this post was submitted on 21 Aug 2024
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Do you value high fidelity (audio, visual, or other)? Do you notice a difference? (lemm.ee)
submitted 3 months ago by Please_Do_Not@lemm.ee to c/asklemmy@lemmy.ml
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It often surprises me to see people with time, money, and knowledge settling for subpar experiences that have night and day differences to me. Even at my brokest (pretty darn broke), speakers, headphones, and glasses were always worth researching and some saving up, and the difference between what I'd end up with and the average always feels like it paid off tenfold.

I've got a surprising number of friends/acquaintances who just don't seem to care, though, and I am trying to understand if they just don't experience the difference similarly or if they don't mind. I know musicians who just continue using generation 1 airpods or the headphones included with their phone, birdwatchers who don't care about their binoculars, people who don't care if they could easily make their food taste better, and more examples of people who, in my opinion, could get 50% better results/experiences by putting in 1% more thought/effort.

When I've asked some friends about it, it sounds as much like they just don't care as they don't experience the difference as starkly as I do, but I have a hard time understanding that, as it's most often an objective sensory difference. Like I experience the difference between different pairs of binoculars and speakers dramatically, and graphical analysis backs up the differences, so how could they sound/look negligibly different to others? Is it just a matter of my priorities not being others' priorities, or do they actually experience the difference between various levels of quality as smaller than I seem to? What's your take on both major and, at the high end, diminishing returns on higher quality sensory experiences?

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[โ€“] PM_ME_VINTAGE_30S@lemmy.sdf.org 13 points 3 months ago* (last edited 3 months ago) (1 children)

So I don't value high fidelity video because I don't see very well even with glasses, so it wouldn't make a difference for me.

I do value high fidelity audio because:

  • I am a musician and producer, although not as much as I used to
  • I have ear training
  • I went to recording school
  • I am autistic with sensitive hearing
  • I have audio and acoustical engineering as special interests
  • I'm doing a master's degree in electrical engineering where I've already designed audio gear for my projects
  • I am teaching myself audio plugin design for fun

But I simply can't afford high fidelity gear for every day listening. For my studio monitors, I spent as much as I could to get the best speakers I could afford so that I can be certain that what I'm hearing is an accurate representation of what I "commit to tape". However, for walking to class or going to the market, I'm not gonna pay for expensive headphones that could get stolen, broken, or lost. It's impractical.

My $20 Bluetooth headphones [1] are sufficient for every day carry. They sound "95% of the way there", they don't get in the way when I'm walking, and if I lose them, I can have an identical pair delivered to my door with a couple days. 95% is good enough for me. Actually, I could probably settle for less.

And then there's storage. My library is already > 110GB in MP3 format, so storing it all in uncompressed formats would be unwieldy.

So in the rare cases that my listening hardware is insufficient, I'll usually consult a software equalizer. For example, on Linux, Easy Effects allows me to apply equalizers, dynamic compression, and a bunch of other plugins in LV2 format to the PipeWire output (and input). It's super convenient for watching YouTube college lectures with questionable microphone quality on my shitty TV speakers. Other than dynamic compression for leveling and an equalizer for frequency effects, I am typically not interested in doing anything else for intelligibility. Said differently, I am not interested in exploiting the nonlinearities in real speaker systems (other than possibly dynamic compression), so I should be able to fix any linear defects (bad frequency response) with a digital equalizer. The nonlinearities in real speaker systems are, for HiFi listening purposes [2], defects.

Also, I'm extremely skeptical of products marketed towards "audiophiles" because there's so much ~~marketing bullshit~~ pseudoscience surrounding the field that all the textbooks that cover loudspeaker design and HiFi audio electronics have paragraphs warning about it as the first thing.

Like I experience the difference between different pairs of binoculars and speakers dramatically, and graphical analysis backs up the differences, so how could they sound/look negligibly different to others?

Next time you do a graphical analysis, check out the magnitudes of the differences in your graphs versus the magnitude of the Just Noticeable Difference in amplitude or frequency. We probably do experience the differences between speakers differently than others. We're outliers.

What's your take on both major and, at the high end, diminishing returns on higher quality sensory experiences?

For personal listening, the point of diminishing returns is basically $20 because I can't afford shit. For listening to something I plan on sharing with others, I'd be willing to put in whatever I can afford. But frankly, I'd be just as likely to straight-up do the math and design my systems myself because I 100% don't trust any """high fidelity""" system that doesn't come with a datasheet and frequency response.

Lastly, I do wear glasses. I typically get my glasses online because, once you have the prescription and your facial measurements, it is the same quality as the stuff you get at the big-box stores.

[1] I acknowledge that Bluetooth sucks, particularly for audio.

[2] As a metal guitarist, I'm not against speaker nonlinearity for guitar speakers, but then again, guitar speakers are really convincingly simulated by impulse responses, which are a core linear systems concept, implying that they are nearly linear devices even at the volumes they are typically played at.

[โ€“] GuyFi@lemmy.sdf.org 2 points 3 months ago (1 children)

I'm a computing/music student thinking of getting into plugin design and similar stuff, I'm curious what plugin design is like and what it consists of. Is there any like quick tips or bits of information you could give me about the field?

[โ€“] PM_ME_VINTAGE_30S@lemmy.sdf.org 6 points 3 months ago* (last edited 3 months ago) (1 children)

It's so hard! Heavily distorted moving image of laughing/crying emoji

It's really hard! But it's really rewarding too. And as a computing/music student [1], you're in a great major to start!

First off, if you just want to make your own effects and you're not really interested in distributing them or making them public, I recommend using JSFX. It's way easier. You can read through the entire spec in a night. JSFX support is built into REAPER, and apparently YSFX allows you to load JSFX code into other DAWs, although I haven't tested it. JSFX plugins are compiled on the fly (unlike VST plugins, which are compiled ahead of time and distributed as DLLs), so you just write them up as text files.

However, their capabilities are limited compared to VST, AU, LV2, AAX [2], and other similar plugin formats. Also, pre-compiled plugins perform better. That's why plugins are released as such.

So if you plan on writing pre-compiled plugins for public consumption, you'll need to do some C++ programming.

IMO the most important thing to learn for plugin design is how to code well, particularly in C++ with Git and JUCE.

If you learn how to code with good practices, you can compensate for all other deficiencies.

Between "music", "engineering", and "software development", plugin design feels the most like "software development".

99.9% of all plugins are written in C++, and most of those are done (both proprietary and FOSS) with the JUCE library. School taught me the basics of C++ but they don't teach you how to code well. Particularly, your DSP code needs to meet a soft real-time constraint. You have to use multithreading because you have a thread for the audio signal (which must NEVER get interrupted) and at least one thread for the GUI.

You also need to figure out which parts of the C++ standard library are real-time safe, and which aren't. Here's a good talk on that.

If you use JUCE or a similar development library then they have well-tested basic DSP functions, meaning you can get by without doing all the math from scratch.

Start watching Audio Developer Conference talks like TV as they come out. JUCE has a tutorial, and MatKat released a video tutorial guiding the viewer through coding a simple EQ plugin [3]. JUCE plugins are basically cross platform, and can typically be compiled as VSTs on Windows, AU plugins on Mac, and LV2 plugins on Linux.

JUCE is a really complicated library even though it vastly simplifies the process (because audio plugin development is inherently hard!). You're going to have to learn to read a LOT of documentation and code.

I also recommend learning as much math as you can stomach. Start with linear algebra, calculus, Fourier analysis, circuit theory, and numerical analysis (especially Padรฉ approximants), in that order. Eventually, you'll want to roll your own math, or at least do something that JUCE doesn't provide out the box. Julius O Smith has some really good free online books on filters, Fourier Analysis, and DSP with a music focus.

If you're willing to ~~sail the high seas to LibGen~~ buy a book, I recommend Digital Audio Signal Processing by Udo Zolzer for "generic" audio signal processing, and DAFX: Digital Audio Effects by Zolzer for coverage of nonlinear effects, which are typically absent from DSP engineering books. I also recommend keeping a copy of Digital Signal Processing by Proakis and Manolakis on hand because of its detailed coverage of DSP fundamentals, particularly the coverage of filter structures, numerical errors, multirate signal processing, and the Z transform.

A little bit of knowledge about machine learning and optimization is good too, because sometimes you need to solve an optimization problem to synthesize a filter, or possibly in a fixed time as your actual output (example: pitch shifting). Deep learning is yielding some seriously magical effects, so I do recommend you learn it at your own pace.

DSP basically requires all the math ever, especially the kind of DSP that we want to do as musicians, so the more you have the better you'll be.

[1] IMO that would have been the perfect major for me, that or acoustical engineering, if anything like that existed in my area when I went to recording school 10 years ago. While my recording degree taught me some really valuable stuff, I kinda wish that they pushed us harder into programming, computing, and electronics.

[2] AAX requires you to pay Avid to develop. So I never use AAX plugins, and I have no intention of supporting the format once I start releasing plugins for public consumption, despite its other technical merits.

[3] Over half of MatKat's tutorial is dedicated towards GUI design, i.e. the audio part is basically done but the interface looks boring and default. GUI design and how your GUI (editor component) interacts with the audio processor component are extremely important and time-consuming parts of plugin design. Frankly, GUI design has been by far the most complicated thing to "pick up", and it's why I haven't released anything yet.

[โ€“] GuyFi@lemmy.sdf.org 3 points 3 months ago

Dude I can't thank you enough this is incredible! Seriously this is really useful :)