Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
10894996efb088c305db07365c1871e02c98a818
1089499
Fix timing frames and FEC conflict
by ilnik
· 8 years ago
83c97da
Only append SPS/PPS to bitstream if supplied out of band.
by philipel
· 8 years ago
548813a
Roll chromium_revision 97f626b505..f81fb7a573 (481106:481183)
by buildbot
· 8 years ago
73e2180
Add webrtc/rtc_base skeleton.
by Henrik Kjellander
· 8 years ago
1b2469b
Fix AVFoundation framework import
by hansknoechel92
· 8 years ago
0789dab
Revert "Support more formats in RTCVideoFrame"
by Anders Carlsson
· 8 years ago
0f15f92
Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface.
by nisse
· 8 years ago
130ca7e
Reland of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2949953003/ )
by zhihuang
· 8 years ago
c2e208a
Revert of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2945233002/ )
by zhihuang
· 8 years ago
9ed1609
Try to fix the binary size increase issue on Chromium.
by zhihuang
· 8 years ago
37aa8ba
Test and fix for huge bwe drop after alr state.
by tschumim
· 8 years ago
bd2220a
Support more formats in RTCVideoFrame
by Anders Carlsson
· 8 years ago
7f84aea
Roll chromium_revision 1b223e58f3..97f626b505 (481050:481106)
by buildbot
· 8 years ago
734e3a7
Roll chromium_revision 2dc643d0f9..1b223e58f3 (480970:481050)
by buildbot
· 8 years ago
5a9c27c
Roll chromium_revision 779c46499b..2dc643d0f9 (480899:480970)
by buildbot
· 8 years ago
556ddc5
Roll chromium_revision fdee024fd5..779c46499b (480836:480899)
by buildbot
· 8 years ago
3814524
Create VideoDecoderFactory interface and implementation.
by Bjorn Mellem
· 8 years ago
b080b46
Create a hardware VideoDecoder implementation using Android MediaCodec.
by Bjorn Mellem
· 8 years ago
fde2116
Use constexpr to avoid a static initializer
by brucedawson
· 8 years ago
26b16f7
Roll chromium_revision b2c019fd75..fdee024fd5 (480797:480836)
by buildbot
· 8 years ago
a2af000
Improve the simulation stats aggregation in neteq_rtpplay
by Henrik Lundin
· 8 years ago
2b3e061
Hotfix for psnr regresion with fec tests caused by timing frames.
by ilnik
· 8 years ago
0bc0ccd
Add Matlab plotting script generator to neteq_rtpplay
by Henrik Lundin
· 8 years ago
4b15afe
Roll chromium_revision 4ae8d1757f..b2c019fd75 (480783:480797)
by buildbot
· 8 years ago
8c51282
Added new AudioProcessing fuzzer
by aleloi
· 8 years ago
be45757
Add henrik.lundin to fuzzers OWNERS
by Henrik Lundin
· 8 years ago
223e3d4
Roll chromium_revision dd87f36fc7..4ae8d1757f (480756:480783)
by buildbot
· 8 years ago
c8ece43
Minor updates to VideoReceiveStream:
by tommi
· 8 years ago
9024248
Roll chromium_revision f1878113fa..dd87f36fc7 (480719:480756)
by buildbot
· 8 years ago
a9b848a
Bugfix:setting capture framerate always defaults to 30fps.
by Daniela
· 8 years ago
e8d2bd6
Roll chromium_revision 8a0e666385..f1878113fa (480667:480719)
by buildbot
· 8 years ago
0463fe5
Roll chromium_revision 0a95e6bab5..8a0e666385 (480594:480667)
by buildbot
· 8 years ago
cadd306
Fix test break by the recent changes in IcerServer
by Emad Omara
· 8 years ago
efcc131
Roll chromium_revision ba61076b34..0a95e6bab5 (480516:480594)
by buildbot
· 8 years ago
6dd77c4
Add reference counter of DxgiDuplicatorController to unload DXGI components
by zijiehe
· 8 years ago
a87675d
Roll chromium_revision f471163c11..ba61076b34 (480455:480516)
by buildbot
· 8 years ago
28e546d
Roll chromium_revision df32089dae..f471163c11 (480415:480455)
by buildbot
· 8 years ago
3352ce9
Android: Modular WebRTC follow-up
by Magnus Jedvert
· 8 years ago
42308f6
Fix uploading of available send bitrate statistics.
by Alex Narest
· 8 years ago
ce433fa
Revert "Adding ANA config event to debug dump."
by Minyue Li
· 8 years ago
bfe45c2
Use uint8 pointer instead of std::vector in NV12Scale.
by Anders Carlsson
· 8 years ago
652abc9
Adding ANA config event to debug dump.
by minyue-webrtc
· 8 years ago
3093ef1
Android JNI: Clean up AndroidVideoTrackSource and NativeHandleImpl
by Magnus Jedvert
· 8 years ago
04f4d12
Implement timing frames.
by ilnik
· 8 years ago
3b921f0
Roll chromium_revision 4b74fa1307..df32089dae (480384:480415)
by buildbot
· 8 years ago
1f7476f
Remove explicit draw call on MTKView.
by Daniela
· 8 years ago
91047e5
Remove redundant std::min from ProbeBitrateEstimator.
by terelius
· 8 years ago
76b20b7
Roll chromium_revision 2390071bb3..4b74fa1307 (480364:480384)
by buildbot
· 8 years ago
bed7a6b
Use information about blacklisted devices in video_quality_loopback_test
by oprypin
· 8 years ago
429d614
Roll chromium_revision add3c68a6c..2390071bb3 (480340:480364)
by buildbot
· 8 years ago
edf2859
Roll chromium_revision ed82d45fc0..add3c68a6c (480324:480340)
by buildbot
· 8 years ago
fb8cf3c
Roll chromium_revision e438353b8b..ed82d45fc0 (480311:480324)
by buildbot
· 8 years ago
0393de4
Roll chromium_revision b032878ebd..e438353b8b (480186:480311)
by kjellander
· 8 years ago
1a610f1
Revert of Opus implementation of the AudioEncoderFactoryTemplate API (patchset #4 id:80001 of https://codereview.webrtc.org/2930243003/ )
by charujain
· 8 years ago
eb2d2d3
Revert of Opus implementation of the AudioDecoderFactoryTemplate API (patchset #1 id:1 of https://codereview.webrtc.org/2942733003/ )
by charujain
· 8 years ago
af62935
Support building WebRTC without audio and video for Android
by zhihuang
· 8 years ago
d053fe4
Opus implementation of the AudioDecoderFactoryTemplate API
by kwiberg
· 8 years ago
fe1aa82
Opus implementation of the AudioEncoderFactoryTemplate API
by kwiberg
· 8 years ago
b8727ae
G722 implementation of the AudioEncoderFactoryTemplate API
by kwiberg
· 8 years ago
b1ed7f0
G722 implementation of the AudioDecoderFactoryTemplate API
by kwiberg
· 8 years ago
0eacd88
Templated AudioDecoderFactory
by kwiberg
· 8 years ago
19b3a55
Fixing incorrect use of erase/remove idiom.
by deadbeef
· 8 years ago
dab1d2d
Enable SNI in ssl adapter.
by Emad Omara
· 8 years ago
43b39de
Roll chromium_revision 175bf817db..b032878ebd (480112:480186)
by buildbot
· 8 years ago
5369a98
Roll chromium_revision 1d3617187c..175bf817db (480056:480112)
by buildbot
· 8 years ago
6531583
Templated AudioEncoderFactory
by kwiberg
· 8 years ago
9fbbdc2
Create the VideoEncoderFactory and implement it.
by Bjorn Mellem
· 8 years ago
c1db79b
Roll chromium_revision 804cd4b03e..1d3617187c (480025:480056)
by buildbot
· 8 years ago
5cb1982
Tune loss-based BWE to be more compatible with the low frequency loss reports of audio streams.
by stefan
· 8 years ago
8fa21c4
Style fixes in rtcp_packet/
by eladalon
· 8 years ago
6b826ef
Add cropping to VIEEncoder to match simulcast streams resolution
by ilnik
· 8 years ago
f79dbad
Add has_value() and value() methods to rtc::Optional.
by terelius
· 8 years ago
0ef8fb9
Roll chromium_revision ff467ab402..804cd4b03e (480004:480025)
by buildbot
· 8 years ago
af35f83
Reduces sensitivity in audio-glitch detector for iOS
by henrika
· 8 years ago
bf5a2fc
Use RaceChecker instead of ThreadChecker in a few places.
by erikvarga
· 8 years ago
bd09ebc
Remove unused #include "libyuv/compare.h"
by eladalon
· 8 years ago
e150058
Move setting switches in AppRTCMobile to Settings screen
by Anders Carlsson
· 8 years ago
7c30390
Roll chromium_revision 2696d5a95c..ff467ab402 (479974:480004)
by buildbot
· 8 years ago
bc061b4
Create AndroidVideoBuffer and allow renderers to consume it.
by Sami Kalliomäki
· 8 years ago
af99b6d
Delete SignalSrtpError.
by nisse
· 8 years ago
92b607b
Roll chromium_revision 0292e51a15..2696d5a95c (479940:479974)
by buildbot
· 8 years ago
2b6e658
Roll chromium_revision 3a8e71c423..0292e51a15 (479885:479940)
by buildbot
· 8 years ago
7ca9e36
Roll chromium_revision 14085ad54d..3a8e71c423 (479854:479885)
by buildbot
· 8 years ago
3fc2350
Support H.264 high profile encoding on Exynos devices.
by glaznev
· 8 years ago
24d00dd
Roll chromium_revision 3a1b0a576e..14085ad54d (479771:479854)
by buildbot
· 8 years ago
38ede13
Support building WebRTC without audio and video.
by zhihuang
· 8 years ago
beae92c
Roll chromium_revision 408bddbca7..3a1b0a576e (479728:479771)
by buildbot
· 8 years ago
65f3150
Roll chromium_revision 5cda18fb97..408bddbca7 (479679:479728)
by buildbot
· 8 years ago
112adf9
Validate references of frames inserted into FrameBuffer2.
by philipel
· 8 years ago
eb02c03
Allow WebRtcMediaEngine to be created from any thread.
by deadbeef
· 8 years ago
d03a578
Roll chromium_revision 47f5ca5642..5cda18fb97 (479659:479679)
by buildbot
· 8 years ago
67561a6
Use the same QP max for tests as in production
by sprang
· 8 years ago
fda496a
Set overuse detector max frame interval based on target frame rate.
by sprang
· 8 years ago
19e087f
This CL finalizes the Conversational Speech tool.
by alessiob
· 8 years ago
b2152b7
Roll chromium_revision 5423b19357..47f5ca5642 (479628:479659)
by buildbot
· 8 years ago
6af9399
ACM: Make AcmReceiver's ownership of NetEq more obvious
by Henrik Lundin
· 8 years ago
f9784f2
Reland of Conversational speech tool, simualtor + unit tests (patchset #1 id:1 of https://codereview.webrtc.org/2925123003/ )
by alessiob
· 8 years ago
f4dd191
Change existing aec dump tests to use webrtc::AecDump.
by aleloi
· 8 years ago
af66f2c
Roll chromium_revision c4920d627d..5423b19357 (479593:479628)
by buildbot
· 8 years ago
08cfc84
Roll chromium_revision e7ba6263e5..c4920d627d (479563:479593)
by buildbot
· 8 years ago
Next »