Log - 42e7d9c929efeba364a455639baa5e41c52ffef3 - platform/external/webrtc

42e7d9c Enable rtc event log in *_loopback tools running with renderers by Ilya Nikolaevskiy · 7 years ago
f8ba95e Add field trial for vp8 cpu speed configuration for arm. by Åsa Persson · 7 years ago
56ef305 Move event logging of config into AudioSendStream. by Oskar Sundbom · 7 years ago
6bf2054 Roll chromium_revision 734e273d43..e842ab5f98 (604373:604874) by chromium-webrtc-autoroll · 7 years ago
aa3c1cc Delete _strnicmp. Uses replaced with abseil functions. by Niels Möller · 7 years ago
41f00de Fix chromium roll by Artem Titov · 7 years ago
6b9dec0 Delete rtc::Pathname by Niels Möller · 7 years ago
d4a68bd Implement Injectable Audio Codecs for the Java SDK. by Lennart Kolmodin · 7 years ago
3e4c77f Fix AGC2 fixed-adaptive gain controllers order. by Alessio Bazzica · 7 years ago
096d016 Update MultiplexEncoderAdapter to use EncoderInfo by Erik Språng · 7 years ago
58df0ad Add guards to VideoCaptureDS::Init for when pins are null by Andreas Pehrson · 7 years ago
9b5b070 Use EncoderInfo in SimulcastEncoderAdapter by Erik Språng · 7 years ago
4eb4112 Plug-in media transport state listener by Piotr (Peter) Slatala · 7 years ago
189013b Update QualityTestVideoEncoder to use GetEncoderInfo() by Erik Språng · 7 years ago
449afd9 Updated ScopedVideoEncoder to use GetEncoderInfo() by Erik Språng · 7 years ago
5e78461 Make the extra seturation margin configurable. by Alex Loiko · 7 years ago
b1e031a JitterEstimator: Remove old LowRate exp and add trial for upper bound. by Erik Språng · 7 years ago
15ca5a9 Add implicit conversion between rtc:PacketTime and int64_t. by Niels Möller · 7 years ago
96965ae Add ability to enable frame dumping decoder via field trial. by Erik Språng · 7 years ago
fe45da4 Remove WebRTC-VP8-GfBoost field trial. by philipel · 7 years ago
af6d741 Makes send time information in feedback non-optional. by Sebastian Jansson · 7 years ago
be837ac Remove RTPFragmentationHeader from LibvpxVp8Encoder. by philipel · 7 years ago
2812763 Remove deprecated AudioProcessing::GetStatistics function by Sam Zackrisson · 7 years ago
4e93329 Properly setup MockPeerConnectionObserver in tests (continued). by Yves Gerey · 7 years ago
dd20c9c Add support for screen sharing with PipeWire on Wayland by Tomas Popela · 7 years ago
7f4dfa4 Remove locks from AEC2 and move it into private_submodules_ by Sam Zackrisson · 7 years ago
59844ce Revert "Use the factory instead of using the builtin code path in `VideoCodecInitializer`." by Qingsi Wang · 7 years ago
7852d29 Improve the documentation of MdnsResponderInterface and rename MDns.* to Mdns.*. by Qingsi Wang · 7 years ago
eb2c641 Delete the default implementations of MediaTransportInterface methods. by Bjorn Mellem · 7 years ago
be14217 Use the factory instead of using the builtin code path in `VideoCodecInitializer`. by Jiawei Ou · 7 years ago
8386435 Roll chromium_revision 6271fcdc14..734e273d43 (604273:604373) by chromium-webrtc-autoroll · 7 years ago
1f6aa9f Add interfaces for using MediaTransport as the transport for data channels. by Bjorn Mellem · 7 years ago
062a691 Roll chromium_revision 9996ac8918..6271fcdc14 (604166:604273) by chromium-webrtc-autoroll · 7 years ago
9f95625 When SDES is used, pass pre-shared key to media transport. by Piotr (Peter) Slatala · 7 years ago
7182286 Allow FakeNetworkPipe to wake up its processing thread by Sebastian Jansson · 7 years ago
693432d Add obj-c mapping from native configuration to RTCConfiguration by Piotr (Peter) Slatala · 7 years ago
e6caa9f export RTCRtpTransceiverInit by Piasy · 7 years ago
ed45c57 Corrects audio overhead correction in Scenario test. by Sebastian Jansson · 7 years ago
69807e8 Depend directly on destination targets. by Yves Gerey · 7 years ago
a8fa2d0 Move some methods from StreamInterface to FifoBuffer by Niels Möller · 7 years ago
21cddff Harmonize paths to dependent targets. by Yves Gerey · 7 years ago
b32bb95 Bugfix: FlexFEC causes retransmit bitrate increase. by Ying Wang · 7 years ago
8b7d206 AEC3: Decrease latency until the delay has been detected by Per Åhgren · 7 years ago
f577ab3 Roll chromium_revision 7e85c0922c..9996ac8918 (604065:604166) by chromium-webrtc-autoroll · 7 years ago
b00b28e Roll chromium_revision 0cb3899c4e..7e85c0922c (603959:604065) by chromium-webrtc-autoroll · 7 years ago
b3f887b Expose key derivation through a simple interface for use in WebRTC. by Benjamin Wright · 7 years ago
1a92cd7 Roll chromium_revision 34bb9a9162..0cb3899c4e (603839:603959) by chromium-webrtc-autoroll · 7 years ago
c78b0ea Create a MediaTransportState enum and add a state callback to MediaTransport. by Bjorn Mellem · 7 years ago
eaf337a Remove event wait logic from DesktopConfigurationMonitor by Emircan Uysaler · 7 years ago
746d46b AGC2: renaming GainCurveApplier to Limiter. by Alessio Bazzica · 7 years ago
fcc3981 Revert "Use only first payload timestamp for RTCP SR generation for audio" by Ilya Nikolaevskiy · 7 years ago
992a868 Fix for clock reset repair. by Christoffer Rodbro · 7 years ago
a2e133d Delete StreamInterface::ReadLine. by Niels Möller · 7 years ago
ed7b8b1 Update media transport settings struct by Piotr (Peter) Slatala · 7 years ago
3e67676 Add support for field trials in peerconnection_client|server by Bjorn Terelius · 7 years ago
9a0662a Use only first payload timestamp for RTCP SR generation for audio by Ilya Nikolaevskiy · 7 years ago
b26cf2f Add field trial to enable the new RTC event log format. by Bjorn Terelius · 7 years ago
97e35ce Revert "Disabled TestPacketBuffer.SeqNumWrapOneFrame test due to clang update" by Artem Titarenko · 7 years ago
0eb7d3ff Always call ConvertToI420 with positive crop_height by Robert Bares · 7 years ago
9862c2e Delete OptionsFile class. Refactored only user, TurnFileAuth. by Niels Möller · 7 years ago
3df6e715 Makes PacketResult::GetSentPacket const. by Sebastian Jansson · 7 years ago
b33168e Roll chromium_revision 89ed1da2c8..34bb9a9162 (603733:603839) by chromium-webrtc-autoroll · 7 years ago
946179c Delete unused function rtc::Flow. by Niels Möller · 7 years ago
42b4315 Add iOS SDK unit tests for nalu_rewriter by Artem Titarenko · 7 years ago
9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 7 years ago
f43bcd4 Remove likely obsolete entries from WATCHLISTS by Oleh Prypin · 7 years ago
0ac98ab Roll chromium_revision 03b56190ff..89ed1da2c8 (603619:603733) by chromium-webrtc-autoroll · 7 years ago
770c32a Roll chromium_revision 55624cc6cd..03b56190ff (603513:603619) by chromium-webrtc-autoroll · 7 years ago
3b149e4 Added myself to the base watchlist to monitor ssl* changes. by Benjamin Wright · 7 years ago
5124a04 Roll chromium_revision 62e33bd2f0..55624cc6cd (603177:603513) by chromium-webrtc-autoroll · 7 years ago
6b9d823 Add TargetBitrate callback to MediaTransportInterface. by Piotr (Peter) Slatala · 7 years ago
c640a93 Fix import of chromium into webrtc. by Artem Titov · 7 years ago
a0677d1 Add MediaTransportSettings struct for configuring media transport. by Piotr (Peter) Slatala · 7 years ago
12048c7 Fix error handling in hex_decode. by Niels Möller · 7 years ago
ef45669 Adds GetSentPacket to PacketResult. by Sebastian Jansson · 7 years ago
449c1c0 Adds unit tests for safe reset trial. by Sebastian Jansson · 7 years ago
7286496 Download aap2 and bundletool as part of required dependencies. by Yves Gerey · 7 years ago
6fcf6ca Modified PressEnterToContinue() to actualy check if Enter is pressed by Danail Kirov · 7 years ago
2c16cc6 Replace some usage of EventWrapper with rtc::Event. by Niels Möller · 7 years ago
88d8d7d Add missing assignment in RTCConfiguration.mm by Piotr (Peter) Slatala · 7 years ago
f3ff14c Properly setup MockPeerConnectionObserver in tests. by Yves Gerey · 7 years ago
22a8f98 Formatted sslidenty.cc and moved non referenced functions into an by Benjamin Wright · 7 years ago
428320c Formatting OpenSSLCertificate and doing some minor code cleanup. by Benjamin Wright · 7 years ago
5d35554 Rename private member functions to use CamelCase. by Benjamin Wright · 7 years ago
61c5cc8 Makes OpenSSL concrete implementations final. by Benjamin Wright · 7 years ago
2616594 Refactor: Make SSLCertChain a final class. by Benjamin Wright · 7 years ago
150a907 FrameEncryption Video End To End Testcase. by Benjamin Wright · 7 years ago
c462a6e Prevent the frame decryptor being set if the channel is stopped. by Benjamin Wright · 7 years ago
625771d Roll chromium_revision a539a24569..62e33bd2f0 (603045:603177) by chromium-webrtc-autoroll · 7 years ago
59ebf23 Refactor structs in rtc_event_log_parser_new.h by Elad Alon · 7 years ago
ff43541 Delta compression efficiency improvement for non-existent base by Elad Alon · 7 years ago
436ebca Fix extra setdscp call introduced by bad merge. by Tim Haloun · 7 years ago
0f08d22 Add a function for enabling the congestion window and pushback controller in the webrtc::SendSideCongestionController. by erikvarga@webrtc.org · 7 years ago
99b71df Use function_video_(en|de)coder_factory from api by Danil Chapovalov · 7 years ago
88c2c50 Use monotonic clock to derive NTP timestamps in RTCP module by Ilya Nikolaevskiy · 7 years ago
fdee701 Add parser and unittests for new RTC event log format. by Bjorn Terelius · 7 years ago
916ae08 Makes critsect_.Leave() more visible in PacedSender. by Sebastian Jansson · 7 years ago
6dd7f91 Remove deprecated deregistration functions in RtcpTransceiver by Danil Chapovalov · 7 years ago
06aa209 Add support to adapt video without preserving aspect ratio by Magnus Jedvert · 7 years ago
9049037 Simplify api/DEPS presubmit check. by Mirko Bonadei · 7 years ago