Stop analog signal from blowing speakers

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zoogoo wrote: so basicly keep the the master volume down on ableton? it just seems that i could have some kind of vst that stops harmful spikes.
This is an issue of gain-staging. If you need a brick wall limiter during tracking, to prevent peaks from blowing up your speakers, you have your iterface's outputs (or speaker inputs) set too loud. I have my modular>DAW>audio Interface setup, so that even if my modular sends out a signal which clips my inputs, the output won't be too loud for my monitors to handle.

The solution to this may be as simple as using a passive attenuator between the output of your audio interface and your speakers. Or you can simply turn the inputs on your speakers down.

After turning your speakers down, if you find that your synth isn't loud enough, check how much headroom you have before the inputs on your audio interface clip. If you still have a ton of headroom, turn up the output from your modular. If you find that you don't have much headroom left at your inputs, then you will need to increase the volume in Ableton Live.

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zoogoo wrote:
dreamkeeper wrote:The only right way is to use an analogue (preferably passive) volume knob between soundcard and monitors, period! If you're controlling volume in the digital domain before the converters, you'll sacrifice resolution (1 bit per 6dB) - plus you'll always run the risk of doing damage IF something goes wrong in software. So ideally you'd use the whole digital range (of course without clipping) and control monitoring volume in the analogue domain.

Though from the OP I get the impression, that input levels should be controlled in this case - which can be done only with an analogue limiter before going into the soundcard. Some interfaces may have inbuilt limiters before AD conversion though.
so basicly keep the the master volume down on ableton?
Me thinks I said the exact opposite (see bold). Let me try again:

Let's say you play a well mastered, reasonably loud song in your DAW and keep all faders at unity, OK? With the full output of your soundcard directly into the monitors, this will be fecking LOUD, right?

Now you turn down your master to reach a good monitoring volume (you'll probably need to turn it down a lot!) and even put a "safety limiter" there. But: there's still a chance that the signal blows up for whatever reason - and you'll end up again with a fecking LOUD blast.

Short excursion: The max output level of your soundcard is limited anyway. It cannot go beyond the equivalent of 0dBFS - at least not much. However a heavily clipped signal is roughly a full scale square wave - which is... yes: fecking LOUD!!!

Back to our reasonably loud song. Put an analogue volume knob between soundcard outputs and monitors, turn it down, leave master fader in your DAW (and all faders in the digital domain, incl. those in the control panel of your soundcard) at unity, play song. Turn volume knob up to get a good, reasonably loud level - LEAVE IT THERE and never worry about any peaks or safety limiters again! Of course you can now also use the digital faders to turn volume further down. But now you can be sure that even the loudest peak cannot damage anything, because the max level is set by the analogue volume knob. HTH
The hole is deeper than the hum of its farts

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zoogoo wrote:
dreamkeeper wrote:The only right way is to use an analogue (preferably passive) volume knob between soundcard and monitors, period! If you're controlling volume in the digital domain before the converters, you'll sacrifice resolution (1 bit per 6dB) - plus you'll always run the risk of doing damage IF something goes wrong in software. So ideally you'd use the whole digital range (of course without clipping) and control monitoring volume in the analogue domain.

Though from the OP I get the impression, that input levels should be controlled in this case - which can be done only with an analogue limiter before going into the soundcard. Some interfaces may have inbuilt limiters before AD conversion though.
so basicly keep the the master volume down on ableton? it just seems that i could have some kind of vst that stops harmful spikes. I like the ICE plugin, but i wasnt able to figure out how to download it. I really dont want to buy anything external to do this right now.

Ableton Live is very safe, it clips the output at +6 dbFS at the Master output, so you can't output more than that from Live.

And you must have reached that +6 dbFS multiple times already (probably several times per day), as a lot of synths and presets are already at 0 dbFS, so just some tracks with those or adding a few FX will easily reach +6 dbFS and clip in Live's Master.

If that +6 dbFS did not blow your speakers already (probably not), then you already know in practice Ableton Live will never blow your speakers.

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zoogoo wrote:
dumbledog wrote:My DAW template has an instance of Fabfilter Pro-L at the end and I just go with that for my own Eurorack stuff.
oh yeah, isnt that just a limiter? its protected your system from spikes?
Yeah. I tend to be careful and run stuff thru a passive attenuators first, and I use an ES-8 for an audio interface so there's lots of headroom.

Honestly my difficulty is often getting things into the DAW loud enough -- with a filter closed well I can be lucky to crack -10dB sometimes. But the limiter's saved my ass a couple times.

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A really simple way would be to build a diode clipping circuit to go between your Eurorack output and your soundcard input. Any signal over a certain voltage threshold (controlled by the diode you use) gets shunted to ground before it gets to your soundcard.

You could also use this as an effect by deliberately exceeding the threshold and thus clipping the signal

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I was gonna say you could build an analog clipper circuit, or hire someone to do it for very little money. You could also build a switch off circuit very easily as well. Iike 10$ in parts or so. Barring that I would recommend just using ice 9 or one of the other ear protection plugins out there.

Outside of all these fixes you may want to reexamine your gain staging through your system. If you have so much gain above normal listening levels that when your synth gets crazy the system has the headroom to do serious damage than you may want to run the input a bit hotter so that these bursts just get clipped. Or maybe turn your speaker amplifier down a bit so that it will clip before it can damage the cones or your ears.

Good luck and be careful. I've got a few friends who have had their careers adversely affected by hearing loss.
Don't F**K with Mr. Zero.

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Ok, so pretty much a brick wall limiter is what i need right? seems like the best solution for me if this is the case. I think i can find some free ones somewhere. Ill look for one, unless others disagree this is what i need.


thanks guys, always a huge help as usual.

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Seems more like you need to understand gain staging properly in order to avoid the problem in the first place. There's plenty of good advice in this thread about how to do that.

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Some people are recommending a brick wall limiter but I still think you would better served by getting your gain staging sorted first.
As others have mentioned, set you speakers and the output of your audio interface to unity gain and then play some commercial (i.e. mastered) music with the media player set to full volume. If that plays back too loud, then turn your speakers down.

Like I said, even if you clip the inputs on your audio interface you shouldn't be at a level where your speakers could be damaged. The output level of your interface and the attentuation of your speakers are what determine maximum loudness. Get your levels right before trying to introduce signal processing to fix an issue with peaks. If you understand gain staging you will never have to worry about destroying your speakers with loud peaks.

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Spot on Justin :tu:

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