Emu XL-7 & MP-7 User's Group

To be honest, there's really no need to use one (I don't think they even 
exist).  Even though the signal is quiet, you can still normalize it to a 
hotter level and have it be cleaner than any analog signal you can record 
from the XX-7.  The only area that the quiet signal is bothersome IMO is 
when you're overdubbing into another sequencer and the other tracks are 
significantly louder.  It can be a pain to hear what you're trying to add, 
no?   In this case, it's best to just turn down the other tracks until 
you're done recording, and then normalize the new XX-7 part.

BTW, if you EVER normalize a digital signal, regardless of if it's the 
XX-7 or anything else, NEVER normalize to 0dB!!!!  There's some very valid 
reasons for this, mainly that some DAC's will actually create a signal 
that clips from intersample modulations it perceives as greater than 0dB!. 
 Sounds impossible, but it's a well known fact.  If you need to normalize 
a track, I recommend always normalizing no higher than -2dBFS.  Same thing 
applies when you're doing a final mix or master on a DAW, the final file 
should never be higher than -0.5dBFS.  Any hotter, and some CD players 
will actually clip the signal resulting in subtle, but noticeable 
distortion.

Just an FYI.
rEalm






The S/PDIF signal isn't suffering from any hum. It's just weak. Does 
anybody know if there are any signal enhancers for this problem?
Maik


 






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Yo rEalm -

another POV on this (my own, altho I am not glued to this method...)

the quality of Normalization does depend on the software that is being
used to Normalize, and of course we all know that in the N process the
audio takes a hit, so to a degree one has to weigh which hit they
prefer - N or A/D.

I find myself continually more drawn to an analog re-recording of the
XL7 - and have not found noise to be an issue. For me the analog
re-recording gives synths a bit more definition and I do like the
sound of the Emu D/A's - but of course this method does require good
A/D's and a mixer or mic pre or other signal booster, and it does add
a coloration of the orginal sound. 

That said - depending on what DAW software is being used one should be
able to boost the XL7 spdif signal before/during the recording.

I hear time and again that the optimum place to aim for in a 24 bit
recording of individual tracks is -6 db (Bob Katz, the Metric Halo
guys et al), for whatever that is worth.

rock forward

Ed


 


--- In xl7@yahoogroups.com, erik_magrini@B... wrote:
> To be honest, there's really no need to use one (I don't think they
even 
> exist).  Even though the signal is quiet, you can still normalize it
to a 
> hotter level and have it be cleaner than any analog signal you can
record 
> from the XX-7.  The only area that the quiet signal is bothersome
IMO is 
> when you're overdubbing into another sequencer and the other tracks are 
> significantly louder.  It can be a pain to hear what you're trying
to add, 
> no?   In this case, it's best to just turn down the other tracks until 
> you're done recording, and then normalize the new XX-7 part.
> 
> BTW, if you EVER normalize a digital signal, regardless of if it's the 
> XX-7 or anything else, NEVER normalize to 0dB!!!!  There's some very
valid 
> reasons for this, mainly that some DAC's will actually create a signal 
> that clips from intersample modulations it perceives as greater than
0dB!. 
>  Sounds impossible, but it's a well known fact.  If you need to
normalize 
> a track, I recommend always normalizing no higher than -2dBFS.  Same
thing 
> applies when you're doing a final mix or master on a DAW, the final
file 

> should never be higher than -0.5dBFS.  Any hotter, and some CD players 
> will actually clip the signal resulting in subtle, but noticeable 
> distortion.
> 
> Just an FYI.
> rEalm
> 
> 
> 
> 
> 
> 
> The S/PDIF signal isn't suffering from any hum. It's just weak. Does 
> anybody know if there are any signal enhancers for this problem?
> Maik
> 
> 
>  
> 
> 
> 
> 
> 
> 
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> [Non-text portions of this message have been removed]

the quality of Normalization does depend on the software that is being 
used to Normalize, and of course we all know that in the N process the 
audio takes a hit, so to a degree one has to weigh which hit they prefer - 
N or A/D.

>>>Hmmm, not sure I agree actually (though not sayig you're wrong either). 
 Normalization is a fairly simple operation, you're just adding more bits 
to the existing signal, not altering it or performing any calculation that 
could introduce truncation errors. <<<

I find myself continually more drawn to an analog re-recording of the XL7 
- and have not found noise to be an issue. 
>>>Me neither, the analog outs are certain fine, and I too use those most 
of the time. <<<

That said - depending on what DAW software is being used one should be 
able to boost the XL7 spdif signal before/during the recording.

>>>This is where you'll start to incure errors though.  Any time you raise 
the gain of a digital signal via software (ala, you raise the input fader 
in your DAW) you're performing calculations on the signal.  Truncation 
errors can occur, so the signal will not be as clean as it you had just 
recorded it straight and then normalized after the fact.  That being said, 
lets keep this all in perspective.  Truncation errors occur at roughly 
round -144dBFS in a 24 bit signal, so it's hardly likely you'd be able to 
hear them.  If nothing else, the dither added during bit reduction later 
in the process will be louder than those errors.  The issue with 
truncation errors is that it's cumulative, so the more operations you do, 
the more likely you are to hear them.  Again, these are VERY small in 
terms of how they degrade the signal quality though. <<<

I hear time and again that the optimum place to aim for in a 24 bit 
recording of individual tracks is -6 db (Bob Katz, the Metric Halo guys et 
al), for whatever that is worth.

>>>For recording individual tracks, yes I agree, you're only losing 1 bit 
of resolution in a 24 bit signal, so it's no big deal (for those that 
don't know each bit in a 24 bit signal is about 6dB of level).  My 
comments were geared ONLY towards normalization, and I used the -2dB as an 
"upper" limit.  To be honest, if I record the XL-7 digitally, I just leave 
it at around the -14dB it's typically was recorded at, and don't even 
bother to normalize. 

When I track individual instruments, I too aim for around the -6dB mark. 
Anyone interested in this geeky kind of stuff should definitely get the 
book "Mastering Audio" by Bob Katz, goes into much more detail on all of 
these topics, very highly recommended. <<<

roll backward,
rEalm


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Good Points rEalm

FWIIW (in my limited experience based on the hardware/software that I
own and work on) - I think some software will perform XXX operations
better than other soft/hard. I have experienced over the years what I
have subjectively considered to be "bad Normalization" and "good
Normalization)....

I am currently working with a Creamware SFP system and the Digital
Gain that is available in the SFP mixer is great (imo) - but I guess
what you are saying is that a more sure-fire QC method is
Normalization, not Digital gain...and I can see the logic there.

Which raises the question: In the E4 we do not have to think about
this exact subject in the same way given the variable HEADROOM setting
in the MASTER section (allows digital signal to be raised or lowered)
- do you know who this type of operation figures into all that we have
been discussing here?

curious to know your thoughts

thanks!

Ed

--- In xl7@yahoogroups.com, erik_magrini@B... wrote:
> the quality of Normalization does depend on the software that is being 
> used to Normalize, and of course we all know that in the N process the 
> audio takes a hit, so to a degree one has to weigh which hit they
prefer - 
> N or A/D.
> 
> >>>Hmmm, not sure I agree actually (though not sayig you're wrong
either). 
>  Normalization is a fairly simple operation, you're just adding more
bits 
> to the existing signal, not altering it or performing any
calculation that 
> could introduce truncation errors. <<<
> 
> I find myself continually more drawn to an analog re-recording of
the XL7 
> - and have not found noise to be an issue. 
> >>>Me neither, the analog outs are certain fine, and I too use those
most 
> of the time. <<<
> 
> That said - depending on what DAW software is being used one should be 
> able to boost the XL7 spdif signal before/during the recording.
> 
> >>>This is where you'll start to incure errors though.  Any time you
raise 
> the gain of a digital signal via software (ala, you raise the input
fader 
> in your DAW) you're performing calculations on the signal.  Truncation 
> errors can occur, so the signal will not be as clean as it you had just 
> recorded it straight and then normalized after the fact.  That being
said, 
> lets keep this all in perspective.  Truncation errors occur at roughly 
> round -144dBFS in a 24 bit signal, so it's hardly likely you'd be
able to 
> hear them.  If nothing else, the dither added during bit reduction
later 
> in the process will be louder than those errors.  The issue with 
> truncation errors is that it's cumulative, so the more operations
you do, 
> the more likely you are to hear them.  Again, these are VERY small in 
> terms of how they degrade the signal quality though. <<<
> 
> I hear time and again that the optimum place to aim for in a 24 bit 
> recording of individual tracks is -6 db (Bob Katz, the Metric Halo
guys et 
> al), for whatever that is worth.
> 
> >>>For recording individual tracks, yes I agree, you're only losing
1 bit 
> of resolution in a 24 bit signal, so it's no big deal (for those that 
> don't know each bit in a 24 bit signal is about 6dB of level).  My 
> comments were geared ONLY towards normalization, and I used the -2dB
as an 
> "upper" limit.  To be honest, if I record the XL-7 digitally, I just
leave 
> it at around the -14dB it's typically was recorded at, and don't even 
> bother to normalize. 
> 
> When I track individual instruments, I too aim for around the -6dB
mark. 
> Anyone interested in this geeky kind of stuff should definitely get the 
> book "Mastering Audio" by Bob Katz, goes into much more detail on
all of 

> these topics, very highly recommended. <<<
> 
> roll backward,
> rEalm
> 
> 
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> to which it is addressed and may contain confidential and/or legally 
> privileged material. Delivery of this message to any person other than 
> the intended recipient(s) is not intended in any way to waive privilege 
> or confidentiality. Any review, retransmission, dissemination or other 
> use of, or taking of any action in reliance upon, this information by 
> entities other than the intended recipient is prohibited. If you 
> receive this in error, please contact the sender and delete the 
> material from any computer.
> 
> For Translation:
> 
> http://www.baxter.com/email_disclaimer
> 
> 
> [Non-text portions of this message have been removed]

I think some software will perform XXX operations better than other 
soft/hard. I have experienced over the years what I have subjectively 
considered to be "bad Normalization" and "good Normalization).

>>>Not disagreeing here, I just don't have enough experience with other 
software to verify.  I personally use Wavelab 4 for all of my audio 
editing, and always have.  In "theory", normalization shouldn't involve 
any calculations that would affect audio quality (you're just tacking on 
extra bits), though certainly each manufacturer can have a different way 
of doing it. <<<

I am currently working with a Creamware SFP system and the Digital Gain 
that is available in the SFP mixer is great (imo) - but I guess what you 
are saying is that a more sure-fire QC method is Normalization, not 
Digital gain...and I can see the logic there.

>>>Pretty much hits it on the head.  I think there are vast differences in 
how various software companies calculate things like gain change, so 
certainly some manufacturers are better than others in this regard.  It's 
all about the processing resultion (is the signal path 24 bit, 32 float, 
64bit, etc), and more importantly, how do they deal with any calculation 
remainders.  Some companies used to just 'throw away' the extra bits 
post-calc, ouch! <<<

Which raises the question: In the E4 we do not have to think about this 
exact subject in the same way given the variable HEADROOM setting in the 
MASTER section (allows digital signal to be raised or lowered) - do you 
know how this type of operation figures into all that we have been 
discussing here?

>>>Without knowing how it's implemented, I can't say. I'm totally 
unfamiliar with the Emu sampler range too.  Again, 'in theory' anytime you 
change the gain digitally, you're affecting the quality of the audio.  But 
as I said in the last email, we're talking about VERY minor amounts of 
distortion.  Much lower than the noise floor of tape for instance. <<<

rEalm


 






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--- erik_magrini@... wrote:
> In "theory", normalization shouldn't involve any calculations
> that would affect audio quality (you're just tacking on extra bits)

Not quite.  Normalization involves a multiply by a non-integral number
which leads to error, albeit down below the level of the original bit
depth.  For example, pretend you have a 2 bit system (4 levels).  If
you normalize a signal of 1,2,1,2 you'd have to multiply by 1.5,
leaving you with 1.5,3,1.5,3 which truncates to 1,3,1,3 or rounds to
2,3,2,3.  Either are wrong to the tune of up to a single bit.  While a
single bit is significant in a 2 bit system, it is negligible in a 16
or 20 or 24 bit system.

Wow, that was technical...

And my standard disclaimer, the only reason to record the analog
instead of digital is because you like the color of the XL-7's D/As,
preamps, or A/Ds you capture with.  The digital will *always* be
cleaner.

-Aaron

__________________________________
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Yes very technical, not a clue what you just said :)  Thanks for that 
though Aaron, I never realized normalization involved a multiplication 
process, points to Ed.

rEalm




Not quite.  Normalization involves a multiply by a non-integral number
which leads to error, albeit down below the level of the original bit
depth.  For example, pretend you have a 2 bit system (4 levels).  If
you normalize a signal of 1,2,1,2 you'd have to multiply by 1.5,
leaving you with 1.5,3,1.5,3 which truncates to 1,3,1,3 or rounds to
2,3,2,3.  Either are wrong to the tune of up to a single bit.  While a
single bit is significant in a 2 bit system, it is negligible in a 16
or 20 or 24 bit system.

Wow, that was technical...


 






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