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-- 96 khz or 120 khz

96 khz or 120 khz

why bother when processing at such high rates if everything goes back down to 44khz

Why bother mixing in 24 bit or float when it's just going to get bounced back down to 16 bit? Because the mixing and mastering process itself can be lossy, so it's better to work at a high resolution.

Almost every financial system on the planet uses 4 decimal points internally, even though every dollar amount is rounded to 2. If you're doing many calculations, you always work at a higher precision than what you need on the output, otherwise you end up with inaccuracies.

I still have yet to see anyone prove that they can hear the difference between 44/48 and 88/96, but the precision of your final result is an entirely separate matter from the precision of your calculations.

If you have both options available, it is always better to just use the higher rate.

However, if only 44 hz is available, don't go panicking and buying more things.

This guy says the device chain in a PA should begin with a 96kHz wave file. Computer Music Tutorial also asserts that, even when converting down to 44kHz, it's best to start with as high a resolution as possible.

This seems to come up every now and then.

According to Bob Katz' mastering book, working at higher bit rates improves the end result when coming back to 16 bit.

Working at higher sample rates doesn't improve things at all when coming back to 44.1 kHz. It'll only use up more hard drive space, RAM and CPU.

Of course, it gives you the luxury of running off a higher resolution version if it's going to go on a Blu-Ray disc or something at some point (if you've recorded external instruments).

quote:

Originally posted by EddieZilker This guy says the device chain in a PA should begin with a 96kHz wave file. Computer Music Tutorial also asserts that, even when converting down to 44kHz, it's best to start with as high a resolution as possible.

Thanks for that video, that was actually really enjoyable to watch

if you export from your daw and convert directly to mp3 without any mastering or anything i dont see how it matters, the daw works in 32bit anyway dont they? of course if u plan to send it to mastering or something like that i see the point but seriously do any of us produce at that level anyway? if youre recording external synths or vocals etc id probably work with 24bit though. im not sure about this but i havent had any problem producing in the box at 44,1 16bit but ive had problems with the oposite. unless you find a good reason to do elsehow id stick to the standards. imo. most of you dont have monitoring (soundcard+monitors) being able to reproduce the microdynamics in 24bit anyway. elitist thinking never get things done.

edit: what im trying to say is that if you are working with software only i dont really see any point in exporting the final result as anything else than 44,1 16bit unless you plan to master it. your soundcard settings should be as good as possible for the best possible monitoring while producing. IE 24bit 96kHz is pretty common. This doesnt mean its any point in rendering at that rate. I dont really know how to explain what I really mean.

edit2: lol. in the end its all about how you produce. if you record stuff and do external mastering etc etc (like a real oldschool band or something), definitly work in as high bitrate and samplerate as possible. But if you, like me, only works ITB with software, and are a noob like most of us, you SHOULD actually export as 44,1 16bit, but monitor at 24bit 96kHz if you have the possibility in your soundcard. send a label a 24bit 96kHz wav for release and they might fuck up the mp3 convertion, it has happened to two of my releases and I know many others had the same problem. Submit 44,1 16bit wav to labels is my advice.

quote:

Originally posted by derail Working at higher sample rates doesn't improve things at all when coming back to 44.1 kHz. It'll only use up more hard drive space, RAM and CPU.

On what do you base this claim? A higher sample rate, like a higher bit rate, more closely emulates the analog signal; a lower bit rate will make a sine wave look more like a square wave, and a lower sample rate will make it look more like a triangle.

Mixing at a higher sample rate allows you to keep those higher harmonics, minimizing any aliasing distortion during the mixing process itself. You'll still get a certain amount of that distortion when down-converting, but that's the same as dithering from 32-bit to 16-bit.

Without adding more to what has already been said (in favour of using a higher resolution), might I just add that without using a sufficient low-pass filter before resampling down to your output sample rate, there is no point in using a higher resolution. Check out Voxengo's r8brain PRO if you're really serious about resampling, especially its LinPhase and ultra-steep modes.

quote:

Originally posted by derail This seems to come up every now and then. According to Bob Katz' mastering book, working at higher bit rates improves the end result when coming back to 16 bit. Working at higher sample rates doesn't improve things at all when coming back to 44.1 kHz. It'll only use up more hard drive space, RAM and CPU. Of course, it gives you the luxury of running off a higher resolution version if it's going to go on a Blu-Ray disc or something at some point (if you've recorded external instruments).

I'd agree with this. Sorry but I don't see the point of sampling at 96khz. Especially when going back to 44.1 since they don't divide evenly and I understand this causes issues.

24 bit I sort of understand though because it means you have more headroom if you decide to perform some dynamics proccesses later.

Tried working in 96khz but the only differences i noticed were a bigger file size for my renders and higher cpu load

quote:

Originally posted by evo8 Tried working in 96khz but the only differences i noticed were a bigger file size for my renders and higher cpu load

my point exactly BUT you should split the discussion into two:

A: during work/monitoring
B: export/rendering and mastering

during A you should have as high as possible without having problems with CPU load etc. if you record external gear this should also be as high as possible, and only then will project-file size be a problem. this is simply your soundcard settings. 24bit 96kHz for example.

when exporting (B) you need to decide whether you plan to send it for mastering or not. incase not you should render as 44,1kHz 16bit as that is what labels, digital download websites, promo CDs etc etc all wants. sending on a different format than standard will in some cases fuck up the whole release and mp3 coding etc etc.

if you plan to send it to mastering you should render your track at whatever the selected mastering company wants it on. if you plan to master yourself in a different software than your sequencer then just render at highest usable format, like 48kHz 24bit or something. then you render another new file with your mastering program at 44,1kHz 16bit.

my point is this: never have your finished product as anything else than 44.1 16bit wav as this is the industry standard. if you send around your 24bit 48kHz (or whatever) all kinds of things might happen to it - it will fuck up. on the other hand you should monitor at the best possible settings.

if you plan on getting your track onto a SACD or surround soundtrack or something like that, you can render two versions. but never send the other version to labels unless they specifically ask for it.

to sum up:
monitoring/recording: 24bit 96kHz
rendering/final: 16bit 44.1kHz

as discussed already it wont make a noticable difference if your working ITB.

however...if your recording 16 tracks of audio from a band - ie 12 mics on a drum kit plus guitars vox etc then having it at double the resolution makes a significant difference.


but it doubles your file sizes... which can be an issue.

when i'm doing ITB stuff i'll always render in 24 bit and dither to 16 bit when burning to CD after mastering.

So my take is that working at higher bit rates (as high as possible) is more important than working at higher sample rates.

That said, if you have the disk space and CPU (which is cheap these days in relative terms) then work as high as possible. Now bear in mind on a pair of PC speakers, working at 96 or 192 is just a futile exercise.

So all things being equal (you have the kit to take advantage of it) working at higher sample rates, is better than lower sample rates.

The main advantage as pointed out, is being able to hear more during the monitoring process, and therefore engineer, mix and produce better.

I've worked with one very well known score mix engineer who will only work at 96k - any thing less and it's a no go. One time our digital desks were being buggy an would not stay in 96k mode, he literally walked out of the mix session and took the gig to another studio.

In terms of recording at higher sample rates, there is evidence to suggest that you are able to capture high harmonics that would otherwise not be recorded. While this does not matter for lower register frequency sounds like vocals or basses, high strings have been shown to benefit from this effect, even if the maximum upper threshold of hearing frequency (20khz) is one quarter below the sample rate 96khz.

But, it will also greatly depend on your soundcard as to whether this this higher sample rate can be taken advantage of; even tiny clock variations at these higher sample rates can mean huge inaccuracies meaning the introducing of artifacts, truncation of waveforms and aliasing.

If you have something like a lavry, metric halo, lynx or prism then you're on the right track but anything below that grade, it's really not worth it all things considered.

Also, beyond a certain point (IMO 96khz) the ratio of file size to perceived improvement is not worth it.

IMO, work at 96khz of you have a pro studio grade soundcard and monitors, otherwise it's just a waste of CPU and disk space.

ive been wondering though, my soundcard doesnt have settings for bitrate. will the DAW use whatever it can when monitoring? like 32 float typically?

quote:

Originally posted by Senator Clay Davis ive been wondering though, my soundcard doesnt have settings for bitrate. will the DAW use whatever it can when monitoring? like 32 float typically?

You're in logic so unless you've changed it from the default setting you should be working in 32bit float.

Your AF4 can do a max of 24bit/96k which is more than enough. Bear in the mind the conversion from a digital signal is done in the AF4 so your DAW provides the mixing environment & headroom (at 32 bit float it's above 1500db) and with 24bit it's close 115db of usable headroom at output conversion - you're never going to get close to that unless you're doing it wrong and like blood dripping from your ears.

quote:

Originally posted by DJ RANN Your AF4 can do a max of 24bit/96k which is more than enough.

but will it do this automatically? the bitrate that is, i know i can adjust the samplerate in the AF console, but the bitrate? will logic just setup the best possible with the available device??

I wonder if certain plug-ins incorporate an 'upsampling' feature where a 22.05 kHz low-pass filter isn't used... To my knowledge so far no such (mastering) plug-ins exist (I haven't noticed a distinct difference between working at 44.1 kHz), but basically the only benefit to using a higher sample rate is if you have content (whether it be existed or generated) past 22.05 kHz.

Think about it: say you boosted the treble at 16 kHz with a wide Q...what happens past 22.05 kHz when working at 44.1 kHz? The plug-in's design would be made so as to roll off around the nyquist frequency with a steep low-pass filter to avoid aliasing, but certain plug-ins could be designed to work more efficiently without this when working at 96 kHz.

quote:

Originally posted by DigiNut On what do you base this claim? A higher sample rate, like a higher bit rate, more closely emulates the analog signal; a lower bit rate will make a sine wave look more like a square wave, and a lower sample rate will make it look more like a triangle. Mixing at a higher sample rate allows you to keep those higher harmonics, minimizing any aliasing distortion during the mixing process itself. You'll still get a certain amount of that distortion when down-converting, but that's the same as dithering from 32-bit to 16-bit.

As I said, I based it on Bob Katz's book "Mastering Audio". That was the conclusion he reached. Working at higher bit rates made for an improved sound when going back to 44.1kHz/16 bit, but working at higher sample rates didn't. I'll dig it out and see what his main point is. Higher sample rates will give you more frequency content above 22 kHz, but this may actually degrade the sound if you aren't using appropriate filters to remove this content when down sampling.

I have a really old computer in any case, so I'd need a strong case that it's going to make a noticeable difference in quality in the final 44.1kHz/16 bit file. I suppose it's easy enough to try out. I'll give it a go.

quote:

Originally posted by Senator Clay Davis but will it do this automatically? the bitrate that is, i know i can adjust the samplerate in the AF console, but the bitrate? will logic just setup the best possible with the available device??

Yes, it's running at 24bit at all times. the sample rate is user selectable so what you're actually hearing from your speakers is 24bit/44.1k conversion quality.

So the digital signal gets to the AF4, and then is converted to analogue using 24bit, 44.1k conversion (unless you change the sample rate).

The internal processing of your daw (32BFP)is just internal digital headroom.

quote:

Originally posted by derail As I said, I based it on Bob Katz's book "Mastering Audio". That was the conclusion he reached. Working at higher bit rates made for an improved sound when going back to 44.1kHz/16 bit, but working at higher sample rates didn't. I'll dig it out and see what his main point is. Higher sample rates will give you more frequency content above 22 kHz, but this may actually degrade the sound if you aren't using appropriate filters to remove this content when down sampling.

As far as I remember it's because bit rate is based on log (not linear) so there is an advantage in recording your original material with 24-bit resolution, as it increases the available dynamic range and also translates into greater headroom, reducing the risk of overloads/ transient clipping.

Samplerate is directly linear, so if you're only going to play back your track at 96k on a 96k system, then great, it will technically sound better but it's doubtfull you can really tell the difference.

If however it's ever going to end up as 44.1k, then my advice if you really want to go to a higher sample rate is to go with 88.2k as it's an even division, not an approximation which can result in waveform truncation.

Ther is an argument to say that higher sample rates are better because if you have 30 tracks of audio, all at 44.1k then they have a give digital inaccuracy over the true analogue source.....but you're mixing 30 of them, which means you're mixing 30 times that inaccuracy together in to your final one track. So if you were to lessen that digital inaccuracy by going to a higher sample rate which was higher quality or a more accurate representation of the source, then combine 30 of them together it's a substantially less inaccuracy than before.

But to be honest it really doesn't make a shade of difference to 99% of us, unless we're recording or using a lot of analogue outboard during the engineering process.

quote:

Originally posted by Atlantis-AR I wonder if certain plug-ins incorporate an 'upsampling' feature where a 22.05 kHz low-pass filter isn't used... To my knowledge so far no such (mastering) plug-ins exist (I haven't noticed a distinct difference between working at 44.1 kHz), but basically the only benefit to using a higher sample rate is if you have content (whether it be existed or generated) past 22.05 kHz. Think about it: say you boosted the treble at 16 kHz with a wide Q...what happens past 22.05 kHz when working at 44.1 kHz? The plug-in's design would be made so as to roll off around the nyquist frequency with a steep low-pass filter to avoid aliasing, but certain plug-ins could be designed to work more efficiently without this when working at 96 kHz.

I believe that many synths do 2x oversampling, so many mastering plugins probably would also, or at least include a mode for this. In that case, there would still be an antialiasing filter, but at twice the nyquist frequency, which should help eliminate unwanted reflections back down into the freqency spectrum

quote:

Originally posted by kitphillips I believe that many synths do 2x oversampling, so many mastering plugins probably would also, or at least include a mode for this. In that case, there would still be an antialiasing filter, but at twice the nyquist frequency, which should help eliminate unwanted reflections back down into the freqency spectrum

I think some of the UAD plugs upsample internally, im sure other plugs do also

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