Ze Frias

If it's a VR/360 video production company, they will specifically ask for "spatial audio" which seems to have become the default nomenclature for this kind of work. They may specify that they want Ambisonics, which is obviously not the only spatial audio format, but it seems to be the most widely adopted one for VR and 360. A professional VR production company would definitely understand the entire pipeline. Now if it's a regular production company that is doing a VR / 360 video project, then it may be a different situation where they know very little to nothing of the workflow and pipeline, but they have some basic understanding of the language. In those cases most of the clients I have worked with will hire a VR sup or tech who guides them through the process and who communicates all the technical requirements for the project. Depends on the project, but for most of the VR / 360 video projects my company and I work on I'm delivering to post: - Multi-track files (sync'ed with video) with all isolated sound sources (mostly dialog or speaking parts, sometimes interesting production sound effects that can't be replaced or reproduced in post) - Wild mono files of isolated sound sources - Wild 1OA A-format files of room tone and ambiences - Wild 1OA A-format files of impulse responses In post, for a linear project like a 360 video we will sync the multi-track file to video. Lay down some to all of the wild sounds, add Foley and sound design as appropriate. We pan, or as we like to call it "spatialize", all of these tracks into a 3OA mix. We convert A- to B-format, and upconvert the 1OA room tone to 3OA and lay them down in the mix. We feed the 1OA impulse responses through a convolver that we use for reverb on the ISOs. Once the 3OA mix is done, it is muxed with the final 360 video for delivery to the platforms that support 360 video playback with Ambisonics (YouTube, Facebook, LittlStar, SamsungVR, etc), or it is packaged into an app for the Oculus store or Steam VR for example. If the project is interactive, then we build the audio through wwide, an audio engine that operates as middleware for a game engines like Unity or Unreal. BTW, 1OA files we typically record with the MixPre-6.

To a degree. It's not so much about knowing how the sound field will sound like without using decoders, as much as it is about knowing what can present as issues when converting the A- to B-format. i.e. wind noise, suspension rumble, if there's any weird noise artifacts in any of the capsules, etc. Most can do both, record and monitor the signal post processing, or record the signal raw and monitor post processing. The latter is recommended as the A- to B-format conversions in recorders use a generic algorithm, and in post you can use better conversion algorithms.

FOA = First Order Ambisonic. Also acronymed as 1OA. HOA = Higher Order Ambisonic. This includes Second Order Ambisonic or SOA / 2OA, Third Order Ambisonic or TOA / 3OA, and above (4OA, 5OA, 6OA, 7OA, etc) The higher the order the better the spatial resolution, which typically translates to better localisation of sound sources within the sound field reproduction, but it also means the file will be larger due to the exponential increase in channels, and it also means it will require more processing power to decode. Most video games live in the 3OA world. VR/360 videos in its current iteration started mostly as 1OA, though other formats were also available like omni binaural, but currently there's a wave of 3OA tools making its way to content creators and post production professionals. Since all you need is 4 channels with mic pres (hopefully all matched with gangable parameters), quite a few field recorders offer a generic A-to-B format converter for 1OA which allows the field recordist to monitor the sound field decoded to stereo for an idea of what it will sound like later, but for the most part and if you've done this enough to understand how to work with ambisonic mics and how they get processed in post, you'll develop a workflow that doesn't require you to monitor the sound field per se, and then it's about gauging levels, making sure all inputs have matched settings, there's no wind noise, etc. So any recorder with gangable inputs will do. HOA recordings are only available through the very few 2OA mics, including the Octomic from Core Sound, or through high array microphones like the Eigenmike, and beamforming / matricing to convert to HOA like 5OA. The latter would require a laptop system to record since it's 32 channels.

I do love the Harpex plug-in, and think it is a great investment. I did not personally get my SPS200 calibrated yet. It was my intention to do so sometime last year, and I had reached out to Len to get it done, but then I got into a string of jobs and couldn't make it happen. These days, I'm more interested in HOA so I haven't put too much mind into it.

Tom Craca, Brian Garfield, Ted Roth are all top notch!

Len will also calibrate other tetrahedral mics like the SoundField SPS200 and deliver calibration files for use with VVMic, VVTetra or VVEncode. Or at least he used to, not sure if he is still doing that. While in theory, matched pre-amps are definitely the way to go, I've found that in the case of generic A-to-B conversion as it happens with the SoundField SPS200 you already lose some spatial resolution and that 1st order ambisonic is already low spatial resolution in comparison to higher order ambisonics anyway. I think your money is better spent in sending your SPS200 to Core Sound (they're based in NJ) for calibration so that you get better B-format files. Or maybe on a HOA microphone solution, though not many exist.

Ambient and TCS slates boast some awesome features, primarily wireless sync if you're using their wireless sync networks. The ambient slate does have a neat handle that makes one hand slating so effortless. I know lots of AC's who love it. The TCS slate (actually made by Denecke) has a second screen below the time code LED screen that allows you to have things like scene and take, and seems like a neat step in the right direction, but unfortunately places the responsibility of labeling the slate away from the camera department / script supervisor, and onto the mixer, which has its pros and cons. All this said, I'm still a fan of my Denecke TS-C slate. It's been my go to for 75% of my jobs. Small size is amazing, and it's built like a tank. I would love to see a similar version that can tether to any one of these wireless sync networks like Ambient's, TCS' or Zaxnet.

This very much depends on a lot of factors, mainly, what is the RF environment like, how many channels of wireless you are running, etc. If you coordinate your frequencies well, and you make sure there's enough physical distance between the transmitter and the receivers to make sure you're not desensitizing the receivers' front end, you should be ok. If you add things like band-pass filters in front of the receivers, then you can make sure that any signal outside of the filter gets attenuated. In short, it can be done, but you have to be meticulously careful. There's pros and cons to either of these choices. 941 seems to be super clean even in really crowded RF environments like Times Square, but I suspect that may not last long as the available spectrum reduces after the post-spectrum auction business comes to a conclusion and manufacturers (like Lectro) continue to make wireless transmitters in other viable TV channels (like 941). I'm sure it won't be long before other wireless manufacturers follow suit, and not just audio, probably video too. VHF is great because it doesn't require too much power to transmit over long distances, and no matter what, you can always find good frequencies to work. Both are definitely viable choices. They do not. The Duet is a fully digital system, whereas the R1a's are not, they're digital hybrid. That's a harder question to answer, as everybody's taste is different. All the options you've mentioned are great. I think ultimately it will come down to money and preference. Option 1 seems like the most frugal choice, and you can definitely make it work. Other options like Comtek are definitely more accessible price-wise. Myself and many other folks I know use Comteks for everyone who does not need to critically listen to the program. That is most of the folks in video village (clients, producers, etc). Usually better IFBs like Lectro IFB / Duet only go to sound department, maybe script supervisor, maybe director. So that may be another option, spend the money on quality IFBs where it's really needed, and the rest get Comteks. Also look at the Sennheiser IEM. They do sound good (definitely better than comtek), and fall somewhere between a Lectro R1a and a Comtek PR-216.

Text me, let's talk. I did something similar for my current company.

I think everyone's college experience is going to be different, and usually, you get out of it what you put in. I also went to NYU, and while I thought it was ridiculously expensive (could have probably gone to a city college and learn all the same for a fraction of the cost), I did value the connections that I made there (I still work with a few of my friends / old classmates who went on to make their own production companies). Chat Gunter's production sound classes were actually some of my favorites, as they were all practical. You actually got hands on gear and set experience. He was definitely eccentric, but from what I remember he does care, and goes above and beyond to make sure you get it. I can't speak about him telling students what rates to go out for, as he never spoke about rates in my class, and in fact, my biggest feedback about his class was that there is no mention of the business side of things (though you could say the same about any college class really). I know a few handfuls of professional working sound mixers who took Chat's classes, so I'd say your experience, while not invalid, is probably very rare. Being a professor/teacher is extremely difficult. I've been on that side as well. Most, if not all, colleges and universities have a rigorous vetting process to make sure you have the appropriate credentials (whether it be school, job experience, or both), and usually require the recommendation from other active professors in the univeristy before you can begin teaching. Not only that, you're usually on a trial period for one or two semesters before they actually hire you full time. Then starts the tenure track. Despite what you may think of him, Chat does have a pretty decent resume, and IMO he does know his shit (at least from a scripted work standpoint). He's been teaching for quite a while now. I'm sure NYU is keeping him on the roster for a reason. Anyway, I digress. Best of luck to you in your endeavors.

Would another sound mixer qualify as an objective 3rd party?

For the most part, I doubt the public will get any actual news or information from either of the parties regarding the status of the suit until the case is done. Unfortunately not much can be done except for letting it run its course.

ZHD will only be available as a single channel of audio. If you wish to have two channels of audio you will need to use the "Stereo" modulation scheme. As you probably can already imagine, not possible unfortunately. Very different guts.

New wideband camera link from Zaxcom: https://zaxcom.com/zaxcom-adds-wide-band-trxcl3-camera-link-transmitter/

Tyler Faison is indeed the creator of this app (along with the TV Timecard app). He's also a working sound mixer based in Dallas. Great guy. I'm sure you can email or call him. His contact info is on his website: http://tylerfaison.com He also roams around these forums every once in a while.

Copy that, thanks for the correction. I can't tell a difference in weight, but I had assumed that because it's smaller it was also lighter. I totally agree with your last sentence.

I'm using the Orca Bags quick releases. Smallest and lightest of the bunch, and very easy to snap in. http://www.orcabags.com/product/orca-boom-pole-quick-release/

Hey Jeff, as I mentioned on the Facebook post, didn't intend to sound dismissive about the Deva 24. Just not the solution I'm looking for (digital mixer with Dante and DC power). Unless it's changed, Dante will only be available as an option through the Mix-16 control surface, which would then interface to the Deva24 via it's connector. While definitely a solution, not what I would personally want as it means you need both the Mix-16 and the Deva 24 to get Dante running. At least when it was the Deva 32, you'd really only need a PC computer to set up the interface software, which you would require anyway to setup Dante, and with something like the Kangaroo PC you could get an inexpensive yet small and powerful solution to do just that, also with DC power.

I've been talking about this for at least two years since Zaxcom announced the now defunct Deva 32 concept. It would've been the first Dante-enabled digital mixer with DC power (and built-in recorder too). Unfortunately for those of us wanting such a product, the Deva 32 transformed into the Deva 24 instead, which is no longer the same thing. I so desperately would like a digital mixing surface with at least 12 banking faders (for most situations, enough, especially considering I only have 10 fingers to mix with, so that should be enough) that fits within a 19" rackmount space (and doesn't weight 50lbs), is DC-powered, and is Dante enabled. P&G faders would be nice, but not a requirement; it would need to be motorized, though. Infinity knobs for trim and settings. I'd be okay with a "Touch-and-go" knob a la Yamaha for trim and other settings. I think having some analog and digital I/O may be useful to connect to other pieces in the cart that are not Dante-compatible (say IFB transmitter), but I don't see having any phantom power mic inputs on the mixer necessary. Most inputs would be via wireless (so Dante), I would only really need one input for my talkback mic, but phantom isn't necessary for that. And if I actually did need phantom power, I could easily interface any Dante I/O unit separately depending on how many mic inputs I need, all of which could be powered easily over ethernet. So in essence, all I would need at the back of the unit is maybe 2-4 mic/line inputs, 4 mic/line/AES outputs, and you could even throw a bigger AES I/O count (8 in / 8 out) via multi-pin connector just in case. Any other serious I/O I would do via Dante. This would hopefully keep the size and weight of the unit down. Flexibility is the most important thing. Being able to do custom routing with multiple buses is key, otherwise Dante is useless. Built-in DSP can be a big plus; EQ, compression, limiter, gate, delay, phase, even an automix algorithm would be welcomed additions. To that end, having a decent sized screen to be able to see settings, routing, effects, etc is essential. I do like the idea of small LED screen per channel that will display channel / track naming, as well as values for variables that are being affected on that particular channel. Unfortunately, as many others have mentioned, I don't know that there are many companies that would be willing to make such a product considering that it wouldn't be many of us buying (I've asked). It's definitely a niche product.

Assuming that the director wants video playback as well, not just audio, then just depending on your Nomad won't do the job. In fact, it is actually another person's whole job to do this exact thing, record takes to playback for director. Usually it means feeding audio via XLR to either camera directly, or to the Video Assist who then dials the proper delay to match and record with video.

Good to know. Thanks Paul!

I also think the MixPre-3 would probably be your best bet.

Tom, I think you keep inferring things from my comments that I'm not implying. All I keep saying is that a camera's clock (as you stated in professional video cameras usually less than 10ppm) is in general less accurate than a recorder's clock (In my experience, professional recorders will have less than 1ppm). Until now, I hadn't stated any specific accuracy values, nor any specific examples. The Canon C300 is considered a professional video camera by many standards, however as you probably already know, it will start drifting very quickly in comparison to a recorder like Sound Devices 6-series. There's obviously exceptions to everything. Arri Alexa is extremely accurate at 0.2ppm. Anyway, the point of all this here being that the Sound Devices MixPre-3/6 could likely work well with professional cameras as well as DSLR-types by having a timecode generator feeding it timecode via Aux In, and having a sync box feed timecode and genlock to the camera (i.e. Canon C300).

I never stated any reasons. Just that in general, video camera clocks are less accurate than those of recorders. I do agree with your statement regarding TC stamping not being enough even with pro video camera when doing longer takes.