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Unison - Detune Simultaneous voices singing approximately the same pitch, but slightly detuned from one another. Inspired by the sonic artifact produced when a choir sings a note in unison without being in perfect tune with each other, this characteristic can be replicated by duplicating a single audio recording and moving the pitch randomly up and down ...


6

Not all processors work the same way. Probably the most-imitated one is the Aphex Aural Exciter, which was the original. The quotes below are from http://www.soundonsound.com/sos/1995_articles/feb95/exciters.html The Aphex Aural Exciter (the first one): ...some of the input signal is diverted, via a side-chain and a high-pass filter, into a harmonics-...


4

In general, listen to whether it sounds good or whether it has something that irritates you. Or whether the sound is fitting or whether it's e.g. weak (in the modern days of dynamics squashing that can be a valid notion that you may want to adjust with compression. Just remember what the dialogue is for, i.e. it has to sound humanly like the person who's ...


3

Yes, audio software, be it a DAW or a live mixing console, do use multithreading and multicore functionality. Example 1 : Pyramix MassCore technology “hides” one or more cores in a multiple CPU computer and then creates a “pipe” directly between the software and those hidden cores to essentially create an Intel-powered DSP based system. Example 2 :...


3

To add to James' answer… I would personally never use a delay to fake the number of voices, I would always just keep adding tracks til I had enough. Secondly, I always pan all BVs away from centre [spread from 10% to 100% width], leaving a hole for the lead vox if there is one, or just a space in the centre if there isn't. So long as it's not too big a ...


3

You can use an Audio to MIDI plugin to generate notes and some MIDI expressions. You can record this as a MIDI file or route its output within your host audio editor to a synthesizer plugin in it. You can start with checking this Audio-to-MIDI plugins: WIDI Midifier (There are audio examples at the end of the page) Since these are plugins and you need ...


3

The point of over sampling is not about hearing frequencies above 20k. The point of over sampling is to make it easier for the filters to filter everything out above the nyquist frequency. When you sample at lower sampling rates such as 44.1 or 48 the quality of the filter matters more. So if you record with an amazing ad converter with a really expensive ...


2

In short, this is a very hard one to answer because even between all our peers, the answer to this is all broadly subjective (taste and nuance) within a narrowly defined objective (calibration and QC specs). I'd love to elaborate on this a little later, so I will be appending to this, but in a nutshell this is my thought/gut reaction, being a dialogue ...


2

That's an extremely hard question to answer, and would result in an answer thick as the bible. But the quintessence of it all is to keep it even, fluent, and remove everything that isn't needed for the scene or feeling, and replace sounds that sound bad or just wrong. Also to never ever use more processing than you actually need. The dialogue editing is more ...


2

Although many good answers have been given already, I would like to point you in the direction of this book. It's not exactly written to teach 'what a voice' should sound like, but goes in depth to teach about dialogue editing and workflow. Furthermore: Directing, recording and editing a voice is an art and I think one should have an obvious talent for it. ...


2

For pitch shifting down you need more than a 0 - ∼22.05kHz bandlimited signal to try to maintain some highs in the downpitched version. For digital recording you need more than 44.1kHz total sampling rate for 1. Making all gear-related conversion and input/output stage signal processing so that audible problems aren't induced to the audible range or the ...


2

I think they generate harmonics to add to the weight of a implied fundamental frequency. (Eg 200hz 400hz of a fundamental 100hz frequency). That said Maxxbass and Rbass supposedly work differently, though I haven't played around enough with it to figure out the specific differences.


2

Antares Kantos did this. However, it was released around 2002, and appears to be no longer available from Antares, so I'm not sure it is possible to obtain a legitimate copy these days. A brief search suggests that there are sites which claim that it is freeware now, and possibly offer it for download, but I cannot vouch for those sites. The general, brief ...


1

It will be lossless. You can avoid the popping sounds by doing very short fade ins/outs


1

The first half of your question states some clear facts. "Gameboy" and "32 4-bit samples played back in a loop". The rest of the question goes off in an unclear direction. Any bandpass filter you use will be determined by the frequency response of the speaker in the device and the sampling rate used by the DAC in the hardware device. To be honest, 32 ...


1

A simple, easy way of doing it is by using the amplify effect in audacity. Just select the audio and increase by a couple of dbs and listen to how it sounds (Remember to enable clipping). If you want it more distorted apply the effect again by different amounts until you get what you want. Really what you are looking for is distortion. Any digital ...


1

Note that audio synthesis algorithms can be inherently highly parallel. You have several buses that can be split into independent sets, such that no audio inputs of one class correlate (i.e. have data dependencies) with others. You render a single buffer and, when no delay is applied immediately, this buffer can be rendered in a embarassingly parallel manner,...


1

For a cheap way out you could just plug some ear-bud headphones into your mic-inputs. Headphones can work in reverse too! then place the earbuds in your ears but facing outwards and you'll get a bin-aural recording. This will work better with actual mics placed on either side of your head. You can simulate the binaural experience using reverbs and eq but ...


1

It sound like the good version has simply been cleaned up using equalization. The humming you speak of is just bad low frequency 'rumble'. It's a bad recording, but in the cleaner version somebody has removed some of the more troublesome frequencies, and maybe boosted some of the brightness. It would be pretty easy to tidy up the 'bad' version using the ...


1

You can add some compression to bring it forward and reverb for reflections - but changing the vocal characteristics of an audio requires recording another take or, alternatively, layering and using effects like vocoding to the original samples - but results might not be as expected


1

I don't know if this will satisfy the "dirty" part, but you could use a free (freeware/open source) digital audio editor to get it done quickly and cheaply. I'd suggest using Audacity. It's a popular, and in my experience, capable open source digital audio editor, which, with a little direction, you could quite easily utilize for all your needs. And because ...


1

A bit late to the party but your best bet would be to use the command line version. https://github.com/paulnasca/paulstretch_python


1

You would rather turn to a program like Matlab or its open source equivalent Octave. Open your mono-channel audio data as a vector (using wavread for instance), and turn each sample value to its square value. Then compute the mean every N samples, N depending on the smoothness/time-precision you need.


1

For me it sounds like ethnic vocals treated with some automated fx. I would try to get a great sustained vocal/speech phrase and chop it up to get some rhythm. Then i would use celemonys melodyne to pitch individual notes around. In melodyne you can change the formants of the vocal without editing the pitch to get the "kittiness" into the vocal. Also be ...


1

Magic sounds is always that we can't hear in real life. For your event try to mix this sounds: 1) Good warm and bright chimes. Try to find them in some orchestral vst instruments. Render a few variations, take 2 of them and pan hard left and right (left -100% right - 100%). 2) Sizzle of fire torch (\Boom Libraries\Cinematic Trailers - Construction Kit\...


1

Hey there. The goal should be to analyze what you hear and determine if it sounds natural. What does that mean? Well, if you hear hi freq or low freq that generally are not there in real life, that is because the mic or actor have those freqs in recording and you need to adjust it to balance it back to the real world as much as you can. Usually involes eqing ...


1

http://www.boomlibrary.com/boomlibrary/products/outdoor-impulse-responses This could also help for Exterior scenes when ADR is recorded in a studio.


1

There's a balance between proper editing of the dialog tracks and proper use of the processing that Jay mentioned, but also the effects of masking when you introduce atmospheres, walla, foley, roomtones, and reverb/early reflections. For instance, even without EQ or compression, the acoustical variations, small amounts of background noise, level differences, ...


1

Another thing to keep in mind when it comes to post audio, especially on the mix stage, is I need everything automated. I usually don't do a record pass to my stems until after we have completed the reel and not all hardware gear is easy to automate. I use pretty much all software for mixing and that keeps my automation contained and also easy to recall. ...


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