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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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7127f342a90a095f678d0c007f4859c1ae67f05b
7127f34
Signal Network route change in fake ice.
by Piotr (Peter) Slatala
· 6 years ago
d95b0a2
Use delta-encoding in new WebRTC event logs
by Elad Alon
· 6 years ago
7246720
Clean up root OWNERS.
by Patrik Höglund
· 6 years ago
e598e6b
Run robolectric tests for Android on several Android API versions
by Artem Titarenko
· 6 years ago
9973fa8
Pass HdrMetadata between VideoFrame and EncodedImage for VP9
by Johannes Kron
· 6 years ago
6c373cc
Add support for audio in latency visualization.
by Bjorn Terelius
· 6 years ago
d8aa9f9
Fix flaky JsepTransportControllerTests.
by Jonas Olsson
· 6 years ago
ad1d9f0
Add RTP header extension for HDR metadata
by Johannes Kron
· 6 years ago
ee45f90
In RTP to NTP estimator do not allow huge jumps in NTP timestamps
by Ilya Nikolaevskiy
· 6 years ago
06f6bc9
Reintroduce missing dependencies in libwebrtc.a library.
by Yves Gerey
· 6 years ago
175aa2e
Implement data channels over media transport.
by Bjorn Mellem
· 6 years ago
c2ebe21
Reland "Use the factory instead of using the builtin code path in `VideoCodecInitializer`"
by Jiawei Ou
· 6 years ago
0393c64
[Win/boringSSL] Add nasm as part of required dependencies.
by Yves Gerey
· 6 years ago
ada077f
Callback changes to media transport interface:
by Piotr (Peter) Slatala
· 6 years ago
87e1619
Add owners for media_transport_interface
by Piotr (Peter) Slatala
· 6 years ago
d3438aa
Add ability to specify if rate controller of video encoder is trusted.
by Erik Språng
· 6 years ago
6528d8a
In Android encoders, cache EncoderInfo in InitEncode.
by Erik Språng
· 6 years ago
260770c
Delete rtc::Filesystem. Move needed functions to filerotatingstream.cc.
by Niels Möller
· 6 years ago
b0550bd
Eliminate use of EventWrapper from mac audio device
by Niels Möller
· 6 years ago
c94b22e
Add magjed/nisse/sprang/brandtr as api/video_codecs owners
by Erik Språng
· 6 years ago
c5dd300
Introduce RtpPacket::GetExtension accessor that return result
by Danil Chapovalov
· 6 years ago
357f596
Split a separate codecs target off of :video_jni
by Jonathan Yu
· 6 years ago
5bb1ed6
Eliminate use of EventWrapper from ios audio device tests
by Niels Möller
· 6 years ago
a33c7af
Tolerate optional chunks in WAV files
by Alessio Bazzica
· 6 years ago
c496d58
Add flag for fast jitter buffer playout in neteq simulation
by Sam Zackrisson
· 6 years ago
e6c2c08
MsanUninitialized: restric type check to msan case.
by Alessio Bazzica
· 6 years ago
c4e9825
Delete classes EventFactory and EventFactoryImpl.
by Niels Möller
· 6 years ago
2a74263
Make the bitrate_allocator param optional to prepare for its removal
by Oleh Prypin
· 6 years ago
cd2e105
Reenable test RampUpTest.AudioTransportSequenceNumber
by Niels Möller
· 6 years ago
694ed17
Add a style rule about not using const optional<T>& arguments
by Karl Wiberg
· 6 years ago
f0e7440
Add missing conditional defines to neteq test and tools targets
by Sam Zackrisson
· 6 years ago
689983f
Deprecate EventFactory and delete all usage.
by Niels Möller
· 6 years ago
54b4924
Update H264 encoder to use GetEncoderInfo
by Erik Språng
· 6 years ago
1060870
Update LibVpxVp8Encoder to use GetEncoderInfo
by Erik Språng
· 6 years ago
727d164
Update VP9 encoder to use GetEncoderInfo
by Erik Språng
· 6 years ago
5473a45
Remove multiple RTX codec entries in GetRtpReceiver/SenderCapabilities
by Florent Castelli
· 6 years ago
75de46a
Update SimulcastEncoderAdapter merging of EncoderInfo
by Erik Språng
· 6 years ago
e6a2d94
Clear FrameBuffer if there were no frames received for 10 minutes
by Ilya Nikolaevskiy
· 6 years ago
b768e88
Reland "Isolating APM API build target: making :api an actual target."
by Alessio Bazzica
· 6 years ago
bdc6c40
Add field trial for target bitrate RTCP XR message.
by Rasmus Brandt
· 6 years ago
d565918
Delete NullEventFactory
by Niels Möller
· 6 years ago
e769ed9
Roll chromium_revision 38dcb5ed01..db720b4ab9 (605924:606025)
by chromium-webrtc-autoroll
· 6 years ago
50f60cb
Rename software codec classes and move them into api/
by Jonathan Yu
· 6 years ago
ff7020a
Remove non-default VideoEncoder::EncoderInfo() ctor
by Erik Språng
· 6 years ago
36d907b
Update MockVideoEncoder with correct methods.
by Erik Språng
· 6 years ago
61c6e56
Revert "Isolating APM API build target: making :api an actual target."
by Alessio Bazzica
· 6 years ago
a7f77a7
Isolating APM API build target: making :api an actual target.
by Alessio Bazzica
· 6 years ago
7553c02
Update ObjCVideoEncoder to use GetEncoderInfo()
by Erik Språng
· 6 years ago
7b3c76b
Reland "Delete rtc::Pathname"
by Niels Möller
· 6 years ago
17fc7e2
Add counter to the end of FakeEncoder frames in order to make them unique.
by Per Kjellander
· 6 years ago
c572ff3
Add default constructor for rtc::Event
by Niels Möller
· 6 years ago
3ea7b83
Resolve the race condition between mDNS name registration and
by Qingsi Wang
· 6 years ago
8770ce7
Roll chromium_revision 03cf97f6d8..38dcb5ed01 (605818:605924)
by chromium-webrtc-autoroll
· 6 years ago
bb091db
Roll chromium_revision 793c8566ab..03cf97f6d8 (605715:605818)
by chromium-webrtc-autoroll
· 6 years ago
2cd3b4c
Fixing bug in SimulatedNetwork where packets stop.
by Sebastian Jansson
· 6 years ago
0f54f21
Removes deprecated GetSentPacket from PacketResult.
by Sebastian Jansson
· 6 years ago
dc98b9b
AEC3: Corrected include
by Per Åhgren
· 6 years ago
c564a7b
Roll chromium_revision 7841106b37..793c8566ab (605607:605715)
by chromium-webrtc-autoroll
· 6 years ago
8ffd710
Update Android encoder to use GetEncoderInfo()
by Erik Språng
· 6 years ago
020e583
AEC3: Compensate comfort noise level for loss due to filter bank
by Gustaf Ullberg
· 6 years ago
83b00f0
AEC3: Computation of comfort noise gains from suppression gains
by Gustaf Ullberg
· 6 years ago
34fc346
Add support for computing iOS code coverage
by Artem Titarenko
· 6 years ago
277b6ea
Isolating APM API build target: adding dummy :api target.
by Alessio Bazzica
· 6 years ago
3ddaf3c
Revert "Add support for screen sharing with PipeWire on Wayland"
by Patrik Höglund
· 6 years ago
82c07ea
Tune huge video frames detection threshold for GetStats googHugeFramesSent stat
by Ilya Nikolaevskiy
· 6 years ago
4f3cc6e
Make VideoSendStreamTest.NoPaddingWhenVideoIsMuted less flaky
by Erik Språng
· 6 years ago
a8f5461
nit: Use make_unique in rtp_video_stream_receiver.cc
by Elad Alon
· 6 years ago
27f3172
Simplify use of events in TestAudioDevice
by Niels Möller
· 6 years ago
361dbc1
Android: Add option to set presentation timestamp in EglRenderer
by Magnus Jedvert
· 6 years ago
967f7d5
Add audio level to CSRC class
by Jonas Oreland
· 6 years ago
df351f4
Update FakeEncoder to use EncoderInfo
by Erik Språng
· 6 years ago
254d3db
Add missing #include to absl/memory/memory.h from audio_encoder_cng.cc
by tzik
· 6 years ago
fbf1683
Add HdrMetadata to VideoFrame
by Johannes Kron
· 6 years ago
4f0f3d5
Remove unused member variable - RTCPSender::using_nack_
by Elad Alon
· 6 years ago
63ada78
Remove outdated TODO
by Sam Zackrisson
· 6 years ago
3ea1878
Add severity into RTC logging callbacks
by Jiawei Ou
· 6 years ago
edfb883
Roll chromium_revision 11d7305a72..7841106b37 (605505:605607)
by chromium-webrtc-autoroll
· 6 years ago
d7db17b
Roll chromium_revision bf7ad46dee..11d7305a72 (605401:605505)
by chromium-webrtc-autoroll
· 6 years ago
a9bbd86
Add a configuration parameter for using the media transport for data channels.
by Bjorn Mellem
· 6 years ago
41b5296
Roll chromium_revision c26ff44a53..bf7ad46dee (605286:605401)
by chromium-webrtc-autoroll
· 6 years ago
ee49f70
Remove VideoEncoder::SetChannelParameters.
by philipel
· 6 years ago
c22f551
Remove locks from AECM and move it into private_submodules_
by Sam Zackrisson
· 6 years ago
e693381
Delete struct rtc::PacketTime.
by Niels Möller
· 6 years ago
0070655
Removing ancient and unused test scripts and data files
by Henrik Lundin
· 6 years ago
fd1a2fb
Set RtpRtcp config receive_only in voe::ChannelReceive
by Niels Möller
· 6 years ago
aed3070
Replace GetScalingSettings() with GetEnocderInfo() in TestEncoder
by Erik Språng
· 6 years ago
f418bcb
Refactor RtpSender to use absl::string_view for payload name.
by Niels Möller
· 6 years ago
2634199
Move MovingAverage to rtc_base/numerics and update it.
by Ilya Nikolaevskiy
· 6 years ago
a1ead6f
Update EncoderProxy to use EncoderInfo
by Erik Språng
· 6 years ago
bf0d0c1
Add IPv6 configuration parameters to iOS API
by Uladzislau Susha
· 6 years ago
842a2a8
Roll chromium_revision 4e7c87b55c..c26ff44a53 (605184:605286)
by chromium-webrtc-autoroll
· 6 years ago
e7547d5
Move MemoryStream to separate source files, and to a test target.
by Niels Möller
· 6 years ago
9f878f6
Roll chromium_revision b58a03341b..4e7c87b55c (605082:605184)
by chromium-webrtc-autoroll
· 6 years ago
671341a
Roll chromium_revision 35f882550d..b58a03341b (604980:605082)
by chromium-webrtc-autoroll
· 6 years ago
1bc0b9d
Roll chromium_revision e842ab5f98..35f882550d (604874:604980)
by chromium-webrtc-autoroll
· 6 years ago
2039ee7
Revert "Delete rtc::Pathname"
by Qingsi Wang
· 6 years ago
273d029
Implement data channel methods in LoopbackMediaTransport.
by Bjorn Mellem
· 6 years ago
0367d1a
Adds a field trial parameter to configure waiting time before sending Nack packets.
by Ying Wang
· 6 years ago
e401863
Change to RtcEvent::Copy
by Elad Alon
· 6 years ago
2365936
Hide the AudioEncoderCng class behind a create function
by Karl Wiberg
· 6 years ago
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