mirror of
https://github.com/markqvist/OpenModem.git
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735 lines
30 KiB
C
735 lines
30 KiB
C
//////////////////////////////////////////////////////
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// First things first, all the includes we need //
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//////////////////////////////////////////////////////
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#include "afsk.h" // We need the header file for the modem
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#include "config.h" // This stores basic configuration
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#include "hardware.h" // Hardware functions are nice to have too :)
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#include <drv/timer.h> // Timer driver from BertOS
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#include <cpu/power.h> // Power management from BertOS
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#include <cpu/pgm.h> // Access to PROGMEM from BertOS
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#include <struct/fifobuf.h> // FIFO buffer implementation from BertOS
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#include <string.h> // String operations, primarily used for memset function
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//////////////////////////////////////////////////////
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// Definitions and some useful macros //
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//////////////////////////////////////////////////////
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// Sine table for Direct Digital Synthesis DAC
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// Since it would be inefficient to calculate a sine value each
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// time we process a sample, we store the values in program memory
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// as a look-up table. We only need to store values for a quarter
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// wave, since we can easily reconstruct the entire 512 values
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// from only these 128 values.
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#define SIN_LEN 512
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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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}; STATIC_ASSERT(sizeof(sin_table) == SIN_LEN / 4);
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// Calculate any sine value from quarter wave sine table
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// The reason we declare this inline is to eliminate an extra
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// call for the code. The code is essentially inserted directly
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// in the calling functions code. This makes stuff faster :)
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INLINE uint8_t sinSample(uint16_t i) {
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// Make sure that the index asked for is in the correct range
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ASSERT(i < SIN_LEN);
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// First we make a new index value, and restrict it to only
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// the first half-wave of the sine.
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uint16_t newI = i % (SIN_LEN/2);
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// We then check if this new index is larger than the first
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// quarter wave. If it is, we don't have the value for this
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// index directly, but we can figure it out by subtracting
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// the new index from a half wave, effectively wrapping us
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// back into the same place on the wave, whithin the quarter
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// wave we have data for, only with the inverse sign. If the
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// index was actually in the first quarter, we don't need to
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// do anything.
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newI = (newI >= (SIN_LEN/4)) ? (SIN_LEN/2 - newI -1) : newI;
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// Now we just need to read the value from program memory
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uint8_t sine = pgm_read8(&sin_table[newI]);
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// And flip the sign (+/-) if the original index was greater
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// than a half wave.
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return (i >= (SIN_LEN/2)) ? (255 - sine) : sine;
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}
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// A very basic macro that just checks whether the last bit
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// of a whatever is passed into it differ. This is used in the
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// next macro.
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#define BITS_DIFFER(bits1, bits2) (((bits1)^(bits2)) & 0x01)
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// This macro is used to look for signal transitions. We need
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// to identify these to keep the phase of our demodulator in
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// sync with the incoming signal. Each time we find a signal
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// transition on the physical medium, we adjust the phase of
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// the demodulator.
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// The macro effectively looks at the two least significant
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// bits in a stream and returns true if they differ.
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#define TRANSITION_FOUND(bits) BITS_DIFFER((bits), (bits) >> 1)
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// We use this macro to check if the signal transitioned
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// from one bit (tone) to another. This is used in the phase
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// synchronisation. We look at the last four bits in the
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// stream of demodulated bits and if they differ in sets of
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// two bits, we assume a signal transition occured. We look
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// at pairs of bits to eliminate false positives where a
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// single erroneously demodulated bit will trigger an
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// incorrect phase syncronisation.
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#define DUAL_XOR(bits1, bits2) ((((bits1)^(bits2)) & 0x03) == 0x03)
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#define SIGNAL_TRANSITIONED(bits) DUAL_XOR((bits), (bits) >> 2)
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// Phase sync constants
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#define PHASE_BITS 8 // How much to increment phase counter each sample
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#define PHASE_INC 1 // Nudge by an eigth of a sample each adjustment
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#define PHASE_MAX (SAMPLESPERBIT * PHASE_BITS) // Resolution of our phase counter = 64
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#define PHASE_THRESHOLD (PHASE_MAX / 2) // Target transition point of our phase window
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// Modulation constants
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#define MARK_FREQ 1200 // The tone frequency signifying a binary one
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#define SPACE_FREQ 2200 // The tone frequency signifying a binary zero
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// We calculate the amount we need to increment the index
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// in our sine table for each sample of the two tones
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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_INC (uint16_t)(DIV_ROUND(SIN_LEN * (uint32_t)SPACE_FREQ, CONFIG_AFSK_DAC_SAMPLERATE))
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// HDLC flag bytes
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#define HDLC_FLAG 0x7E // An HDLC_FLAG is used to signify the start or end of a frame
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#define HDLC_RESET 0x7F // An HDLC_RESET is used to abruptly stop or reset a transmission
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#define AX25_ESC 0x1B // We use the AX.25 escape character for escaping bit sequences in
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// the actual data. This is similar to escaping an " character in a
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// string enclosed by "s.
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// Check that sample rate is divisible by bitrate.
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// If this is not the case, all of our algorithms will
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// fail horribly and we will cry.
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STATIC_ASSERT(!(CONFIG_AFSK_DAC_SAMPLERATE % BITRATE));
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// How many samples it takes to encode or decode one bit
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// on the physical medium.
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#define DAC_SAMPLESPERBIT (CONFIG_AFSK_DAC_SAMPLERATE / BITRATE)
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//////////////////////////////////////////////////////
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// Link Layer Control and Demodulation //
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//////////////////////////////////////////////////////
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// hdlcParse /////////////////////////////////////////
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// This function looks at the raw bits demodulated from
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// the physical medium and tries to parse actual data
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// packets from the bitstream. Note that at this level,
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// we don't really try to discriminate when a packet
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// starts or ends, or where the payload is. We only try
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// to detect that a transmission is taking place, then
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// synchronise to the start and end of the transmitted
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// bytes, and push these up to the data-link layer, in
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// this example the MP.x protocol. It is then the
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// protocols job to actually recreate the full packet.
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// Also note that the data is not "pushed" per se, but
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// stored in a FIFO buffer, that the protocol must
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// continously read to recreate the received packets.
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static bool hdlcParse(Hdlc *hdlc, bool bit, FIFOBuffer *fifo) {
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// Initialise a return value. We start with the
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// assumption that all is going to end well :)
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bool ret = true;
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// Bitshift our byte of demodulated bits to
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// the left by one bit, to make room for the
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// next incoming bit
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hdlc->demodulatedBits <<= 1;
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// And then put the newest bit from the
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// demodulator into the byte.
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hdlc->demodulatedBits |= bit ? 1 : 0;
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// Now we'll look at the last 8 received bits, and
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// check if we have received a HDLC flag (01111110)
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if (hdlc->demodulatedBits == HDLC_FLAG) {
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// If we have, check that our output buffer is
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// not full.
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if (!fifo_isfull(fifo)) {
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// If it isn't, we'll push the HDLC_FLAG into
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// the buffer and indicate that we are now
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// receiving data. For bling we also turn
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// on the RX LED.
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fifo_push(fifo, HDLC_FLAG);
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hdlc->receiving = true;
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LED_RX_ON();
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} else {
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// If the buffer is full, we have a problem
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// and abort by setting the return value to
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// false and stopping the here.
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ret = false;
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hdlc->receiving = false;
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LED_RX_OFF();
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}
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// Everytime we receive a HDLC_FLAG, we reset the
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// storage for our current incoming byte and bit
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// position in that byte. This effectively
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// synchronises our parsing to the start and end
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// of the received bytes.
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hdlc->currentByte = 0;
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hdlc->bitIndex = 0;
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return ret;
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}
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// Check if we have received a RESET flag (01111111)
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// In this comparison we also detect when no transmission
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// (or silence) is taking place, and the demodulator
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// returns an endless stream of zeroes. Due to the NRZ
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// coding, the actual bits send to this function will
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// be an endless stream of ones, which this AND operation
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// will also detect.
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if ((hdlc->demodulatedBits & HDLC_RESET) == HDLC_RESET) {
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// If we have, something probably went wrong at the
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// transmitting end, and we abort the reception.
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hdlc->receiving = false;
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LED_RX_OFF();
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return ret;
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}
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// If we have not yet seen a HDLC_FLAG indicating that
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// a transmission is actually taking place, don't bother
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// with anything.
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if (!hdlc->receiving)
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return ret;
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// First check if what we are seeing is a stuffed bit.
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// Since the different HDLC control characters like
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// HDLC_FLAG, HDLC_RESET and such could also occur in
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// a normal data stream, we employ a method known as
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// "bit stuffing". All control characters have more than
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// 5 ones in a row, so if the transmitting party detects
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// this sequence in the _data_ to be transmitted, it inserts
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// a zero to avoid the receiving party interpreting it as
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// a control character. Therefore, if we detect such a
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// "stuffed bit", we simply ignore it and wait for the
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// next bit to come in.
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//
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// We do the detection by applying an AND bit-mask to the
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// stream of demodulated bits. This mask is 00111111 (0x3f)
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// if the result of the operation is 00111110 (0x3e), we
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// have detected a stuffed bit.
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if ((hdlc->demodulatedBits & 0x3f) == 0x3e)
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return ret;
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// If we have an actual 1 bit, push this to the current byte
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// If it's a zero, we don't need to do anything, since the
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// bit is initialized to zero when we bitshifted earlier.
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if (hdlc->demodulatedBits & 0x01)
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hdlc->currentByte |= 0x80;
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// Increment the bitIndex and check if we have a complete byte
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if (++hdlc->bitIndex >= 8) {
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// If we have a HDLC control character, put a AX.25 escape
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// in the received data. We know we need to do this,
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// because at this point we must have already seen a HDLC
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// flag, meaning that this control character is the result
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// of a bitstuffed byte that is equal to said control
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// character, but is actually part of the data stream.
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// By inserting the escape character, we tell the protocol
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// layer that this is not an actual control character, but
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// data.
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if ((hdlc->currentByte == HDLC_FLAG ||
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hdlc->currentByte == HDLC_RESET ||
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hdlc->currentByte == AX25_ESC)) {
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// We also need to check that our received data buffer
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// is not full before putting more data in
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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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// If it is, abort and return false
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hdlc->receiving = false;
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LED_RX_OFF();
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ret = false;
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}
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}
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// Push the actual byte to the received data FIFO,
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// if it isn't full.
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if (!fifo_isfull(fifo)) {
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fifo_push(fifo, hdlc->currentByte);
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} else {
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// If it is, well, you know by now!
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hdlc->receiving = false;
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LED_RX_OFF();
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ret = false;
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}
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// Wipe received byte and reset bit index to 0
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hdlc->currentByte = 0;
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hdlc->bitIndex = 0;
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} else {
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// We don't have a full byte yet, bitshift the byte
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// to make room for the next bit
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hdlc->currentByte >>= 1;
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}
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return ret;
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}
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// adcISR ////////////////////////////////////////////
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// This is the Interrupt Service Routine for the
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// Analog to Digital Conversion. It is called 9600
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// times each second to analyze the sample taken from
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// the physical medium. The job of this routine is
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// to detect whether we have a "mark" or "space"
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// frequency present on the baseband (the physical
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// medium). The result of this analysis will then
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// be passed to the HDLC parser in form of a 1 or a 0
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void afsk_adc_isr(Afsk *afsk, int8_t currentSample) {
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// To determine the received frequency, and thereby
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// the bit of the sample, we multiply the sample by
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// a sample delayed by (samples per bit / 2).
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// We then lowpass-filter the samples with a
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// Chebyshev filter. The lowpass filtering serves
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// to "smooth out" the variations in the samples.
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afsk->iirX[0] = afsk->iirX[1];
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afsk->iirX[1] = ((int8_t)fifo_pop(&afsk->delayFifo) * currentSample) >> 2;
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afsk->iirY[0] = afsk->iirY[1];
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afsk->iirY[1] = afsk->iirX[0] + afsk->iirX[1] + (afsk->iirY[0] >> 1); // Chebyshev filter
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// We put the sampled bit in a delay-line:
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// First we bitshift everything 1 left
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afsk->sampledBits <<= 1;
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// And then add the sampled bit to our delay line
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afsk->sampledBits |= (afsk->iirY[1] > 0) ? 1 : 0;
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// Put the current raw sample in the delay FIFO
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fifo_push(&afsk->delayFifo, currentSample);
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// We need to check whether there is a signal transition.
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// If there is, we can recalibrate the phase of our
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// sampler to stay in sync with the transmitter. A bit of
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// explanation is required to understand how this works.
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// Since we have PHASE_MAX/PHASE_BITS = 8 samples per bit,
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// we employ a phase counter (currentPhase), that increments
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// by PHASE_BITS everytime a sample is captured. When this
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// counter reaches PHASE_MAX, it wraps around by modulus
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// PHASE_MAX. We then look at the last three samples we
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// captured and determine if the bit was a one or a zero.
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//
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// This gives us a "window" looking into the stream of
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// samples coming from the ADC. Sort of like this:
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//
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// Past Future
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// 0000000011111111000000001111111100000000
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// |________|
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// ||
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// Window
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//
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// Every time we detect a signal transition, we adjust
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// where this window is positioned little. How much we
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// adjust it is defined by PHASE_INC. If our current phase
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// phase counter value is less than half of PHASE_MAX (ie,
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// the window size) when a signal transition is detected,
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// add PHASE_INC to our phase counter, effectively moving
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// the window a little bit backward (to the left in the
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// illustration), inversely, if the phase counter is greater
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// than half of PHASE_MAX, we move it forward a little.
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// This way, our "window" is constantly seeking to position
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// it's center at the bit transitions. Thus, we synchronise
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// our timing to the transmitter, even if it's timing is
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// a little off compared to our own.
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if (SIGNAL_TRANSITIONED(afsk->sampledBits)) {
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if (afsk->currentPhase < PHASE_THRESHOLD) {
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afsk->currentPhase += PHASE_INC;
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} else {
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afsk->currentPhase -= PHASE_INC;
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}
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}
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// We increment our phase counter
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afsk->currentPhase += PHASE_BITS;
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// Check if we have reached the end of
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// our sampling window.
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if (afsk->currentPhase >= PHASE_MAX) {
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// If we have, wrap around our phase
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// counter by modulus
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afsk->currentPhase %= PHASE_MAX;
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// Bitshift to make room for the next
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// bit in our stream of demodulated bits
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afsk->actualBits <<= 1;
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// We determine the actual bit value by reading
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// the last 3 sampled bits. If there is three or
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// more 1's, we will assume that the transmitter
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// sent us a one, otherwise we assume a zero
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uint8_t bits = afsk->sampledBits & 0x07;
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if (bits == 0x07 || // 111
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bits == 0x06 || // 110
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bits == 0x05 || // 101
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bits == 0x03 // 011
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) {
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afsk->actualBits |= 1;
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}
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//// Alternative using five bits ////////////////
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// uint8_t bits = afsk->sampledBits & 0x0f;
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// uint8_t c = 0;
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// c += bits & BV(1);
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// c += bits & BV(2);
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// c += bits & BV(3);
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// c += bits & BV(4);
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// c += bits & BV(5);
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// if (c >= 3) afsk->actualBits |= 1;
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/////////////////////////////////////////////////
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// Now we can pass the actual bit to the HDLC parser.
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// We are using NRZ coding, so if 2 consecutive bits
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// have the same value, we have a 1, otherwise a 0.
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// We use the TRANSITION_FOUND function to determine this.
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//
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// This is smart in combination with bit stuffing,
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// since it ensures a transmitter will never send more
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// than five consecutive 1's. When sending consecutive
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// ones, the signal stays at the same level, and if
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// this happens for longer periods of time, we would
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// not be able to synchronize our phase to the transmitter
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// and would start experiencing "bit slip".
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//
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// By combining bit-stuffing with NRZ coding, we ensure
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// that the signal will regularly make transitions
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// that we can use to synchronize our phase.
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//
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// We also check the return of the Link Control parser
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// to check if an error occured.
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if (!hdlcParse(&afsk->hdlc, !TRANSITION_FOUND(afsk->actualBits), &afsk->rxFifo)) {
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afsk->status |= RX_OVERRUN;
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}
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}
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}
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//////////////////////////////////////////////////////
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// Signal modulation and DAC //
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//////////////////////////////////////////////////////
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// Defines how many consecutive ones we send
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// before we need to "stuff" in a zero
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#define BIT_STUFF_LEN 5
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// A macro for switching what tone is being
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// synthesized by the DAC. We basically just
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// change how quickly we go through the sine
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// table each time we send out a sample. This
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// is done by changing the phaseInc variable
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#define SWITCH_TONE(inc) (((inc) == MARK_INC) ? SPACE_INC : MARK_INC)
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// This function starts the transmission
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static void afsk_txStart(Afsk *afsk) {
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if (!afsk->sending) {
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// Initialize the phase increment to
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// that of the mark frequency (zero)
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afsk->phaseInc = MARK_INC;
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// Reset the phase accumulator to 0
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afsk->phaseAcc = 0;
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// And also the bitstuff counter
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afsk->bitstuffCount = 0;
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// Indicate we are now sending
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afsk->sending = true;
|
|
// And turn on the blingy LED
|
|
LED_TX_ON();
|
|
// We also need to calculate how many HDLC_FLAG
|
|
// bytes we need to send in preamble
|
|
afsk->preambleLength = DIV_ROUND(CONFIG_AFSK_PREAMBLE_LEN * BITRATE, 8000);
|
|
AFSK_DAC_IRQ_START();
|
|
}
|
|
// We make the same calculation for the tail length,
|
|
// but this needs to be atomic, since the txStart
|
|
// function could potentially be called while we
|
|
// are already transmitting.
|
|
ATOMIC(afsk->tailLength = DIV_ROUND(CONFIG_AFSK_TRAILER_LEN * BITRATE, 8000));
|
|
}
|
|
|
|
// This is the DAC ISR, called at sampling rate whenever the DAC IRQ is on.
|
|
// It modulates the data to be transmitted and returns a value directly
|
|
// for output on the DAC
|
|
uint8_t afsk_dac_isr(Afsk *afsk) {
|
|
// Check whether we are at the beginning of a sample
|
|
if (afsk->sampleIndex == 0) {
|
|
// If we are, we should figure out what we are
|
|
// actually going to modulate and transmit :)
|
|
if (afsk->txBit == 0) {
|
|
// txBit is a bitmask that is ANDed to the
|
|
// byte we are sending. It is bitshifted one
|
|
// position left each time we shift the next
|
|
// bit. If it is 0, we know we are at the
|
|
// beginning of the next byte, and nothing
|
|
// has been transmitted yet.
|
|
|
|
// If TX FIFO is empty and tail-length has decremented to 0
|
|
// we are done, stop the IRQ and reset
|
|
if (fifo_isempty(&afsk->txFifo) && afsk->tailLength == 0) {
|
|
AFSK_DAC_IRQ_STOP();
|
|
afsk->sending = false;
|
|
LED_TX_OFF();
|
|
return 0;
|
|
} else {
|
|
// Reset the bitstuff counter if we have just sent
|
|
// a bitstuffed byte
|
|
if (!afsk->bitStuff) afsk->bitstuffCount = 0;
|
|
// Reset bitstuff indicator to true, signifying
|
|
// that it's ok to bit stuff.
|
|
afsk->bitStuff = true;
|
|
|
|
// Check if we are in preamble or tail
|
|
if (afsk->preambleLength == 0) {
|
|
// We are not in preamble
|
|
if (fifo_isempty(&afsk->txFifo)) {
|
|
// If the TX buffer is empty, we must
|
|
// be in the TX tail then.
|
|
// Decrement the tail counter and send
|
|
// a HDLC_FLAG
|
|
afsk->tailLength--;
|
|
afsk->currentOutputByte = HDLC_FLAG;
|
|
} else {
|
|
// If preamble is already transmitted and TX
|
|
// buffer is not empty, we should get a byte
|
|
// for transmission
|
|
afsk->currentOutputByte = fifo_pop(&afsk->txFifo);
|
|
}
|
|
} else {
|
|
// We are in preamble. We'll decrement
|
|
// the preamble counter and transmit a
|
|
// HDLC_FLAG
|
|
afsk->preambleLength--;
|
|
afsk->currentOutputByte = HDLC_FLAG;
|
|
}
|
|
|
|
// This handles escape sequences and control
|
|
// characters. First we check if the current
|
|
// byte is an escape character. If it is, we
|
|
// know the next byte, even though it looks
|
|
// like an HDLC control character, in fact is
|
|
// not. Therefore we'll fetch it and transmit
|
|
// it as data using bit stuffing.
|
|
if (afsk->currentOutputByte == AX25_ESC) {
|
|
// First make sure that the TX buffer is
|
|
// not empty for some strange reason
|
|
if (fifo_isempty(&afsk->txFifo)) {
|
|
AFSK_DAC_IRQ_STOP();
|
|
afsk->sending = false;
|
|
LED_TX_OFF();
|
|
return 0;
|
|
} else {
|
|
// If it is not, fetch the next byte
|
|
afsk->currentOutputByte = fifo_pop(&afsk->txFifo);
|
|
}
|
|
} else if (afsk->currentOutputByte == HDLC_FLAG || afsk->currentOutputByte == HDLC_RESET) {
|
|
// If there was not an escape character and
|
|
// this byte is an HDLC control character,
|
|
// we know that it is an _actual_ control
|
|
// character, and we indicate that it should
|
|
// not be bitstuffed.
|
|
afsk->bitStuff = false;
|
|
}
|
|
}
|
|
// Since we are at the beginning of a byte,
|
|
// we'll initialize the txBit mask to:
|
|
// 00000001. It will then be bit-shifted one
|
|
// position to the left each time we send the
|
|
// next bit. By ANDing this mask to the byte
|
|
// we are sending, we can quickly figure out
|
|
// what tone we should transmit. For example:
|
|
//
|
|
// If we are sending bit number 4 of the
|
|
// byte: 01101011
|
|
// The bit mask would be: 00001000
|
|
// If we AND the byte and the
|
|
// mask, we get: 00001000
|
|
// Since this is not zero, we know we should
|
|
// transmit a one.
|
|
afsk->txBit = 0x01;
|
|
}
|
|
|
|
// First we need to check for bit-stuffing
|
|
if (afsk->bitStuff && afsk->bitstuffCount >= BIT_STUFF_LEN) {
|
|
// If we are allowed to bit-stuff, and we have
|
|
// reached the maximum number of consecutive
|
|
// ones, we'll reset the bit-stuff counter and
|
|
// insert a zero into the bitstream
|
|
afsk->bitstuffCount = 0;
|
|
afsk->phaseInc = SWITCH_TONE(afsk->phaseInc);
|
|
} else {
|
|
// If we don't need to bit-stuff now, we can get
|
|
// on with the actual transmission.
|
|
//
|
|
// We are using NRZ so if we want to transmit a 1
|
|
// the modulated signal will stay the same. For a 0
|
|
// we make the signal transition.
|
|
if (afsk->currentOutputByte & afsk->txBit) {
|
|
// We don't do anything, aka stay on the same
|
|
// tone as before. We have sent one 1, so we
|
|
// increment the bitstuff counter.
|
|
afsk->bitstuffCount++;
|
|
} else {
|
|
// We switch the tone, and reset the bitstuff
|
|
// counter, since we have now transmitted a
|
|
// zero
|
|
afsk->bitstuffCount = 0;
|
|
afsk->phaseInc = SWITCH_TONE(afsk->phaseInc);
|
|
}
|
|
|
|
// Bitshift the mast to allow for the next
|
|
// bit in the byte to be transmitted
|
|
afsk->txBit <<= 1;
|
|
}
|
|
|
|
// We set sampleIndex to DAC_SAMPLESPERBIT,
|
|
// so we will transmit this bit for the number
|
|
// of samples one bit requires to transmit at
|
|
// the chosen bitrate.
|
|
afsk->sampleIndex = DAC_SAMPLESPERBIT;
|
|
}
|
|
|
|
// We increment the phase accumulator
|
|
// by the amount needed for the tone
|
|
afsk->phaseAcc += afsk->phaseInc;
|
|
// We then make sure that we have not
|
|
// exceeded the length of our sine table
|
|
afsk->phaseAcc %= SIN_LEN;
|
|
// Finally we decrement the sample counter
|
|
afsk->sampleIndex--;
|
|
// ... and return the sample to for it to
|
|
// be written out
|
|
return sinSample(afsk->phaseAcc);
|
|
}
|
|
|
|
|
|
//////////////////////////////////////////////////////
|
|
// File operation functions for read/write //
|
|
// These functions make the "class" act like a file //
|
|
// pointer, which can be read from or written to. //
|
|
// Handy for sending and receiving data :) //
|
|
//////////////////////////////////////////////////////
|
|
|
|
// Read from the modem
|
|
static size_t afsk_read(KFile *fd, void *_buf, size_t size) {
|
|
Afsk *afsk = AFSK_CAST(fd);
|
|
uint8_t *buffer = (uint8_t *)_buf;
|
|
|
|
#if CONFIG_AFSK_RXTIMEOUT == 0
|
|
while (size-- && !fifo_isempty_locked(&afsk->rxFifo))
|
|
#else
|
|
while (size--)
|
|
#endif
|
|
{
|
|
#if CONFIG_AFSK_RXTIMEOUT != -1
|
|
ticks_t start = timer_clock();
|
|
#endif
|
|
|
|
while (fifo_isempty_locked(&afsk->rxFifo)) {
|
|
cpu_relax();
|
|
#if CONFIG_AFSK_RXTIMEOUT != -1
|
|
if (timer_clock() - start > ms_to_ticks(CONFIG_AFSK_RXTIMEOUT)) {
|
|
return buffer - (uint8_t *)_buf;
|
|
}
|
|
#endif
|
|
}
|
|
*buffer++ = fifo_pop_locked(&afsk->rxFifo);
|
|
}
|
|
|
|
return buffer - (uint8_t *)_buf;
|
|
}
|
|
|
|
// Write to the modem
|
|
static size_t afsk_write(KFile *fd, const void *_buf, size_t size) {
|
|
Afsk *afsk = AFSK_CAST(fd);
|
|
const uint8_t *buf = (const uint8_t *)_buf;
|
|
|
|
while (size--) {
|
|
while (fifo_isfull_locked(&afsk->txFifo)) {
|
|
cpu_relax();
|
|
}
|
|
|
|
fifo_push_locked(&afsk->txFifo, *buf++);
|
|
afsk_txStart(afsk);
|
|
}
|
|
|
|
return buf - (const uint8_t *)_buf;
|
|
}
|
|
|
|
// Waits for the write operation to finish
|
|
static int afsk_flush(KFile *fd) {
|
|
Afsk *afsk = AFSK_CAST(fd);
|
|
while (afsk->sending) {
|
|
cpu_relax();
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
// Check whether there was any errors
|
|
// while reading or writing
|
|
static int afsk_error(KFile *fd) {
|
|
Afsk *afsk = AFSK_CAST(fd);
|
|
int err;
|
|
ATOMIC(err = afsk->status);
|
|
return err;
|
|
}
|
|
|
|
// Allows resetting the error-state
|
|
static void afsk_clearerr(KFile *fd) {
|
|
Afsk *afsk = AFSK_CAST(fd);
|
|
ATOMIC(afsk->status = 0);
|
|
}
|
|
|
|
|
|
//////////////////////////////////////////////////////
|
|
// Modem Initialization //
|
|
//////////////////////////////////////////////////////
|
|
|
|
void afsk_init(Afsk *afsk, int _adcPin) {
|
|
// Allocate memory for struct
|
|
memset(afsk, 0, sizeof(*afsk));
|
|
|
|
// Configure ADC pin
|
|
afsk->adcPin = _adcPin;
|
|
|
|
// Initialise phase increment to that
|
|
// of the mark frequency
|
|
afsk->phaseInc = MARK_INC;
|
|
|
|
// Initialize FIFO buffers
|
|
fifo_init(&afsk->delayFifo, (uint8_t *)afsk->delayBuf, sizeof(afsk->delayBuf));
|
|
fifo_init(&afsk->rxFifo, afsk->rxBuf, sizeof(afsk->rxBuf));
|
|
fifo_init(&afsk->txFifo, afsk->txBuf, sizeof(afsk->txBuf));
|
|
|
|
// Fill delay FIFO with zeroes
|
|
for (int i = 0; i<SAMPLESPERBIT / 2; i++) {
|
|
fifo_push(&afsk->delayFifo, 0);
|
|
}
|
|
|
|
// Initialize hardware
|
|
AFSK_ADC_INIT(_adcPin, afsk);
|
|
AFSK_DAC_INIT();
|
|
LED_TX_INIT();
|
|
LED_RX_INIT();
|
|
|
|
// And register the modem file-pointer
|
|
// functions for reading from and
|
|
// writing to it.
|
|
DB(afsk->fd._type = KFT_AFSK);
|
|
afsk->fd.write = afsk_write;
|
|
afsk->fd.read = afsk_read;
|
|
afsk->fd.flush = afsk_flush;
|
|
afsk->fd.error = afsk_error;
|
|
afsk->fd.clearerr = afsk_clearerr;
|
|
} |