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-- Waveforms Yay!

Waveforms Yay!

So, This is just a description of what I noticed and maby someone could shed some light on it.

So I was messing with 3osc while staring at an oscilloscope. And I noticed something intresting, for most keys the thing goes wild, no matter the octave [well, not for really low octaves but we will get to that a bit later]

My first wonder is.. Ive noticed is that the key G, in any octave seems to be the most "stable" tone. The oscilloscope just shows a static wav [not squiggly and jumping around.] Why is that? Its also the most soothing to listen to [i'm using a 3osc with one voice actiavted, its a sine]

Another thing, When playing tones at below C0, there is no audible sound, but the meter goes WAY into the like..+90db range or so. Why is it so loud?

I think these are properties of 3osc. Try with another synth set to sine wave.

Re: Waveforms Yay!

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Originally posted by cronodevir So, This is just a description of what I noticed and maby someone could shed some light on it. So I was messing with 3osc while staring at an oscilloscope. And I noticed something intresting, for most keys the thing goes wild, no matter the octave [well, not for really low octaves but we will get to that a bit later] My first wonder is.. Ive noticed is that the key G, in any octave seems to be the most "stable" tone. The oscilloscope just shows a static wav [not squiggly and jumping around.] Why is that? Its also the most soothing to listen to [i'm using a 3osc with one voice actiavted, its a sine] Another thing, When playing tones at below C0, there is no audible sound, but the meter goes WAY into the like..+90db range or so. Why is it so loud?

This is what I speculate, it is because the lower the note, the slower the oscillation and during that slow oscillation, higher and lower peak in amplitude achieved by the waveform, which creates more raw volume. When the notes are in a higher octave, it is oscillating at a faster frequency at a shorter period roundtrip from midpoint to crest to trough, and repeats again.

Re: Re: Waveforms Yay!

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Originally posted by Beatflux I think the "stableness" of the note G comes from the properties of the synth, rather than the note. Although...the waveform could have more instances of consonant overtones due to whatever reason.

No it just depends of the settings of the oscilloscope. the G note appears stable because the period of time in which is measured by the oscilloscope equals (or is close to at least) to the period of the G note (period of a frequency = 1/frequency in Hz).

So let's say you have a frequency note of 440 Hz which corresponds to A3:

- set your oscilloscope to measure a time frame of 1/440 = 0.002272 s = 2.272 ms
- if you play the A3 note, you will see a perfectly still signal on your screen.
- if you play the A4 note (one octave higher, at 880Hz), you will see a perfectly still signal just like before, only difference is the signal will look twice as "squeezed" because 1/880 = 1.136 ms ---> twice as short as A3

Hope you got it.

Re: Re: Re: Waveforms Yay!

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Originally posted by ponsshin No it just depends of the settings of the oscilloscope. the G note appears stable because the period of time in which is measured by the oscilloscope equals (or is close to at least) to the period of the G note (period of a frequency = 1/frequency in Hz). So let's say you have a frequency note of 440 Hz which corresponds to A3: - set your oscilloscope to measure a time frame of 1/440 = 0.002272 s = 2.272 ms - if you play the A3 note, you will see a perfectly still signal on your screen. - if you play the A4 note (one octave higher, at 880Hz), you will see a perfectly still signal just like before, only difference is the signal will look twice as "squeezed" because 1/880 = 1.136 ms ---> twice as short as A3 Hope you got it.

Bingo!

Well explained mate.

Re: Re: Re: Waveforms Yay!

quote:

Originally posted by ponsshin No it just depends of the settings of the oscilloscope. the G note appears stable because the period of time in which is measured by the oscilloscope equals (or is close to at least) to the period of the G note (period of a frequency = 1/frequency in Hz). So let's say you have a frequency note of 440 Hz which corresponds to A3: - set your oscilloscope to measure a time frame of 1/440 = 0.002272 s = 2.272 ms - if you play the A3 note, you will see a perfectly still signal on your screen. - if you play the A4 note (one octave higher, at 880Hz), you will see a perfectly still signal just like before, only difference is the signal will look twice as "squeezed" because 1/880 = 1.136 ms ---> twice as short as A3 Hope you got it.

That would make sense.

So the smooth wavs are just the meter being funny.

What about the slow wavs peaking madly? Mind you, when i say slow, i mean wavs so slow it takes 30 second to go from flat to peak. then another 30 seconds to go back. [two minutes to complete one cycle]

quote:

Originally posted by cronodevir So the smooth wavs are just the meter being funny. What about the slow wavs peaking madly? Mind you, when i say slow, i mean wavs so slow it takes 30 second to go from flat to peak. then another 30 seconds to go back. [two minutes to complete one cycle]

a Tsunami wave? Which beach is this?

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Originally posted by cronodevir What about the slow wavs peaking madly? Mind you, when i say slow, i mean wavs so slow it takes 30 second to go from flat to peak. then another 30 seconds to go back. [two minutes to complete one cycle]

It's all the same, dude. The scope is showing snapshots of the waveform at different points in time. If the fundamental frequency of the waveform happens to be close to the scope's refresh frequency, or some whole multiple/factor of it, then you'll see what looks more like an animation than chaos.

Real scientific oscilloscopes calibrate themselves to the fundamental so the waveform actually appears to be standing still (assuming that it has a constant frequency and amplitude). The software scopes on programs like FL and Winamp are just novelties; they don't tell you anything about the sound.

No, There is no sound coming from the speakers, but the FX or mixer insert is very high. I used a meter that goes up to +96 db and it was even above that. I set the root key to C0 [which is now C4 on the midi keyboard...] then i pressed C0 on the midi keyboard and it did go down even 4 more octaves. But it also made my mixer explode.

Not talking about the anaylyser anymore, I want to know why such a low frequency sound [in truth there is no sound being produced] is causing the mixer to redline.

Also, another thing i'm curious about. Is there really even a "wave"? I know in real instruments the "wave" is produced by blowing or striking or plucking something. In a analog synth the "wave" is the electrical current running through the box, but what is it in a software instrument?

quote:

Originally posted by cronodevir Not talking about the anaylyser anymore, I want to know why such a low frequency sound [in truth there is no sound being produced] is causing the mixer to redline.

Is not this because it is outside the frequency range of the human ear ?

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Originally posted by cronodevir Not talking about the anaylyser anymore, I want to know why such a low frequency sound [in truth there is no sound being produced] is causing the mixer to redline.

Any sound can cause the mixer to redline if it's loud enough. Doesn't matter what frequency it is. And of course, depending on the synthesis method and effects used, some notes can end up being louder than others - it's all to do with harmonics and phase.

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In a analog synth the "wave" is the electrical current running through the box, but what is it in a software instrument?

Samples of the analog waveform. That's what the term "sampling rate" refers to.

It's no different from drawing a line on your computer screen. In reality the line is not continuous, it's several discrete pixels, but they are so small that it appears to be continuous. With sound, the DAC actually interpolates the digital signal into a true analog signal, but it's the same basic idea.

I mean, the "signal" its a string of 1's and 0's though, not electricity? Thus the reason software sounds different than hardware [this was my assumption]

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Originally posted by cronodevir I mean, the "signal" its a string of 1's and 0's though, not electricity? Thus the reason software sounds different than hardware [this was my assumption]

Ones and Zeros calculated to create a sample, if I'm not mistaken.

Yeah that is what 8bit 16bit etc etc refer to, how many 1's and 0's in a string. 32bit wav file uses 32 numbers in every string.

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Originally posted by cronodevir I mean, the "signal" its a string of 1's and 0's though, not electricity? Thus the reason software sounds different than hardware [this was my assumption]

Uh... yeah, it's stored in the hardware registers as a "string of 1s and 0s" but it's read and written as integers or floats. And because it's interpolated by a DAC on the way out, it really doesn't sound different unless the sampling rate or bit depth are really low.

The "string of bits" is called a "word" and is just a regular number in base 2.

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