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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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1979384e40ca85572c1a3a5e94dd0bcd2787254d
1979384
Ensure that PC usage is recorded if a PC is alive for 60 seconds.
by Harald Alvestrand
· 6 years ago
bdee46d
Add functionality to set min bitrate for single stream through RtpEncodingParameters.
by Åsa Persson
· 6 years ago
ff3dd0c
Delete unused rtc::Stream subclasses.
by Niels Möller
· 6 years ago
ac5bbd9
Reland "Enable any address ports by default."
by Mirko Bonadei
· 6 years ago
87e4479
Roll chromium_revision 72ef4e4784..ecf8a6133e (569500:569618)
by Autoroller
· 6 years ago
0d4ee0a
Roll chromium_revision cb8b61b491..72ef4e4784 (569376:569500)
by Autoroller
· 6 years ago
5f5819f
Roll chromium_revision 105c043148..cb8b61b491 (569260:569376)
by Autoroller
· 6 years ago
23c5a99
Fix for VP9 K-SVC video freeze frame when send bandwidth is restricted.
by “Michael
· 6 years ago
43800f9
Generalize SimulcastEncoderAdapter, use for H264 & VP8.
by Sergio Garcia Murillo
· 6 years ago
b6b29e0
Convert video quality test from a TEST_F to a TEST fixture.
by Patrik Höglund
· 6 years ago
97899a0
Roll chromium_revision e1ef7d4b6b..105c043148 (568794:569260)
by Autoroller
· 6 years ago
45fc6df
Aligning time in audio jitter buffer plot to other plots in rtc event log visualizer.
by Minyue Li
· 6 years ago
1ec04f1
Reland "Reland "Injectable logging""
by Paulina Hensman
· 6 years ago
6f440ed
Revert "Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8."
by Mirko Bonadei
· 6 years ago
f341f3f
Add AGC1 fuzzer
by Sam Zackrisson
· 6 years ago
cb76c70
Revert "Roll chromium_revision e1ef7d4b6b..b47e7752c6 (568794:569173)"
by Artem Titov
· 6 years ago
0bc58cf
Replace rtc::Optional with absl::optional in api
by Danil Chapovalov
· 6 years ago
1ff41eb
Revert "NetEq: Deprecate playout modes Fax, Off and Streaming"
by Henrik Lundin
· 6 years ago
07efe43
Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.
by Sergio Garcia Murillo
· 6 years ago
80c4cca
NetEq: Deprecate playout modes Fax, Off and Streaming
by Henrik Lundin
· 6 years ago
c0260b4
Remove third party dependecies that are not more in the source code
by Artem Titov
· 6 years ago
c2a83ee
Remove usage of rtc_base/checks.h in 3pp base64.cc
by Artem Titov
· 6 years ago
c19ab07
Add support for content-hint value "text"
by Harald Alvestrand
· 6 years ago
0a1d189
Replace rtc::Optional with absl::optional in rtc_base
by Danil Chapovalov
· 6 years ago
c806c1d
Fixed crash when PCF is destroyed before MediaStream in ObjC
by Yura Yaroshevich
· 6 years ago
8b23dba
Add RTPVideoHeader const accessor.
by philipel
· 6 years ago
196100e
Replace rtc::Optional with absl::optional
by Danil Chapovalov
· 6 years ago
ae810c1
Create a peer connection factory builder
by Jiawei Ou
· 6 years ago
f1e3cb4
Add OnLogMessage(msg, sev, tag) to logsinks
by Paulina Hensman
· 6 years ago
db38972
Remove nonlinear beamformer API from APM
by Sam Zackrisson
· 6 years ago
7b55c73
Add RTPVideoHeader accessor.
by philipel
· 6 years ago
fd2457b
Roll chromium_revision e1ef7d4b6b..b47e7752c6 (568794:569173)
by Autoroller
· 6 years ago
5b8dd4d
Fix a dangling-pointer bug in P2PTransportChannel unit tests.
by Qingsi Wang
· 6 years ago
b983bae
Remove unused/deprecated DTMF methods
by Steve Anton
· 6 years ago
3d8b171
Removing some TSan suppressions around Thread class.
by Taylor Brandstetter
· 6 years ago
db6af36
Add RNN-VAD to AGC2.
by Alex Loiko
· 6 years ago
87a9353
Add format check to `git cl presubmit`
by Yves Gerey
· 6 years ago
beb2d98
Removing usage of //build/config/compiler:no_size_t_to_int_warning.
by Mirko Bonadei
· 6 years ago
de212ca
Removing some MSVC warning suppression flags.
by Mirko Bonadei
· 6 years ago
b23db02
Add mbonadei@ to build configs OWNERS.
by Mirko Bonadei
· 6 years ago
bc5c934
Adding mbonadei@ to build_files WATCHLIST.
by Mirko Bonadei
· 6 years ago
a97c931
Fix a bug where TestAudioDeviceModule crashes if destroyed uninitialized.
by Sami Kalliomäki
· 6 years ago
aaa483b
Revert "Remove deprecated mac capture code."
by Kári Helgason
· 6 years ago
caca556
Roll chromium_revision 20579735a6..e1ef7d4b6b (568689:568794)
by Autoroller
· 6 years ago
88703e8
Let git-hyper-blame ignore format commit.
by Yves Gerey
· 6 years ago
056a68d
Revert "Enable any address ports by default."
by Mirko Bonadei
· 6 years ago
80c0f06
Init GainControlImpl with correct lock.
by Alex Loiko
· 6 years ago
5565981
Add functionality to set min/max bitrate per simulcast layer through RtpEncodingParameters.
by Åsa Persson
· 6 years ago
f88a22c
Delete pre_decode_callback.
by Niels Möller
· 6 years ago
5297bd2
Fixed crash when PCF is destroyed before PC in ObjC
by Yura Yaroshevich
· 6 years ago
f04148c
Enable any address ports by default.
by Qingsi Wang
· 6 years ago
188301c
Roll chromium_revision 6e14efc13e..20579735a6 (568572:568689)
by Autoroller
· 6 years ago
6109d03
Mark unused/deprecated DTMF methods for removal
by Steve Anton
· 6 years ago
1d4a76d
Fixing flakiness in PeerConnectionIntegrationTest.
by Seth Hampson
· 6 years ago
66cadcc
Replace rtc::Optional with absl::optional in pc
by Danil Chapovalov
· 6 years ago
751a817
Roll chromium_revision c27ef6f9f6..6e14efc13e (568443:568572)
by Autoroller
· 6 years ago
a465344
Return SSRC stats with the old stats API when SSRCs are unsignaled.
by Taylor Brandstetter
· 6 years ago
0a5fe77
Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.
by philipel
· 6 years ago
7e9a619
Add setter method EncodedFrame::SetTimestamp.
by Niels Möller
· 6 years ago
acef18d
Roll chromium_revision 9d565db4c0..c27ef6f9f6 (568343:568443)
by Autoroller
· 6 years ago
fa2b8d7
Add separate native library for instrumentationtests
by Paulina Hensman
· 6 years ago
faf2827
Add Parsing/Building generic frame descriptor extension
by Danil Chapovalov
· 6 years ago
709c822
Add nisse@ as owner of api/video/
by Niels Möller
· 6 years ago
665174f
Reformat the WebRTC code base
by Yves Gerey
· 6 years ago
b602123
Replace rtc::Optional with absl::optional in modules/audio_coding
by Danil Chapovalov
· 6 years ago
bbfcc70
AEC3: Unittests for MovingAverage
by Gustaf Ullberg
· 6 years ago
1998e9d
Drop tools/gyp from dependencies
by Niels Möller
· 6 years ago
f344dbb
Cover AecDump calls in APM fuzzer.
by Alex Loiko
· 6 years ago
8406c43
AEC3: Average the spectrum of multiple nearend frames in the suppressor.
by Gustaf Ullberg
· 6 years ago
5d848f3
Delete picture id and tl0 index from CodecSpecificInfo.
by Niels Möller
· 6 years ago
db9f7ab
Replace rtc::Optional with absl::optional in modules/audio processing
by Danil Chapovalov
· 6 years ago
c66613d
Android: Simlify createOesTextureBuffer() in VideoFrameBufferTest
by Magnus Jedvert
· 6 years ago
8a5edb2
Always enable 'delay-agnostic' in APM fuzzer.
by Alex Loiko
· 6 years ago
790da37
Fuzz AEC field trial killswitches
by Sam Zackrisson
· 6 years ago
af998e2
Remove non-API beamformer references
by Sam Zackrisson
· 6 years ago
024eeff
Roll chromium_revision 9df92afb16..9d565db4c0 (566630:568343)
by Autoroller
· 6 years ago
aac7dee
[desktopCapture Mac]reorder execution order in start/release processing
by braveyao
· 6 years ago
15ac521
Removing unused cricket::Port constructor.
by Taylor Brandstetter
· 6 years ago
6bbeb08
Extract rtc_base/base64.h and rtc_base/base64.cc into separate target.
by Artem Titov
· 6 years ago
6250fdd
Delete FakeWebRtcVcmFactory::OnDestroyed method.
by Niels Möller
· 6 years ago
9394f6f
Stop using the beamformer inside APM
by Sam Zackrisson
· 6 years ago
431abd9
Replace rtc::Optional with absl::optional in test and rtc_tools
by Danil Chapovalov
· 6 years ago
9bf3158
Pass buffer with size when writing rtp header extension
by Danil Chapovalov
· 6 years ago
0040b66
Replace rtc::Optional with absl::optional
by Danil Chapovalov
· 6 years ago
ae18886
Disable new RTC_CHECK unittest
by Jonas Olsson
· 6 years ago
00c7183
Replace rtc::Optional with absl::optional in media, ortc, p2p
by Danil Chapovalov
· 6 years ago
5a9ba68
Add base64 webrtc owned third_party dep
by Artem Titov
· 6 years ago
5adf07d
Make instructions for checkin_chrome_dep a bit clearer.
by Patrik Höglund
· 6 years ago
ce4829a
Adds trial to ignore video pacing for audio packets.
by Sebastian Jansson
· 6 years ago
f8e5c11
Refactor checks to use a copy of the new logging backend.
by Jonas Olsson
· 6 years ago
6a9bd74
Fix a downstream test failure.
by Ying Wang
· 6 years ago
c235a8d
Adds trial to always send padding packets when not sending video.
by Sebastian Jansson
· 6 years ago
fc50110
Remove stringstreams from modules/video_coding/
by Jonas Olsson
· 6 years ago
5c43150
Makes BBR congestion window more similar to QUIC.
by Sebastian Jansson
· 6 years ago
fb4d66b
Improves buffer time calculation in network control tester.
by Sebastian Jansson
· 6 years ago
b9b146c
Replace rtc::Optional with absl::optional in audio, call and video
by Danil Chapovalov
· 6 years ago
e61d72b
Disables congestion window in pacer when CongestionWindowPushback is enabled.
by Ying Wang
· 6 years ago
92b24f0
Delete an unneeded include of pathutils.h.
by Niels Möller
· 6 years ago
fc9dcb6
Remove wire-up for cancelled experement on VAAPI VP8 encoding
by Ilya Nikolaevskiy
· 6 years ago
d264df5
Replace rtc::Optional with absl::optional in modules/rtp_rtcp
by Danil Chapovalov
· 6 years ago
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