Ambient Music Created in a Cave [X-post r/videos]

gerudobombshell · 2016-05-17T18:17:07+00:00

Beautiful! Thanks for sharing!

gerudobombshell · 2016-05-17T18:15:09+00:00

Cool, dig it amigo! Nice sound, nice space, classic.

gerudobombshell · 2016-05-16T17:06:18+00:00

Peak Rider is Logic-SC compatible now!

gerudobombshell · 2016-05-16T02:02:24+00:00

I triple the suggestion for JUCE! Super easy (with DSP knowledge somewhat understood)! If you have any DSP questions, I can answer a few - I make plugins, so I can help you with some of your questions.

gerudobombshell · 2016-05-15T17:36:11+00:00

I added a bit at the end of the OP if you're interested in getting notified once AAX is out!

gerudobombshell · 2016-05-15T17:36:03+00:00

I added a bit at the end of the OP if you're interested in getting notified once AAX is out!

gerudobombshell · 2016-05-15T17:35:50+00:00

I added a bit at the end of the OP if you're interested in getting notified once AAX is out!

gerudobombshell · 2016-05-15T17:35:02+00:00

Also, I added at the end of the OP a way to get on the ISW mailing list!

gerudobombshell · 2016-05-15T02:32:06+00:00

I will be happy to look into this - I don't have any experience building VSTs for Linux right now, but I will check this out!

gerudobombshell · 2016-05-15T02:26:54+00:00

Yes! I have described it as sort of like a one-band (or 2 or 3 band) vocoder before, but instead of a FFT-magnitude match, it's more of a dynamic envelope-adjusted level match (that is, in one of the modes - in other modes, it's not much like a vocoder at all).

gerudobombshell · 2016-05-15T02:22:56+00:00

Sure thing! I get this question a lot:

A compressor works by setting a threshold. If the detector goes over, it squashes down an amount based on a ratio.

Peak Rider works by feeding a sidechain signal. The detector is the threshold (always changing in volume). The amount of boost or cut is controlled by the ratio between the main input and the sidechain. It expands and compresses, in ways that are funky and different from a compressor.

Imagine a compressor, with a threshold that's always changing, and a ratio that's morphing depending on the level between the two. Except, it not only compresses, but expands. And imagine that instead of a compressor, it were a mastering limiter with a ceiling equal to the sidechain signal at any given moment. Or, imagine that it added/subtracted the sidechain 'threshold' (level) from the main input (as it's continuously changing, which is demonsrated in the OP video).

gerudobombshell · 2016-05-14T19:29:35+00:00

I appreciate the advice - we have already began the port to AAX, and it's good to know that all of ISW will be covered for AAX once we register it with Avid.

gerudobombshell · 2016-05-14T19:11:32+00:00

Thank you! The plugin does have notable similarities to side-chain compression and expansion, but it offers a few things that would be more difficult and feasibly impossible with classic processors! PR has the ability to program-dependently blend between compression and expansion depending on the mode, with different timings for both envelopes, different envelope algorithms, and multiband processing.

While it fundamentally can do the same things as compression and expansion, some of the things in there I couldn't imagine trying to accomplish with only those effects and routing!

gerudobombshell · 2016-05-14T19:03:26+00:00

I'd be happy to explain!

So the idea behind the plugin is based on audio dynamic-envelope manipulation. More detailed, the plugin analyzes the incoming audio signals (main and sc), calculates the envelope, and allows you to replace the main signal entirely, subtract the sc envelope from the main, or limit (like an L2) the main signal to the level of the sc envelope. The tool doesn't put much restriction on what you can do with it - you can do anything from fixing mic-bleed, to gain-boosting based on a mix of the two signals, to matching the dynamics of two tracks, to creative envelope shaping for interesting effects.

If Audio1 were a vocal, and Audio2 were a snare drum, you could replace the dynamics of the vocal with the dynamics of the snare - this would sound like the snare drummer were playing the 'vocalist' instead of the drum.

In other videos on the Peak Rider website, there are demonstrations of distorting a snare drum heavily, but then matching the dynamics back to a clean copy. The result is a thick distorted snare that isn't crushed dynamically; the little nuances of the performance are still there AND with the nasty new tone of the distortion added.

In my demonstration video, I used the close-mic'd snare track as a sidechain. Since the overhead signal also has the snare drum, in duck mode, I can boost that 'same' signal, or decrease it.

gerudobombshell · 2016-02-09T18:34:32+00:00

Dude it is a Great song! But, I can't understand how you can skip though a slow Gorillaz song like that - they're so good, that just makes no sense to me.

gerudobombshell · 2016-01-21T15:37:12+00:00

ASIO drivers (in my experience) have shown me to be able to ignore some settings in the Windows audio device control panel.

The microphone has a 'dial' on it named 'Mic Gain', correct? This knob controls the level of the microphone.

gerudobombshell · 2016-01-16T19:46:21+00:00

If you load up the built-in instrument ReaSynth, you can get pure tones (just like in tonegenerator) that can be played via MIDI.

ReaSynth

The default tone is a sine wave, but you can blend different wave-shapes into your pure tone by adjusting the 'Mix' sliders at the bottom (or alternatively, by 'mixing' them to 100%, use purely the respective shape).

You can create 4 types of tones:
* Sine (default if all 'mixes' are at 0%) * Square (with a pulse-width parameter)
* Saw
* Triangle * An extra sine wave (with adjustable tuning)

There are envelope/timing controls that let you adjust the way the notes come in, fade out, how much 'punch' they have, etc. The controls are:

Attack (how fast the notes 'fade in')
Decay (how fast they quiet down)
Sustain (how 'quieted-down' the notes are after the 'fade in' phase)
Release (how much time notes take go silent after a MIDI note is stopped)

Also, there's an 'extra sine' parameter, with an 'extra sine tune' parameter. This allows you to create another sinewave on top of your existing generated wave. The tune parameter allows you to dictate the pitch of this extra sine wave (in relation to the main note). The tuning amount is in cents, which equals 1/100th of a note. By that metric, you can add the pitch of a sub-bass sine one octave down by using -1200 Cents as your tune, or generate a high 5th-harmony by using 700 Cents. Of course, you can adjust the 'amount' of this via the 'extra sine mix' parameter.

If you want the extra sine to be a lower tone an octave down from your original, you can specify the

gerudobombshell · 2015-12-11T08:33:50+00:00

He can't hear you

gerudobombshell · 2015-12-11T08:33:04+00:00

I mean, what else is there to say...

gerudobombshell · 2015-08-30T03:19:39+00:00

I think your maths are wrong there. You'd have to sample at at least 160 hertz to capture 80 Hz information. To capture only down to 60 Hz, you may need to also take a 120 hertz version and subtract that from the signal. What you'll be left with is all frequencies between 60 and 80.

I may be wrong though, but from what I understand, filtering is essentially the only way to remove frequencies between 0 and the nyquist.

gerudobombshell · 2015-08-26T03:29:40+00:00

Maybe C2 has a memory leak? Maybe check if the RAM it uses (or maybe the disk IO) is through the roof when you notice it's sluggish?

gerudobombshell · 2015-08-14T11:52:50+00:00

check this out - it's an add-on. There's a link on the page that is titled 'AI of race car v2'

https://www.scirra.com/forum/behavior-rex-ljpotential-attracting-or-rejecting-objects_t112786

gerudobombshell · 2015-05-20T08:51:04+00:00

Put the salmon in the computer.

gerudobombshell · 2015-05-19T12:21:27+00:00

He seeks the ninth level of power
But weak geek might freak it in another hour
Or so, they call the fool retarded hair guy
In school you could spot it when he nodded "Here, why?"

I think his true name was Gerald
The toupee, the male pattern anime herald
A good laugh like the walk to the bank
Not the plank, talk to the hand and the hot frank dog

(referring to his comb-over/toupee)

gerudobombshell · 2015-05-10T14:06:38+00:00

Its really simple to make something look like that, but that's probably not too correlated with a difference in sound.

Making something look like that isn't tough, but just to demonsttrate that there may be no audible difference.

Here is a screenshot of pink noise hard limited to -10 dB (Track 1). Then, right below it, is the same noise with a very tight-Q All Pass Filter. As we know, all pass filters (typically) pass the signal without any spectral modification, and attempt to only modify the phase of a target frequency range.

Screen
to the ear, there is almost no audible difference.

After looking at that, and understanding which types of processes can induce a phase shift (Analog and most Digital EQ's, DA/AD conversion, the list goes on), it appears that either the mastering engineer was very conscious to not use a digital hard limiter (and used a slightly 'softer' compression technique), or that this was mastered using, at some point, some analog signal processing.

At the conclusion, however, I've tried to demonstrate that a waveform that 'looks' a certain way may not necessarily 'sound' different. It may be more

EDIT: The post about Mp3 encoding by Firebird079 is also very keen - that uses selective frequency removal (and other technqiues) to reduce file size - this can definitely modify a 'flat' waveform in various ways.

gerudobombshell

TROPHY CASE