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what is the purpose of the analog part in the DAC

devlux

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hi
why we can find DACs with simple analog circuit and some with very complex
what is exactly the purpose of the Analog part , as I understand it , the DAC chip output is analog
does it act like a pre amp? even without volume control

thanks
David
 

solderdude

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DAC chips can have a voltage output (rare) and there are DAC chips that are current out.
There are DAC chips that are SE out (so only signal to ground) and balanced out.
Those actual DCA device outputs may have to be SE or balanced so may need conversion.
There are DACs that need different amounts of post filtering, this is an analog filter designed to remove high frequency switching noises.
Some manufacturers want 1V out, others around 2V out, yet others a bit more or a whole lot more and this is very easily done.
Some want tubes in there, others believe in discrete transistors, or FET's, others want no active parts (which is possible) or just some passive filtering.
Some manufacturers want best performance, others have a philosophy or like a certain type of circuit for some reason and use that.
Some manufacturers simply follow DAC chip manufacturer application notes, others (like to) think they know better.

Basically it is the designer' s choice (DAC chip and analog part) and its desired specifications and what the DAC chip(s) put out that determines what components are used after the DAC chip.
There are a thousand possible ways to achieve what designers aim for.
 
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rschoss

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I had a debate with a friend regarding this "analog part" issue.
I said that if it measures as a "black box" (*) better than lets said S/D+N 100 dB & Frequency response flat +- 0.2 dB 20 to 20 Khz there is no way that one "analog part" will have a different sound signature then the other, and he said that different analog parts can have a different sound signatures i.e. stage, image etc. - Your advice please.

(*) In addition he said that different type of capacitors in the analog chain can sound different.

Thanks in advance,
Robert
 
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redshift

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I had a debate with a friend regarding this "analog part" issue.
I said that if it measures better than lets said S/D+N 100 dB, there is no way that one "analog part" will have a different sound signature then the other, and he said that different analog parts can have a different sound signatures i.e. stage, image etc. - Your advice please.

Thanks in advance,
Robert

I’d say that analogue and digital filters most certainly affects the sound. Personally I’d sacrifice some SnR/dynamic range for time accuracy, i.e. optimally dampened step and impulse responses.

I’ve been running my RME ADI 2 FS DAC in unfettered mode (unfiltered) and just blasted the digital noises down the audio pipeline. I scoff at aliasing effects.

A bit of dithering never hurts. :cool:
 

rschoss

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Thanks for the answer redshift, I want to emphasis that we are talking only about the final analog stage in a D/A converter.
 
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Blumlein 88

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I’d say that analogue and digital filters most certainly affects the sound. Personally I’d sacrifice some SnR/dynamic range for time accuracy, i.e. optimally dampened step and impulse responses.

I’ve been running my RME ADI 2 FS DAC in unfettered mode (unfiltered) and just blasted the digital noises down the audio pipeline. I scoff at aliasing effects.

A bit of dithering never hurts. :cool:
You scoff at aliasing, a real effect, in order to get a mythical imagined damped step and impulse response improvement of time accuracy.
uh huh!
 

redshift

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Thanks for the answer redshift, I want to emphasis that we are talking only about the final analog stage in a D/A converter.

You’re welcome. My working hypothesis is that the human auditory system is sort of an I/Q modulator.

https://en.m.wikipedia.org/wiki/In-phase_and_quadrature_components

Sound pressure waves -> ears -> complex data in the auditory system -> subjectively experienced tunes.

A bit of noise doesn’t hurt that much as long as the frequency spectra, phase and amplitude is well behaved for the modulation to work as intended.

A bit of noise, even above the hearing threshold doesn’t sound that bad after all.

For example, I’m not that susceptible to tape hiss, it can be quite nice with a bit of that as it gives away that the music soon is hitting the loudspeakers.

On the other hand: Vinyl pops, scrapes, wow and flutter though. Yuck and suck from the Stone Age of audio reproduction… :facepalm:
 
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redshift

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You scoff at aliasing, a real effect, in order to get a mythical imagined damped step and impulse response improvement of time accuracy.
uh huh!

Ever heard of critically dampened filters? How about Bessel filters that doesn’t screw with the impulse response in a horrific way?

Btw. You can never completely get rid of aliasing effects. Such filters doesn’t exist.
 

restorer-john

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A bit of noise doesn’t hurt that much as long as the frequency spectra, phase and amplitude is well behaved for the modulation to work as intended.

You do know that when CD first came along, even with analogue brickwall filters, many high powered amplifiers were prone to oscillation due to remnants of the sampling frequency being passed to them. Plenty of amplifiers had specific CD inputs to roll-off their response around 40kHz.

Passing HF rubbish down the line is not a good idea.
 

redshift

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You do know that when CD first came along, even with analogue brickwall filters, many high powered amplifiers were prone to oscillation due to remnants of the sampling frequency being passed to them. Plenty of amplifiers had specific CD inputs to roll-off their response around 40kHz.

Passing HF rubbish down the line is not a good idea.

I generally agree that poor amp design isn’t very accommodating to digital “EMI”. But a 400kHz class D switcheroo? Totally oblivious to the DAC racket.
 

restorer-john

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I generally agree that poor amp design isn’t very accommodating to digital “EMI”.

Not poor design at all. Far from it. They are and were designed to be wideband amplifiers. Long before CD came along, we had amplifiers with responses of DC to 600kHz. I have a power amp rated out to unity gain at 3MHz.

Most of the power amps I like the most and use the most have responses from 1Hz->200kHz (-3dB). Even the little AHB-2 Benchmark is a wideband amplifier.

So, you want your sources to reproduce only content you want, not out of band garbage, unrelated to the actual musical content itself.
 

redshift

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Not poor design at all. Far from it. They are and were designed to be wideband amplifiers. Long before CD came along, we had amplifiers with responses of DC to 600kHz. I have a power amp rated out to unity gain at 3MHz.

Most of the power amps I like the most and use the most have responses from 1Hz->200kHz (-3dB). Even the little AHB-2 Benchmark is a wideband amplifier.

So, you want your sources to reproduce only content you want, not out of band garbage, unrelated to the actual musical content itself.

A good loudspeaker XO’s would just present itself as a high impedance load to HF content.

A reasonably designed amp front end likely have some filtering going on to protect itself from the various nasty frequencies of objective reality and crazy people running their DAC’s in unfettered mode.

Besides, a second order Bessel or optimally dampened topology would whack that HF crap into oblivion without ringing like mad.
 
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AnalogSteph

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@solderdude has basically provided the answers - I/V conversion, filtering of out-of-band noise and aliasing, diff/SE conversion, combining multiple channels, and achieving the desired output level.

OP, if you want to know more about the challenges that the output stages tends to face, look at some datasheets for really old parts (found e.g. on Datasheet Archive). CS4303 is a good one, as it shows how much ultrasonic noise a high-order delta sigma modulator will spit out (and these are far from gone, as any CS4272 powered interface will attest - see e.g. Behringer UMC202/204HD, Focusrite Scarlett 2i2). It's about the worst case, as I don't think they ever went beyond 5th-order modulators, instead more modern designs tend to be up to 5 bits with various tricks to preserve linearity at 1-bit level (see e.g. the CS43122 whitepaper for a typical early-2000s kind of design). The function of oversampling filters is briefly touched on as well, though if you want to know more about those, I might recommend DF1700 or SM5803. CS4328 is another good one to look at, the first DAC with all switched-capacitor lowpass filtering (discussed in The Audio Critic, issue 21 - the TDA1547 had had first-order switched-capacitor filtering before but nothing to that extent). For the challenges of traditional resistor ladder DACs, see e.g. PCM56.

I’ve been running my RME ADI 2 FS DAC in unfettered mode (unfiltered) and just blasted the digital noises down the audio pipeline. I scoff at aliasing effects.

A bit of dithering never hurts. :cool:
Give a man a hammer, and he'll find new and innovative ways of hitting his thumb...

The "NOS" / "ultra slow" filter option is only intended for use at 384 or 768 kHz, combined with external upsampling to essentially replace the built-in digital filter. For example, you can use Foobar2000 with the SoX resampler plugin for some of the best upsampling money can buy (but actually doesn't have to since they're free). In fact, I may be inclined to stack two instances of the resampler DSP, the first to e.g. 48 kHz to get rid of any ultrasonics (those needing to cater to their feline or canine audience may have to go to 96k ;)), and the second to 384k or 768k. Whatever aliases remain beyond 360 / 744 kHz would easily be taken care of by analog filtering, which is the entire point of using oversampling in the first place.

Looking at filters in time domain is rarely ever useful, unless you're Julian Dunn and trying to pin down pre-echo (which, however, generally is so small you wouldn't even see it at a normal size). Thankfully Monsieur Fourier is telling us that for any impulse response, there is an exactly equivalent set of frequency and phase responses. What is most prominently seen in impulse response relates to what's happening around filter cutoff, and that in turn generally is not the very most interesting area when we're talking 44.1 kHz and above.
My working hypothesis is that the human auditory system is sort of an I/Q modulator.
I could still see some mid/side going on, but I/Q? Where is hearing supposed to be hiding a product detector? As far as I'm aware the inner ear rather is sort of a frequency binning affair above the bass range (a bit like an FFT, but with variable bin size so frequency perception becomes roughly logarithmic). When it comes to hearing models, I'd recommend starting with Messrs. Zwicker, Fastl and Blauert for the time being.
 
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redshift

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@solderdude has basically provided the answers - I/V conversion, filtering of out-of-band noise and aliasing, diff/SE conversion, combining multiple channels, and achieving the desired output level.

OP, if you want to know more about the challenges that the output stages tends to face, look at some datasheets for really old parts (found e.g. on Datasheet Archive). CS4303 is a good one, as it shows how much ultrasonic noise a high-order delta sigma modulator will spit out (and these are far from gone, as any CS4272 powered interface will attest - see e.g. Behringer UMC202/204HD, Focusrite Scarlett 2i2). It's about the worst case, as I don't think they ever went beyond 5th-order modulators, instead more modern designs tend to be up to 5 bits with various tricks to preserve linearity at 1-bit level (see e.g. the CS43122 whitepaper for a typical early-2000s kind of design). The function of oversampling filters is briefly touched on as well, though if you want to know more about those, I might recommend DF1700 or SM5803. CS4328 is another good one to look at, the first DAC with all switched-capacitor lowpass filtering (discussed in The Audio Critic, issue 21 - the TDA1547 had had first-order switched-capacitor filtering before but nothing to that extent). For the challenges of traditional resistor ladder DACs, see e.g. PCM56.


Give a man a hammer, and he'll find new and innovative ways of hitting his thumb...

The "NOS" / "ultra slow" filter option is only intended for use at 384 or 768 kHz, combined with external upsampling to essentially replace the built-in digital filter. For example, you can use Foobar2000 with the SoX resampler plugin for some of the best upsampling money can buy (but actually doesn't have to since they're free). In fact, I may be inclined to stack two instances of the resampler DSP, the first to e.g. 48 kHz to get rid of any ultrasonics (those needing to cater to their feline or canine audience may have to go to 96k ;)), and the second to 384k or 768k. Whatever aliases remain beyond 360 / 744 kHz would easily be taken care of by analog filtering, which is the entire point of using oversampling in the first place.

Looking at filters in time domain is rarely ever useful, unless you're Julian Dunn and trying to pin down pre-echo (which, however, generally is so small you wouldn't even see it at a normal size). Thankfully Monsieur Fourier is telling us that for any impulse response, there is an exactly equivalent set of frequency and phase responses. What is most prominently seen in impulse response relates to what's happening around filter cutoff, and that in turn generally is not the very most interesting area when we're talking 44.1 kHz and above.

I could still see some mid/side going on, but I/Q? Where is hearing supposed to be hiding an SSB detector? As far as I'm aware the inner ear rather is sort of a frequency binning affair above the bass range (a bit like an FFT, but with variable bin size so frequency perception becomes roughly logarithmic). When it comes to hearing models, I'd recommend starting with Messrs. Zwicker, Fastl and Blauert for the time being.

Give the man a sharp output filter and watch the ringing when he bashes it with impulses and step responses.

Yah, let the analogue electronics downstream the DAC take care of the noises.

It is about time to drop the fallacy that the human auditotory system is a simple frequency analyzer.
 

egellings

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To 'dampen' a filter means to add moisture to it. To 'damp' a filter means to control or alter its output in some way.
 
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