Hacker News Reader

Comment by andrewzah | Hacker News Reader

This is why "audiophile" is a joke term in the audio engineering community.

It is not biologically possible for us (unless one is an alien) to hear above 22kHz. 44.1kHz is enough to record in perfect fidelity [0]. For strictly listening purposes, 192kHz/24bit is wasteful, just extreme overkill because "bigger numbers = more good".

People can't even reliably detect a difference between 320kbps MP3s and source-quality FLAC/ALAC/WAV/PCM. On very good headphone and speaker setups. Good quality MP3/AAC is all one needs for airpods and the bluetooth protocol can easily handle those bitrates.

[0]: https://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampli...

edit: changed record->playback. Here I'm just discussing playback; for audio production purposes 192kHz/24bit is desireable for several reasons. Once one ships that album/song though, it should be downmixed back to 44.1kHz/16bit.

View on HN

It's not necessarily just that we can't hear above 20khz - recording above 44.1khz has its merits. For example, if you want to apply digital distortion or do amplitude modulation then up-sampling or having a high SR stops wrap around in the spectrum and getting all those nasty artefacts. The end user should never need to worry about this and it should be downsampled at the master.

You forgot the #1 reason for recording at high SR; using a lower order low-pass filter without affecting the audible range. Sharp digital low-pass filters have far fewer tradeoffs and better SNR than analog filters, so doing the anti-aliasing low-pass filter digitally has big advantages.

The interesting part is in how actually valuable improvements are rare, while useless "improvements" are common.

I've got all music in 44.1kHz or 48kHz FLAC (only on the server, so I can transcode it to ogg opus for mobile playback reducing the space usage without lossy to lossy artifacts). Similar effects apply to many other such cases.

Audiophiles buy 10'000$ golden HDMI cables... which don't even support HDMI 2.0a. They buy gold-plated toslink cables (!)

Is there something that does automatic transcoding? Though I can hardly give up the discovery aspect of Spotify these days...

Transcoding is a fairly common feature, although typically to mp3. I use Navidrome [0] to self-host my own music, and it lets me set up custom transcoding profiles with ffmpeg.

[0]: https://github.com/navidrome/navidrome

That's what I was looking for, thanks!

KDE's Amarok (semi-dead project) can automatically transcode audio files if you copy them to e.g. a phone, Jellyfin (an Emby fork, which is a Plex clone) can automatically transcode files on the fly during playback.

I've simply written a small python utility to handle everything automatically.

gold plated hdmi cables add a parity bit.

Once I learned the Nyquist frequency rule existed in my EE classes I became incredibly suspicious of anybody who threw the words "lossless" around. Glad to know that this suspicion was well-founded.

Aside, "Nyquist criterion" would be a sick band name.

> …I became incredibly suspicious of anybody who threw the words "lossless" around. Glad to know that this suspicion was well-founded.

What's suspicious about it? Do you take this to mean something other than lossless compression/encoding?

I took it as either:

- not particularly a huge upgrade in audio quality, or

- an attempt to upsell me on some audio-related product

The audio myths workshop, old but still gold: https://youtu.be/BYTlN6wjcvQ

While 192KHz is overkill, there are plenty of instruments that produce non-trivial amounts of energy above 22KHz. While we're unable to hear them some can be generally felt (depending on volume and proximity). A pair of headphones won't reproduce those ultrasonic energies but good speakers can. Even so, this is only germane in a handful of types of music with high fidelity recording and mastering.

The same for higher resolution samples, an audio pipeline using that available dynamic range can do cool stuff.

Both are pointless for 99% of music sold though.

Debunked https://web.archive.org/web/20200310055211/https://people.xi...

This is somewhat true. Also, while human beings cannot hear a dog whistle per se, due to the way the human ear works they can hear the beats produced by the combination of that sound with another, straightforwardly audible-frequency sound. Per Nørgård exploits this in his Symphony No. 5. But examples of this actually affecting enjoyment or even an A/B test in practice are so few they shouldn't really guide one's encoding choices, CD audio is already sufficiently wide in practice and claims that we need more are audiophile wishful thinking.

I'm definitely not arguing for the stupid "audiophile" frequencies. For the tiny fraction of music taking advantage of ultrasonic frequencies and high dynamic range it's nice there's an option now.

I hope the availability of high resolution, or more importantly the spacial audio, will inspire some artists to really explore the space. I'm hoping for some stuff like the Flaming Lips' Zaireeka or Yoshimi album's 5.1 mix.

But does sampling above the Nyquist rate guarantee phase reconstruction though? In a real world (not theoretical ideal) implementation of a reconstruction filter, what’s the phase reproduction ability the nearer you get to the Nyquist rate? I thought that there was no phase reconstruction guarantee by the Shannon reconstruction theorem, only frequency, and that there can be advantages to sampling rates higher than than Nyquist alone would suggest. Can anyone with more authority on this subject speak to this? It’s a bit fuzzy but I remember my DSP professor making that argument once and it always kind of stuck with me. It does make some intuitive sense, as you picture trying to reconstruct a sinusoid close to (but below) the Nyquist rate. It’s not hard to see how a reconstruction filter may get some odd results like incorrect amplitude, oscillating amplitude, or phase shift the closer you get.

Reply on news.ycombinator.com