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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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478cb46435d03b19ab2dcf15c243f6911542d655
478cb46
Add GeneratePadding method to replace TimeToSendPadding
by Erik Språng
· 6 years ago
c2f5686
Extend structures to store updated version of the dependency descriptor
by Danil Chapovalov
· 6 years ago
a3f3ab9
Remove Simple Command Line Parser.
by Mirko Bonadei
· 6 years ago
4ba04b7
Delete RtcEventLogFactory factory as now unused
by Danil Chapovalov
· 6 years ago
36c8ef6
Cleanup Ulpfec receiver: remove 2 blocks RED packets support
by Ilya Nikolaevskiy
· 6 years ago
a36c591
Reland "Reland "Change buffer level filter to store current level in number of samples.""
by Jakob Ivarsson
· 6 years ago
b93af85
Revert "Reland "Change buffer level filter to store current level in number of samples.""
by Jakob Ivarsson
· 6 years ago
2d821c3
Allow VideoTimingExtension to be used with FEC
by Ilya Nikolaevskiy
· 6 years ago
e4ac723
Delete deprecated version of PeerConnectionFactoryInterface::StartAecDump
by Niels Möller
· 6 years ago
bca1485
Enable setting surface_ice_candidates_on_ice_transport_type_changed on the fly.
by Qingsi Wang
· 6 years ago
0ded32d
Reland "Change buffer level filter to store current level in number of samples."
by Jakob Ivarsson
· 6 years ago
4d69516
Don't use angle-bracket #includes for WebRTC's own files
by Oleh Prypin
· 6 years ago
c57b0ee
Fix for NACK retransmission in Scenario tests.
by Sebastian Jansson
· 6 years ago
be0adee
Add resolution bitrate thresholds to EncoderInfo.
by Sergey Silkin
· 6 years ago
2644a70
Delete rtc::TryCritScope as unused
by Danil Chapovalov
· 6 years ago
e8df482
Clean-up after CL #140941
by Elad Alon
· 6 years ago
db59de3
Add optimization to PacketRouter for large number of senders.
by Danil Chapovalov
· 6 years ago
90f3b89
Replace the implementation of `GetContributingSources()` on the video side.
by Chen Xing
· 6 years ago
3472b9a
Delete RTCInboundRTPStreamStats::fraction_lost
by Niels Möller
· 6 years ago
1d20185
Convert file objects to strings, before passing to PresubmitNotifyResult
by Niels Möller
· 6 years ago
7dd9969
Android: Expose getDisplaySize() helper function
by Magnus Jedvert
· 6 years ago
554be4b
Roll chromium_revision c30c3a10ff..27c2f87cf5 (672280:672459)
by chromium-webrtc-autoroll
· 6 years ago
c538506
Enable H.264 temporal scalability in simulcast.
by Johnny Lee
· 6 years ago
0d65fb5
Mass refactoring: Change JNI #includes to use full paths (webrtc/).
by Eric Stevenson
· 6 years ago
6e8bfae
Roll chromium_revision 42482d4f53..c30c3a10ff (672061:672280)
by chromium-webrtc-autoroll
· 6 years ago
a79abd4
Specify min_sdk_version for unittest apks also in GN configs
by Oleh Prypin
· 6 years ago
f97f342
Remove deprecated flags from compare_videos.py.
by Mirko Bonadei
· 6 years ago
75bc70c
Remove flags include from p2p/base/datagram_dtls_adaptor.cc.
by Mirko Bonadei
· 6 years ago
25ca0ac
Also fail CreateOffer and CreateAnswer if there is a session error
by Steve Anton
· 6 years ago
9a5c2e8
Remove unused command_line_parser dependency.
by Mirko Bonadei
· 6 years ago
04cffe3
Switch example peerconnection server to ABSL_FLAG.
by Mirko Bonadei
· 6 years ago
f03b365
Reland "Raise IllegalStateException for calls to retain() or release() on zero ref count"
by Niels Möller
· 6 years ago
3894078
Roll chromium_revision 6c5bbd86c3..42482d4f53 (671957:672061)
by chromium-webrtc-autoroll
· 6 years ago
5767fb8
Stop passing deprecated flags to frame analyzer.
by Mirko Bonadei
· 6 years ago
d3fc161
Revert "Change buffer level filter to store current level in number of samples."
by Jakob Ivarsson
· 6 years ago
87977dd
Change buffer level filter to store current level in number of samples.
by Jakob Ivarsson
· 6 years ago
3391072
Use DefaultTaskQueueFactory in CreatePeerConnectionFactoryForJava
by Danil Chapovalov
· 6 years ago
a95949e
Switch rtp_generator to ABSL_FLAG.
by Mirko Bonadei
· 6 years ago
beff535
Switch frame_editor to ABSL_FLAG.
by Mirko Bonadei
· 6 years ago
6d6b0fb
Switch reference_less_video_analysis to ABSL_FLAG.
by Mirko Bonadei
· 6 years ago
8e032c8
Switch psnr_ssim_analyzer to ABSL_FLAG.
by Mirko Bonadei
· 6 years ago
2308bb2
Switch rgba_to_i420_converter to ABSL_FLAG.
by Mirko Bonadei
· 6 years ago
6bafc8b
Reland "Switch frame_analyzer to ABSL_FLAG."
by Mirko Bonadei
· 6 years ago
90c6d1b
Revert "Switch frame_analyzer to ABSL_FLAG."
by Mirko Bonadei
· 6 years ago
2923d76
Roll chromium_revision c049034cc7..6c5bbd86c3 (671848:671957)
by chromium-webrtc-autoroll
· 6 years ago
3668e9b
Switch frame_analyzer to ABSL_FLAG.
by Mirko Bonadei
· 6 years ago
fa1e7c1
Roll chromium_revision d6c73f3029..c049034cc7 (671737:671848)
by chromium-webrtc-autoroll
· 6 years ago
f4996d4
Roll chromium_revision 163d6b201e..d6c73f3029 (671624:671737)
by chromium-webrtc-autoroll
· 6 years ago
e9cac4f
Improvements to scenario video stats for scenario tests.
by Sebastian Jansson
· 6 years ago
8605fbf
[getStats] Make remote-inbound-rtp.ssrc match outbound-rtp.ssrc.
by Henrik Boström
· 6 years ago
8f7b41a
Roll chromium_revision ec99dbf1aa..163d6b201e (671513:671624)
by chromium-webrtc-autoroll
· 6 years ago
4b8a5b4
Removes unused PacketFeedbackComparator
by Sebastian Jansson
· 6 years ago
f48bca7
Avoid triggering a false error logging when using encryptor and sending DTX.
by Minyue Li
· 6 years ago
59b8654
Switch from RtpPacketSender to RtpPacketPacer interface usage.
by Erik Språng
· 6 years ago
13eb764
Move towards always using packet type instead of priority in RTPSender
by Erik Språng
· 6 years ago
1231419
BalancedDegradationSettings: Add option to configure QP thresholds.
by Åsa Persson
· 6 years ago
88290ae
Reland "Cleanup of RTP references in GoogCC implementation."
by Sebastian Jansson
· 6 years ago
746dd0d
Add support for two-byte RTP header extensions to UsedRtpHeaderExtensionIds
by Johannes Kron
· 6 years ago
2ffcd82
Make DefaultTemporalLayers explicitly request a key frame
by Elad Alon
· 6 years ago
9aa870a
Fixing fuzzer by backing up and restoring `packet_info`.
by Chen Xing
· 6 years ago
214f543
Make useful padding the default.
by Erik Språng
· 6 years ago
2b1d8c8
Roll chromium_revision b827eec031..ec99dbf1aa (671413:671513)
by chromium-webrtc-autoroll
· 6 years ago
4a15833
Roll chromium_revision a8f7a25b6e..b827eec031 (671303:671413)
by chromium-webrtc-autoroll
· 6 years ago
3e3a900
Roll chromium_revision ffeecb2d27..a8f7a25b6e (671198:671303)
by chromium-webrtc-autoroll
· 6 years ago
abf7eba
Roll chromium_revision ef202aa763..ffeecb2d27 (671098:671198)
by chromium-webrtc-autoroll
· 6 years ago
6fd5b9e
Roll chromium_revision 568fb40139..ef202aa763 (670985:671098)
by chromium-webrtc-autoroll
· 6 years ago
6e91646
Roll chromium_revision 5cf576bfbe..568fb40139 (670869:670985)
by chromium-webrtc-autoroll
· 6 years ago
6924d50
Roll chromium_revision 4dfb50605a..5cf576bfbe (670751:670869)
by chromium-webrtc-autoroll
· 6 years ago
d487a55
Revert "Remove sync buffer length from FilteredCurrentDelayMs."
by Jakob Ivarsson
· 6 years ago
a550dad
Minor rtc_event_log_impl cleanup.
by Nikita Zetilov
· 6 years ago
6c42d92
Added video_coding::EncodedFrame copy ctor.
by philipel
· 6 years ago
f00bf42
Add plumbing of RtpPacketInfos to each VideoFrame as input for SourceTracker.
by Chen Xing
· 6 years ago
7953ad5
Revert "Cleanup of RTP references in GoogCC implementation."
by Sebastian Jansson
· 6 years ago
fa79081
Cleanup of RTP references in GoogCC implementation.
by Sebastian Jansson
· 6 years ago
775c02e
Do not use libevent when targeting wasm.
by Mirko Bonadei
· 6 years ago
9da25bd
In PeerConnection unittests set TaskQueueFactory explicitly
by Danil Chapovalov
· 6 years ago
621be83
Roll chromium_revision 516b926bdf..4dfb50605a (670612:670751)
by chromium-webrtc-autoroll
· 6 years ago
e3cc489
Add logging and edit the field trial name for piggyback ICE check
by Qingsi Wang
· 6 years ago
0d92f15
Roll chromium_revision 39e46dcf9e..516b926bdf (670459:670612)
by chromium-webrtc-autoroll
· 6 years ago
5bf5806
Force DefaultTaskQueueFactory in legacy CreatePeerConnectionFactory
by Danil Chapovalov
· 6 years ago
c2a54dc
Roll chromium_revision 6e638caa96..39e46dcf9e (670359:670459)
by chromium-webrtc-autoroll
· 6 years ago
c16289f
Split the build rule for video_frame into a video_rtp_headers part.
by Chen Xing
· 6 years ago
b64ad0e
Using Clock::CurrentTime() where non-test behavior is unchanged.
by Sebastian Jansson
· 6 years ago
18f1f0c
Revert "Raise IllegalStateException for calls to retain() or release() on zero ref count"
by Niels Moller
· 6 years ago
8a959bf
Raise IllegalStateException for calls to retain() or release() on zero ref count
by Niels Möller
· 6 years ago
505bac2
Add default implementation of deprecated StartAecDump method.
by Niels Möller
· 6 years ago
4d504c7
New interface EncodedImageBufferInterface, replacing use of CopyOnWriteBuffer
by Niels Möller
· 6 years ago
0894f0f
Add piggyback acknowledgement of the last ICE check received in outgoing checks.
by Qingsi Wang
· 6 years ago
f9511fc
Roll chromium_revision 619b073720..6e638caa96 (670258:670359)
by chromium-webrtc-autoroll
· 6 years ago
b7a3e3f
Roll chromium_revision 8639acbee7..619b073720 (670130:670258)
by chromium-webrtc-autoroll
· 6 years ago
e8347a8
Roll chromium_revision bf62d746a4..8639acbee7 (669828:670130)
by chromium-webrtc-autoroll
· 6 years ago
1b3f4f9
Allow RtpPacketHistory encapsulator function to abort retransmit
by Erik Språng
· 6 years ago
4cbb4ef
Roll chromium_revision 6ae0f0cd4c..bf62d746a4 (669703:669828) + fix AndroidManifest
by Oleh Prypin
· 6 years ago
b762b5b
Fix potential signed overflow in IntervalBudget::set_target_rate_kbps
by Per Kjellander
· 6 years ago
342f98b
Fixes for flexfec crash in scenario tests.
by Sebastian Jansson
· 6 years ago
58ee187
Add support within PacedSender and pacer queue for owning rtp packets.
by Erik Språng
· 6 years ago
b028c6a
Support __EMSCRIPTEN__ in rtc_base.
by Mirko Bonadei
· 6 years ago
0c0c969
Add/rewrite H264 VUI video signal type description.
by Sergey Silkin
· 6 years ago
449888e
Cleanup of resources from removed remote bitrate estimate test framework.
by Sebastian Jansson
· 6 years ago
2c648f5
Stop running 'bwe_simulations_tests'.
by Mirko Bonadei
· 6 years ago
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