how to convert audio file to these parameters?

shmu26

New member
In order to upload audio files to a certain site, I need to meet these requirements:
Audio: 32 Kbps, 22,050 Hz, 16 Bit, Mono, MP3

Usually, my raw files are in WAV or MP3 format.
How to convert them to the required parameters?
Preferably, free software...
 
If it's kbps that is MP3, just extremely low quality. (This is spoken word, right? Music sounds like crap at this rate.)

Bounce and dither your source from 24-bit to 16-bit to start, if it's not already there. I'd keep it WAV format still.

Then, it depends on your DAW. I can bounce out from Logic to a 64 stereo/32k mono and if I set the Stereo out to Mono I get that 32kbps file.

Audacity can *export* to fixed rate mono that low, and it's free.

Not sure what that 22050Hz means. Is that max frequency? You could lowpass it to guarantee that, I suppose.
 
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If it's kbps that is MP3, just extremely low quality. (This is spoken word, right? Music sounds like crap at this rate.)

Bounce and dither your source from 24-bit to 16-bit to start, if it's not already there. I'd keep it WAV format still.

Then, it depends on your DAW. I can bounce out from Logic to a 64 stereo/32k mono and if I set the Stereo out to Mono I get that 32kbps file.

Audacity can *export* to fixed rate mono that low, and it's free.

Not sure what that 22050Hz means. Is that max frequency? You could lowpass it to guarantee that, I suppose.
Yeah, it's lectures, and even for that, it's low quality, but that's the rules...
Thanks for answer.
 
Samplitude does this very nicely.

See attached, first the .wav is converted to 16 bits at 22050 Hz. I would save that to Dtop then plug it back into Sam to do the MP3 smash.

Unfortunately the free Silver version seems to be stopped (but if you can find a download MAGIX will still activate it) but you could download a months demo of Pro X. Magix Music Studio probably has the same converter in it.

Reaper will almost certainly do it but, compared to Sam, I found the process much more "clunky".

Dave.
 

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Samplitude does this very nicely.

See attached, first the .wav is converted to 16 bits at 22050 Hz. I would save that to Dtop then plug it back into Sam to do the MP3 smash.

Unfortunately the free Silver version seems to be stopped (but if you can find a download MAGIX will still activate it) but you could download a months demo of Pro X. Magix Music Studio probably has the same converter in it.

Reaper will almost certainly do it but, compared to Sam, I found the process much more "clunky".

Dave.

Thanks.
The thing I don't understand is the 22050Hz. If file is converted to 16 bits, will it always be at 22050 Hz?
My AIMP audio converter doesn't have a separate parameter for Hz, but it does the rest.
 
Thanks.
The thing I don't understand is the 22050Hz. If file is converted to 16 bits, will it always be at 22050 Hz?
My AIMP audio converter doesn't have a separate parameter for Hz, but it does the rest.

The only reason I can think of for the 20k sample rate requirement is because the audio will be going through an HF limited system and they want to keep out of band signals out. FM cuts off at 15kHz so a 32kHz rate would be ok. AM radio, bugger all past about 4kHz! Then, I seem to think MP3 has little beyond 10kHz "left" anyway?

Even Samplitude does not do a "one shot" 24 bit, 44.1kHz .wav to the 16bit 2050Hz MP3 form AFAICT hence my stepwise procedure.

Dave.
 
The number is the sample rate - we are pushing it higher usually - 44.1, 48, then we doubled it to 96 and then again to silly figures.

The sample rate has an impact on file size too - so a 20KHz sample rate gives a small file size, but still allows the upper limit to be 10KHz, perfectly high enough for clean speech, which rarely goes much over 3.5/4K tops. The one they have chosen makes a lot of sense for lectures - keeping up and downloads speedy yet not compromising quality too much. the bits per second is just a maths product. I just checked Cubase and it happily does 22050Hz 16 bit mono - it's one of the available options.
 
I discovered that I can do what I need with my free AIMP audio converter.
The quality comes out pretty good for these parameters, and the uploader of the website accepts it. :)

Annotation.png
 
The snag is that if you create a 24bit mp3, not all players can replay them. I don't think any of the ones I use can? EDIT - I created a 320k, 44.1K stereo 24 bit mp3, and all the MP3 players I have on the machine will play it - and all the variations I've tried. So it's perfectly feasible to create mp3s in whatever quality format you wish.
 
Right, mp3 encoding imposes its own bit depth as part of the compression. I would think giving it more information to start with lets the algorithm have more to work with when compressing. But that's just speculation.
 
Hah. I totally missed that 22050 was 1/2 of 44.1. Duh.

Yes, there's no need to go from 24-bit to 16 when converting to MP3, especially for the normal case where you're probably going to 256kbps or 320kbps. assuming the original here was even recorded at 24-bit, but I'd be curious to hear if there's a difference going to 16-bit first (with dither) before compressing to such a low bitrate as required here. You'd be compressing some more "noise" added by the dither, but I guess I wonder if the MP3 algorithm is going to be better at picking what to keep vs. toss if presented with a little less data.

Just rambling...
 
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