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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
34029e209d6d116c338e7d65d6980cc395184658
/
call
/
call.h
78609d5
Reland of BWE allocation strategy
by Alex Narest
· 7 years ago
dc9ca93
Revert "BWE allocation strategy"
by Alex Narest
· 7 years ago
a5fbc23
BWE allocation strategy
by Alex Narest
· 7 years ago
39260c4
Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic."
by Lu Liu
· 7 years ago
54d1da1
BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic.
by Alex Narest
· 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/call.h]
440b6d9
Move video send/receive stream headers to webrtc/call.
by aleloi
· 7 years ago
db2a9fc
Wire up RTP keep-alive in ortc api.
by sprang
· 7 years ago
e5c4a81
Move RTP keep-alive config from VideoSendStream::Config to Call::Config
by sprang
· 7 years ago
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 7 years ago
a80c16a
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 7 years ago
c3771cc
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 7 years ago
38ede13
Support building WebRTC without audio and video.
by zhihuang
· 7 years ago
a5e0df6
Move MinPositive to call.h as discussed here: https://codereview.chromium.org/2888303005/#msg19
by zstein
· 7 years ago
4b97980
Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
by zstein
· 7 years ago
441718e
Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ )
by charujain
· 7 years ago
9641c13
Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.
by zstein
· 7 years ago
7cb69d5
This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008).
by zstein
· 7 years ago
b8a654c
Delete declaration of non-existing function webrtc::Version().
by nisse
· 7 years ago
fb45c6c
Inform jitter buffer about FlexFEC protection.
by brandtr
· 7 years ago
7250b39
Move FlexfecReceiveStream from api/call/ to call/.
by brandtr
· 8 years ago
446fcb6
Clean up FlexfecReceiveStream ctor signatures.
by brandtr
· 8 years ago
f515ab8
Moved call.h and most of api/call/* into call/
by ossu
· 8 years ago