Add global DC-Filter and remove all individual DC-Filters

This commit is contained in:
Andreas Eversberg
2017-01-28 18:18:44 +01:00
parent 71e556e7ff
commit bd7ccc5fa0
9 changed files with 32 additions and 58 deletions

View File

@@ -27,7 +27,7 @@
#define PI M_PI
#define CUT_OFF_H 300.0 /* cut-off frequency for high-pass filters */
#define CUT_OFF_H 100.0 /* cut-off frequency for high-pass filter */
static void gen_sine(double *samples, int num, int samplerate, double freq)
{
@@ -106,8 +106,6 @@ void de_emphasis(emphasis_t *state, double *samples, int num)
double x, y, y_last, factor, amp;
int i;
filter_process(&state->d.hp, samples, num);
y_last = state->d.y_last;
factor = state->d.factor;
amp = state->d.amp;
@@ -126,3 +124,9 @@ void de_emphasis(emphasis_t *state, double *samples, int num)
state->d.y_last = y_last;
}
/* high pass filter to remove DC and low frequencies */
void dc_filter(emphasis_t *state, double *samples, int num)
{
filter_process(&state->d.hp, samples, num);
}

View File

@@ -17,4 +17,5 @@ typedef struct emphasis {
int init_emphasis(emphasis_t *state, int samplerate, double cut_off);
void pre_emphasis(emphasis_t *state, double *samples, int num);
void de_emphasis(emphasis_t *state, double *samples, int num);
void dc_filter(emphasis_t *state, double *samples, int num);

View File

@@ -30,30 +30,26 @@
* audio level calculation
*/
/* return average value (rectified value), that can be 0..1 */
/* Return average value (rectified value)
* The input must not have any dc offset!
* For a perfect rectangualr wave, the result would equal the peak level.
* For a sine wave the result would be factor (2 / PI) below peak level.
*/
double audio_level(sample_t *samples, int length)
{
double bias;
double level;
int sk;
double level, sk;
int n;
/* calculate bias */
bias = 0;
for (n = 0; n < length; n++)
bias += samples[n];
bias = bias / length;
/* level calculation */
level = 0;
for (n = 0; n < length; n++) {
sk = samples[n] - bias;
sk = samples[n];
if (sk < 0)
level -= (double)sk;
if (sk > 0)
level += (double)sk;
}
level = level / (double)length / 32767.0;
level = level / (double)length;
return level;
}
@@ -79,17 +75,10 @@ void audio_goertzel_init(goertzel_t *goertzel, double freq, int samplerate)
*/
void audio_goertzel(goertzel_t *goertzel, sample_t *samples, int length, int offset, double *result, int k)
{
double bias;
double sk, sk1, sk2;
double cos2pik;
int i, n;
/* calculate bias to remove DC */
bias = 0;
for (n = 0; n < length; n++)
bias += samples[n];
bias = bias / length;
/* we do goertzel */
for (i = 0; i < k; i++) {
sk = 0;
@@ -98,7 +87,7 @@ void audio_goertzel(goertzel_t *goertzel, sample_t *samples, int length, int off
cos2pik = goertzel[i].coeff;
/* note: after 'length' cycles, offset is restored to its initial value */
for (n = 0; n < length; n++) {
sk = (cos2pik * sk1) - sk2 + samples[offset++] - bias;
sk = (cos2pik * sk1) - sk2 + samples[offset++];
sk2 = sk1;
sk1 = sk;
if (offset == length)

View File

@@ -281,7 +281,7 @@ cant_recover:
display_wave(inst, samples[i], count);
sender_receive(inst, samples[i], count);
}
/* do pre emphasis towards radio, not wave_write */
/* do pre emphasis towards radio */
if (inst->pre_emphasis)
pre_emphasis(&inst->estate, samples[i], count);
/* set paging signal */
@@ -331,9 +331,11 @@ transmit_later:
/* rx gain */
if (inst->rx_gain != 1.0)
gain_samples(samples[i], count, inst->rx_gain);
/* do de emphasis from radio (then write_wave/wave_read), receive audio, process echo test */
if (inst->de_emphasis)
/* do filter and de-emphasis from radio receive audio, process echo test */
if (inst->de_emphasis) {
dc_filter(&inst->estate, samples[i], count);
de_emphasis(&inst->estate, samples[i], count);
}
if (inst->loopback != 1) {
display_wave(inst, samples[i], count);
sender_receive(inst, samples[i], count);