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gerrit-public.fairphone.software
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platform
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external
/
webrtc
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02ee47c1ae2a7beecae36a886994d82390eb3d57
/
pc
/
rtpsenderreceiver_unittest.cc
02ee47c
Signal track ID correctly when Unified Plan semantics selected
by Steve Anton
· 7 years ago
6077675
Change RtpReceivers to interact with the media channel directly
by Steve Anton
· 7 years ago
24722b3
Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator."
by Seth Hampson
· 7 years ago
8b77aea
Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator."
by Lu Liu
· 7 years ago
d2b912a
Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator.
by Seth Hampson
· 7 years ago
9e6fd2b
Add streams() to RtpReceiverInterface and implementations.
by Henrik Boström
· 7 years ago
36f8f3e
Optional: Use nullopt and implicit construction in /pc
by Oskar Sundbom
· 7 years ago
c61ce0d
Fixing some clang-tidy findings.
by Mirko Bonadei
· 7 years ago
c9e1560
Modernize and cleanup ChannelManager
by Steve Anton
· 7 years ago
b526158
Move the TransportController from p2p/base to pc/.
by Zhi Huang
· 7 years ago
563934e
Clean up dependencies of peerconnection_unittest.
by Patrik Höglund
· 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/pc/rtpsenderreceiver_unittest.cc]
8ffb9c3
Change RtpSender to have multiple stream_ids
by Steve Anton
· 7 years ago
773be36
Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt
by perkj
· 7 years ago
539d104
Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ )
by mbonadei
· 7 years ago
f1377f7
Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread.
by perkj
· 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
eaabdf6
Delete MediaController class, move Call ownership to PeerConnection.
by nisse
· 7 years ago
e814a0d
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
by deadbeef
· 7 years ago
1a2183d
Removing unnecessary parameters from CreateXChannel methods.
by deadbeef
· 8 years ago
757146b
Remove PC factory options param from LocalAudioSource::Create.
by deadbeef
· 8 years ago
112b2e9
Switching some interfaces to use std::unique_ptr<>.
by deadbeef
· 8 years ago
e702b30
Adding C++ versions of currently spec'd "RtpParameters" structs.
by deadbeef
· 8 years ago
20cb0c1
Move DTMF sender to RtpSender (as opposed to WebRtcSession).
by deadbeef
· 8 years ago
7bb87ee
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
by ossu
· 8 years ago
[Renamed (98%) from webrtc/api/rtpsenderreceiver_unittest.cc]
b2cdd93
Remove the dependency of TransportChannel and TransportChannelImpl.
by zhihuang
· 8 years ago
6ce9259
Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ )
by zhihuang
· 8 years ago
5aed06c
make the DtlsTransportWrapper inherit form DtlsTransportInternal
by zhihuang
· 8 years ago
ac22f70
Refactoring of RTCP options in BaseChannel.
by deadbeef
· 8 years ago
f5b251b
Remove BaseChannel's dependency on TransportController.
by zhihuang
· 8 years ago
5214a0a
Add support for content hints to VideoTrack.
by pbos
· 8 years ago
7af91dd
Removing "crypto_required" from MediaContentDescription.
by deadbeef
· 8 years ago
11a9cbf
Refactoring: move ownership of RtcEventLog from Call to PeerConnection
by skvlad
· 8 years ago
ac9f876
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
by kwiberg
· 8 years ago
77eab70
Enable the -Wundef warning for clang
by kwiberg
· 8 years ago
ba29c6a
Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
by Taylor Brandstetter
· 8 years ago
3784b4a
Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ )
by tkchin
· 8 years ago
2d54917
Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
by Taylor Brandstetter
· 8 years ago
1a7162d
Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ )
by deadbeef
· 8 years ago
bc58319
Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
by Taylor Brandstetter
· 8 years ago
5a4a75a
Combining SetVideoSend and SetSource into one method.
by deadbeef
· 8 years ago
db0cd9e
Adding getParameters/setParameters APIs to RtpReceiver.
by Taylor Brandstetter
· 8 years ago
5dd42fd
Fixing a segfault that can occur when changing the track of an RtpSender.
by deadbeef
· 8 years ago
ef8b61e
Enable -Winconsistent-missing-override flag.
by nisse
· 8 years ago
d1fe281
Replace scoped_ptr with unique_ptr in webrtc/api/
by kwiberg
· 8 years ago
2ded9b1
Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value.
by nisse
· 8 years ago
d61bf80
Removed MediaStreamTrackInterface::set_state
by perkj
· 8 years ago
af510af
Use a FakeVideoTrackSource instead of nullptr in all VideoTrack tests.
by nisse
· 8 years ago
dc1c62c
Enable setting the maximum bitrate limit in RtpSender.
by skvlad
· 8 years ago
f0dcfe2
Change VideoRtpReceiver to create remote VideoTrack and VideoTrackSource.
by perkj
· 8 years ago
0d3eef2
Add implementation of VideoTrackSource and make VideoCapturerTrackSource inherit from it.
by perkj
· 8 years ago
1a018dc
Prevent a voice channel from sending data before a source is set.
by Taylor Brandstetter
· 8 years ago
a3ede6c
Renamed VideoSourceInterface to VideoTrackSourceInterface.
by perkj
· 8 years ago
f2880a0
Change webrtc::VideoSourceInterface to inherit rtc::VideoSourceInterface.
by perkj
· 8 years ago
b24317b
Fix license headers in webrtc/api.
by kjellander
· 9 years ago
15583c1
Move talk/app/webrtc to webrtc/api
by Henrik Kjellander
· 9 years ago
[Renamed (97%) from talk/app/webrtc/rtpsenderreceiver_unittest.cc]
8e8908a
Delete FrameInput method and FrameInputWrapper class.
by nisse
· 9 years ago
a96e2d7
Move talk/media to webrtc/media
by kjellander
· 9 years ago
08582ff
Replace uses of cricket::VideoRenderer by rtc::VideoSinkInterface.
by nisse
· 9 years ago
e73afba
New rtc::VideoSinkInterface.
by nisse
· 9 years ago
2098fca
Revert of New rtc::VideoSinkInterface. (patchset #7 id:120001 of https://codereview.webrtc.org/1594973006/ )
by nisse
· 9 years ago
a862d45
New rtc::VideoSinkInterface.
by Niels Möller
· 9 years ago
2d110be
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
by deadbeef
· 9 years ago
e591f93
Storing raw audio sink for default audio track.
by deadbeef
· 9 years ago
6eca7e3
Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :(
by tommi
· 9 years ago
f888bb5
Support for unmixed remote audio into tracks.
by Tommi
· 9 years ago
fac0655
Reland of Adding the ability to create an RtpSender without a track.
by deadbeef
· 9 years ago
5def7b9
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
by deadbeef
· 9 years ago
6834fa1
Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ )
by deadbeef
· 9 years ago
8f46c63
Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
by deadbeef
· 9 years ago
ac9d92c
Adding the ability to create an RtpSender without a track.
by deadbeef
· 9 years ago
d4cec0d
Remove MediaChannel::SetRemoteRenderer().
by solenberg
· 9 years ago
0c4e06b
Use suffixed {uint,int}{8,16,32,64}_t types.
by Peter Boström
· 9 years ago
70ab1a1
Exposing RtpSenders and RtpReceivers from PeerConnection.
by deadbeef
· 9 years ago