mirror of
https://github.com/aicodix/modem.git
synced 2026-04-27 14:30:34 +00:00
366 lines
11 KiB
C++
366 lines
11 KiB
C++
/*
|
|
OFDM modem encoder
|
|
|
|
Copyright 2021 Ahmet Inan <inan@aicodix.de>
|
|
*/
|
|
|
|
#include <iostream>
|
|
#include <cassert>
|
|
#include <cmath>
|
|
#include "common.hh"
|
|
#include "xorshift.hh"
|
|
#include "complex.hh"
|
|
#include "utils.hh"
|
|
#include "bitman.hh"
|
|
#include "decibel.hh"
|
|
#include "fft.hh"
|
|
#include "wav.hh"
|
|
#include "pcm.hh"
|
|
#include "mls.hh"
|
|
#include "psk.hh"
|
|
#include "qam.hh"
|
|
#include "polar_encoder.hh"
|
|
|
|
template <typename value, typename cmplx, int rate>
|
|
struct Encoder : public Common
|
|
{
|
|
typedef int8_t code_type;
|
|
static const int guard_len = rate / 100;
|
|
static const int symbol_len = guard_len * 16;
|
|
DSP::WritePCM<value> *pcm;
|
|
DSP::FastFourierTransform<symbol_len, cmplx, -1> fwd;
|
|
DSP::FastFourierTransform<symbol_len, cmplx, 1> bwd;
|
|
CODE::PolarEncoder<code_type> polar_encoder;
|
|
code_type code[bits_max], perm[bits_max], mesg[bits_max];
|
|
cmplx fdom[symbol_len];
|
|
cmplx tdom[symbol_len];
|
|
cmplx kern[symbol_len];
|
|
cmplx guard[guard_len];
|
|
cmplx tone[tone_count];
|
|
cmplx prev[tone_count];
|
|
value weight[guard_len];
|
|
value papr_min, papr_max;
|
|
|
|
static int bin(int carrier)
|
|
{
|
|
return (carrier + symbol_len) % symbol_len;
|
|
}
|
|
static int nrz(bool bit)
|
|
{
|
|
return 1 - 2 * bit;
|
|
}
|
|
void clipping_and_filtering(value scale, bool limit)
|
|
{
|
|
for (int i = 0; i < symbol_len; ++i) {
|
|
value pwr = norm(tdom[i]);
|
|
if (pwr > value(1))
|
|
tdom[i] /= sqrt(pwr);
|
|
}
|
|
fwd(fdom, tdom);
|
|
for (int i = 0; i < symbol_len; ++i) {
|
|
int j = bin(i + tone_off);
|
|
if (i >= tone_count) {
|
|
fdom[j] = 0;
|
|
} else if (i % block_length == pilot_off) {
|
|
fdom[j] = tone[i];
|
|
} else if (i % block_length == reserved_off) {
|
|
fdom[j] = 0;
|
|
} else {
|
|
fdom[j] *= 1 / (scale * symbol_len);
|
|
cmplx err = fdom[j] - tone[i];
|
|
value mag = abs(err);
|
|
value lim = 0.5 * mod_distance();
|
|
if (limit && mag > lim)
|
|
fdom[j] -= ((mag - lim) / mag) * err;
|
|
}
|
|
}
|
|
bwd(tdom, fdom);
|
|
for (int i = 0; i < symbol_len; ++i)
|
|
tdom[i] *= scale;
|
|
}
|
|
void tone_reservation()
|
|
{
|
|
for (int n = 0; n < 10; ++n) {
|
|
int peak = 0;
|
|
for (int i = 1; i < symbol_len; ++i)
|
|
if (norm(tdom[peak]) < norm(tdom[i]))
|
|
peak = i;
|
|
cmplx orig = tdom[peak];
|
|
if (norm(orig) <= value(1))
|
|
break;
|
|
for (int i = 0; i < symbol_len; ++i)
|
|
tdom[i] -= orig * kern[bin(i-peak)];
|
|
}
|
|
}
|
|
void symbol(bool papr_reduction = true, bool guard_interval = true)
|
|
{
|
|
for (int i = 0; differential && i < tone_count; ++i)
|
|
if (!papr_reduction)
|
|
prev[i] = 1;
|
|
else if (i % block_length != reserved_off)
|
|
prev[i] = tone[i] *= prev[i];
|
|
for (int i = 0; i < symbol_len; ++i)
|
|
fdom[i] = 0;
|
|
for (int i = 0; i < tone_count; ++i)
|
|
fdom[bin(i+tone_off)] = tone[i];
|
|
bwd(tdom, fdom);
|
|
value scale = value(0.5) / std::sqrt(value(tone_count));
|
|
for (int i = 0; i < symbol_len; ++i)
|
|
tdom[i] *= scale;
|
|
if (papr_reduction) {
|
|
clipping_and_filtering(scale, true);
|
|
tone_reservation();
|
|
}
|
|
auto clamp = [](value v){ return v < value(-1) ? value(-1) : v > value(1) ? value(1) : v; };
|
|
for (int i = 0; i < symbol_len; ++i)
|
|
tdom[i] = cmplx(clamp(tdom[i].real()), clamp(tdom[i].imag()));
|
|
for (int i = 0; i < guard_len; ++i)
|
|
guard[i] = DSP::lerp(guard[i], tdom[i+symbol_len-guard_len], weight[i]);
|
|
if (papr_reduction) {
|
|
value peak = 0, mean = 0;
|
|
for (int i = 0; i < symbol_len; ++i) {
|
|
value power(norm(tdom[i]));
|
|
peak = std::max(peak, power);
|
|
mean += power;
|
|
}
|
|
mean /= symbol_len;
|
|
if (mean > 0) {
|
|
value papr(peak / mean);
|
|
papr_min = std::min(papr_min, papr);
|
|
papr_max = std::max(papr_max, papr);
|
|
}
|
|
}
|
|
if (guard_interval)
|
|
pcm->write(reinterpret_cast<value *>(guard), guard_len, 2);
|
|
pcm->write(reinterpret_cast<value *>(tdom), symbol_len, 2);
|
|
for (int i = 0; i < guard_len; ++i)
|
|
guard[i] = tdom[i];
|
|
}
|
|
void leading_noise(int num = 1)
|
|
{
|
|
CODE::MLS noise(0x163);
|
|
for (int j = 0; j < num; ++j) {
|
|
for (int i = 0; i < tone_count; ++i)
|
|
tone[i] = nrz(noise());
|
|
symbol(false);
|
|
}
|
|
}
|
|
void schmidl_cox()
|
|
{
|
|
CODE::MLS seq0(mls0_poly, mls0_seed);
|
|
for (int i = 0; i < tone_count; ++i)
|
|
tone[i] = nrz(seq0());
|
|
symbol(false);
|
|
symbol(false, false);
|
|
}
|
|
cmplx map_bits(code_type *b, int bits)
|
|
{
|
|
switch (bits) {
|
|
case 1:
|
|
return PhaseShiftKeying<2, cmplx, code_type>::map(b);
|
|
case 2:
|
|
return PhaseShiftKeying<4, cmplx, code_type>::map(b);
|
|
case 3:
|
|
return PhaseShiftKeying<8, cmplx, code_type>::map(b);
|
|
case 4:
|
|
return QuadratureAmplitudeModulation<16, cmplx, code_type>::map(b);
|
|
case 6:
|
|
return QuadratureAmplitudeModulation<64, cmplx, code_type>::map(b);
|
|
case 8:
|
|
return QuadratureAmplitudeModulation<256, cmplx, code_type>::map(b);
|
|
}
|
|
return 0;
|
|
}
|
|
value mod_distance()
|
|
{
|
|
switch (mod_bits) {
|
|
case 1:
|
|
return PhaseShiftKeying<2, cmplx, code_type>::DIST;
|
|
case 2:
|
|
return PhaseShiftKeying<4, cmplx, code_type>::DIST;
|
|
case 3:
|
|
return PhaseShiftKeying<8, cmplx, code_type>::DIST;
|
|
case 4:
|
|
return QuadratureAmplitudeModulation<16, cmplx, code_type>::DIST;
|
|
case 6:
|
|
return QuadratureAmplitudeModulation<64, cmplx, code_type>::DIST;
|
|
case 8:
|
|
return QuadratureAmplitudeModulation<256, cmplx, code_type>::DIST;
|
|
}
|
|
return 2;
|
|
}
|
|
void shuffle(code_type *dest, const code_type *src)
|
|
{
|
|
if (code_order == 11) {
|
|
CODE::XorShiftMask<int, 11, 1, 3, 4, 1> seq;
|
|
dest[0] = src[0];
|
|
for (int i = 1; i < 2048; ++i)
|
|
dest[i] = src[seq()];
|
|
} else if (code_order == 12) {
|
|
CODE::XorShiftMask<int, 12, 1, 1, 4, 1> seq;
|
|
dest[0] = src[0];
|
|
for (int i = 1; i < 4096; ++i)
|
|
dest[i] = src[seq()];
|
|
} else if (code_order == 13) {
|
|
CODE::XorShiftMask<int, 13, 1, 1, 9, 1> seq;
|
|
dest[0] = src[0];
|
|
for (int i = 1; i < 8192; ++i)
|
|
dest[i] = src[seq()];
|
|
} else if (code_order == 14) {
|
|
CODE::XorShiftMask<int, 14, 1, 5, 10, 1> seq;
|
|
dest[0] = src[0];
|
|
for (int i = 1; i < 16384; ++i)
|
|
dest[i] = src[seq()];
|
|
} else if (code_order == 15) {
|
|
CODE::XorShiftMask<int, 15, 1, 1, 3, 1> seq;
|
|
dest[0] = src[0];
|
|
for (int i = 1; i < 32768; ++i)
|
|
dest[i] = src[seq()];
|
|
} else if (code_order == 16) {
|
|
CODE::XorShiftMask<int, 16, 1, 1, 14, 1> seq;
|
|
dest[0] = src[0];
|
|
for (int i = 1; i < 65536; ++i)
|
|
dest[i] = src[seq()];
|
|
}
|
|
}
|
|
void tone_reservation_kernel()
|
|
{
|
|
value mag(0.001);
|
|
for (int i = 0; i < symbol_len; ++i)
|
|
fdom[i] = 0;
|
|
for (int i = 0; i < reserved_tones; ++i)
|
|
fdom[bin(i*block_length+tone_off+reserved_off)] = mag;
|
|
bwd(kern, fdom);
|
|
}
|
|
void guard_interval_weights()
|
|
{
|
|
for (int i = 0; i < guard_len / 4; ++i)
|
|
weight[i] = 0;
|
|
for (int i = guard_len / 4; i < guard_len / 4 + guard_len / 2; ++i) {
|
|
value x = value(i - guard_len / 4) / value(guard_len / 2 - 1);
|
|
weight[i] = value(0.5) * (value(1) - std::cos(DSP::Const<value>::Pi() * x));
|
|
}
|
|
for (int i = guard_len / 4 + guard_len / 2; i < guard_len; ++i)
|
|
weight[i] = 1;
|
|
}
|
|
Encoder(DSP::WritePCM<value> *pcm, const char *const *input_names, int input_count, int freq_off, int oper_mode) : pcm(pcm)
|
|
{
|
|
setup(oper_mode);
|
|
int offset = (freq_off * symbol_len) / rate;
|
|
tone_off = offset - tone_count / 2;
|
|
guard_interval_weights();
|
|
papr_min = 1000, papr_max = -1000;
|
|
leading_noise();
|
|
hadamard_encoder(mode, oper_mode);
|
|
for (int input_index = 0; input_index < input_count; ++input_index) {
|
|
const char *input_name = input_names[input_index];
|
|
if (input_count == 1 && input_name[0] == '-' && input_name[1] == 0)
|
|
input_name = "/dev/stdin";
|
|
std::ifstream input_file(input_name, std::ios::binary);
|
|
if (input_file.bad()) {
|
|
std::cerr << "Couldn't open file \"" << input_name << "\" for reading." << std::endl;
|
|
continue;
|
|
}
|
|
for (int i = 0; i < data_bytes; ++i)
|
|
data[i] = std::max(input_file.get(), 0);
|
|
CODE::Xorshift32 scrambler;
|
|
for (int i = 0; i < data_bytes; ++i)
|
|
data[i] ^= scrambler();
|
|
schmidl_cox();
|
|
for (int i = 0; i < data_bits; ++i)
|
|
mesg[i] = nrz(CODE::get_le_bit(data, i));
|
|
crc0.reset();
|
|
for (int i = 0; i < data_bytes; ++i)
|
|
crc0(data[i]);
|
|
for (int i = 0; i < 32; ++i)
|
|
mesg[i+data_bits] = nrz((crc0()>>i)&1);
|
|
polar_encoder(code, mesg, frozen_bits, code_order);
|
|
shuffle(perm, code);
|
|
CODE::MLS seq1(mls1_poly);
|
|
for (int j = 0, k = 0, m = 0; j < symbol_count; ++j) {
|
|
pilot_off = (block_skew * j + first_pilot) % block_length;
|
|
reserved_off = (block_skew * j + first_reserved) % block_length;
|
|
for (int i = 0; i < tone_count; ++i) {
|
|
if (i % block_length == pilot_off) {
|
|
tone[i] = nrz(seq1());
|
|
if (j == 0)
|
|
tone[i] *= mode[m++];
|
|
} else if (i % block_length == reserved_off) {
|
|
tone[i] = 0;
|
|
} else {
|
|
int bits = mod_bits;
|
|
if (oper_mode >= 7 && oper_mode <= 9 && k % 32 == 30)
|
|
bits = 2;
|
|
if (oper_mode >= 21 && oper_mode <= 23 && k % 64 == 60)
|
|
bits = 4;
|
|
tone[i] = map_bits(perm+k, bits);
|
|
k += bits;
|
|
}
|
|
}
|
|
tone_reservation_kernel();
|
|
symbol();
|
|
}
|
|
}
|
|
for (int i = 0; i < tone_count; ++i)
|
|
tone[i] = 0;
|
|
symbol(false);
|
|
std::cerr << "PAPR: " << DSP::decibel(papr_min) << " .. " << DSP::decibel(papr_max) << " dB" << std::endl;
|
|
}
|
|
};
|
|
|
|
int main(int argc, char **argv)
|
|
{
|
|
if (argc < 8) {
|
|
std::cerr << "usage: " << argv[0] << " OUTPUT RATE BITS CHANNELS OFFSET MODE INPUT.." << std::endl;
|
|
return 1;
|
|
}
|
|
|
|
const char *output_name = argv[1];
|
|
if (output_name[0] == '-' && output_name[1] == 0)
|
|
output_name = "/dev/stdout";
|
|
int output_rate = std::atoi(argv[2]);
|
|
int output_bits = std::atoi(argv[3]);
|
|
int output_chan = std::atoi(argv[4]);
|
|
|
|
int freq_off = std::atoi(argv[5]);
|
|
if (freq_off % 100) {
|
|
std::cerr << "Frequency offset must be divisible by 100." << std::endl;
|
|
return 1;
|
|
}
|
|
int input_count = argc - 7;
|
|
int oper_mode = std::atoi(argv[6]);
|
|
if (oper_mode < 0 || oper_mode > 27) {
|
|
std::cerr << "Unsupported operation mode." << std::endl;
|
|
return 1;
|
|
}
|
|
int band_width = 2000;
|
|
if ((output_chan == 1 && freq_off < band_width / 2) || freq_off < band_width / 2 - output_rate / 2 || freq_off > output_rate / 2 - band_width / 2) {
|
|
std::cerr << "Unsupported frequency offset." << std::endl;
|
|
return 1;
|
|
}
|
|
typedef float value;
|
|
typedef DSP::Complex<value> cmplx;
|
|
DSP::WriteWAV<value> output_file(output_name, output_rate, output_bits, output_chan);
|
|
output_file.silence(output_rate);
|
|
switch (output_rate) {
|
|
case 8000:
|
|
delete new Encoder<value, cmplx, 8000>(&output_file, argv+7, input_count, freq_off, oper_mode);
|
|
break;
|
|
case 16000:
|
|
delete new Encoder<value, cmplx, 16000>(&output_file, argv+7, input_count, freq_off, oper_mode);
|
|
break;
|
|
case 44100:
|
|
delete new Encoder<value, cmplx, 44100>(&output_file, argv+7, input_count, freq_off, oper_mode);
|
|
break;
|
|
case 48000:
|
|
delete new Encoder<value, cmplx, 48000>(&output_file, argv+7, input_count, freq_off, oper_mode);
|
|
break;
|
|
default:
|
|
std::cerr << "Unsupported sample rate." << std::endl;
|
|
return 1;
|
|
}
|
|
output_file.silence(output_rate);
|
|
|
|
return 0;
|
|
}
|
|
|