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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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9922016ee44c70eb1c14f647cc4a5ade58a0018c
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aed581a
Made AudioReceiveStream a mixer participant.
by aleloi
· 8 years ago
5f70d3b
Fix org.mockito.Matchers deprecation warnings in DirectRTCClientTest.
by sakal
· 8 years ago
201dfe9
Split audio mixer into interface and implementation.
by aleloi
· 8 years ago
76648da
Add FlexfecReceiveStream.
by brandtr
· 8 years ago
057b8d9
Remove all traces of Dr Memory.
by Henrik Kjellander
· 8 years ago
69034df
Make GN build screenshare_loopback
by palmkvist
· 8 years ago
5a87245
iOS: Optimize video scaling and cropping
by magjed
· 8 years ago
7a97344
Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.
by minyue
· 8 years ago
1cb4823
Android YuvConverter: Use OpenGL Framebuffer instead of EGL pixel buffer
by magjed
· 8 years ago
9ab8a18
Android: Extend functionality of EglRenderer
by magjed
· 8 years ago
ca20e7c
Revert of Delete unused file mediacommon.h. (patchset #1 id:1 of https://codereview.webrtc.org/2437703002/ )
by nisse
· 8 years ago
c1f8ecb
Remove check for numberOfCameras from AppRTC Mobile PeerConnectionClient.
by sakal
· 8 years ago
be4aff7
Suppress deprecation warning in CallFragment.
by sakal
· 8 years ago
aff9ff0
Create .git-blame-ignore-revs and add Java format CL to it.
by sakal
· 8 years ago
e33c5d9
Added a level controller initialization value to MediaConstraints.
by aleloi
· 8 years ago
647915f
Add loopback option and improve UX of AppRTCMobile for Mac.
by denicija
· 8 years ago
725e212
Prevent stripping of C interfaces in framework
by kthelgason
· 8 years ago
e037060
Add to rtc::Optional equality/unequality comparisions with object
by danilchap
· 8 years ago
a34e796
Delete unused file mediacommon.h.
by nisse
· 8 years ago
55928fe
QualityScaler reset bugfix
by kthelgason
· 8 years ago
0489e49
Change RefCountedObject to use perfect forwarding.
by perkj
· 8 years ago
79f0bf3
A variable in ScreenCapturerWinDirectx has a bad name
by zijiehe
· 8 years ago
f04f14e
Revert of Move bitstream parser to more appropriate directory. (patchset #4 id:60001 of https://codereview.webrtc.org/2370853005/ )
by kthelgason
· 8 years ago
cc6817e
Move current bitstream parser to more appropriate directory.
by kthelgason
· 8 years ago
577bc19
Android: Move YuvConverter to its own file
by Magnus Jedvert
· 8 years ago
b6f1fb5
Delete RTPSender::BuildRtpHeader function and all dependencies
by danilchap
· 8 years ago
061ea0d
Remove VideoCodec resolution validation.
by Per
· 8 years ago
e3e411a
Removed perkj@ from video WATCHLIST
by Per
· 8 years ago
73c5d4a
Include ScreenCapturerAndroid in libjingle_peerconnection_java.jar
by sakal
· 8 years ago
0934785
Reland of Make cricket::VideoFrame inherit webrtc::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2402853002/ )
by nisse
· 8 years ago
4e52386
Reland of Add path for recovered packets from internal::Call to RtpStreamReceiver. (patchset #1 id:1 of https://codereview.webrtc.org/2427733002/ )
by brandtr
· 8 years ago
249beee
Remove DesktopRegion parameter from DesktopCapturer::Capture
by zijiehe
· 8 years ago
e8295fc
Roll chromium_revision eb9b71b64b..4c4977aa05 (426008:426117)
by buildbot
· 8 years ago
6a4607e
Deflaky ScreenCapturerTest
by zijiehe
· 8 years ago
1eb1293
Handle BW drop in ALR region and initiate probing
by Irfan Sheriff
· 8 years ago
a9c7cfa
Prepare for introduction of rtc::PacketTransportInterface.
by johan
· 8 years ago
1203066
Compilerwarning possible loss of data in file port.h
by bertholdherrmann08
· 8 years ago
cc555c5
RTCDataChannelStats[1] added, supporting all stats members.
by hbos
· 8 years ago
1394c7b
Fix for flaky test: EndToEndTest.VerifyHistogramStatsWithRtx
by asapersson
· 8 years ago
0b7be9c
Roll chromium_revision c8b7ee41e0..eb9b71b64b (425645:426008)
by buildbot
· 8 years ago
9960bb1
Call OnTransportFeedback just when feedback_observer exist.
by michaelt
· 8 years ago
53fe19d
Set min and max rate on caller and on callee side.
by michaelt
· 8 years ago
64e1a32
Second try to get "Support for video file instead of camera and output video out to file" accepted
by mandermo
· 8 years ago
67a8c98
Revert of Support for video file instead of camera and output video out to file (patchset #17 id:320001 of https://codereview.webrtc.org/2273573003/ )
by kjellander
· 8 years ago
f33970b
Add unittest for I420Buffer::Rotate.
by nisse
· 8 years ago
6ed592d
Rename variables to reflect that DelayBasedBwe lives on the send side rather than receive side.
by terelius
· 8 years ago
5588a13
Now uses rtc::Buffer in AudioDeviceBuffer.
by henrika
· 8 years ago
4466699
Support for video file instead of camera and output video out to file
by mandermo
· 8 years ago
9e83c97
Add rtc::Optional::emplace
by danilchap
· 8 years ago
7a37761
Removed RTPHeader from NetEq's Packet struct.
by ossu
· 8 years ago
553024a
During a fix of an unrelated issue, a bug was introduced in the rtp analyzer tool: when the number of data points was divisible by RTPStatitstics.PLOT_RESOLUTION_MS (which is 50), pyplot.plot was called with arrays of different lengths. One of the arrays could be one element larger.
by aleloi
· 8 years ago
e405d9b
Add a fuzzer for FlexfecReceiver.
by brandtr
· 8 years ago
e6b5829
Extends how AppRTCMobile handles audio focus on Android
by henrika
· 8 years ago
91718a1
Roll script: Update after SVN support was dropped from depot tools
by kjellander
· 8 years ago
5c63989
Import build/config/clang/clang.gni in webrtc/base/BUILD.gn
by ehmaldonado
· 8 years ago
862d74d
Revert of Add path for recovered packets from internal::Call to RtpStreamReceiver. (patchset #2 id:60001 of https://codereview.webrtc.org/2390823009/ )
by honghaiz
· 8 years ago
c4fd23c
Add rtc::Optional::reset
by danilchap
· 8 years ago
9b9910d
Roll chromium_revision 3d5a0fb164..c8b7ee41e0 (425604:425645)
by buildbot
· 8 years ago
7e76560
Enable logging to console in DirectRTCClientTest.
by sakal
· 8 years ago
2f255d8
Replace const -> constexpr for rtcp Packet Type
by danilchap
· 8 years ago
c1f40b7
Remove RtcpPacket dependency on rtcp_utility
by danilchap
· 8 years ago
27c3d5b
Restore thread name consistency for webrtc/p2p/ .
by johan
· 8 years ago
883ad66
Removed the deprecated audioproc executable
by peah
· 8 years ago
e40a7ee
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
by kjellander
· 8 years ago
671534e
Roll chromium_revision 90b4cf429c..3d5a0fb164 (425384:425604)
by buildbot
· 8 years ago
9c4b4b4
Add path for recovered packets from internal::Call to RtpStreamReceiver.
by brandtr
· 8 years ago
e5ddf52
Delete unused file webrtcvideochannelfactory.h.
by nisse
· 8 years ago
285e558
Removed suppressions for the data race inside the APM that is now fixed.
by peah
· 8 years ago
2c4c422
Roll chromium_revision a3c4a78675..90b4cf429c (425286:425384)
by buildbot
· 8 years ago
0d4b129
Roll chromium_revision 00384b2217..a3c4a78675 (425234:425286)
by buildbot
· 8 years ago
8f7cc7e
This CL corrects the emptying of the render queues for the
by peah
· 8 years ago
5d2e58c
Roll chromium_revision 61fb879aaf..00384b2217 (425083:425234)
by buildbot
· 8 years ago
91902cb
Remove DesktopRegion parameter from DesktopCapturer::Capture.
by zijiehe
· 8 years ago
794d535
Roll chromium_revision 50c7b3ce18..61fb879aaf (424992:425083)
by buildbot
· 8 years ago
9ae585d
Cleanup of voice_engine includes.
by aleloi
· 8 years ago
3283cf9
Add asyncstuntcpsocket_unittest.cc to rtc_unittests
by kjellander
· 8 years ago
982bf89
Revert of Add RtcpRttStats to AudioStream (patchset #1 id:1 of https://codereview.webrtc.org/2402333002/ )
by sprang
· 8 years ago
b593bc0
Suggest myself as owner of api/
by solenberg
· 8 years ago
81b8a07
Roll chromium_revision 2cabef4e7d..50c7b3ce18 (424936:424992)
by buildbot
· 8 years ago
0d83857
NetEq: Convert AverageIAT from int to float calculations
by henrik.lundin
· 8 years ago
c9ec875
NetEq: Remove special case for Merge without Expand
by henrik.lundin
· 8 years ago
722b0dc
Revert of Android audio playout now supports non-call media streams (patchset #3 id:10004 of https://codereview.webrtc.org/2411263003/ )
by henrika
· 8 years ago
dd7a1cf
Landmine due to corrupt .pdb files on Windows.
by Henrik Kjellander
· 8 years ago
da3303f
Revert of Remove tools dir from root webrtc target (patchset #1 id:1 of https://codereview.webrtc.org/2412353004/ )
by kjellander
· 8 years ago
91a5759
Roll chromium_revision 316b880c55..2cabef4e7d (421519:424936)
by Henrik Kjellander
· 8 years ago
163b1a2
Remove tools dir from root webrtc target
by charujain
· 8 years ago
614f68f
Remove duplicate entry in webrtc .gn file exec_script_whitelist
by fbarchard
· 8 years ago
db158f9
Fix experiment name in BitrateControllerTest.
by Stefan Holmer
· 8 years ago
77c663d
Give FeedbackTimeout experiment the correct name.
by stefan
· 8 years ago
12a39f4
Don't crash on unexpected stap-a or fu-a.
by stefan
· 8 years ago
75c8fb4
DataChannelInterface default impl of [messages/bytes]_[sent/received].
by hbos
· 8 years ago
84ffdee
DataChannel[Interface]::[message/bytes]_[sent/received]() added.
by hbos
· 8 years ago
73fdc31
Several fixes to screen_capturer_mac.
by erikchen
· 8 years ago
e280cde
Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.
by ossu
· 8 years ago
872f614
Android audio playout now supports non-call media streams.
by henrika
· 8 years ago
116ec6d
Implemented further mixer interface change suggestions from https://codereview.webrtc.org/2386383003/
by aleloi
· 8 years ago
7e30432
Hooking up audio network adaptor to VoE.
by minyue
· 8 years ago
917d4e1
Removed the legacy behavior of stopping playout when setting new receive codecs.
by solenberg
· 8 years ago
e97974d
Cleanup of the mixer interface.
by aleloi
· 8 years ago
73a28ee
The AudioProcessing class is used as an interface
by peah
· 8 years ago
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