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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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e75d96b5bd3a952083b60efb860706845ce0dcb7
e75d96b
Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ )
by terelius
· 7 years ago
2d54784
Reland "Adding ANA config event to debug dump."
by minyue-webrtc
· 7 years ago
17675ce
Enable the injection of an APM into a peerconnection
by peah
· 7 years ago
fa85678
Fix FecTest.FlexfecTest flakiness caused by seq. num. wraparound.
by brandtr
· 7 years ago
697a590
Added the ability to adjust the AEC3 performance for large rooms
by peah
· 7 years ago
4583db4
Enable -Wunused-function warning everywhere.
by Henrik Kjellander
· 7 years ago
1009cfc
More gracefully handle rtp timestamp jumps in the rtp to ntp estimator.
by stefan
· 7 years ago
9addbeb
Remove RtpDemuxer tweak for preventing multiple RSID inspections
by eladalon
· 7 years ago
49085ef
Improves audio-routing in combination with BT in AppRTCMobile on Android.
by henrika
· 7 years ago
0072511
Revert "Update includes for webrtc/{base => rtc_base} rename (3/3)"
by Henrik Kjellander
· 7 years ago
f1c5ebf
Update includes for webrtc/{base => rtc_base} rename (3/3)
by kjellander
· 7 years ago
96d74bb
Opus implementation of the AudioDecoderFactoryTemplate API
by kwiberg
· 7 years ago
3aba2d1
Fix android video_quality_loopback_test
by ehmaldonado
· 7 years ago
d76b753
Disable AudioDeviceTest.StartStopRecording on iOS
by henrika
· 7 years ago
96da011
Opus implementation of the AudioEncoderFactoryTemplate API
by kwiberg
· 7 years ago
41bafb2
Update PRESUBMIT.py for webrtc/{tools => rtc_tools} rename.
by kjellander
· 7 years ago
9aed31c
Temporarily removed the analog gain change detection in AEC3
by Per Åhgren
· 7 years ago
8f9ce1d
Corrected the limit on the allowed API jitter in AEC3
by peah
· 7 years ago
d2b63cf
Move webrtc/{tools => rtc_tools}
by kjellander
· 7 years ago
cb8f045
Fix receiving FlexFEC in video_loopback.
by brandtr
· 7 years ago
5f8b04d
Higher logging severity for RED packets in UlpfecReceiverImpl.
by brandtr
· 7 years ago
2800be3
Roll chromium_revision 7aa4e8bf36..cf58257d56 (483375:483646)
by buildbot
· 7 years ago
1129df2
Always ResetSenderCongestionControlObjects before RegisterEtc...
by ossu
· 7 years ago
88af8b4
Fix -Wcomment warning in webrtcsdp.cc
by kjellander
· 7 years ago
5869f50
Support encrypted RTP extensions (RFC 6904)
by jbauch
· 7 years ago
c9be3d5
Roll chromium_revision 3691cc167a..7aa4e8bf36 (483339:483375)
by buildbot
· 7 years ago
26afe21
Properly export the symbols of video frame-buffer classes for link-time
by VladimirTechMan
· 7 years ago
06b47c5
Listen for Wifi-Direct networks and include them in the network list
by bdodson
· 7 years ago
8cf398d
Roll chromium_revision 53b56ec80b..3691cc167a (483312:483339)
by buildbot
· 7 years ago
a9cc40b
Allow an external audio processing module to be used in WebRTC
by peah
· 7 years ago
3dbfac3
Fix two simple type mismatches thay may cause compilation issues on win.
by sprang
· 7 years ago
960fd5b
Adding a presubmit check for .proto files EOF newline
by Mirko Bonadei
· 7 years ago
f1e3483
Revert "VideoFrameBuffer: Remove deprecated functions"
by Magnus Jedvert
· 7 years ago
bc8feda
Delete SignalThread class.
by nisse
· 7 years ago
428c9e2
VideoFrameBuffer: Remove deprecated functions
by Magnus Jedvert
· 7 years ago
57ca81a
Actually use virtual network in OrtcFactory unit test.
by deadbeef
· 7 years ago
8a90f87
Add SetCodecSettings method for configuring VideoCodec settings.
by asapersson
· 7 years ago
17432ec
Add magjed@ as owner of webrtc/api/video/
by Magnus Jedvert
· 7 years ago
26b1f92
Roll chromium_revision 9dd69e9f64..53b56ec80b (483005:483312)
by buildbot
· 7 years ago
d726a3f
Reland of Only compare sequence numbers from the same SSRC in ForwardErrorCorrection. (patchset #1 id:1 of https://codereview.webrtc.org/2919313005/ )
by brandtr
· 7 years ago
e435019
Don't disable FEC if timing frames are disabled.
by ilnik
· 7 years ago
8c1ee7b
Simplifies StartStopRecording test on iOS.
by henrika
· 7 years ago
8d08a92
Do not copy I420 frames in the decoder when not necessary.
by Sami Kalliomäki
· 7 years ago
b14fad4
Adding newline at the end of .proto files
by Mirko Bonadei
· 7 years ago
f4efb6f
Reland "Move webrtc/{base => rtc_base} (stub headers)
by Henrik Kjellander
· 7 years ago
c036276
Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002)
by Henrik Kjellander
· 7 years ago
ec78f1c
Revert "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002)
by Henrik Kjellander
· 7 years ago
a4c113a
Support building WebRTC without audio and video for IOS.
by zhihuang
· 7 years ago
9588682
Update memcheck suppression for HttpServer.SignalsCloseAfterForcedCloseAll
by Henrik Kjellander
· 7 years ago
9b808e7
Update TSan suppressions for base->rtc_base rename
by Henrik Kjellander
· 7 years ago
6776518
Move webrtc/{base => rtc_base}
by Henrik Kjellander
· 7 years ago
e0eb35d
Roll chromium_revision 8af690d4cd..9dd69e9f64 (482924:483005)
by buildbot
· 7 years ago
86c40a1
Fixing RTCIceCandidatePairStats.nominated for ICE controlling agent.
by deadbeef
· 7 years ago
c3e3e60
nit: Rename RtpDemuxer::sink_ to RtpDemuxer::ssrc_sinks_
by eladalon
· 7 years ago
9f789a4
LowCutFilter::BiqueadFilter::Process: Fix UBSan fuzzer bug
by Alex Loiko
· 7 years ago
d6e9466
No compliation-flag-dependent members in CriticalSection
by eladalon
· 7 years ago
3d0e7bb
Improved thread checking scheme for iOS.
by henrika
· 7 years ago
1330166
Add value_type alias to rtc::Buffer
by Danil Chapovalov
· 7 years ago
c8e0552
Limit the number of simultaneous event logs.
by terelius
· 7 years ago
41b59ca
Fix CQ_INCLUDE_TRYBOTS CL value generated by roll_deps.py
by Henrik Kjellander
· 7 years ago
3635f44
Workaround for hardware encoders crashing timing frames processing
by ilnik
· 7 years ago
03fa534
Support getting external HMAC auth context with libsrtp 2.1.0.
by jbauch
· 7 years ago
8cb4397
Roll chromium_revision 1b96d497c1..8af690d4cd (482761:482924)
by buildbot
· 7 years ago
db3c9b0
Expose ILBC codec in webrtc/api/audio_codecs/
by solenberg
· 7 years ago
cd9dd45
Revert of Roll chromium_revision 1b96d497c1..6e26093ba6 (482761:482873) (patchset #1 id:1 of https://codereview.webrtc.org/2961953002/ )
by kjellander
· 7 years ago
372e587
Fix samplingMatrix for I420Frames converted from VideoFrame.
by Sami Kalliomäki
· 7 years ago
3aa3ea7
Improve HardwareVideoDecoder stability.
by Sami Kalliomäki
· 7 years ago
912b4d5
Fix typo in roll_deps.py
by Henrik Kjellander
· 7 years ago
de4ea0c
Add linux_internal trybot to DEPS autoroll script.
by Henrik Kjellander
· 7 years ago
3dd574a
Ensure Dxgi duplicator works correctly in session 0
by zijiehe
· 7 years ago
8a67175
Roll chromium_revision 1b96d497c1..6e26093ba6 (482761:482873)
by buildbot
· 7 years ago
93a889b
Revert of Roll chromium_revision 1b96d497c1..11b33cca76 (482761:482828) (patchset #1 id:1 of https://codereview.webrtc.org/2955103004/ )
by zhihuang
· 7 years ago
cb09abd
Roll chromium_revision 1b96d497c1..11b33cca76 (482761:482828)
by buildbot
· 7 years ago
fadfc5e
Roll chromium_revision 0591491eb3..1b96d497c1 (482698:482761)
by buildbot
· 7 years ago
696f8ca
Handle the PROTO_TSL when getting the protocol priority.
by zhihuang
· 7 years ago
a7d0df7
Enable libjingle_peerconnection_datachannelonly_so target.
by Henrik Kjellander
· 7 years ago
542407c
Roll chromium_revision eafc049cce..0591491eb3 (482638:482698)
by buildbot
· 7 years ago
14f59e3
Roll chromium_revision 23503cd4bc..eafc049cce (482586:482638)
by buildbot
· 7 years ago
323197a
Attempt to reduce AUDIO_RECORD_START_STATE_MISMATCH error rate on Android.
by henrika
· 7 years ago
471f635
Allow passing in decoder factory to PeerConnectionFactory.
by Sami Kalliomäki
· 7 years ago
8179a7c
Fixing bad use of std::sort in test method.
by deadbeef
· 7 years ago
3764730
Only use 95% of the link capacity if the true link capacity is found by probing.
by terelius
· 7 years ago
4bdced5
Corrected the initialization of the AEC3
by Per Åhgren
· 7 years ago
267041c
Fix deadlock in webrtc_perf_tests
by ilnik
· 7 years ago
4847ae6
Reland of Periodically update codec bit/frame rate settings.
by sprang
· 7 years ago
f0a6fb1
Corrected the computation of the headroom in the AEC3 buffer alignment
by Per Åhgren
· 7 years ago
17c11ec
Fix building RTCCameraVideoCapturereTests with iOS 11 SDK.
by Kári Tristan Helgason
· 7 years ago
2d94f81
Roll chromium_revision 751f0a2995..23503cd4bc (482554:482586)
by buildbot
· 7 years ago
121ea32
Notify delegates about audio glitches in real time
by Anders Carlsson
· 7 years ago
93ad1f7
Reland C++ wrapper for VideoDecoder and VideoDecoderFactory interfaces.
by Sami Kalliomäki
· 7 years ago
37a2350
Revert "C++ wrapper for VideoDecoder and VideoDecoderFactory interfaces."
by Sami Kalliomäki
· 7 years ago
afd52d3
iOS: Run more tests on real devices
by Oleh Prypin
· 7 years ago
9eaf236
Roll chromium_revision 038bd90be9..751f0a2995 (482533:482554)
by buildbot
· 7 years ago
ef4342f
C++ wrapper for VideoDecoder and VideoDecoderFactory interfaces.
by Sami Kalliomäki
· 7 years ago
42c62e6
Roll chromium_revision 9ea908c113..038bd90be9 (482493:482533)
by buildbot
· 7 years ago
e00c279
Roll chromium_revision 1e7ac4345c..9ea908c113 (482425:482493)
by buildbot
· 7 years ago
1897751
Roll chromium_revision 00c0761c39..1e7ac4345c (482341:482425)
by buildbot
· 7 years ago
3ea0e92
Roll chromium_revision 41562d1c4d..00c0761c39 (482284:482341)
by buildbot
· 7 years ago
a52722f
Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ )
by eladalon
· 7 years ago
376b6fd
Roll chromium_revision 54e58e7497..41562d1c4d (482251:482284)
by buildbot
· 7 years ago
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