Gitiles
Code Review
Sign In
gerrit-public.fairphone.software
/
platform
/
external
/
webrtc
/
b8867115a755cf54a75b4123054dae29190441b4
/
video
/
stream_synchronization.cc
675513b
Stop using LOG macros in favor of RTC_ prefixed macros.
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
[Renamed from webrtc/video/stream_synchronization.cc]
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 8 years ago
a80c16a
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 8 years ago
c3771cc
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 8 years ago
3ebbcb5
Stop using VoEVideoSync in Call/VideoReceiveStream.
by solenberg
· 8 years ago
fe50b4d
Make class of static functions in rtp_to_ntp.h: - UpdateRtcpList - RtpToNtp
by asapersson
· 8 years ago
4cd2790
Move RTP for synchroninzation and rename classes, files and variables.
by mflodman
· 8 years ago
f8cdd18
Add histogram stats for AV sync stream offset: "WebRTC.Video.AVSyncOffsetInMs"
by asapersson
· 9 years ago
7623ce4
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
by Peter Boström
· 9 years ago
[Renamed (99%) from webrtc/video_engine/stream_synchronization.cc]
8237abf
Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
by kjellander
· 9 years ago
[Renamed (99%) from webrtc/video/stream_synchronization.cc]
03ef053
Merge webrtc/video_engine/ into webrtc/video/
by Peter Boström
· 9 years ago
[Renamed (99%) from webrtc/video_engine/stream_synchronization.cc]
415d2cd
Use webrtc/base/logging.h for video.
by Peter Boström
· 9 years ago
36a1438
Remove ViEFrameProviderBase.
by Peter Boström
· 10 years ago
66773a0
Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine.
by wu@webrtc.org
· 11 years ago
5574dac
Log Fixit for parts of video_engine folder.
by mflodman@webrtc.org
· 11 years ago
47fadba
Add include stdlib.h to files using abs.
by stefan@webrtc.org
· 11 years ago
12dc1a3
Switch C++-style C headers with their C equivalents.
by pbos@webrtc.org
· 11 years ago
e46c8d3
API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay.
by turaj@webrtc.org
· 12 years ago
f5d4cb1
Include files from webrtc/.. paths in video_engine/
by pbos@webrtc.org
· 12 years ago
d35964a
Fixing AV sync. Increased 2 const to allow for a bigger difference in AV sync.
by pwestin@webrtc.org
· 12 years ago
6311733
Updated the sync module with a slow moving filter
by pwestin@webrtc.org
· 12 years ago
0d8d010
Handle multiple calls to set initial delay
by mikhal@webrtc.org
· 12 years ago
ef9f76a
Adding a receive side API for buffering mode.
by mikhal@webrtc.org
· 12 years ago
14b43be
Move src/ -> webrtc/
by andrew@webrtc.org
· 12 years ago
[Renamed from src/video_engine/stream_synchronization.cc]
64d9dec
Move RtpToNtp functionality to its own file.
by stefan@webrtc.org
· 12 years ago
2dcbcc1
Changing two asserts which should have returned errors instead.
by stefan@webrtc.org
· 12 years ago
7c3523c
Change audio/video sync to be based on mapping RTP timestamps to NTP.
by stefan@webrtc.org
· 12 years ago
5f28498
First step in refactoring audio/video synchronization. Adds unittests.
by stefan@webrtc.org
· 13 years ago