PierMesh
/
direwolf
mirror of https://github.com/wb2osz/direwolf.git
1
0
Fork 0
Code Issues Packages Projects Releases Wiki Activity
direwolf/src/fsk_demod_state.h

550 lines
18 KiB
C
Raw Blame History

/* 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;
// 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
{
/*
* These are set once during initialization.
*/
enum modem_t modem_type; // MODEM_AFSK, MODEM_8PSK, etc.
// enum v26_e v26_alt; // Which alternative when V.26.
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.
/*
* Window type for the various filters.
*/
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_width_sym; /* Length in number of symbol times. */
#define lp_filter_len_bits lp_filter_width_sym // FIXME: temp hack
int lp_filter_taps; /* Size of Low Pass filter, in audio samples. */
#define lp_filter_size lp_filter_taps // FIXME: temp hack
/*
* 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.
* It is more resistant to change when locked on to a signal.
*/
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_sym; // Length in number of symbol times.
#define pre_filter_len_bits pre_filter_len_sym // temp until all references changed.
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)));
float raw_cb[MAX_FILTER_SIZE] __attribute__((aligned(16))); // audio in, need better name.
/*
* The rest are continuously updated.
*/
unsigned int lo_phase; /* Local oscillator for PSK. */
/*
* 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;
/*
* Outputs from the mark and space amplitude detection,
* used as inputs to the FIR lowpass filters.
* Kernel for the lowpass filters.
*/
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.
// Must be 32 bits!!!
// So far, this is the case for every compiler used.
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.
// This is for detecting phase lock to incoming signal.
int good_flag; // Set if transition is near where expected,
// i.e. at a good time.
int bad_flag; // Set if transition is not where expected,
// i.e. at a bad time.
unsigned char good_hist; // History of good transitions for past octet.
unsigned char bad_hist; // History of bad transitions for past octet.
unsigned int score; // History of whether good triumphs over bad
// for past 32 symbols.
int data_detect; // True when locked on to signal.
} slicer [MAX_SLICERS]; // Actual number in use is num_slicers.
// Should be in range 1 .. MAX_SLICERS,
/*
* Version 1.6:
*
* This has become quite disorganized and messy with different combinations of
* fields used for different demodulator types. Start to reorganize it into a common
* part (with things like the DPLL for clock recovery), and separate sections
* for each of the demodulator types.
* Still a lot to do here.
*/
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 //
// //
//////////////////////////////////////////////////////////////////////////////////
// TODO: Continue experiments with root raised cosine filter.
// Either switch to that or take out all the related stuff.
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.
float lp_filter[MAX_FILTER_SIZE] __attribute__((aligned(16))); // Low pass filter.
// New in 1.7 - Polyphase filter to reduce CPU requirements.
float lp_polyphase_1[MAX_FILTER_SIZE] __attribute__((aligned(16)));
float lp_polyphase_2[MAX_FILTER_SIZE] __attribute__((aligned(16)));
float lp_polyphase_3[MAX_FILTER_SIZE] __attribute__((aligned(16)));
float lp_polyphase_4[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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. //
// //
//////////////////////////////////////////////////////////////////////////////////
struct psk_only_s {
enum v26_e v26_alt; // Which alternative when V.26.
float sin_table256[256]; // Precomputed sin table for speed.
// Optional band pass pre-filter before phase detector.
// TODO? put back into common section?
// TODO? Why was I thinking that?
int use_prefilter; // True to enable it.
float prefilter_baud; // Cutoff frequencies, as fraction of baud rate, beyond tones used.
// In the case of PSK, we use only a single tone of 1800 Hz.
// If we were using 2400 bps (= 1200 baud), this would be
// the fraction of 1200 for the cutoff below and above 1800.
float pre_filter_width_sym; /* Length in number of symbol times. */
int pre_filter_taps; /* Size of pre filter, in audio samples. */
bp_window_t pre_window;
float audio_in[MAX_FILTER_SIZE] __attribute__((aligned(16)));
float pre_filter[MAX_FILTER_SIZE] __attribute__((aligned(16)));
// Use local oscillator or correlate with previous sample.
int psk_use_lo; /* Use local oscillator rather than self correlation. */
unsigned int lo_step; /* How much to advance the local oscillator */
/* phase for each audio sample. */
unsigned int lo_phase; /* Local oscillator phase accumulator for PSK. */
// After mixing with LO before low pass filter.
float I_raw[MAX_FILTER_SIZE] __attribute__((aligned(16))); // signal * LO cos.
float Q_raw[MAX_FILTER_SIZE] __attribute__((aligned(16))); // signal * LO sin.
// 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. */
float delay_line_width_sym;
int delay_line_taps; // In audio samples.
float delay_line[MAX_FILTER_SIZE] __attribute__((aligned(16)));
// Low pass filter Second is frequency as ratio to baud rate for FIR.
// TODO? put back into common section?
// TODO? What are the tradeoffs?
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. */
float lp_filter_width_sym; /* Length in number of symbol times. */
int lp_filter_taps; /* Size of Low Pass filter, in audio samples (i.e. filter taps). */
bp_window_t lp_window;
float lp_filter[MAX_FILTER_SIZE] __attribute__((aligned(16)));
} psk;
} u; // end of union for different demodulator types.
};
/*-------------------------------------------------------------------
*
* Name: pll_dcd_signal_transition2
* dcd_each_symbol2
*
* Purpose: New DCD strategy for 1.6.
*
* Inputs: D Pointer to demodulator state.
*
* chan Radio channel: 0 to MAX_CHANS - 1
*
* subchan Which of multiple demodulators: 0 to MAX_SUBCHANS - 1
*
* slice Slicer number: 0 to MAX_SLICERS - 1.
*
* dpll_phase Signed 32 bit counter for DPLL phase.
* Wraparound is where data is sampled.
* Ideally transitions would occur close to 0.
*
* Output: D->slicer[slice].data_detect - true when PLL is locked to incoming signal.
*
* Description: From the beginning, DCD was based on finding several flag octets
* in a row and dropping when eight bits with no transitions.
* It was less than ideal but we limped along with it all these years.
* This fell apart when FX.25 came along and a couple of the
* correlation tags have eight "1" bits in a row.
*
* Our new strategy is to keep a running score of how well demodulator
* output transitions match to where expected.
*
*--------------------------------------------------------------------*/
#include "hdlc_rec.h" // for dcd_change
// These are good for 1200 bps AFSK.
// Might want to override for other modems.
#ifndef DCD_THRESH_ON
#define DCD_THRESH_ON 30 // Hysteresis: Can miss 2 out of 32 for detecting lock.
// 31 is best for TNC Test CD. 30 almost as good.
// 30 better for 1200 regression test.
#endif
#ifndef DCD_THRESH_OFF
#define DCD_THRESH_OFF 6 // Might want a little more fine tuning.
#endif
#ifndef DCD_GOOD_WIDTH
#define DCD_GOOD_WIDTH 512 // No more than 1024!!!
#endif
__attribute__((always_inline))
inline static void pll_dcd_signal_transition2 (struct demodulator_state_s *D, int slice, int dpll_phase)
{
if (dpll_phase > - DCD_GOOD_WIDTH * 1024 * 1024 && dpll_phase < DCD_GOOD_WIDTH * 1024 * 1024) {
D->slicer[slice].good_flag = 1;
}
else {
D->slicer[slice].bad_flag = 1;
}
}
__attribute__((always_inline))
inline static void pll_dcd_each_symbol2 (struct demodulator_state_s *D, int chan, int subchan, int slice)
{
D->slicer[slice].good_hist <<= 1;
D->slicer[slice].good_hist |= D->slicer[slice].good_flag;
D->slicer[slice].good_flag = 0;
D->slicer[slice].bad_hist <<= 1;
D->slicer[slice].bad_hist |= D->slicer[slice].bad_flag;
D->slicer[slice].bad_flag = 0;
D->slicer[slice].score <<= 1;
// 2 is to detect 'flag' patterns with 2 transitions per octet.
D->slicer[slice].score |= (signed)__builtin_popcount(D->slicer[slice].good_hist)
- (signed)__builtin_popcount(D->slicer[slice].bad_hist) >= 2;
int s = __builtin_popcount(D->slicer[slice].score);
if (s >= DCD_THRESH_ON) {
if (D->slicer[slice].data_detect == 0) {
D->slicer[slice].data_detect = 1;
dcd_change (chan, subchan, slice, D->slicer[slice].data_detect);
}
}
else if (s <= DCD_THRESH_OFF) {
if (D->slicer[slice].data_detect != 0) {
D->slicer[slice].data_detect = 0;
dcd_change (chan, subchan, slice, D->slicer[slice].data_detect);
}
}
}
#define FSK_DEMOD_STATE_H 1
#endif
Reference in New Issue View Git Blame Copy Permalink