Werner Althaus

22 hours ago, Domingo said:

I suppose that you mean the 8000 series @Werner Althaus.
The MKH 8060 seems like a wonderful microphone, I'm reading... 83dB SNR and very high output  (63mV/Pa at 25ohm!). Unfortunately my preamp cannot currently handle such high output and low impedance without clipping above ca. 110dBSPL, phantom power absence aside. I'm liking to keep the preamp discrete and to make it capable of 129dBSPL at those levels is very challenging (3.54Vrms input). But I'm sure it sounds beautiful for outdoor nature and details with equipment that can handle it.

 

Thanks for sharing your wide experience on the older models. I guess I should stop worrying about versions/age and find a well maintained one.

If the 8000 series mics are too hot for your preamp then the MKH 60 might fit the bill, it's hotter than a 416 but at 40mV/Pa it's not too sensitive and the pad brings it down to 12.5mV/Pa with a max input of 134dBSPL. I really like those mics as good allrounder. BTW, here's a useful table that helps determining whether a mic is too hot for a micpre.

At my work we have a good collection of T-powered as well as P-48 powered 415's, 416's, 435's, 815's and 816s and I have not noticed any significant difference in noise-floor relative to the powering scheme across this selection spanning close to 50 years. Having said that I am not sure I'd call any of them "low noise" when comparing them with newer MKH series microphones. I love these mics for what they do well but on occasion I had to replace them in the field for quieter, higher output variants.

2 hours ago, inspire said:

Thank you for the teaching that was good, but it was a common bucket with which I tried to prevent the wind from blowing behind. 

 

I'd advise against putting anything at the 180 degree spot/ back of the microphone, especially a bucket which is a very non-linear sound collector of sorts, especially when agitated with wind. The common misconception is that directional mics have a null at 180 degrees or various other angles , depending on the polar pattern. In practice microphones, especially shotgun microphones hardly ever have textbook patterns across the entire frequency range. Schoeps is probably the best at achieving as close as possible a textbook pattern but one quick look at the graph of the MiniCmit shows that at lower frequencies you have a significant rear lobe, meaning that this mic hears a lot of what's going on in the lower mids and bass behind the mic .  500 Hz is only down 8 dB and 250 Hz is only down 7 dB from its on-axis response.

It is very important to consider what is happening "behind" the mic.

getting clean gain vs noisy gain may reveal some audio signals that would otherwise be buried in the noisefloor but it will not give your microphone more "reach"

3 minutes ago, grawk said:

from page 29 of the manual, the tascam x8 has a choice between 24v or 48v phantom power.  

Yes, Phantom power used (+48V, 3mA×2 load), the current rating of the Mini CMIT is 2.3mA, so no real issue there although far from a comfortable margin.

5 hours ago, Throwback said:

I can't think of any circumstance where I would want to run a FetHead with a condenser mic, unless wanting to up the signal at the mic for an extremely long cable run to the preamp....

 

 

 yes, driving a long cable with a hotter signal is definitely one good reason, however there might be another one. I'm not an EE and for a long time the idea of stacking micpres has confused me but the explanation that I found recently made some sense to me, so here it goes. 

Apparently every OP amp's performance is restricted by the "gain bandwidth product" which basically states that for more gain we sacrifice audio bandwidth. If a cheap design is implemented in a prosumer level interface/ mixer then it seems possible that in order to achieve 50 or 60 dB of gain, bandwidth might be sacrificed. This certainly would shed some light on my experience with cheap micpres where the mic only sounds somewhat linear in a very narrow gain range, usually around 20-30 dB.  In that case a second fixed gain preamp might yield better results.

Maybe someone with more electrical engineering background can shed some light on my "cringe-worthy" attempt of an explanation.

FYI, the store is called "Hieber-Lindberg" and "Instrumente-Noten" refers to Instruments and sheet music. I remember the Lindberg mail-order catalogues from my youth. I assume this is that company.

Thanks for the replies. I did find a few working machines locally.

It's 2024 and I can't revive any of our old DA-88 machines that sat in storage since 2000. Our engineering department no longer works on tape machines so I'm out of luck there. I have some tapes that need digitizing but I can't decide where to send them for digitizing. Any recommendations?

Thanks.

I'm still running ProTools 11.3.2 on a RADAR Studio workstation and I can't (and don't want to) upgrade due to the RADAR folks strongly discouraging updating to a higher Windows OS ( It's running a stripped down version of 8.1)

Will Hush Pro work with my old ProTools version and windows OS?

Nobody has mentioned the youtube channel "OneMic" yet. There are lot's of videos done with one AEA stereo ribbon mic. This is one of my favorites

https://youtu.be/kVf8lifkmF8

Post production is done with clever MS processing, driving a reverb with a bandpassed send of the mid signal only.

Full band, wonderfully balanced.

https://youtu.be/af8pTJCpljU

I bought two IZ Technologies RADAR Studio workstations from VK, as well as some other stuff, always first rate, very professional. IMO SW is more for buying  stuff I can't get at my local store, like guitar pedals, recording interfaces, etc.

3 hours ago, PMC said:

THIS!

"When he watches films or TV shows at home, he turns on the subtitles in case of clarity issues – he is far from the only one – and will limit the TV’s dynamic range. (On home TVs the dynamic range is more extreme than in a cinema: this is why you often have to turn up the volume for dialogue, then down again for action."

 

I also turn subtitles on for everything I watch and my hearing is still good. Part of it is the speakers facing down or backwards in modern flatscreen TVs. I do not turn dynamic range control on since it only makes the problem worse IMO. A very dynamic mix with a cheesy AGC or some other "dynamic range control" on it sounds worse to my ears than turning the volume up and down. I have a dim/ mute switch on my remote that drops the volume to half, then full mute, dialog is always full volume, action scenes are at half, works like a charm.

 And on home TVs the dynamic range isn't more extreme, it just seems that way because in general people watch TV at lower levels than movie-goers do it the cinema and lower level dialog quickly disappears into the general domestic noise floor since the overall listening volume is much lower to begin with.

Properly understood, better S/N ratios and additional headroom in digital delivery systems are there to do away with sound-degrading artifacts  (noise floors, compression and limiting artifacts, etc) when recording and reproducing sounds at various levels, they are not for shifting loudness to more extremes on either end, that's just counterproductive.

19 hours ago, sciproductions said:

This might be slightly controversial but I always hated the Zaxcom recorder pre-amps.

 

It always felt so lifeless to me, this is compared to 7-Series Sound Devices, Cantar X-3 and Sonosax R4+

 

They are definitely quiet, but just lack something that I can't quite explain (if that makes sense).

 

I will say that the Nova sounds better than the Nomad or Maxx though.

I think it's not as simple as that. Certain pre amps play well with certain microphones. With mic inputs we're looking for impedance bridging so a ratio of >10:1 is considered ideal. A Shure SM 57 wants to see at least 1.5KOhm and increasing/ decreasing the input impedance will alter the response. I'm unable to get impedance specs on the Zaxcom but some of the SoundDevices pres have input impedances of up to 4 KOhm brining them closer to certain studio preamps like Millennia or even specific Ribbon micpres that go up to 18KOhm or higher. With most modern mics it doesn't matter but without knowing what mic you base your opinion on it is hard to follow along.

20 hours ago, Dan Wake said:

thx for the suggestions I’m trying to figure it out this is new for me. with the objective in mind to avoid phase issues for theatrical 5.1mix maybe is it better to just record 4 separate tracks and do not use plugins? I mean raw 4 tracks straight on my DAW timelines.

If I buy an ambisonic mic it will be the Rode nt-sf1 (the others are too expansive). I tested them plugin with a bunch of sample files, the plugin offer a 5.1 conversion. Should I avoid that? I hope you have tested it maybe you can give me some feedbacks. 
If have time I would be really grateful if you could test the plugin and the samples (at the bottom of the page) here https://it.rode.com/soundfieldplugin

Thx Ramallo. Are little phase issues something unaboidable also using other recording technique for surround ambient recording? 
I'm new to surround recording and editing/mixing, in the past I've worked only in stereo, this is a new journey for me and I have many things to learn about it.

It might help to first understand that Ambisonic B-Format is basically Stereo Mid-Side times 3 ( since it's along 3 axis, left to right, front to back and up to down).and just like MS Stereo it is very compatible with both up and down mixing.Hence my comment that it can be safe. But does it sound good?

I think you misunderstood my comment about plug-ins and phase. Someone will have to use either a plug-in or hardware to first convert A-Format to B-Format and then to create the desired channel based configuration for the mic. On our Calrec MK-4 the hardware controller converts A Format to B-Format and it can  generate various patterns from Omni to figure 8 in stereo ( No 5.1 with the old hardware controller) and everything in between, as well as rotate the mic along the horizontal and vertical plane. However, the analog circuitry involved creates phase issues and the digital plug ins do a much better job at this math but if pushed to extremes you can still get phase artifacts, at least to my ears. Other problems arise from the correction filters in the plug in not matching the mic's physical distances between the capsules.

If you have a specific mic in mind ( the Rode) and use the Rode plug in to generate the channel based output ( 5.1 for example) you'll be fine, if you send raw A-format and the post house uses something else to convert to B-Format and then to channel based audio you might have issues. Remember that A-format is basically useless before conversion to B-Format.

20 hours ago, Dan Wake said:

Is first order Ambisonic* ambient recording safe for 5.1 mix in film theaters? Will brings problems as phase cancellation?

 

Thanks,

 

Dan

 

*mic models as for examples: Sennheiser AMBEO VR, Rode nt-sf1

I think it can be "safe" but I don't think it's a particularly good choice. The spatial resolution and soundstage of FOA isn't very good except at  close range , the sweetspot is very small. Converting to channel based formats ( mono,stereo,5.1, etc) requires a matching plug-in with the appropriate correction for that particular microphone's set of capsules only being "somewhat coincident", so you'd use the Sennheiser Ambeo plug for the Sennheiser Ambeo mic but that plug-in doesn't do conversion to channelbased audio, it only converts A to B format and steers the mic's 0 axis. It's strictly for 360/ VR.

 I've used the Harpex for Ambeo and Soundfield mics. The soundfield plugin sounds better with the Soundfield mic, however, for the Ambeo mic the Harpex sounds better than the soundfield/ Rode plug-in. And while somewhat coincident you can certainly create phase issues with the plug-in decoding to channelbased formats. I can't imagine anyone mixing for theatrical release being too thrilled to be handed B-format audio but I could be wrong about that.  

1 hour ago, Ty Ford said:

so, not just a lot of data compression, real transformation!  That's sort of scary.

 

My understanding is that both start with A2D conversion and limit the bandwidth but while "regular" data compression algorithms ( mp3, etc) rely on psychoacoustics and bandwidth limitations,  in cellphones they use LPC (linear predictive coding) which starts with hard bandlimiting ( <3KHz, just like POTS) but then removes those elements of speech which it can express/ transmit in a much reduced dataset, sending only the residual audio (plosives, sibilance, consonants etc) as actual digital audio. On the receiving end the residue and the data about the formants are synthesized into speech.

The engine driving speech production is represented as an acoustic tube and a buzzer and can be regenerated at the other end if the modifiers (throat shape, etc) are known.

I apologize for this sketchy, limited explanation but I'm trying to wrap my mind around this as well.

It kind of reminds me of generating room tones using IR. I do this in post a lot when roomtone isn't available for one reason or another .I use a little snippet of dead space between words of a clip of dialogue to generate an impulse response and then feed white noise into that IR loaded into an IR reverb. The white noise is the engine driving the synthesis of room tone and wouldn't need to be transmitted, it could  just be generated during reproduction so 1 hr of roomtone could be expressed with very little data, the IR and the metadata describing the level of white noise. I hope this makes sense.

I have very little knowledge about this but I've heard that with cellphones the voice you hear is modeled in 10 ms increments . A quick search revealed this statement from a research paper that makes the point rather well. 

Quote

The parameters that are actually sent from one cell phone to another are vocal tract coefficients

related to the frequency response of the vocal tract and source coefficients

related to the residual signal.

The fact that the vocal tract coefficients are very much related to the

geometric configuration of the vocal tract for each frame of 10 ms of

speech calls for an important conclusion: cell phones, in a way, transmit a

picture of our vocal tract rather than the speech it produces.

It seems logical that music reproduction would  suffer a great deal in this scenario.

13 hours ago, The Immoral Mr Teas said:

...As a dialogue editor, if confronted with room tone recorded on an ambisonic mic, I would:

 

1. swear

2. listen to decide if it was any good ...

 

3. if, as expected, it was no good, swear again, then laugh, then find something in my library

4. if it actually seemed useful, say "oh ok" (possibly swear again whilst laughing) and either choose (if possible) or randomly select a single channel of the 4 (to NINE?!?) available and delete the rest.

 

Although I would expect '3' to be the final stage and if pressed for time just '1'.

 

Different recording techniques for different purposes. Different media even. Buy the mic that suits what you REALLY want to do ...

1 x hyper, or

2 x cardioid, or

1 x ambisonic

... none of these options (for instance) can be a desirable replacement for any of the others.

 

Jez Adamson

 

 

Just out of curiosity, are you saying if someone delivered B-Format room tone to you, you'd swear....OR...anything captured with an Ambisonics mic and decoded as , mono, coincidental stereo, 5.1 or whatever would make you swear?

I'm only asking because, while  I've never used an Ambisonic mic for roomtone, I have used Ambisonics mics extensively to decode in post as needed, mostly for music recording. I find the decoding options of a Soundfield mic to be very useful, kind of like MS, only times three ( which is exactly what it is.)

2 hours ago, Derek H said:

 

what is a “nadir”?

What are the high end choices for Ambisonics mics. Also is “ambisonics“ intellectual property of a certain company or a generic term?

Nadir is the lowest point from the observer, the camera. If you look straight down in 360 videos you'll most likely see a logo, bug or other graphic to cover up the area where the tripod and sound bag would be.

High end choice for Ambisonic mics would be the Soundfield mic.I believe that  Rode bought the company a few years ago but I doubt that their current offering ( NT-SF1) can compete with the Original Soundfield product line.

I use a Calrec Soundfield MK-IV and a Sennheiser Ambeo. The Calrec is absolutely amazing sounding while the Ambeo sounds like a cheap condenser that happens to be ambisonic.

To my knowledge "Ambisonics" is a generic term. It's important to remember that Ambisonics wasn't created to accommodate VR, 360  videos, or anything like that. It just happens to be the perfect format for distributing 360 / VR audio. Here's a nice article from 1979 to fill you in on it's history.

https://www.ambisonic.net/sfexp.html

We filled the position but due to a staff member leaving we have another opening.

https://employment.unl.edu/postings/68334

I have removed various elements of applause with RX 5 but I'm skeptical about removing the entire applause. In my case I had some video of an unveiling event at National Statuary Hall and a few people must have been standing very close to the mic during the applause. I was able to completely isolate and remove those "offending" claps from the track using various passes of de-click. But in general applause is fairly broadband and ambient so removing it will undoubtedly negatively affect the audio you want to preserve.

One thing that I'd try is MS decoding. It is entirely possible that the majority of the applause resides within the Side signal but there's no guarantees, it really depends on the recording. If it does then you could turn down the side signal vs the mid signal during the problem spots. You could also try to add ( sample accurate, please) an inverted and  band passed track to the problem spots ( try to zero in on the most prominent frequencies of the applause), You might be able to null out a good amount of the most noticeable components of the applause. maybe try that on the side signal only.

None of this is guaranteed to work but it might be worth a try.

Good luck

22 hours ago, Dan Brockett said:

 To me, it reminds me of news, which if you are shooting news, great but what if you're shooting higher end interviews with A list talent on a nice set with great lighting?

 

In that situation I always use an overhead mic ( Schoeps MK 41) on a boom, either fixed or handheld, depending on the movements of the talent. Lavs are useful when an overhead isn't feasible due to external noise, bad room acoustics, talent is in motion while talking, etc. If the situation requires use of lavs I try to not hide them unless someone insists that I do. If I have to hide the lav then I try to keep the mic capsule exposed or at least away from layers of fabric.

22 hours ago, Trey LaCroix said:

I have a 20 year old Schoeps and a 4 year old schoeps and did a blind test a while back. I could not for the life of me tell the difference.

The consistent difference I can tell between our mid 80's Schoeps and newer ones is the improved RF shielding. All in all we own a dozen or more and in a difficult RF environment the old ones are more susceptible to interference. I don't know if this is due to preamp design, the capsule or both.

Beyond that I'd say that each of the old capsules now sounds slightly different due to different exposure to the elements, handling, abuse during traveling, etc, while all the newer ones sound identical.

If your 7506s sound too bright it might be time to replace them. That's my main problem with these, they do not age well and age they do fast ( a couple of years).

It creeps up on you very gradually but I force my co-workers to check their old 7506's against a fresh pair on a regular basis and it's always shocking to them how the low end and low mids are just gone after a few years of use. I don't know what it is that makes them age so poorly but when new they're great and for the run and gun style shooting we do they are perfect because you can still move safely without being too isolated from the outside world.