Standard CDs might not be good enough for archival and further production work (where you also need wiggle room for further processing or format changes), but they have plenty of headroom to reproduce any kind of sound for playback purposes. Any more is just placebo.
With 96khz sampling rate, you don't need as a sharp drop off in your filter.
The above is mostly accurate, hopefully someone else will comment and tell me how I'm wrong and correct the inaccuracies.
Edit: to anyone reading this, read the replies if you want the correct explanation. To everyone that replied, thank you :).
A sharply cutting off analog filter is expensive to produce. It has multiple stages to create the multiple poles. High precision resistors and capacitors have to be used to get all those circuit stages to line up. The filter will have phase distortion.
That's the basis of "supersampling": sampling at a higher rate with a simpler filter with less of a cutoff, then completing the job with a digital filter to get to the target sample rate.
This can be done in reverse, in reproducion. Take, say, a 48 KHz signal, and digitally interpolate it to a higher sample rate like 96 KHz. That is fed to the DAC. The filter after the DAC then doesn't need such a steep cutoff after 20 KHz. A greater bit depth can be used; like 16 bit samples interpolated to 24 bit at a higher rate, fed to a 24 bit DAC.
The digital filter or interpolator doesn't care about accurate resistors, capacitors or drift in component values over time or due to heat; it does the same thing with the same data every time.
Digital filters can look at future values also. The state of an analog filter is determined by only the current and past values of the signal; but digital signal processing can delay the signal a little bit and look at a "box" around the current value. I think that is key to preserving phase relationships.
This is incorrect. See my comment above/below.
Its vwry “spherical cow”. It assumes all your filters, DACs, are perfect, and that time is infinite.
And you can use oversampling to correct for the filtering issues. That's pretty standard in modern DACs and ADCs.
Yes, your filters, DAC and speaker wire could still create a less than ideal listening situation, but the fundamental aspect of the Nyquist frequency is not concerned with any of that.
Monty (of xiph) has an article on the subject (well on 24/192 but it applies all the same) which goes into the gory details: https://people.xiph.org/~xiphmont/demo/neil-young.html
There's a space in the audio world for more bits of data. But the final output format isn't where it belongs.
I believe the comment you're responding to is talking about the analog filter that is needed to avoid aliasing -- as the first words of your comment correctly note/explain.
And in particular, the original comment seems to be noting the phase distortion (in frequencies near the cutoff) that analog brick-wall filters will cause. This has been a design contention for decades, really, ever since the CD format was introduced.
It's a big design space, with options for gentler analog filters, followed by very fast digital sampling, and further tricks with filtering in the digital domain, where you don't have to worry about getting great capacitors, etc.
It may be out-of-scope to lay all that out in one paragraph!
You're right that its a big design space. The key takeaway is that "yes, higher sample rates can actually make a difference, but almost entirely down to the filter design, not because Nyquist moves ... and you probably cannot hear the difference."
If you're sampling at 48 kHz, the signal has to be severely attenuated already; it has to go from 20 kHz to deep cutoff in just the space of a few kHz.
At 96 kHz can achieve the effect as if you were sampling at 48 kHz, with a steep filter. You sample at 96 kHz with a milder filter, and then purely in the digital realm, you down-sample to 48 kHz. There is an overall filter consisting of the original analog one plus the digital processing.
That is cheaper and more reliable than doing it all in analog.
An analog filter with a steep cut off will be challenging in mass production because of the strict component tolerances.
Sure, you could use a steep filter with 96 kHz also. Say, a steep filter that starts cutting off at 30 kHz. It would still be a less demanding filtering application because of the margin that you have in the frequency domain. The multiple poles of the filter don't have to be lined up as well. E.g. if the first pole starts rolling off at around 30 kHz, and then next ones at 31, and the third one at 28, ... it doesn't matter because you're still hitting the absolute target of there being next to nothing at 48 kHz, and nearly the full signal at 20 kHz.
The content can be faked, and this is used in low-bandwidth codecs and such.
This is a very simplistic view of things, on many fronts, but I'll try to be as concise as possible.
In a classical wind/bow orchestra, there are more harmonics at play than the 20Hz-20KHz band, and these harmonics affects our perception of the sound and soundstage. However, recreation of this is very hard, because you need both recording and playback chains which can handle these harmonics as well.
To faithfully reproduce such orchestra, you need a speaker for every instrument, ideally with the exact air movement capacity of the instrument you're mirroring. This is not practical. Instead we mic all of them, mix all of them, and add room echo to the mix to capture harmonics as best as we can.
However, this can't replicate some instruments anywhere around its real sound.
Examples are Turkish qanun [0] and Chinese guzheng [1]. These instruments sound bland, flat and shallow on all recordings, but listening them live, directly with your ears is a goose pimlples inducing experience. Higher end stringed instruments have a similar vibe to them.
So, a 16bit/44KHz signal at 20Hz-20KHz band cannot reproduce these instruments with any faith.
Standard CDs, high quality vinyls, lossless files carry a lot of information, but not all information we can process with our ears. Like how even the best digital cameras cannot reproduce the colors we can see with our eyes.
Reaching these resolution levels are neither cheap, nor practical, hence we use what's practical.
As a result, your master-pressed-vinyl possibly can't carry this information either, because the recording chain was not able to capture that amount of information, even if you went all the way to install your own power pole to feed clean power to your impossibly expensive audio equipment with all that capacity to reproduce that sound.
IOW, you can't extract the sound which is not there to begin with.
Your friendly ex-orchestra player reported from its AKAI-AM2850.
It has never been clear to me what the goal of home playback is supposed to be.
Let's simplify from a full orchestra to just a solo piano. Is the goal
1. to sound like it would if that piano was being played in my living room, or
2. to sound like what I'd have heard if I was there in the concert hall sitting in a good seat when the recording was made, or
3. to sound like what it would sound like if a replica of my living room was built inside the concert hall but with walls that do not transmit sound, with my speakers replaced by speaker-sized holes in the wall behind where the speakers normally sit, and I was sitting in that replica living room during the concert?
ABBA’s sound is mastered to fit into AM radio and jukeboxes for example. To make the music broadly listenable, for example.
Today everything is so blurry, because there’s no studio per se. It’s just sequenced, vocal is added, mastered, compressed and released (today’s pop). Rock and other stuff is still track recorded, tho.
At a certain point the speakers really start to interfere with each other and you have to worry about the stability of the image at each point in the room...
It occurred to me at that point that you're really starting to approach 3. You're trying to recreate the sound pressure patterns within the room.
That's about the point I moved on to something simpler. Granted my explorations were super simple, but it was a VERY weird thought that got me thinking about the question you've asked.
It's enough to reproduce well above what we can hear by the normal mechanism that we use for most of our hearing (sound vibrates the eardrum, which vibrates some tiny bones in the middle ear, which cause fluid in the cochlea to ripple which disturb sensory hairs which we perceive as sound).
Ultrasonic sounds can be conducted to the cochlea via the surrounding bone bypassing the eardrum and middle ear. Those bone conducted ultrasonics can be perceived. See [1].
This is probably not important in music except in maybe a few rare cases.
I'd still like to see better frequency handling though, so that our music wouldn't sound terrible to the dogs and cats that are forced to listen with their owners. We can't normally hear much past 20 KHz (more like 12 KHz for many of us), but they can.
There are several musical instruments that have significant ultrasonic output. It would be an interesting experiment to see if dogs and cats like those instruments live better than they do recorded (or dislike live less than they dislike recorded). Then compare to the same experiment but with instruments that don't have ultrasonic output.
(And also much of the digital gear in the 80s when the first wave of CD releases was pretty crappy)
Sorry, but CD (16bit PCM) has better dynamic range than vinyl. The best vinyl systems can have dynamic range of around 80dB. 16bit PCM is 96dB.
Still, you can always take a great analog recording, pipe it into a good ADC, and listen to that sample forever without degradation. I think almost everything can play FLACs now.
Eventually got the better mastered version which was leaked from guitar hero or something.
Yes, but these mastering choices could apply to the vinyl version too. In other words, your response had nothing to do with a comparison of the two media.
They can't, because these choices create enough volume for the needle to go haywire, so mastering always has a better dynamic range.
When you say 'mastering always has a better dynamic range', did you mean CD quality PCM has a better dynamic range?
Without RIAA equalization, it's not possible to keep the needle in the groove with that amount of bass and low end. You can't keep the needle on the track with stonewalling even with RIAA equalization.
As a result, you are limited by the medium itself, hence you have to make milder choices.
Perhaps, but what about optional mastering choices like brick-wall limiting with normalization (which is what I think you meant)? Maybe I'm misunderstanding your point.
A good lossless album, regardless of the medium played through shows a significant difference in soundstage given the system can handle the resolution thrown at it when compared with a MP3/M4A file.
It "upsets a lot of people" in the same sense and for the same reason as telling them drinking alkaline water cures cancer.