Log - 7eb0a5e210ea0884e0c2faae25dd65505686ecce - platform/external/webrtc

7eb0a5e AudioDecoderOpus: Add support for 16 kHz output sample rate by Karl Wiberg · 5 years ago
ed69d41 Remove deprecated RtcEventLog Create functions by Danil Chapovalov · 5 years ago
2f5554d Make KeyFrameRequestSender injectable in RtpVideoStreamReceiver by Niels Möller · 5 years ago
e8e7d7b Move Connection into it's own .h/.cc file. by Jonas Oreland · 5 years ago
28f0eb2 Move H.264 SPS VUI rewriting to FrameEncodeMetadataWriter. by Mirta Dvornicic · 5 years ago
a1d1a1e WebRTC Opus C interface: Add support for non-48 kHz decode sample rate by Karl Wiberg · 5 years ago
232b6a1 Propagate screenshare info into video track and it's source. by Artem Titov · 5 years ago
98266a4 Roll chromium_revision 99181c0bec..d4906ebd49 (664078:664184) by chromium-webrtc-autoroll · 5 years ago
6737841 Add jitterBufferDelay and jitterBufferEmittedCount stats for video by Guido Urdaneta · 5 years ago
e4470cd Roll chromium_revision 9b60f86c15..99181c0bec (663961:664078) by chromium-webrtc-autoroll · 5 years ago
686be20 Fix ICE connection in datagram_transport. by Anton Sukhanov · 5 years ago
44bd71c Create a composite implementation of RtpTransportInternal. by Bjorn A Mellem · 5 years ago
64e97cf Roll chromium_revision 09fae7ef1b..9b60f86c15 (663849:663961) by chromium-webrtc-autoroll · 5 years ago
f94e3d9 Roll chromium_revision 9809faf8ca..09fae7ef1b (663719:663849) by chromium-webrtc-autoroll · 5 years ago
ce33b6a Implement QualityLimitationReasonTracker and expose "reason". by Henrik Boström · 5 years ago
07fc398 Roll chromium_revision 13f6824c51..9809faf8ca (663612:663719) by chromium-webrtc-autoroll · 5 years ago
787f4b2 Fix text logging of ALR detector experiment settings. by Bjorn Terelius · 5 years ago
0b97e17 Cleanup of CongestionWindowDownlinkDelay trial. by Sebastian Jansson · 5 years ago
9ab520e Reland "Avoid encrypting empty audio packet." by Minyue Li · 5 years ago
9a57350 Use ';' to escape '/' characters in path to dumped received video stream by Ilya Nikolaevskiy · 5 years ago
4ffed7c Add field trial for selecting potentially useful packets as padding. by Erik Språng · 5 years ago
a33a860 Deprecate functions returning cricket::DataContentDescription. by Harald Alvestrand · 5 years ago
f2e9cab Fix BWE simulation graph in event log visualization by Bjorn Terelius · 5 years ago
ca2c430 Allow both LNTF to coexist with NACKs and key frame requests by Elad Alon · 5 years ago
3a072de Roll chromium_revision 60cc82f9b7..13f6824c51 (663509:663612) by chromium-webrtc-autoroll · 5 years ago
8b27910 Include downlink delay into congestion window size. by Ying Wang · 5 years ago
2370242 Enable flex fec support in PC quality test framework by Artem Titov · 5 years ago
36bc4f8 Add thread guards to cricket::P2PTransportChannel. by Harald Alvestrand · 5 years ago
2e8d78c Allow overriding subsets of probing field trials by Jonas Olsson · 5 years ago
6019d43 Removes using imports from flexfec_receiver. by Sebastian Jansson · 5 years ago
126f2b3 AudioEncoderOpus: Add support for 16 kHz input sample rate by Karl Wiberg · 5 years ago
883eefc Implement RTCRemoteInboundRtpStreamStats for both audio and video. by Henrik Boström · 5 years ago
6e436d1 [audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
87e3f9d [video] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 5 years ago
e0eb325 AudioEncoderOpusImpl: Remove unused static methods by Karl Wiberg · 5 years ago
87da109 Make ReceiveStatisticsImpl::SetMaxReorderingThreshold apply per ssrc by Niels Möller · 5 years ago
ad44b75 Roll chromium_revision e1ec78e27e..60cc82f9b7 (663034:663509) by chromium-webrtc-autoroll · 5 years ago
15baf5e Remove last mention of ortc from the codebase. by Mirko Bonadei · 5 years ago
3a1b927 Remove rtp_ and rtcp_packet_transport() from the RtpTransport interface. by Bjorn A Mellem · 5 years ago
8b096a0 LogToSderr by default in WebRTC tests by Anton Sukhanov · 5 years ago
34cd485 Delete the remaining ORTC interfaces. by Bjorn A Mellem · 5 years ago
039a714 VP9 screenshare: drop base layer separately by Ilya Nikolaevskiy · 5 years ago
d9b4f33 Cleanup of AudioAllocationSettings flags. by Sebastian Jansson · 5 years ago
4c29546 Add test to cover bug in vp9 wrapper, triggered by field trial by Erik Språng · 5 years ago
4b27648 Avoid the render lock in AudioProcessingImpl::ProcessStream by Oskar Sundbom · 5 years ago
a0e9943 Negotiation of LNTF controls instantiation of RTPSenderVideo::rtp_sequence_number_map_ by Elad Alon · 5 years ago
0730872 Roll chromium_revision 8ae1a64b43..e1ec78e27e (662926:663034) by chromium-webrtc-autoroll · 5 years ago
a8cf3b7 Ensure CpuInfo::DetectNumberOfCores is > 0 and thread safe. by Mirko Bonadei · 5 years ago
fadb181 Negotiate use of RTCP loss notification feedback (LNTF) by Elad Alon · 5 years ago
815b1a6 Use preprocessor to strip H264 implementation. by Mirko Bonadei · 5 years ago
5c18a5f Reland "VP9 screenshare: Don't base layers frame-rate on input frame-rate" by Ilya Nikolaevskiy · 5 years ago
479c055 Let RtpVideoStreamReceiver implement KeyFrameRequestSender by Niels Möller · 5 years ago
f25df35 Reland "Delete STACK_ARRAY macro, and use of alloca" by Niels Möller · 5 years ago
ce72323 Revert "VP9 screenshare: Don't base layers frame-rate on input frame-rate" by Ilya Nikolaevskiy · 5 years ago
eb1754c VP9 screenshare: Don't base layers frame-rate on input frame-rate by Ilya Nikolaevskiy · 5 years ago
f3db34d Revert "Cleanup of video packet overhead calculation." by Sebastian Jansson · 5 years ago
4c55c89 Roll chromium_revision 8b25075ed7..8ae1a64b43 (662811:662926) by chromium-webrtc-autoroll · 5 years ago
316f3ac Datagram Transport Integration by Anton Sukhanov · 5 years ago
c1c0d6d Roll chromium_revision b82a501520..8b25075ed7 (662691:662811) by chromium-webrtc-autoroll · 5 years ago
4163317 [PeerConnection::AddIceCandidate()] Use mid to look up contents in remote descriptions by Guido Urdaneta · 5 years ago
51f5790 Roll chromium_revision 15b783dc7c..b82a501520 (662034:662691) by chromium-webrtc-autoroll · 5 years ago
c1b3666 Revert "Delete STACK_ARRAY macro, and use of alloca" by Henrik Boström · 5 years ago
2988aca Fix chromium autoroller to parse new clang revision format by Artem Titov · 5 years ago
62ce035 RtpVideoSender nits by Elad Alon · 5 years ago
890bc30 Cleanup of video packet overhead calculation. by Sebastian Jansson · 5 years ago
e9a2ee2 Improve NetEq network adaptation in the beginning of the call. by Jakob Ivarsson · 5 years ago
74b373f Delete STACK_ARRAY macro, and use of alloca by Niels Möller · 5 years ago
eb180f8 Fix incorrect libvpx vp9 dynamic rate control settings by Erik Språng · 5 years ago
fe68daa Add option to configure raw RTP packetization per payload type. by Mirta Dvornicic · 5 years ago
a352248 Add a config flag to disable the audio ALR probing request. by Christoffer Rodbro · 5 years ago
e7e3601 Remove hex_encode functions with raw buffer output from the header file by Niels Möller · 5 years ago
39ece6d Delete unused method ModuleRtpRtcpImpl::SendCompoundRTCP by Niels Möller · 5 years ago
2799e63 Add sizes of spatial layer frames to EncodedImage by Sergey Silkin · 5 years ago
4024440 Lowercase windows includes in desktop_capture/. by Noah Richards · 5 years ago
ecd3054 Replace a broken assumption in candidate gathering for mDNS candidates. by Qingsi Wang · 5 years ago
7e7c5c3 WebRTC Opus C interface: Add support for non-48 kHz encode sample rate by Karl Wiberg · 5 years ago
646fda0 Implement RTCMediaSourceStats and friends in standard getStats(). by Henrik Boström · 5 years ago
58c71db Fix for crash in event log when using scenario tests. by Sebastian Jansson · 5 years ago
9ce451a End NetEq simulation if there are no more packets to decode. by Jakob Ivarsson · 5 years ago
4ed7e51 Revert "Add ability to cap the video jitter estimate to a max value." by Stefan Holmer · 5 years ago
040dc43 Fix shadowing of override_field_trials_ in WebRtcVideoEngineTest by Elad Alon · 5 years ago
b32f2c7 Publish rtc event log api and default factory for it in api/ by Danil Chapovalov · 5 years ago
23aff9b Implement RTCOutboundRtpStreamStats.totalEncodedBytesTarget. by Henrik Boström · 5 years ago
04f3924 Delete no longer used windows helpers by Niels Möller · 5 years ago
b5d9183 Add RTP timestamp to contributing sources by Johannes Kron · 5 years ago
afb8d5c Log average decoded and rendered framerate for a VideoReceiveStream. by Åsa Persson · 5 years ago
bb90ccc Roll chromium_revision 1216f271d5..15b783dc7c (661928:662034) by chromium-webrtc-autoroll · 5 years ago
5f19f8f Roll chromium_revision 0c18b1a229..1216f271d5 (661811:661928) by chromium-webrtc-autoroll · 5 years ago
4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 5 years ago
4880e15 Roll chromium_revision 7a39eea5d8..0c18b1a229 (661628:661811) by chromium-webrtc-autoroll · 5 years ago
9c91887 Splits SendTimeHistory::AddAndRemoveOld into Add/Remove. by Sebastian Jansson · 5 years ago
3b112e2 Delete multi-parameter CreateModularPeerConnectionFactory by Danil Chapovalov · 5 years ago
acab559 Adds overuse predictor to GoogCC. by Sebastian Jansson · 5 years ago
c701dec Add GetTransportParametersOffer method for DatagramTransportInterface by Anton Sukhanov · 5 years ago
04a3cc1 Delete rtc_base/unittest_main.cc by Niels Möller · 5 years ago
d703cd0 Revert "Avoid encrypting empty audio packet." by Minyue Li · 5 years ago
19da5ce Formatting of WebRTC-Vp9InterLayerPred field trial. by Sergey Silkin · 5 years ago
3be9da3 Make unpack_aecdump unpack RuntimeSettings by Fredrik Hernqvist · 5 years ago
b0ac943 Avoid encrypting empty audio packet. by Minyue Li · 5 years ago
4d29ef0 Add periodic alive message logging to prevent test infra think, that test is dead by Artem Titov · 5 years ago