Timbre therefore includes those subtle differences in sound quality pursued by audiophiles.

The academic paper linked below, published in 2019, describes an experiment in which subjects were asked to distinguish timbral differences between sounds while brain response was measured via fMRI. The fMRI eliminates some difficulties in experiments that require a conscious verbal response, such as whether there is a response that is subconscious, or whether the subject is insufficiently certain to give a definite verbal response.

The results showed that a (pre-existing) model of timbre constructed from subjective descriptions of timbral differences predicted brain response as measured by fMRI better than 3 models of timbral differences constructed from objective measurements of differences in the sounds themselves, or in lower level auditory processing in the cochlea. That is, at least as of the publication date in 2019, models based on objective measurements didn’t account for the higher level auditory processing in the brain that went into the subjective model.

The best performing among the objective models takes into account not only frequency-based but time-based objective characteristics, which makes a nice counterbalance to some objective measurements that tend to concentrate primarily on the frequency-based characteristics of sound reproduction.

https://www.ncbi.nlm.nih.gov/pmc/articles/PMC5747995/

@jabbr's thread on optical networking is what made me think of creating this topic. That thread is much more subjective and I know others have objective comments they'd like to put forth in this discussion. 

Here is Jabbr's thread - 

Continuing from the discussions on measurement standardization and headphones in particular; @botrytis...

Quote

 

As a person, who has been involved in Analytical Chemistry for years, standardization of methods used for measurement is the key. Even if the initial setup if off, at least all the measurements are done the same way. 

 

I can give an example, from the industry I am consulting in, currently. In Canada and the US, where recreational Cannabis is legal, many producers are shopping for the lab that gives them the highest number for THC, not the most accurate number. This is due to the fact, the regulatory agencies have not really put out any recommendations as to the accepted methods and no blind testing from standard samples. Until standardization of testing methods is done, which measurements does one believe?

 

I would argue potentially not any one of them in the case of headphone testing and correlating to subjective preferences. More important I think is seeing if one can find similarities between multiple methods or relative differences using the same method. Even if we all used a standardized system like Head-Fi's B&K 5128, would that ensure each of us can look at the graph and immediately agree that those lines prove we'll enjoy the headphone from which the measurement was obtained? I don't think it'll ever be that simple... Unless we already have a reference headphone, know the B&K 5128 measurement for that (recognizing that even the same brand and model headphone could have significant variations - look at that Westone W60 samples in the link), and then compare with the one from the new headphone, then we *might* be able to consider that maybe the enhanced mid-range could be more enjoyable, or the extra +3dB bass would be more "fun", etc...

Some things are more straight forward and demand accuracy because there is a linear correlation with effect. For example, knowing exactly how many "honest" milligrams of THC (not just a marketing number like "1000W PMPO!") there is in a gummy might mean that if it's the same price, maybe one could split the gummy and be able to use a box longer, it becomes a meaningful value comparison. But this is more difficult with non-linear correlations (like to subjective preferences). We of course still care that the measurement method is accurate and reliable, but beyond a threshold, there's no longer a subjective importance.

I might be unhappy if my expensive DAC scores a measly 1kHz THD+N of -65dB at best (which likely suggests bad noise floor, but even then unless we look at the detailed FFT, cannot be sure) using a reasonably precise measurement system say +/- 1dB. But once we can measure -100dB, does it really matter to my ears if it's -110dB? I don't think so because the threshold for "clean", "high enough fidelity" sound has likely been achieved (again, I would still like to see the FFT, not just a "one number objectivism"!). For things like SINAD, this is easily done with great accuracy for little cost these days. I doubt headphone testing will ever be as simple, and reductionistic as this; nor should it be!

It occurred to me that he mentions preference, but the experts in music, instrument design, and audio equipment design don't just have a preference among multiple sounds, but evaluate it... in other words describe its properties like (to use a few widely known) brightness, rolled off, bass tightness, etc. It might not be just a choice of two devices, and there might not be a clear preference. For instance someone might say that one device is good with a particular speaker while another device is best for another speaker.

So I'm wondering if any research tries to correlate, say, the appearance of an audio unit with its described sound qualities. 

He mentions that aural memory is only a few hundred milliseconds long. That's surprising to those of us who believe we can hear differences between components when auditioned minutes, hours, or days apart.

I have an informal argument that aural memory can last hours or days in musicians or musical instrument designers, and probably in other experts such as audio engineers, experienced audiophiles, etc.

First, we observe that musicians or musical instruments designers are honing their sound. They often have a unique, recognizable sound, and it's often of very high quality.

Next we observe that it takes a long time to develop that sound. Years of daily practice and experimentation. What are they doing during that time? I suggest they are "navigating"... that is, through past experimentation, they have some concepts of sound they like, and they are trying to move closer and closer to an image of what they would like to sound like in the future.

I observe that to "navigate" somewhere requires that you be able to determine where you are at this moment relative to previous locations. So, imagine that every morning, Yo-yo Ma gets up and starts practicing. Because aural memory only lasts a couple hundred milliseconds, he has no idea where he left off the night before, right? He has no image of the sound he'd like to produce today. And if he does something that sounds better today than it did yesterday, he has no idea, because of course he can't remember what he sounded like yesterday.

As this is an obvious absurdity, we conclude that he does have an enduring memory of his sound and that he can make fine distinctions in sound even in practice sessions that are days apart. By the same logic, so do expert instrument makers, audio engineers, etc.

http://boson.physics.sc.edu/~kunchur/Response_to_ASR_comments.pdf

I had been catching up on this interesting thread but then this comment caused me to take a detour.  I’ve lately been taking a greater interest in psychology as I’ve been curious about what makes people tick.

I have heard it repeatedly often on audio forums that we all suffer from expectation bias - and it is this bias that leads many to imaging things that aren’t there.  I’ve not spent long researching this but I’ve yet to find anything that confirms that.  
 

Expectation bias comes up most often as “researcher bias” and it pertains more to conducting research in a way that achieves an outcome that meets the researcher’s expectations. That’s different than the charge leveled at those who report having heard certain qualities from a component.  It could be argued that it’s researcher bias that leads some to discount observations from listeners that don’t conform to their expectations.
 

I will keep digging into this.  Sorry about going off topic here.  Thanks @GoldenOnefor such an interesting thread.

After reading the thread by the Computer audiophile here and his comment about the importance of a good quality recording mic VS a usb mic, I decided to invest in a sonarworks soundid condenser mic and an Artpro mic preamp.

Both the Sonarworks mic and the mic preamp supports up to 24/192 and I wonder if there is indeed any advantage to record my room measurement using higher sampling rate than 24/48 for use with audiolense convolution filters?

Nowadays, qobuz has more and more albums that are 24/96 so I wonder if it makes sense or perhaps there are technical reasons that measuring the room at 24/96 or higher would be better than sticking with 24/48?

Thanks

Deric

https://phys.org/news/2023-07-acoustics-decompose-accurately-basic-components.html

PC -> intona USB isolator -> 4 port passive USB hub, each USB port connected to 1 dac to my Schiit freya + preamp input -> all converted to balanced output to my power amp to speakers/subs

But before i can do that I need to volume match the 4 DACs, I am thinking of using REW to measure using the built-in SPL meter with a mic at listening position and for each dac to play a tone and record the volume level differences say DAC A shows 75db and DAC B shows 78.5db then in my actual comparison I will simply add back +3.5db to DAC A.

I also have one of those iphone app that uses the iphone as a SPL meter, but not sure how accurate that is...

Any suggestions on whether there is any other way that might be easier for me to do this? 

Cheers

Deric

To me this doesn’t sounds logical at all. Much in audio isn’t. But a fuse ? At $150 ? Crazy 😜 

It must be possible to measure and  document what’s making such a device changing the output of a power supply. Either it’s Paul Hynes SR4T or the Farad. 
 

Can someone come up with some evidence or suggestions why a fuse should change sound ?

If it can’t be documented somehow, should we assume audiophile fuses is just snake oil ?

Can the small resistance make a difference between hot and neutral, and cause an issue in the transformers. If so, just add dual fuses ?

Does a fuse radiate noise ?

https://www.ieee.li/pdf/viewgraphs/overcurrent_protectors.pdf

https://www.allaboutcircuits.com/technical-articles/understanding-the-details-of-fuse-operation-and-implementation/

How Does a Fuse Work?

A fuse is a simple and highly effective way to protect a device from dangerous levels of current:

1. Current flowing through a conductor’s nonzero resistance leads to power dissipation.
2. Power is dissipated in the form of heat.
3. Heat raises the temperature of the conductor.
4. If the combination of current amplitude and duration is sufficient to raise the temperature above the fuse’s melting point, the fuse becomes an open circuit and current flow ceases.

New to this forum, had to register so I could respond to this, hoping the moderator is unbiased.

What the hell?

Ok first off. Doraymon, I assume you mean optical as in SPDIF or similar? because if not, then I think you must have misunderstood something. If you mean that your laptop is taking in pulses of light, like a fibre-optic connection would, and on it's own converting it into data, and that is improving your DAC's audio output, then, really, no offense meant, but all hope is lost.

Speaking with experience amounting to 10 years of music production life coupled with 5, soon 6 years of electronics engineering, the AQVox ethernet switch does nothing, and is a scam. Absolutely nothing about it is improved, except the physical sturdyness which I assume is an accidental byproduct of all the superglue, hot glue and resin compound they added all over the thing.

No components have been swapped out for improved ones in the board, solder jobs have clearly been re-done on the connectors (likely to accompany their own branded overpriced cables while excluding everything else, and the joints are brittle and not anywhere close to professional. That's it. That's all they did, except add in an extra IC that in it's current configuration literally doesn't do anything, and swapping out the xtal for a cheaper one. Under analysis, practically every component introduces more (not less) noise than the ones on the base model, and I consistently encountered bit-loss and checksum errors when analyzing the chain, more so than on the base model they've used.

Not to mention the fact that all of this should have been obvious to anyone who thought twice about it. This is TCP we're talking about. It goes through error-correction, jittercompensation, delay compensation, buffers, and it's even goddamn encrypted. No matter how hard you try, you literally can't make improvements to it that you couldn't by just getting a decent ethernet switch from a reputable brand, which AQvox absolutely isn't. They're hated by practically everyone of note in both the music industry, and in all engineering communities I've managed to find it talked about in. 

And that's not even getting into their marketing, which is filled with outright plain lies, and now there's a pending class action lawsuit coming their way because of it. (which is good news though because you guys can get your money back with interest when, not if, it succeeds.)

But seriously. I can't stress this enough. The components are WORSE. they made the base model WORSE, and slapped a 3000% price increase on it while lying in their marketing. It's more outrageous than if they had just straight up not shipped it to anyone who ordered it.

Buy the base model, if you have the AQvox one, open both up and see for yourself. If you have an oscilloscope you can really see for yourself. This is a scam, and the fact that placebo has worked so well on you guys is a testament to how low this company has fallen in order to extort money out of people who don't know better. I don't blame you guys for falling for it, but I do blame you for defending it when challenged about it.

i am currently eyeing on either the canare 4s11 or the Morgami w3103 but are open to suggestions from others.

thanks

Deric

I’ve long been fascinated by JVC’s K2 system. Some of the XRCDs and other K2-mastered CDs I have sound wonderful, though I have no idea whether the actual technology has anything to do with that. But on a whim I just picked up an affordable old K2 DAC. Yet I still don’t quite understand what the K2 system did (or didn’t do).
 

There are plenty of sources (see page 36 and after) out there, but none are particularly clear. Is it upsampling? Jitter reduction? Noise shaping? Something else?

If any folks here have more knowledge about K2 and can point me to other sources, that would be much appreciated. 

It seems to be quite the business to sell specialized networking hardware (cables, switches & routers, data stream cleaners) for the audiophile minded. I also noticed that this topic seems to divide the community. Unfortunately I find it very hard to build my own educated opinion because most of the postings, blogs, vlogs etc. are opinion based and lack any scientific method to quantify the change in the audio signal and thereby attempt to really proof their point, instead of preaching. If there is a difference then it should be measurable, shouldn't it? So where are these reproducible tests and measurements and proofs?

Kind regards
sine

Am I missing something?

1. Pure analog recording and recording - roughly late 1950's to late 1970s on vinyl records and magnetic tape (the Analog Era)

2. Digital recording and (re)mastering at 16 bits 44,100 on compact discs - early 1980s to approximately 2010 (just to pick a date) on compact discs and digital tape (the Compact Disc era)

3. High Resolution digital recording and (re)mastering at 24 bits and 44,100, 48,000, 88,200, 96,000 (more or less the norm at the moment), and 192,000 - 2010 - present via HiRes downloads and streaming (the HiRes era.

During the Compact disc era, the most common recordings were either pure digital recordings made with digital mics, mastering, and medium (so-called DDD), and analog recordings mastered digitally at 16-44 and distributed digitally (so-called ADD).

During the Hi-Res era, the most common recordings are made digitally at 24-96 and distributed via download, streaming, or audio discs capable of storing the larger sized hi-res recordings.

My question concerns primarily the quality (theoretically speaking) of 16-44 digital recordings made during Compact Disc era as compared to Hi-Res remasters of analog recordings made today. Put another way, if it were possible to make 2 recordings of the same music, with one recorded with analog mics and then mastered at 24-96, and a second recorded with digital mics and mastered at 16-44, which recording should objectively sound better? (I am operating under the assumption that a purely hi-res digital recording and mastering would be superior to both.)

The reason I ask this is because if it were the case that the hi-res remastered analog recordings are superior to the 16-44 digital recordings, then purely as a matter of historical interest, the Compact Disc era could be considered a step backward in terms of audio fidelity. I say this because we can remaster old analog recordings, but we can't improve 16-44 recordings except by upsampling, as to which again, I operate under the assumption does not objectively improve audio quality. Thus recordings made during the Compact Disc era that were considered benchmarks in terms of performance and recording quality at the time cannot be made better in the hi-res era, whereas pre-Compact Disc era recordings can be improved.

Does this question make any sense? Am I missing something? Are any of my underlying assumptions faulty? Has this already been discussed ad naueum? (If so, I apologize and ask that someone might kindly direct me to a prior discussion).

Thanks very much,

Rod

P.S. I am ignoring DSD because it is not directly comparable to the digital-analog comparisons I am interested in here. I am also deliberately ignoring historical differences between performers, venues, producers, etc. that may make objective comparison between recordings over time impossible. Rather, my question goes simply to the theoretical comparison under identical circumstances between hi-res remastered analog vs. 16-44 pure digital recording. If the question has any validity, then one can speculate about whether listeners may have suffered because of the inability to improve recordings made during the Compact Disc era. In other words, would we as listeners be better off today if recordings during the Compact Disc era were made with analog mics?

5 hours ago, manueljenkin said:

... science studies do show humans can discern a time precision way above what a 48khz sampling rate can reliably capture (5micro seconds is what I remember, check some MIT studies and stuff).

This is incorrect. 48 KHz sample rate has a "time precision" in the picosecond range. Here's something I wrote for 44.1 KHz:

The time resolution of a 16 bit, 44.1khz PCM channel is not limited to the 22.7µs time difference between samples. The actual minimum time resolution is equivalent to 1/(2pi * quantization levels * sample rate). For 16/44.1, that is 1/(2pi * 65536 * 44100), which is about 55 picoseconds. To put that in perspective, light travels less than an inch in that time.

Shannon and Nyquist showed that as long as you keep all components of the input signal below half the sampling frequency, you can reconstruct the original signal perfectly - not just in terms of amplitude, but in terms of temporal relationships too. They only addressed sampling, and assumed infinite resolution in amplitude. With a digital signal the precision is limited by the number of amplitude steps, leading to the above formula.

If anyone has difficulty in understanding why the above is true, please post. I'll try and explain it in other terms. It's non-intuitive that an event that occurs between samples can be accurately captured.

If you want to see a real world demonstration of a single event (the edge of a square wave) being accurately sampled between sample points, check out Monty's show and tell at the 20:55 mark. If anyone following this thread hasn't seen the video before, I strongly suggest you take the time to watch it all.

Would appreciate other's contribution as well. What I hope is to be able to write a very succinct, under 2 page entry about how this whole issue evolved; how it started and what keeps it going.

If you are on the opposite side, I would request that you don't post on this thread please. Not because I don't want debate. Its just that I want ideas that will help with my blog and these will be best gotten from people on the skeptics side of the fence. I think it would be a good idea if others from the opposite side of the debate did something similar as well.

Basically my argument is this: the unique characteristics of audio have provided an environment where this phenomenon has been able to gain traction. You would not expect to, and will not find an analogy in photography for example. Due to current legislation the companies who peddle this expensive stuff are able to get away with not proving up their claims. Audiophile land is a bit of a backwater, so this is not surprising. It seems the con has gained enough momentum now for it to have reached the stage where the sheer number of believers has given it an elevated status.

Anyhow look forward to comments. Will likely need help getting the significance of the null testing right and other bits. Will write sections of it up here as I get time and ask for feedback.

However, other skeptics feel free to post your ideas anytime without waiting for me.

https://alpha-audio.net/background/the-mutual-influence-of-speakers-amplifiers-and-cables/