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352ce5c
Expose delayed packet outage as a cumulative metric of samples in the new getStats API.
by Jakob Ivarsson
· 5 years ago
59cfd35
Address vptr race condition while PeerConnection is destructed.
by Yves Gerey
· 5 years ago
09d6588
Change HdrMetadataExtension to ColorSpaceExtension
by Johannes Kron
· 5 years ago
53382cb
Move RtcpStatistics from common_types.h to a new header file
by Niels Möller
· 5 years ago
6b3d181
Remove unused BWE field trial strings.
by Bjorn Terelius
· 5 years ago
65c921c
Add setters to ColorSpace class
by Johannes Kron
· 5 years ago
196c5ba
Specific pacing configuration.
by Christoffer Rodbro
· 5 years ago
ba2840c
Various VP9 high fps fixes
by Ilya Nikolaevskiy
· 5 years ago
af52b68
Populate VideoSendTime extension network2 field when configured
by Danil Chapovalov
· 5 years ago
31a4331
Roll chromium_revision 208bb982f7..81c26a093b (610939:611047)
by chromium-webrtc-autoroll
· 5 years ago
c5095e5
Fix wrong forward declaration namespace.
by Mirko Bonadei
· 5 years ago
4e58444
Roll chromium_revision 00f78b5b14..208bb982f7 (610831:610939)
by chromium-webrtc-autoroll
· 5 years ago
74cdf78
add cstring include need for strncmp
by Michel Promonet
· 5 years ago
e38a5a1
Small cleanup to mediasession_unittest.cc
by Steve Anton
· 5 years ago
a6e034a
Rebase std::is_trivially_* with absl::is_trivially_*
by Jiawei Ou
· 5 years ago
622eeda
Bump variable sizes in response to fuzzer bug
by Jonas Olsson
· 5 years ago
b24c00f
Add AudioProcessingCaptureStats and a level estimator replacement
by Sam Zackrisson
· 5 years ago
2918d4e
Roll chromium_revision 7579fcbc1c..00f78b5b14 (610728:610831)
by chromium-webrtc-autoroll
· 5 years ago
e977199
Delete ChannelSend::RegisterTransport, replacing by construction argument
by Niels Möller
· 5 years ago
b253303
Add magjed as owner of rtc_tools.
by Patrik Höglund
· 5 years ago
856cf22
In ReceiveStatistics use monotonic clock instead of ntp clock
by Danil Chapovalov
· 5 years ago
22027b9
Add a new Task Queue for WinUWP.
by Robin Raymond
· 5 years ago
ff05816
Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric
by Sam Zackrisson
· 5 years ago
8ce0d2b
In ReceiveStatistic require callbacks during construction
by Danil Chapovalov
· 5 years ago
4c0cc5b
Reland Profile 2 to default profiles
by Emircan Uysaler
· 5 years ago
f1c194d
Roll chromium_revision d298cced6c..7579fcbc1c (610627:610728)
by chromium-webrtc-autoroll
· 5 years ago
05aee74
Roll chromium_revision f9be7d3d66..d298cced6c (610432:610627)
by Oleh Prypin
· 5 years ago
9289eda
Revert "Replace the IceConnectionState implementation."
by Alex Loiko
· 5 years ago
4f00075
Remove use of CodecSpecificInfo.codec_name
by Erik Språng
· 5 years ago
1e87b4f
Replace the IceConnectionState implementation.
by Jonas Olsson
· 5 years ago
57f3ad0
Adds stable bandwidth estimate to GoogCC.
by Sebastian Jansson
· 5 years ago
88ce4ef
Don't buffer encoded frames.
by Sergey Silkin
· 5 years ago
885cf60
Moves ProbeBitrateEstimator from DelayBasedBwe.
by Sebastian Jansson
· 5 years ago
e3abb81
Decouple //rtc_base:rtc_base_tests_utils from gunit.
by Mirko Bonadei
· 5 years ago
8af8896
Expose jitter buffer flushes metric in new getStats api.
by Ruslan Burakov
· 5 years ago
b357e54
Add field trial config to disable pacer emergency stops.
by Christoffer Rodbro
· 5 years ago
6d254bc
Delete unused method NetEq::PacketBufferStatistics
by Niels Möller
· 5 years ago
5f2ffee
Clean up deprecated APM stats
by Sam Zackrisson
· 6 years ago
f40150d
Removing ANA enabling field trials.
by Minyue Li
· 5 years ago
2c977b4
Remove RSID from stream configs in new event log format.
by Bjorn Terelius
· 5 years ago
14dfe7f
[GN] Fix dependency rebasing in BUILD.gn files.
by Yves Gerey
· 5 years ago
254d869
Routing BitrateAllocationUpdate to audio codec.
by Sebastian Jansson
· 5 years ago
3890c1a
Roll chromium_revision 1500c78c93..f9be7d3d66 (610314:610432)
by chromium-webrtc-autoroll
· 5 years ago
3955a50
Metal: Don't render into an empty view.
by Peter Hanspers
· 5 years ago
777cf26
AEC3: Clockdrift detection
by Gustaf Ullberg
· 5 years ago
f259078
Use cropping aligning in video quality analysis tool
by Magnus Jedvert
· 5 years ago
ebb50c2
Fix setting max reordering threshold in ReceiveStatistics
by Danil Chapovalov
· 5 years ago
286df00
Add tool for aligning cropped region of video files
by Magnus Jedvert
· 5 years ago
8e66863
Remove cricket::UdpTransport.
by Mirko Bonadei
· 5 years ago
94c9420
Remove cricket::BundleFilter.
by Mirko Bonadei
· 5 years ago
eccfc47
Cleanup AimdRateController and remove RateControlRegion enum.
by Bjorn Terelius
· 5 years ago
42d2e4b
Increase test timeouts in TCPChannelClientTest
by Artem Titarenko
· 5 years ago
00dfe93
Remove superfluous constructor from dltsTransport
by Harald Alvestrand
· 5 years ago
44727b4
Cleanup rtcp StreamStatistician::OnRtpPacket
by Danil Chapovalov
· 5 years ago
af228ee
Disable flaky tests CallPerfTest.CaptureNtpTimeWithNetworkDelay on WIN.
by Alex Loiko
· 5 years ago
5486bcd
Remove SetChannelParameters function from API classes.
by philipel
· 6 years ago
ecd6205
Disable GoogCcNetworkControllerTest.DetectsHighRateInSafeResetTrial
by Alex Loiko
· 5 years ago
8ac05cc
Adds trial to use link capacity estimate in Opus encoder.
by Sebastian Jansson
· 5 years ago
2ff3f49
Move webrtc::CreatePeerConnectionFactory definition next to decl.
by Mirko Bonadei
· 5 years ago
d51b355
Delete unused NetEq Rtcp stats.
by Niels Möller
· 5 years ago
7c36c71
Roll chromium_revision 6931f4c0d0..1500c78c93 (610209:610314)
by chromium-webrtc-autoroll
· 5 years ago
8b5d9d8
Remove the audio/video split for the RTCP report intervals.
by Jiawei Ou
· 5 years ago
4a2dd7a
Roll chromium_revision 5825fead7b..6931f4c0d0 (610108:610209)
by chromium-webrtc-autoroll
· 5 years ago
540ef28
Adds OnReceivedUplinkAllocation method to AudioEncoder.
by Sebastian Jansson
· 5 years ago
6736df1
Moves BitrateAllocationUpdate to api.
by Sebastian Jansson
· 5 years ago
13e5903
Using unit classes in BitrateAllocationUpdate struct.
by Sebastian Jansson
· 5 years ago
e4cccae
Removed ability to set CryptoOptions through PeerConnectionFactory from bindings.
by Benjamin Wright
· 5 years ago
a526ae6
Roll chromium_revision 92f8c5b2a2..5825fead7b (609994:610108)
by chromium-webrtc-autoroll
· 5 years ago
5eae1d9
Remove legacy SetTargetTransferRateObserver
by Piotr (Peter) Slatala
· 5 years ago
37227be
Add check for media transport and bundle policy
by Piotr (Peter) Slatala
· 5 years ago
47dfdca
Create 'MaybeCreateMediaTransport' function
by Piotr (Peter) Slatala
· 5 years ago
64bfcde
Add sakal@ to OWNERS in android tests / aarproject directories.
by Sami Kalliomäki
· 5 years ago
4749e4e
Move HdrMetadata to ColorSpace
by Johannes Kron
· 5 years ago
ecf6315
AGC2 adaptive digital: remove unnecessary flag.
by Alessio Bazzica
· 5 years ago
8da7b35
AGC2 adaptive digital false by default
by Alessio Bazzica
· 5 years ago
cfddbb7
Add ios bindings for PeerConnectionState.
by Jonas Olsson
· 5 years ago
49a7843
Don't restart streams in scenario tests.
by Sebastian Jansson
· 5 years ago
0e4dfcb
Roll chromium_revision 16e6b25329..92f8c5b2a2 (609893:609994)
by chromium-webrtc-autoroll
· 5 years ago
59a01b0
Set Framerate in RTCVideoEncoderH264
by Qiang Chen
· 5 years ago
2b5b0e9
Disabling ScreenDrawerTest.TwoScreenDrawerLocks
by Alex Loiko
· 5 years ago
c4d5642
Revert "Default to dlopening the PipeWire."
by Oleh Prypin
· 5 years ago
c69a56e
Remove more unneeded things from ChannelSend
by Fredrik Solenberg
· 5 years ago
a13be01
Default to dlopening the PipeWire.
by Tomas Popela
· 5 years ago
c68d282
Add test PeerConnectionIntegrationTest.MediaTransportBidirectionalAudio
by Niels Möller
· 5 years ago
89c94b9
Adds target bandwidth to BitrateAllocator.
by Sebastian Jansson
· 5 years ago
66eedce
Roll chromium_revision 7d53bc243c..16e6b25329 (609559:609893)
by chromium-webrtc-autoroll
· 5 years ago
bd04f4a
Increase buffer level threshold in VP8/9 tests.
by Sergey Silkin
· 5 years ago
2222a80
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
by Niels Möller
· 5 years ago
38332cd
Add RTCP and simulcast support for RTCRtpReceiver::getParameters()
by Florent Castelli
· 5 years ago
4bc6045
Add output directory option for audioproc_f data dump files.
by Alessio Bazzica
· 5 years ago
388e4e9
Make RTC_LOG_FILE_LINE use its parameters
by Jonas Olsson
· 5 years ago
c20b82a
Remove unused variables in RtcEventAudioXStreamConfig::Copy()
by Bjorn Terelius
· 5 years ago
22b70ff
Move VideoCodecType from common_types.h to api/video/video_codec_type.h
by Niels Möller
· 5 years ago
22ff1a4
Fix threshold in VideoCodecTestLibvpx.ChangeFramerateVP9.
by Mirko Bonadei
· 5 years ago
6817038
APM audioproc_f: flag for AGC2 adaptive level estimator.
by Alessio Bazzica
· 5 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
· 5 years ago
985a1f3
Add const or GUARDED_BY on a few ChannelSend members
by Niels Möller
· 5 years ago
5f00995
Using unit classes in AimdRateControl.
by Sebastian Jansson
· 5 years ago
50b8426
Roll chromium_revision 2f3cca903d..7d53bc243c (609431:609559)
by chromium-webrtc-autoroll
· 5 years ago
f85b6d2
Roll chromium_revision 9508bd7fec..2f3cca903d (609314:609431)
by chromium-webrtc-autoroll
· 5 years ago
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