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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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7f57788ab7e2f9ee03086ab3c4a14c1ad18b49e6
/
call
02c4f15
Stop using deprecated PacedSender method from RtpTransportControllerSend.
by Sebastian Jansson
· 6 years ago
b275788
Register stat callbacks after rate observer is registered.
by Piotr (Peter) Slatala
· 6 years ago
3d2ed19
Remove Transport implementation from ChannelSend
by Fredrik Solenberg
· 6 years ago
8eeccbe
Delete Start and Stop methods from TestVideoCapturer.
by Niels Möller
· 6 years ago
1618095
Cleanup of RtpTransportControllerSend.
by Sebastian Jansson
· 6 years ago
2701bc9
Signals start rate when registering to TargetTransferRateObserver.
by Sebastian Jansson
· 6 years ago
00672b1
Don't trigger too many probes when max allocated bitrate changes.
by Erik Språng
· 6 years ago
514f084
New statistic added to VideoReceiveStream to determine latency to first decode.
by Benjamin Wright
· 6 years ago
d1d7b23
Include protection bitrate in total max allocated bitrate
by Erik Språng
· 6 years ago
87609be
Merges RtpTransportControllerSend with SendSideCongestionController.
by Sebastian Jansson
· 6 years ago
af2adda
Explicit comparisons on NetworkRoute.
by Sebastian Jansson
· 6 years ago
d0b69a8
Send and receive color space information if available
by Johannes Kron
· 6 years ago
3e70781
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
by Yves Gerey
· 6 years ago
10403ae
Add PeerConnection option to configure minimum audio jitter buffer delay.
by Jakob Ivarsson
· 6 years ago
352ce5c
Expose delayed packet outage as a cumulative metric of samples in the new getStats API.
by Jakob Ivarsson
· 6 years ago
53382cb
Move RtcpStatistics from common_types.h to a new header file
by Niels Möller
· 6 years ago
ff05816
Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric
by Sam Zackrisson
· 6 years ago
8af8896
Expose jitter buffer flushes metric in new getStats api.
by Ruslan Burakov
· 6 years ago
af228ee
Disable flaky tests CallPerfTest.CaptureNtpTimeWithNetworkDelay on WIN.
by Alex Loiko
· 6 years ago
8b5d9d8
Remove the audio/video split for the RTCP report intervals.
by Jiawei Ou
· 6 years ago
6736df1
Moves BitrateAllocationUpdate to api.
by Sebastian Jansson
· 6 years ago
13e5903
Using unit classes in BitrateAllocationUpdate struct.
by Sebastian Jansson
· 6 years ago
c69a56e
Remove more unneeded things from ChannelSend
by Fredrik Solenberg
· 6 years ago
89c94b9
Adds target bandwidth to BitrateAllocator.
by Sebastian Jansson
· 6 years ago
2222a80
Delete unneeded includes of common_types.h and gn deps on webrtc_common.
by Niels Möller
· 6 years ago
179a392
Implement TargetBitrate, NetworkRoute and overhead features of media transport interface.
by Piotr (Peter) Slatala
· 6 years ago
cc8e8bb
Pass the media transport from JsepTransportController to Call.
by Piotr (Peter) Slatala
· 6 years ago
de8e6e6
Refactor bitrate configuration in CallTest
by Niels Möller
· 6 years ago
5571812
Adding rtcp report interval into RTCConfiguration.
by Jiawei Ou
· 6 years ago
c2ebe21
Reland "Use the factory instead of using the builtin code path in `VideoCodecInitializer`"
by Jiawei Ou
· 6 years ago
cd2e105
Reenable test RampUpTest.AudioTransportSequenceNumber
by Niels Möller
· 6 years ago
b768e88
Reland "Isolating APM API build target: making :api an actual target."
by Alessio Bazzica
· 6 years ago
61c6e56
Revert "Isolating APM API build target: making :api an actual target."
by Alessio Bazzica
· 6 years ago
a7f77a7
Isolating APM API build target: making :api an actual target.
by Alessio Bazzica
· 6 years ago
c572ff3
Add default constructor for rtc::Event
by Niels Möller
· 6 years ago
2cd3b4c
Fixing bug in SimulatedNetwork where packets stop.
by Sebastian Jansson
· 6 years ago
56ef305
Move event logging of config into AudioSendStream.
by Oskar Sundbom
· 6 years ago
59844ce
Revert "Use the factory instead of using the builtin code path in `VideoCodecInitializer`."
by Qingsi Wang
· 6 years ago
be14217
Use the factory instead of using the builtin code path in `VideoCodecInitializer`.
by Jiawei Ou
· 6 years ago
7182286
Allow FakeNetworkPipe to wake up its processing thread
by Sebastian Jansson
· 6 years ago
69807e8
Depend directly on destination targets.
by Yves Gerey
· 6 years ago
21cddff
Harmonize paths to dependent targets.
by Yves Gerey
· 6 years ago
992a868
Fix for clock reset repair.
by Christoffer Rodbro
· 6 years ago
9190b82
Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap
by Johannes Kron
· 6 years ago
7d76a31
Use MediaTransportInterface, for audio streams.
by Niels Möller
· 6 years ago
359d60a
Adds target rate to audio send stream stats.
by Sebastian Jansson
· 6 years ago
e2754c9
Fixes bug in AudioPriorityBitrateAllocationStrategy field trial.
by Sebastian Jansson
· 6 years ago
c0e4d45
Adds BitrateAllocation struct to OnBitrateUpdated.
by Sebastian Jansson
· 6 years ago
1803bb2
Fix for clock read race in FakeNetworkPipe.
by Christoffer Rodbro
· 6 years ago
3284b61
Fix for packet loss tracking in network emulation.
by Christoffer Rodbro
· 6 years ago
2506839
Add DCHECK for wrap around in RtpVideoSender::OnBitrateUpdated.
by Bjorn Terelius
· 6 years ago
44a262a
Declares BitrateAllocator methods const.
by Sebastian Jansson
· 6 years ago
362cb50
Remove redundant RTC_DCHECK of max/min RTP header extension id
by Johannes Kron
· 6 years ago
988cc08
[Cleanup] Add missing #include. Remove useless ones.
by Yves Gerey
· 6 years ago
648d28a
Media engine and channel support for per-channel dscp values, specified by RtpParameter
by Tim Haloun
· 6 years ago
2dfa998
Reland "Prefix flag macros with WEBRTC_."
by Mirko Bonadei
· 6 years ago
c538fc7
Revert "Prefix flag macros with WEBRTC_."
by Mirko Bonadei
· 6 years ago
5ccdc13
Prefix flag macros with WEBRTC_.
by Mirko Bonadei
· 6 years ago
192eeec
Enable End-to-End Encrypted Video Frames.
by Benjamin Wright
· 6 years ago
bfb444c
Adds new CryptoOption crypto_options.frame.require_frame_encryption.
by Benjamin Wright
· 6 years ago
fab9129
Get frame type, width and height from the generic descriptor.
by philipel
· 6 years ago
f5e767d
Don't send max allocation probe unless allocation changed.
by Sebastian Jansson
· 6 years ago
1298541
Removing unnecessary dependencies on socket.h.
by Sebastian Jansson
· 6 years ago
40a7a35
Extract functionality of test_main into separate library.
by Artem Titov
· 6 years ago
76ad154
New method for precise packet reception time measurement.
by Christoffer Rodbro
· 6 years ago
88be972
Delete post_encode_callback
by Niels Möller
· 6 years ago
0378997
Adds flags indicating presence in allocation and feedback per packet.
by Sebastian Jansson
· 6 years ago
4ff7214
Using TaskQueue for congestion controller by default.
by Sebastian Jansson
· 6 years ago
75e3647
Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig
by Artem Titov
· 6 years ago
64be7fa
Move FecController to RtpVideoSender.
by Stefan Holmer
· 6 years ago
8ea1e9d
Switch webrtc from deprecated usages of NetworkSimulationInterface
by Artem Titov
· 6 years ago
ae4237e
Set ChannelReceive transport at construction time.
by Niels Möller
· 6 years ago
84583f6
Enable End-to-End Encrypted Audio Payloads.
by Benjamin Wright
· 6 years ago
433eafe
Delete unused includes of assert.h
by Niels Möller
· 6 years ago
32fe3d1
Temporarily increase visibility of pacing and call/rtp_interfaces
by Danil Chapovalov
· 6 years ago
5ca2912
Delete VideoReceiveStream::EnableEncodedFrameRecording
by Niels Möller
· 6 years ago
35fa280
Adds allocated rate without feedback to new congestion controller.
by Sebastian Jansson
· 6 years ago
e0d455b
Remove runtime_enabled_feature.
by Mirko Bonadei
· 6 years ago
156d11d
Adds packet_size to rtc::SentPacket in testing code.
by Sebastian Jansson
· 6 years ago
1f3206c
Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1
by Niels Möller
· 6 years ago
cbcbc22
Reland "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."
by Niels Möller
· 6 years ago
17f4878
Remove deprecated field_trial_default and metrics_default.
by Mirko Bonadei
· 6 years ago
377b26e
Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."
by Sebastian Jansson
· 6 years ago
dc8c981
Makes new congestion controller work with rtp sender tests.
by Sebastian Jansson
· 6 years ago
efb94d5
Revert "Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config.""
by Oleh Prypin
· 6 years ago
7961dc2
Revert "Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config."
by Niels Moller
· 6 years ago
529d0d9
Replace VideoDecoder with VideoDecoderFactory in VideoReceiveStream config.
by Niels Möller
· 6 years ago
49ac595
Add GetSources to VideoRtpReceiver
by Jonas Oreland
· 6 years ago
569397f
Reland "Added field trial WebRTC-GenericDescriptor for the new generic descriptor."
by philipel
· 6 years ago
6f68324
Revert "Added field trial WebRTC-GenericDescriptor for the new generic descriptor."
by Lu Liu
· 6 years ago
3f4a4fa
Added field trial WebRTC-GenericDescriptor for the new generic descriptor.
by philipel
· 6 years ago
a7af021
Disable ulpfec when field trial flag is present
by Emircan Uysaler
· 6 years ago
3a6b729
Cleanup: remove deprecated class shortcuts.
by Artem Titov
· 6 years ago
585d1aa
Register video rtp header extensions in rtp_rtcp by uri
by Danil Chapovalov
· 6 years ago
8c1bf95
Reland "Add initial support for RtpEncodingParameters max_framerate."
by Åsa Persson
· 6 years ago
cb7e1d2
Use SdpVideoFormat in VideoReceiveStream::Decoder
by Niels Möller
· 6 years ago
d52a1a6
Reland "Remove RTPVideoHeader::vp8() accessors."
by Philip Eliasson
· 6 years ago
1811c04
Revert "Remove RTPVideoHeader::vp8() accessors."
by Philip Eliasson
· 6 years ago
af7afc6
Remove RTPVideoHeader::vp8() accessors.
by philipel
· 6 years ago
22c7d69
Enable ULPFEC for kVideoCodecGeneric if GenericPictureId is enabled.
by Sami Kalliomäki
· 6 years ago
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