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gerrit-public.fairphone.software
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platform
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external
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webrtc
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5ba58c6735fd2c8ce285027993bc1e32b74b61cc
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talk
1387149
Adding reduced size RTCP configuration down to the video stream level.
by deadbeef
· 9 years ago
434aca8
Add empty placeholder files for remote audio tracks.
by tommi
· 9 years ago
7623ce4
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
by Peter Boström
· 9 years ago
bda7e0b
Fixing issue with default stream upon setting 2nd remote description.
by deadbeef
· 9 years ago
d02b0fa
Add oldest rotation type option to RTCFileLogger
by haysc
· 9 years ago
1a9d615
Add tracing to public PeerConnection methods.
by Peter Boström
· 9 years ago
7b2f762
Don't call SetPreviewFormat if capturing to textures.
by perkj
· 9 years ago
edd8fef
Add new view that renders local video using AVCaptureLayerPreview.
by haysc
· 9 years ago
246b817
Refactor handling of AudioOptions.
by solenberg
· 9 years ago
8237abf
Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
by kjellander
· 9 years ago
9f45a45
Add tracing to upper-level WebRTC calls.
by Peter Boström
· 9 years ago
03ef053
Merge webrtc/video_engine/ into webrtc/video/
by Peter Boström
· 9 years ago
3868692
Free SCTP data channels asynchronously in PeerConnection.
by deadbeef
· 9 years ago
46ad542
Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ )
by pbos
· 9 years ago
6f28cf0
Implement standalone event tracing in AppRTCDemo.
by Peter Boström
· 9 years ago
84f0970
Reland of "Create rtc::AtomicInt POD struct."
by Peter Boström
· 9 years ago
cd4003f
Use @webrtc.org addresses for OWNERS.
by Peter Boström
· 9 years ago
cf890bc
Roll gtest-parallel.
by Peter Boström
· 9 years ago
9d69c3f
Return a copy of the supported RTP header extensions instead of a reference.
by Stefan Holmer
· 9 years ago
b86d4e4
Prepare the AudioSendStream to be hooked up to send-side BWE.
by Stefan Holmer
· 9 years ago
03f80eb
Refactor EglBase configuration.
by nisse
· 9 years ago
1218d7a
Allow remote fingerprint update during a call
by Guo-wei Shieh
· 9 years ago
86aaa4b
Revert "Allow remote fingerprint update during a call"
by Guo-wei Shieh
· 9 years ago
9c38c2d
Allow remote fingerprint update during a call
by Guo-wei Shieh
· 9 years ago
381b421
Ping backup connection at a slower rate
by Honghai Zhang
· 9 years ago
9e1b992
Clear old decoders after recreating the receiver.
by Peter Boström
· 9 years ago
b572768
- Remove calls to VoEDtmf from WVoE/MC.
by Fredrik Solenberg
· 9 years ago
1a5cf6e
Remove the unused NullMediaEngine (and NullVoiceEngine+NullVideoEngine).
by Fredrik Solenberg
· 9 years ago
9cf0c3d
Removes MAYBE_ from several test case names in JsepPeerConnectionP2PTestClient.
by Ivo Creusen
· 9 years ago
7635684
Fix Mac ObjC PeerConnection API compilation.
by tkchin
· 9 years ago
9462052
In some rare Android systems ConnectivityManager may be null.
by honghaiz
· 9 years ago
3c28d0d
Disable PeerConnectionEndToEndTest.Call on Mac.
by kjellander@webrtc.org
· 9 years ago
1d63dd0
- Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused.
by solenberg
· 9 years ago
ee524f7
Adding Java binding for CreateSender.
by deadbeef
· 9 years ago
7e4e01a
Add header extension filtering for WebRtcVoiceEngine/MediaChannel.
by solenberg
· 9 years ago
2515af2
Removing some unnecessary string manipulation code from VoEBase::GetVersion().
by solenberg
· 9 years ago
d20d247
Fix VideoCaptureAndroid, drop frame when switching camera using textures.
by perkj
· 9 years ago
226a602
Fix problem when drawing to the Android Media encoder surface.
by perkj
· 9 years ago
40455d6
This cl change so that we use EGL14 where it is supported and EGL10 otherwise. The idea is to make this agnostic to an application and for WebRTC except in EGLBase.
by perkj
· 9 years ago
41b0798
Adding CreatePeerConnection method that uses new PC Initialize method.
by deadbeef
· 9 years ago
0de97f1
WebRtcVideoCapturer: SetCaptureState(CS_STOPPED) on Stop and ensure state changes in unittest.
by hbos
· 9 years ago
cb9792e
Fix VideoCapturerAndroidTest.testStartWhileCameraIsAlreadyOpen on Android M.
by perkj
· 9 years ago
14f4144
Add helper KeepRefUntilDone.
by perkj
· 9 years ago
ee69ed5
Add separate event for camera freeze.
by glaznev
· 9 years ago
70c0e29
Disable PeerConnectionEndToEndTest.Call for TSan.
by kjellander@webrtc.org
· 9 years ago
ae54b83
Android SurfaceViewRenderer: Add resetStatistics() method
by magjed
· 9 years ago
2fe1cb0
Don't overwrite audio stats when they're not available.
by andrew
· 9 years ago
26c8c91
Using Rent-A-Codec for static Codec access in WVoE/MC.
by solenberg
· 9 years ago
727dbc2
VideoCapturerAndroid - allow lower frame rate in bad lightning
by Per
· 9 years ago
598242a
Support texture scaling in Androids MediaEncoder.
by Per
· 9 years ago
a3c20bb
Add support for scaling textures in AndroidVideoCapturer.
by Per
· 9 years ago
fac0655
Reland of Adding the ability to create an RtpSender without a track.
by deadbeef
· 9 years ago
444682a
Remove frame time scheduing in IncomingVideoStream
by qiangchen
· 9 years ago
b251472
Add JNI interface for functions to start and stop recording AEC dumps and RTC event logs.
by ivoc
· 9 years ago
4c5eea3
Android SurfaceViewRenderer: Don't rely on widthSpec/heightSpec after onMeasure() returns
by Magnus Jedvert
· 9 years ago
7baf79f
Temporary remove spamming MediaDecoder log
by perkj
· 9 years ago
4f2152e
Android SurfaceViewRenderer: Make sure not to call eglCreateSurface() twice
by Magnus Jedvert
· 9 years ago
9237559
Add SurfaceTextureHelper.disconnect(Handler handler) method
by perkj
· 9 years ago
b5cb19b
Fixing direction attribute in answer for non-RTP protocols.
by deadbeef
· 9 years ago
05816eb
Fix target_arch for ios devices
by wr.wllm
· 9 years ago
1aa6efe
Android ThreadUtils: Make the class public for access outside org.webrtc
by Magnus Jedvert
· 9 years ago
8becec3
talk: remove deprecated *processor.h files
by tfarina
· 9 years ago
87d5845
Fix androidmediadecoder_jni TS logging.
by perkj
· 9 years ago
43edf0f
Require negotiation to send transport cc feedback over RTCP.
by stefan
· 9 years ago
bd13838
Remove SetVideoLogging/SetAudioLogging from ChannelManager and down the stack.
by solenberg
· 9 years ago
5def7b9
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
by deadbeef
· 9 years ago
7add058
Move some receive stream configuration into webrtc::AudioReceiveStream.
by solenberg
· 9 years ago
6834fa1
Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ )
by deadbeef
· 9 years ago
30e9182
This cl add support to encode from textures to MediaCodecVideoEncoder.
by perkj
· 9 years ago
7e63ef0
Allow default audio receive channel to receive on any unsignalled SSRC.
by solenberg
· 9 years ago
17c0aff
Enable VP9 HW decoder on Exynos chips.
by Alex Glaznev
· 9 years ago
7755e20
Chrome has now been updated.
by perkj
· 9 years ago
726b1f7
Removed dummy "mediastreamsignaling.h"
by perkj
· 9 years ago
191c1f9
Disable all JsepPeerConnectionP2PTestClient tests on Mac due to flakiness on Mac Debug bots.
by ivoc
· 9 years ago
ef45323
Android: Make classes non-final
by Magnus Jedvert
· 9 years ago
1503867
Disabled several JsepPeerConnectionP2PTestClient tests on Mac, due to flakiness on Debug Mac trybots.
by ivoc
· 9 years ago
b6755ab
Revert of Adding thread timeout for audio recorer thread in Java (patchset #2 id:20001 of https://codereview.webrtc.org/1444313002/ )
by henrika
· 9 years ago
488e75f
Patchset 1 yet again relands without modification https://codereview.webrtc.org/1422963003/
by Per
· 9 years ago
521ed7b
Reland Convert internal representation of Srtp cryptos from string to int
by Guo-wei Shieh
· 9 years ago
318166b
Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ )
by guoweis
· 9 years ago
2764e10
Convert internal representation of Srtp cryptos from string to int.
by guoweis
· 9 years ago
b7ce964
modules/video_coding/utility: Remove include
by kjellander@webrtc.org
· 9 years ago
ad948c4
Preliminary support of VP9 HW encoder on Android.
by Alex Glaznev
· 9 years ago
2557b86
modules/video_coding refactorings
by Henrik Kjellander
· 9 years ago
4dd7a65
Temporarily disable VERIFY while bug is investigated.
by phoglund
· 9 years ago
2aff615
Remove spammy logging of RTCP delivery failures.
by Peter Boström
· 9 years ago
fd614c2
Adding thread timeout for audio recorer thread in Java
by henrika
· 9 years ago
6f8ce06
common_video: rename interface -> include
by kjellander
· 9 years ago
482b12e
Remove BundleFilter filtering of RTCP.
by pbos
· 9 years ago
3a94154
Move some send stream configuration into webrtc::AudioSendStream.
by solenberg
· 9 years ago
633a3aa
ThreadUtils: Add joinUninterruptibly() with timeout
by magjed
· 9 years ago
3e0f602
Android EglBase: Add support for creating EGLSurface from Surface, not SurfaceHolder
by magjed
· 9 years ago
4a41361
Android SurfaceViewRenderer: Never hold a pending frame indefinitely
by magjed
· 9 years ago
c01c254
Revert of Android MediaCodecVideoDecoder: Manage lifetime of texture frames (patchset #12 id:320001 of https://codereview.webrtc.org/1422963003/ )
by Per
· 9 years ago
cbe9f51
Revert of Remove global list of SRTP sessions. (patchset #4 id:60001 of https://codereview.webrtc.org/1416093010/ )
by phoglund
· 9 years ago
faac497
Fix for scenario where m-line is revived after being set to port 0.
by deadbeef
· 9 years ago
68876f9
Introduces Android API level linting, fixes all current API lint errors.
by Patrik Höglund
· 9 years ago
9576e54
Reland "Prepare MediaCodecVideoEncoder for surface textures.""
by perkj
· 9 years ago
8093d54
Change default SSRC for RTCP receiver reports to not collide with video.
by solenberg
· 9 years ago
5dda80a
Remove webrtc/modules/video_{capture,render}/include
by Henrik Kjellander
· 9 years ago
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