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best software for converting 32 bit sample to 16 bit
Old 2 weeks ago
  #1
Gear Head
best software for converting 32 bit sample to 16 bit

Hey all just got a MPC1000 and i have a lot of loops/songs/samples i want to load on them, but they were exported out of ableton at 32 bit (44.1 khz). The mpc1000 only takes 16 bit (44.1 khz). Whats the best, most lossless way for me to convert (software)? trying to maintain my original sound / quality as much as possible
Old 2 weeks ago
  #2
Gear Maniac
Audio out of sound card -> audio in on mpc
Old 2 weeks ago
  #3
Gear Head
Quote:
Originally Posted by mr shifty View Post
Audio out of sound card -> audio in on mpc
as of now i need to usb transfer wavs. pretty big files/long recordings.
Old 2 weeks ago
  #4
Gear Maniac
Chop em up, sample em, stitch em back together in mpc.

You don’t buy an mpc without being prepared for a bit of hard work.
Old 2 weeks ago
  #5
Lives for gear
 
chaocrator's Avatar
converting from 32 bit to 16 is basically not lossy in any way, because no frequency transformations involved — it's just narrowing the dynamic range (which is something that you want anyway to use a waveform as a sample).

so, any modern audio editor should do it will no problems.
Old 2 weeks ago
  #6
Gear Head
would converting in audacity or ableton be better? i heard audacity doesnt dither and therefore maintains higher quality for bit conversion... im not sure if thats true or false though. can someone confirm? wondering if re-loading the 32 bit wav into ableton and reexporting as 16 bit + triangle dither would be worse/better than just converting 32 bit to 16 bit pcm in audacity
Old 2 weeks ago
  #7
Gear Head
Quote:
Originally Posted by chaocrator View Post
converting from 32 bit to 16 is basically not lossy in any way, because no frequency transformations involved — it's just narrowing the dynamic range (which is something that you want anyway to use a waveform as a sample).

so, any modern audio editor should do it will no problems.
would dithering in ableton mess this up?
Old 2 weeks ago
  #8
Lives for gear
 
syntonica's Avatar
Any sample editing software can easily do this. Normalize and then convert--the software will just throw away the bottom 16 bits.

I like Acoustica or Twisted Wave. They both come with demos that will happily convert all your samples as a one-time thing.
Old 2 weeks ago
  #9
Gear Head
Quote:
Originally Posted by syntonica View Post
Any sample editing software can easily do this. Normalize and then convert--the software will just throw away the bottom 16 bits.

I like Acoustica or Twisted Wave. They both come with demos that will happily convert all your samples as a one-time thing.
do i have to normalize in audacity first before i convert to 16bit?



https://learn.adafruit.com/microcont...ile-conversion

this guide doesnt seem to mention normalization
Old 2 weeks ago
  #10
Lives for gear
 
syntonica's Avatar
Quote:
Originally Posted by spenny23 View Post
do i have to normalize in audacity first before i convert to 16bit?



https://learn.adafruit.com/microcont...ile-conversion

this guide doesnt seem to mention normalization
You should because it will make the sample as loud as possible, properly scaling it, so that when you convert, you don't lose any dynamics. Audacity will do both as part of a batch process. I just don't like the way Audacity works. I think it had to do with converting bittedness, although I don't recall offhand.
Old 2 weeks ago
  #11
Gear Head
Quote:
Originally Posted by syntonica View Post
You should because it will make the sample as loud as possible, properly scaling it, so that when you convert, you don't lose any dynamics. Audacity will do both as part of a batch process. I just don't like the way Audacity works. I think it had to do with converting bittedness, although I don't recall offhand.
would converting in audacity or ableton be better? i heard audacity doesnt dither and therefore maintains higher quality for bit conversion... im not sure if thats true or false though. can someone confirm? wondering if re-loading the 32 bit wav into ableton and reexporting as 16 bit + triangle dither would be worse/better than just converting 32 bit to 16 bit pcm in audacity
Old 2 weeks ago
  #12
Lives for gear
 

ultimately: how does it sound ? if it sounds ok just do it.
Old 2 weeks ago
  #13
I just use AIMP, it uses sox for all of its converting so its top notch. I have it installed in a Wine wrapper but if youre on Windows, its native have at it. The only catch in AIMP, and this might happen because Im using it in WINE I dont know, when you batch convert you HAVE to check the bit rate, even if you dont want it to touch that part. I did a ton of batch converting of my whole sample library recently and wanted it all at 48khz. I left the bitrate box unchecked during the entire thing, it upsampled the entire library to 32bit 48k, all the 16 and 24 bit files one and all. Even after converting down the 96k libs I have, my library got BIGGER because it was all 32 bit now haha. You obviously dont have to worry about that this time but in the future if youre sending like 16/48k to it, make sure you check the bitrate box at 16 or it will upsample all of it heh.

One thing to watch out for when youre converting for an MPC.. If your samples rely on embedded loop points like Live/ACID/Akai samplers do... Converting them might wipe out those loops. I dont know if it does it when its just the bitrate but if you convert the sample rate, theyre gone. The loop points are always set according to the samples in the file. When you convert the sample rate, the number of samples change. IF you use a program like Izotope RX to convert it, you can tell it to leave the metadata alone, and it wont wipe it out. But the length of your loop area doesn't match the area you want looped in the sample anymore because its all shifted.
Old 2 weeks ago
  #14
Lives for gear
 
monomer's Avatar
 

Quote:
Originally Posted by chaocrator View Post
converting from 32 bit to 16 is basically not lossy in any way, because no frequency transformations involved — it's just narrowing the dynamic range (which is something that you want anyway to use a waveform as a sample).
So you're somehow chopping off 8 bits worth of data and you say is not a lossy process???

Of course it's lossy!!!

It's just that you're cutting into the very low level part of the signal and under normal circumstances you won't really miss the data being cut.
Old 2 weeks ago
  #15
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monomer's Avatar
 

Quote:
Originally Posted by spenny23 View Post
i heard audacity doesnt dither and therefore maintains higher quality for bit conversion...
The way you stated this it sounds like utter nonsense.
Maybe audacity maintains the original bit depth until you render the master and only then does it do bit reduction (and dithering) or something. And of course, if you don't do bit reduction up front you won't need to dither up front.

But dither is actually mathematically needed when doing any kind of bit reduction to maintain highest quality. So what you stated (at least, the way you stated it) is categorically wrong.
Old 2 weeks ago
  #16
Lives for gear
 
monomer's Avatar
 

Quote:
Originally Posted by spenny23 View Post
wondering if re-loading the 32 bit wav into ableton and reexporting as 16 bit + triangle dither would be worse/better than just converting 32 bit to 16 bit pcm in audacity
You could test this relatively easily.
(just make sure you disable warp in ableton)

Make a test signal (whatever you feel comfortable with, but clean signals will give you a better impression of what goes on so a simple sine wave might be best).
Set the level of that track so that it peaks at about -80dB.

Render in 32 bits.

Make new projects in ableton and audacity.

Import the 32 bit file.
(again, disable warp in ableton)

Render to 16 bits in both. If different dithering methods are possible you can make several exports to check them out. Just don't forget to name them well.

Import all the resulting wavs in application of choice.

Boost all files by 80dB.

Listen for differences.

You should be able to clearly hear the effects of the truncation and dither.
Old 2 weeks ago
  #17
I think youre probably worrying too much about it. In this day and age its going to sound fine unless you go way out of your way to use some 1999 era conversion software or really screw up the options when you convert down.

Check out the spectral analysis charts over at src.infinitewave.ca. Just look at how horrible the MacOS's own built in conversion was back under OS 10.4 (Tiger).. Funny we all thought Logic sounded just fine, and look at how much noise and distortion was in those converted mix downs we sent off to be pressed to cd. Just convert it and listen to it, you more than likely dont have the supersonic robot ears needed to hear zipper noise at 24000hz.
Old 1 week ago
  #18
Quote:
Originally Posted by spenny23 View Post
Hey all just got a MPC1000 and i have a lot of loops/songs/samples i want to load on them, but they were exported out of ableton at 32 bit (44.1 khz). The mpc1000 only takes 16 bit (44.1 khz). Whats the best, most lossless way for me to convert (software)? trying to maintain my original sound / quality as much as possible
http://sox.sourceforge.net/

Here is how to install it in Linux:

Code:
[email protected][21:11]: sudo apt install sox
To convert whole folder full of 32bit waves first create another folder in it called out (where output will go), then type this:

Code:
for x in ./*.wav; do sox −−norm "$x" −b 16 "out/$x" dither −s; done
This will normalize the sample first. Then it will apply shaped dither and save it as 16 bit.

Hey! You asked for the best!!
Old 1 week ago
  #19
Gear Maniac
 
Qliphoth's Avatar
i noticed, that these young MPC users of today often ask rather "stupid" questions. it's almost funny
Old 1 week ago
  #20
There are no stupid questions.
Old 1 week ago
  #21
Some of the folks on the actual Akai support forums worry me for the future hehe.

Hey Don, that sox command you typed... As far as renaming the converted files, does that just overwrite the old ones as is or???
Thats why I stuck with AIMP, I couldn't figure out exactly what to tell sox to do for a batch haha. If I can do it with sox itself I can skip AIMP now too.
Basically I just want it to convert a batch of files to the same bitrate/samplerate and overwrite the thing instead of making a new file. Saves the tedious cleanup/renaming step afterwards on a 80+ GB library heh. Ill go look up setting the normalization stuff. I like it to hit -3db, not go all the way to 0.

EDIT: Hmm cant get it to work.. Just tells me "sox FAIL formats: can't open input file `16': No such file or directory" when I copy and paste it and run it in a directory I just made.
Old 1 week ago
  #22
Lives for gear
 
Jamie munro's Avatar
Quote:
Originally Posted by Qliphoth View Post
i noticed, that these young MPC users of today often ask rather "stupid" questions. it's almost funny
a question is never stupid, it is a search for information

stupid could be applied to those that prefer to insult or withhold information from the person asking the question while offering nothing of any use
Old 1 week ago
  #23
Lives for gear
 
chaocrator's Avatar
stupid questions are better than no questions.
i'm too lazy to explain why
Old 1 week ago
  #24
Quote:
Originally Posted by Monotremata View Post
EDIT: Hmm cant get it to work.. Just tells me "sox FAIL formats: can't open input file `16': No such file or directory" when I copy and paste it and run it in a directory I just made.
Most likely -b switch has to be elsewhere.

http://sox.sourceforge.net/sox.html
Old 1 week ago
  #25
Ok so stupid questions continuing hahaha.. Finally got that command line to work with a couple catches.. I copied a bunch of files to test on that were 16/44.1 and upsampled them to 24/48.. Command line was this: "for x in ./*.wav; do sox "$x" -b 24 -r 48000 "./48k/$x"; done"

1. You have to make the output folder first, Sox cant create them on its own. It just tells you it cant open the file in the output folder, if the folder isn't already there. Once its created, conversion is smooth as ever. The command as listed above needs one more edit too. The output folder needs ./ in front of it.

2. I want them to overwrite the source file and replace it. When I do batch conversions, I usually do like an entire library at once when I buy it (anywhere from 200MB to what 6GB?). I do not want to have to duplicate all those files and then go back through and delete them all, Id like to just overwrite them once and for all. If I dont specify a separate location for the output files, Sox stops with an EOF error on every wav file and basically kills the whole file and youre left with an 80k file of nothing heh. It has options about "clobbering" the source files and overwriting them but I cant get it to do it here. Only way is to make a separate directory for the converted files. AIMP has a "Remove original file" option and it uses Sox so its gotta be in there somewhere..
Old 1 week ago
  #26
Quote:
Originally Posted by Monotremata View Post
"for x in ./*.wav; do sox "$x" -b 24 -r 48000 "./48k/$x"; done"
Perhaps try this. It gives an extra 100 milliseconds for the safety (if the disk is slow). And then deletes the source:

Code:
for x in ./*.wav; do sox "$x" -b 24 -r 48000 "./48k/$x" && sleep 0.1 && rm $x; done
Of course, backup first. I never tried this. In fact it might delete all your source files.
Old 1 week ago
  #27
Lives for gear
 

Myriad on the Mac is wonderful for batch processing. All that command line stuff is too hard.
Old 1 week ago
  #28
I found Myriad pretty clunky and never could get my head wrapped around it hehe. AudioFinder's batch stuff works for me for the most part, but because of SRC quality its Sox all the way for that one. If you run the test files from src.infinitewave.ca in it and convert their 32bit integer 96k file down to 32bit 44.1k, it doesn't have alot of noise, but it does start to alias at the top of the frequency range. Sox is 99.9% perfect all the way up, just as IzotopeRX, Live, and Logic are nowadays.

Audiofinder can actually use Sox for its conversion, but you still have to know the parameters to enter for it as you would typing it on the command line. I never thought about that, since Audiofinder adds SRBitDepth to a filename anytime it does it anyways, I might as well just use Sox in it since I have to rename everything afterwards. Man I need to stop smoking so much Im getting really slow in my old age hahaha.

Don Ill give that a try and thanks again! Even if I cant get it to work how I want, I got AIMP for it but nonetheless, now I understand how to use it in general so I can when I want to. Im definitely using it next time I convert something in Audiofinder!
Old 1 week ago
  #29
Lives for gear
 
chaocrator's Avatar
Quote:
Originally Posted by HaveItAll View Post
All that command line stuff is too hard.
some people may be surprised, but for those who are familiar with command line stuff, it's faster and very convenient way of doing things.

for example, i never had any GUI archivers or (back in the days) CD/DVD writing apps in my system. moreover, i use a GUI file manager only to transfer files to/from with Android devices, because Midnight Commander does not support MTP.
Old 1 week ago
  #30
Gear Addict
 
tiger001's Avatar
i use sox for those rare convert job, like converting to 12 or 8 bit and 32 or 24.88 Khz (linndrum/SP12/00 territory)
i use wavelab for every 'normal 'conversion
does batch job & has tick box for delete/overwrite original file
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