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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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26968bafc2023039c6f9abfdd4ce8dbed0334a6e
26968ba
Delete unused utf8 conversion utilities
by Niels Möller
· 6 years ago
e8038e9
Adds IP overhead info to PacketInfo.
by Sebastian Jansson
· 6 years ago
74cd1ef
AEC3: Enabling by default the use of the stationarity properties at render at init
by Jesús de Vicente Peña
· 6 years ago
5350d1c
RtcEventLogSource no longer uses deprecated parsing functions.
by Bjorn Terelius
· 6 years ago
499bc6c
Fix race conditions for ReofferDoesNotCallOnTrack test.
by Yves Gerey
· 6 years ago
53e2211
AEC3: Kill kill-switches
by Gustaf Ullberg
· 6 years ago
8b3cc49
Adds default values for feedback/allocation indicators.
by Sebastian Jansson
· 6 years ago
fb226af
Remove some old logging in goog_cc for congestion window.
by Ying Wang
· 6 years ago
a1d9ca4
Revert "Add ability to specify if rate controller of video encoder is trusted."
by Oleh Prypin
· 6 years ago
cdc959f
Compute video freeze metrics on rendered frames instead of on decoded
by Ilya Nikolaevskiy
· 6 years ago
3bdbc84
Moves pushback controller to GoogCC
by Sebastian Jansson
· 6 years ago
f81170b
Add error logs to RtpPacketHistory::GetBestFittingPacket when no packet is found.
by Per Kjellander
· 6 years ago
ade98c9
Adds srte to WATCHLISTS.
by Sebastian Jansson
· 6 years ago
2b15626
Revert "Use unique_ptr and ArrayView in SSLFingerprint"
by Henrik Grunell
· 6 years ago
703259c
Don't CHECK when parsing AEC3 parameters from json
by Sam Zackrisson
· 6 years ago
80bf775
Roll chromium_revision 2499289737..f34485ffde (598606:598711)
by chromium-webrtc-autoroll
· 6 years ago
f7fcaf0
Use zero octets for rtp packet padding
by Danil Chapovalov
· 6 years ago
3d25530
Reland "Export symbols needed by the Chromium component build (part 1)."
by Mirko Bonadei
· 6 years ago
3e335d1
Add ability to specify if rate controller of video encoder is trusted.
by Erik Språng
· 6 years ago
88be972
Delete post_encode_callback
by Niels Möller
· 6 years ago
74f6c7e
AEC3: Cleanup test code for platforms with clock-drift
by Per Åhgren
· 6 years ago
d6b0796
AEC3: Ensure that the usage of stationary signal properties is not unset
by Per Åhgren
· 6 years ago
23b2a25
Remove unlimited retransmission for screenshare experiment code
by Ilya Nikolaevskiy
· 6 years ago
cc21e61
Use unique_ptr and ArrayView in SSLFingerprint
by Steve Anton
· 6 years ago
e8d2b1b
Roll chromium_revision 8afdf16764..2499289737 (598496:598606)
by chromium-webrtc-autoroll
· 6 years ago
f7dd9df
Change TurnPort::Create to return a unique_ptr
by Steve Anton
· 6 years ago
9cfce17
Roll chromium_revision 0d09089dd5..8afdf16764 (598349:598496)
by chromium-webrtc-autoroll
· 6 years ago
0854eb6
Respond to SDP request extmap-allow-mixed.
by Johannes Kron
· 6 years ago
a8f1e56
Change Port::Create methods to return a unique_ptr
by Steve Anton
· 6 years ago
7940da0
Integration of media_transport in JSepTransportController
by Anton Sukhanov
· 6 years ago
6cc9cca
Don't reset streams for the FrameEncryptor / FrameDecryptor unless they changed.
by Benjamin Wright
· 6 years ago
da67c16
Roll chromium_revision 8a25f94ac2..0d09089dd5 (598237:598349)
by chromium-webrtc-autoroll
· 6 years ago
ca27091
Remove rtc_base:rtc_base_approved_generic.
by Mirko Bonadei
· 6 years ago
ede8796
Print per-frame VMAF score instead of average.
by Paulina Hensman
· 6 years ago
b3b0179
Fix backwards logic in rtc::Buffer::OnMovedFrom()
by Karl Wiberg
· 6 years ago
0213786
Add certificate gen/set functionality to bring Android closer to JS API
by Michael Iedema
· 6 years ago
dcc0238
Don't increment timestamp on drop/reencode in LibvpxVp8Encoder.
by Erik Språng
· 6 years ago
5526e45
vp9: change x-google-profile-id to profile-id
by Philipp Hancke
· 6 years ago
028248c
Add `rtc_enable_symbol_export` to incrementally create a WebRTC component.
by Mirko Bonadei
· 6 years ago
b686396
Makes AudioSendStream signal that it's part of allocation.
by Sebastian Jansson
· 6 years ago
99a70a2
Remove rtc_base_approved_objc and introduce rtc_base:logging_mac.
by Mirko Bonadei
· 6 years ago
edc49c1
[Cleanup] Remove unused swap function.
by Yves Gerey
· 6 years ago
a4c8514
Add JSON parsing and corresponding ToString to EchoCanceller3Config
by Sam Zackrisson
· 6 years ago
2558c4e
Remove ortc folder.
by Mirko Bonadei
· 6 years ago
88b68ac
Create field trial for setting a minimum value for Opus encoder packet loss rate
by Jakob Ivarsson
· 6 years ago
f08dd9d
Disable flaky tests on mac perf bot
by Ilya Nikolaevskiy
· 6 years ago
1bca65b
Makes RtpSender indicate allocation and feedback status on packets.
by Sebastian Jansson
· 6 years ago
81125f0
Implement (mostly) standards-compliant RTCIceTransportState.
by Jonas Olsson
· 6 years ago
5f35e96
Roll chromium_revision 476ae6d661..8a25f94ac2 (598136:598237)
by chromium-webrtc-autoroll
· 6 years ago
c87b8c1
Moves GoogCC factory to API.
by Sebastian Jansson
· 6 years ago
0d8c100
AEC3: Decrease the suppression during the echo-only case
by Per Åhgren
· 6 years ago
463c764
Roll chromium_revision cfe6e706d0..476ae6d661 (598018:598136)
by chromium-webrtc-autoroll
· 6 years ago
aabf204
Remove container typedefs from RelayServer
by Steve Anton
· 6 years ago
11358fe
Use unique_ptr in port_unittest
by Steve Anton
· 6 years ago
13d392d
AEC3: Utilize dominant nearend functionality to increase transparency
by Per Åhgren
· 6 years ago
3a3f027
Roll chromium_revision 0cf8926390..cfe6e706d0 (597915:598018)
by chromium-webrtc-autoroll
· 6 years ago
0378997
Adds flags indicating presence in allocation and feedback per packet.
by Sebastian Jansson
· 6 years ago
30e2d6e
Moves locking outside function in RtpSender.
by Sebastian Jansson
· 6 years ago
789f459
Adds fields for unacknowledged data to transport feedback.
by Sebastian Jansson
· 6 years ago
20a49f3
Don't try to use CN if voice codec isn't mono
by Karl Wiberg
· 6 years ago
5fcc4de
Roll chromium_revision f362b3e857..0cf8926390 (597811:597915)
by chromium-webrtc-autoroll
· 6 years ago
759f959
Refactor tests with ConfigurableFrameSizeEncoder
by Niels Möller
· 6 years ago
040f87f
AEC3: Allow a more stable filter during double-talk
by Gustaf Ullberg
· 6 years ago
7730193
Remove SetExecutablePath, simplify ResourcePath
by Patrik Höglund
· 6 years ago
7004571
AEC3: Decrease the modelling of the reverb
by Per Åhgren
· 6 years ago
d76a0fc
Throttle the RTP decryption error messages in the SrtpSession and SrtpTransport
by erikvarga@webrtc.org
· 6 years ago
b674cd1
Enable multithreading in libvpx VP9 decoder.
by Sergey Silkin
· 6 years ago
d0bc462
Check if __IPHONE_OS_VERSION_MAX_ALLOWED is defined before reference
by Joel Sutherland
· 6 years ago
0414040
Fix race condition for SupportsFlexfecWithMultithreadedH264/0 test.
by Yves Gerey
· 6 years ago
bf47198
Roll chromium_revision ba2e073e2c..f362b3e857 (597606:597811)
by chromium-webrtc-autoroll
· 6 years ago
4ff7214
Using TaskQueue for congestion controller by default.
by Sebastian Jansson
· 6 years ago
4b14416
Roll chromium_revision 0cdd2e3eab..ba2e073e2c (597498:597606)
by chromium-webrtc-autoroll
· 6 years ago
e0c2e97
Pass MediaTransportFactory to PeerConnectionFactory.
by Piotr (Peter) Slatala
· 6 years ago
1e05486
Added the new generic descriptor extension to WebRtcVideoEngine::GetCapabilities.
by philipel
· 6 years ago
ab09039
Add comment that xcode version needs to be updated in two places
by Oleh Prypin
· 6 years ago
16fe3f2
Revert "Export symbols needed by the Chromium component build (part 1)."
by Mirko Bonadei
· 6 years ago
99eea42
Reland "Reland "Export symbols needed by the Chromium component build (part 1).""
by Mirko Bonadei
· 6 years ago
2e068e8
Adds RTT based backoff trial to SendSideBandwidthEstimation.
by Sebastian Jansson
· 6 years ago
d2fb1bf
Generate module.modulemap file when building Mac Framework
by Joel Sutherland
· 6 years ago
e6708f3
Notify a rotation about autoroll CLs
by Oleh Prypin
· 6 years ago
75e3647
Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig
by Artem Titov
· 6 years ago
3a74239
Fix compilation issues on media_transport_interface.h
by Niels Möller
· 6 years ago
788c51c
Pass HeaderExtensionMap by reference in rtc_event_log2rtp_dump.
by Bjorn Terelius
· 6 years ago
b6a8942
Fix race condition for GetContributingSources test.
by Yves Gerey
· 6 years ago
666fb32
Rename DefaultNetworkSimulationConfig into BuiltInNetworkBehaviorConfig.
by Artem Titov
· 6 years ago
6a8327f
Roll chromium_revision ccb83d4a55..0cdd2e3eab (597330:597498)
by chromium-webrtc-autoroll
· 6 years ago
7c1744d
Reland "Reland "Using units in SendSideBandwidthEstimation.""
by Sebastian Jansson
· 6 years ago
841c912
Changed FakeVp8Encoder to write dimensions in payload.
by Per Kjellander
· 6 years ago
a4de9c8
Revert "Reland "Using units in SendSideBandwidthEstimation.""
by Sebastian Jansson
· 6 years ago
e2cb26c
Reland "Using units in SendSideBandwidthEstimation."
by Sebastian Jansson
· 6 years ago
917e596
Revert "Using units in SendSideBandwidthEstimation."
by Oleh Prypin
· 6 years ago
2e00abc
Reland "[cleanup] Remove useless includes."
by Yves Gerey
· 6 years ago
4dc66c5
Move EncodedImage class to api/video/
by Niels Möller
· 6 years ago
38537ed
Fix visibility of api/units build targets.
by Mirko Bonadei
· 6 years ago
343f414
Allows copy and assignment of field trial parameters.
by Sebastian Jansson
· 6 years ago
e53341c
Roll chromium_revision 7099444bc9..ccb83d4a55 (597172:597330)
by chromium-webrtc-autoroll
· 6 years ago
35b5e5f
Using units in SendSideBandwidthEstimation.
by Sebastian Jansson
· 6 years ago
9f80b97
Fix fuzzer build failures on Windows
by Jonathan Metzman
· 6 years ago
4d6f605
Roll chromium_revision d62b62d830..7099444bc9 (597059:597172)
by chromium-webrtc-autoroll
· 6 years ago
ef8a3eb
Include NTP value in playout path.
by Niklas Enbom
· 6 years ago
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