Are 24bit/192KHz music files really better than the CD standard?
I am clearly not a DSP guy so I believe almost anything concerning such.
Having now DACs that can handle 24bit/192KHz, I thought of updating a bit of my music collection to 24bit/192KHz level. Looking for a suitable site to buy music in that quality, I noticed this article: 24/192 Music Downloads are Very Silly Indeed . The article concludes in large that 24bit/192KHz music serves no purpose. The article appears very serious and the argumentation is based on sampling according to the Nyquist-Shannon Theorem.
If I had to describe a time-varying signal through sampling, my intuition would urge me to use as many equidistant samplings as possible with as fine a resolution as possible. Without doing any calculations, I would be able to reproduce the original signal quite closely by just repeating the sampled values. The article states that I am not the only person being foolish (luckily there are co-foolish so I do not feel singled out) and that I am wrong.
The way I understand the article, sampling according to the Theorem should ensure full information about the signal up to the limit (half the sampling rate) given by the Theorem, eventually through Fourier calculations.
From search on this forum I am left with the impression that 24bit/192KHz music is a clear improvement. Perhaps one improvement is that Fourier calculations become obsolete, at least in part.
My questions are, do I have a benefit from 24bit/192KHz music compared to my present CDs?
If not, are the many disclosures about new high quality music standards really just a commercial scam? Is a difference the amount of calculations I need to do on samplings according to the Theorem in order to reproduce the signal?
Thanks for any reply. Sorry if I have overlooked existing replies to my questions.
Why not do a listening test for yourself? The 2L label offered a variety of tracks at different bit depths and sample rates of tracks all from the same master. I think nativeDSD did something similar as well. You could then A/B these using foobar or jriver.
Just my understanding and observations, I'm not an expert in the field by any means...
From what I understand the main improvement from using high sample rates is that the recording of higher frequency content is possible.
Due to the above, filtering can happen well outside the audible range (instead of very close to it, affecting some of the audible frequencies). Some kinds of filtering can cause distortion artifacts back into the audible range as well. These issues are mitigated with a higher than needed sample rate.
The other concepts center around our ability to process phase information beyond the frequency to which we can hear, also that we perhaps sense in some other way high frequency content which we cannot, strictly speaking, "hear".
More bits per sample give a higher possible dynamic range. But it seems like the 106dB possible with a 16 bit sample is already overkill for musical content.
Add to this - many "high res" recordings were not recorded or mastered for high res formats. So whatever you might gain in fidelity as already been lost.
Do you have a cat who likes music? If so, 192 kHz is definitely better than 44.1 kHz.
If not, and no other pets either, then it all depends on two things:
1. Whether ultrasonic signals are really always inaudible, also when heard in combination with other signals.
2. Whether the recordings are peak sample normalized and whether your playback chain can handle it when they are.
1) The only experiment I know of that indicates that music may sound better with ultrasonics included is a 1990's Japanese experiment involving EEG equipment. Japanese gamelan players listening to Balinese gamelan recordings had different brain waves when a super tweeter reproducing signals above 26 kHz was turned on than when it was off, even though they could not consciously hear the sound of the super tweeter on its own. From the description of the experiment, it wasn't clear to me whether it was double- or only single-blind, single-blind tests are known to be rather unreliable.
2) Many if not most DACs and digital signal processing chips clip on peak sample normalized music, the Benchmark Media site has some good explanations why. This effect gets less severe with increasing sample rate, but you can solve it completely by digitally reducing the volume before any type of filtering is applied to the signal.
It depends on which era of music you listen to.
If we look at the history of recording, PCM digital first displaced analogue tape around 1970.
The recordings in the early 70s were made at 13 bit resolution, increasing to 14 bits by the end of the decade - and no amount of fiddling with bit rates today will restore the missing information.
The seventies was too early. Most studios didnt buy digital recorders till the early eighties and those were already mostly 48k/16bit.
As already mentioned, it all depends on the masters. Getting 48k copies of 48k masters might be worth it, or hi-res transfers from analogue, though 40 year old tapes have there own problems. The other worry, are they remastered, as in over compressed? My guess is most are not worth it, they may even be worse than the CD.
The first 16-bit recordings were released in the latter half of the 70s.
As I understand it, higher sampling rates are nothing to do with producing very high frequencies for cats (or anyone/anything else) to hear. The Nyquist theorem is sound and can reproduce the human hearing range at the CD sampling frequency of 44.1hKz just fine, but that's not what higher sampling rates are all about.
As ubergeeknz says, the difference it makes is in the filtering in the analog conversion, when the sampling frequency needs to be filtered out. If the file is sampled at 44.1kHz, that's only a little over an octave above 20khz, so the filtering needs to be very steep. But very steep and very clean filters don't exist, so the filter produces distortion that reaches down into the audible range.
With a higher sampling rate, the filter can be less steep - and the higher the sampling rate, the less steep the filter. And the less steep the filter, the less distortion there is extending down into the audible range.
It's the same reason we have upsampling DACs, which resample a 44.1/48kHz source in the digital domain before analog conversion - upsample, then use a gentler filter that produces less distortion.
I'm using a DacMagic Plus upsampling DAC which upsamples to 384kHz before analog conversion. It's a relatively modestly priced DAC, but I can hear a definite improvement over a couple of non-upsampling DACs I've compared it to.
Now, it might be simply that it's a better DAC than the comparisons, but the DacMagic Plus has three different filter settings to choose from, and they definely do sound different - I'm not sure which I prefer, but to my ears two of them are defintely better than the third.
Whether there's any difference between a source sampled at 192kHz and a 44.1kHz source upsampled, I don't know - but I am sceptical.
As for 24 bit samples, from what I understand, the maximum dynamic range that's practically possible extends to only 19 or 20 bits, with any greater resolution being below the thermal noise floor. So 24-bit should be better than 16-bit, but not to the full 24 bits. Whether it really is better? I've only heard a few 24-bit tracks, and I can't hear a difference on my system (which is not high end, so I can't say more than that).
Oscroft's response reminds me that I forgot a third issue:
3. Whether your DAC uses linear-phase interpolation filters with too much passband ripple
3) If the DAC uses a linear-phase interpolation filter with too much passband ripple, this will cause a pre-echo. Unlike the ultrasonic sin(x)/x-shaped pre-ringing that all linear-phase interpolation filters have, this pre-echo is in the audible frequency range. The higher the original sample rate and the less steep the filter, the shorter the time between the pre-echo and the main signal and the better it gets masked.
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