Log - 2ae140ae7e34bd1e102ca832a14b7b0adbb72d15 - platform/external/webrtc

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