008773002/ )
by nisse
· 7 years ago
75204c5
Change reporting of timing frames conditions in GetStats on receive side
by ilnik
· 7 years ago
5c0f6c6
Use RtxReceiveStream.
by nisse
· 7 years ago
1acbd68
Move RtpExtension to api/ directory and config.h/.cc to call/.
by Stefan Holmer
· 7 years ago
26e3abb
Reverse |rtx_payload_types| map, and rename.
by nisse
· 7 years ago
23bdb67
New accessor function VideoReceiveStream::Config::Rtp::AddRtxBinding
by Niels Möller
· 7 years ago
a79cc28
Report max interframe delay over window insdead of interframe delay sum
by ilnik
· 7 years ago
440b6d9
Move video send/receive stream headers to webrtc/call.
by aleloi
· 7 years ago