Playing RTP stream has artifacts and sound problems

[Chuck Crisler]

OK, you say that the stream is RTP with MP2T inside. So you need something
like:

udpsrc <...> !
'application/x-rtp,media=video,payload=33,clock-rate=90000,encoding-name=H264'\
     ! rtpmp2tdepay ! mpegtsdemux ! more stuff.

You probably want a 'T' (tee) to branch off the different programs for
their respective decoders. This isn't complete but should get you started.


On Wed, May 7, 2014 at 7:43 AM, antimatter <antimatter at quantentunnel.de>wrote:

> Hello,
>
> I am a beginner with GStreamer and haven't done any fancy stuff so far. My
> goal is to play a RTP multicast stream from our local network. The video
> should be decoded with Fluendo codec pack (GPU accelerated), because the
> target platforms have potent GPUs but only Atom CPUs. The general problem
> is
> failing sound right after start (manual pipeline) or asynchronous/distorted
> sound and lots of artifacts (playbin).
>
> The stream originates from a DVB-S2 network tuner, which is sending a RTP
> stream with a MPEG transport stream. Inside there's a H.264 720p video
> stream, a 2-channel audio stream as AC3 and MP2. The teletext stream is not
> needed. VLC can play this stream perfectly fine. I provide the stream
> address "rtp://239.5.2.1:6670".
>
> Before I start coding, I'd like to build a prototype pipeline with
> gst-launch-0.10. So I used playbin and provided the URI with "udp" instead
> of "rtp" and created a diagram from the DOT file, to see what's going on
> under the hood. The result is having those problems as described at the
> beginning. Playbin creates an udpsrc, a typefinder, a TSDemux and a
> multiqueue at the beginning. I recreated this part on my own and added the
> Fluendo decoder for video and an audio decoder:
>
> gst-launch-0.10 -v -m \
> udpsrc uri="udp://239.5.2.1:6670" ! typefind ! flutsdemux name=demux \
> multiqueue name=mq \
> demux.video_13ed ! mq.sink0 \
> demux.audio_13f2 ! mq.sink1 \
> mq.src0 ! fluvadec ! fluvasink \
> mq.src1 ! ac3parse ! ffdec_ac3 ! audioconvert ! autoaudiosink
>
> This will display the video with artifacts (especially at moving image
> parts), audio is gone after 1 sec or it is extremely scrambled.
>
> Some other forum posts indicate the use of rtpbin, but I could not quite
> get
> my head around its usage. And I've got the feeling, this is the way to go.
> Can I use rtpbin with all it's glory with gst-launch? How do I connect the
> udpsrc to rtpbin and rtpbin to my decoders? The SSRC and payload type
> parameters on its dynamic source pad got me lost. And there's something
> with
> caps and "application/x-rtp" as mimetype I have seen given to udprsc. Do I
> need this twice (audio + video)?
>
> I was hoping someone could give me the famous push in the right direction.
>
> Best regards,
> Andreas
>
>
>
> --
> View this message in context:
> http://gstreamer-devel.966125.n4.nabble.com/Playing-RTP-stream-has-artifacts-and-sound-problems-tp4666829.html
> Sent from the GStreamer-devel mailing list archive at Nabble.com.
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