mirror of https://github.com/lwvmobile/dsd-fme.git
New Analog Audio Filters WIP;
This commit is contained in:
lwvmobile
parent
df28ebe573
commit
2b7de90474
|
|
@ -86,6 +86,35 @@
|
|||
|
||||
extern volatile uint8_t exitflag; //fix for issue #136
|
||||
|
||||
//new audio filter stuff from: https://github.com/NedSimao/FilteringLibrary
|
||||
typedef struct {
|
||||
float coef[2];
|
||||
float v_out[2];
|
||||
}LPFilter;
|
||||
|
||||
typedef struct {
|
||||
float coef;
|
||||
float v_out[2];
|
||||
float v_in[2];
|
||||
|
||||
}HPFilter;
|
||||
|
||||
typedef struct {
|
||||
LPFilter lpf;
|
||||
HPFilter hpf;
|
||||
float out_in;
|
||||
}PBFilter;
|
||||
|
||||
typedef struct {
|
||||
float alpha;
|
||||
float beta;
|
||||
|
||||
float vin[3];
|
||||
float vout[3];
|
||||
|
||||
}NOTCHFilter;
|
||||
//end new filters
|
||||
|
||||
//group csv import struct
|
||||
typedef struct
|
||||
{
|
||||
|
|
@ -285,6 +314,7 @@ typedef struct
|
|||
int rtl_started;
|
||||
long int rtl_rms;
|
||||
int monitor_input_audio;
|
||||
int analog_only;
|
||||
int pulse_raw_rate_in;
|
||||
int pulse_raw_rate_out;
|
||||
int pulse_digi_rate_in;
|
||||
|
|
@ -630,6 +660,13 @@ typedef struct
|
|||
|
||||
dPMRVoiceFS2Frame_t dPMRVoiceFS2Frame;
|
||||
|
||||
//new audio filter structs
|
||||
LPFilter RCFilter;
|
||||
HPFilter HRCFilter;
|
||||
LPFilter RCFilter8;
|
||||
HPFilter HRCFilter8;
|
||||
// PBFilter PBF;
|
||||
|
||||
char dpmr_caller_id[20];
|
||||
char dpmr_target_id[20];
|
||||
|
||||
|
|
@ -1285,12 +1322,22 @@ void eot_cc(dsd_opts * opts, dsd_state * state); //end of TX return to CC
|
|||
//Generic Tuning Functions
|
||||
void return_to_cc (dsd_opts * opts, dsd_state * state);
|
||||
|
||||
//misc generic audio filtering for analog at 48k/1
|
||||
//misc audio filtering for analog
|
||||
long int raw_rms(short *samples, int len, int step);
|
||||
void analog_deemph_filter(short * input, int len);
|
||||
void analog_preemph_filter(short * input, int len);
|
||||
void analog_dc_block_filter(short * input, int len);
|
||||
void analog_clipping_filter(short * input, int len);
|
||||
void init_audio_filters(dsd_state * state);
|
||||
void lpf(dsd_state * state, short * input, int len);
|
||||
void hpf(dsd_state * state, short * input, int len);
|
||||
void lpf8(dsd_state * state, short * input, int len);
|
||||
void hpf8(dsd_state * state, short * input, int len);
|
||||
//from: https://github.com/NedSimao/FilteringLibrary
|
||||
void LPFilter_Init(LPFilter *filter, float cutoffFreqHz, float sampleTimeS);
|
||||
float LPFilter_Update(LPFilter *filter, float v_in);
|
||||
void HPFilter_Init(HPFilter *filter, float cutoffFreqHz, float sampleTimeS);
|
||||
float HPFilter_Update(HPFilter *filter, float v_in);
|
||||
void PBFilter_Init(PBFilter *filter, float HPF_cutoffFreqHz, float LPF_cutoffFreqHz, float sampleTimeS);
|
||||
float PBFilter_Update(PBFilter *filter, float v_in);
|
||||
void NOTCHFilter_Init(NOTCHFilter *filter, float centerFreqHz, float notchWidthHz, float sampleTimeS);
|
||||
float NOTCHFilter_Update(NOTCHFilter *filter, float vin);
|
||||
|
||||
//csv imports
|
||||
int csvGroupImport(dsd_opts * opts, dsd_state * state);
|
||||
|
|
|
|||
211
src/3.c
211
src/3.c
|
|
@ -321,4 +321,213 @@ uint16_t q_abs_diff(const uint16_t v1, const uint16_t v2)
|
|||
//
|
||||
// Wojciech Kaczmarski, SP5WWP
|
||||
// M17 Project, 29 December 2023
|
||||
//--------------------------------------------------------------------
|
||||
//--------------------------------------------------------------------
|
||||
|
||||
|
||||
//audio filter stuff sourced from: https://github.com/NedSimao/FilteringLibrary
|
||||
//no license / information provided in source code
|
||||
#define PI 3.141592653
|
||||
|
||||
//do we need this for test input sthit?
|
||||
// #define PLOT_POINTS 100
|
||||
// #define SAMPLE_TIME_S 0.001
|
||||
// #define CUTOFFFREQ_HZ 50
|
||||
// #define sine_freq 40
|
||||
// #define sine_freq_1 500
|
||||
// #define HFC 50
|
||||
// #define LFC 100
|
||||
|
||||
|
||||
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;
|
||||
filter->v_out[1]=0.0;
|
||||
|
||||
}
|
||||
|
||||
|
||||
float LPFilter_Update(LPFilter *filter, float v_in){
|
||||
|
||||
filter->v_out[1]=filter->v_out[0];
|
||||
filter->v_out[0]=(filter->coef[0]*v_in) + (filter->coef[1]*filter->v_out[1]);
|
||||
|
||||
return (filter->v_out[0]);
|
||||
|
||||
}
|
||||
|
||||
|
||||
/********************************************************************************************************
|
||||
* HIGH PASS FILTER
|
||||
********************************************************************************************************/
|
||||
void HPFilter_Init(HPFilter *filter, float cutoffFreqHz, float sampleTimeS){
|
||||
float RC=0.0;
|
||||
RC=1.0/(2*PI*cutoffFreqHz);
|
||||
|
||||
filter->coef=RC/(sampleTimeS+RC);
|
||||
|
||||
filter->v_in[0]=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){
|
||||
|
||||
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[0]=filter->coef * (filter->v_in[0] - filter->v_in[1]+filter->v_out[1]);
|
||||
|
||||
|
||||
|
||||
return (filter->v_out[0]);
|
||||
}
|
||||
|
||||
/********************************************************************************************************
|
||||
* BAND PASS FILTER
|
||||
********************************************************************************************************/
|
||||
|
||||
|
||||
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;
|
||||
|
||||
}
|
||||
|
||||
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);
|
||||
|
||||
return (filter->out_in);
|
||||
}
|
||||
|
||||
/********************************************************************************************************
|
||||
* NOTCH FILTER
|
||||
********************************************************************************************************/
|
||||
|
||||
|
||||
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
|
||||
float w0_pw_rps=(2.0/sampleTimeS) * tanf(0.5 * w0_rps * sampleTimeS);
|
||||
|
||||
//computing filter coefficients
|
||||
|
||||
filter->alpha=4.0 + w0_rps*w0_pw_rps*sampleTimeS*sampleTimeS;
|
||||
filter->beta=2.0*ww_rps*sampleTimeS;
|
||||
|
||||
//clearing input and output buffers
|
||||
|
||||
for (uint8_t n=0; n<3; n++){
|
||||
filter->vin[n]=0;
|
||||
filter->vout[n]=0;
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
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];
|
||||
filter->vout[1]=filter->vout[0];
|
||||
|
||||
filter->vin[0]=vin;
|
||||
|
||||
//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]);
|
||||
}
|
||||
|
||||
void init_audio_filters (dsd_state * state)
|
||||
{
|
||||
//still not sure if this is even correct or not, but 48k sounds good now
|
||||
LPFilter_Init(&state->RCFilter, 960, (float)1/(float)48000);
|
||||
HPFilter_Init(&state->HRCFilter, 960, (float)1/(float)48000);
|
||||
|
||||
//still unsure if 8k sounds good or not (may be the walkie-talkie is too close to SDR, even outside)
|
||||
LPFilter_Init(&state->RCFilter8, 160, (float)1/(float)8000);
|
||||
HPFilter_Init(&state->HRCFilter8, 160, (float)1/(float)8000);
|
||||
|
||||
//PBFilter_Init(PBFilter *filter, float HPF_cutoffFreqHz, float LPF_cutoffFreqHz, float sampleTimeS);
|
||||
// void NOTCHFilter_Init(NOTCHFilter *filter, float centerFreqHz, float notchWidthHz, float sampleTimeS);
|
||||
|
||||
|
||||
}
|
||||
|
||||
//FUNCTIONS for handing use of above filters
|
||||
|
||||
//lpf
|
||||
void lpf(dsd_state * state, short * input, int len)
|
||||
{
|
||||
int i;
|
||||
for (i = 0; i < len; i++)
|
||||
input[i] = LPFilter_Update(&state->RCFilter, input[i]);
|
||||
}
|
||||
|
||||
//hpf
|
||||
void hpf(dsd_state * state, short * input, int len)
|
||||
{
|
||||
int i;
|
||||
for (i = 0; i < len; i++)
|
||||
input[i] = HPFilter_Update(&state->HRCFilter, input[i]);
|
||||
}
|
||||
|
||||
//lpf
|
||||
void lpf8(dsd_state * state, short * input, int len)
|
||||
{
|
||||
int i;
|
||||
for (i = 0; i < len; i++)
|
||||
input[i] = LPFilter_Update(&state->RCFilter8, input[i]);
|
||||
}
|
||||
|
||||
//hpf
|
||||
void hpf8(dsd_state * state, short * input, int len)
|
||||
{
|
||||
int i;
|
||||
for (i = 0; i < len; i++)
|
||||
input[i] = HPFilter_Update(&state->HRCFilter8, input[i]);
|
||||
}
|
||||
|
||||
//Generic RMS function derived from RTL_FM (RTL_SDR) RMS code (doesnt' really work correctly outside of RTL)
|
||||
long int raw_rms(int16_t *samples, int len, int step) //use samplespersymbol as len
|
||||
{
|
||||
|
||||
int i;
|
||||
long int rms;
|
||||
long p, t, s;
|
||||
double dc, err;
|
||||
|
||||
p = t = 0L;
|
||||
for (i=0; i<len; i+=step) {
|
||||
s = (long)samples[i];
|
||||
t += s;
|
||||
p += s * s;
|
||||
}
|
||||
/* correct for dc offset in squares */
|
||||
dc = (double)(t*step) / (double)len;
|
||||
err = t * 2 * dc - dc * dc * len;
|
||||
|
||||
rms = (long int)sqrt((p-err) / len);
|
||||
if (rms < 0) rms = 150;
|
||||
return rms;
|
||||
}
|
||||
|
|
@ -195,7 +195,7 @@ getFrameSync (dsd_opts * opts, dsd_state * state)
|
|||
int lmin, lmax, lidx;
|
||||
|
||||
//assign t_max value based on decoding type expected (all non-auto decodes first)
|
||||
int t_max; //maximum values allowed for t will depend on decoding type - NXDN will be 10, others will be more
|
||||
int t_max = 24; //initialize as an actual value to prevent any overflow related issues
|
||||
if (opts->frame_nxdn48 == 1 || opts->frame_nxdn96 == 1)
|
||||
{
|
||||
t_max = 10;
|
||||
|
|
@ -218,6 +218,10 @@ getFrameSync (dsd_opts * opts, dsd_state * state)
|
|||
{
|
||||
t_max = 19; //Phase 2 S-ISCH is only 19
|
||||
}
|
||||
else if (opts->analog_only == 1) //shim to make sure this is set to a reasonable value
|
||||
{
|
||||
t_max = 24;
|
||||
}
|
||||
else t_max = 24; //24 for everything else
|
||||
|
||||
int lbuf[48], lbuf2[48]; //if we use t_max in these arrays, and t >= t_max in condition below, then it can overflow those checks in there if t exceeds t_max
|
||||
|
|
|
|||
|
|
@ -691,6 +691,7 @@ initOpts (dsd_opts * opts)
|
|||
opts->dmr_mute_encR = 1;
|
||||
|
||||
opts->monitor_input_audio = 0; //enable with -8
|
||||
opts->analog_only = 0; //only turned on with -fA
|
||||
|
||||
opts->inverted_p2 = 0;
|
||||
opts->p2counter = 0;
|
||||
|
|
@ -1636,7 +1637,7 @@ main (int argc, char **argv)
|
|||
|
||||
initOpts (&opts);
|
||||
initState (&state);
|
||||
|
||||
init_audio_filters(&state); //audio filters
|
||||
InitAllFecFunction();
|
||||
// CNXDNConvolution_init(); //seems to function better without initting it
|
||||
|
||||
|
|
@ -2103,6 +2104,7 @@ main (int argc, char **argv)
|
|||
opts.dmr_mono = 0;
|
||||
state.rf_mod = 0;
|
||||
opts.monitor_input_audio = 1;
|
||||
opts.analog_only = 1;
|
||||
sprintf (opts.output_name, "Analog Monitor");
|
||||
fprintf (stderr,"Only Monitoring Passive Analog Signal\n");
|
||||
}
|
||||
|
|
|
|||
|
|
@ -17,29 +17,6 @@
|
|||
|
||||
#include "dsd.h"
|
||||
|
||||
long int raw_rms(int16_t *samples, int len, int step) //use samplespersymbol as len
|
||||
{
|
||||
|
||||
int i;
|
||||
long int rms;
|
||||
long p, t, s;
|
||||
double dc, err;
|
||||
|
||||
p = t = 0L;
|
||||
for (i=0; i<len; i+=step) {
|
||||
s = (long)samples[i];
|
||||
t += s;
|
||||
p += s * s;
|
||||
}
|
||||
/* correct for dc offset in squares */
|
||||
dc = (double)(t*step) / (double)len;
|
||||
err = t * 2 * dc - dc * dc * len;
|
||||
|
||||
rms = (long int)sqrt((p-err) / len);
|
||||
if (rms < 0) rms = 150;
|
||||
return rms;
|
||||
}
|
||||
|
||||
int
|
||||
getSymbol (dsd_opts * opts, dsd_state * state, int have_sync)
|
||||
{
|
||||
|
|
@ -301,14 +278,9 @@ getSymbol (dsd_opts * opts, dsd_state * state, int have_sync)
|
|||
sf_write_sync (opts->wav_out_raw);
|
||||
}
|
||||
|
||||
//test running analog audio through a de-emphasis filter
|
||||
analog_deemph_filter(state->analog_out, 960);
|
||||
//and dc_block filter analog_dc_block_filter
|
||||
analog_dc_block_filter(state->analog_out, 960);
|
||||
//test running analog audio through a pre-emphasis filter
|
||||
analog_preemph_filter(state->analog_out, 960);
|
||||
//test running analog short sample clipping filter
|
||||
analog_clipping_filter(state->analog_out, 960);
|
||||
// analog audio filtering
|
||||
// lpf (state, state->analog_out, 960);
|
||||
hpf (state, state->analog_out, 960);
|
||||
|
||||
//seems to be working now, but RMS values are lower on actual analog signal than on no signal but noise
|
||||
if ( (opts->rtl_rms > opts->rtl_squelch_level) && opts->monitor_input_audio == 1 && state->carrier == 0 ) //added carrier check here in lieu of disabling it above
|
||||
|
|
|
|||
137
src/edacs-fme.c
137
src/edacs-fme.c
|
|
@ -66,118 +66,9 @@ void openWavOutFile48k (dsd_opts * opts, dsd_state * state)
|
|||
}
|
||||
}
|
||||
|
||||
//generic rms type function
|
||||
long int gen_rms(short *samples, int len, int step)
|
||||
{
|
||||
|
||||
int i;
|
||||
long int rms;
|
||||
long p, t, s;
|
||||
double dc, err;
|
||||
|
||||
p = t = 0L;
|
||||
for (i=0; i<len; i+=step) {
|
||||
s = (long)samples[i];
|
||||
t += s;
|
||||
p += s * s;
|
||||
}
|
||||
/* correct for dc offset in squares */
|
||||
dc = (double)(t*step) / (double)len;
|
||||
err = t * 2 * dc - dc * dc * len;
|
||||
|
||||
rms = (long int)sqrt((p-err) / len);
|
||||
//make sure it doesnt' randomly feed us a large negative value (overflow?)
|
||||
if (rms < 0) rms = 150; //could also consider returning 0 and making it return the last known good value instead
|
||||
return rms;
|
||||
}
|
||||
|
||||
//super generic pre-emphasis filter with alpha value of 1 for short sample input / output
|
||||
static short pstate = 1;
|
||||
void analog_preemph_filter(short * input, int len)
|
||||
{
|
||||
int i;
|
||||
for (i = 0; i < len; i++)
|
||||
{
|
||||
//debug
|
||||
// fprintf (stderr, " I:%d;", input[i]);
|
||||
|
||||
input[i] = input[i] - 1 * pstate;
|
||||
pstate = input[i];
|
||||
|
||||
//debug
|
||||
// fprintf (stderr, " O:%d;;", input[i]);
|
||||
|
||||
//increase gain value slightly
|
||||
input[i] *= 1.5;
|
||||
|
||||
//debug
|
||||
// fprintf (stderr, " O2:%d;;;", input[i]);
|
||||
}
|
||||
}
|
||||
|
||||
//super generic short sample clipping filter, prevent short values from exceeding tolerable clip value
|
||||
void analog_clipping_filter(short * input, int len)
|
||||
{
|
||||
int i;
|
||||
for (i = 0; i < len; i++)
|
||||
{
|
||||
if (input[i] > 32760)
|
||||
input[i] = 32760;
|
||||
else if (input[i] < -32760)
|
||||
input[i] = -32760;
|
||||
}
|
||||
}
|
||||
|
||||
//modified from version found in rtl_fm
|
||||
static short avg = 0;
|
||||
void analog_deemph_filter(short * input, int len)
|
||||
{
|
||||
// static short avg; // cheating...
|
||||
int i, d;
|
||||
|
||||
// int a = (int)round(1.0/((1.0-exp(-1.0/(96000 * 75e-6))))); //48000 is rate out, but doubling it sounds 'beefier'
|
||||
int a = 8; //result of calc above is 8, so save a cycle on the low end CPUs
|
||||
|
||||
// de-emph IIR
|
||||
// avg = avg * (1 - alpha) + sample * alpha;
|
||||
|
||||
for (i = 0; i < len; i++)
|
||||
{
|
||||
d = input[i] - avg;
|
||||
if (d > 0)
|
||||
avg += (d + a/2) / a;
|
||||
else
|
||||
avg += (d - a/2) / a;
|
||||
|
||||
input[i] = (short)avg;
|
||||
}
|
||||
}
|
||||
|
||||
//modified from version found in rtl_fm
|
||||
static int dc_avg = 0;
|
||||
void analog_dc_block_filter(short * input, int len)
|
||||
{
|
||||
int i, avg;
|
||||
int64_t sum = 0;
|
||||
for (i=0; i < len; i++)
|
||||
sum += input[i];
|
||||
|
||||
avg = sum / len;
|
||||
avg = (avg + dc_avg * 9) / 10;
|
||||
for (i=0; i < len; i++)
|
||||
input[i] -= (short)avg;
|
||||
|
||||
dc_avg = avg;
|
||||
}
|
||||
|
||||
//listening to and playing back analog audio
|
||||
void edacs_analog(dsd_opts * opts, dsd_state * state, int afs, unsigned char lcn)
|
||||
{
|
||||
//reset static states for new channel
|
||||
pstate = 1;
|
||||
avg = 0;
|
||||
dc_avg = 0;
|
||||
|
||||
int i, result;
|
||||
int count = 5; //RMS has a 5 count (5 * 180ms) now before cutting off;
|
||||
short analog1[960];
|
||||
|
|
@ -220,7 +111,7 @@ void edacs_analog(dsd_opts * opts, dsd_state * state, int afs, unsigned char lcn
|
|||
analog3[i] = sample;
|
||||
}
|
||||
//this rms will only work properly (for now) with squelch enabled in SDR++ or other
|
||||
rms = gen_rms(analog3, 960, 1);
|
||||
rms = raw_rms(analog3, 960, 1);
|
||||
}
|
||||
|
||||
//NOTE: The core dumps observed previously were due to SDR++ Remote Server connection dropping due to Internet/Other issues
|
||||
|
|
@ -274,7 +165,7 @@ void edacs_analog(dsd_opts * opts, dsd_state * state, int afs, unsigned char lcn
|
|||
}
|
||||
|
||||
//this rms will only work properly (for now) with squelch enabled in SDR++
|
||||
rms = gen_rms(analog3, 960, 1);
|
||||
rms = raw_rms(analog3, 960, 1);
|
||||
}
|
||||
|
||||
//RTL Input
|
||||
|
|
@ -311,22 +202,14 @@ void edacs_analog(dsd_opts * opts, dsd_state * state, int afs, unsigned char lcn
|
|||
sr += digitize (opts, state, (int)analog1[i]);
|
||||
}
|
||||
|
||||
//test running analog audio through a de-emphasis filter
|
||||
analog_deemph_filter(analog1, 960);
|
||||
analog_deemph_filter(analog2, 960);
|
||||
analog_deemph_filter(analog3, 960);
|
||||
//and dc_block filter analog_dc_block_filter
|
||||
analog_dc_block_filter(analog1, 960);
|
||||
analog_dc_block_filter(analog2, 960);
|
||||
analog_dc_block_filter(analog3, 960);
|
||||
//test running analog audio through a pre-emphasis filter
|
||||
analog_preemph_filter(analog1, 960);
|
||||
analog_preemph_filter(analog2, 960);
|
||||
analog_preemph_filter(analog3, 960);
|
||||
//test running analog short sample clipping filter
|
||||
analog_clipping_filter(analog1, 960);
|
||||
analog_clipping_filter(analog2, 960);
|
||||
analog_clipping_filter(analog3, 960);
|
||||
// analog audio filtering
|
||||
lpf (state, analog1, 960);
|
||||
lpf (state, analog2, 960);
|
||||
lpf (state, analog3, 960);
|
||||
|
||||
hpf (state, analog1, 960);
|
||||
hpf (state, analog2, 960);
|
||||
hpf (state, analog3, 960);
|
||||
|
||||
//reconfigured to use seperate audio out stream that is always 48k short
|
||||
if (opts->audio_out_type == 0 && opts->slot1_on == 1)
|
||||
|
|
|
|||
|
|
@ -1852,6 +1852,12 @@ void encodeM17STR(dsd_opts * opts, dsd_state * state)
|
|||
voice2[i] *= (float)state->aout_gain/ (float)25.0f;
|
||||
}
|
||||
|
||||
//run lpf and hpf
|
||||
// lpf8 (state, voice1, 160);
|
||||
// lpf8 (state, voice2, 160);
|
||||
hpf8 (state, voice1, 160);
|
||||
hpf8 (state, voice2, 160);
|
||||
|
||||
//convert out audio input into CODEC2 (3200bps) 8 byte data stream
|
||||
uint8_t vc1_bytes[8]; memset (vc1_bytes, 0, sizeof(vc1_bytes));
|
||||
uint8_t vc2_bytes[8]; memset (vc2_bytes, 0, sizeof(vc2_bytes));
|
||||
|
|
|
|||
Loading…
Reference in New Issue