Sign in
webrtc
/
src
/
cfe75c12ee04d17e7898ebc0a8ad1051b6627e53
cfe75c1
Roll chromium_revision 236f912c9d..e0ea7be7d1 (728281:728381)
by chromium-webrtc-autoroll
· 5 years ago
f3aa632
Replace the ExperimentalAgc config with the new config format
by Per Åhgren
· 5 years ago
1f74351
Roll chromium_revision 9427b39371..236f912c9d (728171:728281)
by chromium-webrtc-autoroll
· 5 years ago
2257c08
[Cleanup/Optim] Pass IPAddress by const reference.
by Yves Gerey
· 5 years ago
d572748
Run delay tasks on time when using GlobalSimulatedTimeController.
by philipel
· 5 years ago
3828c30
Delete unused member BasicPortAllocator::allow_tcp_listen_
by Niels Möller
· 5 years ago
12e319a
Merge the preambles of the ProcessStream implementations
by Sam Zackrisson
· 5 years ago
0f14db2
Reduce for reallocations the pre-amplifier and high-pass filter
by Per Åhgren
· 5 years ago
df344bb
Roll chromium_revision d4992c6f92..9427b39371 (728071:728171)
by chromium-webrtc-autoroll
· 5 years ago
873610c
Fix updating degradation preference in SetRtpParameters.
by Mirta Dvornicic
· 5 years ago
308bc64
Remove one acquisition of capture lock in APM AudioFrame API
by Sam Zackrisson
· 5 years ago
2bd85ab
Avoid AGC2 runtime allocation and activate it on demand
by Per Åhgren
· 5 years ago
83ee982
Delete p2p/base/packet_transport_interface.h
by Niels Möller
· 5 years ago
c073471
APM: Move the TransientSuppression activation to the apm_config
by Per Åhgren
· 5 years ago
045c36d
Roll chromium_revision 81693dc9aa..d4992c6f92 (727940:728071)
by chromium-webrtc-autoroll
· 5 years ago
65bbcab
[Android] Replace java_files with sources
by Natalie Chouinard
· 5 years ago
29fec66
AEC3: Remove metrics that are not used for analysis
by Per Åhgren
· 5 years ago
d2fb5f5
Fixes WebRtcAudioTrack crash while stopping
by Alex Narest
· 5 years ago
a688d11
Return unavailable rate rather than garbage value.
by Yves Gerey
· 5 years ago
cf4c872
APM: Make the GetStatistics call independent of the locks in APM
by Per Åhgren
· 5 years ago
a43777d
Roll chromium_revision 2d48822491..81693dc9aa (727839:727940)
by chromium-webrtc-autoroll
· 5 years ago
1c34ca7
Roll chromium_revision f19d6cb823..2d48822491 (727734:727839)
by chromium-webrtc-autoroll
· 5 years ago
9d2c2db
Roll chromium_revision 9986f2241a..f19d6cb823 (727633:727734)
by chromium-webrtc-autoroll
· 5 years ago
8ac7912
Roll chromium_revision d814fc7ea9..9986f2241a (727531:727633)
by chromium-webrtc-autoroll
· 5 years ago
f2dc059
Roll chromium_revision 6f7e5e79ce..d814fc7ea9 (727038:727531)
by chromium-webrtc-autoroll
· 5 years ago
26762d0
Add video codec AV1 to the deprecated android decoder/encoder wrappers
by Danil Chapovalov
· 5 years ago
38a55a0
Roll chromium_revision 937a78378f..6f7e5e79ce (726883:727038)
by chromium-webrtc-autoroll
· 5 years ago
a79fc59
Roll chromium_revision e4c6d7fe53..937a78378f (726742:726883)
by Yves Gerey
· 5 years ago
b5159fe
Revert "Reland "Reland "Distinguish between send and receive video codecs"""
by Olga Sharonova
· 5 years ago
f6b875c
Fixed crash on iOS13, methods beginGeneratingDeviceOrientationNotifications and endGeneratingDeviceOrientationNotifications.
by Andrey Efremov
· 5 years ago
4e64e60
Reland "Reland "Distinguish between send and receive video codecs""
by Johannes Kron
· 5 years ago
23df143
Roll chromium_revision 26cf7e7d6c..e4c6d7fe53 (725941:726742)
by Yves Gerey
· 5 years ago
077ee35
Remove unused parameter in RtpFragmentize
by Jiwon Kim
· 5 years ago
41875aa
add rotationOverride for RTCEAGLVideoView
by CZ Theng
· 5 years ago
2e8e1c6
Open up for do the noise suppressor analysis on the linear AEC output
by Per Åhgren
· 5 years ago
9136abb
AEC3: Ensure that the data size in the reverb computer is not fixed
by Per Åhgren
· 5 years ago
c8f3134
Parse max-fr and max-fs from SDP FMTP line
by Johannes Kron
· 5 years ago
5cad55b
Signal requested resolution alignment requirements from sinks to sources.
by Rasmus Brandt
· 5 years ago
c042425
Make the high-pass filter operate in full-band
by Per Åhgren
· 5 years ago
26335a9
Roll chromium_revision 98124fd660..26cf7e7d6c (725465:725941)
by Yves Gerey
· 5 years ago
7ab41e5
Fix typo in abseil-in-webrtc.md.
by Rasmus Brandt
· 5 years ago
ae10029
Prevents probing while paused.
by Erik Språng
· 5 years ago
768c5f4
Roll chromium_revision faed30b47a..98124fd660 (724977:725465)
by Yves Gerey
· 5 years ago
6fd58b3
Add maxFramerate support to SimulcastEncoderAdapter
by Florent Castelli
· 5 years ago
9b540cb
Correctly process disabled streams in FrameEncodeMetadataWriter
by Ilya Nikolaevskiy
· 5 years ago
00a1bcb
Ensure that unset capture timestamp wouldn't cause incorrect SR rtp timestamps
by Ilya Nikolaevskiy
· 5 years ago
f4cf4c7
Don't allow creation of sockets for wild card IPs in emulated networks.
by Sebastian Jansson
· 5 years ago
3a8df88
Add field trial to avoid extra backoffs in AIMD rate control.
by Björn Terelius
· 5 years ago
32fe4ef
Move vp9 rtp depacketization to VideoRtpDepacketizerVp9
by Danil Chapovalov
· 5 years ago
094396f
Use a fake clock for rtc::Thread::PostDelayedTask test
by Steve Anton
· 5 years ago
f9d92ed
Revert "Reland "Distinguish between send and receive video codecs""
by Ilya Nikolaevskiy
· 5 years ago
2697ac1
Stop an SCTP connection when the DTLS transport closes.
by Harald Alvestrand
· 5 years ago
8525a80
Add ability to resize buffers pool in decoder and use it in IVF generator
by Artem Titov
· 5 years ago
77eb338
Reland "Distinguish between send and receive video codecs"
by Johannes Kron
· 5 years ago
5331079
Protect against assigning current_offset_ negative value.
by Raman Budny
· 5 years ago
cebdbf6
switch RtpVideoStreamReceiver to use VideoRtpDepacketizer interface
by Danil Chapovalov
· 5 years ago
0f6bf75
Make video engine tests aware of padding packets
by Danil Chapovalov
· 5 years ago
73eb784
Don't crash the test process when X11 isn't available.
by Patrik Höglund
· 5 years ago
774fb93
Roll chromium_revision cd7700164d..faed30b47a (724740:724977)
by Yves Gerey
· 5 years ago
17ea068
Integration test that verifies that data channels open.
by Harald Alvestrand
· 5 years ago
04158be
Roll chromium_revision da78695105..cd7700164d (724157:724740)
by Yves Gerey
· 5 years ago
95059e0
Moved the legacy noise suppressor to a separate build target
by Per Åhgren
· 5 years ago
eae6896
Move vp8 rtp depacketization to VideoRtpDepacketizerVp8
by Danil Chapovalov
· 5 years ago
41466b7
Revert "Extracts ssrc based feedback tracking from feedback adapter."
by Sebastian Jansson
· 5 years ago
9d06bc2
Replace sequence checker with lock in IvfFrameGemerator.
by Artem Titov
· 5 years ago
b57fe17
Migrate video tests and tool to VideoRtpDepacketizer interface
by Danil Chapovalov
· 5 years ago
c9e532a
Fix PacketBuffer::LastReceivedKeyframePacketMs
by Danil Chapovalov
· 5 years ago
5e9cac9
Don't try to resend packets that were removed out of order.
by Sebastian Jansson
· 5 years ago
d77c829
Fix incorrect log message in FilterNetworks function.
by Sungwoo
· 5 years ago
9338bbc
Replace assert() with RTC_DCHECK
by Jerome Humbert
· 5 years ago
0808a8c
Explicitly set and use WEBRTC_USE_X11 instead of USE_X11.
by Patrik Höglund
· 5 years ago
f2d6fe6
Revert "Reland "Distinguish between send and receive video codecs""
by Johannes Kron
· 5 years ago
26e6afe
Reland "Distinguish between send and receive video codecs"
by Johannes Kron
· 5 years ago
977b265
Reduce some logging at INFO level by moving log statements
by Harald Alvestrand
· 5 years ago
08c46ad
Extracts ssrc based feedback tracking from feedback adapter.
by Sebastian Jansson
· 5 years ago
3a7e8b0
BalancedDegradationSettings: update codec specific settings.
by Åsa Persson
· 5 years ago
22619b3
Allow external initialization of libsrtp.
by Sebastian Jansson
· 5 years ago
6286bb0
Roll chromium_revision ac00cfc7cd..da78695105 (723527:724157)
by Yves Gerey
· 5 years ago
82f33c5
Delete transitional method EncodedImage.maybeRetain
by Niels Möller
· 5 years ago
3927298
Adds queue length setter to simulated network node builder.
by Sebastian Jansson
· 5 years ago
ce91126
Allows creating a test network node builder without manager.
by Sebastian Jansson
· 5 years ago
bcc1a76
Add rtc::Thread::PostDelayedTask
by Steve Anton
· 5 years ago
26fe811
Roll chromium_revision 5939567173..ac00cfc7cd (722888:723527)
by Yves Gerey
· 5 years ago
dcb4fcc
Execute cached video encoder switching request if encoder switching is allowed after the switch request was made.
by philipel
· 5 years ago
1e51a38
Makes padding prefer video SSRCs instead of audio.
by Erik Språng
· 5 years ago
184da52
Fix ASAN bug in iSAC codec
by Sam Zackrisson
· 5 years ago
75bc75c
Optimize FindNaluIndices
by Piasy
· 5 years ago
1154915
video_encoder: Add RTC_EXPORT ScalingSettings
by Hirokazu Honda
· 5 years ago
9a52bd7
STUN PING request
by Jonas Oreland
· 5 years ago
c907d4f
Revert "Ensure loss-based controller is always enabled."
by Jonas Olsson
· 5 years ago
4fc52c8
Make struct SynchronizationDelays more general.
by Åsa Persson
· 5 years ago
3fdb3cb
Remove potential real-time reallocation in PushResampler
by Per Åhgren
· 5 years ago
375eff4
Add guidance to style guide how to reference a bug in a TODO
by Danil Chapovalov
· 5 years ago
3c4fda2
Do not disable metrics by default.
by Mirko Bonadei
· 5 years ago
947a380
Split unit tests out of end-to-end PeerConnection test.
by Sami Kalliomäki
· 5 years ago
a3ecb7a
Migrate tests from RtpDepacketizer to VideoRtpDepacketizer interface
by Danil Chapovalov
· 5 years ago
007915a
Refresh some links in the docs folder.
by Mirko Bonadei
· 5 years ago
b04b2a1
Initial version of ResourceAdaptationProcessor and friends.
by Henrik Boström
· 5 years ago
f18f920
Revert "Moves TransportFeedbackAdapter to TaskQueue."
by Mirko Bonadei
· 5 years ago
89aaeda
Move audioproc_f to rtc_tools.
by Daniel Johansson
· 5 years ago
Next »