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EFFECT to add warmth to DRUMS?

Which do you use?

PO 12 - looking to warm it up and add a little density.

«1

Comments

  • @5pinlink said:
    Cassette deck or reel to reel deck, I overdrive them on to the tape and then sample them.

    Good idea.

    Really good idea

  • You might like Master Record, as well.

    http://audiob.us/get/367/Master-Record

  • TAPE is good. Vintage Warmer in Auria is also GOOD.

  • I'm a big fan of Fabfilter's Saturn, their description says it all:

    http://www.fabfilter.com/products/saturn-multiband-distortion-saturation-plug-in

    Can be found on iOS in Auria, if desktop ain't your thang.

  • I heard a track by @Jocphone a while back and loved the drum sound. He used an amp sim on it. So, what i do now is seperate the drum tracks and feed each one through an amp sim, usually from Cubasis. Can be as subtle or as mental as the songs asks. Sometimes, enough is using the sim but flicking the drive and treble knobs off. In this wee demo of a song I'm working on, (it's for SOTMC so please don't tell em over there) hopefully you can hear the difference in the hats only. The verses have a normal hat clicking away all nice n comfy then in the chorus I've whacked the same hats pattern through the "Axis" amp sim (drive on 2 and treble 5). It distorts them but think you can still hear the rhythm misting through. With some singing on top, hahahaha....... ha, the chorus hats will be less in yer face. If it doesn't, i won't care cos it's all about fun...... as you will hear in my non guitarist guitaring using Guitarism. I've screwed up with the upload, again. Demo is only 3.20 not the 89 hours it plays for. Sorry.

  • edited January 2017

    I like AUM's 'saturation' effect on everything..

    I very often run various things from my full selection of sounds to a bus send which goes through a mono node, low pass (or rolled off high shelf) filter and saturation effect, which is then sent through a bass head and out of a 2x10 cab.

    Makes bass synths and Ruismaker kicks sounds enormous.

    Obviously it's a slightly different process for studio use rather than live, as I'm working with a specific speaker configuration, but I love the sound that this gives me.

  • @1P18 said:
    I'm a big fan of Fabfilter's Saturn, their description says it all:

    http://www.fabfilter.com/products/saturn-multiband-distortion-saturation-plug-in

    Can be found on iOS in Auria, if desktop ain't your thang.

    I have Auria.

    I have the all the synth IAP and pro version, but, never got any of the IAP fab stuff.

    First I gotta open the app

  • @OscarSouth said:
    I like AUM's 'saturation' effect on everything..

    I very often run various things from my full selection of sounds to a bus send which goes through a mono node, low pass (or rolled off high shelf) filter and saturation effect, which is then sent through a bass head and out of a 2x10 cab.

    Makes bass synths and Ruismaker kicks sounds enormous.

    Obviously it's a slightly different process for studio use rather than live, as I'm working with a specific speaker configuration, but I love the sound that this gives me.

    You are a fucking clairvoyant.

    I realized last night I can just do 3 channels for the Audio in AUM.

    I used the Sat along with Haze and a dash of reverb (red one from the comp who makes Viking I forget) with some cool pan / mid arranging

    I forget sometimes when you are not using apps you have much more flexibility with things like this.

    Thanks again

  • I use Blamsoft's 'Zero' reverb myself a lot too actually, has a nice character to it. I like it especially on smooth lead sounds.

  • @RustiK said:

    @OscarSouth said:
    I like AUM's 'saturation' effect on everything..

    I very often run various things from my full selection of sounds to a bus send which goes through a mono node, low pass (or rolled off high shelf) filter and saturation effect, which is then sent through a bass head and out of a 2x10 cab.

    Makes bass synths and Ruismaker kicks sounds enormous.

    Obviously it's a slightly different process for studio use rather than live, as I'm working with a specific speaker configuration, but I love the sound that this gives me.

    You are a fucking clairvoyant.

    I realized last night I can just do 3 channels for the Audio in AUM.

    I used the Sat along with Haze and a dash of reverb (red one from the comp who makes Viking I forget) with some cool pan / mid arranging

    I forget sometimes when you are not using apps you have much more flexibility with things like this.

    Thanks again

    I'd be interested to hear from a 'more knowledgeable person' about how phase would factor into this. I've found that flipping the phase on this bass channel has a favourable effect on the sound, which makes sense according to my rudimentary practical knowledge of phase, but it'd be interesting to hear more from an expert on the topic, regarding how phase comes into play when running a couple of parallel channels (and is it different if running out of the same speaker or different speakers).

  • My favourite for this is Amazing Noises limiter, Moebius lab and virsyn tap delay

  • In Auria, a little MicroWarmer on your drum buss goes a long way.

  • @OscarSouth said:
    I'd be interested to hear from a 'more knowledgeable person' about how phase would factor into this. I've found that flipping the phase on this bass channel has a favourable effect on the sound, which makes sense according to my rudimentary practical knowledge of phase, but it'd be interesting to hear more from an expert on the topic, regarding how phase comes into play when running a couple of parallel channels (and is it different if running out of the same speaker or different speakers).

    No idea if I'm more knowledgeable, and I'm certainly not an expert, but if it sounds good when you play it back in mono, you should be good to go.

  • @eustressor said:
    In Auria, a little MicroWarmer on your drum buss goes a long way.

    Indeed.

  • edited January 2017

    @OscarSouth said
    I'd be interested to hear from a 'more knowledgeable person' about how phase would factor into this. I've found that flipping the phase on this bass channel has a favourable effect on the sound, which makes sense according to my rudimentary practical knowledge of phase, but it'd be interesting to hear more from an expert on the topic, regarding how phase comes into play when running a couple of parallel channels (and is it different if running out of the same speaker or different speakers).

    as already mentioned: no need to be an expert, but trust your ears (listen carefully).
    'Phase' is an annoying term in this context and can be very misleading.
    Technically it means the displacement of a single(!) frequency sine wave in time and the value changes with frequency.

    Since for each frequency a shift of 180 degrees results in an inverterted signal you can in fact yield a 180 degree phase shift for any frequency by simply inverting the signal.
    That's what the so called phase button on a mixer channel is for.
    It's valid only for that specific 180 degree shift.
    Any 'phase value' different from 180 degrees will displace different frequencies (in time) and mix them to a new sound, an effect that's called comb filtering.
    It's often recognized as a 'hollow sound' (sometimes intended, as the famous mid position of a Stratocaster pickups selection switch), but any mixture may occur, depending on frequency content of the signal. There may as well happen a big fat bottom (the opposite) because sone frequencies add.

    But forget about theory and just diy with a record on one track and an identical copy on a 2nd track, which you move slightly in time.
    Ideally you do that in 1 sample steps. On high frequecies (like a hi-hat) you start noticing the difference in sound at roughly 3 samples, a bass needs much more.

    A kickdrum is a good example because it has a rather high frequency click and a deep bottom sound.
    The click (or attack) will be reduced (or smoothed) by shifting the signal in time by small amounts. With increasing values the base wave will become affected, too.
    But: since the wavelength of the high frequency part is much shorter than the bass, it will cycle between extinction and emphasize many times if you do this in single sample steps.

    This is just sound physics, it's by no way a fault or flaw - it's a part of our natural hearing and a fundamental part of any mix. The members of a traditional symphony orchestra are positioned in a way to yield a specific kind of sound. One would hardly recognize the ensemle, if members switched position.
    (there's an obvious coincinence between phase shift and signal runtime)

    Don't try to get too deep into controlling such stuff by it's values, it's way too complex in a real world situation (there's also room reflection...)
    Just know about the mechanism - it makes it a bit easier to understand some stuff you hear (though it may read quite confusing at first glance)

  • @Telefunky said:

    @OscarSouth said
    I'd be interested to hear from a 'more knowledgeable person' about how phase would factor into this. I've found that flipping the phase on this bass channel has a favourable effect on the sound, which makes sense according to my rudimentary practical knowledge of phase, but it'd be interesting to hear more from an expert on the topic, regarding how phase comes into play when running a couple of parallel channels (and is it different if running out of the same speaker or different speakers).

    as already mentioned: no need to be an expert, but trust your ears (listen carefully).
    'Phase' is an annoying term in this context and can be very misleading.
    Technically it means the displacement of a single(!) frequency sine wave in time and the value changes with frequency.

    Since for each frequency a shift of 180 degrees results in an inverterted signal you can in fact yield a 180 degree phase shift for any frequency by simply inverting the signal.
    That's what the so called phase button on a mixer channel is for.
    It's valid only for that specific 180 degree shift.
    Any 'phase value' different from 180 degrees will displace different frequencies (in time) and mix them to a new sound, an effect that's called comb filtering.
    It's often recognized as a 'hollow sound' (sometimes intended, as the famous mid position of a Stratocaster pickups selection switch), but any mixture may occur, depending on frequency content of the signal. There may as well happen a big fat bottom (the opposite) because sone frequencies add.

    But forget about theory and just diy with a record on one track and an identical copy on a 2nd track, which you move slightly in time.
    Ideally you do that in 1 sample steps. On high frequecies (like a hi-hat) you start noticing the difference in sound at roughly 3 samples, a bass needs much more.

    A kickdrum is a good example because it has a rather high frequency click and a deep bottom sound.
    The click (or attack) will be reduced (or smoothed) by shifting the signal in time by small amounts. With increasing values the base wave will become affected, too.
    But: since the wavelength of the high frequency part is much shorter than the bass, it will cycle between extinction and emphasize many times if you do this in single sample steps.

    This is just sound physics, it's by no way a fault or flaw - it's a part of our natural hearing and a fundamental part of any mix. The members of a traditional symphony orchestra are positioned in a way to yield a specific kind of sound. One would hardly recognize the ensemle, if members switched position.
    (there's an obvious coincinence between phase shift and signal runtime)

    Don't try to get too deep into controlling such stuff by it's values, it's way too complex in a real world situation (there's also room reflection...)
    Just know about the mechanism - it makes it a bit easier to understand some stuff you hear (though it may read quite confusing at first glance)

    Thanks very much, very interesting and informative read. Especially loved the analogy about orchestra positioning. As primarily a performer I definitely notice first hand how much tiny things like this can have a huge impact on the sound.

    As you hinted at too, I'm talking about simple 180 phase flips. My understanding (as one of mys producer friends explained a good few years ago!) is that when doubling signals, similar waveforms can cancel each other other out (having a much more profound effect on bass frequencies due to the longer waveform). He recommended that when doubling things I just try flipping the phase while listening carefully to the bass. I find that it normally sounds 'different' rather than obviously wrong, but there's usually a preferable sound between the two options.

    I'm guessing that this is only applicable when everything is coming out of the same speaker(s)?

  • @OscarSouth said:

    @Telefunky said:

    @OscarSouth said
    I'd be interested to hear from a 'more knowledgeable person' about how phase would factor into this. I've found that flipping the phase on this bass channel has a favourable effect on the sound, which makes sense according to my rudimentary practical knowledge of phase, but it'd be interesting to hear more from an expert on the topic, regarding how phase comes into play when running a couple of parallel channels (and is it different if running out of the same speaker or different speakers).

    as already mentioned: no need to be an expert, but trust your ears (listen carefully).
    'Phase' is an annoying term in this context and can be very misleading.
    Technically it means the displacement of a single(!) frequency sine wave in time and the value changes with frequency.

    Since for each frequency a shift of 180 degrees results in an inverterted signal you can in fact yield a 180 degree phase shift for any frequency by simply inverting the signal.
    That's what the so called phase button on a mixer channel is for.
    It's valid only for that specific 180 degree shift.
    Any 'phase value' different from 180 degrees will displace different frequencies (in time) and mix them to a new sound, an effect that's called comb filtering.
    It's often recognized as a 'hollow sound' (sometimes intended, as the famous mid position of a Stratocaster pickups selection switch), but any mixture may occur, depending on frequency content of the signal. There may as well happen a big fat bottom (the opposite) because sone frequencies add.

    But forget about theory and just diy with a record on one track and an identical copy on a 2nd track, which you move slightly in time.
    Ideally you do that in 1 sample steps. On high frequecies (like a hi-hat) you start noticing the difference in sound at roughly 3 samples, a bass needs much more.

    A kickdrum is a good example because it has a rather high frequency click and a deep bottom sound.
    The click (or attack) will be reduced (or smoothed) by shifting the signal in time by small amounts. With increasing values the base wave will become affected, too.
    But: since the wavelength of the high frequency part is much shorter than the bass, it will cycle between extinction and emphasize many times if you do this in single sample steps.

    This is just sound physics, it's by no way a fault or flaw - it's a part of our natural hearing and a fundamental part of any mix. The members of a traditional symphony orchestra are positioned in a way to yield a specific kind of sound. One would hardly recognize the ensemle, if members switched position.
    (there's an obvious coincinence between phase shift and signal runtime)

    Don't try to get too deep into controlling such stuff by it's values, it's way too complex in a real world situation (there's also room reflection...)
    Just know about the mechanism - it makes it a bit easier to understand some stuff you hear (though it may read quite confusing at first glance)

    Yeah some nice explanation in there @Telefunky

    I'm guessing that this is only applicable when everything is coming out of the same speaker(s)?

    I would guess this is not true...perhaps less likely to happen as your own ear position relative to the speakers comes into play...but if you were stood sideways on to a stereo pair of speakers and not moving at all, and your ear receives inverted signals from each speaker at exactly the same time...they would cancel each other out.

  • @AndyPlankton said:

    @OscarSouth said:

    @Telefunky said:

    @OscarSouth said
    I'd be interested to hear from a 'more knowledgeable person' about how phase would factor into this. I've found that flipping the phase on this bass channel has a favourable effect on the sound, which makes sense according to my rudimentary practical knowledge of phase, but it'd be interesting to hear more from an expert on the topic, regarding how phase comes into play when running a couple of parallel channels (and is it different if running out of the same speaker or different speakers).

    as already mentioned: no need to be an expert, but trust your ears (listen carefully).
    'Phase' is an annoying term in this context and can be very misleading.
    Technically it means the displacement of a single(!) frequency sine wave in time and the value changes with frequency.

    Since for each frequency a shift of 180 degrees results in an inverterted signal you can in fact yield a 180 degree phase shift for any frequency by simply inverting the signal.
    That's what the so called phase button on a mixer channel is for.
    It's valid only for that specific 180 degree shift.
    Any 'phase value' different from 180 degrees will displace different frequencies (in time) and mix them to a new sound, an effect that's called comb filtering.
    It's often recognized as a 'hollow sound' (sometimes intended, as the famous mid position of a Stratocaster pickups selection switch), but any mixture may occur, depending on frequency content of the signal. There may as well happen a big fat bottom (the opposite) because sone frequencies add.

    But forget about theory and just diy with a record on one track and an identical copy on a 2nd track, which you move slightly in time.
    Ideally you do that in 1 sample steps. On high frequecies (like a hi-hat) you start noticing the difference in sound at roughly 3 samples, a bass needs much more.

    A kickdrum is a good example because it has a rather high frequency click and a deep bottom sound.
    The click (or attack) will be reduced (or smoothed) by shifting the signal in time by small amounts. With increasing values the base wave will become affected, too.
    But: since the wavelength of the high frequency part is much shorter than the bass, it will cycle between extinction and emphasize many times if you do this in single sample steps.

    This is just sound physics, it's by no way a fault or flaw - it's a part of our natural hearing and a fundamental part of any mix. The members of a traditional symphony orchestra are positioned in a way to yield a specific kind of sound. One would hardly recognize the ensemle, if members switched position.
    (there's an obvious coincinence between phase shift and signal runtime)

    Don't try to get too deep into controlling such stuff by it's values, it's way too complex in a real world situation (there's also room reflection...)
    Just know about the mechanism - it makes it a bit easier to understand some stuff you hear (though it may read quite confusing at first glance)

    Yeah some nice explanation in there @Telefunky

    I'm guessing that this is only applicable when everything is coming out of the same speaker(s)?

    I would guess this is not true...perhaps less likely to happen as your own ear position relative to the speakers comes into play...but if you were stood sideways on to a stereo pair of speakers and not moving at all, and your ear receives inverted signals from each speaker at exactly the same time...they would cancel each other out.

    Ah good to know, thanks. I was curious for a while whether this was an acoustic phenomenon or whether it was connected with the vibration of the cones.

  • @OscarSouth said:

    @AndyPlankton said:

    @OscarSouth said:

    @Telefunky said:

    @OscarSouth said
    I'd be interested to hear from a 'more knowledgeable person' about how phase would factor into this. I've found that flipping the phase on this bass channel has a favourable effect on the sound, which makes sense according to my rudimentary practical knowledge of phase, but it'd be interesting to hear more from an expert on the topic, regarding how phase comes into play when running a couple of parallel channels (and is it different if running out of the same speaker or different speakers).

    as already mentioned: no need to be an expert, but trust your ears (listen carefully).
    'Phase' is an annoying term in this context and can be very misleading.
    Technically it means the displacement of a single(!) frequency sine wave in time and the value changes with frequency.

    Since for each frequency a shift of 180 degrees results in an inverterted signal you can in fact yield a 180 degree phase shift for any frequency by simply inverting the signal.
    That's what the so called phase button on a mixer channel is for.
    It's valid only for that specific 180 degree shift.
    Any 'phase value' different from 180 degrees will displace different frequencies (in time) and mix them to a new sound, an effect that's called comb filtering.
    It's often recognized as a 'hollow sound' (sometimes intended, as the famous mid position of a Stratocaster pickups selection switch), but any mixture may occur, depending on frequency content of the signal. There may as well happen a big fat bottom (the opposite) because sone frequencies add.

    But forget about theory and just diy with a record on one track and an identical copy on a 2nd track, which you move slightly in time.
    Ideally you do that in 1 sample steps. On high frequecies (like a hi-hat) you start noticing the difference in sound at roughly 3 samples, a bass needs much more.

    A kickdrum is a good example because it has a rather high frequency click and a deep bottom sound.
    The click (or attack) will be reduced (or smoothed) by shifting the signal in time by small amounts. With increasing values the base wave will become affected, too.
    But: since the wavelength of the high frequency part is much shorter than the bass, it will cycle between extinction and emphasize many times if you do this in single sample steps.

    This is just sound physics, it's by no way a fault or flaw - it's a part of our natural hearing and a fundamental part of any mix. The members of a traditional symphony orchestra are positioned in a way to yield a specific kind of sound. One would hardly recognize the ensemle, if members switched position.
    (there's an obvious coincinence between phase shift and signal runtime)

    Don't try to get too deep into controlling such stuff by it's values, it's way too complex in a real world situation (there's also room reflection...)
    Just know about the mechanism - it makes it a bit easier to understand some stuff you hear (though it may read quite confusing at first glance)

    Yeah some nice explanation in there @Telefunky

    I'm guessing that this is only applicable when everything is coming out of the same speaker(s)?

    I would guess this is not true...perhaps less likely to happen as your own ear position relative to the speakers comes into play...but if you were stood sideways on to a stereo pair of speakers and not moving at all, and your ear receives inverted signals from each speaker at exactly the same time...they would cancel each other out.

    Ah good to know, thanks. I was curious for a while whether this was an acoustic phenomenon or whether it was connected with the vibration of the cones.

    It is to do with both really, it is all down to the resulting pressure received by your eardrum . That can be affected by the cones (more so in a mono situation, which is why a check in mono is good for detecting phase issues) where the pressure doesn't get generated in the first place, or at your ear in the stereo example, from before, when you get conflicting pressure which cancels itself out.

    If you take a simple sine wave, add together 2 sine waves that are the same and the result is double sized peaks and troughs
    If you add 2 sine waves that are 180 out of phase, the peaks from one will be in the troughs from the other so you end up with nothing.
    Of course there are all the variations in between where some frequencies will match up and others not, which would give the comb filter that @Telefunky mentioned.

    This is in fact how most EQ's work (by shifting phase) and also how MID/SIDE encoding/decoding works.

  • Are you using a specific preset?

    @supanorton said:

    @eustressor said:
    In Auria, a little MicroWarmer on your drum buss goes a long way.

    Indeed.

  • @MusicMan4Christ said:
    Are you using a specific preset?

    @supanorton said:

    @eustressor said:
    In Auria, a little MicroWarmer on your drum buss goes a long way.

    Indeed.

    I usually set it to "tape-normal" and turn the little valve knob up, then play with the other knobs 'till it suits me. Sometimes all I change are the drive and output levels ... typically more drive equals lower output, since I'm not looking to add volume.

  • @Telefunky said:

    @OscarSouth said
    I'd be interested to hear from a 'more knowledgeable person' about how phase would factor into this. I've found that flipping the phase on this bass channel has a favourable effect on the sound, which makes sense according to my rudimentary practical knowledge of phase, but it'd be interesting to hear more from an expert on the topic, regarding how phase comes into play when running a couple of parallel channels (and is it different if running out of the same speaker or different speakers).

    as already mentioned: no need to be an expert, but trust your ears (listen carefully).
    'Phase' is an annoying term in this context and can be very misleading.
    Technically it means the displacement of a single(!) frequency sine wave in time and the value changes with frequency.

    Since for each frequency a shift of 180 degrees results in an inverterted signal you can in fact yield a 180 degree phase shift for any frequency by simply inverting the signal.
    That's what the so called phase button on a mixer channel is for.
    It's valid only for that specific 180 degree shift.
    Any 'phase value' different from 180 degrees will displace different frequencies (in time) and mix them to a new sound, an effect that's called comb filtering.
    It's often recognized as a 'hollow sound' (sometimes intended, as the famous mid position of a Stratocaster pickups selection switch), but any mixture may occur, depending on frequency content of the signal. There may as well happen a big fat bottom (the opposite) because sone frequencies add.

    But forget about theory and just diy with a record on one track and an identical copy on a 2nd track, which you move slightly in time.
    Ideally you do that in 1 sample steps. On high frequecies (like a hi-hat) you start noticing the difference in sound at roughly 3 samples, a bass needs much more.

    A kickdrum is a good example because it has a rather high frequency click and a deep bottom sound.
    The click (or attack) will be reduced (or smoothed) by shifting the signal in time by small amounts. With increasing values the base wave will become affected, too.
    But: since the wavelength of the high frequency part is much shorter than the bass, it will cycle between extinction and emphasize many times if you do this in single sample steps.

    This is just sound physics, it's by no way a fault or flaw - it's a part of our natural hearing and a fundamental part of any mix. The members of a traditional symphony orchestra are positioned in a way to yield a specific kind of sound. One would hardly recognize the ensemle, if members switched position.
    (there's an obvious coincinence between phase shift and signal runtime)

    Don't try to get too deep into controlling such stuff by it's values, it's way too complex in a real world situation (there's also room reflection...)
    Just know about the mechanism - it makes it a bit easier to understand some stuff you hear (though it may read quite confusing at first glance)

    Any 'phase value' different from 180 degrees will displace different frequencies (in time) and mix them to a new sound, an effect that's called comb filtering.

    What determines this rate? "displaces diff frequencies"

  • Full disclosure as MicSwap is an app I helped to create, but as a musician and recording engineer I run drums through MicSwap Pro to warm or dirty them up. With the adjustable gain and 13 different mics there's plenty of options to alter any sound you run through it.

  • @garygary said:
    Full disclosure as MicSwap is an app I helped to create, but as a musician and recording engineer I run drums through MicSwap Pro to warm or dirty them up. With the adjustable gain and 13 different mics there's plenty of options to alter any sound you run through it.

    Good thought....will try.

  • @eustressor said:

    @MusicMan4Christ said:
    Are you using a specific preset?

    @supanorton said:

    @eustressor said:
    In Auria, a little MicroWarmer on your drum buss goes a long way.

    Indeed.

    I usually set it to "tape-normal" and turn the little valve knob up, then play with the other knobs 'till it suits me. Sometimes all I change are the drive and output levels ... typically more drive equals lower output, since I'm not looking to add volume.

    What is this 'little valve knob' of which you speak?

  • @JohnnyGoodyear said:

    @eustressor said:

    @MusicMan4Christ said:
    Are you using a specific preset?

    @supanorton said:

    @eustressor said:
    In Auria, a little MicroWarmer on your drum buss goes a long way.

    Indeed.

    I usually set it to "tape-normal" and turn the little valve knob up, then play with the other knobs 'till it suits me. Sometimes all I change are the drive and output levels ... typically more drive equals lower output, since I'm not looking to add volume.

    What is this 'little valve knob' of which you speak?

    I'm pretty sure @eustressor is thinking about the valve knob on the Old Timer compressor.

  • edited January 2017

    @MusicMan4Christ said:
    Are you using a specific preset?

    @supanorton said:

    @eustressor said:
    In Auria, a little MicroWarmer on your drum buss goes a long way.

    Indeed.

    I start with "drum group", "snare", or one of the "track tape" speeds to get myself going.

    If you're familiar with more standard compressors, think of the speed knob as the attack and the relMul as the release. The others are pretty self explanatory.

    The Auria user guide has a simple description on pages 13 and 14.

    https://s3.amazonaws.com/auria.store/docs/Optional+Plugins.pdf

  • edited January 2017

    @RustiK said:
    Any 'phase value' different from 180 degrees will displace different frequencies (in time) and mix them to a new sound, an effect that's called comb filtering.

    What determines this rate? "displaces diff frequencies"

    it's quite simple as it's mostly about 'musical' frequencies, that have a certain relation.
    A note an octave above has 2 times the frequency, which corresponds to half of the time it takes to complete one wave.
    This gives a regular pattern, as waves overlay, the already mentioned 'comb' because the result looks similiar when graphically displayed.

  • Me, my first step to getting warmer drums is putting the mix in mono. Once that's established, then I use Saturn. I probably should purchase Vintage Warmer too, just to see which one I like more for warming. (Saturn will still get its use via other types of distortion.)

  • @garygary said:
    Full disclosure as MicSwap is an app I helped to create, but as a musician and recording engineer I run drums through MicSwap Pro to warm or dirty them up. With the adjustable gain and 13 different mics there's plenty of options to alter any sound you run through it.

    Congrats on the new "Arrangement"!

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