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gerrit-public.fairphone.software
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platform
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external
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webrtc
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ffdd3d4176b38ed257d6516f8d4e1d4ff6b42cb3
ffdd3d4
Enabling 'gn check' on all //webrtc/modules
by mbonadei
· 8 years ago
1cd7ab3
Roll chromium_revision d02a1d189e..1721879351 (455229:455313)
by buildbot
· 8 years ago
9fa7e4a
Fix error handling in X11 screen capturer
by sergeyu
· 8 years ago
b09b3f9
Add the option to disable IPv6 ICE candidates on WiFi.
by zhihuang
· 8 years ago
03e8af8
Roll chromium_revision 792dd3474f..d02a1d189e (455158:455229)
by buildbot
· 8 years ago
12dfd7f
Roll chromium_revision 37402096ff..792dd3474f (455082:455158)
by buildbot
· 8 years ago
3794c74
Roll chromium_revision bd11b45ad7..37402096ff (455060:455082)
by buildbot
· 8 years ago
cb9ba30
Perform probing on network route change.
by philipel
· 8 years ago
f89a738
Disable failing fullstack test with 15 thumbnail streams
by ilnik
· 8 years ago
bddab23
Roll chromium_revision fd6ebdefe3..bd11b45ad7 (455055:455060)
by buildbot
· 8 years ago
566c43b
Cleaning up full-stack simulcast tests and making them more realistic.
by ilnik
· 8 years ago
a014cc5
Reland of "Added large room scenario to full-stack tests"
by ilnik
· 8 years ago
d461ffc
Avoid overflow in WebRtcSpl_DotProductWithScale
by henrik.lundin
· 8 years ago
57f19cc
Drop VP8 frames in case of duplicates in RtpFrameReferenceFinder.
by philipel
· 8 years ago
2150ac2
Roll chromium_revision 06552870e5..fd6ebdefe3 (455043:455055)
by buildbot
· 8 years ago
21dc189
Replace Clock::CurrentNtp with Clock::CurrentNtpTime
by danilchap
· 8 years ago
92a7a18
Update formatting of AudioLevel class
by henrik.lundin
· 8 years ago
759358c
Remove unused VoiceChannelTransport.
by solenberg
· 8 years ago
7cb0e55
Remove voe_cmd_test.
by solenberg
· 8 years ago
17b958c
Support pipelining codecs in VideoProcessor.
by brandtr
· 8 years ago
ae9ba04
Minor changes in videoprocessor and videoprocessor_integrationtests.h
by asapersson
· 8 years ago
c438bb1
Roll chromium_revision 79d2efda6e..06552870e5 (455031:455043)
by buildbot
· 8 years ago
b5cb14f
Roll chromium_revision 5bdba6cf80..79d2efda6e (455014:455031)
by buildbot
· 8 years ago
4d1cf9f
Roll chromium_revision 3c427be05b..5bdba6cf80 (454978:455014)
by buildbot
· 8 years ago
846e1be
Fix iOS8 crash in background mode.
by sdkdimon
· 8 years ago
fe3e97b
Roll chromium_revision 22a4567d56..3c427be05b (454904:454978)
by buildbot
· 8 years ago
6dfd53a
Rename PeerConnection::OnIceConnectionChange to OnIceConnectionStateChange
by zstein
· 8 years ago
ad94c4c
Replace StunMessage's vector<StunAttribute*>* with a
by zstein
· 8 years ago
d5a2d9a
WebRtcVideoChannel2Test::SetRecvCodecsAcceptsMultipleVideoCodecs passes now.
by zstein
· 8 years ago
eaa9c1d
Remove HAVE_SRTP define and unmaintained code.
by jbauch
· 8 years ago
0744a47
Roll chromium_revision b728d6759a..22a4567d56 (454873:454904)
by buildbot
· 8 years ago
89cf8ae
Roll chromium_revision fe896a85ac..b728d6759a (454855:454873)
by buildbot
· 8 years ago
bfb1245
Revert of Added large room scenario to full-stack tests. Added thumbnail streams functionality to call test/v… (patchset #8 id:140001 of https://codereview.webrtc.org/2730073002/ )
by ilnik
· 8 years ago
b072739
Check __GLIBC_PREREQ availability before use.
by hillma
· 8 years ago
e7811f5
Fix segmentation fault in AudioEncoderOpusTest.EncodeAtMinBitrate.
by michaelt
· 8 years ago
d8bd1b1
Added large room scenario to full-stack tests. Added thumbnail streams functionality to video quality test.
by ilnik
· 8 years ago
a2ef4f9
Enable GN check for webrtc/media
by kjellander
· 8 years ago
f949000
Rename webrtc::PacketInfo to webrtc::PacketFeedback. This resolves ambiguity with a similarly named RTCPReceiver::PacketInformation and RtpPacketizerVp9::PacketInfo.
by elad.alon
· 8 years ago
d1587ad
Android HW decoder: Don't check slice_height for texture output
by magjed
· 8 years ago
87ee970
Roll chromium_revision 88c14863d3..fe896a85ac (454842:454855)
by buildbot
· 8 years ago
8f8d1a0
Adding placeholder for low bandwidth audio test
by kjellander
· 8 years ago
c898825
Add kjellander to OWNERS for *.py in examples/android{app,tests}
by kjellander
· 8 years ago
b8102e0
Reland of Add QP for FFmpeg H264 decoder. (patchset #1 id:1 of https://codereview.webrtc.org/2726973003/ )
by sakal
· 8 years ago
fb1fa44
Remove MockRemoteBitrateObserver (unused)
by elad.alon
· 8 years ago
609ab2b
Make ExtendedReports::target_bitrate() accessor const as it supposed to be
by danilchap
· 8 years ago
5419ac8
Remove unused RemoteBitrateEstimator::IncomingPacketFeedbackVector()
by elad.alon
· 8 years ago
dea7f4f
Ignore aud and sei NALus when parsing bitstream.
by kthelgason
· 8 years ago
5571ee9
Fix memory leaks in Windows core audio
by jens.nielsen
· 8 years ago
3b2fb20
Add PESQ precompiled tool for audio quality testing
by oprypin
· 8 years ago
518161b
Roll chromium_revision ef90396563..88c14863d3 (454837:454842)
by buildbot
· 8 years ago
1993b1d
Reland "Enable GN check for webrtc/examples"
by kjellander
· 8 years ago
dfcab72
Reland: Improve testing of SRTP external auth code paths.
by jbauch
· 8 years ago
156e3af
Optimize ParseRbsp method in H264 bitstream parser.
by kthelgason
· 8 years ago
0a3e69a
Roll chromium_revision c9500f1d64..ef90396563 (454819:454837)
by buildbot
· 8 years ago
1e96815
White noise generator implemented.
by alessiob
· 8 years ago
4610ec7
Noise generator abstract class and identity noise generator implemented.
by alessiob
· 8 years ago
21debb3
APM quality assessment tool, signal processing tools.
by alessiob
· 8 years ago
ef3162c
Roll chromium_revision 3bb4fdb452..c9500f1d64 (454805:454819)
by buildbot
· 8 years ago
58492bb
Roll chromium_revision 6316a55cb9..3bb4fdb452 (454797:454805)
by buildbot
· 8 years ago
17c6f88
Roll chromium_revision 34b92857a5..6316a55cb9 (454795:454797)
by buildbot
· 8 years ago
2280783
Roll chromium_revision 42e64efbfe..34b92857a5 (454794:454795)
by buildbot
· 8 years ago
0ed3bfe
Roll chromium_revision 775aa60705..42e64efbfe (454791:454794)
by buildbot
· 8 years ago
d3e5111
Roll chromium_revision 59566e8814..775aa60705 (454787:454791)
by buildbot
· 8 years ago
d3456cf
Roll chromium_revision 0832734083..59566e8814 (454785:454787)
by buildbot
· 8 years ago
80b16f7
Roll chromium_revision f28287ce59..0832734083 (454779:454785)
by buildbot
· 8 years ago
98e1531
Revert of Enable GN check for webrtc/{ortc,p2p} (patchset #4 id:60001 of https://codereview.webrtc.org/2714263004/ )
by kjellander
· 8 years ago
991f829
Roll chromium_revision 351aa094a2..f28287ce59 (454776:454779)
by buildbot
· 8 years ago
c9515b6
Enable GN check for webrtc/{ortc,p2p}
by kjellander
· 8 years ago
a80e4f6
Roll chromium_revision 16ffcf196c..351aa094a2 (454775:454776)
by buildbot
· 8 years ago
13dc486
Roll chromium_revision 3665df287a..16ffcf196c (454770:454775)
by buildbot
· 8 years ago
7e15cfb
Roll chromium_revision 34e86af9f3..3665df287a (454769:454770)
by buildbot
· 8 years ago
c670200
Roll chromium_revision 6dfeb5c005..34e86af9f3 (454766:454769)
by buildbot
· 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
ca81868
Roll chromium_revision 78ea87a165..6dfeb5c005 (454758:454766)
by buildbot
· 8 years ago
e269fe7
Roll chromium_revision 7791a3fc00..78ea87a165 (454729:454758)
by buildbot
· 8 years ago
9660627
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
by mbonadei
· 8 years ago
3720262
Roll chromium_revision 698727682c..7791a3fc00 (454685:454729)
by buildbot
· 8 years ago
2d9e768
Revert of Setting is_component_build to false by default (patchset #5 id:80001 of https://codereview.webrtc.org/2728643003/ )
by tommi
· 8 years ago
ccf57a7
BlankDetectorDesktopCapturerWrapper to detect a blank DesktopFrame
by zijiehe
· 8 years ago
d3501ad
Create the SrtpTransportInterface.
by zhihuang
· 8 years ago
716e726
Roll chromium_revision d41fc4a850..698727682c (454644:454685)
by buildbot
· 8 years ago
a4fb2e9
Roll chromium_revision 34094a6991..d41fc4a850 (454592:454644)
by buildbot
· 8 years ago
2cb3944
Setting is_component_build to false by default
by mbonadei
· 8 years ago
02455b2
Modify TransportFeedbackPacketLossTrackerTest to use parameterization
by elad.alon
· 8 years ago
7af9357
Packet Loss Tracker - Stream Separation
by elad.alon
· 8 years ago
b85a888
Fixing race between CallbackCanceled and CancelCallback in AsyncInvoker.
by deadbeef
· 8 years ago
a1991c5
Removing HTTPS and SOCKS proxy server code.
by deadbeef
· 8 years ago
78d8977
Roll chromium_revision 84609d29db..34094a6991 (454568:454592)
by buildbot
· 8 years ago
0ebe019
Add unit tests for RTCMTLVideoView.
by denicija
· 8 years ago
446fb13
APM configuration files generator.
by alessiob
· 8 years ago
bf4007d
Evaluation controller, evaluation score abstract class, and score data access.
by alessiob
· 8 years ago
930ff23
Fixing a unittest to allow new Opus version.
by minyue
· 8 years ago
810025f
Readme file with example that explains how the conversational speech generator tool should work.
by alessiob
· 8 years ago
db23ea6
Add performance tracing for PlatformThread and parts of the video code.
by tommi
· 8 years ago
b4ec876
"audioproc_f" wrapper
by alessiob
· 8 years ago
a4b1d31
Simulation controller (shell arguments parser and simulation runner) and libraries (data access, noise generators, evaluation scores).
by alessiob
· 8 years ago
4336d73
Reland of Fix flaky EndToEndTest.TransportSeqNumOnAudioAndVideo (patchset #1 id:1 of https://codereview.webrtc.org/2730893002/ )
by danilchap
· 8 years ago
65cb70d
Fix cyclic deps: rent_a_codec<->audio_coding and rent_a_codec<->neteq
by kwiberg
· 8 years ago
154a7bb
Add Metal video view in AppRTCMobile and Metal availability macro.
by denicija
· 8 years ago
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