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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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7b2c4308759eed65ac94c3003abac530d4c5ca1d
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video_engine
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
f09f7b2
Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test.
by kjellander@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
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
a18c6e5
Removing functionality for inserting pre-encoded frames instead of raw
by mflodman@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
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
695ff2a
Add support for padding in pacer.
by stefan@webrtc.org
· 11 years ago
0016110
Setting SSRC in vie_loopback_test
by mikhal@webrtc.org
· 11 years ago
e3e4615
Use int for FPS instead of size_t.
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
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
5221d1c
Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp.
by andrew@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
39784c4
Improve vie_autotest_rtp_rtcp by reenabling important tests and reducing flakiness.
by stefan@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
460e172
Disabling a flaky expectation in vie_autotest_rtp_rtcp.cc.
by stefan@webrtc.org
· 11 years ago
ad6cade
Make sure GlxRenderer frees its resources.
by pbos@webrtc.org
· 11 years ago
eef4fd5
Adds integration test for RTX and fixes bugs found.
by stefan@webrtc.org
· 11 years ago
8f1d1a9
Fix regression where retransmission bitrate is no longer estimated.
by stefan@webrtc.org
· 11 years ago
42e1fe1
CreateEmptyFrame casts from size_t to int.
by pbos@webrtc.org
· 11 years ago
08f3ca9
FrameGenerator class for future fake capture device.
by pbos@webrtc.org
· 11 years ago
074eb20
Control new VideoEngine tests with gflags.
by pbos@webrtc.org
· 11 years ago
b59962f
Adds print out of incoming resolution.
by henrike@webrtc.org
· 11 years ago
d557734
API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay.
by turaj@webrtc.org
· 11 years ago
b7716d8
- Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp.
by solenberg@webrtc.org
· 11 years ago
141a00c
Remove <iostream> usage from loopback.cc
by pbos@webrtc.org
· 11 years ago
6169712
Suffix VcmCapturer's privates with underscore_
by pbos@webrtc.org
· 11 years ago
cca5086
Log error in ViESender::SendRTCPPacket
by hclam@chromium.org
· 11 years ago
453f9c0
Add functions to ViE API to enable/disable the absolute send time header extension.
by solenberg@webrtc.org
· 11 years ago
b9e5732
Avoid NPE crash on Android platforms that don't support getting preview framerate.
by fischman@webrtc.org
· 11 years ago
9ea8c99
Include gflags properly and X11 include order in VideoEngine.
by pbos@webrtc.org
· 11 years ago
281cff8
Include files from webrtc/.. paths in video_engine/
by pbos@webrtc.org
· 11 years ago
e8dc588
Enable WebRTC demo application on x86 Android
by fischman@webrtc.org
· 11 years ago
5a22c40
Cleanup traces in WebRTC
by hclam@chromium.org
· 11 years ago
2a9108f
New VideoEngine API implementation on top of old one, first steps.
by pbos@webrtc.org
· 11 years ago
7645e4d
Remove SetOverUseDetectorOptions and cleaned ViESharedData.
by mflodman@webrtc.org
· 11 years ago
0425392
Adding a factory to remote bitrate estimator and allow it to be set via config.
by andresp@webrtc.org
· 11 years ago
fb20e53
Fix typo in log statement. witdh should be width.
by fbarchard@google.com
· 11 years ago
d474c13
Add more tracing for key frames.
by justinlin@chromium.org
· 11 years ago
dcfeff7
Increased the limit for KViEMaxCaptureDevices from 10 to 256. See issue 1343.
by vikasmarwaha@webrtc.org
· 11 years ago
08fe40f
Updated WebRTC version to 3.31 TBR=wu@webrtc.org
by elham@webrtc.org
· 11 years ago
9062a9a
Disabled flaky codec test (RunsCodecTestWithoutErrors)
by phoglund@webrtc.org
· 11 years ago
ac6d919
Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation.
by andresp@webrtc.org
· 11 years ago
7d6e2a0
Remove TEXT(x) for BUILDINFO macros.
by pbos@webrtc.org
· 11 years ago
15c0af4
This is the first step to convert building the Android WebRTC demo to a proper GYP target, android ndk toolchains is being used to build the jni cpp files instead of using ndk-build.
by fischman@webrtc.org
· 11 years ago
f22bfed
Since the layout of the Android WebRTC demo application is fixed, if we start the demo application in portrait postion, the activity will be destroyed and then created again, force the demo application to start in landscape position to avoid activity re-creation.
by fischman@webrtc.org
· 11 years ago
af89cc3
WebRTCDemo Android doesn't hangle activity recreation correctly.
by braveyao@webrtc.org
· 11 years ago
4afe86b
Add fischman into OWNERS of WebRTCDemo Android.
by braveyao@webrtc.org
· 11 years ago
7ab7268
Fix compile errors in ViE with latest clang.
by andrew@webrtc.org
· 11 years ago
90f05ed
Clean creation of VideoEngine:
by andresp@webrtc.org
· 11 years ago
06ad384
Trigger a PLI if the duration of non-decodable frames exceeds a threshold.
by stefan@webrtc.org
· 11 years ago
98b2011
Fix clang errors in non-GYP_DEFINES=clang=1 build
by pbos@webrtc.org
· 11 years ago
b6c7447
Updated WebRTC version number to 3.30
by elham@webrtc.org
· 11 years ago
3a3a787
Get rid of some unnecessary copying when sending REMBs.
by solenberg@webrtc.org
· 11 years ago
677ca88
Removing bad code resulting in flaky test.
by pwestin@webrtc.org
· 11 years ago
36bdba4
Adding trace and changing pacing constants
by pwestin@webrtc.org
· 11 years ago
422a124
Bugfix custom call stop.
by pwestin@webrtc.org
· 11 years ago
be6ad88
WebRTCDemo Android app to route audio to headphone when it's plugged in.
by braveyao@webrtc.org
· 11 years ago
166153e
Consolidate common_audio into a single target.
by andrew@webrtc.org
· 11 years ago
4d4ba47
Fixing AV sync. Increased 2 const to allow for a bigger difference in AV sync.
by pwestin@webrtc.org
· 11 years ago
6ed1c8c
Adding buffered mode to loopback test
by mikhal@webrtc.org
· 11 years ago
1a58dd7
Removing vie file related code from vie_custom_call
by mikhal@webrtc.org
· 11 years ago
b13f394
Fixed remaining nits from Stefan
by pwestin@webrtc.org
· 11 years ago
3816c52
Fix the encoder pause logic. BUG=1691
by pwestin@webrtc.org
· 11 years ago
4eac481
Disabling avi file interface
by mikhal@webrtc.org
· 11 years ago
d430f32
Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode."
by stefan@webrtc.org
· 11 years ago
2788107
Add a default RTT to CallStats and use different values for buffered/real-time mode.
by stefan@webrtc.org
· 11 years ago
f13f1fc
Updated the sync module with a slow moving filter
by pwestin@webrtc.org
· 11 years ago
c11933f
Removed unused variable.
by mflodman@webrtc.org
· 11 years ago
bea854a
Fixing Coverity issues.
by mflodman@webrtc.org
· 11 years ago
b6e175d
Ensure build_demo.py run subprocesses with bash shell.
by kjellander@webrtc.org
· 11 years ago
06e8026
New ViE interface.
by mflodman@webrtc.org
· 11 years ago
237fe4f
Remove vim/emacs modelines from .gypi files
by pbos@webrtc.org
· 11 years ago
7bc7e02
Adding a payload type for RTX.
by mflodman@webrtc.org
· 11 years ago
9b53152
Change capture interface to use NTP capture time.
by stefan@webrtc.org
· 11 years ago
f272497
Adding playout buffer status to the voe video sync
by pwestin@webrtc.org
· 11 years ago
de1c434
Updated WebRTC version number to 3.29
by elham@webrtc.org
· 11 years ago
6e816cb
WebRTCDemo: Enable making multiple calls.
by fischman@webrtc.org
· 11 years ago
74472fe
More trace events
by hclam@chromium.org
· 11 years ago
73ebe67
Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps."
by stefan@webrtc.org
· 11 years ago
67879bc
WebRtc_Word32 -> int32_t in video_engine/
by pbos@webrtc.org
· 11 years ago
65deb26
With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps.
by stefan@webrtc.org
· 11 years ago
208a648
Always set render delay in ViEChannel::RegisterExternalDecoder.
by pbos@webrtc.org
· 11 years ago
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