[LAU] open hw soundcard with ext. codec

Martin Homuth-Rosemann linuxaudio at cryptomys.de
Thu Nov 19 17:06:21 EST 2009


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 at 44.1/48 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



More information about the Linux-audio-user mailing list