Log - e7a5f7bfae7e619d2c9711a0b13a17d02e407563 - platform/external/webrtc

e7a5f7b Modifying MediaChannel to accept CopyOnWriteBuffer by value. by Amit Hilbuch · 6 years ago
fe6e50f Allow more than one registered network change callback in MediaTransport by Niels Möller · 6 years ago
eee110d Remove nogncheck from pc/. by Mirko Bonadei · 6 years ago
64b626b Use Abseil container algorithms in pc/ by Steve Anton · 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
29e13fd Delete rtc::PacketTime (was an alias for int64_t) by Niels Möller · 6 years ago
5f8b5fd Use for range loop in pc/channel.cc 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
15ca5a9 Add implicit conversion between rtc:PacketTime and int64_t. by Niels Möller · 6 years ago
9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
98a462c Reland "Reland "Propagate media transport to media channel."" by Anton Sukhanov · 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
366a50c Remove simple stringstream usages. by Jonas Olsson · 6 years ago
ee01a83 Remove MetricsObserverInterface. by Qingsi Wang · 6 years ago
918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 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
365381f Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 6 years ago
5897a6e Adds support for signaling a=msid lines without a=ssrc lines. by Seth Hampson · 6 years ago
0ffe03d Add Deinit() to the destructors of Voice/Video/RtpDataChannel. by Zhi Huang · 6 years ago
e830e68 Use new TransportController implementation in PeerConnection. by Zhi Huang · 6 years ago
95e7dbb Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" by Zhi Huang · 6 years ago
27f3bf5 Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 6 years ago
97d5e5b Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 6 years ago
ea8b62a Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 6 years ago
bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 6 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
42805f3 Revert "Remove nogncheck and add proper dependencies." by Patrik Höglund · 7 years ago
9b045fa Remove nogncheck and add proper dependencies. by Patrik Höglund · 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
aaaf1cf Revert "Remove nogncheck and add proper dependencies." by Patrik Höglund · 7 years ago
eefd543 Remove nogncheck and add proper dependencies. by Patrik Höglund · 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
b1c1de1 Use the SDP ContentInfo helpers to avoid downcasting by Steve Anton · 7 years ago
5634427 Remove unused properties from MediaContentDescription by Steve Anton · 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
d745578 Call SrtpTransport::EnableExternalAuth when enabling SDES. by Zhi Huang · 7 years ago
2a4d70c Make the DtlsSrtpTransport cache the RtpAbsSendTimeHeaderExtension. 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
36f8f3e Optional: Use nullopt and implicit construction in /pc by Oskar Sundbom · 7 years ago
1d88d74 Remove the unused code. by Zhi Huang · 7 years ago
801b868 Remove the CA_UPDATE and related code. by Zhi Huang · 7 years ago
5f5918f Merge MediaChannel's OnTransportOverheadChanged and OnNetworkRouteChanged callbacks. 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
675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
8699a32 Have BaseChannel take MediaChannel as unique_ptr by Steve Anton · 7 years ago
04eaa15 Change the flag when RtpTransport objects send packet. 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.cc]
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 · 7 years ago
7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. 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
d48dbda Add a minimal RtpTransport class for use by BaseChannel. by zstein · 7 years ago
dfcab72 Reland: Improve testing of SRTP external auth code paths. by jbauch · 8 years ago
d81f121 Revert of Improve testing of SRTP external auth code paths. (patchset #2 id:20001 of https://codereview.webrtc.org/2722423003/ ) by jbauch · 8 years ago
ac170d5 Improve testing of SRTP external auth code paths. by jbauch · 8 years ago
d48f488 Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined. by jbauch · 8 years ago
e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 8 years ago
5bd5ca3 Rename "PacketTransportInterface" to "PacketTransportInternal". by deadbeef · 8 years ago
7ce109a Replace the easy cases of VERIFY usage. by nisse · 8 years ago