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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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81e1bf03961535f5e0bcdf510b9ffdb30e548c8e
81e1bf0
Remove using DegradationPreference from scenario_config.h
by Artem Titov
· 5 years ago
6542826
Add new tests with lossy networks on PC test framework
by Artem Titov
· 5 years ago
cd8a6e2
Add writing and parsing of the `abs-capture-time` RTP header extension.
by Chen Xing
· 5 years ago
53d45ba
Make TaskQueueFactory required construction parameter for Call
by Danil Chapovalov
· 5 years ago
84ce3c0
Macro rename s/CS_DEBUG_CHECKS/RTC_CS_DEBUG_CHECKS.
by Mirko Bonadei
· 5 years ago
a2b30d8
Add functions to read from/write to bitstream values with known max value
by Danil Chapovalov
· 5 years ago
9eee121
Switch py_quality_assessment to ABSL_FLAG.
by Mirko Bonadei
· 5 years ago
b60141b
Save and serialize the receive RIDs in MediaContentDescription
by Florent Castelli
· 5 years ago
e8ed830
WebRtcVideoChannel encoder fallback.
by philipel
· 5 years ago
e420c6a
Add missing include for memcpy/memcmp
by Artem Titov
· 5 years ago
6a2c1ba
Roll chromium_revision ba17fd6b36..13b7238371 (674288:674397)
by chromium-webrtc-autoroll
· 5 years ago
8fa7151
Replace the implementation of `GetContributingSources()` on the audio side.
by Chen Xing
· 5 years ago
16661eb
Fix: report video_bwe_stats as bytes per second, as specified in the unit
by Artem Titov
· 5 years ago
443b7ee
Destroy existing encoder instance before creating a new one.
by Sergey Silkin
· 5 years ago
2c5af4f
Add * and / operator into SamplesStatsCounter.
by Artem Titov
· 5 years ago
1d46f9c
Add RtpPacket::IsExtensionReserved().
by Erik Språng
· 5 years ago
6038926
Roll chromium_revision 35be8751a4..ba17fd6b36 (674036:674288)
by chromium-webrtc-autoroll
· 5 years ago
02d7d35
Revert "Add ability to set ssrcs of RtpSender at construction time"
by Amit Hilbuch
· 5 years ago
c442197
Check the rid direction matches the direction in simulcast description
by Florent Castelli
· 5 years ago
238aab9
Fix bug in use_datagram_transport configuration.
by Bjorn A Mellem
· 5 years ago
b073f1c
Only set the RtcEventLog for media transport when it's used for media.
by Bjorn A Mellem
· 5 years ago
73bfc0e
Roll chromium_revision 6f0434662d..35be8751a4 (673926:674036)
by chromium-webrtc-autoroll
· 5 years ago
ed56cf4
Remove deprecated version of Vp8FrameBufferControllerFactory::Create
by Elad Alon
· 5 years ago
e9d6e65
Add ability to set ssrcs of RtpSender at construction time
by Erik Språng
· 5 years ago
5ee6967
Don't reset encoder on max/min bitrate change.
by Sergey Silkin
· 5 years ago
bc70b61
Switch rnn_vad_tool to ABSL_FLAG.
by Mirko Bonadei
· 5 years ago
f1a7bb1
Stop using unnecessary gclient vars
by Oleh Prypin
· 5 years ago
45befc5
Pass FecControllerOverride to Vp8FrameBufferControllerFactory::Create
by Elad Alon
· 5 years ago
14be799
Switch neteq tools to ABSL_FLAG.
by Mirko Bonadei
· 5 years ago
e731a2e
Remove check on supported profile in favor of expilict disabling
by Artem Titov
· 5 years ago
bfd343b
Add totalDecodeTime to RTCInboundRTPStreamStats
by Johannes Kron
· 5 years ago
419aae2
Remove android_tools deps
by Yun Liu
· 5 years ago
ebdf9f8
Roll chromium_revision 097ffaa18d..6f0434662d (673789:673926)
by chromium-webrtc-autoroll
· 5 years ago
6fdfec1
Add overload to CreateIceTransport that takes additional dependencies
by Steve Anton
· 5 years ago
1cf9470
Roll chromium_revision 05067e74f0..097ffaa18d (673689:673789)
by chromium-webrtc-autoroll
· 5 years ago
5985a04
Add a field trial to control datagram transport use.
by Bjorn A Mellem
· 5 years ago
3e8ef94
Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.
by Chen Xing
· 5 years ago
62eb89d
Fixing possible overflow in NetEq buffle level filter.
by Minyue Li
· 5 years ago
5983087
Forced vp8 sw encoder fallback: only use min bitrate config if codec type is vp8.
by Åsa Persson
· 5 years ago
5b2ce12
Roll chromium_revision ce6d12c81b..05067e74f0 (673457:673689)
by chromium-webrtc-autoroll
· 5 years ago
ea95c37
Report freeze_time_ms from PC test framework
by Artem Titov
· 5 years ago
754c952
Don't do ToI420() for each frame while checking is it dummy
by Artem Titov
· 5 years ago
6d9f001
Fix regression in PC quality test.
by Artem Titov
· 5 years ago
a6cb150
Use Default instead of GlobalTaskQueueFactory to create AudioDeviceBuffer for ios
by Danil Chapovalov
· 5 years ago
3e3a6e5
Remove obsolete deps
by Artem Titov
· 5 years ago
896f4b6
Use Default instead of GlobalTaskQueueFactory to create AudioDeviceBuffer for android
by Danil Chapovalov
· 5 years ago
a63aede
Make VideoStreamEncoderInterface::SetFecControllerOverride pure virtual
by Elad Alon
· 5 years ago
65764e4
Add missing overrides in VideoEncoder proxies/adapters
by Elad Alon
· 5 years ago
2ce1da5
Roll chromium_revision dfe5e91525..ce6d12c81b (673350:673457)
by chromium-webrtc-autoroll
· 5 years ago
1e00dbc
Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly
by Min Wang
· 5 years ago
67daf71
Implement RtpVideoSender::SetFecAllowed()
by Elad Alon
· 5 years ago
099b02a
Get rid of deprecated version of NackSender::SendNack [2/2]
by Elad Alon
· 5 years ago
7e00c67
Pass FecControllerOverride to Vp8FrameBufferController
by Elad Alon
· 5 years ago
22896d4
Roll chromium_revision 27c2f87cf5..dfe5e91525 (672459:673350)
by chromium-webrtc-autoroll
· 5 years ago
8f01c4e
Define FecControllerOverride and plumb it down to VideoEncoder
by Elad Alon
· 5 years ago
52e5242
Add trait to Build/Parse DependencyDescriptor rtp header extension
by Danil Chapovalov
· 5 years ago
225842c
Initialize signal processing function pointers statically
by Karl Wiberg
· 5 years ago
a47ba41
Get rid of deprecated version of NackSender::SendNack [1/2]
by Elad Alon
· 5 years ago
a094849
RateControlSettings: add option to set min pixels per frame for libvpx vp8.
by Åsa Persson
· 5 years ago
60bfb3d
NetEQ: BackgroundNoise::Update returns true when the filter is updated
by Alessio Bazzica
· 5 years ago
825aad1
Delete almost all includes of platform_file.h
by Niels Möller
· 5 years ago
767efab
Delete method ReadableWav::Eof, which was used incorrectly.
by Niels Möller
· 5 years ago
71809c6
WindowCapturerWin: properly check return value of GetClassName
by Julien Isorce
· 5 years ago
9407776
Temporarily suppress -Wdeprecated-declarations to update jsoncpp.
by Mirko Bonadei
· 5 years ago
dd4dc7a
Adds additional fields to NetworkStateEstimate.
by Sebastian Jansson
· 5 years ago
49167de
Adds interface for remote network estimates to NetworkControllerInterface.
by Sebastian Jansson
· 5 years ago
2efae77
Add RTCStats for keyFramesEncoded, keyFramesDecoded.
by Rasmus Brandt
· 5 years ago
478cb46
Add GeneratePadding method to replace TimeToSendPadding
by Erik Språng
· 5 years ago
c2f5686
Extend structures to store updated version of the dependency descriptor
by Danil Chapovalov
· 5 years ago
a3f3ab9
Remove Simple Command Line Parser.
by Mirko Bonadei
· 5 years ago
4ba04b7
Delete RtcEventLogFactory factory as now unused
by Danil Chapovalov
· 5 years ago
36c8ef6
Cleanup Ulpfec receiver: remove 2 blocks RED packets support
by Ilya Nikolaevskiy
· 5 years ago
a36c591
Reland "Reland "Change buffer level filter to store current level in number of samples.""
by Jakob Ivarsson
· 5 years ago
b93af85
Revert "Reland "Change buffer level filter to store current level in number of samples.""
by Jakob Ivarsson
· 5 years ago
2d821c3
Allow VideoTimingExtension to be used with FEC
by Ilya Nikolaevskiy
· 5 years ago
e4ac723
Delete deprecated version of PeerConnectionFactoryInterface::StartAecDump
by Niels Möller
· 5 years ago
bca1485
Enable setting surface_ice_candidates_on_ice_transport_type_changed on the fly.
by Qingsi Wang
· 5 years ago
0ded32d
Reland "Change buffer level filter to store current level in number of samples."
by Jakob Ivarsson
· 5 years ago
4d69516
Don't use angle-bracket #includes for WebRTC's own files
by Oleh Prypin
· 5 years ago
c57b0ee
Fix for NACK retransmission in Scenario tests.
by Sebastian Jansson
· 5 years ago
be0adee
Add resolution bitrate thresholds to EncoderInfo.
by Sergey Silkin
· 5 years ago
2644a70
Delete rtc::TryCritScope as unused
by Danil Chapovalov
· 5 years ago
e8df482
Clean-up after CL #140941
by Elad Alon
· 5 years ago
db59de3
Add optimization to PacketRouter for large number of senders.
by Danil Chapovalov
· 5 years ago
90f3b89
Replace the implementation of `GetContributingSources()` on the video side.
by Chen Xing
· 5 years ago
3472b9a
Delete RTCInboundRTPStreamStats::fraction_lost
by Niels Möller
· 5 years ago
1d20185
Convert file objects to strings, before passing to PresubmitNotifyResult
by Niels Möller
· 5 years ago
7dd9969
Android: Expose getDisplaySize() helper function
by Magnus Jedvert
· 5 years ago
554be4b
Roll chromium_revision c30c3a10ff..27c2f87cf5 (672280:672459)
by chromium-webrtc-autoroll
· 5 years ago
c538506
Enable H.264 temporal scalability in simulcast.
by Johnny Lee
· 5 years ago
0d65fb5
Mass refactoring: Change JNI #includes to use full paths (webrtc/).
by Eric Stevenson
· 5 years ago
6e8bfae
Roll chromium_revision 42482d4f53..c30c3a10ff (672061:672280)
by chromium-webrtc-autoroll
· 5 years ago
a79abd4
Specify min_sdk_version for unittest apks also in GN configs
by Oleh Prypin
· 5 years ago
f97f342
Remove deprecated flags from compare_videos.py.
by Mirko Bonadei
· 5 years ago
75bc70c
Remove flags include from p2p/base/datagram_dtls_adaptor.cc.
by Mirko Bonadei
· 5 years ago
25ca0ac
Also fail CreateOffer and CreateAnswer if there is a session error
by Steve Anton
· 5 years ago
9a5c2e8
Remove unused command_line_parser dependency.
by Mirko Bonadei
· 5 years ago
04cffe3
Switch example peerconnection server to ABSL_FLAG.
by Mirko Bonadei
· 5 years ago
f03b365
Reland "Raise IllegalStateException for calls to retain() or release() on zero ref count"
by Niels Möller
· 5 years ago
3894078
Roll chromium_revision 6c5bbd86c3..42482d4f53 (671957:672061)
by chromium-webrtc-autoroll
· 5 years ago
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