A normal person uses the audio system to listen music, while an audiophile uses music to listen to his audio system.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.
Rinse, repeat.
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.
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.
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?
Even if every properly conducted blind placebo test seems to indicate otherwise?
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. :-)
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.
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.)
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.
Bad wiring and a sub-par setup will make that bleed right through your speakers.
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.
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).
search for "brut nature" which for cava, champagne and the like means it's only the sugar in the grapes themselves.
[0]: https://en.wikipedia.org/wiki/Sweetness_of_wine#Residual_sug...
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.
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.
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 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.
For Widows there is Rightmark, maybe you could run the Macbook's output against a decent Windows DAW.
audio.rightmark.org
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.
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.
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.
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?
:-)
(Spoiler alert: Recording studios do not install their own power poles. What they do, though, is install isolated ground circuits.)
https://www.amazon.com/AudioQuest-Diamond-feet-Braided-Cable...
It's an HDMI cable! The video signal has CRC in it and is packetized, it's either going to make it or it isn't. The quality of video delivered at, for instance, 2160p 60fps 10bit color to an expensive 80" 4K TV is going to be literally indistinguishable from the same video delivered by a $16 Amazon house brand HDMI cable.
Or this, it's a $340 cable for 1000BaseT Ethernet:
https://arstechnica.com/gadgets/2015/07/gallery-we-tear-apar...
People actually make the claim that their FLAC "sounds warmer" when transferred through this. How, exactly?
That's not enough? How about the $10,0000 1000BaseT Ethernet cable? https://arstechnica.com/staff/2015/02/to-the-audiophile-this...
I can sort of understand methodology and expensive equipment modifications to separate AC power from amplifiers, and shield AC power lines and such, keep AC power runs away from speaker wires, etc. That has at least some basis in technological reality.
Also: Acoustics are what matters the most (and they are the hardest to get right, just by throwing money at the problem, that is why it is not usually beeing discussed that much).
There's something to be said for a bit of extra effort in filtering, but most of what audiophiles pursue has no basis in electrical engineering reality.
Edit: a commenter below has a link to an actual audiophile SATA cable.
If it’s all-digital and marketed to audiophiles, it’s a scam and/or absurd. You can’t improve upon bit-perfect. (Room calibration being a notable exception.)
But it is incorrect to say that digital transmissions must be at the mercy of physics. Digital transmissions can defy physics. There are numerous techniques which ensure that a signal reaches the destination with precisely zero flipped bits and exact timing. And even when the signal has no integrity mechanisms, in practice the error rate will be low enough to never matter in practice.
And you know what, even if bits were getting flipped and jitter was extreme, you still wouldn't have signal degradation in the ways described by audiophiles; you would get a raised noise floor. Random error is noise. Noise is random error.
Of course this is never an issue. We can send digital signals that are millions of times more complex than digital audio, with zero problems, using equipment that is insanely cheap. The assertion that the extremely low data rate PCM audio signals have some special risk associated with them is delusional.
Sorry, I misformulated my sentence.
It should have read (emphasizing what I left out from the sentence in my previous post):
> You mean like those people, who argue, that: “Everything is a sequence of Zeroes and Ones, therefore the signal either gets transmitted or not!”, while totally leaving out the fact, that this digital signal materializes in the real world via an analogous signal, which is electricity, and therefore each digital data transmission (as long as not optical) is submitted to the physics of electricity, that happen totally outside of the "digital domain"?
Though, I think you are nitpicking, because, from the rest, it should be totally obvious, what I wanted to say. Use the error correction, man ;-)
Therefore:
If an analogous signal's "success" can suffer from the material, through which it travels, then a digital signal will suffer for the same reasons, simply, because it is not a digital signal, but, materialistically, both signals are of the same sort.
Even if we could create the perfect mathematical concept in our brain and have a solution for anything, as soon as we step into the real world, that is, the material side of affairs, many unexpected things can happen, that have to be accounted for at the next time.
In engineering we call this a "race condition". I like the definition on FOLDOC: https://foldoc.org/race%20condition
> ”Anomalous behavior due to unexpected critical dependence on the relative timing of events.“
Or, in other words: chaos!
tl;dr: As soon as the "digital domain" steps into the "physical domain", you can not guarantee a "side effects" free system.
Let's start with a real-life example:
I have a Macbook from 2006, which has a Firewire port. I have an external audio-interface (Onyx Satelite Pro), that has a (Logitech Z4) active speakers system connected via line-out. When I got the Onyx, I had to realize, that the delivered Firewire cable was too short for my setup, so I had to purchase a longer one. I took great care to buy the cheapest one in the store (which was a Radio Shack type), because, hey, digital is digital, or isn't?!.
When I came home and exchanged the two cables the first thing I realized, much to my surprise, was, that the audio sounded different! The old cable sounded more "relaxed", "warm", "analogous". The new sounded "faster", "analytical", a little "sterile". How could that be? All I changed was the cable...?!
Conventional audio wisdom, especially of those, who call audiophiles "audiofools" dictates, that a digital signal is just zeroes and ones. The sequence either get transmitted or not. Add in the timing component, which is either to the point or off, and things should be pretty clear: What goes in is what comes out. No loss of information. Should there be such loss or distortion, we can error correct it, at least up to a certain degree. Therefore, one always get a perfect, or near perfect, signal.
That, at least, is the argument.
If that argument (alone) would hold true, and be sufficient to describe all, that happens in such a system, I would not have been able to hear a difference, by exchanging the digital interconnect, right?
But I did!
Therefore there must be something happening outside of the digital domain, that influences the signal. In my case, clearly the cable!
And now I come to the conclusion, which should answer your question:
> What I don’t understand is the point you’re trying to make and how it relates to real digital audio systems.
Former example covers exactly your request for a real digital audio system. What happens here is, that we step out of the digital domain into the real world domain, which means the physical and chemical domain. Namely, the domain of audio components. These components are made out of matter. In the case of electrical signals (Firewire, USB, etc.), the medium, which transmits our signal, is the same, which would transmit an analogous audio signal. Signals get transmitted as electricity with ever changing voltages (leaving out optical transmissions now). Therefore all, that applies to an analogous signal transmission, applies to digital signal transmission, as well, as long as the transmitting medium is electrical.
Now, you may, say, that that is not of importance, since the information carried is still a sequence of binary data, and it gets transmitted or not, given the theory behind it, it should be perfectly reproducable. But then, I ask you, why the difference in sound?
Given a perfect program (our digital concept), when applying it to the real world (physical/chemical), race conditions may occur, simply, because of unpredicted side effects (you know the importance of this), that come with this domain. Our perfect program, here, would be the theory of lossless, bit-perfect data transmission: the digital domain. Or - Theory. Our real world would be the cables, connectors, building blocks of electronics, which are: the chemical and physical domain. Or - Matter. And there is something happening in the material domain, that makes an impact on the sound. I don't know what that is. But I can clearly hear it on sub 1000€ system. Now imagine, how much more this could be heard on a stereo system, that cost 25.000€ or more...
Closing, I would like to requote the definition of "Race Condition" from http://FOLDOC.org/Race%20Condition:
> Anomalous behavior due to unexpected critical dependence on the relative timing of events.
The anomalous is the key word here. There are side effects, that happen outside of the "digital domain". And it is these side effects, that change the sound. And audiophiles know this. If a cable can change the sound, then capacitors, resistors, whatever, may well change the sound, too.
Placebo effect. 100% guaranteed. It is literally impossible for digital audio signals to degrade in any way that even vaguely aligns with your descriptions. It is analogous to using a different brand of SD card to get richer, more vibrant colours out of your digital camera. It is analogous to changing the ethernet cable on your computer to make web pages more insightful.
Bits don't work that way.
Whereas the placebo effect is very real and very powerful.
Firewire is also a two-way packetized, error corrected protocol. If a cable was not delivering the signal reliably, the consequence would have been a lower maximum transmission rate, not error that could affect sound. If there was any change in the signal due to the cable, your Onyx interface could not maintain a stable connection with your computer and you'd get no sound at all.
Your attempt to describe how digital signals are transmitted is actually just plain wrong.
Furthermore, race conditions have absolutely no relevance to this discussion.
> [Digital] Signals [when not transmitted optically] get transmitted as electricity with ever changing voltages [...]
You wrote:
> Your attempt to describe how digital signals are transmitted is actually just plain wrong.
Wikipedia writes:
> A waveform [...] as a digital signal [...]. The two states are usually represented by some measurement of an electrical property: Voltage is the most common [...]
Therefore I will leave it here.
The latter is very unlikely with asynchronous USB DACs since USB will retransmit corrupted packets, and jitter isn't a concern since the DAC buffers then uses it's own clock.
The former, well yeah, it's impossible to completely eliminate RFI no matter what you do, short of optically isolating the source, putting the analog equipment in a faraday cage, and running it all off of batteries.
The only thing I can agree is better on a digital cable is increased shielding to avoid ground noise.
If you have nanosecond timing requirements, then those might be useful, but for audio they are overkill.
Update: geez, it was satire...
...or was it?
https://serato.com/forum/discussion/1541194
https://www.head-fi.org/threads/flac-vs-wav-format-surprisin...
> Be civil. Don't say things you wouldn't say face-to-face. Don't be snarky. Comments should get more civil and substantive, not less, as a topic gets more divisive.
Sometimes snarky comments will be forgiven if they are substantive. Yours contributes very little beyond what the parent/grandparent comments had already mentioned.
For listening purposes the transmitted data can be buffered on the receiver DAC. This means that as long as the buffer on DAC is not emptied everything else may be failing badly and it will have zero effect on your listening experience.
Professional cables are usually flexible, with robust insulation and connectors. They may also be fire rated. Quality control ensures that cables will always be up to specs because people doing the installation have other things to do than dealing with defective cables.
Like most pro stuff, pro cables are not "better", they are cables that won't be a problem.
This isn't _strictly_ true - I've seen basically the equivalent of noise in a HDMI picture (white dots all over the picture), and eventually tracked it back to a faulty HDMI cable.
I did replace it with the cheapest (<£5) HDMI cable I could find though, so there is that.
Say the HDMI cable is crap and it can only do 100 packets per second, and HDR needs 1000 packets per second, the HDMI chips are going to negotiate a HDMI transfer settings without HDR. Going back further, say the HDMI data path is negotiated to do only basic HDMI, say, DVI-like video. If even that is too much for the cable to handle, say, it constantly drops a few packets, you would get either perfect pixels or missing pixels.
The only difference is the threshold.
Is there any error correcting code/protocol in HDMI spec to fix individual bit flips?
Displayport has packets with a checksum but I don't think there's any resending, only discarding.
" While the receiver may reach a determination sooner, the receiver must determine loss of synchronization at least by the time it has detected 50 consecutive data island packets with ECC errors. The HDCP Receiver must either assert the REAUTH_REQ bit of the RxStatus register or de-assert HDCP_HPD to the upstream transmitter once it determines loss of synchronization. "
We are running FPGAs (with NICs) and were getting lots of packet errors over the copper cable SFP cables. Turns out, we either needed to write out noise cleanup logic into the FPGA, or add in a noise cleanup module which would add an extra 50ns to packet processing to the FPGA. Simple solution was to convert to fiber and the noise and errors all went away.
I'd say if audiophiles truly want unmolested electricity, they ought to use optical fiber for as much of the run as possible.
Now, what I think you mean is that the copper in the cable picks up noise either from the device on the other end or from the surroundings and that has the possibility of affecting the DAC.
Even moderately well designed receiver should be able to galvanically isolate itself from outside sources of noise without having to resort to fiberoptics or wireless. It's just couple of dollars max for extra few components on the board.
There is no such thing as a digital transmission. All signals are analog.
I'm just so tired of all the bullshit where it "Must be a 1 or a 0". That is patently not true. HDMI video, to name one example very relevant to this thread, has no error correction, and bit errors will indeed crop up as the "sparkling" artifacts GGGGGP reported.
We have a cluster with ~1000 nodes and all generations of Infiniband (DDR/QDR/EDR) equipment, and routing all these copper cables without damaging them is not easy.
The quality of digital tranmsission over copper cables is dependent on the size and clarity of "eye", which is the pattern which two sinusodial signals when run through and oscilloscope. Lower quality cables (HDMI/DP/Coax/etc.) has blurry and small eyes, which increases chances of artifacts and drives the error correction modules harder, while better cables are easier on the both sender and receiver, since they produce lower number of errors to correct.
Trivia: Amphenol / Gore has high speed digital interconnect cables called Eye Opener Plus [0], which also used some of the Infiniband cables that we use.
[0]: http://www.spectra-strip.com/Eye_Opener_Plus_Cable.cfm?Nav=E...
The description of that cable though, says Perfect-Surface Technology applied to extreme-purity silver provides unprecedented clarity and dynamic contrast.
An HDMI cable can not affect the "clarity" or contrast of the image or audio it carries. It can just succeed or fail (and failure results in obvious pixel noise, not some overall lowering of contrast or clarity like in analog)
I don't disagree with your main point, but this actually isn't quite true. The HDMI signal is split into 3 distinct interleaved periods: video data, data island and control. Video data is not packetized and the only possible error detection it has is from TMDS signaling, but no such error handling is required by the TMDS spec. You can absolutely get imperfect transmission of an HDMI video signal due to cable or other electrical problems. Auxiliary packets in the data island, including audio data, do have an error correction scheme (BCH + TERC4).
Feel free to check out the spec: https://glenwing.github.io/docs/HDMI-1.4b.pdf
I have seen an HDMI transmission with white blinking pixels at content edges that were not from the original signal. I ran test signal through that source and verified the errors on an HDMI waveform monitor, after which I tested the cable and threw it away.
HDMI video is not magically impervious to error because it is digital. It's actually a pretty straightforward transmission scheme.
This was while working on an HDMI-output device, working with team members who had written the original HDMI specifications (so, yes, I've read them).
It is possible to have subtle degradation in HDMI-delivered content. It's rare, but possible. As one might expect, comprehensive failures (e.g. black screen, periodic HDCP key mismatch) are far more common, as are negotiation/configuration failures (e.g. wrong resolution, wrong framerate, wrong color space).
What does matter is getting a decent HDMI cable. I'm going to get a braided cable which I know supports the bandwidth of HDMI 2.1 (I think that's the current version).
1k? No. Probably more like 25.
The AmazonBasics cable will do 18Gbps, so it can do 2160p, 60fps, 10 bit, but only with chroma subsampling (which is fine because most video sources have chroma subsampling). The Amazon cable may work with a 4:4:4 signal, but it's not certified for it. There's a level above that for 48Gbps cables.
I totally agree with the general idea, though. As long as a cable for a digital signal meets the specifications set by the standards body, you're good.
Yeah that basically describes me, between the ages of 16 and 26. We would spend hours moving gear to different locations (homes), fiddling with cables, different amplifiers and loudspeaker to find the best possible match our (limited) finances would allow.
Tube amplifiers combined with mosfet pre's, magnapans with external bass speakers. Class A amplifiers that would require a dedicated powerfuse per amplifier. First press vinyl from Japan. Van-den-Hul needles. Granite turntable platform.
Every single weekend we'd swap pre amps, amps, turntables, arms, elements, needles and listen to the "audio system via music" :)
What happened to my setup, you ask? I got married. My living room was modelled around the audio experience: it was not an ideal love nest :) I actually gave my hand crafted, tailor made end amplifiers away just a year ago to a young stranger I overheard talking about his audio quest and who is in the phase where he is searching for his ideal setup like I was 30 years ago.
Sound quality is not music quality.