What do you want to know? --- From the other side

[Veracohr]

This sub-forum is too slow. Surely there are artists and/or those interested in production who maybe don't feel inclined to start a new thread, but still have things they would like to know as far as production.

 

This is your opportunity to ask a question. Maybe it's a simple question that only needs a single-post answer so you haven't started a new thread. Maybe it's really in-depth and we'll break it out into a new thread.

 

Even if you're not an artist and don't plan to be, but want to know how someone made a particular sound, ask here!

[Rotwang]

I have a question. Lately I've been attempting to make music using software I wrote. Each channel of a sound is stored as a list of floats representing a waveform, and on demand it will convert such a waveform into a 16-bit 44.1kHz WAV file. Often a waveform will go outside the bounds of what can be stored in 16 bits, i.e. when the wave's displacement (after being converted to an integer) is more than 32767 or less than -32767; in this case the software just truncates the wave (so e.g. a displacement of -40000 is changed to -32767). Obviously this creates distortion, but if a sound is at the right volume in the first place then it doesn't happen much and is barely noticeable.

 

Here's the problem: when creating a whole track out of individual sounds, the waveforms for the different sounds are just added together. Constructive interference means that the resulting waveform needs to be truncated a lot more than the individual sounds need to be. When listening to a track through monitoring headphones it sounds terrible. I can fix this easily enough by just reducing the volume of the whole track, and then it sounds fine - but much quieter than the music on most CD's. So it seems that I either have to have music that's too quiet, or music with too much distortion.

 

Anyone know how to get around this?

[Rotwang]

Also, can anyone tell me how to make

and/or

?

[Guest E-Mantra]

I have a question. Lately I've been attempting to make music using software I wrote. Each channel of a sound is stored as a list of floats representing a waveform, and on demand it will convert such a waveform into a 16-bit 44.1kHz WAV file. Often a waveform will go outside the bounds of what can be stored in 16 bits, i.e. when the wave's displacement (after being converted to an integer) is more than 32767 or less than -32767; in this case the software just truncates the wave (so e.g. a displacement of -40000 is changed to -32767). Obviously this creates distortion, but if a sound is at the right volume in the first place then it doesn't happen much and is barely noticeable.

 

Here's the problem: when creating a whole track out of individual sounds, the waveforms for the different sounds are just added together. Constructive interference means that the resulting waveform needs to be truncated a lot more than the individual sounds need to be. When listening to a track through monitoring headphones it sounds terrible. I can fix this easily enough by just reducing the volume of the whole track, and then it sounds fine - but much quieter than the music on most CD's. So it seems that I either have to have music that's too quiet, or music with too much distortion.

 

Anyone know how to get around this?

 

i dont quite understood everything you wanted to say ..but if you have a lot of tracks (with waves samples or whatever) you should check them all before rendering ..try to put all the samples into an editor like sound forge for example and check them all for high peaks before rendering all tracks as one..and then do the final mixing

regarding the songs..first one seems to me a formant sound ..

[Guest E-Mantra]

the second sound from the link can be easily created with purity vsti for example

[Rotwang]

the second sound from the link can be easily created with purity vsti for example

 

I'll take your word for it. But what I mean is, how does purity vsti (or whatever) create that sound? What kind of waveform or spectrum does it have, and what kind of effects have been applied to it? I'd like to be able to recreate something similar with my own software.

[Guest E-Mantra]

basically most of vsti's can create whatever sounds u want ..i use some delays and reverb on leads for example..and also eq..

[Veracohr]

I have a question. Lately I've been attempting to make music using software I wrote. Each channel of a sound is stored as a list of floats representing a waveform, and on demand it will convert such a waveform into a 16-bit 44.1kHz WAV file. Often a waveform will go outside the bounds of what can be stored in 16 bits, i.e. when the wave's displacement (after being converted to an integer) is more than 32767 or less than -32767; in this case the software just truncates the wave (so e.g. a displacement of -40000 is changed to -32767). Obviously this creates distortion, but if a sound is at the right volume in the first place then it doesn't happen much and is barely noticeable.

 

Here's the problem: when creating a whole track out of individual sounds, the waveforms for the different sounds are just added together. Constructive interference means that the resulting waveform needs to be truncated a lot more than the individual sounds need to be. When listening to a track through monitoring headphones it sounds terrible. I can fix this easily enough by just reducing the volume of the whole track, and then it sounds fine - but much quieter than the music on most CD's. So it seems that I either have to have music that's too quiet, or music with too much distortion.

 

Anyone know how to get around this?

 

Great, there goes the illusion I could give an answer or two.

 

Unfortunately I don't really know much about the inner computations of digital audio, or how your software differs from any other, but I'll give it a shot. I may not entirely know what I'm talking about however.

 

First, what size are your floats?

 

Second, this is what springs to my mind first: "rule of thumb" advice among DAW users is generally to record audio somewhere around -18 to -12 dBFS, because if you record everything as close to 0 as possible you end up having to reduce the level of each track so the mix bus doesn't clip. Relating this to your situation, if you're talking about signals that are very high level, it makes sense that mixing a bunch together will cause problems. This happens in analog audio as well, that's why mixers have faders to reduce the level. In an analog mixer, even though you can reduce the level of the mix output, if that mix bus is being fed from too many signals at too high a level, the bus circuit will still be overdriven even if the output circuit isn't. Hopefully I'm not pulling this out of my ass, but perhaps the concept is kind of the same in digital processing.

 

Also, be aware that all DAW's employ dither algorithms when converting their internal signal format (usually 32-bit float) to an output format of 16 or 24 bits, to reduce distortion. If your software doesn't do this, look into it.

 

The issue of the track being quieter than CD's sounds in the realm of dynamic range. That's where compressors and limiters are your friends.

[Agneton]

Anyone else experiencing troubles with Rubberfilter resetting its parameters in FL Studio each time I click somewhere randomly on my PC screen?

 

U never know somebody might have the answer : p

[Veracohr]

Also, can anyone tell me how to make

and/or

?

 

First one sounds like a single oscillator with formant filter. Possibly oscillator sync but I don't think so. Second one can be accomplished with two oscillators, one detuned a fair amount, short attack, short decay, mid-level sustain, long release.

[Rotwang]

First, what size are your floats?

They can be any size. But I've set it up such that a sine wave with an amplitude of 2.0 is the loudest that can be played with a 16-bit WAV (i.e. a sine wave of floats with amplitude 1.0 will be converted to a sine wave of ints with amplitude 16384 before being turned into a WAV).

 

e: Oh, did you mean how many bits per float? I dunno, whatever Python's default is. [checks] It looks like it varies from implementation to implementation but probably 64 bits. In case you can read computerese, Python says this:

 

sys.floatinfo(max=1.7976931348623157e+308, max_exp=1024, max_10_exp=308, min=2.2250738585072014e-308, min_exp=-1021, min_10_exp=-307, dig=15, mant_dig=53, epsilon=2.2204460492503131e-16, radix=2, rounds=1)

Second, this is what springs to my mind first: "rule of thumb" advice among DAW users is generally to record audio somewhere around -18 to -12 dBFS,

OK. I set 2.0 to be the maximum amplitude because I figured that would make 1.0 a sensible maximum amplitude (this corresponds to -6 dBFS). So maybe I should try reducing the volume of all my tracks by a factor of 2 or so.

 

Also, be aware that all DAW's employ dither algorithms when converting their internal signal format (usually 32-bit float) to an output format of 16 or 24 bits, to reduce distortion. If your software doesn't do this, look into it.

Thanks, I will.

 

The issue of the track being quieter than CD's sounds in the realm of dynamic range. That's where compressors and limiters are your friends.

 

I'll need to Google those.

 

First one sounds like a single oscillator with formant filter. Possibly oscillator sync but I don't think so. Second one can be accomplished with two oscillators, one detuned a fair amount, short attack, short decay, mid-level sustain, long release.

 

Thanks very much (I'll need to Google "formant filter" too). Can you give a quantitative estimate on how much "a fair amount" might be? I already tried adding together some sine waves at frequencies slightly different from harmonics of C but it sounded like shit.

[Veracohr]

e: Oh, did you mean how many bits per float? I dunno, whatever Python's default is. [checks] It looks like it varies from implementation to implementation but probably 64 bits.

Yes, I meant how many bits. Not that I know what to do with that info, I'm not a programmer.

 

So maybe I should try reducing the volume of all my tracks by a factor of 2 or so.

That's basically my answer in a nutshell. I don't know where your expertise lies as far as programming, but maybe you should look up some info specifically on DSP and/or audio DSP.

 

Compressors and limiters in a software environment are typically plugins in formats like VST or Audio Units (Mac). They control the dynamic range of the signal through automatic gain control. I'm sure you could code one yourself, but a lot of history and knowledge goes into what makes a compressor sound good musically.

 

Thanks very much (I'll need to Google "formant filter" too). Can you give a quantitative estimate on how much "a fair amount" might be? I already tried adding together some sine waves at frequencies slightly different from harmonics of C but it sounded like shit.

 

A formant filter can be like a set of parallel bandpass filters, set such that it mimics the resonances of the human voice.

 

I should have also mentioned on the second sound that it's probably two square wave oscillators. Two detuned sines would indeed sound like shit. Most synths have small tuning increments in cents, so the quantitative answer I can give you is 10-15 cents.

[Rotwang]

I don't know where your expertise lies as far as programming, but maybe you should look up some info specifically on DSP and/or audio DSP.

When I started out I made a conscious decision to learn nothing about how audio software usually works since I figured it would be more fun to try to figure it out by myself. But I guess I'll look into that at some point.

 

I should have also mentioned on the second sound that it's probably two square wave oscillators.

 

That works like a charm, thanks. Here is the result of me trying to reproduce the melody from Eggshell, if you're interested.

[Veracohr]

When I started out I made a conscious decision to learn nothing about how audio software usually works since I figured it would be more fun to try to figure it out by myself.

 

That's cool, but I bet there could be something useful to learn regarding general DSP that wouldn't necessarily steer you toward rehashing audio software. I'm not really sure, I'm kind of overreaching with my answers here.

 

Nice job on the sound. You could also try modifying the pulse width to change the sound a little, but if you're doing it additively with sine waves I suppose you might have to work out the amplitudes yourself.

[Rotwang]

Nice job on the sound.

Thanks.

 

You could also try modifying the pulse width to change the sound a little,

I don't know what that means - are you talking about this?

 

but if you're doing it additively with sine waves I suppose you might have to work out the amplitudes yourself.

 

I don't have to do it additively with sine waves; I can tell the program to output whatever kind of function I like. The sounds in the WAV I posted above were made by directly adding together two square waves and multiplying the result by an envelope - I tell the program to do that typing this:

 

(square(880, 4, 0.25) + square(880*1.01, 4, 0.25))*(fade(0.01, 0, 1.3, 0.5)**fade(3.99, 1, 0, 4))

[Panoptes]

This sub-forum is too slow. Surely there are artists and/or those interested in production who maybe don't feel inclined to start a new thread, but still have things they would like to know as far as production.

 

This is your opportunity to ask a question. Maybe it's a simple question that only needs a single-post answer so you haven't started a new thread. Maybe it's really in-depth and we'll break it out into a new thread.

 

Even if you're not an artist and don't plan to be, but want to know how someone made a particular sound, ask here!

 

Its great that you are offering such knowledge. I have several questions which I would honestly prefer to discuss this in person with someone who knows what they are doing haha (Pardon me asking Veracohr/ others offering advice here ;I'm new here) but what experience do you have? I'm not familiar with what musical projects you are affiliated with.

[Veracohr]

I don't know what that means - are you talking about this?

 

I don't have to do it additively with sine waves; I can tell the program to output whatever kind of function I like.

Yes I'm talking about pulse width modulation, but not necessarily continuously modulated. Many standard subtractive synths with a pulse/square waveform allow setting the pulse width manually as well as continuously modulating it with a LFO. A true square wave has sort of a 'hollow' sound, but if you make the pulse width something other than 50%, it contributes some more harmonics. Just more options.

 

Its great that you are offering such knowledge. I have several questions which I would honestly prefer to discuss this in person with someone who knows what they are doing haha (Pardon me asking Veracohr/ others offering advice here ;I'm new here) but what experience do you have? I'm not familiar with what musical projects you are affiliated with.

 

I don't have any particularly extensive or special experience, I just wanted to stimulate some conversation in this subforum. I'm only affiliated with my own craptastic music, but I did go to school to learn audio production. I also pick up info here and there relating to audio that I may not have personal experience with, but if I see something repeated enough I can regurgitate it and sort of sound like I know what I'm talking about.

 

I hope I didn't come across as "Here I am, I have all your answers!" I definitely don't. I was kind of drunk when I started the thread and it sounded like a really good idea then. But I'd be happy to try to answer your question(s) if you're OK with me possibly not having answers.

[Panoptes]

There are few people who would have all the answers and you didn't come off as cocky or anything I was just curious about your experience . I too am going to school for production but i'm still in my first year. Also my production classes revolve more around studio production with more "acoustic" music rather than the psychedelic madness that we listen too haha. I'm currently swamped with work but once i'm done i'll post some of my questions.

[Veracohr]

Also my production classes revolve more around studio production with more "acoustic" music rather than the psychedelic madness that we listen too haha.

So was mine, but if the school is decent and you're paying attention, you should be learning production concepts that can be applicable across many areas.

 

I've also had 15 years to add to my school audio learning.

[Panoptes]

 

So was mine, but if the school is decent and you're paying attention, you should be learning production concepts that can be applicable across many areas.

 

I've also had 15 years to add to my school audio learning.

 

This is true. but my questions revolve around the more in-depth aspects of producing wild psy music. I feel like some of these things you can only learn for the artists themselves. You know what I'm getting at??

[Veracohr]

So you're talking more about composition than production? To me production is technical; composition is artistic.

[Panoptes]

So you're talking more about composition than production? To me production is technical; composition is artistic.

 

Well I consider crafting certain sounds both technical and artistic. When I explore a track I listen for both artistic and technical choices made by the artist which combine to present a certain effect (an effect which one does not simply stumble upon). I would like to explore some track in detail, which would include both the composition and production behind the sound itself.

[pixma]

Which wavesforms and which vst is commonly used in darkpsy? The one thing I cant seem to get right is a phat sounding bassline

[ermize]

Which wavesforms and which vst is commonly used in darkpsy? The one thing I cant seem to get right is a phat sounding bassline

 

...there are no rules/waveforms/vsts to get these sounds... anyway:

 

- I guess saw and pulse are probably the most commons waveforms due to their richness in harmonics

 

- for synth layers in darkpsy, the FM synthesis is your friend and this funny guy could give you some interesting advice about that:

http://www.youtube.com/user/AndysPsyTutorials?feature=watch

 

- My preferred vst is Zebra and I think you can get a lot of interesting-weird-amazing sounds from it.

 

- I can't help you with bassline but a good way to start is Alien303. Try almost closing the filter and then playing with the envelope of the filter to get your sound and sure...eq and compress sidechaining with the kick... put a note that overlaps with the kick if you want more 'vibes' (don't how call it...)

 

hope it helps..

[Veracohr]

Yeah, saw and pulse are the most common waveforms for bass sounds. I'd suggest trying saw first. The rest is all in the filter settings and EQ. I'm not sure what 'phat' means to you, and I'm not familiar enough with darkpsy to guess what sound you're trying for.