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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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e3abb8134f27d4b21c1246b6e5f268f238e203f4
e3abb81
Decouple //rtc_base:rtc_base_tests_utils from gunit.
by Mirko Bonadei
· 6 years ago
8af8896
Expose jitter buffer flushes metric in new getStats api.
by Ruslan Burakov
· 6 years ago
b357e54
Add field trial config to disable pacer emergency stops.
by Christoffer Rodbro
· 6 years ago
6d254bc
Delete unused method NetEq::PacketBufferStatistics
by Niels Möller
· 6 years ago
5f2ffee
Clean up deprecated APM stats
by Sam Zackrisson
· 6 years ago
f40150d
Removing ANA enabling field trials.
by Minyue Li
· 6 years ago
2c977b4
Remove RSID from stream configs in new event log format.
by Bjorn Terelius
· 6 years ago
14dfe7f
[GN] Fix dependency rebasing in BUILD.gn files.
by Yves Gerey
· 6 years ago
254d869
Routing BitrateAllocationUpdate to audio codec.
by Sebastian Jansson
· 6 years ago
3890c1a
Roll chromium_revision 1500c78c93..f9be7d3d66 (610314:610432)
by chromium-webrtc-autoroll
· 6 years ago
3955a50
Metal: Don't render into an empty view.
by Peter Hanspers
· 6 years ago
777cf26
AEC3: Clockdrift detection
by Gustaf Ullberg
· 6 years ago
f259078
Use cropping aligning in video quality analysis tool
by Magnus Jedvert
· 6 years ago
ebb50c2
Fix setting max reordering threshold in ReceiveStatistics
by Danil Chapovalov
· 6 years ago
286df00
Add tool for aligning cropped region of video files
by Magnus Jedvert
· 6 years ago
8e66863
Remove cricket::UdpTransport.
by Mirko Bonadei
· 6 years ago
94c9420
Remove cricket::BundleFilter.
by Mirko Bonadei
· 6 years ago
eccfc47
Cleanup AimdRateController and remove RateControlRegion enum.
by Bjorn Terelius
· 6 years ago
42d2e4b
Increase test timeouts in TCPChannelClientTest
by Artem Titarenko
· 6 years ago
00dfe93
Remove superfluous constructor from dltsTransport
by Harald Alvestrand
· 6 years ago
44727b4
Cleanup rtcp StreamStatistician::OnRtpPacket
by Danil Chapovalov
· 6 years ago
af228ee
Disable flaky tests CallPerfTest.CaptureNtpTimeWithNetworkDelay on WIN.
by Alex Loiko
· 6 years ago
5486bcd
Remove SetChannelParameters function from API classes.
by philipel
· 6 years ago
ecd6205
Disable GoogCcNetworkControllerTest.DetectsHighRateInSafeResetTrial
by Alex Loiko
· 6 years ago
8ac05cc
Adds trial to use link capacity estimate in Opus encoder.
by Sebastian Jansson
· 6 years ago
2ff3f49
Move webrtc::CreatePeerConnectionFactory definition next to decl.
by Mirko Bonadei
· 6 years ago
d51b355
Delete unused NetEq Rtcp stats.
by Niels Möller
· 6 years ago
7c36c71
Roll chromium_revision 6931f4c0d0..1500c78c93 (610209:610314)
by chromium-webrtc-autoroll
· 6 years ago
8b5d9d8
Remove the audio/video split for the RTCP report intervals.
by Jiawei Ou
· 6 years ago
4a2dd7a
Roll chromium_revision 5825fead7b..6931f4c0d0 (610108:610209)
by chromium-webrtc-autoroll
· 6 years ago
540ef28
Adds OnReceivedUplinkAllocation method to AudioEncoder.
by Sebastian Jansson
· 6 years ago
6736df1
Moves BitrateAllocationUpdate to api.
by Sebastian Jansson
· 6 years ago
13e5903
Using unit classes in BitrateAllocationUpdate struct.
by Sebastian Jansson
· 6 years ago
e4cccae
Removed ability to set CryptoOptions through PeerConnectionFactory from bindings.
by Benjamin Wright
· 6 years ago
a526ae6
Roll chromium_revision 92f8c5b2a2..5825fead7b (609994:610108)
by chromium-webrtc-autoroll
· 6 years ago
5eae1d9
Remove legacy SetTargetTransferRateObserver
by Piotr (Peter) Slatala
· 6 years ago
37227be
Add check for media transport and bundle policy
by Piotr (Peter) Slatala
· 6 years ago
47dfdca
Create 'MaybeCreateMediaTransport' function
by Piotr (Peter) Slatala
· 6 years ago
64bfcde
Add sakal@ to OWNERS in android tests / aarproject directories.
by Sami Kalliomäki
· 6 years ago
4749e4e
Move HdrMetadata to ColorSpace
by Johannes Kron
· 6 years ago
ecf6315
AGC2 adaptive digital: remove unnecessary flag.
by Alessio Bazzica
· 6 years ago
8da7b35
AGC2 adaptive digital false by default
by Alessio Bazzica
· 6 years ago
cfddbb7
Add ios bindings for PeerConnectionState.
by Jonas Olsson
· 6 years ago
49a7843
Don't restart streams in scenario tests.
by Sebastian Jansson
· 6 years ago
0e4dfcb
Roll chromium_revision 16e6b25329..92f8c5b2a2 (609893:609994)
by chromium-webrtc-autoroll
· 6 years ago
59a01b0
Set Framerate in RTCVideoEncoderH264
by Qiang Chen
· 6 years ago
2b5b0e9
Disabling ScreenDrawerTest.TwoScreenDrawerLocks
by Alex Loiko
· 6 years ago
c4d5642
Revert "Default to dlopening the PipeWire."
by Oleh Prypin
· 6 years ago
c69a56e
Remove more unneeded things from ChannelSend
by Fredrik Solenberg
· 6 years ago
a13be01
Default to dlopening the PipeWire.
by Tomas Popela
· 6 years ago
c68d282
Add test PeerConnectionIntegrationTest.MediaTransportBidirectionalAudio
by Niels Möller
· 6 years ago
89c94b9
Adds target bandwidth to BitrateAllocator.
by Sebastian Jansson
· 6 years ago
66eedce
Roll chromium_revision 7d53bc243c..16e6b25329 (609559:609893)
by chromium-webrtc-autoroll
· 6 years ago
bd04f4a
Increase buffer level threshold in VP8/9 tests.
by Sergey Silkin
· 6 years ago
2222a80
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
by Niels Möller
· 6 years ago
38332cd
Add RTCP and simulcast support for RTCRtpReceiver::getParameters()
by Florent Castelli
· 6 years ago
4bc6045
Add output directory option for audioproc_f data dump files.
by Alessio Bazzica
· 6 years ago
388e4e9
Make RTC_LOG_FILE_LINE use its parameters
by Jonas Olsson
· 6 years ago
c20b82a
Remove unused variables in RtcEventAudioXStreamConfig::Copy()
by Bjorn Terelius
· 6 years ago
22b70ff
Move VideoCodecType from common_types.h to api/video/video_codec_type.h
by Niels Möller
· 6 years ago
22ff1a4
Fix threshold in VideoCodecTestLibvpx.ChangeFramerateVP9.
by Mirko Bonadei
· 6 years ago
6817038
APM audioproc_f: flag for AGC2 adaptive level estimator.
by Alessio Bazzica
· 6 years ago
44974e1
AEC3: Adding a correction factor for the Erle estimation that depends on the portion of the filter that is currently in use.
by Jesús de Vicente Peña
· 6 years ago
985a1f3
Add const or GUARDED_BY on a few ChannelSend members
by Niels Möller
· 6 years ago
5f00995
Using unit classes in AimdRateControl.
by Sebastian Jansson
· 6 years ago
50b8426
Roll chromium_revision 2f3cca903d..7d53bc243c (609431:609559)
by chromium-webrtc-autoroll
· 6 years ago
f85b6d2
Roll chromium_revision 9508bd7fec..2f3cca903d (609314:609431)
by chromium-webrtc-autoroll
· 6 years ago
b6787bc
Using data unit classes in DelayBasedBwe.
by Sebastian Jansson
· 6 years ago
2e0c655
[Sanitizers] Don't retry failed tests.
by Yves Gerey
· 6 years ago
b22f077
Adds FieldTrialConstrained class.
by Sebastian Jansson
· 6 years ago
76f5750
Roll chromium_revision 3efc758c50..9508bd7fec (609210:609314)
by chromium-webrtc-autoroll
· 6 years ago
85340ce
Move rtc::scoped_refptr to api/.
by Mirko Bonadei
· 6 years ago
52e69d7
Explicitly specify color space enum indices
by Johannes Kron
· 6 years ago
3a83748
New loss-based bandwidth control mechanism.
by Christoffer Rodbro
· 6 years ago
26e88b0
Replace RTC_DCHECK by RTC_DCHECK_RUN_ON for worker thread.
by Niels Möller
· 6 years ago
2058d52
Disabling test StunPortTest.TestPrepareAddressHostname on WIN.
by Alex Loiko
· 6 years ago
eb13484
Remove ChannelSendState
by Fredrik Solenberg
· 6 years ago
c3313a3
Make api:create_peerconnection_factory public.
by Mirko Bonadei
· 6 years ago
c5e8be3
Remove ChannelReceiveState
by Fredrik Solenberg
· 6 years ago
72bba62
Adds shared base class for data units.
by Sebastian Jansson
· 6 years ago
d474672
Make rtc_event_log protos publicly visible.
by Mirko Bonadei
· 6 years ago
78e88fe
Move NetworkStatistics and AudioDecodingCallStats from common_types.h
by Fredrik Solenberg
· 6 years ago
3cf8f3e
Adding empty api:create_peerconnection_factory.
by Mirko Bonadei
· 6 years ago
2ee41fe
Disabling test StunPortTest.TestPrepareAddressHostname on WIN.
by Alex Loiko
· 6 years ago
95adedb
Always compile VP9 source files.
by Mirko Bonadei
· 6 years ago
dced9f6
Delete class ChannelSendProxy
by Niels Möller
· 6 years ago
601504c
in RtcpTransceiver remove workaround for old bug in RtcpReceiver
by Danil Chapovalov
· 6 years ago
c3bd2fb
Roll chromium_revision 92e84c81c1..3efc758c50 (608282:609210)
by chromium-webrtc-autoroll
· 6 years ago
0a8bd9c
Adds clamping to TimeDelta.
by Sebastian Jansson
· 6 years ago
b5f8201
Adds scalar division to DataRate.
by Sebastian Jansson
· 6 years ago
8ef5793
Switch from RTC_DISABLE_VP9 to RTC_ENABLE_VP9.
by Mirko Bonadei
· 6 years ago
bd6ffaf
Fix small issues that stops the Chromium DEPS roll.
by Patrik Höglund
· 6 years ago
179a392
Implement TargetBitrate, NetworkRoute and overhead features of media transport interface.
by Piotr (Peter) Slatala
· 6 years ago
8c1e73b
Don't add empty extension list in event log parser.
by Sebastian Jansson
· 6 years ago
1eebec9
Fix data race in channel_send.cc
by Piotr (Peter) Slatala
· 6 years ago
b5bb513
Disable RTCStatsIntegrationTest.GetsStatsWhileDestroyingPeerConnection
by Yves Gerey
· 6 years ago
6eb8a16
Exposing audio and video engines directly.
by Sebastian Jansson
· 6 years ago
eee3920
Don't poll EncoderInfo from encoder twice per frame
by Erik Språng
· 6 years ago
645a3af
Remove unused/unnecessary things from ChannelSend.
by Fredrik Solenberg
· 6 years ago
a32d7e2
Add default values for PlayoutDelay in RTPVideoHeader.
by Niels Möller
· 6 years ago
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