SPDIF sample rate problem? Please HELP!

[tcornishmn]

I can hear some hiss (after normalization) using analog line out from the MD player. As this is being amplified to 7.5 kW live, I want to make sure that hiss/hum is minimised. You can understand i don't want to record several hours of material to find that there is too much noise for live use. I am probably being over-cautious to be honest.....

To be fair, I'm sure I could live with it, but I just want to make absolutely sure there is as little noise recorded as possible. I have already recorded some analog and (to be fair) it sounds great through headphones - but as I say, there is a small amount if hiss.

Ray.

I can GUARANTEE that you are being over-cautious. A little bit of hiss in this is going to be so lost with all of the other distortion from the venue as well as frequency-response issues with even the best million $$ PA systems. It doesn't matter if it's 7.5Kw or 7.5Mw - higher power doesn't translate into higher quality.



Let me put it this way - if you record with decent gain structure - that is Minidisk player's output fairly high into the Oasys so that the input level meters are fairly hot, then you won't have any more noise in the recording than your Minidisk player has now.

In other words - if your Minidisk player was good enough before, it's going to be just as good being recorded into the Oasys. The Oasys' preamps and A/D converters are far better than the output of your Minidisk player.


I'm a live player. For my weekly gig we have a fairly high-end PA system with very qualified sound guys. BTW, ours is larger than 7500 watts, whatever that means. Compared to my studio monitors, it sounds like garbage - especially acoustic piano sounds.

Occasionally I do larger gigs where we usually play through a L'Acoustics Kudo line array system from a Digidesign Venue board. 100,000 watts easy (again, power has nothing particularly important to do with sound quality). Same deal. Live sounds in live rooms mangle everything.

This excersize may be informative for you, but honestly you're making this harder than it needs to be.

[mozartella]

The Oasys will only work correctly if the clock that it is syncing to is 48kHz (or 96kHz IIRC). If you clock it at 44.1kHz it will appear to work, however everything will be at the incorrect pitch as you've observed (if you hit a key whilst the Oasys is synced this way you'll it plays back at the wrong pitch). I doubt that you can change the sample rate on your MD player to 48Khz so I suspect you'll need to use analog connectivity to record your audio from the MD to the Oasys. What the D/A -> A/D conversion will do to the audio is probably not as bad as what ATRAC has already done to it

Daz.

daz, very good answer, I have a question, if the SPDIF from the Oasys must be at 48'000, then a World clock with the EXB-DI board would be at ?

and, if the Oasys, using only ADAT, and SPDIF, and REC at say 48'000 what external sequencer would be able to truly multi-rec at 44'100 even if Oasys send it at 48'000? do we could have a way, if when recorded at 48'000 to keep the same pitch at 44'100 or no? If I got it right, the clock of an audio card, able to be on Master mode, would be at 48"000, then how to lower it at 44'100 and keeping the same pitch? I jusr recorded a song, on 48'000 and I am able to make it at 44'100 but with a different pitch, and that is no---no...

thanks

[Daz]

daz, very good answer, I have a question, if the SPDIF from the Oasys must be at 48'000, then a World clock with the EXB-DI board would be at ?

The Word Clock connected to the BNC on the EXB-DI must be 48 kHz. Period

The manual confirms this :

Word Clock: The OASYS will use the optional EXBDI’s
BNC WORD CLOCK IN as the master clock. The
incoming clock must be at 48kHz.

and, if the Oasys, using only ADAT, and SPDIF, and REC at say 48'000 what external sequencer would be able to truly multi-rec at 44'100 even if Oasys send it at 48'000? do we could have a way, if when recorded at 48'000 to keep the same pitch at 44'100 or no? If I got it right, the clock of an audio card, able to be on Master mode, would be at 48"000, then how to lower it at 44'100 and keeping the same pitch? I jusr recorded a song, on 48'000 and I am able to make it at 44'100 but with a different pitch, and that is no---no...

If you're using the Oasys digital I/O then the simplest thing to do is to clock everything at 48kHz, including your Sequencer/DAW. That's the only way unless you have any special tools to reclock/resample multiple digital audio devices.

In my case Logic is configured to use 24 bits/48Khz, it is clocked to my RME Fireface (which is configured with a sample rate of 48Khz) and the Oasys is synced to that via the EXB-DI.

What you're suggesting will only lead to tears and frustration

Daz.

[Ray]

I can GUARANTEE that you are being over-cautious. A little bit of hiss in this is going to be so lost with all of the other distortion from the venue as well as frequency-response issues with even the best million $$ PA systems. It doesn't matter if it's 7.5Kw or 7.5Mw - higher power doesn't translate into higher quality.



Let me put it this way - if you record with decent gain structure - that is Minidisk player's output fairly high into the Oasys so that the input level meters are fairly hot, then you won't have any more noise in the recording than your Minidisk player has now.

In other words - if your Minidisk player was good enough before, it's going to be just as good being recorded into the Oasys. The Oasys' preamps and A/D converters are far better than the output of your Minidisk player.


I'm a live player. For my weekly gig we have a fairly high-end PA system with very qualified sound guys. BTW, ours is larger than 7500 watts, whatever that means. Compared to my studio monitors, it sounds like garbage - especially acoustic piano sounds.

Thanks for the encouraging words.

I was talking about this to the drummer earlier and he has the same (old fashioned?) notions as me. i.e. keep all recordings as first generation as possible to minimise noise. Which is sort of strange given when I started playing the digital Casio watch hadn't been invented, let alone digital recording

The PA is JBL with QSC amps/dbx driver and it needs to be fairly loud as well as good quality. The sound from it is great to my ears and I wanted to keep that quality as much as possible.

But I/we take your point. I hereby promise that once I have tried the SPDIF convert (tonight) I will put this to bed......honest.

Regards,

Ray.

[mozartella]

daz, very good answer, I have a question, if the SPDIF from the Oasys must be at 48'000, then a World clock with the EXB-DI board would be at ?

The Word Clock connected to the BNC on the EXB-DI must be 48 kHz. Period

The manual confirms this :

Word Clock: The OASYS will use the optional EXBDI’s
BNC WORD CLOCK IN as the master clock. The
incoming clock must be at 48kHz.

thanks , at least we have the world clock and the SPDIF at the same rate....

and, if the Oasys, using only ADAT, and SPDIF, and REC at say 48'000 what external sequencer would be able to truly multi-rec at 44'100 even if Oasys send it at 48'000? do we could have a way, if when recorded at 48'000 to keep the same pitch at 44'100 or no? If I got it right, the clock of an audio card, able to be on Master mode, would be at 48"000, then how to lower it at 44'100 and keeping the same pitch? I jusr recorded a song, on 48'000 and I am able to make it at 44'100 but with a different pitch, and that is no---no...

If you're using the Oasys digital I/O then the simplest thing to do is to clock everything at 48kHz, including your Sequencer/DAW. That's the only way unless you have any special tools to reclock/resample multiple digital audio devices.

In my case Logic is configured to use 24 bits/48Khz, it is clocked to my RME Fireface (which is configured with a sample rate of 48Khz) and the Oasys is synced to that via the EXB-DI.

What you're suggesting will only lead to tears and frustration

Yes I do know, and I did clock the RME and the Oasys at 48'000 the sequencer was looking for the external clco, and fine, but Imade a REC and, when I saved, it was, of course at 48'000 and after I tried to save it at 44'100 and that where the change of pitch occured, when I was saving the file on MP3 format at 44'100...I miss something here

thanks Daz

[Ray]

I hereby promise that once I have tried the SPDIF convert (tonight) I will put this to bed......honest.

Regards,

Ray.

Tried it, but got mixed up when .wav's required renaming. So reverted to audio.

I am happy

Many thanks for all your assistance.

Kind Regards,

Ray.

[Mike Conway]

Tried it, but got mixed up when .wav's required renaming. So reverted to audio.

That's so you can't name a second file the same name. Just put a "2" or make a little change to the name. That way, when it saves to the folder, you know which one is the 44.1khz file and which is 48khz.

I've actually brought in WAV files songs and sound FX, from the computer to the OASYS. It works fine.