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platform
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external
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webrtc
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dfd5c4b15ea25499fb70dc9fcf24198fe4f69934
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AUTHORS
8905d04
Add ',' between elements in RTCStatsReport::ToJson
by Dirk-Jan C. Binnema
· 6 years ago
74cdf78
add cstring include need for strncmp
by Michel Promonet
· 6 years ago
83aa5ac
Adding Microsoft Corporation (*@microsoft.com) to WebRTC AUTHORS
by James Cadd
· 6 years ago
318da51
Reland "Add support for screen sharing with PipeWire on Wayland"
by Tomas Popela
· 6 years ago
3ddaf3c
Revert "Add support for screen sharing with PipeWire on Wayland"
by Patrik Höglund
· 6 years ago
bf0d0c1
Add IPv6 configuration parameters to iOS API
by Uladzislau Susha
· 6 years ago
dd20c9c
Add support for screen sharing with PipeWire on Wayland
by Tomas Popela
· 6 years ago
0eb7d3ff
Always call ConvertToI420 with positive crop_height
by Robert Bares
· 6 years ago
6fcf6ca
Modified PressEnterToContinue() to actualy check if Enter is pressed
by Danail Kirov
· 6 years ago
d2fb1bf
Generate module.modulemap file when building Mac Framework
by Joel Sutherland
· 6 years ago
289e980
Remove unused var in device info bits from video capture module for Linux
by Jose Antonio Olivera Ortega
· 6 years ago
e0c8b23
Frame marking RTP header extension (PART 1: implement extension)
by Johnny Lee
· 6 years ago
8cec4fb
Use default RTCConfiguration on iOS
by Yuriy Pavlyshak
· 6 years ago
ccee56b
Add certificate generate/set functionality to bring iOS closer to JS API
by Michael Iedema
· 6 years ago
25cc8ad
Fixed issue with BGRA RTCCVPixelBuffer scale and crop
by David Porter
· 6 years ago
e250645
Call callback in IDLE state
by Ing. Jan Kaláb
· 6 years ago
43800f9
Generalize SimulcastEncoderAdapter, use for H264 & VP8.
by Sergio Garcia Murillo
· 6 years ago
6f440ed
Revert "Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8."
by Mirko Bonadei
· 6 years ago
07efe43
Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.
by Sergio Garcia Murillo
· 6 years ago
b90e63c6
Fix: NetEq PacketBuffer logs discarded packet with wrong codec level when new packet replaces the lower level packet
by Peng Yu
· 6 years ago
e7e0602
ObjC: Notify local video track
by Piasy Xu
· 6 years ago
f8d8d6d
Use range-based-for instead of std::for_each and std::mem_fun
by Yusuke Suzuki
· 6 years ago
ebd9abc
Use IFA_LOCAL instead of IFA_ADDRESS over IPv4 network on ANDROID
by Yongje Lee
· 7 years ago
a72b7fc
ObjC: Add missing _lastDrawnFrame assignments
by Maxim Pavlov
· 7 years ago
3cfe9e1
Fixed video capturing on Mac.
by Maksim Khobat
· 7 years ago
2870b0a
Expose a link-local network interfaces enumeration option
by Daniel Lazarenko
· 7 years ago
5e4833c
Add missing stdio.h header in files using scanf/sscanf function.
by Piotr Tworek
· 7 years ago
b887435
RemoteBitrateEstimatorAbsSendTime: check clock is a valid ref
by Miguel Paris
· 7 years ago
d3c642b
Fix typo in the include path of ooura_fft.h
by Jiawei Ou
· 7 years ago
6728003
Skip H246 scaling lists in SPS packets
by Todd Wong
· 7 years ago
5a7508a
Fixed NPE inside org.webrtc.Camera1Session.create
by Yura Yaroshevich
· 7 years ago
1c62ffa
Normalize main(..) routines for WinUWP
by Robin Raymond
· 7 years ago
12e555b
Delete wrapper API ConvertToI420 for YUV conversion to I420
by mallikarjun82
· 7 years ago
149533a
Move rendering code in SurfaceViewRenderer to a separate class.
by Xiaolei Yu
· 7 years ago
e21be1d
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
by philipel
· 7 years ago
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
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