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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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b7e5b2741f9cbd68e7be54e3445d28d266477c92
/
video_engine
b7e5b27
Update makefiles after merge of Chromium at 457b0a1c9412
by Android Chromium Automerger
· 10 years ago
cb45b28
Update makefiles after merge of Chromium at 041843cbf814
by Android Chromium Automerger
· 10 years ago
f1234f3
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 69370488385c14d73e6ae8a3d5001c42884f9275
by Android Chromium Automerger
· 10 years ago
5191730
Partial revert of r7014 (Android APK refactor)
by kjellander@webrtc.org
· 10 years ago
95d2195
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at f8698ce1dacfdcf804809638483adb702760469c
by Android Chromium Automerger
· 10 years ago
b9d6b2b
Android APK tests built from a normal WebRTC checkout.
by kjellander@webrtc.org
· 10 years ago
0de7d38
GN: Implement video_engine, video_capture and video_render.
by kjellander@webrtc.org
· 10 years ago
2b1b7b7
Update makefiles after merge of Chromium at b241671f0248
by Android Chromium Automerger
· 10 years ago
0e2b7ec
Remove Android.mk build files.
by pbos@webrtc.org
· 10 years ago
7a2cfc5
Remove former team members from OWNERS and WATCHLISTS
by kjellander@webrtc.org
· 10 years ago
e0ff458
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 67afd1fc176021f625e064f20ae747e23d87d727
by Android Chromium Automerger
· 10 years ago
225eac0
Bump WebRTC version number. Starting now, we will be setting WebRTC major version numbers to align with Chrome.
by tnakamura@webrtc.org
· 10 years ago
68fe1fc
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c1696da9a74c7ed4ed793ce993352bd370cfc414
by Torne (Richard Coles)
· 10 years ago
c1696da
Small refactor on ViE to remove redudant conditions and long ifdefs.
by andresp@webrtc.org
· 10 years ago
f694796
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c2ef523233552340785557abce1129a0f61537eb
by Android Chromium Automerger
· 10 years ago
d1d198b
Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics().
by stefan@webrtc.org
· 10 years ago
c2ef523
Decreased kMaxOverusesBeforeApplyRampupDelay (from 7 to 4).
by asapersson@webrtc.org
· 10 years ago
5f19242
Update makefiles after merge of Chromium at 288938
by Android Chromium Automerger
· 10 years ago
22c283b
Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798
by henrike@webrtc.org
· 10 years ago
841ee42
Remove the old H264 code now that a new H.264 packetizer has been implemented.
by stefan@webrtc.org
· 10 years ago
8661714
Update makefiles after merge of Chromium at 287308
by Android Chromium Automerger
· 10 years ago
fa50854
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 4a1b3e3a69d349b0d3e91f607f24e02d8b975688
by Android Chromium Automerger
· 10 years ago
31b38da
Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.
by minyue@webrtc.org
· 10 years ago
f3d2702
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 82383d9b14ff8e5fedf5a70229eb0ac6b512909a
by Android Chromium Automerger
· 10 years ago
6111d79
Print an info log instead of return an error if an external encoder is de-registered, but no corresponding internal encoder can be registered automatically.
by stefan@webrtc.org
· 10 years ago
15097fc
Remove the VPM denoiser.
by pbos@webrtc.org
· 10 years ago
9fbd3ec
Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. ---
by tommi@webrtc.org
· 10 years ago
55b0f2e
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
by stefan@webrtc.org
· 10 years ago
c928d36
Cast payload types to int for logging.
by pbos@webrtc.org
· 10 years ago
09da1a7
Remove the send-side cname getter APIs from voice and video engine.
by stefan@webrtc.org
· 10 years ago
477e6bc
Update makefiles after merge of Chromium at 282385
by Android Chromium Automerger
· 10 years ago
8c95e83
Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel.
by andresp@webrtc.org
· 10 years ago
10b9861
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 138adbb0bcdab60afda25a8727e5a071abc4ae36
by Android Chromium Automerger
· 10 years ago
fedbe8b
Thread annotations for vie_encoder.cc/.h
by stefan@webrtc.org
· 10 years ago
f8ec08e
Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel.
by andresp@webrtc.org
· 10 years ago
6aae61c
Some refactoring inside rtp_rtcp/.
by pbos@webrtc.org
· 10 years ago
2fd91bd
Preserve RTP states for restarted VideoSendStreams.
by pbos@webrtc.org
· 10 years ago
2d4a80c
Add boilerplate code for H.264.
by stefan@webrtc.org
· 10 years ago
65afbf3
Configure RTX send status on new modules.
by pbos@webrtc.org
· 10 years ago
c9995bc
Introduces PacedVideoSender to test framework and moves the Pacer to use Clock.
by stefan@webrtc.org
· 10 years ago
c7343a3
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at d13c3753199496aeddc73ec88548da73283c312f
by Android Chromium Automerger
· 10 years ago
07dc4be
Removed old code and default implementations.
by asapersson@webrtc.org
· 10 years ago
65a971a
Possibly fix deadlock happening due to unregister/register modules as switching between AST and TSO estimators.
by andresp@webrtc.org
· 10 years ago
88b558f
Reserve RTP/RTCP modules in SetSSRC.
by pbos@webrtc.org
· 10 years ago
b3f0584
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 9ff0df06bd431ddbf595620f94ae515bbdcde2da
by Android Chromium Automerger
· 10 years ago
07737de
Bump version number to 3.55
by tnakamura@webrtc.org
· 10 years ago
841f8c8
Update makefiles after merge of Chromium at 279716
by Android Chromium Automerger
· 10 years ago
4c21d3a
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c34b9e5d5cd44c31c4f9da649b71d0d3132cf516
by Android Chromium Automerger
· 10 years ago
3610f63
GN: Add BUILD.gn files + kjellander to OWNERS
by kjellander@webrtc.org
· 10 years ago
4ee6348
Add tests of texture frames in video_send_stream_test.
by wuchengli@chromium.org
· 10 years ago
c497bcd
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 68f4c7b51ec6434b302de9e97ee01f5ccdb48aa2
by Android Chromium Automerger
· 10 years ago
ad3bcf4
Update makefiles after merge of Chromium at 278252
by Android Chromium Automerger
· 10 years ago
e5a0f26
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 847dfa535730a30d57cf26d788d31070b70a02af
by Android Chromium Automerger
· 10 years ago
cb4fdd1
Update makefiles after merge of Chromium at 277428
by Android Chromium Automerger
· 10 years ago
c7fcada
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 5fcef2b6df45ceab39ee96a616ab0a4d3c63b83a
by Android Chromium Automerger
· 10 years ago
eddcc63
Add max limit of number for overuses. When limit is reached always apply the rampup delay.
by asapersson@webrtc.org
· 10 years ago
00dffd7
Pass GYP DEPTH variable to isolate.
by kjellander@webrtc.org
· 10 years ago
19f89a1
Enable pacing by default and remove the option to disable it from the new API.
by stefan@webrtc.org
· 10 years ago
f6eaabf
Increased kMaxRampUpDelayMs (120 to 240s).
by asapersson@webrtc.org
· 10 years ago
6845de7
Add APIs to enable padding with redundant payloads.
by stefan@webrtc.org
· 10 years ago
adda09e
Update makefiles after merge of Chromium at 276202
by Android Chromium Automerger
· 10 years ago
9cd8281
Add additional metric (relative standard deviation of encode time) for overuse detection.
by asapersson@webrtc.org
· 10 years ago
f006e8d
Add kjellander@webrtc.org as OWNER for *.isolate
by kjellander@webrtc.org
· 10 years ago
8097a46
Update makefiles after merge of Chromium at 275833
by Android Chromium Automerger
· 10 years ago
20d9f00
Update makefiles after merge of Chromium at 275661
by Android Chromium Automerger
· 10 years ago
daf186d
ViEAutoTestAndroid: Unbreak compile by casting void* to jobject.
by fischman@webrtc.org
· 10 years ago
5101f84
AppRTCDemo(android): support app (UI) & capture rotation.
by fischman@webrtc.org
· 10 years ago
431772f
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 81f8df9af96c6b4bf43234f2a0162146a5da6112
by Android Chromium Automerger
· 10 years ago
6e6292d
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 00d9c49cb076626f711988332749a0ebe8d2a32f
by Android Chromium Automerger
· 10 years ago
903e746
Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC.
by stefan@webrtc.org
· 10 years ago
00d9c49
Android: cleanup gtest_target_type conditions.
by henrike@webrtc.org
· 10 years ago
6038f4c
Update makefiles after merge of Chromium at 274467
by Android Chromium Automerger
· 10 years ago
bf2bd58
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at ff6b4a8eddca609ad2691b54f443b6f1e9342579
by Android Chromium Automerger
· 10 years ago
52dfe97
Update makefiles after merge of Chromium at 273259
by Android Chromium Automerger
· 10 years ago
47475b8
Update makefiles after merge of Chromium at 273188
by Android Chromium Automerger
· 10 years ago
1bdf186
Add support of texture frames for video capturer.
by wuchengli@chromium.org
· 10 years ago
5424828
Revert "Add support of texture frames for video capturer."
by wuchengli@chromium.org
· 10 years ago
a3b8c85
Add support of texture frames for video capturer.
by wuchengli@chromium.org
· 10 years ago
98e1ef1
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at e066d34bb747f730084f1726408ca8348ff25da7
by Android Chromium Automerger
· 10 years ago
4e95436
Added api for getting cpu measures using a struct.
by asapersson@webrtc.org
· 10 years ago
99153ba
First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class.
by asapersson@webrtc.org
· 10 years ago
f290b35
Update makefiles after merge of Chromium at 272740
by Android Chromium Automerger
· 10 years ago
01d8e22
vie_autotest_android.cc: stop referring to undefined functions.
by fischman@webrtc.org
· 10 years ago
271bf09
Update makefiles after merge of Chromium at 272566
by Android Chromium Automerger
· 10 years ago
c78d40d
Revert "Revert "Remove VideoSendStreamInput::PutFrame.""
by pbos@webrtc.org
· 10 years ago
7f54561
Fix deadlock in RegisterPreDecodeImageCallback.
by pbos@webrtc.org
· 10 years ago
dc1a607
Bump WebRTC version number to 3.54 TBR=wu@webrtc.org
by tnakamura@webrtc.org
· 10 years ago
774b3d3
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
by henrike@webrtc.org
· 10 years ago
0a9ed7c
Revert 6202 "Switch to using base/constructormagic.h and remove ..."
by mcasas@webrtc.org
· 10 years ago
28b7c07
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
by henrike@webrtc.org
· 10 years ago
5c9cc90
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 07818d1134a8fc4272ca0dd108d8f35d1753f9c3
by Android Chromium Automerger
· 10 years ago
0af662d
Update makefiles after merge of Chromium at 271215
by Android Chromium Automerger
· 10 years ago
618be31
Update makefiles after merge of Chromium at 270770
by Torne (Richard Coles)
· 10 years ago
4eff397
Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at da7c539c377367da25fc913d4399c5f0f69764ad
by Torne (Richard Coles)
· 10 years ago
bd49ac2
Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main}
by andresp@webrtc.org
· 10 years ago
dd0f8b2
Ignore the return value of UpdateRtcpTimestamp instead of printing warning. Because UpdateRtcpTimestamp may fail when there's no valid RTCP SR, which can happen in the first couple seconds or when the channel is a send only channel. Either case we don't want the warning log.
by wu@webrtc.org
· 10 years ago
2fae0d1
Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe.
by wu@webrtc.org
· 10 years ago
0b8a1c4
Add webrtc field trials API.
by andresp@webrtc.org
· 10 years ago
d658c11
Update makefiles after merge of Chromium at 269467
by Torne (Richard Coles)
· 10 years ago
3468f20
Remove WEBRTC_TRACE uses in video_engine/
by pbos@webrtc.org
· 10 years ago
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