ua1zbe
/
dsd-fme
mirror of https://github.com/lwvmobile/dsd-fme.git
1
0
Fork 0
Code Issues Packages Projects Releases Wiki Activity

Misc Code Organizing; audio_gainA Tweaks;

This commit is contained in:
lwvmobile 2024-03-21 23:39:43 -04:00
parent 3585e7e594
commit 4a63068ce1
2 changed files with 83 additions and 68 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

143
src/3.c
View File

@ -338,24 +338,27 @@ uint16_t q_abs_diff(const uint16_t v1, const uint16_t v2)
// #define LFC 100 // #define LFC 100
void LPFilter_Init(LPFilter *filter, float cutoffFreqHz, float sampleTimeS){ void LPFilter_Init(LPFilter *filter, float cutoffFreqHz, float sampleTimeS)
float RC=0.0; {
RC=1.0/(2*PI*cutoffFreqHz);
filter->coef[0]=sampleTimeS/(sampleTimeS+RC);
filter->coef[1]=RC/(sampleTimeS+RC);
filter->v_out[0]=0.0; float RC=0.0;
filter->v_out[1]=0.0; RC=1.0/(2*PI*cutoffFreqHz);
filter->coef[0]=sampleTimeS/(sampleTimeS+RC);
filter->coef[1]=RC/(sampleTimeS+RC);
filter->v_out[0]=0.0;
filter->v_out[1]=0.0;
} }
float LPFilter_Update(LPFilter *filter, float v_in){ float LPFilter_Update(LPFilter *filter, float v_in)
{
filter->v_out[1]=filter->v_out[0]; filter->v_out[1]=filter->v_out[0];
filter->v_out[0]=(filter->coef[0]*v_in) + (filter->coef[1]*filter->v_out[1]); filter->v_out[0]=(filter->coef[0]*v_in) + (filter->coef[1]*filter->v_out[1]);
return (filter->v_out[0]); return (filter->v_out[0]);
} }
@ -363,34 +366,34 @@ return (filter->v_out[0]);
/******************************************************************************************************** /********************************************************************************************************
* HIGH PASS FILTER * HIGH PASS FILTER
********************************************************************************************************/ ********************************************************************************************************/
void HPFilter_Init(HPFilter *filter, float cutoffFreqHz, float sampleTimeS){ void HPFilter_Init(HPFilter *filter, float cutoffFreqHz, float sampleTimeS)
float RC=0.0; {
RC=1.0/(2*PI*cutoffFreqHz);
filter->coef=RC/(sampleTimeS+RC); float RC=0.0;
RC=1.0/(2*PI*cutoffFreqHz);
filter->v_in[0]=0.0; filter->coef=RC/(sampleTimeS+RC);
filter->v_in[1]=0.0;
filter->v_out[0]=0.0; filter->v_in[0]=0.0;
filter->v_out[1]=0.0; filter->v_in[1]=0.0;
filter->v_out[0]=0.0;
filter->v_out[1]=0.0;
} }
float HPFilter_Update(HPFilter *filter, float v_in){ float HPFilter_Update(HPFilter *filter, float v_in)
{
filter->v_in[1]=filter->v_in[0];
filter->v_in[0]=v_in;
filter->v_in[1]=filter->v_in[0];
filter->v_in[0]=v_in;
filter->v_out[1]=filter->v_out[0]; filter->v_out[1]=filter->v_out[0];
filter->v_out[0]=filter->coef * (filter->v_in[0] - filter->v_in[1]+filter->v_out[1]);
filter->v_out[0]=filter->coef * (filter->v_in[0] - filter->v_in[1]+filter->v_out[1]); return (filter->v_out[0]);
return (filter->v_out[0]);
} }
/******************************************************************************************************** /********************************************************************************************************
@ -398,19 +401,25 @@ return (filter->v_out[0]);
********************************************************************************************************/ ********************************************************************************************************/
void PBFilter_Init(PBFilter *filter, float HPF_cutoffFreqHz, float LPF_cutoffFreqHz, float sampleTimeS){ void PBFilter_Init(PBFilter *filter, float HPF_cutoffFreqHz, float LPF_cutoffFreqHz, float sampleTimeS)
LPFilter_Init(&filter->lpf, LPF_cutoffFreqHz, sampleTimeS); {
HPFilter_Init(&filter->hpf, HPF_cutoffFreqHz, sampleTimeS);
filter->out_in=0.0; LPFilter_Init(&filter->lpf, LPF_cutoffFreqHz, sampleTimeS);
HPFilter_Init(&filter->hpf, HPF_cutoffFreqHz, sampleTimeS);
filter->out_in=0.0;
} }
float PBFilter_Update(PBFilter *filter, float v_in){ float PBFilter_Update(PBFilter *filter, float v_in)
filter->out_in=HPFilter_Update(&filter->hpf, v_in); {
filter->out_in=LPFilter_Update(&filter->lpf, filter->out_in); filter->out_in=HPFilter_Update(&filter->hpf, v_in);
filter->out_in=LPFilter_Update(&filter->lpf, filter->out_in);
return (filter->out_in);
return (filter->out_in);
} }
/******************************************************************************************************** /********************************************************************************************************
@ -418,44 +427,49 @@ return (filter->out_in);
********************************************************************************************************/ ********************************************************************************************************/
void NOTCHFilter_Init(NOTCHFilter *filter, float centerFreqHz, float notchWidthHz, float sampleTimeS){ void NOTCHFilter_Init(NOTCHFilter *filter, float centerFreqHz, float notchWidthHz, float sampleTimeS)
//filter frequency to angular (rad/s) {
float w0_rps=2.0 * PI *centerFreqHz;
float ww_rps=2.0 * PI *notchWidthHz;
//pre warp center frequency //filter frequency to angular (rad/s)
float w0_pw_rps=(2.0/sampleTimeS) * tanf(0.5 * w0_rps * sampleTimeS); float w0_rps=2.0 * PI *centerFreqHz;
float ww_rps=2.0 * PI *notchWidthHz;
//computing filter coefficients //pre warp center frequency
float w0_pw_rps=(2.0/sampleTimeS) * tanf(0.5 * w0_rps * sampleTimeS);
filter->alpha=4.0 + w0_rps*w0_pw_rps*sampleTimeS*sampleTimeS; //computing filter coefficients
filter->beta=2.0*ww_rps*sampleTimeS;
//clearing input and output buffers filter->alpha=4.0 + w0_rps*w0_pw_rps*sampleTimeS*sampleTimeS;
filter->beta=2.0*ww_rps*sampleTimeS;
for (uint8_t n=0; n<3; n++){ //clearing input and output buffers
filter->vin[n]=0;
filter->vout[n]=0; for (uint8_t n=0; n<3; n++)
} {
filter->vin[n]=0;
filter->vout[n]=0;
}
} }
float NOTCHFilter_Update(NOTCHFilter *filter, float vin){ float NOTCHFilter_Update(NOTCHFilter *filter, float vin)
//shifting samples {
filter->vin[2]=filter->vin[1];
filter->vin[1]=filter->vin[0];
filter->vout[2]=filter->vout[1]; //shifting samples
filter->vout[1]=filter->vout[0]; filter->vin[2]=filter->vin[1];
filter->vin[1]=filter->vin[0];
filter->vin[0]=vin; filter->vout[2]=filter->vout[1];
filter->vout[1]=filter->vout[0];
//compute new output filter->vin[0]=vin;
filter->vout[0]=(filter->alpha*filter->vin[0] + 2.0 *(filter->alpha -8.0)*filter->vin[1] + filter->alpha*filter->vin[2]
-(2.0f*(filter->alpha-8.0)*filter->vout[1]+(filter->alpha-filter->beta)*filter->vout[2]))/(filter->alpha+filter->beta); //compute new output
filter->vout[0]=(filter->alpha*filter->vin[0] + 2.0 *(filter->alpha -8.0)*filter->vin[1] + filter->alpha*filter->vin[2]
-(2.0f*(filter->alpha-8.0)*filter->vout[1]+(filter->alpha-filter->beta)*filter->vout[2]))/(filter->alpha+filter->beta);
return (filter->vout[0]); return (filter->vout[0]);
} }
void init_audio_filters (dsd_state * state) void init_audio_filters (dsd_state * state)
@ -473,11 +487,12 @@ void init_audio_filters (dsd_state * state)
//PBFilter_Init(PBFilter *filter, float HPF_cutoffFreqHz, float LPF_cutoffFreqHz, float sampleTimeS); //PBFilter_Init(PBFilter *filter, float HPF_cutoffFreqHz, float LPF_cutoffFreqHz, float sampleTimeS);
//NOTCHFilter_Init(NOTCHFilter *filter, float centerFreqHz, float notchWidthHz, float sampleTimeS); //NOTCHFilter_Init(NOTCHFilter *filter, float centerFreqHz, float notchWidthHz, float sampleTimeS);
//TODO: Make PBFilter initiazation values based on rtl bandwidth values? //NOTE: PBFilter_init also inits a LPF and HPF, but on another set of filters, might be worth
//testing just using the PBFilter by itself and see how it does without the hpf used
//passband filter working (seems to be), notch filter unsure which values to use //passband filter working (seems to be), notch filter unsure which values to use, doesn't have any appreciable affect when used as is
PBFilter_Init(&state->PBF, 8000, 12000, (float)1/(float)1536000); //RTL Sampling at 1536000 S/s. PBFilter_Init(&state->PBF, 8000, 12000, (float)1/(float)1536000); //RTL Sampling at 1536000 S/s.
NOTCHFilter_Init(&state->NF, 1000, 4000, (float)1/(float)1536000); //this one probably isn't set up correctly NOTCHFilter_Init(&state->NF, 1000, 4000, (float)1/(float)1536000);
} }

8
src/dsd_main.c
View File

@ -583,7 +583,7 @@ initOpts (dsd_opts * opts)
opts->mbe_out_fR = NULL; //second slot on a TDMA system opts->mbe_out_fR = NULL; //second slot on a TDMA system
opts->audio_gain = 0; opts->audio_gain = 0;
opts->audio_gainR = 0; opts->audio_gainR = 0;
opts->audio_gainA = 50.0f; //scale of 1 - 100 opts->audio_gainA = 30.0f; //scale of 1 - 100
opts->audio_out = 1; opts->audio_out = 1;
opts->wav_out_file[0] = 0; opts->wav_out_file[0] = 0;
opts->wav_out_fileR[0] = 0; opts->wav_out_fileR[0] = 0;
@ -2177,14 +2177,14 @@ main (int argc, char **argv)
opts.dmr_stereo = 0; opts.dmr_stereo = 0;
opts.dmr_mono = 0; opts.dmr_mono = 0;
state.dmr_stereo = 0; state.dmr_stereo = 0;
// opts.setmod_bw = 12500; // opts.setmod_bw = 16000;
sprintf (opts.output_name, "EDACS/PV"); sprintf (opts.output_name, "EDACS/PV");
fprintf (stderr,"Setting symbol rate to 9600 / second\n"); fprintf (stderr,"Setting symbol rate to 9600 / second\n");
fprintf (stderr,"Decoding only ProVoice frames.\n"); fprintf (stderr,"Decoding only ProVoice frames.\n");
fprintf (stderr,"EDACS Analog Voice Channels are Experimental.\n"); fprintf (stderr,"EDACS Analog Voice Channels are Experimental.\n");
//rtl specific tweaks //misc tweaks
opts.rtl_bandwidth = 24; opts.rtl_bandwidth = 24;
// opts.rtl_gain_value = 36; opts.audio_gainA = 50.0f;
} }
else if (optarg[0] == '1') else if (optarg[0] == '1')
{ {