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Thread: What are your favorite features?

  1. #41
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    Quote Originally Posted by spantini View Post
    It's improving mine. Playing live, I put more emotion into my strumming and just bang away sometimes.
    Sounds good live Sounds like crap recording

    Yeah, definitely. Turning the gain up on a sensitive mic can get ugly. Last night I actually had an idea to take a patch of flannel or denim and affix it somehow to the pickguard, with a bit tucked into the edge of the soundhole.
    Just hold the pick closer to the tip.
    Mike B My new album on CD Baby: Fact and Fiction
    My Bandcamp site: http://mikebirchmusic.bandcamp.com

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  3. #42
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    I hold it that way when playing bass because I use a very light grip which causes the pick to flip out when held otherwise. I've tried this playing acoustic and my forefinger takes the hit with the nail to the knuckle scraping the strings. I have a "normal" grip for acoustic strumming, and staying aware of the situation while playing seems to be the answer, but if I drift off for a second the problem can return. It's not a habit yet.

    Normal Grip

    pick.jpg
    Failure - - the path of least persistence
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  4. #43
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    Gain staging and clipping

    The topic of gain staging has been popping up here lately and I've been reading through some older threads and other material and Kenny's videos - these are interesting. The second one on Gain Staging actually comes before the Clipping video and helps explain what's going on.



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  5. #44
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    So I appreciate and agree with what Kenny’s saying there, but...

    In that gain staging video he keeps saying things like “it’s distorting here, but we’re fixing it over here.” That’s not actually true. If it was actually distorting at one stage, then turning it down further down the line wouldn’t fix anything, it would just turn down the distorted signal. The real point that he’s making is that it’s NOT distorting at the track, FX, or bus. It’s going over 0dbFS at that point, but the internal processing can handle that. It won’t actually distort at all until you try to push it out your DAC or render it to a fixed point file format. 0dbFS is literally defined to be as loud as a fixed point file (and your DAC) can get, but until you actually try to do that, it can get a hell of a lot louder without distorting.

    If you were to look under the hood of that JS Volume plugin, you’d find that the individual samples are not really “talking” in dB. The actual sample values are represented on a linear scale pretty much like a voltage. In fact, they are exactly a ratio of the voltage that would be generated by the DAC to the maximum voltage that DAC can produce. A sample which would make the meter show 0dbFS will be either -1 or 1, and most normal signals exist between those two extremes. If you add 100db of gain, what the plugin actually does is multiply each sample by 100,000. That’s A LOT, and obviously the result is a heck of a lot more that +-1. But the 64 bit floating point environment can handle numbers that are like millions of billions. 10^15 or so (in both directions from 0) which is like 300dbFS. Fixed point files and your DAC only go up to 1, but Reaper has no problem handling numbers way bigger.

    BUT he chose plugins which do not artificially impose their own limits. Many popular plugins do, though. Real analog hardware has hard limits on how much voltage it can pass, and anything emulating such hardware will also emulate those limits. If you try to push a 100dbFS signal through one of those, it’s going to distort, and trying to turn it down after will definitely not “fix” that distortion.

    It’s perhaps worth mentioning that this works in the other direction, too. That is, fixed point files (and DACs) have a limit on the smallest number (think like absolute value - how close to 0 we can get, not how far negative) they can represent. If a signal is too quiet, the area around the zero crossing starts to kind of get chopped out and we get crossover distortion, sometimes called “quantization” distortion. With 24 or even 16 bit fixed point, that’s still so quiet that it’ll be lost in the analog noise floor and you’ll never hear it. But if you were say to render a very quiet file to 16bit and then try to amplify back up, you’ll definitely start to hear it. In the floating point engine, that’s not a thing either. You can turn down your signal by 100db and then turn it back up and (again as long as no plugins are imposing artificial limits) you won’t have lost any information near those zero crossings. There will be no crossover distortion, and it will null perfectly with the original.

    Now we’ve pretty much been talking about “real-time” playback, but what happens if we do something like Glue items, Apply Track FX, or Render stems? All of these actions create new audio files, and what happens there depends on the format of those files. If those files are going to be fixed point, then they will have limits in both directions. Render too hot, it clips. Render too quiet, and you get crossover distortion. Either way, you’ve completely lost information and there’s nothing you can do to get it back. That distortion is baked in and can’t really be fixed.

    If those files are floating point, though, then it’s again just not an issue, almost as though you never left the floating point mix engine to begin with. You can render a floating point file that peaks at +100dbFS, bring it back in and turn it down 100db, and there will be no distortion. Now for some stupid reason, the default for these kinds of renders is “automatic” which I think uses the (fixed point) resolution of your interface, but it almost seems arbitrary and is never actually ideal. I strongly suggest that you go find that setting (in Project Settings) and change it to a floating point format instead and then save that as your default project settings. (Note that this won’t help any previously saved projects). I personally think 32bit is plenty. This way you don’t have to worry if whatever you render is a little too hot or too quiet. You can just render and know that all the information you need will be there.

    In fact, ANYtime you’re rendering ANything other than your final distribution file it should go to floating point. Well, OK, like test mixes or whatever will have to be fixed point if you’re going to play them on other devices, but the mix file that you’re going to bring back into a mastering session should be floating.

  6. #45
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    That's a lot for me to digest. I'll start off slow..

    So.. Kenny's saying "distortion.. or overloading" is not really distortion, as far as the examples in this video go. Just hot.

    I have a question. So we see the track fader going into the red. And Kenny using this JS:Volume/Pan Smooth plugin to reduce output gain. My question.. is the original signal that's showing red in the track fader still hot at it's point in the chain? Or has the plugin actually lowered it's output gain at that point but the fader meter still shows red (doesn't make sense, I know). From your previous statements I would presume the original signal is still hot - just as you say if it was actual distortion it couldn't be corrected further up the chain by that plugin.

    One more.. you're saying the track's hot, un-distorted signal will not distort until it's rendered. When I render to stereo .wav I frequently see the meter in the render window go into the red, even though the original track and master faders are well below. Is that the rendered distortion? My hearing fails me there - I can't detect distortion at those lower levels (in the render window) where it just slides over the line by 1-3 dB.

    I hope I didn't mangle that too badly.
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  7. #46
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    Quote Originally Posted by spantini View Post
    I have a question. So we see the track fader going into the red. And Kenny using this JS:Volume/Pan Smooth plugin to reduce output gain. My question.. is the original signal that's showing red in the track fader still hot at it's point in the chain? Or has the plugin actually lowered it's output gain at that point but the fader meter still shows red (doesn't make sense, I know). From your previous statements I would presume the original signal is still hot - just as you say if it was actual distortion it couldn't be corrected further up the chain by that plugin.
    He uses ReaEQ to amplify the signal coming from the audio item itself by 100db, multiplying it by 100,000, which causes it to go way over 0dbFS. It is that hot going into the JS Volume, but he sets that plugin to attenuate 100db, dividing by 100,000. Since 100,000/100,000 = 1, this puts it right back to the same level that's coming out of the item. If the fader is still showing red, it's only because he didn't clear the clip indicator. Or else I'm misunderstanding your question. It is really hot between ReaEQ and JS Volume, but is normal before ReaEQ and after JS Volume.

    One more.. you're saying the track's hot, un-distorted signal will not distort until it's rendered. When I render to stereo .wav I frequently see the meter in the render window go into the red, even though the original track and master faders are well below. Is that the rendered distortion? My hearing fails me there - I can't detect distortion at those lower levels (in the render window) where it just slides over the line by 1-3 dB.
    I guess I'd have to see what you're doing, but the render should peak at the same point as the master. Anyway, if that render meter does show red from going over 0dbFS, AND you're rendering to fixed point format, then yes, there is distortion/clipping. Anything above 0dbFS just gets lopped off and lost - just becomes 0dbFS period - and cannot be replaced or fixed.

  8. #47
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    My Project Settings Media shows render @ 24-bit PCM. As soon as I saw that I went off looking.. I found this great explanation for 16-24-32 Bit. The figures on floating point headroom are impressive.

    32-Bit Float Files Explained - Sound Devices
    Failure - - the path of least persistence
    And, uh, oh - hire a decorator to come in here quick, 'cause... DAMN.

  9. #48
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