My input settings were 48khz, 24 bit PCM, Stereo (just as recorded) and Little Endian. I have tried values 0, 1 and 2 (even 3) in the "start byte offset" space.

What I get is an almost 4-hour long audio file with 85% ultra-saturated sound at 0db (or almost) and the other 15% are chunks of good audio with normal levels. Where levels are just a little below 0db I can tell those are takes that I recorded just fine--Apart from this one take I'm trying to recover, I can play all other takes on my computer as wav files and they all have normal levels.

When I change the value to 0, 1 or 2 the chunks that sound good are all different, but still, together they comprise about 15% of the entire file. When I changed the value to 3 I got a split second file.

Anyone know a solution? Thanks in advance.

PS: I have also tried the chkdsk solution to recover the 0 bytes file, but the .chk files are not showing up on my drive event though the "show hidden files" option is checked.

WAVI AutoMUX lets you choose directories which contain WAV and AVI files- and it will mux each wav into an avi with matching name.

I coded this some days ago to deal with the tedious muxing of mute AVIs which are created by Resolves processing of DNG sequences, which were recorded with a separate WAV file, like in most DNG recording cameras like Blackmagic, Sony FS700 or Magic Lantern. (at the moment just WAV/AVI, other formats might be added by request)

you can get it at: http://www.fatpigtures.com/wavi-automux.html you will need ffmpeg in its directory.

Let me know what you think.