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Post by svart on Apr 30, 2024 9:38:01 GMT -6
Which model Bumblebees do you have? I am intrigued. RM6 and RM7
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Post by svart on Apr 30, 2024 8:46:42 GMT -6
I love my R121. I have a friend who had a R10 and it never sounded "royer" to me.
I have a couple Bumblebee ribbons too. Love them as well but while they look like a royer, they are their own beasts. Much more hi-fi sounding than the royer, but still ribbon-y.
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Post by svart on Apr 30, 2024 8:42:27 GMT -6
Generally start with a 57. Pick a speaker in the cab, put the 57 right on the cloth, pointed right at where the cone and the dust cap meet. Turn your amp to mostly mid settings and then listen in the control room for a minute while someone plays. Now go move the mic away from the cap slightly, towards the middle of the cone for less highs and more mids. I generally end up somewhere near the cap, usually within about 3". Now go back and tweak the amp.
Most people tweak the amp first and what happens is that they dial in tons of high end to get clarity and it ends up waaaaaay too fizzy and then they want a different mic.. Then they end up chasing the tail of the dragon. Amps and speakers sound very different to the ear in the room than they do to a mic on a speaker. Set the amp how it sounds best through the mic. Players tend to set way too much top and bottom end because it sounds good in the room, but this is the opposite of what you want for recording. When you end up doing HPF/LPF on the track later because it's all fizz and rumble, you'll be left with nothing.
Set the mic for best balance first, then tweak, never the other way around.
Don't even worry about choosing the best speaker in the cab until you've learned how to identify where to put the mic. Then once you found the best spot on ONE speaker and the best tone for the amp, then try other speakers in the same spot and listen to the differences. Some speakers will have drastically different tones. Even the same speaker have different tones from one side of the cone to the other. That's why I tend to make people play through my cabs, if possible, because I've spent a lot of time over the years moving mics around on them and I sort of know where they sound best to me.
The other magic here is a R121. Right in the middle of the dust cap. Move it back about a fist-width from the cloth. This will work for most clean tones. Put it at the seam between the cap and cone if you're using distortion, and then move it around some. You'll get a lot less variation than with the 57, but it'll be noticeable.
I used to multi-mic cabs, but I've lost interest in doing that. You end up messing around with blending way too much and the phase issues become too apparent, too often, and usually only one mic gets chosen anyway.
One other piece of advice.. Never sit in front of the amp trying to find the right spot while the player is playing. Even with headphones your ears will get fatigued and the rumble and leakage will make it seem different than it is.
Most musicians will tell you it sounds great at the time, but that's because their ears are blown too. They like loud. Loud is fun. So don't ask them if they approve..
Loud is not good for your ears, nor your recording. An amp will be just above talking volume for most recordings. If it's shaking the walls, then something wrong is being overcompensated for.
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Post by svart on Apr 26, 2024 8:48:29 GMT -6
And while this was "cheap enough to try", it's a little discouraging that this is how the plugin industry is these days. I have probably 50+ plugs I've bought over the years and stopped using. Some of them were expensive and now virtually worthless and not really re-sellable without a lot of effort and discounting.
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Post by svart on Apr 26, 2024 8:45:01 GMT -6
Though I bought deBleed (for GAS reasons) I still haven't even tried it. I seriously have a purchasing problem. I already have so many plugins that "I had to buy" and I haven't even used them. Have you ever gone to purchase a new plugin only to find that you already purchased it during the last sale, yet never used it? That's my issue. I use Silencer on drums, mostly on the toms. I go pretty light on snare. Maybe I will get a chance to finally play with deBleed in the next few days. I one way, hardware wins. After I get hardware, it's always in front of me and always gets used (or gets sold). I don't tend to do endless searching for the best hardware 1176, but I do that with plugins. How many can I possibly need? Obviously... one more. Can't say I've tried to buy two of the same exact thing, but I have been guilty of buying multiples of the same style, I.E., Waves SSL channel and then SSL Native, etc.. And once the GAS is over, I can't really tell the difference. The absolute settings might be different, but overall they do the same things.
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Post by svart on Apr 26, 2024 6:33:36 GMT -6
So I took the plunge with this for everyone. I'm a sucker for trying to find magical/mythical gates.
Anyway, it's roughly the same thing as Silencer. It has mostly the same controls but adds a HOLD control. It even looks mostly the same, GUI-wise.
So I was able to set up Silencer and DeBleed on a pretty noisy snare and then A/B them.
Honestly, they sound mostly the same. Setting them to the same settings does not result in the same sound, but you can tweak a few knobs and get mostly similar amounts of bleed through them. I don't think either one worked better than the other. I think they both resulted in similar amounts of bleed or choked snare sound.
I can't see needing both. Pick one or the other and you'll never miss the one you didn't buy.
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Post by svart on Apr 25, 2024 9:47:24 GMT -6
The only thing I care about is whether or not my mix translates. I don't even bother with adjusting the crossover on my sub, I just turned it up until I felt like the sub-bass in my mixes was too low and then backed off. I think they can also be very deceiving because once they sound good, you end up pushing the low end a lot more than things like earbuds can handle well.
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Post by svart on Apr 23, 2024 13:36:31 GMT -6
Yeah it sounds bad right now.. But in 5 years this will be so good that they'll use it everywhere. Every movie will be AI generated and have a single person act and voice every character and just have someone digital create all the characters, etc.
And then albums will be AI generated with the same kind of stuff.
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Post by svart on Apr 23, 2024 13:03:39 GMT -6
I still like ValhallaRoom. Slate verbsuite is ok if you get the add on lex and bricasti libraries.
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Post by svart on Apr 23, 2024 9:32:23 GMT -6
John, maybe rather than a sideway step, you should push the boat out and get your final keeper pair of monitors. ATC 25’s or 45’s Or similar. Then you’re done and dusted forever with monitors Man, I gotta be honest…I don’t get ATCs. I had the baby scm12’s, but I don’t think those are particularly representative of the line. And I’ve never mixed on the 25s. So maybe I’m completely missing the boat. But every time I listen to them I always think they sound boxy with no low end. I definitely lack experience with them. And as I’m drinking my morning coffee with almond milk because there’s no milk and that pisses me off because almond milk tastes like the tears of real milk…but I digress…as I drink my morning coffee, it leads me to this riddle: how can something have “better” midrange? Wouldn’t that just be “louder” midrange? Is it just that it’s like a quicker slew to the transient or something? As far as why I want to? Because I’m a glutton for punishment. lol. It’s a hobby. I love the excitement of the deal. The possibility of taking a chance and finding something killer is more fun than the slight hassle that I gotta go find some used Amphions again lol.
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Post by svart on Apr 23, 2024 6:20:37 GMT -6
I dunno. I think the sound of plugs has been good enough for a long time. I've narrowed down the ones I use to just a handful and I can't really see how they can be bested for what I use them for.
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Post by svart on Apr 23, 2024 6:18:15 GMT -6
Just the shelf.
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Post by svart on Apr 23, 2024 6:15:43 GMT -6
The waveguide vs. No waveguide thing makes me hesitate. I don't think I can work on monitors that don't have super wide tweeter dispersion anymore.
Granted, I haven't listened to the 3's but I didn't like the 1's and these seem to be of the same or lesser pedigree..
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Post by svart on Apr 22, 2024 10:50:58 GMT -6
I've had nothing but problems with monitors in the room while recording. I've received a number of tracks from artists with similar concerns, not being able to use headphones, etc and the vocal tracks are usually very dirty with phasey trash in the background.
Usually the rumble is the worst though and an aggressive HPF goes a long way to helping.
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Post by svart on Apr 19, 2024 15:26:21 GMT -6
5H? wow that's a lot. It would have to be really small wires to coil that much inductance. Like Eric said, you could find a smaller value/size inductor and then scale the cap values to get the same frequencies, but the Q of the inductor would likely be different and the rolloffs might be different. The value was revised to 0.5 Henry. Oh ok, that's better. There's a handful of them in eBay
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Post by svart on Apr 19, 2024 14:43:34 GMT -6
5H? wow that's a lot. It would have to be really small wires to coil that much inductance.
Like Eric said, you could find a smaller value/size inductor and then scale the cap values to get the same frequencies, but the Q of the inductor would likely be different and the rolloffs might be different.
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Post by svart on Apr 19, 2024 13:13:33 GMT -6
Got it. It's interesting. I can't hear any difference without doing a null test and honestly, the major things that remain ifrom the null test are modulated sounds. If this is a mix of yours, then I think the modulation effects added are non-deterministic which would render a null test impossible. Also, I do hear a phasing sound from the null test as well which tells me that one of the clocks is wandering around slightly.
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Post by svart on Apr 19, 2024 11:04:56 GMT -6
Here are some files if anyone wants to compare a few different clocking scenarios(including blind files): www.dropbox.com/home/Seawell%20Studios%20YouTube%20Channel/Mutec%20Audio%20FilesA few different clocking scenarios, Mutec master clock, Avid and Lynx, etc.. all explained in the video below if you'd like to check that out. My conclusion is that a master clock absolutely makes a difference, you just have to decide if that difference sounds better to you or not. I continue to use the Mutec as a master clock more for stability than anything but I do also enjoy what it does to the sound 👍🏼 Can't access without logging in..
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Post by svart on Apr 19, 2024 11:03:34 GMT -6
Jitter can absolutely cause harmonic distortion. The primary clock frequency being offset for any small amount of time would be jitter and cause sidebanding, but that offset amount does not necessarily need to be an multiple of the primary frequency. It can modulate any fractional offset onto the clock and that offset need not be related to the primary frequency at all. A lot of PLLs use fractional dividers to keep complexity down, but these cause sidebands at strange fractions of the primary output that aren't necessarily related in harmonicity. You are describing inharmonic distortion. Harmonic distortion: having to do with multiples or evenly divided fractional multiples of a tone. Inharmonic distortion: Having to do with harminics that do not fall on multiples or evenly divided fractional multiples of tones. Sidebands can be ANYwhere in relation to the primary tone, therefor can be harmonic or inharmonic depending on where they fall. Jitter can have either, or both.
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Post by svart on Apr 19, 2024 9:46:53 GMT -6
FWIW, jitter doesn't cause harmonic distortion for a richer sound. It causes inharmonic distortion for a harsher sound. I doubt that is what people like about external clocking. But that inharmonic distortion is usually so slight that it's hard to hear. I remember when Eric Valentine posted a comparison with printed files of 192 I/Os with and without an Atomic Clock. The audible difference was that the Atomic Clock files had less crispness in the top end, making them seem smoother and warmer, basically "less digital" and more like analog tape. A touch of HF smearing. It was subtle, and I felt that I could engineer my way around it either way. But there are so many different converter designs and studio configurations that we can't really generalize about what audible effect every combo of clock and converter will have. We have to try it out with our setups and make choices. The most important thing I learned when I tested it was to print files for listening tests because it takes too much time to switch clocks and that makes memory too much of a factor. Expectation bias was especially powerful in this context, so blind testing was essential. And brutal honesty. Jitter can absolutely cause harmonic distortion. The primary clock frequency being offset for any small amount of time would be jitter and cause sidebanding, but that offset amount does not necessarily need to be an multiple of the primary frequency. It can modulate any fractional offset onto the clock and that offset need not be related to the primary frequency at all. A lot of PLLs use fractional dividers to keep complexity down, but these cause sidebands at strange fractions of the primary output that aren't necessarily related in harmonicity.
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Post by svart on Apr 19, 2024 6:58:46 GMT -6
So modern converters don't need external clocking at all unless you need some semblance of sync between devices, which I don't think is necessary in any normal use case because DAWs allow you to nudge tracks to line up in phase anyway.
Huh. No sync necessary?
Recently I was trying to reamp a guitar through a digital Katana amplifer, sending the clean signal through its built-in soundcard, and then recording the miced up amp through my soundcard. The audio started to get out of sync as soon as ~10 seconds in. Luckily, I was able to fix it by lossless stretching, but I was surprised by the amount of clock drift.
Is this not a common occurrence, then, using multiple interfaces that are not clocked together?
I guess that would be a good example of a time you might want to sync, but it seems somewhat extreme. Easily fixed by using channels on the same interface if you need extreme phase alignment. But that's also not a symptom of jitter, that's frequency being off. One or more of the clocks is not precisely on the right frequency. It's a common problem if the division factor is low and the reference crystal is not temperature stable or is old or poor quality.
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Post by svart on Apr 18, 2024 12:33:15 GMT -6
Without taking scientific view, just going on using my ears (crazy as that seems these days) I clock my system from my HEDD 192 because … Again more craziness …. it’s sounds better. Just my 2 cents. I don't think anybody is saying it's not different.. But I think a lot of people actually like MORE jitter than less jitter. More jitter would cause more harmonic excitement and would sound "wider" and more "full" and probably have more "detail" in the form of small amounts of distortion.
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Post by svart on Apr 18, 2024 11:33:17 GMT -6
It won't. External clocks were used to sync a bunch of different devices. At some point people seemed to like what they did to the audio so they started using them to "fix" the sound.
But actual measurements show that external clocking, with the cable/connector/termination impairments and parasitics make EVERY external clock technically worse than internal clocking.
What I've found over the years is that deterministic phase noise is not necessarily a problem in any modern system in the sense that the primary tone of interest is quite stable with sideband power being very low and easily filtered out. Almost all sigma-delta converter ICs need 128-512fs clocking, and any residual phase noise would also be divided by those amounts before the sampling hardware sees the clock signal. Again, very, very low chance that jitter is the "cause" of a difference in sound.
However, in external WC cabling, there is a possibility of random external noise being introduced.
There's also the question of *what exactly is happening to the WC signal* once it's brought into the device.
1. Used directly.. Which would be more rare today since sigma-delta converters simply can't use the low frequencies directly. 2. Upconverted by PLL/DPLL to 128-512fs for use by the sigma-delta converters.. Which is more likely but also a worst-case scenario. 3. #2 but following the PLL is a "jitter cleaner" which is another PLL with a specific design to decouple the dirty reference clock from the output.
#2 is a bad situation alone. Taking a noisy WC signal and using it as a direct reference for a PLL would MULTIPLY the jitter by the amount that you'd need to multiply the signal to get 128-512fs for the converter clocking. Imagine multiplying your jitter 512x..
So #3 would be the only viable situation, by multiplying the WC signal up to say 100MHz with an aggressive loop filter(low pass feedback) then using that as a reference for a jitter cleaner IC that divides back down to something like 22.579Mhz for the converter ICs to use.
You can see that the converter IC clocks are now extremely decoupled from the WC input and other than being in sync, have very little to nothing in common with other devices also being clocked in a system. It also has very little to nothing to do with the quality of the WC signal.
So modern converters don't need external clocking at all unless you need some semblance of sync between devices, which I don't think is necessary in any normal use case because DAWs allow you to nudge tracks to line up in phase anyway.
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Post by svart on Apr 17, 2024 9:26:56 GMT -6
if you didn’t like earlier more expensive Barefoots, why would you like these with their probably cheaper drivers and underbuilt cabinets? Because… Damnit, I don’t know,…you’re killing the dream lol…but the answer to that is maybe three ways are different? I’m not unhappy with my Amphions - just wondering if there’s something idk I’m missing out there… I gotta say, 3 ways are night and day different from 2 ways in how they present the audio. Tonally they may or may not be all that different but having a midrange driver just makes a huge difference in details. that being said, I agree that if you didn't like the higher end barefoots, there's probably no realistic expectation that these would suddenly be better. They're designed to a lesser price point and likely voiced by the same folks with the same ears, so I'd expect them to be similar in overall sound to the rest of the barefoot lineup.
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Post by svart on Apr 16, 2024 7:44:49 GMT -6
I find myself travling a bit due to my wifes schedule, which means I have Pro Tools on my laptop (Macbook Air M2) and don't have my Carbon with me. I don't really have a small portable interface like and UA Arrow Solo or something along those lines, but I do have a set of Beyer Dynamic DT1770 Pro's, which personally I think are decent, but not top-shelf perhaps. Assuming good original source that I am mixing on this setup, is an interface really needed or is the headphone out good enough? I think I already know the answer to this question, at least I know what my assumption is, but wanted to get some broader opinions on it. Appreciate whatever you can offer. Thanks, BHM. I know some folks will say it absolutely matters, but I don't think so. You're always going to get a decent picture of the sound. If you don't, then it's probably the headphones. I think the transducers (mics or speakers/headphones) will always be more important than the converters.
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