mirror of https://github.com/wb2osz/direwolf.git
323 lines
9.0 KiB
C
323 lines
9.0 KiB
C
/* fsk_demod_state.h */
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#ifndef FSK_DEMOD_STATE_H
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#include "rpack.h"
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#include "audio.h" // for enum modem_t
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/*
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* Demodulator state.
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* The name of the file is from we only had FSK. Now we have other techniques.
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* Different copy is required for each channel & subchannel being processed concurrently.
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*/
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// TODO1.2: change prefix from BP_ to DSP_
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typedef enum bp_window_e { BP_WINDOW_TRUNCATED,
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BP_WINDOW_COSINE,
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BP_WINDOW_HAMMING,
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BP_WINDOW_BLACKMAN,
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BP_WINDOW_FLATTOP } bp_window_t;
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struct demodulator_state_s
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{
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/*
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* These are set once during initialization.
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*/
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enum modem_t modem_type; // MODEM_AFSK, MODEM_8PSK, etc.
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char profile; // 'A', 'B', etc. Upper case.
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// Only needed to see if we are using 'F' to take fast path.
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int play_it_again_sample; // Enable new synchronous demod in version 1.6.
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#define TICKS_PER_PLL_CYCLE ( 256.0 * 256.0 * 256.0 * 256.0 )
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int pll_step_per_sample; // PLL is advanced by this much each audio sample.
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// Data is sampled when it overflows.
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int ms_filter_size; /* Size of mark & space filters, in audio samples. */
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/* Started off as a guess of one bit length */
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/* but about 2 bit times turned out to be better. */
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/* Currently using same size for any prefilter. */
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int m2_filter_size;
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int s2_filter_size; /* Size of mark & space filters, in audio samples */
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/* for the synchronous demodulator. I'm expecting */
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/* smaller, perhaps just over 1 bit time here. */
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int lp2_filter_size; /* FSK resampling - Size of Low Pass filter, in audio samples. */
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#define MAX_FILTER_SIZE 320 /* 304 is needed for profile C, 300 baud & 44100. */
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/*
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* Filter length for Mark & Space in bit times.
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* e.g. 1 means 1/1200 second for 1200 baud.
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*/
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float ms_filter_len_bits;
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float m2_filter_len_bits;
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float s2_filter_len_bits;
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float lp_delay_fract;
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/*
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* Window type for the various filters.
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*/
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bp_window_t pre_window;
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bp_window_t ms_window;
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bp_window_t lp_window;
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bp_window_t ms2_window; /* New in 1.6. */
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/*
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* Alternate Low pass filters.
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* First is arbitrary number for quick IIR.
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* Second is frequency as ratio to baud rate for FIR.
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*/
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int lpf_use_fir; /* 0 for IIR, 1 for FIR. */
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float lpf_iir; /* Only if using IIR. */
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float lpf_baud; /* Cutoff frequency as fraction of baud. */
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/* Intuitively we'd expect this to be somewhere */
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/* in the range of 0.5 to 1. */
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/* In practice, it turned out a little larger */
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/* for profiles B, C, D. */
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float lp_filter_len_bits; /* Length in number of bit times. */
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int lp_filter_size; /* Size of Low Pass filter, in audio samples. */
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/* Previously it was always the same as the M/S */
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/* filters but in version 1.2 it's now independent. */
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int lp_filter_delay; /* Number of samples that the low pass filter */
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/* delays the signal. */
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/* New in 1.6. */
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/*
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* Automatic gain control. Fast attack and slow decay factors.
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*/
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float agc_fast_attack;
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float agc_slow_decay;
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/*
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* Use a longer term view for reporting signal levels.
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*/
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float quick_attack;
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float sluggish_decay;
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/*
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* Hysteresis before final demodulator 0 / 1 decision.
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*/
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float hysteresis;
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int num_slicers; /* >1 for multiple slicers. */
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/*
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* Phase Locked Loop (PLL) inertia.
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* Larger number means less influence by signal transitions.
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*/
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float pll_locked_inertia;
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float pll_searching_inertia;
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/*
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* Optional band pass pre-filter before mark/space detector.
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*/
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int use_prefilter; /* True to enable it. */
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float prefilter_baud; /* Cutoff frequencies, as fraction of */
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/* baud rate, beyond tones used. */
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/* Example, if we used 1600/1800 tones at */
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/* 300 baud, and this was 0.5, the cutoff */
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/* frequencies would be: */
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/* lower = min(1600,1800) - 0.5 * 300 = 1450 */
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/* upper = max(1600,1800) + 0.5 * 300 = 1950 */
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float pre_filter_len_bits; /* Length in number of bit times. */
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int pre_filter_size; /* Size of pre filter, in audio samples. */
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float pre_filter[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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/*
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* Kernel for the mark and space detection filters.
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*/
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float m_sin_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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float m_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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float s_sin_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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float s_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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/*
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* Same for the synchronous re-demodulator.
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*/
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float m2_sin_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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float m2_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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float s2_sin_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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float s2_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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float lp2_filter[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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/*
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* These are for PSK only.
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* They are number of delay line taps into previous symbol.
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* They are one symbol period and + or - 45 degrees of the carrier frequency.
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*/
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int boffs; /* symbol length based on sample rate and baud. */
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int coffs; /* to get cos component of previous symbol. */
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int soffs; /* to get sin component of previous symbol. */
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unsigned int lo_step; /* How much to advance the local oscillator */
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/* phase for each audio sample. */
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int psk_use_lo; /* Use local oscillator rather than self correlation. */
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/*
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* The rest are continuously updated.
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*/
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unsigned int lo_phase; /* Local oscillator for PSK. */
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/*
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* Most recent raw audio samples, before/after prefiltering.
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*/
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float raw_cb[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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/*
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* Use half of the AGC code to get a measure of input audio amplitude.
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* These use "quick" attack and "sluggish" decay while the
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* AGC uses "fast" attack and "slow" decay.
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*/
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float alevel_rec_peak;
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float alevel_rec_valley;
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float alevel_mark_peak;
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float alevel_space_peak;
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/*
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* Input to the mark/space detector.
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* Could be prefiltered or raw audio.
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*/
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float ms_in_cb[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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/*
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* Outputs from the mark and space amplitude detection,
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* used as inputs to the FIR lowpass filters.
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* Kernel for the lowpass filters.
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*/
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float m_amp_cb[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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float s_amp_cb[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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float lp_filter[MAX_FILTER_SIZE] __attribute__((aligned(16)));
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float m_peak, s_peak;
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float m_valley, s_valley;
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float m_amp_prev, s_amp_prev;
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/*
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* For the PLL and data bit timing.
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* starting in version 1.2 we can have multiple slicers for one demodulator.
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* Each slicer has its own PLL and HDLC decoder.
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*/
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/*
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* Version 1.3: Clean up subchan vs. slicer.
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*
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* Originally some number of CHANNELS (originally 2, later 6)
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* which can have multiple parallel demodulators called SUB-CHANNELS.
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* This was originally for staggered frequencies for HF SSB.
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* It can also be used for multiple demodulators with the same
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* frequency but other differing parameters.
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* Each subchannel has its own demodulator and HDLC decoder.
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*
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* In version 1.2 we added multiple SLICERS.
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* The data structure, here, has multiple slicers per
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* demodulator (subchannel). Due to fuzzy thinking or
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* expediency, the multiple slicers got mapped into subchannels.
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* This means we can't use both multiple decoders and
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* multiple slicers at the same time.
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*
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* Clean this up in 1.3 and keep the concepts separate.
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* This means adding a third variable many places
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* we are passing around the origin.
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*
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*/
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struct {
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signed int data_clock_pll; // PLL for data clock recovery.
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// It is incremented by pll_step_per_sample
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// for each audio sample.
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signed int prev_d_c_pll; // Previous value of above, before
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// incrementing, to detect overflows.
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int prev_demod_data; // Previous data bit detected.
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// Used to look for transitions.
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float prev_demod_out_f;
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/* This is used only for "9600" baud data. */
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int lfsr; // Descrambler shift register.
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} slicer [MAX_SLICERS]; // Actual number in use is num_slicers.
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// Should be in range 1 .. MAX_SLICERS,
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/*
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* Special for Rino decoder only.
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* One for each possible signal polarity.
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* The project showed promise but fell by the wayside.
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*/
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#if 0
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struct gr_state_s {
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signed int data_clock_pll; // PLL for data clock recovery.
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// It is incremented by pll_step_per_sample
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// for each audio sample.
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signed int prev_d_c_pll; // Previous value of above, before
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// incrementing, to detect overflows.
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float gr_minus_peak; // For automatic gain control.
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float gr_plus_peak;
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int gr_sync; // Is sync pulse present?
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int gr_prev_sync; // Previous state to detect leading edge.
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int gr_first_sample; // Index of starting sample index for debugging.
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int gr_dcd; // Data carrier detect. i.e. are we
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// currently decoding a message.
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float gr_early_sum; // For averaging bit values in two regions.
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int gr_early_count;
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float gr_late_sum;
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int gr_late_count;
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float gr_sync_sum;
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int gr_sync_count;
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int gr_bit_count; // Bit index into message.
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struct rpack_s rpack; // Collection of bits.
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} gr_state[2];
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#endif
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};
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#define FSK_DEMOD_STATE_H 1
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#endif |