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gerrit-public.fairphone.software
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platform
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external
/
webrtc
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2ae140ae7e34bd1e102ca832a14b7b0adbb72d15
/
pc
/
peerconnection_integrationtest.cc
8e545ee
Revert "Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32."
by Tommi
· 7 years ago
6780c51
Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32.
by Joachim Bauch
· 7 years ago
70b820f
Implemented the GetRemoteAudioSSLCertificate method.
by Zhi Huang
· 7 years ago
74255ff
Add PeerConnection interop integration tests
by Steve Anton
· 7 years ago
194939b
Added UMA counters for SDES vs DTLS key agreement
by Harald Alvestrand
· 7 years ago
d367921
Configure media flow correctly with Unified Plan
by Steve Anton
· 7 years ago
1532477
Convert PeerConnection integration tests to the track-based API
by Steve Anton
· 7 years ago
389a97c
Fixing leaked reference from SCTP transport to DTLS/ICE transport.
by Taylor Brandstetter
· 7 years ago
b1c1de1
Use the SDP ContentInfo helpers to avoid downcasting
by Steve Anton
· 7 years ago
4ab68ee
Move sessiondescription.h/cc from p2p/base to pc/
by Steve Anton
· 7 years ago
a3a92c2
Replace string type with SdpType enum
by Steve Anton
· 7 years ago
c61ce0d
Fixing some clang-tidy findings.
by Mirko Bonadei
· 7 years ago
de93943
Revert "Revert "Encode log events periodically instead of for every event.""
by Bjorn Terelius
· 7 years ago
83119dd
Fix and re-enable flaky PeerConnectionIntegrationTests
by Steve Anton
· 7 years ago
675513b
Stop using LOG macros in favor of RTC_ prefixed macros.
by Mirko Bonadei
· 7 years ago
1c9faee
Disable several flaky PeerConnectionIntegration tests.
by Ivo Creusen
· 7 years ago
ff52f1b
Fix flake in AddMediaToConnectedBundleDoesNotRestartIce test
by Steve Anton
· 7 years ago
4f167df
Adds new DisableAndEnableAudioRecording integration test to Peerconnection.
by henrika
· 7 years ago
5f6bf24
Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II)
by henrika
· 7 years ago
990d6b8
Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API"
by Mirko Bonadei
· 7 years ago
90bace0
Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
by henrika
· 7 years ago
074dece
Fix flaky DataChannel integration test
by Steve Anton
· 7 years ago
da6c095
Rewrite WebRtcSession data channel tests as PeerConnection tests
by Steve Anton
· 7 years ago
6f25b09
Reland "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests"
by Steve Anton
· 7 years ago
b49b661
Revert "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests"
by Olga Sharonova
· 7 years ago
096e367
Rewrite WebRtcSession BUNDLE tests as PeerConnection tests
by Steve Anton
· 7 years ago
1b0eae3
Don't call deprecated CreatePeerConnectionFactory() overloads
by Karl Wiberg
· 7 years ago
ede9ca5
Rewrite WebRtcSession ICE integration tests as PeerConnection tests
by Steve Anton
· 7 years ago
99c3fe5
Add PeerConnection::StartRtcEventLog version that takes RtcEventLogOutput as parameter
by Elad Alon
· 7 years ago
bdcee28
TurnCustomizer - an interface for modifying stun messages sent by TurnPort
by Jonas Oreland
· 7 years ago
604427b
Revert "TurnCustomizer - an interface for modifying stun messages sent by TurnPort"
by Guido Urdaneta
· 7 years ago
b23ed7f
TurnCustomizer - an interface for modifying stun messages sent by TurnPort
by Jonas Oreland
· 7 years ago
8c0f7a7
Add GetRemoteAudioSSLCertificate() to PeerConnection
by Steve Anton
· 7 years ago
94286cb
Add base fixture and PeerConnection wrapper for unit tests
by Steve Anton
· 7 years ago
4e2deab
Only return stats for the most recent unsignaled audio stream.
by deadbeef
· 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/peerconnection_integrationtest.cc]
73276ad
- Removes voe_conference_test.
by Fredrik Solenberg
· 7 years ago
b1a15d7
In PC integration tests, create tracks/streams with random IDs.
by deadbeef
· 7 years ago
4389b4d
Add a PeerConnection integration test for adding an audio track mid-call
by deadbeef
· 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
8b7e9ad
Support "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP" profile strings.
by deadbeef
· 7 years ago
f816493
Add media related stats (audio level etc.) to unsignaled streams.
by zhihuang
· 7 years ago
98e186c
Remove VirtualSocketServer's dependency on PhysicalSocketServer.
by deadbeef
· 7 years ago
7145280
Unflaking PeerConnectionIntegrationTest.DtmfSenderObserver.
by deadbeef
· 7 years ago
7914b8c
Negotiate the same SRTP crypto suites for every DTLS association formed.
by deadbeef
· 7 years ago
30952b4
Add "ice-option:trickle" to generated offers/answers.
by deadbeef
· 7 years ago
d8ad788
Adding integration test for unsignaled inbound RTP stream stats.
by deadbeef
· 7 years ago
2f425aa
Fix SDP stream ID mismatch issue when a track's stream changes.
by deadbeef
· 7 years ago
8d609f6
Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
by hbos
· 7 years ago
fbcc5cb
Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
by olka
· 7 years ago
292084c
Added the GetSources() to the RtpReceiverInterface and implemented
by zhihuang
· 7 years ago
c964d0b
Fixing some case-sensitive codec name comparisons.
by deadbeef
· 7 years ago
1dcb164
Rewrite PeerConnection integration tests using better testing practices.
by deadbeef
· 7 years ago