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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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4fed3c0b6f74acbdce2f7c7395463772a19dc739
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AUTHORS
bdbc889
Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
by philipel
· 7 years ago
f1e08d0
Fix the video buffer size should take rtt into consideration
by gustavogb
· 7 years ago
f3a48ab
Delete unused field from AndroidVideoTrackSource
by korniltsev.anatoly
· 7 years ago
ff7acb1
Reset isFirstFrameRendered on init of SurfaceViewRenderer
by tserng
· 7 years ago
c43f68e
Fix do not unregister bluetooth receiver if it was not registered
by Gustavo Garcia
· 7 years ago
1b2469b
Fix AVFoundation framework import
by hansknoechel92
· 7 years ago
8e857d1
Adding capture device selection capability for video_loopback. It will help to select any capture device to test the utility. In future we can add screen share as capture device.
by Tarun Chawla
· 7 years ago
ace5c88
This CL adds RTCMTLVideoView.h and RTCCameraVideoCapturer.h to WebRTC.h
by hewwatt
· 8 years ago
a1fa491
Fix invalid output buffer usage
by steweg
· 8 years ago
0d335c7
Fixed that RTCCameraPreviewView did not rotate the video on device rotation.
by meetAkshay99
· 8 years ago
9d65f39
Added support for changing the volume of AudioTrack as discussed in BUG=webrtc:6533
by dax
· 8 years ago
0642b32
Remove duplicate entries from AUTHORS file
by henrik.lundin
· 8 years ago
9f2c18e
Changed OLA window for neteq. Old code didnt work well with 48khz
by soren
· 8 years ago
4b37127
Fix compilation issues of std::unique_ptr
by steweg
· 8 years ago
28dc285
Adding cbr support for Opus
by soren
· 8 years ago
0248e7c
Re-add author accidentally removed in https://codereview.webrtc.org/2534843002.
by solenberg
· 8 years ago
846e1be
Fix iOS8 crash in background mode.
by sdkdimon
· 8 years ago
228c268
Support 4 channel mic in Windows Core Audio
by jens.nielsen
· 8 years ago
0d1305e
Added support for changing the volume of RTCAudioSource as discussed in BUG=webrtc:6533
by frederik.riedel
· 8 years ago
8a855d6
Allow any unsignalled SSRC changes on default video receive channel.
by mzanaty
· 8 years ago
b11fb25
Protect APM in webkit builds.
by agouaillard
· 8 years ago
888874f
Allow GCC 4.9 to compile Chromium
by floppymaster
· 8 years ago
e5dc62a
PRESUBMIT: Add authorized-authors check + AUTHORS e-mails.
by kjellander
· 8 years ago
ba7e71b
remove googViewLimitedResolution stat
by philipp.hancke
· 8 years ago
bbfed52
Set OPENSSL_EC_NAMED_CURVE explicitly on EC key so that certificate has ASN1 OID and NIST curve info. Without this openSSL handshake negotiation fails throwing NO_SHARED_CIPHER error. the change made is along the lines of openssl behavior documented here: https://wiki.openssl.org/index.php/Elliptic_Curve_Diffie_Hellman#ECDH_and_Named_Curves
by ssaroha
· 8 years ago
610c454
Add Datachannel support to Android AppRTCMobile
by hekra01
· 8 years ago
bcc5d87
Add a GN target for unit tests, get them working again and added a test.
by adam.fedor
· 8 years ago
a264ecc
Copy RTCAudioSource.h and RTCMediaSource.h with other public header files when building the WebRTC framework for iOS / Mac
by VladimirTechMan
· 8 years ago
86ccd7b
Revert of Add field_trial_default dependency to libjingle_peerconnection (patchset #3 id:40001 of https://codereview.webrtc.org/2120673004/ )
by sakal
· 8 years ago
a7a01df
Add field_trial_default dependency to libjingle_peerconnection
by arlolra
· 8 years ago
96b6b83
iOS: add type to peer connection local streams
by vopatop.skam
· 8 years ago
3f70562
Fix WebRtc ninja x86 build using Visual Studio 2015 (set GYP_MSVS_VERSION=2015).
by conceptgenesis
· 9 years ago
bedc17b
Fixing integer underflow in FileAudioDevice (webrtc issue 4554)
by A.Brauckmann
· 9 years ago
978244e
Adding a bunch of Agora IO team members to the watch lists
by yujie.mao
· 9 years ago
f70568c
So long and thanks for all the code reviews!
by andrew
· 9 years ago
bb79127
Add Riku Voipio to AUTHORS.
by Andrew MacDonald
· 9 years ago
88799d9
RTCEAGLVideoView: Fix GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT error.
by christoffer
· 9 years ago
92068ee
Android: Guard against switching camera on stopped camera
by colin
· 9 years ago
4de6622
Fix a bug in computing audio delay on ios device. Converts seconds to
by Jiawei Ou
· 9 years ago
fcfdb08
Update AUTHORS file.
by tkchin
· 9 years ago
4988ca5
Removed unused variables and the need to include the d3dx9.h file.
by dkirovbroadsoft
· 9 years ago
3ee4fe5
Re-land: Add API to get negotiated SSL ciphers
by pthatcher@webrtc.org
· 10 years ago
2bf0e90
Revert 8275 "This CL adds an API to the SSL stream adapters and ..."
by tommi@webrtc.org
· 10 years ago
1d11c82
This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer.
by pthatcher@webrtc.org
· 10 years ago
db1ebf6
Add jakehilton@gmail.com to AUTHORS
by tnakamura@webrtc.org
· 10 years ago
0ba1533
Added support for an Origin header in STUN messages.
by pthatcher@webrtc.org
· 10 years ago
ee9d61c
This fixes a small memory leak (found using Xcode/Instruments on iOS) in
by tkchin@webrtc.org
· 10 years ago
c569a49
Unit tests for SSLAdapter
by tkchin@webrtc.org
· 10 years ago
31c285b
Update AUTHORS file.
by henrike@webrtc.org
· 10 years ago
ddb85ab
Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07
by jiayl@webrtc.org
· 10 years ago
d798095
replace inline assembly WebRtcNsx_PrepareSpectrumNeon by intrinsics.
by andrew@webrtc.org
· 10 years ago
0402515
Implement command line flags for peerconnection client example on Windows
by kjellander@webrtc.org
· 10 years ago
7c82ada
AppRTCDemo was blocking the main thread for network requests. This fixes it by making the background queue serial instead of using @synchronize to make the background operations serial.
by fischman@webrtc.org
· 11 years ago
ceffdbc
Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord()
by henrike@webrtc.org
· 11 years ago
82387e4
Add ability to receive calls for iOS BUG=2701 R=fischman@webrtc.org
by fischman@webrtc.org
· 11 years ago
a9bdee6
Add Christophe Dumez to AUTHORS.
by andrew@webrtc.org
· 11 years ago
af320fd
The designated initializer method declaration in the Objective-C headers for RTCICEServer does't match its implementation.
by fischman@webrtc.org
· 11 years ago
eb7def2
Fix compilation errors on Fedora 20.
by fischman@webrtc.org
· 11 years ago
7b2f955
Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README.
by henrike@webrtc.org
· 11 years ago
efdf778
BUG=1351
by mflodman@webrtc.org
· 12 years ago
17b867a
compile fix for get_nprocs() with uClibc
by phoglund@webrtc.org
· 12 years ago
5140e24
MIPS optimizations for Signal Processing Library patch01
by andrew@webrtc.org
· 12 years ago
73a702c
This is a change in the iOS audio device to use VoiceProcessingIO API instead of RemoteIO. This way we don't need to use WebRTC EC and NS because it happens on the device hardware.
by andrew@webrtc.org
· 12 years ago
bcb7174
.gitignore: Add *.mk, created as part of ChromiumOS build
by andrew@webrtc.org
· 12 years ago
1b6da28
Bugfix for NACK behavior. Current code sends a number of duplicate NACK requests.
by pwestin@webrtc.org
· 12 years ago
ef62929
Landing http://review.webrtc.org/914006/
by niklas.enbom@webrtc.org
· 12 years ago
01ad758
ilbc: Mark untouched input arrays as const
by turaj@webrtc.org
· 12 years ago
e22beab
[MIPS] Adding support for MIPS architecture for WebRTC.
by andrew@webrtc.org
· 12 years ago
87c50f0
Adding author
by niklas.enbom@webrtc.org
· 13 years ago
3a9680b
Adding author
by niklas.enbom@webrtc.org
· 13 years ago
da159d6
by niklase@google.com
· 13 years ago