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WAV file settings guidance for resampling

Posted on 2008-10-13
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Last Modified: 2013-12-01
I have a large amount of WAV files for download (500GB). I am running out of disk space on our web server and was looking at different solutions. One was to reduce the size of the Wav files by resampling. However I am no expert and was looking for some guidance. Quality must not alter much but there are the options of frequency and bit rate, what do these achieve and can i reduce the file size much? I know you can't hear much difference between 128kbps and 192 kbps MP3's unless you have a great stereo system but i wondered if i can do the same for wav's?
Thank you.
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Question by:suroma
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by:garycase
ID: 22719536
I certainly disagree with your premise about the quality difference between 128kb and 192kb MP3's => 192kb MP3's are FAR better than 128kb.   And ANY MP3 has much lower quality than the original .WAV source files.

But as for your question -- assuming you want to retain the full original quality, your best alternative is to use a lossless compression codec that will reduce the size of the .WAV files by roughly 1/2.   I keep all of my original audio in Monkey's Audio format [http://www.monkeysaudio.com/ ].   This will reduce your storage requirements by roughly 50%.

If you don't care about the quality -- or if you consider MP3 quality "good enough" -- you could simply convert the .WAV's to MP3's ==> this would reduce the space requirements a LOT (by as much as 90%) ... but you will lose a lot of quality and cannot recover it.   [A Monkey's audio file can be restored to the EXACT original .WAV file.]
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by:Merete
ID: 22736569
You could try super. just disable the video aspect
It supports audio and you set the different bitrates and sampling frequencies
convert many at once.
http://www.erightsoft.net/faqt.html
http://www.erightsoft.com/Superdc.html
snapshot of the interface so you can see how it looks

Super.jpg
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by:suroma
ID: 22737279
Thank you for your answers so far. I may have not explained what I needed to know correctly. I know that changing the frequency and/or bits will reduce the file size but i don't know how they affect the music? So what does changing the frequency from the standard 44100 to 32000 do? Are there certain tracks this works better for etc? Same for bits. What does 8bit do instead of 16bit and what type of tracks can you get away with doing this?
Thank you.
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garycase earned 500 total points
ID: 22738101
Digital signaling requires at least double the sampling rate of the highest frequency you want to reproduce.   So a 44.1kHz sample is used to digitize frequencies up to 22KHz.   If you reduce the signaling frequency, you degrade the signal to only include frequencies up to 1/2 the new sampling frequency ==> so, for example, 32khz would only retain information on sounds up to 16khz.   In addition, due to the nature of sampling, re-sampling will further degrade the signals, since the original waveform is not perfectly recreated (the exception is if you reduce the sampling rate to 1/2, 1/4th, etc. => you'll still reduce the highest frequency info as noted; but the quality isn't reduced).

Changing from 16 bits to 8 bits is NOT something you want to do.   The # of bits determines the quantitization detail contained in the samples ==> 8 bits only allows 256 values;  16 bits allows 65536.   Imagine a light switch with only 2 levels:  on and off;  and another with 256 levels -- essentially a continuously variable dimmer.   That kind of difference (except with regard to sound levels instead of light levels) is the kind of difference you'll get between 8 bit and 16 bit samples.

For a situation where you don't have (or don't want to buy) enough storage to store everything in its original uncompressed form, I would do one of two things:

(a)  If a ~50% reduction in space is good enough, use a lossless code as I noted before.   This will result in NO loss of fidelity (i.e. the sound is identical to the original .WAV file, since the original .WAV is actually reconstructed on playback) ... and will take about 1/2 the storage space of the original files.

or

(b)  simply convert everything to MP3's at a bit rate that retains good fidelity.   I'd use a good variable bit rate code [Lame works well & is free] with a max bit rate of 256 or 320 (these result in excellent quality for MP3's) ... but 192 works fine if you're not listening on a high quality stereo.
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Author Closing Comment

by:suroma
ID: 31505621
Thank you exactly what i wanted to know.
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