Nitpick: “audiophile-quality sound” it seems, is becoming the new “military-grade encryption.”
I don’t have many other comments to make other than I am surprised rust-analyzer was only mentioned in passing.
I don’t have many other comments to make other than I am surprised rust-analyzer was only mentioned in passing.
Beyond that I wish the article had explain a bit better why it chose these "better-than-std" crates. I'm actually using all the std variants in my projects, I'm curious to know if I'm missing out or if I just happen not to hit on their limitations.
At least for parking_lot, its README has a long list with its advantages over std: https://github.com/Amanieu/parking_lot/blob/master/README.md
That being said since they're drop-in replacements for the most part I suppose I could just try to rebuild my project with this crate and see if I notice a difference performance-wise.
With crossbeam for example, you can hit roadblocks with std since their channels are MPSC, whereas crossbeam supports MPMC channels (and is faster than std in every meaningful measurement last I checked).
Reading the description it almost seemed too good to be true but if it's indeed objectively better in basically every situation I should probably give it a try.
Rust leaves a lot of improvements to its standard library to the community, so these improvements start off as separate libraries for faster iteration. The most recent example I remember is the hashbrown crate replacing the standard HashMap.
To clarify, we're targeting "transparent" sounding audio, not "FLACs or bust" audio. Right now we send stereo 48kHz 96kb/s Opus (CELT, not SILK) that we found hit the voice transparency sweet-spot compared to the lossless audio source. We had used higher bitrates in the past, and could easily go back to them, but quality plateaued at around 96k in our experimentation.
More than choosing sane transparent-sounding encoding parameters, the biggest difference in fidelity by far was choosing the correct microphones and speakers for accurate reproduction of voices.
Are you using 48khz for a specific reason?
Audiophile grade at least has roots in high fidelity.
Does it though? Audiophiles generally seem to eschew fidelity in favour of something that sounds subjectively nice, including the psychoacoustic effects of spending a lot of money.
Eg. they seem very fond of "warmth". If you asked me to make something sound "warm", I'd be applying some soft clipping and dampening the top end, not eliminating sources of distortion.
Edit: If you actually wanted high fidelity, you'd use studio headphones / monitors, which are designed to be "unflattering", so you can be confident you'll hear any issues when mixing / mastering. People don't normally listen for pleasure with those, because they become fatiguing after a few hours.
Choosing equipment because you like the sound is a very reasonable thing to do, but it's not the same as pursuing fidelity.
And some are all about accuracy and measurements.
For instance, I use Sennheiser HD600[0], which I strongly recommend, attached to Topping DX3 Pro (old model)[1], which I cannot recommend, as the v2 model shipping now is garbage[2], a consequence of a redesign to work around high fault rates. Mine is fine as problem units fail within weeks, and I've had it for years.
[0]: https://reference-audio-analyzer.pro/en/report/hp/sennheiser...
[1]: https://www.audiosciencereview.com/forum/index.php?threads/r...
[2]: https://www.audiosciencereview.com/forum/index.php?threads/m...
Also warmth is just a single quality. I have a pair of very accurate “cold” headphones that I prefer for music and a pair of “warm” headphones for electronic music and gaming.
Past the headphones, it is not so much warmth as it is space in the sound for me. My headphone amplifier sounds effortless and that’s the best way I can describe the quality of what I hear.
The characteristic of warmth is related to amplification of certain harmonics as well as equalization in the signal. This is fairly well understood by now.
I don't really know what, if anything, that means. But if we're talking about fidelity, surely the ideal would be no sound signature? If a particular "sound signature" makes it sound "warm", surely it's decreasing the fidelity?
That might sound good! But it's a less-than-perfect reproduction of the source signal.
If there's a better explanation than what I've come across every time I've search for this, I'm all ears and honestly open to being corrected.
You're saying that I ought to judge the merits of audiophile equipment by the subjective measure of whether I like the sound of it. Which is the metric I said audiophiles would favour.
> If apply "some soft clipping" it will sound bad
Soft clipping often sounds nice, which is why it's very commonly applied to music. You're saying that eg. the sound of a classic Vox amp is bad, which I guess you're free to believe if that's what your ears tell you, but it's certainly not an objective truth.
The attached documents have additional information on implementation and (non) usage, including deadline to migrate legacy military systems. It's sadly quite cumbersome to go through the tens of PDF to find the relevant information.
It's too bad they didn't explain it. I expected they meant allowance for "full bandwidth" audio (possibly including music you can listen to).
Video conferencing systems generally use voice-only codecs compressed to shit, full of artifacts in the voice range and utterly dead outside of it.
https://web.archive.org/web/20200310174634/https://people.xi...
The trick is that our hearing systems are logarithmic (we can't hear a quiet sound next to a loud sound--that's what compression relies on), so they map to floating point numbers better (ie. 16-bit floating point is way more than enough).
24-bits is effectively for recording engineers so they have lots of headroom and don't have to worry about clipping basically at all (6dbm per bit implies about 18dbm of extra headroom which is a LOT).
However, when you calculate non-linear audio effects, you want extra bit depth (generally floating point) because cancellation and multiplication in your intermediate results can really move your noise floor up into bits that humans can actually hear.
The effect of bit depth has little to do with how you perceive the sound; what adding more bits does is allowing for more dynamic range, i.e. more difference between the loudest possible and the quietest possible sound. More bits brings down the noise floor. This means that for example the final part of a fade-out retains more detail at 24 bits than at 16, but this difference is not something that you would be able to observe in normal listening conditions.
If you like to learn more about the effects of bit depth, I would recommend “Digital Show & Tell” by Xiph Mont at https://www.xiph.org/video/.
That really doesn't make any sense. The bit depth provides for a dynamic range, meaning the difference between the loudest and quietest sounds which can be encoded. 16 bits is enough to go from "mosquito in the room" to "jackhammer right in your ear". Congratulation, 24 bits let you go up to "head in the output nozzle of a rocket taking off" with room to spare, that's… not very useful?
Now what might make sense — aside from plain placebo — is a difference in mastering. For instance lots of SACD comparisons at the time were really comparing differences in mastering, with the SACD converted to regular CDDA turning out way superior to the CD version because the mastering of the CD was so much worse.
The "Loudness Wars" is an especially bad period of horrible mastering, and it went from the mid 90s to the early-mid 2010s (which doesn't mean that regular-CD has gone back to "super awesome", just that you're unlikely to have clipping throughout a piece these days).
When people talk about 24 bits (and >48kHz) in the context of "audiophilia", it's generally about the data at rest and "HD audio" (aka 24 bit music files and downloads). Not about the bit depth of the processing pipeline for which it's generally acknowledged that yes, >16 bit depth does make sense for the audio processing pipeline (as well as the original recording).
Nobody said it would hurt so I’m not sure why you’re pointing out the consensus like it’s some sort of profound statement.
> If your recording is not loud, you lose your dynamic range
If your sound engineer is wasting your dynamic range, maybe get a better sound engineer? And if they manage to fuck up something at the core of their job, there’s no reason they wouldn’t fuck up just as much with 24 bits to waste.
> So it is true in both senses.
In no meaning of “true” and “both” in common use.
Unless this was a double-blind study and the audio levels were exactly the same between runs, this is useless data. Even a 0.1dbSPL difference between runs is noticeable (people gravitate to louder sounds as better).
> every time I switch sound cards to 24 bits
This may be related to the sound card. I use an external DAC, not a soundcard, as most soundcards that come with computers are not up to par.
Changing 16 bits to 24 bits should not change the audio in a way that is discernible to the human ear.
> This may be related to the sound card. I use an external DAC, not a soundcard, as most soundcards that come with computers are not up to par.
For simplicity, I did not talk about them separately, BTW, following your logic there is no point in bying DAC, unless there was a double-blind study, comparing these DACs to cheaper sound-cards. Both are 16-bit/48000, are not they?
> Changing 16 bits to 24 bits should not change the audio in a way that is discernible to the human ear.
This a bold statement, which begs a proof itself.
Only if one doesn't understand what those bits mean or what they correspond to.
These bits are important for quantization, which is the process of converting analog sound into digital numbers. On a graph, X = time and Y = amplitude. The higher the bits, the higher the resolution.
A 16bit recording has 2^16 steps (discrete values) available for amplitude (65,536) and a 24bit recording is 2^24 or 16,777,216 steps.
So why is this important? Well, a 24-bit recording can more finely record differences in amplitude. Given that 1bit = 6dB: a regular 16-bit recording already has a dynamic range of 96dB. A 24-bit recording has a dynamic range of >144db. At ~125-130dB SPL is where hearing loss (permanent) begins.
You do not hear the difference because if you were listening to a 24-bit recording on a 24-bit capable system at sound levels loud enough to actually discern a difference, you would have permanently damaged your ears. Actually, I believe that applies to 20-bit, let alone 24-bit.
So why do 24-bit or higher recordings even exist? They are useful for people mixing and working with the raw audio, before it gets processed down to 16bit audio for distribution. At 24-bit resolution you have a larger amount of headroom before you start clipping, so it's easier to work with considering you have X amount of bits that are just part of the noise floor.
This is also assuming your input files are actually 24-bit to begin with. The vast majority of files are 16-bit because there is literally no point as a consumer to have larger file sizes for no humanly audible benefit.
44.1kHz 16-bit files are all that you need as a human consumer of audio. 48kHz has to do with video and is not better than 44.1kHz because you (a human) cannot hear the difference. 44.1kHz is 22.5kHz x 2. Humans hear sound from 20hz to 20kHz -at best-. This is assuming perfect hearing with no degradation. We sample at 44.1kHz due to the Nyquist-Shannon sampling theorem, and 22kHz gives us just a bit of headroom to apply filters to avoid aliasing. [2]
So I reiterate my initial assumption: flicking a switch to change from 16bit to 24bit should not magically change the quality of audio (in a humanly discernible manner). Assuming the file being played is 24bit lossless audio in the first place.
> BTW, following your logic there is no point in bying DAC
We're talking about dedicated external equipment vs an onboard soundcard+amp which are generally neglected. Not -all- onboard cards suck of course, the Realtek ALC1220 chip on my mobo seems to be comparable or better than entry level DACs from the specs I'm seeing. This is assuming no interference is happening, which is more likely to happen around unshielded electrical components. If you don't believe this is a thing, ask why the audio industry uses thick XLR [shielded AND grounded] cables as standard.
Certain headphones require equipment that can drive them properly, whether it's an onboard soundcard+amp or a DAC+amp. For example, my sennheiser hd600s are 300Ω but some models go up to 600Ω. And yes, the quality of the amp/preamp does make a huge difference.
If one can prove that a component is unable to drive a component, or is sub-par mathematically, one doesn't exactly need double abx trials. Those are for tests like "Monster says their $200 cable is better than <X> standard cable?", or "Is a McIntosh amp better than a $<amount> competitor?".
I don't need to do a double ABX study to realize that beats headphones are drastically worse in performance than sennheiser hd600s: [3], [4], [5]
[0]: https://www.mojo-audio.com/blog/the-24bit-delusion/
[1]: https://web.archive.org/web/20200202124704/https://people.xi...
[2]: https://en.wikipedia.org/wiki/44,100_Hz#Origin
[3]: https://reference-audio-analyzer.pro/en/report/hp/monster-be...
[4]: https://reference-audio-analyzer.pro/en/report/hp/sennheiser...
[5]: https://reference-audio-analyzer.pro/en/report/hp/audio-tech...
It is so believed (although there's a lack of supporting evidence, and knowledge that human hearing has excellent dynamic range), but only as long as the mastering work was well done. 24bit allows for much less destructive human error and is very welcome. Much more so than absurdly high sample rates (96KHz, reproducing sounds up to 48KHz as per Niquist), which are of dubious value.
>my sennheiser hd600s are 300Ω
At some frequencies. At some others, it's more like 600Ω. Impedance is seldom stable across the frequency range in headphones.
Amplifier design should account for this and still provide enough power[0].
Output impedance of headphone jacks should be low enough (1:10 is commonly cited, which means <2Ω in practice as 20-30Ω headphones are very common) relative to the low end of the headphone impedance range, in order to prevent the impairment of frequency response.
>Not -all- onboard cards suck of course
But most do. The design of audio circuitry in motherboards doesn't get that much attention. None of my motherboards have good sound. Flaws vary. Some are lowpassed (greedy anti-aliasing filter). Some are noisy. Most have excessive output impedance (typically more than 6Ω, and at times higher than 15Ω). None can output enough power[0] for hd600 (my favourite pair).
Dynamic range is not loudest sound / quitest sound ratio (as would one expect), but loudest sound / noise level ratio. Otherwise you would need to count additional bits to encode quietest sound with low enough quantization noise.
Threshold of hearing could be as low as -9 dB SPL, so one wound want noise level below that. Therefore with 96 dB dynamic range from 16 bits the loudest representable sound would be say 86 dB SPL. But symphonic orchestra music may have peaks way above 100 dB.
The audiophile world would do well to adopt the concept of double-blind study.
“Listening fatigue” when you know which is which is simply placebo.