ECMC PVC example file chordmapper2


 Example PVC script used to create /sflib/x soundfile >> chordmapper2.wav <
# ECMC pvc example  chordmapper2: this example creatres a four note
# "harmonization" from soprano tone /sflib/voice/sop1.b3. The four
# chord tone clones created from the analysis of this source soundfile
# are pitched at fs3, c4, f4 and b4. The output amplitudes of these 4 notes decrease from the lowest to highest note, in attempt to
# "balance" the chord.
 # echo \
# 7.11   1  7.06   0  .3  0    1 0  0 0 0 0 \
# 7.11   1  8.00   0  .3  -3   1 0  0 0 0 0 \
# 7.11   1  8.05   0  .3  -6   1 0  0 0 0 0 \
# 7.11   1  8.11   0  .3  -9  1 0  0 0 0 0 \
# > $SFDIR/chordmapfile 
# The low .3 values in parameter 5 (partial bandwidth as proportion  of
# fundamental frequency) simplify the timbre, reducing higher frequencies
# To obtain better spectral resolution and reduce buzzing artifacts,
# the FFT size is increase from the default 1024 to 4096:
 #  FFT_length=4096 # 2048 # 1024 
# To eliminate clipping that would be caused by 4 clones of a tone the
# master gain is reduced by 10 dB 
 # MASTER_gain_in_decibels=-10   # in dB, int, float or FUNC 
# Notice the exaggeration in the quavering amplitude variations of the
# original soprano tone that result when four pitch-shifted clones
# of this tone sound simultaneously.



#******************************************************
#************** CHORDMAPPER ***************************
#******************************************************
     # ******ECMC CHANGES & ADDITIONS: ***************
#******** INPUT & OUTPUT SOUNDFILES *************
  # input soundfile, can be aiff or wave format on Linux, aiff only on SGI
inputsf=/sflib/voice/sop1.b3.wav
outputsf=chordmapper2.wav  # output soundfile
# ********************* #
   ##### Cmusic function file generator tempates #####
#   gen0  normalizes function files previously created with other gen routines
# gen0 -Llength  max < inputfuncfile > outputfuncfile
#   gen1 creates linear {straight line} segments, like Csound gen 7
# gen1 -Llength t1 v1 ... tN vN
#   gen2 generates harmonic waveforms from sine {a} & cosine {b} amps
# gen2 -Llength [-o (default) or -c] a1 ... aN b0 ... bM N
#  gen3 generates amp values & linear connections at equally spaced time points
# gen3 -Llength v1 v2 ... vN
#   gen4 generates exponenetial segments; "a" values determine shape &
#  depth of curve: 0 = linear, neg. = exponential, pos. = inverse expo.
# gen4 -Llength t1 v1 a1 ... tN vN
#  gen5 is like Csound gen 9 : harmonic1/amp/phase harmonic2/amp/phase
# gen5 -Llength h1 a1 p1 ... hN aN pN
#     gen6 generates a table of random numbers between +1 and -1
# gen6 -Llength
#    cspline: smooth curve {cubic spline} interpolator
# cspline len_flag [flags] x0 y0 x1 y1 ... xN yN
#   genraw reads in a previously created function file
# genraw -LN filename    (where N is the length of the output function.)
# For a usage summary of "reshape" type  "reshape"  with no arguments.
   ##### End of gen routine function generator tempates #####
#......................................................
output_data_format=1
   #  0=  Same as input file, 1 = integers 2 = rescaled floats 

#******** BEGIN/END TIMES *****************************
   # beginning and end times within input soundfile for analysis/resynthesis
begintime=0    # time in input soundfile to begin analysis/resynthesis
endtime=0      # (-1 or 0 end time defaults to end of file)

#======================================================
#*** ANALYSIS PARAMETERS ******************************
FFT_length=1024 # must be power-of-2, usually 1024, 512 or 2048
      # do not set FFT_length lower than 1024 if samp. rate = 96000
window_type=2
    # window type: 0 = Hamming,1 = rectangular, 2 = Blackman (DEFAULT),
    #  3 = Bartlett triangular, 4-12 = Kaiser windows for alpha = 4-12
    # generally recommended: 2 or 8
windowsize=0    # default 0 sets windowsize (in samples) to 2 * FFT_length if
  #  samp. rate <= 48000 or to 4 * FFT_length if samp. rate > 48000
frames_per_second=200   # generally 200, occasionally 400 or 600 when time stretching
#======================================================
#*** RESYNTHESIS PARAMETERS ***************************

time_expansion_contraction_factor=1
       # OUTPUT CHANNEL(S) .......................
output_channel=0
   # channels are numbered from 1 to the maximum; 0 = all channels
#.............OSCIL THRESHOLD ........................
oscillator_resynthesis_threshold_in_dB=-80
     #( Usually -60 to -80 unless dropouts become audible. )

#****************** MODIFICATIONS *********************

#.................. DECIBELS ..........................

MASTER_gain_in_decibels=-10   # in dB, int, float or FUNC

#********** SOURCE ************************************
#.................. DECIBELS ..........................
SOURCE_gain_in_dB=-96  # in dB, int, float or FUNC

#.................. PITCH .............................

SOURCE_frequency_shift=0  # int, float or FUNC
SOURCE_pitch_transposition_in_semitones=0 # int, float or FUNC

#======================================================
#**************** TARGETS *****************************
#======================================================
#.................. DECIBELS ..........................

TARGETS_gain_in_dB=-0  # int, float or FUNC

#.................. PITCH .............................
TARGETS_frequency_shift=0   # int, float or FUNC
TARGETS_pitch_transposition_in_semitones=0  # int, float or FUNC

#............ SPECTRUM WARPSHAPE ......................

spectrum_warpshape_index=0  # int, float or FUNC

#............ AMPLITUDE RESPONSE ......................
attack_time_in_seconds=0   # int, float or FUNC
release_time_in_seconds=0  # int, float or FUNC

#======================================================
#***** LOW/HIGH SHELF EQ (TARGETS ONLY) ***************

LOW_SHELF_EQ_gain_in_decibels=-0
LOW_SHELF_EQ_frequency=100

HIGH_SHELF_EQ_gain_in_decibels=0
HIGH_SHELF_EQ_frequency=1000

#======================================================
#...MASTER INTERPOLATION CONTROLS ......................

# (Using (0-1) control functions, these 3 variables provide
#  master controls for time varying manipulation of the amp, 
#  freq and null phase values of a tone. With the respective P-field 
#  of a tone set to 1 (on), the controls interpolate 
#  between the source value and target setting. For example, a 
#  function moving between 0 and 1 would cause the frequency to 
#  shift from the orginal source value to the tone's target 
#  setting. Setting the P-field to .5 would start the shift halfway 
#  between source and target values. Negative
#  produce the same as positive values except that they
#  invert the control value in its 0-1 range before using it. 

#.....AMPLITUDE: 
TARGETS_amp_interpolation_control__0_to_1=1
#....FREQUENCY:  
TARGETS_freq_interpolation_control__0_to_1=1
#....NULL PHASE: 
TARGETS_nullphase_interpolation_control__0_to_1=1

#======================================================

#........Two more TUNING conrtols: ........................

# Most of you will never use these variables (A.S.)
# There are two tuning control values; the first for the tone
#  and the second global value that sets the tuning base for
#  all tones. The partial_tuning_control determines the current tuning
#  of the tone by mapping its 0-1 value or function into the difference
#  between the base and tone tuning settings.  For example, 
#  a function progressing from 0 to 1, placed in the tuning control 
#  parameter would case the tuning to move from the base value to the 
#  target tuning.

TARGETS_partial_tuning_control=1 
   # (0-1 control value)

TARGETS_partial_tuning_base_value=0
   # 0 = original source tuning,  1 = tuned to target partial
  # < 0 = source tuning deviation amplified, 
  # > 1 = inverse amplified source tuning deviation

#======================================================
TARGETS_shift_file=$SFDIR/chordmapfile

#..............SOURCE POINT FORMAT ....................

SOURCE_POINT_DATA_FORMAT=1
 # 0 = pitch specified in frequency used in chordmapfile
 # 1 = pitch specified in octave.pitchclass used in chordmapfile

#.............. SHIFT POINT FORMAT ....................

SHIFT_DATA_FORMAT=3
# 0 = frequency multiplier  # 1 = partial shift adder 
# 2 = map to frequency ) 3 = map to octave.pitchclass 
 # (A.S. : I don't know what this parameter does)

#======================================================
#***** DATA SET for CHORDMAPPER "TONES" **********
# Create 12 values in the data file for each output tone.
  # Line 1 below
# 1)  source freq. or oct.pch (depending on SOURCE_DATA_FORMAT above)
# 2) lowest partial number  : 1 or higher
# 3) output freq. or oct.pch (depending on SOURCE_DATA_FORMAT above
# 4) number of partials above lowest  (0 = all below Nyquist)
# 5) partial bandwidth as proportion  of fundamental frequency 
  #  higher values = brighter timbres
# 6) gain or attenuation in dB
  # Line 2 below : these parameters used less frequently
# 7) partial spacing as proportion of fundamental,negative = rejectmode 
# 8) decibel rolloff per octave  beginning with lowest partial.
# Postive rolloff produces gain.rejectmode measure begins from fundamental
# 9) Amplitude interpolation switch/degree.
#	0 = off, 1 = on, -1 = inverse on, .5 = half on
# 10) Frequency interpolation switch/degree.
#	0 = off, 1 = on, -1 = inverse on,  .5 = half on
# 11) Tuning factor
#	0 = original source tuning,  1 = tuned to target partial
#	< 0 = source tuning-deviation amplified, 
#	> 1 = inverse of amplified source-tuning-deviation
# 12) Null phase factor, 0 = off,  1 = null phase,  
#	.5 = halfway to null phase
#======================================================
# Create CHORDMAPPER FUNCTION HERE:

# 4 note harmonization at fs3, c4, f4, b4:
 echo \
7.11   1  7.06   0  .3  0    1 0  0 0 0 0 \
7.11   1  8.00   0  .3  -3   1 0  0 0 0 0 \
7.11   1  8.05   0  .3  -6   1 0  0 0 0 0 \
7.11   1  8.11   0  .3  -9  1 0  0 0 0 0 \
> $SFDIR/chordmapfile 
#*****************************************************
# An * (asterisk) at the beginning of a data set will comment it out.
# Do not include any blank lines, # or other characters in this function
# definition. Be very careful - -chordmapper is VERY picky

#........... RESCALE for floating point only ......
rescale_level_in_decibels=0
   # set to 1 to rescale to peak of input file; do not do this if input amplitude
   # is low
#********** AMPLITUDE STATISTICS ********************** 

print_amplitude_statistics_0_no__1_yes=1
amplitude_statistics_time_interval=.25

#====================================================
# COMMAND LINE SETUP -- OFFICE USE ONLY
#      (DO NOT WRITE BELOW THIS LINE)
#====================================================
# *****  TKLA CHANGES: ******* #
cd $SFDIR 
 SR=`sfsr $inputsf | awk '{print $1}'`
if ( ( [ `expr "$SR" \> "48000"` == 1 ] ) &&  ( [ "$windowsize" == "0" ] ) ) ; then 
       # if SR is > 48000 & windowsize is set to 0 increase default windowsize
  windowsize=`expr $FFT_length \* 4`
fi
# Determine if input soundfile is 24 bit. If so, compile 32 bit float outputs, then 
# convert to 24 bit ints
WORDSIZE=`/usr/local/bin/sfbits "${inputsf}" | awk '{print $1}'`  
if [ "$WORDSIZE" == "24" ] ; then rm -f pvcin ; 24tofloat $inputsf pvcin ; input_file=pvcin
  if  [ "$output_data_format" != 2 ] ; then
    output_file=pvcout  # temporary floating point output soundfile for 24-bit inputs
    rm -f  pvcout;  output_data_format=2
  else  # 24 bit input but float output requested
    output_file=$outputsf ; fi
else # not 24 bit input ; 16 bit int or 32 bit float input
    input_file=$inputsf ;  output_file=$outputsf
fi
#  ****** end of TKLA changes & additions ****** #
pvroutine=chordmapper 

PVFLAGS="\
\
-N$FFT_length \
-M$windowsize \
-w$window_type \
-D$frames_per_second \
-I$time_expansion_contraction_factor \
\
-C$output_channel \
-t$oscillator_resynthesis_threshold_in_dB \
\
-b$begintime \
-e$endtime \
\
-A$MASTER_gain_in_decibels \
\
-G$SOURCE_gain_in_dB \
-a$SOURCE_frequency_shift \
-P$SOURCE_pitch_transposition_in_semitones \
\
-q$TARGETS_frequency_shift \
-X$TARGETS_pitch_transposition_in_semitones \
-m$TARGETS_gain_in_dB \
-F$TARGETS_shift_file \
\
-L$release_time_in_seconds \
-l$attack_time_in_seconds \
\
\
-c$TARGETS_amp_interpolation_control__0_to_1 \
-d$TARGETS_freq_interpolation_control__0_to_1 \
\
-B$TARGETS_nullphase_interpolation_control__0_to_1 \
-K$TARGETS_partial_tuning_control \
-J$TARGETS_partial_tuning_base_value \
\
\
-Z$SHIFT_DATA_FORMAT \
-Q$SOURCE_POINT_DATA_FORMAT \
-W$spectrum_warpshape_index \
\
\
-S$LOW_SHELF_EQ_gain_in_decibels \
-U$LOW_SHELF_EQ_frequency \
\
-T$HIGH_SHELF_EQ_gain_in_decibels \
-V$HIGH_SHELF_EQ_frequency \
\
\
-_$output_data_format \
-=$rescale_level_in_decibels \
\
-p$print_amplitude_statistics_0_no__1_yes \
-i$amplitude_statistics_time_interval \
\
"
echo "\n$pvroutine $PVFLAGS  $input_file $output_file"

$pvroutine  $PVFLAGS $input_file $output_file   ; 
   # *****  TKLA CHANGES & ADDITIONS: ******* #
if ( ( [ "$WORDSIZE" == "24" ] ) &&  ( [ "$output_file" == "pvcout" ] ) ) ; then
      echo "Converting temporary float output file pvcout to $outputsf"
      echo " -------------------------------------------------------"
      floatto24 pvcout $outputsf 2> /dev/null  ; rm -f pvcin pvcout 
   echo " -------------------------------------------------------"
else
   echo " -------------------------------------------------------"
    echo -e -n "Output soundfile: " ;  sfinfo -s $outputsf
    if (  ( test -f "pvcin" ) ) ; then   rm -f pvcin ; fi
   echo " -------------------------------------------------------"
fi
#  ****** end of TKLA changes & additions ****** #

# If you have created any gen function files above delete them below:
rm $SFDIR/chordmapfile