Re: [LAU] What audio interface to use for a Linux-powered surround preamp?

On 12/18/2011 03:28 PM, Johan Herland wrote:

>> out of curiousity, what is the benefit of upsampling? is there something
>> peculiar about power dacs that would make it useful or mandatory?
>
> It's not really a requirement, but I thought it'd make sense for two reasons:
>
> 1. Making sure there's enough resolution to do room correction, volume
> control, etc. without losing details.

with these very simple and mostly linear operations, you will gain
nothing at all from upsampling. certainly nothing that would justify
doubling the cpu and throughput expense of your system.

preventing the full fidelity of material produced at 96k is another
issue, but such material is rare outside of studio workflows.

<snip>
>> plus you will need to think about the clocking structure. usually
>> this will mean that your audio card will have to slave itself to the
>> incoming hdmi/spdif/whatever.
>
> Hmm. I don't really know much about clocking. How would you organize
> the system to minimize clocking issues, and maximize fidelity?

all digital gear in the signal chain must run from the same clock,
unless you insert a sample-rate converter.
in a studio, there is a common wordclock which is distributed to all
players, processors, and output DACs.
consumer equipment will generally not be able to deal with external
clocking, so your source will have to be the clock master. the clock is
then distributed embedded in the signal - hdmi audio, spdif and aes/ebu
are all self-clocking.

but this also means that if you switch from blue-ray player a to
blue-ray player b, your sound card must change its clocking source from
input a to input b. which might or might not cause an audible click or
thump.

as to "maximizing fidelity", this is digital pcm. it either works
perfectly or not at all. the only way to slightly degrade the signal is
to have a lossy codec in between (such as ac3 or dts), or when you're
forced to insert a SRC. but the latter should be pretty close to perfect
if it's a good one. no longer bit-transparent, obviously.

>>> - A suitable audio interface with at least 8 digital outputs.
>>
>> tough one. there are a number of cheap options with ADAT, but you will need
>> two at 96k due to s/muxing, and four at 192k. for the latter, the only
>> option i know of is the rme raydat.
>
> I've been doing some reading on 96kHz vs. 192kHz, and most people seem
> to think that there is no audible difference, and that it's a lot of
> marketing hype, so until I'm convinced otherwise, I'm not going to
> spend extra money on 192kHz equipment. So what other equipment is
> available with 2 ADAT outputs?

i tend to agree on the 192k issue...
iirc, there is the rme hdsp 9636 which should fit your needs (2 adats).
or a hdsp hammerfall with the digiface break-out (3 adats). there might
be cheaper alternatives from other manufacturers, maybe others will
comment. m-audio used to be a linux-friendly choice...

<snip>
> It really comes
> down to how cheaply I can convert from ADAT to either AES/EBU or SPDIF
> (either of which is what most digital amps will take as input).

haven't done a proper survey, but i'd use RME gear for this job. which
means the combination of adat card plus external AES/EBU bridge will be
more expensive and less elegant than the hdsp/aes card.

> When it comes to kernel compilation, I'm not too scared, as my
> background is in Linux and software.

good :)

have fun,

jörn

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Received on Mon Dec 19 12:15:02 2011