Any use of going beyond 44100?

[snooky]

anyway, if you want to be an engineer, then care, that is your job. However dont think that just because you got a 96 khz card that you are making better sounding recordings than someone who "only" uses 44.1 khz...

Well, that depends what you mean with better "sounding"

If you're talking musically then no, it makes no dif.
But if you're talking technically, then a recording made at a higher samperate and bitrate than the same one using lower quality will of cource sound better...Or?

So maybe we are both drawing the same conclusion here- better is subjective, possibly in the ear of the beholder?

Yeah, of course, but better qualit is always better quality, no?

no matter if you can hear/appriciate it or not...

[Armadillo]

.........and while the geeks are arguing for why 96k is better, us 44.1k boys are writing yet another song

I've done 2 within the last 2 weeks, how about you?
But of course since mine are only at 44.1k they are of course not great songs.

[monkeymandarin]

ehm, not really, even playing hearable freqs get's me a less sharper and a smoother, rounder sound...

Perhaps I have bat hearing?

I think this may be 'cos you are losing the higher harmonics which give you the sharpness....?

[snooky]

perhaps, but I shouldn't hear any harmonics above 20khz...but the sound still is smoother using 88.1 or 96 khz

[Z3R0T0N1N]

@ SAP

you are right, of course. I was being lazy
yeah, s.r. is across, b.r. is up.
Thanx for correcting that, SAP. I should be accurate if I am going to post about this stuff.

[pHz]

anyone know anything about psychoacoustics ???

slainte rob

[monkeymandarin]

perhaps, but I shouldn't hear any harmonics above 20khz...but the sound still is smoother using 88.1 or 96 khz

oh. Sorry, am better at my medical knowledge than my musical.

[kritikon]

anyone know anything about psychoacoustics ???

I used to know a psycho that listened to punk alot.

[DevonB]

Ok, you tell me I'm wrong, but won't tell me why? Try again.

You are wrong and I did explain why. Contrast then gave another explaination why you are wrong. I'll try again if you like?

Please, do so. I'm referring specifically to how 44,100 samples taken in 1 second time interval of the amplitude of the signal is NOT playing connect the dots.

Devon

Hi Devon;
no dis' meant here; you know much more than I do about computer audio; However, in this particular case it is more like 'climb the stairs' than 'connect the dots', if I am not mistaken. This goes for both amplitude (bitrate) and frequency (samplerate).

Ok, with some further research, I think I'm starting to catch on to what's happening. It's taking the data and mathematically recreating the analog signal. The math behind it is a bit beyond my comprehension without someone really explaining it to me, to be quite honest. But based off the information stored, the DAC can recreate the waveform based off the given information stored in the file. It isn't a complete plot file and 'connect the dots' as we think of that we did as kids, but it uses the given information to create the waveform based on sinc reconstruction. The more information the DAC can get its hands on 'ahead of time', the more accurately it can recreate the sound.

Taking it from this angle, I can see how increasing the sample rate won't increase the quality below the given sample rate, because the necessary information for frequencies at that sample rate are there to do a proper reconstruction. Increasing the sample rate just provides information to recreate higher frequencies properly based off of the given formula.

Those of you in the calculus know, please correct me further if I'm off in the weeds still.

Been very enlightening experience.

Devon

[stefancrs]

ehm, not really, even playing hearable freqs get's me a less sharper and a smoother, rounder sound...

Perhaps I have bat hearing?

or you use synths that produces aliasing noise (which will get inaudible at higher sampling rates).

[UltraJv]

So, to sum up - there is not much point in going beyond 44.1 as no one who listens will generally be aware (or care) unless they are listening under geek conditions.

[munchkin]

:Old codger mode on:

I remember when CD's first came out. They sounded incredibly different to tape. Much sharper but not necessarily clearer. They sounded very clinical to my ears.

The CD has developed over the years and now I'm used to the sound. When I play a tape or listen to vinyl the difference is still noticable to me. I've never grown accustomed to the difference in sound between analogue and digital.

When listening to a digital recording at the different frequencies and bit rates I don't think I've ever been able to tell the difference. A 16/44 song sounds the same to me as 24/96. Technically there is a difference but audibly there isn't IMO.

I suppose the proof is in the pudding. Millions of people around the world listen to mp3's daily. The consumer is unfazed by the lower bit depth. People willingly convert their CD's to mp3 and buy downloadable mp3's. I doubt whether many will be able to distinguish an 192Kbps mp3 from a 16/44 wav file. The ear is a lot more forgiving than the eye when it comes to distinguishing between different digital codecs.

I think CD's were foisted upon us by multi-nationals desperate to market new technology. In order to compete with other technologies High Definition TV/DVD/Audio will be forced upon us. I believe this has little to do with improving media quality for the consumer and more to do with marketing. The multinationals are fighting a marketing war to ensure the technology that their R&D have spent millions developing becomes the standard. If quality was at the forefront of these multinationals concerns then Betamax would've never died a death. The consumer couldn't tell the difference in that case either.

Is there a difference between 16/44 and 24/96? Technically yes but it doesn't make a difference when it comes to making music (great, shit or mediocre.)

Shit! I can't switch the old codger mode off.

[dharmawan]

hmm... feedback should work better at higher sample rates since unit delays are smaller, no?

[Christian Schüler]

I don't know much about this, but I can give a specifc example. I have found that Reaktor instruments almost always sound better at higher sampling rates. I generally bump up the sampling rate to 88.1 within Reaktor, even if the project sampling rate is lower; I downsample when I render to audio, or later if I'm running the host at a higher sampling rate. Specifically, I've noticed on Steampipe, the physical modeling synth, that the tuning is less accurate (flatter) at lower sampling rates, espeically, for some reason, when I've closed the project and opened it again later. The tuning is corrected when I bump up the sampling rate. For whatever reason, after Reaktor parts are rendered and downsampled to 44.1, they still sound better than if they start out at 44.1 or whatever.
-c

Seams like that Steampipe must stick to waveguides with integer delay units to avoid some of the waveguide patents. This means they cannot use fractional delays to get the pitch right.


As for Reason running better at 88.1, it's probably due to simulation of non-linear effects.

For example, if you have a 5 kHz sine running through a distortion, and your sample rate is 44.1, the added harmonics are 10kHz, 15kHz, 20kHz, 25kHz, 30kHz, etc, but any of these above the Nyquist limit (in this case, 22050Hz) would reflect back and appear as 19100 Hz, 14100 Hz, 9100 Hz, 4100 Hz, etc.

That why a non-linear process, like distortion, FM, or a physical simulation, benefits from oversampling. But for mere reproduction, like recording and playback, with added (linear) reverb or a rompler with just filter and eq, 44.1 or 48 is theoretically enough.

So a useful thing would be, running Reason at a higher rate, but not actually storing a 88.1 kHz recording on disk.

[kritikon]

I'm definitely a codger in Munchkin's camp, but I'm willing to try anything once (as long as it doesn't hurt). It was mentioned here earlier that somebody renders at top quality (at least sensible number high quality - which to me is 88.2/32). 88.2 is a direct multiple of 44.1, and 32bit will negate any need to dither (actually I generally don't dither anyway....but that's another story)

So what do you do exactly? 1) Render everything at 88.2/32 and mix the whole lot at that level until the final mix/master when you reduce it down to 44.1/16?
Or...2) render at 88.2/32 and then reduce down each audio piece to 44.1/16 and then mix as usual?

Because doing it the first way is going to put an awful lot of strain on my PC, and the files are going to be f**king ginormous! (is that spelled correctly...it doesn't look right ). And doing it the second way - Not only do I save the initial files, I also have to save the reduced files, so the whole project is going to be even ginormouser ( ), although certainly the mixing will ease up on my PC.

I'm willing to give it a go as long as my PC doesn't go up in a puff of smoke...not sure I've got enough carrots to feed the hamsters inside it? Personally I don't feel I'll get any better mixes from it, but I'm prepared to be proved wrong by my own doing. An open mind means lots of nice sounds echoing round in it