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Werner Althaus

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

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  1. 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.
  2. 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.
  3. 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.
  4. 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.
  5. 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.
  6. 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. It seems logical that music reproduction would suffer a great deal in this scenario.
  7. 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.)
  8. 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
  9. We filled the position but due to a staff member leaving we have another opening. https://employment.unl.edu/postings/68334
  10. 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
  11. 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.
  12. 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.
  13. 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.
  14. I had to use RX 5 a few times to deal with those AC generated harmonics and if the dedicated module didn't get it, spectral repair surely did, but I see no need to argue about it. Whatever gets results is the correct choice.. I agree about FabFilter BTW. We're only talking about this because the OP's initial recordings of the 40 and 50 sounded gutted in the low end. I think we all agree on that.
  15. you mean on movie sets or in general? I don't hear a problem in the flat recording of the 435 provided here. RX 5 (or 6 or 7) does a good job with those harmonics if they are a problem but if hi-passing it works for you that's all that matters. I should clarify my earlier remarks regarding HPFs and ADCs for the benefit of the OP because they read like a contradiction to me now. Sorry if I'm repeating myself. If the HPF is a digital filter ( I believe the Zoom H6 or F8 recorders are like that) then I wouldn't bother with it because anything applied in the digital domain can be done in post as well. If low end is a problem I'd prefer to use the built-in lo-cut of the mic or some inline analog HPF. If you actually mix on location then use them if necessary to achieve a good mix . But many times a recorder just records ISOs so why bother committing to a digital HPF at 80 if you can do it just as well in post. If the filter is before the ADC then the capture of ISOs as well as the mix can of course benefit from a good HPF because if the low frequency energy is excessive ( windy outdoors or in proximity of heavy machinery or whatnot) then the headroom at the ADC is affected. Limiters will also not work too well if confronted with a ton of unnecessary low frequency content. But on a recording like the examples provided by the OP ( typical low end rumble of HVAC) I don't think the ADC is affected by the low frequency energy. here I'd rather get full range recordings because not all HPFs are created the same with regards to phase shift, ringing, overshoot, etc. The recording of the 435 flat proves this in my mind. It just sounds best when recorded flat, inebriated subject notwithstanding. If it's not needed in the field I'd prefer to do it in post. But other microphones behave differently, Our Calrec Soundfield MK IV flat will turn even moderate HVAC rumble into earthshaking low end, given a capable playback system. In this case you'd really want to record with HPF, although not at 80Hz.
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