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Tom Duffy

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    Los Angeles, CA
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    R&D Manager, TASCAM, mainly professional products.
  • Interested in Sound for Picture

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  1. <belated> Those are pictures of the prototype when it was called the HD-P8. The LCD section was on a lever lock like the old Sony video tape decks, but that didn't work well at all orientations. TASCAM kept that for the rack mount HS series, but the HS-P82 went to a stiffer friction hold. I am no longer with TASCAM, I am currently with Apple in Los Angeles.
  2. SD and CF manufacturers world wide are facing a shortage of flash chips this year. All chip fabs doing flash have shifted the majority of their factory lines to producing 3D NAND chips to meet the demand for high density SSD products, so the smaller size chips are in short supply -> SD card producers have to rev their designs often to use whatever chips they can get in quantity. 3D NAND has a different access speed to single layer NAND, so the specs on the SD cards are changing greatly with each revision, and are currently targeting the 4K video recording markets, which is a single file stream scenario - multiple file streams for audio recorders are not a priority. Remember that all flash media hits a wall in write speed once all the blocks have been written once - it's vital to do a full erase format that lets the card go back to "factory spec speed", before each use. If you don't, you can run into the case where a morning session was fine, but the afternoon resulted in underruns and write errors. "Quick Format" doesn't do this, and not all products have a full erase format feature, in which case you need to use a computer and the SD Consortium's formatter app. This is why media is approved - the write speed during the worst case scenario is the key, the write speed when new is not important, but that's what's written on the card. For this to be foolproof, it would have to record constantly until the card was full, then delete the files, then record until full again... That's the only way you'll see the worst case write speed. Bonus for doing a full erase format after that and another pass to double check that the worst case write speed is improved. For a 64GB card, back of the envelope calculation -> >16Hrs per pass for a 8 channel recorder.
  3. Notes from my previous research (low budget hobby usage): * The availability of a dual receiver in the Sony UPX-P03D if you want to reduce kit as much as possible. * G3 widely available used. * DIY upgrade G3 to SMA antennas) * Mics wired differently, can't get both... * Sony has USB external power, G3 requires a DIY mod or battery adapter.
  4. If by "in general", you mean only DSLR cameras have inaccurace frame rates based on poor clocks. I've never been able to measure the accuracy of say an older VHS-C camcorder, but those manufacturers knew the specs required accurate clocks. It's only when video became an afterthought that we really see the bad ones. A C200 should be at spec and much better than a 5D.
  5. Correct, but for the wrong reason. The clock accuracy for video streams is actually very precisely specified, and is under 10ppm. But DSLR cameras were never designed for video, so they only have 50ppm or (much) worse clocks. Video drift is bad on DSLRs, and there's obviously no Genlock on that class of product. Pro video cameras are at 10ppm or better, so are fine for short clips, but for long days, they're still not accurate enough to match the low ppm drifts boasted by the newer audio recorders. -> TC timestamping is still not enough. You can't completely solve the drift unless you lock the audio recorder to the DSLR somehow. This is something TASCAM considered deeply for the DR-701D, and we have a patent pending on our method as well.
  6. DR-10L coming in White. No pictures yet though. https://www.bhphotovideo.com/c/product/1331924-REG/tascam_dr_10lw_dr_10l_mini_portabl_recorder.html
  7. Tiny bit of perspective - Patents exist not to enforce a monopoly, but to allow the inventor to profit from their invention, possibly a subtle disintinction. A patent lawsuit puts into motion the first step, which is discussions to settle the dispute before going before a judge. The outcome of those discussions can be one of 3 things - pay the inventor a mutually acceptable license fee, modify the product to no longer infringe (worst case drop the product from the market), or go to the next step of going in front of a Judge, i.e. a Trial. Going in front of a judge can be either with or without a jury, and that's the point at which you get to argue whether the patent is enforceable, and/or how much infringement is actually taking place, and thus the judge decides for you what you couldn't decide on in the first place. I'm going to defend Glenn here a little bit as well - the totality of the patent does cover a unique way of solving a problem that existed - commanding all the remote wireless recorders to playback their content again, in sync, so you can capture or re-do the mix if the first mix had problems such as rx hits or just a bad mix. You'd be unlikely to do that now, you'd just grab the files from each recorder and send them to post and say "here you go", but if your job was to provide a mix, this equipment provides the ability to re-do. The way patents are worded, it's built up in steps, so at some point you have the description "recorder with timecode", and then you start to argue whether the patent covers that, or just the use of that component as part of the whole.
  8. Due to the nature of litigation, no-one involved will be able to make any comments here... so we're left with fruitless banter. If there's anything useful left to discuss, this thread should be moved to General Discussion.
  9. From experience, there are plenty of gotchas when tuning a real-time MP3 encoder. TASCAM has been doing it since the PocketStudio 5 in 2003, when the available processors were seriously lacking in power compared to what is in use now. Squirrily sound can be caused by not filtering LF, only recording one channel when using Joint Stereo or Stereo, (should use dual mono); or recording at low level, which can confuse the encoder. 128kbps (stereo) should be enough, 320kbps should be pristine. It's only when you go down to 96k or below that the result is never going to be good enough.
  10. Yes, the difference being pre PT10, you ended up with the split-out mono files in the working directory. From PT10, it could use the file directly. No difference to the PT operator, but would probably have affected what files you end up with when you export the project for Media Composer etc. (Actually, wouldn't poly files in PT10+ prevent you from dragging and slipping individual tracks if you wanted to correct for mic distances?
  11. Shipping from/to the Philiipines to the US for repair might/probably be expensive, compared to just buying a new unit. It might be just a couple of blown capacitors, or it might be damaged op-amps, so it's up to you whether you trust a local repair place to fix it cheaply.
  12. ProTools added Poly Wav file support in version 10. Before that you need to use an external app to break them out. ProTools native has a 32 channel input limit which I believe means 64 channel poly files are not supported - you need ProTools HD or HDX for >32 channels.
  13. The conversion from 96kHz to 48kHz is a mathematical process that can be done with as much precision as you desire, given the prcessing power and time available on a PC. So the resulting file will have the same "qualities" as the bottom 24kHz of the signal if it was recorded at 48kHz. So the only way for a recording at 96kHz that is converted to 48kHz to be better than a recording at straight 48kHz is if the 96kHz converter has better specs (S/N, Dynamic Range, linearity, clock jitter) than the 48kHz one. A lot of the time, you're looking at the same converter just running it at different frequencies, so the specs will be mostly the same. Since all common ADC converter chips are delta-sigma types, outputting at 96kHz can actually result in the chip doing less decimation stages, and possibly having better specs at the higher rate than the lower one. So in theory, recording at 96kHz and doing a high quality down-sample to 48kHz can result in a better recording, but only if the ADC (and analog front end) actually have better specs to start with. The downside of having to do the extra work on the computer usually puts this process out of the realm of day-to-day work, but can provide a little extra quality if it's a concert recording, for example. The comparison with video @ 4K shouldn't be made, because the capacity of a 48kHz 16 bit recording is already great enough to cover the human ear in terms of dynamic range and hearing range, so it's already down to the subjective level. (24 bit recording is the standard because you don't want to compromise both S/N and Dynamic range if you are recording quiety, and you don't want to dither that early on in the signal chain). Video 2D downsampling and motion compensated frame rate adjustment are much heavier processes where the trade offs in processing time and cost are still a large part of the decision.
  14. The TEAC DA-P20 was a rebadged Casio DA-7. The R&D team said they could do better, and thus the DA-P1 was designed. They used the combined experience from the rackmount DAT decks and the portastudio series, everyone was happy with the result.
  15. Yep. Can't beat a >2.5MHz wide waterfall to see where those signals are. which UI? HDSDR or SDRSharp ? HDSDR doesn't have any bandpass filtering on its FM decoder, so it sounds bad when picking out wireless Tx. SDRSharp works on my home PC but not my work PC for some reason.
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