]> git.ipfire.org Git - thirdparty/asterisk.git/commit
func_jitterbuffer: Add audio/video sync support.
authorJoshua Colp <jcolp@digium.com>
Fri, 6 Sep 2019 13:18:55 +0000 (13:18 +0000)
committerJoshua Colp <jcolp@digium.com>
Wed, 18 Sep 2019 20:22:50 +0000 (20:22 +0000)
commit7298a785ad998ac9e5aff36a3d6efa8e4dc82a47
treef74e7ce0e9ce2b886629119862b8c30ba6ca560a
parentac8db871ec3aafb41bc4edcac57ae5c72c31cdcc
func_jitterbuffer: Add audio/video sync support.

This change adds support to the JITTERBUFFER dialplan function
for audio and video synchronization. When enabled the RTCP SR
report is used to produce an NTP timestamp for both the audio and
video streams. Using this information the video frames are queued
until their NTP timestamp is equal to or behind the NTP timestamp
of the audio. The audio jitterbuffer acts as the leader deciding
when to shrink/grow the jitterbuffer when adaptive is in use. For
both adaptive and fixed the video buffer follows the size of the
audio jitterbuffer.

ASTERISK-28533

Change-Id: I3fd75160426465e6d46bb2e198c07b9d314a4492
doc/CHANGES-staging/func_jitterbuffer_video.txt [new file with mode: 0644]
funcs/func_jitterbuffer.c
include/asterisk/abstract_jb.h
include/asterisk/rtp_engine.h
main/abstract_jb.c
main/rtp_engine.c
res/res_rtp_asterisk.c