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7abfd56
Improve CPU utilization when encoding VP8 with two temporal layers
by Elad Alon
· 5 years ago
599d592
Extend RemoteEstimatorProxy to support feedback on sender request.
by Johannes Kron
· 5 years ago
a89800c
Parse params of 3rd spatial layer from command line.
by Sergey Silkin
· 5 years ago
d8d3248
Reland "Delete test/constants.h"
by Elad Alon
· 5 years ago
1925b5a1
Delete deprecated version of AudioCodingModule::IncomingPacket
by Niels Möller
· 5 years ago
1431572
Roll chromium_revision 0f484ff968..55c117dd14 (633071:633171)
by chromium-webrtc-autoroll
· 5 years ago
ffd1f93
Revert "Tests for multi-stream Opus."
by Mirko Bonadei
· 5 years ago
7131880
Don't block the signaling thread during the call.
by Mirko Bonadei
· 5 years ago
0e1a1f9
Add verbose logging to encoder bitrate adjuster
by Erik Språng
· 5 years ago
4f36b7a
Revert "Delete test/constants.h"
by Oleh Prypin
· 5 years ago
06c5145
Adds support for VP9 scalability layers to scenario tests.
by Sebastian Jansson
· 5 years ago
9c31ac2
Tests for multi-stream Opus.
by Alex Loiko
· 5 years ago
f2727fb
Adds slides support to scenario tests.
by Sebastian Jansson
· 5 years ago
e9652ca6
Android: Add video processing interface
by Magnus Jedvert
· 5 years ago
4a2d57a
Don't include video_bitrate_allocation.h from encoded_image.h
by Niels Möller
· 5 years ago
71aee3a
Reland "Propagate VideoFrame::UpdateRect to encoder"
by Ilya Nikolaevskiy
· 5 years ago
f873cd9
Roll chromium_revision 26c36e3408..0f484ff968 (632825:633071)
by chromium-webrtc-autoroll
· 5 years ago
bf47495
Update remaining audio test code to not use WebRtcRTPHeader.
by Niels Möller
· 5 years ago
a0b1fb9
Pass H264 profile/level settings to codec.
by Sergey Silkin
· 5 years ago
3073c72
Fix AndroidVideoDecoderTest for new Robolectric version.
by Sami Kalliomäki
· 5 years ago
e049eba
Revert "Add Sender and Receiver interfaces for MediaTransport audio"
by Sergey Silkin
· 5 years ago
d2f0436
Make sdk/android:{audio,video}_api_java publicly visible.
by Mirko Bonadei
· 5 years ago
0d8eed6
Add Sender and Receiver interfaces for MediaTransport audio
by Niels Möller
· 5 years ago
6e1402b
Skip SSIM calculation in real time mode.
by Sergey Silkin
· 5 years ago
afb5dbb
Update ACM to use RTPHeader instead of WebRtcRTPHeader
by Niels Möller
· 5 years ago
389b167
Delete test/constants.h
by Elad Alon
· 5 years ago
8d2e228
Add thread safety annotations for PeerConnection::*_state_
by Karl Wiberg
· 5 years ago
e45c688
Remove webrtc::ProtoString.
by Mirko Bonadei
· 5 years ago
eaf6a8c
Adding src/third_party/androidx to the DEPS file.
by Mirko Bonadei
· 5 years ago
7ea4605
Add latency to remote source api.
by Ruslan Burakov
· 5 years ago
86f0974
Roll chromium_revision 7df1a5ba86..26c36e3408 (632711:632825)
by chromium-webrtc-autoroll
· 5 years ago
c664314
Clean up implementation in stream_params
by Steve Anton
· 5 years ago
ca890ee
Revert "Fix getStats() freeze bug affecting Chromium but not WebRTC standalone."
by Mirko Bonadei
· 5 years ago
ca3c801
Minor eventlogvisualizer tweaks.
by Konrad Hofbauer
· 5 years ago
429b67d
Revert "Propagate VideoFrame::UpdateRect to encoder"
by Mirko Bonadei
· 5 years ago
675b47d
Roll chromium_revision bf2d75ba40..7df1a5ba86 (632595:632711)
by chromium-webrtc-autoroll
· 5 years ago
9775a58
Plot bitrate allocation per layer based on RTCP XR target bitrate.
by Bjorn Terelius
· 5 years ago
b03ab71
Add thread safety annotation for PeerConnection::event_log_
by Karl Wiberg
· 5 years ago
744310f
Add thread safety annotation for PeerConnection::observer_ and factory_
by Karl Wiberg
· 5 years ago
7c974e6
Plot RTCP types for incoming and outgoing RTCP packets.
by Bjorn Terelius
· 5 years ago
c39f462
Move RtcEventProbeClusterCreated to the network controller.
by Piotr (Peter) Slatala
· 5 years ago
6255af9
Fix RateCounter to don't fail if there are too small amount of events
by Artem Titov
· 5 years ago
efa72a1
Propagate VideoFrame::UpdateRect to encoder
by Ilya Nikolaevskiy
· 5 years ago
3a656d1
Tune bitrates and minQP thresholds for high-fps screenshare.
by Ilya Nikolaevskiy
· 5 years ago
c8221fc
Roll chromium_revision d1f68eb66e..bf2d75ba40 (632477:632595)
by chromium-webrtc-autoroll
· 5 years ago
075f687
Add struct for feedback request to RTPHeaderExtension
by Johannes Kron
· 5 years ago
05d43c6
Fix getStats() freeze bug affecting Chromium but not WebRTC standalone.
by Henrik Boström
· 5 years ago
914351d
Reland "Always offer transport sequence number header extension for audio""
by Per Kjellander
· 5 years ago
106d92d
Add thread safety annotation for PeerConnection::SignalDataChannelCreated_
by Karl Wiberg
· 5 years ago
13bc871
PostMessageWithFunctor() added.
by Henrik Boström
· 5 years ago
397c06f
Revert "Always offer transport sequence number header extension for audio"
by Ying Wang
· 5 years ago
7e0e44f
Move video-related MediaTransport interfaces to their own file and target
by Niels Möller
· 5 years ago
054db54
Remove an absl::WrapUnique usage without absl/memory/memory.h include
by tzik
· 5 years ago
22997d6
Roll chromium_revision 2ad52fb2a4..d1f68eb66e (632357:632477)
by chromium-webrtc-autoroll
· 5 years ago
1c9c9fc
Replace replace_substrs with Abseil
by Steve Anton
· 5 years ago
bf9e01a
Add support of fast media sending in peer connection e2e test
by Artem Titov
· 5 years ago
ceba6ae
Return a copy, becase GetPercentile in SamplesStatsCounter isn't const
by Artem Titov
· 5 years ago
cf8405e
Add generic packet rates to event_log_visualizer.
by Piotr (Peter) Slatala
· 5 years ago
15653f9
Roll chromium_revision 78de17c053..2ad52fb2a4 (632252:632357)
by chromium-webrtc-autoroll
· 5 years ago
aa58415
Reland "Enabling Simulcast use via AddTransceiver."
by Amit Hilbuch
· 5 years ago
aec9794
Fix DCHECK when encoding GenericPacket* events using the legacy RTC event log format.
by Piotr (Peter) Slatala
· 5 years ago
9e2692c
Roll chromium_revision 9a34b2cc2d..78de17c053 (632146:632252)
by chromium-webrtc-autoroll
· 5 years ago
d036c65
Clarify and unify outgoing and incoming packet loss rate plots.
by Konrad Hofbauer
· 5 years ago
663844d
Update test code to use EncodedImage::Allocate
by Niels Möller
· 5 years ago
fd965c0
Always offer transport sequence number header extension for audio
by Per Kjellander
· 5 years ago
92e7c69
Revert "Update VP9EncoderImpl to use EncodedImage::Allocate"
by Niels Moller
· 5 years ago
8e847ee
Make recv_deltas optional in TransportFeedback packets
by Johannes Kron
· 5 years ago
69fb6c8
Allow DtlsTransport::Information() to be called off-thread
by Harald Alvestrand
· 5 years ago
068fc35
Break out parameters from EventLogAnalyzer to AnalyzerConfig struct.
by Bjorn Terelius
· 5 years ago
f0c366b
Cleanup of scenario test video stream setup.
by Sebastian Jansson
· 5 years ago
d00045e
Changing command line flag for scenario logs root directory.
by Sebastian Jansson
· 5 years ago
dac03d9
Move MediaConstraintsInterface to sdk/, and make it a concrete class
by Niels Möller
· 5 years ago
1d7bf89
Add LS_VERBOSE logging for target bitrate in GoogCC
by Evan Shrubsole
· 5 years ago
0179a3d
Roll chromium_revision bfd7bcf815..9a34b2cc2d (632040:632146)
by chromium-webrtc-autoroll
· 5 years ago
da825b1
Replace NOTREACHED with a break.
by Piotr (Peter) Slatala
· 5 years ago
1290fc7
Remove old accessor in GenericAckReceived
by Piotr (Peter) Slatala
· 5 years ago
788f577
Update the resolution check for VP8 simulcast.
by Mirta Dvornicic
· 5 years ago
6b88a8f
Introduce default video quality analyzer
by Artem Titov
· 5 years ago
b1ea48c
Roll chromium_revision 103665932a..bfd7bcf815 (631883:632040)
by chromium-webrtc-autoroll
· 5 years ago
5ae259e
Use a provider in rtc::Network to access the mDNS responder.
by Qingsi Wang
· 5 years ago
616b233
Add FullStackTest with simulated encoder overshooting
by Erik Språng
· 5 years ago
6c02541
Revert "Delete video source proxying in WebRtcVideoSendStream"
by Christian Fremerey
· 5 years ago
3588394
Roll chromium_revision d026ac796d..103665932a (631722:631883)
by chromium-webrtc-autoroll
· 5 years ago
dfd5c4b
Parse XR, FIR and PLI in rtc_event_log_parser.cc
by Bjorn Terelius
· 5 years ago
3c119fb
Handle HKDF key derivation when building with OpenSSL.
by Sergey Sablin
· 5 years ago
5e2aad1c9
Support GenericPacketReceived/Sent/AckReceived event logs.
by Piotr (Peter) Slatala
· 5 years ago
975a899
Roll chromium_revision aa7b61fdc4..d026ac796d (631597:631722)
by chromium-webrtc-autoroll
· 5 years ago
4a68fb9
Separate base minimum delay and minimum delay.
by Ruslan Burakov
· 5 years ago
69bb3af
Update EncodedFrameForMediaTransport to use Retain() rather than set_buffer + memcpy.
by Niels Möller
· 5 years ago
14a7cf9
Adds CallEncoder to ChannelSend.
by Sebastian Jansson
· 5 years ago
cbf5949
Update MultiplexEncoderAdapter to use EncodedImage::Allocate
by Niels Möller
· 5 years ago
448c387
IceTransportWithTransportChannel: Initialize |thread_checker_| in declaration
by Raphael Kubo da Costa
· 5 years ago
2bd54a1
Ensure TestPeers are destroyed at the end of Run.
by Mirko Bonadei
· 5 years ago
6aca0b7
Add |update_rect| field and UpdateRect struct to VideoFrame.
by Ilya Nikolaevskiy
· 5 years ago
7f24fb9
Add settings to turn off VP8 base layer qp limit
by Erik Språng
· 5 years ago
98bcd32
Remove always_passing_unittest.cc.
by Mirko Bonadei
· 5 years ago
b4f7ab1
Fix -Wunused-result warnings
by Hans Wennborg
· 5 years ago
eedb0a1
Roll chromium_revision 23b4d2134b..aa7b61fdc4 (631425:631597)
by chromium-webrtc-autoroll
· 5 years ago
a795c3b
Roll chromium_revision d366835eb8..23b4d2134b (631269:631425)
by chromium-webrtc-autoroll
· 5 years ago
dcbdd2c
Add Foundation.framework to cocoa_threading target
by Jiawei Ou
· 5 years ago
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