Jump to content

All Activity

This stream auto-updates     

  1. Past hour
  2. while I agree that the fix it in post mindset is often problematic, I can think of situations where recorder with dual ADC and 32bit float would be useful. like for example I've done some acoustic concert filming/recording as one man crew for friends, so I couldn't adjust the trims all the time, so I had to record at rather low gain settings where the quiet parts were recorded on very low levels while I still hit the limiters for the peaks. with a properly designed dual ACD 32bit recorder, I really wouldn't have to worry about this at all since the high gain preamp/ADC would capture all the quiet parts and the low gain circuit would take capture the loud parts and all would be saved in a file format that makes it impossible to clip or distort, no matter what I set the recorder level at, so it's basically set and forget (and even the set part is reduced to setting the mics and not the levels). of course there would be extra work involved in post, but even if I have my full attention on riding the trims during the performance, I'll still have to do additional levelling anyway and might even fight with the variable levels that are burnt into the recording. I also agree that we can make perfectly fine recordings with the current gear, but personally I think the main reason that prevents 32bit float from being useful is just the post workflow/tools are not established yet because the idea of having a recorder which can basically capture the full output range of any mic without any clipping or distortion sure sounds intriguing to me. chris
  3. Today
  4. There posts about it the Zaxcom forums. I’ve never used Zaxcom but many other CF card speced hardware. And there are quite a lot of the thicker Type II adapters but only a few thinner Type I adapters and I’ve tried many of them and some are really unreliable in low temps or does not write fast enough while others are stable and good performers. One of the better are the Type I in plastic from Delock. They are rock solid down to -25C during winters as well as high humidity and temps in the 40C. Ans they work great for WiFi cards like the FlashAir if that kind of function is needed. https://www.delock.com/produkte/S_62637/merkmale.html
  5. I see, that wasn't my intention. I'm with you and think it is very likely that they will use dual 24bit ADCs and stitch their data together in a 32bit float file. Initially 0dBFS of this file would line up with the clipping point of the more insensitive ADC. At this point 0dBFS is of course still the maximum signal level you can record. It is only after the initial AD conversion that you could push values above 0dBFS with the digital gain. But this would then be recoverable because of the 32bit float.
  6. Been using one on the end of a pole for a week now. Much lighter and lower profile than the connector it replaced. Haven't noticed any issues I can trace to the connector, which seems pretty legit. Ran into one issue with humidity, but I believe that was the mic as other mics were fine. It was a little tedious to solder as you have to strip only very short lengths, then muscle the cap on. There's not a lot of wiggle room for mistakes. Also, I had some trouble getting thicker mic cable into the housing. I had to ream out the hole a little bit, but that was straightforward.
  7. Hey Guys, Just starting to research the new SMWB transmitters and I'm wondering if I'm missing something in regards to the software on the transmitter itself and updating. Is what is available at this link: https://www.lectrosonics.com/Support/category/98-smwb-smdwb.html the software running on the transmitter (the OS) or is this something more low level than that? If so, are OS updates available to do yourself or do you have to send them in? Thanks
  8. On initial ZMT3 transmitter tests, both "Remote Standby" and "LOW2" offer about a 2:1 battery life savings.
  9. Yesterday
  10. Yes! I find it cleaner than the mixpre 10t with sm7b mics. And as a matter of convenience I use it with my DPA mics when recording video, but in general either one can be used just fine. I think the max is lower noise and cleaner but that’s not saying that the mix pre is not low noise because it is very low noise. I do love the Wingman app and that’s very nice feature being able to easily change track names with an iPad. For Ultimate quality and where never clip comes in to play it will be the max but for general low-cost high speed recording the mixpre 10 is sweet.
  11. my understanding has always been that any bits get added to the bottom of the scale. unless something changed i would think that those extra 8bits get placed at the bottom of the scale not in the 0 dbfs or above the 0 dbfs. its really not like tape where you can bias the machine differently along with different formulations. i compare it to overxposing film/sensor. at some point, if its overexposed you lose all details and sich. the magic halpens when you add more shadow details. its the same thing here i feel like. perhaps i’m wrong and 32 bit float goves you bits above 0dbfs. i never record at 32bit fixed or float. 24bits has great dynamic range as it is.
  12. Another video from NAB was posted. Seems they've been using them on Marvelous Mrs Maisel. Gotham Sound TV NAB Shure
  13. osa


    Glenn in the Gotham video mentioned the option of using sd card adapters in the cf slots - is this something nomad users do as well or is it recommended to stick with approved cf cards
  14. The RSM 190 and RSM 191 are exactly the same microphone. Neumann didn't sell these microphones separately, but only in sets with accessories and a carrying case, and it's the accessories (mainly the required matrix box) that differ by "generation". There have been three different types of matrix box (MTX 190, MTX 191 and MTX 191 A), with differences in cabling and powering arrangements between the MTX 190 on the one hand, and the two MTX 191 models on the other. But any of these matrix boxes can be used equally well with a microphone labeled either as "RSM 190" or as "RSM 191", as long as you use the right type of cable between the microphone and the matrix box. The main functional difference among the matrix boxes is that the MTX 190 requires external 48 Volt phantom powering, while the MTX 191 [A] offers a compartment for a 9V battery for when phantom powering isn't available, plus a toggle switch for battery vs. phantom operation. Specifications are identical for both types of powering. Battery life is claimed to be 8 hours for an alkaline, which I haven't tested. The MTX 191 has a low battery LED, while the MTX 191 A has a "battery test" position on its battery-vs.-phantom toggle switch. The MTX 191 and 191 A also have a 10 dB pad switch, a switch to shift the (always on) low-cut filter up from 40 to 80 or 200 Hz, and a switch that reverses the L vs. R outputs in either X/Y or M/S mode--none of which the MTX 190 has. Speaking of low-cut filters, there is also a gradual rolloff filter for frequencies below about 150 Hz for the "S" channel (figure-8) only, in the body of the microphone itself. This filter can be bypassed by unsoldering a bridge on the circuit board. Bypassing the filter makes the stereo pickup more spacious-sounding, but of course it also increases the risk of wind and handling noise. The filter is engaged when the mike comes from the factory, but if you buy a used RSM 190 or 191, this solder bridge should be checked (pages 9 and 10 of the instruction manual) to make sure that it's set the way you need it. --best regards
  15. Nonobstant everything that's being said about the 32bit technicals, I'm kind of curious to try it when it launches just for form factor; the idea of only riding levels instead of gain+levels usual combo, and a center lcd on a not so wide unit is a good thing for one hand operation while the other holds the boom.
  16. I wonder wether the detuning effect caused by the body in the near field has been taking into account. That effect should be maybe a little less for higher frequencies. I remember when I mentioned the 60 GHz band, with a wavelength of 5 mm the body would be in the far field. I guess the detuning effect is worse than absorption? I remember two years ago I saw some flat magnetic antennas designed to sit on metal enclosures. It was a discreet antenna (so it could be used in vending machines without stupid users breaking them off for fun) and indeed they didn't tune properly if placed on a wooden table (according to my toy VNA). But they worked pretty well when attached to a metal surface. The same manufacturer has a model or two designed to be taped on glass as well.
  17. This is correct, but we are talking about the 32-bit floating point in a >>recording<< application when it comes to the F6. Your original comment I quoted made it seem that you can still >>record<< above 0dBFS: "32bit float on the other hand allows the signal to go above 0dBFS and can still capture it just fine". Unless Zoom is doing some magic I'm unaware of here, their double ADCs are still bound to a fixed point / integer math, and unless they calibrate 0dBFS to not represent the full scale of their ADCs, then 0dBFS does in fact represent the the maximum signal you can >>record<< before clipping.
  18. @DSatz thanks for the detailed reply. Update: I wasn’t crazy. I had someone at the local shop replicate the problem with me, and sent it back in for repair for a second time. It was highly unlikely that the 48v supplied was the problem as we used multiple preamps to test. This time the sennheiser technician heard the problem and repaired it. He wasn’t on his lunch break. I’m afraid I didn’t ask for their detailed repair procedure in the end... I just needed to get back to work! cheers adam
  19. Please, do the math properly. It's not hard. 32 bits / 8 equals 4 bytes. With a 8 channel recording, and sample rate of 48000 samples per second, your data rate is 8 hannels * 48000 samples per second * 4 bytes per sample = 1536000 bytes per second 1536000 / 1024 (to get from bytes to KiloBytes) = 1500 Divide again by 1024 to get from KiloBytes to MegaBytes, and you have 1.46 MB/sec Then you have to add a tiny bit for header / metadata info, but (unless you have a Cantar) that is just a few KB.
  20. That's true for when you are capturing audio. In the process of analog to digital conversion, 0dBFS is really the maximum signal level you can record without clipping. But once you have your signal in the digital domain and in 32bit float you can go above 0dBFS. This has to do with how 32bit float is encoding the data. You can try it yourself: Take any sound file in your DAW. Your DAW has to work in 32 bit float. Then apply a lot of digital gain to your audio, so that it goes above 0dBFS. Export this sound file as 24bit int and also as 32bit float. Import these two files again and use digtal gain to lower their amplitude. You will see that the audio in the 24bit file is clipped at 0dBFS and with the 32bit float file you can recover the audio above 0dBFS just fine.
  21. As John B. said, the DC converter circuitry in modern microphones is part of this. First-generation phantom powered microphones generally took the incoming 48 Volts and routed that through a high-value resistor to polarize the capsule. Those mikes generally had a single FET as their only active device, with an output transformer--miniaturized so as to fit into a 20 or 21 mm-diameter housing--that brought the output impedance down into the standard 150 or 200 Ohm range. Transformers that small, however, saturate rather easily, especially at low frequencies. They restrict the maximum output voltage and thus the maximum SPL of the microphone. More modern condenser microphones generally add an active output stage which is direct coupled, i.e. transformerless. That arrangement requires substantially more operating current, but also offers much better headroom, and greatly improves the ability to drive long cable runs. The DC converter that John mentioned improves the sensitivity of the microphone, since all other things being equal, the sensitivity is proportional to the capsule's polarization voltage, and those converters typically put out around 60 Volts. They're also almost a necessity in a modern 48-Volt microphone, since the increased current draw of the output stage causes a larger voltage drop across the 6.8 kOhm resistor pair in the phantom supply. Thus a microphone that draws 4 mA, for example (2 mA per resistor -> 13.6 V drop across 6.8 kOhms), actually receives a voltage in the low-to-middle 30s rather than 48. It would cause a major step backwards in sensitivity if such a low voltage were used to polarize the capsule. So: The original, analog version of the Neumann KMR 81 is one of the last remaining holdovers from their fet 80 series, which began with the KM 84 microphone in 1966. It features the older, simpler, lower-current, transformer-output type of circuit, with lower headroom (as a wild guess, maybe 6 to 10 dB lower) than it could have with more modern circuitry. It still does well for its age, though--it can put out about 900 mV (when lightly loaded) if it has to, for a maximum SPL of 128 dB (again, when lightly loaded). It's a nice-sounding microphone in my opinion. I don't know how well it does in high humidity, though; it's a traditional DC-polarized condenser, and for situations with any risk of moisture condensation, RF condenser microphones are generally considered more reliable. --best regards
  22. The KM 180 series uses a DC converter to boost the 48-Volt phantom supply voltage up to 60 Volts to polarize the capsule. The converter basically is a radio frequency oscillator that drives a tiny step-up transformer; its output is then rectified and smoothed. With proper powering and a properly functioning microphone, the frequency of that oscillator is well above the audio range. However, if the microphone is defective or the phantom powering isn't up to specification, the oscillator frequency can dip down into the audible range. You can hear this if you connect the mike to an outboard phantom supply, run it for a minute, then turn the powering off while you continue listening to the microphone's output. As the stored energy in the supply and the microphone ebbs away, the oscillator will drop in frequency, and you will hear it descend through the audible range until it dies out completely. Since Sennheiser didn't find any problem with the microphone itself, I suspect that it wasn't being powered correctly when you ran it. There have been two different versions of the KM 180-series circuitry; the changeover occurred around 2002. The original KM 180-series microphones required 2.3 mA; the later version requires 3.2 mA (in return for which they are 3 dB quieter, while keeping the same sensitivity and maximum SPL). I suggest that you try your microphone with a different preamp, mixer or recorder, or with a known good outboard 48-Volt phantom power supply. The original version of the phantom powering standard set 2 mA as the recommended limit, and a lot of older equipment (and even some that's newer) falls out of spec trying to deliver the 3 to 5 mA that microphones commonly require today, let alone the 7 to 8 mA required in some extreme cases. --best regards
  23. The KM 183 is a "diffuse-field equalized" omni, which means that it was designed for distant miking. When used at medium or close range, it will have a considerable on-axis elevation in its high-frequency response--6 to 8 dB. That evens out when the mike is used at a distance in a reverberant space, since when you get far away enough, the sound reaches the microphone at more or less random angles, while that high-frequency elevation only affects the front quadrant. On the plus side, this type of microphone would have lower sensitivity to wind and handling noise than any directional microphone. But due to the pickup pattern, you would have to get almost twice (1.7 times, technically) as close to your sound source in order to pick up the same direct/reflected mixture as even a cardioid would give you. --best regards
  24. The factory can update older CMC 6-- amplifiers, but it's not cheap; it involves an entire circuit board replacement, not merely installation of that shield plate in the connector well. In effect it's a trade-in. There's no sonic difference, nor any change in performance or powering or anything else. And clearly, not everyone absolutely needs the more recent version. Still, I'm glad to have it because I record mostly live classical concerts, and I can't exactly tap the conductor on the shoulder and ask for a retake. --best regards
  25. I just think the 32 bit thing is kind of pointless, seeing as people have been able to get good audio out of the zooms in the past and other brands, even on tape. I don't believe it will have a strong impact on our workflow. As of now it's just a selling point. However, it's fun to see that zoom are strong believers of "we'll fix it in post". Because if anything, 32 bit workflow and being able to not gain stage properly will only mean more work afterwards.
  26. SndGuy

    Nomad firmware

    Thanks for your responses. I emailed Howy and he sent me the file. Also, when my brain woke up, I went into Time Machine and found a copy in one of the older backup folders!
  1. Load more activity
  • Create New...