How To Cleanup Distorted Audio Using Audacity?

Welp, this 1 ought to be interesting:

I’m curious to know if there’s any guides out there on how to cleanup distorted audio (not remove audio) using Audacity? :thinking: I made a post on the LTT forum about this topic & someone there suggested Audacity even though I’ve never used it before. In case anyone is wondering, I’m asking because I’m still working on that personal project of mine & I noticed that with some of the tracks that I had to convert, there’s a tiny bit of distortion to the audio. (To be clear on something: these tracks aren’t my own; I got permission from each artist/Youtube channel before downloading them. Even though I can hear each individual track just fine.) So if anyone knows anything or can point me out to a guide, that would be greatly appreciated! :smiley:

If anyone happens to need more details, or wishes to ask me anything additional, feel free to do so & I’ll do my best to fill you in! :slight_smile: Thanks in advance!

Distortion can only kind off be fixed after recording.

There should be a clipping removal tool somewhere, let that run over the recording.
Edit: Highlight the section (or entire track) → Effects → Clip Fix (towards the bottom)

I see what you’re saying, but I forgot to mention in my post that these tracks I acquired aren’t my works because I’m not a musician; they’re actually for a project I’m currently working on. I will edit the post just in case anyone else decides to comment with something helpful…

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The quick and dirty way I do it is to precisely highlight the segment of the audio where the noise is and just turn down the volume on that particular segment. It will be tedious if there are multiple distortions particularly if it happens every few seconds, but for occasional noise, chirps and scratches, thats how I do it.

If the audio is already mixed, there is not much that you will be able to do to fix the distortion with out effecting everything around it. If you can find the frequencies that the distortion lives on, you can attack those specific frequencies for the duration of the problem, but it will also affect anything else that lives on those frequencies.

I am not familiar with the projects but I assume this is dealing with already mixed music. Is this distortion like in the musical sense or are we just talking about audio artifacts?

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It’s been a long time since I’ve used Audacity, but I can speak to what I would do in Logic (a similar if not slightly more robust Digital Audio Workstation DAW). Generally when you hear distortion, it’s a signal clipping (jargon for exceeding the signal strength of the medium). If you think about audio, it’s a wave (think of a sine wave for some intuition). Distortion typically occurs when the troughs and peaks of a wave exceed the allowable range of the medium. This causes the medium to turn your perfectly curved signal to smash into a square. Literally, the peak and trough of the wave gets forced into a square as it’s run out of room the medium can provide and generates a sound that’s generally unpleasant to the listener.

While I’m sure there are many places, the two main places you introduce distortion are on input and in the recording process. On the input, your signal might exceed the limits of the source device (microphone, input jack, sound card, etc), as well as an intermediate device in that signal chain (pre-amp, or some signal processing device). These input devices typically have a signal limit and once exceed, typically will simply truncate your wave to a square when it exceeds input limits. Unless you’re using good quality input devices, it’s entirely possible the act of recording the signal introduced distortion. Even with the best equipment, you can still introduce distortion, it’s just a bit harder as the input devices will typically have higher input ceilings. In this case, you can’t ever really remove the distortion. As it’s baked into the recording itself. You can only take steps to reduce how audible it is. The solutions here would involve what other folks are talking about: trimming volumes (check out ducking and side chaining), applying compression or eq either globally or at certain frequencies at moments of high distortion. You can do this by hand and it’s super tedious. I’m not sure about Audacity, but there are typically plugins for DAWs that will do this semi-automatically and reduce the tedium greatly (Izotope Audio Repair is considered the industry standard for this sort of thing).

You can also use approaches like side chaining, where you copy the track twice. The first track will be heard, the second won’t. But you can use the audio from the second track (particularly when it exceeds some threshold), and apply a volume reduction or compression to the first track. Typically this is used in professional audio and the reason you can always hear a kick drum in a mix which you likely wouldn’t hear if you listened to the music live (aside, the reason you hear a kick drum at a concert is they simply mic and amplify that kick multiple times louder so it’s louder than everything else). In recordings, the kick drum is almost always side chained so the instant the kick drum hits, the volume of most other instruments is lowered momentarily to make space for it.

The second area distortion is most often seen is within the daw. If you take a signal and max out the gain/volume, your signal will clip in the DAW itself. In these cases, the easiest thing to do is turn the volume or gain down and see if that removes the distortion. If it does, you lucked out and have a clean signal. Solving this problem will create a second, your audio will now be too quiet. Fortunately, there are audio engineering tricks around this. Typically, you will run your signal through processing like a limiter or compressor. The compressor will “normalize” the signal by lowering the high volume and raising the low volumes. This will make the audio signal more consistent and allow you to raise the volume again without “clipping”. Alternatively, you may look into a “limiter” which only compresses signals when a signal exceeds a limit. At extreme settings, the compressor and limiter will both make a wave a square and recreate that nasty distortion sound. However, at more moderate settings they will perform a ratio reduction (for example in the case of the limiter, reduce a signal 4 decibels for every decibel over some thresholds), this will, ideally, cause your signal not to hit the “ceiling” and turn square, but not flatten the peaks so much they become “square” due to signal processing either.

Hopefully this was helpful and gave you the tools to get yourself unstuck.


That is some seriously good audio engineering explanation. I actually felt I learned something reading this. Well done!


if its the conversion process thats causing the distortion. try a different mp3/raw converter.
use a 24bit bitrate 48khz max.

although 48khz seems low, if you use 96khz or higher you just add electrical noise into the signal thats beyond our ability to hear, but does add to the file size and reduces overall sound quality.

24bit as thats the typical bitrate of digital amps and audio interfaces that dont cost silly money.

most of the time though i just convert weba’s via a website, and get an mp3 back.

WEBA to MP3 | CloudConvert

^^ works well enough for me.

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