Clean out old obsolete demodulators.

src/audio.h

@@ -173,9 +173,6 @@ struct audio_s {
 
 	    int upsample;		/* Upsample by this factor for G3RUH. */
 
-	    int interleave;		/* If > 1, interleave samples among multiple decoders. */
-					/* Quick hack for experiment. */
-
             int mark_freq;		/* Two tones for AFSK modulation, in Hz. */
 	    int space_freq;		/* Standard tones are 1200 and 2200 for 1200 baud. */
 

src/demod.c

@@ -120,7 +120,6 @@ int demod_init (struct audio_s *pa)
 	   * This can be increased by:
 	   *	Multiple frequencies.
 	   *	Multiple letters (not sure if I will continue this).
-	   *	New interleaved decoders.
 	   *
 	   * num_slicers is set to max by the "+" option.
 	   */
@@ -313,48 +312,6 @@ int demod_init (struct audio_s *pa)
 
 	        save_audio_config_p->achan[chan].num_subchan = num_letters;
 		
-/*
- * Quick hack with special case for another experiment.
- * Do this in a more general way if it turns out to be useful.
- */
-	        save_audio_config_p->achan[chan].interleave = 1;
-	        if (strcasecmp(save_audio_config_p->achan[chan].profiles, "EE") == 0) {
-	          save_audio_config_p->achan[chan].interleave = 2;
-	          save_audio_config_p->achan[chan].decimate = 1;
-	        }
-	        else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "EEE") == 0) {
-	          save_audio_config_p->achan[chan].interleave = 3;
-	          save_audio_config_p->achan[chan].decimate = 1;
-	        }
-	        else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "EEEE") == 0) {
-	          save_audio_config_p->achan[chan].interleave = 4;
-	          save_audio_config_p->achan[chan].decimate = 1;
-	        }
-	        else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "EEEEE") == 0) {
-	          save_audio_config_p->achan[chan].interleave = 5;
-	          save_audio_config_p->achan[chan].decimate = 1;
-	        }
-	        else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "GG") == 0) {
-	          save_audio_config_p->achan[chan].interleave = 2;
-	          save_audio_config_p->achan[chan].decimate = 1;
-	        }
-	        else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "GGG") == 0) {
-	          save_audio_config_p->achan[chan].interleave = 3;
-	          save_audio_config_p->achan[chan].decimate = 1;
-	        }
-	        else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "GGG+") == 0) {
-	          save_audio_config_p->achan[chan].interleave = 3;
-	          save_audio_config_p->achan[chan].decimate = 1;
-	        }
-	        else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "GGGG") == 0) {
-	          save_audio_config_p->achan[chan].interleave = 4;
-	          save_audio_config_p->achan[chan].decimate = 1;
-	        }
-	        else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "GGGGG") == 0) {
-	          save_audio_config_p->achan[chan].interleave = 5;
-	          save_audio_config_p->achan[chan].decimate = 1;
-	        }
-
 		if (save_audio_config_p->achan[chan].num_subchan != num_letters) {
 		  text_color_set(DW_COLOR_ERROR);
 		  dw_printf ("INTERNAL ERROR, %s:%d, chan=%d, num_subchan(%d) != strlen(\"%s\")\n",
@@ -383,7 +340,7 @@ int demod_init (struct audio_s *pa)
 	            dw_printf ("        %d.%d: %c %d & %d\n", chan, d, profile, mark, space);
 	          }
 
-	          demod_afsk_init (save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec / (save_audio_config_p->achan[chan].decimate * save_audio_config_p->achan[chan].interleave), 
+	          demod_afsk_init (save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec / save_audio_config_p->achan[chan].decimate, 
 			    save_audio_config_p->achan[chan].baud,
 		            mark, 
 	                    space,

src/demod_9600.c

@@ -159,12 +159,6 @@ void demod_9600_init (int samples_per_sec, int baud, struct demodulator_state_s
 	    D->pll_locked_inertia = 0.89;
 	    D->pll_searching_inertia = 0.67;
 
-	    D->play_it_again_sample = 0;		// TODO: 1.6 experiment.
-							// assuming lp_filter_size > lp2_filter_size
-
-	    D->lp2_filter_size =  samples_per_sec / baud;	// samples for 1 bit
-
-
 //	    break;
 //	}
 
@@ -216,12 +210,6 @@ void demod_9600_init (int samples_per_sec, int baud, struct demodulator_state_s
 
 	(void)gen_lowpass (fc, D->lp_filter, D->lp_filter_size, D->lp_window, 0);
 
-// Go back and resample where bit is expected.
-
-	fc = (float)baud * 1 / (float)samples_per_sec;
-
-	(void)gen_lowpass (fc, D->lp2_filter, D->lp2_filter_size, D->lp_window, 0);
-
 	/* Version 1.2: Experiment with different slicing levels. */
 
 	for (j = 0; j < MAX_SUBCHANS; j++) {
@@ -524,32 +512,8 @@ inline static void nudge_pll (int chan, int subchan, int slice, float demod_out_
 
 	  /* Overflow.  Was large positive, wrapped around, now large negative. */
 
-
-	  if (D->play_it_again_sample) {	// New experiment in 1.6.
-
-// FIXME: double check position and draw picture.
-
-	    int offset = ( D->lp_filter_size - D->lp2_filter_size ) / 2;
-
-	    float amp = convolve (D->raw_cb + offset, D->lp2_filter, D->lp2_filter_size);
-
-	    int resampled;
-
-	    if (D->num_slicers > 1) {
-	      resampled = amp - slice_point[slice] > 0;;
-	    }
-	    else {
-	      resampled = amp > 0;
-	    }
-
-	    hdlc_rec_bit (chan, subchan, slice, resampled, 1, D->slicer[slice].lfsr);
-	  }
-	  else {
-
-// traditional
 	  hdlc_rec_bit (chan, subchan, slice, demod_out_f > 0, 1, D->slicer[slice].lfsr);
 	}
-	}
 
 /*
  * Zero crossing?

src/demod_afsk.c

@@ -197,79 +197,6 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 
 	switch (profile) {
 
-	  case 'A':
-
-		/* Original.  52 taps, truncated bandpass, IIR lowpass */
-		/* 'F' is the fast version for low end processors. */
-		/* It is a special case that works only for a particular */
-		/* baud rate, tone pair, and sampling rate. */
-
-	    D->use_prefilter = 0;		
-
-	    D->ms_filter_len_bits = 1.415;		/* 52 @ 44100, 1200 */
-	    D->ms_window = BP_WINDOW_TRUNCATED;
-
-	    //D->bp_window = BP_WINDOW_TRUNCATED;
-
-	    D->lpf_use_fir = 0;
-	    D->lpf_iir = 0.195;
-
-	    D->agc_fast_attack = 0.250;		
-	    D->agc_slow_decay = 0.00012;
-	    D->hysteresis = 0.005;
-
-	    D->pll_locked_inertia = 0.700;
-	    D->pll_searching_inertia = 0.580;
-	    break;
-
-	  case 'B':
-
-		/* Original bandpass.  Use FIR lowpass instead. */
-
-	    D->use_prefilter = 0;		
-
-	    D->ms_filter_len_bits = 1.415;		/* 52 @ 44100, 1200 */
-	    D->ms_window = BP_WINDOW_TRUNCATED;
-
-	    //D->bp_window = BP_WINDOW_TRUNCATED;
-
-	    D->lpf_use_fir = 1;
-	    D->lpf_baud = 1.09;
-	    D->lp_filter_len_bits = D->ms_filter_len_bits;		
-	    D->lp_window = BP_WINDOW_TRUNCATED;
-
-	    D->agc_fast_attack = 0.370;		
-	    D->agc_slow_decay = 0.00014;
-	    D->hysteresis = 0.003;
-
-	    D->pll_locked_inertia = 0.620;
-	    D->pll_searching_inertia = 0.350;
-	    break;
-
-	  case 'C':
-
-		/* Cosine window, 76 taps for bandpass, FIR lowpass. */
-
-	    D->use_prefilter = 0;		
-
-	    D->ms_filter_len_bits = 2.068;		/* 76 @ 44100, 1200 */
-	    D->ms_window = BP_WINDOW_COSINE;
-
-	    //D->bp_window = BP_WINDOW_COSINE;
-
-	    D->lpf_use_fir = 1;
-	    D->lpf_baud = 1.09;
-	    D->lp_filter_len_bits = D->ms_filter_len_bits;		
-	    D->lp_window = BP_WINDOW_TRUNCATED;
-
-	    D->agc_fast_attack = 0.495;		
-	    D->agc_slow_decay = 0.00022;
-	    D->hysteresis = 0.005;
-
-	    D->pll_locked_inertia = 0.620;
-	    D->pll_searching_inertia = 0.350;
-	    break;
-
 	  case 'D':
 
 		/* Prefilter, Cosine window, FIR lowpass. Tweeked for 300 baud. */
@@ -335,73 +262,6 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 	    D->pll_searching_inertia = 0.50;
 	    break;
 
-	  case 'G':
-
-		/* 1200 baud - Started out same as E but add 3 way interleave. */
-		/* Version 1.3 - EXPERIMENTAL - Needs more fine tuning. */
-
-	    //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 = 0.15;
-	    D->pre_filter_len_bits = 128 * 1200. / (44100. / 3.);
-	    D->pre_window = BP_WINDOW_TRUNCATED;
-
-	    D->ms_filter_len_bits = 25 * 1200. / (44100. / 3.);
-	    D->ms_window = BP_WINDOW_COSINE;
-
-	    D->lpf_use_fir = 1;
-	    D->lpf_baud = 1.16;
-	    D->lp_filter_len_bits = 21 * 1200. / (44100. / 3.);
-	    D->lp_window = BP_WINDOW_TRUNCATED;
-
-	    D->agc_fast_attack = 0.130;
-	    D->agc_slow_decay = 0.00013;
-	    D->hysteresis = 0.01;
-
-	    D->pll_locked_inertia = 0.73;
-	    D->pll_searching_inertia = 0.64;
-	    break;
-
-	  case 'H':
-
-		/* Experiment in Version 1.6 			*/
-		/* 1200 baud - Started out as a copy of E but	*/
-		/* will probably have little tweaks after the	*/
-		/* major experiment.				*/
-		/* Enhancements: 				*/
-		/*  + Look back and sample the bit position.	*/
-		/*  + Avoid smearing by long filter and low pass. */
-
-	    D->use_prefilter = 1;		/* first, a bandpass filter. */
-	    D->prefilter_baud = 0.21;
-	    D->pre_filter_len_bits = 184 * 1200. / 44100.;
-	    D->pre_filter_len_bits = 235 * 1200. / 44100.;
-	    D->pre_window = BP_WINDOW_TRUNCATED;
-
-	    D->ms_filter_len_bits = 65 * 1200. / 44100.;	// Just over 2 bit times.
-	    D->ms_window = BP_WINDOW_COSINE;
-
-	    /* New for synchronous re-demod in 1.6. */
-	    D->play_it_again_sample = 1;
-	    D->m2_filter_len_bits = 44 * 1200. / 44100.;	// a little more than 1 bit time.
-	    D->s2_filter_len_bits = 48 * 1200. / 44100.;	// a little more than 1 bit time.
-	    D->ms2_window = BP_WINDOW_TRUNCATED;
-	    D->lp_delay_fract = 0.53;		// FIXME:  This is backwards.
-
-	    D->lpf_use_fir = 1;
-	    D->lpf_baud = 1.10;
-	    D->lp_filter_len_bits = 64 * 1200. / 44100.;
-	    D->lp_window = BP_WINDOW_TRUNCATED;
-
-	    D->agc_fast_attack = 0.820;
-	    D->agc_slow_decay = 0.000214;
-	    D->hysteresis = 0.001;
-
-	    D->pll_locked_inertia = 0.765;
-	    D->pll_searching_inertia = 0.44;
-	    break;
-
 	  default:
 
 	    text_color_set(DW_COLOR_ERROR);
@@ -457,8 +317,6 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 
 	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->m2_filter_size = (int) round( D->m2_filter_len_bits * (float)samples_per_sec / (float)baud );
-	D->s2_filter_size = (int) round( D->s2_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. */
@@ -469,12 +327,6 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 #ifdef TUNE_MS_FILTER_SIZE
 	D->ms_filter_size = TUNE_MS_FILTER_SIZE;
 #endif
-#ifdef TUNE_M2_FILTER_SIZE
-	D->m2_filter_size = TUNE_M2_FILTER_SIZE;
-#endif
-#ifdef TUNE_S2_FILTER_SIZE
-	D->s2_filter_size = TUNE_S2_FILTER_SIZE;
-#endif
 #ifdef TUNE_LP_FILTER_SIZE
 	D->lp_filter_size = TUNE_LP_FILTER_SIZE;
 #endif
@@ -503,24 +355,7 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 	  exit (1);
 	}
 
-	if (D->m2_filter_size * 2 > MAX_FILTER_SIZE)
+
-	{
-	  text_color_set (DW_COLOR_ERROR);
-	  dw_printf ("Calculated filter size of %d is too large.\n", D->m2_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->s2_filter_size * 2 > MAX_FILTER_SIZE)
-	{
-	  text_color_set (DW_COLOR_ERROR);
-	  dw_printf ("Calculated filter size of %d is too large.\n", D->s2_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) 
 	{
@@ -585,13 +420,6 @@ void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
 
 	  gen_ms (space_freq, samples_per_sec, D->s_sin_table, D->s_cos_table, D->ms_filter_size, D->ms_window);
 
-	  // Note that these are twice as long so we can try matching different phases.
-
-	  if (D->play_it_again_sample) {
-	    gen_ms (mark_freq, samples_per_sec, D->m2_sin_table, D->m2_cos_table, D->m2_filter_size * 2, D->ms2_window);
-	    gen_ms (space_freq, samples_per_sec, D->s2_sin_table, D->s2_cos_table, D->s2_filter_size * 2, D->ms2_window);
-	  }
-
 /*
  * Now the lowpass filter.
  * I thought we'd want a cutoff of about 0.5 the baud rate 
@@ -764,9 +592,6 @@ void demod_afsk_process_sample (int chan, int subchan, int sam, struct demodulat
 // FIXME:  calculate how much we really need.
 
 	int extra = 0;
-	if (D->play_it_again_sample) {
-	  extra = D->lp_filter_size;
-	}
 
 	if (D->use_prefilter) {
 	  float cleaner;
@@ -1015,77 +840,8 @@ 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. */
-
-// In version 1.6 we will try a new experiment.
-// The tone filters are about 2 bit times wide so this smears the signal.
-// I originally expected this to be about 1 bit time but 2 turned out
-// to give the best results after extensive experimentation.
-// We are looking at half of each adjacent bit, not just the one we want.
-// The low pass filter also smears the signal.
-//
-// We will try to look back in time and re-demodulate only the time period of the bit.
-//
-
-	  if (D->play_it_again_sample) {	// New in 1.6.  Currently 'H' demod profile.
-
-// FIXME: double check position and draw picture.
-
-#if 0
-	    // This provided a slight benefit.
-
-	    int offset = ( D->ms_filter_size - D->m2_filter_size ) / 2 + D->lp_filter_delay;
-
-	    float m_sum1 = convolve (D->ms_in_cb + offset, D->m2_sin_table, D->m2_filter_size);
-	    float m_sum2 = convolve (D->ms_in_cb + offset, D->m2_cos_table, D->m2_filter_size);
-	    float m_amp = sqrtf(m_sum1 * m_sum1 + m_sum2 * m_sum2);
-
-	    offset = ( D->ms_filter_size - D->s2_filter_size ) / 2 + D->lp_filter_delay;
-
-	    float s_sum1 = convolve (D->ms_in_cb + offset, D->s2_sin_table, D->s2_filter_size);
-	    float s_sum2 = convolve (D->ms_in_cb + offset, D->s2_cos_table, D->s2_filter_size);
-	    float s_amp = sqrtf(s_sum1 * s_sum1 + s_sum2 * s_sum2);
-#else
-	    // Here we try something completely new.
-	    // Rather than taking the vector magnitude of the I+Q components,
-	    // correlate it with only a sine wave.  We don't know the phase so
-	    // we will try matching with a bunch of different phases and take the best match.
-
-	    // Trying match with cosine as well could be beneficial for lower sample rates.
-
-	    int j;
-	    float m_amp = 0;
-	    float s_amp = 0;
-
-	    int offset = ( D->ms_filter_size - D->m2_filter_size ) / 2 + D->lp_filter_delay;
-
-	    for (j = 0; j <= D->m2_filter_size; j++) {
-	      float match = fabsf(convolve (D->ms_in_cb + offset, D->m2_sin_table + j, D->m2_filter_size));
-	      if (match > m_amp) m_amp = match;
-	    }
-
-	    offset = ( D->ms_filter_size - D->s2_filter_size ) / 2 + D->lp_filter_delay;
-
-	    for (j = 0; j <= D->s2_filter_size; j++) {
-	      float match = fabsf(convolve (D->ms_in_cb + offset, D->s2_sin_table + j, D->s2_filter_size));
-	      if (match > s_amp) s_amp = match;
-	    }
-#endif
-
-	    int resampled;
-	    if (D->num_slicers > 1) {
-	      resampled = m_amp > s_amp * space_gain[slice];
-	    }
-	    else {
-	      resampled = m_amp > s_amp;
-	    }
-
-	    hdlc_rec_bit (chan, subchan, slice, resampled, 0, -1);
-	  }
-	  else {
-	    // Traditional way, after the low pass filter.
 	  hdlc_rec_bit (chan, subchan, slice, demod_data, 0, -1);
 	}
-	}
 
 	// Even if we used alternative method to extract the data bit,
 	//  we still use the low pass output for the PLL.

src/fsk_demod_state.h

@@ -33,8 +33,6 @@ struct demodulator_state_s
 	char profile;			// 'A', 'B', etc.	Upper case.
 					// Only needed to see if we are using 'F' to take fast path.
 
-	int play_it_again_sample;	// Enable new synchronous demod in version 1.6.
-
 #define TICKS_PER_PLL_CYCLE ( 256.0 * 256.0 * 256.0 * 256.0 )
 
 	int pll_step_per_sample;	// PLL is advanced by this much each audio sample.
@@ -46,13 +44,6 @@ struct demodulator_state_s
 					/* but about 2 bit times turned out to be better. */
 					/* Currently using same size for any prefilter. */
 
-	int m2_filter_size;
-	int s2_filter_size;		/* Size of mark & space filters, in audio samples */
-					/* for the synchronous demodulator.  I'm expecting */
-					/* smaller, perhaps just over 1 bit time here. */
-
-	int lp2_filter_size;		/* FSK resampling - Size of Low Pass filter, in audio samples. */
-
 
 #define MAX_FILTER_SIZE 320		/* 304 is needed for profile C, 300 baud & 44100. */
 
@@ -61,8 +52,6 @@ struct demodulator_state_s
  * e.g.  1 means 1/1200 second for 1200 baud.
  */
 	float ms_filter_len_bits;
-	float m2_filter_len_bits;
-	float s2_filter_len_bits;
 	float lp_delay_fract;
 
 /* 
@@ -72,7 +61,6 @@ struct demodulator_state_s
 	bp_window_t pre_window;
 	bp_window_t ms_window;
 	bp_window_t lp_window;
-	bp_window_t ms2_window;		/* New in 1.6. */
 
 
 /*
@@ -158,17 +146,6 @@ struct demodulator_state_s
 	float s_sin_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
 	float s_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
 
-/*
- * Same for the synchronous re-demodulator.
- */
-
-	float m2_sin_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
-	float m2_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
-
-	float s2_sin_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
-	float s2_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
-
-	float lp2_filter[MAX_FILTER_SIZE] __attribute__((aligned(16)));
 
 /*
  * These are for PSK only.

src/multi_modem.c

@@ -306,7 +306,6 @@ void multi_modem_process_sample (int chan, int audio_sample)
 {
 	int d;
 	int subchan;
-	static int i = 0;	/* for interleaving among multiple demodulators. */
 
 // Accumulate an average DC bias level.
 // Shouldn't happen with a soundcard but could with mistuned SDR.
@@ -334,22 +333,10 @@ void multi_modem_process_sample (int chan, int audio_sample)
 	/* Formerly one loop. */
 	/* 1.2: We can feed one demodulator but end up with multiple outputs. */
 
-
-	if (save_audio_config_p->achan[chan].interleave > 1) {
-
-// TODO: temp debug, remove this.
-
-	  assert (save_audio_config_p->achan[chan].interleave == save_audio_config_p->achan[chan].num_subchan);
-	  demod_process_sample(chan, i, audio_sample);
-	  i++;
-	  if (i >= save_audio_config_p->achan[chan].interleave) i = 0;
-	}
-	else {
 	/* Send same thing to all. */
 	for (d = 0; d < save_audio_config_p->achan[chan].num_subchan; d++) {
 	  demod_process_sample(chan, d, audio_sample);
 	}
-	}
 
 	for (subchan = 0; subchan < save_audio_config_p->achan[chan].num_subchan; subchan++) {
 	  int slice;