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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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4f6ef1894e7cae0102ac295cb9eab2c86d6cb1cb
4f6ef18
Added underscore to dtls_transport_unittest.cc.
by Benjamin Wright
· 5 years ago
2c79648
Remove rtc::TimeMillis() call from ALR detector.
by Sebastian Jansson
· 5 years ago
7307824
Rolling third_party/winsdk_samples_v71.
by Mirko Bonadei
· 5 years ago
493a650
Propagate base minimum delay from video jitter buffer to webrtc/api.
by Ruslan Burakov
· 5 years ago
48e7065
Remove default IDs for RTP extensions from rtp_parameters.h
by Elad Alon
· 6 years ago
1a7a4af
Fix encoded image data injectors.
by Artem Titov
· 5 years ago
aec663e
Fix video_loopback tool with different TL numbers in simulcast streams
by Ilya Nikolaevskiy
· 5 years ago
28221de
Fix more -Wextra-semi.
by Mirko Bonadei
· 5 years ago
5cceaa1
Remove iOS 9 support from mb config
by Artem Titarenko
· 5 years ago
db42ed2
Add RELATIVE_ARRIVAL_DELAY histogram mode to DelayManager.
by Jakob Ivarsson
· 5 years ago
d00405f
Drop support for link-time injection of the rtc::TaskQueue::Impl
by Danil Chapovalov
· 5 years ago
dda5fdc
Fix vp8 simulcast screenshare and perf tests for it
by Ilya Nikolaevskiy
· 5 years ago
08f6a6c
Import proto_library.gni when rtc_enable_protobuf is true
by Kimmo Kinnunen
· 5 years ago
e12a1c7
Adding GetStats APIs for senders/receivers.
by Peter Hanspers
· 5 years ago
7b3f4a2
Remove unused |keyframe_interval| from codec tests.
by Rasmus Brandt
· 5 years ago
f54e30b
Add const to variables in openssl_stream_adapter.cc that can use it.
by Benjamin Wright
· 5 years ago
619b294
RtpSender's RtpParameters were invalidated in a call to SLD/SRD.
by Amit Hilbuch
· 5 years ago
d6f61dd
Add ::Connect method to the media transport interface
by Piotr (Peter) Slatala
· 5 years ago
6a7baa7
Remove VCMEncodedFrameCallback and VCMGenericEncoder
by Erik Språng
· 5 years ago
c9d0b08
Respects min ALR probing interval.
by Sebastian Jansson
· 5 years ago
1a16da1
Remove deprecated CreateMediaTransport method
by Piotr (Peter) Slatala
· 5 years ago
39b69cc
Add field trial for adding remote ufrag CreatePermission
by Jonas Oreland
· 5 years ago
546ee61
clang-tidy helper script, with clang static analyzer included.
by Yves Gerey
· 5 years ago
b7cb7b5
Remove VCMEncoderDataBase and put remaining code into VideoStreamEncoder
by Erik Språng
· 5 years ago
695af94
Add reentrancy comment for critical section.
by Ruslan Burakov
· 5 years ago
fee13e8
Log pacer values to verbose log
by Evan Shrubsole
· 5 years ago
12ae4f4
Introduce possibility to poll stats and notify analyzers.
by Mirko Bonadei
· 5 years ago
2684ab3
Test default TaskQueue implementation via TaskQueueBase interface
by Danil Chapovalov
· 5 years ago
22dab11
Remove Legacy ADM from AppRTC mobile
by Paulina Hensman
· 5 years ago
0bf4c29
Add support of auto IP generation in network emulation manager.
by Artem Titov
· 5 years ago
9595d1b
Roll chromium_revision 15651144f3..ec3bf6e607 (635345:635450)
by chromium-webrtc-autoroll
· 5 years ago
e2da931
Remove a leftover audio codec poison immutinty declaration
by Karl Wiberg
· 5 years ago
f2889bb
Add option to inject YuvConverter to SurfaceTextureHelper.
by Åsa Persson
· 5 years ago
b4f0393
Roll chromium_revision a55c7bb989..15651144f3 (635189:635345)
by chromium-webrtc-autoroll
· 5 years ago
bd0deca
Ban absl::StrSplit and absl::StrJoin
by Karl Wiberg
· 5 years ago
7572bb4
Fix -Wextra-semi warnings in webrtc fuzzers.
by Nico Weber
· 5 years ago
c35a72c
Roll chromium_revision 81fda909f3..a55c7bb989 (635067:635189)
by chromium-webrtc-autoroll
· 5 years ago
b000b71
Wiring up RIDs from the video engine to the RTP Sender.
by Amit Hilbuch
· 5 years ago
98335f8
Fixing webrtc::IceTransportState.
by Seth Hampson
· 5 years ago
5cbc528
Revert "Remove VCMEncoderDataBase and put remaining code into VideoStreamEncoder"
by Sami Kalliomäki
· 5 years ago
7d6a4c0
Connect LossNotificationController to RtpRtcp
by Elad Alon
· 5 years ago
a497d12
Avoids PostTask to repost a repeated task.
by Sebastian Jansson
· 6 years ago
ce7a4fb
Adding possibility to save an RTCEventLog of the call.
by Mirko Bonadei
· 5 years ago
99f5d5f
Roll chromium_revision 95a23eca14..81fda909f3 (634895:635067)
by chromium-webrtc-autoroll
· 5 years ago
d37307c
Reland "Adds resource path support for video files in scenario tests."
by Sebastian Jansson
· 5 years ago
715c476
Remove VCMEncoderDataBase and put remaining code into VideoStreamEncoder
by Erik Språng
· 5 years ago
2b08e31
Adds CoDel implementation to network simulation.
by Sebastian Jansson
· 5 years ago
418dd0b
Stop using special RTT value for DelayBasedBwe.
by Sebastian Jansson
· 5 years ago
76a74e6
Delay bug during audio receiver stream recreation.
by Ruslan Burakov
· 5 years ago
c4dd730
Fix -Wextra-semi warnings.
by Mirko Bonadei
· 5 years ago
3812fa9
Delete VideoCodecTestParameterized.
by Rasmus Brandt
· 5 years ago
19d0104
Make RtpRtcp::Configuration::field_trials ptr const
by Per Kjellander
· 5 years ago
a9cfa47
Revert "Delete rtc_task_queue_impl build target"
by Mirko Bonadei
· 5 years ago
74f0a51
Move kFeedbackMessageType from Remb to Psfb
by Elad Alon
· 5 years ago
56973e6
Delete rtc_task_queue_impl build target
by Danil Chapovalov
· 5 years ago
8721bb3
Roll chromium_revision e7ecd1bfc2..95a23eca14 (634731:634895)
by chromium-webrtc-autoroll
· 5 years ago
e1e789b
Removing non-const RtpSenderInterface::GetParameters().
by Amit Hilbuch
· 6 years ago
f58e43e
Add an OpenChannel method to MediaTransportInterface and call it whenever PeerConnection opens a new data channel.
by Bjorn Mellem
· 5 years ago
8f096d0
Map clat devices to cellular on Android
by Jeroen de Borst
· 5 years ago
e19a6da
Roll chromium_revision a77f654a3c..e7ecd1bfc2 (634608:634731)
by chromium-webrtc-autoroll
· 5 years ago
487c09b
Adds FakeNetworkPipeTest to rtc_unittests.
by Sebastian Jansson
· 5 years ago
29f9cd9
Synchronize replaceRegion calls.
by Anders Carlsson
· 5 years ago
7ef34f8
Replace field trials with WebRtcKeyValueConfig in PacedSender
by Per Kjellander
· 5 years ago
ce8e867
Add support for TransportSequenceNumberV2 in SDP negotiation
by Johannes Kron
· 5 years ago
14f96d1
Roll chromium_revision f39a1b8992..a77f654a3c (634190:634608)
by chromium-webrtc-autoroll
· 5 years ago
8aa00f0
Add missing absl/memory/memory.h to rtc_event_generic_ack_received.cc
by tzik
· 6 years ago
b4643ad
Rename "OnReceivedFrame" to "OnAssembledFrame"
by Elad Alon
· 5 years ago
d7329ca
Remove VideoSender and fold code into VideoStreamEncoder
by Erik Språng
· 5 years ago
10874b2
Create LossNotificationController
by Elad Alon
· 5 years ago
b75d9e9
Allow IceConnectionState to become failed without ever connecting.
by Jonas Olsson
· 5 years ago
d209cd1
Lower SSIM thresholds.
by Sergey Silkin
· 5 years ago
6543881
2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883
by Alex Loiko
· 5 years ago
caa499b
PFFFT C++ wrapper for APM
by Alessio Bazzica
· 5 years ago
45af00f
Revert "Adds resource path support for video files in scenario tests."
by Sergey Silkin
· 5 years ago
4ae6347
Use `final` so that the compiler will be able to inline calls
by Karl Wiberg
· 5 years ago
5966c50
Add thread safety annotations for PeerConnection::configuration_
by Karl Wiberg
· 5 years ago
8306a73
Adds resource path support for video files in scenario tests.
by Sebastian Jansson
· 5 years ago
96fccfe
Make sure RTC_SUPPORTS_METAL is set in AppRTCMobile.
by Anders Carlsson
· 5 years ago
735f823
CreateAudioProcessor: do not propagate an unset echo control factory to the AudioProcessing instance
by Jesús de Vicente Peña
· 5 years ago
bed8604
Adding entry point for the v2 stats API.
by Peter Hanspers
· 5 years ago
2645193
DtlsTransport::ice_transport is const and can be called off thread
by Harald Alvestrand
· 5 years ago
ee95f3e
Roll chromium_revision 94ca2b10d8..f39a1b8992 (634089:634190)
by chromium-webrtc-autoroll
· 5 years ago
54047be
Reland "Extend TransportSequenceNumber RTP header extension"
by Johannes Kron
· 5 years ago
1eb3d7e
Refactor DelayManager into separate Histogram class and make it injectable for testing purposes.
by Jakob Ivarsson
· 5 years ago
fa52efa
Migrate stdlib task queue to TaskQueueBase interface
by Danil Chapovalov
· 5 years ago
e11b7d2
Replace field trials with WebRtcKeyValueConfig in RtpRtcpModule
by Per Kjellander
· 5 years ago
aa1a43e
AEC3: Use minimum ERLE during onsets
by Gustaf Ullberg
· 5 years ago
d6c6f16
Update RTP packet and header fuzzers to support additional extensions
by Johannes Kron
· 5 years ago
3256225
"Remove" loophole in rtc::Thread::ScopedDisallowBlockingCalls
by Karl Wiberg
· 5 years ago
826f2e7
Migrate win task queue to TaskQueueBase interface
by Danil Chapovalov
· 6 years ago
bb05369
Delete unused class FakeCandidatePair.
by Niels Möller
· 6 years ago
00c57e3
Delete unused class RtpTransportInternalAdapter
by Niels Möller
· 6 years ago
17c147c
Feed PacedSender with RTP packet size
by Per Kjellander
· 6 years ago
252725d
Rename RtpPacketHistory::PacketState::payload_size -> packet_size
by Per Kjellander
· 6 years ago
1b801e0
Roll chromium_revision 55c441e653..94ca2b10d8 (633987:634089)
by chromium-webrtc-autoroll
· 5 years ago
feef8f5
Roll chromium_revision 919d2e8241..55c441e653 (633811:633987)
by chromium-webrtc-autoroll
· 5 years ago
32232e9
Add spatial layers support to video analyze pipeline.
by Artem Titov
· 6 years ago
8e68920
Roll chromium_revision 554be8c5f4..919d2e8241 (633687:633811)
by chromium-webrtc-autoroll
· 6 years ago
47cf5ea
Migrate gcd task queue implementation to TaskQueueBase interface
by Danil Chapovalov
· 6 years ago
f5d8808
Remove Analyzers struct.
by Mirko Bonadei
· 6 years ago
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