]> 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 20:26:00 +0000 (15:26 -0500)
commit6647be69acc1d662efe4c458a7522e5a70bc8276
treef84dd9bea99104a03d4046846b01ae2c50c89adb
parentf9f17f3bfe9820bbe0ff7314eb53c16557b6e40e
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