Log - 17fc7e2320413eb995a17fd8602d1974994d9121 - platform/external/webrtc

17fc7e2 Add counter to the end of FakeEncoder frames in order to make them unique. by Per Kjellander · 7 years ago
c572ff3 Add default constructor for rtc::Event by Niels Möller · 7 years ago
3ea7b83 Resolve the race condition between mDNS name registration and by Qingsi Wang · 7 years ago
8770ce7 Roll chromium_revision 03cf97f6d8..38dcb5ed01 (605818:605924) by chromium-webrtc-autoroll · 7 years ago
bb091db Roll chromium_revision 793c8566ab..03cf97f6d8 (605715:605818) by chromium-webrtc-autoroll · 7 years ago
2cd3b4c Fixing bug in SimulatedNetwork where packets stop. by Sebastian Jansson · 7 years ago
0f54f21 Removes deprecated GetSentPacket from PacketResult. by Sebastian Jansson · 7 years ago
dc98b9b AEC3: Corrected include by Per Åhgren · 7 years ago
c564a7b Roll chromium_revision 7841106b37..793c8566ab (605607:605715) by chromium-webrtc-autoroll · 7 years ago
8ffd710 Update Android encoder to use GetEncoderInfo() by Erik Språng · 7 years ago
020e583 AEC3: Compensate comfort noise level for loss due to filter bank by Gustaf Ullberg · 7 years ago
83b00f0 AEC3: Computation of comfort noise gains from suppression gains by Gustaf Ullberg · 7 years ago
34fc346 Add support for computing iOS code coverage by Artem Titarenko · 7 years ago
277b6ea Isolating APM API build target: adding dummy :api target. by Alessio Bazzica · 7 years ago
3ddaf3c Revert "Add support for screen sharing with PipeWire on Wayland" by Patrik Höglund · 7 years ago
82c07ea Tune huge video frames detection threshold for GetStats googHugeFramesSent stat by Ilya Nikolaevskiy · 7 years ago
4f3cc6e Make VideoSendStreamTest.NoPaddingWhenVideoIsMuted less flaky by Erik Språng · 7 years ago
a8f5461 nit: Use make_unique in rtp_video_stream_receiver.cc by Elad Alon · 7 years ago
27f3172 Simplify use of events in TestAudioDevice by Niels Möller · 7 years ago
361dbc1 Android: Add option to set presentation timestamp in EglRenderer by Magnus Jedvert · 7 years ago
967f7d5 Add audio level to CSRC class by Jonas Oreland · 7 years ago
df351f4 Update FakeEncoder to use EncoderInfo by Erik Språng · 7 years ago
254d3db Add missing #include to absl/memory/memory.h from audio_encoder_cng.cc by tzik · 7 years ago
fbf1683 Add HdrMetadata to VideoFrame by Johannes Kron · 7 years ago
4f0f3d5 Remove unused member variable - RTCPSender::using_nack_ by Elad Alon · 7 years ago
63ada78 Remove outdated TODO by Sam Zackrisson · 7 years ago
3ea1878 Add severity into RTC logging callbacks by Jiawei Ou · 7 years ago
edfb883 Roll chromium_revision 11d7305a72..7841106b37 (605505:605607) by chromium-webrtc-autoroll · 7 years ago
d7db17b Roll chromium_revision bf7ad46dee..11d7305a72 (605401:605505) by chromium-webrtc-autoroll · 7 years ago
a9bbd86 Add a configuration parameter for using the media transport for data channels. by Bjorn Mellem · 7 years ago
41b5296 Roll chromium_revision c26ff44a53..bf7ad46dee (605286:605401) by chromium-webrtc-autoroll · 7 years ago
ee49f70 Remove VideoEncoder::SetChannelParameters. by philipel · 7 years ago
c22f551 Remove locks from AECM and move it into private_submodules_ by Sam Zackrisson · 7 years ago
e693381 Delete struct rtc::PacketTime. by Niels Möller · 7 years ago
0070655 Removing ancient and unused test scripts and data files by Henrik Lundin · 7 years ago
fd1a2fb Set RtpRtcp config receive_only in voe::ChannelReceive by Niels Möller · 7 years ago
aed3070 Replace GetScalingSettings() with GetEnocderInfo() in TestEncoder by Erik Språng · 7 years ago
f418bcb Refactor RtpSender to use absl::string_view for payload name. by Niels Möller · 7 years ago
2634199 Move MovingAverage to rtc_base/numerics and update it. by Ilya Nikolaevskiy · 7 years ago
a1ead6f Update EncoderProxy to use EncoderInfo by Erik Språng · 7 years ago
bf0d0c1 Add IPv6 configuration parameters to iOS API by Uladzislau Susha · 7 years ago
842a2a8 Roll chromium_revision 4e7c87b55c..c26ff44a53 (605184:605286) by chromium-webrtc-autoroll · 7 years ago
e7547d5 Move MemoryStream to separate source files, and to a test target. by Niels Möller · 7 years ago
9f878f6 Roll chromium_revision b58a03341b..4e7c87b55c (605082:605184) by chromium-webrtc-autoroll · 7 years ago
671341a Roll chromium_revision 35f882550d..b58a03341b (604980:605082) by chromium-webrtc-autoroll · 7 years ago
1bc0b9d Roll chromium_revision e842ab5f98..35f882550d (604874:604980) by chromium-webrtc-autoroll · 7 years ago
2039ee7 Revert "Delete rtc::Pathname" by Qingsi Wang · 7 years ago
273d029 Implement data channel methods in LoopbackMediaTransport. by Bjorn Mellem · 7 years ago
0367d1a Adds a field trial parameter to configure waiting time before sending Nack packets. by Ying Wang · 7 years ago
e401863 Change to RtcEvent::Copy by Elad Alon · 7 years ago
2365936 Hide the AudioEncoderCng class behind a create function by Karl Wiberg · 7 years ago
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