Am I missing something? [Audio phase modulation]

cs5947 Member Posts: 28 Member
edited December 2022 in Building With Reaktor

Planning to create an audio phase-modulation effect. This is different from a delay effect, in which the delay is in absolute time and the phase mod is in relative time. Like this plugin, from the creators of setBfree and Whirl

But I am struggling to determine the wavelength/time shifting amount from a signal alone. Say, a sine wave with a varying rate is not at all being still enough for the delay time to be able to match the phase, or a complex audio signal with lots of harmonics has virtually the same effect.


  • errorsmith
    errorsmith Member Posts: 25 Helper

    phase rotate isn't realised with a delay. if you set a phase rotator to 90° ALL frequencies of the incoming signal get a phase shift of 90°.

    the closest thing to a phase rotator in the reaktor library is the 'phase splitter' used in the frequency shifter (search for 'freq shift'). its shift amount is an unadjustable 90° between 'im' and 're' out. send a sine wave in and connect a scope to 'im' and 're'. you will see this shift regardless of the frequency. The shift is between 'im' and 're' not in relation to the input (like in x42 i assume). so this might not what you are looking for.

    Note that the x42 has pre ringing as noted on their website. that tells me that it is realised with a FIR filter. i have no idea how to calculate the FIR coefficients for a phase rotator. i would send an impulse into x42 and look at the impulse response and see if i would come up with a function for the coefficients that approximates the impulse response.

  • Jean louis P
    Jean louis P Member Posts: 7 Member

    With a delay :

    If you want to work with phase of a pitch, you need to put the phase delay modulator on the oscillator level.

    And you set the full phase time to the reverse of the pitch.

    if your note is 1000 Hz, then you phase length is 0.001 s , then the delay you want is = 0.001 * phase-delay.

    With low frequency ( LFO on phase instead of time ) you can have some distortion in the signal, then on audio level it will not be very good , else may be if you work with over sampling, for a better definition, the more you have of samples along the phase the better your signal will be.

Back To Top