Log - baaf911c80e15b51ed7cfa52b873ba8c0049a883 - platform/external/webrtc

10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from pc/peerconnectioninterface_unittest.cc]
b8b3c99 Clean up visibility and dependencies of RTC event log build targets. by Bjorn Terelius · 6 years ago
ab64e8a Using fully qualified names for using declarations. by Mirko Bonadei · 6 years ago
3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
2ff3f49 Move webrtc::CreatePeerConnectionFactory definition next to decl. by Mirko Bonadei · 6 years ago
37227be Add check for media transport and bundle policy by Piotr (Peter) Slatala · 6 years ago
89f874e Add offer_extmap_allow_mixed to RTCConfiguration by Johannes Kron · 6 years ago
a9bbd86 Add a configuration parameter for using the media transport for data channels. by Bjorn Mellem · 6 years ago
4e93329 Properly setup MockPeerConnectionObserver in tests (continued). by Yves Gerey · 6 years ago
59844ce Revert "Use the factory instead of using the builtin code path in `VideoCodecInitializer`." by Qingsi Wang · 6 years ago
be14217 Use the factory instead of using the builtin code path in `VideoCodecInitializer`. by Jiawei Ou · 6 years ago
f3ff14c Properly setup MockPeerConnectionObserver in tests. by Yves Gerey · 6 years ago
aa1e7c2 Allow 'use_media_transport' to be modified on PeerConnection before local/remote description are set. by Piotr (Peter) Slatala · 6 years ago
16e27a1 Reland "Delete leftover includes and declarations for MediaConstraintsInterface" by Niels Möller · 6 years ago
ec4a060 Revert "Delete leftover includes and declarations for MediaConstraintsInterface" by Niels Moller · 6 years ago
a1e4ae2 Delete leftover includes and declarations for MediaConstraintsInterface by Niels Möller · 6 years ago
f06f923 Delete almost all use of MediaConstraintsInterface in the PeerConnection API by Niels Möller · 6 years ago
d9e4a06 Add CreateSessionDescription overload which takes a cricket::SessionDescription by Steve Anton · 6 years ago
0f5400a [Unified Plan] Implement FiredDirection for RtpTransceiver by Steve Anton · 6 years ago
918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
e12c1fe Removing warning suppression flags from pc/. by Mirko Bonadei · 6 years ago
0756373 [Unified Plan] Avoid offering two senders with the same ID by Steve Anton · 6 years ago
b983bae Remove unused/deprecated DTMF methods by Steve Anton · 6 years ago
66cadcc Replace rtc::Optional with absl::optional in pc by Danil Chapovalov · 6 years ago
665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
fc9dcb6 Remove wire-up for cancelled experement on VAAPI VP8 encoding by Ilya Nikolaevskiy · 6 years ago
97b4ee5 Wire up VAAPI VP8 experimental support in WebRTC. by Ilya Nikolaevskiy · 6 years ago
b330688 Fix build errors when rtc_use_builtin_sw_codecs is set to false. by Anders Carlsson · 6 years ago
0c4f7be New api struct BitrateSettings. by Niels Möller · 6 years ago
6753795 Built in video codec factories. by Anders Carlsson · 6 years ago
c79268f Add IsClosed checks to various PeerConnection methods by Steve Anton · 6 years ago
a2d6067 Reland "Add thread checker to PortAllocator and its subclasses and fix a bug causing memory contention by threads." by Qingsi Wang · 6 years ago
3dc4106 Revert "Add thread checker to PortAllocator and its subclasses and fix a bug" by Patrik Höglund · 6 years ago
fc43d11 Add thread checker to PortAllocator and its subclasses and fix a bug by Qingsi Wang · 6 years ago
83d676b Bug fix for applying a remote description twice without stream IDs. by Seth Hampson · 6 years ago
fd350d7 By default, don't use SRTP_AES128_CM_SHA1_32 protection profile. by Taylor Brandstetter · 6 years ago
5897a6e Adds support for signaling a=msid lines without a=ssrc lines. by Seth Hampson · 6 years ago
644fde4 Add nullptr check in SctpTransport. by Zhi Huang · 6 years ago
5b4f075 Reland "Reland "Adds support for multiple or no media stream ids."" by Seth Hampson · 6 years ago
e830e68 Use new TransportController implementation in PeerConnection. by Zhi Huang · 6 years ago
191bf5c Revert "Reland "Adds support for multiple or no media stream ids."" by Tomas Gunnarsson · 6 years ago
f351c34 Reland "Adds support for multiple or no media stream ids." by Seth Hampson · 6 years ago
bc609ea Revert "Adds support for multiple or no media stream ids." by Emircan Uysaler · 6 years ago
1550292 Adds support for multiple or no media stream ids. by Seth Hampson · 6 years ago
13b8bad Final name changing of MediaStreamInterface.label() to id(). by Seth Hampson · 7 years ago
845e878 Name change from stream label to stream id for spec compliance. by Seth Hampson · 7 years ago
5a26a3a Remove public sync_label from StreamParams by Steve Anton · 7 years ago
fc8d26b Reland "Moved BitrateConfig out of Call::Config." by Sebastian Jansson · 7 years ago
6e22137 Enable Unified Plan tests that were blocked on the stats collector by Steve Anton · 7 years ago
e4bf600 Revert "Moved BitrateConfig out of Call::Config." by Lu Liu · 7 years ago
5897fe2 Moved BitrateConfig out of Call::Config. by Sebastian Jansson · 7 years ago
36da6ff Parameterize PeerConnection interface tests for Unified Plan by Steve Anton · 7 years ago
57858b3 Reland "Update RTCStatsCollector to work with RtpTransceivers" by Steve Anton · 7 years ago
ee2388f Revert "Update RTCStatsCollector to work with RtpTransceivers" by Guido Urdaneta · 7 years ago
56bae8d Update RTCStatsCollector to work with RtpTransceivers by Steve Anton · 7 years ago
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
93a8439 Bind the structured ICE logging with P2PTransportChannel. by Qingsi Wang · 7 years ago
1d7ecd2 Rename a few MediaConfig::Video flags for consistency. by Niels Möller · 7 years ago
6539f69 Add VideoSendStream::Config::EncoderSettings::experiment_cpu_load_estimator. by Niels Möller · 7 years ago
c72af93 Reland "Move stats ID generation from SSRC to local ID" by Harald Alvestrand · 7 years ago
c0092c3 Revert "Move stats ID generation from SSRC to local ID" by Erik Språng · 7 years ago
e357a4d Move stats ID generation from SSRC to local ID by Harald Alvestrand · 7 years ago
62337e5 Use AudioProcessingBuilder everywhere AudioProcessing is created. by Ivo Creusen · 7 years ago
2d6c76a Switch to using AddTrack with stream labels by Steve Anton · 7 years ago
8906187 Pivot generation of stats to iterate senders/receivers by Harald Alvestrand · 7 years ago
b1c1de1 Use the SDP ContentInfo helpers to avoid downcasting by Steve Anton · 7 years ago
74cefe1 Removing dependency on JsepTransport from DtlsTransport tests. by Taylor Brandstetter · 7 years ago
a3a92c2 Replace string type with SdpType enum by Steve Anton · 7 years ago
4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
b95fd2c Optional: Use nullopt and implicit construction in /pc/peerconnectioninterface_unittest.cc by Oskar Sundbom · 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
b2d355e Reland: Reject the description with fewer m= sections. by Zhi Huang · 7 years ago
6f25b09 Reland "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests" by Steve Anton · 7 years ago
8d3444d Reland "Rewrite WebRtcSession media tests as PeerConnection tests" by Steve Anton · 7 years ago
f2662f0 Revert "Rewrite WebRtcSession media tests as PeerConnection tests" by Olga Sharonova · 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
3df5dca Rewrite WebRtcSession media tests as PeerConnection tests by Steve Anton · 7 years ago
1b0eae3 Don't call deprecated CreatePeerConnectionFactory() overloads by Karl Wiberg · 7 years ago
919dc2e Removes fallback from Linux PulseAudio to ALSA. by henrika · 7 years ago
589ae45 Revert "Reject the subsequent offer with fewer m= sections." by Tommi · 7 years ago
a8264db Reject the subsequent offer with fewer m= sections. by Zhi Huang · 7 years ago
f1c6db1 Rewrite WebRtcSession ICE 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
9e6565b Fix PeerConnectionInterfaceTest_StartAndStopLoggingAfterPeerConnectionClosed by Elad Alon · 7 years ago
94286cb Add base fixture and PeerConnection wrapper for unit tests by Steve Anton · 7 years ago
02e7a19 Remove unnecessary video factory references in PeerConnectionFactory by Magnus Jedvert · 7 years ago
835cc0c Remove unnecessary audio references in PeerConnectionFactory by Magnus Jedvert · 7 years ago
b19012e Remove the support of fallback from DTLS to SDES. by zhihuang · 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/peerconnectioninterface_unittest.cc]
2a5e426 Reject the descriptions that attempt to change the order of m= sections by Zhi Huang · 7 years ago
db45ca8 Change PeerConnection test helpers to take unique_ptr by Steve Anton · 7 years ago
141aacb Fix the Chromium crash when creating an answer without a remote description. by zhihuang · 7 years ago
d7850b2 Use fake audio device in peerconnectioninterface_unittest.cc. by deadbeef · 7 years ago
1c378ed Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago