RScottATL

In my experience, the Canatrans knock out a good block on either side of them and their bandwidth gives you significant range reduction for about 1-2 blocks away. With 3 running at 1/4 watt, there are small hits on all my whip antennas unless I'm quite close to talent. I do like to have the systems in my freq finder as 200khz custom bands but it is mainly for having them stored. If you set their full bandwidth in freq finder you will likely have no Intermod free freqs on your transmitter list.

dBFS refers to levels of a digital recording relative to 0 which is the maximum recordable value. The only variable is the reference level relative to 0 that your average program level and tone relates to, and if you calibrate a playback system you will have a dBSPL level that is tied to -20 dBFS so that the bits actually correspond to a sound pressure. So, 0 dBFS is just a max level of your recording track, but it has never really been tied to anything other than a -20 tone. The only reason for the tone is so that we can coordinate peaking across all recording mediums so that none overload. The 788T can certainly use the exact same process of recording content at a lower level and get the same "headroom" as the Neverclip system relative to peaking, but it does not have the extra benefits from dual A/D converters to lower converter noise. Both systems write to 24bit files, so all would be equal in workflow, but since you're writing your sound content closer to the system's self noise, anything you can do to limit self noise of your input stage will improve the hiss heard on the track. Yes if you're recording content around -40dBFS and you get huge spikes, your ears are certainly going to hurt. Your headphone limiter will help a bit, but I would enjoy a variable headphone compressor or limiting system so you can set your own preferred level of hearing protection when recording very dynamic content. It would also be great to have a toggle headphone setting to heavily compress the content for cranking up headphones when analyzing the ambient noise of a scene without exposing your ears to shouts at amped up levels. Doesn't relate to NeverClip, but it's been a little wish of mine for a while.

The system can work fine with 24 bits of resolution in the file. 24 bits gives enough room for 144dB of material, and dithering can always be used to extend this resolution even further, not that dithering is needed in this case for a 137dB system. The improved quality of the audio is there when it is recorded. Whether you tell the mix engineer or not, he will still do the same thing with the ISO tracks that he's been doing for a long time now -- set them to the appropriate level in the mix. Telling him is just the nice way to make sure he knows WHY they are quieter, but it has no bearing on whether or not the full quality of your recordings can be used. It's already there.

Oops!! Sorry, I was thinking mainly about bodypack use with lav mics. You're right, the stereo adapters have been used for a long time for hops.

Two A/D converters are addressing the signal, one working at a lower setting to address the quieter half of the audio from the preamp, one working to address the louder half of the audio from the preamp. The processor stitches these two signals together to give you an A/D converter with a much higher dynamic range. As the audio fluctuates from a low voltage to a high voltage, the processor transitions from Converter 1 to Converter 2, and then back. The entire point of the system is to overcome the fact that the converter's range degrades the preamp's quality quite a bit. It has hiss at the very quietest part of its range. And, it cannot write anything over 0dBFS, so it must use a limiter to prevent this. Consider that the "Converter 2" at the upper end of the range will only ever be addressed during brief transitions to louder modulation, and will otherwise not be heard. The only hiss from the entire A/D system will be heard from the converter working on the quieter bits, and this hiss will be pushed down 20dB. So there you have it. Your converter now has 20dB less noise. It gives you three distinct benefits if you use it as designed. First, now you have a huge space at the quieter end of your converter where you can record great audio. You wouldn't have used it before because the converter hiss lived there. So, your NeverClip converter system is free to record the preamp at its full working range. It can only do this if you write the file so that the recording is quieter than before. Instead of your audio modulating around -20dBFS, by nature, it must be written at -40dBFS (or someplace inbetween). When the post mixer gets the files, he will have to either apply +20dB of gain OR mix with his system playing back 20dB hotter. This is nothing new. Comparing the calibrations between films, TV, commmercials and CD music, we can already see plenty of places in the industry where different levels of calibration are used. If the post mixer understands what he is working with, he should be able to work with the NeverClip files. It's your job (and Glenn's) to convince him that it is worth his time to reap the rewards of cleaner tracks. The second bonus is that you avoid your input limiters. Your preamp gets recorded in all it's glory 20dB lower on your converters, and will probably not need the assistance of a limiter to keep from hitting 0dBFS. I think that it is confusing to hear Glenn's explanation of the system (and from this thread, it seems many of you guys are confused as well), because he emphasizes so much about how the limiters (and distortion) are avoided and how there is more headroom, but then we hear a track where the most obvious thing to jump out is less hiss. Limiter distortion will be avoided, but in most instances, the limiters are pretty good at their jobs. Yep, we will get better sounding car door slams and oops protection, but it's confusing if you don't separate the benefits from hiss and limiters into two separate things. The third bonus to this system is now you have a preamp/converter system that you do not have to gainstage as frequently. Once you set the preamp to the proper gain, your ISO settings are done. You don't have to adjust due to the input limiter engaging. And you will basically never have converter hiss to avoid. I think this is one of the primary reasons you see the Nomad designed with faders but not gain controls as dedicated physical knobs. If you are using the system with NeverClip, you will probably not have the need to tweak your preamp gain after you initially set it. Your entire input system now has performance that is pretty much "perfect." That's not entirely accurate, but it makes your input system's performance in terms of dynamic range and signal:noise so good that it is far from the weakest link in your gain-staging audio chain. The microphone itself, in the case of a boom and the wireless system in the case of radio mics will be the major determining factors that add a significant amount of hiss or distortion to your recordings. There are quite good boom mics out there that this system will excel with. That leaves us with radio mics. These systems have always had a less-than-ideal working range, and I think that is one of the areas that Glenn intends to address in the future. His systems already have the ability to transmit two channels of audio on a single pack, but for the most part this is useless to us (Edit: for bodypack use with lavs.) The core concept of NeverClip can be used in many things, and lav mics are a great place for improvement. Consider the future beneifts of similarly stitching together the audio from two lav capsules running into a dual-channel wireless pack. Dual-element lavs currently only exist to provide a failsafe, but the potential is there to use this technology to remove hiss and minimize gain-tweaking at the lav capsule end of our audio chain. The core idea of NeverClip is what is revolutionary -- using two systems to deliver amazing gains in dynamic range and freedom from noise in our tools. It's not exactly a new concept, but it is one of the biggest implementations of that kind of system into our daily working toolset. For those that scoff at the concept as being of little use, it helps to understand that the ENTIRE audio chain must be improved one part at a time. NeverClip converters improve one step of the chain. NeverClip wireless technology will address another step of the chain. If microphone developers can embrace dual-capsule microphones in a similar fashion, that will improve another part of the chain. It's a gradual process to improve our entire toolset, but it takes companies that are willing to push down that long path one step at a time. Hope this helps understand the system. Please correct any places where I got it wrong. -Scott

Why wouldn't you be able to use Zaxcom's wireless with your Sound Devices rig? http://www.zaxcom.com/ifb100 The IFB100 takes SMPTE timecode in or is set to free run. It transmits control signal and IFB over Zaxnet. **Edit, looks like my post is late.

Zoom H2 has 5.1. I learned that in the video. Somehow I thought it took more money to make a movie in surround sound but I found out no! Just BOOM AND ZOOM all the way to savings.

This little camera has come a long way, even considering how great the first version was. The footage on the 3 is incredible for such a small and affordable piece of kit.

A show I worked on did not want a slate in as the interviewee was discussing a battle with cancer and it was a very delicate interview. They were rolling the 5D with an audio feed patched into it. I offered to send a small burst of tone from my mixer at the head of each take. This worked out great for the producer because it gave a solid sync point on all of the audio tracks but was silent to everyone on set. Pluraleyes was going to be the preferred post method. I did not hear anything back from post.

Interesting retune, John. I would love a Canatran option as well, if only for being able to have the Canatran freqs on file so I can stay the heck away from them. I'm working on a show with three of them right now and it EATS my range. Usually if you're a block or two away it will give you the best performance, but in general, they are spitting out so much stuff that one transmitter indoors at 1/4 watt was giving me minor problems across all of my wireless. When we get all three running at 1/2 watt, my range ends up being less than 20 feet from talent (on wireless one block away from one of the transmitters, the others are further out). If I stand within 3-5 feet of a Canatran it cuts that range down by half.

The "Special connection" is a soundman with gear. I hear it is a handy connection to have. It helps make your talkie.

I'm a huge fan of the Neverclip idea as it eliminates the issue we've had for the longest time, setting our equipment to the "sweet spot" between noise and clipping. Yes, I understand that gain staging is something that we all pride ourselves on as professionals and one of the core tenants to our work, but it is JUST a workflow designed around solving equipment limitations. I don't see any reason why having the limitations removed is a bad thing. Yeah, if we are feeding a 137db recording stage with a mic that can achieve a much lower dynamic range (let's say 100db), then we may not use those 137db all the time. But it gives far more flexibility for us to place our 100db in the precise dynamic space that it should occupy and mix things at an exact loudness. Most of our dialog is recorded at -20, so that loud dialog comes in around -10 and screaming kicks the limiters a bit. That leaves no room for car door slams, coughs, screaming and the rest. I'm happy to move on to a point where our reference is -30 or -40 and we have no noise floor on the recording stage. When it's applicable, it seems like there is room for other technologies to employ the exact same methods Zaxcom is promoting to extend the dynamic range of our other tools. 137db is crazy talk if it's the only stage of the process that can use it. But what if our mics begin to be built with double capsules operating as a Neverclip-style system? What about when our radio systems use similar approaches to deliver the same dynamic range wirelessly? Then we get to spend more of our time tackling other issues and can let the gain control issues that we are so familiar with become yesterday's problems.

I'd just compensate the analog signals (hardwired boom) by the delay of the digital systems (lectro hybrid systems). In my case, 3.2ms. I would not bother much to compensate for physical differences or even attempt to change this on the fly... or ever. When the two microphones are focussed at the same talent and are mixed, if there is no compensation, the resulting sound will be hollow sounding, somewhat as if standing right next to a hard wall. If the boom is flying over talent further away, I don't perceive a big difference. The problem is only a huge deal if your final mix is what is going to air and you cannot tolerate this type of coloration. In most cases, ISO tracks mean we don't use the production mix and other means (putting a plug transmitter on the boom) can also fix the problem. If you must never accept this phasing issue and cannot deal with it in another manner, then the Nomad and 788 are great digital recorders that provide the featureset you require. As always, many great product choices makes it a good time to be shopping for a recorder. Pick the one that suits you and go make a living.

Just put them in your pocket and keep on working. I still haven't heard anyone discuss putting a loudspeaker 40 feet away and cranking it up.....

Generally when I'm working, I find natural delay by distance to be far less troublesome as I am usually trying to find a happy place for my mic 1-2 feet from the subject at most. Any signifcant (3-5 foot or 3-5ms) delay also drops the level quite a lot, and the time alignment problem is not as pronounced. I'm rarely mixing a mic a foot away with one equal level as one 5 feet away. When I'm working with two close mics and the audio is good and clean on both, I find a delay much more cumbersome and obvious a problem. So, for instance an analog path on boom paired with a digital path on lav, with both mics 1-2 feet away is not great sounding to me.

Yep. XLR shorties, usually with right angle female ends so that the weight of the mixer doesn't sit on the cable itself. Neutrik right angle connectors can be disassembled and the orientation can be changed so that the cable exits to the proper side of the bag. I leave headphones built into each bag.

It's the H4T. A revolutionary new product inspired by the Senator himself. 4 channel timecode recorder/mixer. All features have previously been discussed, nested mostly in other threads. There is no manual for the product, but all inquiries can be directed to Mike through a personal private message, or on the H4T forums as this is obviously not the appropriate place to discuss the products that we use. The product is noted for its revolutionary Hat button on the side that offers the user assistance when he is unsure of proper operating procedures using the device. When "Hatting the H4T", a user will see an impossibly prompt, extensive text readout that offers no useful information whatsoever and will usually be worded in a way that implies stupidity on the part of the user.

Our amazing team of rising stars is making the Next Big Thing! Do you have what it takes to give us your audio?

I'd love a block-selectable unit! Turn the bug into a feature and then it's pure gold! I'll just bet now that it's not that simple.

Been using the mix in with 302's mix out for 7 full-featured channels when the 442 just isn't enough. It works pretty well. I put the 302 HP out into the Return B in for monitoring PFL on the other 3 inputs.

If it is barely big enough, but is just offering too much resistance, I've been putting some WD40 on a paper towel and rubbing the tip of the wire to allow the boot and rubber seal to slide over the cable.

Larry, thank you so much for taking the time to write this. I'm still learning as a soundman and coming to terms with exactly how RF issues work, so having this kind of detail spelled out for the layman (me) is invaluable. The Lectro user manuals have long been my go-to source for RF and wireless knowledge, though a good 50% of it is still way over my head. I'm just thankful that there are folks around willing to put it out there for the sake of giving a hand up. Thanks again.

I'm pretty sure that line wouldn't have lowered the bizzaro factor.

Zoom H4N with a boom and lav makes me think that at least they are considering price. It's the ones that say "we are looking for a soundman who can bring his own gear -- email us everything you have" that really let you know they've abandoned all sanity a long time ago.

That is consistant with Lectrosonics 400 series, if I recall correctly. I believe 3.2ms as a um400-ucr411 chain.