Need more info about 14/15

[niceboy]

yes, all Scope products work at 96k.
ADAT light pipe is 44.1k or 48k and can actually work at any samplerate up to 48k. YOU have to choose the samplerate. ADAT light pipe can do 96k in S-MUX mode.

W

ADAT light pipe is one alesis interface right or ???.
but cant we use spdif for monitring from our SCOPE or maybe anologe monitoring from that then when mixing and mastering ,
analog will work right but you say that lightpipe can be connected to other ADAT s in 48 khz when in S-MUX mode 96 khz .

Is that one ADAT INTERFACE with built in S-MUX ? to be connected to SCOPE ADAT in and out.

I am now going to Google

can we then continue monitoring in 48 KHZ with the other ADAT interfaces.
If so it might be one option to record to disk in 96KHZ our newest song thats only on drums midifile here,
but I want to know what you think about RME ADI 648 with 8 ADAT is that a good ADAT to MADI converter

It seems that M-Audio have one light pipe
We are using M-POWERED and Pro Fire so we might find out about this.

Thank you.

[niceboy]

yes, all Scope products work at 96k.
ADAT light pipe is 44.1k or 48k and can actually work at any samplerate up to 48k. YOU have to choose the samplerate. ADAT light pipe can do 96k in S-MUX mode.

W

ADAT light pipe is one alesis interface right or ???.
but cant we use spdif for monitring from our SCOPE or maybe anologe monitoring from that then when mixing and mastering ,
analog will work right but you say that lightpipe can be connected to other ADAT s in 48 khz when in S-MUX mode 96 khz .

Is that one ADAT INTERFACE with built in S-MUX ? to be connected to SCOPE ADAT in and out.

I am now going to Google

can we then continue monitoring in 48 KHZ with the other ADAT interfaces.
If so it might be one option to record to disk in 96KHZ our newest song thats only on drums midifile here,
but I want to know what you think about RME ADI 648 with 8 ADAT is that a good ADAT to MADI converter

It seems that M-Audio have one light pipe
We are using M-POWERED and Pro Fire so we might find out about this.

Thank you.

[niceboy]

well, there are (exist) devices that handle multiple samplerates simultaneously (realtime samplerate conversion)
no idea if the system you connect to has something like that - at least it's not exotic, in fact it's standard in onboard audio codecs and streaming content (not that great implementation usually, tho...).
Anyway, an A16 MK2 doesn't cost an arm and a leg and you could solve it easily with a 2nd unit if someone insists on 96k and if it keeps things more simple.

But frankly speaking that's a nerd attitude...
There IS a difference between 44.1 and 96k that's easily detected in a listening room or a studio - but under live conditions ? C'mon... that must be a f*cking great treated venue ...
Hey, this is DSP processing - they oversample internally where required (Minimax filter for example)...
No need to waste the juice all the time.

cheers, Tom

I want to find out about the 96KHZ because we still use one old plugin
DSPFX Aurial Activator together with Scope
and it can raise the diskant on the Cymbals to the level to sound good in the next step.
I get as much diskant as I want but I wonder how we can prove to our self if we need this 96 KHZ
when mixing,
We have to record the same thing in both 48KHZ and in 96 KHZ to be abel to listen to the difference.
We must be ready with our new song so we dont have much time,
so I wonder has anyone really tested it like that.
Or how did you test that then.
Maybe I am wrong about this, 96 is not about latency
and neither about volym on frequencies thats the bit rate.
I am confused how did you hear any difference.
I can hear the difference between 12% above and 30% above
when I am singing and both is okey in intonation
but this seems to be one aria thats not so much tested.
Otherwise feel free to tell me how you did test that

[astroman]

that's a very simple story and it's totally obvious.
It's not even related to any specific material - it's the simple fact that so called 'aliasing products' which are part of every digital processing of analog sources 'flip back' into the audible range with 48k, while they end way beyond in 96k.
This lack of aliasing is perceived as a more clear or defined tone - that simple.
If it's worth the effort or resources is up to each own's estimation.

That's a simple matter of fact, but here it ends with simplicity...

If you go 'real world' it's indeed hard to 'objectively' compare because the hardware isn't identical.
Any processing element colours the sound, even am AD or DA converter. It makes a significant difference if an amp stage is built from discrete parts or integrated opamps, if it's transformer balanced or not and so on.
You probably know devices that just 'sound good' - regardless of their technical performance.
Who wants a Linn Drum with a 24bit converter ?
Imo that fact sheet data crap is totally overestimated, just trust your ears...

cheers, Tom

[niceboy]

that's a very simple story and it's totally obvious.
It's not even related to any specific material - it's the simple fact that so called 'aliasing products' which are part of every digital processing of analog sources 'flip back' into the audible range with 48k, while they end way beyond in 96k.
This lack of aliasing is perceived as a more clear or defined tone - that simple.
If it's worth the effort or resources is up to each own's estimation.

That's a simple matter of fact, but here it ends with simplicity...

If you go 'real world' it's indeed hard to 'objectively' compare because the hardware isn't identical.
Any processing element colours the sound, even am AD or DA converter. It makes a significant difference if an amp stage is built from discrete parts or integrated opamps, if it's transformer balanced or not and so on.
You probably know devices that just 'sound good' - regardless of their technical performance.
Who wants a Linn Drum with a 24bit converter ?
Imo that fact sheet data crap is totally overestimated, just trust your ears...

cheers, Tom

For us as a Band it seems to be ,the option not to play in thouse places where they want to use 96 KHZ then,
because it seems to be thouse people using Venue Boards but not knowing what they are doing.
Then we dont have to play .
In the Studio it seems to be no need to use even 48 we can use 24 bit 44,1 maybe
48 might be more practical for live then, if most Front of the House guys use that maybe.
Thats what SSL told me in the Headquarters in Oxford.
I dont know about US as one State sort of.
I must admit thought that I dont fully understand your answere ( That's a simple matter of fact ) ???
We have the Waves plugins and we tried the L316 preset Ben Mink solid TV mix Threshold -4,7
and it has 24 bit Dithering but when we listen to that it sounded very strong but that was for what ??? 24 bit diethering
on the intro we had just four Hiat counting in, and we could not use it there but the rest of the song sounded great
but the same using 16 bit as for mastering to CD the diethering was just too much for only Hiat on intro,
so I thought you might mean that dithering from 96KHZ with for example L316 or some other plugins
gives so much more effects in the diethering process coming from 96KHZ to 44,1 KHZ to the CD,
but I am then probarbly out in the blue or because the Diethering was all about going from 24 bit to 16 bit or
What do you acctually mean with That's a simple matter of fact, but here it ends with simplicity,
I have a Bachelor Degree in Music from America ,but no Degree in Sound Technic though ,
and English is not my first language.
Best Regards Bjorn

[astroman]

it's a simple matter of fact because it applies to each and every digital processing system, without exception.
Of course it needs a switchable samplerate, so you can actually hear both versions in the same environment.
It doesn't make sense to campare Scope 48k to Avid 96k to RME 44.1k etc as each system has a different analog circuit design.
But in a real world application you frequently have to do just that because the gear is setup this way... and that's not so simple then - so simplicity ends.
Btw the lack of alias frequencies in the audible range (samplerates > 64k) is the reason for high rates in typical consumer gear (home entertainment).
It greatly simplifies filter design. An anti-alias filter is essential(!) with low sample rates and one of the most expensive parts of a CD player for example...

Dithering is a completely different thing and not releated to the sample rate.
It's supposed to smooth out rounding errors when signal dynamic is processed.
I question you can hear that on a single source at all.
I even question that you can tell a 16bit from a 20bit single(!) source if mixed randomly.
The improvement is (probably) noticable in a mix of many channels, but to be honest - I don't care.
Maybe I would if involved in classical recordings of symphony orchestras and opera...
But definitely not in what's considered 'contemporary', pop, funk, alternative and I dunno-what.

cheers, Tom