Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
6753795409a242d2cac9288e2d62256cd111f07b
/
pc
/
peerconnectioninterface_unittest.cc
« Previous
42a4263
Making candidate pool size behave as decided in JSEP.
by deadbeef
· 8 years ago
7f06766
Delete deprecated PeerConnection methods, and corresponding using declarations.
by nisse
· 8 years ago
6038e97
Adding RTCErrorOr class to be used by ORTC APIs.
by deadbeef
· 8 years ago
112b2e9
Switching some interfaces to use std::unique_ptr<>.
by deadbeef
· 8 years ago
087bd34
Move AudioDecoder and related stuff to the api/ directory
by kwiberg
· 8 years ago
7798501
Fix the Chrome crash caused by RtcEventLog
by zhihuang
· 8 years ago
d1f5fda
Allow changing the minimal ICE ping timeout with PeerConnection.SetConfiguration.
by skvlad
· 8 years ago
63b14b7
Add override declarations to PeerConnectionObserver subclasses, and delete obsolete methods.
by nisse
· 8 years ago
1e4e8cb
Add CreatePeerConnectionFactory overloads that take audio codec factory args
by kwiberg
· 8 years ago
f534659
Adding ability for BaseChannel to use PacketTransportInterface.
by deadbeef
· 8 years ago
1b54a5f
Relanding: Removing #defines previously used for building without BoringSSL/OpenSSL.
by deadbeef
· 8 years ago
7bb87ee
Create //webrtc/api:libjingle_peerconnection_api + refactorings.
by ossu
· 8 years ago
[Renamed (99%) from webrtc/api/peerconnectioninterface_unittest.cc]
8662f94
Only set certificate on DTLS transport if fingerprint is found in SDP.
by deadbeef
· 8 years ago
f33491e
Revert of Removing #defines previously used for building without BoringSSL/OpenSSL. (patchset #2 id:20001 of https://codereview.webrtc.org/2640513002/ )
by deadbeef
· 8 years ago
eaa826c
Removing #defines previously used for building without BoringSSL/OpenSSL.
by deadbeef
· 8 years ago
293e926
Reland of: Adding error output param to SetConfiguration, using new RTCError type.
by deadbeef
· 8 years ago
953c2ce
Reland of: Separating SCTP code from BaseChannel/MediaChannel.
by deadbeef
· 8 years ago
c0dad89
Revert of Separating SCTP code from BaseChannel/MediaChannel. (patchset #14 id:240001 of https://codereview.webrtc.org/2564333002/ )
by deadbeef
· 8 years ago
67b3bbe
Separating SCTP code from BaseChannel/MediaChannel.
by deadbeef
· 8 years ago
1e23461
Revert of Adding error output param to SetConfiguration, using new RTCError type. (patchset #4 id:60001 of https://codereview.webrtc.org/2587133004/ )
by deadbeef
· 8 years ago
7a5fa6c
Adding error output param to SetConfiguration, using new RTCError type.
by deadbeef
· 8 years ago
fe4a8a4
Implement current/pending session description methods.
by deadbeef
· 8 years ago
6de92f9
Don't allow changing ICE pool size after SetLocalDescription.
by deadbeef
· 8 years ago
d1a38b5
Implement the "needs-ice-restart" logic for SetConfiguration.
by deadbeef
· 8 years ago
3edec7c
Adding error enum to be used by PeerConnectionInterface methods.
by deadbeef
· 8 years ago
c63b894
Modify the parameter type of PeerConnectionObserver callback OnAddTrack.
by zhihuang
· 8 years ago
4dfb8ce
Make the default value of rtcp-mux policy to required.
by zhihuang
· 8 years ago
81c3a03
Added a callback function OnAddTrack to PeerConnectionObserver
by zhihuang
· 8 years ago
46c7389
Adding GetConfiguration to PeerConnection.
by deadbeef
· 8 years ago
71a1b61
WebRTC: Fix and enable -Woverloaded-virtual warnings.
by kjellander
· 8 years ago
e9e94c3
Return false if PeerConnection::GetStats() is called on invalid tracks
by zhihuang
· 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
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
· 8 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
· 8 years ago
a6bdb09
Add virtual Initialize methods to PortAllocator and NetworkManager.
by Taylor Brandstetter
· 8 years ago
a601f5c
Separating internal and external methods of RtpSender/RtpReceiver.
by deadbeef
· 8 years ago
d79599d
Turning FakeDtlsIdentityStore into FakeRTCCertificateGenerator.
by Henrik Boström
· 8 years ago
d03c23b
Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface.
by Henrik Boström
· 8 years ago
6034705
Add a flag to filter out high-cost networks.
by honghaiz
· 8 years ago
98cde26
Use scoped_refptr for On(Add|Remove)Stream and OnDataChannel.
by Taylor Brandstetter
· 8 years ago
d7973cc
Revert of Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. (patchset #2 id:20001 of https://codereview.webrtc.org/2013523002/ )
by hbos
· 8 years ago
400781a
Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface.
by Henrik Boström
· 8 years ago
417eebe
Fixing the behavior of the candidate filter with pooled candidates.
by Taylor Brandstetter
· 8 years ago
e9021a3
Propogate network-worker thread split to api
by danilchap
· 8 years ago
fd8be34
Remove webrtc/base/scoped_ptr.h
by kwiberg
· 8 years ago
a1c3035
Relanding: Implement RTCConfiguration.iceCandidatePoolSize.
by Taylor Brandstetter
· 8 years ago
c55fb30
Revert of Implement RTCConfiguration.iceCandidatePoolSize. (patchset #7 id:120001 of https://codereview.webrtc.org/1956453003/ )
by deadbeef
· 8 years ago
48e9d05
Implement RTCConfiguration.iceCandidatePoolSize.
by Taylor Brandstetter
· 8 years ago
dc4eb8c
Refactoring some tests in peerconnectioninterface_unittest.cc.
by Taylor Brandstetter
· 8 years ago
3fe372d
Fix all -Wnon-virtual-dtor warnings.
by Henrik Kjellander
· 8 years ago
6ab3db2
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
by kwiberg
· 8 years ago
65fc62e
Remove webrtc/base/scoped_ptr.h
by kwiberg
· 8 years ago
8f65cdf
Only generate one CNAME per PeerConnection.
by zhihuang
· 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
7ff1737
Re-enabling tests that were disabled for Windows debug builds.
by Taylor Brandstetter
· 8 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
· 8 years ago
d45b95c
Making new unit test assertions use the standard timeout.
by Taylor Brandstetter
· 8 years ago
85e46a8
Fix PeerConnectionInterfaceTest.CloseAndTestStreamsAndStates
by Per
· 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
eec21bd
Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
by jbauch
· 8 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
· 8 years ago
944c390
Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
by jbauch
· 8 years ago
102362b
Truly disable tests.
by Stefan Holmer
· 8 years ago
55d6e7c
Disable tests due to issue 5659.
by Stefan Holmer
· 8 years ago
2bbff99
Helpers in peer connection unit tests: Use scoped_ptr instead of raw pointers
by kwiberg
· 8 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
f475d36
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
Next »