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TranceAddict Forums > DJing / Production / Promotion > Production Studio > Rendering in 24bit
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zodiac9
Supreme tranceaddict



Registered: Oct 2005
Location: Houston, TX

I'm working on a solo black metal project, and at first I was recording the guitar parts in 16 bit, but then I decided to record in 32 bit, at 44.1 kHz. My soundcard doesn't even give me the choice to record in 24 bit, my only choices are 16 bit and 32 bit floating point. I rendered the master in 32 bit last night, haven't listened to it yet. I normally render in 16 bit, for my electronic music. I was wondering if it made any difference if you render the master in a higher bitrate.

A buddy of mine told me that when he bounces synths to audio, he uses 24 bit. That makes sense, since you are making a recording. I suppose he then renders in 24 bit, but I'm not sure. Seems if you are using any 24 bit samples, you will want to render in 24 bit. Since it's all reverted back to 16 bit, I'm still not sure if there is any point though. It seems to me, by rendering in higher bitrates, there will be less degregation in the rendering process. When you bump it down to 16 bit, you won't be losing anything.

Old Post Feb-09-2008 01:03  United States
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Derivative
Bipolar Bear



Registered: Jun 2004
Location: Dublin

Well when you quantise an electrical AC signal, a ADC (analogue to digital convertor) samples the amplitude of the input signal a certain number of times per second. i.e. if the samplerate of the ADC is 44,100hz then it will sample the input signal 44,100 times per second. It works like a camera taking snap shots and stores each 'coordinate' as digital data.

So if you imagine a single full cycle sine wave going from peak to trough - this is a function describing voltage (with variable amplitude) over time. An electrical signal (variation in voltage over time) is infinitely continuous. When you quantise it, you literally recreate it as a series of discrete points that are interpolated to give the illusion that it is continuous when it is in fact, not. The space between points is the 'quantisation error' and this error becomes greater with fewer bits and fewer samples. The samplerate of the convertor determines the horizontal resolution. Bit depth describes the vertical resolution. More bits = more vertical resolution. More samples = more horizontal resolution but more of both is not necessarily better.

Dan Lavry made a point that 192khz convertors are completely unnecessary and even goes so far as to make high end convertors that dont clock convertors higher than 96khz. He says it is because 192khz describes a frequency range far in excess of the limits of human hearing, uses up alot of storage space to save all the data and convertors operating at such high frequencies will not be particularly accurate because of the delays involved in crucial electronic components - i.e. it takes time for a capacitor to charge or a logic gate to open/close.

Thus far we are looking exclusively at multi bit convertors which seem to be the best compromise. For an alternative you can check out Sony's proprietary DSD format which performs conversion at 1 bit but with a ridiculously high samplerate (in the megahertz range). Goggle/wiki some of the advantages and disadvantages of this format. It is worth emphasising once again that more bits and more samples is not necessarily 'better'.

Whether you notice any difference at higher bit depths in multi bit convertors depends on lots of things - how loud you are listening to the signal when it is converted back via a DAC (digital to analogue convertor) and how many bits it was originally quantised with. You can convert a 16/44.1 sample to 24/44.1 and there would be absolutely no difference just the same as you can save a 24 bit .bmp as an 8 bit .bmp and then resave it as a 24 bit .bmp again. You don't regain the resolution that was lost when you truncated bits.

The extra 8 bits you get on top of a 16 bit recording are all lower order bits, the last 4 or so expressing low level noise (pretty much). If you don't listen loud enough to be able to hear very low level sound in the recording then you won't notice any difference.

Also, I don't think there is a multibit ADC that operates at more than 20 bits anyway.

The difference is mathematical. If you have a fast enough PC to do everything at 24 bit I don't see why you wouldnt want to. On the other hand, if your PC is really struggling with the load of heaping post processing on 24 bit samples then you aren't exactly going to lose a whole lot by setting your DAW to operate at 16 bit instead.

Natively workign at 24 bits in your DAW even when using nothing but 16 bit samples will produce a difference though as all the post processing you do (compression, equalisation, reverberation, everything) will be at the higher resolution (provided of course that they can operate at the higher resolution). But as mentioned before the difference is very very small and only really apparant at ear shattering levels. For the sake of your ears I doubt very much that it is in anyone's interests to listen so loud.

I think what is more important is the attitude of the person making this kind of decision. If you are like 'fuck it, CDs are 16 bit anyway' then I think thats a bad attitude to have since you clearly havent done your homework. And if you havent done your homework then you are making an uninformed decision. And that leads to all sorts of questions about how many other uninformed decisions you make when putting together your songs.

I firmly believe such laziness shows in your work even if the difference here is almost purely academic.

Last edited by Derivative on Feb-11-2008 at 14:13

Old Post Feb-11-2008 13:49  Ireland
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