This approach is to left/right channel processing is similar to the
amplitude-envelope following instrument
FOLLOWER.
This instrument is similar in some respects to
PVOC,
but it is a channel vocoder using a bank of band-pass filters
instead of an FFT analysis (like with phase vocoders).
This kind of instrument was originally
designed for cross-synthesis work -- see also
VOCODE and
VOCODESYNTH.
p-fields:
/* VOCODE2 - channel vocoder
p0 = output start time
p1 = input start time (must be 0 for aux bus)
p2 = duration
p3 = amplitude multiplier (post-processing)
p4 = number of filters (greater than 0)
p5 = lowest filter center frequency (in Hz or oct.pc)
p6 = center frequency spacing multiplier (greater than 1)
(multiplies each cf by this to get next higher cf)
p7 = amount to transpose carrier filters (in Hz or oct.pc)
p8 = filter bandwidth proportion of center frequency (greater than 0)
p9 = filter response time (seconds) [optional; default is 0.01]
Determines how often changes in modulator power are measured.
p10 = amount of high-passed modulator signal to mix with output
(amplitude multiplier) [optional; default is 0]
p11 = cutoff frequency for high pass filter applied to modulator.
This pfield ignored if p10 is zero. [optional, default is 5000 Hz]
p12 = amount of noise signal to mix into carrier before processing
(amplitude multiplier applied to full-scale noise signal)
[optional; default is 0]
p13 = specifies how often (in samples) to get new random values from
the noise generator. This pfield is ignored if p12 is zero.
[optional; default is 1 -- a new value every sample]
p14 = percent to left channel [optional, default is 0.5]
Assumes function table 1 is an amplitude curve for the note. (Try gen 18.)
Or you can just call setline. If no setline or function table 1, uses a
flat amplitude curve. This curve, combined with the amplitude multiplier,
affect the signal after processing by the filter bank.
NOTES:
- Currently in RTcmix it's not possible for an instrument to take
input from both an "in" bus and an "aux in" bus at the same time.
So if you want the modulator to come from a microphone -- which
must enter via an "in" bus -- and the carrier to come from a
WAVETABLE instrument via an "aux" bus, then you must route the
mic into the MIX instrument as a way to convert it from "in" to
"aux in". See "VOCODE2_1.sco" for an example.
- The "left" input channel comes from the bus with the lower number;
the "right" input channel from the bus with the higher number.
- When specifying center frequencies...
p6 = 2.0 will make a stack of octaves
p6 = 1.5 will make a stack of perfect (Pythagorian) fifths
Use this to get stacks of an equal tempered interval (in oct.pc):
p6 = cpspch(interval) / cpspch(0.0)
- More about p13:
When this is greater than 1 sample, successive noise samples are
connected using linear interpolation. This acts as a low-pass filter;
the higher the interval, the lower the cutoff frequency.
*/
Sample scorefile:
rtsetparams(44100, 2)
load("VOCODE2")
rtinput("/snd/beckett/furniture1.aiff")
/* carrier */
bus_config("MIX", "in 0", "aux 0 out")
inskip = 0
amp = 1
dur = DUR() - inskip
MIX(0, inskip, dur, amp, 0)
/* modulator */
bus_config("MIX", "in 0", "aux 1 out")
inskip = 0
dur = DUR() - inskip
amp = 1
MIX(0, inskip, dur, amp, 0)
/* -------------------------------------------------------------------------- */
bus_config("VOCODE2", "aux 0-1 in", "out 0-1")
amp = 1.0
numfilt = 0
makegen(2, 2, numpitches = 8,
7.00,
7.07,
8.02,
8.09,
9.04,
9.11,
10.06,
11.01
)
transp = 0.07
freqmult = 2.02
cartransp = -0.02
bw = 0.008
resp = 0.00
hipass = 0.2
hpcf = 5000
noise = 0.01
noisubsamp = 5
dur = dur + 2
setline(0,1, dur-1,1, dur,0)
VOCODE2(0, 0, dur, amp, numfilt, transp, freqmult, cartransp, bw,
resp, hipass, hpcf, noise, noisubsamp, 1)
transp = transp + 0.002
VOCODE2(0, 0, dur, amp, numfilt, transp, freqmult, cartransp, bw,
resp, hipass, hpcf, noise, noisubsamp, 0)