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Hi all! It's been a long time since i last posted: Cheers!

A question: I've heard a lot about the sonic improvements in recording at 24bit vs 16bit ( and it sounds really logical ) so I decided to try.

How can I get 24bit from my Yamaha02r ADAT port into my Pulsar II? Am I Supposed to have a "BitSplitter" or this mixer is already capable of outputting 24 bit over a digital ADAT Channel?

Thks a lot in advance!

You should be able to load a 24 bit driver (Wav or Asio) in SFP and connect the ADAT output from the O2R to your Pulsar II.

You will benefit from extra headroom at 24 bit, so you won't have to record at high levels.

Hope this helps.

The 02r outputs 24 bit from its ADAT ports so just connect it to the Pulsar and off you go.

yes,be sure and use the adat source and destination modules.......

Dear all,thks for your kind answer!

My real problem is that I'm going to use my SFP to dimostrate that it's possible to create with it the core of a professional studio.

But I MUST be sure that I'm actually selling to my future customers data sampled with a REAL 24 bit depth, almost a Standard today !

Actually, I can Load "Asio 24Source/Dest" into my SFP project and Record "44.1Khz@24Bit" with Cubase Sx directly from my 02r.

But I can also record "44.1Khz@16Bit" without any kind of dither inserted and it sounds correctly, withou pops/crackles/etc etc.

Yamaha 02r's internal path is 20 bit. How can it samples analog sound @24bit? Does it split the word into two ADAT channel? The first 16 bit on channel 1 and the remaining 8 on channel 2? Something similar to S/MUX?

So I can only assume that bit from 17 to 24 from my 02r are dithered or SPF does the dithering in real time or Cubase does.

Where's the truth? Can Someone help and explain me? If the topic is too wide to be discussed here please , at least send me interesting links!

Thks in advance to all, I would not know wherelse I could ask this kind of question and
I do apologize if this is not the right forum!

Cheers

On 2003-08-27 05:52, raboz wrote:
...Yamaha 02r's internal path is 20 bit...

I'm not familiar with the various 02rs, but this is almost certainly not the case or you'd run into real trouble with the mixer's headroom.

There was a model with 20bit converters for analog input, but the internal number of bits the mixer processes is at least 24bit, possibly 32bit or even higher.

Which means if your customer comes up with 24-bit recording feeded digitally into the 02r, then this data is transferred as 24bit digital data to Pulsar Adat In from the mixers Adat Out.

If you record via the analog input of your mixer then of course the resolution is at best 20 bit, probably padded.
The AD/DA converters of your Pulsar are very high quality 24-bit types, while SFP internally processes all data as 32bit.

cheers, Tom

Thks!

So we can state that a 02r mixer is ** NOT ** able to sample Analog data with 24bit depth but can be a "pass-through" for "pre-recorded" 24bit samples.

This explain me why I can send samples 44.1Khz@24bit to my 02r ADAT IN via SFP "Asio 24 Source".

The last un-solved question is : why I'm able to record via "ASIO24 Dest" a signal sampled from my 02r analog input with no distortions/pops usually created by frequencies/bit rate not-compatible? I can't really understand what the bit 17>24 means in my sampled data transferred via ADAT!

I can even record data 44.1Khz@16bit via "ASIO 24 Dest"! So i suspect only the first 16 bit are sampled ...

Hope you could be so kind to ansewr me once more!

Cheers

if you are using 24bit i/o and putting a 16 bit signal in,you have 8 bits headroom.

From page 204 in the 02R v.2 manual:


Sorry,

Kim.

P.S.
If I remember it correctly, the 02R card uses the original Alesis chips, which by design are capable of transmitting only 20bits per. channel...


<font size=-1>[ This Message was edited by: kimgr on 2003-08-27 15:46 ]</font>

Hi Kimgr, thks for your kind and detailed answer.

I have a 02RV2.Pdf but it's Italian and there's no mention at all! If you search for "24 bit" you only get statement related to dithering, that's what scares me: I suppose datas are simply dithered to 24 bit NOT sampled!

So my main question remain, as your last post shows: if have have an Alesis Analog to Digital converter with a 20-bit wordlength, how can I Sample data at 24 bit?

Don't shoot me please! It's a real matter of extreme importance!

Cheers

I thought Apogee made a card for the O2r that was 24bit i/o