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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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9ac3c9118a15fd64efb6913df015590b425537cb
9ac3c91
Refactor of extmap-allow-mixed in SessionDescription
by Johannes Kron
· 6 years ago
cae8802
Delete force_mic_volume_max.
by Patrik Höglund
· 6 years ago
83bd37c
Add field trials for configuring Opus encoder packet loss rate.
by Jakob Ivarsson
· 6 years ago
fcebe0e
in RtpPacketizers separate case 'frame fits into single packet'.
by Danil Chapovalov
· 6 years ago
1a35fbd
Add field trial for normalized simulcast size.
by Åsa Persson
· 6 years ago
09256c1
Remove ios32_sim_ios9_dbg from CQ.
by Mirko Bonadei
· 6 years ago
147038c
cq: explicitly mark presubmit tryjob as not re-usable in CQ.
by Oleh Prypin
· 6 years ago
9c18d21
Remove rtc_base/Dummy.java.
by Mirko Bonadei
· 6 years ago
28887a5
Roll chromium_revision 03013c95df..0df2607f98 (599460:599562)
by chromium-webrtc-autoroll
· 6 years ago
37cf245
Revert "Propagate media transport to media channel."
by Oleh Prypin
· 6 years ago
f409246
Roll chromium_revision 3b54b6aa8b..03013c95df (599343:599460)
by chromium-webrtc-autoroll
· 6 years ago
8c16f74
Propagate media transport to media channel.
by Anton Sukhanov
· 6 years ago
dbc2ea7
Roll chromium_revision c12ec9eedc..3b54b6aa8b (599188:599343)
by chromium-webrtc-autoroll
· 6 years ago
55cd3ac
Modernize rtc::SSLCertificate
by Steve Anton
· 6 years ago
47f3240
Reland: Use unique_ptr and ArrayView in SSLFingerprint
by Steve Anton
· 6 years ago
5e23a41
Removes backwards compatability CryptoOptions support.
by Benjamin Wright
· 6 years ago
23e48fb
Move expectations from eventlog unittests to helper functions.
by Bjorn Terelius
· 6 years ago
f7fee39
Remove rtc_base:rtc_base_generic.
by Mirko Bonadei
· 6 years ago
b354f74
Roll chromium_revision d47784f23e..c12ec9eedc (599082:599188)
by chromium-webrtc-autoroll
· 6 years ago
6af1c92
Add mock_video_encoder.h to api/test
by Erik Språng
· 6 years ago
3b4b4f5
Mitigate miscalculation of rtp packet size
by Danil Chapovalov
· 6 years ago
781b2bd
Restore "device type" for iOS internal.client.webrtc
by Artem Titarenko
· 6 years ago
62b1345
Get rid of thread_darwin file.
by Kári Tristan Helgason
· 6 years ago
c34cf71
Revert "Remove old video_bitrate_allocator.h"
by Oleh Prypin
· 6 years ago
93428bf
Move SdpType from/to string definition close to declaration.
by Mirko Bonadei
· 6 years ago
55d1af1
Remove support for microsecond resolution in RtcEventLogs.
by Bjorn Terelius
· 6 years ago
4529fbc
Move TemporalLayers to api/video_codecs.
by Erik Språng
· 6 years ago
28d200c
Roll chromium_revision 37b6d53f02..d47784f23e (598967:599082)
by chromium-webrtc-autoroll
· 6 years ago
a54daf1
Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h"
by Benjamin Wright
· 6 years ago
edd204e
Roll chromium_revision 9d052f4b6f..37b6d53f02 (598839:598967)
by chromium-webrtc-autoroll
· 6 years ago
8f4bc41
Revert "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h"
by Oleh Prypin
· 6 years ago
1cd39fa
make CreateOffer/CreateAnswer use ice credentials of pooled sessions.
by Jonas Oreland
· 6 years ago
df1bf00
Headers shouldn't include themselves.
by Yves Gerey
· 6 years ago
ac2f3d1
Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h
by Benjamin Wright
· 6 years ago
8285841
Adds handling of untracked data to congestion controller.
by Sebastian Jansson
· 6 years ago
ca51189
Roll chromium_revision f34485ffde..9d052f4b6f (598711:598839)
by chromium-webrtc-autoroll
· 6 years ago
0d399a8
Removes socket addresses from PacketInfo struct.
by Sebastian Jansson
· 6 years ago
20ad254
Adds tracking of allocated but unacknowledged bitrate.
by Sebastian Jansson
· 6 years ago
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
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