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Werner Althaus

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About Werner Althaus

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

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  1. I see a lot of references made to the Neumann RSM 191 in this thread. I have 2 of them that are in need of repair. Does Sennheiser USA do a good job with those? I agree, they are wonderful sounding but unfortunately not very good in humid conditions.
  2. Regarding dynamic range I'm still a little confused by the importance placed on digital levels and the ability to work with "overs" when writing audio files (processing is of course an entirely different matter) AFAIK 0 dBFS is not a defined level, it doesn't correspond to a signal voltage, it is just a digital value hopefully corresponding to the analog limits of the system at conversion under some agreeable standard. For example a properly designed AD converter for US broadcast will correspond to the required analog reference of 1.228 Volt / +4dBu = 0VU = -20dBFS. It then follows that 0 dBFS = +24dBu, so the analog maximum input level better be equal or greater than +24dBu, balanced <1% THD, otherwise there will be analog clipping before we hit 0dBFS. (BTW, In reality this happens all the time because manufacturers like to fudge the numbers and claim max input levels without properly specifying THD, that's why many analog front ends sound distorted/ pinched, non-linear way before the 0dBFS value is reached) Now compare that to a piece of gear that uses .775 Volt / 0dBu = 0VU = -20dBFS, it only has to have a max input level of +20dBu and will be cheaper to build. If you use .316 Volts /-10dBv as analog reference and define a digital reference level of -14 dBFS then a max input level of around +6 dBu = 0dBFS will suffice, even cheaper. Slapping a 32 bit converter into the unit adds pennies to the cost vs the amounts it would cost to build a studio level device that meets all the criteria. Note that only talking about the digital level expressed in dBFS and how they are expressed in bits tells us nothing about the analog capabilities / limitations of the system and those are the ones that will determine the signal to noise ratio way more than 24 vs 32 bits because while the analog reference levels are in flux as of late even on pro looking devices (seeing an XLR input no longer tells us anything about the reference used) the analog noise floor is still the same and if you operate at lower levels you're closer to the noise floor. I don't believe that the AD converter built around lower operating levels ( cheaper ) will rival the performance of the AD converter built to handle professional level audio signals when it comes to S/N, and even the best of those only achieve 21 bit of dynamic range.
  3. I know this isn't really relevant to regular location sound work but I'd never record phase correlated material across multiple recorders, even when they are "locked", "synched" or whatever. If you have ever listened to the " Auto-align" plug in do it's sample offset thing to align phase you'll see that even one or two samples will massively mess up phase between 2 correlated microphones , that's 1/24thousands of a second, or 0.04 ms, nothing purist about hearing that. I doubt that any of the recorders mentioned here can be locked with 100% sample accuracy
  4. I'm curious about your experience with the Neumann (which one?) I can't speak to the 104 ( a condenser) but I do have the BCM 705, the first (and hopefully last) dynamic mic Neumann has made. In my view it ranks among the worst microphones ever made. When it arrived at my place of employment it was supposed to "fix room problems". The mic had a tight pickup pattern but sounded extremely unnatural, harsh, sibilant and with a wonky midrange. As soon as I could I pulled it out of service and fixed the room's problems instead. I proceeded to store the mic in its original container in a dry, room temperature environment. A few years later I pulled it out just for fun and it now sounded completely broken, very hollow. The diaphragm doesn't seem to age very well. I suspect that this happens due to material breakdown of the plastics or glues involved. Not Neumann's finest moment and not anywhere comparable to the other mics listed in this thread.
  5. What a great post, I love the "talk to the tiger" bit. A little off topic but are you talking RCA 77A or the much smaller 77 DX? Which polar pattern do you prefer? I recently had 2 RCA 77DX serviced by Clarence Kane for exactly the reasons you state, we normally use U87Ai for Voice Over work but I always felt that I needed a more natural sounding alternative and we had a few of them lying around with the leads clipped off. I really like the 77DX but am still in the experimental stages regarding a proper mic pre. You mention a "high gain pre amp", do you have any recommendations that don't involve vintage OP-6s or replicas? I am reasonably happy with the Aphex 1788A (input transformer set to "un-loaded" per RCAs old manuals) but given what a huge difference transformer vs electrically balanced inputs and impedance in general makes when using this mic I was wondering if you have any recommendations. Thanks
  6. Myself also being far from an expert I recall that the bitrate has nothing to do with the kind of anti-aliasing filter being employed. In this day and age I'm under the impression that this particular nut (pre and post ringing, phase shift) has been cracked, a big reason why many of the past's most ardent devotees of analog gear have adopted a ITB (In The Box) workflow as of late. As you delve into the more budget friendly gear this may not be the case yet. Dropping the quantization noise to even lower theoretical levels seems irrelevant when the analog circuitry (the weakest link in the chain as far as noise is concerned) can't possibly exceed 21 bit dynamic range, and that is state of the art analog circuitry at low gain. I've heard it expressed like this: it makes no difference if your calculator computes the amount owed to you to the 20th decimal point, in the end you'll still only receive dollars and whole cents, the currency's maximum "analog" physical resolution. I am still curious about the mic pres and (lack of) gain staging in the F6, the pres are supposedly the same as the F8. Are they fixed gain at 10dB, the lowest setting on the F8 or continuous like stagetec's true-match ADCs? How do I monitor a mic signal without sufficient analog gain applied during normal operation, cranking the fader which controls the record level? Remember there are no prefade ISOs here, yes, the recorded file may be clean but the isos will have all potentially faulty fader moves baked in. If then confronted with excessive spl will the headphone amp clip if my fader is too high? Can I trust that the recording was unclipped despite clipping the headphone amp? In a highend digital mixing console with tons of internal headroom these things are computed in real time to guarantee the output and monitoring remains free of clipping at all time regardless how the internal levels are, is that the case here? How will the super low level files integrate into a realistic post workflow? How susceptible to RFI/EMI are the un-amplified mic level signals within this tightly packed "portable computer" really? I mean the reason we apply gain via mic pres is to optimize audio levels in terms of noise before they are distributed and manipulated. And lastly , what's keeping me from using the same theoretical approach of feeding "raw" mic signal into a conventional 24 bit converter and enjoy what has previously believed to be unobtainable 144 dB of dynamic range?
  7. I'm not familiar with the claim that 32 bit converters have better filters (why?) or lower self noise so I can't really comment on that. Instead I'll be paraphrasing (and borrowing , quoting from) Dan Lavry here and would recommend looking him up whenever questions about digital audio conversion arise: One should always differentiate between processing bit depth and conversion bit depth. The lower bits out of a 24 bit converter carry noise, not audio data, due to mic and micpre generated noise, so in reality 20 bits is about as good as one can expect. on the DA conversion side it's the same since "real hardware can't even do 24 bits, because the lower bits are buried in noise......fundamentally there is nothing to be gained by inputting more then 24 bits. More bits would be a waste of space, and no sonic improvement". 24 bits is more than enough for conversion yet not enough for processing to maintain a full 24 bit resolution.. because the same mics (dpa) into SD 664s line input during an identical setup didn't clip or sound "pinched". The dpa's can easily handle the SPL. As far as actual clipping goes the answer is yes but like I mentioned, even on passages that didn't outright clip the Zoom pres sounded non-linear, raspy, edgy, etc. which makes me wonder how figures for max input signals are measured over at Zoom. I've experienced this with pretty much every budget interface/ recorder, the sound quality suffers as you get closer to actual outright clipping.
  8. I used a pair of old Omni 4066s , mounted them inside a Steinway into a Zoom F8 (not the F8n, do they have different pres?). Couldn't use them due to clipping so I changed them out for a pair of MKE-2 Gold. Even before the clip lights come on those pres sounded "pinched". The same setup into a SD 664 line input with phantom power (brilliant feature BTW) sounded wonderful and clean. I haven't used the MixPre, I only know the MP-1, 442, 552, 633 and 664 pres which all sound different but are all miles ahead of the Zoom IMHO. I use the Zoom F8 for ambisonic recording now, with the Sennheiser Ambeo, it's a decent match. Don't get me wrong, I think Zoom makes a great product at the price , we have H2n, H6 and the F8, they are all very useful and have their place. And yes, AES would be nice.
  9. That Zoom guy lost me when he said that 24 bit records high volume at much better quality than low volume. The fact that cheap prosumer recorders sound worse at recording levels that are low (as well as too high despite being encoded below 0 dBFS ) has nothing to do with the supposedly insufficient 24 bit rate and everything to do with the quality of the analog circuitry used. For them to state that the mic will distort before the mic pres will is also a rather bold claim, it all comes down to the degree with which they give you honest specs. The Zoom F8 at its minimal mipre gain (10dB) cannot handle high output mics at all, my DPA miniature mics for example will over modulate the mic pres in those things. Depending on the source even a hot dynamic mic can over modulate the mic input of our old Soundcraft 800 series console with it's max input level of +37 dBu, no way can the F6 handle levels like that without distorting, 32 bit notwithstanding.
  10. SM7B is a classic great choice and affordable but it has really low output due to lack of output transformer so you'll need a really good preamp with lots of clean gain (>60dB) OR you can get something like a cloudlifter that will bump up the output by up to 25 dB, depending on the preamp design. Cascading preamps (cloudlifter is a fixed gain preamp) does make sense for improving S/N ratio vs a single mediocre preamp cranked due to the way that OP amps operate. There's a reason the SM7B / cloudlifter combo is so popular. yeah, 441s are very pricey. RE 20s are variable D, meaning they should not exibit any proximity effect but I'm not sure I'm buying that claim. I haven't used an RE 20 in decades but I have an EV 668 (precursor to the RE 20, also variable D) and it definitely has proximity effect.
  11. Sennheiser 416 is indeed a very popular choice for voice work, both in a booth or in a larger room. It has this thunderous low end response and presence boost that gives it that larger than life character and it has enough reach to stay out of the way of scripts. Unfortunately what makes it so popular is also what makes it so despised, the fact that it super-imposes itself onto the voice it's supposed to capture. Many complain that it eliminates any personality from the actor. Then there is the less than ideal off-axis response so make sure your booth is really dead or record in a room with high ceilings and proper treatment. I really like the classic Sennheiser dynamics for voice over in small spaces, MD 421 and 441 (my favorite) will sound better ( to my ears anyway) in many instances than spitty LDCs like the Neumann 103. If you spend a grand on an LDC there are better choices out there. Never tried the slate but also am very skeptical.
  12. My education in digital audio is probably a bit outdated so I'd welcome some input regarding my comments below: "Recording 32-bit floating point audio files" seems pointless to me as long as the dynamic range of the analog I/O falls way below the dynamic range limitations of 24 bit. In these prosumer recorders this is most certainly the case. 32 bit Floating Point processing is useful once inside a DAW since it allows the mixer more leeway regarding gainstaging as long as the output is free of clipping. Some mixing engines boast internal headroom in the 5 figure dB range due to 32 or 64 bit processing but that doesn't change the fact that during acquisition the analog stage (including the A part of the AD converter) largely determines the S/N ratio. Putting a 32 bit converter after an input stage that can only handle moderate maximum input levels and exhibits a high noisefloor is not giving us the ability to record a wider range of dynamics. We seem to be living in an age where theoretical dynamic range moves one way (up) while actual hardware based dynamic range moves down due to cheaper front and back ends, lower analog reference levels ( 0 dBu being the "new" +4, etc). Just for fun put up an audio file recorded with the mic pres up halfway and no mic connected to the recorder, then gain it up 50 dB in your DAW. What you'll hear is hiss where theoretically there should be another 60dB of S/N under 24 bits theoretical specs. 24 bit resolution exceeds all but but the most over-engineered analog stages in any mixer so why bother with higher wordlength? I wish they'd go for better sounding mic pres instead. Just my opinion.
  13. Terrific recording, really engaging both visually and aurally, that's the way it's done IMHO. I hope the shoot went well for the OP. I've done a few of these myself and the #1 thing is to realize that the board feed is only going to include sources that need reinforcement, if the instrument is loud enough acoustically it will likely not be miked, so for those sources and audience / ambience you have to use your own mics. these days I usually ask if the club/ venue has one of those ubiquitous Behringer X-32 rigs or similar that allows for ISO recordings of every micpre via FW or built-in SD card. If they do your day is going to be very easy, just ask to plug your supplemental mics (audience, etc) into FOH mixer as well (without sending them to the PA of course) and record with your laptop and DAW of choice ( I prefer Reaper for this, by far the most resilient for live recording) or onto SD card. You do put your fate into the hands of the FOH soundguy somewhat but if he/ she can't get clean signals through the FOH board then it's going to ruin the recording one way or another anyway, either by clipping or feedback or both.
  14. That's a lot of recommendations here. Instead of listing my favorite budget mic (I don't really have one) I'll just point out what I believe to be the most important feature for indoor use. Most mics recommended offer good enough specs in most areas except the most important one and that is off-axis response. (Lack of) Off-axis coloration is what separates Schoeps from the rest of the pack, and if I was looking for a cheap alternative I'd find the mic with the least amount of off-axis coloration.
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