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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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00c57e39294a2d2100623629202115b55639e5a4
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
· 6 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
1925b5a
Delete deprecated version of AudioCodingModule::IncomingPacket
by Niels Möller
· 6 years ago
1431572
Roll chromium_revision 0f484ff968..55c117dd14 (633071:633171)
by chromium-webrtc-autoroll
· 6 years ago
ffd1f93
Revert "Tests for multi-stream Opus."
by Mirko Bonadei
· 6 years ago
7131880
Don't block the signaling thread during the call.
by Mirko Bonadei
· 6 years ago
0e1a1f9
Add verbose logging to encoder bitrate adjuster
by Erik Språng
· 6 years ago
4f36b7a
Revert "Delete test/constants.h"
by Oleh Prypin
· 6 years ago
06c5145
Adds support for VP9 scalability layers to scenario tests.
by Sebastian Jansson
· 6 years ago
9c31ac2
Tests for multi-stream Opus.
by Alex Loiko
· 6 years ago
f2727fb
Adds slides support to scenario tests.
by Sebastian Jansson
· 6 years ago
e9652ca
Android: Add video processing interface
by Magnus Jedvert
· 6 years ago
4a2d57a
Don't include video_bitrate_allocation.h from encoded_image.h
by Niels Möller
· 6 years ago
71aee3a
Reland "Propagate VideoFrame::UpdateRect to encoder"
by Ilya Nikolaevskiy
· 6 years ago
f873cd9
Roll chromium_revision 26c36e3408..0f484ff968 (632825:633071)
by chromium-webrtc-autoroll
· 6 years ago
bf47495
Update remaining audio test code to not use WebRtcRTPHeader.
by Niels Möller
· 6 years ago
a0b1fb9
Pass H264 profile/level settings to codec.
by Sergey Silkin
· 6 years ago
3073c72
Fix AndroidVideoDecoderTest for new Robolectric version.
by Sami Kalliomäki
· 6 years ago
e049eba
Revert "Add Sender and Receiver interfaces for MediaTransport audio"
by Sergey Silkin
· 6 years ago
d2f0436
Make sdk/android:{audio,video}_api_java publicly visible.
by Mirko Bonadei
· 6 years ago
0d8eed6
Add Sender and Receiver interfaces for MediaTransport audio
by Niels Möller
· 6 years ago
6e1402b
Skip SSIM calculation in real time mode.
by Sergey Silkin
· 6 years ago
afb5dbb
Update ACM to use RTPHeader instead of WebRtcRTPHeader
by Niels Möller
· 6 years ago
389b167
Delete test/constants.h
by Elad Alon
· 6 years ago
8d2e228
Add thread safety annotations for PeerConnection::*_state_
by Karl Wiberg
· 6 years ago
e45c688
Remove webrtc::ProtoString.
by Mirko Bonadei
· 6 years ago
eaf6a8c
Adding src/third_party/androidx to the DEPS file.
by Mirko Bonadei
· 6 years ago
7ea4605
Add latency to remote source api.
by Ruslan Burakov
· 6 years ago
86f0974
Roll chromium_revision 7df1a5ba86..26c36e3408 (632711:632825)
by chromium-webrtc-autoroll
· 6 years ago
c664314
Clean up implementation in stream_params
by Steve Anton
· 6 years ago
ca890ee
Revert "Fix getStats() freeze bug affecting Chromium but not WebRTC standalone."
by Mirko Bonadei
· 6 years ago
ca3c801
Minor eventlogvisualizer tweaks.
by Konrad Hofbauer
· 6 years ago
429b67d
Revert "Propagate VideoFrame::UpdateRect to encoder"
by Mirko Bonadei
· 6 years ago
675b47d
Roll chromium_revision bf2d75ba40..7df1a5ba86 (632595:632711)
by chromium-webrtc-autoroll
· 6 years ago
9775a58
Plot bitrate allocation per layer based on RTCP XR target bitrate.
by Bjorn Terelius
· 6 years ago
b03ab71
Add thread safety annotation for PeerConnection::event_log_
by Karl Wiberg
· 6 years ago
744310f
Add thread safety annotation for PeerConnection::observer_ and factory_
by Karl Wiberg
· 6 years ago
7c974e6
Plot RTCP types for incoming and outgoing RTCP packets.
by Bjorn Terelius
· 6 years ago
c39f462
Move RtcEventProbeClusterCreated to the network controller.
by Piotr (Peter) Slatala
· 6 years ago
6255af9
Fix RateCounter to don't fail if there are too small amount of events
by Artem Titov
· 6 years ago
efa72a1
Propagate VideoFrame::UpdateRect to encoder
by Ilya Nikolaevskiy
· 6 years ago
3a656d1
Tune bitrates and minQP thresholds for high-fps screenshare.
by Ilya Nikolaevskiy
· 6 years ago
c8221fc
Roll chromium_revision d1f68eb66e..bf2d75ba40 (632477:632595)
by chromium-webrtc-autoroll
· 6 years ago
075f687
Add struct for feedback request to RTPHeaderExtension
by Johannes Kron
· 6 years ago
05d43c6
Fix getStats() freeze bug affecting Chromium but not WebRTC standalone.
by Henrik Boström
· 6 years ago
914351d
Reland "Always offer transport sequence number header extension for audio""
by Per Kjellander
· 6 years ago
106d92d
Add thread safety annotation for PeerConnection::SignalDataChannelCreated_
by Karl Wiberg
· 6 years ago
13bc871
PostMessageWithFunctor() added.
by Henrik Boström
· 6 years ago
397c06f
Revert "Always offer transport sequence number header extension for audio"
by Ying Wang
· 6 years ago
7e0e44f
Move video-related MediaTransport interfaces to their own file and target
by Niels Möller
· 6 years ago
054db54
Remove an absl::WrapUnique usage without absl/memory/memory.h include
by tzik
· 6 years ago
22997d6
Roll chromium_revision 2ad52fb2a4..d1f68eb66e (632357:632477)
by chromium-webrtc-autoroll
· 6 years ago
1c9c9fc
Replace replace_substrs with Abseil
by Steve Anton
· 6 years ago
bf9e01a
Add support of fast media sending in peer connection e2e test
by Artem Titov
· 6 years ago
ceba6ae
Return a copy, becase GetPercentile in SamplesStatsCounter isn't const
by Artem Titov
· 6 years ago
cf8405e
Add generic packet rates to event_log_visualizer.
by Piotr (Peter) Slatala
· 6 years ago
15653f9
Roll chromium_revision 78de17c053..2ad52fb2a4 (632252:632357)
by chromium-webrtc-autoroll
· 6 years ago
aa58415
Reland "Enabling Simulcast use via AddTransceiver."
by Amit Hilbuch
· 6 years ago
aec9794
Fix DCHECK when encoding GenericPacket* events using the legacy RTC event log format.
by Piotr (Peter) Slatala
· 6 years ago
9e2692c
Roll chromium_revision 9a34b2cc2d..78de17c053 (632146:632252)
by chromium-webrtc-autoroll
· 6 years ago
d036c65
Clarify and unify outgoing and incoming packet loss rate plots.
by Konrad Hofbauer
· 6 years ago
663844d
Update test code to use EncodedImage::Allocate
by Niels Möller
· 6 years ago
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