So, i filmed a performance recently and couldn't hear how poor the mix was coming off of the soundboard during the recording.
I'm having a hard time mixing in the room mic in post to get a good sound, like i normally do.

My questions are:
Do most boards have a pre mix out? Is that a better/preferred option in smaller rooms that have a lopsided mix?
I understand that some boards can have a second mix . . . is that a major problem for the operator and is that a better option than a premix out? (understanding that it is too much to ask the operator to monitor a second mix during the performance).

So, i filmed a performance recently and couldn't hear how poor the mix was coming off of the soundboard during the recording.
I'm having a hard time mixing in the room mic in post to get a good sound, like i normally do.

My questions are:
Do most boards have a pre mix out? Is that a better/preferred option in smaller rooms that have a lopsided mix?
I understand that some boards can have a second mix . . . is that a major problem for the operator and is that a better option than a premix out? (understanding that it is too much to ask the operator to monitor a second mix during the performance).

I have assumed that the mix that was being done was to achieve the best for for the audience, often people take a mix of the console and expect a "perfect" broadcast mix. Most front of house mixes are exactly that... "for front of house" and are there for "sound reinforcement" of the things occurring on the stage, if that's the case it will NOT be a full and complete mix.
Most desks can / could be able to deliver other variations of a mix if the operator is aware of what you want and they are capable of doing it, If that is what you want talk to them, days in advance so they can configure the desk to your needs, don't just turn up with 1/2 hour before the performance and expect them to reconfigure the console. Be aware you have just doubled the operators work and doubled to room for errors .... and you want them to do it for free and out of the goodness of their hearts.....right? And you wonder why they might not be happy with your request.
To do it properly mic everything, split every mic one feed going to the FOH mixer and the other feed going to the audio mixer (for video shoot), bigger events will also have an additional split for artist fold-back.

It really is a big burden on the FOH sound man, and is often something he can't realistically provide.

I've been in situations where I was running FOH in a 2000+ seat auditorium. Obviously I'm listening with my ears live to the room sound. I can't keep putting on headphones and listening to a recording feed at the same time... that would divert from my main (paid) job of providing a good mix for the house. I'm not going to compromise the job I'm supposed to be doing, because of a job someone else hopes I can do to help him out.

You might get away with it in a very simplistic situation (e.g. a very small number of performers, not moving around on the stage, and hopefully not playing at ear-splitting levels). Otherwise, as Brian points out, you really need two (or three) mixing boards and two (or three) operators, each dedicated to one specific task.

Except for trivial situations, it is for all practical purposes IMPOSSIBLE for a single person (no matter how skilled) to do a good FOH mix AND CONCURRENTLY a good recording mix. The requirements are substantially different, and monitoring the results properly are MUTUALLY EXCLUSIVE.

That is why we typically use a duplicate mixing console with all the same input sources available to create a "recording mix". Furthermore, almost always we need ADDITIONAL sources (microphones, etc.) to create a reasonable mix suitable for an audience that isn't sitting in the venue for the live event.

Unless it is something simple like a lecture, etc. it is unreasonable to expect the FOH person to provide you a decent recording mix.

If you have time for trial and error sound checks and playbacks, sometimes a 'usable' mix may be had using the board's pre-fader (or post-fader) Aux. sends. Of course there's many variables and caveats. Live mics should be available as well on separate tracks.

In smaller/simpler shows, it's been my experience that it's possible for a single person to create two mixes. However, many live sound engineers just aren't that familiar with what's needed for a recording mix. Those who've had some formal education are familiar, but, especially in smaller venues, you run into a lot of soundies who can't easily set up gain structures for a recording mix, much less manage two simultaneous mixes.

In a loud venue, like a club, you can't really do a separate mix on headphones because there's a lot of leakage of the direct sound into your cans.

Recommendations above for a split and dedicated 2nd mixer and recording engineer are good, but often not possible for reasons of money, time, and the scale of the project. But, they are often the best way!

OTOH, in Portland, I can rent a 24 track recorder for $75/day. With some prearrangement with the FOH engineer, I can get an unmixed feed for each board channel. Take his FOH mix too, and add a preamp and an audience mic. This has worked well for me - I'm something of a soundie myself, and have worked out post syncing & etc.

The cheep and fast method is to get a stereo mic in just the right spot to hear a good mix off the stage and PA. Of course, that frequently is impossible with a mic stand, because it might be right in the middle of audience or dancers. Even then, it can be worth the time for a ceiling rig.

In making choices about approaches, much depends on who is sponsoring the recording and how important it is to them. Will they block off seats for you? Allow budget for dedicated mixer, engineer and split? Access for rigging? Call (and pay) the FOH engineer a couple hours early to help?

Seth, those all sound like good suggestions. Absolutely the best option is to rent (or own) the multi-track and record every channel separately, then do a mix at your leisure in post.

I have tried finding the "right spot" for a stereo mic. In theory that should work. In practice, I've run into two issues.

First, people often expect to hear vocals sounding as if they were close-miced. I have never successfully found a spot anywhere in the house, where room sound has a close-miced sound to it. So, try as I may, I always end up with a recording that sounds as if it was "recorded in the room" (however accurate it may be) and not "studio-like" or "broadcast-like" close micing.

Second, our brains do a lot of processing to what our ears pick up. You can demonstrate this by having someone stand 30 feet away from you in a fairly live room, and speaking. When you listen with both ears, your brain does some amazing time/phase processing, and the speaker will be quite intelligible. If you plug one ear (with your thumb, an earplug, etc.) then you will suddenly hear a lot more of the room reverberation and echo, and the speech will sound much more distant and the intelligibility will suffer.

The relevance of the second point is that, when you walk around the room and try to find the "perfect spot" for your mic, your brain is doing a lot of processing to the signal from your ears. If you put your mic there, and later play back the recording on speakers, you will hear a lot more "room" and the recording will sound muddy and reverberant, compared to what you expected. (Indeed, if place a binaural pair of mics in that "perfect spot" and then playback over headphones, it will sound exceedingly realistic. But that recording won't do you much good in a conventional mix intended for loudspeaker playback.)

So, in fact, when looking for the "perfect spot" I often walk around with my thumb in my ear. I end up closer to the sound source (stage or speakers as the case may be) but the recording tends to be better. Still, the first point (above) applies, and vocals still sound distant, and lack clarity and presence, compared to that close-miced "broadcast" or "studio" sound that people want.

Yeah, I'll second the suggestion for a multi-track recorder and mixdown in post.

Best method - mic splits, dedicated desk with each source recorded separately. Standard practice for broadcast of a live event. Downside - expense and needs multiple staff, and somewhere to be able to hear what you are recording. Almost impossible if the event is loud.

Anything else is compromise, and unless you can hear what you are recording, it's fingers crossed until you get to the edit.

I have to stand with everyone who says the board op is there for room sound. No way do they have the time, or the isolation to be able to make recording decisions in a live environment. They also are getting paid to provide the service. If people want a dedicated mix, which most mixers could physically do - are they getting extra? It is possible to create a separate mix for video - but it means an extra person - and space is obviously tricky if two people are trying to use one space for two jobs.

It is possible to run a multi-track from the desk = something we do reasonably often, but it means some compromises. The sharing of the pre-amp means that the live sound op cannot change the pre-amp gain during the show as this impacts on the recording. A proper split at the stage end and two mixers solves this one, of course.

...people often expect to hear vocals sounding as if they were close-miced. I have never successfully found a spot anywhere in the house, where room sound has a close-miced sound to it. So, try as I may, I always end up with a recording that sounds as if it was "recorded in the room" (however accurate it may be) and not "studio-like" or "broadcast-like" close micing...
I agree 100% with Greg's observation here, I meant to include something like "this method will usually produce a recording that has more indirect sound than we might want."

So much depends on the room! A club doesn't *have* to be a reverberant echo chamber, but frequently is just that.

I started to get into what a producer's role might include in making these decisions. Greg's observation points out that usually we're trying to balance three factors:
1. Cheap
2. Fast
3. Good

Most often we only get to have two of them. That's an oversimplification, perhaps, but there's more than a little truth to it.

I'd reframe Paul's comment: "Anything else is compromise...", to, "It's all a compromise!" Even with a split before the FOH preamps, we're adding more pieces, more labor, more time, more potential points of failure. But, yes, I agree that when we can support that approach, it's best. Better include a backup recorder, too, because they can be flakey!

Remember also, whether doing the "recording" mix live, or multi-tracking and mixing down in post-production, except in rare circumstances, you will need additional ambient microphones to pick up things that aren't comprehended in the FOH mix. At minimum, an audience/ambience mic/track (or two). And maybe several others depending on the source(s) of the sounds of interest.

I am taking a class in Live Sound right now and we're taught that a separate recording mixer is always needed, as everyone has said in this thread.

But one solution that I think is a good compromise is to close-mic a speaker cabinet on or near the stage. I learned this from the wedding pros. Only requires a single dynamic mic, or two for stereo. You can even use a video mic like the RE50 or 635 if you already have one.

That way you get the close-miced sound of the vocals from the speakers, plus the drums and other less-amplified instruments get captured in roughly the same ratios as the audience would hear.

It's not perfect in that you do get some room reflections and the degradation of an "unnecessary" roundtrip from electrical to acoustic energy, but it's quick to set up. Because it requires little to no attention it's much more reliable than depending on the FOH mix.

I'm still working out how to place such a mic unobtrusively but stably. I've heard of wedding guys using drum mic clamps to clamp a mic onto the speaker, but modern speakers often don't have lips to clamp onto. Obviously one could use a low boom mic stand like musicians typically use for micing bass cabinets, but that seems visually distracting and unstable if there are standing audience members nearby.

What I'm now thinking is to try a heavy desk stand for floor speakers. Atlas makes one designed for micing kick drums. For stand-mounted speakers some kind of huge rubber band or velcro strap (with padding to prevent the mic from rattling against the speaker) seems like it might be the least obtrusive solution.

Putting a mic infront of a PA speaker box should be regarded as desperation audio...
It's like using a piece of string to hold your trousers up, yes it works but it's not the best way to go about it.

Desperation audio is a pretty strong term for something that's more reliable than the alternatives, but I would agree this is more for documentary type work than music quality.

But one solution that I think is a good compromise is to close-mic a speaker cabinet on or near the stage. (snip) That way you get the close-miced sound of the vocals from the speakers
Actually, what you get is the close-miced sound of PA speakers, which are reproducing everything in the mix; not a close-miced sound of the vocals. And if they're typical PA speakers, this will sound significantly different from the actual close-miced sound of the performers.

It's much more difficult to make accurate transducers than to make accurate electronics. Of all the transducers, high-level loudspeakers are perhaps the most difficult to make accurate. And PA people (with the exception of the Grateful Dead's "wall of speakers") are always making some sort of compromise. Consider, too, that speakers have multiple drivers, covering different parts of the frequency range, with varying phase and timing errors related to crossover design and cabinet construction; that makes it almost impossible to find the exact location where the sound reaching the mic will have good frequency and phase response.

Now add into the equation the fact that a lot of PA operators use really bad EQ, or over-drive the power amps and add an audible amount of distortion to the signal. Take the sum of all these unknowns, and muddy it up by running it through a set of (often marginal) PA speakers.

Nope, IMHO that will sound like a recording of PA speakers. (There's a good example of this, in another thread in this forum... someone claiming to have close-miced wedding participants, while playback of his demo video reveals that he actually has a recording of PA speakers... some of it apparently made with an on-camera mic. It is not pretty.)

For stand-mounted speakers some kind of huge rubber band or velcro strap (with padding to prevent the mic from rattling against the speaker) seems like it might be the least obtrusive solution.
And which one of the drivers did you want to record? The woofer? The midrange horn? The tweeter? The drivers can be several inches to several feet apart; in some big touring rigs they can even be in separate (stacked) cabinets. This idea will only make a marginal sound into a truly terrible sound.

Desperation audio is a pretty strong term for something that's more reliable than the alternatives, but I would agree this is more for documentary type work than music quality.

Not at all... Desperation Audio is absolutely what I mean, I don't want people to read this forum and think putting a mic in front of a speaker is even remotely an acceptable practice.
Many people have gone down the DSLR path for the quality pictures it can deliver, yet good audio practices are seeming to be ignored.
There are so many times recently have I seen interviews at events being done with a DSLR with just a mic on top of the camera about a metre away from the person being interviewed (and often in a noisy environment), the physics of sound just don't change, the results will be just awful.

I am self employed as a freelance sound engineer doing most of my work for Broadcasting Networks its interesting to see the number of new people coming into the industry that have done their education at institutions that accept poor audio as normal on video shoots, it then becomes a very steep learning curve to get the material they shoot up to acceptable broadcast standards.

I thought the role of a forum like this is to educate, inform, inspire and improve those interested in the subject.

Not at all... Desperation Audio is absolutely what I mean, I don't want people to read this forum and think putting a mic in front of a speaker is even remotely an acceptable practice....I thought the role of a forum like this is to educate, inform, inspire and improve those interested in the subject.

+10 Absolutely. Putting a mic in front of the venue reinforcement system is only one step above the absolute minimum: just using the microphone on top of the camera. I got into video (from exclusively audio) primarily because of the pathetic state of audio for video. "Desperation audio" is a perfect description of the practice. NOT recommended unless you absolutely have no better option.

Okay okay, so y'all are correct that micing a speaker is not broadcast quality, not hifi, probably not even musical.

But I still think for the original poster, who doesn't seem to know how FOH mixers work, micing the speakers would give a more consistently reliable and useful result than anything involving mixers. If you disagree, I'd like to hear how to get better audio without messing with a mixer (other than "hire someone who knows mixers"),

The person running the mixer probably knows enough about it to at least give you a feed for a simple production (talking head, etc.) Of course, you must communicate and negotiate in advance (even if "advance" means 90 minutes early.)

And if you are shooting something more complex, then you (or somebody) must make the production trade-off decision how good does the audio need to be? If you are shooting B-roll for the 11pm news, then throwing a mic in front of the house speakers for a 15 second clip is probably acceptable.

Okay okay, so y'all are correct that micing a speaker is not broadcast quality, not hifi, probably not even musical.

But I still think for the original poster, who doesn't seem to know how FOH mixers work, micing the speakers would give a more consistently reliable and useful result than anything involving mixers. If you disagree, I'd like to hear how to get better audio without messing with a mixer (other than "hire someone who knows mixers"),

A better way to do a show might be take a feed from the FOH desk (knowing it is NOT a full mix) put 2 or 3 short S/G mics on desk stands on the front of stage, mix the FOH desk o/p and the front of stage mics with a mixer and send it to TK 1 of the camera.... take the Camera mic off the camera mic and put it on a stand slightly away from the camera to avoid the zoom / focus noise for audience reaction / applause this signal goes to TK 2.
Do the final mix of stage sound and audience in post production.
The mixer needed is only a simple mixer you don't need to go top shelf like a Sound Devices even a mixer like a Behringer Behringer XENYX 1002B - Battery Operated 10 Channel 1002B B&H would work a treat, the mixer could be set up on a chair at the base of the tripod and then fed into the camera.
For very little cost the sound of your project would be improved beyond your wildest dreams.... (I'm not saying its broadcast level but it would much better than a mic in front of a speaker)

If you were to do it this way make sure that you monitor the mix of the audio signals... I often use 2 sets of headphones in setups like this, one set on the mixer and the other set on the camera (to make sure its actually getting to the camera through out the shoot)
I have seen so many times people not even monitoring the mix..... would you use a camera without a view finder...NO, so why would anyone use an audio mixer without listening to whats being mixed?

There is absolutely no point in micing the PA speakers, except that someone who knows absolutely nothing about audio can do it. It will be the wrong mix at a poor quality level.

If you're willing to settle for the wrong mix (what a mistake!) then you'd be better to just get a split off the board feed to the power amps. That would avoid any distortion from the power amps and all the unavoidable distortions from the speakers.

But since the OP has originally said he's not satisfied with the board feed, he won't be satisfied with the board feed through speakers either... it will sound worse than that he's already tried. Micing the speakers is entirely illogical in regard to the OP's question. And is a bad idea in general.

Thanks to all for the feedback. I've learned quite a bit from this . . . long way to go, though.

My clients are performers that rent the hall and pay the staff for the gig, so it's not like the engineers are doing me a favor, they are performing the job they are being paid to do, even though it's their second priority. If they require more pay for a longer soundcheck/setup, alright . . . me too!
Most of the time an elaborate and costly setup is not in the budget. Let alone a post mix from a multitrack recording - no way.

So this is my 'boil down' - increasing in cost and complexity. (for my needs)

1- basic: Room mic (on-camera or better positioning) +FOH mix
2- better: Room mic + Secondary mix determined at soundcheck -no engineer monitoring
3- a little mo' better: Room mic + second mixer monitored by me (i use a pair of Bose over-the-ear headphones that have pretty good isolation & the shows are usually not that loud)
4- best: Room mic + second mixer monitored by second engineer

So, next time, i'll tell the client/performer that we need to get with the SE to get a plan together on a separate mix. If we have to get a more accomplished engineer or a second engineer, that's the client's expenditure and decision.

I have a Mackie 14 channel mixer (1402-VLZ pro).
Would this be an option for use if the board doesn't have a second mix capability? It only has 6 XLR inputs. My clients frequently have Piano, Drums, Bass, 3 pc Brass, vocals.

It was obtained in a package deal years ago and largely sits idle and dusty. I've never been able to get clean audio out of it using 1/4" patch cords from Roland V-drums and other sources(background hum), but i'm not a sound guy and i've never taken the time to learn how to use it properly.

What do you think about taking just the vocal feed out of the board? The clients are vocalists and that's what i'm usually trying to 'clarify'. The band usually sounds okay on the room mic but the vocals lack that sharpness. Maybe this would be a safer way to go. Or at least feed it to the second camera as an option.

I know the Rode NTG-2 shotgun mic is not the ideal mic for this application, can you suggest an economical mic that would do a better job?
Any suggestions on books or educational sources to get me more up to speed on the basics of soundboards?

Thanks again, Soundies!

Ok. So, i was just thinking. I manage multiple cameras, sometimes locking off one angle to run to another position to adjust the second or third for a given variable. Obviously not the ideal, but necessary given a production's limited budget. Timed properly, it's a reasonable tradeoff.

So, why exactly can't a Sound Engineer check in on a separate mix once the house mix is stable?
Are they actively moving faders throughout every song in the setlist?
I'm not looking for perfection, but if this was performed even 2 or 3 times during a show it could have helped avoid glaring problems in my mix (ie. individual instruments way out front or not audible at all).

Would a secondary mix (is this the right term?) on one soundboard affect the FOH mix, when adjusted during the show? No, right? That's why it's a separate mix. I just want to get this straight.

5- Even better: Room mics + second mixer and mixed by second engineer in a isolated 'control room'

I don't think the band (or vocalists) would be happy with a vocals only mix. ( though it could be blended with the room mics in post)

The house mixer 'may' be able to send sub-mixes (stems) to your 1402.. they would likely be line-level, so you would probably need a 1/4" TRS <to> 1/4" TRS snake. That could give you 10 mono input channels on the 1402. Twelve if you also used the Aux. returns.

"( though it could be blended with the room mics in post)" - Yes, absolutely, that's what i meant.

"The house mixer 'may' be able to send sub-mixes" - so, if there's 5 mics on the drumset, this sub could aggregate the drum kit to one feed to the 1402, right?

Any other things to look out for? . . . .stupid question, of course there are!
What's the protocol on hand mic + lav mic . . . would they necessitate an individual feed? Or could that be setup in a sub mix -preshow. ie. set levels for each on the main board, then sent out to one line to 1402?

I guess this is where an engineer has to know the setlist and jump on muting that line after the song and switch to the other, huh?

"The house mixer 'may' be able to send sub-mixes" - so, if there's 5 mics on the drum set, this sub could aggregate the drum kit to one feed to the 1402, right?"
-- Right, IF.. the 'house' console has enough aux. sends, matrix or other assignable outputs.. and the engineer has the knowledge to set it up properly.

-- Lav mics are 'generally' not used in club/concert type music venues... there's no law against trying it, though a usual dialog-sensitivity lav may not be able to handle the high SPLs.

I guess this is where an engineer has to know the set list and jump on muting that line after the song and switch to the other, huh?
-- Not exactly sure what you mean, but in a band scenario, most inputs are usually not muted unless there's noise issue. (ground hums or EMI and such) BTW, I used to be (way-back when) a house engineer/mixer in a nightclub and dealt with unfamiliar bands almost every night w/o knowing their original songs.. OR even having a set-list from most cover bands for that matter.

Sorry, i didn't mean lav mic - they use a combination of handmics, mic's on boom (piano), and those mics on a wire that are fixed to their faces, headset? What do you call those?

That's what i mean . . . they walk off stage with a headset and come out with a handset, does the performer mute it on the transmitter or does the engineer kill it?
If the performer does it, then i wouldn't have to be fumbling in the dark looking for the mute button on my board while managing the camera. In any case, I guess a gooseneck led is mandatory for a second board under my tripod if i go that route.

yeah, "Headset mic".
"does the performer mute it on the transmitter or does the engineer kill it?"
Either or, whoever can be more relied upon.

"If the performer does it, then i wouldn't have to be fumbling in the dark looking for the mute button on my board while managing the camera. In any case, I guess a gooseneck led is mandatory for a second board under my tripod if i go that route."
-- Oh yeah, always have some kind of a mounted light. But I would suspect you wouldn't want to worry about muting and un-muting inputs when you have a camera to operate. Depending on your sub-mix feeds, it may not be possible anyway. The talent or the house-engineer should do it since if would also affect the house mix. Any semi-skilled FOH mixer should be able to handle it, tell him/her to "mute the headsets when they're off-stage" (or have H/Hs)
I assume your dealing with some kind of 'musical' and not a 4 or 5 piece garden variety rock band.
Good luck.

So, why exactly can't a Sound Engineer check in on a separate mix once the house mix is stable?
Because often there is no such thing as "stable."

Are they actively moving faders throughout every song in the setlist?
In many cases, yes they are. It depends on the instrumentation, individual performers, type of music, venue, etc.

Let's just consider one imaginary scenario. Let's say the trumpet player starts getting too loud, or too close to his mic. The FOH engineer will pull down the fader for that mic until the mix sounds right to him. It's a very dynamic process.

OK, now what about your recording pre-mix? Pulling down the fader for the PA does not help your pre-mix; the trumpet is still too loud. The engineer just pulled down the fader and adjusted "by ear," but how many dB was that? He didn't take the time to look at the scale on the fader before and after he made the adjustment. So... how much should he turn down the rotary pot that's controlling the recording mix? How many dB? We don't know.

Should he put on his earphones and listen to your pre-mix while adjusting that rotary pot for the trumpet? Uh, the clarinet's just starting a solo, he needs to re-adjust the faders for the PA mix again. No time to piddle around with earphones and a pre-mix. He doesn't want to get reamed later because he screwed up the PA mix. You (with the recording mix) are just SOL.

Why don't the musicians put a comic book on their music stand and read that while they're playing the gig? Because playing is very intense business and it requires 100% of their attention and skill to do the best job they can. That's the difference between an artist and some schmo driving a garbage truck. The same thing applies to the FOH engineer. Doing his job is a very intense business and it requires 100% of his attention and skill. He's an artist, too. (It's the same reason you don't want your surgeon talking on his cellphone while he's chopping out your spleen. It's the same reason you shouldn't text and drive at the same time. You concentrate on the one job that you're supposed to do, period.)

Yes, there may be some instances, at some times, with some bands, with some types of shows, when the FOH engineer has more flexibility than in the above example. But there's no guarantee if or when this magical moment will occur. The engineer needs to keep his eyes on the band, his fingers on the faders, and his ears open at all times. Anything less than that is asking for increased mistakes with the FOH mix. Believe me, I've gotten distracted and I've made those mistakes and it is very ugly!

There is some great info on this thread and some really questionable info as well...

A couple points. Micing a pa speaker, while usually not a good idea, has one situation where it will be fine- when you are dealing with talking heads only. This could be a wedding where you have just a couple people talking and no other music or live performers. It could be lectures, graduations or other similar situations. When a band is involved, as others have said, this is useless for good sound.

Taking feeds from a FOH console is a "do at your own risk" kind of proposition. If you are taking the mix out, don't expect good sound. As others have said, the FOH engineer's job is to provide good sound for the hall. Period. Notice that good sound for the vidiot that showed up 10 minutes before showtime expecting a feed isn't on that list of priorities. With few exceptions, taking direct outs of the channels of the consoles will also be a post fader proposition. When the mixer moves a fader, changes EQ, inserts a comp or anything, you get that on your mix whether you want it or not. Remember, they are mixing for the house- not for you so don't expect great sound there. It will likely be better than the mix out, but you cannot count on it. The exceptions I mentioned? A couple analog consoles have mods that can give pre-fade outputs (a lot of Midas consoles come to mind). A couple others (like soundcraft large consoles) will give you the ability to turn aux 1 into a prefade out with level control. A cool feature, but your FOH mixer needs time to set it up.

Ironically, the larger the venue, the more likely your FOH feed will be good quality. The more space to fill, the less effect the stage volume has on the sound in the house. Therefore, the PA mix has more to it. In a small club, you may find that the drums or a guitars are way low in the mix because of the stage volume providing the sound. In a big venue, it all goes out there over the PA so you end up with a better product.

The most reliable way to get good sound for a band is a "real" record rig. Split all mics, bring preamps and/or a console and make a recording. Means you need other people working with you and a lot more prep time, but if the audio matters at all to you, this is what you have to do.

--Ben

the larger the venue, the more likely your FOH feed will be good quality. The more space to fill, the less effect the stage volume has on the sound in the house. Therefore, the PA mix has more to it.
That's a good point and I tend to agree with that.

Micing a pa speaker, while usually not a good idea, has one situation where it will be fine- when you are dealing with talking heads only. This could be a wedding where you have just a couple people talking...
Well, in this case the mix will probably be fine. But the audio quality will still be marginal, IMHO. You will get a recording of a PA system, and not a realistic close-miced recording of the persons speaking. The PA speakers will have some characteristic sound to them, they might have EQ added that is not the best, there may be some distortion from the amplifiers. (And there is bound to be some amount of added room noise, crowd noise, and reverberation.)

In this scenario, I think you would be better off to get a split of the PA mix coming out of the board, before it goes to the EQ, power amps, and speakers. It will be exactly the same mix that you hear from the speakers, but much cleaner and more realistic than you'd get with a mic pointed at the speaker stacks.

The best method I've seen is:
Channel 1: Off camera mic on a mic stand. This will give you ambient and audience applause.
Channel 2: Mic at front of stage sent back to the camera via wireless. This gives you ambient with minimal audience. This is also a good safety source (let say there is a screaming baby at front of your other mic)

Mix in post will give you nice stereo sound.

You can also substitute board feed into channel 2. I let the Sound guy know in advance I need a board feed if the client requests it, I also come early and do a levels test. Despite doing all this, there is a good chance that the board feed will been unusable, so don't rely on it. If the board feed is that important another person has to be put in charge and test need to be done in advance of the performance.

Most clients think all that is involved is running cable from the board to your camera and I used to think the same thing. I now try to temper the clients expectations of relying on a board feed. Its an extra that if it works can enhance the ambient sound.

The best method I've seen is:
Channel 1: Off camera mic on a mic stand. This will give you ambient and audience applause.
Channel 2: Mic at front of stage sent back to the camera via wireless. This gives you ambient with minimal audience. This is also a good safety source (let say there is a screaming baby at front of your other mic)

Mix in post will give you nice stereo sound.

You can also substitute board feed into channel 2. I let the Sound guy know in advance I need a board feed if the client requests it, I also come early and do a levels test. Despite doing all this, there is a good chance that the board feed will been unusable, so don't rely on it. If the board feed is that important another person has to be put in charge and test need to be done in advance of the performance.

Most clients think all that is involved is running cable from the board to your camera and I used to think the same thing. I now try to temper the clients expectations of relying on a board feed. Its an extra that if it works can enhance the ambient sound. Mixing those two mics might give you two-channel sound but it won't be stereo sound. The mere fact something is coming from speakers on the left and right doesn't make it stereo.Stereo requires that the mics be arrayed in an arrangement that will recreate the sound stage of a person listening from the optimal "sweet spot" in the audience with sound from one side of the stage coming from one speaker and sound from the other side of the stage coming from the other.

It gives a fuller sound with depth that is more pleasing to my ear than one mono mic. Whether it meets the definition of stereo is not of great concern to me.

This is very slightly related to a technique that was proposed, tried, and abandoned back in the '50s in the early days of two-track recorders.

I forget who originally proposed this system... seems to me it was someone who had other audio credentials. A well-known name like Avery Fisher, or Henry Kloss, for example. But I don't think it was either of those two. If I research it and find the answer I'll update this post.

The idea was to place one mic front and center in the sweet spot, to record the main musical information; then to place the other mic far back in the hall to pick up mainly reverberant information. These would then be played back on a two-channel system, with one speaker front and center relative to the listener, and one speaker behind the listener. In other words, a mono main channel and a mono "surround" channel.

If you recorded it and played it like that, it was somewhat realistic. Yes, the orchestra was in front of you, but all 40+ musicians were located in one narrow point source. And yes, the hall's reverberation was behind you, but all in one spot, behind the center of your head. Indeed, it was "fuller" than mono, but it proved to be a lot less realistic than left/right stereo... that's why it was quickly abandoned.

(In fact I made some recordings like this as a test at the time. I was recording for AM radio broadcast, so I did my best to get a good mono recording with one front mic... we used that channel only for broadcast and that was fine. I recorded the reverb mic on the other track of the tape, and I played that experimentally at home on my L/R stereo system. If I sat between the speakers it was acceptable, but not nearly as good as real stereo. After all, when you listen to a symphony, you expect the violins on the left and the bass fiddles on the right and everyone else spread in between; hearing a realistic stereophonic image is much more important than hearing reverb coming from a single point behind your head.)

Of course if you listened to one of those front/back recordings over a normal left/right pair of speakers, it would sound as if you were sitting in the middle of the audience, facing directly toward one of the side walls, with the orchestra either to your left or to your right, depending on the channel assignment. That was very unrealistic. Yet it seems as if this is what Mr. Cofrancesco is recommending. Or do I misunderstand you, Mr. Cofrancesco?.

Of course if you listened to one of those front/back recordings over a normal left/right pair of speakers, it would sound as if you were sitting in the middle of the audience, facing directly toward one of the side walls, with the orchestra either to your left or to your right, depending on the channel assignment.

I think such a configuration should be mixed as if it were a middle/side setup. The microphone near the stage gets mixed equal between left and right channels and the ambient mike is added to the left channel and subtracted from the right.

Eric, that's a very interesting suggestion!

It would certainly have good mono compatibility, because the ambience mic would disappear in a mono mixdown. No problems there.

I wonder about having all that out-of-phase ambience information, though. In general, I don't think it's wise to have too much info with opposite phase in the L and R stereo channels. But if you mixed it in at a relatively low level, compared with the main front mic, it might work out OK.

In fact, a similar technique is sometimes used to create "artificial stereo" by using a short delay to produce the second track, then adding and subtracting it from the main track (which is mathematically the same as mixing it as M/S).

I'm wishing I had a pair of tracks here to try this right now... I think the result might, in fact, be audibly interesting. Next time I have a chance, I will try to capture a pair of tracks in this configuration.

I'm also wondering whether there's a way to process that ambience track, to get two resulting tracks that have a smaller phase difference between them... say 90º apart, rather than the 180º you'd get by simply using M/S math. Then you make one playback channel = front + unshifted ambience, and the other playback channel = front + 90º shifted ambience. That would lessen the out-of-phase problems, but would still give a nice stereo feel to the ambience. That way, some ambience would remain in the mono mixdown.

I don't recall seeing any constant-phase-shift features in Audition. (I have schematics for an old CBS analog circuit somewhere, I think it was part of the SQ matrix system.) Perhaps I'll have to dig deeper for a software equivalent.

I don't recall seeing any constant-phase-shift features in Audition. (I have schematics for an old CBS analog circuit somewhere, I think it was part of the SQ matrix system.) Perhaps I'll have to dig deeper for a software equivalent.

There is no "constant-phase-shift" except at a fixed frequency. Do the math. It seems horribly retrogressive and devolutionary to be discussing creating artificial "stereo" when we have the means to record the real thing. Artifical stereo was never convincing or even satisfactory except for the most undiscerning hearer.

This is very slightly related to a technique that was proposed, tried, and abandoned back in the '50s in the early days of two-track recorders...
In my experience with near-coincident stereo recording techniques for acoustic performance, in a good hall, a mic or array in the sweet spot can be wonderful.

OTOH, many "music" venues don't really have a spot for this kind of technique.

It comes back to the traditional chamber music and concert venues. In the early days of recording, smart and innovative people devised direct-to-stereo techniques that worked really well. Prior to general availability of sound reinforcement, jazz, pop, and folk performers all depended on achieving acoustic balance within their groups. Performance venue acoustics *had* to support this approach.

Today, most pop venues are supporting an entirely different sound, (mostly) meant to replicate close-miced studio recordings. Most venues aren't concerned with room acoustics unless it affects their bottom line.

As someone pointed out above, a sweet-spot recording might be better than some other methods, but usually will not reflect the close-miced aesthetic.

It seems horribly retrogressive and devolutionary to be discussing creating artificial "stereo" when we have the means to record the real thing.

Over half the televisions in use feature monaural audio through a single speaker. In today's world DVD audio is also played through the built-in speakers of a notebook computer, sound bars, and in theater setups consisting of 5 tiny speakers and one monaural woofer. The number of people who employ a well-positioned pair of high-fidelity speakers suitable for music reproduction is surprisingly small.

Sony has handicams that record 5.1 surround sound, but has still not produced a prosumer model with even 4 XLR inputs. If all you have is a monaural stage track and a monaraul ambience track, it seems quite resonable to center the stage track and use the ambience track to create fake stereo. It might even be reasonable to create a 5.1 soundtrack with such sources.

There is no "constant-phase-shift" except at a fixed frequency. Do the math.

I cannot do the math, that's way beyond me. But I did build the circuit (back around 1978) and tested it with a dual trace scope, and it did, in fact, produce a 90º phase shift across the audio band.

I remember this very distinctly. I was trying to build a variable speed drive for a hysteresis synchronous motor. The motor in question was designed, nominally, for 60 Hz. There were two separate windings, one driven directly from the power source (12 VAC in this case) and the other winding driven through a series capacitor, whose value was selected to produce a 90º phase shift... that's that makes the motor turn.

I was trying to figure out some digital way to do this, which would have been pretty complicated in 1978. Then I remembered reading some description of the Columbia SQ quadraphonic system, which encoded four audio channels onto two phonograph channels, using phase relationship to "steer" the playback signal. The system ran one of the channels (rear???) through an analog circuit that produced a constant 90º phase shift, before recording onto the LP groove.

I wrote to someone at Columbia, whose name I had found in the relevant magazine article. Luckily I got a reply from the right person. (Perhaps the fact that I had a prestigious-sounding government job helped with that.) After some discussion, he mailed me a set of the schematics, I built the thing, and it worked as described. I could feed in the signal from an audio oscillator, view the two outputs on a dual-trace scope, and while I swept the oscillator across the audio band the phase shift did -- miraculously, to my mind -- remain very close to 90º.

I went on to build a few prototype circuits. Incidentally, we fed the output of the phase-shifter into the two channels of a Crown D-75 power amp, and used the amp output to drive the two motor windings (directly, without the phase-shift capacitor, of course). The thing worked as designed. But by then the "higher-ups" had changed their mind about how the show would go, and we didn't need any of this stuff any more. Shortly thereafter I decided I'd rather work elsewhere. The last I knew, there were a dozen or more unused D-75s just gathering dust... your tax dollars at work.

If I can find any further documentation about this, I'll post it. I have some files that old stored in the attic, but no guarantee that I still have this particular info.

I cannot do the math, that's way beyond me...
If you could, you would realize that it is impossible unless you have invented time-shifting as well.

If you could, you would realize that it is impossible unless you have invented time-shifting as well.

Quadraphonic sound - Wikipedia, the free encyclopedia (http://en.wikipedia.org/wiki/Quadraphonic_sound#SQ_.2F_Stereo_Quadraphonic)

Yes, and perhaps the primary reason quad sound never really made it off the ground was because of the quite poor performance of the "QUASI" phase shift scheme.

You could also have cited analog color television (both NTSC and PAL/SECAM) as popular communication channels that depend on phase-shift. But note very carefully that in THOSE cases, they were dealing with a FIXED frequency (3.579545 MHz for NTSC, and 4.43361875 MHz for PAL) which made phase-shift trivial.

And modern high-capacity digital communication (like ATSC and QAM) are also completely dependent on sophisticated phase modulation and demod. But AGAIN they are dealing with FIXED frequencies.

The people who developed SQ, et.al. knew that it was impossible, but they kludged a scheme the best that could be achieved in the real world. Alas, it was nowhere good enough even for casual consumer use. The same could be said for artificial stereo by the phase-shift method. Certainly there are better ways of simulating "stereo" from a monaural source here in the digital era. But you can't do broad-band "phase shift" with digital technology, either.

Quadraphonic sound never made it off the ground for several reasons. For example, I suspect most non-audiophile "normal" folks didn't want to instantly double the cost of their audio system, by adding two more speakers and two more channels of amplification.

There were two quadraphonic-LP systems. The JVC system used a very high frequency subcarrier on the disc, which required a special stylus and preamp to play back. It was not reliable even with new discs, and of course repeated playings decreased the amplitude of the subcarrier signal, so that it became less and less reliable as the discs aged.

The Columbia-SQ system also wasn't workable, simply because music already contains a lot of random phase information. So simply using phase information to move sound between the two front speakers and two rear speakers really didn't work. There was a lot of "crosstalk" which was basically random in nature. yes, you got sound out of all four speakers, but it did not accurately reproduce the sound field.

Interestingly, years later, Dolby used a similar phase-encoding system with their analog sound tracks on 35mm release prints. Stereo music tracks were recorded, basically unchanged, to the left and right tracks on the film. Mono dialog was recorded equally on left and right tracks. Surround information was recorded out of phase on the left and right film tracks. The reason this worked (and Columbia-SQ did not) was that by the time Dolby was doing it, advanced DSP was available. The Dolby system performed a sophisticated phase-relationship analysis of the information on the two film tracks. If the system decided that the information was mostly in phase and equal, it "steered" it to the center stage speaker (by means of adjusting playback gains). If the system decided the information was mostly not correlated, it was "steered" to the left and right stage speakers. If the information was mostly out of phase, it was "steered" to the rear surround channel. I've listened to analog optical tracks played back over this system, and the steering worked quite well. (Columbia-SQ probably would have worked better than it did, if sophisticated DSP had done the steering. But maybe not. SQ was trying to reproduce four channels of continuous information: two "stage" speakers plus two "surround" speakers with reverberant audio at all times. That's a lot harder problem to solve than Dolby theatre sound, where you rarely have stereo music, mono dialog, and mono surround all at once. The typical motion picture track would be easier to "steer" than continuous quadraphonic music, like the SQ problem.)

Be that as it may, regardless of consumer acceptance of these flakey quadraphonic systems, and regardless of whether you can comprehend it and believe it, I assure you that the 90º phase shift filter did work the way it was supposed to. I confess that, when I first heard of it, my reaction was the same as yours: "that's not mathematically possible." What that really meant was that I didn't know enough math to comprehend how it worked. But after talking with some people who know a lot more about filters and math than I do, I was convinced to try it. And they were right... it did work. I believe it's related to an "allpass" filter, but it may be more complex than that. I'll try to find some more convincing information when I have time.

Minor update:

I've just found a lengthy discussion of SQ-quad which includes this relevant sentence: "CBS encoders and prototype consumer decoders used precision aligned 10-Pole phase shift networks that were accurate +1° over a 20-20kHz bandwidth." The phase shift was, indeed, 90º.

It's getting rather far OT but for anyone interested in some history, the thread containing the discussion is here:
New Technology SQ Decoder discussion (http://www.quadraphonicquad.com/forums/showthread.php?14334-New-Technology-SQ-Decoder-discussion)

Then they must have some special definition for multi-frequency "phase shift" for audio. It is still a mathematical impossibility no matter how you slice it.

The delay required to produce a "90° phase-shift" at 1 KHz will produce a 180° phase-shift at 2 KHz. and a 360° phase-shift at 4 KHz. But if you could somehow separate the various frequencies and delay each of them for long enough to produce exactly 90° phase-shift (or whatever), you will end up smearing the sound all over the map. We typically try to AVOID doing things like that.

It gives a fuller sound with depth that is more pleasing to my ear than one mono mic. Whether it meets the definition of stereo is not of great concern to me.

Actually, it should be of tremendous concern to you. A true stereo recording will mix down to mono much better than this will. One of the things that so many people fail to take into account is the speed of sound. Sound travels slowly and if you're sitting there pulling microphones from two completely different locations, you are going to have a really bad lag between the two channels. When that sums to mono, you are going to end up with phasing and comb filtering- both qualities of sound that any good sound engineer tries very hard to avoid.

Never mind the issues of sync to your video. Sound travels approximately 1ms per foot. That means that if you are 40 feet away and you're shooting 24 frames, you're about a frame out on that microphone. If you are shooting a faster frame rate, you'll be further out of sync in relation to your frame rate.

If you want that to come close to working as a mono recording, you will need to compensate for the time lag in post.

--Ben

Then they must have some special definition for multi-frequency "phase shift" for audio. It is still a mathematical impossibility no matter how you slice it.

I am quite impressed that you know more about mathematics than Ben Bauer and all the other engineers at CBS Labs who were involved in development of the SQ-quadraphonic system.

The delay required to produce a "90° phase-shift" at 1 KHz will produce a 180° phase-shift at 2 KHz. and a 360° phase-shift at 4 KHz.

No argument there. The point is that an "all pass filter" does not use a fixed time delay.

if you could somehow separate the various frequencies and delay each of them for long enough to produce exactly 90° phase-shift (or whatever), you will end up smearing the sound all over the map.

I won't debate that point, either. It certainly does seem that the resulting waveform would be changed significantly, for anything except single-frequency tones. As we both agree, the SQ system did not gain consumer acceptance. Engineering tests showed that it had very limited channel separation, and many reviewers mentioned "artifacts" which might have been a result of such time smearing.

If you read the lengthy reference I posted, you'll have noted that the encoder, and the prototype decoders, had very accurate 10-pole filters (to produce the phase shift). Those apparently sounded acceptable. But many consumer decoders had only 2- or 3-pole filters. Those would produce a very inaccurate phase shift, and undoubtedly made the whole system sound much worse.

Be that as it may, I have never made any claims about SQ's audio quality. I have only stated, and here reaffirm, that it is possible to make a filter which produces a reasonably accurate 90º phase shift across the audio spectrum which by CBS's definition was 20Hz - 20kHz. You denied that it is possible to make such a filter. I stand by my original statement: it is possible. CBS Labs did it, and I built a prototype (based on a schematic from CBS) which I did observe to work as stated. So such a filter is, indeed, possible... whether you or I can explain the math, or not.

Then we will have to agree to disagree. I am unwilling to accept "I can't explain the math" or "it's patented" or "it's magic". No, I have no respect for the technical competence of the USPTO. You would be amazed at what they have granted patents for.

Actually, it should be of tremendous concern to you. A true stereo recording will mix down to mono much better than this will. One of the things that so many people fail to take into account is the speed of sound. Sound travels slowly and if you're sitting there pulling microphones from two completely different locations, you are going to have a really bad lag between the two channels. When that sums to mono, you are going to end up with phasing and comb filtering- both qualities of sound that any good sound engineer tries very hard to avoid.

Ben, everything you say there is entirely true. Mixing a close mono mic and a distant mono mic could give you some rather strange results. That is probably less of an issue if the "reverb" mic is mixed in at a relatively low level. For example if the level from the "reverb" mic is 1/10 the level (-20dB) of the "close" mic, then the comb filtering would not go to zero, it would go only to 90% (-.915dB). That doesn't make it a good idea, but if used very judiciously it would be less of a problem.

However, I notice that the scenario suggested above by Eric Olsen is a special case. He suggests mixing the "reverb" mic as if it were the "side" mic in an M/S setup. Let's look at the math. We'll call the mic close to the stage the M mic, and the distant (reverb) mic the S mic.

So if he mixes the two together in a typical M/S matrix,

L = M + S
R = M - S

Now if you further mix those together equally, to get a mono signal, you get:

Mono = (L) + (R) = (M + S) + (M - S) = 2M. In other words, in the mono mix the "reverb" mic, which we're calling the "side" mic or "S" in this equation, disappears completely. In this special case only you end up with complete mono compatibility, although with no reverb mixed in... just a close-up mono recording from the mic closest to the stage.

Be that as it may, I would not choose to do this, for three reasons. First, you're taking pressure-related signal from the "reverb" mic and putting it into the two stereo channels completely out of phase. If the level is at all significant, it could result in that "hollow" or "sound inside one's head" effect. Second, if the "reverb" level is at all significant, you will have some comb filtering. And finally, as mentioned above, the reverb will completely drop out of the mono mix.

IMHO if I wanted a close mono mic, with additional ambience mixed in, I'd use a stereo ambience mic, mixed L/R like a normal stereo source, and mix the close mic signal to the center, after delaying it so it is coincident with the ambience mic. Not ideal, but perhaps workable. Just my opinion... deposit 2¢ please.