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Audition - How to record at 24bit via M-Audio 2496?
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| Rikki |
Greets peeps.
Just a curiosity question, how do I record at 24bit from my Audiophile? When making a new wave it allows only 16 and 32 bits to be chosen?
Since the card wont do 32bit I reckon it will just use 16bit and interpolate up the way. Anyone got an answer to this one?
Thanks
Rikki |
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| Diginerd |
Oh boy...
Short version is what kinds of bits are you referring to.. 32bit floatingpoint (as touted by Steinberg and by the looks of things Adobe too) is roughly equivalent to 24 Bit Fix Point maths in terms of sonic precision. (To the anoracks I can argue the merits of both and their subtle differences, but not worth it here). Floating point mathis is what you use on your PC / Mac if you are doing native processing (which you are by the looks of things), Fix point maths is what you run into on DSP based systems and DSP based Synths and outboard FX.
16 bit should in general be avoided as it limits your dynamic range, though that general statement doesn't take into account the limits of the equipment you have conencted.
If you have a behringer mixer then recording at a greater bit depth is not going to help you much as most of the lower 8 bits will be random noise.
The key to any digital system is to have the dynamic range of your digital system exceed the range of what you have infront. ie you want the noise floor of digital system to be lower than the gear in front.
A good analogy to use here is think of a picture on your screen that is sized to 640 x 480 with 256 colors (8bit). Pointing a digital camera at it and filling the whole field of view with the picture on the computer screen with a resolution of say 3200 X 2400 and 24 bit (Millions of colors, as most are) and taking a picture is not going to give you (all things being equal with optics etc) any better picture than a camera with resolution of 1280 x 1024 with a bit depth of 16 bits (thousands of colors). There are only 640 x 480 pixels and 256 colors in the orginal picture, so anything more is unneccesary information to give an accurate representation of the original(for the anoracks I know I'm touching on Nyquest Therum here too)
So to answer your question (or not as the case maybe!) it depends.
If you are just recording a stereo track from Vinyl or CD and not doing much processing then 16 bit will actually probabbly do you well, as the extra quality gained by higher bit depth prolly won't help much.
On the other hand, if you have decent gear and are applying lots of internal effects, are super concerned about quality and (MOST IMPORTANTLY if most of the above doesn't apply) are willing to live with files that are twice as large and live with nearly half the plugin processing power (twice as many bits = big cpu hit) then go with 32 bit.
Sorry not to give a black or white answer, but you are inadvertedly touching on a very complex subject. |
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| Diginerd |
oh to add. The card is feeding the Audition 24 bit fixed math, the software is working with 32 bit, and capturing files that way.
The next question (I'm surprised it wasn't the first) should probabbly be about sample rate.. ;-) |
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| Rikki |
Thanks for the reply, I think I caught most of it LOL.
What I want to do is record my sets in the highest quality possible and you are right, I will be applying post filtering and effects to it (normalise, chopping certain bits in and out, fades, flange etc).
I have figured out (right or wrongly) that 88.2KHz is the recording frequency to go for to avoid harsh harmonics creeping in and once all is done I can mix down to 44.1 without inheriting any resampling errors as its a straight divide by 2 - maybe Im over simplyfying there LOL.
So the answer to my question would be, use 32bit recording in Audition and let the software worry about it all :)
R. |
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| dEEkAy |
I still dont see a reason why to record @ any rate higher than 44.1khz (or 48khz max.) as most tracks, even the vinyl ones come as 44.1 (the masterfile of the tracks pressed on vinyl are usually coming as audio CD to the pressing plant).
It wont change anything on your sound... 24bit, alright, maybe, but not increasing the sampling rate. If you got "harsh harmonics" creeping in or whatsoever, I'd rather check the external hardware (e.g. turntables/mixer and whatever the audiosignal runs through). |
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| Rikki |
| quote: | Originally posted by dEEkAy
I still dont see a reason why to record @ any rate higher than 44.1khz (or 48khz max.) as most tracks, even the vinyl ones come as 44.1 (the masterfile of the tracks pressed on vinyl are usually coming as audio CD to the pressing plant).
It wont change anything on your sound... 24bit, alright, maybe, but not increasing the sampling rate. If you got "harsh harmonics" creeping in or whatsoever, I'd rather check the external hardware (e.g. turntables/mixer and whatever the audiosignal runs through). |
Recording at higher bitrate means any post production ( as I mentioned I wanted to do ) will be carried out with more dynamic range.
R. |
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| dEEkAy |
| I was referring to the sampling rate. |
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| Rikki |
| quote: | Originally posted by dEEkAy
I was referring to the sampling rate. |
The same would hold true for both, post production. Just the same as why studio stuff is at 192k and not 44.1k
R. |
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| dEEkAy |
That's certainly wrong. I'm working at a company who is into Audio Production of all kinds (Radio Commercials, Soundlogos, Voice Recording for movies) and the Fairlight Desk and DSPs are running at 48khz.
However the meters and all stuff are adjusted at -6db.
If it was just for the bits and samplingrate,...why dont we all produce at 64bits and a megaherz? :rolleyes:
:thepirate |
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| Diginerd |
Ok, Confusion is abounding..
Higher sampling rates increase frequency response bandwidth (ie the greater the sampling frequency the higher pitched sound that can be captured). That is all.
Number of bits is dynamic range. (ie Amplitude)
In photographic terms they also correlate. Sampling rate is analogous to resolution, bit depth is the same (amount of colours in your palette).
To reccap, for a digital system to be correctly balanced to the analog part you need enough bits to put the noise floor of the digital equipment below that of the analog. Where that floor is depends on your source material.
For all practical purposes 24bits of fixed float precision, and 32 bits of floating point precision have a dynamic range that exceeds even an SSL or Neve analog console. More bits than this will do nothing for your sound, and consume correspondingly increased CPU/DSP resources. So it's unlikely you'll see 64bit recording files unless it's in a marketer's wet dream.
On sampling rate there are 3 key root standards, one of which is falling by the wayside. I say "Root" standards, as the higher rate sampling frequencies are exact multiples of these for reasons that you are aware of.
32Khz (Old Long play digital format) Used in older broadcast systems, but pretty much retired unless it's field recording by journalists doing speech, and even then it's dying.
44.1Kz (Pop quiz, anyone know why this specific Frequency, and not say 44.2KHz?) the CD standard, and one of the most common sampling rates out there. If you ever want your stuff to go to CD then this is the one (or a multiple of it, say 88.2KHz) you want to be at to avoid ikky sample rate conversion.
48KHz TV broadcast standard. I you're doing sound for picture you'll almost certainly be working here or some multiple of it)
As for high sampling rates, this is where things get controversial FAST.
I'll begin by saying that Nyquist Theorem states that to accurately capture any waveform simply requires sampling at 2x the frequency of the waveform.
This theorem has stood up to an awful lot of scrutiny, so with that in mind and all things being equal recording at 44.1KHz should be enough for anyone (or 48KHz if you need the standard), as only a young child's hearing goes up to 20Khz or so. At my age it really stops around 15-16Khz (and less if you have hearing damage.
Not only that but 44.1Kz should sound the same as 48KHz, or 192KHz or even 2Ghz.
The problem is, all things are not equal, and there is a lot more going on than simple sampling rate.
You've disovered this with your "Harsh harmonics".
What's happening is that part of every A/D Converter there is a Lowpass filter. This ensures that nothing out of the Niquist range gets into the signal and cause audiable artifacts (Alaising) to be generated.
With a 44.1KHz or 48KHz sampling rate the filter cutoff has to be pretty sharp, and doing it right is very very expensive, and also requires an awful lot of clever design. This is a major reason why people will spend buckets of cash on Apogee and Prism (to name but 2) converters over your regular joe soundcard converters.
Too sharp a filter, one with a non-optimal response or one that is poorly designed in general will have a significant impact on the sound passing through it. You already know about filters and tonality if you have played with more than a couple fo synths. The effects of this are very real.
Now, if you sample at is higher rate, say 96KHz or 192KHz what the designer can do is to create a filter with a much more gentle cutoff slope, which is much easier to design, and also (and more importantly) has substantially less sonic impact (Ringing) than a steeper filter.
This, and not the extra samples is why your interface sounds better at 88.1KHz instead of 44.1KHz. Remember, your hearing at best only goes up to 20Khz, which means that a 44.1KHz sampling rate is more than 2x so you're clear under Nyquist Theorem even at 44.1Kz.
Anyone who tells you that 881.Khz sounds better than 44.1KHz because there are more datapoints doesn't have a handle on the maths..
As for which sampling rate you should use, that depends on your predicted output format. You've already worked out that 88.2KHz is better than 96KHz if you are going to CD as your ultimate destination. The reverse is true if you are working to picture.
What every you do though you really REALLY want to avoid sample rat conversion. It's ugly at the bes of times, and going from non multiples to another is just plain bad. If you have two decent sets of AD DA converters it is ofeten more desirable to step back through analog than it is to perform even the best SRC.
Finally, there's another factor as yet undiscussed that has an impact too. That's clocking accuracy, also known as Jitter. Think of it as sloppy drummer Vs a drum machine, this can produce awful artefacts that cannot be fully undone. I think I've blabbered anough about this subject so far, so I'll leave jitte for another time (if ever). |
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| nhibberd |
A simpler answer to your question;
M-Audio comes with it's own ASIO drivers. Depending on your software it should be possible to select these drivers in stead of the standard ones you got with the software. I would recomend you use the M-Audio drivers because they are especialy designed to make use of everything the card has to offer. With M-Audio you will find you have zero latency and 24-bit recording possibilities. Your recording software should adapt to the 24-bit recording quality automaticaly when you activate the ASIO drivers.
I must say that recording at high quality though can take up to 30MB of diskspace per minute recorded and an entire set wouldn't even fit on a DVD. I would question if the quality is realy worth it.
kind regards,
Charlie D |
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