mirror of
https://github.com/eried/portapack-mayhem.git
synced 2024-10-01 01:26:06 -04:00
319 lines
10 KiB
C++
319 lines
10 KiB
C++
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/*
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* Copyright (C) 1996 Thomas Sailer (sailer@ife.ee.ethz.ch, hb9jnx@hb9w.che.eu)
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* Copyright (C) 2012-2014 Elias Oenal (multimon-ng@eliasoenal.com)
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* Copyright (C) 2015 Jared Boone, ShareBrained Technology, Inc.
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* Copyright (C) 2016 Furrtek
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* Copyright (C) 2023 Kyle Reed
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*
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* This file is part of PortaPack.
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2, or (at your option)
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* any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; see the file COPYING. If not, write to
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* the Free Software Foundation, Inc., 51 Franklin Street,
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* Boston, MA 02110-1301, USA.
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*/
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#include "proc_fsk_rx.hpp"
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#include "event_m4.hpp"
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#include <algorithm>
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#include <cmath>
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#include <cstdint>
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#include <cstddef>
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using namespace std;
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using namespace dsp::decimate;
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namespace {
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/* Count of bits that differ between the two values. */
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uint8_t diff_bit_count(uint32_t left, uint32_t right) {
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uint32_t diff = left ^ right;
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uint8_t count = 0;
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for (size_t i = 0; i < sizeof(diff) * 8; ++i) {
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if (((diff >> i) & 0x1) == 1)
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++count;
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}
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return count;
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}
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} // namespace
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/* AudioNormalizer ***************************************/
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void AudioNormalizer::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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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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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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}
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void AudioNormalizer::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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/* FSKRxProcessor ******************************************/
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void FSKRxProcessor::clear_data_bits() {
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data = 0;
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bit_count = 0;
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}
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void FSKRxProcessor::handle_sync(bool inverted) {
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clear_data_bits();
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has_sync_ = true;
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inverted = inverted;
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word_count = 0;
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}
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void FSKRxProcessor::process_bits(const buffer_c8_t& buffer) {
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// Process all of the bits in the bits queue.
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while (buffer.count > 0) {
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// Wait until data_ is full.
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if (bit_count < data_bit_count)
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continue;
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// Wait for the sync frame.
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if (!has_sync_) {
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if (diff_bit_count(data, sync_codeword) <= 2)
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handle_sync(/*inverted=*/false);
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else if (diff_bit_count(data, ~sync_codeword) <= 2)
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handle_sync(/*inverted=*/true);
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continue;
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}
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}
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}
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/* FSKRxProcessor ***************************************/
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FSKRxProcessor::FSKRxProcessor() {
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}
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void FSKRxProcessor::execute(const buffer_c8_t& buffer) {
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if (!configured) {
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return;
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}
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// Decimate by current decim 0 and decim 1.
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const auto decim_0_out = decim_0.execute(buffer, dst_buffer);
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const auto decim_1_out = decim_1.execute(decim_0_out, dst_buffer);
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feed_channel_stats(decim_1_out);
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spectrum_samples += decim_1_out.count;
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if (spectrum_samples >= spectrum_interval_samples) {
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spectrum_samples -= spectrum_interval_samples;
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channel_spectrum.feed(decim_1_out, channel_filter_low_f,
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channel_filter_high_f, channel_filter_transition);
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}
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// process_bits();
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// Update the status.
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samples_processed += buffer.count;
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if (samples_processed >= stat_update_threshold) {
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// send_packet(data);
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samples_processed -= stat_update_threshold;
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}
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}
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void FSKRxProcessor::on_message(const Message* const message) {
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switch (message->id) {
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case Message::ID::FSKRxConfigure:
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configure(*reinterpret_cast<const FSKRxConfigureMessage*>(message));
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break;
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case Message::ID::UpdateSpectrum:
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case Message::ID::SpectrumStreamingConfig:
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channel_spectrum.on_message(message);
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break;
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case Message::ID::SampleRateConfig:
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sample_rate_config(*reinterpret_cast<const SampleRateConfigMessage*>(message));
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break;
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case Message::ID::CaptureConfig:
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capture_config(*reinterpret_cast<const CaptureConfigMessage*>(message));
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break;
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default:
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break;
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}
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}
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void FSKRxProcessor::configure(const FSKRxConfigureMessage& message) {
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// Extract message variables.
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deviation = message.deviation;
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channel_decimation = message.channel_decimation;
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// channel_filter_taps = message.channel_filter;
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channel_spectrum.set_decimation_factor(1);
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}
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void FSKRxProcessor::capture_config(const CaptureConfigMessage& message) {
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if (message.config) {
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audio_output.set_stream(std::make_unique<StreamInput>(message.config));
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} else {
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audio_output.set_stream(nullptr);
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}
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}
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void FSKRxProcessor::sample_rate_config(const SampleRateConfigMessage& message) {
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const auto sample_rate = message.sample_rate;
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// The actual sample rate is the requested rate * the oversample rate.
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// See oversample.hpp for more details on oversampling.
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baseband_fs = sample_rate * toUType(message.oversample_rate);
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baseband_thread.set_sampling_rate(baseband_fs);
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// TODO: Do we need to use the taps that the decimators get configured with?
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channel_filter_low_f = taps_200k_decim_1.low_frequency_normalized * sample_rate;
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channel_filter_high_f = taps_200k_decim_1.high_frequency_normalized * sample_rate;
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channel_filter_transition = taps_200k_decim_1.transition_normalized * sample_rate;
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// Compute the scalar that corrects the oversample_rate to be x8 when computing
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// the spectrum update interval. The original implementation only supported x8.
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// TODO: Why is this needed here but not in proc_replay? There must be some other
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// assumption about x8 oversampling in some component that makes this necessary.
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const auto oversample_correction = toUType(message.oversample_rate) / 8.0;
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// The spectrum update interval controls how often the waterfall is fed new samples.
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spectrum_interval_samples = sample_rate / (spectrum_rate_hz * oversample_correction);
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spectrum_samples = 0;
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// For high sample rates, the M4 is busy collecting samples so the
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// waterfall runs slower. Reduce the update interval so it runs faster.
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// NB: Trade off: looks nicer, but more frequent updates == more CPU.
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if (sample_rate >= 1'500'000)
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spectrum_interval_samples /= (sample_rate / 750'000);
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switch (message.oversample_rate) {
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case OversampleRate::x4:
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// M4 can't handle 2 decimation passes for sample rates needing x4.
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decim_0.set<FIRC8xR16x24FS4Decim4>().configure(taps_200k_decim_0.taps);
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decim_1.set<NoopDecim>();
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break;
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case OversampleRate::x8:
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// M4 can't handle 2 decimation passes for sample rates <= 600k.
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if (message.sample_rate < 600'000) {
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decim_0.set<FIRC8xR16x24FS4Decim4>().configure(taps_200k_decim_0.taps);
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decim_1.set<FIRC16xR16x16Decim2>().configure(taps_200k_decim_1.taps);
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} else {
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// Using 180k taps to provide better filtering with a single pass.
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decim_0.set<FIRC8xR16x24FS4Decim8>().configure(taps_180k_wfm_decim_0.taps);
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decim_1.set<NoopDecim>();
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}
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break;
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case OversampleRate::x16:
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decim_0.set<FIRC8xR16x24FS4Decim8>().configure(taps_200k_decim_0.taps);
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decim_1.set<FIRC16xR16x16Decim2>().configure(taps_200k_decim_1.taps);
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break;
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case OversampleRate::x32:
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decim_0.set<FIRC8xR16x24FS4Decim4>().configure(taps_200k_decim_0.taps);
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decim_1.set<FIRC16xR16x32Decim8>().configure(taps_16k0_decim_1.taps);
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break;
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case OversampleRate::x64:
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decim_0.set<FIRC8xR16x24FS4Decim8>().configure(taps_200k_decim_0.taps);
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decim_1.set<FIRC16xR16x32Decim8>().configure(taps_16k0_decim_1.taps);
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break;
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default:
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chDbgPanic("Unhandled OversampleRate");
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break;
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}
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// Update demodulator based on new decimation. Todo: Confirm this works.
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size_t decim_0_input_fs = baseband_fs;
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size_t decim_0_output_fs = decim_0_input_fs / decim_0.decimation_factor();
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size_t decim_1_input_fs = decim_0_output_fs;
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size_t decim_1_output_fs = decim_1_input_fs / decim_1.decimation_factor();
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// size_t channel_filter_input_fs = decim_1_output_fs;
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// size_t channel_filter_output_fs = channel_filter_input_fs / channel_decimation;
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size_t demod_input_fs = decim_1_output_fs;
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send_packet((uint32_t)demod_input_fs);
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// Set ready to process data.
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configured = true;
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}
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void FSKRxProcessor::flush() {
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// word_extractor.flush();
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}
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void FSKRxProcessor::reset() {
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clear_data_bits();
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has_sync_ = false;
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inverted = false;
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word_count = 0;
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samples_processed = 0;
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}
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void FSKRxProcessor::send_packet(uint32_t data) {
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data_message.is_data = true;
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data_message.value = data;
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shared_memory.application_queue.push(data_message);
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}
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/* main **************************************************/
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int main() {
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EventDispatcher event_dispatcher{std::make_unique<FSKRxProcessor>()};
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event_dispatcher.run();
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return 0;
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}
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