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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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2057673c11f91affd59c79aa5074ea490f0170f3
2057673
Roll chromium_revision 4029fd18b6..f2f5c1896e (639320:639430)
by chromium-webrtc-autoroll
· 6 years ago
abea6e5
Delete always-true member is_media_transport_factory_enabled_
by Niels Möller
· 6 years ago
c0c3e96
Revert "DCHECK feedback_rtt is positive"
by Yves Gerey
· 6 years ago
3368721
Revert "Change clip_name -> clip_path in VideoQualityTestFixture::Params::Video."
by Yves Gerey
· 6 years ago
32f887b
Roll chromium_revision e4e8741478..4029fd18b6 (639216:639320)
by chromium-webrtc-autoroll
· 6 years ago
fe07c42
Roll chromium_revision ecf963f5ce..e4e8741478 (639097:639216)
by chromium-webrtc-autoroll
· 6 years ago
e25f595
Guard preferred_dscp with the network interface lock
by Steve Anton
· 6 years ago
9a071d1
Roll chromium_revision e328c33c20..ecf963f5ce (638961:639097)
by chromium-webrtc-autoroll
· 6 years ago
42d8c93
Revert "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current"
by Yves Gerey
· 6 years ago
cad95b8
Reland "Tune vp9 screenshare bitrate and framerate of spatial layers"
by Ilya Nikolaevskiy
· 6 years ago
62c7b39
Allow suppression of padding check in RtpHeaderParser.
by Sebastian Jansson
· 6 years ago
44dd9f2
Adds ChannelSend specific encoder task queue.
by Sebastian Jansson
· 6 years ago
e01857c
Revert "Reland "Tune vp9 screenshare bitrate and framerate of spatial layers""
by Ilya Nikolaevskiy
· 6 years ago
12abf67
Reland "Tune vp9 screenshare bitrate and framerate of spatial layers"
by Ilya Nikolaevskiy
· 6 years ago
304e9d2
Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current
by Danil Chapovalov
· 6 years ago
37d4f91
DCHECK feedback_rtt is positive
by Evan Shrubsole
· 6 years ago
184f6d5
Change clip_name -> clip_path in VideoQualityTestFixture::Params::Video.
by Rasmus Brandt
· 6 years ago
159e53a
Fix LibvpxVp8Encoder::FrameDropThreshold
by Elad Alon
· 6 years ago
dac7aa0
Roll chromium_revision c71bb6f5f1..e328c33c20 (638607:638961)
by chromium-webrtc-autoroll
· 6 years ago
f0cbcd3
Use stdlib TaskQueue implementation in webrtc fuzzers
by Danil Chapovalov
· 6 years ago
3caf50d
Make ChangeBitrateVP9 unittest a bit more lenient.
by Yves Gerey
· 6 years ago
7f1c589
Adding new top-level directory crypto/
by Benjamin Wright
· 6 years ago
1109b59
Revert "Tune vp9 screenshare bitrate and framerate of spatial layers"
by Jeroen de Borst
· 6 years ago
2c7b982
Revert "Delete CodecSpecificInfo argument from VideoDecoder::Decode"
by Jeroen de Borst
· 6 years ago
7e70291
Fix unscoped variable in test/scenario/BUILD.gn.
by Mirko Bonadei
· 6 years ago
74350db
Roll chromium_revision 4205483be6..c71bb6f5f1 (638505:638607)
by chromium-webrtc-autoroll
· 6 years ago
aaf3cb3
Tune vp9 screenshare bitrate and framerate of spatial layers
by Ilya Nikolaevskiy
· 6 years ago
39d3a7d
Delete CodecSpecificInfo argument from VideoDecoder::Decode
by Niels Möller
· 6 years ago
1c90cab
Fix UpdateRect handling for native buffers in VideoStreamEncoder
by Ilya Nikolaevskiy
· 6 years ago
6f0aafa
Add PrintResults to VideoCodecTest.
by Rasmus Brandt
· 6 years ago
d5af402
Add overhead observers to MediaTransportInterface
by Niels Möller
· 6 years ago
06b77f9
Use min allocatable bitrate as lower bound for target bitrate.
by Sebastian Jansson
· 6 years ago
ffe9376
Bump iOS min supported version to 10.0
by Kári Tristan Helgason
· 6 years ago
b859b32
Update more VideoEncoder implementations to drop CodecSpecificInfo input
by Niels Möller
· 6 years ago
6318f13
Stop using rtc::TaskQueue::Current in RtcpTransceiver
by Danil Chapovalov
· 6 years ago
dc62ae4
Cleanup of constraints configuration in GoogCcNetworkController.
by Sebastian Jansson
· 6 years ago
78b7d49
Roll chromium_revision 1af146a0f6..4205483be6 (638325:638505)
by chromium-webrtc-autoroll
· 6 years ago
8a1e35c
Finally delete deprecated mac capturer.
by Kári Tristan Helgason
· 6 years ago
87e2d78
Prepare for splitting FrameType into AudioFrameType and VideoFrameType
by Niels Möller
· 6 years ago
0b69826
Don't inject worker queue into send streams.
by Sebastian Jansson
· 6 years ago
de3360e
Create Vp8FrameBufferController
by Elad Alon
· 6 years ago
610c763
Add target bitrate headroom signal to VideoStreamEncoder.
by Erik Språng
· 6 years ago
e49d64e
Roll chromium_revision 3eb6e6ce76..1af146a0f6 (638159:638325)
by chromium-webrtc-autoroll
· 6 years ago
7276b97
Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC.
by Benjamin Wright
· 6 years ago
4423c36
Migrate RepeatingTask to take raw pointer to TaskQueueBase instead of TaskQueue
by Danil Chapovalov
· 6 years ago
11e55ee
Renaming min_pacing_rate to min_total_allocated_bitrate.
by Sebastian Jansson
· 6 years ago
7b41225
Throttle frame-rate In VP8 encoder in steady state for screenshare
by Ilya Nikolaevskiy
· 6 years ago
2ecc8c8
Roll chromium_revision 99baeeafe2..3eb6e6ce76 (638035:638159)
by chromium-webrtc-autoroll
· 6 years ago
8672cac
Trigger audio bitrate allocation update on overhead change.
by Sebastian Jansson
· 6 years ago
ee5ccbc
Move ownership of RTPSenderAudio to ChannelSend.
by Niels Möller
· 6 years ago
232b3fd
Expose relative packet arrival delay metric in stats API.
by Jakob Ivarsson
· 6 years ago
67f862e
Guard against calls to OnEncodedFrame after Release.
by Sami Kalliomäki
· 6 years ago
6117068
Throttle frame-rate In VP9 encoder in steady state for screenshare
by Ilya Nikolaevskiy
· 6 years ago
0cb858c
New VCMPacket constructor without WebRtcRTPHeader argument
by Niels Möller
· 6 years ago
7bc331f
Android: Use android_deps directly
by Peter Wen
· 6 years ago
c44f6cc
Modernize RtpRtcp factory function: use unique_ptr as return type
by Danil Chapovalov
· 6 years ago
ede7cb2
Rewrite video_loopback to use new mac capturer.
by Kári Tristan Helgason
· 6 years ago
c8d2e73
Delete CodecSpecificInfo argument from VideoEncoder::Encode
by Niels Möller
· 6 years ago
1e42761
Removes verbose extension warning in Scenario tests.
by Sebastian Jansson
· 6 years ago
110c64b
Delete unused key WebRTC-Audio-SendSideBwe-For-Video.
by Christoffer Rodbro
· 6 years ago
745cfb9
use link_deps in ana_debug_dump_proto
by Takuto Ikuta
· 6 years ago
d738071
Refactor FakeEncoder to avoid writing to a const EncodedImage
by Niels Möller
· 6 years ago
9df3353
Generic Frame Descriptor (GFD) VP8 templates.
by philipel
· 6 years ago
8fb1a6a
Delete a few return values from audio streams and video send streams.
by Niels Möller
· 6 years ago
0da25a1
Update TransportSequenceNumberV2 extension to support fixed size
by Johannes Kron
· 6 years ago
f441ea9
Minor cleanup of probe_controller.cc
by Jonas Olsson
· 6 years ago
200feba
Make AEC3 the default desktop AEC option in WebRTC
by Per Åhgren
· 6 years ago
359c9b9
Roll chromium_revision 49f30ad2d0..99baeeafe2 (637641:638035)
by chromium-webrtc-autoroll
· 6 years ago
be7af93
Add dsymutil as a mac cipd dependency.
by Yves Gerey
· 6 years ago
b443dfe
Use metrics::Samples in a couple pc/ tests
by Steve Anton
· 6 years ago
e2a284d
Adding metrics to measure usage of simulcast API.
by Amit Hilbuch
· 6 years ago
4eb5c14
Reland "Remove field trial include from decision logic."
by Jakob Ivarsson
· 6 years ago
07a4f2b
Merge rtc_task_queue(_api|_impl)? build targets into one
by Danil Chapovalov
· 6 years ago
4450708
Add relative_packet_arrival_delay and jitter_buffer_packets_received statistics.
by Jakob Ivarsson
· 6 years ago
1aa7581
Replace all usage of rtc::NewClosure with webrtc::ToQueuedTask
by Danil Chapovalov
· 6 years ago
c1e6e86
Add metrics::Samples to facilitate easier testing
by Steve Anton
· 6 years ago
d36c086
Add support for simulcast streams in QualityAnalyzingVideoDecoder.
by Artem Titov
· 6 years ago
6ec2f54
Fix mis-spelled TODO items
by Niels Möller
· 6 years ago
7949f21
Revert "Removes lock from ChannelSend."
by Sebastian Jansson
· 6 years ago
9ef5e05
Fix target bitrate handling for a single layer VP9 screenshare
by Ilya Nikolaevskiy
· 6 years ago
977b335
Injecting Clock into audio streams.
by Sebastian Jansson
· 6 years ago
f23f161
Roll chromium_revision 5fe3fd14f6..49f30ad2d0 (637538:637641)
by chromium-webrtc-autoroll
· 6 years ago
423bae4
Revert "Remove field trial include from decision logic."
by Jakob Ivarsson
· 6 years ago
bd7ed4b
Include sign for infinity in ToString for data units.
by Sebastian Jansson
· 6 years ago
44ce4b4
Adding a placeholder audio_buffer build target
by Per Åhgren
· 6 years ago
15845af
Reland "Another mock for GetSctpTransport" (and add test)
by Harald Alvestrand
· 6 years ago
9b93447
Removes lock from ChannelSend.
by Sebastian Jansson
· 6 years ago
c8eeb18
Fixes parsing of logs where receive time info is missing.
by Sebastian Jansson
· 6 years ago
07316a6
Roll chromium_revision 4229a4b64d..5fe3fd14f6 (637436:637538)
by chromium-webrtc-autoroll
· 6 years ago
1f8e445
Roll chromium_revision 5afa522447..4229a4b64d (637301:637436)
by chromium-webrtc-autoroll
· 6 years ago
572c60f
Injecting Clock into video senders.
by Sebastian Jansson
· 6 years ago
8026d60
Injecting Clock in video receive.
by Sebastian Jansson
· 6 years ago
ef50b25
Remove lock in WebRtcVideoEngine
by Steve Anton
· 6 years ago
4cde9ad
Fix some typos found in ivf_file_writer.cc
by Elad Alon
· 6 years ago
4e5f5ed
Allow Clock injection in Call.
by Sebastian Jansson
· 6 years ago
5fe9510
Move ownership of RTPSenderVideo one more level up, to RtpVideoSender
by Niels Möller
· 6 years ago
ac6cf7f
Roll chromium_revision e65d7afd91..5afa522447 (637200:637301)
by chromium-webrtc-autoroll
· 6 years ago
da6806c
Injecting Clock into BitrateAllocator.
by Sebastian Jansson
· 6 years ago
d0f3d84
Wire UpdateRect signalling in test frame generators
by Ilya Nikolaevskiy
· 6 years ago
acd8ae7
Reinstate old iceConnectionState "completed" behavior
by Jonas Olsson
· 6 years ago
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