mirror of
https://github.com/markqvist/OpenModem.git
synced 2024-10-01 03:15:46 -04:00
529 lines
12 KiB
C
529 lines
12 KiB
C
#include "afsk.h"
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#include "config.h"
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#include "hardware.h"
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#include <drv/timer.h>
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#include <cfg/module.h>
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#define HDLC_FLAG 0x7E
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#define HDLC_RESET 0x7F
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#define AX25_ESC 0x1B
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#include <cfg/log.h>
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#include <cpu/power.h>
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#include <cpu/pgm.h>
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#include <struct/fifobuf.h>
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#include <string.h> /* memset */
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#define PHASE_BIT 8
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#define PHASE_INC 1
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#define PHASE_MAX (SAMPLESPERBIT * PHASE_BIT)
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#define PHASE_THRES (PHASE_MAX / 2)
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// Modulator constants
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#define MARK_FREQ 1200
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#define MARK_INC (uint16_t)(DIV_ROUND(SIN_LEN * (uint32_t)MARK_FREQ, CONFIG_AFSK_DAC_SAMPLERATE))
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#define SPACE_FREQ 2200
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#define SPACE_INC (uint16_t)(DIV_ROUND(SIN_LEN * (uint32_t)SPACE_FREQ, CONFIG_AFSK_DAC_SAMPLERATE))
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//Ensure sample rate is a multiple of bit rate
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STATIC_ASSERT(!(CONFIG_AFSK_DAC_SAMPLERATE % BITRATE));
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#define DAC_SAMPLESPERBIT (CONFIG_AFSK_DAC_SAMPLERATE / BITRATE)
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/**
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* Sine table for the first quarter of wave.
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* The rest of the wave is computed from this first quarter.
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* This table is used to generate the modulated data.
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*/
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static const uint8_t PROGMEM sin_table[] =
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{
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128, 129, 131, 132, 134, 135, 137, 138, 140, 142, 143, 145, 146, 148, 149, 151,
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152, 154, 155, 157, 158, 160, 162, 163, 165, 166, 167, 169, 170, 172, 173, 175,
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176, 178, 179, 181, 182, 183, 185, 186, 188, 189, 190, 192, 193, 194, 196, 197,
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198, 200, 201, 202, 203, 205, 206, 207, 208, 210, 211, 212, 213, 214, 215, 217,
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218, 219, 220, 221, 222, 223, 224, 225, 226, 227, 228, 229, 230, 231, 232, 233,
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234, 234, 235, 236, 237, 238, 238, 239, 240, 241, 241, 242, 243, 243, 244, 245,
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245, 246, 246, 247, 248, 248, 249, 249, 250, 250, 250, 251, 251, 252, 252, 252,
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253, 253, 253, 253, 254, 254, 254, 254, 254, 255, 255, 255, 255, 255, 255, 255,
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};
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#define SIN_LEN 512 ///< Full wave length
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STATIC_ASSERT(sizeof(sin_table) == SIN_LEN / 4);
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/**
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* Given the index, this function computes the correct sine sample
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* based only on the first quarter of wave.
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*/
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INLINE uint8_t sin_sample(uint16_t idx)
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{
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ASSERT(idx < SIN_LEN);
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uint16_t new_idx = idx % (SIN_LEN / 2);
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new_idx = (new_idx >= (SIN_LEN / 4)) ? (SIN_LEN / 2 - new_idx - 1) : new_idx;
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uint8_t data = pgm_read8(&sin_table[new_idx]);
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return (idx >= (SIN_LEN / 2)) ? (255 - data) : data;
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}
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#define BIT_DIFFER(bitline1, bitline2) (((bitline1) ^ (bitline2)) & 0x01)
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#define EDGE_FOUND(bitline) BIT_DIFFER((bitline), (bitline) >> 1)
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/**
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* High-Level Data Link Control parsing function.
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* Parse bitstream in order to find characters.
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*
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* \param hdlc HDLC context.
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* \param bit current bit to be parsed.
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* \param fifo FIFO buffer used to push characters.
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*
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* \return true if all is ok, false if the fifo is full.
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*/
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static bool hdlc_parse(Hdlc *hdlc, bool bit, FIFOBuffer *fifo)
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{
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bool ret = true;
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hdlc->demod_bits <<= 1;
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hdlc->demod_bits |= bit ? 1 : 0;
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/* HDLC Flag */
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if (hdlc->demod_bits == HDLC_FLAG)
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{
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if (!fifo_isfull(fifo))
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{
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fifo_push(fifo, HDLC_FLAG);
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hdlc->rxstart = true;
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}
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else
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{
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ret = false;
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hdlc->rxstart = false;
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}
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hdlc->currchar = 0;
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hdlc->bit_idx = 0;
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return ret;
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}
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/* Reset */
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if ((hdlc->demod_bits & HDLC_RESET) == HDLC_RESET)
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{
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hdlc->rxstart = false;
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return ret;
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}
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if (!hdlc->rxstart)
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return ret;
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/* Stuffed bit */
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if ((hdlc->demod_bits & 0x3f) == 0x3e)
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return ret;
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if (hdlc->demod_bits & 0x01)
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hdlc->currchar |= 0x80;
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if (++hdlc->bit_idx >= 8)
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{
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if ((hdlc->currchar == HDLC_FLAG
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|| hdlc->currchar == HDLC_RESET
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|| hdlc->currchar == AX25_ESC))
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{
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if (!fifo_isfull(fifo))
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fifo_push(fifo, AX25_ESC);
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else
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{
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hdlc->rxstart = false;
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ret = false;
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}
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}
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if (!fifo_isfull(fifo))
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fifo_push(fifo, hdlc->currchar);
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else
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{
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hdlc->rxstart = false;
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ret = false;
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}
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hdlc->currchar = 0;
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hdlc->bit_idx = 0;
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}
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else
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hdlc->currchar >>= 1;
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return ret;
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}
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/**
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* ADC ISR callback.
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* This function has to be called by the ADC ISR when a sample of the configured
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* channel is available.
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* \param af Afsk context to operate on.
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* \param curr_sample current sample from the ADC.
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*/
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void afsk_adc_isr(Afsk *af, int8_t curr_sample)
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{
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AFSK_STROBE_ON();
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/*
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* Frequency discriminator and LP IIR filter.
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* This filter is designed to work
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* at the given sample rate and bit rate.
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*/
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STATIC_ASSERT(SAMPLERATE == 9600);
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STATIC_ASSERT(BITRATE == 1200);
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/*
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* Frequency discrimination is achieved by simply multiplying
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* the sample with a delayed sample of (samples per bit) / 2.
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* Then the signal is lowpass filtered with a first order,
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* 600 Hz filter. The filter implementation is selectable
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* through the CONFIG_AFSK_FILTER config variable.
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*/
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af->iir_x[0] = af->iir_x[1];
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#if (CONFIG_AFSK_FILTER == AFSK_BUTTERWORTH)
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af->iir_x[1] = ((int8_t)fifo_pop(&af->delay_fifo) * curr_sample) >> 2;
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//af->iir_x[1] = ((int8_t)fifo_pop(&af->delay_fifo) * curr_sample) / 6.027339492;
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#elif (CONFIG_AFSK_FILTER == AFSK_CHEBYSHEV)
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af->iir_x[1] = ((int8_t)fifo_pop(&af->delay_fifo) * curr_sample) >> 2;
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//af->iir_x[1] = ((int8_t)fifo_pop(&af->delay_fifo) * curr_sample) / 3.558147322;
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#else
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#error Filter type not found!
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#endif
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af->iir_y[0] = af->iir_y[1];
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#if CONFIG_AFSK_FILTER == AFSK_BUTTERWORTH
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/*
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* This strange sum + shift is an optimization for af->iir_y[0] * 0.668.
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* iir * 0.668 ~= (iir * 21) / 32 =
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* = (iir * 16) / 32 + (iir * 4) / 32 + iir / 32 =
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* = iir / 2 + iir / 8 + iir / 32 =
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* = iir >> 1 + iir >> 3 + iir >> 5
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*/
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af->iir_y[1] = af->iir_x[0] + af->iir_x[1] + (af->iir_y[0] >> 1) + (af->iir_y[0] >> 3) + (af->iir_y[0] >> 5);
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//af->iir_y[1] = af->iir_x[0] + af->iir_x[1] + af->iir_y[0] * 0.6681786379;
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#elif CONFIG_AFSK_FILTER == AFSK_CHEBYSHEV
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/*
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* This should be (af->iir_y[0] * 0.438) but
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* (af->iir_y[0] >> 1) is a faster approximation :-)
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*/
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af->iir_y[1] = af->iir_x[0] + af->iir_x[1] + (af->iir_y[0] >> 1);
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//af->iir_y[1] = af->iir_x[0] + af->iir_x[1] + af->iir_y[0] * 0.4379097269;
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#endif
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/* Save this sampled bit in a delay line */
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af->sampled_bits <<= 1;
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af->sampled_bits |= (af->iir_y[1] > 0) ? 1 : 0;
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/* Store current ADC sample in the af->delay_fifo */
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fifo_push(&af->delay_fifo, curr_sample);
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/* If there is an edge, adjust phase sampling */
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if (EDGE_FOUND(af->sampled_bits))
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{
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if (af->curr_phase < PHASE_THRES)
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af->curr_phase += PHASE_INC;
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else
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af->curr_phase -= PHASE_INC;
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}
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af->curr_phase += PHASE_BIT;
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/* sample the bit */
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if (af->curr_phase >= PHASE_MAX)
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{
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af->curr_phase %= PHASE_MAX;
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/* Shift 1 position in the shift register of the found bits */
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af->found_bits <<= 1;
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/*
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* Determine bit value by reading the last 3 sampled bits.
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* If the number of ones is two or greater, the bit value is a 1,
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* otherwise is a 0.
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* This algorithm presumes that there are 8 samples per bit.
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*/
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STATIC_ASSERT(SAMPLESPERBIT == 8);
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uint8_t bits = af->sampled_bits & 0x07;
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if (bits == 0x07 // 111, 3 bits set to 1
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|| bits == 0x06 // 110, 2 bits
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|| bits == 0x05 // 101, 2 bits
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|| bits == 0x03 // 011, 2 bits
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)
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af->found_bits |= 1;
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/*
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* NRZI coding: if 2 consecutive bits have the same value
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* a 1 is received, otherwise it's a 0.
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*/
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if (!hdlc_parse(&af->hdlc, !EDGE_FOUND(af->found_bits), &af->rx_fifo))
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af->status |= AFSK_RXFIFO_OVERRUN;
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}
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AFSK_STROBE_OFF();
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}
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static void afsk_txStart(Afsk *af)
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{
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if (!af->sending)
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{
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af->phase_inc = MARK_INC;
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af->phase_acc = 0;
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af->stuff_cnt = 0;
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af->sending = true;
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af->preamble_len = DIV_ROUND(CONFIG_AFSK_PREAMBLE_LEN * BITRATE, 8000);
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AFSK_DAC_IRQ_START(af->dac_ch);
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}
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ATOMIC(af->trailer_len = DIV_ROUND(CONFIG_AFSK_TRAILER_LEN * BITRATE, 8000));
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}
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#define BIT_STUFF_LEN 5
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#define SWITCH_TONE(inc) (((inc) == MARK_INC) ? SPACE_INC : MARK_INC)
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/**
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* DAC ISR callback.
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* This function has to be called by the DAC ISR when a sample of the configured
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* channel has been converted out.
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*
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* \param af Afsk context to operate on.
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*
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* \return The next DAC output sample.
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*/
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uint8_t afsk_dac_isr(Afsk *af)
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{
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AFSK_STROBE_ON();
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/* Check if we are at a start of a sample cycle */
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if (af->sample_count == 0)
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{
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if (af->tx_bit == 0)
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{
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/* We have just finished transimitting a char, get a new one. */
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if (fifo_isempty(&af->tx_fifo) && af->trailer_len == 0)
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{
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AFSK_DAC_IRQ_STOP(af->dac_ch);
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af->sending = false;
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AFSK_STROBE_OFF();
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return 0;
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}
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else
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{
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/*
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* If we have just finished sending an unstuffed byte,
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* reset bitstuff counter.
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*/
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if (!af->bit_stuff)
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af->stuff_cnt = 0;
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af->bit_stuff = true;
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/*
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* Handle preamble and trailer
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*/
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if (af->preamble_len == 0)
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{
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if (fifo_isempty(&af->tx_fifo))
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{
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af->trailer_len--;
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af->curr_out = HDLC_FLAG;
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}
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else
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af->curr_out = fifo_pop(&af->tx_fifo);
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}
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else
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{
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af->preamble_len--;
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af->curr_out = HDLC_FLAG;
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}
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/* Handle char escape */
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if (af->curr_out == AX25_ESC)
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{
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if (fifo_isempty(&af->tx_fifo))
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{
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AFSK_DAC_IRQ_STOP(af->dac_ch);
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af->sending = false;
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AFSK_STROBE_OFF();
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return 0;
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}
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else
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af->curr_out = fifo_pop(&af->tx_fifo);
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}
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else if (af->curr_out == HDLC_FLAG || af->curr_out == HDLC_RESET)
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/* If these chars are not escaped disable bit stuffing */
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af->bit_stuff = false;
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}
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/* Start with LSB mask */
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af->tx_bit = 0x01;
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}
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/* check for bit stuffing */
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if (af->bit_stuff && af->stuff_cnt >= BIT_STUFF_LEN)
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{
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/* If there are more than 5 ones in a row insert a 0 */
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af->stuff_cnt = 0;
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/* switch tone */
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af->phase_inc = SWITCH_TONE(af->phase_inc);
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}
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else
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{
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/*
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* NRZI: if we want to transmit a 1 the modulated frequency will stay
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* unchanged; with a 0, there will be a change in the tone.
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*/
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if (af->curr_out & af->tx_bit)
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{
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/*
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* Transmit a 1:
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* - Stay on the previous tone
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* - Increase bit stuff counter
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*/
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af->stuff_cnt++;
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}
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else
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{
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/*
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* Transmit a 0:
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* - Reset bit stuff counter
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* - Switch tone
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*/
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af->stuff_cnt = 0;
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af->phase_inc = SWITCH_TONE(af->phase_inc);
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}
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/* Go to the next bit */
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af->tx_bit <<= 1;
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}
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af->sample_count = DAC_SAMPLESPERBIT;
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}
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/* Get new sample and put it out on the DAC */
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af->phase_acc += af->phase_inc;
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af->phase_acc %= SIN_LEN;
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af->sample_count--;
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AFSK_STROBE_OFF();
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return sin_sample(af->phase_acc);
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}
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static size_t afsk_read(KFile *fd, void *_buf, size_t size)
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{
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Afsk *af = AFSK_CAST(fd);
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uint8_t *buf = (uint8_t *)_buf;
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#if CONFIG_AFSK_RXTIMEOUT == 0
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while (size-- && !fifo_isempty_locked(&af->rx_fifo))
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#else
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while (size--)
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#endif
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{
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#if CONFIG_AFSK_RXTIMEOUT != -1
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ticks_t start = timer_clock();
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#endif
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while (fifo_isempty_locked(&af->rx_fifo))
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{
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cpu_relax();
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#if CONFIG_AFSK_RXTIMEOUT != -1
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if (timer_clock() - start > ms_to_ticks(CONFIG_AFSK_RXTIMEOUT))
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return buf - (uint8_t *)_buf;
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#endif
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}
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*buf++ = fifo_pop_locked(&af->rx_fifo);
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}
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return buf - (uint8_t *)_buf;
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}
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static size_t afsk_write(KFile *fd, const void *_buf, size_t size)
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{
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Afsk *af = AFSK_CAST(fd);
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const uint8_t *buf = (const uint8_t *)_buf;
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while (size--)
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{
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while (fifo_isfull_locked(&af->tx_fifo))
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cpu_relax();
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fifo_push_locked(&af->tx_fifo, *buf++);
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afsk_txStart(af);
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}
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return buf - (const uint8_t *)_buf;
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}
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static int afsk_flush(KFile *fd)
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{
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Afsk *af = AFSK_CAST(fd);
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while (af->sending)
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cpu_relax();
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return 0;
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}
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static int afsk_error(KFile *fd)
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{
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Afsk *af = AFSK_CAST(fd);
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int err;
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ATOMIC(err = af->status);
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return err;
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}
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static void afsk_clearerr(KFile *fd)
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{
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Afsk *af = AFSK_CAST(fd);
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ATOMIC(af->status = 0);
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}
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/**
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* Initialize an AFSK1200 modem.
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* \param af Afsk context to operate on.
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* \param adc_ch ADC channel used by the demodulator.
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* \param dac_ch DAC channel used by the modulator.
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*/
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void afsk_init(Afsk *af, int adc_ch, int dac_ch)
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{
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#if CONFIG_AFSK_RXTIMEOUT != -1
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MOD_CHECK(timer);
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#endif
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memset(af, 0, sizeof(*af));
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af->adc_ch = adc_ch;
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af->dac_ch = dac_ch;
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|
|
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fifo_init(&af->delay_fifo, (uint8_t *)af->delay_buf, sizeof(af->delay_buf));
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fifo_init(&af->rx_fifo, af->rx_buf, sizeof(af->rx_buf));
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|
|
|
/* Fill sample FIFO with 0 */
|
|
for (int i = 0; i < SAMPLESPERBIT / 2; i++)
|
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fifo_push(&af->delay_fifo, 0);
|
|
|
|
fifo_init(&af->tx_fifo, af->tx_buf, sizeof(af->tx_buf));
|
|
|
|
AFSK_ADC_INIT(adc_ch, af);
|
|
AFSK_DAC_INIT(dac_ch, af);
|
|
AFSK_STROBE_INIT();
|
|
//LOG_INFO("MARK_INC %d, SPACE_INC %d\n", MARK_INC, SPACE_INC);
|
|
|
|
DB(af->fd._type = KFT_AFSK);
|
|
af->fd.write = afsk_write;
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|
af->fd.read = afsk_read;
|
|
af->fd.flush = afsk_flush;
|
|
af->fd.error = afsk_error;
|
|
af->fd.clearerr = afsk_clearerr;
|
|
af->phase_inc = MARK_INC;
|
|
}
|