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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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8718afb283b7bf6b071d4a65ad15ab51d438123f
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call
ef0627f
Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
by Mirko Bonadei
· 5 years ago
17608dc
RtpRtcp modules and below: Make media, RTX and FEC SSRCs const
by Erik Språng
· 5 years ago
fbde32e
Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
by Niels Möller
· 5 years ago
51bf200
Reduce number of RTPVideoSender::SendVideo parameters
by Danil Chapovalov
· 5 years ago
28214cd
Fix handling of large packets in RtxReceiveStream
by Niels Möller
· 5 years ago
dc34a25
Adds RTPSenderVideo::Config struct with red/ulpfec config
by Erik Språng
· 5 years ago
b9bfe65
Delete VCMEncodedFrame::VerifyAndAllocate
by Niels Möller
· 5 years ago
6cf554e
Reduces locking in RtpSenderVideo.
by Erik Språng
· 5 years ago
44db436
Propagate task queue to create test::DirectTransport by TaskQueueBase interface
by Danil Chapovalov
· 5 years ago
538ca57
Converts const methods in BitrateAllocator to non-member functions.
by Sebastian Jansson
· 5 years ago
f34116e
Replacing bandwidth adaptation trial with stable target in Opus encoder.
by Sebastian Jansson
· 5 years ago
40de3cc
Propagating TargetRate struct to BitrateAllocator.
by Sebastian Jansson
· 5 years ago
0a5ed89
Adds remote estimates to rtc event log.
by Sebastian Jansson
· 5 years ago
93b1ea2
Using struct for bitrate allocation limits.
by Sebastian Jansson
· 5 years ago
53227cc
Remove webrtc::MinPositive from api/.
by Mirko Bonadei
· 5 years ago
738bfa7
Remove api/bitrate_constraints.h.
by Mirko Bonadei
· 5 years ago
4d461ba
Reusing MediaStreamAllocationConfig struct in ObserverConfig.
by Sebastian Jansson
· 5 years ago
317a1f0
Use std::make_unique instead of absl::make_unique.
by Mirko Bonadei
· 5 years ago
7bf7a42
Delete flag VideoReceiveStream::Config::Rtp::remb
by Niels Möller
· 5 years ago
eaaaf41
Introduce api/crypto/BUILD.gn.
by Mirko Bonadei
· 5 years ago
65f17ca
Move MediaTransportInterface out of the libjingle_peerconnection_api target
by Niels Möller
· 5 years ago
fcfeefe
Move rtc_error.{h,cc} to its own build target.
by Mirko Bonadei
· 5 years ago
0987273
Add option to enable retransmission for all temporal layers in the constructor for rtp_sender_video.
by Andrei Dumitru
· 5 years ago
cc62b16
Add qualityLimitationResolutionChanges stat
by Evan Shrubsole
· 5 years ago
6516f76
Deprecate SingleThreadedTaskQueueForTesting class.
by Yves Gerey
· 5 years ago
a837030
Split out RtpSource from libjingle_peerconnection_api
by Niels Möller
· 5 years ago
f13df86
Delete audio methods SignalNetworkState
by Niels Möller
· 5 years ago
55dd72c
Remove lock for process thread pointer from PacedSender.
by Tommi
· 5 years ago
25eb47c
Make the RtpHeaderParserImpl available to tests and tools only.
by Tommi
· 5 years ago
b4a6128
Delete unneeded dependencies on libjingle_peerconnection_api
by Niels Möller
· 5 years ago
6dcd4dc
New target for api/rtp_parameters.h and api/media_types.h.
by Niels Möller
· 5 years ago
fac7e31
Removes TransportSequenceNumberAllocator
by Erik Språng
· 5 years ago
0c141c5
Fix frames dropped statistics
by Johannes Kron
· 5 years ago
d77cc24
New const method StreamStatistician::GetStats
by Niels Möller
· 5 years ago
4e615d5
Wire the stable target bitrate from GoogCC to the BitrateAllocator
by Florent Castelli
· 5 years ago
224c69d
Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo
by Niels Möller
· 5 years ago
54d5d2c
Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc
by Erik Språng
· 5 years ago
6b117a5
Make the callbacks to PollStats for RampUp* tests more regular.
by Tommi
· 5 years ago
93f5189
Remove some usage of RtpRtcp::SetSSRC()
by Erik Språng
· 5 years ago
6950b30
Fix thread naming in Call constructor
by Erik Språng
· 5 years ago
c14b233
Disable the most flaky tests on iOS.
by Sami Kalliomäki
· 5 years ago
eea605d
Make fake network degradation work also for sent audio
by Erik Språng
· 5 years ago
48b48e5
Enable thread check in Call::GetStats().
by Tommi
· 5 years ago
78a7138
Remove MediaTransport from Call.
by Tommi
· 5 years ago
891d393
Call Call::GetStats() from the correct thread in ProbingEndToEndTest.
by Tommi
· 5 years ago
5b5d97c
Reland of "Reporting of decoding_codec_plc events""
by Alex Narest
· 5 years ago
83bbe91
Delete deprecated rtc_event_log header
by Danil Chapovalov
· 5 years ago
c24a5b1
Fix CallPerfTests to call Call::GetStats() from the right thread.
by Tommi
· 5 years ago
c648819
DegradedCall: fake network using TaskQueue instead of ProcessThread
by Erik Språng
· 5 years ago
5e005f4
Fix RampUp tests to call Call::GetStats() from the right thread - and remove the need for a dedicated polling thread.
by Tommi
· 5 years ago
bedb7a8
Revert "Reporting of decoding_codec_plc events"
by Mirko Bonadei
· 5 years ago
0a88ea0
Reporting of decoding_codec_plc events
by Alex Narest
· 5 years ago
425d6aa
Add RtpPacketPacer interface for pacer control
by Erik Språng
· 5 years ago
463d44a
Don't crash when simulcast layer count is different from RID count
by Florent Castelli
· 5 years ago
aa59eca
Move RtpPacketSender and merge it with RtpPacketPacer.
by Erik Språng
· 5 years ago
e1795f4
Adds remote estimate RTCP packet.
by Sebastian Jansson
· 5 years ago
2ab97f6
Migrate WebRTC test infra to ABSL_FLAG.
by Mirko Bonadei
· 5 years ago
0182a03
Reland "Remove the injectable bitrate allocation strategy API."
by Jonas Olsson
· 5 years ago
e95b57c
Revert "Remove the injectable bitrate allocation strategy API."
by Mirko Bonadei
· 5 years ago
80cb3f6
Remove the injectable bitrate allocation strategy API.
by Jonas Olsson
· 5 years ago
a4d8737
Format almost everything.
by Jonas Olsson
· 5 years ago
d2c336f
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
by Henrik Boström
· 5 years ago
53d45ba
Make TaskQueueFactory required construction parameter for Call
by Danil Chapovalov
· 5 years ago
bfd343b
Add totalDecodeTime to RTCInboundRTPStreamStats
by Johannes Kron
· 5 years ago
67daf71
Implement RtpVideoSender::SetFecAllowed()
by Elad Alon
· 5 years ago
8f01c4e
Define FecControllerOverride and plumb it down to VideoEncoder
by Elad Alon
· 5 years ago
4ba04b7
Delete RtcEventLogFactory factory as now unused
by Danil Chapovalov
· 5 years ago
3472b9a
Delete RTCInboundRTPStreamStats::fraction_lost
by Niels Möller
· 5 years ago
59b8654
Switch from RtpPacketSender to RtpPacketPacer interface usage.
by Erik Språng
· 5 years ago
4d504c7
New interface EncodedImageBufferInterface, replacing use of CopyOnWriteBuffer
by Niels Möller
· 5 years ago
3d61ab1
Adds send time to ReceivedPacket struct.
by Sebastian Jansson
· 5 years ago
8aba8fe
Reland "Populate the GFD-00 for H264 and generic codecs."
by philipel
· 5 years ago
607a6f1
Moves conversion to ReceivedPacket from RtpPacketReceived to Call.
by Sebastian Jansson
· 5 years ago
04e129a
Revert "Populate the GFD-00 for H264 and generic codecs."
by Philip Eliasson
· 5 years ago
d3c6f9c
Populate the GFD-00 for H264 and generic codecs.
by philipel
· 5 years ago
08fa953
Reland "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
by Danil Chapovalov
· 5 years ago
370f93a
Reland "Inform VideoEncoder of negotiated capabilities"
by Elad Alon
· 5 years ago
fd5166c
Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory"
by Philip Eliasson
· 5 years ago
fc96135
Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory
by Danil Chapovalov
· 5 years ago
49d661a
Revert "Inform VideoEncoder of negotiated capabilities"
by Philip Eliasson
· 5 years ago
11dfff0
Inform VideoEncoder of negotiated capabilities
by Elad Alon
· 5 years ago
4ad51d8
Removes SendSideCongestionController.
by Sebastian Jansson
· 5 years ago
cf41eb1
Reland "Cleanup of video packet overhead calculation."
by Sebastian Jansson
· 5 years ago
5d24b16
Prepare for splitting the api/video:video_frames build rule.
by Chen Xing
· 5 years ago
27a7e9f
Remove myself from OWNERS in a few places.
by Fredrik Solenberg
· 5 years ago
f6c914a
Revert "Reland "Cleanup of video packet overhead calculation.""
by Sebastian Jansson
· 5 years ago
35d4e43
Reland "Cleanup of video packet overhead calculation."
by Sebastian Jansson
· 5 years ago
b64af4b
Add retransmission_allowed flag to encoder output
by Elad Alon
· 5 years ago
845c6aa
Add support for early loss detection using transport feedback.
by Erik Språng
· 5 years ago
6737841
Add jitterBufferDelay and jitterBufferEmittedCount stats for video
by Guido Urdaneta
· 5 years ago
ce33b6a
Implement QualityLimitationReasonTracker and expose "reason".
by Henrik Boström
· 5 years ago
8b27910
Include downlink delay into congestion window size.
by Ying Wang
· 5 years ago
6e436d1
[audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo
by Henrik Boström
· 5 years ago
87e3f9d
[video] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo
by Henrik Boström
· 5 years ago
a0e9943
Negotiation of LNTF controls instantiation of RTPSenderVideo::rtp_sequence_number_map_
by Elad Alon
· 5 years ago
fadb181
Negotiate use of RTCP loss notification feedback (LNTF)
by Elad Alon
· 5 years ago
f3db34d
Revert "Cleanup of video packet overhead calculation."
by Sebastian Jansson
· 5 years ago
62ce035
RtpVideoSender nits
by Elad Alon
· 5 years ago
890bc30
Cleanup of video packet overhead calculation.
by Sebastian Jansson
· 5 years ago
fe68daa
Add option to configure raw RTP packetization per payload type.
by Mirta Dvornicic
· 5 years ago
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