AAC/RTP streaming

[Martin Vachovski]

Hello everybody,


I am trying to implement AAC streaming overt RTP, but the receiving pipeline doesn't play any audio (although it says it goes to PLAYING state).

?I am using the following pipelines with gstreamer 1.12.2:


TX:

gst-launch-1.0.exe -v audiotestsrc ! audioconvert ! avenc_aac ! rtpmp4apay ! udpsink


RX:

gst-launch-1.0.exe -v udpsrc ! application/x-rtp,clock-rate=44100 ! rtpjitterbuffer ! rtpmp4adepay ! avdec_aac ! audioconvert ! autoaudiosink


(I have also tried replacing the "avdec_aac" element with a "decodebin").

The pipelines don't report any errors, but the second one doesn't play any sound.

I wonder if there is any other setting, caps I need to specify? I know that the Vorbis codec,

for instance, needs the DCT coefficients of the transform as extra data when streaming over the network?


Also, if I merge the two pipelines together (omitting the udpsink/udpsrc pair), the pipeline works:

audiotestsrc volume=0.1 ! audioconvert ! avenc_aac ! rtpmp4apay ! application/x-rtp,clock-rate=44100 ! rtpjitterbuffer ! rtpmp4adepay ! avdec_aac ! audioconvert ! wavescope ! videoconvert ! autovideosink


I have also tried to supply all the CAPS as reported by the SRC pad of "rtpmp4apay " before the "rtpjitterbuffer", still no luck

/GstPipeline:pipeline0/GstRtpMP4APay:rtpmp4apay0.GstPad:src: caps = application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)MP4A-LATM, cpresent=(string)0, config=(string)40002410adca00, payload=(int)96, ssrc=(uint)3854776221, timestamp-offset=(uint)466007625, seqnum-offset=(uint)12210


Many thanks!

Martin

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