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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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e2da931307480d4675750dc08ad4baf9fc47e640
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
22f9925
webrtc: Remove semicolons.
by Nico Weber
· 6 years ago
af623ae
Delete unused file mock_video_codec_interface.h
by Niels Möller
· 6 years ago
d36a815
Remove the deprecated CreateProbeClusters method
by Piotr (Peter) Slatala
· 6 years ago
8b3db59
Revert "Reland of https://webrtc-review.googlesource.com/c/src/+/114883"
by Alex Loiko
· 6 years ago
01fe309
Do not use RtcEventLogs in media transport when used only for data channel.
by Piotr (Peter) Slatala
· 6 years ago
ce27875
[AndroidAudioRecord] Added audio format parameter to configure AudioRecord - JavaAudioDeviceModule
by Alvaro Martinez
· 6 years ago
5341aac
Reland of https://webrtc-review.googlesource.com/c/src/+/114883
by Alex Loiko
· 6 years ago
d5e02f0
Delete redundant members from VCMPacket.
by Niels Möller
· 6 years ago
4d2367a
Removes broken frame matching code in scenario quality stats.
by Sebastian Jansson
· 6 years ago
b35bacc
Fix NetEq minimum and maximum delay always reset on acm creation.
by Ruslan Burakov
· 6 years ago
8073db6
Roll chromium_revision 4b3282a5d6..554be8c5f4 (633587:633687)
by chromium-webrtc-autoroll
· 6 years ago
76d7ce2
Disabling flaky RecievesVp8SimulcastFrames test.
by Sebastian Jansson
· 6 years ago
dd1cc98
Reland "Update VP9EncoderImpl to use EncodedImage::Allocate"
by Niels Möller
· 6 years ago
109b5fb
Revert "Extend TransportSequenceNumber RTP header extension"
by Mirko Bonadei
· 6 years ago
28c7362
Extend TransportSequenceNumber RTP header extension
by Johannes Kron
· 6 years ago
3f6bf3a
Clarify that pacing rate is based on raw target rate
by Evan Shrubsole
· 6 years ago
5fbebd5
Adds support for VP8 simulcast to scenario tests.
by Sebastian Jansson
· 6 years ago
ccb9b75
Create version 01 of Generic Frame Descriptor - with discardability flag
by Elad Alon
· 6 years ago
0b2150c
Add a task queue into pc e2e fixture implementation
by Artem Titov
· 6 years ago
e82643f
Fix FFT output size to avoid incorrect band energy computation
by Alessio Bazzica
· 6 years ago
cc26fef
Use a CopyOnWriteBuffer to back EncodedImage data
by Niels Möller
· 6 years ago
0d4869c
Roll chromium_revision d723882358..4b3282a5d6 (633435:633587)
by chromium-webrtc-autoroll
· 6 years ago
ea7ef2a
Refactoring RtpSenderInternal to share implementation for Audio & Video.
by Amit Hilbuch
· 6 years ago
ba63caf
Roll chromium_revision 086bdb74b2..d723882358 (633288:633435)
by chromium-webrtc-autoroll
· 6 years ago
2297d33
Rejected simulcast layers will no longer appear in GetParameters().
by Amit Hilbuch
· 6 years ago
4e7058e
desktopCaptuer: exempt to overlapping between hidden taskbar and maximized target
by braveyao
· 6 years ago
0e44907
Roll chromium_revision 55c117dd14..086bdb74b2 (633171:633288)
by chromium-webrtc-autoroll
· 6 years ago
7abfd56
Improve CPU utilization when encoding VP8 with two temporal layers
by Elad Alon
· 6 years ago
599d592
Extend RemoteEstimatorProxy to support feedback on sender request.
by Johannes Kron
· 6 years ago
a89800c
Parse params of 3rd spatial layer from command line.
by Sergey Silkin
· 6 years ago
d8d3248
Reland "Delete test/constants.h"
by Elad Alon
· 6 years ago
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