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PiFmRds
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switched the filter to a much better sounding one 

previouse filter was described by:
y[n] = bx[n]-bx[n-1]+ay[n-1]
filter is now described by:
y(n) = x(n) - a x(n-1)
with 'a' = old 'b'
even i have no idea if that coefficient 'a' is selected correctly, it sounds much better than before
This commit is contained in:
Florian Schöck 2016-03-12 18:16:45 +01:00
parent e0cc4fe10a
commit 2cc5d96e62
1 changed files with 8 additions and 10 deletions
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18
src/fm_mpx.c
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@ -66,11 +66,13 @@ float fir_buffer_stereo[FIR_SIZE] = {0};
int fir_index = 0;
int channels;
//3.185kHz for Europe, 2.120kHz for US
const float PREEMPHASIS_US = 2120;
const float PREEMPHASIS_EU = 3185;
float *last_buffer_val;
float preemphasis_corner_freq;
float preemphasis_prewarp;
float preemphasis_coefficient_a;
float preemphasis_coefficient_b;
float preemphasis_coefficient;
SNDFILE *inf;
@ -126,11 +128,10 @@ int fm_mpx_open(char *filename, size_t len) {
last_buffer_val = (float*) malloc(sizeof(float)*channels);
for(int i=0;i<channels;i++) last_buffer_val[i] = 0;
preemphasis_corner_freq = 2120; //3.185kHz for Europe, 2,120kHz for US
preemphasis_corner_freq = PREEMPHASIS_EU;
preemphasis_prewarp = tan(PI*preemphasis_corner_freq/in_samplerate);
preemphasis_coefficient_a = (1.0 - preemphasis_prewarp)/(1.0 + preemphasis_prewarp);
preemphasis_coefficient_b = (1.0 + preemphasis_coefficient_a)/2.0;
printf("Created preemphasis with \n%lf \n%lf \n%lf \n%lf", preemphasis_corner_freq, preemphasis_prewarp, preemphasis_coefficient_a, preemphasis_coefficient_b);
preemphasis_coefficient = (1.0 + (1.0 - preemphasis_prewarp)/(1.0 + preemphasis_prewarp))/2.0;
printf("Created preemphasis with with cutoff at %.1f Hz\n", preemphasis_corner_freq);
// Create the low-pass FIR filter
@ -204,11 +205,8 @@ int fm_mpx_get_samples(float *mpx_buffer) {
for(k=0;k<audio_len;k+=channels) {
for(l=0;l<channels;l++) {
tmp = audio_buffer[k+l];
audio_buffer[k+l] = preemphasis_coefficient_b*audio_buffer[k+l] -
preemphasis_coefficient_b*last_buffer_val[l] +
preemphasis_coefficient_a*last_buffer_val[l];
audio_buffer[k+l] = audio_buffer[k+l] - preemphasis_coefficient*last_buffer_val[l];
last_buffer_val[l] = tmp;
}
}