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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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ec22183f4368bbd6f0e6938417ce4ddfa7566485
ec22183
Revert lock for logging to CriticalSection
by Danil Chapovalov
· 5 years ago
9982efa
MediaStreamInterface: introduce encoded sinks.
by Markus Handell
· 5 years ago
b86a177
Expose ABGRToI420 in YuvHelper.
by Sami Kalliomäki
· 5 years ago
e835fc0
Add UMA counter for audio interruptions
by Henrik Lundin
· 5 years ago
c4e506e
Fix writing into closed FrameWriter when dumping video in PC framework
by Artem Titov
· 5 years ago
22ae169
Remove dependencies on Chromium's webrtc_overrides.
by Mirko Bonadei
· 5 years ago
108a2f0
Preventively fix missing braces warnings.
by Mirko Bonadei
· 5 years ago
09c452e
Split P2PTransportChannel
by Jonas Oreland
· 5 years ago
d51cc7b
Add absolute capture time property to rtp sources.
by Ruslan Burakov
· 5 years ago
bb55e0b
Clarifies identification of default communication device in ADM2
by henrika
· 5 years ago
eec1d04
Roll chromium_revision 8b5b046f02..022da7ca60 (715044:716991)
by chromium-webrtc-autoroll
· 5 years ago
6dd488b
RTC_EXPORT RTCStatsMember's specialized members.
by Mirko Bonadei
· 5 years ago
dca1449
Makes RoundRobinPacketQueue use same field trials as PacingController.
by Erik Språng
· 5 years ago
5831dda
Introduce IVF file reader
by Artem Titov
· 5 years ago
1721de1
Add STUN_ATTR_GOOG_MISC_INFO
by Jonas Oreland
· 5 years ago
2f385d2
Manual chromium roll: Compile using JDK 11
by Yves Gerey
· 5 years ago
3b1a8bb
Account for rounding errors in dyanmic pacing mode.
by Erik Språng
· 5 years ago
83b2862
Protect against NumberOfEnumeratedDevices and Get[In|Out]putDeviceNames returning inconsistent results.
by Tim Haloun
· 5 years ago
4dd56a3
ACM: Adding unittests for the remixing functionality
by Per Åhgren
· 5 years ago
0e3198e
Refactoring of the analog AGC functionality to add multichannel support
by Per Åhgren
· 5 years ago
f3fcde3
Store delay measurements as struct instead of std::pair
by Björn Terelius
· 5 years ago
fa7a8ca
Revert "Prepares PacingController for simplified packet queue."
by Erik Språng
· 5 years ago
0660cee
Add scale and compare methods to VideoFrame::UpdateRect
by Ilya Nikolaevskiy
· 5 years ago
332274d
Adding GetInDtx to WebRTC Opus Interface.
by Minyue Li
· 5 years ago
67e5bd3
[Android SDK] Update to version 'Q' (29) to unblock autoroller/roll_deps.py
by Yves Gerey
· 5 years ago
063c7d1
In dependency descriptor remove extended fields indicator
by Danil Chapovalov
· 5 years ago
fe04775
Fix a bug in interruption metrics
by Henrik Lundin
· 5 years ago
29e07e5
Add @Nullable annotations to quiet errorprone.
by Yves Gerey
· 5 years ago
0197887
Stop using DEPRECATED_SingleThreadedTaskQueueForTesting in MultiStreamTester
by Danil Chapovalov
· 5 years ago
3574d05
EventLogVisualizer pacer delay plot fix.
by philipel
· 5 years ago
287e464
Change VideoAdapter::OnResolutionFramerateRequest to VideoAdapter::OnSinkWants
by Rasmus Brandt
· 5 years ago
8fe22fa
StreamSynchronizationTest: Replace class Time with SimulatedClock.
by Åsa Persson
· 5 years ago
db82cd7
Make android_compile_x64_rel actually compile in x64.
by Yves Gerey
· 5 years ago
351173c
Tests that all available audio devices can be selected and used by the ADM.
by henrika
· 5 years ago
8ae70f6
Enable WebRTC-Bwe-MaxRttLimit by default.
by Nikita Zetilov
· 5 years ago
56d9452
Move stun.h to api/.
by Patrik Höglund
· 5 years ago
aa3f5da
Fork VCMPacket for PacketBuffer into own struct
by Danil Chapovalov
· 5 years ago
acdc22d
Prepares PacingController for simplified packet queue.
by Erik Språng
· 5 years ago
ccf12c6
Reland "Add AV1 RtpDepacketizer class"
by Danil Chapovalov
· 5 years ago
cb0b874
Add source-side perf upload script for WebRTC.
by Patrik Höglund
· 5 years ago
4186603
Always record timestamp of keyframe request.
by Markus Handell
· 5 years ago
92dd35d
Reland "Force Chromium deps on the WebRTC component."
by Mirko Bonadei
· 5 years ago
9f99175
Revert "Add AV1 RtpDepacketizer class"
by Yves Gerey
· 5 years ago
49470c2
Add AV1 RtpDepacketizer class
by Danil Chapovalov
· 5 years ago
2419dc9
Revert "Force Chromium deps on the WebRTC component."
by Mirko Bonadei
· 5 years ago
313a10e
p2p_transport_channel_unittest : put frequently used checks into functions.
by Jonas Oreland
· 5 years ago
39bab5a
Add missing assert.h for win no-test build
by Jerome Humbert
· 5 years ago
408cb4b
Make SCTPtransport enter "closed" state when DTLStransport does.
by Harald Alvestrand
· 5 years ago
e6eded3
VideoFrame: Store a reference to an encoded frame
by Markus Handell
· 5 years ago
ef6fe0c
Use GetDefaultAudioEndpoint for the default communications device as well as the vanilla default device
by Tim Haloun
· 5 years ago
25ec888
Make ICE transports injectable.
by Qingsi Wang
· 5 years ago
53e157d
Force Chromium deps on the WebRTC component.
by Mirko Bonadei
· 5 years ago
bb55c5e
Correct the upmixing of mono to stereo in ACM2
by Per Åhgren
· 5 years ago
fd0e32a
Fix filtering of small packets in delay-based BWE
by Björn Terelius
· 5 years ago
7b46e17
In rtc::ByteBuffer drop support for ORDER_HOST as unused
by Danil Chapovalov
· 5 years ago
74f35e4
Add support for dynamic processing mode in PacedSender.
by Erik Språng
· 5 years ago
02fac7d
Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
by Mirko Bonadei
· 5 years ago
eb48799
Prepares PacingController for scheduled send tasks.
by Erik Språng
· 5 years ago
3527a4f
In PacketBuffer split logic for detecting frame boundaries and assembling frame.
by Danil Chapovalov
· 5 years ago
01294f0
Don't configure video codec switching if no video stream has been created.
by philipel
· 5 years ago
bb0aac2
Reduce verbosity of logging around playout underrun count on iOS.
by Noah Richards
· 5 years ago
f11b46b
Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""
by Mirko Bonadei
· 5 years ago
f181137
Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
by Mirko Bonadei
· 5 years ago
1b66890
Stop using DEPRECATED_SingleThreadedTaskQueueForTesting in call tests
by Danil Chapovalov
· 5 years ago
60bd1ae
Detach thread checker in RtcEventLogImpl destructor.
by Sebastian Jansson
· 5 years ago
048b10a
Correcting the ACM upmixing from mono/stereo to surround
by Per Åhgren
· 5 years ago
50f8686
Providing EmulatedRoute instances when creating TcpMessageRoute
by Sebastian Jansson
· 5 years ago
017c84f
Synchronize is_screencast_ state in AndroidVideoTrackSource.
by Jakob Ivarsson
· 5 years ago
c20a19c
Allow extracting the linear AEC output
by Per Åhgren
· 5 years ago
06a394e
Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""
by Mirko Bonadei
· 5 years ago
57d53cf
Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
by Mirko Bonadei
· 5 years ago
759f161
Use explicit template decl/def for RTCNonStandardStatsMember.
by Mirko Bonadei
· 5 years ago
c5ec54e
Add SetIsScreencast method to VideoSource.
by Jakob Ivarsson
· 5 years ago
c1dac38
Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""
by Mirko Bonadei
· 5 years ago
f3dccc6
Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
by Mirko Bonadei
· 5 years ago
1242d9c
Reland Cleanup MultiStreamTester
by Danil Chapovalov
· 5 years ago
d4002a7
RTC_EXPORT missing symbols for Chromium's component build.
by Mirko Bonadei
· 5 years ago
f47c2ab
RTC_EXPORT webrtc::RTCNonStandardStatsMember<T>.
by Mirko Bonadei
· 5 years ago
4ed17ff
Roll chromium_revision 3b0967fe5b..0717a4dc31 (714463:714607)
by chromium-webrtc-autoroll
· 5 years ago
86b33e0
Don't ask for the friendly name of a default device if we failed to enumerate it.
by Tim Haloun
· 5 years ago
dcc910a
Adds simulated TCP message route for testing.
by Sebastian Jansson
· 5 years ago
b4463ee
Add ability to specify custom video source for PC framework.
by Artem Titov
· 5 years ago
815e00c
Revert "Reset RtpFrameReferenceFinder on long pause"
by Ilya Nikolaevskiy
· 5 years ago
059daa4
Don't leak device moniker when BindToStorage fails.
by Tim Haloun
· 5 years ago
2bd2d85
Roll chromium_revision 3861228399..3b0967fe5b (714324:714463)
by chromium-webrtc-autoroll
· 5 years ago
fe31b79
Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""
by Mirko Bonadei
· 5 years ago
93f7907
Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
by Mirko Bonadei
· 5 years ago
051cae5
RTC_EXPORT IceServer, StatsReport::{IdBase,Value}.
by Mirko Bonadei
· 5 years ago
bd27b14
Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""
by Mirko Bonadei
· 5 years ago
ba916b7
Mark scoped_refptr move and swap operations as noexcept
by Danil Chapovalov
· 5 years ago
bcec8a7
Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
by Mirko Bonadei
· 5 years ago
8fa616f
Add 2 more RTC_EXPORT.
by Mirko Bonadei
· 5 years ago
c65de42
Fix NetworkEmulationManagerTest.ThroughputStats flakiness (again).
by Yves Gerey
· 5 years ago
229035d
Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""
by Mirko Bonadei
· 5 years ago
d1f40f3
Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
by Mirko Bonadei
· 5 years ago
8cbb33c
Roll chromium_revision e2cf3bc09c..3861228399 (714186:714324)
by chromium-webrtc-autoroll
· 5 years ago
251b0dc
Simplified throughput estimator
by Björn Terelius
· 5 years ago
62a19d0
RTC_EXPORT RTCStatsMember::StaticType and VideoFrame::Builder.
by Mirko Bonadei
· 5 years ago
3a65f39
Expose NetEqDecodingTest for re-use in chromium tests.
by Yves Gerey
· 5 years ago
64e07f4
Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""
by Mirko Bonadei
· 5 years ago
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