How to use a the Studio FF Compressor as a De-esser?

No, this means that the compressor will only react to the frequencies in that range. As far as I know it is not possible to deess in the Axe Fx. Only wideband compression is possible.
 
No, this means that the compressor will only react to the frequencies in that range. As far as I know it is not possible to deess in the Axe Fx. Only wideband compression is possible.
so you mean if the audio spikes in the 5-7k range , it'll compress the whole 20-20khz range? just to make sure
 
https://forum.fractalaudio.com/threads/deesser.147303/post-1743145

How would I go about using the EQ settings? Is it like this since I (Men) usually around the 5-7k range?View attachment 95736

Does this mean the compressor will only compress around th 5-7k range I have set it at?
Have you tried the multi-band compressor? Seems like that would make the most sense? Maybe use an EQ in Solo to find the frequency you want to De-Ess, and then use the multi-band to duck it however much you need.
 
Many de-essers do wideband compression based on a narrow band sidechain, such as you show above.

More advanced de-essers do limited band compression, but then the sidechain band and the compression band should be decoupled, and that is not possible with our MBC block.
 
Many de-essers do wideband compression based on a narrow band sidechain, such as you show above.

More advanced de-essers do limited band compression, but then the sidechain band and the compression band should be decoupled, and that is not possible with our MBC block.
yeah that's probably the term I'm looking for , limited band compression , if so then that kind of sucks that the Axe fx doesn't let us do that @FractalAudio is there any possible solution to just compressing a specific range of frequency
 
If you’ll share what your scenario is I can probably give you a valid workaround that will allow you to compress/limit just the offending HF spikes.
 
If you’ll share what your scenario is I can probably give you a valid workaround that will allow you to compress/limit just the offending HF spikes.
I’m automating vocal processing (EQ>Compressor>Deesser) using the Axe Fx that sends the audio to a Camera so I don’t have to edit the audio in post


The crossover frequency in the multiband compressor block limits you to having to use double the frequency value if you try to compress the 5khz to 7khz range

The mid crossover freq I choose is 5khz then the high crossover frequency range automatically changes to 10khz , if I lower the high crossover freq then the mid crossover freq lowers with it

TL: DR Basically just need to compress only a certain range of frequencies , in my case 5khz to 7khz
 
Basically the Andy Sneap Compressor trick for taming the palm muted guitars of the 80-250 hz range ….but in the higher frequency ranges for vocal sibilant consonants (s,sh,ch,z,t…)
 
Actually, what you really want is to detect in the 5-7k range and compress the entire high frequency range from 3-5k and all the way up. One way would be to use the MBC and just use the high band with a cross-over at 4k. You could also just compress the 4-8k band.

Just using the normal FF studio compressor with a narrow detection band will also be just fine. Many many de-essers work that way. I would go this way.

If you want to go ballistic, you can split the frequency band using filters and then insert a wideband compressor on the high frequencies (still with narrow detection band). Then you would recombine the frequencies afterwards. But I think, this is way more complication than you need.
 
I think this is a silly subject/thread.
The Axe-Fx is a 'GUITAR' processor..
Last time I looked, guitars do not nor did they ever require De-essing.
just my 2cents.
 
I think this is a silly subject/thread.
The Axe-Fx is a 'GUITAR' processor..
Last time I looked, guitars do not nor did they ever require De-essing.
just my 2cents.
You’d be surprised the amount of people who use it otherwise

The process I’m talking about can be used for the guitar aswell to dynamically reduce a frequency range

Also I clearly said in the post I’ll be using for my voice
 
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Actually, what you really want is to detect in the 5-7k range and compress the entire high frequency range from 3-5k and all the way up. One way would be to use the MBC and just use the high band with a cross-over at 4k. You could also just compress the 4-8k band.

Just using the normal FF studio compressor with a narrow detection band will also be just fine. Many many de-essers work that way. I would go this way.

If you want to go ballistic, you can split the frequency band using filters and then insert a wideband compressor on the high frequencies (still with narrow detection band). Then you would recombine the frequencies afterwards. But I think, this is way more complication than you need.
Compressing 4-8k might be a bit much but can’t hurt to try

I did think of your ballistic method and might try it out but that whole combining of frequencies will probably introduce artefacts I don’t want

I appreciate the help and thank you for your time
 
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Let me make it a little clearer for you...
MY point is...it seems unreasonable for anyone to 'EXPECT 'something that is clearly designed for guitar, to assume the job of this type of vocal processing...
This is not a one size fits all machine... Might I suggest acquiring a de-essor.
Nuff said.
 
Let me make it a little clearer for you...
MY point is...it seems unreasonable for anyone to 'EXPECT 'something that is clearly designed for guitar, to assume the job of this type of vocal processing...
This is not a one size fits all machine... Might I suggest acquiring a de-essor.
Nuff said.
A digital signal processor is a DSP is a DSP, regardless of what you may choose to use it for. The power inside the FX3 surpasses a large number of "dedicated" audio processors , and it's effects processing has a wide variety of sonic applications regardless of the source.

If, as you previously posted, you think this thread is "silly" then WTF are you doing here? Move on.
 
I’m automating vocal processing (EQ>Compressor>Deesser) using the Axe Fx that sends the audio to a Camera so I don’t have to edit the audio in post


The crossover frequency in the multiband compressor block limits you to having to use double the frequency value if you try to compress the 5khz to 7khz range

The mid crossover freq I choose is 5khz then the high crossover frequency range automatically changes to 10khz , if I lower the high crossover freq then the mid crossover freq lowers with it

TL: DR Basically just need to compress only a certain range of frequencies , in my case 5khz to 7khz
Sorry for the delay in replying. – super busy day.

What you want to consider doing is what us Old Sound Guys refer to as parallel (or "split") processing – i.e. set up two (or more) separate processing "pathways" for a single source (so you can deal with them as separate signals as required) and then mix them back together. In your case (wanting a de-esser that operates in the 5-7kHz range on a voice source) you need to take your vocal source and split it into two "rows" on your AFX – let's call them "Path A" and "Path B". Once you've set these up, you'll want to:

1) Use your EQ block of choice (mine would be PEQ parametric) to reduce the amount of the offending frequency band (as you've already identified, 5-7kHz) in Path A. I'm guessing about 12-15dB of cut should be sufficient, maybe even less. You don't need to completely get rid of it, just reduce the 5-7kHz down to a level where the peaks are well below being offensive. This will mean, of course, that the overall Path A signal is now too "dark" to be used by itself as-is.

2) Use another instance of PEQ block on Path B, but set it so that it reduces everything EXCEPT the 5-7kHz range you're reducing in Path A. This time I'd set up a lo-cut shelf at 5kHz and a hi-cut shelf at 7kHz – your results may vary.

3) Put a compressor/limiter processing block after the EQ on Path B and set if for fairly fast attack/release (since it's dealing only with HF) and adjust the threshold as needed to even out the response of the 5-7kHz range.

4) Mix Path A and Path B back together, adjusting their levels to get the amount of HF you wish – only now without the signal peaks that you want to get rid of. Always keep in mind that you can fiddle about with all of the settings while listening to either or both of the Paths, as well as monkeying around with the relative levels of same.

5) Obviously, using Snapshots so you can compare differently settings of all of this can be pretty useful to determine the best overall result. I'd also be inclined to play around with using the different types of filters in the PEQ block: "BLOCK" would be tempting, but the "brick-wall" characteristics might be a little harsh, while the SHELVING/PEAKING filters and their wider bandwidths might sound better. Do share and let us know what you find works best.

Voila! You've just created a (programmable, no less!) de-esser the same way we used to do it in the days before plug-ins when we didn't have a dedicated de-esser in the rack but did have an extra comp/limiter and the board's channel EQs. As an aside, we'd typically use something like an 1176 with a pretty stiff ratio (20:1) on the HF side of the chain and a dbx160 "vca" compressor (set with low ratio and a low threshold so it's pretty much "on" all the time) on the post-mix signal.

FWIW, this same concept of channel splits can be used to get a powerful tight drum sound (via strong-ish overall compression to create a solid "base" mix level) without losing the dynamics and "life" that more lightly-compressed parallel channel processing can provide.

Hope this helps! Now that I've said all this, I would strongly encourage you to solve the problem where it's starting – mic selection and placement are the usual culprits.
 
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I should have added that, because you're doing all of this digitally, you may experience some latency-based sonic side effects (comb filtering, "hollow" sound, etc.) – especially if you have more processing blocks in Path A vs. Path B. You may need to adjust the timing of the two paths to get the cleanest final mix of paths.
 
Sorry for the delay in replying. – super busy day.

What you want to consider doing is what us Old Sound Guys refer to as parallel (or "split") processing – i.e. set up two (or more) separate processing "pathways" for a single source (so you can deal with them as separate signals as required) and then mix them back together. In your case (wanting a de-esser that operates in the 5-7kHz range on a voice source) you need to take your vocal source and split it into two "rows" on your AFX – let's call them "Path A" and "Path B". Once you've set these up, you'll want to:

1) Use your EQ block of choice (mine would be PEQ parametric) to reduce the amount of the offending frequency band (as you've already identified, 5-7kHz) in Path A. I'm guessing about 12-15dB of cut should be sufficient, maybe even less. You don't need to completely get rid of it, just reduce the 5-7kHz down to a level where the peaks are well below being offensive. This will mean, of course, that the overall Path A signal is now too "dark" to be used by itself as-is.

2) Use another instance of PEQ block on Path B, but set it so that it reduces everything EXCEPT the 5-7kHz range you're reducing in Path A. This time I'd set up a lo-cut shelf at 5kHz and a hi-cut shelf at 7kHz – your results may vary.

3) Put a compressor/limiter processing block after the EQ on Path B and set if for fairly fast attack/release (since it's dealing only with HF) and adjust the threshold as needed to even out the response of the 5-7kHz range.

4) Mix Path A and Path B back together, adjusting their levels to get the amount of HF you wish – only now without the signal peaks that you want to get rid of. Always keep in mind that you can fiddle about with all of the settings while listening to either or both of the Paths, as well as monkeying around with the relative levels of same.

5) Obviously, using Snapshots so you can compare differently settings of all of this can be pretty useful to determine the best overall result. I'd also be inclined to play around with using the different types of filters in the PEQ block: "BLOCK" would be tempting, but the "brick-wall" characteristics might be a little harsh, while the SHELVING/PEAKING filters and their wider bandwidths might sound better. Do share and let us know what you find works best.

Voila! You've just created a (programmable, no less!) de-esser the same way we used to do it in the days before plug-ins when we didn't have a dedicated de-esser in the rack but did have an extra comp/limiter and the board's channel EQs. As an aside, we'd typically use something like an 1176 with a pretty stiff ratio (20:1) on the HF side of the chain and a dbx160 "vca" compressor (set with low ratio and a low threshold so it's pretty much "on" all the time) on the post-mix signal.

FWIW, this same concept of channel splits can be used to get a powerful tight drum sound (via strong-ish overall compression to create a solid "base" mix level) without losing the dynamics and "life" that more lightly-compressed parallel channel processing can provide.

Hope this helps! Now that I've said all this, I would strongly encourage you to solve the problem where it's starting – mic selection and placement are the usual culprits.
Damn you the real MVP for writing this out , I’ll try this out when I can !

Much appreciated and thanks a million for your time and sharing knowledge on the subject!
 
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