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rtp: gstreamer: Don't count on buffer DTS for capture time

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With GStreamer 1.18, the old behaviour of storing the capture time in
DTS is gone (which is reasonable, since the semantics really don't
match). So instead, we get a capture timestamp when the buffer is being
pushed from udpsrc. This should eventually move into udpsrc, and the
timestamp should come from the cmsg instead of the clock.

We still fallback to the DTS if the meta isn't available, as the meta
might be dropped in older versions of rtpL16pay due to a bug.
This commit is contained in:
Arun Raghavan 2020-10-22 10:06:40 -04:00
parent 3fc2ac10c6
commit 5f12dde2f5
3 changed files with 51 additions and 6 deletions
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2
meson.build
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@ -678,7 +678,7 @@ if webrtc_dep.found()
cdata.set('HAVE_WEBRTC', 1)
endif
gst_dep = dependency('gstreamer-1.0', required : get_option('gstreamer'))
gst_dep = dependency('gstreamer-1.0', version : '>= 1.14', required : get_option('gstreamer'))
gstapp_dep = dependency('gstreamer-app-1.0', required : get_option('gstreamer'))
gstrtp_dep = dependency('gstreamer-rtp-1.0', required : get_option('gstreamer'))