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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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c9cb79815b494eeed0f4f3213cf6a1565b2c95ec
c9cb798
Use 3 threads for higher than 720p resolutions
by fbarchard@google.com
· 11 years ago
5b9adb0
Add a log message to see video delay break down
by hclam@chromium.org
· 11 years ago
b82ee51
Make ScreenCapturerMac work in versions of OSX before Lion.
by sergeyu@chromium.org
· 11 years ago
9ca71b1
Enable ScreenCapturer unittests
by sergeyu@chromium.org
· 11 years ago
ba458e2
Use intptr_t to represent window IDs on all platforms.
by sergeyu@chromium.org
· 11 years ago
69f7605
Wire up pacer-based padding.
by stefan@webrtc.org
· 11 years ago
67ca2b4
Revert r4145 "Revert 4127 "Switch frame list implementation to std::map.""
by stefan@webrtc.org
· 11 years ago
376ae3e
Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de...""
by stefan@webrtc.org
· 11 years ago
b06dd93
Fix AV sync issue
by hclam@chromium.org
· 11 years ago
9540e2a
Log current and target AV delay in ViESyncModule
by hclam@chromium.org
· 11 years ago
8dbc8ab
Merge more tests into modules_{unit,integration}tests.
by kjellander@webrtc.org
· 11 years ago
6828566
WebRTCDemo: ensures that using front and back camera work as expected.
by henrike@webrtc.org
· 11 years ago
19b339a
Fixes linker issue with no op trace.
by henrike@webrtc.org
· 11 years ago
7b2c430
Risk of division by zero.
by turaj@webrtc.org
· 11 years ago
cefb004
Revert 4211 "Build all java files into jar for each module on An..."
by fischman@webrtc.org
· 11 years ago
8ed5369
Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test.
by kjellander@webrtc.org
· 11 years ago
83163e0
Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test.
by kjellander@webrtc.org
· 11 years ago
789e98b
Fix breakage due to test_fec conversion to gtest.
by kjellander@webrtc.org
· 11 years ago
f04f54a
Convert test_fec to gtest
by kjellander@webrtc.org
· 11 years ago
f09f7b2
Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test.
by kjellander@webrtc.org
· 11 years ago
6ab2b1f
G722_1/G722_1C codecs won't instantiate
by tina.legrand@webrtc.org
· 11 years ago
46cec2a
Reorganize test targets in WebRTC
by kjellander@webrtc.org
· 11 years ago
0604490
Build all java files into jar for each module on Android
by fischman@webrtc.org
· 11 years ago
e8c9ecd
Allow the screen capturer to capture oversized cursors and cursors without alpha channel (Windows).
by alexeypa@chromium.org
· 11 years ago
a463a4b
Landing binary cursor image files to be used in a follow up CL.
by alexeypa@chromium.org
· 11 years ago
f1bcae0
Updated WebRTC version to 3.33
by elham@webrtc.org
· 11 years ago
b4c89a4
Making no NACK mode work again in VideoEngine.
by mflodman@webrtc.org
· 11 years ago
63988b2
RW lock access to ssrc maps in VideoCall.
by pbos@webrtc.org
· 11 years ago
203d656
Add back the WEBRTC_DIRECT_TRACE flag.
by solenberg@webrtc.org
· 11 years ago
0a86a9f
AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout()
by braveyao@webrtc.org
· 11 years ago
1125d89
Revert some variables to uint32_t to fix compile errors on Mac gcc.
by andrew@webrtc.org
· 11 years ago
64e2651
Allow audio devices with up to 64 channels on Mac.
by andrew@webrtc.org
· 11 years ago
0e7cd85
Fixed Rtp/Rtcp tests
by pwestin@webrtc.org
· 11 years ago
cfe2a74
Fix relative path to .gitignore and other minor changes.
by andrew@webrtc.org
· 11 years ago
a18c6e5
Removing functionality for inserting pre-encoded frames instead of raw
by mflodman@webrtc.org
· 11 years ago
029c3f4
Add script for appending entries to .gitignore.
by andrew@webrtc.org
· 11 years ago
9aeef32
Fix size_t to int conversion error on Win64.
by andrew@webrtc.org
· 11 years ago
a653113
Remove fake screen capturer because it's not used anywhere.
by sergeyu@chromium.org
· 11 years ago
5f545ff
Fix for STL vector function data not available.
by pwestin@webrtc.org
· 11 years ago
4aa9f1a
Connect ACM with RTP module for audio NACK.
by pwestin@webrtc.org
· 11 years ago
d0631e3
Nack for audio.
by turaj@webrtc.org
· 11 years ago
31fad0b
Fix leaks in DesktopRegion
by sergeyu@chromium.org
· 11 years ago
69bab25
Implement DetectNumberOfCores on Android and make it consistent on Linux and Android
by fischman@webrtc.org
· 11 years ago
b8171ff
Wire up Nack for Voe
by pwestin@webrtc.org
· 11 years ago
6fb2ca3
Fix init list for VideoSendStream::Config::Rtp.
by pbos@webrtc.org
· 11 years ago
6f1c3ef
Stats+Config moved into VideoSend/ReceiveStreams.
by pbos@webrtc.org
· 11 years ago
53304e8
Merge webrtc_utility_unittests into modules_unittests.
by kjellander@webrtc.org
· 11 years ago
266fc69
Restore relative include paths to libyuv.
by andrew@webrtc.org
· 11 years ago
9b82368
Issue 1847, memcopy is wrong and unnecessary, it is sufficient to store the pointer before clearing the instance, and write back the pointer.
by turaj@webrtc.org
· 11 years ago
ae05178
resolve b9050210. Avoid pushing sync packet before any packet received. Do not turn on AV-sync if initial delay is zero.
by turaj@webrtc.org
· 11 years ago
6c82a7e
Move screen capturers from chromium to webrtc.
by sergeyu@chromium.org
· 11 years ago
12bce3b
Refactor padding and rtp header functionality.
by stefan@webrtc.org
· 11 years ago
d8ecee5
Update the remote bitrate estimator before passing the packet to the RTP module.
by stefan@webrtc.org
· 11 years ago
e54928f
Remove XvRenderer.
by pbos@webrtc.org
· 11 years ago
751253d
Fix build error introduced with r4168.
by stefan@webrtc.org
· 11 years ago
695ff2a
Add support for padding in pacer.
by stefan@webrtc.org
· 11 years ago
026d1ce
Include files from webrtc/.. paths in common_video/
by pbos@webrtc.org
· 11 years ago
cff5c03
Include files from webrtc/.. paths in tools/
by pbos@webrtc.org
· 11 years ago
1cc4ed7
Disable neteq_unittests on Win x64 in code.
by kjellander@webrtc.org
· 11 years ago
ec5caf3
Disable audio_decoder_unittests on Win x64 in code.
by kjellander@webrtc.org
· 11 years ago
f4fc8ba
Disable audio_coding_unittests on Win x64 in code.
by kjellander@webrtc.org
· 11 years ago
3b35ec6
Do not hold a lock when calling VCMReceiveCallback::FrameToRender.
by fischman@webrtc.org
· 11 years ago
aec1bc8
Optimized DesktopRegion implementation.
by sergeyu@chromium.org
· 11 years ago
ad9ee0d
Removed unused class members to enable clang=1 android build.
by fischman@webrtc.org
· 11 years ago
0016110
Setting SSRC in vie_loopback_test
by mikhal@webrtc.org
· 11 years ago
915ca75
Fix error in mixing test for supported sample rates.
by andrew@webrtc.org
· 11 years ago
a80d94b
Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling.
by wu@webrtc.org
· 11 years ago
92bfbbd
Replace the old resampler with SincResampler in the voice engine signal path.
by andrew@webrtc.org
· 11 years ago
379dce7
Remove ancient and unused CNG test.
by andrew@webrtc.org
· 11 years ago
2753b76
Add dummy audio NACK APIs
by niklas.enbom@webrtc.org
· 11 years ago
40954f0
Prevent excessive logging in jitter buffer
by hclam@chromium.org
· 11 years ago
8bf7456
Revert 4104 "Refactor jitter buffer to use separate lists for de..."
by tnakamura@webrtc.org
· 11 years ago
884ff69
Revert 4127 "Switch frame list implementation to std::map."
by tnakamura@webrtc.org
· 11 years ago
1aa406c
MIPS optimizations for the following functions:
by andrew@webrtc.org
· 11 years ago
e477574
VCM/Timing: Setting clear names to members & methods
by mikhal@webrtc.org
· 11 years ago
9f62516
Fixes the frameRate stats by grouping the frames by timestamp.
by jiayl@webrtc.org
· 11 years ago
e3e4615
Use int for FPS instead of size_t.
by pbos@webrtc.org
· 11 years ago
cbd78ae
Include files from webrtc/.. paths in rtp_rtcp/
by pbos@webrtc.org
· 11 years ago
54b6ebc
Correctly set SSRCs for extra send RTP modules.
by stefan@webrtc.org
· 11 years ago
23e3f44
Remove assert for aborting FrameGeneratorCapturer.
by pbos@webrtc.org
· 11 years ago
c1506a2
Fake VideoCapturer based on FrameGenerator
by pbos@webrtc.org
· 11 years ago
4e5f983
Fix a return value mismatch introduced in r4129.
by stefan@webrtc.org
· 11 years ago
50a4d9f
Remove #pragma once
by pbos@webrtc.org
· 11 years ago
6696fba
Breaking out RTP header parsing from the RTP module.
by stefan@webrtc.org
· 11 years ago
4988d94
Break video_engine/new_include/common.h into smaller parts.
by pbos@webrtc.org
· 11 years ago
afe587e
Switch frame list implementation to std::map.
by stefan@webrtc.org
· 11 years ago
5221d1c
Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp.
by andrew@webrtc.org
· 11 years ago
2dcf742
Add comment about test_packet_masks_metrics.
by marpan@webrtc.org
· 11 years ago
3990df2
Updated WebRTC version to 3.32 TBR=wu@webrtc.org
by elham@webrtc.org
· 11 years ago
e3b52e6
Don't return an estimated receive BW for channels not receiving video.
by mflodman@webrtc.org
· 11 years ago
bb6bef5
Include gflags with "gflags/gflags.h" instead of <>
by pbos@webrtc.org
· 11 years ago
8838f68
Include "gtest/gtest.h", not by full path, on WEBRTC_ANDROID_PLATFORM_BUILD
by pbos@webrtc.org
· 11 years ago
39784c4
Improve vie_autotest_rtp_rtcp by reenabling important tests and reducing flakiness.
by stefan@webrtc.org
· 11 years ago
0c836bf
Include files from webrtc/.. paths in audio_conference_mixer/
by pbos@webrtc.org
· 11 years ago
9fb1613
Include files from webrtc/.. paths in audio_processing/
by pbos@webrtc.org
· 11 years ago
b2d1a40
Default constructors for new VideoEngine structs.
by pbos@webrtc.org
· 11 years ago
5437a2c
Remove libvpx_intrinsics_sse4_1.a in Android.mk since this target is no longer generated in libvpx
by fischman@webrtc.org
· 11 years ago
f40e9b6
- Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup.
by solenberg@webrtc.org
· 11 years ago
a93cbbf
Adding Mac test renderer, some test refactoring and made cpplint pass.
by mflodman@webrtc.org
· 11 years ago
c6d6fed
Include files from webrtc/.. paths in system_wrappers/
by pbos@webrtc.org
· 11 years ago
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