/* 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. 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. // 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 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. */ /* * Filter length for Mark & Space in bit times. * 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; /* * Window type for the various filters. */ bp_window_t pre_window; bp_window_t ms_window; bp_window_t lp_window; bp_window_t ms2_window; /* New in 1.6. */ /* * 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. */ int lp_filter_delay; /* Number of samples that the low pass filter */ /* delays the signal. */ /* New in 1.6. */ /* * 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))); /* * 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. * 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