😱 paste & copy from JRiver site :
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Because the audio path is fully 64 bit, any adjustments to volume, bitdepth and sample-rate are mathematically lossless.
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I understand their position, their paying customers are looking for such statements.
Volume control in 64 bit produces same first 24 bits as volume control in 32 bits - actually the same first 32 bits.
Not embeded in their free (just for music) version : Jbox ?
It's an odd story, it's like if you writed cutting an air in 4 is getting you 4 airs !
It's an odd story, it's like if you writed cutting an air in 4 is getting you 4 airs !
Not embeded in their free (just for music) version : Jbox ?
Probably the same code
It's an odd story, it's like if you writed cutting an air in 4 is getting you 4 airs !
??
S/N ratio due to bit loss may be preferable to a physical volume control depending on the noise characteristics of the preamplifier. Turning a pot counter clockwise also degrades S/N ratio, the quantum of which will depend on the noise floor of the successive stages.
This is even before you consider the distortion added by these stages.
Most pro soundcards and interfaces use hardware mixing and volume anyway, with a high enough internal resolution that S/N loss is not encountered unless your system gainstaging is horribly off. For the average -20 to -10dB listening range, there should be no issues - at least not to guys who are making the music.
I prefer digital volume control when done correctly. Preamplifiers tend to be very fiddly circuits to do properly, and pots don't interface really well with cables when used 'passively'.
I do think though that if the system has correct gain structure, either method works well enough. Most commercial products have too much gain for either method though. I have a Marantz receiver that promises 100w/ch, delivers about 40% of that number, and has 30dB of gain. I'm not sure why a 40W amp would need 30dB of gain.
This is even before you consider the distortion added by these stages.
Most pro soundcards and interfaces use hardware mixing and volume anyway, with a high enough internal resolution that S/N loss is not encountered unless your system gainstaging is horribly off. For the average -20 to -10dB listening range, there should be no issues - at least not to guys who are making the music.
I prefer digital volume control when done correctly. Preamplifiers tend to be very fiddly circuits to do properly, and pots don't interface really well with cables when used 'passively'.
I do think though that if the system has correct gain structure, either method works well enough. Most commercial products have too much gain for either method though. I have a Marantz receiver that promises 100w/ch, delivers about 40% of that number, and has 30dB of gain. I'm not sure why a 40W amp would need 30dB of gain.
Ok thanks, it's all I wanted to know : their x64 bits process doesn't change anything on low bits material like the 16 bits ! So If you use the soft volume attenuation : you loose bits into your 16 bits recccording ! 🙂
They use the same statment iirc with Audirvana and Mac !
Well in any case putt the soft volume at O dB and use a physical analog plot...
They use the same statment iirc with Audirvana and Mac !
Well in any case putt the soft volume at O dB and use a physical analog plot...
Because the audio path is fully 64 bit, any adjustments to volume, bitdepth and sample-rate are mathematically lossless. [/COLOR]
Inmho the important part of the statement is not the 64 bit nor the lossless stuff, but rather the option to increase by resampling thr bit depth of that 16bit material to match the 24 or 32 capability of your soundcard and then apply soft volume control.
If it actually works that way i find it's worth using it... and sell the preamp...😀
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😀 does a 16 bits hairs cuted in 4 parts equal = 64 bits ?
I keep my analog pre....as far the opposite can be proved !
I keep my analog pre....as far the opposite can be proved !
If it sounds too loud, i personally prefer the easy way: soft volume control. Just too lazy to worry about a handfull of bits lost...😛
😀 does a 16 bits hairs cuted in 4 parts equal = 64 bits ?
I keep my analog pre....as far the opposite can be proved !
4 is 2 squared, so 16 bits cut in 4 should be 18 bits. Unless I'm missing your question. As has been stated, 16 bit audio, upsampled and fed through a properly dithered volume control will likely be better performing than your preamp. DAC chips are really good nowadays.
Just make sure your end to end gain is set wisely so as to get the most out of your input. (That's both analog and digital)
Sorry my English :
It's not about upsampling, it's about to know if a 16 bits material is loosing bit depth when such softs (JRiver, Audirvana) are claimiming they "convert" to 64 bits a 16 bits material to allow volume control without bits loss...
It's not about upsampling, it's about to know if a 16 bits material is loosing bit depth when such softs (JRiver, Audirvana) are claimiming they "convert" to 64 bits a 16 bits material to allow volume control without bits loss...
64bits is just the precision of the calculator, not the bits of the material. It is like using pi=3'14159 or pi=3,14: if the measured diameter or radius of your circle has a precision of only one decimal, using 5 decimals for pi is completely useless. Of course you will get from the calculator a 5 decimals result, but the last 4 have no interest at all. You can apply 64 bits corrections to 16 bits material, but it will still be a 16bits material.
Upsampling 16bits to 64 bits is a totally different question, and btw bit depth is never increased so much, just enough to match a value that the dac can physically handle, usually 24, and maybe 32. Then, if attenuation make you lose 4, 6 o 8bits, there will be 16bits left, so no apparent loss of resolution. ( Nor increase in resolution, btw...)
Upsampling 16bits to 64 bits is a totally different question, and btw bit depth is never increased so much, just enough to match a value that the dac can physically handle, usually 24, and maybe 32. Then, if attenuation make you lose 4, 6 o 8bits, there will be 16bits left, so no apparent loss of resolution. ( Nor increase in resolution, btw...)
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So you guess it's not BS marketing, sorry to incist : if there a los of bit depth when playing attenuation with such a soft with a genuine 16 bits reccording... whatever the dac chip : 16, 20, 24 or 32...
Is the concept valid and it is just about the dac bit possibility ? When i talked about hairs cuted in 4 , it was an image of course to say can we multiplicate and attenuate to refall on the same number of hair (the original STILL 16 bits depths but after digital attenuation ?)
To rephrase : can we rize the bit depth of a 16 bits materials ? Is it working differently ?
My English again, I didn't understand if someone said Yes it's possible or Not it's not possible, or at which conditions...
Is the concept valid and it is just about the dac bit possibility ? When i talked about hairs cuted in 4 , it was an image of course to say can we multiplicate and attenuate to refall on the same number of hair (the original STILL 16 bits depths but after digital attenuation ?)
To rephrase : can we rize the bit depth of a 16 bits materials ? Is it working differently ?
My English again, I didn't understand if someone said Yes it's possible or Not it's not possible, or at which conditions...
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I also think there is some hair by 4 cutting in this story, because resampling will not increase the actual resolution of your material. The resolution will be the same , but with a thinner scale.
Inmho, the only point in doing this is not a math or software question, but rather that the hardware, the dac, will work better when fed 24 or 32 bits stuff than 16 bits.
Btw, there is some weird discussion on the web about the non sense of HD digital material, some guys arguing that maths will be maths, and 44100/16 is enough. Imho, the problem is not the math nor the sampling theorem, simply the question is that cheap consumer grade 44/16 hardware suck!😀
Inmho, the only point in doing this is not a math or software question, but rather that the hardware, the dac, will work better when fed 24 or 32 bits stuff than 16 bits.
Btw, there is some weird discussion on the web about the non sense of HD digital material, some guys arguing that maths will be maths, and 44100/16 is enough. Imho, the problem is not the math nor the sampling theorem, simply the question is that cheap consumer grade 44/16 hardware suck!😀
🙂, btw today the bad consumer chips are all minimum 24 bits and cold as the stone... but not sure it is the just the chips alone !
I setuped my 16 bit dacs on very difficult instruments ( mainly : saxo, cello, violin and harmonica) as matrice and it works pretty good as far the clock has a good phase noise and jitter is reduced enough before the chip...
I was just aking myself if I could give up my pre for a one in all pc only !
We are agree : no as far you have not at minima 24 bit depth material played on a 24 bits dac to loose some extra bits not too much important in relation of the gain of your amp (so if not too much bits are used for digital attenuation).
But I was not sure totally about these softs 😱
I setuped my 16 bit dacs on very difficult instruments ( mainly : saxo, cello, violin and harmonica) as matrice and it works pretty good as far the clock has a good phase noise and jitter is reduced enough before the chip...
I was just aking myself if I could give up my pre for a one in all pc only !
We are agree : no as far you have not at minima 24 bit depth material played on a 24 bits dac to loose some extra bits not too much important in relation of the gain of your amp (so if not too much bits are used for digital attenuation).
But I was not sure totally about these softs 😱
Resampling to 24 bits and doing volume control there will obviously not gain you any "new" resolution as that has already been quantized, BUT, ostensibly, you can park that 16 bits of data within the range of the 24 bits, (i.e. a 8 bit volume control), but that assumes that you aren't losing the LSB's to noise anyhow, which you are (probably more like 20 bits of SNR).
But dithering can get you very far, too.
But dithering can get you very far, too.
But dithering can get you very far, too.
Yes, attenuation is bit depth reduction so that dither should be applied.
Btw, what is dither, random noise? 🙄
So what is this all about, mimic carbon (or whatever) track potentiometer noise behaviour to sound more... analogue? 🙄
Some report that different brands/models of potentiometers / attenuators sound better than others... Different tastes in added noise, the Alps, TKD, Noble's taste or whatever...? 🙄
All dithers sound the same, or dither is some kind of perfect noise?😀
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Resampling to 24 bits and doing volume control there will obviously not gain you any "new" resolution as that has already been quantized, BUT, ostensibly, you can park that 16 bits of data within the range of the 24 bits, (i.e. a 8 bit volume control), but that assumes that you aren't losing the LSB's to noise anyhow, which you are (probably more like 20 bits of SNR).
But dithering can get you very far, too.
Spot on explanation.
And you can get even further if you have that special kind of DAC which can do the attenuation itself immediately before analogue output.
GDO- Well, that depends on the dithering algorithm you choose. But, yes, you can push the noise into the stop band, and make it awfully benign to the user. And given the final bits of a sota 24 but DAC are in the noise floor, I'd say it's not going to impart much of any flavor onto the overall response.
And tracking is far better than an n-ganges pot.
Edit-agreed, gwing2!
And tracking is far better than an n-ganges pot.
Edit-agreed, gwing2!
Y
So what is this all about, mimic carbon (or whatever) track potentiometer noise behaviour to sound more... analogue? 🙄
No. But Wikipedia has a good explanation of what dither is.
GDO- Well, that depends on the dithering algorithm you choose. But, yes, you can push the noise into the stop band, and make it awfully benign to the user.
Sure, but know what..., i am not really a purist and believe that straight bit depth truncation of a 16 bit material by soft volume control, even without dither applied, is also benign. Especially with Youtube material i am so much fond of...😀
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