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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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8e032c8b7aa0474e6e25c44e76257ce5927dbf7a
8e032c8
Switch psnr_ssim_analyzer to ABSL_FLAG.
by Mirko Bonadei
· 5 years ago
2308bb2
Switch rgba_to_i420_converter to ABSL_FLAG.
by Mirko Bonadei
· 5 years ago
6bafc8b
Reland "Switch frame_analyzer to ABSL_FLAG."
by Mirko Bonadei
· 5 years ago
90c6d1b
Revert "Switch frame_analyzer to ABSL_FLAG."
by Mirko Bonadei
· 5 years ago
2923d76
Roll chromium_revision c049034cc7..6c5bbd86c3 (671848:671957)
by chromium-webrtc-autoroll
· 5 years ago
3668e9b
Switch frame_analyzer to ABSL_FLAG.
by Mirko Bonadei
· 5 years ago
fa1e7c1
Roll chromium_revision d6c73f3029..c049034cc7 (671737:671848)
by chromium-webrtc-autoroll
· 5 years ago
f4996d4
Roll chromium_revision 163d6b201e..d6c73f3029 (671624:671737)
by chromium-webrtc-autoroll
· 5 years ago
e9cac4f
Improvements to scenario video stats for scenario tests.
by Sebastian Jansson
· 5 years ago
8605fbf
[getStats] Make remote-inbound-rtp.ssrc match outbound-rtp.ssrc.
by Henrik Boström
· 5 years ago
8f7b41a
Roll chromium_revision ec99dbf1aa..163d6b201e (671513:671624)
by chromium-webrtc-autoroll
· 5 years ago
4b8a5b4
Removes unused PacketFeedbackComparator
by Sebastian Jansson
· 5 years ago
f48bca7
Avoid triggering a false error logging when using encryptor and sending DTX.
by Minyue Li
· 5 years ago
59b8654
Switch from RtpPacketSender to RtpPacketPacer interface usage.
by Erik Språng
· 5 years ago
13eb764
Move towards always using packet type instead of priority in RTPSender
by Erik Språng
· 5 years ago
1231419
BalancedDegradationSettings: Add option to configure QP thresholds.
by Åsa Persson
· 5 years ago
88290ae
Reland "Cleanup of RTP references in GoogCC implementation."
by Sebastian Jansson
· 5 years ago
746dd0d
Add support for two-byte RTP header extensions to UsedRtpHeaderExtensionIds
by Johannes Kron
· 5 years ago
2ffcd82
Make DefaultTemporalLayers explicitly request a key frame
by Elad Alon
· 5 years ago
9aa870a
Fixing fuzzer by backing up and restoring `packet_info`.
by Chen Xing
· 5 years ago
214f543
Make useful padding the default.
by Erik Språng
· 5 years ago
2b1d8c8
Roll chromium_revision b827eec031..ec99dbf1aa (671413:671513)
by chromium-webrtc-autoroll
· 5 years ago
4a15833
Roll chromium_revision a8f7a25b6e..b827eec031 (671303:671413)
by chromium-webrtc-autoroll
· 5 years ago
3e3a900
Roll chromium_revision ffeecb2d27..a8f7a25b6e (671198:671303)
by chromium-webrtc-autoroll
· 5 years ago
abf7eba
Roll chromium_revision ef202aa763..ffeecb2d27 (671098:671198)
by chromium-webrtc-autoroll
· 5 years ago
6fd5b9e
Roll chromium_revision 568fb40139..ef202aa763 (670985:671098)
by chromium-webrtc-autoroll
· 5 years ago
6e91646
Roll chromium_revision 5cf576bfbe..568fb40139 (670869:670985)
by chromium-webrtc-autoroll
· 5 years ago
6924d50
Roll chromium_revision 4dfb50605a..5cf576bfbe (670751:670869)
by chromium-webrtc-autoroll
· 5 years ago
d487a55
Revert "Remove sync buffer length from FilteredCurrentDelayMs."
by Jakob Ivarsson
· 5 years ago
a550dad
Minor rtc_event_log_impl cleanup.
by Nikita Zetilov
· 5 years ago
6c42d92
Added video_coding::EncodedFrame copy ctor.
by philipel
· 5 years ago
f00bf42
Add plumbing of RtpPacketInfos to each VideoFrame as input for SourceTracker.
by Chen Xing
· 5 years ago
7953ad5
Revert "Cleanup of RTP references in GoogCC implementation."
by Sebastian Jansson
· 5 years ago
fa79081
Cleanup of RTP references in GoogCC implementation.
by Sebastian Jansson
· 5 years ago
775c02e
Do not use libevent when targeting wasm.
by Mirko Bonadei
· 5 years ago
9da25bd
In PeerConnection unittests set TaskQueueFactory explicitly
by Danil Chapovalov
· 5 years ago
621be83
Roll chromium_revision 516b926bdf..4dfb50605a (670612:670751)
by chromium-webrtc-autoroll
· 5 years ago
e3cc489
Add logging and edit the field trial name for piggyback ICE check
by Qingsi Wang
· 5 years ago
0d92f15
Roll chromium_revision 39e46dcf9e..516b926bdf (670459:670612)
by chromium-webrtc-autoroll
· 5 years ago
5bf5806
Force DefaultTaskQueueFactory in legacy CreatePeerConnectionFactory
by Danil Chapovalov
· 5 years ago
c2a54dc
Roll chromium_revision 6e638caa96..39e46dcf9e (670359:670459)
by chromium-webrtc-autoroll
· 5 years ago
c16289f
Split the build rule for video_frame into a video_rtp_headers part.
by Chen Xing
· 5 years ago
b64ad0e
Using Clock::CurrentTime() where non-test behavior is unchanged.
by Sebastian Jansson
· 5 years ago
18f1f0c
Revert "Raise IllegalStateException for calls to retain() or release() on zero ref count"
by Niels Moller
· 5 years ago
8a959bf
Raise IllegalStateException for calls to retain() or release() on zero ref count
by Niels Möller
· 5 years ago
505bac2
Add default implementation of deprecated StartAecDump method.
by Niels Möller
· 5 years ago
4d504c7
New interface EncodedImageBufferInterface, replacing use of CopyOnWriteBuffer
by Niels Möller
· 5 years ago
0894f0f
Add piggyback acknowledgement of the last ICE check received in outgoing checks.
by Qingsi Wang
· 5 years ago
f9511fc
Roll chromium_revision 619b073720..6e638caa96 (670258:670359)
by chromium-webrtc-autoroll
· 5 years ago
b7a3e3f
Roll chromium_revision 8639acbee7..619b073720 (670130:670258)
by chromium-webrtc-autoroll
· 5 years ago
e8347a8
Roll chromium_revision bf62d746a4..8639acbee7 (669828:670130)
by chromium-webrtc-autoroll
· 5 years ago
1b3f4f9
Allow RtpPacketHistory encapsulator function to abort retransmit
by Erik Språng
· 5 years ago
4cbb4ef
Roll chromium_revision 6ae0f0cd4c..bf62d746a4 (669703:669828) + fix AndroidManifest
by Oleh Prypin
· 5 years ago
b762b5b
Fix potential signed overflow in IntervalBudget::set_target_rate_kbps
by Per Kjellander
· 5 years ago
342f98b
Fixes for flexfec crash in scenario tests.
by Sebastian Jansson
· 5 years ago
58ee187
Add support within PacedSender and pacer queue for owning rtp packets.
by Erik Språng
· 5 years ago
b028c6a
Support __EMSCRIPTEN__ in rtc_base.
by Mirko Bonadei
· 5 years ago
0c0c969
Add/rewrite H264 VUI video signal type description.
by Sergey Silkin
· 5 years ago
449888e
Cleanup of resources from removed remote bitrate estimate test framework.
by Sebastian Jansson
· 5 years ago
2c648f5
Stop running 'bwe_simulations_tests'.
by Mirko Bonadei
· 5 years ago
e181440
Fix documentation in BitrateAdjuster.
by Sami Kalliomäki
· 5 years ago
0f557fe
Removes unused dependency on RTP/RTCP from loss based controller.
by Sebastian Jansson
· 5 years ago
61d8ee1
Roll chromium_revision 2ce8c83798..6ae0f0cd4c (669595:669703)
by chromium-webrtc-autoroll
· 5 years ago
23026ee
Adds SortedByReceiveTime to TransportPacketsFeedback.
by Sebastian Jansson
· 5 years ago
873a7a9
Fix event_log_visualizer help text and default profile.
by Bjorn Terelius
· 5 years ago
9c771c2
Add TrySendPacket() method to RTP modules.
by Erik Språng
· 5 years ago
d9c900f
Add Clone() to Vp8FrameBufferControllerFactory
by Elad Alon
· 5 years ago
1dee91a
Roll chromium_revision d958b08217..2ce8c83798 (669491:669595)
by chromium-webrtc-autoroll
· 5 years ago
e93454a
Removes AddAndRemoveOld from SendTimeHistory
by Sebastian Jansson
· 5 years ago
7b99fa8
Roll chromium_revision 447b80d261..d958b08217 (669391:669491)
by chromium-webrtc-autoroll
· 5 years ago
2f42dab
Roll chromium_revision e5d3b43486..447b80d261 (669285:669391)
by chromium-webrtc-autoroll
· 5 years ago
ef5e35d
Roll chromium_revision be90998a40..e5d3b43486 (669184:669285)
by chromium-webrtc-autoroll
· 5 years ago
e112bb8
Adds support for abs send time extension in scenario tests.
by Sebastian Jansson
· 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
bdb6b39
Let HardwareVideoEncoder cache result from codec.getOutputBuffers()
by Niels Möller
· 5 years ago
5e953d7
Insert startcodes for H264 single NALU packets.
by philipel
· 5 years ago
856ca19
Delete unused method ReportBlockStats::AggregateAndStore
by Niels Möller
· 5 years ago
0292e97
Roll chromium_revision f165a6d5de..be90998a40 (669076:669184)
by chromium-webrtc-autoroll
· 5 years ago
607a6f1
Moves conversion to ReceivedPacket from RtpPacketReceived to Call.
by Sebastian Jansson
· 5 years ago
98cbb22
Moved AsyncInvoker to be destructed first in WebRtcVideoSendStream.
by philipel
· 5 years ago
7f2a67f
Cleanup names and extra checks for errors in PC test framework
by Artem Titov
· 5 years ago
12d64de
Remove sequence_number from RtpPacketInfo.
by Chen Xing
· 5 years ago
ad82e8e
Fix: restore disabling PC smoke test on iOS
by Artem Titov
· 5 years ago
19a1d50
Refactor WavWriter to use FileWrapper rather than PlatformFile
by Niels Möller
· 5 years ago
04e129a
Revert "Populate the GFD-00 for H264 and generic codecs."
by Philip Eliasson
· 5 years ago
1a285e0
Roll chromium_revision 96eca2d491..f165a6d5de (668951:669076)
by chromium-webrtc-autoroll
· 5 years ago
a9a2a75
Roll chromium_revision 22f062d604..96eca2d491 (668845:668951)
by chromium-webrtc-autoroll
· 5 years ago
c5effc2
Remove DataContentDescription shim
by Harald Alvestrand
· 5 years ago
ef3fd9c
Add support for simulcast with Vp8 from caller into PC level quality tests.
by Artem Titov
· 5 years ago
6751260
Roll chromium_revision b08bd9b643..22f062d604 (668716:668845)
by chromium-webrtc-autoroll
· 5 years ago
d3c6f9c
Populate the GFD-00 for H264 and generic codecs.
by philipel
· 5 years ago
9e25f74
Update visibility for JNI targets in sdk/android.
by Sami Kalliomäki
· 5 years ago
5894b6a
Add kPayloadTypeGeneric to CallTest and use it in VideoQualityTest.
by Rasmus Brandt
· 5 years ago
5740afa
Removes SimulatedTimeClient
by Sebastian Jansson
· 5 years ago
6fd67f0
Pass java EncodedImage over jni to VideoEncoderWrapper::OnEncodedFrame
by Niels Möller
· 5 years ago
f3f5770
Using full scenario test client for loss based control test.
by Sebastian Jansson
· 5 years ago
4284828
Remove deprecated version of RtpPacket::SetPadding that used to randomize padding
by Danil Chapovalov
· 5 years ago
5a8f860
Prepare for deletion of the NO_MAIN_THREAD_WRAPPING preprocessor define
by Niels Möller
· 5 years ago
50dd80b
Remove data channel only .so-file.
by Sami Kalliomäki
· 5 years ago
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