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

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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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