Downloaded digital redux planet machine
Drum- extensive machine drum library..

Is high quality too-24bit

.. I replaced a few exact sample hits with
The higher quality than I had-16bit

..turns out for me-too good quality:
Much harder to eq the Roland 626 clap
Today in 24bit - hard to pick a high
Freq. band to boost, much more so
Than when I was using it originally
As 16 bit I thought

The greater detail/quality made it harder to
Pin point high bands to boost

..and now I think I just prefer the lo-fi
Sound of 16bit anyway, has a rawer edge

But I'll continue on anyway, might stretch the
Ear capability and concentration : )

Greater detail? there shouldn't be greater details... The ONLY thing you get with 24 bit is a lower noise floor due to smaller quantisation error thereby allowing greater headroom.

Do sound louder as such, would the noise floor
Colour the sounds, def 'sounds' brighter..

Probbly nothing to do with bt depth. 24-bit is really only about noise floor.

Maybe these particular samples were just pre-processed in a way that left little room for further tweaking.

well, the 24 bits are dry, maybe the 16 bits had been messed with-
def much easier with the 16 bit to go whatever freq. 12 khz ya I'll boost a bit,
took longer to know whats more ideal with the 24bit.

like u mentioned and etc perhaps the 16 bits had some processing......

I know some of the other parts of such kits had def. to my ears.

It was alot harder, i felt anyway.

The difference you're hearing is unrelated to the bit depth, which simply lowers the noise floor giving more headroom.

What you're hearing is the difference between the samples - a different issue altogether! So the new samples have been recorded/sythesised with more definition or whatever, and you appear not to like them as much as your old samples - which is valid but unrelated to bit depth.

Bob

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www.bickerton.co.nz

Try converting your 24 bit material to 16 bit and see what happens. No doubt the differences you hear will persist because the material is different. As others have said, it has nothing to do with 24 v 16 bit. And this will affirm that you have good ears.

The only real value in using 24-bit files is to provide headroom for digital effects and processors, so it's more about number crunching than what you can actually hear. (32-bit float is just about providing a more convenient number value for certain processor/effect calculations I think.)

I'm not sure if it is worth having 24-bit files for amps that have built-in effects and processors, but the effect is probably negligible due to the rest of the equipment.

I had assumed 24-bit renderings benefited vinyl prints, but Hugh informed me that 16-bit converted to 24-bit by the engineer is standard practice. (I'd assumed 24-bit files would have been useful to the engineer because of any digital processors that may be used.)

Higher sample-rates may be useful for recording (close up) sounds (using a high-frequency mic or DI signal) that are intended to be slowed down, so that high frequency harmonic content is present.

A 96kHz sample rate translates into 48kHz (potentially anyway), not too far from the 'best-case' analog system output, but most equipment wouldn't be able to capture or reproduce these frequencies anyway. I can only hear up to about 18kHz now in any case, although people talk of high frequency effects on the body.

For converstion purposes, there are those who argue that 88.2kHz is better than 96kHz because it only entails a division by two into 44.1kHz.

There are so many other factors affecting quality anyway: soundcards, preamp, cables, speakers, etc, etc.

Quote J.A.S:

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For converstion purposes, there are those who argue that 88.2kHz is better than 96kHz because it only entails a division by two into 44.1kHz.

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Those people are mad, of course, because sample rate convert is never a divide by 2!!

Quote narcoman:

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Those people are mad, of course, because sample rate convert is never a divide by 2!!

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Thanks. Has this got any substance at all...?

http://theproaudiofiles.com/3-reasons-to-record-at-88-2-khz/

Not really. Point one only works if there is no content at more than half the target sample rate... largely not likely unless it was a prior up sample. Best to use interpolation and resample method with the required band limiting.

88.2 is pretty good because it's the nearest widespread commercial format that gets the filters way outside of the audio band. Lavry mentions using an idealised 60khz sample rate...

Quote J.A.S:

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Has this got any substance at all...?

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Not really.

Synchronous sample rate conversion went the way of the dodo a long time ago. it was a relatively simple approach but with limited resolution in terms of the pracical wordlengths achievable, and the approach wastes computational effort calculating millions of sample values no one has any interest in.

Modern asynchronous sample rate conversion is far more sophisticated and achieves a technical performance that is significantly in excess of any real-world converter and very close to the 24-bit theoretical level. And that's with any ratio of input to output sample rate. There is no measurable difference in performance between using simple interger ratios or complex ones.

In fact, some of the best performing D-A and A-D converters from the likes of Benchmark, Crookwood, Cranesong, Dramwer and others all use non-integer sample rate conversion as an inherent part of their jitter-isolation. In the case of the D-As these designs typically up-sample the incoming digital audio at whatever rate it is to something odd, like 210kHz, because that's the rate where the actual D-A converter chip achieves its best performance figures!

There is a (weak) argument for sampling original material at a rate higher than 44.1kHz in some cases because a lot of A-D systems are designed with pretty poor anti-alias filtering. Often it's only 6dB down at half the sampling rate -- it's done that way because it makes the A-D' s digital filtering a lot easier -- and this could result in aliasing if there is a lot of strong harmonics at the extreme HF end. Close miked cymbals, strings and brass instruments can all generate enough HF energy to cause this problem in some situations with some equipment, in which case switching to a higher sample rate will sound noticeably sweeter. The issue then is how to down-convert to 44.1kHz to cure the problem... but that requires a properly designed digital filter -- some SRCs do this properly and some don't. Check out this website to see what I mean!

The 176kHz / 192kHz thing is entertaining, and really is all about 'mine's bigger than yours'. As Dan Lavry point's out, the higher the sample rate, the greater the proportion of error in sampling time and the lower the actual resolution. As Narco says, Lavry argues (very sensibly in my opinion) that the optimal sample rate is actually 60kHz. 96kHz can be useful for the rasons mentioned above, and Lavry gear offers 96k rates. But the quad rates are a folly and Lavry refuses to support them!

Hugh


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Technical Editor, Sound On Sound

Quote J.A.S:

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I had assumed 24-bit renderings benefited vinyl prints, but Hugh informed me that 16-bit converted to 24-bit by the engineer is standard practice. (I'd assumed 24-bit files would have been useful to the engineer because of any digital processors that may be used.)

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I'm not sure where that came from, but I think some misinterpretation has occured!

The practical dynamic range capability of vinyl is far less than 16 bit, so 16 bit material could be used as the source for a vinyl record without penalty. However, it is normal to work with 24 bit files because that's what most DAWs expect and, as you say, with an appropriate gain structure it allows headroom for signal processing.

hugh

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Technical Editor, Sound On Sound

RE: I'm not sure where that came from, but I think some misinterpretation has occured!

Quote:

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RE: Headphones + LF Loudpeakers for more realism?

Quote J.A.S: So if people were making prints of compilations (of other people's) music it would be better to use a 24-bit file then?

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Quote:

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It would be sensible to compile the 16 bit source tracks into a 24 bit master to provide plenty of headroom and an insignificant noise floor within which the source material can then be rebalanced and equalised to make them sound like they belong together -- all the things a mastering engineer normally does -- and then produce a 16 master of the final compilation for the end consumer. But there would be little if any practical advantage in starting with 24 bit source files.

hugh

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Sorry, I thought you meant "mastering engineer" as in the vinyl mastering engineer, the 16-bit tracks being the songs and the master being the album. You meant the mastering engineer in the production studio and mixing 16-bit multitracks into a 24-bit final stereo digital master. Thanks

Actually, no that's not right is it? Sorry, I've not had much sleep so I'm a bit addled today.

You really meant 16-bit song tracks compiled into 24-bit album master to provide headroom to make the songs more sonically matched with eachother (via EQ, etc)? But that's by the digital production stage not the vinyl preperation stage?

Cheers.

In most cases, everything is prepared digitally in a DAW and then tweaked and output to the appropriate medium, be that download, MP3, CD, DVD, or vinyl.

Hugh

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Technical Editor, Sound On Sound