I'm a tad confused how this module works. I can't follow how the same timbre flows through the two filters at the same time to produce a sound different from the serial filter configuration.
What types of sounds or tricks is the parallel filter best used for?
Radias parallel filter?
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- axxim
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Hi platypus,
it's simple: If you use first a low pass and then a hi pass in series what you get at their output is the band between them. If you set them in parallel, then what you get is either the inverse, the bands below and above them. It depends on what filters you use and where the cutoff is.
For example if you set a low pass to 1kHz and a hi pass to 3kHz, in series you will get nearly nothing. In parallel you will get all but the band between 1 and 3kHz what in this case is a notch filter (supposing the filters are very steep)
The use of the parallel filters could be used to enhance/limit more some frequencies of the sound depending on what filters you select. In the parallel and serial mode both osc's are mixed before the filters. In the individual mode each osc is routed to a filter and then mixed. This lets you select a sepparate frequency spectrum for each osc.
it's simple: If you use first a low pass and then a hi pass in series what you get at their output is the band between them. If you set them in parallel, then what you get is either the inverse, the bands below and above them. It depends on what filters you use and where the cutoff is.
For example if you set a low pass to 1kHz and a hi pass to 3kHz, in series you will get nearly nothing. In parallel you will get all but the band between 1 and 3kHz what in this case is a notch filter (supposing the filters are very steep)
The use of the parallel filters could be used to enhance/limit more some frequencies of the sound depending on what filters you select. In the parallel and serial mode both osc's are mixed before the filters. In the individual mode each osc is routed to a filter and then mixed. This lets you select a sepparate frequency spectrum for each osc.
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https://www.youtube.com/channel/UCyi189 ... UTEpsykkIg
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- Timo
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Just made a quick pic:

'Drive' (if used) can be either Pre- or Post-filter, but not both.
Understanding how the filters are routed is one of the key aspects of programming any synthesiser, I feel.
Hope this helps.

'Drive' (if used) can be either Pre- or Post-filter, but not both.
Understanding how the filters are routed is one of the key aspects of programming any synthesiser, I feel.

Hope this helps.
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Ah! The inverse! That makes sense now. Thank you axxim for the clarification. So in a sense the parallel filters are a glorified notch filter (or at least when running the oscs mixed together)?
Timo, I'm highly visual in nature so that diagram was a good aid as I thought about the audio path and attenuated frequencies that would be used. Thanks!
While we're on the subject, axxim used Hertz when discussing frequencies. What are the ranges for the OSCs and LFOs on the radias? I can't find it anywhere in the book or on the net (the display just says 0-127).
Timo, I'm highly visual in nature so that diagram was a good aid as I thought about the audio path and attenuated frequencies that would be used. Thanks!
While we're on the subject, axxim used Hertz when discussing frequencies. What are the ranges for the OSCs and LFOs on the radias? I can't find it anywhere in the book or on the net (the display just says 0-127).
- axxim
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No, I haven't found any info about this. I know that the comb filter values are tuned to the keyboard scale. A frequency value of 60 and a key follow of +1.00 will match the chromatic tuning. The next matches are every 12 steps above or below. A frequency value of 79 makes a nice 7th harmonics over the original. I think the other filters are scaled similar.platypus wrote: While we're on the subject, axxim used Hertz when discussing frequencies. What are the ranges for the OSCs and LFOs on the radias? I can't find it anywhere in the book or on the net (the display just says 0-127).
The best way to test this comb filter is noise=full, filter1 on bypass or low pass full open, filter2 freq=60, keyfollow=+1.00, reso=full. You will get a sound like blowing in a metall tube matching the kb tones. This is a good setting to experiment with the other settings of the filter (env follow, lfo->frequency, etc.)
What I have measured out two days ago, are the center frequencies of the vocoder bands, which I will post later in this forum
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Only a Radias, VP-770, SP-170S and iPad2
http://www.axxim.de/r3dias
https://www.youtube.com/channel/UCyi189 ... UTEpsykkIg
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https://www.youtube.com/channel/UCyi189 ... UTEpsykkIg
- Timo
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Regards the filter frequencies, you can measure the rough frequency (Hz) value of each filter cutoff step by whacking the resonance up and sweeping the cutoff while looking at a FFT display and homing in on the resonant spike.
However filters aren't "surgical". Instead of frequencies being surgically cut after the cutoff frequency, they gently fall away. LPF24 uses a -24dB/octave slope, and LPF12 uses -12dB/oct slope.
But no filters are equal. Not even if they specify the slope (as above).
The Access Virus -12dB/oct and -24dB/oct low-pass filters sound nothing like the Radias' filters of the same -12dB/oct or -24dB/oct type, even when the cutoff values for both Virus and Radias are calibrated and matched.
When you calibrate both Radias and Virus filters to the same cutoff frequency by using maximum resonance and a FFT display, as soon as you lower the resonance they sound completely different again as their slopes and character attenuate different frequencies.
Therefore I'd say it's potentially pointless trying to measure the discrete Hz value for the filter cutoffs as the overall filter character is more subjective and important instead. "If it sounds good, it is good." ~Joe Meek
Noticed you mentioned LFO frequencies. These are wholly different to filters. LFOs merely 'modulate' something else. They don't make any sound themselves. They are controllers. They can control the pitch of an oscillator cyclically making it go up and down, as one example.
LFOs are shown in Hz in the menus. They go from 0.01Hz (one cycle every 100 seconds) to 100Hz (100 times a second).
However filters aren't "surgical". Instead of frequencies being surgically cut after the cutoff frequency, they gently fall away. LPF24 uses a -24dB/octave slope, and LPF12 uses -12dB/oct slope.
But no filters are equal. Not even if they specify the slope (as above).
The Access Virus -12dB/oct and -24dB/oct low-pass filters sound nothing like the Radias' filters of the same -12dB/oct or -24dB/oct type, even when the cutoff values for both Virus and Radias are calibrated and matched.
When you calibrate both Radias and Virus filters to the same cutoff frequency by using maximum resonance and a FFT display, as soon as you lower the resonance they sound completely different again as their slopes and character attenuate different frequencies.
Therefore I'd say it's potentially pointless trying to measure the discrete Hz value for the filter cutoffs as the overall filter character is more subjective and important instead. "If it sounds good, it is good." ~Joe Meek
Noticed you mentioned LFO frequencies. These are wholly different to filters. LFOs merely 'modulate' something else. They don't make any sound themselves. They are controllers. They can control the pitch of an oscillator cyclically making it go up and down, as one example.
LFOs are shown in Hz in the menus. They go from 0.01Hz (one cycle every 100 seconds) to 100Hz (100 times a second).
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Thanks guys, this all helps a lot.
Much of the literature I've found talking about what frequencies were best used for basic tone or adding passing effects (such as using growl for the beginning of a brass sound) always gave it to me in Hz. So my reason for asking was basically to find out so I could get better results for physically modeled instruments (I know, I know, the radias doesn't excel here but I experiment with it anyway).
Much of the literature I've found talking about what frequencies were best used for basic tone or adding passing effects (such as using growl for the beginning of a brass sound) always gave it to me in Hz. So my reason for asking was basically to find out so I could get better results for physically modeled instruments (I know, I know, the radias doesn't excel here but I experiment with it anyway).