diff --git a/src/atest.c b/src/atest.c
index 5c19775..7e8f03b 100644
--- a/src/atest.c
+++ b/src/atest.c
@@ -236,7 +236,7 @@ int main (int argc, char *argv[])
my_audio_config.achan[channel].space_freq = DEFAULT_SPACE_FREQ;
my_audio_config.achan[channel].baud = DEFAULT_BAUD;
- strlcpy (my_audio_config.achan[channel].profiles, "E", sizeof(my_audio_config.achan[channel].profiles));
+ strlcpy (my_audio_config.achan[channel].profiles, "A", sizeof(my_audio_config.achan[channel].profiles));
my_audio_config.achan[channel].num_freq = 1;
my_audio_config.achan[channel].offset = 0;
@@ -430,19 +430,21 @@ int main (int argc, char *argv[])
/* We have similar logic in direwolf.c, config.c, gen_packets.c, and atest.c, */
/* that need to be kept in sync. Maybe it could be a common function someday. */
- if (my_audio_config.achan[0].baud == 100) {
+ if (my_audio_config.achan[0].baud == 100) { // What was this for?
my_audio_config.achan[0].modem_type = MODEM_AFSK;
my_audio_config.achan[0].mark_freq = 1615;
my_audio_config.achan[0].space_freq = 1785;
- strlcpy (my_audio_config.achan[0].profiles, "D", sizeof(my_audio_config.achan[0].profiles));
+ //strlcpy (my_audio_config.achan[0].profiles, "A", sizeof(my_audio_config.achan[0].profiles));
}
- else if (my_audio_config.achan[0].baud < 600) {
+ else if (my_audio_config.achan[0].baud < 600) { // e.g. HF SSB packet
my_audio_config.achan[0].modem_type = MODEM_AFSK;
my_audio_config.achan[0].mark_freq = 1600;
my_audio_config.achan[0].space_freq = 1800;
- strlcpy (my_audio_config.achan[0].profiles, "D", sizeof(my_audio_config.achan[0].profiles));
+ // Previously we had a "D" which was fine tuned for 300 bps.
+ // In v1.7, it's not clear if we should use "B" or just stick with "A".
+ //strlcpy (my_audio_config.achan[0].profiles, "B", sizeof(my_audio_config.achan[0].profiles));
}
- else if (my_audio_config.achan[0].baud < 1800) {
+ else if (my_audio_config.achan[0].baud < 1800) { // common 1200
my_audio_config.achan[0].modem_type = MODEM_AFSK;
my_audio_config.achan[0].mark_freq = DEFAULT_MARK_FREQ;
my_audio_config.achan[0].space_freq = DEFAULT_SPACE_FREQ;
@@ -460,7 +462,7 @@ int main (int argc, char *argv[])
my_audio_config.achan[0].space_freq = 0;
strlcpy (my_audio_config.achan[0].profiles, "", sizeof(my_audio_config.achan[0].profiles));
}
- else if (my_audio_config.achan[0].baud == 12345) {
+ else if (my_audio_config.achan[0].baud == 12345) { // Hack for different use of 9600
my_audio_config.achan[0].modem_type = MODEM_AIS;
my_audio_config.achan[0].baud = 9600;
my_audio_config.achan[0].mark_freq = 0;
@@ -473,7 +475,7 @@ int main (int argc, char *argv[])
// Will make more precise in afsk demod init.
my_audio_config.achan[0].mark_freq = 2083; // Actually 2083.3 - logic 1.
my_audio_config.achan[0].space_freq = 1563; // Actually 1562.5 - logic 0.
- strlcpy (my_audio_config.achan[0].profiles, "D", sizeof(my_audio_config.achan[0].profiles));
+ strlcpy (my_audio_config.achan[0].profiles, "A", sizeof(my_audio_config.achan[0].profiles));
}
else {
my_audio_config.achan[0].modem_type = MODEM_SCRAMBLE;
diff --git a/src/config.c b/src/config.c
index 08f71a3..93b92a1 100644
--- a/src/config.c
+++ b/src/config.c
@@ -1337,7 +1337,7 @@ void config_init (char *fname, struct audio_s *p_audio_config,
// Will make more precise in afsk demod init.
p_audio_config->achan[channel].mark_freq = 2083; // Actually 2083.3 - logic 1.
p_audio_config->achan[channel].space_freq = 1563; // Actually 1562.5 - logic 0.
- // ? strlcpy (p_audio_config->achan[channel].profiles, "D", sizeof(p_audio_config->achan[channel].profiles));
+ // ? strlcpy (p_audio_config->achan[channel].profiles, "A", sizeof(p_audio_config->achan[channel].profiles));
}
else {
p_audio_config->achan[channel].modem_type = MODEM_SCRAMBLE;
diff --git a/src/demod.c b/src/demod.c
index 281367b..3f032b4 100644
--- a/src/demod.c
+++ b/src/demod.c
@@ -198,46 +198,57 @@ int demod_init (struct audio_s *pa)
assert (num_letters == (int)(strlen(just_letters)));
/*
- * Pick a good default demodulator if none specified.
+ * Pick a good default demodulator if none specified.
+ * Previously, we had "D" optimized for 300 bps.
+ * Gone in 1.7 so it is always "A+".
*/
if (num_letters == 0) {
+ strlcpy (just_letters, "A", sizeof(just_letters));
+ num_letters = strlen(just_letters);
- if (save_audio_config_p->achan[chan].baud < 600) {
-
- /* This has been optimized for 300 baud. */
-
- strlcpy (just_letters, "D", sizeof(just_letters));
-
- }
- else {
-#if __arm__
- /* We probably don't have a lot of CPU power available. */
- /* Previously we would use F if possible otherwise fall back to A. */
-
- /* In version 1.2, new default is E+ /3. */
- strlcpy (just_letters, "E", sizeof(just_letters)); // version 1.2 now E.
- if (have_plus != -1) have_plus = 1; // Add as default for version 1.2
+ if (have_plus != -1) have_plus = 1; // Add as default for version 1.2
// If not explicitly turned off.
- if (save_audio_config_p->achan[chan].decimate == 0) {
- if (save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec > 40000) {
- save_audio_config_p->achan[chan].decimate = 3;
- }
- }
-#else
- strlcpy (just_letters, "E", sizeof(just_letters)); // version 1.2 changed C to E.
- if (have_plus != -1) have_plus = 1; // Add as default for version 1.2
- // If not explicitly turned off.
-#endif
- }
- num_letters = 1;
}
+/*
+ * Special case for ARM.
+ * The higher end ARM chips have loads of power but many people
+ * are using a single core Pi Zero or similar.
+ * (I'm still using a model 1 for my digipeater/IGate!)
+ * Decreasing CPU requirement has a negligible impact on decoding performance.
+ *
+ * atest -PA- 01_Track_1.wav --> 1002 packets decoded.
+ * atest -PA- -D3 01_Track_1.wav --> 997 packets decoded.
+ *
+ * Someone concerned about 1/2 of one percent difference can add "-D 1"
+ */
+#if __arm__
+ if (save_audio_config_p->achan[chan].decimate == 0) {
+ if (save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec > 40000) {
+ save_audio_config_p->achan[chan].decimate = 3;
+ }
+ }
+#endif
+
+/*
+ * Number of filter taps is proportional to number of audio samples in a "symbol" duration.
+ * These can get extremely large for low speeds, e.g. 300 baud.
+ * In this case, increase the decimation ration. Crude approximation. Could be improved.
+ */
+ if (save_audio_config_p->achan[chan].decimate == 0 &&
+ save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec > 40000 &&
+ save_audio_config_p->achan[chan].baud < 600) {
+
+ // Avoid enormous number of filter taps.
+
+ save_audio_config_p->achan[chan].decimate = 3;
+ }
- assert (num_letters == (int)(strlen(just_letters)));
/*
* Put it back together again.
*/
+ assert (num_letters == (int)(strlen(just_letters)));
/* At this point, have_plus can have 3 values: */
/* 1 = turned on, either explicitly or by applied default */
@@ -286,7 +297,7 @@ int demod_init (struct audio_s *pa)
if (save_audio_config_p->achan[chan].decimate == 0) {
save_audio_config_p->achan[chan].decimate = 1;
- if (strchr (just_letters, 'D') != NULL && save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec > 40000) {
+ if (strchr (just_letters, 'B') != NULL && save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec > 40000) {
save_audio_config_p->achan[chan].decimate = 3;
}
}
diff --git a/src/demod_9600.c b/src/demod_9600.c
index 2f98983..ef45e4c 100644
--- a/src/demod_9600.c
+++ b/src/demod_9600.c
@@ -221,7 +221,7 @@ void demod_9600_init (enum modem_t modem_type, int samples_per_sec, int baud, st
//dw_printf ("demod_9600_init: call gen_lowpass(fc=%.2f, , size=%d, )\n", fc, D->lp_filter_size);
- (void)gen_lowpass (fc, D->lp_filter, D->lp_filter_size, D->lp_window, 0);
+ gen_lowpass (fc, D->lp_filter, D->lp_filter_size, D->lp_window);
/* Version 1.2: Experiment with different slicing levels. */
diff --git a/src/demod_afsk.c b/src/demod_afsk.c
index 7a007d1..c34d8bc 100644
--- a/src/demod_afsk.c
+++ b/src/demod_afsk.c
@@ -1,7 +1,7 @@
//
// This file is part of Dire Wolf, an amateur radio packet TNC.
//
-// Copyright (C) 2011, 2012, 2013, 2014, 2015 John Langner, WB2OSZ
+// Copyright (C) 2011, 2012, 2013, 2014, 2015, 2020 John Langner, WB2OSZ
//
// This program is free software: you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
@@ -23,8 +23,7 @@
// #define DEBUG3 1 /* print carrier detect changes. */
// #define DEBUG4 1 /* capture AFSK demodulator output to log files */
-
-// #define DEBUG5 1 /* capture 9600 output to log files */
+ /* Can be used to make nice plots. */
/*------------------------------------------------------------------
@@ -62,15 +61,32 @@
#define MIN(a,b) ((a)<(b)?(a):(b))
#define MAX(a,b) ((a)>(b)?(a):(b))
+#define TUNE(envvar,param,name,fmt) { \
+ char *e = getenv(envvar); \
+ if (e != NULL) { \
+ param = atof(e); \
+ text_color_set (DW_COLOR_ERROR); \
+ dw_printf ("TUNE: " name " = " fmt "\n", param); \
+ } }
-/* Quick approximation to sqrt(x*x+y*y) */
+
+// Cosine table indexed by unsigned byte.
+static float fcos256_table[256];
+
+#define fcos256(x) (fcos256_table[((x)>>24)&0xff])
+#define fsin256(x) (fcos256_table[(((x)>>24)-64)&0xff])
+
+static void nudge_pll (int chan, int subchan, int slice, float demod_out, struct demodulator_state_s *D, float amplitude);
+
+
+/* Quick approximation to sqrt(x*x + y*y) */
/* No benefit for regular PC. */
-/* Should help with microcomputer platform. */
+/* Might help with microcomputer platform??? */
-#if 0 // not using anymore
__attribute__((hot)) __attribute__((always_inline))
-static inline float z (float x, float y)
+static inline float fast_hypot(float x, float y)
{
+#if 0
x = fabsf(x);
y = fabsf(y);
@@ -80,8 +96,11 @@ static inline float z (float x, float y)
else {
return (y * .941246f + x * .41f);
}
-}
+#else
+ return (hypotf(x,y));
#endif
+}
+
/* Add sample to buffer and shift the rest down. */
@@ -96,22 +115,31 @@ static inline void push_sample (float val, float *buff, int size)
/* FIR filter kernel. */
__attribute__((hot)) __attribute__((always_inline))
-static inline float convolve (const float *__restrict__ data, const float *__restrict__ filter, int filter_size)
+static inline float convolve (const float *__restrict__ data, const float *__restrict__ filter, int filter_taps)
{
float sum = 0.0f;
int j;
-
//#pragma GCC ivdep // ignored until gcc 4.9
- for (j=0; j<filter_size; j++) {
+ for (j=0; j<filter_taps; j++) {
sum += filter[j] * data[j];
}
-
return (sum);
}
-/* Automatic gain control. */
-/* Result should settle down to 1 unit peak to peak. i.e. -0.5 to +0.5 */
+// Automatic Gain control - used when we have a single slicer.
+//
+// The first step is to create an envelope for the peak and valley
+// of the mark or space amplitude. We need to keep track of the valley
+// because it does not go down to zero when the tone is not present.
+// We want to find the difference between tone present and not.
+//
+// We use an IIR filter with fast attack and slow decay which only considers the past.
+// Perhaps an improvement could be obtained by looking in the future as well.
+//
+
+// Result should settle down to 1 unit peak to peak. i.e. -0.5 to +0.5
+
__attribute__((hot)) __attribute__((always_inline))
static inline float agc (float in, float fast_attack, float slow_decay, float *ppeak, float *pvalley)
@@ -130,13 +158,47 @@ static inline float agc (float in, float fast_attack, float slow_decay, float *p
*pvalley = in * slow_decay + *pvalley * (1.0f - slow_decay);
}
+#if 1
+ float x = in;
+ if (x > *ppeak) x = *ppeak; // experiment: clip to envelope?
+ if (x < *pvalley) x = *pvalley;
+#endif
if (*ppeak > *pvalley) {
- return ((in - 0.5f * (*ppeak + *pvalley)) / (*ppeak - *pvalley));
+
+ return ((x - 0.5f * (*ppeak + *pvalley)) / (*ppeak - *pvalley)); // my original AGC
+
+ //return (( x - 0.5f * (*ppeak + *pvalley )) * ( *ppeak - *pvalley )); // see note below.
+ //return (x - 0.5f * (*ppeak + *pvalley)); // not as good either.
}
return (0.0f);
}
+// K6JQ pointed me to this wonderful article:
+// Improved Automatic Threshold Correction Methods for FSK by Kok Chen, W7AY.
+// http://www.w7ay.net/site/Technical/ATC/index.html
+//
+// The stated problem is a little different, selective fading for HF RTTY, but the
+// general idea is the similar: Compensating for imbalance of the two tones.
+//
+// The stronger tone probably has a better S/N ratio so we apply a larger
+// weight to it. Effectively it is comparing power rather than amplitude.
+// This is the optimal method from the article referenced.
+//
+// Interesting idea but it did not work as well as the original AGC in this case.
+// For VHF FM we are not dealing with rapid deep selective fading of one tone.
+// Instead we have an imbalance which is the same for the whole frame.
+// It might be interesting to try this with HF SSB packet which is much like RTTY.
+//
+// I use the term valley rather than noise floor.
+// After a little algebra, it looks remarkably similar to the function above.
+//
+// return (( x - valley ) * ( peak - valley ) - 0.5f * ( peak - valley ) * ( peak - valley ));
+// return (( x - valley ) - 0.5f * ( peak - valley )) * ( peak - valley ));
+// return (( x - 0.5f * (peak + valley )) * ( peak - valley ));
+
+
+
/*
* for multi-slicer experiment.
*/
@@ -162,11 +224,7 @@ static inline float agc (float in, float fast_attack, float slow_decay, float *p
*
* D - Pointer to demodulator state for given channel.
*
- * Outputs: D->ms_filter_size
- * D->m_sin_table[]
- * D->m_cos_table[]
- * D->s_sin_table[]
- * D->s_cos_table[]
+ * Outputs:
*
* Returns: None.
*
@@ -180,6 +238,10 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
{
int j;
+
+ for (j = 0; j < 256; j++) {
+ fcos256_table[j] = cosf((float)j * 2.0f * (float)M_PI / 256.0f);
+ }
memset (D, 0, sizeof(struct demodulator_state_s));
D->num_slicers = 1;
@@ -188,122 +250,156 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
dw_printf ("demod_afsk_init (rate=%d, baud=%d, mark=%d, space=%d, profile=%c\n",
samples_per_sec, baud, mark_freq, space_freq, profile);
#endif
-
-#ifdef TUNE_PROFILE
- profile = TUNE_PROFILE;
-#endif
+ D->profile = profile;
+ switch (D->profile) {
- D->profile = profile; // so we know whether to take fast path later.
+ case 'A': // Official name
+ case 'E': // For compatibility during transition
- switch (profile) {
+ D->profile = 'A';
- case 'D':
-
- /* Prefilter, Cosine window, FIR lowpass. Tweeked for 300 baud. */
+ /* New in version 1.7 */
+ /* This is a simpler version of what has been used all along. */
+ /* Rather than convolving each sample with a pre-computed mark and */
+ /* space filter, we have two free running local oscillators. */
+ /* Also see if we can do better with a Root Raised Cosine filter */
+ /* which supposedly reduces intersymbol interference. */
D->use_prefilter = 1; /* first, a bandpass filter. */
- D->prefilter_baud = 0.87;
- D->pre_filter_len_bits = 1.857;
- D->pre_window = BP_WINDOW_COSINE;
- D->ms_filter_len_bits = 1.857; /* 91 @ 44100/3, 300 */
- D->ms_window = BP_WINDOW_COSINE;
-
- //D->bp_window = BP_WINDOW_COSINE;
+ if (baud > 600) {
+ D->prefilter_baud = 0.155;
+ // Low cutoff below mark, high cutoff above space
+ // as fraction of the symbol rate.
+ // Intuitively you might expect this to be about
+ // half the symbol rate, e.g. 600 Hz outside
+ // the two tones of interest for 1200 baud.
+ // It turns out that narrower is better.
- D->lpf_use_fir = 1;
- D->lpf_baud = 1.10;
- D->lp_filter_len_bits = D->ms_filter_len_bits;
- D->lp_window = BP_WINDOW_TRUNCATED;
+ D->pre_filter_len_sym = 383 * 1200. / 44100.; // about 8 symbols
+ D->pre_window = BP_WINDOW_TRUNCATED;
+ }
+ else {
+ D->prefilter_baud = 0.87; // TOTO: fine tune
+ D->pre_filter_len_sym = 1.857;
+ D->pre_window = BP_WINDOW_COSINE;
+ }
+
+ // Local oscillators for Mark and Space tones.
- D->agc_fast_attack = 0.495;
- D->agc_slow_decay = 0.00022;
- D->hysteresis = 0.027;
+ D->u.afsk.m_osc_phase = 0;
+ D->u.afsk.m_osc_delta = round ( pow(2., 32.) * (double)mark_freq / (double)samples_per_sec );
- D->pll_locked_inertia = 0.620;
- D->pll_searching_inertia = 0.350;
- break;
+ D->u.afsk.s_osc_phase = 0;
+ D->u.afsk.s_osc_delta = round ( pow(2., 32.) * (double)space_freq / (double)samples_per_sec );
- case 'F': // removed obsolete. treat as E for now.
- case 'E':
+ D->u.afsk.use_rrc = 1;
+ TUNE("TUNE_USE_RRC", D->u.afsk.use_rrc, "use_rrc", "%d")
- /* 1200 baud - Started out similar to C but add prefilter. */
- /* Version 1.2 */
- /* Enhancements: */
- /* + Add prefilter. Previously used for 300 baud D, but not 1200. */
- /* + Prefilter length now independent of M/S filters. */
- /* + Lowpass filter length now independent of M/S filters. */
- /* + Allow mixed window types. */
+ if (D->u.afsk.use_rrc) {
+ D->u.afsk.rrc_width_sym = 2.80;
+ D->u.afsk.rrc_rolloff = 0.20;
+ }
+ else {
+ D->lpf_baud = 0.14;
+ D->lp_filter_width_sym = 1.388;
+ D->lp_window = BP_WINDOW_TRUNCATED;
+ }
- //D->bp_window = BP_WINDOW_COSINE; /* The name says BP but it is used for all of them. */
-
- D->use_prefilter = 1; /* first, a bandpass filter. */
- D->prefilter_baud = 0.23;
- D->pre_filter_len_bits = 156 * 1200. / 44100.;
- D->pre_window = BP_WINDOW_TRUNCATED;
-
- D->ms_filter_len_bits = 74 * 1200. / 44100.;
- D->ms_window = BP_WINDOW_COSINE;
-
- D->lpf_use_fir = 1;
- D->lpf_baud = 1.18;
- D->lp_filter_len_bits = 63 * 1200. / 44100.;
- D->lp_window = BP_WINDOW_TRUNCATED;
-
- //D->agc_fast_attack = 0.300;
- //D->agc_slow_decay = 0.000185;
D->agc_fast_attack = 0.820;
D->agc_slow_decay = 0.000214;
- D->hysteresis = 0.01;
+ D->agc_fast_attack = 0.45;
+ D->agc_slow_decay = 0.000195;
+ D->agc_fast_attack = 0.70;
+ D->agc_slow_decay = 0.000090;
- //D->pll_locked_inertia = 0.57;
- //D->pll_searching_inertia = 0.33;
D->pll_locked_inertia = 0.74;
D->pll_searching_inertia = 0.50;
break;
+ case 'B': // official name
+ case 'D': // backward compatibility
+
+ D->profile = 'B';
+
+ // Experiment for version 1.7.
+ // Up to this point, I've always used separate mark and space
+ // filters and compared the amplitudes.
+ // Another technique for an FM demodulator is to mix with
+ // the center frequency and look for the rate of change of the phase.
+
+ D->use_prefilter = 1; /* first, a bandpass filter. */
+
+ if (baud > 600) {
+ D->prefilter_baud = 0.19;
+ // Low cutoff below mark, high cutoff above space
+ // as fraction of the symbol rate.
+ // Intuitively you might expect this to be about
+ // half the symbol rate, e.g. 600 Hz outside
+ // the two tones of interest for 1200 baud.
+ // It turns out that narrower is better.
+
+ D->pre_filter_len_sym = 8.163; // Filter length in symbol times.
+ D->pre_window = BP_WINDOW_TRUNCATED;
+ }
+ else {
+ D->prefilter_baud = 0.87; // TOTO: fine tune
+ D->pre_filter_len_sym = 1.857;
+ D->pre_window = BP_WINDOW_COSINE;
+ }
+
+ // Local oscillator for Center frequency.
+
+ D->u.afsk.c_osc_phase = 0;
+ D->u.afsk.c_osc_delta = round ( pow(2., 32.) * 0.5 * (mark_freq + space_freq) / (double)samples_per_sec );
+
+ D->u.afsk.use_rrc = 1;
+ TUNE("TUNE_USE_RRC", D->u.afsk.use_rrc, "use_rrc", "%d")
+
+ if (D->u.afsk.use_rrc) {
+ D->u.afsk.rrc_width_sym = 2.00;
+ D->u.afsk.rrc_rolloff = 0.40;
+ }
+ else {
+ D->lpf_baud = 0.5;
+ D->lp_filter_width_sym = 1.714286; // 63 * 1200. / 44100.;
+ D->lp_window = BP_WINDOW_TRUNCATED;
+ }
+
+ // For scaling phase shift into normallized -1 to +1 range for mark and space.
+ D->u.afsk.normalize_rpsam = 1.0 / (0.5 * abs(mark_freq - space_freq) * 2 * M_PI / samples_per_sec);
+
+ D->pll_locked_inertia = 0.74;
+ D->pll_searching_inertia = 0.50;
+
+ D->alevel_mark_peak = -1; // FIXME: disable display
+ D->alevel_space_peak = -1;
+ break;
+
default:
text_color_set(DW_COLOR_ERROR);
- dw_printf ("Invalid filter profile = %c\n", profile);
+ dw_printf ("Invalid AFSK demodulator profile = %c\n", profile);
exit (1);
}
-#ifdef TUNE_PRE_WINDOW
- D->pre_window = TUNE_PRE_WINDOW;
-#endif
-#ifdef TUNE_MS_WINDOW
- D->ms_window = TUNE_MS_WINDOW;
-#endif
-#ifdef TUNE_MS2_WINDOW
- D->ms2_window = TUNE_MS2_WINDOW;
-#endif
-#ifdef TUNE_LP_WINDOW
- D->lp_window = TUNE_LP_WINDOW;
-#endif
+ TUNE("TUNE_PRE_BAUD", D->prefilter_baud, "prefilter_baud", "%.3f")
+ TUNE("TUNE_PRE_WINDOW", D->pre_window, "pre_window", "%d")
+
+ TUNE("TUNE_LPF_BAUD", D->lpf_baud, "lpf_baud", "%.3f")
+ TUNE("TUNE_LP_WINDOW", D->lp_window, "lp_window", "%d")
+
+ TUNE("TUNE_RRC_ROLLOFF", D->u.afsk.rrc_rolloff, "rrc_rolloff", "%.2f")
+ TUNE("TUNE_RRC_WIDTH_SYM", D->u.afsk.rrc_width_sym, "rrc_width_sym", "%.2f")
+
+ TUNE("TUNE_AGC_FAST", D->agc_fast_attack, "agc_fast_attack", "%.3f")
+ TUNE("TUNE_AGC_SLOW", D->agc_slow_decay, "agc_slow_decay", "%.6f")
+
+ TUNE("TUNE_PLL_LOCKED", D->pll_locked_inertia, "pll_locked_inertia", "%.2f")
+ TUNE("TUNE_PLL_SEARCHING", D->pll_searching_inertia, "pll_searching_inertia", "%.2f")
-#if defined(TUNE_AGC_FAST) && defined(TUNE_AGC_SLOW)
- D->agc_fast_attack = TUNE_AGC_FAST;
- D->agc_slow_decay = TUNE_AGC_SLOW;
-#endif
-#ifdef TUNE_HYST
- D->hysteresis = TUNE_HYST;
-#endif
-#if defined(TUNE_PLL_LOCKED) && defined(TUNE_PLL_SEARCHING)
- D->pll_locked_inertia = TUNE_PLL_LOCKED;
- D->pll_searching_inertia = TUNE_PLL_SEARCHING;
-#endif
-#ifdef TUNE_LPF_BAUD
- D->lpf_baud = TUNE_LPF_BAUD;
-#endif
-#ifdef TUNE_PRE_BAUD
- D->prefilter_baud = TUNE_PRE_BAUD;
-#endif
-#ifdef TUNE_LP_DELAY_FRACT
- D->lp_delay_fract = TUNE_LP_DELAY_FRACT;
-#endif
/*
* Calculate constants used for timing.
@@ -321,77 +417,28 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
D->pll_step_per_sample = (int) round((TICKS_PER_PLL_CYCLE * (double)baud) / ((double)samples_per_sec));
}
-/*
- * Convert number of bit times to number of taps.
- */
-
- D->pre_filter_size = (int) round( D->pre_filter_len_bits * (float)samples_per_sec / (float)baud );
- D->ms_filter_size = (int) round( D->ms_filter_len_bits * (float)samples_per_sec / (float)baud );
- D->lp_filter_size = (int) round( D->lp_filter_len_bits * (float)samples_per_sec / (float)baud );
-
-/* Experiment with other sizes. */
-
-#ifdef TUNE_PRE_FILTER_SIZE
- D->pre_filter_size = TUNE_PRE_FILTER_SIZE;
-#endif
-#ifdef TUNE_MS_FILTER_SIZE
- D->ms_filter_size = TUNE_MS_FILTER_SIZE;
-#endif
-#ifdef TUNE_LP_FILTER_SIZE
- D->lp_filter_size = TUNE_LP_FILTER_SIZE;
-#endif
-
- //assert (D->pre_filter_size >= 4);
- assert (D->ms_filter_size >= 4);
- //assert (D->lp_filter_size >= 4);
-
- if (D->pre_filter_size > MAX_FILTER_SIZE)
- {
- text_color_set (DW_COLOR_ERROR);
- dw_printf ("Calculated filter size of %d is too large.\n", D->pre_filter_size);
- dw_printf ("Decrease the audio sample rate or increase the baud rate or\n");
- dw_printf ("recompile the application with MAX_FILTER_SIZE larger than %d.\n",
- MAX_FILTER_SIZE);
- exit (1);
- }
-
- if (D->ms_filter_size > MAX_FILTER_SIZE)
- {
- text_color_set (DW_COLOR_ERROR);
- dw_printf ("Calculated filter size of %d is too large.\n", D->ms_filter_size);
- dw_printf ("Decrease the audio sample rate or increase the baud rate or\n");
- dw_printf ("recompile the application with MAX_FILTER_SIZE larger than %d.\n",
- MAX_FILTER_SIZE);
- exit (1);
- }
-
-
-
- if (D->lp_filter_size > MAX_FILTER_SIZE)
- {
- text_color_set (DW_COLOR_ERROR);
- dw_printf ("Calculated filter size of %d is too large.\n", D->pre_filter_size);
- dw_printf ("Decrease the audio sample rate or increase the baud rate or\n");
- dw_printf ("recompile the application with MAX_FILTER_SIZE larger than %d.\n",
- MAX_FILTER_SIZE);
- exit (1);
- }
-
/*
* Optionally apply a bandpass ("pre") filter to attenuate
* frequencies outside the range of interest.
- * This was first used for the "D" profile for 300 baud
- * which uses narrow shift. We expect it to have significant
- * benefit for a narrow shift.
- * In version 1.2, we will also try it with 1200 baud "E" as
- * an experiment to see how much it actually helps.
*/
if (D->use_prefilter) {
- float f1, f2;
- f1 = MIN(mark_freq,space_freq) - D->prefilter_baud * baud;
- f2 = MAX(mark_freq,space_freq) + D->prefilter_baud * baud;
+ // odd number is a little better
+ D->pre_filter_taps = ((int)( D->pre_filter_len_sym * (float)samples_per_sec / (float)baud )) | 1;
+
+ TUNE("TUNE_PRE_FILTER_TAPS", D->pre_filter_taps, "pre_filter_taps", "%d")
+
+ if (D->pre_filter_taps > MAX_FILTER_SIZE) {
+ text_color_set (DW_COLOR_ERROR);
+ dw_printf ("Calculated pre filter size of %d is too large.\n", D->pre_filter_taps);
+ dw_printf ("Decrease the audio sample rate or increase the decimation factor or\n");
+ dw_printf ("recompile the application with MAX_FILTER_SIZE larger than %d.\n", MAX_FILTER_SIZE);
+ D->pre_filter_taps = (MAX_FILTER_SIZE - 1) | 1;
+ }
+
+ float f1 = MIN(mark_freq,space_freq) - D->prefilter_baud * baud;
+ float f2 = MAX(mark_freq,space_freq) + D->prefilter_baud * baud;
#if 0
text_color_set(DW_COLOR_DEBUG);
dw_printf ("Generating prefilter %.0f to %.0f Hz.\n", f1, f2);
@@ -399,121 +446,67 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
f1 = f1 / (float)samples_per_sec;
f2 = f2 / (float)samples_per_sec;
- gen_bandpass (f1, f2, D->pre_filter, D->pre_filter_size, D->pre_window);
+ gen_bandpass (f1, f2, D->pre_filter, D->pre_filter_taps, D->pre_window);
}
-/*
- * Filters for detecting mark and space tones.
- */
-
-#if DEBUG1
- text_color_set(DW_COLOR_DEBUG);
- dw_printf ("%s: \n", __FILE__);
- dw_printf ("%d baud, %d samples_per_sec\n", baud, samples_per_sec);
- dw_printf ("AFSK %d & %d Hz\n", mark_freq, space_freq);
- dw_printf ("spll_step_per_sample = %d = 0x%08x\n", D->pll_step_per_sample, D->pll_step_per_sample);
- dw_printf ("D->ms_filter_size = %d = 0x%08x\n", D->ms_filter_size, D->ms_filter_size);
- dw_printf ("\n");
- dw_printf ("Mark\n");
- dw_printf (" j shape M sin M cos \n");
-#endif
-
-
- gen_ms (mark_freq, samples_per_sec, D->m_sin_table, D->m_cos_table, D->ms_filter_size, D->ms_window);
-
-#if DEBUG1
- text_color_set(DW_COLOR_DEBUG);
-
- dw_printf ("Space\n");
- dw_printf (" j shape S sin S cos\n");
-#endif
-
- gen_ms (space_freq, samples_per_sec, D->s_sin_table, D->s_cos_table, D->ms_filter_size, D->ms_window);
-
/*
* Now the lowpass filter.
- * I thought we'd want a cutoff of about 0.5 the baud rate
- * but it turns out about 1.1x is better. Still investigating...
+ * In version 1.7 a Root Raised Cosine filter is added as an alternative
+ * to the generic low pass filter.
+ * In both cases, lp_filter and lp_filter_taps are used but the
+ * contents will be generated differently. Later code does not care.
*/
+ if (D->u.afsk.use_rrc) {
- if (D->lpf_use_fir) {
- float fc;
- fc = baud * D->lpf_baud / (float)samples_per_sec;
- D->lp_filter_delay = gen_lowpass (fc, D->lp_filter, D->lp_filter_size, D->lp_window, D->lp_delay_fract);
+ assert (D->u.afsk.rrc_width_sym >= 1 && D->u.afsk.rrc_width_sym <= 16);
+ assert (D->u.afsk.rrc_rolloff >= 0. && D->u.afsk.rrc_rolloff <= 1.);
+
+ D->lp_filter_taps = ((int) (D->u.afsk.rrc_width_sym * (float)samples_per_sec / baud)) | 1; // odd works better
+
+ TUNE("TUNE_LP_FILTER_TAPS", D->lp_filter_taps, "lp_filter_taps (RRC)", "%d")
+
+ if (D->lp_filter_taps > MAX_FILTER_SIZE) {
+ text_color_set(DW_COLOR_ERROR);
+ dw_printf ("Calculated RRC low pass filter size of %d is too large.\n", D->lp_filter_taps);
+ dw_printf ("Decrease the audio sample rate or increase the decimation factor or\n");
+ dw_printf ("recompile the application with MAX_FILTER_SIZE larger than %d.\n", MAX_FILTER_SIZE);
+ D->lp_filter_taps = (MAX_FILTER_SIZE - 1) | 1;
+ }
+
+ assert (D->lp_filter_taps > 8 && D->lp_filter_taps <= MAX_FILTER_SIZE);
+ (void)gen_rrc_lowpass (D->lp_filter, D->lp_filter_taps, D->u.afsk.rrc_rolloff, (float)samples_per_sec / baud);
}
else {
- // D->lp_filter_delay =
- // Only needed for looking back and I don't expect to use IIR in that case.
+ D->lp_filter_taps = (int) round( D->lp_filter_width_sym * (float)samples_per_sec / (float)baud );
+
+ TUNE("TUNE_LP_FILTER_TAPS", D->lp_filter_taps, "lp_filter_taps (FIR)", "%d")
+
+ if (D->lp_filter_taps > MAX_FILTER_SIZE) {
+ text_color_set (DW_COLOR_ERROR);
+ dw_printf ("Calculated FIR low pass filter size of %d is too large.\n", D->lp_filter_taps);
+ dw_printf ("Decrease the audio sample rate or increase the decimation factor or\n");
+ dw_printf ("recompile the application with MAX_FILTER_SIZE larger than %d.\n", MAX_FILTER_SIZE);
+ D->lp_filter_taps = (MAX_FILTER_SIZE - 1) | 1;
+ }
+
+ assert (D->lp_filter_taps > 8 && D->lp_filter_taps <= MAX_FILTER_SIZE);
+
+ float fc = baud * D->lpf_baud / (float)samples_per_sec;
+ gen_lowpass (fc, D->lp_filter, D->lp_filter_taps, D->lp_window);
}
+
/*
- * A non-whole number of cycles results in a DC bias.
- * Let's see if it helps to take it out.
- * Actually makes things worse: 20 fewer decoded.
- * Might want to try again after EXPERIMENTC.
+ * Starting with version 1.2
+ * try using multiple slicing points instead of the traditional AGC.
*/
-
-#if 0
-#ifndef AVOID_FLOATING_POINT
-
-failed experiment
-
- dc_bias = 0;
- for (j=0; j<D->ms_filter_size; j++) {
- dc_bias += D->m_sin_table[j];
- }
- for (j=0; j<D->ms_filter_size; j++) {
- D->m_sin_table[j] -= dc_bias / D->ms_filter_size;
- }
-
- dc_bias = 0;
- for (j=0; j<D->ms_filter_size; j++) {
- dc_bias += D->m_cos_table[j];
- }
- for (j=0; j<D->ms_filter_size; j++) {
- D->m_cos_table[j] -= dc_bias / D->ms_filter_size;
- }
-
-
- dc_bias = 0;
- for (j=0; j<D->ms_filter_size; j++) {
- dc_bias += D->s_sin_table[j];
- }
- for (j=0; j<D->ms_filter_size; j++) {
- D->s_sin_table[j] -= dc_bias / D->ms_filter_size;
- }
-
- dc_bias = 0;
- for (j=0; j<D->ms_filter_size; j++) {
- dc_bias += D->s_cos_table[j];
- }
- for (j=0; j<D->ms_filter_size; j++) {
- D->s_cos_table[j] -= dc_bias / D->ms_filter_size;
- }
-
-#endif
-#endif
-
-/*
- * In version 1.2 we try another experiment.
- * Try using multiple slicing points instead of the traditional AGC.
- */
-
space_gain[0] = MIN_G;
float step = powf(10.0, log10f(MAX_G/MIN_G) / (MAX_SUBCHANS-1));
for (j=1; j<MAX_SUBCHANS; j++) {
space_gain[j] = space_gain[j-1] * step;
}
-#if 0
- text_color_set(DW_COLOR_DEBUG);
- for (j=0; j<MAX_SUBCHANS; j++) {
- float db = 20.0 * log10f(space_gain[j]);
- dw_printf ("G = %.3f, %+.1f dB\n", space_gain[j], db);
- }
-#endif
-
-} /* fsk_gen_filter */
+} /* demod_afsk_init */
@@ -531,16 +524,10 @@ failed experiment
*
* Returns: None
*
- * Descripion: We start off with two bandpass filters tuned to
- * the given frequencies. In the case of VHF packet
- * radio, this would be 1200 and 2200 Hz.
+ * Descripion: First demodulate the AFSK signal.
*
- * The bandpass filter amplitudes are compared to
- * obtain the demodulated signal.
- *
- * We also have a digital phase locked loop (PLL)
- * to recover the clock and pick out data bits at
- * the proper rate.
+ * A digital phase locked loop (PLL) recovers the symbol
+ * clock and picks out data bits at the proper rate.
*
* For each recovered data bit, we call:
*
@@ -552,145 +539,21 @@ failed experiment
* of the function to be called for each bit recovered
* from the demodulator. For now, it's simply hard-coded.
*
+ * Evolution: The simple version works less well when there is a substantial difference
+ * in amplitude of the two tones. e.g. When de-emphasis cuts the
+ * higher tone down to about half the amplitude. We overcome that
+ * by boosting the space amplitude by varying amounts before slicing.
+ *
+ * In version 1.7 an entirely different approach is added, an FM
+ * discriminator which produces a result proportional to the
+ * frequency.
+ *
*--------------------------------------------------------------------*/
-
-inline static void nudge_pll (int chan, int subchan, int slice, int demod_data, struct demodulator_state_s *D);
-
-__attribute__((hot))
-void demod_afsk_process_sample (int chan, int subchan, int sam, struct demodulator_state_s *D)
-{
- float fsam;
- //float abs_fsam;
- float m_sum1, m_sum2, s_sum1, s_sum2;
- float m_amp, s_amp;
- float m_norm, s_norm;
- float demod_out;
-#if DEBUG4
- static FILE *demod_log_fp = NULL;
- static int seq = 0; /* for log file name */
-#endif
-
-
- //int j;
- int demod_data;
-
-
- assert (chan >= 0 && chan < MAX_CHANS);
- assert (subchan >= 0 && subchan < MAX_SUBCHANS);
-
-/*
- * Filters use last 'filter_size' samples.
- *
- * First push the older samples down.
- *
- * Finally, put the most recent at the beginning.
- *
- * Future project? Can we do better than shifting each time?
- */
-
- /* Scale to nice number, TODO: range -1.0 to +1.0, not 2. */
-
- fsam = sam / 16384.0f;
-
- //abs_fsam = fsam >= 0.0f ? fsam : -fsam;
-
-
-/*
- * Optional bandpass filter before the mark/space discriminator.
- */
-
-// FIXME: calculate how much we really need.
-
- int extra = 0;
-
- if (D->use_prefilter) {
- float cleaner;
-
- push_sample (fsam, D->raw_cb, D->pre_filter_size);
- cleaner = convolve (D->raw_cb, D->pre_filter, D->pre_filter_size);
- push_sample (cleaner, D->ms_in_cb, D->ms_filter_size + extra);
- }
- else {
- push_sample (fsam, D->ms_in_cb, D->ms_filter_size + extra);
- }
-
-/*
- * Next we have bandpass filters for the mark and space tones.
- */
-
-/*
- * find amplitude of "Mark" tone.
- */
- m_sum1 = convolve (D->ms_in_cb, D->m_sin_table, D->ms_filter_size);
- m_sum2 = convolve (D->ms_in_cb, D->m_cos_table, D->ms_filter_size);
-
- m_amp = sqrtf(m_sum1 * m_sum1 + m_sum2 * m_sum2);
-
-/*
- * Find amplitude of "Space" tone.
- */
- s_sum1 = convolve (D->ms_in_cb, D->s_sin_table, D->ms_filter_size);
- s_sum2 = convolve (D->ms_in_cb, D->s_cos_table, D->ms_filter_size);
-
- s_amp = sqrtf(s_sum1 * s_sum1 + s_sum2 * s_sum2);
-
-
-/*
- * Apply some low pass filtering BEFORE the AGC to remove
- * overshoot, ringing, and other bad stuff.
- *
- * A simple IIR filter is faster but FIR produces better results.
- *
- * It is a balancing act between removing high frequency components
- * from the tone dectection while letting the data thru.
- */
-
- if (D->lpf_use_fir) {
-
- push_sample (m_amp, D->m_amp_cb, D->lp_filter_size);
- m_amp = convolve (D->m_amp_cb, D->lp_filter, D->lp_filter_size);
-
- push_sample (s_amp, D->s_amp_cb, D->lp_filter_size);
- s_amp = convolve (D->s_amp_cb, D->lp_filter, D->lp_filter_size);
- }
- else {
-
- /* Original, but faster, IIR. */
-
- m_amp = D->lpf_iir * m_amp + (1.0f - D->lpf_iir) * D->m_amp_prev;
- D->m_amp_prev = m_amp;
-
- s_amp = D->lpf_iir * s_amp + (1.0f - D->lpf_iir) * D->s_amp_prev;
- D->s_amp_prev = s_amp;
- }
-
-/*
- * Version 1.2: Try new approach to capturing the amplitude for display.
- * This is same as the AGC above without the normalization step.
- * We want decay to be substantially slower to get a longer
- * range idea of the received audio.
- */
-
- if (m_amp >= D->alevel_mark_peak) {
- D->alevel_mark_peak = m_amp * D->quick_attack + D->alevel_mark_peak * (1.0f - D->quick_attack);
- }
- else {
- D->alevel_mark_peak = m_amp * D->sluggish_decay + D->alevel_mark_peak * (1.0f - D->sluggish_decay);
- }
-
- if (s_amp >= D->alevel_space_peak) {
- D->alevel_space_peak = s_amp * D->quick_attack + D->alevel_space_peak * (1.0f - D->quick_attack);
- }
- else {
- D->alevel_space_peak = s_amp * D->sluggish_decay + D->alevel_space_peak * (1.0f - D->sluggish_decay);
- }
-
-
/*
* Which tone is stronger?
*
* In an ideal world, simply compare. In my first naive attempt, that
- * worked perfectly with perfect signals. In the real world, we don't
+ * worked well with perfect signals. In the real world, we don't
* have too many perfect signals.
*
* Here is an excellent explanation:
@@ -724,51 +587,207 @@ void demod_afsk_process_sample (int chan, int subchan, int sam, struct demodulat
*
* This is similar to my observations of local signals, from the speaker.
* The amplitude ratio varies from 1.48 to 3.41 with a median of 2.70.
- *
- * Rather than only two filters, let's try slicing the data in more places.
*/
- /* Fast attack and slow decay. */
- /* Numbers were obtained by trial and error from actual */
- /* recorded less-than-optimal signals. */
- /* See fsk_demod_agc.h for more information. */
- m_norm = agc (m_amp, D->agc_fast_attack, D->agc_slow_decay, &(D->m_peak), &(D->m_valley));
- s_norm = agc (s_amp, D->agc_fast_attack, D->agc_slow_decay, &(D->s_peak), &(D->s_valley));
+__attribute__((hot))
+void demod_afsk_process_sample (int chan, int subchan, int sam, struct demodulator_state_s *D)
+{
+#if DEBUG4
+ static FILE *demod_log_fp = NULL;
+ static int seq = 0; /* for log file name */
+#endif
- if (D->num_slicers <= 1) {
+ assert (chan >= 0 && chan < MAX_CHANS);
+ assert (subchan >= 0 && subchan < MAX_SUBCHANS);
- /* Normal case of one demodulator to one HDLC decoder. */
- /* Demodulator output is difference between response from two filters. */
- /* AGC should generally keep this around -1 to +1 range. */
+/*
+ * Filters use last 'filter_taps' samples.
+ *
+ * First push the older samples down.
+ *
+ * Finally, put the most recent at the beginning.
+ *
+ * Future project? Can we do better than shifting each time?
+ */
- demod_out = m_norm - s_norm;
+ /* Scale to nice number. */
- /* Try adding some Hysteresis. */
- /* (Not to be confused with Hysteria.) */
+ float fsam = (float)sam / 16384.0f;
+
+ switch (D->profile) {
+
+ case 'E':
+ default:
+ case 'A': {
+ /* ========== New in Version 1.7 ========== */
+
+ // Cleaner & simpler than earlier 'A' thru 'E'
+
+ if (D->use_prefilter) {
+ push_sample (fsam, D->raw_cb, D->pre_filter_taps);
+ fsam = convolve (D->raw_cb, D->pre_filter, D->pre_filter_taps);
+ }
+
+ push_sample (fsam * fcos256(D->u.afsk.m_osc_phase), D->u.afsk.m_I_raw, D->lp_filter_taps);
+ push_sample (fsam * fsin256(D->u.afsk.m_osc_phase), D->u.afsk.m_Q_raw, D->lp_filter_taps);
+ D->u.afsk.m_osc_phase += D->u.afsk.m_osc_delta;
+
+ push_sample (fsam * fcos256(D->u.afsk.s_osc_phase), D->u.afsk.s_I_raw, D->lp_filter_taps);
+ push_sample (fsam * fsin256(D->u.afsk.s_osc_phase), D->u.afsk.s_Q_raw, D->lp_filter_taps);
+ D->u.afsk.s_osc_phase += D->u.afsk.s_osc_delta;
+
+ float m_I = convolve (D->u.afsk.m_I_raw, D->lp_filter, D->lp_filter_taps);
+ float m_Q = convolve (D->u.afsk.m_Q_raw, D->lp_filter, D->lp_filter_taps);
+ float m_amp = fast_hypot(m_I, m_Q);
+
+ float s_I = convolve (D->u.afsk.s_I_raw, D->lp_filter, D->lp_filter_taps);
+ float s_Q = convolve (D->u.afsk.s_Q_raw, D->lp_filter, D->lp_filter_taps);
+ float s_amp = fast_hypot(s_I, s_Q);
+
+/*
+ * Capture the mark and space peak amplitudes for display.
+ * It uses fast attack and slow decay to get an idea of the
+ * overall amplitude.
+ */
+ if (m_amp >= D->alevel_mark_peak) {
+ D->alevel_mark_peak = m_amp * D->quick_attack + D->alevel_mark_peak * (1.0f - D->quick_attack);
+ }
+ else {
+ D->alevel_mark_peak = m_amp * D->sluggish_decay + D->alevel_mark_peak * (1.0f - D->sluggish_decay);
+ }
+
+ if (s_amp >= D->alevel_space_peak) {
+ D->alevel_space_peak = s_amp * D->quick_attack + D->alevel_space_peak * (1.0f - D->quick_attack);
+ }
+ else {
+ D->alevel_space_peak = s_amp * D->sluggish_decay + D->alevel_space_peak * (1.0f - D->sluggish_decay);
+ }
+
+ if (D->num_slicers <= 1) {
+
+ // Which tone is stonger? That's simple with an ideal signal.
+ // However, we don't see too many ideal signals.
+ // Due to mismatching pre-emphasis and de-emphasis, the two
+ // tones will often have greatly different amplitudes so we use
+ // automatic gain control (AGC) to scale each to the same range
+ // before comparing.
+ // This is probably over complicated and could be combined with
+ // the signal amplitude measurement, above.
+ // It works so let's move along to other topics.
+
+ float m_norm = agc (m_amp, D->agc_fast_attack, D->agc_slow_decay, &(D->m_peak), &(D->m_valley));
+ float s_norm = agc (s_amp, D->agc_fast_attack, D->agc_slow_decay, &(D->s_peak), &(D->s_valley));
+
+ // The normalized values should be around -0.5 to +0.5 so the difference
+ // should work out to be around -1 to +1.
+ // This is important because nudge_pll uses the demod_out amplitude to assign
+ // a quality or confidence score to the symbol.
+
+ float demod_out = m_norm - s_norm;
+
+ // Tested and it looks good. Range of about -1 to +1.
+ //printf ("JWL DEBUG demod A with agc = %6.2f\n", demod_out);
+
+ nudge_pll (chan, subchan, 0, demod_out, D, 1.0);
+
+ }
+ else {
+ // Multiple slice case.
+ // Rather than trying to find the best threshold location, use multiple
+ // slicer thresholds in parallel.
+ // The best slicing point will vary from packet to packet but should
+ // remain abount the same or a given packet.
+
+ // We are not performing the AGC step here but still want the envelope
+ // for caluculating the confidence level (or quality) of the sample.
+
+ (void) agc (m_amp, D->agc_fast_attack, D->agc_slow_decay, &(D->m_peak), &(D->m_valley));
+ (void) agc (s_amp, D->agc_fast_attack, D->agc_slow_decay, &(D->s_peak), &(D->s_valley));
+
+ for (int slice=0; slice<D->num_slicers; slice++) {
+ float demod_out = m_amp - s_amp * space_gain[slice];
+ float amp = 0.5f * (D->m_peak - D->m_valley + (D->s_peak - D->s_valley) * space_gain[slice]);
+ if (amp < 0.0000001f) amp = 1; // avoid divide by zero with no signal.
+
+ // Tested and it looks good. Range of about -1 to +1 relative to amp.
+ // Biased one way or the other depending on the space gain.
+ //printf ("JWL DEBUG demod A with slicer %d: %6.2f / %6.2f = %6.2f\n", slice, demod_out, amp, demod_out/amp);
+
+ nudge_pll (chan, subchan, slice, demod_out, D, amp);
+ }
+ }
+ }
+ break;
+
+ case 'D':
+ case 'B': {
+ /* ========== Version 1.7 Experiment ========== */
+
+ // New - Convert frequency to a value proportional to frequency.
+
+ if (D->use_prefilter) {
+ push_sample (fsam, D->raw_cb, D->pre_filter_taps);
+ fsam = convolve (D->raw_cb, D->pre_filter, D->pre_filter_taps);
+ }
+
+ push_sample (fsam * fcos256(D->u.afsk.c_osc_phase), D->u.afsk.c_I_raw, D->lp_filter_taps);
+ push_sample (fsam * fsin256(D->u.afsk.c_osc_phase), D->u.afsk.c_Q_raw, D->lp_filter_taps);
+ D->u.afsk.c_osc_phase += D->u.afsk.c_osc_delta;
+
+ float c_I = convolve (D->u.afsk.c_I_raw, D->lp_filter, D->lp_filter_taps);
+ float c_Q = convolve (D->u.afsk.c_Q_raw, D->lp_filter, D->lp_filter_taps);
+
+ float phase = atan2f (c_Q, c_I);
+ float rate = phase - D->u.afsk.prev_phase;
+ if (rate > M_PI) rate -= 2 * M_PI;
+ else if (rate < -M_PI) rate += 2 * M_PI;
+ D->u.afsk.prev_phase = phase;
+
+ // Rate is radians per audio sample interval or something like that.
+ // Scale scale that into -1 to +1 for expected tones.
+
+ float norm_rate = rate * D->u.afsk.normalize_rpsam;
+
+ // We really don't have mark and space amplitudes available in this case.
+
+ if (D->num_slicers <= 1) {
+
+ float demod_out = norm_rate;
+ // Tested and it looks good. Range roughly -1 to +1.
+ //printf ("JWL DEBUG demod B single = %6.2f\n", demod_out);
+
+ nudge_pll (chan, subchan, 0, demod_out, D, 1.0);
- if (demod_out > D->hysteresis) {
- demod_data = 1;
}
- else if (demod_out < (- (D->hysteresis))) {
- demod_data = 0;
- }
else {
- demod_data = D->slicer[subchan].prev_demod_data;
+
+ // This would be useful for HF SSB where a tuning error
+ // would shift the frequency. Multiple slicing points would
+ // then compensate for differences in transmit/receive frequencies.
+ //
+ // Where should we set the thresholds?
+ // I'm thinking something like:
+ // -.5 -.375 -.25 -.125 0 .125 .25 .375 .5
+ //
+ // Assuming a 300 Hz shift, this would put slicing thresholds up
+ // to +-75 Hz from the center.
+
+ for (int slice=0; slice<D->num_slicers; slice++) {
+
+ float offset = -0.5 + slice * (1. / (D->num_slicers - 1));
+ float demod_out = norm_rate + offset;
+
+ //printf ("JWL DEBUG demod B slice %d, offset = %6.3f, demod_out = %6.2f\n", slice, offset, demod_out);
+
+ nudge_pll (chan, subchan, slice, demod_out, D, 1.0);
+ }
}
- nudge_pll (chan, subchan, 0, demod_data, D);
- }
- else {
- int slice;
-
- for (slice=0; slice<D->num_slicers; slice++) {
- demod_data = m_amp > s_amp * space_gain[slice];
- nudge_pll (chan, subchan, slice, demod_data, D);
}
+ break;
}
-
-
+
#if DEBUG4
if (chan == 0) {
@@ -805,9 +824,6 @@ void demod_afsk_process_sample (int chan, int subchan, int sam, struct demodulat
} /* end demod_afsk_process_sample */
-__attribute__((hot))
-inline static void nudge_pll (int chan, int subchan, int slice, int demod_data, struct demodulator_state_s *D)
-{
/*
* Finally, a PLL is used to sample near the centers of the data bits.
@@ -839,6 +855,9 @@ inline static void nudge_pll (int chan, int subchan, int slice, int demod_data,
* because this happens for each transition from the demodulator.
*/
+__attribute__((hot))
+static void nudge_pll (int chan, int subchan, int slice, float demod_out, struct demodulator_state_s *D, float amplitude)
+{
D->slicer[slice].prev_d_c_pll = D->slicer[slice].data_clock_pll;
// Perform the add as unsigned to avoid signed overflow error.
@@ -850,12 +869,19 @@ inline static void nudge_pll (int chan, int subchan, int slice, int demod_data,
if (D->slicer[slice].data_clock_pll < 0 && D->slicer[slice].prev_d_c_pll > 0) {
/* Overflow - this is where we sample. */
- hdlc_rec_bit (chan, subchan, slice, demod_data, 0, -1);
+ // Assign it a confidence level or quality, 0 to 100, based on the amplitude.
+ // Those very close to 0 are suspect. We'll get back to this later.
+
+ int quality = fabsf(demod_out) * 100.0f / amplitude;
+ if (quality > 100) quality = 100;
+
+ hdlc_rec_bit (chan, subchan, slice, demod_out > 0, 0, quality);
pll_dcd_each_symbol2 (D, chan, subchan, slice);
}
// Transitions nudge the DPLL phase toward the incoming signal.
+ int demod_data = demod_out > 0;
if (demod_data != D->slicer[slice].prev_demod_data) {
pll_dcd_signal_transition2 (D, slice, D->slicer[slice].data_clock_pll);
diff --git a/src/demod_psk.c b/src/demod_psk.c
index f01ee21..b953add 100644
--- a/src/demod_psk.c
+++ b/src/demod_psk.c
@@ -501,7 +501,7 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
*/
float fc = correct_baud * D->u.psk.lpf_baud / (float)samples_per_sec;
- gen_lowpass (fc, D->u.psk.lp_filter, D->u.psk.lp_filter_taps, D->u.psk.lp_window, 0);
+ gen_lowpass (fc, D->u.psk.lp_filter, D->u.psk.lp_filter_taps, D->u.psk.lp_window);
/*
* No point in having multiple numbers for signal level.
diff --git a/src/dsp.c b/src/dsp.c
index 6ba7094..4a5f4a8 100644
--- a/src/dsp.c
+++ b/src/dsp.c
@@ -43,7 +43,6 @@
#include "dsp.h"
-//#include "fsk_demod_agc.h" /* for M_FILTER_SIZE, etc. */
#define MIN(a,b) ((a)<(b)?(a):(b))
#define MAX(a,b) ((a)>(b)?(a):(b))
@@ -127,7 +126,7 @@ float window (bp_window_t type, int size, int j)
*----------------------------------------------------------------*/
-int gen_lowpass (float fc, float *lp_filter, int filter_size, bp_window_t wtype, float lp_delay_fract)
+void gen_lowpass (float fc, float *lp_filter, int filter_size, bp_window_t wtype)
{
int j;
float G;
@@ -175,54 +174,7 @@ int gen_lowpass (float fc, float *lp_filter, int filter_size, bp_window_t wtype,
lp_filter[j] = lp_filter[j] / G;
}
-
-// Calculate the signal delay.
-// If a signal at level 0 steps to level 1, this is the time that it would
-// take for the output to reach 0.5.
-//
-// Examples:
-//
-// Filter has one tap with value of 1.0.
-// Output is immediate so I would call this delay of 0.
-//
-// Filter coefficients: 0.2, 0.2, 0.2, 0.2, 0.2
-// "1" inputs Out
-// 1 0.2
-// 2 0.4
-// 3 0.6
-//
-// In this case, the output does not change immediately.
-// It takes two more samples to reach the half way point
-// so it has a delay of 2.
-
- float sum = 0;
- int delay = 0;
-
- if (lp_delay_fract == 0) lp_delay_fract = 0.5;
-
- for (j=0; j<filter_size; j++) {
- sum += lp_filter[j];
-#if DEBUG1
- dw_printf ("lp_filter[%d] = %.3f sum = %.3f lp_delay_fract = %.3f\n", j, lp_filter[j], sum, lp_delay_fract);
-#endif
- if (sum > lp_delay_fract) {
- delay = j;
- break;
- }
- }
-
-#if DEBUG1
- dw_printf ("Low Pass Delay = %d samples\n", delay) ;
-#endif
-
-// Hmmm. This might have been wasted effort. The result is always half the number of taps.
-
- if (delay < 2 || delay > filter_size - 2) {
- text_color_set(DW_COLOR_ERROR);
- dw_printf ("Internal error, %s %d, delay %d for size %d\n", __func__, __LINE__, delay, filter_size);
- }
-
- return (delay);
+ return;
} /* end gen_lowpass */
@@ -369,4 +321,86 @@ void gen_ms (int fc, int sps, float *sin_table, float *cos_table, int filter_siz
} /* end gen_ms */
+
+
+
+/*------------------------------------------------------------------
+ *
+ * Name: rrc
+ *
+ * Purpose: Root Raised Cosine function.
+ * Why do they call it that?
+ * It's mostly the sinc function with cos windowing to taper off edges faster.
+ *
+ * Inputs: t - Time in units of symbol duration.
+ * i.e. The centers of two adjacent symbols would differ by 1.
+ *
+ * a - Roll off factor, between 0 and 1.
+ *
+ * Returns: Basically the sinc (sin(x)/x) function with edges decreasing faster.
+ * Should be 1 for t = 0 and 0 at all other integer values of t.
+ *
+ *----------------------------------------------------------------*/
+
+__attribute__((const))
+float rrc (float t, float a)
+{
+ float sinc, window, result;
+
+ if (t > -0.001 && t < 0.001) {
+ sinc = 1;
+ }
+ else {
+ sinc = sinf(M_PI * t) / (M_PI * t);
+ }
+
+ if (fabsf(a * t) > 0.499 && fabsf(a * t) < 0.501) {
+ window = M_PI / 4;
+ }
+ else {
+ window = cos(M_PI * a * t) / ( 1 - powf(2 * a * t, 2));
+ // This made nicer looking waveforms for generating signal.
+ //window = cos(M_PI * a * t);
+ // Do we want to let it go negative?
+ // I think this would happen when a > 0.5 / (filter width in symbol times)
+ if (window < 0) {
+ //printf ("'a' is too large for range of 't'.\n");
+ //window = 0;
+ }
+ }
+
+ result = sinc * window;
+
+#if DEBUGRRC
+ // t should vary from - to + half of filter size in symbols.
+ // Result should be 1 at t=0 and 0 at all other integer values of t.
+
+ printf ("%.3f, %.3f, %.3f, %.3f\n", t, sinc, window, result);
+#endif
+ return (result);
+}
+
+// The Root Raised Cosine (RRC) low pass filter is suppposed to minimize Intersymbol Interference (ISI).
+
+void gen_rrc_lowpass (float *pfilter, int filter_taps, float rolloff, float samples_per_symbol)
+{
+ int k;
+ float t;
+
+ for (k = 0; k < filter_taps; k++) {
+ t = (k - ((filter_taps - 1.0) / 2.0)) / samples_per_symbol;
+ pfilter[k] = rrc (t, rolloff);
+ }
+
+ // Scale it for unity gain.
+
+ t = 0;
+ for (k = 0; k < filter_taps; k++) {
+ t += pfilter[k];
+ }
+ for (k = 0; k < filter_taps; k++) {
+ pfilter[k] = pfilter[k] / t;
+ }
+}
+
/* end dsp.c */
diff --git a/src/dsp.h b/src/dsp.h
index 5d5b882..e0dbd24 100644
--- a/src/dsp.h
+++ b/src/dsp.h
@@ -5,9 +5,13 @@
float window (bp_window_t type, int size, int j);
-int gen_lowpass (float fc, float *lp_filter, int filter_size, bp_window_t wtype, float lp_delay_fract);
+void gen_lowpass (float fc, float *lp_filter, int filter_size, bp_window_t wtype);
void gen_bandpass (float f1, float f2, float *bp_filter, int filter_size, bp_window_t wtype);
void gen_ms (int fc, int samples_per_sec, float *sin_table, float *cos_table, int filter_size, int wtype);
+
+__attribute__((const)) float rrc (float t, float a);
+
+void gen_rrc_lowpass (float *pfilter, int filter_taps, float rolloff, float samples_per_symbol);
diff --git a/src/fsk_demod_agc.h b/src/fsk_demod_agc.h
deleted file mode 100644
index 95c8079..0000000
--- a/src/fsk_demod_agc.h
+++ /dev/null
@@ -1,2 +0,0 @@
-#define TUNE_MS_FILTER_SIZE 140
-#define TUNE_PRE_BAUD 1.080
diff --git a/src/fsk_demod_state.h b/src/fsk_demod_state.h
index f7b5650..33f7901 100644
--- a/src/fsk_demod_state.h
+++ b/src/fsk_demod_state.h
@@ -20,6 +20,31 @@ typedef enum bp_window_e { BP_WINDOW_TRUNCATED,
BP_WINDOW_BLACKMAN,
BP_WINDOW_FLATTOP } bp_window_t;
+// Experimental low pass filter to detect DC bias or low frequency changes.
+// IIR behaves like an analog R-C filter.
+// Intuitively, it seems like FIR would be better because it is based on a finite history.
+// However, it would require MANY taps and a LOT of computation for a low frequency.
+// We can use a little trick here to keep a running average.
+// This would be equivalent to convolving with an array of all 1 values.
+// That would eliminate the need to multiply.
+// We can also eliminate the need to add them all up each time by keeping a running total.
+// Add a sample to the total when putting it in our array of recent samples.
+// Subtract it from the total when it gets pushed off the end.
+// We can also eliminate the need to shift them all down by using a circular buffer.
+
+#define CIC_LEN_MAX 4000
+
+typedef struct cic_s {
+ int len; // Number of elements used.
+ // Might want to dynamically allocate.
+ short in[CIC_LEN_MAX]; // Samples coming in.
+ int sum; // Running sum.
+ int inext; // Next position to fill.
+} cic_t;
+
+
+#define MAX_FILTER_SIZE 404 /* 401 is needed for profile A, 300 baud & 44100. Revisit someday. */
+
struct demodulator_state_s
{
@@ -39,30 +64,12 @@ struct demodulator_state_s
// Data is sampled when it overflows.
- int ms_filter_size; /* Size of mark & space filters, in audio samples. */
- /* Started off as a guess of one bit length */
- /* but about 2 bit times turned out to be better. */
- /* Currently using same size for any prefilter. */
-
-
-#define MAX_FILTER_SIZE 320 /* 304 is needed for profile C, 300 baud & 44100. */
-
-/*
- * Filter length for Mark & Space in bit times.
- * e.g. 1 means 1/1200 second for 1200 baud.
- */
- float ms_filter_len_bits;
- float lp_delay_fract;
-
/*
* Window type for the various filters.
*/
- bp_window_t pre_window;
- bp_window_t ms_window;
bp_window_t lp_window;
-
/*
* Alternate Low pass filters.
* First is arbitrary number for quick IIR.
@@ -78,16 +85,13 @@ struct demodulator_state_s
/* In practice, it turned out a little larger */
/* for profiles B, C, D. */
- float lp_filter_len_bits; /* Length in number of bit times. */
+ float lp_filter_width_sym; /* Length in number of symbol times. */
- int lp_filter_size; /* Size of Low Pass filter, in audio samples. */
- /* Previously it was always the same as the M/S */
- /* filters but in version 1.2 it's now independent. */
+#define lp_filter_len_bits lp_filter_width_sym // FIXME: temp hack
- int lp_filter_delay; /* Number of samples that the low pass filter */
- /* delays the signal. */
-
- /* New in 1.6. */
+ int lp_filter_taps; /* Size of Low Pass filter, in audio samples. */
+
+#define lp_filter_size lp_filter_taps // FIXME: temp hack
/*
@@ -111,6 +115,7 @@ struct demodulator_state_s
/*
* Phase Locked Loop (PLL) inertia.
* Larger number means less influence by signal transitions.
+ * It is more resistant to change when locked on to a signal.
*/
float pll_locked_inertia;
float pll_searching_inertia;
@@ -129,23 +134,17 @@ struct demodulator_state_s
/* lower = min(1600,1800) - 0.5 * 300 = 1450 */
/* upper = max(1600,1800) + 0.5 * 300 = 1950 */
- float pre_filter_len_bits; /* Length in number of bit times. */
+ float pre_filter_len_sym; // Length in number of symbol times.
+#define pre_filter_len_bits pre_filter_len_sym // temp until all references changed.
- int pre_filter_size; /* Size of pre filter, in audio samples. */
+ bp_window_t pre_window; // Window type for filter shaping.
+
+ int pre_filter_taps; // Calculated number of filter taps.
+#define pre_filter_size pre_filter_taps // temp until all references changed.
float pre_filter[MAX_FILTER_SIZE] __attribute__((aligned(16)));
-
-/*
- * Kernel for the mark and space detection filters.
- */
-
- float m_sin_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
- float m_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
-
- float s_sin_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
- float s_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
-
+ float raw_cb[MAX_FILTER_SIZE] __attribute__((aligned(16))); // audio in, need better name.
/*
* The rest are continuously updated.
@@ -154,11 +153,6 @@ struct demodulator_state_s
unsigned int lo_phase; /* Local oscillator for PSK. */
-/*
- * Most recent raw audio samples, before/after prefiltering.
- */
- float raw_cb[MAX_FILTER_SIZE] __attribute__((aligned(16)));
-
/*
* Use half of the AGC code to get a measure of input audio amplitude.
* These use "quick" attack and "sluggish" decay while the
@@ -170,24 +164,14 @@ struct demodulator_state_s
float alevel_mark_peak;
float alevel_space_peak;
-/*
- * Input to the mark/space detector.
- * Could be prefiltered or raw audio.
- */
- float ms_in_cb[MAX_FILTER_SIZE] __attribute__((aligned(16)));
-
/*
* Outputs from the mark and space amplitude detection,
* used as inputs to the FIR lowpass filters.
* Kernel for the lowpass filters.
*/
- float m_amp_cb[MAX_FILTER_SIZE] __attribute__((aligned(16)));
- float s_amp_cb[MAX_FILTER_SIZE] __attribute__((aligned(16)));
-
float lp_filter[MAX_FILTER_SIZE] __attribute__((aligned(16)));
-
float m_peak, s_peak;
float m_valley, s_valley;
float m_amp_prev, s_amp_prev;
@@ -266,6 +250,113 @@ struct demodulator_state_s
union {
+//////////////////////////////////////////////////////////////////////////////////
+// //
+// AFSK only - new method in 1.7 //
+// //
+//////////////////////////////////////////////////////////////////////////////////
+
+
+ struct afsk_only_s {
+
+ unsigned int m_osc_phase; // Phase for Mark local oscillator.
+ unsigned int m_osc_delta; // How much to change for each audio sample.
+
+ unsigned int s_osc_phase; // Phase for Space local oscillator.
+ unsigned int s_osc_delta; // How much to change for each audio sample.
+
+ unsigned int c_osc_phase; // Phase for Center frequency local oscillator.
+ unsigned int c_osc_delta; // How much to change for each audio sample.
+
+ // Need two mixers for profile "A".
+
+ float m_I_raw[MAX_FILTER_SIZE] __attribute__((aligned(16)));
+ float m_Q_raw[MAX_FILTER_SIZE] __attribute__((aligned(16)));
+
+ float s_I_raw[MAX_FILTER_SIZE] __attribute__((aligned(16)));
+ float s_Q_raw[MAX_FILTER_SIZE] __attribute__((aligned(16)));
+
+ // Only need one mixer for profile "B". Reuse the same storage?
+
+//#define c_I_raw m_I_raw
+//#define c_Q_raw m_Q_raw
+ float c_I_raw[MAX_FILTER_SIZE] __attribute__((aligned(16)));
+ float c_Q_raw[MAX_FILTER_SIZE] __attribute__((aligned(16)));
+
+ int use_rrc; // Use RRC rather than generic low pass.
+
+ float rrc_width_sym; /* Width of RRC filter in number of symbols. */
+
+ float rrc_rolloff; /* Rolloff factor for RRC. Between 0 and 1. */
+
+ float prev_phase; // To see phase shift between samples for FM demod.
+
+ float normalize_rpsam; // Normalize to -1 to +1 for expected tones.
+
+ } afsk;
+
+//////////////////////////////////////////////////////////////////////////////////
+// //
+// Baseband only, AKA G3RUH //
+// //
+//////////////////////////////////////////////////////////////////////////////////
+
+
+ struct bb_only_s {
+
+ float rrc_width_sym; /* Width of RRC filter in number of symbols. */
+
+ float rrc_rolloff; /* Rolloff factor for RRC. Between 0 and 1. */
+
+ int rrc_filter_taps; // Number of elements used in the next two.
+
+// FIXME: TODO: reevaluate max size needed.
+
+ float audio_in[MAX_FILTER_SIZE] __attribute__((aligned(16))); // Audio samples in.
+
+// FIXME: use lp_filter
+ float rrc_filter[MAX_FILTER_SIZE] __attribute__((aligned(16))); // RRC Low pass filter.
+
+ float lp_1_iir_param; // very low pass filters to get DC offset.
+ float lp_1_out;
+
+ float lp_2_iir_param;
+ float lp_2_out;
+
+ float agc_1_fast_attack; // Signal envelope detection.
+ float agc_1_slow_decay;
+ float agc_1_peak;
+ float agc_1_valley;
+
+ float agc_2_fast_attack;
+ float agc_2_slow_decay;
+ float agc_2_peak;
+ float agc_2_valley;
+
+ float agc_3_fast_attack;
+ float agc_3_slow_decay;
+ float agc_3_peak;
+ float agc_3_valley;
+
+ // CIC low pass filters to detect DC bias or low frequency changes.
+ // IIR behaves like an analog R-C filter.
+ // Intuitively, it seems like FIR would be better because it is based on a finite history.
+ // However, it would require MANY taps and a LOT of computation for a low frequency.
+ // We can use a little trick here to keep a running average.
+ // This would be equivalent to convolving with an array of all 1 values.
+ // That would eliminate the need to multiply.
+ // We can also eliminate the need to add them all up each time by keeping a running total.
+ // Add a sample to the total when putting it in our array of recent samples.
+ // Subtract it from the total when it gets pushed off the end.
+ // We can also eliminate the need to shift them all down by using a circular buffer.
+ // This only works with integers because float would have cummulated round off errors.
+
+ cic_t cic_center1;
+ cic_t cic_above;
+ cic_t cic_below;
+
+ } bb;
+
//////////////////////////////////////////////////////////////////////////////////
// //
// PSK only. //
diff --git a/test/scripts/check-modem1200 b/test/scripts/check-modem1200
index 2b97587..0c79af9 100755
--- a/test/scripts/check-modem1200
+++ b/test/scripts/check-modem1200
@@ -1,5 +1,7 @@
@CUSTOM_SHELL_SHABANG@
@GEN_PACKETS_BIN@ -n 100 -o test12.wav
-@ATEST_BIN@ -F0 -PE -L64 -G72 test12.wav
-@ATEST_BIN@ -F1 -PE -L70 -G75 test12.wav
+@ATEST_BIN@ -F0 -PA -L66 -G72 test12.wav
+@ATEST_BIN@ -F1 -PA -L72 -G78 test12.wav
+@ATEST_BIN@ -F0 -PB -L66 -G72 test12.wav
+@ATEST_BIN@ -F1 -PB -L70 -G76 test12.wav
\ No newline at end of file
diff --git a/test/scripts/check-modem300 b/test/scripts/check-modem300
index da37dd2..6d6e643 100755
--- a/test/scripts/check-modem300
+++ b/test/scripts/check-modem300
@@ -1,5 +1,7 @@
@CUSTOM_SHELL_SHABANG@
@GEN_PACKETS_BIN@ -B300 -n 100 -o test3.wav
-@ATEST_BIN@ -B300 -F0 -L68 -G69 test3.wav
-@ATEST_BIN@ -B300 -F1 -L71 -G75 test3.wav
+@ATEST_BIN@ -B300 -PA -F0 -L65 -G71 test3.wav
+@ATEST_BIN@ -B300 -PA -F1 -L69 -G75 test3.wav
+@ATEST_BIN@ -B300 -PB -F0 -L69 -G75 test3.wav
+@ATEST_BIN@ -B300 -PB -F1 -L73 -G79 test3.wav