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Gear / Technical Help => Post-Processing, Computer / Streaming / Internet Devices & Related Activity => Topic started by: Teen Wolf Blitzer on October 02, 2010, 11:02:33 AM
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I just got a ton of mastered recordings from Pickathon this year that one of our crew guys got. They are all 32 bit 48 hz though. Why is this? How do I get them to 24/48? Or do they need to be redone?
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You could load them into Audacity and then export them to whatever format you want. I'd assume that other DAWs can also work with floating-point files.
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So you're saying I haven't lost any quality in the files I received? I know nothing of 32 bit. Never used it. Honestly I don't even get what it is even after reading about it online. Hence my question. ;D
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They could be 32bfp <1> if they were recorded direct to computer and the recording software uses a 32bfp workspace to accommodate 24-bit files (like Adobe Audition), or <2> if they were already manipulated by an editing program and output as 32bfp.
At any rate, leave them in 32bfp until you finish any processing. (Your s/w probably supports 32bfp, so it shouldn't be an issue.) Then when you're done, dither down 24-bit or SRC/dither down to 16/44.
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Well that would be #2 Brian. ;D They were all like 8-10 track onto a Tascam DR680. I did not do the mixdowns. I just received the solid waves for tracking/downsampling and dithering. So how would I dither to 24 using Izotope? Any special settings?
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A large majority of mixing and sound modifying programs today use 32bit float internally when processing sound. It is a very good format format for this and it keeps a really large part of the sound information intact. Quite a few playback programs handles 32 bit today, not all though.
When converting between formats the signal should as a general rule be dithered. If you go to 24 bit the choice is not sensitive at all in my experience, simply select one. If you go to 16 bits, it can in my experience make a tiny difference when working on finished mixes, but I let my ears judge there.
What you have to look out for however when going from 32bit float to 24 or 16 bit is that the signal never goes above 0dB. The float format allows signals to go above that level, but the fixed formats cannot handle it. The simple solution is to normalize the files first, leaving a small margin up to 0dB, I generally leave 1 dB. There are a lot of technical reasons for leaving a margin, a bit too involved to describe right here. Regardless one or a few dB-s in lowered volume really does not make much of a difference.
// Gunnar
Gunnar
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I just got a ton of mastered recordings from Pickathon this year that one of our crew guys got. They are all 32 bit 48 hz though. Why is this? How do I get them to 24/48? Or do they need to be redone?
Wow, I had no idea that this had thrown you for a loop. I guess I just figured that you knew what the 32 bit stuff was. BUt anyway, I do all editing/processing in the 32 bit realm. I also leave the mixdown masters that way it makes opening them back up for editing too ;D