ECMC PVC example file plainpv6-1 Example PVC script used to create /sflib/x soundfile >> plainpv6-1.wav < # Example PVC script used to create /sflib/x soundfile >> plainpv6-1 << # ECMC pvc example "plainpv6-1," which illustrates use of time # varying functions applied to both pitch and spectral warp index # of an oboe tone. # For details on this example, type : pvcex plainpv6 # This script produced the first tone in /sflibx example "plainpv6 #****************************************************** #****** script to run PLAINPV (ECMC version) ********** #****************************************************** #******** INPUT & OUTPUT SOUNDFILES ************* inputsf=/sflib/wind/oboe.d4.wav outputsf=$SFDIR/plainpv6-1.wav # ********************* # # 2 function files created to control time varying frequency spectrum and pitch: # gen3 generates amp values & linear connections at equally spaced time points # gen3 -Llength v1 v2 ... vN gen3 -L1000 5 5 -5 -5 5 > $SFDIR/spectrumfunc # 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 gen4 -L1000 0 -2 1 \ .25 -2 1 \ .5 4 1 \ .8 4 1 \ 1 2 > $SFDIR/pitchfunc ##### 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=300 # generally 200, occasionally 400 or 600 when time stretching #====================================================== time_expansion_contraction_factor=1.5 #**** OUTPUT CHANNEL(S) ******************************* resynthesis_channel_1_to_max__0_all=0 # (channels are numbered from 1-maximum) # (0 = all channels) oscillator_resynthesis_threshold_in_dB=-80 #****** RESYNTHESIS PARAMETERS ************************** frequency_shift=0 # detune partials, in herz; int, float or FUNC gain_in_decibels=-0 # in dB, int, float or FUNC pitch_transposition_in_semitones=$SFDIR/pitchfunc # int, float or FUNC #************ENVELOPE RESPONSE ************************ release_time_in_seconds=0 # int, float or FUNC attack_time_in_seconds=0 # int, float or FUNC #********** SPECTRUM WARPSHAPE **************************** spectrum_warpshape_index=$SFDIR/spectrumfunc # int, float or FUNC #******** BRICKWALL BAND OR REJECT FILTER WINDOW ************ # (-1 selects respective lowest or highest) FILTER_TYPE__0_bandpass__1_bandreject=1 BRICKWALL_FILTER_window_low_frequency=500 BRICKWALL_FILTER_window_high_frequency=1200 #*************** LOW/HIGH SHELF EQ ************************* LOW_SHELF_EQ_gain_in_decibels=0 LOW_SHELF_EQ_frequency=1000 HIGH_SHELF_EQ_gain_in_decibels=-0 HIGH_SHELF_EQ_frequency=1100 #******** TERMINAL DISPLAY **************************** TERMINAL_DISPLAY_0_off__1_phase__2_magnitude__3_both=0 TERMINAL_DISPLAY_low_bin=10 TERMINAL_DISPLAY_high_bin=30 TERMINAL_DISPLAY_number_of_frames=1 #====================================================== #........... 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=`/usr/local/bin/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=$outputsf fi # ****** end of TKLA changes & additions ****** # pvroutine=plainpv PVFLAGS="\ \ -N$FFT_length \ -M$windowsize \ -D$frames_per_second \ -I$time_expansion_contraction_factor \ \ -a$frequency_shift \ -P$pitch_transposition_in_semitones \ -A$gain_in_decibels \ \ -C$resynthesis_channel_1_to_max__0_all \ -t$oscillator_resynthesis_threshold_in_dB \ \ -b$begintime \ -e$endtime \ \ -H$LOW_SHELF_EQ_gain_in_decibels \ -m$LOW_SHELF_EQ_frequency \ \ -X$HIGH_SHELF_EQ_gain_in_decibels \ -R$HIGH_SHELF_EQ_frequency \ \ -L$release_time_in_seconds \ -l$attack_time_in_seconds \ \ -W$spectrum_warpshape_index \ \ -T$FILTER_TYPE__0_bandpass__1_bandreject \ -f$BRICKWALL_FILTER_window_low_frequency \ -F$BRICKWALL_FILTER_window_high_frequency \ \ -S$TERMINAL_DISPLAY_0_off__1_phase__2_magnitude__3_both \ -u$TERMINAL_DISPLAY_low_bin \ -U$TERMINAL_DISPLAY_high_bin \ -n$TERMINAL_DISPLAY_number_of_frames \ \ -p$print_amplitude_statistics_0_no__1_yes \ -i$amplitude_statistics_time_interval \ " echo "\n\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/spectrumfunc $SFDIR/pitchfunc