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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
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Kyle Reed
GitHub
parent
2435ee780f
commit
4819a2f4e2
9 changed files with 305 additions and 230 deletions
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@ -26,107 +26,87 @@
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#ifndef __PROC_POCSAG2_H__
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#define __PROC_POCSAG2_H__
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#include "audio_output.hpp"
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#include "baseband_processor.hpp"
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#include "baseband_thread.hpp"
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#include "rssi_thread.hpp"
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#include "dsp_decimate.hpp"
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#include "dsp_demodulate.hpp"
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#include "pocsag_packet.hpp"
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#include "pocsag.hpp"
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#include "dsp_iir_config.hpp"
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#include "message.hpp"
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#include "audio_output.hpp"
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#include "pocsag.hpp"
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#include "pocsag_packet.hpp"
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#include "portapack_shared_memory.hpp"
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#include "rssi_thread.hpp"
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#include <cstdint>
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#include <bitset>
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using namespace std;
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// Class used to smooth demodulated waveform prior to decoding
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// -----------------------------------------------------------
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template <class ValType, class CalcType>
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class SmoothVals {
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protected:
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ValType* m_lastVals; // Previous N values
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int m_size; // The size N
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CalcType m_sumVal; // Running sum of lastVals
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int m_pos; // Current position in last vals ring buffer
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int m_count; //
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/* Takes audio stream and automatically normalizes it to +/-1.0f */
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class AudioNormalizer {
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public:
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SmoothVals()
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: m_lastVals(NULL), m_size(1), m_sumVal(0), m_pos(0), m_count(0) {
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m_lastVals = new ValType[m_size];
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}
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// --------------------------------------------------
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// --------------------------------------------------
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virtual ~SmoothVals() {
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delete[] m_lastVals;
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}
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SmoothVals(const SmoothVals<float, float>&) {
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}
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SmoothVals& operator=(const SmoothVals<float, float>&) {
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return *this;
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}
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// --------------------------------------------------
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// Set size of smoothing
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// --------------------------------------------------
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void SetSize(int size) {
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m_size = std::max(size, 1);
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m_pos = 0;
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delete[] m_lastVals;
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m_lastVals = new ValType[m_size];
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m_sumVal = 0;
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}
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// --------------------------------------------------
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// Get size of smoothing
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// --------------------------------------------------
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int Size() { return m_size; }
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// --------------------------------------------------
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// In place processing
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// --------------------------------------------------
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void Process(ValType* valBuff, int numVals) {
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ValType tmpVal;
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if (m_count > (1024 * 10)) {
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// Recalculate the sum value occasionaly, stops accumulated errors when using float
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m_count = 0;
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m_sumVal = 0;
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for (int i = 0; i < m_size; ++i) {
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m_sumVal += (CalcType)m_lastVals[i];
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}
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void execute_in_place(const buffer_f32_t& audio) {
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// Decay min/max every second (@24kHz).
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if (counter_ >= 24'000) {
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// 90% decay factor seems to work well.
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// This keeps large transients from wrecking the filter.
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max_ *= 0.9f;
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min_ *= 0.9f;
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counter_ = 0;
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calculate_thresholds();
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}
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// Use a rolling smoothed value while processing the buffer
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for (int buffPos = 0; buffPos < numVals; ++buffPos) {
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m_pos++;
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if (m_pos >= m_size) {
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m_pos = 0;
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counter_ += audio.count;
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for (size_t i = 0; i < audio.count; ++i) {
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auto& val = audio.p[i];
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if (val > max_) {
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max_ = val;
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calculate_thresholds();
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}
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if (val < min_) {
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min_ = val;
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calculate_thresholds();
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}
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m_sumVal -= (CalcType)m_lastVals[m_pos]; // Subtract the oldest value
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m_lastVals[m_pos] = valBuff[buffPos]; // Store the new value
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m_sumVal += (CalcType)m_lastVals[m_pos]; // Add on the new value
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tmpVal = (ValType)(m_sumVal / m_size); // Scale by number of values smoothed
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valBuff[buffPos] = tmpVal;
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if (val >= t_hi_)
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val = 1.0f;
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else if (val <= t_lo_)
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val = -1.0f;
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else
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val = 0.0;
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}
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m_count += numVals;
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}
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private:
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void calculate_thresholds() {
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auto center = (max_ + min_) / 2.0f;
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auto range = (max_ - min_) / 2.0f;
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// 10% off center force either +/-1.0f.
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// Higher == larger dead zone.
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// Lower == more false positives.
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auto threshold = range * 0.1;
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t_hi_ = center + threshold;
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t_lo_ = center - threshold;
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}
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uint32_t counter_ = 0;
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float min_ = 99.0f;
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float max_ = -99.0f;
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float t_hi_ = 1.0;
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float t_lo_ = 1.0;
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};
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// --------------------------------------------------
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// Class to process base band data to pocsag frames
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// --------------------------------------------------
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// How to detect clock signal across baud rates?
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// Maybe have a bit extraction state machine that reset
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// then watches for the clocks, but there are multiple
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// clock and the last one is the right one.
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// So keep updating clock until a sync?
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class BitExtractor {};
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class WordExtractor {};
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class POCSAGProcessor : public BasebandProcessor {
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public:
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void execute(const buffer_c8_t& buffer) override;
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@ -137,28 +117,54 @@ class POCSAGProcessor : public BasebandProcessor {
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private:
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static constexpr size_t baseband_fs = 3072000;
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std::array<complex16_t, 512> dst{};
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const buffer_c16_t dst_buffer{
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dst.data(),
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dst.size()};
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std::array<float, 32> audio{};
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const buffer_f32_t audio_buffer{
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audio.data(),
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audio.size()};
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dsp::decimate::FIRC8xR16x24FS4Decim8 decim_0{};
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dsp::decimate::FIRC16xR16x32Decim8 decim_1{};
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dsp::decimate::FIRAndDecimateComplex channel_filter{};
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dsp::demodulate::FM demod{};
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SmoothVals<float, float> smooth = {};
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AudioOutput audio_output{};
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bool configured = false;
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pocsag::POCSAGPacket packet{};
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static constexpr uint8_t stat_update_interval = 10;
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static constexpr uint32_t stat_update_threshold =
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baseband_fs / stat_update_interval;
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void configure();
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void send_stats() const;
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// Set once app is ready to receive messages.
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bool configured = false;
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// Buffer for decimated IQ data.
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std::array<complex16_t, 512> dst{};
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const buffer_c16_t dst_buffer{dst.data(), dst.size()};
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// Buffer for demodulated audio.
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std::array<float, 32> audio{};
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const buffer_f32_t audio_buffer{audio.data(), audio.size()};
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// Decimate to 48kHz.
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dsp::decimate::FIRC8xR16x24FS4Decim8 decim_0{};
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dsp::decimate::FIRC16xR16x32Decim8 decim_1{};
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// Filter to 24kHz and demodulate.
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dsp::decimate::FIRAndDecimateComplex channel_filter{};
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dsp::demodulate::FM demod{};
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// LPF to reduce noise.
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// scipy.signal.butter(2, 1800, "lowpass", fs=24000, analog=False)
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IIRBiquadFilter lpf{{{0.04125354f, 0.082507070f, 0.04125354f},
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{1.00000000f, -1.34896775f, 0.51398189f}}};
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// Squelch to ignore noise.
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FMSquelch squelch{};
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uint64_t squelch_history = 0;
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// Attempts to de-noise signal and normalize to +/- 1.0f.
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AudioNormalizer normalizer{};
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// Handles writing audio stream to hardware.
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AudioOutput audio_output{};
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// Holds the data sent to the app.
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pocsag::POCSAGPacket packet{};
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bool has_been_reset = true;
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uint32_t samples_processed = 0;
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//--------------------------------------------------
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// ----------------------------------------
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// Frame extractraction methods and members
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@ -169,8 +175,6 @@ class POCSAGProcessor : public BasebandProcessor {
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int numBits;
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};
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#define BIT_BUF_SIZE (64)
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void resetVals();
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void setFrameExtractParams(long a_samplesPerSec, long a_maxBaud = 8000, long a_minBaud = 200, long maxRunOfSameValue = 32);
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@ -207,7 +211,8 @@ class POCSAGProcessor : public BasebandProcessor {
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uint32_t m_maxSymSamples_1024{0};
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uint32_t m_maxRunOfSameValue{0};
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bitset<(size_t)BIT_BUF_SIZE> m_bits{0};
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static constexpr long BIT_BUF_SIZE = 64;
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std::bitset<64> m_bits{0};
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long m_bitsStart{0};
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long m_bitsEnd{0};
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@ -216,8 +221,7 @@ class POCSAGProcessor : public BasebandProcessor {
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int m_numCode{0};
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bool m_inverted{false};
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FMSquelch squelch_{};
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uint64_t squelch_history = 0;
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//--------------------------------------------------
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/* NB: Threads should be the last members in the class definition. */
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BasebandThread baseband_thread{baseband_fs, this, baseband::Direction::Receive};
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