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September 13th, 2005, 06:45 AM | #1 |
Major Player
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what is "signed" audio?
can someone tell me what the difference is between "signed" and "unsigned" audio?
also, if a sound is captured at sample rate of 10k, it doesn't add quality to resample at 48k, correct? but when editing, should all the sounds have the same sample rate? what about the same bit size? - does it matter? Know of any good resources for these kind of basic audio questions? thanks. |
September 13th, 2005, 07:03 AM | #2 | |||
Inner Circle
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Quote:
Quote:
If you are recording analog things into a digital format, then you may want to oversample the original audio. So a 12khz tone is better if it's sampled at 48khz instead of 24khz. Quote:
I believe all sound programs will convert all the sounds into one format and then process sound in that format. Some programs aren't very good at converting between sample rates (i.e. 44.1khz and 48khz), so you get lower quality. For basic questions, Jay Rose's Audio Postproduction for Digital Video or Great Sound for Digital Video are good books. One is more post-oriented, the other is more oriented towards capturing good sound (but has some info on post). |
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September 13th, 2005, 10:19 AM | #3 |
Inner Circle
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signed and unsigned means the numbers used to represent audio use a different scale.
if you use 16 bits audio you can go from value 0 to 65536, that is unsigned. you can express this as -32768 to +32768 that is signed. Unfortunately as a bit is a bit, there is now way for a computer to know if you use one format or another, that is why you have to tell him (this is written in the header of the wav file) there is not much difference and you should be able to convert easily from one to another. sample rate and bitsize is totally different thing. bit rate is the number of sample per seconde (usually 44 ou 48 K/s for CD or DV) Most of serious sound editing station works now on 96Khz, since a mathematical law states that you should use twice as much samples per unity of time than your final format. the bitsize express the number of level you can use to convert the analog signal to digital. 2 bits would mean on/off (not really usefull for sound). 8 bits means 256 level (0 to 255) bad radio quality. 10 to 16 bit is ok for sound, but usually 16 bit is the standard, while 12 bit is found in some DV mode. |
September 16th, 2005, 04:59 AM | #4 |
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thanks for the responses.
as to whether you can combine audio sampled at 22k with audio sampled at 48k - will this cause a computer problem? or an audible problem? I've used some sound effects that are sampled at 8k in a 48k project, and they seem to be working fine. Is the program resampling them on it's own? I noticed the program says "conforming audio file" sometimes - is this what it's doing? Thanks again |
September 16th, 2005, 07:09 AM | #5 |
Inner Circle
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converting audio files can give strange effect, specially when the multiplication factor is not round.
from 8khz to 22 or from 11 to 32. 22 to 44 is ok , while 22 to 48 is not. some programs are dealing well with different sformat mixed. my opinion is to convert all your sources to one format (the final one) so you are going easy on recompression while working and can detect problem immediately. |
September 17th, 2005, 01:10 PM | #6 |
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"my opinion is to convert all your sources to one format (the final one) so you are going easy on recompression while working and can detect problem immediately"
when you say "convert to one format", what do you mean by "format"? sampling rate? or just number of bits? or the difference between .wav and .au? I started resampling the files up to 48k (to be used on a DVD ultimately) thanks again. |
September 17th, 2005, 04:56 PM | #7 |
Inner Circle
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yes , when i mean format is frequency,bit,sign and file format.
(premiere for example is a pain to mix mp3 and wav, even if they are same frequency) |
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