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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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ad88c886d7f4cee4c76be740da207d812a086efa
ad88c88
Add API for returning a webrtc::DtlsTransport for a MID on a PC
by Harald Alvestrand
· 6 years ago
3e70781
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
by Yves Gerey
· 6 years ago
0f9c38e
Add new names of perf bots that will be migrated to LUCI
by Oleh Prypin
· 6 years ago
3312092
Keep bitrate constraints.
by Sergey Silkin
· 6 years ago
ff088a1
Reland "Run robolectric tests for Android on several Android API versions"
by Artem Titarenko
· 6 years ago
d5696fb
Add video support to LoopbackMediaTransport
by Niels Möller
· 6 years ago
bb9f4c1
Delete ssrc book-keeping in NetEq
by Niels Möller
· 6 years ago
48fcf94
React to changes in either width or height in iOS Metal renderer.
by Anders Carlsson
· 6 years ago
071edf3
Add missing files to AAR.
by Sami Kalliomäki
· 6 years ago
648a7ce
Delete method EncodedFrame::GetBitstream, part 1
by Niels Möller
· 6 years ago
0cc11b4
Android: Bump stack trace logging severity from debug to warning
by Magnus Jedvert
· 6 years ago
68478b8
Added user-defined predicate to filter video codec implementations.
by Yura Yaroshevich
· 6 years ago
7f7e973
Roll chromium_revision 82a8b043ef..28d6168850 (611537:611644)
by chromium-webrtc-autoroll
· 6 years ago
de10eea
AEC3: Fix ENR in the dominant nearend detection
by Gustaf Ullberg
· 6 years ago
cf69d22
AEC3: Optimizing the Update method of the FilterAnalyzer class.
by Jesús de Vicente Peña
· 6 years ago
154a262
Don't clear self.videoFrame when setting up OpenGL in the EAGL video view.
by Aaron Golden
· 6 years ago
2ba65c3
Fix webrtc-internal ios json config
by Artem Titarenko
· 6 years ago
ce1b140
Adding WinUWP compilation support to WebRTC.
by Robin Raymond
· 6 years ago
3bc24bc
Roll chromium_revision b04e513f82..82a8b043ef (611432:611537)
by chromium-webrtc-autoroll
· 6 years ago
5ec2c94
Reland "Delay call to Destroy until after SignalDone has finished firing."
by Zach Stein
· 6 years ago
0d007d7
Roll chromium_revision aed902039c..b04e513f82 (611312:611432)
by chromium-webrtc-autoroll
· 6 years ago
72d2ddd
Fix raddr on srflx and relay candidates
by Jeroen de Borst
· 6 years ago
889b587
Revert "Delay call to Destroy until after SignalDone has finished firing."
by Qingsi Wang
· 6 years ago
ab20b3f
Roll chromium_revision 81c26a093b..aed902039c (611047:611312)
by chromium-webrtc-autoroll
· 6 years ago
c7b8da4
Delay call to Destroy until after SignalDone has finished firing.
by Zach Stein
· 6 years ago
14f252a
AEC3: Add metrics for API call jitter
by Per Åhgren
· 6 years ago
10403ae
Add PeerConnection option to configure minimum audio jitter buffer delay.
by Jakob Ivarsson
· 6 years ago
c7f1a0a
Fix webrtc-internal configs to run perf tests on separate bots
by Artem Titarenko
· 6 years ago
8b55602
Batch RTC event log output if using the new wire format.
by Bjorn Terelius
· 6 years ago
352ce5c
Expose delayed packet outage as a cumulative metric of samples in the new getStats API.
by Jakob Ivarsson
· 6 years ago
59cfd35
Address vptr race condition while PeerConnection is destructed.
by Yves Gerey
· 6 years ago
09d6588
Change HdrMetadataExtension to ColorSpaceExtension
by Johannes Kron
· 6 years ago
53382cb
Move RtcpStatistics from common_types.h to a new header file
by Niels Möller
· 6 years ago
6b3d181
Remove unused BWE field trial strings.
by Bjorn Terelius
· 6 years ago
65c921c
Add setters to ColorSpace class
by Johannes Kron
· 6 years ago
196c5ba
Specific pacing configuration.
by Christoffer Rodbro
· 6 years ago
ba2840c
Various VP9 high fps fixes
by Ilya Nikolaevskiy
· 6 years ago
af52b68
Populate VideoSendTime extension network2 field when configured
by Danil Chapovalov
· 6 years ago
31a4331
Roll chromium_revision 208bb982f7..81c26a093b (610939:611047)
by chromium-webrtc-autoroll
· 6 years ago
c5095e5
Fix wrong forward declaration namespace.
by Mirko Bonadei
· 6 years ago
4e58444
Roll chromium_revision 00f78b5b14..208bb982f7 (610831:610939)
by chromium-webrtc-autoroll
· 6 years ago
74cdf78
add cstring include need for strncmp
by Michel Promonet
· 6 years ago
e38a5a1
Small cleanup to mediasession_unittest.cc
by Steve Anton
· 6 years ago
a6e034a
Rebase std::is_trivially_* with absl::is_trivially_*
by Jiawei Ou
· 6 years ago
622eeda
Bump variable sizes in response to fuzzer bug
by Jonas Olsson
· 6 years ago
b24c00f
Add AudioProcessingCaptureStats and a level estimator replacement
by Sam Zackrisson
· 6 years ago
2918d4e
Roll chromium_revision 7579fcbc1c..00f78b5b14 (610728:610831)
by chromium-webrtc-autoroll
· 6 years ago
e977199
Delete ChannelSend::RegisterTransport, replacing by construction argument
by Niels Möller
· 6 years ago
b253303
Add magjed as owner of rtc_tools.
by Patrik Höglund
· 6 years ago
856cf22
In ReceiveStatistics use monotonic clock instead of ntp clock
by Danil Chapovalov
· 6 years ago
22027b9
Add a new Task Queue for WinUWP.
by Robin Raymond
· 6 years ago
ff05816
Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric
by Sam Zackrisson
· 6 years ago
8ce0d2b
In ReceiveStatistic require callbacks during construction
by Danil Chapovalov
· 6 years ago
4c0cc5b
Reland Profile 2 to default profiles
by Emircan Uysaler
· 6 years ago
f1c194d
Roll chromium_revision d298cced6c..7579fcbc1c (610627:610728)
by chromium-webrtc-autoroll
· 6 years ago
05aee74
Roll chromium_revision f9be7d3d66..d298cced6c (610432:610627)
by Oleh Prypin
· 6 years ago
9289eda
Revert "Replace the IceConnectionState implementation."
by Alex Loiko
· 6 years ago
4f00075
Remove use of CodecSpecificInfo.codec_name
by Erik Språng
· 6 years ago
1e87b4f
Replace the IceConnectionState implementation.
by Jonas Olsson
· 6 years ago
57f3ad0
Adds stable bandwidth estimate to GoogCC.
by Sebastian Jansson
· 6 years ago
88ce4ef
Don't buffer encoded frames.
by Sergey Silkin
· 6 years ago
885cf60
Moves ProbeBitrateEstimator from DelayBasedBwe.
by Sebastian Jansson
· 6 years ago
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
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