Werner Althaus

via the internet of course. I wish I knew guys like that personally, so much to learn from guys who really have their own sound. I knew of him through his work with Kathleen Edwards. I was fortunate enough to learn about guitar amps, tone, etc from a deceased german guitar player who played his "Peter Gunn" style Blackguard Tele ( Bigsby and PAF style PU in neck position) and his '58 Flying Vee through a Dynacord S62 tape echo into a white Fender Bandmaster head with two 2x12" cabs. Once you hear that kinda sound in person at close range during practice there's no way that you'll ever go the modeler route. But for pro players who travel a lot, sure, absolutely a viable option. But hobbyists like myself don't have to worry about such things.

I'm digging the Sansamp a lot, just not for guitar. I use it in post mixing, anytime I need a recent recording to sound old I use Sansamp and/ or AudioEase Speakerphone. For guitar amplification I wouldn't touch modelers or plug ins with a 10-foot pole. main reason is that they feel sterile to me and latency, even small amounts of it are a deal breaker. I've tried a few and didn't get anything useable out of them despite or because of the convenience and ease of use. then there's the fact that if I'm not driving a guitar speaker cabinet and a monitor instead it all feels and acts wrong. I could use guitar cabs with modelers but why bother, I already have a box that has all that built in, it's called a tube guitar combo amplifier, lol. Even the current loadboxes with IR seem ill-conceived to me because they tend to make every recorded track sound the same. I demoed the UA Oxbox ( I think that's what it's called) and it sounded so formulaic and boring. The Filmosound thing is cool, I blame Blake Mills for making them popular but I first learned about them from Colin Cripps, a Canadian player who worked with Brian Adams, Blue Rodeo and Kathleen Edwards. He has fantastic tones.

I like the Sansamp plug in for ProTools, I use it everyday in audio post production. So there, this thread should now be relevant to this forum, check!😃😃😃 When I moved to the states with nothing but 2 suitcases in 1990 I really wanted a white piggyback bassman with presence control so I went to this great store in Dekalb, IL (called "Ax in hand") and they had six !!! sets of them. I proceeded to try them and they all either sounded terrible or broke down with smoke coming out of them. After the 3rd of 4th amp I was beginning to feel a bit uneasy but the guy in the store (not the owner, I'm sure) just told me to keep going, that way he can identify the amps that need fixing. After being somewhat disappointed with those particular specimen I turned my attention to my 2nd option, a brown Fender Concert with four 10" speakers. they had five !!! of those. I picked the best one, an early '60 model 5G12 (making it a late 50's circuit) with Alnico Jensen P10Qs. To me that is THE clean tone, all midrange, thick and beefy, especially when running 7591As for tubes. For Dirt I love my '59 tweed bassman , those are usually not too dirty until pushed hard but mine is equipped with a solid state rectifier and a 12AX7 in V1 instead of the lower gain 12 AY7. Those amps don't need reverb or anything for that matter, a guitar, a cable and that amp, that's it. For EL-84 tones I love my 18 watt '74 WEM Dominator MK III, that was my first "real" amp when I was a kid, before that I used my dads UHER Report Reel-to-reel field recorder. It's a poor mans' AC-15 of sorts, not very widely known in the US although anybody over 50 has probably seen that ugly red logo in old concert footage from Europe. As far as new amps go my favorite is the Teixeira Bernie amp, a modern take on the old Filmosound 385 filmprojector I've never seen one in person but the sound samples on their website are out of this world.

Hi, Izen Yes, I am familiar with the rerouting trick, I have no use for it since I don't own BF/ SF Fenders. I had a '65 deluxe reverb once, wasn't all it's cracked up to be, at least not for me, I traded it for a washing machine and I believe I came out of that ok as long as I ignore the fact that the washer is long gone and old BFDRs sell for stupid money now. I haven't tried the mid 70s tanks, sounds like a very different animal. And yes, that Spotnicks stuff is great, the sound of my misspent youth.

I don't think Twin Reverbs are more surfy at all. They have that scooped midrange that all Blackface/ Silverface Fenders have and the Reverb can't "drip" nearly as good as the three-knob 6G15 Reverb Unit. It needs to be a brownface style higher powered Fender to get "that" sound. That '62 Twin amp the OP is mentioning would be very surfy if paired with the 6G15. Our school also had two 70's Fender twin Reverbs ( the 135 watt model) with JBL speakers, those amps sounded good with midrangey Humbuckers but painfully piercing with single coils and nobody is playing surf music on a 335 or les paul AFAIK. the "old" bassman was a tweed then? Awesome amps, I'm lucky enough to own one, but not the sound of surf music IMO. EL 84 amps were used by those Euro surf bands I was mentioning earlier, Shadows, etc. BTW, does anyone know what amps the ventures used on their Japan Tour? This pic shows some odd contraptions for amps.

Yeah, I linked to that video myself in my earlier post. The footage appears to be staged ( one microphone downstage) but the audio might be live with enhancements. I think drums are overdubbed, they sound too good to be true, especially when compared to the very limited bandwidth of the audience cheering at the beginning which then crossfades very quickly to full fidelity ( canned????) Cheers. At 3:07 one guy's lone shadow crosses from right to left .Weird stuff.

Oh, what a fun thread, not sure how it fits with the rest of the forum but so what, I love it. When I think surf I thing Ventures. I always love the sound of those large brown face Fender amps, the big white piggybacks or brown combos with presence control. That and an old 6G15 is a match made in tone heaven. Here's a great video of the Ventures doing "wipeout" live ( not sure what kind of post treatment this recording got but anyway, it's great Ventures - Wipeout live in Japan 1966 Being from Europe I have a fondness of what I'd call Euro Surfbands like the Shadows or the Spotnicks from Sweden Check out these outfits. The Spotnicks - Rocket Man ( 1962) In the end it doesn't matter if you use Fender amps or Vox AC-15s and 30s like Hank Marvin did The Shadows - Apache (1960) There are tons of contemporary Surf bands, most of whom I don't care for. These guys are an exception. Amps are Fender Princeton (nonreverb model), Fender Tweed deluxe and 6G15 Reverb Blue Stingrays - Gold Finger I agree that Surf is more a style than a genre, something that can creep in and enrich all kinds of music, kinda like Blues which by itself doesn't do that much for me but I like it as a style. I like the adaptations of old traditional tunes like Miserlou, Hava Nagila. I play a little guitar on the side so I tried my hands at a version of the Klezmer classic "Di Zilberne Khasene" ( (The Silver Wedding ) I recorded this a long time ago with very basic recording gear ( 2 mics and an interface), but I thought it'd be fun to post so be gentle. 1960 Fender Concert 5G12, 1962 Reverb Unit 6G15, Old blackguard tele made from parts. Di Zilberne Khasene

I think your assumption of ISO mic feeds to the truck is correct. It appears that the PA sound mixer either was not paying attention (very unlikely!!!) or that he was fighting technical issues with his console or they put him somewhere where he couldn't see anything other than what a monitor shows him. My money is on console problems and/ or line-of-sight issues. Digital consoles are great until they decide to crash, lock up, loose tactile control or whatnot, during or right before showtime, been there, done that. What I found much more baffling were the up-cuts, missed cues and unbearable amount of extraneous noise you mentioned, how does that happen during a debate mostly moving at glacial speed??? I don't think it was auto mix, using automixing would explain the unwanted sounds getting in (Dugan and others like it don't differentiate between wanted and unwanted/ unintentional audio) but there'd be no upcuts/ missed cues. My benefit of the doubt goes to technical equipment issues because I believe that these events hire top-notch talent but man, that was hard to watch. I remember working a national issues convention hosted by PBS during the '96 campaign and it was the most professional , high quality event I've ever seen to date. They hired the audio consultant that does events like the Superbowl, they designed a PA specific to the venue, ShowCo brought in a monitor engineer who just came off the road with the Stones to run Sound-reinforcement from a platform high above the audience and talent, mixing on a beautiful Harrisson console, they had 144 Q&A mics on auto-mixers throughout the audience, the truck side of things was a rather simple affair, using an all analog Mountain Mobile Truck. The more these events move into the digital domain the more problems there seem to be, at least that's my impression. The ability to make simple things immensely complex sometimes comes back to bite you.

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.

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.

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

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.

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

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.

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.

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.

+1

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

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".

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.

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.

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.

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.

I don't mean it needs amplification on the receiving end, it just needs the correct signal strength into a balanced input with the right impedance. Sounds pretty analog to me. I am under the impression that connecting 2 pieces of audio gear via 110 or 75 Ohm AES3 involves analog circuitry despite the fact that the signal is digital. So we can say that even these digital inputs are subject to analog circuits before getting encoded. To then claim that there are no analog circuits involved when connecting an analog microphone to an ADC seems utterly implausible.

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.