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an open-source digital signal processing and sound synthesis language
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DCBLOCK -- remove (most of) DC bias from input signal
quick syntax:DCBLOCK(outsk, insk, dur, AMP)
CAPITALIZED parameters are pfield-enabled for table or dynamic control (see the maketable or makeconnection scorefile commands). Parameters after the [bracket] are optional and default to 0 unless otherwise noted.
p0 = output start time (seconds) p1 = input start time (seconds) p2 = duration (seconds) p3 = amplitude multiplier (relative multiplier of input signal) p3 (amplitude) can receive dynamic updates from a table or real-time control source. Author: John Gibson
DCBLOCK uses a simple one-pole/one-zero filter set to remove a DC (0 Hz) offset component in the output signal. The recursive filter equation used for this object is:
y[n] = x[n] - x[n-1] + 0.99*y[n-1]where y[n] and y[n-1] are the current and previous outputs of the equation, respectively, and x[n] and x[n-1] are the current and previous sample inputs to the filter equation.
Usage NotesDCBLOCK processes N input channels to N output channels, e.g. mono to mono, stereo to stereo, quad to quad, etc.
The sound itself should be relatively unchanged by DCBLOCK.
Sample Scoresvery basic:
rtsetparams(44100, 2) load("DCBLOCK") rtinput("mysound.aif") DCBLOCK(0, 0, DUR(), 1.0)
See AlsoELL, EQ, FIR, IIR, JFIR, MULTEQ