slacknote wrote:
It's actually pretty simple: the more bits you have the greater the possible difference between quiet and loud in your tracks.
futures_untold wrote:Even when using VSTs and samples that are not 24k, your computer/software/soundcard will interpolate the empty points between each sample in the audio stream as it upsamples to 24kbit, in theory giving a smoother sound.
Be careful here as you're mixing terms/axes and confusing things a bit.
There are no 'empty points between each sample' when you're talking about bit depth, as you are there. I know you're referring to sample rate, but talking about 24 bit. Then again you're using 24kbit, which doesn't really mean anything at all in this conversation, unless we're talking the worst mp3s ever

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To separate and clarify;
In terms of bit depth, the computer will populate the new, empty bits (y axis, essentially) with zeroes when changing a 16-bit sample to 24-bit. This does not ‘give a smoother sound’. It makes no difference whatsoever to the sound at that point. What it does do is enable subsequent calculations/processing to be done with a lower noise floor, ie more accurately. It doesn’t give you anything back, that’s already lost. It just helps to stop things getting more f#cked up when you process further.
For example, if you get a great 24-bit drum loop, resample it at 8-bit, and then resample
that at 24-bit, you don’t get the initial drum loop back. You have an 8-bit sounding version, but at 24-bit resolution. This principle is one reason why most plugins work at a higher internal resolution than the native resolution, it reduces the noise floor
for that stage of processing.
When it comes to sample rate, it’s not dissimilar when over/upsampling – here the gaps between samples are filled with zeroes, and it doesn’t do anything inherently to the sound. Loads of higher frequencies that were lost at 44.1kHz don’t come flooding back. But computers don’t do this by default, they work at a native/host sample rate and any upsampling etc is handled internally by plugins (when a plugin upsamples). The purpose is much the same as with the bit depth case, better accuracy, but in this case it matters more at the high end of the spectrum, be it for the purpose of HF decramping in eqs, or for better compression response/handling etc. Point being it still comes down to the native sample rate before being passed back to the host, just with potentially less inaccurate results.
All this bollocks is kind of missing the point of the OP’s question though…
Perrrrrrrsonally I feel that if you work 100% in the box, unless you’re going to DVD or another hi-res output then it is better to avoid any sample rate conversion and work at 24/44.1. I’m just a big one for avoiding any unnecessary sample rate conversions, cos it’s a damaging process unless you use the absolute premium SRC ( = £££££).
If you’re recording real stuff a lot then you should experiment and find out what sounds best.
Bigger numbers don’t necessarily mean better performance. A cheap soundcard with high jitter will not sound any better at 96kHz, it will be even more wobbly and less accurate on a per sample basis. A well-designed, high quality convertor will sound better at 44.1 than a cheap one at 96kHz, despite what the numbers say. Accuracy is equally (if not more) important as number of samples taken, ie . Therefore experiment and see what works best for you with your equipment.
Anyway, back to work…
