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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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d059f2c4467042ef2fbff7affef6a1f45dcb5315
d059f2c
Add steveanton@ as api/ and ortc/ OWNER
by Steve Anton
· 6 years ago
d1003d7
A new PeerConnection level perf test.
by Seth Hampson
· 6 years ago
42f0d78
Roll back checking in the third_party directory
by Artem Titov
· 6 years ago
67c8bcf
Revert two instances of num_active_spatial_layers.
by “Michael
· 6 years ago
bcf9180
Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.
by Alex Narest
· 6 years ago
81f5197
Fix pylint presubmit errors and warnings from untouched modules.
by Artem Titov
· 6 years ago
d971109
Extract fft to separate target to be able to move it to third_party
by Artem Titov
· 6 years ago
2c74d85
Adds enum field trial parser.
by Sebastian Jansson
· 6 years ago
b3f5aed
Remove the flag PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS.
by Qingsi Wang
· 6 years ago
d6eb71e
Use the sparse histogram in RTC_HISTOGRAM_ENUMERATION_SPARSE.
by Qingsi Wang
· 6 years ago
901e0ff
Add bit depth information to PlanarYuvBuffer
by Emircan Uysaler
· 6 years ago
0756373
[Unified Plan] Avoid offering two senders with the same ID
by Steve Anton
· 6 years ago
1bc9716
[Unified Plan] Do not initialize recvonly transceivers with any send streams
by Steve Anton
· 6 years ago
e58bd8a
AEC3: Reverb modeling: Including the freq shape of the tails when modeling the reverberation
by Jesús de Vicente Peña
· 6 years ago
fb8e7ef
Implement PayloadUnion as variant instead of pair of optionals
by Danil Chapovalov
· 6 years ago
72f52a1
Delete unused copy constructors for VCMEncodedFrame and VCMFrameBuffer.
by Niels Möller
· 6 years ago
fe288eb
Don't call deprecated FFmpeg API.
by Sergey Silkin
· 6 years ago
df3bcdb
Extract fft4g into separate build target
by Artem Titov
· 6 years ago
58cd385
Fix potential division by zero in VP9 VideoCodecTest.
by Rasmus Brandt
· 6 years ago
63c82a7
Style fixes in event log unittest.
by Bjorn Terelius
· 6 years ago
52f53d5
Revert "Add Timestamp accessor methods to the EncodedImage class."
by Björn Terelius
· 6 years ago
c9ac93f
Adding NetEq lifetime stats to event log visualizer.
by Minyue Li
· 6 years ago
762289e
Fix overflow in digital AGC1
by Sam Zackrisson
· 6 years ago
f4db542
Rewrite the RtcEventLog unit test.
by Bjorn Terelius
· 6 years ago
712678b
Delete unused class TransformAdapter.
by Niels Möller
· 6 years ago
f34d467
Add Timestamp accessor methods to the EncodedImage class.
by Niels Möller
· 6 years ago
f7789c6
Limiting increment in timestamps with neteq simulation.
by Minyue Li
· 6 years ago
44b98f9
Re-introduce a read of a bool in APM fuzzers
by Sam Zackrisson
· 6 years ago
8491693
Update packetsLost and jitter stats any time a packet is received.
by Taylor Brandstetter
· 6 years ago
111fdfd
Refactor RtpSender to take the sender ID as a constructor argument
by Steve Anton
· 6 years ago
d5b8ee1
Re-enable PeerConnectionEndToEndTest.Call on TSan.
by Mirko Bonadei
· 6 years ago
ae82ffa
Delete unused methods for replacing the "default filesystem".
by Niels Möller
· 6 years ago
8396e34
Remove APM limiter in Audio Mixer.
by Alex Loiko
· 6 years ago
01d2a67
Adding jitter buffer plots for all SSRCs in event log visualizer.
by Minyue Li
· 6 years ago
91280e4
Extract third party part of g722 codec into separate target
by Artem Titov
· 6 years ago
3ecec17
Extract third party part of g711 codec into separate target
by Artem Titov
· 6 years ago
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
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