I just converted a bunch of low-quality WAVs (they were from old recordings from the 20s), around 25 bps I think. When they were converted into high-quality mp3s, they came out 2.5 times BIGGER than the originals.

Any idea why this might be? I couldn't find this discussed anywhere else.

In general, will converting a low-quality WAV to a high-quality MP3 (192kpbs VBR in this case) produce an MP3 as good as the original WAV?

- Tim

Well, you should check out what is the bit-depth and frequency of original. I bet it was 8bit 22050 or 11000 khz wavs and you converted them into 44100 16bit mp3's.
Sound? Well, they can't sound much worse, can they?

H.

Too late to know now! But they were definitely 8-bit.
I was just curious about the technicalities involved. I guess it's just
the storage... 16 bits take up twice the space, etc. I thought there might be
some other reason. :-)



They COULD... they could get all echoey and wow-y the way some bad mp3s do. That was my worry.

Thanks for the help, tho'!

- Tim

Did you delete the source files?

WAV is just a container, so your WAV file might be an MP3 or some other lossy format wrapped with RIFF header.

I'm guessing you set the bitrate of the MP3 to be 2.5x the bitrate of the source WAV file.



No, and that would be fairly silly. If your source material is is not 44.1KHz, then then typical bitrates do not make much sense. I would probably use a lossless codec since file sizes are already so low, and it sounds like you are archieving. Otherwise, you'll need to look into what settings should be used for 11khz recording (or whatever you're using) and probably do some testing.

i dunno why no one else said this... but

you probably don't want to transcode/encode these at all unless there is some sort of compatibility issue with playback - but if you do, you probably want to use something else besides mp3 for this low of bitrate/samplerate

something like speex, aotuv vorbis, he-aac, etc.


later

Yes, I already deleted the sources. They weren't that important to me, so I didn't mind if they didn't sound as good.

The reason I wanted to do it was to have everything that I wasn't keeping for archive purposes (eg. my CD collection) in MP3 format.

Also, I'm using MediaMonkey to manage my audio files, and it got messed up and started writing erroneous album/artist info into files where album and/or artist was blank (it's database was corrupted -- the row for blank artists was set to a non-blank value), and I wanted to clear the fields. For MP3s were easy to clear, but MediaMonkey seemed to be storing some Artist/Album info in nonstandard places in my WAV files, as all the WAV properties editors I tried on it couldn't find this one bit of tag info that was left in the files after the mishap.

I found that converting from WAV->WAV seemed to get rid of this erroneous info, or WAV->FLAC.

- Tim

No such thing as a 16-bit mp3... mp3s have no bit depth.