"digital audio"

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-I am unsure of the concept but I will list my assumptions. Please elaborate and correct. . .

-when audio is converted to digital there is a 'sample' of that audio taken 44.1k times a second (cd quality) with 16 bits of information per sample. So when I send audio from my mixer to my soundcard, the soundcard does the sampling/converting. So this signal is susceptive to degradation while travelling down the analog cables, and therefore the signal that the soundcard is sampling is not the same that the mixer is sending.

-so if I want to send a digital signal to the soundcard, the 'sampling' is done by A/D converter within the mixer, correct? So this way, the signal does not degrade while traveling down the cable to the computer.

-my question is this: Does it really make that much of a difference, considering the sampling that needs to be done is still 44.1kHz/16-bit/stereo? Is the resulting sample really that much better than doing it analog style, since the only difference is the length of the analog path? If I got 2 inch audio cables from my mixer to my puter (hypothetical) would that be somewhat similar? Are there different 'techniques'/'methods'/'algorithms' for converting analog audio to digital audio or is there one standard?

-also I remember reading about 'Super Audio CD' (SACD) format which, I think, was 9x.x kHz/2-bit/stereo, and how it is competing with the DVD audio format which is . . .? Does anyone know what the dealio is between these two?
 
There is more to it that that:

For starters, ANY conversion "degrades" the signal- meaning that if you take the analog output of a digital device and record it with another digital device, it is not an EXACT digital copy- and even digital interfaces such as S/PDIF are prone to jitter (where the word clock is slightly out of sync...

Not to mention that cheap convertors tend to be noisy, PCs introduce noise, etc... which is why people might spend thousands on an Apogee external set of convertors- and I'm talking about just the convertors!

I'm of the belief that once a signal is digital, it should STAY digital- and I record at 32 bits- with a noticeable improvement over 16.
 
thanks. . .

I was referring to the digital output of the digital device going straight to the sound card (digitally). Is this signal (sound quality) really noticeably better than analog audio to soundcard to digital? I mean, how much do analog audio cables degrade the signal?

if you record at 32 bits, and you want to put it on cd, you still need to convert to 16 bits, correct? How is that different than recording at 16 bits?
 
I love you, fitz da schitz! :D

. . .I think I will be subscribing to the 'sound on sound' site. . .hopefully I will have time to come back here every now and then :rolleyes:
 
is better than what?

when you record in 24 bits, don't you need to convert to 16 bits to burn it to CD/minidisc? I don't understand the concept of converting 24 bits to 16 bits, rather than recording at 16 bits. What are the 24 to 16 bits, just higher quality bits than the just 16 bits? (pssst, derogatory, I know a bit is a bit)
 
CD audio is 16 bit so yes you have to convert to 16 before you burn to CD.

Here is why 24bit gives you an advantage.

Whenever you process audio you lose dynamic range and quality. This includes everything you do after recording all the way up to mastering. So if you start out with 24bits you have more dynamic range to "give". Then you finally can commit to 16 bit and transfer to CD.

I hope I am making sense.


peace,

fitz
 
fitz said:

I hope I am making sense.
Click to expand...

. . .totally, but

should the sample rate for recording always be 44.1kHz, because I tried recording a mix in SF one time at 48kHz and then when I converted it. . .it . .was. .a . . little. . slower
 
fitz said:

I hope I am making sense.
Click to expand...

. . .totally, but

should the sample rate for recording always be 44.1kHz, because I tried recording a mix in SF one time at 48kHz and then when I converted it. . .it . .was. .a . . little. . slower

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