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authorArun Raghavan <>2020-10-22 10:06:40 -0400
committerArun Raghavan <>2020-10-23 12:11:35 -0400
commit5f12dde2f5dbc85c918ac469ef64866a96b70fa8 (patch)
tree076d40491eb3982525ec4f2b13db51fa268d98af /
parent3fc2ac10c68a57b1fbdc8c464c6a181d4ab31053 (diff)
rtp: gstreamer: Don't count on buffer DTS for capture time
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.
Diffstat (limited to '')
1 files changed, 1 insertions, 1 deletions
diff --git a/ b/
index 02c4d99fe..86af7243e 100644
--- a/
+++ b/
@@ -678,7 +678,7 @@ if webrtc_dep.found()
cdata.set('HAVE_WEBRTC', 1)
-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'))