Did you know that you probably just are listening to 19 or 18 bit sound?

so you've been programming digital audio since the 80's and you've just discovered that if your signal is not near 0dbfs you are not getting maximum bit-dpeth? My favorite part is your only point of reference is a roland vs880! Hey boray, would you please answer my monitor question.
 
Axel, Yes, it's all about the monitor knob. That part about that a volume control could be used as an D/A doesn't have anything to do with it at all. (it's not about the monitor knob, but about old computers). Not that I understand your comment about it, but anyway... ;) Check the VSPlanet thread, it's a lot better: http://www.vsplanet.com/ubb/ultimatebb.php?ubb=get_topic&f=1&t=013032

sweetnubs, It's about if the monitor knob is before or after the D/A. ....VS880???? I don't have one?????

/Anders
 
Clueless..........

So you think that when you perform any gain-related DSP functions, you are affecting the inherent bit-depth of the recorded digital signal????????????

bwa-ha-ha!!!!!!!!!!!!!!!!!!!!! :rolleyes:

That really is funny..........!

(It also once more demonstrates your complete and total ignorance on the subject.............)
 
Blue Bear Sound said:
So you think that when you perform any gain-related DSP functions, you are affecting the inherent bit-depth of the recorded digital signal????????????
Click to expand...

Not of the recorded signal, but when it reaches the D/A converter... Then it won't use it's whole dynamic range if it's not played loud enough. Don't you agree?
 
No - I don't agree..........

...you are confusing amplitude manipulation (which faders control) with bit-usage during the sampling process.....

Two entirely SEPARATE functions that have nothing to do with each other.

Still no surprise that you can't grasp this concept...................... :rolleyes:
 
From the software developper point of view...

It seems to depend...

A simple implementation of any master fader in the digital domain would be a simple software multiplication of the signal.

If you weaken a signal by multiplying it with number small than the equivalent of 1, the numbers get smaller. Smaller numbers mean less bits, that's not really new. So the digital noise is at a higher ratio to the signal... Or the S/N is worse...

And that might be the problem... I'm pretty sure that the fader act in fact like a multiplication and alas, I now am convinced that the monitor level knob is still in digital domain...

A very simple way to test this would be to record something, turn the monitor out to the minimum val and the amp up the same way. The signal should get more and more harsh, as the digital disortion via quantization is increased...

But Boray, you're really right - it is easier to understand what you meant in the other thread... I'll post you something over there... The situation in here does not seem to be too lucky, simply as both sides are too tense to be able to really try to understand what the other one says...

Ciao,

aXel

BTW: I've thought about it and I won't care too much. I only record at 16bit/44.1k so I'm quite unlikely to hear 19bit :D
But it might explain why I had some strange noise once when I had my amp too loud and was just turning down the monitor volume... But my ears are not too good, so it might have been anything else, too :D

aXel
 
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Can ya alls remind me never to purchase any device that has a master fader that is pre DAC in the scheme. I want something that does a DAC before the master fader so Im not losing bits.

Why is a bit that represents silence thought of as unused bit therefore it can be trucated? If...in certain cases that the software is designed to recalculate a different bit depth for each position of a fader, the software must be able to know what zero is silence or below a zero crossing. Most digital mixers that Im a aware of run through the DAC prior to any output meant to feed an analog device. That would be headphones and L/R Master, control room out and Aux sends. Boray is trying to tell me that VS' can real time bit convert on mixdown, how do you do a fade on mixdown?
Ive never seen a redbook cd with less than 16 bits?

Im still waiting for someone to tell me how to lose bits without a bit downconversion? If this goes on much longer Im going to get Bob Katz. That will finish it..

SoMm
 
Son of Mixerman said:
Can ya alls remind me never to purchase any device that has a master fader that is pre DAC in the scheme. I want something that does a DAC before the master fader so Im not losing bits.

Why is a bit that represents silence thought of as unused bit therefore it can be trucated? If...in certain cases that the software is designed to recalculate a different bit depth for each position of a fader, the software must be able to know what zero is silence or below a zero crossing. Most digital mixers that Im a aware of run through the DAC prior to any output meant to feed an analog device. That would be headphones and L/R Master, control room out and Aux sends. Boray is trying to tell me that VS' can real time bit convert on mixdown, how do you do a fade on mixdown?
Ive never seen a redbook cd with less than 16 bits?

Im still waiting for someone to tell me how to lose bits without a bit downconversion? If this goes on much longer Im going to get Bob Katz. That will finish it..

SoMm
Click to expand...

Your first point with the master fader --- the perfect bit depth would be kept with an analog master fader... But I go directly to my CD burner so I would not be able to use it at all. No matter what, m VS880EX was my first real machine (except from my tascam 4-track very analogue cassette thingy and my completely untimed PC in which I had to find the correct timing for every single track manually). From there to looking into the internet took me some time... (I have to admin that my first thoughts were that it was the machine not me that made every thing sound so bad :D - then I started buying some books and then I came to VSplanet and to Homerecording)


I think I understood what Boray meant, so I trying to put in into other words... I can understand your point of view that silent parts that are missing the higher order bits cannot be seen as parts with lower bit depth. But if you have whole songs that way and you normalize them in order to get them onto the CD, you MIGHT have some lower order bits that are never used (depending on how low your volume was). I assume that a Roland master fader gets an amplitude reduction by factor 4 simply by shifting the (signless part) of the signal two bits down. The two lowest bit are simply discarded this way. If you normalize afterwards, You'll have in fact a 14 bit signal made from a (lossless MAS at the VS) recording of a 16 bit CD. You'll even get the same signal forms if you do the same shift down, use a DAC and then reamp by the factor 4... Do you understand what I mean? Sometimes it is a little hard to explain all my thoughts in English...

So it's not the VS doing a resampling, but the whole process... The problem should be similar when using other techniques for getting the master volume of the burnt CD back to its original level...

(If I misunderstood you, Boray, then please tell me...)

I hope I could help clarifying the thoughts...

aXel
 
volltreffer said:

... Do you understand what I mean?
Click to expand...
I think your telling me essentially that VS software is moving what it considers to be the LSB and truncating it down to a smaller bit depth. Once its been truncated you can't get it back except through redithering. ATRAC tries to do sort of the samething, but it does it based on frequency overlap etc, which is bad in alot of cases. The scary thing is, not all "0" are silent, how does the software know what bits to move to a lesser significance before truncating. So far Id say VS has got some serious problem. If your monitoring anything less than 100% of the source then you risk mixing something different thats going to CD? Or even worse, your making tracking descisions by monitoring something unknown. Am I alone in these perceptions?

SoMm
 
volltreffer and Son of Mixerman:

What I ment was not that the least significant bits of the recorded sound gets unused (=truncation), but that the most significant bits of the Digital-to-Analog converter remains zero/unused. But this is however not true! It would be if negative/signed values was represented as I thought they were. Or for an A/D designed to only set positive values, to for example to set a voltage between 0 and 5 volts. But it isn't this way for audio data. Not for signed values and not even for waveforms represented with just unsigned values (having the 0 volt offset at half of the range). I thought specific bits was unused with signed values, but it isn't - because all the bits gets inverted when representing a negative value.

So IF a waveform was only made up of positive values, like "half" of an waveform, then you could use that "0 to 5 volts converter" and then bits would really get unused and always zero when all the values are lower than half of the resolution. So, you can say that by only playing at half volume, then at any time, the audio COULD have been represented with one bit less, even if not one specific bit always is unused.

That is if negative values are represented according to standards.

The most important thing is that you just get access to half of the dynamic resolution by playing at 50% of the total volume. So if you have your monitor knob at 33, then you are only using 33% of the total dynamic range of your converter!

/Anders
 
Boray said:
volltreffer and Son of Mixerman:

What I ment was not that the least significant bits of the recorded sound gets unused (=truncation), but that the most significant bits of the Digital-to-Analog converter remains zero/unused. But this is however not true! It would be if negative/signed values was represented as I thought they were. Or for an A/D designed to only set positive values, to for example to set a voltage between 0 and 5 volts. But it isn't this way for audio data. Not for signed values and not even for waveforms represented with just unsigned values (having the 0 volt offset at half of the range). I thought specific bits was unused with signed values, but it isn't - because all the bits gets inverted when representing a negative value.

So IF a waveform was only made up of positive values, like "half" of an waveform, then you could use that "0 to 5 volts converter" and then bits would really get unused and always zero when all the values are lower than half of the resolution. So, you can say that by only playing at half volume, then at any time, the audio COULD have been represented with one bit less, even if not one specific bit always is unused.

That is if negative values are represented according to standards.

The most important thing is that you just get access to half of the dynamic resolution by playing at 50% of the total volume. So if you have your monitor knob at 33, then you are only using 33% of the total dynamic range of your converter!

/Anders
Click to expand...
...except on Tuesdays and Saturdays, when the exact opposite of this is true...

Oh - and also... during a full moon.... and every 2nd total eclipse of the sun....

:rolleyes:
 
Son of Mixerman said:
Or even worse, your making tracking descisions by monitoring something unknown. Am I alone in these perceptions?

SoMm
Click to expand...

I'm afraid that this is the case; but I'm also afraid most 'cheaper' all-in-one workstaiotns do this. In fact I assume that even most digital mixers work this way, but I would not swear on this...

I was really often not too fond of the VSs MTK sounds and thought that they were too harsh. I'm no longer perfectly sure if it was them or the decreased monitoring volume I used...

But I also have to admit: I'm definitely no pro, so I'll keep on recording and mixing and I still think that my little machines give me possibilities of recording I could only dream of some 15years ago... I simply have the possibility of recording and mixing with a quality that is close to pro, and the worst mistakes come from my missing experience...

aXel
 
boray,i am highly confused.

you are saying that fader levels change the bit res right?

so if this is true than if i do a mix with all the faders low i would be mixing in a lower bit res than if i were mixing at a higher level regardless of my coverters?
 
MaskedMan.....

I suggest you take ANYTHING posted by our king of misinformation here (Boray) with a HUGE grain of salt unless proven correct by a far more reliable source...........
 
maskedman72 said:
boray,i am highly confused.

you are saying that fader levels change the bit res right?

so if this is true than if i do a mix with all the faders low i would be mixing in a lower bit res than if i were mixing at a higher level regardless of my coverters?
Click to expand...

Not neccesarily if your mixer uses floating point numbers and you boost the signal again before it reaches your D/A or ends up on your CD. If your mixer don't use floating points (or doesn't use a bit depth greater than your recordings), then setting your fader very low, bounce it to another track and then boost it to the same level again would degrade the sound.

If you record 24 bit sound with a 24 bit A/D, decrease the volume to 50% and play it through a 24 bit D/A converter, then this D/A converter will only (at most) use half of it's dynamic range = the sound could just as well had been represented with 23 bits instead of 24.

Check the VSPlanet thread instead, it's much better and more interesting: http://www.vsplanet.com/ubb/ultimatebb.php?ubb=get_topic&f=1&t=013032

/Anders
 
If your mixer don't use floating points (or doesn't use a bit depth greater than your recordings), then setting your fader very low, bounce it to another track and then boost it to the same level again would degrade the sound.



boray, i thought that there was no loss of sound quality while doing digital bouncing.
90%of the things you are talking about i do not have the knowledge to challenge(although they sound pretty wild) but if you are correct than a few of the basic concepts i(and many others) have come to know are totally wrong.

at this point either you could be insane...or i could be really dumb.

bluebear-i am still confused(mabye even more now)

so if i am using 24 bit recorders and my mixer is less than 24 bit the recorded audio is not 24 bit?

if my mixer and recorders are both 24 bit than i have to have all of the faders at a certan level to get those 24 bits to play back?(cause fader moves do effect the bit res you are saying)
how can you mix if your faders have to be at a certan point?
wouldnt having great converters be pointless than?
 
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Boray said:
Check the VSPlanet thread instead, it's much better and more interesting: http://www.vsplanet.com/ubb/ultimatebb.php?ubb=get_topic&f=1&t=013032
Click to expand...
Most of it reminds me of what chimps might sound like theorizing on the intricacies of human evolution.......... no facts, no basis in theory, no rational reasoning - just a few buzzwords interspersed with a lot of nonsense and conjecture, mostly by Boray.

What's really funny is that you've managed to get some people to believe your crap..... which is both sad and humourous at the same time....

:rolleyes:
 
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