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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
4241cf5ed5d8ce0531b6e0b3933ba665e4b656f0
/
modules
/
include
f93eda1
Move some video codec constants to separate file.
by Erik Språng
· 6 years ago
988cc08
[Cleanup] Add missing #include. Remove useless ones.
by Yves Gerey
· 6 years ago
276827c
Export symbols needed by the Chromium component build (part 3).
by Mirko Bonadei
· 6 years ago
44b384d
Delete support for VoIP metrics (RFC 3611 4.7)
by Niels Möller
· 6 years ago
db12856
Cleanup modules_common_types
by Danil Chapovalov
· 6 years ago
01a8990
Remove deprecated type alias for RtpVideoCodecTypes.
by Kári Tristan Helgason
· 6 years ago
9a29c03
Fix random crashes - invariant broken in LinkedSet (LRU) implementation.
by Yves Gerey
· 6 years ago
a12c42a
Delete root header file typedef.h.
by Niels Möller
· 6 years ago
1a4746a
Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file.
by philipel
· 6 years ago
5ab67a5
Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
by philipel
· 6 years ago
011dc64
Remove RTPTypeHeader union and use RTPVideoHeader directly.
by philipel
· 6 years ago
8b23dba
Add RTPVideoHeader const accessor.
by philipel
· 6 years ago
196100e
Replace rtc::Optional with absl::optional
by Danil Chapovalov
· 6 years ago
7b55c73
Add RTPVideoHeader accessor.
by philipel
· 6 years ago
0a5fe77
Clean up in module_common_types.h by removing the unused struct RTPAudioHeader.
by philipel
· 6 years ago
665174f
Reformat the WebRTC code base
by Yves Gerey
· 6 years ago
8643b78
Moved NackModule and VCMPacket to their own targets
by Ilya Nikolaevskiy
· 6 years ago
520ca4e
Delete enum RtpVideoCodecTypes, replaced with VideoCodecType.
by Niels Möller
· 6 years ago
c6c4426
Moves network control interface to API.
by Sebastian Jansson
· 6 years ago
ae8d8a1
Remove audio_frame.h from module_common_types.h
by Fredrik Solenberg
· 7 years ago
0dd1b0a
Revert "Revert "Enables PeerConnectionFactory using external fec controller""
by Ying Wang
· 7 years ago
0073301
Revert "Enables PeerConnectionFactory using external fec controller"
by Taylor Brandstetter
· 7 years ago
4f07bdb
Enables PeerConnectionFactory using external fec controller
by Ying Wang
· 7 years ago
d377f04
Move AudioFrame to its own header file and target in api/.
by Niels Möller
· 7 years ago
9bb8f05
Cleanup of unused RTP structs and packetizer for stereo codec
by Emircan Uysaler
· 7 years ago
3e11343
Fix circular dependencies in webrtc_common.
by Patrik Höglund
· 7 years ago
90612a6
Reland "Add stereo codec header and pass it through RTP"
by Emircan Uysaler
· 7 years ago
deb8663
Revert "Add stereo codec header and pass it through RTP"
by Philip Eliasson
· 7 years ago
20f2133
Add stereo codec header and pass it through RTP
by Emircan Uysaler
· 7 years ago
248ccf8
Optional: Use nullopt and implicit construction in /
by Oskar Sundbom
· 7 years ago
e40468b
Move some numeric utility code from rtc_base/ to rtc_base/numerics/
by Karl Wiberg
· 7 years ago
feee08f
Marked UnwrapWithoutUpdate function as const
by Sebastian Jansson
· 7 years ago
558cabf
Refactor RtpToNtpEstimator and MovingMedianFilter
by Ilya Nikolaevskiy
· 7 years ago
a32dd01
Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface."
by Fredrik Solenberg
· 7 years ago
d4404c2
Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface."
by Fredrik Solenberg
· 7 years ago
34cdd2d
Remove AudioDeviceObserver and make ADM not inherit from the Module interface.
by Fredrik Solenberg
· 7 years ago
c7b4a45
Remove various IDs:
by solenberg
· 7 years ago
e423a9de
Revert of Remove various IDs (patchset #7 id:120001 of https://codereview.webrtc.org/3019543002/ )
by solenberg
· 7 years ago
2d0f775
Remove various IDs:
by solenberg
· 7 years ago
fb08994
Adding time profiling support to AudioFrame
by henrika
· 7 years ago
7120742
Adding NOLINT for typedefs.h and common_types.h
by Mirko Bonadei
· 7 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 7 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 7 years ago