Jose Frias

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About Jose Frias

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    www.chapterfour.tv
    www.josefrias.com

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  1. 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.
  2. I was not aware of this, since as you say, it's not documented on the manual. Would love it if someone from SD could confirm this. Thanks for bringing it to my attention!
  3. On the recorder? Just TC stamp. The HDMI connection on the MixPre only provides Timecode as well. Only wordclock will control the speed at which the recorder samples. But considering that our recorders sample at a much faster rate than cameras create frames, they tend to have more accurate clocks, and I would be more concerned about the video drifting first. You can have a sync box on camera that supplies both Timecode and Genlock.
  4. You can use the Aux input on the MixPre to feed it Timecode for syncing. This opens up a few options. You can either use sync box(es) and feed timecode to both recorders, or you can have the 702T as master feeding Timecode from it's Lemo connector to the 3.5mm aux input on the MixPre, for example.
  5. Some of these adapters may have a voltage regulator built in to step down the voltage from whatever the battery or distro is supplying to the voltage that the USB or L-mount batteries typically supply.
  6. DPA is hosting an all-day microphone masterclass next week, Tuesday 6/27 from 10:30 to 19:00 in Brooklyn, NY While the class is geared more towards music and theater, it may be of interest to some of us here. More details below: https://www.eventbrite.com/e/microphone-masterclass-new-york-city-tickets-34621342367 They are also doing one in Los Angeles two days later (6/29). Details for that one on the link below: https://www.eventbrite.com/e/microphone-masterclass-los-angeles-tickets-35035077861?aff=es2
  7. Phasing is not a product of combining two specific brands of microphones, but rather the product of combining the signals of any two microphones capturing the same sound source (whether on axis or off, though the former may accentuate phasing more strongly than the latter), but that may show slight differences in timing on the time domain. These time differences can be caused by the differences in distance between the microphones and the sound source. You can even try this yourself. Take the exact same two microphones, point them at your mouth or someone else's, and have one be slightly further away from the sound source. Listen to each signal individually and then mix them together, and voilà, hear the phasing in action. You can notice how much phasing is affecting the final mixed signal by moving one of the microphones in and out, and/or on and off axis, while the sound source continues to produce sound. Additionally any wireless system will add latency to the signal, further adding to the potential issue of phasing. Any professional location sound recorder will allow you to dial in the latency for any specific input, allowing you to better mix these inputs while providing less likelyhood of phasing. This is an audible artifact that you can deal with on location. However, without the visual aid of waveforms and spectral representation as you have in post, it is a much more difficult task to do during production. As long as you provide isolated tracks for each input to post, they can absolutely deal with any phasing issues by sliding any of the tracks forward or backwards until they have a match, even down to the sample. To be honest, though, these are some pretty basic concepts that IMO every sound engineer and sound mixer should possess, and the idea that you're inquiring about this while referring to two high end microphones that are typically found in professionals' kits seems odd to me. Assuming that you are recording the same sound source using both a boom mic and a lavalier mic, I also question why you need to mix both of these inputs, when one of them probably suffices, and in all likelyhood, sounds more than amazing on its own. This is not to say that there aren't situations where you won't find a need to mixing both, because I have in some occasions, but for the most part, the MKH50 will deliver amazing sound on its own, and will eradicate any phasing issue you could come across. As the owner of multiple MKH50s and DPA mics, I can attest to how well these two mics can be mixed together when done properly, but for a single subject interview for example, I let the $1200 mic do the heavy lifting, not the $500 one. Lastly, I will close by saying that I don't mean to discourage your questions in my response, as I always encourage folks to ask questions. I still ask many questions and will never stop doing so in my lifetime. But I do expect for folks asking questions that they do their due diligence. You could've easily found your answer by doing some searches online and doing some reading. Hope this helps. Best of luck. Cheers, José
  8. Save for a few beta units, I believe they start shipping next week. The unit I tested sounded amazing, never clip or not. I think it's a no brainer regardless.
  9. So I'll second the idea that it definitely needs to be bigger than the prototype that is being pictured. I need to be able to comfortably mix on it, and for someone with big hands, the prototype I tried out at the Pro Sound booth at NAB or AES (can't remember which one) a year or two ago was not comfortable to mix in. Aside from that, a few functions that I think are important for our line of work: - Transport controls - Independent slate and com controls - Infinity trim encoders - Metadata input shortcuts - Headphone jack and encoder a plus - User assignable buttons a plus I think that's it for now. I'm sure I'll think of other things. I'll report back when I do.
  10. Unfortunately, with the mentioned tools, there is no proper way to *accurately* monitor what the sound field being captured sounds like. Both A- and B-format need to be processed one way or another for you to be able to listen to the recorded sound field. Basically, with a microphone like the SoundField SPS200, the Sennheiser AMBEO or the Core Sound Tetramic, your recorder needs to be able to decode both A-format and B-format for you to do this, and only a few recorders can decode B-format (SD 788T, SD 744T, Aeta 4Minx, Nagra VI), and even fewer can do A-format (only Aeta 4Minx and Nagra VI that I'm aware of). I would be even cautious about trusting a recorder with encoding A-format into B-format, as the recorder's A-format to B-format converter would need to be calibrated to account for the distance between the capsules and as well as the frequency response of each capsule for the specific mic you are using, otherwise the spatial resolution in the B-format file suffers. The SoundField ST450 mkii comes with its own A-format to B-format converter box that's specifically calibrated to the particular mic it comes with, and the box also has a headphone jack that will let you monitor in stereo. You would record the B-format output from the box. This is IMO the best (and the most expensive) first order ambisonic setup you can get, especially if you want to monitor live the sound field as it will be reproduced in the final product. That being said, if your goal is merely to gauge levels, proximity, etc, then you can either PFL each of the inputs, or sum some or all of the channels into a headphone mix, though the latter runs the risk of phasing. Hope that helps.
  11. I got a chance to test this out at NAB. I was blown away by the great sound quality and amazing range. This is definitely a home run for Lectro. Strongly considering this purchase for my cart.
  12. Kelly, it was truly a pleasure to meet you. Thank you for having me in one of your videos. Though I must admit, it is quite awkward for me to be in front of the camera :-P
  13. I got a chance to see this in person yesterday. Very tiny yet powerful unit. Love the size, functionality, ability to link with Wingman via Bluetooth, and price point of these units. The Aux input is programmable for a few different functions, including LTC Timecode input. Only two things I dislike is that it has no hirose conector for power, only USB (but you can get an backplate adapter for AAs or Sony L-batts), and that you can't link all the inputs (which would've been lovely for my ambisonic mics). The latter shouldn't be difficult to accomplish via firmware update I imagine (hint hint Sound Devices).
  14. Typically, though dependent on the specific recorder you're using, user bits don't automatically change at midnight. But when using TOD timecode, the timecode will reset to 00:00:00:00 at midnight. This will lead to takes done past midnight to have earlier timecode then takes done before midnight, and when organizing files in an NLE by timecode, it can lead to confusion. Having done some post supervision now, I can tell you that it is generally good practice to do TOD minus 12 when doing overnight shoots.
  15. Zaxcom ZMT3 PH - with a Sound Guys Solutions mount.