Mirror/aicodix___modem
1
0
Fork 0
mirror of https://github.com/aicodix/modem.git synced 2026-04-27 14:30:34 +00:00
Code Issues Projects Releases Packages Wiki Activity
aicodix___modem/encode.cc
Ahmet Inan 20fd39c280 added option for longer leading noise
2025-07-01 23:47:14 +02:00

498 lines
13 KiB
C++
Raw Blame History

/*
OFDM modem encoder
Copyright 2021 Ahmet Inan <inan@aicodix.de>
*/
#include <iostream>
#include <cassert>
#include <cmath>
#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 "crc.hh"
#include "psk.hh"
#include "qam.hh"
#include "polar_tables.hh"
#include "polar_encoder.hh"
#include "hadamard_encoder.hh"
template <typename value, typename cmplx, int rate>
struct Encoder
{
typedef int8_t code_type;
static const int guard_len = rate / 100;
static const int symbol_len = guard_len * 16;
static const int bits_max = 65536;
static const int data_max = 4096;
static const int mls0_poly = 0x331;
static const int mls0_seed = 214;
static const int mls1_poly = 0x25;
static const int data_tones = 256;
static const int pilot_tones = 32;
static const int reserved_tones = 32;
static const int tone_count = data_tones + pilot_tones + reserved_tones;
static const int block_length = 10;
static const int block_skew = 3;
static const int pilot_offset = 4;
static const int reserved_offset = 9;
DSP::WritePCM<value> *pcm;
DSP::FastFourierTransform<symbol_len, cmplx, -1> fwd;
DSP::FastFourierTransform<symbol_len, cmplx, 1> bwd;
CODE::CRC<uint32_t> crc0;
CODE::HadamardEncoder<6> hadamardenc;
CODE::PolarEncoder<code_type> polarenc;
uint8_t input_data[data_max];
code_type code[bits_max], perm[bits_max], mesg[bits_max];
int8_t mode[32];
cmplx fdom[symbol_len];
cmplx tdom[symbol_len];
cmplx temp[symbol_len];
cmplx kern[symbol_len];
cmplx guard[guard_len];
cmplx tone[tone_count];
value weight[guard_len];
value papr_min, papr_max;
const uint32_t *frozen_bits;
int mod_bits;
int data_bits;
int data_bytes;
int code_order;
int tone_off;
int symbol_count;
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(temp, tdom);
for (int i = 0; i < symbol_len; ++i) {
if (norm(fdom[i])) {
temp[i] *= 1 / (scale * symbol_len);
cmplx err = temp[i] - fdom[i];
value mag = abs(err);
value lim = 0.1 * mod_distance();
if (limit && mag > lim)
temp[i] -= ((mag - lim) / mag) * err;
} else {
temp[i] = 0;
}
}
bwd(tdom, temp);
for (int i = 0; i < symbol_len; ++i)
tdom[i] *= scale;
}
void tone_reservation()
{
for (int n = 0; n < 100; ++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; 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;
clipping_and_filtering(scale, papr_reduction);
if (papr_reduction)
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]);
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 2:
return PhaseShiftKeying<4, 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 2:
return PhaseShiftKeying<4, 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 setup(int oper_mode, int freq_off)
{
switch (oper_mode) {
case 0:
symbol_count = 1;
break;
case 1:
mod_bits = 2;
symbol_count = 4;
code_order = 11;
data_bits = 1024;
frozen_bits = frozen_2048_1056;
break;
case 2:
mod_bits = 2;
symbol_count = 8;
code_order = 12;
data_bits = 2048;
frozen_bits = frozen_4096_2080;
break;
case 3:
mod_bits = 2;
symbol_count = 16;
code_order = 13;
data_bits = 4096;
frozen_bits = frozen_8192_4128;
break;
case 4:
mod_bits = 2;
symbol_count = 32;
code_order = 14;
data_bits = 8192;
frozen_bits = frozen_16384_8224;
break;
case 5:
mod_bits = 4;
symbol_count = 4;
code_order = 12;
data_bits = 2048;
frozen_bits = frozen_4096_2080;
break;
case 6:
mod_bits = 4;
symbol_count = 8;
code_order = 13;
data_bits = 4096;
frozen_bits = frozen_8192_4128;
break;
case 7:
mod_bits = 4;
symbol_count = 16;
code_order = 14;
data_bits = 8192;
frozen_bits = frozen_16384_8224;
break;
case 8:
mod_bits = 4;
symbol_count = 32;
code_order = 15;
data_bits = 16384;
frozen_bits = frozen_32768_16416;
break;
case 9:
mod_bits = 6;
symbol_count = 11;
code_order = 14;
data_bits = 8192;
frozen_bits = frozen_16384_8224;
break;
case 10:
mod_bits = 6;
symbol_count = 22;
code_order = 15;
data_bits = 16384;
frozen_bits = frozen_32768_16416;
break;
case 11:
mod_bits = 6;
symbol_count = 44;
code_order = 16;
data_bits = 32768;
frozen_bits = frozen_65536_32800;
break;
case 12:
mod_bits = 8;
symbol_count = 4;
code_order = 13;
data_bits = 4096;
frozen_bits = frozen_8192_4128;
break;
case 13:
mod_bits = 8;
symbol_count = 8;
code_order = 14;
data_bits = 8192;
frozen_bits = frozen_16384_8224;
break;
case 14:
mod_bits = 8;
symbol_count = 16;
code_order = 15;
data_bits = 16384;
frozen_bits = frozen_32768_16416;
break;
case 15:
mod_bits = 8;
symbol_count = 32;
code_order = 16;
data_bits = 32768;
frozen_bits = frozen_65536_32800;
break;
default:
return;
}
data_bytes = data_bits / 8;
int offset = (freq_off * symbol_len) / rate;
tone_off = offset - tone_count / 2;
}
void tone_reservation_kernel(int roff)
{
value mag = 1 / value(10 * reserved_tones);
for (int i = 0; i < symbol_len; ++i)
temp[i] = 0;
for (int i = 0; i < reserved_tones; ++i)
temp[bin(i*block_length+tone_off+roff)] = mag;
bwd(kern, temp);
}
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), crc0(0x8F6E37A0)
{
setup(oper_mode, freq_off);
guard_interval_weights();
papr_min = 1000, papr_max = -1000;
leading_noise();
hadamardenc(mode, oper_mode);
if (!oper_mode) {
schmidl_cox();
CODE::MLS seq1(mls1_poly);
for (int i = 0, m = 0; i < tone_count; ++i) {
if (i % block_length == pilot_offset) {
tone[i] = nrz(seq1()) * mode[m++];
} else {
tone[i] = 0;
}
}
symbol(false);
}
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)
input_data[i] = std::max(input_file.get(), 0);
CODE::Xorshift32 scrambler;
for (int i = 0; i < data_bytes; ++i)
input_data[i] ^= scrambler();
schmidl_cox();
for (int i = 0; i < data_bits; ++i)
mesg[i] = nrz(CODE::get_le_bit(input_data, i));
crc0.reset();
for (int i = 0; i < data_bytes; ++i)
crc0(input_data[i]);
for (int i = 0; i < 32; ++i)
mesg[i+data_bits] = nrz((crc0()>>i)&1);
polarenc(code, mesg, frozen_bits, code_order);
shuffle(perm, code);
CODE::MLS seq1(mls1_poly);
for (int j = 0, k = 0; j < symbol_count; ++j) {
int poff = (block_skew * j + pilot_offset) % block_length;
int roff = (block_skew * j + reserved_offset) % block_length;
for (int i = 0, m = 0; i < tone_count; ++i) {
if (i % block_length == poff) {
tone[i] = nrz(seq1()) * mode[m++];
} else if (i % block_length == roff) {
tone[i] = 0;
} else {
int bits = mod_bits;
if (oper_mode >= 9 && oper_mode <= 11 && k % 64 == 60)
bits = 4;
tone[i] = map_bits(perm+k, bits);
k += bits;
}
}
tone_reservation_kernel(roff);
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 < 7) {
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 != !input_count) {
std::cerr << "Using operation mode " << oper_mode << " but " << input_count << " input file" << (input_count == 1 ? "" : "s") << " provided." << std::endl;
return 1;
}
if (oper_mode < 0 || oper_mode > 15) {
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;
}
Reference in a new issue View git blame Copy permalink