/* * Copyright (C) 2014 Jared Boone, ShareBrained Technology, Inc. * Copyright (C) 2016 Furrtek * * 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. */ #include "proc_wfm_audio.hpp" #include "portapack_shared_memory.hpp" #include "audio_output.hpp" #include "dsp_fft.hpp" #include "event_m4.hpp" #include "audio_dma.hpp" #include void WidebandFMAudio::execute(const buffer_c8_t& buffer) { if (!configured) { return; } const auto decim_0_out = decim_0.execute(buffer, dst_buffer); const auto channel = decim_1.execute(decim_0_out, dst_buffer); // TODO: Feed channel_stats post-decimation data? feed_channel_stats(channel); spectrum_samples += channel.count; if (spectrum_samples >= spectrum_interval_samples) { spectrum_samples -= spectrum_interval_samples; channel_spectrum.feed(channel, channel_filter_low_f, channel_filter_high_f, channel_filter_transition); } /* 384kHz complex[256] * -> FM demodulation * -> 384kHz int16_t[256] */ /* TODO: To improve adjacent channel rejection, implement complex channel filter: * pass < +/- 100kHz, stop > +/- 200kHz */ auto audio_oversampled = demod.execute(channel, work_audio_buffer); /* 384kHz int16_t[256] * -> 4th order CIC decimation by 2, gain of 1 * -> 192kHz int16_t[128] */ auto audio_4fs = audio_dec_1.execute(audio_oversampled, work_audio_buffer); /* 192kHz int16_t[128] * -> 4th order CIC decimation by 2, gain of 1 * -> 96kHz int16_t[64] */ auto audio_2fs = audio_dec_2.execute(audio_4fs, work_audio_buffer); // Input: 96kHz int16_t[64] // audio_spectrum_decimator piles up 256 samples before doing FFT computation // This sends an AudioSpectrum every: sample rate/buffer size/refresh period = 3072000/2048/50 = 30 Hz // When audio_spectrum_timer expires, the audio spectrum computation is triggered // 0~3: feed continuous audio // 4~31: ignore, wrap at 31 audio_spectrum_timer++; if (audio_spectrum_timer == 50) { audio_spectrum_timer = 0; audio_spectrum_state = FEED; } switch (audio_spectrum_state) { case FEED: // Convert audio to "complex" just so the FFT can be done :/ for (size_t i = 0; i < 64; i++) { complex_audio[i] = {(int16_t)(work_audio_buffer.p[i] / 32), (int16_t)0}; } audio_spectrum_decimator.feed( complex_audio_buffer, [this](const buffer_c16_t& data) { this->post_message(data); }); break; case FFT: // Spread the FFT workload in time to avoid making the audio skip // "8" comes from the log2() of the size of audio_spectrum: log2(256) = 8 if (fft_step < 8) { fft_c_preswapped(audio_spectrum, fft_step, fft_step + 1); fft_step++; } else { const size_t spectrum_end = spectrum.db.size(); for (size_t i = 0; i < spectrum_end; i++) { // const auto corrected_sample = spectrum_window_hamming_3(audio_spectrum, i); const auto corrected_sample = audio_spectrum[i]; const auto mag2 = magnitude_squared(corrected_sample * (1.0f / 32768.0f)); const float db = mag2_to_dbv_norm(mag2); constexpr float mag_scale = 5.0f; const unsigned int v = (db * mag_scale) + 255.0f; spectrum.db[i] = std::max(0U, std::min(255U, v)); } AudioSpectrumMessage message{&spectrum}; shared_memory.application_queue.push(message); audio_spectrum_state = IDLE; } break; default: break; } /* 96kHz int16_t[64] * -> FIR filter, <15kHz (0.156fs) pass, >19kHz (0.198fs) stop, gain of 1 * -> 48kHz int16_t[32] */ auto audio = audio_filter.execute(audio_2fs, work_audio_buffer); /* -> 48kHz int16_t[32] */ audio_output.write(audio); } void WidebandFMAudio::post_message(const buffer_c16_t& data) { // This is called when audio_spectrum_decimator is filled up to 256 samples fft_swap(data, audio_spectrum); audio_spectrum_state = FFT; fft_step = 0; } void WidebandFMAudio::on_message(const Message* const message) { switch (message->id) { case Message::ID::UpdateSpectrum: case Message::ID::SpectrumStreamingConfig: channel_spectrum.on_message(message); break; case Message::ID::WFMConfigure: configure(*reinterpret_cast(message)); break; case Message::ID::CaptureConfig: capture_config(*reinterpret_cast(message)); break; default: break; } } void WidebandFMAudio::configure(const WFMConfigureMessage& message) { constexpr size_t decim_0_input_fs = baseband_fs; constexpr size_t decim_0_output_fs = decim_0_input_fs / decim_0.decimation_factor; constexpr size_t decim_1_input_fs = decim_0_output_fs; constexpr size_t decim_1_output_fs = decim_1_input_fs / decim_1.decimation_factor; constexpr size_t demod_input_fs = decim_1_output_fs; spectrum_interval_samples = decim_1_output_fs / spectrum_rate_hz; spectrum_samples = 0; decim_0.configure(message.decim_0_filter.taps); decim_1.configure(message.decim_1_filter.taps); channel_filter_low_f = message.decim_1_filter.low_frequency_normalized * decim_1_input_fs; channel_filter_high_f = message.decim_1_filter.high_frequency_normalized * decim_1_input_fs; channel_filter_transition = message.decim_1_filter.transition_normalized * decim_1_input_fs; demod.configure(demod_input_fs, message.deviation); audio_filter.configure(message.audio_filter.taps); audio_output.configure(message.audio_hpf_config, message.audio_deemph_config); channel_spectrum.set_decimation_factor(1); configured = true; } void WidebandFMAudio::capture_config(const CaptureConfigMessage& message) { if (message.config) { audio_output.set_stream(std::make_unique(message.config)); } else { audio_output.set_stream(nullptr); } } int main() { audio::dma::init_audio_out(); EventDispatcher event_dispatcher{std::make_unique()}; event_dispatcher.run(); return 0; }