Log - 54047bea1bf00aae0d8da31eb07dd3663d5fb545 - platform/external/webrtc

54047be Reland "Extend TransportSequenceNumber RTP header extension" by Johannes Kron · 5 years ago
1eb3d7e Refactor DelayManager into separate Histogram class and make it injectable for testing purposes. by Jakob Ivarsson · 5 years ago
fa52efa Migrate stdlib task queue to TaskQueueBase interface by Danil Chapovalov · 5 years ago
e11b7d2 Replace field trials with WebRtcKeyValueConfig in RtpRtcpModule by Per Kjellander · 5 years ago
aa1a43e AEC3: Use minimum ERLE during onsets by Gustaf Ullberg · 5 years ago
d6c6f16 Update RTP packet and header fuzzers to support additional extensions by Johannes Kron · 5 years ago
3256225 "Remove" loophole in rtc::Thread::ScopedDisallowBlockingCalls by Karl Wiberg · 5 years ago
826f2e7 Migrate win task queue to TaskQueueBase interface by Danil Chapovalov · 5 years ago
bb05369 Delete unused class FakeCandidatePair. by Niels Möller · 5 years ago
00c57e3 Delete unused class RtpTransportInternalAdapter by Niels Möller · 5 years ago
17c147c Feed PacedSender with RTP packet size by Per Kjellander · 5 years ago
252725d Rename RtpPacketHistory::PacketState::payload_size -> packet_size by Per Kjellander · 5 years ago
1b801e0 Roll chromium_revision 55c441e653..94ca2b10d8 (633987:634089) by chromium-webrtc-autoroll · 5 years ago
feef8f5 Roll chromium_revision 919d2e8241..55c441e653 (633811:633987) by chromium-webrtc-autoroll · 5 years ago
32232e9 Add spatial layers support to video analyze pipeline. by Artem Titov · 5 years ago
8e68920 Roll chromium_revision 554be8c5f4..919d2e8241 (633687:633811) by chromium-webrtc-autoroll · 5 years ago
47cf5ea Migrate gcd task queue implementation to TaskQueueBase interface by Danil Chapovalov · 5 years ago
f5d8808 Remove Analyzers struct. by Mirko Bonadei · 5 years ago
22f9925 webrtc: Remove semicolons. by Nico Weber · 5 years ago
af623ae Delete unused file mock_video_codec_interface.h by Niels Möller · 5 years ago
d36a815 Remove the deprecated CreateProbeClusters method by Piotr (Peter) Slatala · 5 years ago
8b3db59 Revert "Reland of https://webrtc-review.googlesource.com/c/src/+/114883" by Alex Loiko · 5 years ago
01fe309 Do not use RtcEventLogs in media transport when used only for data channel. by Piotr (Peter) Slatala · 5 years ago
ce27875 [AndroidAudioRecord] Added audio format parameter to configure AudioRecord - JavaAudioDeviceModule by Alvaro Martinez · 5 years ago
5341aac Reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 5 years ago
d5e02f0 Delete redundant members from VCMPacket. by Niels Möller · 5 years ago
4d2367a Removes broken frame matching code in scenario quality stats. by Sebastian Jansson · 5 years ago
b35bacc Fix NetEq minimum and maximum delay always reset on acm creation. by Ruslan Burakov · 5 years ago
8073db6 Roll chromium_revision 4b3282a5d6..554be8c5f4 (633587:633687) by chromium-webrtc-autoroll · 5 years ago
76d7ce2 Disabling flaky RecievesVp8SimulcastFrames test. by Sebastian Jansson · 5 years ago
dd1cc98 Reland "Update VP9EncoderImpl to use EncodedImage::Allocate" by Niels Möller · 5 years ago
109b5fb Revert "Extend TransportSequenceNumber RTP header extension" by Mirko Bonadei · 5 years ago
28c7362 Extend TransportSequenceNumber RTP header extension by Johannes Kron · 5 years ago
3f6bf3a Clarify that pacing rate is based on raw target rate by Evan Shrubsole · 5 years ago
5fbebd5 Adds support for VP8 simulcast to scenario tests. by Sebastian Jansson · 5 years ago
ccb9b75 Create version 01 of Generic Frame Descriptor - with discardability flag by Elad Alon · 5 years ago
0b2150c Add a task queue into pc e2e fixture implementation by Artem Titov · 5 years ago
e82643f Fix FFT output size to avoid incorrect band energy computation by Alessio Bazzica · 5 years ago
cc26fef Use a CopyOnWriteBuffer to back EncodedImage data by Niels Möller · 5 years ago
0d4869c Roll chromium_revision d723882358..4b3282a5d6 (633435:633587) by chromium-webrtc-autoroll · 5 years ago
ea7ef2a Refactoring RtpSenderInternal to share implementation for Audio & Video. by Amit Hilbuch · 5 years ago
ba63caf Roll chromium_revision 086bdb74b2..d723882358 (633288:633435) by chromium-webrtc-autoroll · 5 years ago
2297d33 Rejected simulcast layers will no longer appear in GetParameters(). by Amit Hilbuch · 5 years ago
4e7058e desktopCaptuer: exempt to overlapping between hidden taskbar and maximized target by braveyao · 5 years ago
0e44907 Roll chromium_revision 55c117dd14..086bdb74b2 (633171:633288) by chromium-webrtc-autoroll · 5 years ago
7abfd56 Improve CPU utilization when encoding VP8 with two temporal layers by Elad Alon · 5 years ago
599d592 Extend RemoteEstimatorProxy to support feedback on sender request. by Johannes Kron · 5 years ago
a89800c Parse params of 3rd spatial layer from command line. by Sergey Silkin · 5 years ago
d8d3248 Reland "Delete test/constants.h" by Elad Alon · 5 years ago
1925b5a Delete deprecated version of AudioCodingModule::IncomingPacket by Niels Möller · 5 years ago
1431572 Roll chromium_revision 0f484ff968..55c117dd14 (633071:633171) by chromium-webrtc-autoroll · 5 years ago
ffd1f93 Revert "Tests for multi-stream Opus." by Mirko Bonadei · 5 years ago
7131880 Don't block the signaling thread during the call. by Mirko Bonadei · 5 years ago
0e1a1f9 Add verbose logging to encoder bitrate adjuster by Erik Språng · 5 years ago
4f36b7a Revert "Delete test/constants.h" by Oleh Prypin · 5 years ago
06c5145 Adds support for VP9 scalability layers to scenario tests. by Sebastian Jansson · 5 years ago
9c31ac2 Tests for multi-stream Opus. by Alex Loiko · 5 years ago
f2727fb Adds slides support to scenario tests. by Sebastian Jansson · 5 years ago
e9652ca Android: Add video processing interface by Magnus Jedvert · 5 years ago
4a2d57a Don't include video_bitrate_allocation.h from encoded_image.h by Niels Möller · 5 years ago
71aee3a Reland "Propagate VideoFrame::UpdateRect to encoder" by Ilya Nikolaevskiy · 5 years ago
f873cd9 Roll chromium_revision 26c36e3408..0f484ff968 (632825:633071) by chromium-webrtc-autoroll · 5 years ago
bf47495 Update remaining audio test code to not use WebRtcRTPHeader. by Niels Möller · 5 years ago
a0b1fb9 Pass H264 profile/level settings to codec. by Sergey Silkin · 5 years ago
3073c72 Fix AndroidVideoDecoderTest for new Robolectric version. by Sami Kalliomäki · 5 years ago
e049eba Revert "Add Sender and Receiver interfaces for MediaTransport audio" by Sergey Silkin · 5 years ago
d2f0436 Make sdk/android:{audio,video}_api_java publicly visible. by Mirko Bonadei · 5 years ago
0d8eed6 Add Sender and Receiver interfaces for MediaTransport audio by Niels Möller · 5 years ago
6e1402b Skip SSIM calculation in real time mode. by Sergey Silkin · 5 years ago
afb5dbb Update ACM to use RTPHeader instead of WebRtcRTPHeader by Niels Möller · 5 years ago
389b167 Delete test/constants.h by Elad Alon · 5 years ago
8d2e228 Add thread safety annotations for PeerConnection::*_state_ by Karl Wiberg · 5 years ago
e45c688 Remove webrtc::ProtoString. by Mirko Bonadei · 5 years ago
eaf6a8c Adding src/third_party/androidx to the DEPS file. by Mirko Bonadei · 5 years ago
7ea4605 Add latency to remote source api. by Ruslan Burakov · 5 years ago
86f0974 Roll chromium_revision 7df1a5ba86..26c36e3408 (632711:632825) by chromium-webrtc-autoroll · 5 years ago
c664314 Clean up implementation in stream_params by Steve Anton · 5 years ago
ca890ee Revert "Fix getStats() freeze bug affecting Chromium but not WebRTC standalone." by Mirko Bonadei · 5 years ago
ca3c801 Minor eventlogvisualizer tweaks. by Konrad Hofbauer · 5 years ago
429b67d Revert "Propagate VideoFrame::UpdateRect to encoder" by Mirko Bonadei · 5 years ago
675b47d Roll chromium_revision bf2d75ba40..7df1a5ba86 (632595:632711) by chromium-webrtc-autoroll · 5 years ago
9775a58 Plot bitrate allocation per layer based on RTCP XR target bitrate. by Bjorn Terelius · 5 years ago
b03ab71 Add thread safety annotation for PeerConnection::event_log_ by Karl Wiberg · 5 years ago
744310f Add thread safety annotation for PeerConnection::observer_ and factory_ by Karl Wiberg · 5 years ago
7c974e6 Plot RTCP types for incoming and outgoing RTCP packets. by Bjorn Terelius · 5 years ago
c39f462 Move RtcEventProbeClusterCreated to the network controller. by Piotr (Peter) Slatala · 5 years ago
6255af9 Fix RateCounter to don't fail if there are too small amount of events by Artem Titov · 5 years ago
efa72a1 Propagate VideoFrame::UpdateRect to encoder by Ilya Nikolaevskiy · 5 years ago
3a656d1 Tune bitrates and minQP thresholds for high-fps screenshare. by Ilya Nikolaevskiy · 5 years ago
c8221fc Roll chromium_revision d1f68eb66e..bf2d75ba40 (632477:632595) by chromium-webrtc-autoroll · 5 years ago
075f687 Add struct for feedback request to RTPHeaderExtension by Johannes Kron · 5 years ago
05d43c6 Fix getStats() freeze bug affecting Chromium but not WebRTC standalone. by Henrik Boström · 5 years ago
914351d Reland "Always offer transport sequence number header extension for audio"" by Per Kjellander · 5 years ago
106d92d Add thread safety annotation for PeerConnection::SignalDataChannelCreated_ by Karl Wiberg · 5 years ago
13bc871 PostMessageWithFunctor() added. by Henrik Boström · 5 years ago
397c06f Revert "Always offer transport sequence number header extension for audio" by Ying Wang · 5 years ago
7e0e44f Move video-related MediaTransport interfaces to their own file and target by Niels Möller · 5 years ago
054db54 Remove an absl::WrapUnique usage without absl/memory/memory.h include by tzik · 5 years ago
22997d6 Roll chromium_revision 2ad52fb2a4..d1f68eb66e (632357:632477) by chromium-webrtc-autoroll · 5 years ago
1c9c9fc Replace replace_substrs with Abseil by Steve Anton · 5 years ago