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gerrit-public.fairphone.software
/
fp2-dev
/
platform
/
external
/
chromium_org
/
third_party
/
webrtc
/
5b4a29777dab756293048ea4de1a37dbafab5c79
/
video_engine
/
vie_rtp_rtcp_impl.h
49e6306
Preserve RTP states for restarted VideoSendStreams.
by pbos@webrtc.org
· 10 years ago
508e748
Add APIs to enable padding with redundant payloads.
by stefan@webrtc.org
· 10 years ago
37f807f
Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams.
by solenberg@webrtc.org
· 10 years ago
9420a1f
Implement minimum transmit bitrate.
by pbos@webrtc.org
· 10 years ago
ecee063
Adds APIs for reporting pacer queuing delay.
by jiayl@webrtc.org
· 10 years ago
2fa9f7e
Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR).
by asapersson@webrtc.org
· 10 years ago
a68379b
Add stats of incoming frame delays for debugging bandwidth estimation.
by jiayl@webrtc.org
· 10 years ago
3fe2e7f
Added API for enabling/disabling RTCP Receiver Reference Time extension.
by asapersson@webrtc.org
· 11 years ago
13a4d31
Interface changes to old api, for use by new api transition.
by sprang@webrtc.org
· 11 years ago
967320b
Add functions to ViE API to enable/disable the absolute send time header extension.
by solenberg@webrtc.org
· 11 years ago
f2e6fb3
Include files from webrtc/.. paths in video_engine/
by pbos@webrtc.org
· 11 years ago
864f9d7
Remove SetOverUseDetectorOptions and cleaned ViESharedData.
by mflodman@webrtc.org
· 11 years ago
2b2e78c
Adding a factory to remote bitrate estimator and allow it to be set via config.
by andresp@webrtc.org
· 11 years ago
06077c9
Adding a payload type for RTX.
by mflodman@webrtc.org
· 11 years ago
4db69af
Adding a receive side API for buffering mode.
by mikhal@webrtc.org
· 11 years ago
d6739c8
Adding a send side API for streaming
by mikhal@webrtc.org
· 11 years ago
a7b57da
Move src/ -> webrtc/
by andrew@webrtc.org
· 12 years ago