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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
2bc1ea0b36528168dc9a582b26956fee002e0e49
/
media
0bad15f
Remove the noise_suppression() pointer to submodule interface
by saza
· 5 years ago
8038541
Update the header extensions capabilities with mid, rid and rrid
by Florent Castelli
· 5 years ago
ac0a4cb
Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
by Niels Möller
· 5 years ago
41478c7
Remove AudioProcessing::gain_control() getter
by Sam Zackrisson
· 5 years ago
35214fc
Add missing RTC_EXPORT for the component build.
by Mirko Bonadei
· 5 years ago
ef0627f
Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
by Mirko Bonadei
· 5 years ago
fbde32e
Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
by Niels Möller
· 5 years ago
80f53b7
Extend WebRTC-Video-MinVideoBitrate to experiment per-codec
by Elad Alon
· 5 years ago
5740f3e
Clarify expectation on GlobalLock
by Danil Chapovalov
· 5 years ago
ff27da5
Add/remove receive streams with SSRC 0 from media channels
by Saurav Das
· 5 years ago
f4e0c29
SimulcastEncoderAdapter: support per layer fallback and single encoder proxying
by Erik Språng
· 5 years ago
9d7eb28
Don't limit simulcast layers number for screenshare based on resolution
by Ilya Nikolaevskiy
· 5 years ago
09f1195
Always pass arguments to INSTANTIATE_TEST_SUITE_P.
by Mirko Bonadei
· 5 years ago
27b0e0d
Remove obsolete todo comment in simulcast.h
by Åsa Persson
· 5 years ago
e942b14
New build target api:media_interface
by Niels Möller
· 5 years ago
1b83a9e
Only handle each RTCP once.
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
317a1f0
Use std::make_unique instead of absl::make_unique.
by Mirko Bonadei
· 5 years ago
d9cc8c0
Encoder switching based on network and/or resolution conditions.
by philipel
· 5 years ago
73ceed5
Update simulcast bitrate calculations for non-standard resolutions.
by Ilya Nikolaevskiy
· 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
70dd165
Delete CoreAudio include from media_engine.h
by Niels Möller
· 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
cc62b16
Add qualityLimitationResolutionChanges stat
by Evan Shrubsole
· 5 years ago
0bd2eff
Reland "New build target p2p:stun_types"
by Niels Möller
· 5 years ago
91c824f
Revert "New build target p2p:stun_types"
by Hannes Landeholm
· 5 years ago
66d6c3b
Buffers non atomic message send with usrsctp lib.
by Seth Hampson
· 5 years ago
8c5520c
Reland "Make the min video bitrate in VideoSendStream configurable."
by Ying Wang
· 5 years ago
1d2149c
Revert "Make the min video bitrate in VideoSendStream configurable."
by Alessio Bazzica
· 5 years ago
b2fb0b9
Make the min video bitrate in VideoSendStream configurable.
by Ying Wang
· 5 years ago
a837030
Split out RtpSource from libjingle_peerconnection_api
by Niels Möller
· 5 years ago
5b4fcb5
New build target p2p:stun_types
by Niels Möller
· 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
4271afb
Fix the bug and reland "Make min video target bitrate configurable."
by Ying Wang
· 5 years ago
0c141c5
Fix frames dropped statistics
by Johannes Kron
· 5 years ago
7e896d0
Revert "Make min video target bitrate configurable."
by Mirko Bonadei
· 5 years ago
a471e79
Make min video target bitrate configurable.
by Ying Wang
· 5 years ago
d77cc24
New const method StreamStatistician::GetStats
by Niels Möller
· 5 years ago
224c69d
Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo
by Niels Möller
· 5 years ago
b689af4
Changes to enable use of DatagramTransport as a data channel transport.
by Bjorn A Mellem
· 5 years ago
70efdde
Set local ssrc at construction of Rtp module
by Erik Språng
· 5 years ago
587991c
Remove jeroendb@webrtc.org from OWNERS
by Steve Anton
· 5 years ago
6b43086
Reland "[GetStats] Expose video codec implementation in standardized metrics."
by Henrik Boström
· 5 years ago
df625f4
Revert "[GetStats] Expose video codec implementation in standardized metrics."
by Henrik Andreassson
· 5 years ago
2b9fa09
[GetStats] Expose video codec implementation in standardized metrics.
by Henrik Boström
· 5 years ago
bbeb109
Reporting audio device underrun counter
by Alex Narest
· 5 years ago
71c6b56
Allow sending abs-send-time for audio streams.
by Sebastian Jansson
· 5 years ago
78a7138
Remove MediaTransport from Call.
by Tommi
· 5 years ago
5b5d97c
Reland of "Reporting of decoding_codec_plc events""
by Alex Narest
· 5 years ago
2d2bbb1
Filter out duplicate receive codecs in the media engine
by Steve Anton
· 5 years ago
83bbe91
Delete deprecated rtc_event_log header
by Danil Chapovalov
· 5 years ago
f40a340
Remove deprecated code related to AEC2
by Per Åhgren
· 5 years ago
d2845f8
Removes unused AudioAllocationSettings from voice engine.
by Sebastian Jansson
· 5 years ago
9b1700c
Enable field trial LegacySimulcastLayerLimit by default
by Florent Castelli
· 5 years ago
d7ee76c
Wire up field trials for some experimental screenshare settings
by Erik Språng
· 5 years ago
8bbdb5b
Update VideoBitrateAllocator allocate to take a struct with more fields
by Florent Castelli
· 5 years ago
da4f093
Reland "Only include payload in bytes sent/received."
by Bjorn A Mellem
· 5 years ago
bedb7a8
Revert "Reporting of decoding_codec_plc events"
by Mirko Bonadei
· 5 years ago
bcd068d
Revert "Only include payload in bytes sent/received."
by Bjorn Mellem
· 5 years ago
0a88ea0
Reporting of decoding_codec_plc events
by Alex Narest
· 5 years ago
a9fbb22
Add a field trial for older applications to reduce the simulcast layer count
by Florent Castelli
· 5 years ago
e1795f4
Adds remote estimate RTCP packet.
by Sebastian Jansson
· 5 years ago
74a1b4b
Only include payload in bytes sent/received.
by Bjorn A Mellem
· 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
0bb0881
Add VideoEncoderFactory::GetImplementations function.
by philipel
· 5 years ago
66b3860
Remove WebRTC-SimulcastScreenshare and enable it by default
by Florent Castelli
· 5 years ago
41300af
Poison default task queue factory
by Danil Chapovalov
· 5 years ago
80cb3f6
Remove the injectable bitrate allocation strategy API.
by Jonas Olsson
· 5 years ago
495a1ae
Remove cricket::WebRtcMediaEngineFactory as now unused
by Danil Chapovalov
· 5 years ago
a4d8737
Format almost everything.
by Jonas Olsson
· 5 years ago
668ce0c
Remove trial WebRTC-SimulcastMaxLayers and make its behavior default
by Florent Castelli
· 5 years ago
fdf74bd
Remove non implemented function from WebRtcVideoChannel.
by philipel
· 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
e8ed830
WebRtcVideoChannel encoder fallback.
by philipel
· 5 years ago
5ee6967
Don't reset encoder on max/min bitrate change.
by Sergey Silkin
· 5 years ago
bfd343b
Add totalDecodeTime to RTCInboundRTPStreamStats
by Johannes Kron
· 5 years ago
5983087
Forced vp8 sw encoder fallback: only use min bitrate config if codec type is vp8.
by Åsa Persson
· 5 years ago
65764e4
Add missing overrides in VideoEncoder proxies/adapters
by Elad Alon
· 5 years ago
8f01c4e
Define FecControllerOverride and plumb it down to VideoEncoder
by Elad Alon
· 5 years ago
2efae77
Add RTCStats for keyFramesEncoded, keyFramesDecoded.
by Rasmus Brandt
· 5 years ago
4ba04b7
Delete RtcEventLogFactory factory as now unused
by Danil Chapovalov
· 5 years ago
90f3b89
Replace the implementation of `GetContributingSources()` on the video side.
by Chen Xing
· 5 years ago
3472b9a
Delete RTCInboundRTPStreamStats::fraction_lost
by Niels Möller
· 5 years ago
c538506
Enable H.264 temporal scalability in simulcast.
by Johnny Lee
· 5 years ago
f00bf42
Add plumbing of RtpPacketInfos to each VideoFrame as input for SourceTracker.
by Chen Xing
· 5 years ago
98cbb22
Moved AsyncInvoker to be destructed first in WebRtcVideoSendStream.
by philipel
· 5 years ago
6e9c2fd
Delete StartRtcEventLog and StopRtcEventLog methods from FakeVoiceEngine
by Niels Möller
· 5 years ago
a9952cb
Uncomment "override" in simulcast_encoder_adapter_unittest.cc
by Elad Alon
· 5 years ago
370f93a
Reland "Inform VideoEncoder of negotiated capabilities"
by Elad Alon
· 5 years ago
e8e4dc4
Change StartAecDump methods to work with FILE* and FileWrapper
by Niels Möller
· 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
5d24b16
Prepare for splitting the api/video:video_frames build rule.
by Chen Xing
· 5 years ago
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