I have this soundtrack (courtesy of bishop743). The point is, it contains almost all music that is written by Visual Music and James Hannigan, but it totally misses out these bar themes, written by Andrew Sega.

Additional Spam:
I'm not trying to return files to they're original state, I'm trying to make the music a bit more listenable. Just for the sake of musical experience (:

In that case I sympathise with your plight. I still say there's nothing significant you can do, but I do sympathise.

I'm sorry, but no, the sample's still clipping, you need to cut it alot more than that. Check Parjay's sample.

If I to cut it a bit more, there will be not much of a music left. Couple of clicks here and there are okay by me (:

Anyway, it's the way original file was and I really don't want to mess with that.

So what's the point of restoring then? Good luck not messing with it if you want to restore it.

I work with audio all the time on a daily basis as part of my work, and Im not new to doing remastering and restoring. Maybe I can help you out?

Can any of you guys briefly explain to me how you restore an audio file like this? I figure you guys use Adobe Audition, but what exactly is it you do you remove clipping and such?

While there are ways to reduce the static sound that results from a clipped waveform, there is no true way to recover the lost information. It's no different than an overexposed photograph: you can darken the while areas, but you will never recover the textural detail; the area can only become a gray mess.

Looking at the waveform, I am fairly disgusted. It reminds me of Californication. A frequency analysis reveals that all frequencies above 10kHz are nonexistent. (Of course, considering the sampling rate). I didn't check the RMS of the file; I assume it must be very high. Tsk tsk! There's not much that can be done here besides basic clip restoration.

music_bar_ku04.flac

I could apply some EQ, a little dehiss and rumble filters and -- if you wanted to -- generate some of the missing frequencies. But it takes many hours.

You can't truly create information that was never there in the first place. This is especially true for complex sounds such as transients and instrument overtones. In this file, the upper frequencies simply do not exist. What would you generate them from? (I'm actually curious here.)

Any way, I don't think the file needs any more filters applied to it than already have been. It's had enough, I think.

I have a piece of software that does it, but it takes forever and its expensive as hell lol.

But what I'm wondering is, how does this software work? How does it "know" what kind of information to add in? For example, let's say we have a 22050Hz waveform audio file of a violin playing. At this sampling rate, frequencies above 11025Hz cannot be reproduced because the sampling frequency was lower than the frequency of the sounds in question. The timbre of a violin is very complex, especially in the upper treble frequencies. Would this software be able to determine the nature of the sound in question and add in the proper harmonic content? How does it determine this? What is it working from? Wouldn't any high frequencies generated by this program only be extrapolations based on the available data?

I'm not trying to be argumentative here, really. I guess I'm just a skeptic at heart. I have no reason to believe that you're lying, but I just can't see how such a program would (or could) produce realistic results.

Ask the people who made the Creative Sound Bullshit X-Fi. It has the same "Restore your 128k mp3s to studio quality with our 24bit crystalizer!!!1!" bullshit.

It's impossible to restore audio this way, no more then you can make a 200x200 px image wallpaper sized. You can increase the percieved quality with filtering and whatnot, but thats only an approximate and may sound (or look) completely different then what it originally was.

I'm kinda wondering why this topic is still open.