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Reducing DAC volume doesn't necessarily decrease bit-depth of audio data.

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DAC means digital to analog converter. Reducing DAC volume can mean reducing voltage level instead of meddling with digital bits before bits are converted to analog signals.

Because DAC produces analog signals, it just needs to produce weaker analog signals instead of meddling with digital bits if DAC volume is reduced.

Reducing application volume reduces bit-depth of digital audio data. Thus, it is a good idea to use 24bit or 32bit audio sampling format in operating system.
 

ZolaIII

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Let's say that there are 64 bit floating point implemented volume controls in certain software player's which scale back to 24 - 32 bit integer with out loos send to DAC. 64 bit floating point is best (up to date) format to do any DSP modifications to the signal thanks to their wide hardware availability (any general purpose CPU cores with FP unit or SIMD 10+ year's back) and easy programability. There were setbacks from integer overflows when rounding back thanks to errors in IEEE floating point standard (which is fixed) and bad hardware implementations (which couldn't be fixed but such are thing of the past now). That software lags heavily behind hardware and that there is no really suitable DSP developer platform (a SoC - SoM developer board with reaal good and good documented DSP and everything else and being part of Linux mainline) to propel this more into right direction is another pair of slippers.
 

earlevel

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Not sure what you're going for here—I guess to stir up some discussion. OK.

In the first two paragraphs, you seem to be considering "reducing DAC volume" to mean turning down the post-conversion analog signal with and analog pot and circuit. OK. But we often turn down DACs with a digital control, and while that does reduce resolution, there isn't much of a practical penalty, because typical listening levels don't use all of the bits anyway.

The last paragraph: We have 24-bit sampling, which is a bit better than we can really achieve, much less need. We don't have 32-bit sampling, and if we did it wouldn't be better than 24-bit, due to fundamental limitations (of things like electrons, and atoms at temperatures we stay alive in—limitations of our ears too, but I'm just referring to the limitations for making a 32-bit DAC). Also, while it's true that 24-bit is better (I like having 24-bit sources myself), 16-bit is no reason to preclude digital volume control, if that's what you have. 16-bit is good enough for full volume listening, and you'd want your gain staging to take advantage of all of it when listening loud, but from there turning down doesn't make anything worse, it actually makes it slightly better as there is more resolution at the quiet extreme of listening.

OK, I bit ;)

PS—I think the "digital vs analog volume control" topic has been discussed ad nauseam in other threads, I probably wouldn't have commented except I'm apparently on a mission to stop people from asking for 32-bit converters—if enough people ask, marketing (not engineering) will surely fulfill your requirement.
 
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antcollinet

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DAC means digital to analog converter. Reducing DAC volume can mean reducing voltage level instead of meddling with digital bits before bits are converted to analog signals.

Because DAC produces analog signals, it just needs to produce weaker analog signals instead of meddling with digital bits if DAC volume is reduced.

Reducing application volume reduces bit-depth of digital audio data. Thus, it is a good idea to use 24bit or 32bit audio sampling format in operating system.
What would be the mechanism for a DAC to "produce weaker analog signals" in order to change volume? Are you talking about a buffered analogue potentiometer (In other words, and analog pre-amp) on the output of the DAC?
 

solderdude

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I think @kenji means digital volume control lowers the output voltage level (which of course it does) but not the noise floor.
Of course, when the noise floor of the DAC is below an audible level this is all moot and of no importance.
It is better than that of the vast majority of amplifiers anyway.
With the vast majority of DACs the noise floor of the DAC is inconsequential anyway even at very high SPL.
Noise floor of a DAC is dwarfed by that of any audio recording anyway.
besides the resolution of DACs can be higher than that of its own self noise. This has been shown to be the case by averaging steady state signals of 24 bit resolution while when looking at the actual scope shot that signal cannot even be seen recognized in the actual noise from the device and also cannot be heard.

From a signal fidelity P.O.V. more bits means higher accuracy.

It is not the wisest thing to use say -40dB digital attenuation and feed that to a 16 bit DAC (certainly not without dithering at 16 bit). Not a real issue with decent DS DAC chips of today.
 
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What would be the mechanism for a DAC to "produce weaker analog signals" in order to change volume? Are you talking about a buffered analogue potentiometer (In other words, and analog pre-amp) on the output of the DAC?
Maybe, a digitally controlled stepped attenuator at the output?
 
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After thinking for a while, I came up with two possibilities.
  • A digitally controlled stepped attenuator at the output of DAC, but this could be expensive?
  • 64-bit floating point numbers. 32-bit or 50-bit integers. This is cheaper than a stepped attenuator.
 
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@tonycollinet As I wrote above, a DAC can theoretically have a digital stepped attenuator at the output, or it can just use 32~64 bits for digital attenuation.

Since using 32bits or more for digital attenuation is cheaper, I expect this to be the standard for most DACs.

FiiO K5 Pro ESS has a digital stepped attenuator actuated by a volume knob on it. In theory, the volume knob can be replaced by its own internal volume setting exposed to operating system.
 

antcollinet

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Theoritically it can have a stepped attenuator - are you aware of any DACS that do? (The RME has a crude one)


Regarding 32/64 bit floating point (or even fixed point) processing - even if you do use them for the DSP, the values still have to be converted to 24bit fixed point before the actual conversion to analogue by the DAC chip. So at that point, you sill lose resolution (-48dB volume reduction will result in reduction from a 24 bit resolution to 16 bit resolution)

However, the main point is that even with that resolution reduction, it doesn't matter since the noise floor of any half way decent DAC will still be inaudible.
 
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even if you do use them for the DSP, the values still have to be converted to 24bit fixed point before the actual conversion to analogue by the DAC chipo. So at that point, you sill lose resolution
If setting DAC volume to -48dB means you lose roughly 8 bits, then you lose 8 bits in the 32/64bit DSP. 32-8 = 24bits. Those 24bit leftovers are going to be converted to 24bits, then.
 

antcollinet

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If setting DAC volume to -48dB means you lose roughly 8 bits, then you lose 8 bits in the 32/64bit DSP. 32-8 = 24bits. Those 24bit leftovers are going to be converted to 24bits, then.
No, the DSP processing will be floating point, so there is no loss of resolution.

When you convert to FIXED POINT for the DAC, you lose the resolution. If you output a signal at 24bit full scale, to the DAC chip, then the DAC chip will ouput the full volume (no reduction). To get the 48dB reduction, all the 8 most significant bits of the 24 bit value must be zero, so your signal is only in the least significant 16 bits.
 
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Why do you assume that DAC has to use 24-bit fixed point numbers for analog conversion? Is there any hidden non-man-made natural law that enforces this?

What if I want to use 32-bit integers for analog conversion?
 

MaxwellsEq

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It's worth searching the forum since this discussion topic comes up very frequently.

If you vary the signal once it's been converted from data to a voltage, you have reinvented the "preamp". This is a volume control, usually buffered, with a potentiometer, stepped resistance or VCA.
 

antcollinet

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Why do you assume that DAC has to use 24-bit fixed point numbers for analog conversion? Is there any hidden non-man-made natural law that enforces this?

What if I want to use 32-bit integers for analog conversion?
DAC Chips are hardware devices that take in fixed point values and converts them to analogue.

It's actually much more complicted than 16 bit, or 24bit, or 32 bit - since in delta sigma converters (vast majority) the PWM data is upsampled and bit reduced, then digitally filtered before being ouput as a bit stream into the external low pass filter.

It may be that the equivalent resolution of this process is better than 24 bits - but again, it doesn't matter. 24 bit would be fine for digtal volume control witn inaudible impact.

See:


However - in all these cases, the volume control is digital - manipulating the digital bits. It is not a case of keeping the 'bits the same and producing a weaker analogue signal.' Which is your statement that started this discussion.
 
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It may be that the equivalent resolution of this process is better than 24 bits - but again, it doesn't matter. 24 bit would be fine for digtal volume control witn inaudible impact
-48dB may not be enough for a power amplifier or for an integrated amplifier at its max volume. Paul McGowan said at least some DACs use 50-bit words internally.

I would want at least 32 bits for a power amplifier.

How many bits are used in Topping D10s?
 
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Topping D10s comes with ES9038Q2M DAC which seems to process 32-bit audio samples internally.

With 32-bits, I would have 16-bits at -96dB of DAC volume.
 

Hayabusa

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DAC means digital to analog converter. Reducing DAC volume can mean reducing voltage level instead of meddling with digital bits before bits are converted to analog signals.

Because DAC produces analog signals, it just needs to produce weaker analog signals instead of meddling with digital bits if DAC volume is reduced.

Reducing application volume reduces bit-depth of digital audio data. Thus, it is a good idea to use 24bit or 32bit audio sampling format in operating system.
if the dac was transparent it stays transparent when digitally lowering volume
 

Sokel

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Why all this debate when Amir presents it to us visually?

Here's an example,at -30db you get 94-95db SINAD.
More than enough:


index.php


Edit:I'm not sure about the -30db but that's his usual attenuation with this test.
 
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danadam

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-48dB may not be enough for a power amplifier
You mean you need the output from the DAC to be attenuated by more than 48 dB before it enters the amplifier? If yes, then it still doesn't matter, because the resolution of the output of any DAC is limited by physics to 22 or 23 bits, no matter how many bits it is using internally. You will cut into the 16-bit of the source signal whether the DAC is using 24 or 32 bits.
 
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