;(in-package :clm)

#| 

This version of pvoc is the same as Michael's version
just ported for clm-2 (by Marco Trevisani)

just added: envelope for distance, and degree


pvoc - A feature complete phase vocoder instrument. Based in part on sms
instrument and the phase vocoder section in F. Richard Moore's "Elements of
Computer Music"

Performs analysis and oscillator bank resynthesis. 
time - specifies the time dilation ratio
pitch - specifies the pitch transposition ratio
gate - specifies a resynthesis gate in dB (partials with amplitudes lower than
the gate value will not be synthesized)
pitchoffset - a pitch offset in Hz (creates inharmonic spectra)

todo: improve speed, reduce number of temporary arrays
      start and end points (currently chops off some analysis that should be
      performed) 
      read and write analysis files (in formats such as Beuachamp's sndan,
      csound pvoc, etc...) 
      improve quality when input and output sample rates differ (still sounds
      bad) 
      smarter resynthesis gating (time based gating)
      plug-in system for spectral modification

Michael Klingbeil
March, 1999   Urbana, IL

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

Modifications by Michael Edwards (michael@ccrma.stanford.edu), January 2000:

The basic pv algorithm is unchanged here, but I added some features that I
thought would be nice to have:

duration: If duration is given then the given time-stretch will be ignored and
will be instead re-calculated to fit the given duration.   Now you have the
choice. 

input-start/input-end: The original instrument always transformed the whole
sound file.  With these arguments (in seconds) you can read whichever portion
you want to.  If input-end is not specified, then it will read to the end of
the file as normal.

time -> time-stetch
pitch -> pitch-scaler 
pitchoffset -> pitch-offset
These arguments all do the same as the original, but I changed their names when
I was feeling pedantic and/or confused.

pitch-scaler-env is like pitch-scaler but instead of being a fixed constant is
now envelope controllable.  pitch-scaler is the scaler to this envelope. 

pitch-offset-env is similar again in that you can control the offset with an
envelope using y values given in Hz.  Note that pitch-offset is an offset to
this envelope so whatever the envelope returns will be added to pitch-offset.

harmonic-mirror-env is an envelope also in Hz, and is similar to
pitch-offset-env except that instead of adding an offset to all harmonics, it
SUBTRACTS the offset from the lower half of the harmonics and ADDS the offset
to the upper half, thus creating a symmetry around the central harmonic(s) and
some kind of expansion of the frequency space.  Note that you can't have a
pitch-offset-env and a harmonic-mirror-env at the same time.

amp/amp-env: The standard things that were missing in the original.

degree/rev-amt/distance: The original was mono only (outa) so I've inserted the
standard loc structure for stereo capability.

rev-env:  a reverb envelope that plays with the loc structure to create
time-varying reverb amount.

For those wondering about the gate argument, it wasn't clear to me how it
worked at first, but I assume now that Michael Klingbeil means amplitudes lower
than a real positive DB will not be resynthesised, which was confusing to me,
because I always tend to think on that negative DB scale that we see on DAT
players.  Using a value of 70 or so worked for me (and I think it reduced the
resynth time--can that be?).

Final words:  Nice instrument, shame it's so damned slow!  I'd really like to
see the analysis/resynthesis parts separated so that analysis data can be saved
and reloaded, but I'm too lazy/confused to go into that at the moment.

|#

#|

(with-sound (:scaled-to .99 :statistics t) 
  (pvoc "b6.wav" 0
        :input-start .145
        :input-end 5.4
        :time-stretch 10
        :duration 7
        ;;:pitch-offset 100
        ;;:pitch-offset-env '(0 0 20 0 50 100 70 100 80 0 100 0)
        ;;:harmonic-mirror-env '(0 0 20 0 50 1000 70 1000 80 0 100 0)
        ;;:pitch-scaler-env '(0 1 50 2 100 1)
        ;;:amp-env '(0 1 40 0 50 1 60 0 100 1)
        ;;:gate 70
        :fftsize 512))

|#

(definstrument pvoc
    (file time &key 
          (duration nil)
          (input-start 0)
          (input-end nil)
          (fftsize 512) 
          (overlap 4) 
          (time-stretch 1.0)
          (pitch-scaler 1.0)
          (pitch-scaler-env '(0 1 100 1))
          (pitch-offset 0.0)  ;; in Herz!
          (pitch-offset-env '(0 0 100 0)) ;; Also in Herz
          (harmonic-mirror-env nil) ;; again in hz
          (degree 0)
	  (degree-env '(0 .5 100 .5))
          (rev-amt 0)
          (rev-env '(0 0 100 1))
          (distance 1)
	  (distance-env '( 0 1 1 1))
          (gate 0.0) 
          (amp 1.0)
          (amp-env '(0 1 100 1)))
  (when (and pitch-offset-env harmonic-mirror-env)
    (warn "pitch-offset-env and harmonic-mirror-env cannot be used together. ~
           Using harmonic-mirror-env only."))
  (when (and duration time-stretch)
    (warn "duration and time-stretch cannot be used together.  Using duration ~
           to calculate new time-stretch"))
  (let* ((fil (open-input* file))
         (file-duration (sound-duration file)) 
         (file-sampling-rate (sound-srate file))
         (input-start-sample (* input-start *srate*))
         (input-end-sample (floor (if input-end 
                                      (* input-end *srate*)
                                    (* file-duration *srate*))))
         (input-dur-samples (- input-end-sample input-start-sample))
         (input-dur-secs (/ (float input-dur-samples) *srate*))
         (2ids (* 2 input-dur-secs))
         (filptr input-start-sample)           ;; index into the file
         (ts (if duration (/ duration input-dur-secs) time-stretch))
         (dur (* input-dur-secs ts))
         (start (floor (* time *srate*)))
         (end (+ start (floor (* dur *srate*))))
         
         (pi2 (* 2 pi)) ;; handy constant
         (N fftsize)  ;; fft size
         (N2 (floor N 2))  ;; half the fft size
         (N4 (floor N 4))  ;; quarter the fft size
         (Nw fftsize) ;; window size -- currently restricted to the fftsize
         ;; decimation factor (how often do we take an fft)
         (D (floor fftsize overlap)) 
         ;; interpolation factor how often do we synthesize
         (interp (* (floor fftsize overlap) 
                    (/ *srate* file-sampling-rate)
                    ts))      
         ;; compensates for window normalization
         (windowsum 0.0)    
        
         ;; take a resynthesis gate specificed in dB, convert to linear
         ;; amplitude  
         (syngate (if (= 0.0 gate) 0.0 (expt 10 (/ (- (abs gate)) 20))))
         
         ;;(pitch-offset (in-hz pitch-offset))
         (psenv (make-env :envelope pitch-scaler-env :scaler pitch-scaler :duration 2ids))
         (poenv (make-env :envelope pitch-offset-env 
                          :offset (in-hz pitch-offset)
                          :scaler (in-hz 1)
                          :duration 2ids))
         (hmenv (and harmonic-mirror-env
		     (make-env :envelope harmonic-mirror-env 
			       :scaler (in-hz 1)
			       :duration 2ids)))
	 (degrenv (make-env :envelope degree-env :duration dur))
	 (distenv (make-env :envelope distance-env :duration dur))
	 (ampf (make-env :envelope amp-env :scaler amp :duration dur))
         (loc (make-locsig  :distance distance :degree degree
			    :reverb rev-amt))
         (renv (make-env :envelope rev-env :scaler rev-amt :duration dur))
         (window (make-double-array Nw)) ; array for the window
         (fdr (make-double-array N)) ; buffer for real fft data
         (fdi (make-double-array N)) ; buffer for imaginary fft data
         
         (lastphase (make-double-array N2)) ;; last phase change
         (lastamp (make-double-array N2)) ;; last sampled amplitude
         (lastfreq (make-double-array N2)) ;; last sampled frequency
         ;; amplitude interpolation increment
         (ampinc (make-double-array N2))
         ;; frequency interpolation increments
         (freqinc (make-double-array N2))
         
         ;; expresses the fundamental in terms of radians per OUTPUT sample
         (fundamental (* (/ file-sampling-rate N) (/ pi2 *srate*)))
         ;; (fundamental (in-hz (/ file-sampling-rate N)))

         (sum 0.0)                      ;; output sample sum
         ;;(poval 0.0)
         ;;(psval 0.0)
         (output (floor interp))      ;; count of samples that have been output
         (resynth-oscils (make-array N2))
         ;; how often to print out the percentage complete message
         (nextpct 10.0)
	 (stereo (>= (mus-channels *output*) 2))
	 (dist-scaler 0.0))
    ;; zero out the arrays (lastamp and lastphase will be set from fdr and fdi)
    (clear-array lastphase)
    (clear-array fdr)
    (clear-array fdi)

    ;; setup oscillators
    (dotimes (i N2)
      (setf (aref resynth-oscils i) (make-oscil :frequency 0)))
    (when duration (format t "~&time-stretch = ~a~%" ts))

    ;; set-up the analysis window here
    (setf windowsum 0.0)
    ;; create a Hamming window (Moore p. 251)
    (dotimes (k Nw)
      (let ((val (- 0.54 (* 0.46 (cos (* 2 pi (/ k (- Nw 1))))))))
	(setf (aref window k) val)
	(incf windowsum val)))
    ;; normalize window
    (setf windowsum (/ 2.0 windowsum))
    ;; loop over normalizing the window
    (dotimes (k Nw)
      (setf (aref window k) (* (aref window k) windowsum)))
    
    (run
     (loop for i from start to end do
       (declare (type :float dist-scaler nextpct sum fundamental syngate pi2 interp)
		(type :double* lastphase fdr fdi lastamp lastfreq ampinc freqinc window)
		(type :boolean stereo)
		(type :integer output N2 N4 N D file-sampling-rate filptr))

       ;; if all the samples have been output then do the next frame
       (if (>= output interp)
	   (let ((buffix (mod filptr N)))
	     ;; buffix is the index into the input buffer
	     ;; it wraps around circularly as time increases in the input
	     
	     ;; reset the output sample counter
	     (setf output 0)
	     
	     ;; save the old amplitudes and frequencies
	     (dotimes (k N2)
	       (setf (aref lastamp k) (aref fdr k))
	       (setf (aref lastfreq k) (aref fdi k)))
	     
	     (dotimes (k N)
	       ;; apply the window and then stuff into the input array
	       (setf (aref fdr buffix) (* (aref window k)
					  (ina filptr fil)))
	       (incf filptr)
	       
	       ;; increment the buffer index with wrap around
	       (incf buffix)
	       (if (>= buffix N) (setf buffix 0)))

	     ;; rewind the file for the next hop
	     (decf filptr (- N D))
	     ;; no imaginary component input so zero out fdi
	     (clear-array fdi)
	     ;; compute the fft
	     
	     (fft fdr fdi N 1)

	     ;; now convert into magnitude and interpolated frequency
	     (dotimes (k N2)
	       (let* ((a (aref fdr k))
		      (b (aref fdi k))
		      (mag (* (sqrt (+ (* a a) (* b b)))))
		      (phase 0.0) 
		      (phasediff 0.0))
		 (setf (aref fdr k) mag)    ;; current amp stored in fdr
		 ;; mag is always positive
		 ;; if it is zero then the phase difference is zero
		 (if (> mag 0)
		     (progn
		       (setf phase (- (atan b a)))
		       (setf phasediff (- phase (aref lastphase k)))
		       (setf (aref lastphase k) phase)
		       ;; frequency wrapping from Moore p. 254
		       #-clisp (loop while (> phasediff pi) do (decf phasediff pi2)) ; pi2 = 2*pi
		       #-clisp (loop while (< phasediff (- pi)) do (incf phasediff pi2))
		       #+clisp (if (> phasediff pi) (decf phasediff pi2))
		       #-clisp (if (< phasediff (- pi)) (incf phasediff pi2))
		       ))

		 ;; current frequency stored in fdi
		 ;; scale by the pitch transposition
		 (setf (aref fdi k) 
		       (* (env psenv) 
			  (+ (/ (* phasediff file-sampling-rate)
				(* D *srate*))
			     (* k fundamental)
			     (if (env? hmenv)
				 (if (> k N4)
				     (env hmenv)
				   (- (env hmenv)))
			       (env poenv)))))

		 ;; resynthesis gating
		 (if (< (aref fdr k) syngate)
		     (setf (aref fdr k) 0))

		 ;; take (aref lastamp k) and count up to (aref fdr k)
		 ;; interpolating by ampinc
		 (setf (aref ampinc k) (/ (- (aref fdr k) 
					     (aref lastamp k)) 
					  interp))
		 ;; take (aref lastfreq k) and count up to (aref fdi k)
		 ;; interpolating by freqinc
		 (setf (aref freqinc k)
		       (/ (- (aref fdi k) (aref lastfreq k)) interp))))
	     
	     ;; we can now output samples
	     ;; set the interpolation increment
	     ;; (setf interp-inc (/ 1.0 interp))
	     ;; set the interpolation factor
	     ;; (setf interp-factor 0.0)
	     ))

       ;; loop over the partials
       (setf sum 0.0)
       (dotimes (k (floor N 2))
	 ;; interpolate frequency and amplitude
	 ;; it would be faster to do this with increments
	 
	 (incf (aref lastamp k) (aref ampinc k))
	 (incf (aref lastfreq k) (aref freqinc k))
	 
	 ;; nearly 75% of the compute time is spent here
	 ;; clm-2: (setf sum (oscil-bank lastamp resynth-oscils lastfreq (floor N 2)))
	 
	 (if (> (aref lastamp k) 0)
	     (incf sum (* (aref lastamp k)
			  (oscil (aref resynth-oscils k)
				 (aref lastfreq k))))))
       
       (setf dist-scaler (/ 1.0 (env distenv)))
       (setf (locsig-ref loc 0) (* dist-scaler (- 1.0 (env degrenv))))
       (when stereo
	 (setf (locsig-ref loc 1) (* dist-scaler (env degrenv))))
       (when *reverb* (setf (locsig-reverb-ref loc 0)
			    (* (sqrt dist-scaler)(env renv))))
       (locsig loc i (* (env ampf) sum))
       ;; print out a message
       (let ((pct (* 100.0 (/ (- i start) (- end start)))))
	 (if (>= pct nextpct) (progn
				(clm-print "~&~F%" pct)
				(incf nextpct 10))))
       

       ;; we just output 1 sample
       (incf output)))))

