/* fsk_demod_state.h */
#ifndef FSK_DEMOD_STATE_H
#include "rpack.h"
#include "audio.h" // for enum modem_t
/*
* Demodulator state.
* The name of the file is from we only had FSK. Now we have other techniques.
* Different copy is required for each channel & subchannel being processed concurrently.
*/
// TODO1.2: change prefix from BP_ to DSP_
typedef enum bp_window_e { BP_WINDOW_TRUNCATED,
BP_WINDOW_COSINE,
BP_WINDOW_HAMMING,
BP_WINDOW_BLACKMAN,
BP_WINDOW_FLATTOP } bp_window_t;
struct demodulator_state_s
{
/*
* These are set once during initialization.
*/
enum modem_t modem_type; // MODEM_AFSK, MODEM_8PSK, etc.
char profile; // 'A', 'B', etc. Upper case.
// Only needed to see if we are using 'F' to take fast path.
#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.
// 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 somewhat longer 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;
/*
* 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.
* Second is frequency as ratio to baud rate for FIR.
*/
int lpf_use_fir; /* 0 for IIR, 1 for FIR. */
float lpf_iir; /* Only if using IIR. */
float lpf_baud; /* Cutoff frequency as fraction of baud. */
/* Intuitively we'd expect this to be somewhere */
/* in the range of 0.5 to 1. */
/* In practice, it turned out a little larger */
/* for profiles B, C, D. */
float lp_filter_len_bits; /* Length in number of bit 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. */
/*
* Automatic gain control. Fast attack and slow decay factors.
*/
float agc_fast_attack;
float agc_slow_decay;
/*
* Use a longer term view for reporting signal levels.
*/
float quick_attack;
float sluggish_decay;
/*
* Hysteresis before final demodulator 0 / 1 decision.
*/
float hysteresis;
int num_slicers; /* >1 for multiple slicers. */
/*
* Phase Locked Loop (PLL) inertia.
* Larger number means less influence by signal transitions.
*/
float pll_locked_inertia;
float pll_searching_inertia;
/*
* Optional band pass pre-filter before mark/space detector.
*/
int use_prefilter; /* True to enable it. */
float prefilter_baud; /* Cutoff frequencies, as fraction of */
/* baud rate, beyond tones used. */
/* Example, if we used 1600/1800 tones at */
/* 300 baud, and this was 0.5, the cutoff */
/* frequencies would be: */
/* 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. */
int pre_filter_size; /* Size of pre filter, in audio samples. */
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)));
/*
* These are for PSK only.
* They are number of delay line taps into previous symbol.
* They are one symbol period and + or - 45 degrees of the carrier frequency.
*/
int boffs; /* symbol length based on sample rate and baud. */
int coffs; /* to get cos component of previous symbol. */
int soffs; /* to get sin component of previous symbol. */
unsigned int lo_step; /* How much to advance the local oscillator */
/* phase for each audio sample. */
int psk_use_lo; /* Use local oscillator rather than self correlation. */
/*
* The rest are continuously updated.
*/
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
* AGC uses "fast" attack and "slow" decay.
*/
float alevel_rec_peak;
float alevel_rec_valley;
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;
/*
* For the PLL and data bit timing.
* starting in version 1.2 we can have multiple slicers for one demodulator.
* Each slicer has its own PLL and HDLC decoder.
*/
/*
* Version 1.3: Clean up subchan vs. slicer.
*
* Originally some number of CHANNELS (originally 2, later 6)
* which can have multiple parallel demodulators called SUB-CHANNELS.
* This was originally for staggered frequencies for HF SSB.
* It can also be used for multiple demodulators with the same
* frequency but other differing parameters.
* Each subchannel has its own demodulator and HDLC decoder.
*
* In version 1.2 we added multiple SLICERS.
* The data structure, here, has multiple slicers per
* demodulator (subchannel). Due to fuzzy thinking or
* expediency, the multiple slicers got mapped into subchannels.
* This means we can't use both multiple decoders and
* multiple slicers at the same time.
*
* Clean this up in 1.3 and keep the concepts separate.
* This means adding a third variable many places
* we are passing around the origin.
*
*/
struct {
signed int data_clock_pll; // PLL for data clock recovery.
// It is incremented by pll_step_per_sample
// for each audio sample.
signed int prev_d_c_pll; // Previous value of above, before
// incrementing, to detect overflows.
int prev_demod_data; // Previous data bit detected.
// Used to look for transitions.
float prev_demod_out_f;
/* This is used only for "9600" baud data. */
int lfsr; // Descrambler shift register.
} slicer [MAX_SLICERS]; // Actual number in use is num_slicers.
// Should be in range 1 .. MAX_SLICERS,
/*
* Special for Rino decoder only.
* One for each possible signal polarity.
* The project showed promise but fell by the wayside.
*/
#if 0
struct gr_state_s {
signed int data_clock_pll; // PLL for data clock recovery.
// It is incremented by pll_step_per_sample
// for each audio sample.
signed int prev_d_c_pll; // Previous value of above, before
// incrementing, to detect overflows.
float gr_minus_peak; // For automatic gain control.
float gr_plus_peak;
int gr_sync; // Is sync pulse present?
int gr_prev_sync; // Previous state to detect leading edge.
int gr_first_sample; // Index of starting sample index for debugging.
int gr_dcd; // Data carrier detect. i.e. are we
// currently decoding a message.
float gr_early_sum; // For averaging bit values in two regions.
int gr_early_count;
float gr_late_sum;
int gr_late_count;
float gr_sync_sum;
int gr_sync_count;
int gr_bit_count; // Bit index into message.
struct rpack_s rpack; // Collection of bits.
} gr_state[2];
#endif
};
#define FSK_DEMOD_STATE_H 1
#endif