Sign in
webrtc
/
src
/
bfd398ccda27550629ec2440888f4083e4510069
/
webrtc
/
api
/
peerconnectioninterface_unittest.cc
bfd398c
Add a switch to redetermine role when ICE restarts.
by Honghai Zhang
· 8 years ago
9763d56
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
by zhihuang
· 8 years ago
907abe4
Revert of Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage. (patchset #8 id:280001 of https://codereview.webrtc.org/2166873002/ )
by deadbeef
· 8 years ago
34b54c3
Modified PeerConnection and WebRtcSession for end-to-end QuicDataChannel usage.
by zhihuang
· 8 years ago
29ff844
Add PeerConnection IsClosed check.
by zhihuang
· 8 years ago
f8e6577
Add virtual Initialize methods to PortAllocator and NetworkManager.
by Taylor Brandstetter
· 9 years ago
ba8d433
Revert of Add virtual Initialize methods to PortAllocator and NetworkManager. (patchset #4 id:60001 of https://codereview.webrtc.org/2097653002/ )
by deadbeef
· 9 years ago
a6bdb09
Add virtual Initialize methods to PortAllocator and NetworkManager.
by Taylor Brandstetter
· 9 years ago
a601f5c
Separating internal and external methods of RtpSender/RtpReceiver.
by deadbeef
· 9 years ago
d79599d
Turning FakeDtlsIdentityStore into FakeRTCCertificateGenerator.
by Henrik Boström
· 9 years ago
d03c23b
Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface.
by Henrik Boström
· 9 years ago
6034705
Add a flag to filter out high-cost networks.
by honghaiz
· 9 years ago
98cde26
Use scoped_refptr for On(Add|Remove)Stream and OnDataChannel.
by Taylor Brandstetter
· 9 years ago
d7973cc
Revert of Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. (patchset #2 id:20001 of https://codereview.webrtc.org/2013523002/ )
by hbos
· 9 years ago
400781a
Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface.
by Henrik Boström
· 9 years ago
417eebe
Fixing the behavior of the candidate filter with pooled candidates.
by Taylor Brandstetter
· 9 years ago
e9021a3
Propogate network-worker thread split to api
by danilchap
· 9 years ago
fd8be34
Remove webrtc/base/scoped_ptr.h
by kwiberg
· 9 years ago
a1c3035
Relanding: Implement RTCConfiguration.iceCandidatePoolSize.
by Taylor Brandstetter
· 9 years ago
c55fb30
Revert of Implement RTCConfiguration.iceCandidatePoolSize. (patchset #7 id:120001 of https://codereview.webrtc.org/1956453003/ )
by deadbeef
· 9 years ago
48e9d05
Implement RTCConfiguration.iceCandidatePoolSize.
by Taylor Brandstetter
· 9 years ago
dc4eb8c
Refactoring some tests in peerconnectioninterface_unittest.cc.
by Taylor Brandstetter
· 9 years ago
3fe372d
Fix all -Wnon-virtual-dtor warnings.
by Henrik Kjellander
· 9 years ago
6ab3db2
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
by kwiberg
· 9 years ago
65fc62e
Remove webrtc/base/scoped_ptr.h
by kwiberg
· 9 years ago
8f65cdf
Only generate one CNAME per PeerConnection.
by zhihuang
· 9 years ago
ef8b61e
Enable -Winconsistent-missing-override flag.
by nisse
· 9 years ago
d1fe281
Replace scoped_ptr with unique_ptr in webrtc/api/
by kwiberg
· 9 years ago
7ff1737
Re-enabling tests that were disabled for Windows debug builds.
by Taylor Brandstetter
· 9 years ago
71bdda0
Add RTCConfiguration getter and setter methods. The immediate plan is to move some flags into an embedded MediaConfig (https://codereview.webrtc.org/1818033002/), which will be possible after Chrome is updated to use these new setter methods.
by Niels Möller
· 9 years ago
d45b95c
Making new unit test assertions use the standard timeout.
by Taylor Brandstetter
· 9 years ago
85e46a8
Fix PeerConnectionInterfaceTest.CloseAndTestStreamsAndStates
by Per
· 9 years ago
d61bf80
Removed MediaStreamTrackInterface::set_state
by perkj
· 9 years ago
af510af
Use a FakeVideoTrackSource instead of nullptr in all VideoTrack tests.
by nisse
· 9 years ago
eec21bd
Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
by jbauch
· 9 years ago
194e3bc
Revert of Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. (patchset #4 id:60001 of https://codereview.webrtc.org/1785713005/ )
by kjellander
· 9 years ago
944c390
Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
by jbauch
· 9 years ago
102362b
Truly disable tests.
by Stefan Holmer
· 9 years ago
55d6e7c
Disable tests due to issue 5659.
by Stefan Holmer
· 9 years ago
2bbff99
Helpers in peer connection unit tests: Use scoped_ptr instead of raw pointers
by kwiberg
· 9 years ago
aac2dea
Changed defaults for CreateAnswer in non-constraint mode
by hta
· 9 years ago
a3ede6c
Renamed VideoSourceInterface to VideoTrackSourceInterface.
by perkj
· 9 years ago
a2a49d9
This CL provides interfaces that do not use constraints for
by hta
· 9 years ago
0db023a
Move suspend_below_min_bitrate from VideoOptions to MediaConfig.
by nisse
· 9 years ago
0ed85b2
Track pending ICE restarts independently for different media sections.
by deadbeef
· 9 years ago
51542be
Introduce struct MediaConfig, with construction-time settings.
by nisse
· 9 years ago
9b8df25
Move talk/session/media -> webrtc/pc
by kjellander@webrtc.org
· 9 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 (98%) from talk/app/webrtc/peerconnectioninterface_unittest.cc]
dfb769d
Remove deprecated PeerConnectionObserver::OnStateChange and OnIceComplete
by perkj
· 9 years ago
a96e2d7
Move talk/media to webrtc/media
by kjellander
· 9 years ago
884f585
Storing raw audio sink for default audio track.
by deadbeef
· 9 years ago
e1f9d83
Adding AddTrack/RemoveTrack to native PeerConnection API.
by deadbeef
· 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
f475d365
Properly handle different transports having different SSL roles.
by Taylor Brandstetter
· 9 years ago
37ebcf0
Reland "Add APK targets to build libjingle tests for Android."
by phoglund
· 9 years ago
0c7e9f5
Removing webrtc::PortAllocatorFactoryInterface.
by Taylor Brandstetter
· 9 years ago
bd7d8f7
Adding a MediaStream parameter to createSender.
by deadbeef
· 9 years ago
0eb15ed
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
by kwiberg
· 9 years ago
eb45981
Restoring behavior where PeerConnection tracks changes to MediaStreams.
by deadbeef
· 9 years ago
bc14164
Revert of Add APK targets to build libjingle tests for Android. (patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/ )
by stefan
· 9 years ago
a78c021
Add APK targets to build libjingle_peerconnection_unittests for Android.
by perkj
· 9 years ago
bda7e0b
Fixing issue with default stream upon setting 2nd remote description.
by deadbeef
· 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
653b8e0
Reland of Adding the ability to change ICE servers through SetConfiguration. (patchset #1 id:1 of https://codereview.webrtc.org/1424803004/ )
by deadbeef
· 9 years ago
18a944b
Revert of Adding the ability to change ICE servers through SetConfiguration. (patchset #7 id:120001 of https://codereview.webrtc.org/1391013007/ )
by deadbeef
· 9 years ago
d3b26d9
Adding the ability to change ICE servers through SetConfiguration.
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
c80741f
Fixing some issues with the direction attribute of m-lines in offers.
by deadbeef
· 9 years ago
5e97fb5
Don't create remote streams if m-line direction doesn't include "send".
by deadbeef
· 9 years ago
ab9b2d1
Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ )
by deadbeef
· 9 years ago
fc648b6
Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ )
by deadbeef
· 9 years ago
97c3929
Moving MediaStreamSignaling logic into PeerConnection.
by deadbeef
· 9 years ago
0c4e06b
Use suffixed {uint,int}{8,16,32,64}_t types.
by Peter Boström
· 9 years ago
0a6c4ca
Catching more errors when parsing ICE server URLs.
by deadbeef
· 9 years ago
5e56c59
DtlsIdentityStoreInterface added and the implementation is called DtlsIdentityStoreImpl (previously named without the -Impl bit and without an interface).
by Henrik Boström
· 9 years ago
fabe2c9
Remove deprecated functions.
by jbauch
· 10 years ago
61e00b0
Create a in-memory DTLS identity store that keeps a free identity generated in the background.
by jiayl@webrtc.org
· 10 years ago
be77872
Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background."
by jiayl@webrtc.org
· 10 years ago
369f682
Create a in-memory DTLS identity store that keeps a free identity generated in the background.
by jiayl@webrtc.org
· 10 years ago
8ad9660
Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background."
by jiayl@webrtc.org
· 10 years ago
df512cc
Create a in-memory DTLS identity store that keeps a free identity generated in the background.
by jiayl@webrtc.org
· 10 years ago
005b6ff
Convert some EXPECTs to ASSERTs to avoid crashes when object creation fails.
by pkasting@chromium.org
· 10 years ago
5f93d0a
Update libjingle license statements at top of talk files for consistency
by jlmiller@webrtc.org
· 10 years ago
c2dd5ee
Prepare for removal of PeerConnectionObserver::OnError.
by perkj@webrtc.org
· 10 years ago
34f2a9e
Initialize SSL in unittest_main.cc.
by pbos@webrtc.org
· 10 years ago
a09a999
(Auto)update libjingle 73222930-> 73226398
by buildbot@webrtc.org
· 10 years ago
d4e598d
(Auto)update libjingle 72097588-> 72159069
by buildbot@webrtc.org
· 10 years ago
908f57e
Disable GetStatsForInvalidTrack while I rewrite it.
by tommi@webrtc.org
· 11 years ago
001fd2d
Fire OnRenegotiationNeeded only for the first SCTP DataChannel.
by jiayl@webrtc.org
· 11 years ago
f875f15
(Auto)update libjingle 64709629-> 64813990
by buildbot@webrtc.org
· 11 years ago
61c1b8e
(Auto)update libjingle 64585415-> 64594651
by buildbot@webrtc.org
· 11 years ago
db41b4d
Remove the deprecated GetStats method from PeerConnectionInterface.
by jiayl@webrtc.org
· 11 years ago
67ee6b9
Update talk to 60923971
by mallinath@webrtc.org
· 11 years ago
a576faf
Enable SCTP and use OPENSSL on Anroid and NSS on other platforms.
by jiayl@webrtc.org
· 11 years ago
0dac537
Revert 5447 "Update talk to 60420316."
by mallinath@webrtc.org
· 11 years ago
Next »