Re: [LAU] open hw soundcard with ext. codec

From: Martin Homuth-Rosemann <linuxaudio@email-addr-hidden>
Date: Fri Nov 20 2009 - 00:06:21 EET

Am Mittwoch, 18. November 2009 schrieb Karl Hammar:
> Martin Homuth-Rosemann:
> ...
>
> > Hi Karl, hi LAU users
>
> Hello and welcome to the discussion.
>
> > I've followed the discussion about timing and synchronisation - what do
> > you think about separation of number crunching and communication
> > (ATNGW100) from the "dirty business" of ADC.
>
> Shall take that as a question (you have no ?)?
I asked you (and the LAU audience) to hear your/their opinion about my (maybe
silly) idea. I see this discussion in the early project status mainly as a
kind of brainstorming.
To speed up the process of prototyping and to allow many participants I
suggested the use of "ready mades" - don't reinvent the wheel!
The atmel board ATNGW100 is easy available and not expensive, no time
consuming soldering (and hw debuging) needed - this will be a standard
platform for colaboration.
The same goes for the ADC, if we use available units with an (open) standard
communication protocol like AES-3, ADAT or MADI we can concentrate on the
difficult and more exciting part - finding new solutions / algorithms for syncing
different sources, internet transfer, ...
>
> Don't you always have to separate the digital and the analog domains?
>
> My plan is to build a card frame based system with one main power
> module, one cpu card, with the possibility to add a lot of different
> i/o cards. One such card could be for audio input/output. (Although
> my main interest is industrial measurement and control.)
Ok, a slightly different focus.
>
> With this the "dirty business" of ADC is separated to another card
> like an ordinary old soundcard you attached to your motherboard.
>
> Do we need more separation? Could it possible be because of:
> . space constraints
> . noise and audio quality
> . power constraints
> . economical factors
> . "time-to-market"
> etc.
>
> What are the key factors for you ?
See above ^
>
> > We need the codec, some kind of amplification, a clean power supply etc.
> > to get a good S/N ratio - and we need it for a lot of channels.
>
> Do you have a spec. which you'd like to discuss ?
> E.g. how many channels are you regulary using, what s/n ratio is a
> minimal requirement for you ?
We (no pluralis maiestatis, I summed my impression from some postings of the
LAU audience) need more than two channels, more than 16 bit and more than
100dB S/N at minimal 48 kHz, preferably 96 kHz.
>
> > There exist many (more or less) pro-audio devices with well documented
> > interfaces (SPDIF/AES-3; ADAT; MADI)
>
> Is your point, that the system should behave as an spdif etc.
> device instead of delivering the audio over ethernet?
No - just the other way round - I thought of replacing the chips TLC4545ID or
AD7762 (SPI or parallel interface) with a "black box" (AES-3, ADAT or MADI
interface) - just another way to get audio samples into our communication
processor which delivers them via ethernet into the linux computer.
>
> SPDIF [1], seems to be able to carry 20bit (maybe 24) 2 or 4 channels
> at 44.1 or 48kHz (possible other) sampling rates.
>
> AES-3 [2], seems to have the similar (24bit though) carrying capacity.
>
> ADAT [3], seems to be limited to 8 channels at 48 kHz, 24 bit.
>
> MADI [4], seems to be limited to 64 channels at 96kHz, 24 bit.
>
> If this project shall implement any of theese interfaces it might
> then be the ADAT or MADI, since I see no reason to implement the
> smaller interfaces.
But AES-3 (or AES-42 for digital microphone) is a standard for digital audio
connection.
>
> But if we successfully implement adat or madi, we are still missing
> the adat/madi part on the pc. So we still have a problem...
No, our "LAU-interface" is this part on the pc.
>
> And if we get i/o capacity problems with ethernet, we could easily add
> another ethernet card at relatively low cost. But then you might find
> that the rest of the computer is to small.
>
> > - a cheap one is e.g. the Behringer
> > ADA8000 for about 200 ¤ [1] with eight mic (phantom power) or line inputs
> > and eight line outputs. The codecs are 24bit@email-addr-hidden/48 kHz [2]
> >
> > [1] http://www.thomann.de/gb/behringer_ultragain_pro8_digital_ada8000.htm
> > [2] http://images4.thomann.de/pics/prod/164573_manual_eng.pdf
>
> Are you suggesting that that unit's spec is something to aim at ?
Of course not - as Fons stated^^ Behringer rhymes with "beware of" ;)
But it may give simpler and quicker results for testing than soldering small
smd ics onto veroboard ;)
(Frederick Brooks; The Mythical Man-Month: "plan to throw one away; you will,
anyhow.")
>
> Or is your point that it would be better to do a ADAT, or MADI
> interface for the pc instead of doing a "soundcard" ?
Not better but different (brainstorming....)
>
> Doing a adat/madi interface for the pc is outside of the scope of
> my projet, so I cannot help you there.
Ok! Sorry Karl for buggin' you.
>
> Regards,
> /Karl
>
> [1] http://en.wikipedia.org/wiki/S/PDIF
> [2] http://en.wikipedia.org/wiki/AES/EBU
> [3] http://en.wikipedia.org/wiki/ADAT
> [4] http://en.wikipedia.org/wiki/MADI

Ciao Martin
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Received on Fri Nov 20 00:15:02 2009

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