Oh please! You can't have "aliasing and misrepresentation" of the recorded waveform otherwise it's going to be a recording full of complete rubbish and no one would be using digital audio. This is why
all ADCs have an anti-alias filter, it's not optional, it is an absolute requirement and it's not based on bit depth but purely the Nyquist Point, IE. roughly half the sampling frequency. What college did you go to that taught you the rubbish you are spouting?
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Originally Posted by CDBacklash /img/forum/go_quote.gif
As for your precious 16bit, a lot of studios are now recording in 24bit and then converting 16bit for CD media (oh no!).
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That's hardly a suprise to me, as you would know if you'd bothered to read this thread before trolling! I started using high bit rates for recording when you were probably still in diapers so read the damn thread!
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Originally Posted by CDBacklash /img/forum/go_quote.gif
Digital storage mediums will _never_ do audio true justice because of the quantisation that occurs... You do know amplitude quantisation occurs in all digital media where each waveform is not represented as a complete picture, but merely as a snapshot, right? ...
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And you do know that a dithering quantiser was invented probably before you were born which converts the interpolation errors during quantisation into un-correlated noise and results in a perfectly linear reproduction of the original waveform ("complete picture"), right? Obviously not!
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Originally Posted by CDBacklash /img/forum/go_quote.gif
There are more ticks to snap to in a 24bit digital recording and it will be a more accurate representation of the sound if done correctly.
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Now you are completely getting yourself screwed up. Ticks are the basic timing units derived from the BPM (PPQN) of your sequencer and have nothing to do with sampling rates, bit depth or indeed digital audio! Or maybe you are confusing the quantising of MIDI data with the quantising of digital audio, the same term but totally unrelated processes. If you mean there are more quantisation levels in 24bit then yes there are, as explained in my original post!!!
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Originally Posted by CDBacklash /img/forum/go_quote.gif
Most engineers will apply a brickwall (that is a compression filter with a small attack and release time) and then increase the volume of the track.
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You've even got this wrong. I take it you are refering to brickwall compression, which if you knew anything about audio engineering is actually called limiting. The difference between a limiter and a compressor is not the attack and release times but the ratio, which for a limiter needs to be infinity:1. And you've really displayed your ignorance as if you compress or limit the track you are decreasing the dynamic range and therefore require a lower bit depth to encode it. I would call the college you attended and ask for your money back!!
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Originally Posted by CDBacklash /img/forum/go_quote.gif
24bit audio has the capacity to store orchestra hits or heaven forbid cannon blasts (from a particular epic) far more realisticly because of the dynamic range...
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Exactly, so hands up anyone out there that wants to put on headphones and hear the true volume of what a cannon being fired from 3ft away actually sounds like?
To be honest, it's hard to find a single sentence in your post which bares any resemblance to how digital audio actually works. My guess is that you are a kid who has just left college and who has completely mis-heard or mis-understood what your lecturer has tried (but obviously failed) to teach you!
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