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fp2-dev
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platform
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chromium_org
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third_party
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webrtc
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e0df4d7230216a6495b16e754bc4ee1ac8136792
e0df4d7
Remove update_resources.py as it's no longer used.
by kjellander@webrtc.org
· 11 years ago
0aa16d7
Replace disabled logging with a restricted logging mode.
by andrew@webrtc.org
· 11 years ago
c4a7861
Updated WebRTC version to 3.46
by elham@webrtc.org
· 11 years ago
6196a56
Fix for video_processor_intergration_tests to run in parallel.
by marpan@webrtc.org
· 11 years ago
685e91a
Update getUserMedia W3C conformance tests.
by kjellander@webrtc.org
· 11 years ago
1dc0158
Sending status fix for module.
by asapersson@webrtc.org
· 11 years ago
c359e28
Add missing dependencies to .isolate files
by kjellander@webrtc.org
· 11 years ago
4e0ea6a
Fix broken build on x86 Android
by fischman@webrtc.org
· 11 years ago
65e4415
Removed unused code.
by asapersson@webrtc.org
· 11 years ago
e3709a8
Make video quality analysis unittests print to log instead of stdout.
by kjellander@webrtc.org
· 11 years ago
06977ab
Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
by sheu@chromium.org
· 11 years ago
f5fdd0c
Remove extra copy in VideoCaptureImpl::IncomingFrameI420
by sheu@chromium.org
· 11 years ago
a4a5bf2
Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
by sheu@chromium.org
· 11 years ago
987587e
Remove extra copy in VideoCaptureImpl::IncomingFrameI420
by sheu@chromium.org
· 11 years ago
565f991
Address Clag Analyzer issues.
by turaj@webrtc.org
· 11 years ago
f1262f3
Propagate estimated RTT from receivers to rtt observer.
by asapersson@webrtc.org
· 11 years ago
3c97268
Video bandwidth not reported correctly
by sprang@webrtc.org
· 11 years ago
0f78f7b
Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc
by sergeyu@chromium.org
· 11 years ago
893c229
Remove unused make_scoped_ptr which causes an "ambiguous" error with chromium build.
by wu@webrtc.org
· 11 years ago
be03ea6
Add delay limit to ChokeFilter.
by solenberg@webrtc.org
· 11 years ago
77c834d
Logging for BWE test framework.
by solenberg@webrtc.org
· 11 years ago
9a1635a
Make video/ only depend on video_engine_core.
by pbos@webrtc.org
· 11 years ago
6671434
Stop DirectTransports in VideoSendStreamTests.
by pbos@webrtc.org
· 11 years ago
267f694
Avoid a leak in AudioCodingModuleTest.TestIsac. The leak was caught by LSAN.
by turaj@webrtc.org
· 11 years ago
9ce61d4
Adding tl0idx consideration for continuity
by mikhal@webrtc.org
· 11 years ago
56290ed
Fix build/isolate.gypi path in webrtc_tests.gypi.
by pbos@webrtc.org
· 11 years ago
8e3e298
Drop ViEDecoderObserver::DecoderTiming impl now that WebRtcDecoderObserver rolled in r5038.
by fischman@webrtc.org
· 11 years ago
b581c90
Separate Call API/build files from video_engine/.
by pbos@webrtc.org
· 11 years ago
d4ec1f5
Removing the threshold from the auto-mute APIs
by henrik.lundin@webrtc.org
· 11 years ago
4043e7e
Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.
by sprang@webrtc.org
· 11 years ago
b397091
Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h
by xians@webrtc.org
· 11 years ago
d080e35
Added a "interleaved_" flag to webrtc::AudioFrame.
by xians@webrtc.org
· 11 years ago
e2f3ebc
Prefix MOVE_ONLY_TYPE_FOR_CPP_03 with WEBRTC_.
by andrew@webrtc.org
· 11 years ago
ae2b602
Change the low-bitrate handling in BitrateControllerImpl
by henrik.lundin@webrtc.org
· 11 years ago
7af2f81
Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals.
by fischman@webrtc.org
· 11 years ago
d6b231e
Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined.
by andrew@webrtc.org
· 11 years ago
cddf2b1
Add an extended filter option to audioproc.
by andrew@webrtc.org
· 11 years ago
a881576
Fix for incorrect RTT estimation. A too low RTT value could be estimated.
by asapersson@webrtc.org
· 11 years ago
ce21c82
Porting auto mute to new ViE API
by henrik.lundin@webrtc.org
· 11 years ago
7606f43
Fixing broken tests in voe_auto_test extended
by tina.legrand@webrtc.org
· 11 years ago
2ba95be
Upgrade scoped_ptr to Chromium's latest version.
by andrew@webrtc.org
· 11 years ago
e5fd264
Roll chromium_revision 228675:229708
by kjellander@webrtc.org
· 11 years ago
a597fc2
WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN.
by andrew@webrtc.org
· 11 years ago
4535225
Add CurrentLayerId() to temporal layers.
by marpan@webrtc.org
· 11 years ago
1e1938a
Updated WebRTC version to 3.45
by elham@webrtc.org
· 11 years ago
61e533c
Framework for testing bandwidth estimation.
by solenberg@webrtc.org
· 11 years ago
97d0fc6
Changing the bitrate clamping in BitrateControllerImpl
by henrik.lundin@webrtc.org
· 11 years ago
0c6fa57
AutoMute: Adding channel_id parameter to callback.
by henrik.lundin@webrtc.org
· 11 years ago
3ba57eb
Implement I420FrameCallbacks in Call.
by pbos@webrtc.org
· 11 years ago
dea5a74
Make sure the first frame isn't dropped.
by pbos@webrtc.org
· 11 years ago
05ee189
Move audio_e2e_harness into include_tests==1 condition.
by kjellander@webrtc.org
· 11 years ago
8ee76b9
Add audio_e2e_test target to tools.gyp
by kjellander@webrtc.org
· 11 years ago
6c0739e
Protect _transportPtr, which can be accessed by different threads in the case of external transport. This change avoid the potential use-after-free, e.g. the case in the reported bug.
by wu@webrtc.org
· 11 years ago
9b307a7
Have padding decay to zero if no frames are being captured.
by stefan@webrtc.org
· 11 years ago
6be4250
Disable the -Wno-unused-const-variable Clang warning on Mac
by kjellander@webrtc.org
· 11 years ago
c4579f3
Minor comment fix after clang reformat.
by andrew@webrtc.org
· 11 years ago
b746a33
MouseCursorMonitor implementation for OSX and Windows.
by sergeyu@chromium.org
· 11 years ago
224c0f5
Fix tsan failures in channel.cc regarding to the volume settings.
by wu@webrtc.org
· 11 years ago
1e6493d
Check the number of playout channels instead of the send channels in StopPlayout()
by xians@webrtc.org
· 11 years ago
b1ef0d7
Compound/reduced-size RTCP in VideoReceiveStream.
by pbos@webrtc.org
· 11 years ago
de16548
Remove unused kPowTableFrac which causes anroid clang build failure.
by wu@webrtc.org
· 11 years ago
6133dd5
Fixed issue with how MTU is calculated.
by sprang@webrtc.org
· 11 years ago
e44b42d
Don't pad if only one stream is sent, except if auto muted.
by stefan@webrtc.org
· 11 years ago
a19dab9
Revert "Disable tests for TSan v2"
by kjellander@webrtc.org
· 11 years ago
4fe8543
Wired up max packet size and added simple test.
by sprang@webrtc.org
· 11 years ago
55ca27e
Run FullStack tests without render windows.
by pbos@webrtc.org
· 11 years ago
1c83344
Remove TSan v2 disabled test in condition_variable_unittest.cc
by kjellander@webrtc.org
· 11 years ago
976adc0
Open file in binary in CreateFromYuvFile().
by pbos@webrtc.org
· 11 years ago
28f6166
Add MouseCursorRenderer.
by sergeyu@chromium.org
· 11 years ago
92da5d7
Add MouseCursorCapturer interface with implementation for X11.
by sergeyu@chromium.org
· 11 years ago
0f281aa
Roll chromium_revision 226126:228675 and fix clang warnings
by kjellander@webrtc.org
· 11 years ago
7865560
Make RtpData and RtpFeedback destructors public.
by stefan@webrtc.org
· 11 years ago
2390091
Move ChromaGenerator to common_video/.
by pbos@webrtc.org
· 11 years ago
8dc840d
Compile out unused kMinTrustedDelayMs.
by andrew@webrtc.org
· 11 years ago
c1b7718
Android: Fixes WebRTCDemo build (missing Java code).
by henrike@webrtc.org
· 11 years ago
53fa5da
NetEq4: Removing templatization for AudioVector
by henrik.lundin@webrtc.org
· 11 years ago
f24a93f
Remove empty line in SharedXDisplay::RemoveEventHandler.
by sergeyu@chromium.org
· 11 years ago
a171eae
Android OpenSlDemo: remove some usages of deprecated APIs that is breaking the bots.
by henrike@webrtc.org
· 11 years ago
5e8b020
Add event handling in SharedXDisplay.
by sergeyu@chromium.org
· 11 years ago
a6295d3
Add DesktopCaptureOptions class.
by sergeyu@chromium.org
· 11 years ago
4b795a1
WebRTCDemo: Fixes warning for devices with pre-17 API level. Also fixes broken build build.xml and project.properties.
by henrike@webrtc.org
· 11 years ago
0b7aefe
Reorganize GYP targets to make webrtc.gyp more usable.
by kjellander@webrtc.org
· 11 years ago
f50f118
clang-format audio_processing/aec/*
by andrew@webrtc.org
· 11 years ago
fd03cb1
Add a parameter to audioproc for overriding the delay.
by andrew@webrtc.org
· 11 years ago
472d9a7
Updated WebRTC version to 3.44 TBR=wu@webrtc.org
by elham@webrtc.org
· 11 years ago
96ea7ac
Revert r4934 "Add a tool for parsing an RTP file and outputting the BWE relevant fields."
by stefan@webrtc.org
· 11 years ago
2dd26d8
Fix build error in r4934.
by stefan@webrtc.org
· 11 years ago
6dea08f
Add a tool for parsing an RTP file and outputting the BWE relevant fields.
by stefan@webrtc.org
· 11 years ago
e3976cf
Wrap ACM2 code inside acm2 namespace. This gurantees that one ACM would not use components of others by accident.
by turaj@webrtc.org
· 11 years ago
e9a3119
Accounting for wrap-around of timestamps.
by turaj@webrtc.org
· 11 years ago
7a4ff8a
VPM: Fixing namespace
by mikhal@webrtc.org
· 11 years ago
d4b124a
Android: enable camera video stabilization when available.
by fischman@webrtc.org
· 11 years ago
7de0054
Add owners to [webrtc,talk]/build and *.isolate (take 2)
by kjellander@webrtc.org
· 11 years ago
b7768d5
Remove unused Android dummy APK
by kjellander@webrtc.org
· 11 years ago
9670be6
Add APK and isolate target for video_engine_tests
by kjellander@webrtc.org
· 11 years ago
28ea6f8
Clean up AudioProcessing defaults and errors.
by andrew@webrtc.org
· 11 years ago
9466714
Add owners to [webrtc,talk]/build and *.isolate
by kjellander@webrtc.org
· 11 years ago
d64f84f
Only declare kDelayDiffOffset when used.
by andrew@webrtc.org
· 11 years ago
a299658
Unbreaks Android build after r4915.
by henrike@webrtc.org
· 11 years ago
b17cc30
Revert r4913 that reverts r4911. Original CL description:
by andresp@webrtc.org
· 11 years ago
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