Audio transmission over USB with cheap converters and poorly chosen levels can result in loss of sound quality. I'm pretty sure you can tell the difference between 8 bit, 12 bit, and 16 bit sound. Choosing levels poorly can reduce your resolution.

Also, there are cheap analog to digital converters which use voltage comparison, which basically linearly "search" for the voltage they should be encoding. So those fundamentally do introduce jitter. (Large changes in signal voltage get encoded a bit later than small ones.) I'm sure there are ways that jitter can be introduced the other way around by cheap DACs. However, if you add more bits and increase the sampling speed, those should largely go away.

Devices like DACs also include a small amplifier, and if those aren't properly designed, there can be some distortion introduced there as well.

That said, there are times when people think they hear a difference, and it's woo. There are other times when the difference is real, however. Audio is a real field with some real science and know how behind it. Please don't grant yourself instant superior expertise just because you've read some stuff online.

(On more than one occasion, I've gotten the "all audio is woo" knee-jerk from programmers who were just reacting to hearing the words "frequency response" not knowing what it was, but labeling it as woo. Ironically, they did remember stuff about Fourier transforms. Go figure.)

Integration type ADCs are not necessarily cheap. And you always put a sample-and-hold circuit in front of integration type ADCs, which completely eliminates the sampling time variance.

Also many people are unaware that the Nyquist theorem states, that the _average_ sampling frequency must be at least twice the highest signal frequency. The sampling points may indeed jitter around a little bit, as long as the jitter is distributed evenly. As a matter of fact, adding a little bit of sample jitter can be used as a poor man's dithering, to shift the LSB noise spectrum above the reconstruction filter cutoff frequency.

Jitter is mostly a problem with sample clock distribution. Too much jitter and the sample clock no longer arrives with the right delay with respect to the data, which may create large transients, e.g. whenever the numeric value rolls over some power of 2 (a lot of bits change); this problem could be avoided using Gray codes, though.

Oh, and then there's Sigma-Delta Modulation.

However audiophiles are a whole different level. People spend tens of thousands of dollars on items that have questionable difference in quality. I guess people do the same with cars though, a Toyota Corolla will get you to work just as well as a Rolls Royce Phantom.

Wine loaded with sugar??? AFAIK, in a dry wine all the sugar is converted to alcohol- even in the case it's been added before the fermentation, which is still pretty rare, at least in major wine producing countries. That is definitely not the difference between a $5 and $20 bottle of wine (in Italy you can still get excellent wine for $5 a bottle if you buy it from the producer, btw).

[0]: https://en.wikipedia.org/wiki/Sweetness_of_wine#Residual_sug...

search for "brut nature" which for cava, champagne and the like means it's only the sugar in the grapes themselves.

If you just want to put together a cheap car with a good engine, that's a different thing from wanting to build a comfortable or speedy car with a good engine. Someone who isn't in the market for that extra level will just be a sceptic and say "this Audi is just a volkswagen with a prettier body", but if you sit in one you'll feel the difference. (not sure if Audi's have VW engines)

Digital to analog conversion is a solved problem, with inexpensive commodity parts. Even if you buy an expensive DAC, the studio gear used to produce the music you listen to has gone through dozens of A->D->A conversions, all done using standard commodity parts. No one builds fancy audiophile-approved DACs and ADCs into mixer or studio gear.

In my home setup, I'm using a DBX DriveRack PX as a DSP EQ and crossover between my main monitors and subs. It's almost a decade old, but its signal/noise ratio is on par with the newest gear you can buy. Any competently built pro gear from the last ~10-15 years with 24-bit conversion can be plugged right into any analog signal chain and be 100% audibly transparent.

People worry way too much about the electronics, and way too little about their speakers and rooms.

An Audi also has just a couple thousand of extra love worth of physical cost added to it, but the retail price is easily upwards of $20.000 above the equivalent VW car. That's because if you sell fewer units, the r&d and marketing costs per unit are much higher, as well as the required profit per car sold.

If you get a good deal on your hardware, you might have that good DAC in a nice box with a pretty manual for $50 physical costs, then put it on a prime shelf in a nice Hi-Fi store for $130 (30% of $400), maybe $50 to target and prime your customer, and then use $150 to pay your audio engineer, your CEO, maybe outsource the rest of the company infrastructure, and then 8% interest to your lenders. That's 1.7M profit at 10.000 units sold (assuming that you do all of this within a year).

I have no idea if 10.000 units is close to reasonable for an indie hardware startup, but calculations become a little more complex if we're going to have to take a multi year ramp up in account..

edit: Sorry I got caught up a little in the startup aspect of it. Your system sounds great, but your DAC (and DSP+crossover system) also cost $400, so you're not making a point?

The DAC I use cost less than $30, yet performs well beyond the limits of human hearing. The DriveRack did have an MSRP around $300 when it was new, but usually $200 on sale, but it does do a whole lot more than a DAC, and the base price level for pro gear is somewhat higher than for consumer gear.

As I said, people focus way too much on electronics, when even basic devices are more than good enough. I don't even have to use the DAC anymore, after I switched to DisplayPort for my monitor, which has a perfectly good analog stereo output. The only reason I keep the Toslink DAC hooked up is pure laziness.

There are definitely things which can perceptually detract from audio which people just don't think about.

I'm about to buy a couple power conditioners for my electronics because the wiring in my new apartment is garbage and every little spike in voltage comes through as a loud pop in my speakers and headphones.

But more subtly, my audio is constantly being degraded because I have a few cheap dimmers which cause a constant interference pattern in my electrical circuit that weren't a problem in my last apartments, and I can even hear fluctuations caused by my refrigerator if I listen closely.

And all of these things which can subtly degrade audio are 100x more of an issue when you're recording audio. Run any given song through a spectrogram analyzer and you'll usually see a horizontal line or two which could be anything from a little cable noise to a CRT screen.

I got the APC Line-R 1200VA Line Conditioner With AVR installed in front of the UPS and couldn't be happier now - it steps down the voltage and gives the UPS peace of mind.

Try the APC, not the most expensive thing but it's worked well for me.

https://www.scorptec.com.au/product/UPS/UPS/44933-VALUE1500E...

I also have one of their smaller BRICs for my HTPC / home NAS JBOD server: https://www.scorptec.com.au/product/UPS/UPS/40279-BR850ELCD

The voltage in that part of the house appears to be better, but the circuit is easily tripped when running toaster + micro or similar, so having these UPSes has taken a lot of frustration away.

Actually just checked my invoice and I got them both in 2012, which is a little while ago now, and they're still good with their original batteries.

Bad wiring and a sub-par setup will make that bleed right through your speakers.

Most of the perceived gains in quality of higher end equipment (DACs, Amps etc.) come from the higher voltage of the outputs (where you connect your speakers and headphones), this translates into an increased loudness/volume level.

So if you want to do blind tests between different kind of equipment, you have to measure that voltage and make sure it's equal across different pieces of equipment (you can adjust in realtime by volume knobs), otherwise the louder one will always win and perceived as higher quality.

The difference is not one of loudness but of 'energy'. A snare hit played from the high-end cd player sounds like someone actually hitting a snare; you can feel there's actual energy in the hit, while the macbook out sounds flat, even at higher loudness.

I'm not sure what is the exact cause but I've always guessed that it comes from the qualities of the built-in amps in the DACs.

I think the general problem is mid-range speakers and reasonable settings is all it really takes to have quality sound, but people want to be above average and therefore look for ways to discriminate between better sound and worse sound differences they can't "yet" hear. From there you get into placebo and random correlations.

It can also be your room. If you're not exactly located in the same listening position the listening experience can change a lot, even with the same setup.

Your room is basically an equalizer and should be regarded as a part of your audio system. That's how some Hifi sellers can trick you into more expensive setups, they just change the speaker positions or your location to optimized defaults. It's crazy how they can trick people.

A harder hitting snare can be caused by changes to the upper mids and lower highs, by anything that is located between you and the speakers, and then the later reflections of the room which also add up.

Aren't those the same? Put more joules into the speaker wire and more joules of sound pressure waves will come out of the speaker cone, which just means the sound will be louder (until you reach the mechanical limits of the speaker, of course, but no DAC in the world could possibly change those).

Not really. If you take some of the transients out of recorded drums they will sound lifeless, without "energy", but not necessarily quieter.

I never benchmarked my Yamaha CD-S300 to other devices, but its flyers and web site never says it's completely flat on the analog outputs. The reviews of the player notes that it has the distinct Yamaha signature too.

Also "Monitor" and "Flat" devices as their name implies sounds pretty flat and dull.

All-in-all you should consider taking response graphs of your devices rightmark.

For Widows there is Rightmark, maybe you could run the Macbook's output against a decent Windows DAW.

audio.rightmark.org

In principle a low enough quality DAC certainly could affect "the stage", but in reality even cheap ones are likely good enough.

> There is no way to convince these people they are wrong

I'm pretty sure many of them don't want to be convinced since it would make their hobby pretty boring. I also wouldn't be surprised if industry players actively spread myths like these to sell more (expensive) products.

There’s certainly woo and outright fraud in audio. But there’s also attention to detail.

Professional studio engineers will spend $3000 on a high-precision digital clock for their DACs because it makes a clearly audible difference.

Software people don’t understand that human hearing, and human sensitivity to timbre and distortion, is spectacularly non-linear.

Hearing isn’t a linear counting device with x volume levels. It’s an incredibly sophisticated neural network that performs real time classification and source separation based on spatial and ambient cues, spectral analysis, predictive modelling, and entropy estimation.

Some people’s brains and ears are better at all of this than others. Those people can hear things other people can’t.

In music, someone with a trained ear can name the notes in a chord. Someone with exceptional skill can name the notes in a random cluster of pitches on the lowest octave of a piano.

Audio quality has analogous levels of sensitivity. People who have neither shouldn’t be telling people who do what they can and can’t perceive.

I don't know what kind of frequency stability you really need (please enlighten me), but I checked on one of the components retailer websites, and found that even the most stable MEMS oscillators are only a few bucks a piece, any DAC or ADC manufacturer that claims that's worth a $3000 markup is running a scam.

> Software people don’t understand that human hearing, and human sensitivity to timbre and distortion, is spectacularly non-linear. Hearing isn’t a linear counting device with x volume levels.

As a "software person" I take exception to this. Who exactly of my colleagues claimed that it was? Not even the cheapest, most terribly sounding, incompetently designed child's toy's "what sound do cows make" 8-bit 8kHz DAC produces output with x discrete volume levels, it produces a continuous signal. A bad sounding signal perhaps, but a continuous one nonetheless. Plenty of software people know this, but do audiophiles?

No, this is pure steaming BS. $0.2 crystal has 50 ppm stability.

>Some people’s brains and ears are better at all of this than others.

Better than radio receivers operating at GHz ranges I guess.

Well, a lot of this Audiophile stuff is voodoo but if it makes them happy, who cares. But I am skeptical about your statement. The easiest way is to always do a blind or double blind study. A German computer magazine did this once with CD vs. mp3:

https://www.heise.de/ct/artikel/Kreuzverhoertest-287592.html

The guy who was best in spotting the difference and assign it correctly had an ear damage. Since mp3 was developed for "healthy" ears this is not too surprising.

"People who have neither shouldn’t be telling people who do what they can and can’t perceive. "

Let's agree on the scientific method. Anything that shows a significant difference in a double blind study, even for a specific individual I accept.

By the way, I hear my music on my computer via mp3 and send it into my super old tube amplifier. :-)

Even if every properly conducted blind placebo test seems to indicate otherwise?

I don't mean to question your expertise on the subject, I'm just after an explanation what are the exact reasons why the setups with separate DAC and amp boxes have made me find new musical instruments on my audio files. And of course it would be nice to know if there is an inexpensive way of getting the analogue signal to a good amplifier without somehow ruining it.

It would be super interesting to compare an iPhone analog vs iPhone with a DAC.

I guess you'd have to use an old iPhone with an audio jack though :(

But I usually go with onboard sound these days. I did get a FiiO E10K when I got new headphones recently, but I've put it away even though the sound is noticably smoother and less compressed-sounding than my onboard sound - it's another cable and associated hassle.

The reason to go for pricier stuff in my mind is to get more I/O. Good audio interfaces for recording can run you hundreds, and you're paying there mostly for having more channels and some recording features. There's more of an expectation of getting what you paid for.

Rinse, repeat.

Case in point: when I used a USB DAC with my NUC there's an intermittent electrical noise that's actually quite audible and distracting (even when nothing is playing). I ended up switching to another cheap DAC that accepts optical input, and the noise completely disappeared. I'm not entirely sure what caused the electrical noise to begin with, but that it existed means there's some potential for USB to mess with the audio signal.

Is this a real thing, or should I upgrade to a better brand of crack?

It's hard for our brain to comprehend that such a tiny gadget can sound so good, so I suspect that accounts for a lot of the perceived difference. Also, some music can sound better with a poorer output device by e.g. smearing noise in the treble or rolling off excessive bass.

The difference is not much. iPhone has slightly better highs and slightly tighter lows. The headphones I'm using is Sennheiser MM70i (They use the same buds with CX400-II, I also have these).

I didn't found the difference with "critical listening". One night I just started listening it, and it sounded different to my iPhone. Then I played the same song on the iPhone and found out the difference.

The difference maybe due to different generations of DACs, different OPAMPS, different power budget, different decoders, etc. As I said before, the difference is not that much, but normal for the evolution of these types of electronics. Nano honestly sounds impressive for a device that compact.

P.S.: I used to play euphonium in a large band and double bass in a symphony orchestra, so I have trained my ears for separation and tracking of instruments and rhythm. This is why I'm a bit more sensitive to these things than most of the people around.

You also don't understand that these people have found a pastime that makes them happy and engaged. Who cares about the actual difference in sound compared to that?

:-)