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Post by jmoose on May 5, 2021 14:21:33 GMT -6
Hmmm...
Probably me and my dbx subharmonic synth are in the wrong thread...
Might also scare some people to know that I'll run bass through a Pultec and crank 30Hz.
Next on my plate is a stoner rock album where the goal is to blow up speakers.
Ok I'll see myself out now. lol
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Post by Ward on May 5, 2021 15:10:16 GMT -6
I filter the top and bottom of pretty much every track in my sessions. Often filtering lows much more than most of you here. Ward saying he filters something at 29Hz is funny as hell to me. He's at 29 and I'm probably at 100. I came with jokes!! LOL What's a banjo good for? Kindling for an accordion fire! i'm here all week, try the veal and leave a tip!
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Post by Martin John Butler on May 5, 2021 15:28:01 GMT -6
I used to HPF more than i do now. In my untreated apartment, it helps a lot to HPF the vocal track below 30Hz or 50 Hz.
But I've realized those low frequencies are a huge part of the ambient sound, so now I make sure I'm just cutting out noise and not part of an instrument's sound.
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Post by Pueblo Audio on May 5, 2021 15:38:14 GMT -6
HPFs have infinite attenuation at 0Hz. Let’s recognize that “infinite” is a cosmic amount of action! For every action there is an equal and opposite reaction. So what is that reaction; that price we pay? PHASE SHIFTS. Shifts reaching as far as 10x above the cut-off frequency. For example, a 40Hz filter’s recoil may disturb spectra up to 400Hz. A lot of music lives in that region, right? (And minimal phase filters do not get you out of jail free)
So what does that mean to music producers? Phase shifts cause overtones to become misaligned with their fundamental causing internal comb filtering. The result being hollow timbres, ghost-like bass and smeared highs. This, consequently, may lead engineers to apply more processing to try to re-solidify the sound, leading to more phase shift. It can become a tragic race to nowhere.
That’s an ugly sonority penalty to pay if the HPF does not provide a tangible, material benefit. The kind you can hear from down the hall. Certainly there will be circumstances where they will be the appropriate tool. But it seems to me most skillfully tracked signals shouldn’t contain subs or other “garbage” with enough energy to upset a mix . I listen to original masters of the “history of recorded music” at the mastering studio all day, everyday. The most impressive and musical stuff nary saw a HPF at all. I’ve recorded decades of live shows (32 track) with my Pueblo preamps (which go down to 0Hz DC). Never needed HP filtering and the sonorities sound like the artists. I really like that.
I would think, test, then think again before adopting an all-channels-with-HPF default template
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Post by Quint on May 5, 2021 16:20:35 GMT -6
The better the shock mounts, the less need for hpf's. The more you compress, the more you need hpf's....and shock mounts. The smaller and less treated the recording environment, the more hpf's may fix things. DI's are notorious for huge subsonic junk from guitarists and bassists beating on the instrument. Yes, tape is an automatic steep hpf, but lower than many people are talking about here. Are the tape sim's doing that? I've never used one. I'm almost always through 1940-1950's preamps, and that stuff goes way lower than tape ever did. These late 1930's RCA's I've been using are flat to 12Hz, -1dB at 10Hz. I'm a believer in this approach. Get the stuff you don't want out of the signal chain as early as possible through the all of the sort of methods mentioned above. Then you don't have to introduce potential phase shift issues via a HPF. This is one of the reasons that I like to use EV RE-series mics on drums. Less bleed, due to their hypercardioid pattern, as well as lack of proximity bump, means that you're not filtering out a lot of stuff you don't want. Also, so much of this is also genre dependent and should be a part of this discussion. An Americana album doesn't need the same sort of tightness and filtering as a metal album. Different goals and restrictions.
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Post by Deleted on May 5, 2021 16:33:42 GMT -6
The better the shock mounts, the less need for hpf's. The more you compress, the more you need hpf's....and shock mounts. The smaller and less treated the recording environment, the more hpf's may fix things. DI's are notorious for huge subsonic junk from guitarists and bassists beating on the instrument. Yes, tape is an automatic steep hpf, but lower than many people are talking about here. Are the tape sim's doing that? I've never used one. I'm almost always through 1940-1950's preamps, and that stuff goes way lower than tape ever did. These late 1930's RCA's I've been using are flat to 12Hz, -1dB at 10Hz. I'm a believer in this approach. Get the stuff you don't want out of the signal chain as early as possible through the all of the sort of methods mentioned above. Then you don't have to introduce potential phase shift issues via a HPF. This is one of the reasons that I like to use EV RE-series mics on drums. Less bleed, due to their hypercardioid pattern, as well as lack of proximity bump, means that you're not filtering out a lot of stuff you don't want. Also, so much of this is also genre dependent and should be a part of this discussion. An Americana album doesn't need the same sort of tightness and filtering as a metal album. Different goals and restrictions. RE 20 and higher tuning is the best kick mic for double kicks because no proximity effect for rolls and blasts and no muddy buildup like the recent AKG D12 knockoffs (D112 and the D12 VR) are infected with. The beta 52 is is good too but the Audix D6 is just boom click and needs a lot of work to be normal. Depends on the metal record. Most of these guys don’t tune the kick high enough for what they want to play and honestly can’t play it in mot of the tracks I’ve received. That old school Florida death metal sound has zero bass. Neither does modern commercial “metal” except for subkick samples sometimes and a di bass. If they just tuned higher and screwed everything down better, there would be less filtering needed and better drums. 60 something hz double kick rolls on a mic with proximity effect and some muddy response is lol.
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Post by gravesnumber9 on May 5, 2021 16:46:53 GMT -6
HPFs have infinite attenuation at 0Hz. Let’s recognize that “infinite” is a cosmic amount of action! For every action there is an equal and opposite reaction. So what is that reaction; that price we pay? PHASE SHIFTS. Shifts reaching as far as 10x above the cut-off frequency. For example, a 40Hz filter’s recoil may disturb spectra up to 400Hz. A lot of music lives in that region, right? (And minimal phase filters do not get you out of jail free) So what does that mean to music producers? Phase shifts cause overtones to become misaligned with their fundamental causing internal comb filtering. The result being hollow timbres, ghost-like bass and smeared highs. This, consequently, may lead engineers to apply more processing to try to re-solidify the sound, leading to more phase shift. It can become a tragic race to nowhere. That’s an ugly sonority penalty to pay if the HPF does not provide a tangible, material benefit. The kind you can hear from down the hall. Certainly there will be circumstances where they will be the appropriate tool. But it seems to me most skillfully tracked signals shouldn’t contain subs or other “garbage” with enough energy to upset a mix . I listen to original masters of the “history of recorded music” at the mastering studio all day, everyday. The most impressive and musical stuff nary saw a HPF at all. I’ve recorded decades of live shows (32 track) with my Pueblo preamps (which go down to 0Hz DC). Never needed HP filtering and the sonorities sound like the artists. I really like that. I would think, test, then think again before adopting an all-channels-with-HPF default template I've always felt that there must be something I'm losing by removing too much low end. Whenever I try the approaches discussed in this thread, my recordings start sounding less like people playing music. On the other hand, I'm not exactly a master of a well balanced low end so I've got lots of room to grow. My question for you, Pueblo Audio, is does this issue persist even if you're using a low cut at the tracking stage either on the mic itself or on the pre-amp/strip if it has HPF. I generally avoid this as well but I've been experimenting with tracking with some compression going in and there's something to be said for less low end triggering the threshold.
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Post by Martin John Butler on May 5, 2021 21:43:07 GMT -6
This thread made me wonder if using no HPF except on the 2 bus would have the same phase shift effect as when using an HPF on the tracks.
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Post by EmRR on May 5, 2021 22:00:44 GMT -6
This thread made me wonder if using no HPF except on the 2 bus would have the same phase shift effect as when using an HPF on the tracks. Not really apples to apples, but if it was (same filter on every track), yes. As soon as a speaker interacts with an environment you have phase shift.
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Post by trakworxmastering on May 6, 2021 9:13:28 GMT -6
I'm beginning to thing the phase shift introduced by HPF is more detrimental than I thought. You are correct. HPFs come at a cost that few people seem to be aware of. Mixes can lose impact and weight. They sound more "digital" and lack thunder. IMO HPFs are vastly overused. It's a little distressing to see so many posters here saying they HPF nearly everything by default. There is no EQ setting that should ever be a default. I think everyone should try disabling all their HPFs as an experiment just to see what happens. 'Might be surprised. The more years I spend at this the less I use filters. I rarely HPF anything anymore and my mixes and masters do not suffer from the things that the internet imagines they will. Headroom is not a problem as long as the frequency response is balanced. Mud or lack of space is not a problem if the lows are shaped well. The bottom doesn't need to be amputated. Audio forums have perpetuated an over-reliance on HPFs when bells and shelves would work better in many cases. No offense to anyone, but HPFs are kind of a crutch. www.google.com/search?q=stop+the+high+pass+filter+madness&oq=stop+the+high+pass+filter+madness&aqs=chrome..69i57.9520j1j7&sourceid=chrome&ie=UTF-8
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Post by jmoose on May 6, 2021 10:07:24 GMT -6
I've always felt that there must be something I'm losing by removing too much low end. Whenever I try the approaches discussed in this thread, my recordings start sounding less like people playing music. On the other hand, I'm not exactly a master of a well balanced low end so I've got lots of room to grow. My question for you, Pueblo Audio , is does this issue persist even if you're using a low cut at the tracking stage either on the mic itself or on the pre-amp/strip if it has HPF. I generally avoid this as well but I've been experimenting with tracking with some compression going in and there's something to be said for less low end triggering the threshold. My general thought, which goes back to what I learned early on working as an assistant is to use the earliest/best filter (and pad if needed) available. Usually this means the one on the microphone. Occasionally the next downstream sounds or works better, meaning the pad/filter on the preamp but always try the mic first. Like if I need to cut out LF rumble on overheads? I have a pair of AKG460's where one of the filter options is 150Hz. WhamO. Done. From there its EQ as normal. Vocals can be a little trickier depending on how severe the plosives are. Maybe you use the mic filter. Maybe the mic is flat and you whip it at the preamp or an EQ patched after. But once the filter is in you shouldn't have to use it again? With an electric guitar... someone above mentioned LF junk in DI tracks and sure, I've experienced that. Get a DI for reamping and there's a big old thump of pick attack that's un natural and needs to be filtered. But if we had plugged that guitar right into an amp, any amp... the amp and speaker would've acted as a natural filter. So if we're tracking an amp with a mic... pick any mic. An SM57, U87, 121, Earthworks... whatever. Unless you dial the amp in with a huge low end bump there's really not going to be any subsonic junk to filter out because there wasn't any in the signal to begin with. Rinse and repeat for snare drums or trumpets & whatnot. Certainly agree with the thought & experience that if you pick & place mics well, place them in front of a good sounding source then there's really no need to high pass everything just because. Obviously YMMV!
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Post by gravesnumber9 on May 6, 2021 10:16:51 GMT -6
I've always felt that there must be something I'm losing by removing too much low end. Whenever I try the approaches discussed in this thread, my recordings start sounding less like people playing music. On the other hand, I'm not exactly a master of a well balanced low end so I've got lots of room to grow. My question for you, Pueblo Audio , is does this issue persist even if you're using a low cut at the tracking stage either on the mic itself or on the pre-amp/strip if it has HPF. I generally avoid this as well but I've been experimenting with tracking with some compression going in and there's something to be said for less low end triggering the threshold. My general thought, which goes back to what I learned early on working as an assistant is to use the earliest/best filter (and pad if needed) available. Usually this means the one on the microphone. Occasionally the next downstream sounds or works better, meaning the pad/filter on the preamp but always try the mic first. Like if I need to cut out LF rumble on overheads? I have a pair of AKG460's where one of the filter options is 150Hz. WhamO. Done. From there its EQ as normal. Vocals can be a little trickier depending on how severe the plosives are. Maybe you use the mic filter. Maybe the mic is flat and you whip it at the preamp or an EQ patched after. But once the filter is in you shouldn't have to use it again? With an electric guitar... someone above mentioned LF junk in DI tracks and sure, I've experienced that. Get a DI for reamping and there's a big old thump of pick attack that's un natural and needs to be filtered. But if we had plugged that guitar right into an amp, any amp... the amp and speaker would've acted as a natural filter. So if we're tracking an amp with a mic... pick any mic. An SM57, U87, 121, Earthworks... whatever. Unless you dial the amp in with a huge low end bump there's really not going to be any subsonic junk to filter out because there wasn't any in the signal to begin with. Rinse and repeat for snare drums or trumpets & whatnot. Certainly agree with the thought & experience that if you pick & place mics well, place them in front of a good sounding source then there's really no need to high pass everything just because. Obviously YMMV! This makes sense to me. I wonder though... from a physics of sound perspective... if I switch on the filter on a mic will that also cause phase issues? I think I don't really know enough about how these filters work on microphones. For example, I use an SM7b a lot and it can sometimes be too thick in a final mix... should I change my settings on the mic? My view has always been "you can take frequencies out, but you can't really add them" so I've avoided using too many cut switches on mics. I'm rethinking that.
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Post by Guitar on May 6, 2021 11:03:04 GMT -6
Yes, a filter on a microphone will still have "phase issues." They do not defy the laws of physics. Even the high boost filter on an SM7B will have "phase issues" but people love it.
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Post by stormymondays on May 6, 2021 11:22:33 GMT -6
I think there's pure gold in this thread, both in the HPF camp and the no HPF camp. Thanks everyone for the contributions.
I got "addicted" to HPF when I recorded and mixed a live music TV show on location. The rooms were untreated, the preamps were what they were, and cleaning up the low end mess made a lot of sense. My monitoring wasn't as good by then either.
Does it make sense to apply the same philosophy to stuff carefully recorded in my studio, with state-of-the-art sources, mics and preamps? I think not!
I think it's one of those cases of "what got you here won't get you there". HPF has served me well, but it might be time for us to part ways...
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Post by stratboy on May 6, 2021 12:14:38 GMT -6
“This shouldn't make your mix smaller at all, just free it from the emotional baggage it was dragging around so it can dance on the speakers.“ What a beautiful way to say it. Thanks, monkeyxx!
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Post by Guitar on May 6, 2021 12:41:09 GMT -6
If you use EQ of any kind during a mix, you are using a filter, maybe even a few dozen of them. IMO, the issue is a bit exaggerated. And I thank you stratboy.
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Post by Deleted on May 6, 2021 12:45:16 GMT -6
Hmmm... Probably me and my dbx subharmonic synth are in the wrong thread... Might also scare some people to know that I'll run bass through a Pultec and crank 30Hz. Next on my plate is a stoner rock album where the goal is to blow up speakers. Ok I'll see myself out now. lol HA! I have on occasion worked with 8 string guitars with a 23hz root fundamental. Just like everything it depends on what you're trying to achieve / what you're using, I don't often (if ever) record others anymore so I can take a more relaxed direction and focus on what I was taught at college in terms of sonic "superiority". When you have people tapping mic stands, proclaiming the song needs to be louder etc. then the only real choice is to excessively HPF and LPF everything whilst trying to use harmonics to fill in the missing fundamentals. Studies have showcased that this doesn't always work either so I spent a lot of time harnessing lower volume overtones to fill in the gap. With rock / metal bands there were so many DI parts, as others have mentioned these do add subsonic junk into a mix. With the classic way of doing things everything is mic'd so none of this was an issue, guitar cabs mainly reproduce 75 hz - 5Khz, bass cabs 55 onwards. The only thing I'd ever consider HPF'ing in a classically recorded mix is the kick (maybe). Even then I'd probably just MB compress it like the guitars and often that would do the trick. Phase is something to be avoided like the plague, for e.g. my first encounter with PT Native resulted in some of THE worst mixes every made. You could literally send a dry signal to a dry aux and they would be out of time, I researched this and asked around DUC for months. The end result? True delay compensation is a PTHD "feature".. Hmm, well this doesn't happen in Logic, Cubase, Samplitude (or insert most DAW's here).. Weirdest thing is the doubling effect initially sounded better to my ears until I realised what was going on (I worked with consoles before PT8). Nowadays I try to follow as many classical styles or techniques as I can, I'll mic everything / avoid VST's or plugins wherever possible. Yes, I know there are many awesome pieces of software out there but I've had the odd plugins with bugs that were clearly introducing phase and aliasing. Won't deny sometimes I mistook that as an "improvement". Plus I'm not afraid to commit EQ / compression and even verb to a track when recording, I won't go overboard and do small improvements ITB with UAD or something else. Still, once you're used to the old ways it's amazing how simple things become. I often fought tooth and nail ITB only / direct, I'm not saying ITB is bad I'm saying I completely suck at it. Ultimately my point is, I try to avoid over processing and phase where I can nowadays. Although I'll do whatever is necessary, if a track still requires a HPF to clear it up I will do so.. If it doesn't need it then I won't.
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Post by theshea on May 6, 2021 12:45:52 GMT -6
hell, i even hpf my mixes at 35hz! in addition to hpf single tracks!
once i started using lpf in solo (tdr nova) and went way down to 50, 40, 30, 20 ... to listen whats going on down there in isolation. and there‘s a lot of stuff going on down there! but i still hpf pretty aggressively.
but ... after reading some post here i will experiment without hpf or a lightly „abused“ version of it.
thanks for the good input everyone!
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Post by gravesnumber9 on May 6, 2021 13:07:50 GMT -6
Yes, a filter on a microphone will still have "phase issues." They do not defy the laws of physics. Even the high boost filter on an SM7B will have "phase issues" but people love it. I don't really understand what I'm envisioning in my mind on this stuff. I think since I grew up in the digital age I see so much visualization of the frequency range that I kind of feel like "you know, just cut off the stuff on the left, right?" Haha. It's strange to me that plugins and whatnot have to follow the same limitations as real gear in this respect. Anyway, in general I'm a "leave it in unless it sucks" type guy in every sense so the HPF on everything has never been my bag anyway.
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Post by gravesnumber9 on May 6, 2021 13:12:14 GMT -6
I'm not saying ITB is bad I'm saying I completely suck at it. This thread is filled with great quotes! Love this. ITB/OTB debate could be solved if we all just admitted that we're doing the best we can and the smartest of us are avoiding things we're not good at.
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Post by jmoose on May 6, 2021 13:28:40 GMT -6
This makes sense to me. I wonder though... from a physics of sound perspective... if I switch on the filter on a mic will that also cause phase issues? I think I don't really know enough about how these filters work on microphones. For example, I use an SM7b a lot and it can sometimes be too thick in a final mix... should I change my settings on the mic? My view has always been "you can take frequencies out, but you can't really add them" so I've avoided using too many cut switches on mics. I'm rethinking that. There's a few things wrapped together here... First when we're talking about phase its not "time domain" based like most people think of. Its frequency based phase, phase rotation. They're absolutely related but completely different. Think of time aligning mics in the DAW vs using a Little Labs IBP. Technically "phase" is just another form of EQ whether its time based or the phase shift from a Neve or API EQ. The question is, and its entirely subjective, is what's more pleasing? And then maybe, what causes more/less problems down the line? And when I say down the line... try to think all the way out to the actual release of the music. Past recording, past mixing & mastering... go all the way to some random person listening to the music out in the wild. Big picture shit. Remember this is all cumulative! If someone says to me the vocal is too thick in the mix, then usually I'm going to see what's happening around 200Hz and maybe rip out some energy there. Way above "rumble filter" range. Too thick could also mean it doesn't have enough articulation and needs more energy on the top end. Either way what's happening at 60Hz is pretty irrelevant. Taking out vs adding... I tend to capture & print sounds as I want to hear them. Always mixing. If I feel the snare is kinda boxy and need to suck out 400Hz and put a lift up top then I'm gonna do it right at the starting line. I have plenty of outboard gear and no fear about using it and committing on the front end. The key there is knowing where I'm going. If I need to dump a bunch of low end or whatever to slot something into the overall song then I'm not going to put off that decision until later. All those things add up to shape and define the songs as they're recorded. Think of it this way, if you make excellent decisions about your sources... put some care into how they're recorded... then there really isn't going to be any "baggage" to sort out on the back end. That kind of thinking and workflow is completely different from the "track it flat" and sort it out later deal that, for various reasons, often happens at home studios. Sometimes its lack of experience. Doubt about instincts and/or the overall direction of the production and just as often... lack of equipment. Hence my earlier comment that a lot of "LF garbage" is naturally filtered through the equipment... transformer coupling & whatnot vs plugging a microphone right into an interface and recording DC to light with the hopes & dreams of sorting it out later.
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Post by Deleted on May 6, 2021 15:18:49 GMT -6
This makes sense to me. I wonder though... from a physics of sound perspective... if I switch on the filter on a mic will that also cause phase issues? I think I don't really know enough about how these filters work on microphones. For example, I use an SM7b a lot and it can sometimes be too thick in a final mix... should I change my settings on the mic? My view has always been "you can take frequencies out, but you can't really add them" so I've avoided using too many cut switches on mics. I'm rethinking that. There's a few things wrapped together here... First when we're talking about phase its not "time domain" based like most people think of. Its frequency based phase, phase rotation. They're absolutely related but completely different. Think of time aligning mics in the DAW vs using a Little Labs IBP. Technically "phase" is just another form of EQ whether its time based or the phase shift from a Neve or API EQ. The question is, and its entirely subjective, is what's more pleasing? And then maybe, what causes more/less problems down the line? Actually that's somewhat incorrect, it is both time / distance and frequency domain dependant. Okay in terms of frequency domain the most basic repeatable use case is two sine waves flipped to cancel each other out. In terms of time, what we're told to do as AE's is avoid incorrect travel distance between two source collections, in a 3:1 we'd place a mic at exactly three times the distance of another source mic. Why? The delay of positive and negative frequencies clashing will induce phase and again often unpredictably cancel each other out. This can even be done with reverbs.. Dry mic'ing single instruments one at a time (without moving the mic) should be fine (but I can explain several scenario's that say otherwise). Although generally the rule of thumb is the more mic's involved the more phase you'll come across. P.S I get your point but you gotta be careful especially when you're multi-mic'ing because time / distance can play a massive factor in phase and crap recordings.
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Post by jcoutu1 on May 6, 2021 15:20:46 GMT -6
I tend to capture & print sounds as I want to hear them. Always mixing. If I feel the snare is kinda boxy and need to suck out 400Hz and put a lift up top then I'm gonna do it right at the starting line. I have plenty of outboard gear and no fear about using it and committing on the front end. The key there is knowing where I'm going. If I need to dump a bunch of low end or whatever to slot something into the overall song then I'm not going to put off that decision until later. All those things add up to shape and define the songs as they're recorded. Think of it this way, if you make excellent decisions about your sources... put some care into how they're recorded... then there really isn't going to be any "baggage" to sort out on the back end. I'm with you there (even though I still filter like a mad man after the fact).
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Post by gwlee7 on May 6, 2021 19:01:54 GMT -6
Another thread that needs bookmarking by me.
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Post by jmoose on May 6, 2021 20:20:29 GMT -6
Actually that's somewhat incorrect, it is both time / distance and frequency domain dependant. Okay in terms of frequency domain the most basic repeatable use case is two sine waves flipped to cancel each other out. In terms of time, what we're told to do as AE's is avoid incorrect travel distance between two source collections, in a 3:1 we'd place a mic at exactly three times the distance of another source mic. Why? The delay of positive and negative frequencies clashing will induce phase and again often unpredictably cancel each other out. This can even be done with reverbs.. Dry mic'ing single instruments one at a time (without moving the mic) should be fine (but I can explain several scenario's that say otherwise). Although generally the rule of thumb is the more mic's involved the more phase you'll come across. P.S I get your point but you gotta be careful especially when you're multi-mic'ing because time / distance can play a massive factor in phase and crap recordings. Huh..? Possibly I'm missing something. Who was talking about multiple microphones..? Only reason I referred to time domain was because that's probably an easier variation of "phase" for some to understand... Using an equalizer doesn't flip things, it creates ripples in frequency response & phase coherence above and below the center frequency. In the parlance of our times that's the crux of the biscuit.
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