kopia lustrzana https://github.com/f4exb/sdrangel
382 wiersze
14 KiB
C++
382 wiersze
14 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2019-2020, 2022-2023 Edouard Griffiths, F4EXB <f4exb06@gmail.com> //
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// Copyright (C) 2022 Jon Beniston, M7RCE <jon@beniston.com> //
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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 as version 3 of the License, or //
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// (at your option) 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 V3 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. If not, see <http://www.gnu.org/licenses/>. //
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///////////////////////////////////////////////////////////////////////////////////
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#include <stdio.h>
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#include <complex.h>
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#include <QTime>
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#include <QDebug>
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#include "dsp/basebandsamplesink.h"
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#include "dsp/datafifo.h"
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#include "pipes/datapipes.h"
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#include "maincore.h"
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#include "rdsparser.h"
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#include "bfmdemodsink.h"
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const Real BFMDemodSink::default_deemphasis = 50.0; // 50 us
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const int BFMDemodSink::default_excursion = 750000; // +/- 75 kHz
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BFMDemodSink::BFMDemodSink() :
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m_channel(nullptr),
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m_channelSampleRate(48000),
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m_channelFrequencyOffset(0),
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m_audioSampleRate(48000),
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m_audioBufferFill(0),
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m_audioFifo(48000),
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m_pilotPLL(19000/384000, 50/384000, 0.01),
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m_deemphasisFilterX(default_deemphasis * 48000 * 1.0e-6),
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m_deemphasisFilterY(default_deemphasis * 48000 * 1.0e-6),
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m_fmExcursion(default_excursion)
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{
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m_magsq = 0.0f;
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m_magsqSum = 0.0f;
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m_magsqPeak = 0.0f;
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m_magsqCount = 0;
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m_squelchLevel = 0;
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m_squelchState = 0;
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m_interpolatorDistance = 0.0f;
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m_interpolatorDistanceRemain = 0.0f;
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m_interpolatorRDSDistance = 0.0f;
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m_interpolatorRDSDistanceRemain = 0.0f;
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m_interpolatorStereoDistance = 0.0f;
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m_interpolatorStereoDistanceRemain = 0.0f;
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m_spectrumSink = nullptr;
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m_m1Arg = 0;
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m_rfFilter = new fftfilt(-50000.0 / 384000.0, 50000.0 / 384000.0, filtFftLen);
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m_deemphasisFilterX.configure(default_deemphasis * m_audioSampleRate * 1.0e-6);
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m_deemphasisFilterY.configure(default_deemphasis * m_audioSampleRate * 1.0e-6);
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m_phaseDiscri.setFMScaling(384000/m_fmExcursion);
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m_audioBuffer.resize(1<<14);
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m_audioBufferFill = 0;
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m_demodBuffer.resize(1<<13);
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m_demodBufferFill = 0;
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applySettings(m_settings, true);
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applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
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}
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BFMDemodSink::~BFMDemodSink()
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{
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delete m_rfFilter;
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}
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void BFMDemodSink::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end)
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{
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Complex ci, cs, cr;
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fftfilt::cmplx *rf;
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int rf_out;
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double msq;
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Real demod;
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m_sampleBuffer.clear();
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for (SampleVector::const_iterator it = begin; it != end; ++it)
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{
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Complex c(it->real() / SDR_RX_SCALEF, it->imag() / SDR_RX_SCALEF);
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c *= m_nco.nextIQ();
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rf_out = m_rfFilter->runFilt(c, &rf); // filter RF before demod
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for (int i =0 ; i <rf_out; i++)
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{
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msq = rf[i].real()*rf[i].real() + rf[i].imag()*rf[i].imag();
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m_magsqSum += msq;
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if (msq > m_magsqPeak) {
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m_magsqPeak = msq;
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}
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m_magsqCount++;
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if (msq >= m_squelchLevel)
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{
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if (m_squelchState < m_settings.m_rfBandwidth / 10) { // twice attack and decay rate
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m_squelchState++;
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}
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}
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else
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{
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if (m_squelchState > 0) {
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m_squelchState--;
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}
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}
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if (m_squelchState > m_settings.m_rfBandwidth / 20) { // squelch open
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demod = m_phaseDiscri.phaseDiscriminator(rf[i]);
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} else {
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demod = 0;
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}
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if (!m_settings.m_showPilot) {
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m_sampleBuffer.push_back(Sample(demod * SDR_RX_SCALEF, 0.0));
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}
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if (m_settings.m_rdsActive)
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{
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//Complex r(demod * 2.0 * std::cos(3.0 * m_pilotPLLSamples[3]), 0.0);
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Complex r(demod * 2.0 * std::cos(3.0 * m_pilotPLLSamples[3]), 0.0);
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if (m_interpolatorRDS.decimate(&m_interpolatorRDSDistanceRemain, r, &cr))
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{
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bool bit;
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if (m_rdsDemod.process(cr.real(), bit))
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{
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if (m_rdsDecoder.frameSync(bit)) {
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m_rdsParser.parseGroup(m_rdsDecoder.getGroup());
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}
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}
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m_interpolatorRDSDistanceRemain += m_interpolatorRDSDistance;
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}
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}
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Real sampleStereo = 0.0f;
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// Process stereo if stereo mode is selected
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if (m_settings.m_audioStereo)
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{
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m_pilotPLL.process(demod, m_pilotPLLSamples);
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if (m_settings.m_showPilot) {
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m_sampleBuffer.push_back(Sample(m_pilotPLLSamples[1] * SDR_RX_SCALEF, 0.0)); // debug 38 kHz pilot
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}
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if (m_settings.m_lsbStereo)
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{
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// 1.17 * 0.7 = 0.819
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Complex s(demod * m_pilotPLLSamples[1], demod * m_pilotPLLSamples[2]);
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if (m_interpolatorStereo.decimate(&m_interpolatorStereoDistanceRemain, s, &cs))
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{
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sampleStereo = cs.real() + cs.imag();
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m_interpolatorStereoDistanceRemain += m_interpolatorStereoDistance;
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}
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}
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else
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{
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Complex s(demod * 1.17 * m_pilotPLLSamples[1], 0);
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if (m_interpolatorStereo.decimate(&m_interpolatorStereoDistanceRemain, s, &cs))
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{
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sampleStereo = cs.real();
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m_interpolatorStereoDistanceRemain += m_interpolatorStereoDistance;
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}
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}
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}
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Complex e(demod, 0);
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if (m_interpolator.decimate(&m_interpolatorDistanceRemain, e, &ci))
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{
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if (m_settings.m_audioStereo)
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{
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Real deemph_l, deemph_r; // Pre-emphasis is applied on each channel before multiplexing
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m_deemphasisFilterX.process(ci.real() + sampleStereo, deemph_l);
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m_deemphasisFilterY.process(ci.real() - sampleStereo, deemph_r);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(deemph_l * (1<<12) * m_settings.m_volume);
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m_audioBuffer[m_audioBufferFill].r = (qint16)(deemph_r * (1<<12) * m_settings.m_volume);
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}
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else
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{
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Real deemph;
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m_deemphasisFilterX.process(ci.real(), deemph);
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quint16 sample = (qint16)(deemph * (1<<12) * m_settings.m_volume);
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m_audioBuffer[m_audioBufferFill].l = sample;
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m_audioBuffer[m_audioBufferFill].r = sample;
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}
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m_demodBuffer[m_demodBufferFill++] = m_audioBuffer[m_audioBufferFill].l;
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m_demodBuffer[m_demodBufferFill++] = m_audioBuffer[m_audioBufferFill].r;
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++m_audioBufferFill;
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if (m_audioBufferFill >= m_audioBuffer.size())
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{
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std::size_t res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], std::min(m_audioBufferFill, m_audioBuffer.size()));
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if (res != m_audioBufferFill)
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{
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qDebug("BFMDemodSink::feed: %lu/%lu audio samples written", res, m_audioBufferFill);
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m_audioFifo.clear();
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}
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m_audioBufferFill = 0;
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}
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if (m_demodBufferFill >= m_demodBuffer.size())
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{
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QList<ObjectPipe*> dataPipes;
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MainCore::instance()->getDataPipes().getDataPipes(m_channel, "demod", dataPipes);
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if (dataPipes.size() > 0)
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{
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QList<ObjectPipe*>::iterator it = dataPipes.begin();
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for (; it != dataPipes.end(); ++it)
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{
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DataFifo *fifo = qobject_cast<DataFifo*>((*it)->m_element);
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if (fifo) {
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fifo->write((quint8*) &m_demodBuffer[0], m_demodBuffer.size() * sizeof(qint16), DataFifo::DataTypeCI16);
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}
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}
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}
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m_demodBufferFill = 0;
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}
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m_interpolatorDistanceRemain += m_interpolatorDistance;
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}
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}
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}
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if (m_spectrumSink && (m_sampleBuffer.size() != 0))
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{
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m_spectrumSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), true);
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m_sampleBuffer.clear();
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}
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}
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void BFMDemodSink::applyAudioSampleRate(int sampleRate)
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{
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if (sampleRate < 0)
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{
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qWarning("BFMDemodSink::applyAudioSampleRate: invalid sample rate: %d", sampleRate);
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return;
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}
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qDebug("BFMDemodSink::applyAudioSampleRate: %u", sampleRate);
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m_interpolator.create(16, m_channelSampleRate, m_settings.m_afBandwidth);
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m_interpolatorDistanceRemain = (Real) m_channelSampleRate / sampleRate;
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m_interpolatorDistance = (Real) m_channelSampleRate / (Real) sampleRate;
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m_interpolatorStereo.create(16, m_channelSampleRate, m_settings.m_afBandwidth);
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m_interpolatorStereoDistanceRemain = (Real) m_channelSampleRate / sampleRate;
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m_interpolatorStereoDistance = (Real) m_channelSampleRate / (Real) sampleRate;
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m_deemphasisFilterX.configure(default_deemphasis * sampleRate * 1.0e-6);
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m_deemphasisFilterY.configure(default_deemphasis * sampleRate * 1.0e-6);
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m_audioSampleRate = sampleRate;
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}
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void BFMDemodSink::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
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{
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qDebug() << "BFMDemodSink::applyChannelSettings:"
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<< " channelSampleRate: " << channelSampleRate
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<< " channelFrequencyOffset: " << channelFrequencyOffset;
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if((channelFrequencyOffset != m_channelFrequencyOffset) ||
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(channelSampleRate != m_channelSampleRate) || force)
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{
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m_nco.setFreq(-channelFrequencyOffset, channelSampleRate);
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}
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if ((channelSampleRate != m_channelSampleRate) || force)
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{
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m_pilotPLL.configure(19000.0/channelSampleRate, 50.0/channelSampleRate, 0.01);
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m_interpolator.create(16, channelSampleRate, m_settings.m_afBandwidth);
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m_interpolatorDistanceRemain = (Real) channelSampleRate / m_audioSampleRate;
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m_interpolatorDistance = (Real) channelSampleRate / (Real) m_audioSampleRate;
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m_interpolatorStereo.create(16, channelSampleRate, m_settings.m_afBandwidth);
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m_interpolatorStereoDistanceRemain = (Real) channelSampleRate / m_audioSampleRate;
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m_interpolatorStereoDistance = (Real) channelSampleRate / (Real) m_audioSampleRate;
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m_interpolatorRDS.create(4, channelSampleRate, 600.0);
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m_interpolatorRDSDistanceRemain = (Real) channelSampleRate / 250000.0;
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m_interpolatorRDSDistance = (Real) channelSampleRate / 250000.0;
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Real lowCut = -(m_settings.m_rfBandwidth / 2.0) / channelSampleRate;
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Real hiCut = (m_settings.m_rfBandwidth / 2.0) / channelSampleRate;
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m_rfFilter->create_filter(lowCut, hiCut);
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m_phaseDiscri.setFMScaling(channelSampleRate / m_fmExcursion);
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}
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m_channelSampleRate = channelSampleRate;
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m_channelFrequencyOffset = channelFrequencyOffset;
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}
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void BFMDemodSink::applySettings(const BFMDemodSettings& settings, bool force)
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{
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qDebug() << "BFMDemodSink::applySettings: MsgConfigureBFMDemod:"
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<< " m_inputFrequencyOffset: " << settings.m_inputFrequencyOffset
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<< " m_rfBandwidth: " << settings.m_rfBandwidth
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<< " m_afBandwidth: " << settings.m_afBandwidth
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<< " m_volume: " << settings.m_volume
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<< " m_squelch: " << settings.m_squelch
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<< " m_audioStereo: " << settings.m_audioStereo
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<< " m_lsbStereo: " << settings.m_lsbStereo
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<< " m_showPilot: " << settings.m_showPilot
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<< " m_rdsActive: " << settings.m_rdsActive
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<< " m_audioDeviceName: " << settings.m_audioDeviceName
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<< " m_streamIndex: " << settings.m_streamIndex
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<< " m_useReverseAPI: " << settings.m_useReverseAPI
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<< " force: " << force;
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if ((settings.m_audioStereo && (settings.m_audioStereo != m_settings.m_audioStereo)) || force) {
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m_pilotPLL.configure(19000.0/m_channelSampleRate, 50.0/m_channelSampleRate, 0.01);
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}
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if ((settings.m_afBandwidth != m_settings.m_afBandwidth) || force)
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{
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m_interpolator.create(16, m_channelSampleRate, settings.m_afBandwidth);
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m_interpolatorDistanceRemain = (Real) m_channelSampleRate / m_audioSampleRate;
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m_interpolatorDistance = (Real) m_channelSampleRate / (Real) m_audioSampleRate;
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m_interpolatorStereo.create(16, m_channelSampleRate, settings.m_afBandwidth);
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m_interpolatorStereoDistanceRemain = (Real) m_channelSampleRate / m_audioSampleRate;
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m_interpolatorStereoDistance = (Real) m_channelSampleRate / (Real) m_audioSampleRate;
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m_interpolatorRDS.create(4, m_channelSampleRate, 600.0);
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m_interpolatorRDSDistanceRemain = (Real) m_channelSampleRate / 250000.0;
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m_interpolatorRDSDistance = (Real) m_channelSampleRate / 250000.0;
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m_lowpass.create(21, m_audioSampleRate, settings.m_afBandwidth);
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}
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if ((settings.m_rfBandwidth != m_settings.m_rfBandwidth) || force)
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{
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Real lowCut = -(settings.m_rfBandwidth / 2.0) / m_channelSampleRate;
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Real hiCut = (settings.m_rfBandwidth / 2.0) / m_channelSampleRate;
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m_rfFilter->create_filter(lowCut, hiCut);
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m_phaseDiscri.setFMScaling(m_channelSampleRate / m_fmExcursion);
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}
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if ((settings.m_squelch != m_settings.m_squelch) || force) {
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m_squelchLevel = std::pow(10.0, settings.m_squelch / 10.0);
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}
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m_settings = settings;
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}
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