13 Commits

Author SHA1 Message Date
stefan@webrtc.org
7c3523c1a4 Change audio/video sync to be based on mapping RTP timestamps to NTP.
Video Engine:
- Instead compensate for video capture delay by modifying RTP timestamps.
- Calculate the relative offset between audio and video by converting
  RTP timestamps to NTP and comparing receive time.

RTP/RTCP module:
- Removes the awkward video modification of NTP to compensate
  for video capture delay.
- Adjust RTCP RTP timestamp generation in rtcp_sender to have the same offset
  as packets being sent from rtp_sender.

BUG=
TEST=trybots,steam_synchronization_unittest

Review URL: https://webrtc-codereview.appspot.com/669010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2733 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-09-11 07:00:42 +00:00
andrew@webrtc.org
6f8db36e04 Reorganize voice_engine/.
The usual changes:
voice_engine/main/source -> voice_engine/
voice_engine/main/interface -> voice_engine/include
voice_engine/main/test -> voice_engine/test
Include path changes.

BUG=none
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/705004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2535 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-27 21:49:28 +00:00
mflodman@webrtc.org
ab2610ffd9 Removed the last lint warnings in video_engine.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/670006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2468 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-29 10:05:28 +00:00
stefan@webrtc.org
5f28498149 First step in refactoring audio/video synchronization. Adds unittests.
BUG=
TEST=stream_synchronization_unittest

Review URL: https://webrtc-codereview.appspot.com/669005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2455 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-28 07:51:16 +00:00
mflodman@webrtc.org
139c4678c1 Fixed a/v sync issue.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2402 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-14 11:08:51 +00:00
pwestin@webrtc.org
2853dde520 Refactor the internal API to the rtp/rtcp module.
Combination of previous CLs in revisions 2211, 2212, 2214, 2215, 2216.
Review URL: https://webrtc-codereview.appspot.com/570008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2231 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-11 11:08:54 +00:00
turaj@webrtc.org
3c383abd27 Revert 2211 - Refactor the internal API to the rtp/rtcp module.
Review URL: https://webrtc-codereview.appspot.com/568004

A series of CL:s by Patrik W. is breaking the auto-test. It started with CL 2211, but the later CL:s seems dependent on another. So I decided to go in reverse order and revert all of them.

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/563011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2226 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-10 23:01:04 +00:00
pwestin@webrtc.org
0774838f3d Refactor the internal API to the rtp/rtcp module.
Review URL: https://webrtc-codereview.appspot.com/568004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2211 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-10 12:33:50 +00:00
leozwang@webrtc.org
39e9659fc6 Correct wrong usage of WebRtc_Word8 in video enigne
BUG=http://code.google.com/p/webrtc/issues/detail?id=311&sort=-id
TEST=build on all platforms

Review URL: https://webrtc-codereview.appspot.com/428002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1815 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-01 18:22:48 +00:00
mflodman@webrtc.org
d32c44738a Changed constructor used for CriticalSectionScoped in ViE.
Only changed:
- Name of some of the critsects.
- All critsects (but one) are now scoped_ptr.
- Use of ptr constructor of CriticalSectionScoped instead of reference version.

BUG=184
TEST=vie_auto_test

Review URL: http://webrtc-codereview.appspot.com/330015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1291 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-22 14:17:53 +00:00
mflodman@webrtc.org
d2ee5d989d Changed sync bug introduced in refactoring.
Review URL: http://webrtc-codereview.appspot.com/319001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1129 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-12-08 09:25:11 +00:00
mflodman@webrtc.org
511f82eee9 Refactored ViESyncModule.
Only style changes, will follow up with references/ptrs.

Review URL: http://webrtc-codereview.appspot.com/291007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1068 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-30 18:31:36 +00:00
mflodman@webrtc.org
94ea32ef60 Move video_engine/source* to video_engine/. No code changes except paths in gyp-files.
Review URL: http://webrtc-codereview.appspot.com/283002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@984 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-21 14:49:31 +00:00