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About borjam

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    Bilbaina Jazz Club

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  1. Although limited, you can use two cardioids for the S channel. Place them perpendicularly to the axis of the M microphone, one pointing to the left and another one to the right. S will be the sum of the left microphone and the right microphone with the polarity reversed. Try to place the diaphragms as close as possible to minimize phasing issues though. Sound Devices added polarity reversal to the MixPre 3, so you can do this trick with it.
  2. Rycote offers a really excellent selection of spare lyres with different degrees of stiffness. Last month I ordered some to modify a Rode Blimp. I'll need to make a couple of small spacer adapters but it's certainly easy. I ordered the spares from Audiosense (Belgium) but I imagine any Rycote dealer can supply them. This is the reference pages for the Lyres and other upgrade parts, http://mymic.rycote.com/products/lyre/ And, for example, Audiosense lists them on their web page. They had a 3 week lead time though, in my case they didn't stock the parts. Confusingly, they list them as "Shockmounts and Suspensions" rather than "Spares" or "Upgrades". Still, really useful and I found the prices more than reasonable. https://www.audiosense.be/en/store/microphone-accessories.html?brand=rycote&category=shockmounts-suspensions&limit=96
  3. Oh well, the signal integrity requirements are certainly different. Digital signals tolerate certain amounts of noise that would make a transmission channel unusable for quality analog transmission. But of course at the end of the day signals are signals and common transmission line problems can ruin your day. Exhibit one, a silly example. Unterminated RS485 serial bus (note the overshots). It worked but before terminating resistors were installed it tended to fail sometimes. I guess when someone went near the unshielded twisted pair cable with a mobile phone. The second image shows a better terminated bus. Now, modern networks at high speeds enter the realm of microwaves, where circuit design needs to take Maxwell's equations into account rather than simple Kirchoff's Laws Try to assemble a digital circuit on a simple breadboard. It will work at 1 MHz. Now try 100 MHz I understand that he means "no analog gain stages" or even "no active components in the analog signal path".
  4. Hehe, a recorder with a single tape reel letting the tape loose (or, better yet, shooting it away at a speed of 15 ips) would fit in this collection (An arrangement of eraser head right after the recording head wouldn't be so obvious for a casual observer!)
  5. 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.
  6. Thank you, my information source was indeed the FCC internal photos. When looking at the ADC specs I was indeed a bit surprised! So that explains it
  7. Hope I wasn't too blunt! I was just thinking about beginners getting confused with the terms. That would make sense, modern A/D converters implementing better digital filtering. This one is used in MixPre-3: https://www.akm.com/akm/en/file/datasheet/AK5558VN.pdf And it has some nice features such as four different types of digital filters. I was puzzled when I read that they used 32 bit converters, maybe that's the reason. Also I noticed that it can use a neat trick (summation) to increase S/N ratio. Cirrus Logic explain it very well here. I had never heard of this, you can learn new tricks everyday https://statics.cirrus.com/pubs/appNote/AN331REV1.pdf
  8. Hmm. I think you are confusing bit rate with resolution. Bit rate is the product of resolution (bits/sample) by sampling frequency (sample/second). But both factors are not interchangeable. Increasing resolution won't affect the Nyquist frequency which is determined *only* by the sampling rate. It's true that perfect analog filters do not exist, so the behavior of the filter is not so good at frequencies close to the Nyquist frequency. So yes, oversampling is a solution to that. You can use a simpler analog filter with the cutoff frequency much higher than the intended maximum frequency of your application, and add a digital filter (which is cheaper and more effective to implement) to "clean" the bandwidth portion between your intended maximum frequency and the actual Nyquist frequency of the oversampled converter. Trying to make it clearer. Let's imagine an A/D converter with an intended band pass of 20 KHz (good old audio) and a "visible" sampling frequency of 48 KHz. We can make it the straightforward way, adding an analog filter in front of the converter. Let's say the cutoff frequency is 24 KHz. But it will affect the phase of a signal being sampled depending on its frequency. The closer to the limit, the worse. And of course it won't cut frequencies right above the cutoff frequency very well. Now, let's do 2x oversampling. We sample at 96 KHz, so we can put the filter on, say, 48 KHz. So in the digital domain we have a sampled signal with a bandwidth of 48 KHz. We still need to clean up it before down converting to 48 KHz sampling or aliasing will happen. The analog filter is always necessary. Aliasing is an irreversible phenomenon. Once a signal outside of the Nyquist limit has been sampled it becomes indistinguishable of a signal below it. That's the reason why ultrasound can create aliasing. But with oversampling you can relax the analog filter specifications a lot and rely on a cheaper to implement and more flexible digital filter. Reality is a bit complicated and the actual problem with aliasing is to pretend to sample frequencies *both* above and below the Nyquist limit, but this phenomenon mostly has applications in other fields. You can search for information about "Nyquist Zones" but unless you are into software defined radio or other similar applications it won't have much interest for you The first Nyquist zone would be from 0 to Fs (Fs = sampling frequency), the second zone would be Fs to 2Fs, etc. So, you can sample signals between Fs and 2Fs as long as your anti aliasing filter is a (Fs, 2Fs) band pass filter.
  9. But bear in mind that the MD441 is maybe the most expensive dynamic microphone currently available and the RE-20 is in second or third place in the ranking. The RE20 has a clever design that minimizes proximity effect despite being directional. It’s more popular in USA as far as I know and it was designed for radio stations. i have only tried it on a double bass and the result was pretty good.
  10. borjam


    It doesn't look like the Kashmir preamp at all. Remember that the Scorpio offers adjustable limiters while the MixPres have a fixed time constant and threshold level.
  11. At least in Spain the MD441 is the standard microphone at radio stations. I love it for live concerts, especially horns. It also saved my life with a violin once in a very difficult concert. A friend who had a voice over studio in Spain was a big fan of the CAD e300, which is not expensive. And speaking of "emulations", Austrian Audio, from the ashes of AKG, have announced a really curious beast: the OC 818. https://austrian.audio/produkt/oc818/ It's a dual diaphragm LDC which allows the user to record the output of both capsules, including a plugin to process them and adjust a custom polar pattern with programmable crossover points. The idea sounds interesting.
  12. Speaking of recommended mics, there is a Belgian seller on eBay offering AKG SE300B+CK93 at a very good price. I purchased a SE330B+CK93 combo and an additional CK93 and they are in perfect condition. Only complaint, the seller insisted on charging for the shipping of _each_ item.
  13. Now that I think of it, the main advantage of those 32 bits will be a better limiter implementation in software without degrading resolution.
  14. Yes and no As I intended to say, I can imagine an instrumentation microphone emulating the "personality" of another one intended for music. With limits of course (off axis response would be impossible!). But I am really sure my old SE2A microphones won't emulate a Schoeps! Don't get me wrong, *part* of a microphone characteristics come down to some eq, but there is a lot of reluctance against it and many multi microphone lockers could be almost equivalent to a lesser collection. In hindsight I think in the past I heard about audio plugins purportedly emulating different microphones.
  15. I think I heard of them some time ago. Interesting to somewhat copy the frequency response and distortion of popular microphone models. But what about the off-axis response which can be so important in an untreated room? Or, as they show the EV RE-20 as one of the emulated microphones, how to reproduce the lack of proximity effect? Will the plugin have a control to state the distance between speaker and microphone? I am really skeptical about this, I guess you can achieve similar results using some EQ in post. Last month I tried to explain a physicist friend what's the difference between the sound of different microphone models and why it's so important. I explained him that not only the on axis frequency response (which would be reasonably easy to replicate with EQ) matters, but also the off axis response and distortion. Of course it was trivial for him to understand frequency responses, noise floors and maximum SPL. But there are many sound characteristics that we are unable to represent just with a bunch of numbers. After all distortion is a non linear complex phenomenon and a mere percentage doesn't tell the whole story.
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