portapack-mayhem/firmware/baseband/proc_pocsag2.hpp
Kyle Reed 4819a2f4e2
Decode status widget (#1431)
* Initial cleanup of pocsag beta, using DSP filters

* Better filter params

* Better filter

* Add signal diagnostics widgets

* POCSAG procs sends stats messages

* Only draw 32 bits

* Add AudioNormalizer filter
2023-09-03 21:49:44 -07:00

232 lines
6.7 KiB
C++

/*
* Copyright (C) 1996 Thomas Sailer (sailer@ife.ee.ethz.ch, hb9jnx@hb9w.che.eu)
* Copyright (C) 2012-2014 Elias Oenal (multimon-ng@eliasoenal.com)
* Copyright (C) 2015 Jared Boone, ShareBrained Technology, Inc.
* Copyright (C) 2016 Furrtek
* Copyright (C) 2023 Kyle Reed
*
* This file is part of PortaPack.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2, or (at your option)
* any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; see the file COPYING. If not, write to
* the Free Software Foundation, Inc., 51 Franklin Street,
* Boston, MA 02110-1301, USA.
*/
#ifndef __PROC_POCSAG2_H__
#define __PROC_POCSAG2_H__
#include "audio_output.hpp"
#include "baseband_processor.hpp"
#include "baseband_thread.hpp"
#include "dsp_decimate.hpp"
#include "dsp_demodulate.hpp"
#include "dsp_iir_config.hpp"
#include "message.hpp"
#include "pocsag.hpp"
#include "pocsag_packet.hpp"
#include "portapack_shared_memory.hpp"
#include "rssi_thread.hpp"
#include <cstdint>
/* Takes audio stream and automatically normalizes it to +/-1.0f */
class AudioNormalizer {
public:
void execute_in_place(const buffer_f32_t& audio) {
// Decay min/max every second (@24kHz).
if (counter_ >= 24'000) {
// 90% decay factor seems to work well.
// This keeps large transients from wrecking the filter.
max_ *= 0.9f;
min_ *= 0.9f;
counter_ = 0;
calculate_thresholds();
}
counter_ += audio.count;
for (size_t i = 0; i < audio.count; ++i) {
auto& val = audio.p[i];
if (val > max_) {
max_ = val;
calculate_thresholds();
}
if (val < min_) {
min_ = val;
calculate_thresholds();
}
if (val >= t_hi_)
val = 1.0f;
else if (val <= t_lo_)
val = -1.0f;
else
val = 0.0;
}
}
private:
void calculate_thresholds() {
auto center = (max_ + min_) / 2.0f;
auto range = (max_ - min_) / 2.0f;
// 10% off center force either +/-1.0f.
// Higher == larger dead zone.
// Lower == more false positives.
auto threshold = range * 0.1;
t_hi_ = center + threshold;
t_lo_ = center - threshold;
}
uint32_t counter_ = 0;
float min_ = 99.0f;
float max_ = -99.0f;
float t_hi_ = 1.0;
float t_lo_ = 1.0;
};
// How to detect clock signal across baud rates?
// Maybe have a bit extraction state machine that reset
// then watches for the clocks, but there are multiple
// clock and the last one is the right one.
// So keep updating clock until a sync?
class BitExtractor {};
class WordExtractor {};
class POCSAGProcessor : public BasebandProcessor {
public:
void execute(const buffer_c8_t& buffer) override;
void on_message(const Message* const message) override;
int OnDataFrame(int len, int baud);
int OnDataWord(uint32_t word, int pos);
private:
static constexpr size_t baseband_fs = 3072000;
static constexpr uint8_t stat_update_interval = 10;
static constexpr uint32_t stat_update_threshold =
baseband_fs / stat_update_interval;
void configure();
void send_stats() const;
// Set once app is ready to receive messages.
bool configured = false;
// Buffer for decimated IQ data.
std::array<complex16_t, 512> dst{};
const buffer_c16_t dst_buffer{dst.data(), dst.size()};
// Buffer for demodulated audio.
std::array<float, 32> audio{};
const buffer_f32_t audio_buffer{audio.data(), audio.size()};
// Decimate to 48kHz.
dsp::decimate::FIRC8xR16x24FS4Decim8 decim_0{};
dsp::decimate::FIRC16xR16x32Decim8 decim_1{};
// Filter to 24kHz and demodulate.
dsp::decimate::FIRAndDecimateComplex channel_filter{};
dsp::demodulate::FM demod{};
// LPF to reduce noise.
// scipy.signal.butter(2, 1800, "lowpass", fs=24000, analog=False)
IIRBiquadFilter lpf{{{0.04125354f, 0.082507070f, 0.04125354f},
{1.00000000f, -1.34896775f, 0.51398189f}}};
// Squelch to ignore noise.
FMSquelch squelch{};
uint64_t squelch_history = 0;
// Attempts to de-noise signal and normalize to +/- 1.0f.
AudioNormalizer normalizer{};
// Handles writing audio stream to hardware.
AudioOutput audio_output{};
// Holds the data sent to the app.
pocsag::POCSAGPacket packet{};
bool has_been_reset = true;
uint32_t samples_processed = 0;
//--------------------------------------------------
// ----------------------------------------
// Frame extractraction methods and members
// ----------------------------------------
void initFrameExtraction();
struct FIFOStruct {
unsigned long codeword;
int numBits;
};
void resetVals();
void setFrameExtractParams(long a_samplesPerSec, long a_maxBaud = 8000, long a_minBaud = 200, long maxRunOfSameValue = 32);
int processDemodulatedSamples(float* sampleBuff, int noOfSamples);
int extractFrames();
void storeBit();
short getBit();
int getNoOfBits();
uint32_t getRate();
uint32_t m_averageSymbolLen_1024{0};
uint32_t m_lastStableSymbolLen_1024{0};
uint32_t m_samplesPerSec{0};
uint32_t m_goodTransitions{0};
uint32_t m_badTransitions{0};
uint32_t m_sampleNo{0};
float m_sample{0};
float m_valMid{0.0f};
float m_lastSample{0.0f};
uint32_t m_lastTransPos_1024{0};
uint32_t m_lastSingleBitPos_1024{0};
uint32_t m_nextBitPosInt{0}; // Integer rounded up version to save on ops
uint32_t m_nextBitPos_1024{0};
uint32_t m_lastBitPos_1024{0};
uint32_t m_shortestGoodTrans_1024{0};
uint32_t m_minSymSamples_1024{0};
uint32_t m_maxSymSamples_1024{0};
uint32_t m_maxRunOfSameValue{0};
static constexpr long BIT_BUF_SIZE = 64;
std::bitset<64> m_bits{0};
long m_bitsStart{0};
long m_bitsEnd{0};
FIFOStruct m_fifo{0, 0};
bool m_gotSync{false};
int m_numCode{0};
bool m_inverted{false};
//--------------------------------------------------
/* NB: Threads should be the last members in the class definition. */
BasebandThread baseband_thread{baseband_fs, this, baseband::Direction::Receive};
RSSIThread rssi_thread{};
};
#endif /*__PROC_POCSAG2_H__*/