How To Normalize A batch of WAV files

Pardon me if this was mentioned before but I am looking for a piece of SW (preferably freeware) that will allow me to process a batch of wav files (extracted from various CDs) simultaneously in order to make a compilation disc. The software should allow me to normalize all tracks to the same volume level and not normalize one wav file by itself.

Any insight and help will be greatly appreciated. I tried Audacity but could not accomplish this because either this feature was not there or the user interface too confusing for newbies like myself.


I haven’t tried the Normalize on either of these software but both have the option to do it.
EAC or MediaCoder.
I also gave Audacity a try at this .I just did a couple of .wav files but it did them both at the same time.
With Audacity just open the first wav file.
Then Project/Import Audio Import the rest of the wav files as a batch.
File/Export Multiple
Be sure to select a different folder when Exporting & make sure to select wav.
If you don’t use a differet folder it will Overwrite your current files or if you uncheck Overwrite it will put a second set of #2 files in the original folder.

Thank you for the instructions on Audacity. I will try it when I get home from work this evening. BTW, do you notice any compression after Audacity normalized the batch of wav files?

Thank you and have a great weekend!

I was curious myself about the compression so I did a whole album I had on the HD as .wav files to see.No compression the folders were the same size.
The instructions are for Audacity version 1.2.6
Also for .wav files you can burn to a CD -R make sure the
Project Rate(HZ) in the lower left corner is set to 44100.
I also have Audacity 1.3.4-beta(Unicode) the instructions for it are a little different.
There is no Project/Import Audio instead this version uses:
The newest version is 1.3.7 & it’s a beta too I probably won’t update right now & stable 1.2.6 is still available.
If you have a version somewhere in between & can’t figure it out I will do what I can.

You could use foobar2000 to encode to FLAC, perform ReplayGain analysis and then decode back to WAV while applying ReplayGain gain to the audio to achieve the same perceived loudness for each file.

Best of all, foobar2000 is free.

…Audacity is free-er…