Sign in
webrtc
/
src
/
HEAD
« Previous
0874530
Add gn configs to remove the dependency to audio and video codecs.
by Jiawei Ou
· 5 years ago
3329be4
Roll chromium_revision b847e52039..d366835eb8 (631155:631269)
by chromium-webrtc-autoroll
· 5 years ago
464a557
Adds audio priority bitrate field trial parameter.
by Sebastian Jansson
· 5 years ago
eb81b47
Update H264EncoderImpl to use EncodedImage::Allocate
by Niels Möller
· 5 years ago
d3666b2
Introduce cross traffic for emulated network layer.
by Artem Titov
· 5 years ago
5c4ddad
Delete obsolete usage of FakeConstraints
by Niels Möller
· 5 years ago
9bf67ea
AEC3: Fix delay hysteresis validation
by Gustaf Ullberg
· 5 years ago
99b9149
Enable padding bit in TransportFeedback packets
by Johannes Kron
· 5 years ago
2ce0cb0
Add missing 'explicit' specifier to GainControlImpl
by Sam Zackrisson
· 5 years ago
eb17524
Migrate libevent task queue implementation to TaskQueueBase interface
by Danil Chapovalov
· 5 years ago
675e5aa
Roll chromium_revision b6a69427be..b847e52039 (631040:631155)
by chromium-webrtc-autoroll
· 5 years ago
a93b8b0
Update SimulcastTestFixtureImpl to use EncodedImage::Allocate
by Niels Möller
· 5 years ago
40027b1
Roll chromium_revision e4a7c15e8a..b6a69427be (630925:631040)
by chromium-webrtc-autoroll
· 5 years ago
a887c32
Roll chromium_revision 92e6bfe90b..e4a7c15e8a (630806:630925)
by chromium-webrtc-autoroll
· 5 years ago
26d0876
Roll chromium_revision 339f6a582b..92e6bfe90b (630696:630806)
by chromium-webrtc-autoroll
· 5 years ago
871e144
Revert "Reland "Partial frame capture API part 1""
by Ilya Nikolaevskiy
· 5 years ago
421c859
Remove crit_render_ lock from webrtc::GainControlImpl
by Sam Zackrisson
· 5 years ago
00f9400
Dump histogram data in AEC3 delay estimator
by Sam Zackrisson
· 5 years ago
271195f
Fix potential crash when building rtx packet
by Danil Chapovalov
· 5 years ago
501bfba
Split rtp_receiver for readability.
by Ruslan Burakov
· 5 years ago
b66003c
Delete video source proxying in WebRtcVideoSendStream
by Niels Möller
· 5 years ago
6df89cc
Revert "Partial frame capture API part 2"
by Ilya Nikolaevskiy
· 5 years ago
b00eb19
Removes Start/Stop on network emulation manager.
by Sebastian Jansson
· 5 years ago
eb7589e
Revert "Partial frame capture API part 3"
by Ilya Nikolaevskiy
· 5 years ago
fd5d473
Revert "Partial frame capture API part 6"
by Ilya Nikolaevskiy
· 5 years ago
85fc325
Revert "Partial frame capture API part 5"
by Ilya Nikolaevskiy
· 5 years ago
02f4e32
Make some new rtc_base targets publicly visible
by Karl Wiberg
· 5 years ago
f13c2cd
Roll chromium_revision eb2aa6ea6a..339f6a582b (630596:630696)
by chromium-webrtc-autoroll
· 5 years ago
61b4f74
Fix PeerConnectionInterface::StartRtcEventLog documentation.
by Mirko Bonadei
· 5 years ago
1a1c52b
H.264 temporal layers w/frame marking (PART 2/3)
by Johnny Lee
· 5 years ago
e556768
Roll chromium_revision eead273f0c..eb2aa6ea6a (630484:630596)
by chromium-webrtc-autoroll
· 5 years ago
157540a
Stop hard-coding default IDs for RTP extensions
by Elad Alon
· 5 years ago
efc9a14
Make UniqueNumberGenerator::AddKnownId() return a value
by Elad Alon
· 5 years ago
6ba2738
Roll chromium_revision d60317bbda..eead273f0c (630357:630484)
by chromium-webrtc-autoroll
· 5 years ago
5699142
Use c=IN IP4 <hostname> to support the presence of hostname candidates.
by Qingsi Wang
· 5 years ago
7832343
Revert "Enabling Simulcast use via AddTransceiver."
by Emircan Uysaler
· 5 years ago
836fee1
Calculate next process time in simulated network.
by Sebastian Jansson
· 5 years ago
f6adac8
Add rtc event generic packet sent and received.
by Piotr (Peter) Slatala
· 5 years ago
50930a6
Roll chromium_revision 46a21d8d05..d60317bbda (630250:630357)
by chromium-webrtc-autoroll
· 5 years ago
1d13b37
Update LibvpxVp8Encoder to use EncodedImage::Allocate
by Niels Möller
· 5 years ago
b7edf69
Delete rtc::File, usage replaced with FileWrapper
by Niels Möller
· 5 years ago
9f3aabb
Delete obsolete class cricket::VideoCapturer
by Niels Möller
· 5 years ago
494ff28
Delete unused media constraints
by Niels Möller
· 5 years ago
a8d48ab
Fix incorrect FPS measure when frame dropper kicks in
by Erik Språng
· 5 years ago
bdfadd6
Adds Stop methods to media streams in scenario framework.
by Sebastian Jansson
· 5 years ago
85eab49
Simplify peer connection smoke test to remove flakiness for now.
by Artem Titov
· 5 years ago
3dd473b
Refactor of RtpPacket constructor
by Johannes Kron
· 5 years ago
7ff164e
Plumbing of feedback on request setting
by Johannes Kron
· 5 years ago
5f6abcf
Fix for RttBackoff when sending of packets with TWCC stops.
by Christoffer Rodbro
· 5 years ago
dcba72b
Resume rolling buildtools, now as chromium/src/buildtools
by Oleh Prypin
· 5 years ago
b76b9ba
Set WEBRTC_USE_H264 in common_config
by Johannes Kron
· 5 years ago
3f171df
Add support for building iOS simulator code for iOS 11 and 12
by Artem Titarenko
· 5 years ago
52e9e8d
Remove now-unused iOS CI config files
by Oleh Prypin
· 5 years ago
51aa82d
Roll chromium_revision 6f2fb1192a..46a21d8d05 (630145:630250)
by chromium-webrtc-autoroll
· 5 years ago
9f97c9a
Add starting of VideoQualityAnalyzer in the e2e peer connection level test
by Artem Titov
· 5 years ago
5963fdd
Pass-by-reference instead of value to initWithNativeEncodedImage
by Dillon Cower
· 5 years ago
108f20f
Fix color space bug in wrapper of H264 decoder
by Johannes Kron
· 5 years ago
a8cb366
Add field trial for forced software decoder fallback.
by Åsa Persson
· 5 years ago
587c5d1
Roll chromium_revision 34f99c21a3..6f2fb1192a (630023:630145)
by chromium-webrtc-autoroll
· 5 years ago
ec3b9ff
Move audio-related MediaTransport interfaces to their own file and target
by Niels Möller
· 5 years ago
e12778c
Update VP9EncoderImpl to use EncodedImage::Allocate
by Niels Möller
· 5 years ago
f9a5561
Roll chromium_revision ee5dfb2215..34f99c21a3 (629907:630023)
by chromium-webrtc-autoroll
· 5 years ago
d7180cc
Also check the pending remote description when generating MIDs for legacy remote offers
by Steve Anton
· 5 years ago
ce470aa
Enabling Simulcast use via AddTransceiver.
by Amit Hilbuch
· 5 years ago
a6a273d
Introduce PeerConnectionE2EQualityTestFixture implementation.
by Artem Titov
· 5 years ago
c363a53
Define RtpGenericFrameDescriptorExtension00
by Elad Alon
· 5 years ago
260a71d
Delete deprecated method PeerConnectionFactory::CreateVideoSource
by Niels Möller
· 5 years ago
59ab1cf
Move ownership of RTPSenderVideo and RTPSenderAudio one level up
by Niels Möller
· 5 years ago
938dd9f
Add owned data buffer to EncodedImage
by Niels Möller
· 5 years ago
e6f6a0c
Add missing operator= and extra methods to the SamplesStatsCounter.
by Artem Titov
· 5 years ago
710f3d3
Use task queue factory factory as parameter for TaskQueueTest
by Danil Chapovalov
· 5 years ago
0041fe5
Roll chromium_revision 1a597bc4e4..ee5dfb2215 (629788:629907)
by chromium-webrtc-autoroll
· 5 years ago
cdab13d
Roll chromium_revision c27b32b2fd..1a597bc4e4 (629510:629788)
by Oleh Prypin
· 5 years ago
86c8ad9
Pause rolling buildtools
by Oleh Prypin
· 5 years ago
ef288dd
Reland: Remove dead code from stream_params.h
by Steve Anton
· 5 years ago
e1dcce2
Remove HAVE_WEBRTC_VOICE.
by Fredrik Solenberg
· 5 years ago
e7b9e6b
Move RtpSenderVideo tests to separate file.
by Niels Möller
· 5 years ago
d70a114
Delete MediaTransport method SetNetworkChangeCallback
by Niels Möller
· 5 years ago
fe6e50f
Allow more than one registered network change callback in MediaTransport
by Niels Möller
· 5 years ago
3e61888
Roll chromium_revision 9d5d0c6635..c27b32b2fd (629245:629510)
by Oleh Prypin
· 5 years ago
7ca375c
Implement encoder overshoot detector and rate adjuster.
by Erik Språng
· 5 years ago
e98954c
Prevent updating state in the delay manager if the packet was reordered.
by Jakob Ivarsson
· 5 years ago
9025bd5
Separate AndroidVideoTrackSource::OnFrameCaptured from adaptation
by Magnus Jedvert
· 5 years ago
bb87f8a
Delete unused/unsupported RetransmissionMode constants
by Niels Möller
· 5 years ago
0859142
Add events processing to GetIceEvents.
by Sebastian Jansson
· 5 years ago
4092d6f
Fix autoroller to skip entries without @revision in them
by Oleh Prypin
· 5 years ago
6cfb403
Fix test FrameGenerator to work with a single file source
by Ilya Nikolaevskiy
· 5 years ago
cf416e4
Revert "Remove dead code from stream_params.h"
by Oleh Prypin
· 5 years ago
2fb7999
Replace implicit int->char->string conversion
by Oleh Prypin
· 5 years ago
57d4ac9
Add more unit tests for RateControlSettings.
by Rasmus Brandt
· 5 years ago
3b50f9f
Propagate base minimum delay to audio_receiver_stream
by Ruslan Burakov
· 5 years ago
9ce800d
Add PRESUBMIT to enforce usage of new Googletest APIs.
by Mirko Bonadei
· 5 years ago
12d1285
Use the new TEST_SUITE GoogleTest API (regression).
by Mirko Bonadei
· 5 years ago
38c83b9
Remove unused file.
by Fredrik Solenberg
· 5 years ago
3f408d0
Remove dead code from stream_params.h
by Steve Anton
· 5 years ago
d1b6206
Roll chromium_revision 3b81a4d714..9d5d0c6635 (629131:629245)
by chromium-webrtc-autoroll
· 5 years ago
65835be
Allow logging of char* null pointer.
by Niels Möller
· 5 years ago
99b275d
Introduce class that handles native wrapping of AndroidVideoTrackSource
by Magnus Jedvert
· 5 years ago
b3032b6
Revert "Partial frame capture API part 4"
by Ilya Nikolaevskiy
· 5 years ago
7752ad6
Partial frame capture API part 6
by Ilya Nikolaevskiy
· 5 years ago
Next »