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gerrit-public.fairphone.software
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fp2-dev
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platform
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external
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chromium_org
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third_party
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webrtc
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964d78e5c80ec8e4cbaeece5f119806b1ef9ea22
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modules
09b40ec
Add experimental noise suppression dummy API.
by aluebs@webrtc.org
· 11 years ago
8b0791c
Fix DesktopAndCursorComposer to restore frames to the original state.
by sergeyu@chromium.org
· 11 years ago
eb45a20
Adding back main() to the test. Now it is possible to choose between ACM1 and ACM2, furthermore, the test can simulate a channel with packet loss and FEC can be activated. Packet loss pattern is based on channel implementation in Channel{.cc,.h}, which currently is a determenistic pattern with 1 every 3rd packet is discarded.
by turaj@webrtc.org
· 11 years ago
4590177
Rename AutoMute to SuspendBelowMinBitrate
by henrik.lundin@webrtc.org
· 11 years ago
e028410
Hook up audio/video sync to Call.
by stefan@webrtc.org
· 11 years ago
69b1aa4
Increment RTP timestamps for padding packets
by henrik.lundin@webrtc.org
· 11 years ago
b748c9d
Fix for RTX in combination with pacing.
by stefan@webrtc.org
· 11 years ago
591be3b
Reimplementing NetEq4's AudioVector
by henrik.lundin@webrtc.org
· 11 years ago
f4fbef3
Parse next RTCP XR report block after an unsupported block type.
by asapersson@webrtc.org
· 11 years ago
26f5492
Reducing opus_test runtime to pass Android test
by minyue@webrtc.org
· 11 years ago
ff4fc2b
MIPS optimizations for AECM audio processing module
by andrew@webrtc.org
· 11 years ago
731a87b
Move audio_processing dependencies to a variable.
by andrew@webrtc.org
· 11 years ago
9965e3a
Remove ".." from include_dirs in build/common.
by pbos@webrtc.org
· 11 years ago
3051ff7
Remove unnecessary include_dirs from audio_processing.
by andrew@webrtc.org
· 11 years ago
7e97e4c
Fix for making sure that the packet in order checks are done prior to updating the last received packet state.
by stefan@webrtc.org
· 11 years ago
b4d7835
Fix for video_processor_intergration_tests to run in parallel.
by marpan@webrtc.org
· 11 years ago
93bf70f
Add missing dependencies to .isolate files
by kjellander@webrtc.org
· 11 years ago
690a03c
Fix broken build on x86 Android
by fischman@webrtc.org
· 11 years ago
1bd9a7b
Removed unused code.
by asapersson@webrtc.org
· 11 years ago
af92d3e
Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
by sheu@chromium.org
· 11 years ago
a191cb0
Remove extra copy in VideoCaptureImpl::IncomingFrameI420
by sheu@chromium.org
· 11 years ago
6baaf30
Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
by sheu@chromium.org
· 11 years ago
7773eec
Remove extra copy in VideoCaptureImpl::IncomingFrameI420
by sheu@chromium.org
· 11 years ago
a9a788e
Address Clag Analyzer issues.
by turaj@webrtc.org
· 11 years ago
72cc32a
Propagate estimated RTT from receivers to rtt observer.
by asapersson@webrtc.org
· 11 years ago
1d76e9b
Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc
by sergeyu@chromium.org
· 11 years ago
bd5d9fa
Add delay limit to ChokeFilter.
by solenberg@webrtc.org
· 11 years ago
c2b6166
Logging for BWE test framework.
by solenberg@webrtc.org
· 11 years ago
367af84
Avoid a leak in AudioCodingModuleTest.TestIsac. The leak was caught by LSAN.
by turaj@webrtc.org
· 11 years ago
991d66a
Adding tl0idx consideration for continuity
by mikhal@webrtc.org
· 11 years ago
4ce7590
Removing the threshold from the auto-mute APIs
by henrik.lundin@webrtc.org
· 11 years ago
2714c79
Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.
by sprang@webrtc.org
· 11 years ago
77035fd
Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h
by xians@webrtc.org
· 11 years ago
2ea3645
Added a "interleaved_" flag to webrtc::AudioFrame.
by xians@webrtc.org
· 11 years ago
191f4fe
Change the low-bitrate handling in BitrateControllerImpl
by henrik.lundin@webrtc.org
· 11 years ago
ecfef19
Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals.
by fischman@webrtc.org
· 11 years ago
84c1485
Add an extended filter option to audioproc.
by andrew@webrtc.org
· 11 years ago
a2d942a
Fix for incorrect RTT estimation. A too low RTT value could be estimated.
by asapersson@webrtc.org
· 11 years ago
221798a
Upgrade scoped_ptr to Chromium's latest version.
by andrew@webrtc.org
· 11 years ago
d7e9041
WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN.
by andrew@webrtc.org
· 11 years ago
45b5167
Add CurrentLayerId() to temporal layers.
by marpan@webrtc.org
· 11 years ago
4b3ff2d
Framework for testing bandwidth estimation.
by solenberg@webrtc.org
· 11 years ago
4633e15
Changing the bitrate clamping in BitrateControllerImpl
by henrik.lundin@webrtc.org
· 11 years ago
c5b5ad1
Make sure the first frame isn't dropped.
by pbos@webrtc.org
· 11 years ago
127d8ad
Minor comment fix after clang reformat.
by andrew@webrtc.org
· 11 years ago
2873c4c
MouseCursorMonitor implementation for OSX and Windows.
by sergeyu@chromium.org
· 11 years ago
fbd6969
Remove unused kPowTableFrac which causes anroid clang build failure.
by wu@webrtc.org
· 11 years ago
ba6d56c
Add MouseCursorRenderer.
by sergeyu@chromium.org
· 11 years ago
af54d4b
Add MouseCursorCapturer interface with implementation for X11.
by sergeyu@chromium.org
· 11 years ago
9653397
Roll chromium_revision 226126:228675 and fix clang warnings
by kjellander@webrtc.org
· 11 years ago
aa445e7
Make RtpData and RtpFeedback destructors public.
by stefan@webrtc.org
· 11 years ago
bec453d
Compile out unused kMinTrustedDelayMs.
by andrew@webrtc.org
· 11 years ago
9b1b525
NetEq4: Removing templatization for AudioVector
by henrik.lundin@webrtc.org
· 11 years ago
438ae6f
Remove empty line in SharedXDisplay::RemoveEventHandler.
by sergeyu@chromium.org
· 11 years ago
cbde20c
Android OpenSlDemo: remove some usages of deprecated APIs that is breaking the bots.
by henrike@webrtc.org
· 11 years ago
07e0f6c
Add event handling in SharedXDisplay.
by sergeyu@chromium.org
· 11 years ago
91685dc
Add DesktopCaptureOptions class.
by sergeyu@chromium.org
· 11 years ago
cb90617
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
2f40af3
clang-format audio_processing/aec/*
by andrew@webrtc.org
· 11 years ago
17fdf2a
Add a parameter to audioproc for overriding the delay.
by andrew@webrtc.org
· 11 years ago
757a950
Revert r4934 "Add a tool for parsing an RTP file and outputting the BWE relevant fields."
by stefan@webrtc.org
· 11 years ago
244d629
Fix build error in r4934.
by stefan@webrtc.org
· 11 years ago
73063f3
Add a tool for parsing an RTP file and outputting the BWE relevant fields.
by stefan@webrtc.org
· 11 years ago
0a1c75a
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
bda9cbe
Accounting for wrap-around of timestamps.
by turaj@webrtc.org
· 11 years ago
0640850
VPM: Fixing namespace
by mikhal@webrtc.org
· 11 years ago
3213616
Android: enable camera video stabilization when available.
by fischman@webrtc.org
· 11 years ago
bf1da46
Add APK and isolate target for video_engine_tests
by kjellander@webrtc.org
· 11 years ago
73dacd4
Only declare kDelayDiffOffset when used.
by andrew@webrtc.org
· 11 years ago
fae046e
Unbreaks Android build after r4915.
by henrike@webrtc.org
· 11 years ago
3f02f98
Revert r4913 that reverts r4911. Original CL description:
by andresp@webrtc.org
· 11 years ago
4b14e5a
Android standalone: remove some usages of deprecated APIs and prevent further regressions.
by fischman@webrtc.org
· 11 years ago
81cd5ca
VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android.
by fischman@webrtc.org
· 11 years ago
e98a3de
Revert 4911 "Adding temporal layer strategy that keeps base laye..."
by turaj@webrtc.org
· 11 years ago
b576a69
Reformatting VPM: First step - No functional changes.
by mikhal@webrtc.org
· 11 years ago
03ced52
Adding temporal layer strategy that keeps base layer framerate at an acceptable value.
by andresp@webrtc.org
· 11 years ago
a8532a8
Disable Receiver unittests on Android.
by turaj@webrtc.org
· 11 years ago
85cdc39
ACM test are modified to run with both ACM1 and ACM2.
by turaj@webrtc.org
· 11 years ago
22a2893
Fix include of isolate.gypi
by kjellander@webrtc.org
· 11 years ago
1b59234
Android OpenSL: Fixes faulty assertion in jni-code.
by henrike@webrtc.org
· 11 years ago
59e1db1
Remove templatization of the AudioVector test
by henrik.lundin@webrtc.org
· 11 years ago
6583dff
APK for opensl loopback.
by henrike@webrtc.org
· 11 years ago
d6da239
Added support for sending and receiving RTCP XR packets:
by asapersson@webrtc.org
· 11 years ago
053d45a
Update sampling rate and number of channels of NetEq4 if decoder is changed.
by turaj@webrtc.org
· 11 years ago
b9421ac
Remove include_dirs from utility.
by pbos@webrtc.org
· 11 years ago
2934af5
Reset audio bufer if codec changes, b/10835525.
by turaj@webrtc.org
· 11 years ago
37da9ab
Ensure adjusted "known delay" doesn't drop below zero.
by andrew@webrtc.org
· 11 years ago
0e9c399
NetEq4: Removing templatization for AudioMultiVector
by henrik.lundin@webrtc.org
· 11 years ago
24f0702
Support for CELT in NetEq4.
by turaj@webrtc.org
· 11 years ago
d4e1329
Remove include_dirs from video_render.
by pbos@webrtc.org
· 11 years ago
76a6ffb
Remove include_dirs from video_capture.
by pbos@webrtc.org
· 11 years ago
0d4d51b
Revert 4876 "Support for CELT in NetEq4."
by tina.legrand@webrtc.org
· 11 years ago
76238f6
Propagate AutoMuter interface out to VideoCodingModule
by henrik.lundin@webrtc.org
· 11 years ago
cd5c882
Support for CELT in NetEq4.
by turaj@webrtc.org
· 11 years ago
2b35b95
Change the parameters of calculating maximum decode time.
by wuchengli@chromium.org
· 11 years ago
9e035d2
Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from
by wu@webrtc.org
· 11 years ago
b503d1e
Only use -lm on Linux in ISAC.
by andrew@webrtc.org
· 11 years ago
424e0e4
With ACM2 and NetEq4, VoE fuzz test very often fails.
by minyue@webrtc.org
· 11 years ago
44f030c
Implemented AutoMuter in MediaOptimization
by henrik.lundin@webrtc.org
· 11 years ago
dc1f7e9
Remove include_dirs from pacing.
by pbos@webrtc.org
· 11 years ago
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