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Old 06-29-2007, 06:08 AM   #1
rawdio
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Default Retaining highest possible quality when using mp3s / rate conversion

Normally my projects run at 48 or 96khz. Now I have to use a vocal mp3 which is recorded at 44.1.
I'd like to keep the quality as high as possible and save any unnessecary quality loss during conversions.

Does my following approach make sense?

1. I convert the mp3 to 32bitfp wav (keeping the 44.1khz)
2. Set my project to 88.2khz and import that wav (because some plugins I use sound better than with 44.1)
3. Use r8brain to go back to 44.1 after mixdown

Question 2: When I use a 44.1khz mp3 in a 48khz project and later do a mixdown to 44.1... does Reaper convert it twice? Probably yes because the "audio flow" and effects are using 48khz I think?

Question 3: Does selecting 192khz in the rendering dialog have the same result as having the project run with 192khz all the time (apart from any recordings of course)?

EDIT: Question4: (very similar)
When I import a 44.1khz file into a 48khz project and later set the whole project back to 44.1khz do I have the same results as when I had started with 44.1 (because the reapeaks are still the ones that were generated for the 48khz proejct)?

Thanks for any help!

Last edited by rawdio; 06-29-2007 at 06:32 AM.
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Old 06-29-2007, 08:06 PM   #2
Staccato
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Hi, mp3's don't have a sample rate, they have a bit rate. (the original wav, if it was a wav, prior to conversion had a sample rate) I would be hesitant to convert an mp3 back to wav, because it is during this conversion where the chances of losing quality are the highest.

One of the nice things about Reaper is it's ability to import multiple formats, so I would just import the mp3.

HTH
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Old 06-29-2007, 09:18 PM   #3
Measuring Man
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Any sample rate change will potentially affect the sound of the file. You'd be best served by starting a session at whatever the destination sample rate would be (ie 44.1 for CD, 48 for TV/DVD, 96k for DVD-Audio-very rare), and converting the file to that. There are some minor tangible benefits to upsampling for processing, but only if the processing, plug ins that is, can take advantage of the upsampling. Some high end EQ's and plug ins upsample in order to reduce quantization distortion, but that requires adapting anti-aliasing filters and other technical goodies, cheaper plugins wont be so finicky. Otherwise the upsampling and downsampling will only needlessly tie up computer resources by doubling or even quadrupling the CPU and hard drive access and space usage.

Plus, 32bit floating point is good for recording the output of processing from plugins and VSTi, but if the file was originally 16 bit, there's no need to convert it to 32bit for playback. You would want to potentially record the output of your final mix at 32bit, but only if the mastering facility can handle that format, many can't, or don't bother as it's still only a 24 bit word in the file anyway. Recording a 24 bit output that isn't clipped from being too loud by turning down the master fader is the same thing.

The crap you learn in working at a mastering facility...

The end result is I mix 24bit/44.1 unless it's goignt o DVD video, then it's 24bit 48k. Get many more plugins and track playback that way.
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