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

src/amps/amps.c

@@ -450,7 +450,7 @@ int amps_create(int channel, enum amps_chan_type chan_type, const char *audiodev
 	}
 
 	/* init audio processing */
-	rc = dsp_init_sender(amps, (de_emphasis == 0), tolerant);
+	rc = dsp_init_sender(amps, tolerant);
 	if (rc < 0) {
 		PDEBUG(DAMPS, DEBUG_ERROR, "Failed to init audio processing!\n");
 		goto error;

src/amps/dsp.c

@@ -178,12 +178,11 @@ static void dsp_init_ramp(amps_t *amps)
 static void sat_reset(amps_t *amps, const char *reason);
 
 /* Init FSK of transceiver */
-int dsp_init_sender(amps_t *amps, int high_pass, int tolerant)
+int dsp_init_sender(amps_t *amps, int tolerant)
 {
 	sample_t *spl;
 	int i;
 	int rc;
-	double RC, dt;
 	int half;
 
 	/* attack (3ms) and recovery time (13.5ms) according to amps specs */
@@ -256,14 +255,6 @@ int dsp_init_sender(amps_t *amps, int high_pass, int tolerant)
 	amps->test_phaseshift256 = 256.0 / ((double)amps->sender.samplerate / 1000.0);
 	PDEBUG(DDSP, DEBUG_DEBUG, "test_phaseshift256 = %.4f\n", amps->test_phaseshift256);
 
-	/* use this filter to remove dc level for 0-crossing detection
-	 * if we have de-emphasis, we don't need it, so high_pass is not set. */
-	if (high_pass) {
-		RC = 1.0 / (CUT_OFF_HIGHPASS * 2.0 *3.14);
-		dt = 1.0 / amps->sender.samplerate;
-		amps->highpass_factor = RC / (RC + dt);
-	}
-
 	/* be more tolerant when syncing */
 	amps->fsk_rx_sync_tolerant = tolerant;
 
@@ -808,8 +799,7 @@ static void sender_receive_audio(amps_t *amps, sample_t *samples, int length)
 	int max, pos;
 	int i;
 
-	/* SAT detection */
+	/* SAT / signalling tone detection */
-
 	max = amps->sat_samples;
 	spl = amps->sat_filter_spl;
 	pos = amps->sat_filter_pos;
@@ -853,25 +843,10 @@ static void sender_receive_audio(amps_t *amps, sample_t *samples, int length)
 void sender_receive(sender_t *sender, sample_t *samples, int length)
 {
 	amps_t *amps = (amps_t *) sender;
-	double x, y, x_last, y_last, factor;
-	int i;
 
-	/* high pass filter to remove 0-level
+	/* dc filter required for FSK decoding and tone detection */
-	 * if factor is not set, we should already have 0-level. */
+	if (amps->de_emphasis)
-	factor = amps->highpass_factor;
+		dc_filter(&amps->estate, samples, length);
-	if (factor) {
-		x_last = amps->highpass_x_last;
-		y_last = amps->highpass_y_last;
-		for (i = 0; i < length; i++) {
-			x = (double)samples[i];
-			y = factor * (y_last + x - x_last);
-			x_last = x;
-			y_last = y;
-			samples[i] = y;
-		}
-		amps->highpass_x_last = x_last;
-		amps->highpass_y_last = y_last;
-	}
 
 	switch (amps->dsp_mode) {
 	case DSP_MODE_OFF:

src/amps/dsp.h

@@ -1,6 +1,6 @@
 
 void dsp_init(void);
-int dsp_init_sender(amps_t *amps, int high_pass, int tolerant);
+int dsp_init_sender(amps_t *amps, int tolerant);
 void dsp_cleanup_sender(amps_t *amps);
 void amps_set_dsp_mode(amps_t *amps, enum dsp_mode mode, int frame_length);
 

src/cnetz/dsp.c

@@ -812,6 +812,8 @@ void unshrink_speech(cnetz_t *cnetz, sample_t *speech_buffer, int count)
 
 	/* 4. de-emphasis is done by cnetz code, not by common code */
 	/* de-emphasis is only used when scrambler is off, see FTZ 171 TR 60 Clause 4 */
+	if (cnetz->de_emphasis)
+		dc_filter(&cnetz->estate, speech_buffer, count);
 	if (cnetz->de_emphasis && !cnetz->scrambler)
 		de_emphasis(&cnetz->estate, speech_buffer, count);
 	/* 3. descramble */

src/common/emphasis.c

@@ -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);
+}
+

src/common/emphasis.h

@@ -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);
 

src/common/goertzel.c

@@ -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, sk;
-	double level;
-	int 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)

src/common/sender.c

@@ -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 */
+			/* do filter and de-emphasis from radio receive audio, process echo test */
-			if (inst->de_emphasis)
+			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);

src/test/test_emphasis.c

@@ -64,6 +64,7 @@ int main(void)
 
 	for (i = 31.25; i < 4001; i = i * sqrt(sqrt(2.0))) {
 		gen_samples(samples, (double)i);
+		dc_filter(&estate, samples, SAMPLERATE);
 		de_emphasis(&estate, samples, SAMPLERATE);
 		level = get_level(samples);
 		printf("%s%.0f Hz: %.1f dB", debug_db(level), i, level2db(level));