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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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66b0c5d4787a4c878200b86001e28b7b7f3bf9a6
/
video_engine
87d8f2d
Updated version number to 3.25
by elham@webrtc.org
· 11 years ago
3da576e
Update integration tests for idempotent RTP header settings.
by bemasc@google.com
· 11 years ago
1dcba31
Destroy VCM and VPM instead of delete.
by mflodman@webrtc.org
· 11 years ago
ca65c51
Handle multiple calls to set initial delay
by mikhal@webrtc.org
· 11 years ago
213217c
Stop and restart fix.
by mflodman@webrtc.org
· 11 years ago
2325284
Fixed typo in vie_autotest_loopback.cc.
by pbos@webrtc.org
· 11 years ago
cb139b1
Rename webrtc::StatsObserver to webrtc::CallStatsObserver
by fischman@webrtc.org
· 11 years ago
432bc1a
fixing nack list size calculation
by mikhal@webrtc.org
· 11 years ago
39eb955
Updated version number to 3.24
by elham@webrtc.org
· 11 years ago
85e2e0e
Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings.
by stefan@webrtc.org
· 11 years ago
ce3f2ca
Add VoE interface to VieRTP test
by mikhal@webrtc.org
· 11 years ago
4db69af
Adding a receive side API for buffering mode.
by mikhal@webrtc.org
· 11 years ago
64506e2
Roll Chromium revision 176094:182149
by kjellander@webrtc.org
· 11 years ago
e740a7b
Remove MultiStreamMode from test.
by stefan@webrtc.org
· 11 years ago
4c6689a
Reset ssrc when calling SetSendCodec.
by mflodman@webrtc.org
· 11 years ago
33c6e92
Sync libvpx and its gyp wrapper from Chromium.
by andrew@webrtc.org
· 11 years ago
1fb8372
Increase maximum resolution to 4k x 3k.
by fbarchard@google.com
· 11 years ago
9c4707e
Android NDK build tools
by kjellander@webrtc.org
· 11 years ago
4da62e0
Set SingleStream BWE in unittests.
by stefan@webrtc.org
· 11 years ago
6cd34e5
Updates to send side streaming mode:
by mikhal@webrtc.org
· 11 years ago
6bcf2ab
Update version number to 3.23
by tnakamura@webrtc.org
· 11 years ago
75e6669
Made it possible to render custom call output to file.
by phoglund@webrtc.org
· 11 years ago
89c3de3
Don't report an error for GetEstimatedReceiveBandwidth if there is no valid
by mflodman@webrtc.org
· 11 years ago
34d1110
Enable indefinitely running vie_auto_test option
by kjellander@webrtc.org
· 11 years ago
db325e2
Updated version number to 3.22
by elham@webrtc.org
· 11 years ago
cc895d1
Fixing/disabling Windows x64 warnings
by kjellander@webrtc.org
· 11 years ago
d6739c8
Adding a send side API for streaming
by mikhal@webrtc.org
· 11 years ago
a7761c7
Fix mismatch between different NACK list lengths and packet buffers.
by stefan@webrtc.org
· 11 years ago
3442158
Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages.
by stefan@webrtc.org
· 11 years ago
fd2dd1a
Set frame length for frame converting in external renderer
by braveyao@webrtc.org
· 11 years ago
8d759af
VP8: Making key frame interval a tunnable parameter
by mikhal@webrtc.org
· 11 years ago
b4575c1
Fix webrtc compilation errors for Chrome Win64
by andrew@webrtc.org
· 11 years ago
ceca869
Moving ViE test files and deleting files no longer used.
by mflodman@webrtc.org
· 12 years ago
3d7848b
Updated version number to 3.21
by elham@webrtc.org
· 12 years ago
81cfcb5
Remove '<(library)' in gyp files.
by wjia@webrtc.org
· 12 years ago
fc37398
Convert psnr and ssim to strings before printing them.
by stefan@webrtc.org
· 12 years ago
087c593
Removing outdated comment
by mikhal@webrtc.org
· 12 years ago
32ad4a4
Made ViEToFileRenderer use a separate thread for rendering frames to file.
by stefan@webrtc.org
· 12 years ago
1b23416
logical 'and' of mutually exclusive tests is always false in ViECodecImpl::CodecValid()
by braveyao@webrtc.org
· 12 years ago
ee92f9d
Disable full stack PSNR/SSIM triggers on Mac and Win for now due to flakiness. Adding plots of PSNR and SSIM.
by stefan@webrtc.org
· 12 years ago
1d4568f
Disable PSNR/SSIM thresholds for the Gilber-Elliot test.
by stefan@webrtc.org
· 12 years ago
b011c6a
Generalized mechanism for excluding gtests on platforms, disabled broken tests on mac.
by phoglund@webrtc.org
· 12 years ago
c702d28
Disabled GQoS since it breaks ViE auto test.
by henrika@webrtc.org
· 12 years ago
8be968f
Enable external encoders with internal picture source.
by stefan@webrtc.org
· 12 years ago
e91de87
Using Convert in lieu of ExtractBuffer: Less error prone (as we don't need to compute buffer sizes etc.). This cl is first in a series (doing all of WebRtc would make it quite a big cl). While at it, fixing a few headers.
by mikhal@webrtc.org
· 12 years ago
8be556d
Updated version number to 3.20
by elham@webrtc.org
· 12 years ago
5e22650
Removed spaces from full stack test labels, consolidated graphs
by phoglund@webrtc.org
· 12 years ago
c54e675
Changed assert to log.
by mflodman@webrtc.org
· 12 years ago
42264f2
Make protection method, filename and resolution configurable for FullStackTest.
by stefan@webrtc.org
· 12 years ago
c302ff2
vie auto test: Adding a constructor for NetworkParameters
by mikhal@webrtc.org
· 12 years ago
b8029db
ViE autotest: Adding loss models to the external transport
by mikhal@webrtc.org
· 12 years ago
b4e5d10
Updated version number to 3.19
by elham@webrtc.org
· 12 years ago
b36efe3
Added API to get receive side video delay.
by mflodman@webrtc.org
· 12 years ago
da80bad
Remove latency excl network and add render time diff stats.
by stefan@webrtc.org
· 12 years ago
2188300
Fix for buffer overflow, WebRTC issue 1196
by elham@webrtc.org
· 12 years ago
4c8b31e
Added jitter to fake network pipe.
by mflodman@webrtc.org
· 12 years ago
2b6c051
Track the actual render time rather than the decode time.
by stefan@webrtc.org
· 12 years ago
f0bf6f6
Will now only require near-perfect PSNR and SSIM.
by phoglund@webrtc.org
· 12 years ago
7940bbb
Revert 3269
by andrew@webrtc.org
· 12 years ago
6db19bd
Will now only require near-perfect PSNR and SSIM.
by phoglund@webrtc.org
· 12 years ago
c1b0f7d
Use TRACE_EVENT to track time spent in VP8 encoding
by hclam@chromium.org
· 12 years ago
fb537e2
Add a third full stack test and support for random jitter in ext transport.
by stefan@webrtc.org
· 12 years ago
2082b3a
Adding a simple fake network pipe to use for testing. Next CL will contain an external transport implementation using this link and I'll follow up later making this more advanced.
by mflodman@webrtc.org
· 12 years ago
3e5b30b
Add more audio codec information into codec list
by leozwang@webrtc.org
· 12 years ago
9470f64
Added auto-call feature to WebRTCDemo.
by fischman@webrtc.org
· 12 years ago
e5a2710
Adds two full stack performance metrics for end-to-end delay.
by stefan@webrtc.org
· 12 years ago
0cf911a
First pass of MediaCodecDecoder which uses Android MediaCodec API.
by dwkang@webrtc.org
· 12 years ago
3f4b16d
Delete {start,stop}CPULoad() since they're broken.
by fischman@webrtc.org
· 12 years ago
bc23d31
Enable building WebRTCDemo apk using Release webrtc libs, take 2.
by fischman@webrtc.org
· 12 years ago
8adceb0
Corrected .h path.
by phoglund@webrtc.org
· 12 years ago
a7920db
Fixed standard PSNR/SSIM test.
by phoglund@webrtc.org
· 12 years ago
2255427
Properly remove the bitrate observer when ViEEncoder is destructed.
by stefan@webrtc.org
· 12 years ago
0c9d201
Disabled some more flaky tests. Memcheck vie_auto_test should be very stable after this.
by phoglund@webrtc.org
· 12 years ago
4bbb260
Revert 3190 - Enable building WebRTCDemo apk using Release webrtc libs.
by fischman@webrtc.org
· 12 years ago
4a2fab0
Enable building WebRTCDemo apk using Release webrtc libs.
by fischman@webrtc.org
· 12 years ago
5dea525
Remove ringtone from test app
by leozwang@webrtc.org
· 12 years ago
4591d9b
Fixing vie and voe auto test project paths for test execution.
by kjellander@webrtc.org
· 12 years ago
73d3490
Updated version number to 3.18
by elham@webrtc.org
· 12 years ago
6318790
Wire up CallStats to provide modules with correct RTT.
by mflodman@webrtc.org
· 12 years ago
0993f8b
Fixes (or at least reduces) the flakiness in the full stack test by making sure the different frame monitors are registered and deregistered in the right order. Also makes sure only local preview frames which are actually transmitted are rendered by moving the local preview rendering to an effect filter.
by stefan@webrtc.org
· 12 years ago
1ec1bc9
Removed codec comparison test: it didn't work and probably never will.
by phoglund@webrtc.org
· 12 years ago
b743278
Remove ViE lint warnings that should have been caught at upload time.
by mflodman@webrtc.org
· 12 years ago
a39ac68
Reorganize gyp for Android
by leozwang@webrtc.org
· 12 years ago
020b350
Fix possible race condition and access into an empty list.
by stefan@webrtc.org
· 12 years ago
d51d166
Move SSRC list to RemoteBitrateEstimator.
by stefan@webrtc.org
· 12 years ago
8a8517a
Adding ViE CallStats to keep track of call statistics. As a start, only rtt is handled.
by mflodman@webrtc.org
· 12 years ago
36fdd24
Replaced remb unittest sleep with fake clock.
by mflodman@webrtc.org
· 12 years ago
8dd4b98
Revert 3111 (revert of a revert).
by tommi@webrtc.org
· 12 years ago
b159bd8
Revert 3105 - Don't crash the unit test host when tests fail.
by mikhal@webrtc.org
· 12 years ago
6be5b2f
Don't crash the unit test host when tests fail.
by tommi@webrtc.org
· 12 years ago
c06c66d
Fixed test memory leak + disabled base test.
by phoglund@webrtc.org
· 12 years ago
68783ae
Add libpaced_sender to Android makefile
by leozwang@webrtc.org
· 12 years ago
32f05a7
Enable paced sender. Review URL: https://webrtc-codereview.appspot.com/965016
by pwestin@webrtc.org
· 12 years ago
c4f9c04
Fixes an incorrect if statement in vie_sync_module.cc.
by stefan@webrtc.org
· 12 years ago
b95093a
Re-initialize enough state on "Stop Call" to be able to stop/start multiple calls in succession.
by fischman@webrtc.org
· 12 years ago
3f78c6c
Add Android OWNER files
by leozwang@webrtc.org
· 12 years ago
7121008
Can now fully control custom calls from the command line.
by phoglund@webrtc.org
· 12 years ago
215428c
Adding codecType to OnIncomingCapturedEncodedFrame partially reverting r3013.
by mikhal@webrtc.org
· 12 years ago
bf4bba9
Made TickTime immutable, rewrote tick utils to be fakeable.
by phoglund@webrtc.org
· 12 years ago
77085af
Removed ViEBaseObserver.
by mflodman@webrtc.org
· 12 years ago
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