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Serious Question: How can DAC's have a SOUND SIGNATURE if they measure as transparent? Are that many confused?

I know well the formal protocol, but for example putting one 8030 and another G Three (measure the same in spinorama made by Genelec), adjusting gain and verifying levels with a mic, they sound way different in tonality, if it was subtle I would have some doubts but not at all… Not any need for blind test

Not talking about testing speakers, you were talking about DACs.
 
Not talking about testing speakers, you were talking about DACs.
In my example I used to illustrate my experience, as the attenuator was on the 8030C amp, but equally mentioned the attenuator (resistive in this case, I opened the Focusrite) of my audio interface, and have another third example I didn’t mentioned which is the Ifi Zen Signature DAC V2 (which in reality is a DAC with a volume knob, better made than the one of Focusrite judging by more stability on sound varying voltage).

But I don’t need more examples, all sorts of attenuation (passive resistive are the most noticeable) change tone. Even reamping ones, that are quite expensive cannot match perfectly the impedance curve of a speaker.

So as a conclusion, in my particular doubt between DAC transparency and my perceptions with my own DAC-preamps, I have solved the contradiction: I was comparing one DAC + attenuator with another DAC + another attenuator and which were also working at different levels of gain to be compared at same signal output.

The DAC were equal, the harmonics after attenuation were the difference. Finished enigma for me…

P.S: the Ifi Zen has a fixed line output switch that bypass the attenuation circuits, then offering another different tonality but unfortunately I couldn’t compensate by master volume of the amp, its voltage was to high (around 7 V pp)
 
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So as a conclusion, in my particular doubt between DAC transparency and my perceptions with my own DAC-preamps, I have solved the contradiction: I was comparing one DAC + attenuator with another DAC + another attenuator and which were also working at different levels of gain to be compared at same signal output.

The DAC were equal, the harmonics after attenuation were the difference. Finished enigma for me…
Convincing yourself is easy ... convincing other ASR members is the difficult part :D
 
they sound way different in tonality, if it was subtle I would have some doubts but not at all… Not any need for blind test
Why do you think perceptive biases can't create unsubtle differences in sound?

In fact, in a world of modern electronics where differences (if they really exist at all) are going to be incredibly subtle to the point of being impossible to hear - the fact you are hearing unsubtle differences just convinces more that they are simply the results of perceptive bias etc.
 
Convincing yourself is easy ... convincing other ASR members is the difficult part :D

You’re totally right :)

But is enough for me, there are a tone of articles explaining how an attenuator works in electronics (not only audio, you can find them on almost every electronic field).

But this is not concluding the main point, most of DACs send a fixed output (usually 2 Vrms to active speakers, 1 Vrms to headphones, 4 Vrms to XLR balanced inputs) so in this case I guess is impossible to find a difference between them.

My beginning in which I call “independent components audio” (before I bought homogenous equipments that were matched by a single brand) was with DACs with analogue master volume apart form digital ones, which sounded quite different to me. This was my confusion initially
 
Why do you think perceptive biases can't create unsubtle differences in sound?

In fact, in a world of modern electronics where differences (if they really exist at all) are going to be incredibly subtle to the point of being impossible to hear - the fact you are hearing unsubtle differences just convinces more that they are simply the results of perceptive bias etc.
Not in this case, the issue of attenuation is perfectly known and audible, apart from the fact that is not subtle if the component is simply a resistor… high frequencies are more attenuated than lower ones, they suck tone and feel flat. Reactive ones have same issues

Analogue attenuators are suggested by many brands as Genelec do to improve SINAD and dynamic range matching properly the DAC output with the amp input.

What they don’t mentioned is the process is not transparent, here in ASR you can read an article called “The Signal Path” or “Understanding the signal Path” where a member shows a mathematical model introducing a noise and harmonics in a three blocks system (DAC, preamp, amp) and tracking the SNR, THD and SINAD functions when varying gains on each block.

The fact that if this is audible or not relay ultimately on the proper SNR and THD of the attenuation stage, but nor Ifi nor Focusrite give information of this component.

This late is a guess, but I think is not cheap to make good attenuators, and this is the reason because low price DACs only measure SINAD at full voltage, and more priced DACs simply don’t incorporate any analogue knob on the device.

Post edit: sorry, the latest parsgraph is inconsistent: more on the side that well made DACs usually don’t have analogue attenuators. All DACs usually measure SINAD at the highest voltage, I wrote it too fast
 
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Not in this case, the issue of attenuation is perfectly known and audible, apart from the fact that is not subtle if the component is simply a resistor… high frequencies are more attenuated than lower ones, they suck tone and feel flat. Reactive ones have same issues

Analogue attenuators are suggested by many brands as Genelec do to improve SINAD and dynamic range matching properly the DAC output with the amp input.

What they don’t mentioned is the process is not transparent, here in ASR you can read an article called “The Signal Path” or “Understanding the signal Path” where a member shows a mathematical model introducing a noise and harmonics in a three blocks system (DAC, preamp, amp) and tracking the SNR, THD and SINAD functions when varying gains on each block.

The fact that if this is audible or not relay ultimately on the proper SNR and THD of the attenuation stage, but nor Ifi nor Focusrite give information of this component.

This late is a guess, but I think is not cheap to make good attenuators, and this is the reason because low price DACs only measure SINAD at full voltage, and more priced DACs simply don’t incorporate any analogue knob on the device.

Post edit: sorry, the latest parsgraph is inconsistent: more on the side that well made DACs usually don’t have analogue attenuators. All DACs usually measure SINAD at the highest voltage, I wrote it too fast
Sure - you can screw it up if you get the gain staging badly wrong. But with modern electronics, you really have to work hard to make it audible - and that is noise, not distortion. This belief you have that attenuation automatically results in audible defects - is what is causing you to hear audible defects.

Just IMO.
 
Not in this case, the issue of attenuation is perfectly known and audible, apart from the fact that is not subtle if the component is simply a resistor… high frequencies are more attenuated than lower ones, they suck tone and feel flat. Reactive ones have same issues
Show us the measurements (of your own devices used in the listening test you reported as "too obvious to need blind testing").
 
Show us the measurements (of your own devices used in the listening test you reported as "too obvious to need blind testing").
Is a good idea, but after reading further articles I realized that what change is harmonic distortion and signal to noise ratio, not frequency response.

The previous articles on the topic of attenuation pots I red are biased by the perception of a cleanest sound as “flat” and tone sucked when in fact was the harmonic distortion on the amp and the DAC operating at low digital volume that produced the “brilliant” signature.

So to me it still remains unexplained why WiiM Ultra (RCA outputs), Focusrite Scarlett 2i2 4th gen and Ifi Zen Signature V2 sound different to me.

Which is quite obvious but is not the explanation that I hoped to provide, is the fact that operating in correct output levels for input gains produces a better sound. But this is not the topic of the thread…
 
Is a good idea, but after reading further articles I realized that what change is harmonic distortion and signal to noise ratio, not frequency response.

The previous articles on the topic of attenuation pots I red are biased by the perception of a cleanest sound as “flat” and tone sucked when in fact was the harmonic distortion on the amp and the DAC operating at low digital volume that produced the “brilliant” signature.

So to me it still remains unexplained why WiiM Ultra (RCA outputs), Focusrite Scarlett 2i2 4th gen and Ifi Zen Signature V2 sound different to me.

Which is quite obvious but is not the explanation that I hoped to provide, is the fact that operating in correct output levels for input gains produces a better sound. But this is not the topic of the thread…
You cannot make the determinations you are posting with sighted listening.
 
You cannot make the determinations you are posting with sighted listening.
No, that’s true…

Which determinations are you talking concretely? I posted more than one.

The fact that matching levels with analogue attenuators improves the sound is one of them, this is not a subjective impression only, it was measured and modeled many times…

The statement that my 3 DACs may sound different because their respective attenuation pots or internals is that probably doesn’t holds and should verify with measurements.

Later on the day I will measure Focusrite Scarlett against WiiM Ultra once carefully set the first one to 1 Vrms output. Here is to soon for the required SPL level to do the measurements.

My system is far from optimal, it only consist on a mic coupled to the WiiM room correction software, but it plots quite consistently the measured values and the suggested corrections.

I chose Focusrite versus WiiM because they are the two ones I perceive more different on signature. WiiM has ben measured by Amir but I cannot find Focusrite measurements. The brand claims “audiophile quality DAC” for its 4th generation Scarlett model, since apparently precedent versions have received some bad critics by the users.
 
No, that’s true…

Which determinations are you talking concretely? I posted more than one.
Every single one.
The fact that matching levels with analogue attenuators improves the sound is one of them, this is not a subjective impression only, it was measured and modeled many times…
Measured and modeled how? Remember even if one is a bit better both are probably below audibility. Sighted tests of audibility are mostly a waste of time.
The statement that my 3 DACs may sound different because their respective attenuation pots or internals is that probably doesn’t holds and should verify with measurements.
Then do the measurements. Just listening and coming to conclusions is not convincing or reliable to find the truth.
Later on the day I will measure Focusrite Scarlett against WiiM Ultra once carefully set the first one to 1 Vrms output. Here is to soon for the required SPL level to do the measurements.

My system is far from optimal, it only consist on a mic coupled to the WiiM room correction software, but it plots quite consistently the measured values and the suggested corrections.

I chose Focusrite versus WiiM because they are the two ones I perceive more different on signature. WiiM has ben measured by Amir but I cannot find Focusrite measurements. The brand claims “audiophile quality DAC” for its 4th generation Scarlett model, since apparently precedent versions have received some bad critics by the users.
Why would you play the music and measure with a microphone when you have a Focusrite? You can play the various configurations you use and measure the results using REW or Multitone. Or record the music outputs and compare with Deltawave. Using a microphone will place significant constraints on how clean, reliable and repeatable the measurements are. Do these measurement with the ADC in the Focusrite.

Here is a review with measurements of the 4th gen 2i2.
 
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Which determinations are you talking concretely?
Let me help.

With sighted listening, you cannot state if what you are percieving even exists in the signal at all.

Even if it is, you certainly can't make any assessment of the causes of that perception being distortion, or noise, or frequency response. For that you need measurements. Anything else is just speculation.

And. your descriptions of how "attenuation" impacts on frequency response, noise and distortion are just plain wrong.
 
Measured and modeled how? Remember even if one is a bit better both are probably below audibility. Sighted tests of audibility are mostly a waste of time.
Here you have a quite simple yet effective modeling of different harmonic distortions in signal chain elements, is quite technical but I will try to quote a simpler one.

As summary, if you have a targeted SPL and different elements of gain (roughly DAC, preamp, amp and speakers) you can substantially improve nonlinear elements just by calculating how noise and distortions are propagated and produced in the signal chain.

 
Every single one.

Measured and modeled how? Remember even if one is a bit better both are probably below audibility. Sighted tests of audibility are mostly a waste of time.

Then do the measurements. Just listening and coming to conclusions is not convincing or reliable to find the truth.

Why would you play the music and measure with a microphone when you have a Focusrite? You can play the various configurations you use and measure the results using REW or Multitone. Or record the music outputs and compare with Deltawave. Using a microphone will place significant constraints on how clean, reliable and repeatable the measurements are. Do these measurement with the ADC in the Focusrite.

Here is a review with measurements of the 4th gen 2i2.
And here you have a more comprehensive explanation of SNR and THD along the signal chain.

 
Every single one.

Measured and modeled how? Remember even if one is a bit better both are probably below audibility. Sighted tests of audibility are mostly a waste of time.

Then do the measurements. Just listening and coming to conclusions is not convincing or reliable to find the truth.

Why would you play the music and measure with a microphone when you have a Focusrite? You can play the various configurations you use and measure the results using REW or Multitone. Or record the music outputs and compare with Deltawave. Using a microphone will place significant constraints on how clean, reliable and repeatable the measurements are. Do these measurement with the ADC in the Focusrite.

Here is a review with measurements of the 4th gen 2i2.
And the Genelec website recommendation for analogue attenuation on signal chain to keep dynamic range as high as possible, minimizing noise

 
As summary, if you have a targeted SPL and different elements of gain (roughly DAC, preamp, amp and speakers) you can substantially improve nonlinear elements just by calculating how noise and distortions are propagated and produced in the signal chain.
There is nothing in that paper that isn't already pretty well understood around here.

The point you are missing, is that you can completely skip that calculation in many cases, simply because the noise and distortion produced by the devices in question are so low as to be completely inaudible - even after passing through subsequent gain stages**. Especially when listening to music - and even more especially when listening via speakers.

This is certainly the case for the DACs under discussion in this particular subthread.

**Always assuming, of course, you are not doing something really stupid with them such as 60dB of digital attenuation compensated further down the chain by 60dB of extra gain. This would fall under the category of "working hard to screw it up"
 
There is nothing in that paper that isn't already pretty well understood around here.

The point you are missing, is that you can completely skip that calculation in many cases, simply because the noise and distortion produced by the devices in question are so low as to be completely inaudible - even after passing through subsequent gain stages**. Especially when listening to music - and even more especially when listening via speakers.

This is certainly the case for the DACs under discussion in this particular subthread.

**Always assuming, of course, you are not doing something really stupid with them such as 60dB of digital attenuation compensated further down the chain by 60dB of extra gain. This would fall under the category of "working hard to screw it up"
I agree that DACs are essentially similar and barely linear, but I feel the difference between having it at higher or lower volumes.

More concretely the Genelec 8030C has a by default sensitivity of 106 dB @ 1V rms at 1 m. To listen at a reasonable 70 dB SPL at 1 m I have to decrease by 50 dBFS the DAC which has balanced outputs 4 Vrms.

I don’t know if is a psychological phenomenon or not, but having the same speaker (called G Three but is the same, once verifying by Genelec technical support) at 86 dB @ 1 Vrms connected to the WiiM by RCA outputs 2 Vrms operating at -24 dBFS to reach same volume, the result is way better in terms of distinguish instruments in orchestra, natural sound of the piano (which I played since 40 years and know very well) and clarity of voices (specially in english which is not my native language so little bit more transparency help me to understand the words).

There are not 60 dB but 26 dB of difference on the DAC side, so I keep the speaker with low sensitivity having actually same cost (in the past G Three were 100€ more expensive but prices have changed).

Genelec 8030C can be lowered by 12 dB input gain, but still are too sensitive.

I know is not the topic of the thread, but regulating gain stages and digital volume it matters to better listening, doing it randomly is a choice but I prefer searching some better combinations.
 
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