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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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05d43c6f7fe497fed0f2c8714e2042dd07a86df2
05d43c6
Fix getStats() freeze bug affecting Chromium but not WebRTC standalone.
by Henrik Boström
· 6 years ago
914351d
Reland "Always offer transport sequence number header extension for audio""
by Per Kjellander
· 6 years ago
106d92d
Add thread safety annotation for PeerConnection::SignalDataChannelCreated_
by Karl Wiberg
· 6 years ago
13bc871
PostMessageWithFunctor() added.
by Henrik Boström
· 6 years ago
397c06f
Revert "Always offer transport sequence number header extension for audio"
by Ying Wang
· 6 years ago
7e0e44f
Move video-related MediaTransport interfaces to their own file and target
by Niels Möller
· 6 years ago
054db54
Remove an absl::WrapUnique usage without absl/memory/memory.h include
by tzik
· 6 years ago
22997d6
Roll chromium_revision 2ad52fb2a4..d1f68eb66e (632357:632477)
by chromium-webrtc-autoroll
· 6 years ago
1c9c9fc
Replace replace_substrs with Abseil
by Steve Anton
· 6 years ago
bf9e01a
Add support of fast media sending in peer connection e2e test
by Artem Titov
· 6 years ago
ceba6ae
Return a copy, becase GetPercentile in SamplesStatsCounter isn't const
by Artem Titov
· 6 years ago
cf8405e
Add generic packet rates to event_log_visualizer.
by Piotr (Peter) Slatala
· 6 years ago
15653f9
Roll chromium_revision 78de17c053..2ad52fb2a4 (632252:632357)
by chromium-webrtc-autoroll
· 6 years ago
aa58415
Reland "Enabling Simulcast use via AddTransceiver."
by Amit Hilbuch
· 6 years ago
aec9794
Fix DCHECK when encoding GenericPacket* events using the legacy RTC event log format.
by Piotr (Peter) Slatala
· 6 years ago
9e2692c
Roll chromium_revision 9a34b2cc2d..78de17c053 (632146:632252)
by chromium-webrtc-autoroll
· 6 years ago
d036c65
Clarify and unify outgoing and incoming packet loss rate plots.
by Konrad Hofbauer
· 6 years ago
663844d
Update test code to use EncodedImage::Allocate
by Niels Möller
· 6 years ago
fd965c0
Always offer transport sequence number header extension for audio
by Per Kjellander
· 6 years ago
92e7c69
Revert "Update VP9EncoderImpl to use EncodedImage::Allocate"
by Niels Moller
· 6 years ago
8e847ee
Make recv_deltas optional in TransportFeedback packets
by Johannes Kron
· 6 years ago
69fb6c8
Allow DtlsTransport::Information() to be called off-thread
by Harald Alvestrand
· 6 years ago
068fc35
Break out parameters from EventLogAnalyzer to AnalyzerConfig struct.
by Bjorn Terelius
· 6 years ago
f0c366b
Cleanup of scenario test video stream setup.
by Sebastian Jansson
· 6 years ago
d00045e
Changing command line flag for scenario logs root directory.
by Sebastian Jansson
· 6 years ago
dac03d9
Move MediaConstraintsInterface to sdk/, and make it a concrete class
by Niels Möller
· 6 years ago
1d7bf89
Add LS_VERBOSE logging for target bitrate in GoogCC
by Evan Shrubsole
· 6 years ago
0179a3d
Roll chromium_revision bfd7bcf815..9a34b2cc2d (632040:632146)
by chromium-webrtc-autoroll
· 6 years ago
da825b1
Replace NOTREACHED with a break.
by Piotr (Peter) Slatala
· 6 years ago
1290fc7
Remove old accessor in GenericAckReceived
by Piotr (Peter) Slatala
· 6 years ago
788f577
Update the resolution check for VP8 simulcast.
by Mirta Dvornicic
· 6 years ago
6b88a8f
Introduce default video quality analyzer
by Artem Titov
· 6 years ago
b1ea48c
Roll chromium_revision 103665932a..bfd7bcf815 (631883:632040)
by chromium-webrtc-autoroll
· 6 years ago
5ae259e
Use a provider in rtc::Network to access the mDNS responder.
by Qingsi Wang
· 6 years ago
616b233
Add FullStackTest with simulated encoder overshooting
by Erik Språng
· 6 years ago
6c02541
Revert "Delete video source proxying in WebRtcVideoSendStream"
by Christian Fremerey
· 6 years ago
3588394
Roll chromium_revision d026ac796d..103665932a (631722:631883)
by chromium-webrtc-autoroll
· 6 years ago
dfd5c4b
Parse XR, FIR and PLI in rtc_event_log_parser.cc
by Bjorn Terelius
· 6 years ago
3c119fb
Handle HKDF key derivation when building with OpenSSL.
by Sergey Sablin
· 6 years ago
5e2aad1
Support GenericPacketReceived/Sent/AckReceived event logs.
by Piotr (Peter) Slatala
· 6 years ago
975a899
Roll chromium_revision aa7b61fdc4..d026ac796d (631597:631722)
by chromium-webrtc-autoroll
· 6 years ago
4a68fb9
Separate base minimum delay and minimum delay.
by Ruslan Burakov
· 6 years ago
69bb3af
Update EncodedFrameForMediaTransport to use Retain() rather than set_buffer + memcpy.
by Niels Möller
· 6 years ago
14a7cf9
Adds CallEncoder to ChannelSend.
by Sebastian Jansson
· 6 years ago
cbf5949
Update MultiplexEncoderAdapter to use EncodedImage::Allocate
by Niels Möller
· 6 years ago
448c387
IceTransportWithTransportChannel: Initialize |thread_checker_| in declaration
by Raphael Kubo da Costa
· 6 years ago
2bd54a1
Ensure TestPeers are destroyed at the end of Run.
by Mirko Bonadei
· 6 years ago
6aca0b7
Add |update_rect| field and UpdateRect struct to VideoFrame.
by Ilya Nikolaevskiy
· 6 years ago
7f24fb9
Add settings to turn off VP8 base layer qp limit
by Erik Språng
· 6 years ago
98bcd32
Remove always_passing_unittest.cc.
by Mirko Bonadei
· 6 years ago
b4f7ab1
Fix -Wunused-result warnings
by Hans Wennborg
· 6 years ago
eedb0a1
Roll chromium_revision 23b4d2134b..aa7b61fdc4 (631425:631597)
by chromium-webrtc-autoroll
· 6 years ago
a795c3b
Roll chromium_revision d366835eb8..23b4d2134b (631269:631425)
by chromium-webrtc-autoroll
· 6 years ago
dcbdd2c
Add Foundation.framework to cocoa_threading target
by Jiawei Ou
· 6 years ago
0874530
Add gn configs to remove the dependency to audio and video codecs.
by Jiawei Ou
· 6 years ago
3329be4
Roll chromium_revision b847e52039..d366835eb8 (631155:631269)
by chromium-webrtc-autoroll
· 6 years ago
464a557
Adds audio priority bitrate field trial parameter.
by Sebastian Jansson
· 6 years ago
eb81b47
Update H264EncoderImpl to use EncodedImage::Allocate
by Niels Möller
· 6 years ago
d3666b2
Introduce cross traffic for emulated network layer.
by Artem Titov
· 6 years ago
5c4ddad
Delete obsolete usage of FakeConstraints
by Niels Möller
· 6 years ago
9bf67ea
AEC3: Fix delay hysteresis validation
by Gustaf Ullberg
· 6 years ago
99b9149
Enable padding bit in TransportFeedback packets
by Johannes Kron
· 6 years ago
2ce0cb0
Add missing 'explicit' specifier to GainControlImpl
by Sam Zackrisson
· 6 years ago
eb17524
Migrate libevent task queue implementation to TaskQueueBase interface
by Danil Chapovalov
· 6 years ago
675e5aa
Roll chromium_revision b6a69427be..b847e52039 (631040:631155)
by chromium-webrtc-autoroll
· 6 years ago
a93b8b0
Update SimulcastTestFixtureImpl to use EncodedImage::Allocate
by Niels Möller
· 6 years ago
40027b1
Roll chromium_revision e4a7c15e8a..b6a69427be (630925:631040)
by chromium-webrtc-autoroll
· 6 years ago
a887c32
Roll chromium_revision 92e6bfe90b..e4a7c15e8a (630806:630925)
by chromium-webrtc-autoroll
· 6 years ago
26d0876
Roll chromium_revision 339f6a582b..92e6bfe90b (630696:630806)
by chromium-webrtc-autoroll
· 6 years ago
871e144
Revert "Reland "Partial frame capture API part 1""
by Ilya Nikolaevskiy
· 6 years ago
421c859
Remove crit_render_ lock from webrtc::GainControlImpl
by Sam Zackrisson
· 6 years ago
00f9400
Dump histogram data in AEC3 delay estimator
by Sam Zackrisson
· 6 years ago
271195f
Fix potential crash when building rtx packet
by Danil Chapovalov
· 6 years ago
501bfba
Split rtp_receiver for readability.
by Ruslan Burakov
· 6 years ago
b66003c
Delete video source proxying in WebRtcVideoSendStream
by Niels Möller
· 6 years ago
6df89cc
Revert "Partial frame capture API part 2"
by Ilya Nikolaevskiy
· 6 years ago
b00eb19
Removes Start/Stop on network emulation manager.
by Sebastian Jansson
· 6 years ago
eb7589e
Revert "Partial frame capture API part 3"
by Ilya Nikolaevskiy
· 6 years ago
fd5d473
Revert "Partial frame capture API part 6"
by Ilya Nikolaevskiy
· 6 years ago
85fc325
Revert "Partial frame capture API part 5"
by Ilya Nikolaevskiy
· 6 years ago
02f4e32
Make some new rtc_base targets publicly visible
by Karl Wiberg
· 6 years ago
f13c2cd
Roll chromium_revision eb2aa6ea6a..339f6a582b (630596:630696)
by chromium-webrtc-autoroll
· 6 years ago
61b4f74
Fix PeerConnectionInterface::StartRtcEventLog documentation.
by Mirko Bonadei
· 6 years ago
1a1c52b
H.264 temporal layers w/frame marking (PART 2/3)
by Johnny Lee
· 6 years ago
e556768
Roll chromium_revision eead273f0c..eb2aa6ea6a (630484:630596)
by chromium-webrtc-autoroll
· 6 years ago
157540a
Stop hard-coding default IDs for RTP extensions
by Elad Alon
· 6 years ago
efc9a14
Make UniqueNumberGenerator::AddKnownId() return a value
by Elad Alon
· 6 years ago
6ba2738
Roll chromium_revision d60317bbda..eead273f0c (630357:630484)
by chromium-webrtc-autoroll
· 6 years ago
5699142
Use c=IN IP4 <hostname> to support the presence of hostname candidates.
by Qingsi Wang
· 6 years ago
7832343
Revert "Enabling Simulcast use via AddTransceiver."
by Emircan Uysaler
· 6 years ago
836fee1
Calculate next process time in simulated network.
by Sebastian Jansson
· 6 years ago
f6adac8
Add rtc event generic packet sent and received.
by Piotr (Peter) Slatala
· 6 years ago
50930a6
Roll chromium_revision 46a21d8d05..d60317bbda (630250:630357)
by chromium-webrtc-autoroll
· 6 years ago
1d13b37
Update LibvpxVp8Encoder to use EncodedImage::Allocate
by Niels Möller
· 6 years ago
b7edf69
Delete rtc::File, usage replaced with FileWrapper
by Niels Möller
· 6 years ago
9f3aabb
Delete obsolete class cricket::VideoCapturer
by Niels Möller
· 6 years ago
494ff28
Delete unused media constraints
by Niels Möller
· 6 years ago
a8d48ab
Fix incorrect FPS measure when frame dropper kicks in
by Erik Språng
· 6 years ago
bdfadd6
Adds Stop methods to media streams in scenario framework.
by Sebastian Jansson
· 6 years ago
85eab49
Simplify peer connection smoke test to remove flakiness for now.
by Artem Titov
· 6 years ago
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