Log - e34fb878b9fadd89237c2e3ac80b7667ff7620eb - platform/external/webrtc

ff27da5 Add/remove receive streams with SSRC 0 from media channels by Saurav Das · 5 years ago
1b83a9e Only handle each RTCP once. by Sebastian Jansson · 5 years ago
01be33b Using lambdas instead of rtc::Bind in BaseChannel. by Sebastian Jansson · 5 years ago
65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 5 years ago
be2e5f7 Make payload type demux conditional on media direction by Steve Anton · 5 years ago
3a1b927 Remove rtp_ and rtcp_packet_transport() from the RtpTransport interface. by Bjorn A Mellem · 5 years ago
4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
5fc28b1 Reland "Reland "Version 2 "Refactoring DataContentDescription class""" by Harald Alvestrand · 5 years ago
46afbf9 Revert "Reland "Version 2 "Refactoring DataContentDescription class""" by Steve Anton · 5 years ago
37f2b43 Reland "Version 2 "Refactoring DataContentDescription class"" by Harald Alvestrand · 5 years ago
141c0ad Revert "Version 2 "Refactoring DataContentDescription class"" by Harald Alvestrand · 6 years ago
14b2758 Version 2 "Refactoring DataContentDescription class" by Harald Alvestrand · 6 years ago
bcd39d4 Creating Simulcast offer and answer in Peer Connection. by Amit Hilbuch · 6 years ago
309aafe Add 'AudioPacket' notification to media transport interface. by Piotr (Peter) Slatala · 6 years ago
c2c733e Remove unused methods from cricket::BaseChannel. by Mirko Bonadei · 6 years ago
10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
179a392 Implement TargetBitrate, NetworkRoute and overhead features of media transport interface. by Piotr (Peter) Slatala · 6 years ago
dd9390c Prevent channels being set on stopped transceiver. by Amit Hilbuch · 6 years ago
e693381 Delete struct rtc::PacketTime. by Niels Möller · 6 years ago
98a462c Reland "Reland "Propagate media transport to media channel."" by Anton Sukhanov · 6 years ago
2bff543 Removes undefined declarations in channel.h. by Sebastian Jansson · 6 years ago
9accc9f Revert "Reland "Propagate media transport to media channel."" by Oleh Prypin · 6 years ago
da65ed2 Reland "Propagate media transport to media channel." by Anton Sukhanov · 6 years ago
37cf245 Revert "Propagate media transport to media channel." by Oleh Prypin · 6 years ago
8c16f74 Propagate media transport to media channel. by Anton Sukhanov · 6 years ago
a54daf1 Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Benjamin Wright · 6 years ago
8f4bc41 Revert "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Oleh Prypin · 6 years ago
ac2f3d1 Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h by Benjamin Wright · 6 years ago
e41c433 Move sigslot to proper third_party directory by Artem Titov · 6 years ago
ee01a83 Remove MetricsObserverInterface. by Qingsi Wang · 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
0e36a72 Delete unused class CurrentSpeakerMonitor. by Niels Möller · 6 years ago
0327c2d Move VideoStreamEncoderInterface to api/. by Niels Möller · 6 years ago
c6ce9c5 New file api/video/BUILD.gn by Niels Möller · 6 years ago
365381f Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 7 years ago
e830e68 Use new TransportController implementation in PeerConnection. by Zhi Huang · 7 years ago
95e7dbb Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" by Zhi Huang · 7 years ago
27f3bf5 Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 7 years ago
97d5e5b Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 7 years ago
ea8b62a Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 7 years ago
db67ba1 Report SRTP error codes to UMA by Steve Anton · 7 years ago
0807d15 Remove more dead code from BaseChannel by Steve Anton · 7 years ago
f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
ba37b4b Change return type of RtpSenderInterface::SetParameters from bool to RTCError by Zach Stein · 7 years ago
e2a9318 Delete ConnectionMonitor. by Niels Möller · 7 years ago
0228485 Delete MediaMonitor. by Niels Möller · 7 years ago
053c1f8 Delete unused signal VoiceChannel::SignalAudioMonitor. by Niels Möller · 7 years ago
47136dd Change RtpSenders to interact with the media channel directly by Steve Anton · 7 years ago
6077675 Change RtpReceivers to interact with the media channel directly by Steve Anton · 7 years ago
dc8b5ab Remove dead code for media channel errors by Steve Anton · 7 years ago
9e19403 Move videosourceinterface to api. by Patrik Höglund · 7 years ago
be214a2 Move videosinkinterface.h to common_video to solve a circular dep. by Patrik Höglund · 7 years ago
593e325 Change RTCStatsCollector to only access channels from signaling thread by Steve Anton · 7 years ago
9a44f96 Delete rtc_base/window.h. by Niels Möller · 7 years ago
3828c06 Replace cricket::ContentAction with webrtc::SdpType by Steve Anton · 7 years ago
2dfc42d Prepare to make BaseChannel depend on RtpTransportInternal only. by Zhi Huang · 7 years ago
cd3fc5d Use the DtlsSrtpTransport in BaseChannel. by Zhi Huang · 7 years ago
4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
1d88d74 Remove the unused code. by Zhi Huang · 7 years ago
942bc2e Reland: Replaced the SignalSelectedCandidatePairChanged with a new signal. by Zhi Huang · 7 years ago
8c316c1 Revert "Replaced the SignalSelectedCandidatePairChanged with a new signal." by Zhi Huang · 7 years ago
7167745 Replaced the SignalSelectedCandidatePairChanged with a new signal. by Zhi Huang · 7 years ago
c99b6c7 Remove the SetEncryptedHeaderExtensionIds methods. by Zhi Huang · 7 years ago
8699a32 Have BaseChannel take MediaChannel as unique_ptr by Steve Anton · 7 years ago
6b63cd5 Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests by Steve Anton · 7 years ago
b526158 Move the TransportController from p2p/base to pc/. by Zhi Huang · 7 years ago
cf990f5 Reland: Completed the functionalities of SrtpTransport. by Zhi Huang · 7 years ago
eb23e17 Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ ) by zhihuang · 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/channel.h]
18ee1d5 Move SDP m= line matching from BaseChannel to WebRtcSession by Steve Anton · 7 years ago
e683c68 Completed the functionalities of SrtpTransport. by zhihuang · 7 years ago
398c3fd Introduce RtpTransportInternal and SrtpTransport. by zstein · 7 years ago
634977b SignalPacketReceived should pass packet as a pointer instead of a non-const reference. by zstein · 7 years ago
e8ab543 Make BaseChannel::rtp_transport_ a unique_ptr. by zstein · 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
5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 7 years ago
38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
f79ade1 Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )" by stefan · 7 years ago
3dcf0e9 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 7 years ago
d72098a Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ ) by charujain · 7 years ago
e80f4c9 Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. by Stefan Holmer · 7 years ago
56162b9 Move ready to send logic from BaseChannel to RtpTransport. by zstein · 8 years ago
7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 8 years ago
8d609f6 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
fbcc5cb Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
292084c Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
d48dbda Add a minimal RtpTransport class for use by BaseChannel. by zstein · 8 years ago
5bd5ca3 Rename "PacketTransportInterface" to "PacketTransportInternal". by deadbeef · 8 years ago
f534659 Adding ability for BaseChannel to use PacketTransportInterface. by deadbeef · 8 years ago
b2cdd93 Remove the dependency of TransportChannel and TransportChannelImpl. by zhihuang · 8 years ago
6ce9259 Revert of make the DtlsTransportWrapper inherit form DtlsTransportInternal (patchset #11 id:320001 of https://codereview.webrtc.org/2606123002/ ) by zhihuang · 8 years ago
5aed06c make the DtlsTransportWrapper inherit form DtlsTransportInternal by zhihuang · 8 years ago
bad5dad More minor improvements to BaseChannel/transport code. by deadbeef · 8 years ago
ac22f70 Refactoring of RTCP options in BaseChannel. by deadbeef · 8 years ago
f5b251b Remove BaseChannel's dependency on TransportController. by zhihuang · 8 years ago
953c2ce Reland of: Separating SCTP code from BaseChannel/MediaChannel. by deadbeef · 8 years ago