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gerrit-public.fairphone.software
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webrtc
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d6f67435fd0265d0765c80a91880c0465723ade2
d6f6743
CQ: Add linux_ubsan to default trybots.
by kjellander@webrtc.org
· 9 years ago
2080196
Cleanup of webrtc::VideoFrame.
by nisse
· 9 years ago
c63f79a
Fix ubsan warning in byteio_unittest
by sprang
· 9 years ago
e31dc95
Make pbos owner of additional video files.
by nisse
· 9 years ago
10cd6ff
Roll chromium_revision 7542f07..38664e7 (377632:377790) + set SDK 10.11 on Mac
by kjellander
· 9 years ago
686a8ef
Replace scoped_ptr with unique_ptr in webrtc/media/
by kwiberg
· 9 years ago
029e220
Removes use of DeRegister Rtp Header Extension for video
by Danil Chapovalov
· 9 years ago
74622e0
Revert of Removed unused cricket::VideoCapturer methods (patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/ )
by perkj
· 9 years ago
8067068
iSAC entropy coder: Avoid signed integer overflow
by kwiberg
· 9 years ago
db25d2e
Make VideoTrack and VideoTrackRenderers implement rtc::VideoSourceInterface.
by nisse
· 9 years ago
fc59c44
Fix lowPowerModeEnabled crash on iOS8
by tkchin
· 9 years ago
e9c0cdf
Removed unused cricket::VideoCapturer methods:
by perkj
· 9 years ago
0e40f7c
Remove incorrect reinterpret_cast from const.
by Peter Boström
· 9 years ago
6b03995
Compile rtc_api_objc on Mac.
by hjon
· 9 years ago
103c534
Roll chromium_revision 1008ac1..7542f07 (377550:377632)
by kjellander
· 9 years ago
7324eb9
Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ )
by kjellander
· 9 years ago
3dd5d1d
Remove PacketRouter sender distinction.
by Peter Boström
· 9 years ago
13041cf
Add CopyOnWriteBuffer class
by jbauch
· 9 years ago
99b345c
Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.
by kjellander@webrtc.org
· 9 years ago
a5d8e4e
Build SharedExclusiveLock in Chromium.
by Peter Boström
· 9 years ago
a2644c0
Disable tests failing under UBSan to enable deployment to main waterfall.
by kjellander@webrtc.org
· 9 years ago
75869a1
Roll chromium_revision a5f2cf0..1008ac1 (377481:377550)
by kjellander
· 9 years ago
a26ac92
Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ )
by pbos
· 9 years ago
da33a8a
Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ )
by torbjorng
· 9 years ago
91c5b56
Remove DCHECK on duplicate packets in RemoteEstimatorProxy.
by Stefan Holmer
· 9 years ago
bf66ae6
Remove thread check from PacketRouter::SendFeedback.
by Stefan Holmer
· 9 years ago
9ccedc3
Reland: Prevent data race in MessageQueue.
by jbauch
· 9 years ago
0c74ae1
MB: Fix typo in device mixin.
by kjellander@webrtc.org
· 9 years ago
b210506
Roll chromium_revision b97dbeb..a5f2cf0 (376966:377481)
by kjellander
· 9 years ago
f99af6b
Fix the gain calculation in IntelligibilityEnhancer
by aluebs
· 9 years ago
6140fcc
Move RTCFileLogger to webrtc/base/objc.
by Jon Hjelle
· 9 years ago
65c8fd7
Remove the 'audioDebugRecording' media constraint and the aec_dump AudioOptions flag.
by solenberg
· 9 years ago
861dcb7
MB: Add initial mb_config.pyl configuration file.
by kjellander@webrtc.org
· 9 years ago
4cc9f98
Fix bug 574524: DtlsTransportChannel crashes after SSL closes remotely
by guoweis
· 9 years ago
615fabb
Add looping sound button to AppRTCDemo
by Zeke Chin
· 9 years ago
f6ff971
Fix division by zero in FindTMMBRBoundingSet
by danilchap
· 9 years ago
07fb9be
Move RTCP histograms from vie_channel to video channel stats proxies.
by sprang
· 9 years ago
f14c47a
Remove ignored return code from modules.
by Peter Boström
· 9 years ago
985177c
Keep disabled RtpRtcp modules registered.
by Peter Boström
· 9 years ago
c379fcb
Break out pacer thread from CongestionController to increase testability.
by Stefan Holmer
· 9 years ago
23353ab
Increase encoder-overuse thresholds for HW.
by Peter Boström
· 9 years ago
3e60bf0
Adds low complexity audio mode for single core CPUs
by henrika
· 9 years ago
c2b785d
Replace scoped_ptr with unique_ptr in webrtc/common_audio/
by kwiberg
· 9 years ago
837b39e
Fix ubsan warnings in BWE tests.
by Stefan Holmer
· 9 years ago
f01633e
Replace scoped_ptr with unique_ptr in webrtc/modules/audio_device/
by kwiberg
· 9 years ago
58e08cb
Reset indexer upon initialization in AudioLoop.
by minyue
· 9 years ago
2346c5a
Add tools/mb to .gitignore
by Henrik Kjellander
· 9 years ago
0665f05
Fix OOB read in pacing test.
by stefan
· 9 years ago
12f4cda
Histograms for H264EncoderImpl/H264DecoderImpl initialization and errors.
by hbos
· 9 years ago
0ab8e81
Move histograms for rtp receive counters to ReceiveStatisticsProxy
by sprang
· 9 years ago
b7261fd
iSAC float: Check for end of input buffer while decoding
by kwiberg
· 9 years ago
b01c781
Added functional variants of Buffer::SetData and Buffer::AppendData.
by ossu
· 9 years ago
17849fc
Reland of Add tools/mb to setup_links.py (patchset #1 id:1 of https://codereview.webrtc.org/1691723003/ )
by kjellander
· 9 years ago
3e8043b
Roll chromium_revision 4d6ba6e..b97dbeb (376909:376966)
by kjellander
· 9 years ago
1cfe940
Added the agc digital_agc.c file to the ubsan blacklist
by peah
· 9 years ago
f75d008
Bitrate controller for VideoToolbox encoder.
by tkchin
· 9 years ago
0ed85b2
Track pending ICE restarts independently for different media sections.
by deadbeef
· 9 years ago
8df5d4f
Moved the AEC C code to be built using C++.
by peah
· 9 years ago
e80f9d0
Revert of Replace scoped_ptr with unique_ptr in webrtc/common_audio/ (patchset #4 id:60001 of https://codereview.webrtc.org/1712513002/ )
by kjellander
· 9 years ago
9788534
Removing some redundant ostringstreams declarations.
by Taylor Brandstetter
· 9 years ago
71d9721
iOS: Fix JSON for tryserver configurations.
by kjellander@webrtc.org
· 9 years ago
fffa42b
Replace scoped_ptr with unique_ptr in webrtc/audio/
by kwiberg
· 9 years ago
f4d8441
Disabled flaky tests
by philipel
· 9 years ago
77f3e0d
Screen was flickering when the picker for desktop medias showed up in Windows platform. Keeping track of window size for each window so that BitBlt() instead of PrintWindow() will be called for windows with unchanged sizes.
by gyzhou
· 9 years ago
b1eaa8d
Only average positive quality stats.
by Peter Boström
· 9 years ago
b68e02f
Revert of CQ: Disable linux_baremetal pending installation fix. (patchset #1 id:1 of https://codereview.webrtc.org/1710363002/ )
by kjellander
· 9 years ago
80e1207
Move congestion controller to a separate module.
by Stefan Holmer
· 9 years ago
ba3e25e
Simple RTCP receiver fuzzer.
by Peter Boström
· 9 years ago
79d7a49
Replace scoped_ptr with unique_ptr in webrtc/common_audio/
by kwiberg
· 9 years ago
0be9df4
Roll chromium_revision aa04eb9..4d6ba6e (376768:376909)
by Henrik Kjellander
· 9 years ago
dc0e381
Add more camera resolutions to camera scaling slider.
by Alex Glaznev
· 9 years ago
18fcbcf
Use VAD to get a better speech power estimation in the IntelligibilityEnhancer
by Alejandro Luebs
· 9 years ago
67b81f9
Tune QP thresholds for HW H.264 encoder.
by Alex Glaznev
· 9 years ago
18f9ddd
Roll chromium_revision 14bbbf2..aa04eb9 (376710:376768)
by kjellander
· 9 years ago
a094fd1
RTT intermediate calculation use ntp time instead of milliseconds.
by Danil Chapovalov
· 9 years ago
723ead8
Move simple RtpRtcp calls to VideoSendStream.
by Peter Boström
· 9 years ago
2e67ae1
Roll chromium_revision fbc4ecf..14bbbf2 (376680:376710)
by kjellander
· 9 years ago
eee7d9e
iOS: Promote iOS simulator testing to main waterfall.
by kjellander@webrtc.org
· 9 years ago
7ddc9de
Reduce the scope of rtc::Event::Wait() locking.
by Peter Boström
· 9 years ago
d1f718b
Changes in the wav_file implementation in order to
by peah
· 9 years ago
253d8fa
Simplified the function for detecting whether capture data is modified.
by peah
· 9 years ago
ada8fe5
iOS: Don't run modules_unittests on iOS simulator
by kjellander@webrtc.org
· 9 years ago
0077272
Roll chromium_revision 4101b15..fbc4ecf (376664:376680)
by kjellander
· 9 years ago
da1d656
Roll chromium_revision 789f25d..4101b15 (376663:376664)
by kjellander
· 9 years ago
b9f943d
Roll chromium_revision 1120bd3..789f25d (376660:376663)
by kjellander
· 9 years ago
a18f638
Include "sharedexclusivelock.cc" in Chromium GN build.
by jbauch
· 9 years ago
fa830dc
Roll chromium_revision 9dc1788..1120bd3 (376655:376660)
by kjellander
· 9 years ago
bf81175
Roll chromium_revision 5618e25..9dc1788 (376642:376655)
by kjellander
· 9 years ago
330d3d8
Roll chromium_revision fa5d546..5618e25 (376142:376642)
by kjellander
· 9 years ago
b9dd7c5
Remove GetTransport() from TransportChannelImpl
by mikescarlett
· 9 years ago
9bf5cde
Update build_ios_libs.sh script to build new Objective-C API and gather header files.
by hjon
· 9 years ago
91fe304
vp9: Adjust parameter for a test in videoprocessor_integrationtest.cc
by Marco
· 9 years ago
a9d0892
Add initial bitrate and frame resolution parameters to quality scaler.
by Alex Glaznev
· 9 years ago
0013dcc
Simplify SSRC usage inside ViEEncoder.
by Peter Boström
· 9 years ago
7254890
Nuke SetSenderBufferingMode.
by Peter Boström
· 9 years ago
da9ae0c
Revert of CQ: Change Android trybots to not run device tests. (patchset #1 id:1 of https://codereview.webrtc.org/1715643002/ )
by kjellander
· 9 years ago
e2d83d6
Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()
by sprang
· 9 years ago
45c44f0
Simplify EncoderStateFeedback.
by Peter Boström
· 9 years ago
9674d7c
Revert of Prevent data race in MessageQueue. (patchset #3 id:40001 of https://codereview.webrtc.org/1675923002/ )
by jbauch
· 9 years ago
fc968a2
Fix sequence-number replay race for padding.
by Peter Boström
· 9 years ago
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