There is no point to distributing music in 24-bit/192kHz format. • Hacker News Board

Re: There is no point to distributing music in 24-bit/192kHz format.

#301

Post by eatmyshorts » Tue, Mar 06, 2012, 3:16 PM UTC

dedward wrote: post #204

eatmyshorts wrote: post #136

Earlier quoted context omitted.

Yes, but if you sample the frequency to create a step wave, then neglect to filter the results, you will end up reproducing tons of high frequencies. That is why we need to filter the output for signals >20KHz...to remove these harmonics that result from reproducing the square wave. Of course, filters aren't perfect, and result in phase shift and roll-off. So we over-sample the signal to create a signal with a much h…

Yup... people need to get very clear in their heads the difference between the recording/sampling/mixing/mastering stages, where high bitrate/width/gear/knowledge is helpful, and playback, which is a completely different thing. (not for eatmyshorts -you get this I gather) - everyone gets that "upsampling" can't add detail to a recording right? You can't get more than you've got.... no matter what you do. There is no…

You got it 100% correct. You upsample simply to move the frequency of the analog filter higher, with a gentle rolloff (and ideally a 1st order filter, so you introduce no phase effects) to get your final signal.

Re: There is no point to distributing music in 24-bit/192kHz format.

#302

Post by eatmyshorts » Tue, Mar 06, 2012, 3:27 PM UTC

Anechoic wrote: post #189

eatmyshorts wrote: post #165

Earlier quoted context omitted.

Yes, but the effects of interference patterns between multiple ultrasonic frequencies is the same, and definitely does affect the audible spectrum. This is why we must filter the square wave that comes out of a DAC. And the limitations of filters (phase shifts and roll-off) are why modern CD players oversample the signal--so that the filtering can be performed well beyond the audible spectrum.

This: Yes, but the effects of interference patterns between multiple ultrasonic frequencies is the same, and definitely does affect the audible spectrum has nothing to do with this: This is why we must filter the square wave that comes out of a DAC The only reason that square waves "must" be filtered is to reduce the potential of damaging tweeters. If you want to record a square wave with the purpose of later reprodu…

OK, if you say so. I think you're misunderstanding a fundamental concept of digital to analog converters. But if you think it's just to prevent blowing your speakers, that's OK.

The reason that square wave sucks is because it introduces tons of high frequency content (your amp probably won't reproduce the high frequency content anyway, so I don't think most Japanese consumer amps will damage your speakers--that is, the amp will act like a filter anyway). That high frequency content then creates alias effects (think of moire patterns when looking at super high-res photos that are scaled down without anti-aliasing). Those alias effects sound like shit to the human ear.

The point of filtering is to anti-alias the resulting analog signal after conversion from digital to analog. The point of upsampling is to move that filter well beyond the audible range, so you can use a 1st-order filter (gentle slope, but it introduces no phase effects). The fact that a square wave hurts your speakers is inconsequential--the amp will effectively filter the signal anyway. Unfortunately, it will filter the signal without anti-aliasing, which introduces those nasty interference patterns within the audible spectrum (that is, if you feed a straight 44.1KHz sampled square wave to your speakers without upsampling/filtering).

Re: There is no point to distributing music in 24-bit/192kHz format.

#303

Post by goblin89 » Tue, Mar 06, 2012, 3:55 PM UTC

cbr wrote: post #299

goblin89 wrote: post #288

Earlier quoted context omitted.

Well, actually it was that long cymbal sounds “fade” quicker and just sound different with lossy music. IIRC I distinguished an mp3 encoded by iTunes with bit rate 192 or 256 kbps from its original in Apple Lossless (both played on same cheap acoustic system). I probably should test with AAC or Ogg, too. Although I have a feeling that it's pretty much impossible to keep intact those rich in high frequencies cymbals w…

"I distinguished" Was this in a blind test?

Please refer to my comment above[0]. Yes, it was a blind test. The person helping me might've looked up bit rates, so it was not a double-blind experiment, but I could not (nor did I want to) see what's being played, and relied only on hearing.

[0] http://news.ycombinator.com/item?id=3669893

Re: There is no point to distributing music in 24-bit/192kHz format.

#304

Post by nullc » Tue, Mar 06, 2012, 3:57 PM UTC

ars wrote: post #28

wickedchicken wrote: post #14

For an article containing a lot of "well, if you knew signal processing..." there are two fairly major oversights: 1) Any well-designed system is going to have headroom. Period. Just because 48kHz can capture the frequencies the human hear theoretically, it's always good to have a little wiggle room. This comes into play even more with interactive situations: humans are particularly sensitive to jitter. Having an "ov…

0.03ms is 33kHz - you can't, no matter how much you want to, make a granular timing that is faster than at least one cycle of the frequency you are using. 0.005ms is 200kHz BTW.

This isn't true. Sample a bandlimited impulse. The exact timing is encoded into the gibbs oscillations of the signal. So long as you have a high enough SNR you can have timing as precise as you want. (and because the ear doesn't work with ultrasonics— it is itself bandlimited— it uses the same phenomena for timing)

Re: There is no point to distributing music in 24-bit/192kHz format.

#305

Post by tcarnell » Tue, Mar 06, 2012, 4:06 PM UTC

Has anyone had a look at their hi-fi amp recently? If probably probably doesn't handle much more than 80 kHz and your speakers probably dont respond to anything over 20 kHz. So yes, 192 kHz is pointless UNLESS you intend using it for studio quality editing/mixing - and I'm sure Steve Jobs would not have encouraged this!

Re: There is no point to distributing music in 24-bit/192kHz format.

#306

Post by tcarnell » Tue, Mar 06, 2012, 4:07 PM UTC

b0rsuk wrote: post #290

sjwright wrote: post #78

I must say I get rather irritated when people spend time worrying about dubious 'tweaker' methods to improve their audio, when the most under-performing component of most people's sound equipment also has the lowest-hanging fruit: The room itself. When people ask me where they should spend money to improve the quality of their hi-fi or home theater system, in nearly every case my response will be something like "get…

We have a lot in common. It is said that in most cases 192 kbit .mp3 is indistinguishable from >192, and blind tests support that. Granted, there are instruments like castanets which make it easier to hear the difference. In general though, I can't distinguish 128 from 192 and I listen to music a lot . Also it's unlikely that my hearing is already damaged because I try to keep volume low. But I've noticed that where…

Is the 192 the bitrate or the frequency response? I thought 192 in MP3 was the bit rate, not the maximum frequency response...

Re: There is no point to distributing music in 24-bit/192kHz format.

#307

Post by nullc » Tue, Mar 06, 2012, 4:30 PM UTC

sirwitti wrote: post #132

aiscott wrote: post #45

Earlier quoted context omitted.

[edited: I was wrong]

i think what jaylevitt is referencing to is that there is interpolation going on in the dac. that could mean (i'm no dac expert, so not sure) that the dac could guess more granular than the sampling rate would allow the start points (of transient e.g.) but the question for me is how exact that guessing is. correct me if i'm wrong but, that interpolation happens twice: when recording by the adc and on playback by the…

This is the core secret of the sampling theorem. It says if you have signals of a particular type (bandlimited) you can do a certain kind of interpolation and recover the original exactly. This is no more surprising than the fact that you can recover the coefficients for an N degree polynomial using any N points on it, though the computation is easier.

It turns out that if you reproduce a digital signal using stair steps you get an infinite number of harmonics— but _all_ of them are above the nyquist frequency. The frequencies below the nyquist are undisturbed. Then you apply a lowpass filter to the signal to remove these harmonics— after all, we said at the start that the signal was bandlimited— you get the original back unmolested.

Because analog filters are kinda sucky (and because converters with high bit depth aren't very linear), modern ADCs and DACs are oversampling— they internally resample the signal to a few MHz and apply those reconstruction filters digitally with stupidly high precision. Then they only need a very simple analog filter to cope with their much higher frequency sampling.

Re: There is no point to distributing music in 24-bit/192kHz format.

#308

Post by Derbasti » Tue, Mar 06, 2012, 5:24 PM UTC

sjwright wrote: post #291

Derbasti wrote: post #237

Earlier quoted context omitted.

You will still need a aliasing filter that cuts off between, say, 18 and 22.5 kHz to avoid aliasing noise. That is one sharp filter no matter how you look at it. You can use a high quality, long, linear-phase FIR filter, but you can't cheat physics: sharp filters necesserily introduce distortion, and such a sharp filter so close to the hearing threshold does not go unnoticed.

I don't see how a sharp filter could be needed if the DAC is oversampling.

Obviously. Analogue audio does not have a sampling rate. The ADC however can oversample all it wants, but if the output is 44.1 kHz, it needs an aliasing filter that cuts off at 22.05 kHz.

Re: There is no point to distributing music in 24-bit/192kHz format.

#309

Post by libraryatnight » Tue, Mar 06, 2012, 6:04 PM UTC

sjwright wrote: post #78

I must say I get rather irritated when people spend time worrying about dubious 'tweaker' methods to improve their audio, when the most under-performing component of most people's sound equipment also has the lowest-hanging fruit: The room itself. When people ask me where they should spend money to improve the quality of their hi-fi or home theater system, in nearly every case my response will be something like "get…

Alan Parson's makes this point, too: http://boingboing.net/2012/02/10/alan-parsons-on-audiophiles...

Re: There is no point to distributing music in 24-bit/192kHz format.

#310

Post by jwatte » Tue, Mar 06, 2012, 6:42 PM UTC

eatmyshorts wrote: post #145

cmer wrote: post #4

There's a lot of scientific-sounded content in this, but unfortunately most of it couldn't be further from the truth. I'm an ex-audio engineer and studied digital and analog audio engineering; this has been debated to death over the last 15 years. Digitally recording a triangle is the best example of why 48kHz is very limiting. The distinct sound of the triangle constitutes of a high fundamental frequency, ballpark 5…

Yes, but our ears only hear 20Hz-20KHz. So, according to Nyquist theory, you can recreate the entire signal that the human ear hears by recording those sonic artifacts that result from interference between supersonic harmonics. So while it's true that the human ear can't hear well above ~18KHz, and the interference between high order harmonics are audible, it's also true that a properly recorded signal, sampled at 44…

Nyquist is true for static signals. Music is not static. Brick wall at 20 kHz and get audible phasing artifacts! (Even if your filter is phase linear)