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Werner Althaus

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About Werner Althaus

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  • Location
    Lincoln, NE
  • Interested in Sound for Picture
  • About
    Audio supervisor for statewide Public TV network

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  1. Thanks for that video, my info might be outdated as it goes back to the G-Prime/ Gotham days when they co-owned MG, at that point they only supplied parts for microphones made since the early 90's according to Jerry Graham (owner of G-Prime/ Gotham). Anyway, I have used a few MG mics and liked them, I just wasn't a big fan of some of the claims made regarding their part in the fabled Neumann saga.
  2. Totally agree, the reason I mentioned this is that some people believe MG has a direct link to their past, like other historic brands due to some deceptive advertising on their part early on, linking their brand to the Neumann legacy.
  3. MKH 40 and 50 are both great mics, the 50 with a little more "zip" on top due to its polar pattern. The only mic that gets very close to a consistent HF and upper mids response with varying capsules/ polar patterns is Schoeps IMHO. And why do folks keep referring to the MK 41 as hypercardioid, it's super cardioid. Speaking about MG, they are a great company as long as you don't expect them to service vintage Gefell mics, they simply don't support them IIRC. Anyway, can't go wrong with the MKH 40/50s
  4. I don't think it's what he said at all. he said this Mic sensitivity defines the electric output signal against a reference sound pressure level of 1 pascal/ 94 dB, usually expressed in -dB or mV/Pa, in a nutshell the higher the voltage , the less gain need be applied = more sensitive. I have recorded violins, violas, cellos, double-basses and a host of other stringed instruments with lavs, mostly dpa and pillepalle is correct, those mics can be too "sensitive", meaning that their output level as it relates to the SPL to be recorded exceeds the input stages' (mic pre) maximum input. It has nothing to do with the max SPL the mic can handle and everything to do with the maximum input the recording device can handle. Unless you have either external PSUs and pads or a mixer that lets you apply a pad on an input capable of phantom powering (like SD 6xx set to line with phantom power) you have to reach for a mic with lower sensitivity. I also would point out that the placement on the instrument does not yield a pleasant reproduction of the sound of the instrument. It is great for providing detail and perspective but without a main mic setup it sounds like fingernails on chalkboard. The sound of these instruments "comes into its own" only at a distance, curiously enough the higher quality of a concert instrument we're dealing with the more "zerklueftet" ( hard to translate: "riddled with peaks and valleys" might be close) the timbre becomes at close range, only at a distance does it become whole. To answer the OP's question, I'd suggest mounting the lav on the player, preferably somewhere on the head (assuming the head doesn't move wildly during playing) like behind/ above the ear or the forehead / hairline. That way you'll get some distance between the mic and the instrument while staying within the critical distance.
  5. In recording nature it's definitely the recording chains' noise that creates the problem. Last time I went out I had to immediately ditch the 418 I was using in favor of an unwieldy MS rig consisting of an MKH 30 and 60 because the hiss was overwhelming in relation to the wind, water, insects, etc. In post we still use a lot of the old Hollywood edge SFX library and those recordings are full of system noise (mics, pre amps, recorders, tape, etc), sometimes the noise is within 10dB or less of the sounds they were going after. The location didn't contribute anything, it's the application and how it relates to your systems' S/N ratio
  6. Are you sure it's bleed from the monitors and not the mains? If it's a combination of the two then you could experiment with throwing the monitors out of phase with the PA and find the sweet spot where they both cancel at the mic. I'm not a fan of using 416s because of the huge bass response in the rear of the mic, it tends to pick up PA rumble too much for my taste. There's a lot of different ways to do this but IMHO, usually having to post my own stuff I would advocate to use multiple pairs of different reach and speed from as close to a time aligned position with respect to the PA array. I like MKH 70s for the distant , a pair of cardioid (MKH 40) for the front and a pair of dynamic cardioid or super cardioid added to taste to get something with a slower response to blend. All three pairs should have the PA array in their null angle and should be in the same plane as the PA to avoid the delay that tends to wash out the dry signal. I know you can time-align in post but it's not the same. I'd then have the house signal up to a base level at all times and chase the audience responses with single mics as long as I don't mess up the image too much.
  7. I hope this isn't too basic a question but a search didn't reveal too much info on this subject so here it goes. These days I don't get out into the field much anymore but when I do I usually borough a co-workers kit. When I do I usually find out about what gear needs replacing due to age or defects of all kinds. I am talking about things that creep up on the daily user without them necessarily noticing. Example, I'm always shocked when I listen to our headphones of choice (Sony 7506) after they've been in use for a while. The low end on these seems to disappear and they start sounding rather tinny. So the other day I did a few sit-down interviews in a reasonably quiet location and I used a really old (30 years) Schoeps CMC3/ MK 41 on a boom overhead and a MKE-2 lav (the 48V wired variant) that is probably 15 years old. the recorder was a SD 633. For the second interview I switched out the Schoeps for a brand new Sanken CS3e because I didn't like the sound of that particular Schoeps. But in terms of noise the Schoeps was still doing very well, while the MKE-2 was noticeably noisy/ hissy. I switched to a second MKE-2, same thing. I did test a few more, all of similar age and they all seem to be getting noisy. My question is whether this is normal and why it seems that the noise decreases when I use an older transformer balanced 442 . We're pretty much all SD 633 or 664 now and I need to find lavs that do better in terms of noise. Should I try new MKE-2s (in truth I never really cared for those to begin with) or are there better choices out there in terms of noise. Soundwise I like the Sanken COS11ds but I've never tried a hardwired version. My experience with our old dpa miniature mics is that they tend to be awesome for everything except for use as lavs , the cable on these old ones are too rigid and they "hear too much" if that makes any sense. As you can see I haven't really paid attention to lavs much since MKE-2s were just the standard. Any input would be appreciated.
  8. I can only speak for myself but the main reason that I prefer M/S is that my M/S rigs (Sennheiser 418 or 30/60 and sometimes 30/70 combo) are easily used in mono (M only) no drawbacks, no changing mics/ poles, just record and monitor the M signal and pretend you're working with a mono shotgun mic until you need stereo. The 30/60 or 30/70 are a bit heavy though. In post I like the control and compatibility. I used to mix in 5.1 for a little while (not anymore) and the M/S tracks worked great through all the down-mixing, no surprises. Dedicated XY rigs, which I also use sometimes, are nice but not practical in a run and gun situation, same for AB and Ambisonics.
  9. Maybe, who knows? we can speculate all day but the video from NAB gives me a few clues about what's really going on here. The Zoom person claims something along the lines of "24 bit audio converters do really well recording at higher levels but not so well at lower levels" or something to that effect. This statement is misleading if not false. What exactly are these lower levels, what makes them "record bad" and what does the word length have to do with it? I don't have experience with current super high end converters but If I record with my Euphonix AM703 into the MADI input of my RADAR Studio I can record "low levels" just as good as high levels, no quality difference as a result of low recording levels whatsoever, at least to my ears. Same is true for really high levels, no problem, high and low level content will be recorded without negative side effects as long as I stay away from the maximum analog input level that corresponds to 0 dBFS. If I'm recording with a cheap interface or recorder the same is not true even though the cheap interface also has a 24bit converter but low levels sound anemic and noisy while levels close to 0dBFS sound distorted and edgy. There's a smallish range where the audio sounds reasonably good but both hot and low levels tend to be less than good sounding. What this tells me is that the difference is in the analog realm, the power supply, the op amps, the clock, all part of the analog topology surrounding the ADC. If low levels fed into a 24bit ADC sound "bad" it's probably a function of those components rather than the word length used to encode the audio since it is already (theoretically) capable of a staggering dynamic range of 144 dB
  10. I dunno if I'd go along with that, instead i'd offer this quote from Barry Henderson of IZ technologies (RADAR) " All digital signals are analog signals and have to be treated as analog signals".
  11. Yes, even though true match mic inputs provide up to 70 dB of gain per Stagetec's website. " Gain Up to 70 dB (clickfree digital adjustment in 1-dB steps) Anyway, my guess is that the analog circuit inside the XMIC+ is a big, if not the most important part of the equation. It also should be noted that digital gain, while in theory "free" is never able to increase the dynamic range at capture and that depends on the analog circuitry used.
  12. I'm not calling copper wire analog circuitry, I'm calling transformers, resistors, op amps , etc. analog circuitry. And that argument was made by me with regards to digital signals (PCM) being carried over copper, with the Zoom F6 it's analog signals that will pass through analog circuitry and it will affect the bandwidth linearity and dynamic range whereas with digital signals the analog circuitry will not affect the analog audio signal per se, just the quality of transmission, error rate, jitter, etc.
  13. Yes, I can see that I am not getting my point across. I never said that there was any subsequent conversion, my point is that just because it's PCM audio doesn't mean it doesn't involve analog circuitry. Every AES3 input utilizes some form of analog components, transformers, resistors, etc. that's all I am saying. For me this matters in the context of the claim that there's no analog circuitry involved at the input of the Zoom F6. The point being even if there were digital inputs on the F6 it would still involve analog circuitry and with analog inputs there certainly are. As such the analog circuitry dealing with mic signals certainly does affect the dynamic range and linearity, regardless of whether it's a mic pre or just a line amp. I hope I'm making sense now. I get the concept, it's been around since the early 2000s with Stagetec's "true match" converters. But in this context it's made to sound like mic preamps only add noise, therefor eliminating them will reduce noise and give you the microphones true dynamic range. The reality is a bit more complicated and you can already do this anyway with any 24 bit encoder to get 144 dB dynamic range, ah if it were only this easy.
  14. It is what Patrick was supposedly told by Zoom "While this is all true for traditionally designed recorders, it doesn't really apply to the F6. There are no preamps or analog circuitry before the converters which would saturate into nonlinearity. The mic signal directly feeds the converters." This info contradicts what was said in the video that states that the mic pres are the same as in the F8. Even if these pres are set to minimum gain (10 dB IIRC) or no gain at all they will exhibit non linearities and limited dynamic range that do not exceed 24 bit. If not I'd call that a miracle.
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