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gerrit-public.fairphone.software
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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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9105cbd737daaf96bc198b7d5816b224dff09d1b
9105cbd
Set up SSRCs correctly after switching codec.
by pbos@webrtc.org
· 11 years ago
b8b2a23
Recommit CL5184
by bjornv@webrtc.org
· 11 years ago
151cd25
Refactor Remote Estimators Test into a more reusable form.
by solenberg@webrtc.org
· 11 years ago
66d634f
Revert 5184 "Small refactoring change in delay_estimator."
by bjornv@webrtc.org
· 11 years ago
2c75d4e
Small refactoring change in delay_estimator.
by bjornv@webrtc.org
· 11 years ago
801822c
Ensure that no packet stays in the pacer queue for longer than 2 seconds.
by stefan@webrtc.org
· 11 years ago
8f9da30
Create default implementation to fix issue in libjingle
by sprang@webrtc.org
· 11 years ago
4a9843f
Implement and test EncodedImageCallback in new ViE API.
by sprang@webrtc.org
· 11 years ago
2622be1
Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api.
by asapersson@webrtc.org
· 11 years ago
58b912b
Remove const in vie_rtp_rtcp, where there is conflict with
by sprang@webrtc.org
· 11 years ago
e7270f5
Faster implementation of BitRateStats.
by mikhal@webrtc.org
· 11 years ago
1a5aa03
Updated WebRTC version to 3.47 TBR=wu@webrtc.org
by elham@webrtc.org
· 11 years ago
5892ce5
Made video quality toolchain more configurable.
by phoglund@webrtc.org
· 11 years ago
c86d1c6
Add include stdlib.h to files using abs.
by stefan@webrtc.org
· 11 years ago
586becf
Add test for automatically disabling padding when no video is being captured.
by stefan@webrtc.org
· 11 years ago
5ae14be
Clear empty video frames in unittest so DrMemory will allow them to be read without an uninitialized read error.
by fbarchard@google.com
· 11 years ago
e8f79c5
Don't register iSAC-swb and iSAC-fb in NetEqDecodingTest.
by turaj@webrtc.org
· 11 years ago
8bdb87f
Fix MouseCursor to MouseCursorShape conversion in ScreenCapturerWin.
by sergeyu@chromium.org
· 11 years ago
5fd393f
Fix issues with sequence number wrap-around in jitter statistics.
by turaj@webrtc.org
· 11 years ago
6508af1
Distinguish instances of ACM1 from ACM2 by a version string. This is fpr testing purposes and will be removed when the experiment is done and ACM1 is fade out.
by turaj@webrtc.org
· 11 years ago
44b21e7
Replace VideoFrameI420 with I420VideoFrame.
by pbos@webrtc.org
· 11 years ago
51fa6ac
Don't reset the AEC filter in extended mode.
by andrew@webrtc.org
· 11 years ago
ce4a0b8
Add exception handling when configuring MediaCodc in order to prevent break in the new sdk release.
by dwkang@webrtc.org
· 11 years ago
970c5e5
Renaming ViEEncoderObserver::VideoSuspended
by henrik.lundin@webrtc.org
· 11 years ago
a706baf
Protect reads of ViEEncoder::video_suspended_.
by pbos@webrtc.org
· 11 years ago
9c15a62
Increase size of pacer window to 500 ms as that better matches the encoder.
by stefan@webrtc.org
· 11 years ago
d7d60c8
Connect pacer/padding to SuspendBelowMinBitrate
by henrik.lundin@webrtc.org
· 11 years ago
0e6558b
Lock access to ModuleRtpRtcpImpl::simulcast_.
by pbos@webrtc.org
· 11 years ago
f8486d0
Rename DestroyStream methods to include Video.
by pbos@webrtc.org
· 11 years ago
b87f528
Fix issues with sequence number wrap-around in jitter statistics
by henrik.lundin@webrtc.org
· 11 years ago
3c3a953
Add -Wnon-virtual-dtor warning for C++ code.
by pbos@webrtc.org
· 11 years ago
e92aec9
Make interface destructor virtual
by sprang@webrtc.org
· 11 years ago
3fe2e7f
Added API for enabling/disabling RTCP Receiver Reference Time extension.
by asapersson@webrtc.org
· 11 years ago
402f34c
Increase run-time for full stack test for the rtt to be added reliably to the delay measurement.
by asapersson@webrtc.org
· 11 years ago
fa7ac56
Typo in vie_autotest_win.cc
by braveyao@webrtc.org
· 11 years ago
36fb531
Fixes a crash in VoE when unregistering JNI hooks.
by henrike@webrtc.org
· 11 years ago
13a4d31
Interface changes to old api, for use by new api transition.
by sprang@webrtc.org
· 11 years ago
b06a926
Added ViE API for getting overuse measure.
by asapersson@webrtc.org
· 11 years ago
04bcc9d
Deliver I420VideoFrames from VideoRender module.
by pbos@webrtc.org
· 11 years ago
c2162d1
Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module).
by asapersson@webrtc.org
· 11 years ago
f3b4602
Rename newapi::Transport::SendRTP()->SendRtp().
by pbos@webrtc.org
· 11 years ago
60108c2
Rename RTP-extension constants.
by pbos@webrtc.org
· 11 years ago
48cc9dc
Rename video streams' start/stop methods.
by pbos@webrtc.org
· 11 years ago
162021c
Rename Call::Create{Receive,Send}Stream().
by pbos@webrtc.org
· 11 years ago
4bfa866
Add experimental noise suppression dummy API.
by aluebs@webrtc.org
· 11 years ago
2b9794b
Fix DesktopAndCursorComposer to restore frames to the original state.
by sergeyu@chromium.org
· 11 years ago
dbc2a35
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
8fdf191
Rename AutoMute to SuspendBelowMinBitrate
by henrik.lundin@webrtc.org
· 11 years ago
8d2354a
Fix test broken with r5128.
by stefan@webrtc.org
· 11 years ago
26a736f
Hook up audio/video sync to Call.
by stefan@webrtc.org
· 11 years ago
5eca0c7
Fix breakage after introducing new test.
by stefan@webrtc.org
· 11 years ago
f8c47a1
Improve Call tests for RTX.
by stefan@webrtc.org
· 11 years ago
90e2fdd
Increment RTP timestamps for padding packets
by henrik.lundin@webrtc.org
· 11 years ago
8f2997c
Implement VideoSendStream::SetCodec().
by pbos@webrtc.org
· 11 years ago
764b28e
Disable all vie_auto_tests on Linux for now (take 2)
by kjellander@webrtc.org
· 11 years ago
d8dc0f5
Disable all automated vie_auto_tests on Linux for now
by kjellander@webrtc.org
· 11 years ago
04281a4
Fix for RTX in combination with pacing.
by stefan@webrtc.org
· 11 years ago
8dda8d2
Inject config when creating channels to override the existing one.
by turaj@webrtc.org
· 11 years ago
7cba612
Reimplementing NetEq4's AudioVector
by henrik.lundin@webrtc.org
· 11 years ago
0db738b
Parse next RTCP XR report block after an unsupported block type.
by asapersson@webrtc.org
· 11 years ago
c824f2c
Reducing opus_test runtime to pass Android test
by minyue@webrtc.org
· 11 years ago
e5efa32
MIPS optimizations for AECM audio processing module
by andrew@webrtc.org
· 11 years ago
7821bd1
Move audio_processing dependencies to a variable.
by andrew@webrtc.org
· 11 years ago
590c60f
Remove ".." from include_dirs in build/common.
by pbos@webrtc.org
· 11 years ago
01966bb
Remove unnecessary include_dirs from audio_processing.
by andrew@webrtc.org
· 11 years ago
6dc6e03
Remove unneeded includes from trace_posix.cc.
by andrew@webrtc.org
· 11 years ago
e9274ae
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
6106bbc
Fix log build error for Chromium builds.
by henrikg@webrtc.org
· 11 years ago
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
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