Werner Althaus

I know that AES 3 is PCM but the electrical (as opposed to optical) signal travels over copper and has an analog signal strength, it is subject to limitations imposed by cable length, capacitance, EMI/ RFI etc. It needs to be plugged into either a balanced input (110 Ohm) or unbalanced input (75 Ohm). The fact that it is a PCM stream doesn't change the fact that it needs analog circuitry to be distributed. My point is that the claim that there is no analog circuitry involved at the input doesn't seem plausible to me. Again, if someone with more knowledge could clear up some of these issues that would be useful.

Yeah, that would be helpful. I'm not an EE and am going way outside of my comfort zone but I am under the impression that even AES3 digital inputs (except TOSLINK) , balanced or unbalanced require analog circuitry at the input. As long as copper is involved you are plugging into analog circuitry, the difference is whether the signal itself is analog (as in microphone signals) or digital, if it's analog then the circuit / op amp(s) determine the gain, signal to noise ratio and audio bandwidth. Please correct me if I am wrong but that is my layman's understanding.

The zoom guy in the videos clearly states that the pres are the same as in the F8 and that the mic/ line switch needs to be in the correct position. The idea that there is "no analog circuitry before the converter" is not really believable IMO. In order to connect a microphone you need some balanced analog circuitry at the front end of the AD converter and the mic needs to see a bridging impedance for optimum signal transfer.

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.

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.

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

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.

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

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?

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

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

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.

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.

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.