I'm still mad that they high-quality voice audio possible through ISDN didn't become ubiquitous. It's ridiculous to hear the clipped frequency range of a plain old telephone line during radio interviews in 2025.
I'm even more mad that sound quality in radio interviews actually decreased. People used to use landlines for calling, given that very recognizable sound "quality" (which is of course nowhere close to what ISDN could do). Now as (almost) everybody uses cell phones, quality is sometimes good, sometimes very poor depending on network conditions. It's not rare to hear the consequences of dropped packets. Sometimes it also sounds bad with apparently no drops, not sure why: maybe are people using the phone in "hands-free" mode so they don't need to keep it in their hands?
I think it was ubiquitous. NPR in the 90s and 2000s used ISDN to allow many of their commentators to work from home. I think where you hear those crazy 8KHz clipped calls is where it's not an option. Mostly these days remote radio and podcast interviews I hear remote participants sound more like they're on mobile phones: variable voice quality with an almost unbearable latency.
They really do sound alot better. It always reminds me of the first time I ever made a FaceTime call, in 2010, and the high quality audio was just as interesting as the video.
It is the reason why we switched to using FaceTime audio. The sound quality is so much better than over the normal voice line. I don’t know how to reliably get HD Voice.
If course it has to be pre-planned, someone needs to have the hardware with them. So sometimes there's a spontaneous connection over normal mobile phone. That's something that everyone has with them at all times.
24 [bit] * 192 [kHz] = 4.608 [Mbps]. Maybe not sensible to do so, but many people could have Discord calls in uncompressed "high-res" WAV. It's crazy that there aren't even 16bit/44.1k modes in most voice call apps.
That's an absurd bit rate for humans, especially talking about voice.
24-bit samples is ridiculous overkill. That's a huge dynamic range that's completely unnecessary.
At 192KHz you'd be able to capture 96KHz signals, far, FAR outside the range of human hearing. Human hearing peaks at 22KHz so you only need a sample rate of 44KHz to capture the total range of human hearing.
For human voice you don't really need better than 16-bit samples at 12KHz or so. That's for great quality voice.
The only reason audio mastering is done at huge sample sizes and sampling frequencies is to prevent aliasing during mixing and to preserve higher frequency harmonics. There's absolutely no need for such rates delivering to human beings.
Also higher fidelity audio sampling is available for phone calls. The issue is more political than technical. Cellular carriers don't like to negotiate higher quality calls between one another so inter-carrier calls tend to fall back on the lowest common denominator AMR-NB codec. Intra-carrier calls don't even reliably pick AMR-WB let alone EVS available with VoLTE.
There is zero point in sampling higher than 48khz. That captures all frequencies up to 20khz or so. 192khz is just a waste of bandwidth for no actual gain.
The economics probably aren't that great to send uncompressed voice into the data center. If you have a business that gets charged XX cents (or more likely .00XX cents) per GB of traffic and you can cut that in half by using compression... I think people will opt to use compression.
nyquist-shannon means that you only need to sample at 192khz if you need to encode signals up to 96khz.
humans can’t hear above 20khz. adult humans can’t hear above 16khz or so, we lose the top end before age 20. this means that the standard 48khz sampling rate covers the entire human hearing range and then some (0-24khz). any sampling rate over 48khz for sound intended for human hearing is a total waste.
Why do 96 or 128khz sampled audio files sound better than 48khz ones? I blind tested and could always tell the difference between them, but not between 128 and 192
Typically, high sampling rate files are part of a different mastering process than what is published as a 44.1kHz cd audio or 48kHz dvd audio.
Also, you might possibly be sensitive to resampling artifacts if your output device runs at 44.1kHz and your file is 48kHz or vice versa.
Audio testing is hard, and testing on yourself is tricky... But if you have a sample that you're convinced sounds better at high rates than lower rates, I would urge you to put it through a tool to resample it down to lower rates and see if/when you can tell the difference. If the rate isn't an even multiple, it's worth using a tool that can dither; dithered resampling artifacts are less abrasive than undithered... I had some voice recordings to play over the phone, and everything needed to be 8kHz u-law; the 48kHz original recordings sounded better than 44.1kHz original recordings because one is even multiple and the other isn't, but either way, the waveforms looked worse than it sounded.
> If the rate isn't an even multiple, it's worth using a tool that can dither; dithered resampling artifacts are less abrasive than undithered...
This seems to be mixing up two things; proper interpolation and dithering.
If you have limited bit depth (in practice, 16 bits or worse), you should pretty much always dither, ideally also noise shape. This is independent of the interpolation you're using; having a rational relationship between the original and downsampled signal makes some of the implementation a bit easier, but even for something like 48000 -> 24000, you'll end up with effectively a float signal that you need to convert to your chosen bit depth somehow, and that should be done better than just truncating/rounding.
And even for interpolating between two prime rates, or even variable-rate interpolation, you can and should get great interpolation (typically by picking out polyphase filtering coefficients from a windowed sinc of some sort).
Oh come on. I have handheld recorders that do 192khz.
"Headroom"
And the idea that humans can't hear over 20khz is like "humans taste 'sweet' on the tip of the tongue, and 'bitter' on the sides"
As we get older the hairs in out ears break or whatever and our perception decreases, but I could hear the fly backs in my old monitors, I used to be able to see the flicker in 3khz pwm LEDs, and my induction hob drives my kids crazy but it's merely midly annoying to me.
Get a real soundcard and some young people and play square(pwm) and sine tones starting at 16khz and find out where they can't hear it anymore. I find studio monitors with tweeters that are not paper are the best.
If you think you can hear ultrasound, it's nearly always due to nonlinearities in your system producing non-ultrasound when you try to play it. Seriously. (You can sometimes hear above-20 if it's very loud and/or you are pressing the source against your skull. Above-40 would be completely insane.)
The extra headroom can indeed be useful for some kinds of processing, but you can safely discard it for actual listening.
ime 48/96/128kHz 16/24bit through a modern DAC and well warmed Class AB amp and barely more than okay headphones can be told apart in a double blind test
but you do need phile-enough gears(minus the gilded pebbles hot glued onto circuit breakers)
sampling theorem only applies to sine waves. the rate is bit like the order of (fourier)series expansion and so approximations deviate as rate reduces. how many orders is enough depends and is situational
I have no problems with Opus or mp3 at low bitrate, in the same way I have no problems with microwaved food. I just think it's crazy that no one is doing 4Mbps audio while we're routinely streaming 20Mbps video as cheap distractions.
4Mbps audio doesn't really make sense when you can just about perfectly replicate any human-audible sound in ~1.4Mbps
EDIT: I do agree that lossless (or at least high bitrate modern lossy, like 256k Opus which is basically transparent) should be available in many more situations though.
For some context, wikipedia has a good table and diagram of codecs that do well at low bitrates, many of which are very old (G.722 from the late 1988, Speex from 2003, Opus from 2012 for a few examples)
I had ISDN for around $75/mo in '94 or '95, flat rate for "local" calls. It was fantastic! But then the phone company switched it to a metered service (in Omaha Nebraska, USA) and the price would have been over $300/mo after that change. I was grandfathered in, as long as I didn't move.
I was talking to a sales rep, at the time I worked for USWest or QWest, whatever they were called at that time, which may have helped, and the sales rep told me "we are being told to actively discourage people from buying ISDN".
I get the impression that the phone company hated consumer modem use of any kind, because it tied up CO equipment 24x7, and they liked the returns on investment they got with people paying $25/mo for resources that were used an hour or less a day, sometimes with extra revenue from long distance calling. And ISDN was just another representation of that.
I think it requires SDP negotiation between your UE, your EPC, their EPC, and their UE. Assuming all parties agree on the use of AMR-WB, then you're good-to-go.
