I am trying to figure out the proper gstreamer element to use to transmit AAC audio over RTP.
By dumping the dot graph of a playbin on the file I can conclude that the caps coming out of the tsdemux is audio/mpeg,mpegversion:2,stream-format:adts .
If I use the following pipeline
gst-launch-1.0 -v filesrc location=$BA ! tsdemux ! audio/mpeg ! rtpmpapay ! filesink location=/tmp/test.rtp
it fails:
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
/GstPipeline:pipeline0/GstCapsFilter:capsfilter1: caps = audio/mpeg
WARNING: from element /GstPipeline:pipeline0/GstTSDemux:tsdemux0: Delayed linking failed.
Additional debug info:
/var/tmp/portage/media-libs/gstreamer-1.12.3/work/gstreamer-1.12.3/gst/parse/grammar.y(510): gst_parse_no_more_pads (): /GstPipeline:pipeline0/GstTSDemux:tsdemux0:
failed delayed linking some pad of GstTSDemux named tsdemux0 to some pad of GstRtpMPAPay named rtpmpapay0
ERROR: from element /GstPipeline:pipeline0/GstTSDemux:tsdemux0: Internal data stream error.
Additional debug info:
/var/tmp/portage/media-libs/gst-plugins-bad-1.12.3/work/gst-plugins-bad-1.12.3/gst/mpegtsdemux/mpegtsbase.c(1613): mpegts_base_loop (): /GstPipeline:pipeline0/GstTSDemux:tsdemux0:
streaming stopped, reason not-linked (-1)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
Which gstreamer element should I be using to wrap AAC audio in an RTP packet?
I guess its rtpmp4apay: RTP MPEG4 audio payloader. Maybe you want/need aacparse before the payloader as well.