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73cf80a932e82f5fcc54347f00dd9a256fc44888
73cf80a
Fixes incorrect feedback to EncoderBitrateAdjuster [perf note]
by Erik Språng
· 4 years ago
ef036cd
[Stats] Cleanup obsolete stats - isRemote & deleted
by Di Wu
· 4 years ago
56db9ff
VideoStreamEncoder: Don't map kNative video frame buffers.
by Henrik Boström
· 4 years ago
5cf8c2c
Fix unspecified time origin for `lastPacketReceivedTimestamp`
by Alessio Bazzica
· 4 years ago
9054aa8
Update WebRTC code version (2021-03-24T04:02:05).
by webrtc-version-updater
· 4 years ago
f7b1b95
Add `RTCRemoteOutboundRtpStreamStats` for audio streams
by Alessio Bazzica
· 4 years ago
26abdaf
AV1: Use Default TX type for encoding
by Fyodor Kyslov
· 4 years ago
2f71b61
Make sure "remote-inbound-rtp.jitter" and "packetsLost" is exposed to JS
by Henrik Boström
· 4 years ago
ca18809
Move RtpFrameObject and EncodedFrame out of video_coding namespace.
by philipel
· 4 years ago
93ee168
Allow empty video layer allocation extension
by Jonas Oreland
· 4 years ago
fa4db49
Make GL errors thrown by checkNoGLES2Error inherit GLException.
by Sami Kalliomäki
· 4 years ago
7cbe887
Change default adaptive ptime min bitrate to 16kbps.
by Jakob Ivarsson
· 4 years ago
f0adf38
Fix timestamps for the remote outbound audio stream stats
by Alessio Bazzica
· 4 years ago
3889de1
Support native scaling of VideoFrameBuffers in LibvpxVp8Encoder.
by Henrik Boström
· 4 years ago
6a67150
Move RtpFrameReferenceFinder out of video_coding namespace.
by philipel
· 4 years ago
2ba32f3
Delete AsyncInvoker usage in TurnServer
by Niels Möller
· 4 years ago
a9ba450
stats: add address as alias for ip
by Philipp Hancke
· 4 years ago
e2ac591
Update WebRTC code version (2021-03-23T04:03:37).
by webrtc-version-updater
· 4 years ago
1cdbabd
Update WgcCaptureSession to handle portrait oriented screen capture.
by Austin Orion
· 4 years ago
e0059dc
Roll chromium_revision b7f0a0c111..c0436807ae (865116:865247)
by chromium-webrtc-autoroll
· 4 years ago
c366d51
Fix unit for inbound RTP stat `lastPacketReceivedTimestamp` (s -> ms)
by Alessio Bazzica
· 4 years ago
c303f82
Add new owners for sdk/android.
by Sami Kalliomäki
· 4 years ago
464bcd4
Revert "Reland "[Battery]: Delay start of TaskQueuePacedSender.""
by Etienne Pierre-Doray
· 4 years ago
efad89c
Roll chromium_revision 23141e38f1..b7f0a0c111 (864556:865116)
by chromium-webrtc-autoroll
· 4 years ago
bd9e4a9
Support native scaling of VideoFrameBuffers in LibvpxVp9Encoder.
by Henrik Boström
· 4 years ago
2ff25db
Update apply-iwyu tool to report compile errors
by Harald Alvestrand
· 4 years ago
ffb7603
Delete TurnPort usage of AsyncInvoker
by Niels Möller
· 4 years ago
eb28298
Update rsid and mid spec links from draft to release version
by Danil Chapovalov
· 4 years ago
47350c2
Reland "Triggering CI."
by Mirko Bonadei
· 4 years ago
f412976
Provide a default implementation of NV12BufferInterface::CropAndScale.
by Henrik Boström
· 4 years ago
50d79ba
Revert "Triggering CI."
by Mirko Bonadei
· 4 years ago
0a104c4
Delete obsolete method EncodedImage::Retain()
by Niels Möller
· 4 years ago
6f7e205
Delete AsyncInvoker usage from StunProber
by Niels Möller
· 4 years ago
ebd2010
Check if ifa_addr field is null.
by Björn Terelius
· 4 years ago
76b51e2
Improve thread annotations for TurnServer
by Niels Möller
· 4 years ago
2f5f5fa
standalone ice transport: dont use component 0
by Philipp Hancke
· 4 years ago
c732576
Triggering CI.
by Mirko Bonadei
· 4 years ago
8bf1cd1
Rename (packets|bytes)_dropped to (packets|bytes)_discarded_no_receiver
by Artem Titov
· 4 years ago
eecc4f5
Fix: when SamplesStatsCounter is empty it's not propagated to the Histogram perf output
by Artem Titov
· 4 years ago
2bab0ef
Update WebRTC code version (2021-03-20T04:03:21).
by webrtc-version-updater
· 4 years ago
490c150
Delete unowned buffer in EncodedImage.
by Niels Möller
· 4 years ago
2b250734
Delete FakeIceTransport usage of AsyncInvoker
by Niels Möller
· 4 years ago
ef7d61e
Update WebRTC code version (2021-03-19T04:04:06).
by webrtc-version-updater
· 4 years ago
ba3e6c2
Roll chromium_revision 5f1d8e0c95..23141e38f1 (864439:864556)
by chromium-webrtc-autoroll
· 4 years ago
18b0947
Roll chromium_revision 6bb9b62b86..5f1d8e0c95 (864335:864439)
by chromium-webrtc-autoroll
· 4 years ago
3328761
Roll chromium_revision 74fb21b370..6bb9b62b86 (864218:864335)
by chromium-webrtc-autoroll
· 4 years ago
d6d2a29
Roll chromium_revision a1e978b5ab..74fb21b370 (864105:864218)
by chromium-webrtc-autoroll
· 4 years ago
c780605
Make num_encoded_channels_ atomic
by Gustaf Ullberg
· 4 years ago
049e611
Add missing EXPECT_CALL for `RTCStatsCollectorTest` tests
by Alessio Bazzica
· 4 years ago
0848994
Replace AsyncInvoker with PostDelayedTask, in DtmfSender
by Niels Möller
· 4 years ago
9243088
Add thread annotations to FakeIceTransport
by Niels Möller
· 4 years ago
0357477
Delete use of AsyncInvoker from FakePacketTransport
by Niels Möller
· 4 years ago
cbadb8b
Expose offerExtmapAllowMixed in iOS SDK.
by Yura Yaroshevich
· 4 years ago
426d679
Update WebRTC code version (2021-03-18T04:03:50).
by webrtc-version-updater
· 4 years ago
b9a6c03
Roll chromium_revision 345f2fb2f5..a1e978b5ab (863976:864105)
by chromium-webrtc-autoroll
· 4 years ago
5a23b3d
Roll chromium_revision ca2293bd17..345f2fb2f5 (863854:863976)
by chromium-webrtc-autoroll
· 4 years ago
f8776cb
Revert "AV1: Use Default TX type for encoding"
by Fyodor Kyslov
· 4 years ago
db01a82
Roll chromium_revision 5371070da9..ca2293bd17 (863745:863854)
by chromium-webrtc-autoroll
· 4 years ago
b0dc518
AV1: Use Default TX type for encoding
by Fyodor Kyslov
· 4 years ago
dd4d5e3
Reland "[Battery]: Delay start of TaskQueuePacedSender."
by Etienne Pierre-doray
· 4 years ago
32af25b
Disable more flaky PeerConnectionIntegrationTests on Windows
by Rasmus Brandt
· 4 years ago
cf93670
sctp: Finish sending partial messages before sending stream reset events
by Florent Castelli
· 4 years ago
92a768a
Roll chromium_revision 47b94319df..5371070da9 (863625:863745)
by chromium-webrtc-autoroll
· 4 years ago
4173614
Expose enableImplicitRollback in iOS SDK.
by Yura Yaroshevich
· 4 years ago
1827483
Update WebRTC code version (2021-03-17T04:02:27).
by webrtc-version-updater
· 4 years ago
2ba6435
Roll chromium_revision 7f72620d8b..47b94319df (863490:863625)
by chromium-webrtc-autoroll
· 4 years ago
861a0d1
Roll chromium_revision 67d2a9f799..7f72620d8b (863354:863490)
by chromium-webrtc-autoroll
· 4 years ago
f19aec8
Updates ulpfec reader to accept padding on media packets.
by Erik Språng
· 4 years ago
87dbe9a
Roll chromium_revision 432f33c810..67d2a9f799 (863160:863354)
by chromium-webrtc-autoroll
· 4 years ago
d19e3b9
Reland "Reland "Enable quality scaling when allowed""
by Sergey Silkin
· 4 years ago
e37fa19
Delete unused class DummyDtmfObserver
by Niels Möller
· 4 years ago
ab63350
Delete RtpRtcp::RemoteRTCPStat in favor of GetLatestReportBlockData
by Danil Chapovalov
· 4 years ago
19775cb
Reland "Reduce complexity in the APM pipeline when the output is not used"
by Per Åhgren
· 4 years ago
15179a9
Allowing reduced computations in the noise suppressor when the output is not used
by Per Åhgren
· 4 years ago
8ee1ec8
Allowing reduced computations in the AEC3 when the output is not used
by Per Åhgren
· 4 years ago
3e774f6
Make AndroidNetworkMonitor::Start() create a new task safety flag
by Niels Möller
· 4 years ago
a776f51
Avoid two consecutive version updates.
by Mirko Bonadei
· 4 years ago
9d1e070
Increase wait-for-lost-packet from 10 to 100 msec in MTU test
by Harald Alvestrand
· 4 years ago
596ba4c
Roll chromium_revision 0b0b620d02..432f33c810 (863050:863160)
by chromium-webrtc-autoroll
· 4 years ago
785e23b
Drop # of video tracks in renegotiate-many-videos to 8
by Harald Alvestrand
· 4 years ago
0855302
Update WebRTC code version (2021-03-16T04:03:07).
by webrtc-version-updater
· 4 years ago
f172706
Roll chromium_revision d935055b21..0b0b620d02 (862883:863050)
by chromium-webrtc-autoroll
· 4 years ago
bff6489
AV1: Disable several intra coding tools.
by Fyodor Kyslov
· 4 years ago
995c5c8
Roll chromium_revision e4fd023c85..d935055b21 (862756:862883)
by chromium-webrtc-autoroll
· 4 years ago
db5d728
Add refined handling of the internal scaling of the audio in APM
by Per Åhgren
· 4 years ago
b315951
Remove incorrect DCHECKs from LibaomAv1Encoder::SetRates.
by philipel
· 4 years ago
fdd6099
Rework transient suppressor configuration in audioproc_f
by Gustaf Ullberg
· 4 years ago
685be14
Disable flaky AddMediaToConnectedBundleDoesNotRestartIce on tsan
by Rasmus Brandt
· 4 years ago
e657d87
Allow port 53 as a TURN port.
by Harald Alvestrand
· 4 years ago
c88bdad
Roll chromium_revision c3fb27225e..e4fd023c85 (861941:862756)
by chromium-webrtc-autoroll
· 4 years ago
6ca955a
Reland "Fix problem with ipv4 over ipv6 on Android"
by Jonas Oreland
· 4 years ago
7087b83
Test that SCTP succeeds with one MTU and fails with a lower MTU
by Harald Alvestrand
· 4 years ago
0e42cf7
Reland "Parse encoded frame QP if not provided by encoder"
by Sergey Silkin
· 4 years ago
b6bc357
turn: add logging for long usernames
by Philipp Hancke
· 4 years ago
6097b0f
Delete use of AsyncInvoker from PeerConnectionIntegrationWrapper
by Niels Möller
· 4 years ago
13118a7
Update WebRTC code version (2021-03-15T04:05:00).
by webrtc-version-updater
· 4 years ago
55bc077
Add one frame (10 ms) of silence in APM output after unmuting
by Per Åhgren
· 4 years ago
1e60490
Revert "Fix problem with ipv4 over ipv6 on Android"
by Taylor Brandstetter
· 4 years ago
bc1c93d
Add remote-outbound stats for audio streams
by Alessio Bazzica
· 4 years ago
c80f955
Avoid log spam when decoder implementation changes
by Erik Språng
· 4 years ago
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