mirror of https://github.com/wb2osz/direwolf.git
Minor cleanups.
This commit is contained in:
parent
d85abe214f
commit
7a88785fad
18
demod.c
18
demod.c
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@ -403,8 +403,8 @@ int demod_init (struct audio_s *pa)
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/* For signal level reporting, we want a longer term view. */
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// TODO: Should probably move this into the init functions.
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D->quick_attack = D->agc_fast_attack * 0.2;
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D->sluggish_decay = D->agc_slow_decay * 0.2;
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D->quick_attack = D->agc_fast_attack * 0.2f;
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D->sluggish_decay = D->agc_slow_decay * 0.2f;
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}
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}
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else if (have_plus) {
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@ -458,8 +458,8 @@ int demod_init (struct audio_s *pa)
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/* For signal level reporting, we want a longer term view. */
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D->quick_attack = D->agc_fast_attack * 0.2;
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D->sluggish_decay = D->agc_slow_decay * 0.2;
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D->quick_attack = D->agc_fast_attack * 0.2f;
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D->sluggish_decay = D->agc_slow_decay * 0.2f;
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}
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else {
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int d;
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@ -511,8 +511,8 @@ int demod_init (struct audio_s *pa)
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/* For signal level reporting, we want a longer term view. */
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D->quick_attack = D->agc_fast_attack * 0.2;
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D->sluggish_decay = D->agc_slow_decay * 0.2;
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D->quick_attack = D->agc_fast_attack * 0.2f;
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D->sluggish_decay = D->agc_slow_decay * 0.2f;
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} /* for each freq pair */
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}
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@ -687,7 +687,7 @@ int demod_init (struct audio_s *pa)
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dw_printf ("The ratio of audio samples per sec (%d) to data rate in baud (%d) is %.1f\n",
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save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec,
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save_audio_config_p->achan[chan].baud,
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ratio);
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(double)ratio);
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if (ratio < 3) {
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text_color_set(DW_COLOR_ERROR);
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dw_printf ("There is little hope of success with such a low ratio. Use a higher sample rate.\n");
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@ -718,8 +718,8 @@ int demod_init (struct audio_s *pa)
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/* For signal level reporting, we want a longer term view. */
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D->quick_attack = D->agc_fast_attack * 0.2;
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D->sluggish_decay = D->agc_slow_decay * 0.2;
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D->quick_attack = D->agc_fast_attack * 0.2f;
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D->sluggish_decay = D->agc_slow_decay * 0.2f;
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}
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break;
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32
demod_9600.c
32
demod_9600.c
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@ -87,21 +87,21 @@ __attribute__((hot)) __attribute__((always_inline))
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static inline float agc (float in, float fast_attack, float slow_decay, float *ppeak, float *pvalley)
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{
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if (in >= *ppeak) {
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*ppeak = in * fast_attack + *ppeak * (1. - fast_attack);
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*ppeak = in * fast_attack + *ppeak * (1.0f - fast_attack);
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}
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else {
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*ppeak = in * slow_decay + *ppeak * (1. - slow_decay);
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*ppeak = in * slow_decay + *ppeak * (1.0f - slow_decay);
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}
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if (in <= *pvalley) {
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*pvalley = in * fast_attack + *pvalley * (1. - fast_attack);
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*pvalley = in * fast_attack + *pvalley * (1.0f - fast_attack);
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}
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else {
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*pvalley = in * slow_decay + *pvalley * (1. - slow_decay);
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*pvalley = in * slow_decay + *pvalley * (1.0f - slow_decay);
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}
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if (*ppeak > *pvalley) {
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return ((in - 0.5 * (*ppeak + *pvalley)) / (*ppeak - *pvalley));
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return ((in - 0.5f * (*ppeak + *pvalley)) / (*ppeak - *pvalley));
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}
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return (0.0);
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}
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@ -144,8 +144,8 @@ void demod_9600_init (int samples_per_sec, int baud, struct demodulator_state_s
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D->lp_filter_len_bits = 76 * 9600.0 / (44100.0 * 2.0);
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// Works best with odd number in some tests. Even is better in others.
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//D->lp_filter_size = ((int) (0.5 * ( D->lp_filter_len_bits * (float)samples_per_sec / (float)baud ))) * 2 + 1;
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D->lp_filter_size = (int) (( D->lp_filter_len_bits * (float)samples_per_sec / baud) + 0.5);
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//D->lp_filter_size = ((int) (0.5f * ( D->lp_filter_len_bits * (float)samples_per_sec / (float)baud ))) * 2 + 1;
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D->lp_filter_size = (int) (( D->lp_filter_len_bits * (float)samples_per_sec / baud) + 0.5f);
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D->lp_window = BP_WINDOW_HAMMING;
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D->lpf_baud = 0.62;
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@ -199,7 +199,7 @@ void demod_9600_init (int samples_per_sec, int baud, struct demodulator_state_s
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/* Version 1.2: Experiment with different slicing levels. */
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for (j = 0; j < MAX_SUBCHANS; j++) {
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slice_point[j] = 0.02 * (j - 0.5 * (MAX_SUBCHANS-1));
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slice_point[j] = 0.02f * (j - 0.5f * (MAX_SUBCHANS-1));
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//dw_printf ("slice_point[%d] = %+5.2f\n", j, slice_point[j]);
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}
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@ -333,17 +333,17 @@ void demod_9600_process_sample (int chan, int sam, struct demodulator_state_s *D
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// TODO: probably no need for this. Just use D->m_peak, D->m_valley
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if (amp >= D->alevel_mark_peak) {
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D->alevel_mark_peak = amp * D->quick_attack + D->alevel_mark_peak * (1. - D->quick_attack);
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D->alevel_mark_peak = amp * D->quick_attack + D->alevel_mark_peak * (1.0f - D->quick_attack);
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}
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else {
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D->alevel_mark_peak = amp * D->sluggish_decay + D->alevel_mark_peak * (1. - D->sluggish_decay);
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D->alevel_mark_peak = amp * D->sluggish_decay + D->alevel_mark_peak * (1.0f - D->sluggish_decay);
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}
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if (amp <= D->alevel_space_peak) {
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D->alevel_space_peak = amp * D->quick_attack + D->alevel_space_peak * (1. - D->quick_attack);
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D->alevel_space_peak = amp * D->quick_attack + D->alevel_space_peak * (1.0f - D->quick_attack);
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}
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else {
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D->alevel_space_peak = amp * D->sluggish_decay + D->alevel_space_peak * (1. - D->sluggish_decay);
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D->alevel_space_peak = amp * D->sluggish_decay + D->alevel_space_peak * (1.0f - D->sluggish_decay);
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}
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/*
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@ -544,10 +544,10 @@ static void inline nudge_pll (int chan, int subchan, int slice, float demod_out_
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fprintf (demod_log_fp, "Audio, Peak, Valley, Demod, SData, Descram, Clock\n");
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}
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fprintf (demod_log_fp, "%.3f, %.3f, %.3f, %.3f, %.2f, %.2f, %.2f\n",
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0.5 * fsam + 3.5,
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0.5 * D->m_peak + 3.5,
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0.5 * D->m_valley + 3.5,
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0.5 * demod_out + 2.0,
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0.5f * fsam + 3.5,
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0.5f * D->m_peak + 3.5,
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0.5f * D->m_valley + 3.5,
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0.5f * demod_out + 2.0,
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demod_data ? 1.35 : 1.0,
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descram ? .9 : .55,
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(D->data_clock_pll & 0x80000000) ? .1 : .45);
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26
demod_afsk.c
26
demod_afsk.c
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@ -528,7 +528,7 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
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for (j=0; j<D->ms_filter_size; j++) {
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float am;
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float center;
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float shape = 1; /* Shape is an attempt to smooth out the */
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float shape = 1.0f; /* Shape is an attempt to smooth out the */
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/* abrupt edges in hopes of reducing */
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/* overshoot and ringing. */
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/* My first thought was to use a cosine shape. */
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@ -536,16 +536,16 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
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/* windows mentioned in the literature. */
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/* http://en.wikipedia.org/wiki/Window_function */
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center = 0.5 * (D->ms_filter_size - 1);
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am = ((float)(j - center) / (float)samples_per_sec) * ((float)mark_freq) * (2 * M_PI);
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center = 0.5f * (D->ms_filter_size - 1);
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am = ((float)(j - center) / (float)samples_per_sec) * ((float)mark_freq) * (2.0f * (float)M_PI);
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shape = window (D->ms_window, D->ms_filter_size, j);
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D->m_sin_table[j] = sin(am) * shape;
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D->m_cos_table[j] = cos(am) * shape;
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D->m_sin_table[j] = sinf(am) * shape;
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D->m_cos_table[j] = cosf(am) * shape;
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Gs += D->m_sin_table[j] * sin(am);
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Gc += D->m_cos_table[j] * cos(am);
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Gs += D->m_sin_table[j] * sinf(am);
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Gc += D->m_cos_table[j] * cosf(am);
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#if DEBUG1
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dw_printf ("%6d %6.2f %6.2f %6.2f\n", j, shape, D->m_sin_table[j], D->m_cos_table[j]) ;
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for (j=0; j<D->ms_filter_size; j++) {
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float as;
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float center;
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float shape = 1;
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float shape = 1.0f;
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center = 0.5 * (D->ms_filter_size - 1);
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as = ((float)(j - center) / (float)samples_per_sec) * ((float)space_freq) * (2 * M_PI);
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as = ((float)(j - center) / (float)samples_per_sec) * ((float)space_freq) * (2.0f * (float)M_PI);
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shape = window (D->ms_window, D->ms_filter_size, j);
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D->s_sin_table[j] = sin(as) * shape;
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D->s_cos_table[j] = cos(as) * shape;
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D->s_sin_table[j] = sinf(as) * shape;
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D->s_cos_table[j] = cosf(as) * shape;
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Gs += D->s_sin_table[j] * sin(as);
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Gc += D->s_cos_table[j] * cos(as);
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Gs += D->s_sin_table[j] * sinf(as);
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Gc += D->s_cos_table[j] * cosf(as);
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#if DEBUG1
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dw_printf ("%6d %6.2f %6.2f %6.2f\n", j, shape, D->s_sin_table[j], D->s_cos_table[j] ) ;
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32
demod_psk.c
32
demod_psk.c
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@ -293,9 +293,9 @@ void demod_psk_init (enum modem_t modem_type, int samples_per_sec, int bps, char
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D->ms_filter_len_bits = 1.25; // Delay line > 13/12 * symbol period
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D->coffs = (int) round( (11. / 12.) * (float)samples_per_sec / (float)correct_baud );
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D->coffs = (int) round( (11.f / 12.f) * (float)samples_per_sec / (float)correct_baud );
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D->boffs = (int) round( (float)samples_per_sec / (float)correct_baud );
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D->soffs = (int) round( (13. / 12.) * (float)samples_per_sec / (float)correct_baud );
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D->soffs = (int) round( (13.f / 12.f) * (float)samples_per_sec / (float)correct_baud );
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}
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else {
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@ -380,18 +380,18 @@ void demod_psk_init (enum modem_t modem_type, int samples_per_sec, int bps, char
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D->ms_filter_len_bits = 1.25; // Delay line > 10/9 * symbol period
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D->coffs = (int) round( (8. / 9.) * (float)samples_per_sec / (float)correct_baud );
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D->coffs = (int) round( (8.f / 9.f) * (float)samples_per_sec / (float)correct_baud );
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D->boffs = (int) round( (float)samples_per_sec / (float)correct_baud );
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D->soffs = (int) round( (10. / 9.) * (float)samples_per_sec / (float)correct_baud );
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D->soffs = (int) round( (10.f / 9.f) * (float)samples_per_sec / (float)correct_baud );
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}
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if (D->psk_use_lo) {
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D->lo_step = (int) round( 256. * 256. * 256. * 256. * carrier_freq / (float)samples_per_sec);
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D->lo_step = (int) round( 256. * 256. * 256. * 256. * carrier_freq / (double)samples_per_sec);
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assert (MAX_FILTER_SIZE >= 256);
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for (j = 0; j < 256; j++) {
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D->m_sin_table[j] = sinf(2. * M_PI * j / 256.);
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D->m_sin_table[j] = sinf(2.f * (float)M_PI * j / 256.f);
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}
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}
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@ -491,11 +491,11 @@ void demod_psk_init (enum modem_t modem_type, int samples_per_sec, int bps, char
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f2 = carrier_freq + D->prefilter_baud * correct_baud;
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#if 0
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text_color_set(DW_COLOR_DEBUG);
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dw_printf ("Generating prefilter %.0f to %.0f Hz.\n", f1, f2);
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dw_printf ("Generating prefilter %.0f to %.0f Hz.\n", (double)f1, (double)f2);
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#endif
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if (f1 <= 0) {
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text_color_set (DW_COLOR_ERROR);
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dw_printf ("Prefilter of %.0f to %.0f Hz doesn't make sense.\n", f1, f2);
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dw_printf ("Prefilter of %.0f to %.0f Hz doesn't make sense.\n", (double)f1, (double)f2);
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f1 = 10;
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}
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@ -642,7 +642,7 @@ void demod_psk_process_sample (int chan, int subchan, int sam, struct demodulato
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/* 256 units/cycle makes modulo processing easier. */
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/* Make sure it is positive before truncating to integer. */
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id = ((int)((delta / (2.f * M_PI) + 1.f) * 256.f)) & 0xff;
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id = ((int)((delta / (2.f * (float)M_PI) + 1.f) * 256.f)) & 0xff;
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if (D->modem_type == MODEM_QPSK) {
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demod_phase_shift = ((id + 32) >> 6) & 0x3;
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@ -686,7 +686,7 @@ void demod_psk_process_sample (int chan, int subchan, int sam, struct demodulato
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}
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#else
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a = my_atan2f(I,Q);
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int id = ((int)((a / (2.f * M_PI) + 1.f) * 256.f)) & 0xff;
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int id = ((int)((a / (2.f * (float)M_PI) + 1.f) * 256.f)) & 0xff;
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// 128 compensates for 180 degree phase shift due
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// to 1 1/2 carrier cycles per symbol period.
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demod_phase_shift = ((id + 128) >> 6) & 0x3;
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@ -697,7 +697,7 @@ void demod_psk_process_sample (int chan, int subchan, int sam, struct demodulato
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int idelta;
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a = my_atan2f(I,Q);
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idelta = ((int)((a / (2.f * M_PI) + 1.f) * 256.f)) & 0xff;
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idelta = ((int)((a / (2.f * (float)M_PI) + 1.f) * 256.f)) & 0xff;
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// 32 (90 degrees) compensates for 1800 carrier vs. 1800 baud.
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// 16 is to set threshold between constellation points.
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demod_phase_shift = ((idelta - 32 - 16) >> 5) & 0x7;
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@ -827,16 +827,20 @@ static void inline nudge_pll (int chan, int subchan, int slice, int demod_bits,
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/*
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* If demodulated data has changed,
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* pull the PLL phase closer to zero.
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* Pull the PLL phase closer to zero.
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* Use "floor" instead of simply casting so the sign won't flip.
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* For example if we had -0.7 we want to end up with -1 rather than 0.
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*/
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// TODO: demod_9600 has an improved technique. Would it help us here?
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if (demod_bits != D->slicer[slice].prev_demod_data) {
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if (hdlc_rec_gathering (chan, subchan, slice)) {
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D->slicer[slice].data_clock_pll = (int)floor((double)(D->slicer[slice].data_clock_pll) * D->pll_locked_inertia);
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D->slicer[slice].data_clock_pll = (int)floorf((float)(D->slicer[slice].data_clock_pll) * D->pll_locked_inertia);
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}
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else {
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D->slicer[slice].data_clock_pll = (int)floor((double)(D->slicer[slice].data_clock_pll) * D->pll_searching_inertia);
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D->slicer[slice].data_clock_pll = (int)floorf((float)(D->slicer[slice].data_clock_pll) * D->pll_searching_inertia);
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}
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}
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Binary file not shown.
2
morse.c
2
morse.c
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@ -272,6 +272,8 @@ int morse_send (int chan, char *str, int wpm, int txdelay, int txtail)
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time_units, morse_units_str(str));
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}
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audio_flush(ACHAN2ADEV(chan));
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return (txdelay +
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(int) (TIME_UNITS_TO_MS(time_units, wpm) + 0.5) +
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txtail);
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11
telemetry.c
11
telemetry.c
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@ -835,7 +835,16 @@ void telemetry_bit_sense_message (char *station, char *msg, int quiet)
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}
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}
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/* Skip comma if first character of comment field. */
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/*
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* Skip comma if first character of comment field.
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*
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* The protocol spec is inconsistent here.
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* The definition shows the Project Title immediately after a fixed width field of 8 binary digits.
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* The example has a comma in there.
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*
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* The toolkit telem-bits.pl script does insert the comma because it seems more sensible.
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* Here we accept it either way. i.e. Discard first character after data values if it is comma.
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*/
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if (msg[n] == ',') n++;
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@ -1,6 +1,7 @@
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APRS TO-CALL VERSION NUMBERS 14 Nov 2016
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APRS TO-CALL VERSION NUMBERS 6 Dec 2016
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-------------------------------------------------------------------
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WB4APR
|
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16 Dec 16 added APYSxx for W2GMD's Python APRS
|
||||
14 Nov 16 Added APINxx for PinPoint by AB0WV
|
||||
09 Nov 16 added APNICx for SQ5EKU http://sq5eku.blogspot.com/
|
||||
24 Oct 16 added APTKPT TrackPoint N0LP, removed APZTKP
|
||||
|
@ -137,7 +138,7 @@ a TOCALL number series:
|
|||
APRRTx RPC electronics
|
||||
APRS Generic, (obsolete. Digis should use APNxxx instead)
|
||||
APRXxx >40 APRSmax
|
||||
APRXxx <39 for OH2MQK's RX-igate
|
||||
APRXxx <39 for OH2MQK's igate
|
||||
APRTLM used in MIM's and Mic-lites, etc
|
||||
APRtfc APRStraffic
|
||||
APRSTx APRStt (Touch tone)
|
||||
|
@ -176,6 +177,7 @@ a TOCALL number series:
|
|||
APY008 Yaesu VX-8 series
|
||||
APY350 Yaesu FTM-350 series
|
||||
APYTxx for YagTracker
|
||||
APYSxx for W2GMD's Python APRS
|
||||
APZ APZxxx Experimental
|
||||
APZ247 for UPRS NR0Q
|
||||
APZ0xx Xastir (old versions. See APX)
|
||||
|
|
Loading…
Reference in New Issue