7

As heard within amplifiers, the frequency can be 100/120Hz (2nd harmonic of 50 or 60Hz) because of the rectification process when converting AC to DC to power said amplifiers. There can also be a component that is the third harmonic of 50/60Hz as well and this is magnetically "coupled" from the AC transformer inside an amplifier. There are all sorts of ...


5

I believe you are looking for REAPER It has an API and its own plugin language you can use. Above and beyond that it is one of the most customizable DAWs out there, and affordable to boot.


4

There are definitely tendencies - and these mainly appear through the use of the same types of instruments in a genre, some more explicit than others, e.g. Drum and bass, Funk and A capella. Almost all modern electronic dance music uses a steady repeating kick drum and bass pattern - and as you note, this defines a good portion of the frequency ...


3

You need to understand sound sources (oscillators, noise generators, and samples), modulation (LFO in particular), sound shaping (envelopes), and then signal processing (read: math and audio physics). And honestly I'm being really high level here. There are a million details within each thing I mentioned. Its not really difficult, but I think it does ...


3

It's a 50 or 60hz sine wave (depending on power frequency). You aren't going to get more accurate than that. The harmonics are going to depend on the particular power generation and will vary based on uncontrolled variations in both the grid and how the hum is actually being introduced in to the signal.


3

The fastest way to solve this problem is simply sample a mains hum / ground loop and work with that. I would start by recording 5 minutes of ground hum. Then find a section of about a minute long that loops nicely (make sure that the start and end points are aligned to either both a rising or falling zero-crossing). Then progressively shorten the sample ...


2

Apart from regular tools (DAW, audio editor etc...), it is a common thing for a sound designer-composer to learn/use programming tools (like Unity or FMOD) to be able to implement-test audio myself during the development? It's not common, because game (and sound) programming requires a certain level of experience in programming to make the ...


2

Most people are sound or audio designers. There aren't that many of us who are audio coders; I think we're a small community. Audio programming roles aren't normally done by a specialist - normally they do other jobs like AI or Physics or Graphics... I spent about a decade working for a company that specialised in sound cards before that market faded. We ...


2

A couple of years ago, one of my students created a Max/MSP patch that would read a Twitter feed and would give a live sonification, via an additive/granular synthesis engine. This is along similar lines to what you are talking about. You can find further details on his blog at: http://samharman.com/2011/03/twinthesis-twitter-powered-synthesis/ I think ...


2

Signed and Unsigned refer to how numbers are stored for the samples. A sample corresponds to voltage above or below 0. Signed data is a format that can store negative numbers, so 0 is no voltage and positive or negative numbers can be used to describe the direction of the voltage. In unsigned data, all numbers are positive and the middle of the range ...


2

To answer your first question, yes. The resource cost of an unladen Bus is negligible. The main practical limitations lie with running too many CPU-hungry ugens at once. You can run 50 Buses easily, but not 50 reverbs. If you're talking about large numbers of audio inputs or outputs, SuperCollider can handle that easily. If you want to deal with ...


2

You randomly pick 9 of your 35 speakers you want to be able to "randomly" play something different. Don't tell your test subject which 9 you have chosen.You then connect the remaining 26 speakers to the 10th output of your focusrite interface. Bobs your uncle.


1

Suggestion. Two channels of sounds, with two amplifiers. A switch using relays, the speaker either plays left or right channel. You need this: stereo output, stereo amplifier, 35 relays (single pole, double throw, one position to left sound, other to right sound), a digital output card controlling which relay to energize.


1

Reaktor macros are implemented graphically and not programmatically as in via text. You would need to identify some sort of macro block that allows the text transform you are suggesting then route the result to a parameter pin.


1

You basically need a software with numerical processing of MIDI input. It's like the next thing to writing a driver, but some software offers very direct manipulation of MIDI input: MIDIOX, Pd, Logic, Max (mentioned in the comments). You could achieve it with Ableton Live as well, although you would have to jump through some hoops.


1

While Cthulhu is great for comming up with new and creative patterns, you could very well just us the in-build chord tracks in Cubase (or the similar chorder in Studio One) for simple harmonies and circle of fifths stuff. It's acctualy a time saver when working with fixed harmonies music - like those in folk or indie genres. Tutorial video on this chord ...


1

Max has Gen which allows writing code for more low-level DSP operations. I'm also going to push for SuperCollider, because it exactly fits your descriptions -- in particular your disdain for visual programming and interest in writing audio algorithms interactively. There's no way to make a VST plugin with it, however, so in your case it could just serve as ...


1

The best skill for audio programmers would be - strong coding skills (i guess C++ syntax) and good knowledge of middleware such as Wwise and FMod. Knowledge of audio editors (SoundForge, Audio Audition, Audacity etc.) and DAWs (ProTools, Logic, Cubase etc.) is most of the a time a well recieved plus Modern audio technique can also be useful like - using ...


1

I took @maxime.bochon 's advice and posted to signal processing and got the answer there, so I'm posting it here as well. Full question and answer can be found here : https://dsp.stackexchange.com/questions/25830/how-can-i-accurately-represent-an-mp3s-wave-form-in-respects-to-time It turns out that drawStep had to be a double instead of a float. The ...


1

Whislt I agree it's more a programming question than sound based I'll answer it from a Music Technician point of view. Short Answer Not easily. Slightly Longer Answer: Through hangouts itself you can't do it. To get access to all the feeds you'd either need to be at Google itself monitoring every source, then use a panning tool in a DAW or environment like ...


1

if( Mix_PlayChannel( 1, test1, 0 ) == -1 ) // Tests for error return value -1 { return 1; // Returns 1 to something? } if( Mix_PlayChannel( 2, test2, 0 ) == -1 ) // Tests for error return value -1 { return 1; // Returns 1 to something? } Without seeing what your function is the above code seems like it would not be playing sound at ...


1

Here is an online series of tuts http://interactiveaudio.wikiaudio.org


1

Peter Batchelor's got some good tutorials that got me started in the music arena. http://www.peterbatchelor.com/maxTuts.html


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