how to remove audio from live stream

[Marianna Smidth Buschle]

>> Can you play your stream with gstreamer?
>> gst-launch-1.0 srtsrc uri=srt://<IP>:33334 ! decodebin ! autovideosink
> I have no video on this remote vm.

You can change fx to a fakesink to see if you get data.

gst-launch-1.0 srtsrc uri=srt://<IP>:33334 ! decodebin ! video/x-raw ! fakesink dump=1

Then you should see it dumping a lot of data.


Alternatively you can try installing gstreamer on the Windows machine.

And see if you can play with Gstreamer there.

>> As a test you can also try changing to RTP, since that should work with
>> VLC:
>> - Send with
>> gst-launch-1.0 srtsrc uri=srt://:55556?mode=listener ! rtpmp2tpay !
>> rtpbin ! udpsink port=7000 host=127.0.0.1 sync=0
>>
>> - Receive it:
>> vlc rtp://@:7000
>> udpsrc address=127.0.0.1 port=7000 !
>> application/x-rtp,media=video,payload=33,clock-rate=90000,encoding-
>> name=MP2T
>> ! rtpbin ! rtpmp2tdepay ! decodebin ! autovideosink
>>
>  From what I have read about the udpsink in the gst-inspect, it looks like it is pushing the stream to 'host', so I do not think I can get this to work, since I am testing with a local (windows) vlc and a remote vm. I think the problem must be somewhere between demux  and mux, maybe some 'header' or other setting is being removed, and that is why vlc has problems with 'detecting' the live stream.

You can also do multicast in udpsink with fx "|auto-multicast=1 
host=224.0.0.1|". But you might also need to enable multicast on the 
remote VM. And you need to setup the correct eth interface if it is not 
the default one.


But I agree with you, since you say that [2] (gst-launch-1.0 srtsrc 
uri=srt://:55556?mode=listener ! srtsink uri=srt://:33334) works it must 
be something on the CAPS.

So I would suggest trying to compare the CAPS you get with -v for both 
[2] and [4] and see if you can spot any differences.

Look specially for the MPEGTS and H264 CAPS. I have had issues in some 
case with things like: stream-format and alignment.


You can also try adding some tsparse to see if it makes any difference:

gst-launch-1.0 srtsrc uri=srt://:55556?mode=listener ! tsparse ! tsdemux ! queue ! video/x-h264 ! h264parse ! mpegtsmux name=mux ! tsparse ! srtsink uri=srt://:33334


-- 
Best regards / Med venlig hilsen
“Marianna Smidth Buschle”

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