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gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
2870b0a57e68be61331ba9b8b38e762d41a242ba
/
common_types.h
d7ae3c3
Reland "Rename stereo video codec to multiplex"
by Emircan Uysaler
· 7 years ago
1204448
Revert "Reland "Rename stereo video codec to multiplex""
by Taylor Brandstetter
· 7 years ago
4954a77
Reland "Rename stereo video codec to multiplex"
by Emircan Uysaler
· 7 years ago
6bc7bb6
Revert "Rename stereo video codec to multiplex"
by Ivo Creusen
· 7 years ago
bbdabe5
Rename stereo video codec to multiplex
by Emircan Uysaler
· 7 years ago
f6464c9
Adds the active field to SimulcastStream and VideoCodec structs.
by Seth Hampson
· 7 years ago
0f17f9c
Revert "Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer."
by Lu Liu
· 7 years ago
18c4261
Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer.
by Seth Hampson
· 7 years ago
3e11343
Fix circular dependencies in webrtc_common.
by Patrik Höglund
· 7 years ago
c7c4191
Declare the RTCP packets_lost field as signed in the API.
by Harald Alvestrand
· 7 years ago
4f6b6c2
Delete MediaFile support for unused fileformats.
by Niels Möller
· 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
01f2ec3
Add a new function to BitrateAllocation: HasBitrate.
by erikvarga@webrtc.org
· 7 years ago
4bece9a
Set RTPVideoHeader picture id in PayloadRouter if forced fallback for VP8 is enabled.
by Åsa Persson
· 7 years ago
4332d09
Reland "Reland "Remove WEBRTC_TRACE.""
by Fredrik Solenberg
· 7 years ago
39cefdb
Revert "Reland "Remove WEBRTC_TRACE.""
by Fredrik Solenberg
· 7 years ago
68007e9
Reland "Remove WEBRTC_TRACE."
by Fredrik Solenberg
· 7 years ago
729b910
Revert "Remove WEBRTC_TRACE."
by Fredrik Solenberg
· 7 years ago
2209b90
Remove WEBRTC_TRACE.
by Fredrik Solenberg
· 7 years ago
b0a0207
Added RTCMediaStreamTrackStats.jitterBufferDelay for audio
by Gustaf Ullberg
· 7 years ago
9a2e906
Added RTCMediaStreamTrackStats.concealmentEvents
by Gustaf Ullberg
· 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
5740135
by niklase@google.com
· 13 years ago
5988675
Replacing kTraceVqe with kTraceAudioProcessing.
by ajm@google.com
· 13 years ago
da159d6
by niklase@google.com
· 13 years ago