Fascination Street says reamp/di signal is too low.....

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jeez... looks like I miss-read the OP...
I was thinking that that was all about the final wet signal from the Axe...
so please feel totally free to ignore my crap more than usual...

if this is all about the signal level of the dry guitar.. as it enters the Axe
surely the Axe has to record this at the level receieved..
so that when it's played back into the Axe it is exactly the same..
so the Axe sounds exactly the same as if you jacked your guitar straight into it...

you'd only want a much hotter level if you were to stuff it from DAW / tape out to a real amp for reamping
again though.. if you actually needed to do this, you can normalise the dry guitar to heat up the level prior to stuffing it into an amp..
and if you're reamping back into the Axe, just leave the dry guitar level as is...

are some of you guys implying that the reamped dry signal would lack the definition of a real guitar jacked into the front panel due to the signal level being quite low???

lil' note: I don't actually do any of this...
I jack into my guitar striaght into the AI and record a great big signal from guitar to DAW [whilst monitoring the Axe]
when I reamp I don't use the USB.. AI -> RedEye -> Axe Instr input
and I mess with the signal level from the DAW to achieve the same / very similar red tickling action as if I jacked into the front...
to my ears [ok.. so they're a bit fkd] it sounds fine..
 
Who cares ! :)
I cranc the converters no matter what I record. Just to get the best soundquality.
I'm not talking noise figures , but correct way of copying the guitarsignal with minimum bit crushing distortion or what ever
anybody wanna call it.
That can be achieved with the axe II and some creative routing.
No matter what anybody says (even including Cliff) I will still record att full blast.
There's only one guy on this thread who gets my point anyway.

No offence

:D:D
 
Hmm but from what I figured out the reamp level is lowered by the same ammount you are adding in the input trim.
If you have it tickling red it should mean it hits full scale digital, then if you look at the reamp waveform it actually never reaches that value.

No, tickling the red means full-scale from the converters. However, this is not the same as full-scale digital representation. The Axe-Fx II is very clever in that it allows you to slam the converters for best SNR but the internal binary value is unchanged.

When using USB or Digital In the meters represent the level re. full-scale so this is normal.

I've tested various guitars and I get the yellow input LED to light when reamping. This indicates -20 dB re. FS. This means that you have 124 dB of dynamic range which far exceeds the dynamic range of any guitar.
 
No matter what anybody says (even including Cliff) I will still record att full blast.
There's only one guy on this thread who gets my point anyway.

No offence

:D:D

No matter what logic says, you'll continue doing it your way? No offense :razz

There's more than friggin' 144 dB's of dynamic range in 24 bits. Are you aware of how much more than you'd ever need this is?? A signal peaking at -48 dBFS is a full 16 bits/ 96 dB, still more than you need from an electric guitar. In the old 16 bit days it was a good idea to stay pretty close to max when recording raw tracks. It is pointless with 24 bits. Seriously. It's not even a good idea, as the analog stages will likely not sound their best at that point.

Or are you going the analog route? If so yes, it's a good idea to raise the signal level before sending it out, but that would be to improve s/n ratio in analog stages.
 
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We went from 16 to 24 of a reason. I'm gonna use them all .
Period !
(Do what ever you want)
Check how sound engineers record.
With high gain sounds we amplify the guitarsignal
100000 of times. We better get the lowest levels right before we magnify them that much.

:)
 
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By the way , how can you possible know how much "dynamic range" my ears need ?
It's just a bunch of numbers.

No matter what logic says, you'll continue doing it your way? No offense :razz

There's more than friggin' 144 dB's of dynamic range in 24 bits. Are you aware of how much more than you'd ever need this is?? A signal peaking at -48 dBFS is a full 16 bits/ 96 dB, still more than you need from an electric guitar. In the old 16 bit days it was a good idea to stay pretty close to max when recording raw tracks. It is pointless with 24 bits. Seriously. It's not even a good idea, as the analog stages will likely not sound their best at that point.

Or are you going the analog route? If so yes, it's a good idea to raise the signal level before sending it out, but that would be to improve s/n ratio in analog stages.
 
Thomas - just trying to understand where you're coming from..

are you suggesting that a signal recorded at a lower level will have less detail as a direct result of it's lower level?
 
Yes ! :)
That's what i'm saying.
If you zoom in on a picture with a low number of pixels
the picture will look like a chess board or something.

If you strum a chord on a guitar I bet the voltagelevel is
only parts of a millivolt when it rings out (maybe even less, I can meassure it if somebody is interested).

Then we amplify it thousands of times in high gain sounds.
If the computor are using resourses to record 24 bits its only stupid to not
use them.

Of course there are high slope lp-filters in the converters that takes away the
"chess board effect" on the signal. But still , why not use all "pixels" ?

So , yes , less detail I guess . I'm not concerned about signal to noise ratio.
Maybe my ears can't here the difference . Anyway , I am not taking the chance as long as
my computer are doing the job of recording 24-bits.

Thomas - just trying to understand where you're coming from..

are you suggesting that a signal recorded at a lower level will have less detail as a direct result of it's lower level?
 
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The volume should not make any difference, either you are capturing a digital representation at 24 bits or you are not. If you turn down, it does not start using less bits that is not how digital works.
 
I think Thomas is worried about not getting enough input into the converters so that the maximum dynamic range from his guitar, uses all 24 bits.

There are other reasons for using 24 bits or higher beside input resolution (think upsampling). But I think that's outside of the scope of this thread.
 
ok... here's how I understand this.. [maybe someone from FAS that really knows this stuff can chime in and set a few things straight]..

sampling..
when a signal is sampled, the waveform's level is being measured..
a sample rate of 48 kHz means that the waveform's level [essentially volume] is being measured 48,000 times per second..
the encoding bit depth determines just how many values there are that can represent level
24-bit encoding means that there are 16.7 million levels on the scale from no level at all, right up to the maximum possible level that can be measured
so... 48,000 times in a second you get...
- measure how loud the signal is..
- find where this falls on my scale of 16.7 million levels
- turn this level into a binary number [encoding]
- now measure the next slice of the waveform

then at the other end...
- I have this binary number
- on my scale of 16.7 million values this means it's 'this loud' [decoding]
- create this slice of the waveform and place it next to the previous slice

all these slices placed one after the other reconstruct the waveform..
-

if all of the detail [the contours] of the waveform can be measured [and therefore encoded], it should not matter if there is a difference in level
because the contours of the waveform will be exactly the same and the combination of sample rate and bit depth will be able to represent these contours just the same..

that is unless.......

the signal level is so high that the peaks of the highest levels are above the maximum level that can be represented by the maximum of our 16.7 million possible values
the result here will be to chop off the tops of these peaks.. effectively making tall spikes into stunted, squared off stumps.. this is digital clipping..

or

if the signal level is so low that the extreme troughs in the waveform's contours are clipped off in exactly the same way..
this means that the extreme troughs are chopped off and so there is no level at all at the moments

I think what Cliff is saying, is that the dry waveform from the guitar is well within the range of values to faithfully measure and therefore represent all of the contours of the waveform..
so the fact that one waveform has more level than another should be arbitrary provided that the peaks and troughs can all be measured correctly
all of the waveform's details are therefore still captured..

so... provided that the level is within the required dynamic range.. there should be no issue with detail

EDIT: a thing to listen for Thomas. on the recorded dry waveform, do you hear any spurious artefacts?
if you did, then this would certainly prove that the source audio was not being correctly sampled..
thing is... you don't hear these issues... which proves that the source audio is being correctly sampled and encoded..
remember - digital audio is not forgiving like analogue.. things don't gradually start sounding bad and then get gradually worse..
things that are 'wrong' are wrong in a hugely obvious way because the encoding will be wrong..
clicks and pops, strange momentary metallic sounds...
 
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I think Thomas is worried about not getting enough input into the converters so that the maximum dynamic range from his guitar, uses all 24 bits.

There are other reasons for using 24 bits or higher beside input resolution (think upsampling). But I think that's outside of the scope of this thread.

no.. you'll always be using 24 bits...
Thomas is concerned that a low level will squeeze the detail out of the recorded signal..
if the level is extremely low then yes this can happen...
but I think that this is not the case for what is being discussed here because the signal level falls well within the limits that could cause issues..
 
Yes , great explanation ! That sounds (pretty) right !
Anybody that has been studying a/d conversion can chime in :) !

One thing though !
The lower the signal gets the less approximate levels will be left for
"copying the shape" of it (or representing it) (like pixels in a picture) .

Some professional recording equipment is now using 32 bit resolution.
The reason for that can not be to use as much unnecesary computorpower as possible :) .

Some people thinks that an electric guitar needs less sound quality than other signals.
I can't see why. Who knows what pickups and wood everybody is using. Think about how many times
it is being amplified.

After everytime thinking about this
I still come to the same conclusion:

We record 24 bits . Why not let the signal touch all of them.
The only drawback is that the signal has to be attenuated afterwards.
(Before being sent to an amps input/ axe amp sim)
But that's everyday -stuff for guys in the signal bussiness.


ok... here's how I understand this.. [maybe someone from FAS that really knows this stuff can chime in and set a few things straight]..

sampling..
when a signal is sampled, the waveform's level is being measured..
a sample rate of 48 kHz means that the waveform's level [essentially volume] is being measured 48,000 times per second..
the encoding bit depth determines just how many values there are that can represent level
24-bit encoding means that there are 16.7 million levels on the scale from no level at all, right up to the maximum possible level that can be measured

if all of the detail [the contours] of the waveform can be measured [and therefore encoded], it should not matter if there is a difference in level
because the contours of the waveform will be exactly the same and the combination of sample rate and bit depth will be able to represent these contours just the same..

that is unless.......

the signal level is so high that the peaks of the highest levels are above the maximum level that can be represented by the maximum of our 16.7 million possible values
the result here will be to chop off the tops of these peaks.. effectively making tall spikes into stunted, squared off stumps.. this is digital clipping..

or

if the signal level is so low that the extreme troughs in the waveform's contours are clipped off in exactly the same way..
this means that the extreme troughs are chopped off and so there is no level at all at the moments

I think what Cliff is saying, is that the dry waveform from the guitar is well within the range of values to faithfully measure and therefore represent all of the contours of the waveform..
so the fact that one waveform has more level than another should be arbitrary provided that the peaks and troughs can all be measured correctly
all of the waveform's details are therefore still captured..

so... provided that the level is within the required dynamic range.. there should be no issue with detail
 
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what's an example of a case where the signal might fall beneath these limits? the tail end of a cymbol crash?
 
The lower the signal gets the less approximate levels will be left for
"copying the shape" of it (or representing it) (like pixels in a picture)

I don't think this is strictly true
it's not a case of 'what is left' because you only need to capture and represent the contours of the waveform for their entire maximum dynamic range
provided you can do this without introducing any distortion then the contours will be the same..
if the same waveform is sampled again at a higher level, the contours will be the same.. just louder..
all those possible values that fall below the contours of the waveform just won't make any difference
because you've essentially just drawn exactly the same 'picture' of this waveform but it's higher up the scale..
all you need to do is to be able to capture all of the peaks and troughs..the full dynamic range..
if the low levels lose any granularity at all, would it be lost on any information in the source audio that was meaningful / noticeable??
like the first few millionths of a second when the pick strikes the string maybe?? would this even be perceivable to the human ear / brain??

Some professional recording equipment is now using 32 bit resolution.
The reason for that can not be to use as much unnecesary computorpower as possible :) .

the whole point of very high sample rates and bit depths is because the audio will be manipulated [processed]
if source audio is recorded at 12-bit / 44.1k and again at 32-bit / 96k, provided that no processing takes place, I doubt you'd be able to perceive a difference because beyond 12-bit, a human ear cannot distinguish differences...
however... when you are processing the audio [changing bass, treble, adding reverb etc], you are essentially slinging the binary representation of your source audio through a calculator.. at 12-bit / 44.1k you will certainly hear the numbers being rounded up / rounded down in the form of artefacts / aliasing..
the higher resolutions push these round-up / round-down quantisations out to so many decimal places that when the processed audio is finally converted back into 16 bit / 44.1k you'll not hear these artefacts.. basically, you hide all the errors by making them extremely tiny..
 
Clarky, expanded this somewhat, but as long as the signal is within the range, and you sample with 24 bits, you will always use 24 bits. Do you think that if the volume is dropped the sample will use less than 24 bits ?

no.. the bit depth that audio is encoded at remains constant irrespective of level..
it's fixed..
think of it this way...

2-bit = 2 to the power of 2 = 4
this means that you could only measure 4 possible levels [min, max and 2 in between]
and would clearly sound nasty..

24 bit is 2 to the power of 24 = 16.7 million
this is lots of possible levels on the 'scale' you use to take the measurements..

you do not change the bit depth in real time.. it would be like switching languages mid-way through a conversation..
ahh... jeez.. I know know people that do this... including me.. lmao..

24-bit is always 24 bit.. how loud you play, how high or low the signal is will not effect or alter this...
it's a constant

and likewise, a CD is 16-bit.. and that never changes either
 
no.. the bit depth that audio is encoded at remains constant irrespective of level..
it's fixed..
think of it this way...

2-bit = 2 to the power of 2 = 4
this means that you could only measure 4 possible levels [min, max and 2 in between]
and would clearly sound nasty..

24 bit is 2 to the power of 24 = 16.7 million
this is lots of possible levels on the 'scale' you use to take the measurements..

you do not change the bit depth in real time.. it would be like switching languages mid-way through a conversation..
ahh... jeez.. I know know people that do this... including me.. lmao..

24-bit is always 24 bit.. how loud you play, how high or low the signal is will not effect or alter this...
it's a constant

and likewise, a CD is 16-bit.. and that never changes either

Clarky that is exactly what I mean whatever bit rate a digital sample is it is invariable. If it is 16 bit its 16 bit, if its 24 bit its 24 bit.

However I am asking Thomas, does he think by volume lowering bit rate changes as that is what he seems to be saying.
 
absolutely...
that's not quite what's being debated here though..

Thomas is talking about the loss of detail due to low signal levels...
and I totally agree that if the signal is way too low that detail can be lost [or rather, the signal is not correctly captured]..
but this ain't 'a bit softer'.. this is extremely hugely low level...
but I think that in this case, no detail is lost during the reamping process.. and if any is lost, it is too minute to be meaningful..

jeez... I reamp analogue anyhow... so I'm not even involved in any of this... lmao...
 
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