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gerrit-public.fairphone.software
/
fp2-dev
/
platform
/
external
/
chromium_org
/
third_party
/
webrtc
/
f89ce467d1977ea1d867be4e5be2ccd9a206113c
/
video
/
video_send_stream.cc
f89ce46
Implements start bitrate for new video API.
by mflodman@webrtc.org
· 10 years ago
19f89a1
Enable pacing by default and remove the option to disable it from the new API.
by stefan@webrtc.org
· 10 years ago
6845de7
Add APIs to enable padding with redundant payloads.
by stefan@webrtc.org
· 10 years ago
bdfcddf
Make VideoSendStream/VideoReceiveStream configs const.
by pbos@webrtc.org
· 10 years ago
903e746
Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC.
by stefan@webrtc.org
· 10 years ago
c78d40d
Revert "Revert "Remove VideoSendStreamInput::PutFrame.""
by pbos@webrtc.org
· 10 years ago
f3f2dba
Revert "Remove VideoSendStreamInput::PutFrame."
by pbos@webrtc.org
· 10 years ago
0c60ee8
Remove VideoSendStreamInput::PutFrame.
by pbos@webrtc.org
· 10 years ago
66b1bf8
Fix Win VideoSendStream::...::ToString() compiles.
by pbos@webrtc.org
· 10 years ago
7e68693
Add ToString() to VideoSendStream::Config.
by pbos@webrtc.org
· 10 years ago
c476e64
Add thread annotations to Call API.
by pbos@webrtc.org
· 10 years ago
16a058a
Rename Start/Stop in Video{Send,Receive}Streams.
by pbos@webrtc.org
· 10 years ago
1d61e3a
Simplify pacer interface.
by pbos@webrtc.org
· 10 years ago
e2a7a77
Remove internal codecs from VideoSendStream.
by pbos@webrtc.org
· 10 years ago
3f83f9c
Implement minimum transmit bitrate.
by pbos@webrtc.org
· 10 years ago
c766098
Adding a new ramp-up-down-up test
by henrik.lundin@webrtc.org
· 10 years ago
8ef6548
Add configuration for cpu overuse detection to video send stream.
by asapersson@webrtc.org
· 11 years ago
0a29815
Drop early packets when not sending in TransportAdapter.
by sprang@webrtc.org
· 11 years ago
c71929d
Wire up RTX in VideoReceiveStream.
by pbos@webrtc.org
· 11 years ago
75e7da3
Permitting double start/stopping of streams.
by pbos@webrtc.org
· 11 years ago
49812e6
Wire up statistics in video send stream of new video engine api
by sprang@webrtc.org
· 11 years ago
46f7288
Revert r5294 to re-roll r5293.
by pbos@webrtc.org
· 11 years ago
c5a5713
Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."
by turaj@webrtc.org
· 11 years ago
3bbc91e
Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
by solenberg@webrtc.org
· 11 years ago
7ff4089
Adding REMB to receive stream configuration, the send side will always
by mflodman@webrtc.org
· 11 years ago
dadfc9e
Make sure channels in the same call are in the same channel group.
by mflodman@webrtc.org
· 11 years ago
c33d37c
Add SwapFrame() to VideoSendStreamInput.
by pbos@webrtc.org
· 11 years ago
66f4394
Set up SSRCs correctly after switching codec.
by pbos@webrtc.org
· 11 years ago
2e98d45
Implement and test EncodedImageCallback in new ViE API.
by sprang@webrtc.org
· 11 years ago
b9f1eb8
Connect pacer/padding to SuspendBelowMinBitrate
by henrik.lundin@webrtc.org
· 11 years ago
346dbe7
Rename RTP-extension constants.
by pbos@webrtc.org
· 11 years ago
7f9f840
Rename video streams' start/stop methods.
by pbos@webrtc.org
· 11 years ago
4590177
Rename AutoMute to SuspendBelowMinBitrate
by henrik.lundin@webrtc.org
· 11 years ago
6904054
Implement VideoSendStream::SetCodec().
by pbos@webrtc.org
· 11 years ago
24e2089
Separate Call API/build files from video_engine/.
by pbos@webrtc.org
· 11 years ago
[Renamed (93%) from video_engine/internal/video_send_stream.cc]
4ce7590
Removing the threshold from the auto-mute APIs
by henrik.lundin@webrtc.org
· 11 years ago
6036f56
Porting auto mute to new ViE API
by henrik.lundin@webrtc.org
· 11 years ago
63301bd
Implement I420FrameCallbacks in Call.
by pbos@webrtc.org
· 11 years ago
44bb62a
Fixed issue with how MTU is calculated.
by sprang@webrtc.org
· 11 years ago
6c9c551
Wired up max packet size and added simple test.
by sprang@webrtc.org
· 11 years ago
0011252
Enable FEC for VideoSendStream.
by pbos@webrtc.org
· 11 years ago
1ddd57f
Break out glue for old->new Transport.
by pbos@webrtc.org
· 11 years ago
041d54b
Implement NACK over RTX for VideoSendStream.
by pbos@webrtc.org
· 11 years ago
bfad17e
Implement 'abs-send-time' extension in VideoSendStream.
by pbos@webrtc.org
· 11 years ago
990c5e3
Implement 'toffset' extension in VideoSendStream.
by pbos@webrtc.org
· 11 years ago
c179706
Remove newapi:: namespace for typenames without overlap.
by pbos@webrtc.org
· 11 years ago
206c4a5
Enabling and testing RTCP CNAME in new API.
by pbos@webrtc.org
· 11 years ago
55afdbe
Adds two tests for verifying padding and ramp-up behavior.
by stefan@webrtc.org
· 11 years ago
043f6a8
Fix implicit int->bool conversion in VideoSendStream::DeliverRtcp.
by pbos@webrtc.org
· 11 years ago
ce85109
Glue code and tests for NACK in new VideoEngine API.
by pbos@webrtc.org
· 11 years ago
bf76ae2
Hooking up first simple CPU adaptation version.
by mflodman@webrtc.org
· 11 years ago
2f02da8
Initial port of FullStackTest to new VideoEngine API.
by pbos@webrtc.org
· 11 years ago
6f1c3ef
Stats+Config moved into VideoSend/ReceiveStreams.
by pbos@webrtc.org
· 11 years ago
08f3ca9
FrameGenerator class for future fake capture device.
by pbos@webrtc.org
· 11 years ago
2a9108f
New VideoEngine API implementation on top of old one, first steps.
by pbos@webrtc.org
· 11 years ago