Converting .wav to .mp3

24kbps @ 22kHz


I’m trying to get wave files converted to mp3 at 24kbps & 22kHz (mono) but the n-track converter doesn’t do it and all the converters I’ve looked at don’t either.

It’s pretty important to get this format as I’m looking to use it for playing files on a flash player on a web-site I’m working on for my wife. Problem is that 11kHz wont work on the flash player (needs to be 22 kHz and above) and 32kbps and above is really too fast for the dial-up connection version.

At present I’m using 32kbps @ 22kHz but it’s going to be pretty off-putting for the dial-up user as the player tends to pause in the early stages of streaming the file and my feeling is that will prove unacceptable.

Does anyone know of a converter that would convert .wav to .mp3 at 24kbps & 22kHz?



AEDTools will do it, but none are free.

I use Advanced Encode Decode Tools most the time.

I’m no expert, but you might check out DbPowerAmp. I use WMA, myself, but i believe it does MP3 too.


I just checked, dbpoweramp won’t go below 32kbps.
Who the heck would listen to such a crappy file anyhow?

dbpoweramp dont have the mp3 option for free, due to license troubles. You can look for Razorlame and lame.exe. They will do a good job for free.



I believe Audacity will create an mp3 at the bit rate you need.

Thank you one and all; much appreciated!


Ok, I’ve found one that will do 24kbps @ 22Khz.

I take your point teryeah that the sound quality is poor but as this particular conversion is for dial-up users I’ll have to live with it - broadband users will get a better one.

My question now relates to mixing the wav file for mp3 conversion. I seem to recall that certain parts of the audio spectrum suffer more than others and I’d like to reduce the “phaser” effect I seem to be getting on the 24kbps version.

Is it the high frequencies that suffer during conversion, I can’t remember?




A rule of thumb is that the absolute minimum sampling frequency has to be at least twice as high as the highest frequency that you want to reproduce in order to prevent aliasing, a very severe type of distortion. (I think that this is called the Nyquist frequency.) Normal hearing extends up to frequencies of around 20kHz – I believe that is why CD’s are recorded using a 44.1 kHz sampling frequency. You may be able to reduce the phaser effect that you are talking about by filtering out frequencies above 10kHz, as this would eliminate aliasing, but would retain frequencies that are necessary for speech intelligibility (1.5 to 7kHz). Obviously this will make music dull-sounding, but I suspect that the trade-off will favor using the filtered sound.


Thanks tspringer,

Much appreciate!


Good tips there Springer.

Also I remember certain sites that do the convertions for
low-fi streams themselves mention what sampling frequency the original file was in before conversion to MP3.
At one point I was using 48000 because that was the highest my given card would record at. Then those files where converted to the highest bit MP3 I could produce. 320 I think? Or think the site required 256?
But when they were streamed at low-fi they warbled. The websites tips said that you should convert the wav. to 44000 first then do the Mp3 convertion to avoid that.
To my recolection it worked.
Most of my files don’t warble at low-fi stream but they ALL seem to be a bit slower and the pitch is different than the original.

Keep shinin’



You seem to have figured this out, but to add my 2 pence worth…

Compression algorithms are designed for a target bitrate. There is no one-size fits all compression. MP3 is designed for bitrates at and above 128Kbps.

Other streaming compression algorithms will be tailored for the low-bitrates you are talking about