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fallback, it doesnt work

This commit is contained in:
Kuba 2023-11-07 12:53:13 +00:00
parent 0a8820ed97
commit b4ef6f4d32
3 changed files with 49 additions and 44 deletions
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85
src/pifmrds/fm_mpx.c
View File

@ -65,7 +65,6 @@ float fir_buffer_left[FIR_TAPS] = {0};
float fir_buffer_right[FIR_TAPS] = {0};
int fir_index = 0;
int channels;
int in_samplerate;
float left_max=1, right_max=1; // start compressor with low gain
SNDFILE *inf;
@ -82,7 +81,7 @@ float *alloc_empty_buffer(size_t length) {
}
int fm_mpx_open(char *filename, size_t len, int raw, int rawSampleRate, int rawChannels) {
int fm_mpx_open(char *filename, size_t len, int raw, double preemphasis, int rawSampleRate, int rawChannels, float cutoff_freq) {
length = len;
raw_ = raw;
@ -113,7 +112,7 @@ int fm_mpx_open(char *filename, size_t len, int raw, int rawSampleRate, int rawC
}
}
in_samplerate = sfinfo.samplerate;
int in_samplerate = sfinfo.samplerate;
downsample_factor = 228000. / in_samplerate;
printf("Input: %d Hz, upsampling factor: %.2f\n", in_samplerate, downsample_factor);
@ -125,6 +124,48 @@ int fm_mpx_open(char *filename, size_t len, int raw, int rawSampleRate, int rawC
printf("1 channel, monophonic operation.\n");
}
// Choose a cutoff frequency for the low-pass FIR filter
if(in_samplerate/2 < cutoff_freq) cutoff_freq = in_samplerate/2 * .8;
// Create the low-pass FIR filter, with pre-emphasis
double window, firlowpass, firpreemph , sincpos;
// IIR pre-emphasis filter
// Reference material: http://jontio.zapto.org/hda1/preempiir.pdf
double tau=preemphasis;
double delta=1/(2*PI*20000);//double delta=1.96e-6;
double taup, deltap, bp, ap, a0, a1, b1;
taup=1.0/(2.0*(in_samplerate*FIR_PHASES))/tan( 1.0/(2*tau*(in_samplerate*FIR_PHASES) ));
deltap=1.0/(2.0*(in_samplerate*FIR_PHASES))/tan( 1.0/(2*delta*(in_samplerate*FIR_PHASES) ));
bp=sqrt( -taup*taup + sqrt(taup*taup*taup*taup + 8.0*taup*taup*deltap*deltap) ) / 2.0 ;
ap=sqrt( 2*bp*bp + taup*taup );
a0=( 2.0*ap + 1.0/(in_samplerate*FIR_PHASES) )/(2.0*bp + 1.0/(in_samplerate*FIR_PHASES) );
// a1=(-2.0*ap + 1/(in_samplerate*FIR_PHASES) )/(2.0*bp + 1/(in_samplerate*FIR_PHASES) ); //ORI
// b1=( 2.0*bp + 1/(in_samplerate*FIR_PHASES) )/(2.0*bp + 1/(in_samplerate*FIR_PHASES) ); //ORI
a1=(-2.0*ap + 1.0/(in_samplerate*FIR_PHASES) )/(2.0*bp + 1.0/(in_samplerate*FIR_PHASES) );
b1=( 2.0*bp - 1.0/(in_samplerate*FIR_PHASES) )/(2.0*bp + 1.0/(in_samplerate*FIR_PHASES) );
double x=0,y=0;
for(int i=0; i<FIR_TAPS; i++) {
for(int j=0; j<FIR_PHASES; j++) {
int mi=i*FIR_PHASES + j+1;// match indexing of Matlab script
sincpos = (mi)-(((FIR_TAPS*FIR_PHASES)+1.0)/2.0); // offset by 0.5 so sincpos!=0 (causes NaN x/0 )
//printf("%d=%f \n",mi ,sincpos);
firlowpass = sin(2 * PI * cutoff_freq * sincpos / (in_samplerate*FIR_PHASES) ) / (PI * sincpos) ;
y=a0*firlowpass + a1*x + b1*y ; // Find the combined impulse response
x=firlowpass; // of FIR low-pass and IIR pre-emphasis
firpreemph=y; // y could be replaced by firpreemph but this
// matches the example in the reference material
window = (.54 - .46 * cos(2*PI * (mi) / (double) FIR_TAPS*FIR_PHASES )) ; // Hamming window
low_pass_fir[j][i] = firpreemph * window;
}
}
printf("Created low-pass FIR filter for audio channels, with cutoff at %.1f Hz\n", cutoff_freq);
if( 0 )
{
printf("f = [ ");
@ -152,43 +193,7 @@ int fm_mpx_open(char *filename, size_t len, int raw, int rawSampleRate, int rawC
// samples provided by this function are in 0..10: they need to be divided by
// 10 after.
int fm_mpx_get_samples(float *mpx_buffer, int drds, float compressor_decay, float compressor_attack, float compressor_max_gain_recip, int disablestereo, float gain, int enablecompressor, int rds_ct_enabled, float rds_volume, int paused, double preemphasis, float cutoff_freq) {
// Choose a cutoff frequency for the low-pass FIR filter
if(in_samplerate/2 < cutoff_freq) cutoff_freq = in_samplerate/2 * .8;
// Create the low-pass FIR filter, with pre-emphasis
double window, firlowpass, firpreemph , sincpos;
// IIR pre-emphasis filter
// Reference material: http://jontio.zapto.org/hda1/preempiir.pdf
double tau=preemphasis;
double delta=1/(2*PI*20000);//double delta=1.96e-6;
double taup, deltap, bp, ap, a0, a1, b1;
taup=1.0/(2.0*(in_samplerate*FIR_PHASES))/tan( 1.0/(2*tau*(in_samplerate*FIR_PHASES) ));
deltap=1.0/(2.0*(in_samplerate*FIR_PHASES))/tan( 1.0/(2*delta*(in_samplerate*FIR_PHASES) ));
bp=sqrt( -taup*taup + sqrt(taup*taup*taup*taup + 8.0*taup*taup*deltap*deltap) ) / 2.0 ;
ap=sqrt( 2*bp*bp + taup*taup );
a0=( 2.0*ap + 1.0/(in_samplerate*FIR_PHASES) )/(2.0*bp + 1.0/(in_samplerate*FIR_PHASES) );
// a1=(-2.0*ap + 1/(in_samplerate*FIR_PHASES) )/(2.0*bp + 1/(in_samplerate*FIR_PHASES) ); //ORI
// b1=( 2.0*bp + 1/(in_samplerate*FIR_PHASES) )/(2.0*bp + 1/(in_samplerate*FIR_PHASES) ); //ORI
a1=(-2.0*ap + 1.0/(in_samplerate*FIR_PHASES) )/(2.0*bp + 1.0/(in_samplerate*FIR_PHASES) );
b1=( 2.0*bp - 1.0/(in_samplerate*FIR_PHASES) )/(2.0*bp + 1.0/(in_samplerate*FIR_PHASES) );
double x=0,y=0;
for(int i=0; i<FIR_TAPS; i++) {
for(int j=0; j<FIR_PHASES; j++) {
int mi=i*FIR_PHASES + j+1;// match indexing of Matlab script
sincpos = (mi)-(((FIR_TAPS*FIR_PHASES)+1.0)/2.0); // offset by 0.5 so sincpos!=0 (causes NaN x/0 )
//printf("%d=%f \n",mi ,sincpos);
firlowpass = sin(2 * PI * cutoff_freq * sincpos / (in_samplerate*FIR_PHASES) ) / (PI * sincpos) ;
y=a0*firlowpass + a1*x + b1*y ; // Find the combined impulse response
x=firlowpass; // of FIR low-pass and IIR pre-emphasis
firpreemph=y; // y could be replaced by firpreemph but this
// matches the example in the reference material
window = (.54 - .46 * cos(2*PI * (mi) / (double) FIR_TAPS*FIR_PHASES )) ; // Hamming window
low_pass_fir[j][i] = firpreemph * window;
}
}
int fm_mpx_get_samples(float *mpx_buffer, int drds, float compressor_decay, float compressor_attack, float compressor_max_gain_recip, int disablestereo, float gain, int enablecompressor, int rds_ct_enabled, float rds_volume, int paused) {
int stereo_capable = (channels > 1) && (!disablestereo); //chatgpt
if(!drds) get_rds_samples(mpx_buffer, length, stereo_capable, rds_ct_enabled, rds_volume);

4
src/pifmrds/fm_mpx.h
View File

@ -21,6 +21,6 @@
along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
extern int fm_mpx_open(char *filename, size_t len, int raw, int rawSampleRate, int rawChannels);
extern int fm_mpx_get_samples(float *mpx_buffer, int drds, float compressor_decay, float compressor_attack, float compressor_max_gain_recip, int disablestereo, float gain, int enablecompressor, int rds_ct_enabled, float rds_volume, int paused, double preemphasis, float cutoff_freq);
extern int fm_mpx_open(char *filename, size_t len, int raw, double preemphasis, int rawSampleRate, int rawChannels, float cutoff_freq);
extern int fm_mpx_get_samples(float *mpx_buffer, int drds, float compressor_decay, float compressor_attack, float compressor_max_gain_recip, int disablestereo, float gain, int enablecompressor, int rds_ct_enabled, float rds_volume, int paused);
extern int fm_mpx_close();

4
src/pifmrds/pi_fm_rds.cpp
View File

@ -117,7 +117,7 @@ int tx(uint32_t carrier_freq, char *audio_file, uint16_t pi, char *ps, char *rt,
int disablestereo = 0;
// Initialize the baseband generator
if(fm_mpx_open(audio_file, DATA_SIZE, raw, rawSampleRate, rawChannels) < 0) return 1;
if(fm_mpx_open(audio_file, DATA_SIZE, raw, preemp, rawSampleRate, rawChannels, cutoff_freq) < 0) return 1;
// Initialize the RDS modulator
char myps[9] = {0};
@ -223,7 +223,7 @@ int tx(uint32_t carrier_freq, char *audio_file, uint16_t pi, char *ps, char *rt,
}
}
if( fm_mpx_get_samples(data, drds, compressor_decay, compressor_attack, compressor_max_gain_recip, disablestereo, gaim, enablecompressor, rds_ct_enabled, rds_volume, 0, cutoff_freq, preemp) < 0 ) {
if( fm_mpx_get_samples(data, drds, compressor_decay, compressor_attack, compressor_max_gain_recip, disablestereo, gaim, enablecompressor, rds_ct_enabled, rds_volume, 0) < 0 ) {
terminate(0);
}
data_len = DATA_SIZE;