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Werner Althaus

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Everything posted by Werner Althaus

  1. 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
  2. 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?
  3. 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.
  4. 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.
  5. 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.
  6. 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.
  7. 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.
  8. 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.
  9. 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.
  10. 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.
  11. yes, but this isn't about Zaxcom, I'm not up to the latest specs on digital outputs of receivers but if the AES output is 44.1, 32 or anything else other than 48KHz I'd use the analog output unless I had a chance to put the mixers' SRC circuit through its paces. Not all SRCs are created equal. If the manufacturer includes the SRC circuit in their design chances are that it'll perform well though. On the other side, the analog output of our lectro SRc and 411A receivers is not significantly degraded by a second pass of AD conversion so I trust it more than certain SRC scenarios, especially 44.1 to 48 or something similar. 32 or 96 to 48 might be okay, the math would support that. My concerns are not only related to immediately audible audio degradation but also how it affects audio restoration software like RX 6, etc.
  12. I couldn't find the relevant posts and didn't feel like digging because it was a rather unpleasant exchange IIRC. Anyway, my take on this would be that if the output of the digital receiver is anything but 48K I'd treat it like an analog device just to be safe. YMMV.
  13. Maybe I didn't word it correctly, so to clarify, nobody suggested that up sampling would improve audio bandwidth, I'm NOT asking about the merits of 32KHz vs 48KHz sampling, I'm wondering if it would make sense to connect a 32 KHz AES output from a Zaxcom receiver (32 or 96KHz, correct? or do they have 48KHz output options?) and have the AES input of the mixer/recorder do the SRC to 48KHz (which of course would add no HF audio information, just "useless data" and possible HF garbage) versus connecting the analog Zaxcom out (the "real world scenario" or, as you put it..."it would be stupid...") to an analog in and do a straight AD conversion. Would you be comfortable running varying sample rates digitally into a mixer handling all the SRC? Just curious.
  14. I remember a discussion regarding Zaxcom's 32 KHz sampling rate and how it's irrelevant because, among other things "in the real world" (so I was told) the analog output would be used and converted to 48KHz digital via the recorder. This raises the question whether SRC (32 to 48 KHz) is preferable to DAC/ ADC both in terms of quality and latency. Thanks
  15. When I record outside FX it's usually either a Schoeps XY rig or a Sennheiser MKH60/30 MS combo. Neumann RSM 191 was probably my favorite but a bit clunky in the field due to it's power/ steering box. I used 416s for this in the past but find self noise and comparatively low output to be a problem with natural ambiences. Having said that I would not dismiss the 416 outright since there are things that IMHO only a 416 can do. I find that impact sounds, punches, ball bounces and hits, baseball glove catches, pretty much anything at a rodeo, etc. really benefit from the thunderous response of a 416, the 60 and others don't even come close. There'll always be a place for the good old 416.
  16. No, it is not, that's not how frequency responses were listed, read it again: "The frequency response at 15ips was spec’d at 30Hz to 15kHz (+/-2dB)." Translation: From 30 Hz to 15KHz the frequency response falls within +/- 2dB of the input signal, what comes out matches what went in within +/- 2dB. Outside those frequencies the frequency response is less linear and what comes out can't match the input signal within the +/- 2dB range deemed acceptable for a professional machine but there's definitely frequencies above 15KHz being recorded and reproduced, unlike in digital where a filter with a ultra steep slope keeps everything above sample-rate divided by 2 out of the signal path to avoid aliasing.
  17. The board's specs likely exceeded those of the tape recorder. The tape machines' honest specs are actually quite impressive for its vintage. Analog tape machines are usually lined up at 100 Hz, 1KHz and 10 KHz, if those can be lined up reasonably flat then it is assumed that the machines are reasonably flat, Note that what happens outside of 100 Hz to 10KHz isn't included in this at all. Tape machines' frequency response depends, among other things, on the tape speed, in a nutshell you get better bass response at 15IPS and more high end at 30 IPS. But the "Low end head bump" that all machines exhibit ,usually falls below 100Hz and it can can really mess with the low end of a mix on playback, much more so than a gentle roll off of the highs, just ask Bob Clearmountain or pretty much any engineer who cut his teeth on analog tape. The fact that this machine is spected within 2dB from 30 Hz to15 KHz is quite impressive and in addition it doesn't mean that there's nothing happening above 15KHz, it just means that it's high frequency response rolls off more than 2dB at this point, unlike the Zaxcom (or any other) digital gear where there's nothing happening above, 16KHz or 20 KHz, or whatever the cut-off is. Analog vs digital, apples and oranges. And regarding what anybody can hear, there's a famous story about Geoff Emmerick's complaint about one single channel in the custom Neve console build for George Martin, it turns out that the channel in question was oscillating at 54KHz and Geoff heard it.
  18. I forgot to mention, I went to the Nashville location, I think it's much newer and maybe in better shape. Got to listen to some fine mics but nothing at VK could touch the mics I heard at Shannon Rhoades' "mic-rehab", he's Blackbirds "mic curator" and makes his own capsules for Mitek CV4 mods. That's where I'd spend my money if I was in the market.
  19. Are you referring to the video or the regular setup at the store? When I was there I found the setup with the foot switches really useful. What I remember most was the fact that there were 2 mics that really stood out from the crowd and no, they were not U47 clones. Everybody I was with that day picked those 2 as the standouts. In this video I don't hear anything conclusive, other than that the U47fet sounds rather underwhelming on vocal,(there's a reason you only see those in front of kick drums) and that the cheaper variants, while generally holding their own, appear more sibilant. Interesting also the differences in pickup pattern (all are cardioid, I assume), you can really hear the room differently in all those mics when the guy sings. I was surprised about the U87AI, the guy must not be hitting it very hard since those mics are so easily clipped rather unpleasantly by strong voices.
  20. I get it but I felt that the Sony decks -20 dBFS reference = +4dBu with a front end that could take it all the way up to 0dBFS was a beautiful thing in the field and a great match to fieldmixers that actually were capable of outputting clean +24dBu. In the meantime companies like SD have moved the goalposts and now have made 0dBu the defacto reference with a max of +20dBu output.Not crazy about that one. . For post to sacrifice 8-10 dB headroom at the top of the scale due to broadcast delivery specs had probably more to do with legacy technology in the analog transmission realm where 10dB of headroom was all you got. I still have to mix live with a -10dBFS peak ceiling for certain networks, not my favorite thing in the world but I get where they are coming from and why they are reluctant to adopt loudness normalized TruePeak levels of -2dBFS.
  21. That's really interesting, I never really thought about S/N vs dynamic range in analog vs digital systems and what it means exactly , so thank you for that insight. S/N is a confusing term anyway in this day and age when people refer to a microphone's isolation of wanted audio against unwanted audio as "Signal to noise ratio", arguing that directional mics have "better S/N", etc. I also don't think the theoretical dynamic range of a digital system as determined by some formula is of much use in determining the usable dynamic range of a given system. That's where the ears come in. Take 16bit, giving us a range that should be adequate for anything you throw at it. I used to record a few orchestral performances and operas in the late 90's/ early 2000's on Tascam DA-88s (16bit). A few years earlier I was working some with blackface ADATs (16bit) and the usable dynamic range of these two systems struck me as vastly different because the ADATs sounded pretty ratty when pushed close to 0 dBFS while the Tascam handled it much better IMO. On the other end however they both exhibited a high degree of quantization noise, dither/ noise shaping wasn't what it is today. So while the 16bit machines theoretically had a dynamic range of 96dB plus change, in reality they performed (much) worse than a good analog recorder with SR. Even today those numbers tossed around regarding dynamic range don't warrant much attention IMO ("ADC A is better because it has 137 dB of dynamic range, ADC B is bad because it only has 127" etc.) . The Zoom F8 (sorry for picking on this one, it's actually a nice machine for the money) claims a dynamic range of 120 dB, in reality I treat it as little more than half that. Maybe I'm wrong but the only 3 digit number in decibels I care about is the maximum SPL a mic can handle, that number actually matters as we all know. Thanks again.
  22. Maybe I didn't explain clearly what I was trying to say but my point was that in practice I can gain up a nice Schoeps mic to give me "good healthy and clean" levels (peaks around -3 or 4 dBFS on my Zoom F8) only to find that the mic sounds "pinched" or 'slightly distorted and shrill"while backing off the mic pre's gain and effectively lowering the ISO levels will improve the response significantly but going too far the other direction into "super-safe" territory will result in a lifeless, dull recording. These linearity issues are a fairly common problem with budget audio interfaces and mixers. I tend to gain stage according to the gear I'm using, the better the chain, the hotter (or lower, not a paradox since improved linearity affects both ends of the dynamic range) I feel I can print, within reason. The frontend on SD 442/ 664/ 552 or 633 , while all sounding different, have never given me any grief in this area but the Zoom definitely has as have the mic and line inputs of cheap cameras. I'll hit a Sony PDW700 different than a Canon C 100 and I remember the Sony HDW-F900 sounding way better when hit hard than any camcorder on the market today. So IMHO proper gain staging is one thing but knowing how your gear responds to varying levels is also important.
  23. I don't necessarily agree with this, it sounds like the old "don't waste any bits" argument of the 16 bit era. I have found that it really depends on the analog front end and ADC. Mic gain and ISO recording levels are linked (if pre-fade ISOs are recorded) so if you have a cheap mixer/ recorder (Zoom F8, etc) gaining up to "use every bit" may result in a "pinched" sound from the pres giving out before the ADC clips while under-cranking the mic gain may result in an "anemic" sound. I usually shoot for peaks in the -18-12dBFS range and as a post mixer I expect to make up 10 dB of gain or more on ISOs, no big deal as long as there's good S/N ratio, which again can be a problem with cheap mixers/ recorders when printing low levels, regardless of 24bit resolution. On quality mixers/ recorders that most here use this is less of an issue since a good quality pre and ADC will be linear across a wide range but many cheap interfaces and mixers/ recorders underperform in this area so it is something to be mindful of. I wouldn't say that a recording approaching 0 dBFS would constitute a "perfect recording" unless I knew the recording chain.
  24. I don't know if they "squeeze the intelligibility out of the tracks" or whether that's even possible. The 3rd clip sounds to me like watching Colombo on "MeTV" or some other secondary TV channel geared towards nostalgic viewers. I've heard it said many times here that shows like "Colombo" sounded really good because of competent actors projecting well and great boom ops vs lavs everywhere, however these stations, for reasons I can only guess, have turned the soundtrack into an unintelligible mess due to excessive audio compression. The 3rd scene's noisy location may not nearly be as problematic if they only laid off the compression a bit, but the fact that the dialog is still intelligible is a testament to the mixers ability to "proof" his mixes to withstand the abuses the audio may be exposed to downstream. I do agree with the comments regarding audiences expectations vs creative decisions, like I said, IMO it's done for effect, not to help with intelligibility. hard to listen to nonetheless.
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