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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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ee84d39fce1ec4f000b86138a84e56293a9c126a
ee84d39
AEC3: Downmix multichannel signals before delay estimation
by Gustaf Ullberg
· 5 years ago
d181ee7
Adds peer scenario connection interface.
by Sebastian Jansson
· 5 years ago
0cd61b6
MultiCodecReceiveTest: fix for flaky test.
by Åsa Persson
· 5 years ago
b3f1487
Add ability to provide TEXT hint only when requested in PC framework
by Artem Titov
· 5 years ago
9509d95
Add empty build target modules/video_coding:video_coding_legacy
by Niels Möller
· 5 years ago
c77df78
Revert "Improve performance of RtpPacketHistory"
by Qingsi Wang
· 5 years ago
487f9a1
Reland "Refactor SCTP data channels to use DataChannelTransportInterface."
by Bjorn A Mellem
· 5 years ago
116ffe7
Switch to compiling WebRTC -std=c++14 by default
by Danil Chapovalov
· 5 years ago
a0e6ded
Roll chromium_revision 75cf3925c2..e25e764221 (694706:694813)
by chromium-webrtc-autoroll
· 5 years ago
9e380fd
Improve performance of RtpPacketHistory
by Erik Språng
· 5 years ago
a5d952f
Reland "Refactor FEC code to use COW buffers"
by Ilya Nikolaevskiy
· 5 years ago
4d7dac6
Remove usage of RtpRtcp::SetSSRC() in RtpRtcpImplTest
by Erik Språng
· 5 years ago
0987273
Add option to enable retransmission for all temporal layers in the constructor for rtp_sender_video.
by Andrei Dumitru
· 5 years ago
cc62b16
Add qualityLimitationResolutionChanges stat
by Evan Shrubsole
· 5 years ago
a8336d3
Connect the stable target rate to the video encoders
by Florent Castelli
· 5 years ago
ddef8d1
Add support of displaying video during the PC level test
by Artem Titov
· 5 years ago
2aaf66e
Roll chromium_revision 82910f739a..75cf3925c2 (694601:694706)
by chromium-webrtc-autoroll
· 5 years ago
f7cb16f
Check input parameter in RemoteEstimatorProxy::IncomingPacket without lock
by Per Kjellander
· 5 years ago
ee3d995
New class VideoReceiver2, a trimmed-down vcm::VideoReceiver
by Niels Möller
· 5 years ago
4d6b269
Adds setAudio[Track/Record]StateCallback interfaces to the Java ADM
by henrika
· 5 years ago
81a08a7
Roll chromium_revision b67ee864bb..82910f739a (694439:694601)
by chromium-webrtc-autoroll
· 5 years ago
be2e5f7
Make payload type demux conditional on media direction
by Steve Anton
· 5 years ago
3146ea0
Roll chromium_revision 50f07bc317..b67ee864bb (694314:694439)
by chromium-webrtc-autoroll
· 5 years ago
20232a9
Use obfuscated IPs in logging in p2p/ and pc/.
by Qingsi Wang
· 5 years ago
25a4b06
Roll chromium_revision df72a9a5c5..50f07bc317 (694187:694314)
by chromium-webrtc-autoroll
· 5 years ago
b64d65e
Fix NetworkEmulationManagerTest.ThroughputStats flakiness.
by Yves Gerey
· 5 years ago
e611f19
Remove completed TODOs
by Sam Zackrisson
· 5 years ago
30323e2
VP9 screenshare: fix incorrect assumptions on buffer contents
by Ilya Nikolaevskiy
· 5 years ago
e15c10a
Fix for rare read of uninitialized value in remote estimate test.
by Sebastian Jansson
· 5 years ago
2b9dba3
Implement stable rate support in SimulcastRateAllocator
by Erik Språng
· 5 years ago
059a0b7
Fix for deadlock in AudioUsesAbsSendTimeExtension test.
by Sebastian Jansson
· 5 years ago
a3baf2a
Add one more BasicPortAllocator constructor
by Niels Möller
· 5 years ago
cc7ea18
Roll chromium_revision 3cccc3ec06..df72a9a5c5 (694083:694187)
by chromium-webrtc-autoroll
· 5 years ago
19f9c2a
Refactor video analyzer injection helper
by Artem Titov
· 5 years ago
0bd2eff
Reland "New build target p2p:stun_types"
by Niels Möller
· 5 years ago
662e31f
Prepare to move packet_socket_factory to api/.
by Patrik Höglund
· 5 years ago
50b0baf
[tsan] Guard TestStatictics against concurrent access.
by Yves Gerey
· 5 years ago
75e2290
Rollback to strict audio codec tests for libopus on android (neon).
by Yves Gerey
· 5 years ago
ba5f8e9
Revert "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5."
by Guido Urdaneta
· 5 years ago
c2accf2
[tsan] Guard OwnerThread against concurrent access.
by Yves Gerey
· 5 years ago
eea6ced
Roll chromium_revision 675968a8c6..3cccc3ec06 (693954:694083)
by chromium-webrtc-autoroll
· 5 years ago
91c824f
Revert "New build target p2p:stun_types"
by Hannes Landeholm
· 5 years ago
f6ec68d
Roll chromium_revision 7e5c36432b..675968a8c6 (693630:693954)
by chromium-webrtc-autoroll
· 5 years ago
e715301
Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5.
by Trevor Hayes
· 5 years ago
bbbae42
Refactor video analyzer injection helper
by Artem Titov
· 5 years ago
d4e6904
AEC3: Reducing the complexity and heap usage of the adaptive filter
by Per Åhgren
· 5 years ago
f294d26
Delete deprecated method StreamStatistician::GetStatistics
by Niels Möller
· 5 years ago
cfb9497
Add multi-channel to FftBuffer
by Sam Zackrisson
· 5 years ago
5b728cc
Revert "Make relative arrival delay mode default in NetEq delay manager."
by Alessio Bazzica
· 5 years ago
8dcbdd2
Roll chromium_revision e96090c328..7e5c36432b (693514:693630)
by chromium-webrtc-autoroll
· 5 years ago
f3f6159
Rename VectorBuffer->SpectrumBuffer, MatrixBuffer->BlockBuffer, BlockBuffer->Aec2BlockBuffer
by Sam Zackrisson
· 5 years ago
77c71d1
Make relative arrival delay mode default in NetEq delay manager.
by Jakob Ivarsson
· 5 years ago
a81c09d
Make VectorBuffer in AEC3 multi-channel
by Sam Zackrisson
· 5 years ago
9305d11
Delete deprecated rtc_event_log_factory_interface.h
by Danil Chapovalov
· 5 years ago
24b945d
Add support of AudioRecord.Builder in the ADM for Android
by henrika
· 5 years ago
065dd27
Roll chromium_revision c27b8dde0e..e96090c328 (693394:693514)
by chromium-webrtc-autoroll
· 5 years ago
e333505
Roll chromium_revision b931c7fd8b..c27b8dde0e (693252:693394)
by chromium-webrtc-autoroll
· 5 years ago
bbca6dd
Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH.
by Mirko Bonadei
· 5 years ago
c51b4e3
Roll chromium_revision f661d57809..b931c7fd8b (693124:693252)
by chromium-webrtc-autoroll
· 5 years ago
09dcafd
Revert "Always create output_dir in setup_apprtc.py."
by Mirko Bonadei
· 5 years ago
cf9cbf5
Add support for stable bitrate target in SvcRateAllocator
by Erik Språng
· 5 years ago
1067d31
Make the stable target rate always less or equal than the target rate
by Florent Castelli
· 5 years ago
52a8da3
Always create output_dir in setup_apprtc.py.
by Mirko Bonadei
· 5 years ago
d9f98cd
Roll chromium_revision 248662b1b8..f661d57809 (693000:693124)
by chromium-webrtc-autoroll
· 5 years ago
77d197f
Make video_capture_internal_impl publicly visible.
by Mirko Bonadei
· 5 years ago
e74156f
Removes string support in field trial parser.
by Sebastian Jansson
· 5 years ago
3247244
Delete unused method AudioCodingModule::GetDecodingCallStatistics
by Niels Möller
· 5 years ago
a33dc01
AEC3: Propagate the number of channels to the adaptive filters
by Per Åhgren
· 5 years ago
1a3859c
Simplify book-keeping of lost packets
by Niels Möller
· 5 years ago
340e0c5
Delete old version of PeerConnection::SetConfiguration
by Niels Möller
· 5 years ago
59e1464
Fix 28 ClangTidy - Readability findings in modules/rtp_rtcp/
by Danil Chapovalov
· 5 years ago
38350b1
Roll chromium_revision d74690feb1..248662b1b8 (692875:693000)
by chromium-webrtc-autoroll
· 5 years ago
e007ad1
Roll chromium_revision 5e84fd2515..d74690feb1 (692730:692875)
by chromium-webrtc-autoroll
· 5 years ago
da10032
Roll chromium_revision da46a51bc2..5e84fd2515 (692597:692730)
by chromium-webrtc-autoroll
· 5 years ago
7cdcda9
Use the sanitized pair when surfacing the candidate pair change event.
by Qingsi Wang
· 5 years ago
66d6c3b
Buffers non atomic message send with usrsctp lib.
by Seth Hampson
· 5 years ago
8c5520c
Reland "Make the min video bitrate in VideoSendStream configurable."
by Ying Wang
· 5 years ago
1d2149c
Revert "Make the min video bitrate in VideoSendStream configurable."
by Alessio Bazzica
· 5 years ago
23003a2
Add saza to audio watchlists
by Sam Zackrisson
· 5 years ago
b2fb0b9
Make the min video bitrate in VideoSendStream configurable.
by Ying Wang
· 5 years ago
1aa7e2f
Roll chromium_revision 8304ddd943..da46a51bc2 (692489:692597)
by chromium-webrtc-autoroll
· 5 years ago
6516f76
Deprecate SingleThreadedTaskQueueForTesting class.
by Yves Gerey
· 5 years ago
f2773b5
Add webrtc_apprtc_browsertest.cc resource dir to .gitignore.
by Mirko Bonadei
· 5 years ago
082696e
Revert "Refactor FEC code to use COW buffers"
by Ilya Nikolaevskiy
· 5 years ago
ce202a0
Reland "Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3.""
by Per Åhgren
· 5 years ago
a77a1f9
Roll chromium_revision 78591f12ff..8304ddd943 (692389:692489)
by chromium-webrtc-autoroll
· 5 years ago
5056af0
Make sure link allocation is at least as large as bitrate sum.
by Erik Språng
· 5 years ago
a837030
Split out RtpSource from libjingle_peerconnection_api
by Niels Möller
· 5 years ago
d112c75
Revert "Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3.""
by Per Åhgren
· 5 years ago
65024d9
Remove clock drift metric from NetEq.
by Jakob Ivarsson
· 5 years ago
5b4fcb5
New build target p2p:stun_types
by Niels Möller
· 5 years ago
eec5fff
Refactor FEC code to use COW buffers
by Ilya Nikolaevskiy
· 5 years ago
a66395e
Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3."
by Per Åhgren
· 5 years ago
8b7c5e4
Add empty build target p2p:stun_types
by Niels Möller
· 5 years ago
54c0326
Roll chromium_revision a42eacf137..78591f12ff (692288:692389)
by chromium-webrtc-autoroll
· 5 years ago
602942f
Filter out small packets from delay-based overuse detection.
by Bjorn Terelius
· 5 years ago
f660e81
Revert "Simplify pacer queue"
by Erik Språng
· 5 years ago
ce6a0c8
Roll chromium_revision 54ad211b04..a42eacf137 (692182:692288)
by chromium-webrtc-autoroll
· 5 years ago
ed2fc50
Roll chromium_revision 291798b89f..54ad211b04 (692040:692182)
by chromium-webrtc-autoroll
· 5 years ago
af3afff
Roll chromium_revision ea980c903b..291798b89f (691937:692040)
by chromium-webrtc-autoroll
· 5 years ago
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