Gstreamer - Jack - sending audio over WiFi net.

[Mario Sottile]

Hi, there. I need to make a setup like this:



Two computers using Jack, with mic and speaker working and connected to PureData, both connected over WiFi network, sending/receiving audio and only one-way video.

There are several ways to do it: NetJack, jack.trip, VNC, X over SHH, etc. I'm testing them. But here I'm asking about GStreamer.

Can I encode audio with Gstreamer-Jack plugin and put it in IP address to access it from the other computer and use it in Pd? I would need to do the same in the other way.

(For the moment, for one-way video I would use VNC. If there's a better way with GStreamer, it will be in a future mail.)

I would like to use gst-launch/gst-launch-1.0, because I'm not developing an aplication, but a working system with already-done-software/commands.

Thanks in advance.

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[Sebastian Dröge-3]

On Mo, 2016-09-12 at 19:24 -0300, Mario Sottile wrote:

> Hi, there. I need to make a setup like this:
>
>
>
> Two computers using Jack, with mic and speaker working and connected
> to PureData, both connected over WiFi network, sending/receiving
> audio and only one-way video.
>
> There are several ways to do it: NetJack, jack.trip, VNC, X over SHH,
> etc. I'm testing them. But here I'm asking about GStreamer.
>
> Can I encode audio with Gstreamer-Jack plugin and put it in IP
> address to access it from the other computer and use it in Pd? I
> would need to do the same in the other way.

Yes, just use the GStreamer jack plugin, which contains a source and
sink. I'm not sure what you mean with encoding though, currently it
only supports only raw audio but if JACK can do more you could add
support for that.

> (For the moment, for one-way video I would use VNC. If there's a
> better way with GStreamer, it will be in a future mail.)
>
> I would like to use gst-launch/gst-launch-1.0, because I'm not
> developing an aplication, but a working system with already-done-
> software/commands.

gst-launch is a debugging / testing application and should not be used
in production.

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Sebastian Dröge, Centricular Ltd · http://www.centricular.com
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[Mario Sottile]

> Yes, just use the GStreamer jack plugin, which contains a source and
> sink. I'm not sure what you mean with encoding though, currently it
> only supports only raw audio but if JACK can do more you could add
> support for that.
I mean:
Jack (uncompressed audio) -> GSt Jack Plugin -> Encoding (compression,
GSt) -> Put in IP (GSt) -> Transmision to other computer in local
network -> Take from IP (GSt) -> Decoding (uncompression, GSt) -> GSt
Jack Plugin -> Jack (uncompressed audio)

I would want to compress the audio because I'm trying to use WiFi
network and it should be stable as possible (knowing that a WiFi is not
stable).

Could this setup be possible?

> gst-launch is a debugging / testing application and should not be used
> in production.
... for the moment, I can't go beyond this. I don't have the time to
learn writting a program to do that.

Thank you very much.
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[Sebastian Dröge-3]

On Di, 2016-09-13 at 08:44 -0300, Mario Sottile wrote:

> >
> > Yes, just use the GStreamer jack plugin, which contains a source and
> > sink. I'm not sure what you mean with encoding though, currently it
> > only supports only raw audio but if JACK can do more you could add
> > support for that.
> I mean:
> Jack (uncompressed audio) -> GSt Jack Plugin -> Encoding (compression, 
> GSt) -> Put in IP (GSt) -> Transmision to other computer in local 
> network -> Take from IP (GSt) -> Decoding (uncompression, GSt) -> GSt 
> Jack Plugin -> Jack (uncompressed audio)
>
> I would want to compress the audio because I'm trying to use WiFi 
> network and it should be stable as possible (knowing that a WiFi is not 
> stable).
>
> Could this setup be possible?

Yes, that's really a standard setup. You capture on one side, encode,
transmit. Receive on the other side, decode, play back.

You only need to decide on which codec to use and which streaming
protocol you want to use here.

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[Mario Sottile]

> You only need to decide on which codec to use and which streaming
> protocol you want to use here.
I think using UDP as protocol and... don't know which codec. I want to
send only voice with a regular quality. Which one do you recommend? Opus?
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[Mario Sottile]

> You only need to decide on which codec to use and which streaming
> protocol you want to use here.
I think using UDP as protocol and... don't know which codec. I want to
send only voice with a regular quality. Which one do you recommend? Opus?

Do you know what the latency will be, aproximatly?

Maybe I'm asking so much... but is there any example? I found some here:
https://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/tests/examples/rtp?h=0.10 
(I didn't try them yet). Any others?

Thanks.
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[Sebastian Dröge-3]

On Do, 2016-09-15 at 10:29 -0300, Mario Sottile wrote:

> >
> > You only need to decide on which codec to use and which streaming
> > protocol you want to use here.
> I think using UDP as protocol and... don't know which codec. I want to 
> send only voice with a regular quality. Which one do you recommend? Opus?
>
> Do you know what the latency will be, aproximatly?
>
> Maybe I'm asking so much... but is there any example? I found some here: 
> https://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/tests/examples/rtp?h=0.10 
> (I didn't try them yet). Any others?

Why did you look at the 0.10 version of those? But yes, those are
examples of how you can use RTP. There should also be one doing Opus
there. Opus over RTP seems like a good choice in this scenario.

Latency depends on too many factors in this scenario, you would have to
measure it. And not just end-to-end but also for the different parts
individually to get an idea where to tune things.

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Sebastian Dröge, Centricular Ltd · http://www.centricular.com
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