I wrote what you said (it's not possible to attach again a file so here below
it is the code) and i receive this error:
"oscc_cr_sr_autonomous1-monoed_no_vector4.cpp:144:44: error: cannot convert
'float' to 'float*' in assignment buffer = feedforward(buffer + i, i,a0,a1,1);"
the 144:44 is the place of the number of samples of the delay, here is simply
1. Don't understand what's wrong."
It is the place of the number 1, the last place:
buffer = feedforward(buffer + i ,i ,a0 ,a1 ,1 );
#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#include <sndfile.h>
//This is me on windows:
//g++ -o a oscc_cr_sr_autonomous1-monoed_no_vector4.cpp libsndfile-1.dll
//a oscc_cr_sr_autonomous1-monoed_no_vector2.wav 0.5 300 10
//g++ oscc_cr_sr_autonomous1.cpp -lsndfile
//# oscillator sndfile.wav amp freq(hz) dur(s)
//./a.out ross_oscc_1.wav 0.5 300 10
///////////////////////////LIBSNDFILE//////////////////////////////
///functions to open, write, and close a sound file.
SNDFILE *soundout_open(char* name, int chans, float sr){
// initialise the SF_INFO structure
SF_INFO info;
info.samplerate = (int) sr;
info.channels = chans;
info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
// open the file
return sf_open(name, SFM_WRITE, &info);
}
//vector where are written the values
int soundout(SNDFILE *psf_out, float *buffer, int vecsize){
return (int) sf_writef_float(psf_out, buffer, vecsize);
}
void soundout_close(SNDFILE *psf_out){
sf_close(psf_out);
}
/////////////////////OSCILLATOR////////////////////////////////////////////////
float oscc(float *output, float amp, float freq, float *table, float *index,
/*float phase,*/ int tablength,float sr)
{
float incr = freq*tablength/sr, frac, fracsq, fracc; // increment
float pos,a,b,c,d, tmp;
// cubic interpolation
frac = *index - (int)*index;
a = (int) *index > 0 ? table[(int)(*index)-1] : table[tablength-1];
b = table[(int)(*index)];
c = table[(int)(*index)+1];
d = table[(int)(*index)+2];
fracsq = frac*frac;
fracc = frac*fracsq;
//int i;
*output = amp*(fracc*(d-a)/8 + fracsq*(c-a)/4 + frac*(b-a) + a); //output[i]
*index += incr;
while(*index >= tablength) *index -= (tablength);
while(*index < 0) *index += tablength;
return *output;
}
float feedforward(const float * in, int index, double a0, double a1, int delay){
if (index >= delay)
return a0 * in [index] + a1 * in [index - delay];
else
return a0 * in [index];
}
////////////////////////////////TABLE /////////////////////////////////////
const double pi = 4*atan(1.);
float* sinus_table(int length, float phase){
float *table = new float[length+2];
phase *= (float)pi*2;
for(int n=0; n < length+2; n++)
table[n] = (float) cos(phase+n*2*pi/length);
return table;
}
float *buzz_table(){
#define BUFFER_SIZE 44100
float *wtab = new float[BUFFER_SIZE];
// initialize buffer to zero, since we'll be summing into it
for (int i=0; i < BUFFER_SIZE; ++i)
wtab[i] = 0.0f;
const int num_harmonics = 50; // compute this as needed, highest harmonic
// must not have period smaller than 2 samples in the buffer.
// normalize the output amplitude to 1.0...
double amp_scale = 1.0 / num_harmonics;
// outer loop: iterate through harmonics from 1 ... num_harmonics
for (int j=1; j <= num_harmonics; ++j) {
// inner loop: compute the jth harmonic, sum into buffer
for (int i=0; i < BUFFER_SIZE; ++i)
wtab[i] += cos (i * j * (2.0 * pi / (double)BUFFER_SIZE)) *
amp_scale;
}
return wtab;
}
int main(int argc, char** argv){
// NOTE: all of these could be variables instead of #defines
#define CHANS 1
#define SR 44100.0
#define WAVE_SIZE 44100
SNDFILE *psf;
float *buffer;
int bufs;
float amp, freq, *wave, cs, ndx=0,*wtab;
if(argc != 5){
printf("usage: oscillator sndfile.wav amp freq(hz) dur(s)\n");
return 1;
}
amp = (float) atof(argv[2]);
freq = (float) atof(argv[3]);
bufs = (int) (atof(argv[4])*SR);
// allocate buffer & table memory
buffer = new float[bufs];
wave = buzz_table();//buzz_table();//sinus_table(WAVE_SIZE, 0.0);
// now we open the file
if(!(psf = soundout_open(argv[1], CHANS, SR))){
printf("error opening output file\n");
exit(-1);
}
double a0 = 0.5, a1 = 0.5;
for(int i=0; i < bufs; i++){
oscc(buffer, amp, freq, wave, &ndx, WAVE_SIZE, SR);
//feedforward(const float * in, int index, double a0, double a1, int
delay){
buffer = feedforward(buffer + i, i,a0,a1,1);
soundout(psf, buffer, 1);
}
// close file & free memory
soundout_close(psf);
delete[] buffer;
delete[] wave;
return 0;
}
________________________________
From: music-dsp <[email protected]> on behalf of Stefano D'Angelo
<[email protected]>
Sent: Tuesday, November 12, 2024 6:29 AM
To: [email protected] <[email protected]>
Subject: Re: [MUSIC-DSP] feedforward filter
Il 12/11/24 07:23, Stefano D'Angelo ha scritto:
Il 12/11/24 05:28, Pablo Frank ha scritto:
Hi all, in the .cpp file attached there is an oscillator that works ok, and a
feedforward filter (line 64) that i don't know how to insert between the
oscillator (line 142) and the soundout function (line 143) to get it working.
I'm sure it's a simple programming matter that most of you can solve it
immediatelly. Thanks in advance for any help,
Probably something like:
feedforward(&buffer, i, a0, a1, delay);
where a0 and a1 are the filter coefficients and delay is the distance between
current input sample and delayed input tap in number of samples.
Sorry, bad copy/paste
buffer = feedforward(buffer + i, i, a0, a1, delay);
However, this will never work and actually crash as buffer is 1 sample wide.
You need to either allocate it as
buffer = new float[bufs];
or you need to implement and operate on a circular buffer...
Stefano
