• Welcome to ASR. There are many reviews of audio hardware and expert members to help answer your questions. Click here to have your audio equipment measured for free!

Master Thread: Are Measurements Everything or Nothing?

Regarding section1 "complex load”:
A buffered op‑amp input is high‑impedance, but it is not a purely resistive, zero‑interaction load. The DAC still drives: finite input impedance, cable capacitance and inductance, dielectric absorption, the op‑amp’s input bias network, and any RF filtering at the input. Those elements form a real transfer function that affects HF phase, settling behavior, and noise modulation. That’s why different line stages and cable/interface combinations can measure identically at 1 kHz yet differ in wideband behavior and time‑domain performance.

and regarding number 2 “no preamp is always more accurate”:
“DAC → amp is always truer” is only valid in an ideal system with zero output impedance, infinite input impedance, zero cable reactance, and a DAC output stage designed purely as a line driver. In practice, many DACs have modest output stages optimized for short, easy to drive loads, not for driving longer interconnects or more complex input networks. A dedicated preamp with lower output impedance, higher current capability, better common‑mode and RF rejection, and cleaner wideband behavior can reduce interface‑induced errors (HF phase shift, noise modulation, level‑dependent distortion). It doesn’t “add information”; it reduces the degradation that occurs when the DAC is forced to do both precision conversion and heavy line‑driving at the same time. That is why my Gustard R26 DAC replaced my SMSL SU-10 DAC. It sounded better.

I would purchase a Rockna Wavedream DAC, but I ain't rich like you guys.
I'm not a wimp, but you're right about the Rockna. I currently have the Rockna Reference STD. I previously had the Audio GD, then the RME Adi-2 DAC fs, then the Linn Selekt DSM/3 with Utopik and Oragnik, but even with Katalist (which is where I started), the Adi didn't have any real connection to the Linn. Then I had the Mola Mola Tambaqui, and now the Rockna Reference Standard. I also had the Signature (Reference) at home for comparison. The order goes, the progression goes. The Tambaqui and Linn have different sound characteristics. It's up to you. The Signature, however, sounds incredible. I also enjoyed the Linn Klimax DSM/3 at home. The friend who sold it currently has the Signature. It's interesting to compare the excellent Tambaqui with my Reference. The Tambaqui is very clean, precise, and incredibly resolving. This resolution is specified differently in the Tambaqui and Reference STD. The Reference's backgrounds and backgrounds are better gradated in amplitude, resulting in a completely different presentation. The Rockna is generally better in depth and space. It feels more real and authentic despite the Tambaqui's incredible detail. The Signature, on the other hand, is a step higher. Higher resolution and greater dynamics mean greater realism.
 
I'm not a wimp, but you're right about the Rockna. I currently have the Rockna Reference STD. I previously had the Audio GD, then the RME Adi-2 DAC fs, then the Linn Selekt DSM/3 with Utopik and Oragnik, but even with Katalist (which is where I started), the Adi didn't have any real connection to the Linn. Then I had the Mola Mola Tambaqui, and now the Rockna Reference Standard. I also had the Signature (Reference) at home for comparison. The order goes, the progression goes. The Tambaqui and Linn have different sound characteristics. It's up to you. The Signature, however, sounds incredible. I also enjoyed the Linn Klimax DSM/3 at home. The friend who sold it currently has the Signature. It's interesting to compare the excellent Tambaqui with my Reference. The Tambaqui is very clean, precise, and incredibly resolving. This resolution is specified differently in the Tambaqui and Reference STD. The Reference's backgrounds and backgrounds are better gradated in amplitude, resulting in a completely different presentation. The Rockna is generally better in depth and space. It feels more real and authentic despite the Tambaqui's incredible detail. The Signature, on the other hand, is a step higher. Higher resolution and greater dynamics mean greater realism.
Welcome to ASR. Uncontrolled subjective descriptions of likely inaudible phenomena are generally not well received here, as this is one of very few places on the audio-internet that endorses a research-backed approach. Below is a synthesized version of advice for newcomers:

People at ASR tend to view good sound as (for electronics) fidelity to signal, and speaker output conforming to Toole and Olive's research. If you like the sound distorted, less accurate, or prefer some other speaker presentation, that's fine, but own it, don't pretend a) lesser fidelity is greater accuracy or b) there must be something wrong with these standards because of your personal preferences or c)you can hear something that can't be measured. There's no need to rationalize your tastes.

Most of us also believe that the way to test for *strictly audible* differences is by
properly executed and level-matched double blind procedures, or through taking measurements and recording a result above audible thresholds. The fact that you noticed a difference outside of these conditions simply isn't evidence of a difference in signal quality at your ears. Even if it is a difference in the signal, as opposed to some sighted bias, it is likely to be a difference in amplitude rather than something more subtle.

Finally, all of the above mistakes are simply human. No human being is so "experienced" or "trained', or "sensitive" as to be able to make sighted comparisons objectively.


The thread from which this is taken is longer but I do suggest you check out the links.

 
Years ago I used to try the occasional passive preamplifier or DAC with the volume control.
My subjective perception inevitably went like this: it sounded a bit more clean and transparent (I was typically bypassing a tube preamplifier, so there may be something to this).

But also, it sounded a bit more wimpy. A softer less punchy sound, almost like the system was being slightly under driven.

After a while, I always went back to a regular preamplifier, which seemed overall more satisfying.

And I remember encountering numerous threads in forums afterwards where plenty of audiophiles reported similar experiences.

Could be a bias effect, of course, but beyond that ….Is there anything to this? Something that would technically explain this?

My current Benchmark LA4 preamp seems to technically an audibly act, something like the classic straight wire with gain. It doesn’t sound wimpy like the passives I tried before.
(Though my tube preamp - a CJ preamp paired with my CJ amplifiers - can sound a teeny but more dense and punchy)
 
Years ago I used to try the occasional passive preamplifier or DAC with the volume control.

Those two categories are not remotely similar. A passive pre amp will have a potentiometer directly connected to the output. This will result in:
1 - Very high output impedance - probably 10x or more what you might expect from an active device.
2 - Output impedance varying with volume.

These two things might (though by no means certain) interact with longer or high capacitance interconnect to create an audible difference.


A DAC with volume control has neither of these probelms and will generate an audibly perfect output at any volume assuming you cannot hear DAC noise (hiss) in the silences between tracks (this might be possible if - for example - you were to turn the DAC volume very low - say below -50dB - then turn up the gain after the DAC (somehow) to get it back to a listening level)


But by far most likely explanation (IMO) for your perceptions is eliminating distortion from your tube preamp (which might create a "thicker" sound that you are used to) or good old sighted bias.
 
Those two categories are not remotely similar. A passive pre amp will have a potentiometer directly connected to the output. This will result in:
1 - Very high output impedance - probably 10x or more what you might expect from an active device.
2 - Output impedance varying with volume.

These two things might (though by no means certain) interact with longer or high capacitance interconnect to create an audible difference.


A DAC with volume control has neither of these probelms and will generate an audibly perfect output at any volume assuming you cannot hear DAC noise (hiss) in the silences between tracks (this might be possible if - for example - you were to turn the DAC volume very low - say below -50dB - then turn up the gain after the DAC (somehow) to get it back to a listening level)


But by far most likely explanation (IMO) for your perceptions is eliminating distortion from your tube preamp (which might create a "thicker" sound that you are used to) or good old sighted bias.

Thanks very much for your explanation. Especially the difference between the passive preamplifier and a DAC with a volume control.

I’ve blind-tested cables, DACs/CDPs, music servers, preamps and other things, but I never did get around to blind testing that gear. So I never could be sure.

If I still had a passive pre I’d probably get around to it. But I’m done buying audio gear. Just coasting on fumes at this point. :-)
 
...I find myself still flailing about a bit but half deaf anyway so what is point...?!?...

...remember when Amir recommended speakers that measure pooly because he liked how they sounded and were inexpensive...???...

...methinks he be well aware that, although vital, measurements are not everything...
 
...I find myself still flailing about a bit but half deaf anyway so what is point...?!?...

...remember when Amir recommended speakers that measure pooly because he liked how they sounded and were inexpensive...???...

...methinks he be well aware that, although vital, measurements are not everything...
Funny and provocative.

But do you have the link?
 
 

Conclusions
I went into the listening tests prebiased with good looks and poor measurements of the speaker. What I found was that with a bit of EQ, this becomes a very nice speaker to enjoy on the desktop. It leaves far behind any plastic toy computer speaker. It certainly did justify to any track I threw at it, leaving me wanting to sit there and keep listening to it!

I don't often make dispensation for price but here, it is remarkable how good of a sound one can get from well packages speaker and with some correction per above. Even without EQ, if one turns down the treble control it will likely be quite listenable.

Not sure this tells you much. Basically “with EQ, it’s pretty really good for $99”.
 
Last edited:
...I find myself still flailing about a bit but half deaf anyway so what is point...?!?...

...remember when Amir recommended speakers that measure pooly because he liked how they sounded and were inexpensive...???...

...methinks he be well aware that, although vital, measurements are not everything...
No I don't. I remember when he figured out that a bit of EQ turns a pair of very inexpensive speakers into a decent value proposition in its (very low) price category
 
  • Like
Reactions: MAB
Oh wow. You overturned everything with just two posts. ;)
 
...I find myself still flailing about a bit but half deaf anyway so what is point...?!?...

...remember when Amir recommended speakers that measure pooly because he liked how they sounded and were inexpensive...???...

...methinks he be well aware that, although vital, measurements are not everything...

I think in this case the measurements only tell part of the story, because Amir does not did not show us all the measurements that were needed:

1. What is the likely limitation of small speakers like that Edifier? Answer: it's going to be volume limited because of its size. To his credit, he did test it at 96dB, but he showed distortion only. And it was horrible. But what was really needed was formal compression testing. Does the speaker deliver 96dB when commanded to deliver 20dB above 76dB? My guess is likely not.

2. That horizontal directivity looks pretty terrible, but look closely. Every time it widens, it's because the on-axis SPL is louder. What does the directivity look like when the on-axis is normalized to 0dB? I am sure it would look a lot better.

3. The typical use case for this speaker will be on a computer desktop, so it will have some desk bounce. The amount of desk bounce is fixed by the height of the drivers, assuming the user doesn't deploy stands. This will usually boost the low frequencies a bit more, which is already quite respectable for such a small speaker (it starts to roll off at 70Hz).

I suspect that Amir listened to the speaker under "typical use case", i.e. sitting not far from it, with the speakers on his desktop. They should sound fairly decent provided they are not pushed to deliver too much SPL.

So no, this is not a case of measurements meaning nothing. It's a case of insufficient measurements being shown.
 
Those two categories are not remotely similar. A passive pre amp will have a potentiometer directly connected to the output. This will result in:
1 - Very high output impedance - probably 10x or more what you might expect from an active device.
2 - Output impedance varying with volume.

These two things might (though by no means certain) interact with longer or high capacitance interconnect to create an audible difference.
The basic concept is sound. It can be a potentiometer or switching of fixed resistors to create a resistive voltage divider.

Say you have an input resistance of 10Kohm. At 6dB attenuation the output resistance is 2 X 5Kohm resistors yielding around 2.5Kohm to AC ground (source is low impedance).

2.5Kohm to ground is low enough to handle some cable capacitance. But say at 30pF/ft for 100 feet cable the bandwidth is 21KHz (with 45-degree phase shift…). At 20 feet, such cable would yield 106KHz. At all other attenuation settings (not -6dB) the bandwidth extends to higher frequencies. So far so good, but:

2.5Kohm to ground driving an unbalance cable is very susceptible to pick up interference (electromagnetic waves). The cable acts as an antenna. The shield itself acts like an antenna. So cable length may be the main issue.

But the concept of using resistors for analog attenuation is solid. Digital attenuation means ending up with less bits (quantization). Take a dynamic range of 96dBFS (16-bit, CD quality) and attenuate digitally by 36dB. The resulting DR is 60dB or 10 bits audio. Analog attenuation offers quantization free attenuation.

My gold DA converters are aimed at professional studios, so the main output is fixed (24/18dBu, Balanced/unbalanced). But often there is a need for lower voltage (power amplifier and more). I did use the resistor attenuator concept in my DA converters, the attenuation is analog, the control and display are digital. It was very handy to use the same resister divider at different attenuation settings for each of the three inputs (offering quick comparison of channels with different gains). Needless to say the DA Monitor output buffers the divider from the output (XLR connectors, Balanced/Unbalanced).

I don’t have anything against “passive preamplifiers”. It is fine if it works. It does require attention to the input impedance of the power amp (or other destination). The higher the impedance, the better…
 
A resistor only unit looks great – no power supply, extremely low distortions, robust, reliable... But using a passive preamp does not provide freedom form power supply and limitations due to semiconductors (or tubes). It puts the burden on other units, mostly on the destination such as power amp or other analog input.

At 25Kohm resistance, the cable length between the passive pre-amp output becomes a very limiting factor. The input cable side can be much longer, it acts like low AC impedance to GND. So why the high values? (Most DA converters and analog gear can drive much lower resistors than 100K or 10Kohm).

So why the undesirable high values? It is about resistor values especially at high attenuation:

10K and 1ohm divider yields -80dB attenuation. The 1ohm can be 1% accurate. The 10K accuracy needs to be .01%. For -79dB the required resistor value is 1.122 Ohm. For -78dB the value is 1.259 ohm. It is impractical using a pot or fixed switched resistors.

All the limitations can be overcome when incorporating powered analog circuits. My gold DA offers a secondary output for monitoring. It has 0-80dB range with accurate .25dB steps. Unlike a potentiometer, or a passive switched resistor design, each of the three DA inputs has a precise attenuation setting (with display and nonvolatile memory). I did not have to use 320 resistors for 320 setting, or an potentiometer's attenuation limited range.

Using a passive attenuator is like using cents for currency. Using active circuits enables breaking the task into sections, analogues to cents, dimes, quarters, dollars…

Again, I don’t have anything against passive preamp when it works. (In my opinion it would be more accurate to call it “passive attenuator”).
 
Oh wow. You overturned everything with just two posts. ;)
...certainly not trying to overturn anything, just trying to point out the reality of the world n' shit...

...soma youse guys get almost as wound-up as those audiophool dudes...I can understand that with the $$$ some of you have in measuring equipment...

...IMHO it is much like genetics being an important factor in breeding bulldogs, or corn, or marijuana...

...are they important...???...of course...without good genetics you have poor specimens...

...but they are not everything...
 
Last edited:
...certainly not trying to overturn anything, just trying to point out the reality of the world n' shit...

...soma youse guys get almost as wound-up as those audiophool dudes...I can understand that with the $$$ some of you have in measuring equipment...

...IMHO it is much like genetics being an important factor in breeding bulldogs, or corn, or marijuana...

...are they important...???...of course...without good genetics you have poor specimens...

...but they are not everything...
Yes, measurements, like bulldogs’ corn or marijuana and are not the only thing. I ran a ton of measurements, but as a human can miss something. I make sure to carefully verify a design by ear. I mean ears of top mastering engineers, not my 80 years old ears.

I come from a musical family, I play (piano and accordion) every day. I love listening to good music. But I am not thinking about Beethoven 9th or country music when I need to focus on making sure that each sample value is really accurate and timed correctly…

But I know that if I do the job right, the conversion between analog and digital and back will be done without sonic alteration. It is done to provide a digital file format (for easy to handle and process audio). A perfectly executed AD followed directly by DA will produce the exact output waveform as the input (with latency and possible different gain).

For me, the “golden ears” are part of measurements. They are the human measurement tool. I have yet to ask AI to do listening test for me…
 
...I am going to like this even though I am struggling to fully comprehend what I am reading here...
 
Hello Hapo

What is it that you are not clear about?

I have to rely on a lot of measurements, and with many years of design I developed a lot of confidence that I am doing it right. But only an arrogant fool will forgo feedback from people that use the gear. (I will not forget, it was an ear person in the 70’s that made me aware of phase linearity issues). Sending gear to be tested in the field (in different environments) can solve a lot of headaches.

But my main point is that following the theory (as precisely as possible) yields good accurate conversion. The “digital sound” is the same as the analog, but with a different convenient format. You convert it back and the result is the SAME. Designing an AD is about precision an accuracy, which is an EE territory. I designed the Savitr when I was 72-75 years old with less than 10KHz hearing bandwidth. For me it was about electronic design. Needless to say, the ears had to confirm the measurements and the operation of the unit.

My early AD designs were not for audio. Testing the early MRI converters (100KHz max) began with feeding the AD an ultra linear analog ramp (constant current into a capacitor). The output was a “staircase”. A 14-bit AD has 16384 steps, but we focused the scope on one small section, 16 steps at a time (using the trigger threshold and feeding the LSB's to through a 4 bit DA ).

The goal was clear and visible. The step size (vertical) and time (horizontal) should be the same. It is called differential linearity. If the step size (vertical) changes over the whole range it is called integral non linearity (for audio it is the cause of most distortions).

It is not all that different for audio. We have better test tools that are more suitable for audio testing. It is a different language, but the cause of “audio distortions” and “integral non linearity” for MRI, instrumentation or a weighing scale are circuit design issues, aiming at accurate conversion involving voltage, current, electronic parts. Measurements are the tool; the ears are the “security guard rails”.
 
Back
Top Bottom