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5

For a live-sound design production of "King Lear" we constructed a rain machine that consisted of a long, narrow (5 foot by 7 inch) wooden box on a teeter-totter fulcrum. The box had dozens of nails sticking out of the bottom and we filled it with dried beans. As you rocked the seesaw back and forth you could get more or less continual rainfall and also ...


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If I would approach it, I would go and try the following: First of all, FM radio broadcasting has an upper limit of 15 KHz, and lower limit of 30 Hz. So cut all frequencies outside that scope to start with a very steep filter. Then a "little desktop / clock radio" would use a very small speaker which would not reproduce any sound lower than 200 Hz, so roll ...


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It is certainly possible to broadcast over any distance of your choosing, it's just a matter of what equipment you need and what licenses you will require. When conditions are right, it's even possible to bounce a signal around the world at the proper frequencies. For FM, you will need a transmitter with sufficient power, the necessary local clearances to ...


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I'm in full agreement with audionuma's answer, but just to add that a 3 grand mic preamp will not fix a 10 buck microphone. As you didn't mention in your question what mic & preamps you currently have, this is a bit of guesswork, but assuming it's a fairly basic setup, you can gain a reasonable amount of sound quality by using a USB pre-amp - they ...


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Actually, the mic pre amp you are mentioning is a pre amp plus a processing chain (a dynamics processor and some filters and eqs). You cannot emulate the pre amp itself in software as amplifying the mic signal is inherently done in the analog domain. The cost of a mic pre amp is related to the analog components included, their selection and association to ...


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Those are sampling artifacts. The sample rate is too low to cope with the frequencies involved so you get aliasing distortion, where the artificial elements of the digital signal (high frequency components) are aliased with the real signal giving you some very odd outputs. (From Nyquist Theorem, the sampling rate must be at least double the highest frequency ...


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That's a combination of effects. Firstly, the obvious ones: use an EQ or a bandpass filter to remove the low frequencies and a little off the high end and add a touch of distortion. There are some filters with presets that include "Telephone", that's a good place to start. Now the ones I think you're looking for: There seems to be some fast tremolo ...


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The manual for this device states it provides stereo line out (>250mV) on a pair of RCA (phono) connectors. If the microphone signal is routed to the line out you can record that with no need for a receiver. (I'd expect the microphone signal should be available on the line out but the "manual" is very short on details.) The microphones are VHF ...


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You are approaching this from the wrong direction. You are not going to be able to successfully reverse engineer this RF link. Could be any frequency, even 2.4GHz. Could be any modulation. You are not going to be able to find out without a significant amount of effort. Solution is to replace the RF link with your own. Give the talent a different radio mic, ...


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I just went and had a quick read up on sox (I've never heard of it before) and here're my thoughts based on the effects available to you: Start by using the bandpass filter to reject a lot of the sound outside 2KHz - 4KHz - This will imitate the small mic and speakers in the chain. Next use the compander to reduce the differences between the loud and soft ...


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I did this using a convolution effect. Basically, I passed a sweep tone through a narrow band FM radio (basically a CB), recorded it and used the result to generate an impulse response that represented the signal path. This is a blog post that describes the process. Feel free to adapt to your own purposes. https://www.greensideproductions.com.au/2013/01/04/...


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What you are proposing cannot be done. Your only option is to use something like the 'dialogue isolate' feature of Izotope RX to isolate the dialogue in each of the files, then compare one result with the other to pick the best outcome. It is not possible to 'extract a pure version' of the dialogue because although the dialogue is 'similar', there is no ...


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Although designed for music jams over internet, Jamulus is a handy solution for your purpose. It runs on Windows, MacOSX and Linux. From the website: The Jamulus software enables musicians to perform real-time jam sessions over the internet. There is one server running the Jamulus server software which collects the audio data from each Jamulus client, mixes ...


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The simplest way to achieve what you are looking for is to undertake the following workflow: identify a similar type of audio codec encode a stream of audio implement an algorithm to randomly corrupt the data-stream in varying levels of severity decode the corrupted data stream to audio listen.


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My approach (fast version) would be to use an amp-simulation-VST like Voxengo Boogex (which is a great free plugin) & to use the internal EQ settings (cutting the frequencies outside that well described scope above :) ). https://www.voxengo.com/product/boogex/ I also like it to turn up the "pre-gain" settings while keeping the post-gain down (...


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Get a set of NS-10's... But more to the point (to elaborate on the post above), One of the more deceiving issues in a recording studio is that they tend to have some of the best speakers in the world installed in some of the best listening spaces ever made. This is all great until you consider the fact that (especially today with the iPod etc) most music ...


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Performance will get you a long ways towards your goal. They need to project, make sure their headphone feeds aren't too loud. Are they supposed to be getting interviewed? If not keep the mic farther away in more of a boom position. Make sure you don't have any room reflections. Some of it depends on how far you want to go - believable vs "radio" and ...


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Most radio spots are done at 48K sampling rate, audio post in general really. Here in the US there aren't any broadcast level specs, most people just set their output limiters at -1 or 2. From there you just mix with your monitors at the correct level and make a nice sounding mix. The stations themselves handle the final compression before the signal is ...


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