Re: Soundboard Pre Mix
 

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

Re: Soundboard Pre Mix
 

Re: Soundboard Pre Mix
 

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.

Re: Soundboard Pre Mix
 

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.

Re: Soundboard Pre Mix
 

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

Re: Soundboard Pre Mix
 

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.

Re: Soundboard Pre Mix
 

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

Re: Soundboard Pre Mix
 

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.

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

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.

Re: Soundboard Pre Mix
 

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.

Re: Soundboard Pre Mix
 

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.

Re: Soundboard Pre Mix
 

Richard, we can agree to disagree. I know the filter I described is workable, and have found a few references to it in places like The Audio Cyclopedia, but I don't understand the supporting math (I checked out of math after first year calculus). But I built one, and tested it, and saw it work as predicted. No offense intended, but I give more weight to my own measurements and observations than I give to anyone's blanket statement that something is "impossible." After all, manned flight was once considered impossible. Just because something seems impossible to you does not make it so.

Having said that, I don't know why you keep trying to obfuscate the issue. First you wanted to use "SQ was never successful" to prove that a 90º phase shift filter is not possible. There is no logical connection between the economic success of the SQ system, and the feasibility of this filter (which was only one very small part of the SQ scheme).

Now you are invoking the USPTO to prove that a 90º phase shift filter is impossible. In the first place, I haven't made any claims, one way or the other, about the filter's being patented. Actually, I doubt that the filter, itself, can be patented, any more than a two-way 12dB/octave crossover can be patented. Ohm's Law isn't patented, but we accept it as workable. Thevenin's Theorem isn't patented but we accept it as workable. OTOH, at one point phonographs, motion pictures, and FM radio were patented and we know that they work, too. A given device can work, whether or not it is patented... there is no logical relationship between the two. So whether this particular filter is patented or not, there is no logical relationship between the patent status and the question of whether such a filter can exist and function as predicted. Your continued obfuscation simply makes me feel that you are arguing from a position that can not be logically supported. IMHO that doesn't strengthen your argument; if anything it weakens your credibility. Although you might make a good politician.

At the present time, this seems to be all we really have to go on: I have built one and have seen it work and believe it is possible; you have not built one and have not seen one work and believe it is not possible.

Re: Soundboard Pre Mix
 

Wow this subject has gone off in a different direction than the OP must have wanted...

Phase from what I grasp is frequency and there fore time dependant, many years ago in the early days of stereo TV broadcasting we ran a "Howe phase chaser" in the control rooms when we were producing music clip shows, they corrected any head alignment errors that occurred in the VTR machines.
They were slow acting but worked VERY well for fixing small alignment errors but caused problems when used when with MS mics as there often sensed the side components as an error and tried to correct it, with some disastrous on-air audio results. (And hence my personal dislike of MS mics for broadcasting)
Here is a link to the patent for the Phase Chaser. www.google.com/patents/US4890065.pdf

The role of a patent office is NOT to determine the validity or practicality of an idea but purely a place to lodge an idea and to have it documented.

But I have to agree it would be difficult to produce a 90 deg phase shift on ALL frequencies of 20Hz -20k Hz

Re: Soundboard Pre Mix
 

Brian,

Yes, we have gone off on a tangent. When I first briefly mentioned a 90º constant phase shift, I never could have imagined that someone would argue so vehemently and persistently that such a thing was "an impossibility," and would continue to do so when there is so much literature available online supporting their existence.

You're certainly correct in saying that phase is frequency- and time-dependant... no question about that. That much is intuitively obvious to me.

Yet there are a lot of complex engineering principles that are not intuitively obvious, which certainly are true. For example, I don't understand anything about thermodynamics. I can't prove mathematically that an internal combustion engine works; yet I've driven my car, so I know that it does work. I am certainly not a nuclear scientist and couldn't design an atomic bomb or atomic power plant; yet there is enough evidence to convince me that both of these things exist. Likewise, I can't begin to explain how a constant phase-shift filter works; yet I've built and tested one and I know that such a thing is possible. My point is simply that, although a given person doesn't understand the theory behind some device, it doesn't mean that device can't possibly exist. (Heck, I can't explain mathematically how we are here on the earth, orbiting the sun, which is in the milky way... etc. Yet obviously we are here, despite my limited mathematical comprehension.)

Since this discussion began, I've been reading many articles about constant phase-shift circuits. They are not intuitively obvious to me. I suppose if I had stuck with calculus for a few more years, and had taken some filter design courses, then I might begin to understand how the things actually work. I will readily admit that I don't have that deep of an engineering background. Nevertheless, there are plenty of relevant articles online. These filters seem to find a lot of uses; the most recent one I've read about was demodulating SSB (single sideband) radio transmissions. They are out there.

(I decided, at age 18, that I wanted to have a "hands-on" career. I did not want to spend my life playing with calculus and mathematical theory. But my hat's off to the people who do that for a living, and who are smart enough to design filters with this degree of complexity... filters that I really can't comprehend.)

Yes, Brian, you are clearly correct in saying that "it is difficult to produce a 90 deg phase shift on ALL frequencies of 20Hz - 20kHz." It is difficult, but not impossible. The detailed article I read about the SQ system (of which the phase shift is only one part) says that the encoders, and prototype decoders, needed 10-pole filters, with unusually precise components, to produce a phase shift which was accurate to 1º over that frequency range. A 10-pole filter is a very complex circuit, compared to most typical audio filters, crossover networks, and filters that I've worked with. Indeed, it is difficult.

I'll freely admit that I wouldn't have even thought of such a thing, if I hadn't read the article about SQ, and I certainly couldn't have built one if someone hadn't given me a detailed schematic. Yet here we are on the earth, orbiting the sun, flying through space... with cars, nuclear power, and constant phase-shift filters.

Thanks for your interest and input.

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Nothing can kill a documentary, or any other type of film, faster than crappy, amateurish sound. Just because the equipment availability has been democratized doesn't mean that the standards the finished product should adhere to have come down - if anything, the fact that everyone today has grown up listening to "that Hollywood Sound" means the standards are higher today that they ever were in previous generations. I don't care if you're recorder costs $500, $5,000, or $25,000, your finished sound track must adhere to the highest level of quality that is possible to deliver or you're not doing your job as a filmmaker properly. The only time it is acceptable to compromise on quality is when it's a situation so desperate that it becomes a choice of settling for second best or coming home with nothing at all.

Rather than micing a speaker, you need to bite the bullet and do whatever it takes to mic the stage properly for your recording mix. Don't have a second mixer? Buy one. Don't have someone to operate it? Hire one. Can't afford to do it right:? Don't do it at all and spare the world from the presence of yet another POC production.

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There is nothing done so well that it couldn't be done better. Not having an additional sound person for recording the band does not prevent most videographers from covering a wedding reception. The goal is to get the best sound possible given the constrains imposed on the shoot.

The original post considers what a single shooter using a single video camera can do to improve sound without going dual system or bringing so much equipment that it takes a crew to move it. Having said this, an inexpensive improvement could be made using reasonable recording techniques along with something like a $500 Zoom R24 which can record 8 simultaneous 48khz wav files through XLR using 6 AA batteries.

Zoom — R24

Re: Soundboard Pre Mix
 

Thanks Eric,
That could be the best response to my original query.
So, $500 for the unit, another $180 for splitters and patch cords.
And probably an additional 3-4 hours of post for mixing and it should be the best possible solution.

A Couple questions though . . .
Say the performance is:
2 Vocalists (using hand mics & headsets for different songs)
Keyboard (piano + synth)
Drums
Bass
Brass (3pcs)

-do i request a submix from the board combining the different mics for each vocalist?
-same for drums
-same for keyboards

I guess all XLR cables aren't made the same . . . what should i look out for?
I don't specialize in this niche . . . but this is getting close to a solution. I'll have to raise my rates for this setup.

Thanks again.

Re: Soundboard Pre Mix
 

Consider also the trick of using channel inserts as recording outputs:

http://www.hosatech.com/product/0/DOC-106/_/



This will look less suspicious than a bunch of Y-cables and then you don't need to worry about isolating yourself from the phantom power. I would also suggest using one well placed stereo microphone to make sure the brass players get recorded and as a backup. What's possible in terms of sub-mixes depends on the exact mixing desk and level of cooperation.

If you have a notebook computer and don't mind bringing it along, you may prefer using a USB interface such as the Tascam US-1800 for multi-track sound recording.

Product: US-1800 | TASCAM

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Your biggest headache is going to be the drums. In many venues drums won't be mic'ed at all and their only presence in the mix for your recording would be through bleed into other performers mics. You're going to need a full set of mic's for the kit, 5 to 7 or even more mics just for the drum kit, each going to their own input on your mixer.

Electric guitars and basses might not be mic'ed either, depending on their cabinets to provide enough volume for the audience. DI boxes to split their signals to your recording mix might be necessary for them.

As been said before, mic'ing a group for recording and mic'ing them for a stage performance are two very different things

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Dead End, for most budgets i'm associated with.

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I've used an overhead boundary microphone with success. Rumor has it that some drummers will allow you to gaffer-tape a boundary microphone to their chest, but I don't know about that. Anyway, without a crew it needs to be simple and not require too much time to set up.

Crown International / Knowledgebase / Which PZM should I choose for drums?

Re: Soundboard Pre Mix
 

Oh boy... where to even begin. I guess I'll leave it at this because there is so much wrong with this I don't know where to start. As an audio engineer, this really makes me cringe.

The problem with this whole line of thought is that this is not mid-side. Mid Side is a very useful stereo microphone technique, especially when it comes to mono compatability. A ambient mic does not make a side mic. the whole point of that mic technique is sum and difference. The middle mic has everything that a figure eight mic (as the side) does not have. To take a sum and a difference of those two mics makes a stereo image. Two individual microphones in different locations will never make mid and side.

I don't mean to be blunt or rude here, but I do need to try to clarify a correct usage of a technique. Two mics as originally described is mono only. It cannot be stereo.

--Ben

Re: Soundboard Pre Mix
 

Ben:

I think you and I are actually in agreement. In my next to last paragraph (as quoted above) you'll note that I say I would not do what the OP suggests (namely using a mono mic -- presumably a pressure mic -- deep in the house, then mixing it in as if it were a "side" mic). And I give my reasons for disagreeing with that proposed technique.

Indeed, I think the only time the M/S matrix really produces true stereo results is with a coincident pair of proper M/S mics (i.e. the Side mic is figure-8 facing sideways). That is not what was proposed by the OP so the results will be entirely artificial and technically prone to numerous problems.

--

But I did find it an interesting sidelight that, if one followed the OPs proposed setup, the "reverb" mic would be added to the left channel, and subtracted from the right channel; therefore it would entirely drop out of the mono mix. (That's what I was trying to illustrate with the math.) So this particular proposed means of getting a bogus stereo mix, would in fact produce a clean mono mix, consisting of only the front mic.

--

Be that as it may, I agree with you, the proposed mix is not the best way to go about things.

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No offense, however, I was not suggesting that mixing the ambient mic and monaural stage mike as M/S was correct usage, only that this might be better than mixing it as an X/Y pair. One way to create pseudo stereo is by adding a delayed signal to the left channel and subtracting it from the right. This is called the complimentary comb filter technique and similar in spirit to mixing the ambient mic as if it were M/S.

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Yes, exactly. That is a known technique. If you cannot record more than two channels that are perfectly locked in time, and if you keep the level of the "reverb" mic relatively low, compared to the "stage" mic, you will get similar results: it will have some apparent stereo spread, although it won't be true stereo.

Of course it would be much better if you had a recorder with at least three tracks, so you could put a stereo mic out in the house. Better yet if you could then delay the stage mic to be coincident in time with the house mic.

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Going back a tad to this..
If the live sound people have not miked up the kit with individual mics, it's because there was no need - if this is the case, then for a recording there is no need to individually mic the kit - unless maybe it's a heavy rock sound, and if that was the case, then the kit would be miked for the audience too!

Boundary mics above work rather like an omni, or very, very wide cardioid - they cannot work as a boundary mic unless they're on a large surface - other than that, they're not boundaries. Sure - you can try clever placement on the drummer, but why not, if you really must mic the drums, just use an overhead, or an overhead with snare, or the verity common kick, snare overhead system. To complicated for a casual video without real sound people to work it!

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There's no need because the audience can hear the drums directly, true enough. But the RECORDING has no direct sound - if the kit isn't picked up by a mic somewhere it doesn't exist in the recording. Close mic'ing some instruments while relying on bleed or a house mic to capture others is asking for trouble. If you're recording the entire ensemble with a well-placed stereo mic then you are correct ... you're picking up whast the audience is hearing. But if you want high-quality concert footage you're going to have to mic up the group as if you were in the studio laying down a track destined for CD

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That is NOT what I'm saying. I'm merely pointing out there is no absolute requirement to stick a mic on every drum. The style of music dictates the drum mic technique you use. Individually eq'd and processed rock drums with perhaps sampled sound slavered over need isolation, whereas many other drums styles just need a more modest amount of mics to sound good. I'm quite aware that there won't be drums or other instruments in the audience mix, so you need to replace them - but a 3 mic approach works really well for most styles, and for big band, just one or two is quite sufficient. The reality is that most video people do not have sufficient facilities to hand to manage multi-mic audio mixing themselves. As it happens, I do - and can record to hard drive 24 channels, or 16-32 on a computer based system. There is no 'standard' solution, it's worked on fluidly to get the best results for each project. My point is simply that you don't need to mic up every single thing just because you can.