I'd like to share a great studio technique that I call the "echo demo". I hope you find it as useful as I have.
How do you determine optimum mike placement? By ear, of course, but how do you hear just the recorded sound without any live sound? The big-budget answer: have your assistant fiddle with the mike out in the tracking room while you listen in the control room. Well, I ain't got no assistant. I ain't got no control room. All I got is a one-room home studio with a PC and a DAW.
So, here's what I do: I set up my DAW to send the incoming audio from the mike directly out through the monitors, without actually recording it. (In Protools, this involves pushing the track's record button, and disabling low-latency monitoring in the operations menu.) In other words, I use my DAW like a PA.
I want to emphasize the need for caution at this point. BE CAREFUL! Studio mikes are very sensitive and often prone to feedback. Never put the mike close to the monitors, or point it right at them. Use a gobo, if you have one. I recommend first turning the monitor volume all the way down, then setting eveything up, then finally slowly turning up the monitors.
Next I put a long delay plugin on the master track and set it for 3-4 seconds. Now the fun begins. I hold the mike up to the instrument (or amp) and have the musician play a short test riff (perhaps an arpeggio). Then we listen to the riff echo in the monitors. I move the mike around and he/she plays the exact same riff, and we listen again. Over & over until we zero in on the best possible sound. Notice I said "we". A big advantage the echo demo has over the control-room method is that the musician can offer his feedback, too.
The preceding example is for a melody instrument like a sax. For brief percussion sounds, say a cowbell, using a shorter delay greatly speeds up the process.
Now, there's just one little problem. A sensitive mike will pick up the monitor echo and send it off to the DAW, to be sent out through the monitors again as a second, fainter echo. These secondary echoes can be confusing and annoying. There's three ways to get rid of them:
1. Use headphones.
2. If by chance you have a mike muting footswitch (called a "cough drop" or a "short stop"), just mute the mike during the echo.
3. The best solution: use a gate. Put it on the master track, right before your delay plugin . Set the threshold low enough for the test riff to get through, but high enough to keep out the monitor echo. Set your attack as fast as possible, so you don't chop off the instrument's attack. Give yourself lots of hold and decay. Set them long enough so you don't chop off the end of the riff, but short enough so that the gate shuts down before the echo comes back. For a 2-second test riff through a 3-second delay, try setting each to a half-second. Ideally, use a gate with a side chain, and send a slightly-less-delayed signal to the side chain, so that the gate opens up a bit before the note.
You can demo multiple-mike setups this way, too. Just set up your DAW to output all the mikes. The one gate and one delay will handle everything, though you may have to adjust the gate threshold and/or the monitor volume. With the plugins on the master track, I think of it as gating/delaying the speakers, not the mikes.
Here's a fun idea for rhythm instruments. Set the delay to correspond to a common tempo, say 2 seconds for one bar of 120 bpm. Get the musician to alternate playing a bar and listening for a bar. He can often fall into a groove and "jam with himself". You may need to use a shorter decay on the gate. You can also double the delay (use a second plugin if you have to) and play/listen for two bars. This transforms a tedious process into something fun and helps keep your clients fresh for recording.
Okay, now that you've determined how best to place a given mike for a given instrument, how do you choose which mike to use? You demo them, of course, but how do you listen back? The usual way is to record the musician with several mikes at once, then loop the recorded results, using the solo buttons to switch back and forth between samples. Workable, but cumbersome.
Let's use echo-demoing: First, set up your DAW so that each mike has its own track, output directly to the monitors as before. Then put a gate and a delay on the master track, as before. Then put another delay on the second mike track, two delays on the third, three on the fourth, etc. Now each mike has its own unique delay time. Play a riff, and you will hear a series of echoes, one for each mike. This allows you to instantly evaluate your mikes. You can even tweak the mike placement further at this point. Be sure to play both high and low riffs, to test the full range of the instrument.
One more application: demoing effects. To do this, set up as for demoing mike placement, and add a send to an aux track. Put your effect on the aux track, along with another delay. Now when you play a riff, you'll hear a raw echo followed by a processed echo. If you want to compare different settings, set up several aux tracks, each with its own effects chain. Put one delay on the second, two on the third, etc. Remember to mute the raw signal. Now you'll hear a series of echoes, each one processed differently. This is especially useful for experimenting with the order of plugins.
Echo-demoing really excels in situations where the musician is also the recording engineer. A musician with a home studio can learn a lot about recording himself. It also works for experimenting with effects used for his live sound. But what if you are in a full studio and you want to take advantage of the speed and directness of echo-demoing? You have two choices: put the musician and the mike(s) in the control room, or put your monitors in the tracking room. Which way you go depends on the particulars of your set-up.
These are just some of the uses of this method. It's a great way to explore the potential of your studio. Any time you want to experiment with different options and quickly evaluate the results, consider the echo demo.
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Hi Madmax, thanks for reading! "Just be aware that every echo r
Hi Madmax, thanks for reading!
"Just be aware that every echo repeat, you get phase shift and comb filtering that changes the frequency response of the next loop back/echo... and over the course of say... 3-5 seconds you can expect to loose up to 50% of the original frequencies."
I'm no expert on phase shift and comb filtering, but all the echoing does is take the digitized signal from the mike and store it temporarily in the computer's memory, then play it back. If you instead record the same signal to hard disk, and then play it back a minute later or a day later, you are listening to the exact same ones and zeros. Digital is digital. That's why no-one talks about how one brand of hard drive sounds vs. some other brand. So if echoing adds phase shift and comb filtering, so would straightforward recording.
The only way I can make sense of what you're saying is if you're talking about the secondary echoes. If you use a gate as outlined in the article, there is only one loopback/echo, and there are no secondary echoes.
Or maybe you think the echo overlaps the live sound? It doesn't, the live sound should end before the echo begins.
Let me make something clear: the echo effect is not part of the finished recording. It's just a way to immensely speed up the process of making a test recording and listening back to it.
"So, experience is probably the best teacher... and the more you do, the better you become at listening and making better mic position decisions." This method is about much more than mike placement.
It does sound like a lot of nonsense to those who actually know
It does sound like a lot of nonsense to those who actually know what they are doing. Reminds me of how complicated someone makes life when they are in the beginning stages of learning something. Finally, you go, AHH ! you mean all I have to do is listen to that and put it here.
And I do understand this is what you are saying by using your method but,
Audio and creative talent is of course more subjective. You can go to school all you want but that won't make you a natural musician or talented. All the tricks that makes us what we are really come down to a talent and the ability to hear something I personally feel you are born with.
And thats what most of the pro engineers whom are on this site all have in common. So, this really isn't the best place to tell us about something like this. Its kind of silly to me but I admire innovative thinking. Sometimes the most obvious is right there and we don't see it.
Reminds me of round trip processing.
Good luck on this one.
TallKite, post: 409100 wrote: You can also demo one mike vs. ano
TallKite, post: 409100 wrote: You can also demo one mike vs. another mike, or one effect vs. another effect, or one effect setting vs. another setting, or different order of the same effects, etc.
The most Mike's I've ever had in one band, is two... and they got along just fine, so I dunno why they'd be fighting each other.
As far as mic(rophone) choices though, I'm sorry, but that's the producer's role to make those choices, and I damn sure wouldn't expect a final mic choice to be made in a tracking room that is for all practical purposes, totally inappropriate for making tracking or mix decisions in. I'd rather make the most informed decision I could... in the control room/mix position with real vocals, guitars, hammond, horn, drums, etc. coming through my monitors.
(Same goes for any timebase fx or verbs.) Other than to make a space for the instrument(s) to be placed in during tracking... and that should be done with the same monitoring environment that the musician is going to be listening to during the actual tracking session... whether it be foldback monitors, or cans.
audiokid, post: 409101 wrote: Audio and creative talent is of co
audiokid, post: 409101 wrote: Audio and creative talent is of course more subjective. You can go to school all you want but that won't make you a natural musician or talented. All the tricks that makes us what we are really come down to a talent and the ability to hear something I personally feel you are born with.
And thats what most of the pro engineers whom are on this site all have in common.
I'm not talking about about talent and ability. I'm talking about a simple way to greatly speed up the process of making a test recording and listening back to it. Of course you have to have the ears to be able to use what you hear. But those that do, such as yourself and all the talented pros on this site, can use this method to work faster.
audiokid, post: 409101 wrote: So, this really isn't the best place to tell us about something like this.
Is this the wrong subforum? My apologies if so! This is the home recording subforum, labeled "Help and Advise for Home Studio", and this method absolutely rocks for people without control rooms.
Are you saying I don't have enough talent and ability to have a worthy idea? Not being sarcastic, honestly trying to understand why I shouldn't be sharing my idea with you and the other pro engineers on this site. I do hope I'm misunderstanding you, so easy to do on the internet. But why else would you use the word "talent" three times in one paragraph, before telling me not to post about this here?
Or are you saying that you pros are so talented and experienced that you never make test recordings? From the subforum description: "How to get started? Beginner recording questions. How to hook gear up. What is compression, interface, levels, gain staging, budget gear, USB mics more.." Certainly the people reading this forum for guidance aren't above making test recordings!
MadMax, post: 409103 wrote: As far as mic(rophone) choices thoug
MadMax, post: 409103 wrote: As far as mic(rophone) choices though, I'm sorry, but that's the producer's role to make those choices, and I damn sure wouldn't expect a final mic choice to be made in a tracking room that is for all practical purposes, totally inappropriate for making tracking or mix decisions in. I'd rather make the most informed decision I could... in the control room/mix position with real vocals, guitars, hammond, horn, drums, etc. coming through my monitors.
This is the home studio subforum. Are you saying that every home studio has a producer and a control room?
You sound like a very generous and kind man so I don't want you
You sound like a very generous and kind man so I don't want you to get the impression that we are both poking at you.
I use the word talent because there are some things talented people just do and it works, apposed to others that over process everything, or are always trying to reinvent the wheel to something that doesn't need reinventing or patch something that doesn't need patching when it really only needed to be put together right in the first place.
The concept may be cool for newbies but why are we wanting newbie to do something that pro's would never do. And to be honest, I actually wish there were a lot less people goofing around in this business than there is today. Affordable recording has turned the industry really stupid. This is part of affordable recording. So yes, its a perfect concept for the home recording forum.
You are welcome to continue sharing your concept but I have a feeling you will be explaining yourself until you are blue in the face.
My appologies if I don't get it too. I'll keep an eye on this.
You should also post this over on(Dead Link Removed)
Cheers!
audiokid, post: 409106 wrote: ...why are we wanting newbie to do
audiokid, post: 409106 wrote: ...why are we wanting newbie to do something that pro's would never do.
Because newbies don't have the luxury of a control room and an assistant. And here's the thing: we all start out as newbies.
How did you learn to record? You made test recordings and listened to them, right? And then you experimented with different mikes, different placement, etc. This method just speeds up that process immensely. You hear the result of your experiment immediately, without the press-record-run-for-five-seconds-press-stop-press-play-listen-press-stop-press-undo-to-delete part.
audiokid, post: 409106 wrote: ...patch something that doesn't need patching...
The traditional method works fine, it's just slow.
[quote=audiokid, post: 409106]You should also post this over on(Dead Link Removed)
This has gotten a great reception at Gearslutz:
[[url=http://[/URL]="http://www.gearslut…"]Great trick for demoing mikes - Gearslutz.com[/]="http://www.gearslut…"]Great trick for demoing mikes - Gearslutz.com[/]
MadMax, post: 409093 wrote: Just be aware that every echo repeat
MadMax, post: 409093 wrote: Just be aware that every echo repeat, you get phase shift and comb filtering that changes the frequency response of the next loop back/echo... and over the course of say... 3-5 seconds you can expect to loose up to 50% of the original frequencies.
Demo'ing mic's like this is far from exempt from the same frequency shifting/masking.
Hey MadMax, you never addressed my reply:
TallKite, post: 409098 wrote: all the echoing does is take the digitized signal from the mike and store it temporarily in the computer's memory, then play it back. If you instead record the same signal to hard disk, and then play it back a minute later or a day later, you are listening to the exact same ones and zeros. Digital is digital. That's why no-one talks about how one brand of hard drive sounds vs. some other brand. So if echoing adds phase shift and comb filtering, so would straightforward recording.
So does straightforward recording cause phase shift and comb filtering? Or am I misunderstanding something?
TallKite, post: 409105 wrote: This is the home studio subforum.
TallKite, post: 409105 wrote: This is the home studio subforum. Are you saying that every home studio has a producer and a control room?
Yes...
There's someone making decisions for mic's, mic placement, FX, Verbs, comps, gates, etc... In a home based studio, that's usually the owner. And there is at least something called a mix position... whether it be in a one room studio with headphone monitoring, or a separate room used as a control room using monitors.
And if you'll go back and halfway pay attention to your own original post, you're method specifically cites references to multi-room setups, indicating a tracking room and control room... So, I'm not exactly sure what you're implying.
TallKite, post: 409108 wrote: How did you learn to record? by
TallKite, post: 409108 wrote:
How did you learn to record?
by recording. here's how it's done. get a good song. get a good performer or performers. put it in a good sounding room. put up a mic. (anywhere will do. some will sound better but if you have the first three elements it's not that important. keep it simple.
[quote=TallKite, post: 409108]This has gotten a great reception at Gearslutz:
(Dead Link Removed)
ohh! well that makes all the difference.
now i really feel i should look into it more.
As far as phase shifting... you do understand that time=distance
As far as phase shifting... you do understand that time=distance, and in relationship from one track to another, as in tracking a drum kit, you get phase differences between any two microphones due to distance. I mean, this is pretty well established as a reality of physics.
Whether you are .0001mS or 10.00 Seconds distant between any two signals, if the signal durations are sufficient to be captured at the same time and both sources are combined at the point of hearing, there is a potential, and (IIRC) inversely proportional to the square of the distance, chance for an effect of acoustic, or electronically generated harmonic modulation that nulls frequencies... commonly referred to as "comb filtering".
An odd phenomena occurs when a signal diminishes over time in a reverberant environment... In that because of absorption, any reflected energy will be reduced by the amount of energy absorbed at the frequency(s) of absorption.
Delay plug-in's and time based delay units do this function to the extreme. Most, if not all, delay units have the inherent nature of these absorption coefficients built into their delay algorithms. About the only units that I've ever seen that are specifically designed not to add any of the coefficients to the delayed signal are line delays used to time speakers in large installations such as for speech intelligibility, or short delay plug-ins.
I'll reiterate what audiokid said... What you do sure is a buttload of extra work and while it may save you personally some time now... it would take me a lot of extra time to setup and undo to get to the point that I would be ready for tracking. Maybe in super small operations it would be of some advantage, but I just don't see it being of any use to bedroom folks... They're already using headphones to track, and most likely to mix. This method only really makes sense for two room (or more) situations... and the source of my curiosity, as to how this is faster... especially if you're running something like a small analog console or something like a Presonus... which again, you would likely already have a much more straight forward foldback system in place...
So, the only "advantage" I see your methodology is a small 2 room shop that's running a DAW solution that's seriously under powered, or an analog solution suffereing the same inadequacies. For those folks... maybe it works... and good for you for helping these folks out with a twerky fix... but for the vast majority of multiroom home studio rats... I'd suggest you just dig in and learn to do things the right way.
The issue isn't just slappin' the signal routing around... it's not what most professionals would do for the sake of the artist we're recording. Having to fiddle fart around with a gate and a delay just for mic placement is more of an opportunity to interact with a client, but it also exponentially increases the the time I'm not concentrating on the client as it has to be spent working on the gear for the sake of the gear... NOT the client. NOT COOL.
From the time the client comes in, that person should be your sole focus.
There is a certain amount of academic methodology to mic placement and mic selection... if you're doing any amount of recording that warrants a large one room, or a two room (or larger) facility, you've likely already discovered this. And tweaking with mic positioning is pretty critical to know when you've gone PAST the optimal point. I cannot honestly say that I can trust anything I'm hearing until I'm in the mix position... at what should be the most accurate place in the studio to make those decisions.
For the few times that you do get stumped on making that decision in a timely manner... That's just part of learning the craft of recording.
Humm?? This should have been titled Trick to be read by newbie o
Humm?? This should have been titled Trick to be read by newbie only. smoke
I think it's a clever Idea but flawed in the chances to execute perfectly ! The perfect execution would happen If you use a delay plugin which plays back some seconds, only once, without degradation or modification of the signal and assuming you are in an ideal listening position which would be hard to do if you are bending down to a bass drum.
Now let's not only disaprouve but suggest an alternative.
Let's assume that you are alone with only one musician so no help for the mic placement.
What you could do hit record and make the placement tests but before any notes on the instrument use your voice to record a description of the exact position.
(ex. placement 1 ; 2 inches from source angle 45% ... etc....)
Once you've recorded all the placements you can think of, sit down with the musician at your mixing position and compare all the positionning and then decide which one to use.
How's that.. ?
Next I put a long delay plug-in on the master track and set it f
Next I put a long delay plug-in on the master track and set it for 3-4 seconds. Now the fun begins. I hold the mike up to the instrument (or amp) and have the musician play a short test riff (perhaps an arpeggio). Then we listen to the riff echo in the monitors. I move the mike around and he/she plays the exact same riff, and we listen again.
all the echoing does is take the digitized signal from the mike and store it temporarily in the computer's memory, then play it back. If you instead record the same signal to hard disk, and then play it back a minute later or a day later, you are listening to the exact same ones and zeros. Digital is digital. That's why no-one talks about how one brand of hard drive sounds vs. some other brand. So if echoing adds phase shift and comb filtering, so would straightforward recording.
Then, as PC mentioned above, why not just record to your DAW to begin with, and then analyze and adjust from there?
With your process, you're assuming that a delay plug on the 2-bus will repeat exactly the same as the incoming signal in terms of diagnostic fidelity, when with delay plugs - or any plugs for that matter - there are other parameters involved...such as hi/lo freq damping, diffusion, phase, ratio and dispersal...not to mention the ongoing debate as to what plug ins truly sound like in terms of sonic integrity...sure, they can be used successfully as their intended purpose, which is as an effect, but they weren't designed to be used as a diagnostic tool, which is what your process assumes.
I understand you being low budget and not having a control room, assistant, secretary, kiereg coffee machine, teak wood floors, LOL etc.
But it seems it would be just as easy, (probably easier) for you to simply tell the player to reach down/over/across and adjust the mic for you... on that level I don't know of any musician who is arrogant enough to say "no" to that request when paying $10 bucks an hour (or whatever budget rate they are paying...and even if they were big stars I can't believe that any musician wouldn't be helpful in that scenario, regardless of their success level.)
I applaud your motivation in trying to make a process a bit easier, and I suppose that if it works for you, then that's all that matters. Hell, I'm all for whatever works if it saves me time, and as a 30+ year veteran in these crazy trenches on all levels, I've never learned enough to where I turn away from a good idea, regardless of the experience level of the person suggesting it - if it works and if it saves me time - but honestly, your process sounds more complicated than it needs to be, and I'm not sure I believe that the returning signal is the exact same as that of the incoming signal.
IMHO of course.
Here's what I don't understand... is why these manufacturer's do
Here's what I don't understand... is why these manufacturer's don't have direct monitoring.
It's actually pretty lame that they make such inadequate gear...
oh wait...
yes they do...
It's there in all the high end gear...
I guess that's the price the industry personnel pay for the glut of cheap gear being dumped on the prosumer...
creating a false market for the manufacturers to gain huge profits on the backs of those who are willing to dump their ducat's in search of the elusive "hit record", when there's little hope of that happening...
completely bypassing all logic of providing a professional level of product that IS necessary to operate with any kind of sustainable source of revenue...
except to say they make these products for the "home hobbiest" with "professional" quality components...
to wit,
that is an indication that professional operations standards are lowered??
or that the technology is so much better that it's cheap to make professional level gear??
Which the logical answer is that it is obviously some combination...
until you start dancing with the big boys.
Then all of a sudden we revert to "good shit ain't cheap, and cheap shit ain't always good"...
Which at just one step up from the level of "cheap gear", you get a product level that DOES have adequate track monitoring that allows one to live monitor tracks that negates even having to do many test tracks at all.
Hell, even my old Smackie 24•8 Bus let me do live monitoring with my ADAT rig just fine.
MadMax, post: 409132 wrote: or that the technology is so much b
MadMax, post: 409132 wrote:
or that the technology is so much better that it's cheap to make professional level gear??
i think anybody who has paid 1k ish or far more for a darn mic knows this to not be true. cheap gear has always been the same, affordable, w/ compromises. take the mackie big knob, feature laden as any berr.. pos, yet extremely colored, and (degrading imo).
i dunno i'm sticking to my level of knowledge and starting w/ a dynamic just off the grill, in the middle of the speaker (between the cone and egde), and a dynamic a few inches off the snare. its enough work going back and forth to the room(s), and geez how about amp settings. a typical 57 in that place shouldn't sound very bad in general unless the guitar/amp does.
i say keep it simple sir. i recently recorded terrible guitar tones, to show them how terrible they sound, and it worked, they don't like their own guitar sounds anymore. i'm not some master at all, but i do know the less in the way the better (unless you really know what your doing), and it shouldn't be difficult to move the mic. whether your comparing a recorded signal, and making changes, or real time listening, its hard enough to make decent sounding reproductions, lets not add more to the process.
as far as mike goes, i place him behind the kit, or somewhere at the bar. mike likes the atmosphere.
MadMax, post: 409132 wrote: Here's what I don't understand... is
MadMax, post: 409132 wrote: Here's what I don't understand... is why these manufacturer's don't have direct monitoring.
There is budget gear with direct monitoring. Focusrite 2i2 and others in the same categorie. Thing is, I wound not trust it anyway. Chances are what you hear live and what is been sent to the computer via electronics and converters are not equal.
So to me, record and playback to make decision is the way to go.
What got me at first impression was what kind of projects is the
What got me at first impression was what kind of projects is the poster doing as a sole proprietor in a single room environment that doesn't allow for time to really set the mic placement?
I DO see now that I must learn to trust my echo's much much more. I'll bet there's at least twenty or thirty of em in my plugs that I've never used at all........ ;-}
Sorry...being VERY facetious. Max explained it. The very nature of the algorithms written into DAW based delays would indicate that many of the acoustic anomalies occuring in real time have been provided for in the sample.
So......what you are hearing back aint necessarily whats going in........jus sayin
Just be aware that every echo repeat, you get phase shift and co
Just be aware that every echo repeat, you get phase shift and comb filtering that changes the frequency response of the next loop back/echo... and over the course of say... 3-5 seconds you can expect to loose up to 50% of the original frequencies.
Demo'ing mic's like this is far from exempt from the same frequency shifting/masking.
It's definitely a handy trick for those starting out and with little experience.
What is actually the easiest, and smartest way, is to get the musician to play/tiner/futz with the tone/sound until they're happy with it, and while they're dialing in on their sound, you actually use those two floppy things on the side of your head that are hopefully connected to the big gray mushy thing inside that coconut that sits on your shoulders, to determine where the optimal sound of said instrument to be recorded is, in your environment. THEN use your headphone monitoring/foldback to do rough placement, and then record and playback on the reference system... as that's where the rubber meets the road... and takes anyone with nominal experience considerably less time.
Just as an example, after 20+ years of recording and mixing, I'm usually within' 1/8 to 1/4" of optimal placement when I set mic's - if not on it right out of the gate. 10 years ago, I was probably able to hit the mark or be within a 1/4"-1/2"... 15 years ago, I was probably within' half an inch or so most of the time. So, experience is probably the best teacher... and the more you do, the better you become at listening and making better mic position decisions.