Audio Repair

[NightFire]

I have an audio tape that when played the audio fluctuates up and down. I guess the best way to describe this is like the Amplitude Modulations in Sound Forge (Sine would be the closest). Does anybody know a way to fix this so that the captured WAV file doesn't have the up and down volume like this? If a clip helps I'll post one. Thanks

[Soopafresh]

Post a clip

[redwudz]

You are probably looking for 'normalization'. This from Wikipedia: http://en.wikipedia.org/wiki/Audio_normalization

Or you may find a 'leveler' plugin that works similarly. I use Audacity for that. Apparently they call it a 'leveler in the newest versions.

[bourboner]

Normalising doesn't do what the OP is asking. From wiki your link -

Typically normalization increases the amplitude of the audio waveform to the maximum level without introducing any distortion... Specifically, normalization applies a constant amount of gain to the selected region of the recording to bring the highest peak to a target level, usually 98% (-0.3 dB) or 100% (0 dB). This differs from dynamics compression, which applies varying levels of gain over a recording to fit the level within a minimum and maximum range.

Compression might be more useful here, but I wouldn't expect miracles for a fluctuating tape.

[redwudz]

Good point. I haven't tried it with that particular situation. But the idea of the OP posting a sample is still a good idea as we have some very talented audio people here. Worth a try anyway.

[bourboner]

Sample would be good. Post some wav, NightFire, up on sendspace for us!

[NightFire]

Okay here is a 20 second clip.

The first thing I tried was normalizing it in Soundforge but it didn't make much of a difference. I think I will try playing with the Sine plugin to see what results I get. Probably some kind of reverse Sine would work. Anyway if anybody has any ideas it would be appreciated.

[NightFire]

Well I haven't had any luck using Soundforge or Goldwave. I have been trying to map the sine exactly to the wave file but haven't had any luck. Does anybody know if there is a way to map exactly to the amplitude of the wave file? When the amplitude is high the sine will be at bottom and when file amplitude is low the sine will be high. I have the feeling though that it's unrecoverable because the sound goes so low at it's lowest point that 500% db volume gain still doesn't fully help. This is from an audio tape captured from a Nak BX-300, so if anybody has any hardware ideas that would be great too.

[Cornucopia]

Ok,
what I'd do would be:
1st -- Normalize to ~ -6db. Obviously there will be some sections that'll still seem quiet.
2nd -- Apply a LowPass Filter (VERRY LOWW, like ~ 1-5Hz) to a copy of that file. This will be your Amplitude Modulation sine (perhaps MORE complicated waveform than just sine). The good thing is that there really wouldn't likely be any "music" left in that range anyway.
3rd -- Invert the signal. Save a copy. Now the parts that need to be "boosted" will match up with the high portion of the wave at that point, and the boost would be proportional as well.
4th -- Use an amplitude modulator filter that accepts an external signal (using the inverted signal as the "external" source). If you don't specifically have that, you may be able to use a compressor instead (one that also accepts external signal). This would be used as a "ducker". Try the filtered signal on this (synced to play at the same time as the main soundfile), not the inverted one. You'll need to have an app that works in multitrack mode and has side busses (like CoolEdit/Audition or ProTools). Render/save the result.
5th -- Normalize again, this time to -0.1db.
6th -- Apply some levelling (compression + limiting) only at this point.
Done.

Scott

[insearchofanswers1]

I have some audio that is mildly clipping...not enough to cover your ears and run lol but enough to be slightly annoying and tell it wasn't recorded quite right. I have Adobe Audition but I can't seem to figure out how to restore it. Does anyone know how to do this? I appreciate any help. Thanks so much!

[Soopafresh]

There's a free plugin for Audacity that deals with clipped peak reduction. You have to download Audacity AND the LADSPA plugins.

[NightFire]

Ok,
what I'd do would be:
1st -- Normalize to ~ -6db. Obviously there will be some sections that'll still seem quiet.
2nd -- Apply a LowPass Filter (VERRY LOWW, like ~ 1-5Hz) to a copy of that file. This will be your Amplitude Modulation sine (perhaps MORE complicated waveform than just sine). The good thing is that there really wouldn't likely be any "music" left in that range anyway.
3rd -- Invert the signal. Save a copy. Now the parts that need to be "boosted" will match up with the high portion of the wave at that point, and the boost would be proportional as well.
4th -- Use an amplitude modulator filter that accepts an external signal (using the inverted signal as the "external" source). If you don't specifically have that, you may be able to use a compressor instead (one that also accepts external signal). This would be used as a "ducker". Try the filtered signal on this (synced to play at the same time as the main soundfile), not the inverted one. You'll need to have an app that works in multitrack mode and has side busses (like CoolEdit/Audition or ProTools). Render/save the result.
5th -- Normalize again, this time to -0.1db.
6th -- Apply some levelling (compression + limiting) only at this point.
Done.

Scott

Thanks for the help. I tried following the steps but I am getting tripped up at 4. Do you know any plugins that accept external sources for amplitude modulation? Also for the second alternative to 4 can you be more specific or do a little step by step. Also in step 2 should I get a very low amplitude file (not audible)? I am somewhat familiar, beginner-intermediate, with audio editing but I think this maybe beyond my skills. Anyway any more help or elaboration would be appreciated.