Re: DAT transfer problems

[Neil Rutman]

I don't think it's necessarily a clock problem. I've had similar issues
trying to digitally transfer from my DAT (DA 30) to my computer (Layla 24).
Sometimes I get a static like crackle (frying) sound, sometimes it works
fine. I haven't been able to pinpoint the problem but I never had a digital
transfer problem with any other piece of equipment except my DAT machine.
Could be the antiquated technology isn't up to todays tasks.

A solution is to come out of your DAT analog. You'll never notice the
difference.

Neil R

"Walter Harley" > wrote in message
...
> I've got a DAT that I recorded, of a live show I mixed this week. I'm
> trying to burn a CD for the band. This is something I've done
successfully
> in the past, but it's been a few months. The problem I'm having is that
my
> transfers sound noisy (a weird "frying" sort of noise, perhaps 30dB below
> the main signal) and grainy. The question is "is this a clock problem,
and
> if so, how do I solve it?"
>
> Here are the details:
> =======================
>
> I recorded the DAT off the AES/EBU digital output of a TC Finalizer, onto
a
> Fostex D-5. Monitoring the D-5 analog outputs during recording and in
> playback afterwards, through Mackie 824's, everything sounded clean and
nice
> (at least, given the constraints of the material I was recording). Sample
> rate was 44.1kHz.
>
> I then put the DAT into a Tascam DA-P1 and used its S/PDIF out to go into
a
> M Audio Delta Audiophile sound card (I don't remember the precise model),
on
> a Win2k machine. I'm using a good-quality RCA cable and the connections
> appear solid. I set the Audiophile settings to lock to the clock of the
> incoming signal, and it indicated that it was successfully locked. I then
> recorded the signal, in SoundForge 6.0. Finally, I copied the .wav file
to
> another computer (running WinXP) for editing and playback. (I'm
temporarily
> without monitors on the first computer.)
>
> Playing back the .wav on the second computer, it sounded "okay" but bad
> enough that I thought "did I really fool myself into being happy with that
> on the night of the show!??". The main problem is a noticeable "frying"
> noise in the background, but it also sounds overall slightly grainy and
> distorted. There are no obvious glitches or clipped peaks (and generally
> the recording levels are around -12dBFS, with very rare peaks up to 0; I
was
> using the Finalizer very gently as a compressor and limiter).
>
> So, I tried the transfer again. This time I went out of the AES/EBU
output
> of the Fostex, into the Finalizer (set to "bypass"), and out of the
> Finalizer S/PDIF into the Audiophile. Results better, but still not as
good
> as I think the actual recording is. (I have monitored the Fostex
headphone
> output and it sounds good, but I don't have the Fostex connected directly
to
> monitors at the moment.)
>
>
> Here is the question:
> ======================
>
> Am I right that this is likely a problem with the sound card clock not
> properly syncing to the input signal? And if so, does anyone have
> suggestions on how I could fix this problem?
>
> Thanks!
> -walter
>
>

[Michael R. Kesti]

Neil Rutman wrote:

>I don't think it's necessarily a clock problem. I've had similar issues
>trying to digitally transfer from my DAT (DA 30) to my computer (Layla 24).
>Sometimes I get a static like crackle (frying) sound, sometimes it works
>fine. I haven't been able to pinpoint the problem but I never had a digital
>transfer problem with any other piece of equipment except my DAT machine.
>Could be the antiquated technology isn't up to todays tasks.

You wanna talk antiquated? Heh! I have transfered using AES I/O between
a DA30 and a computer equipped with an original version AdB MultiWav Pro
(digital only) interface card for years, without the benefit of external
clock, without even one occasion of noise or dropout.

In fact, one reason I'm still using a 350 MHz PII and Win98 is that I'd have
to replace my AdB and Music Quest MQX-32m, as they require ISA slots and
have no drivers for later OSes. They keep on working so well that I cannot
justify doubling the cost of a new computer because I would have to replace
them!

>A solution is to come out of your DAT analog. You'll never notice the
> difference.

This is not bad advice. If the digital transfer fails, for whatever
reason, an analog transfer is not an unreasonable fallback. Of course,
the better the AtoD converters the better the result.

I suppose I owe Walter an apology, though, as I haven't been able to offer
any advise that might help solve his problem!

--
================================================== ======================
Michael Kesti | "And like, one and one don't make
| two, one and one make one."
| - The Who, Bargain

[Michael R. Kesti]

Neil Rutman wrote:

>I don't think it's necessarily a clock problem. I've had similar issues
>trying to digitally transfer from my DAT (DA 30) to my computer (Layla 24).
>Sometimes I get a static like crackle (frying) sound, sometimes it works
>fine. I haven't been able to pinpoint the problem but I never had a digital
>transfer problem with any other piece of equipment except my DAT machine.
>Could be the antiquated technology isn't up to todays tasks.

You wanna talk antiquated? Heh! I have transfered using AES I/O between
a DA30 and a computer equipped with an original version AdB MultiWav Pro
(digital only) interface card for years, without the benefit of external
clock, without even one occasion of noise or dropout.

In fact, one reason I'm still using a 350 MHz PII and Win98 is that I'd have
to replace my AdB and Music Quest MQX-32m, as they require ISA slots and
have no drivers for later OSes. They keep on working so well that I cannot
justify doubling the cost of a new computer because I would have to replace
them!

>A solution is to come out of your DAT analog. You'll never notice the
> difference.

This is not bad advice. If the digital transfer fails, for whatever
reason, an analog transfer is not an unreasonable fallback. Of course,
the better the AtoD converters the better the result.

I suppose I owe Walter an apology, though, as I haven't been able to offer
any advise that might help solve his problem!

--
================================================== ======================
Michael Kesti | "And like, one and one don't make
| two, one and one make one."
| - The Who, Bargain

[Michael R. Kesti]

Neil Rutman wrote:

>I don't think it's necessarily a clock problem. I've had similar issues
>trying to digitally transfer from my DAT (DA 30) to my computer (Layla 24).
>Sometimes I get a static like crackle (frying) sound, sometimes it works
>fine. I haven't been able to pinpoint the problem but I never had a digital
>transfer problem with any other piece of equipment except my DAT machine.
>Could be the antiquated technology isn't up to todays tasks.

You wanna talk antiquated? Heh! I have transfered using AES I/O between
a DA30 and a computer equipped with an original version AdB MultiWav Pro
(digital only) interface card for years, without the benefit of external
clock, without even one occasion of noise or dropout.

In fact, one reason I'm still using a 350 MHz PII and Win98 is that I'd have
to replace my AdB and Music Quest MQX-32m, as they require ISA slots and
have no drivers for later OSes. They keep on working so well that I cannot
justify doubling the cost of a new computer because I would have to replace
them!

>A solution is to come out of your DAT analog. You'll never notice the
> difference.

This is not bad advice. If the digital transfer fails, for whatever
reason, an analog transfer is not an unreasonable fallback. Of course,
the better the AtoD converters the better the result.

I suppose I owe Walter an apology, though, as I haven't been able to offer
any advise that might help solve his problem!

--
================================================== ======================
Michael Kesti | "And like, one and one don't make
| two, one and one make one."
| - The Who, Bargain

[Mike Rivers]

In article > writes:

> I don't think it's necessarily a clock problem. I've had similar issues
> trying to digitally transfer from my DAT (DA 30) to my computer (Layla 24).
> Sometimes I get a static like crackle (frying) sound, sometimes it works
> fine.

I haven't had a problem transferring between DAT and DAT and between
DAT and CD recorder digitally. When I was transferring between DAT and
computer, I didn't have a card in the computer with digital I/O so my
transfers were always analog.

However, ground loops work strange magic with digital signals. It's
possible that you have one between your Layla and the DAT and that's
what's making the intermittent noises. When phases are right in the
wrong kind of way, you could be losing a couple of cycles of data.


--
I'm really Mike Rivers )
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo

[Mike Rivers]

In article > writes:

> I don't think it's necessarily a clock problem. I've had similar issues
> trying to digitally transfer from my DAT (DA 30) to my computer (Layla 24).
> Sometimes I get a static like crackle (frying) sound, sometimes it works
> fine.

I haven't had a problem transferring between DAT and DAT and between
DAT and CD recorder digitally. When I was transferring between DAT and
computer, I didn't have a card in the computer with digital I/O so my
transfers were always analog.

However, ground loops work strange magic with digital signals. It's
possible that you have one between your Layla and the DAT and that's
what's making the intermittent noises. When phases are right in the
wrong kind of way, you could be losing a couple of cycles of data.


--
I'm really Mike Rivers )
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo

[Mike Rivers]

In article > writes:

> I don't think it's necessarily a clock problem. I've had similar issues
> trying to digitally transfer from my DAT (DA 30) to my computer (Layla 24).
> Sometimes I get a static like crackle (frying) sound, sometimes it works
> fine.

I haven't had a problem transferring between DAT and DAT and between
DAT and CD recorder digitally. When I was transferring between DAT and
computer, I didn't have a card in the computer with digital I/O so my
transfers were always analog.

However, ground loops work strange magic with digital signals. It's
possible that you have one between your Layla and the DAT and that's
what's making the intermittent noises. When phases are right in the
wrong kind of way, you could be losing a couple of cycles of data.


--
I'm really Mike Rivers )
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo

[Paul Stamler]

"Walter Harley" > wrote in message
...

> I then put the DAT into a Tascam DA-P1 and used its S/PDIF out to go into
a
> M Audio Delta Audiophile sound card (I don't remember the precise model),
on
> a Win2k machine. I'm using a good-quality RCA cable and the connections
> appear solid. I set the Audiophile settings to lock to the clock of the
> incoming signal, and it indicated that it was successfully locked. I then
> recorded the signal, in SoundForge 6.0. Finally, I copied the .wav file
to
> another computer (running WinXP) for editing and playback. (I'm
temporarily
> without monitors on the first computer.)

I think the key may be "I'm using a good-quality RCA cable". If the cable
isn't specifically a 75-ohm *video* cable, you may be getting reflections
that are lousing up the signal. Try it with an RCA-RCA video coax cable.

Peace,
Paul

[Paul Stamler]

"Walter Harley" > wrote in message
...

> I then put the DAT into a Tascam DA-P1 and used its S/PDIF out to go into
a
> M Audio Delta Audiophile sound card (I don't remember the precise model),
on
> a Win2k machine. I'm using a good-quality RCA cable and the connections
> appear solid. I set the Audiophile settings to lock to the clock of the
> incoming signal, and it indicated that it was successfully locked. I then
> recorded the signal, in SoundForge 6.0. Finally, I copied the .wav file
to
> another computer (running WinXP) for editing and playback. (I'm
temporarily
> without monitors on the first computer.)

I think the key may be "I'm using a good-quality RCA cable". If the cable
isn't specifically a 75-ohm *video* cable, you may be getting reflections
that are lousing up the signal. Try it with an RCA-RCA video coax cable.

Peace,
Paul

[Paul Stamler]

"Walter Harley" > wrote in message
...

> I then put the DAT into a Tascam DA-P1 and used its S/PDIF out to go into
a
> M Audio Delta Audiophile sound card (I don't remember the precise model),
on
> a Win2k machine. I'm using a good-quality RCA cable and the connections
> appear solid. I set the Audiophile settings to lock to the clock of the
> incoming signal, and it indicated that it was successfully locked. I then
> recorded the signal, in SoundForge 6.0. Finally, I copied the .wav file
to
> another computer (running WinXP) for editing and playback. (I'm
temporarily
> without monitors on the first computer.)

I think the key may be "I'm using a good-quality RCA cable". If the cable
isn't specifically a 75-ohm *video* cable, you may be getting reflections
that are lousing up the signal. Try it with an RCA-RCA video coax cable.

Peace,
Paul

[Art Cohen]

In article >, says...
> Neil Rutman wrote:
>
> >I don't think it's necessarily a clock problem. I've had similar issues
> >trying to digitally transfer from my DAT (DA 30) to my computer (Layla 24).
> >Sometimes I get a static like crackle (frying) sound, sometimes it works
> >fine. I haven't been able to pinpoint the problem but I never had a digital
> >transfer problem with any other piece of equipment except my DAT machine.
> >Could be the antiquated technology isn't up to todays tasks.
>
> You wanna talk antiquated? Heh! I have transfered using AES I/O between
> a DA30 and a computer equipped with an original version AdB MultiWav Pro
> (digital only) interface card for years, without the benefit of external
> clock, without even one occasion of noise or dropout.
>
> In fact, one reason I'm still using a 350 MHz PII and Win98 is that I'd have
> to replace my AdB and Music Quest MQX-32m, as they require ISA slots and
> have no drivers for later OSes. They keep on working so well that I cannot
> justify doubling the cost of a new computer because I would have to replace
> them!
So I'm not the only one still using an old computer (180MHz Pentium Pro)
and an AdB card because it just keeps working.:)

Art Cohen

[Art Cohen]

In article >, says...
> Neil Rutman wrote:
>
> >I don't think it's necessarily a clock problem. I've had similar issues
> >trying to digitally transfer from my DAT (DA 30) to my computer (Layla 24).
> >Sometimes I get a static like crackle (frying) sound, sometimes it works
> >fine. I haven't been able to pinpoint the problem but I never had a digital
> >transfer problem with any other piece of equipment except my DAT machine.
> >Could be the antiquated technology isn't up to todays tasks.
>
> You wanna talk antiquated? Heh! I have transfered using AES I/O between
> a DA30 and a computer equipped with an original version AdB MultiWav Pro
> (digital only) interface card for years, without the benefit of external
> clock, without even one occasion of noise or dropout.
>
> In fact, one reason I'm still using a 350 MHz PII and Win98 is that I'd have
> to replace my AdB and Music Quest MQX-32m, as they require ISA slots and
> have no drivers for later OSes. They keep on working so well that I cannot
> justify doubling the cost of a new computer because I would have to replace
> them!
So I'm not the only one still using an old computer (180MHz Pentium Pro)
and an AdB card because it just keeps working.:)

Art Cohen

[Paul Gitlitz]

My 1st.guess would always be a bad cable. By the way I have an AdB 8
channel digital converter that still works fine on my computer. It
only works with the Yamaha DSP2416 cards which are somewhat obsolete
now. I would replace them with a Hammerfall or Lynx but it means
replacing the converters as well.

[Paul Gitlitz]

My 1st.guess would always be a bad cable. By the way I have an AdB 8
channel digital converter that still works fine on my computer. It
only works with the Yamaha DSP2416 cards which are somewhat obsolete
now. I would replace them with a Hammerfall or Lynx but it means
replacing the converters as well.

[Rob Reedijk]

Art Cohen > wrote:
>>
>> In fact, one reason I'm still using a 350 MHz PII and Win98 is that I'd have
>> to replace my AdB and Music Quest MQX-32m, as they require ISA slots and
>> have no drivers for later OSes. They keep on working so well that I cannot
>> justify doubling the cost of a new computer because I would have to replace
>> them!
> So I'm not the only one still using an old computer (180MHz Pentium Pro)
> and an AdB card because it just keeps working.:)

PII-300/Win98SE with Core II...runs Wavelab 4.0 very nicely.

I have to admit, the occasional clicks that mysteriously appeared on programs
may all have come from DAT transfers.

Rob R.

[Rob Reedijk]

Art Cohen > wrote:
>>
>> In fact, one reason I'm still using a 350 MHz PII and Win98 is that I'd have
>> to replace my AdB and Music Quest MQX-32m, as they require ISA slots and
>> have no drivers for later OSes. They keep on working so well that I cannot
>> justify doubling the cost of a new computer because I would have to replace
>> them!
> So I'm not the only one still using an old computer (180MHz Pentium Pro)
> and an AdB card because it just keeps working.:)

PII-300/Win98SE with Core II...runs Wavelab 4.0 very nicely.

I have to admit, the occasional clicks that mysteriously appeared on programs
may all have come from DAT transfers.

Rob R.

[Mike]

"Walter Harley" > wrote in message >...
> I've got a DAT that I recorded, of a live show I mixed this week. I'm
> trying to burn a CD for the band. This is something I've done successfully
> in the past, but it's been a few months. The problem I'm having is that my
> transfers sound noisy (a weird "frying" sort of noise, perhaps 30dB below
> the main signal) and grainy. The question is "is this a clock problem, and
> if so, how do I solve it?"
>
> Here are the details:
> =======================
>
> I recorded the DAT off the AES/EBU digital output of a TC Finalizer, onto a
> Fostex D-5. Monitoring the D-5 analog outputs during recording and in
> playback afterwards, through Mackie 824's, everything sounded clean and nice
> (at least, given the constraints of the material I was recording). Sample
> rate was 44.1kHz.
>
> I then put the DAT into a Tascam DA-P1 and used its S/PDIF out to go into a
> M Audio Delta Audiophile sound card (I don't remember the precise model), on
> a Win2k machine. I'm using a good-quality RCA cable and the connections
> appear solid. I set the Audiophile settings to lock to the clock of the
> incoming signal, and it indicated that it was successfully locked. I then
> recorded the signal, in SoundForge 6.0. Finally, I copied the .wav file to
> another computer (running WinXP) for editing and playback. (I'm temporarily
> without monitors on the first computer.)
>
> Playing back the .wav on the second computer, it sounded "okay" but bad
> enough that I thought "did I really fool myself into being happy with that
> on the night of the show!??". The main problem is a noticeable "frying"
> noise in the background, but it also sounds overall slightly grainy and
> distorted. There are no obvious glitches or clipped peaks (and generally
> the recording levels are around -12dBFS, with very rare peaks up to 0; I was
> using the Finalizer very gently as a compressor and limiter).
>
> So, I tried the transfer again. This time I went out of the AES/EBU output
> of the Fostex, into the Finalizer (set to "bypass"), and out of the
> Finalizer S/PDIF into the Audiophile. Results better, but still not as good
> as I think the actual recording is. (I have monitored the Fostex headphone
> output and it sounds good, but I don't have the Fostex connected directly to
> monitors at the moment.)
>
>
> Here is the question:
> ======================
>
> Am I right that this is likely a problem with the sound card clock not
> properly syncing to the input signal? And if so, does anyone have
> suggestions on how I could fix this problem?
>
> Thanks!
> -walter

I'd record in the digital input to the windows recorder option
bypassing the Soundforge or whatever you are using. It should be able
to use the digital in by setting the sound card up as the windows
sound device. I think maybe eliminating one extra link in the chain
you might get it straight.

Mike http://www.mmeproductions.com

[Mike]

"Walter Harley" > wrote in message >...
> I've got a DAT that I recorded, of a live show I mixed this week. I'm
> trying to burn a CD for the band. This is something I've done successfully
> in the past, but it's been a few months. The problem I'm having is that my
> transfers sound noisy (a weird "frying" sort of noise, perhaps 30dB below
> the main signal) and grainy. The question is "is this a clock problem, and
> if so, how do I solve it?"
>
> Here are the details:
> =======================
>
> I recorded the DAT off the AES/EBU digital output of a TC Finalizer, onto a
> Fostex D-5. Monitoring the D-5 analog outputs during recording and in
> playback afterwards, through Mackie 824's, everything sounded clean and nice
> (at least, given the constraints of the material I was recording). Sample
> rate was 44.1kHz.
>
> I then put the DAT into a Tascam DA-P1 and used its S/PDIF out to go into a
> M Audio Delta Audiophile sound card (I don't remember the precise model), on
> a Win2k machine. I'm using a good-quality RCA cable and the connections
> appear solid. I set the Audiophile settings to lock to the clock of the
> incoming signal, and it indicated that it was successfully locked. I then
> recorded the signal, in SoundForge 6.0. Finally, I copied the .wav file to
> another computer (running WinXP) for editing and playback. (I'm temporarily
> without monitors on the first computer.)
>
> Playing back the .wav on the second computer, it sounded "okay" but bad
> enough that I thought "did I really fool myself into being happy with that
> on the night of the show!??". The main problem is a noticeable "frying"
> noise in the background, but it also sounds overall slightly grainy and
> distorted. There are no obvious glitches or clipped peaks (and generally
> the recording levels are around -12dBFS, with very rare peaks up to 0; I was
> using the Finalizer very gently as a compressor and limiter).
>
> So, I tried the transfer again. This time I went out of the AES/EBU output
> of the Fostex, into the Finalizer (set to "bypass"), and out of the
> Finalizer S/PDIF into the Audiophile. Results better, but still not as good
> as I think the actual recording is. (I have monitored the Fostex headphone
> output and it sounds good, but I don't have the Fostex connected directly to
> monitors at the moment.)
>
>
> Here is the question:
> ======================
>
> Am I right that this is likely a problem with the sound card clock not
> properly syncing to the input signal? And if so, does anyone have
> suggestions on how I could fix this problem?
>
> Thanks!
> -walter

I'd record in the digital input to the windows recorder option
bypassing the Soundforge or whatever you are using. It should be able
to use the digital in by setting the sound card up as the windows
sound device. I think maybe eliminating one extra link in the chain
you might get it straight.

Mike http://www.mmeproductions.com

[Len Moskowitz]

Walter Harley > wrote:

>I recorded the DAT off the AES/EBU digital output of a TC Finalizer, onto a
>Fostex D-5. Monitoring the D-5 analog outputs during recording and in
>playback afterwards, through Mackie 824's, everything sounded clean and nice
>(at least, given the constraints of the material I was recording). Sample
>rate was 44.1kHz.
>
>I then put the DAT into a Tascam DA-P1...

There may be tape registration problems here. Try playing it back on
the D-5 to see if it sounds better.

--
Len Moskowitz PDAudio, Binaural Mics, Cables, DPA, M-Audio
Core Sound http://www.stealthmicrophones.com
Teaneck, New Jersey USA http://www.core-sound.com
Tel: 201-801-0812, FAX: 201-801-0912

[Len Moskowitz]

Walter Harley > wrote:

>I recorded the DAT off the AES/EBU digital output of a TC Finalizer, onto a
>Fostex D-5. Monitoring the D-5 analog outputs during recording and in
>playback afterwards, through Mackie 824's, everything sounded clean and nice
>(at least, given the constraints of the material I was recording). Sample
>rate was 44.1kHz.
>
>I then put the DAT into a Tascam DA-P1...

There may be tape registration problems here. Try playing it back on
the D-5 to see if it sounds better.

--
Len Moskowitz PDAudio, Binaural Mics, Cables, DPA, M-Audio
Core Sound http://www.stealthmicrophones.com
Teaneck, New Jersey USA http://www.core-sound.com
Tel: 201-801-0812, FAX: 201-801-0912

[M. Im]

Though it's difficult to say without having hands on...

My guess is that there is a compatibility problem between having the
DAT recorded on the Fostex and playing back on the Tascam.

We assume that DAT is DAT and all is standard. But there are many
things that can go wrong and many conditions that can cause this
result. One, is that the tape may track differently on the heads of 2
different recorders resulting is high error rates and possible damage
to the tape.

Do you know the condition of the Fostex D5? Head usage? How many times
were the heads cleaned and how? Has the alignment been checked? Are
you able to read error rates on the Fostex?

Same questions for the Tascam..

Also, do you see any damage or lines, creases or crinkles on the DAT
tape itself? What brand of DAT tape is it? What is the length of the
DAT tape? Was the tape new or used? Did you fully fast forward and
rewind the tape before recording and before playing on the different
recorders?

What I'm getting at, is that there are many ways to screw up a DAT
recording and if the above conditions haven't been met or checked, the
result can be a very high error rate, (even possible damage to the
tape), and strange noises in digital transfer, but not necessarily in
analog listening as the error correction circuits in the DAT machines
may fix lost data though the ones on your computer card may not do it
as well.

The bottom line: record and playback on the same DAT recorder, make
sure it is well maintained and heads are clean, healthy and tracking
correctly. Use a new DAT tape that's not too long in length and a
known professionally tested brand. Always fast forward and rewind new
tapes before recording and sometimes before playback.

Mike

"Walter Harley" > wrote in message >...
> I've got a DAT that I recorded, of a live show I mixed this week. I'm
> trying to burn a CD for the band. This is something I've done successfully
> in the past, but it's been a few months. The problem I'm having is that my
> transfers sound noisy (a weird "frying" sort of noise, perhaps 30dB below
> the main signal) and grainy. The question is "is this a clock problem, and
> if so, how do I solve it?"
>
> Here are the details:
> =======================
>
> I recorded the DAT off the AES/EBU digital output of a TC Finalizer, onto a
> Fostex D-5. Monitoring the D-5 analog outputs during recording and in
> playback afterwards, through Mackie 824's, everything sounded clean and nice
> (at least, given the constraints of the material I was recording). Sample
> rate was 44.1kHz.
>
> I then put the DAT into a Tascam DA-P1 and used its S/PDIF out to go into a
> M Audio Delta Audiophile sound card (I don't remember the precise model), on
> a Win2k machine. I'm using a good-quality RCA cable and the connections
> appear solid. I set the Audiophile settings to lock to the clock of the
> incoming signal, and it indicated that it was successfully locked. I then
> recorded the signal, in SoundForge 6.0. Finally, I copied the .wav file to
> another computer (running WinXP) for editing and playback. (I'm temporarily
> without monitors on the first computer.)
>
> Playing back the .wav on the second computer, it sounded "okay" but bad
> enough that I thought "did I really fool myself into being happy with that
> on the night of the show!??". The main problem is a noticeable "frying"
> noise in the background, but it also sounds overall slightly grainy and
> distorted. There are no obvious glitches or clipped peaks (and generally
> the recording levels are around -12dBFS, with very rare peaks up to 0; I was
> using the Finalizer very gently as a compressor and limiter).
>
> So, I tried the transfer again. This time I went out of the AES/EBU output
> of the Fostex, into the Finalizer (set to "bypass"), and out of the
> Finalizer S/PDIF into the Audiophile. Results better, but still not as good
> as I think the actual recording is. (I have monitored the Fostex headphone
> output and it sounds good, but I don't have the Fostex connected directly to
> monitors at the moment.)
>
>
> Here is the question:
> ======================
>
> Am I right that this is likely a problem with the sound card clock not
> properly syncing to the input signal? And if so, does anyone have
> suggestions on how I could fix this problem?
>
> Thanks!
> -walter

[M. Im]

Though it's difficult to say without having hands on...

My guess is that there is a compatibility problem between having the
DAT recorded on the Fostex and playing back on the Tascam.

We assume that DAT is DAT and all is standard. But there are many
things that can go wrong and many conditions that can cause this
result. One, is that the tape may track differently on the heads of 2
different recorders resulting is high error rates and possible damage
to the tape.

Do you know the condition of the Fostex D5? Head usage? How many times
were the heads cleaned and how? Has the alignment been checked? Are
you able to read error rates on the Fostex?

Same questions for the Tascam..

Also, do you see any damage or lines, creases or crinkles on the DAT
tape itself? What brand of DAT tape is it? What is the length of the
DAT tape? Was the tape new or used? Did you fully fast forward and
rewind the tape before recording and before playing on the different
recorders?

What I'm getting at, is that there are many ways to screw up a DAT
recording and if the above conditions haven't been met or checked, the
result can be a very high error rate, (even possible damage to the
tape), and strange noises in digital transfer, but not necessarily in
analog listening as the error correction circuits in the DAT machines
may fix lost data though the ones on your computer card may not do it
as well.

The bottom line: record and playback on the same DAT recorder, make
sure it is well maintained and heads are clean, healthy and tracking
correctly. Use a new DAT tape that's not too long in length and a
known professionally tested brand. Always fast forward and rewind new
tapes before recording and sometimes before playback.

Mike

"Walter Harley" > wrote in message >...
> I've got a DAT that I recorded, of a live show I mixed this week. I'm
> trying to burn a CD for the band. This is something I've done successfully
> in the past, but it's been a few months. The problem I'm having is that my
> transfers sound noisy (a weird "frying" sort of noise, perhaps 30dB below
> the main signal) and grainy. The question is "is this a clock problem, and
> if so, how do I solve it?"
>
> Here are the details:
> =======================
>
> I recorded the DAT off the AES/EBU digital output of a TC Finalizer, onto a
> Fostex D-5. Monitoring the D-5 analog outputs during recording and in
> playback afterwards, through Mackie 824's, everything sounded clean and nice
> (at least, given the constraints of the material I was recording). Sample
> rate was 44.1kHz.
>
> I then put the DAT into a Tascam DA-P1 and used its S/PDIF out to go into a
> M Audio Delta Audiophile sound card (I don't remember the precise model), on
> a Win2k machine. I'm using a good-quality RCA cable and the connections
> appear solid. I set the Audiophile settings to lock to the clock of the
> incoming signal, and it indicated that it was successfully locked. I then
> recorded the signal, in SoundForge 6.0. Finally, I copied the .wav file to
> another computer (running WinXP) for editing and playback. (I'm temporarily
> without monitors on the first computer.)
>
> Playing back the .wav on the second computer, it sounded "okay" but bad
> enough that I thought "did I really fool myself into being happy with that
> on the night of the show!??". The main problem is a noticeable "frying"
> noise in the background, but it also sounds overall slightly grainy and
> distorted. There are no obvious glitches or clipped peaks (and generally
> the recording levels are around -12dBFS, with very rare peaks up to 0; I was
> using the Finalizer very gently as a compressor and limiter).
>
> So, I tried the transfer again. This time I went out of the AES/EBU output
> of the Fostex, into the Finalizer (set to "bypass"), and out of the
> Finalizer S/PDIF into the Audiophile. Results better, but still not as good
> as I think the actual recording is. (I have monitored the Fostex headphone
> output and it sounds good, but I don't have the Fostex connected directly to
> monitors at the moment.)
>
>
> Here is the question:
> ======================
>
> Am I right that this is likely a problem with the sound card clock not
> properly syncing to the input signal? And if so, does anyone have
> suggestions on how I could fix this problem?
>
> Thanks!
> -walter

[Peter Larsen]

Mike wrote:

Original post not found, I have to quote it all ...

> "Walter Harley" > wrote in message >...

>> I've got a DAT that I recorded, of a live show I mixed this
>> week. I'm trying to burn a CD for the band. This is something
>> I've done successfully in the past, but it's been a few months.
>> The problem I'm having is that my transfers sound noisy (a weird
>> "frying" sort of noise, perhaps 30dB below the main signal)
>> and grainy.

What is the error count? - if you are short of luck it could be a tape
contact problem in recording, if you are lucky in playback only.

>> The question is "is this a clock problem,

Probably not.

>> and if so, how do I solve it?"

Leaving this to others. Generally speaking: a simple crude indication of
whether the digital transfer is OK is that the durations are identical.

>> I recorded the DAT off the AES/EBU digital output of a TC
>> Finalizer, onto a Fostex D-5. Monitoring the D-5 analog
>> outputs during recording and in playback afterwards, through
>> Mackie 824's, everything sounded clean and nice (at least,
>> given the constraints of the material I was recording).
>> Sample rate was 44.1kHz.

OK, tape plays back on the machine it was recorded on. What is the error
count typically and worst case?

>> I then put the DAT into a Tascam DA-P1 and used its S/PDIF
>> out to go into a M Audio Delta Audiophile sound card (I don't
>> remember the precise model), on a Win2k machine.

You set it to sync to the SPdif and you set it to expect the relevant
sample rate? - the Delta cards are not at all touchy in my experience,
mine happily works with my Sony DAT, the output of which is unacceptable
to my DiO 2448.

>> I'm using a good-quality RCA cable and the connections
>> appear solid. I set the Audiophile settings to lock to
>> the clock of the incoming signal, and it indicated that it
>> was successfully locked.

It needs to know the sample rate, remember: trust is good, but control
is better.

>> I then recorded the signal, in SoundForge 6.0.

There is no reason why this should be a problem. It is a default advice
to avoid too muich happening on screen and on disk while making a
realtime transfer.

>> Finally, I copied the .wav file to another computer
>> (running WinXP) for editing and playback. (I'm temporarily
>> without monitors on the first computer.)

Is the nearby audioshoppe out of Sennheisers, has Harvey discontinued
the MoreME's?

>> Playing back the .wav on the second computer, it sounded
>> "okay" but bad enough that I thought "did I really fool myself
>> into being happy with that on the night of the show!??".
>> The main problem is a noticeable "frying" noise in the
>> background, but it also sounds overall slightly grainy and
>> distorted.

What is the error count. You do not want to make a DAT to computer
transfer without having the DAT's error count display on, at least not
if there is a problem to investigate.

>> There are no obvious glitches or clipped peaks (and generally
>> the recording levels are around -12dBFS, with very rare peaks up
>> to 0; I was using the Finalizer very gently as a compressor and
>> limiter).

Why to the latter, sincerely: why do "something" when you know that you
will probably do something more on the DAW anyway?

>> So, I tried the transfer again. This time I went out of the AES/EBU output
>> of the Fostex, into the Finalizer (set to "bypass"), and out of the
>> Finalizer S/PDIF into the Audiophile.

What did you try to remedy by going through the finalizer? - I asks this
because a bypass in the digital domain is less of a bypass than a bypass
in the analog domain, there could be some gentle influence on the sound.
But nothing drastic, and this is about a drastic problem.

>> Results better, but still not as good as I think the actual
>> recording is. (I have monitored the Fostex headphone output
>> and it sounds good, but I don't have the Fostex connected directly
>> to monitors at the moment.)

My guess is a tape to head contact issue.

>> Am I right that this is likely a problem with the sound
>> card clock not properly syncing to the input signal?

I don't think so. The distortion from missing samples is strange indeed,
it can be very audible on long drawn-out tones, but it is next to
inaudible on things like cymbals.

>> And if so, does anyone have suggestions on how I
>> could fix this problem?

Check the error count in playback. If it is high, then try a SV3800, it
has very good error correction properties or try a "small drum" DAT, say
a small "consumer" Sony portable, they can be very good at playing back
problem tapes.

Please note: I don't know that it is a tape to head contact problem, but
such can sound like you describe it, and step #1 in this troubleshooting
is to find the target to shoot at.

> I'd record in the digital input to the windows recorder option
> bypassing the Soundforge or whatever you are using.

Have you actually tried using the Windows sound recorder for recording
anything of duration? - also, assuming that he uses capable software,
there is time - and a layer of dsp action - to save by recording
directly to 32 bit file on the computer.

> It should be able o use the digital in by setting the sound card
> up as the windows sound device. I think maybe eliminating one
> extra link in the chain you might get it straight.

I think the questionee should change to DDS 2 tapes for lower overall
error count and I wonder just how many hours of work the DAT decks in
question have done since last service overhaul and cleaning .... not
that a completely new tape on a completely clean machine is not also
able (7-9-13) to have or cause "A problem" ....

> Mike http://www.mmeproductions.com


Kind regards

Peter Larsen

--
*******************************************
* My site is at: http://www.muyiovatki.dk *
*******************************************

[andrewunix]

8 Jun 2004 18:46:41 GMT, suggested:
: "Peter Larsen" > wrote in message
: ...
:>
:> It needs to know the sample rate, remember: trust is good, but control
:> is better.
:
: Is that true? I had assumed that I would be best off having the sound card
: lock to the incoming signal, rather than to its own clock. After all, the
: DAT doesn't have a clock input, so it's going to send whatever it wants; I
: don't want the sound card skipping samples just because of clock drift. Is
: there more going on than I thought?

You're correct; you *do* want the sound card to lock to the incoming
signal, or else you will just get garbage. I think that Peter was
suggesting that you set the sample rate manually and THEN switch the clock
source to the input, just in case the software or hardware screws up. I've
had it happen before.

: I'm using Maxell HS-4/60s. That's DDS; I don't know what DDS 2 is, though.
: Can you name a model?

DDS-2 is a grade of DAT that is used for computer backup machines. I've
not heard it recommended over DDS before, though. DDS-2 runs at a higher
data density than DDS, but I don't know if the media is of a higher grade
than DDS.

--
agreenbu @ nyx . net andrew michael greenburg

[Peter Larsen]

andrewunix wrote:

> 8 Jun 2004 18:46:41 GMT, suggested:
> : "Peter Larsen" > wrote in message
> : ...

> :> It needs to know the sample rate, remember: trust is good,
> :> but control is better.

> : Is that true? I had assumed that I would be best off having
> : the sound card lock to the incoming signal, rather than to its
> : own clock.

Yes, with a midiman delta you select lock to sp-dif and tell it what to
expect. The latter for monitoring frequency response reasons.

> : After all, the DAT doesn't have a clock input, so it's going
> : to send whatever it wants; I don't want the sound card skipping
> : samples just because of clock drift. Is there more going on than
> : I thought?

This is well described in the manual.

> You're correct; you *do* want the sound card to lock to
> the incoming signal, or else you will just get garbage.

Yes.

> I think that Peter was suggesting that you set the sample rate
> manually and THEN switch the clock source to the input, just
> in case the software or hardware screws up. I've had it happen before.

We are talking about a Midiman Delta card here. It needs to be told what
to expect ref. the manual.

> agreenbu @ nyx . net andrew michael greenburg


Kind regards

Peter Larsen

--
*******************************************
* My site is at: http://www.muyiovatki.dk *
*******************************************

[andrewunix]

Mon, 14 Jun 2004 12:13:08 +0200, suggested:
: andrewunix wrote:
:
:> I think that Peter was suggesting that you set the sample rate
:> manually and THEN switch the clock source to the input, just
:> in case the software or hardware screws up. I've had it happen before.
:
: We are talking about a Midiman Delta card here. It needs to be told what
: to expect ref. the manual.

Hmmm. I have an Audiophile 2496 (which they list as being part of their
Delta series) and when I switch the clock source to the SPDIF input, it
automatically changes the sample rate to match the source material.

--
agreenbu @ nyx . net andrew michael greenburg

[Peter Larsen]

andrewunix wrote:

> Hmmm. I have an Audiophile 2496 (which they list as being part
> of their Delta series) and when I switch the clock source to the
> SPDIF input, it automatically changes the sample rate to match
> the source material.

True, but you need to set codec sample rate manually for proper playback
frequency response ... at least with my 1010lt.

> agreenbu @ nyx . net andrew michael greenburg


Med venlig hilsen

Peter Larsen

--
*******************************************
* My site is at: http://www.muyiovatki.dk *
*******************************************