kopia lustrzana https://github.com/f4exb/sdrangel
399 wiersze
14 KiB
C++
399 wiersze
14 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2019 Edouard Griffiths, F4EXB //
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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 <QTime>
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#include <QDebug>
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#include "audio/audiooutputdevice.h"
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#include "dsp/dspengine.h"
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#include "dsp/dspcommands.h"
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#include "dsp/devicesamplemimo.h"
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#include "dsp/spectrumvis.h"
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#include "dsp/datafifo.h"
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#include "device/deviceapi.h"
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#include "util/db.h"
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#include "util/messagequeue.h"
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#include "maincore.h"
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#include "ft8demodsink.h"
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const int FT8DemodSink::m_ssbFftLen = 1024;
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const int FT8DemodSink::m_agcTarget = 3276; // 32768/10 -10 dB amplitude => -20 dB power: center of normal signal
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FT8DemodSink::FT8DemodSink() :
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m_agc(12000, m_agcTarget, 1e-2),
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m_agcActive(false),
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m_agcClamping(false),
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m_agcNbSamples(12000),
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m_agcPowerThreshold(1e-2),
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m_agcThresholdGate(0),
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m_squelchDelayLine(2*48000),
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m_audioActive(false),
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m_spectrumSink(nullptr),
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m_audioFifo(24000),
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m_ft8SampleRate(12000)
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{
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m_Bandwidth = 5000;
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m_LowCutoff = 300;
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m_volume = 2.0;
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m_spanLog2 = 3;
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m_channelSampleRate = 48000;
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m_channelFrequencyOffset = 0;
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m_audioBuffer.resize(1<<14);
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m_audioBufferFill = 0;
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m_undersampleCount = 0;
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m_sum = 0;
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m_demodBuffer.resize(1<<12);
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m_demodBufferFill = 0;
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m_usb = true;
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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_agc.setClampMax(SDR_RX_SCALED/100.0);
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m_agc.setClamping(m_agcClamping);
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SSBFilter = new fftfilt(m_LowCutoff / m_ft8SampleRate, m_Bandwidth / m_ft8SampleRate, m_ssbFftLen);
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applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
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applySettings(m_settings, true);
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}
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FT8DemodSink::~FT8DemodSink()
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{
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delete SSBFilter;
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}
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void FT8DemodSink::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end)
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{
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if (m_channelSampleRate == 0) {
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return;
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}
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Complex ci;
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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(), it->imag());
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c *= m_nco.nextIQ();
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if (m_interpolatorDistance < 1.0f) // interpolate
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{
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while (!m_interpolator.interpolate(&m_interpolatorDistanceRemain, c, &ci))
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{
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processOneSample(ci);
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m_interpolatorDistanceRemain += m_interpolatorDistance;
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}
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}
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else
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{
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if (m_interpolator.decimate(&m_interpolatorDistanceRemain, c, &ci))
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{
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processOneSample(ci);
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m_interpolatorDistanceRemain += m_interpolatorDistance;
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}
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}
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}
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}
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void FT8DemodSink::processOneSample(Complex &ci)
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{
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fftfilt::cmplx *sideband;
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int n_out = 0;
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int decim = 1<<(m_spanLog2 - 1);
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unsigned char decim_mask = decim - 1; // counter LSB bit mask for decimation by 2^(m_scaleLog2 - 1)
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n_out = SSBFilter->runSSB(ci, &sideband, m_usb);
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for (int i = 0; i < n_out; i++)
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{
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// Downsample by 2^(m_scaleLog2 - 1) for SSB band spectrum display
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// smart decimation with bit gain using float arithmetic (23 bits significand)
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m_sum += sideband[i];
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if (!(m_undersampleCount++ & decim_mask))
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{
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Real avgr = m_sum.real() / decim;
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Real avgi = m_sum.imag() / decim;
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m_magsq = (avgr * avgr + avgi * avgi) / (SDR_RX_SCALED*SDR_RX_SCALED);
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m_magsqSum += m_magsq;
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if (m_magsq > m_magsqPeak)
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{
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m_magsqPeak = m_magsq;
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}
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m_magsqCount++;
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m_sampleBuffer.push_back(Sample(avgr, avgi));
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m_sum.real(0.0);
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m_sum.imag(0.0);
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}
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float agcVal = m_agcActive ? m_agc.feedAndGetValue(sideband[i]) : 0.1;
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fftfilt::cmplx& delayedSample = m_squelchDelayLine.readBack(m_agc.getStepDownDelay());
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m_audioActive = delayedSample.real() != 0.0;
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m_squelchDelayLine.write(sideband[i]*agcVal);
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fftfilt::cmplx z = m_agcActive ? delayedSample * m_agc.getStepValue() : delayedSample;
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Real demod = (z.real() + z.imag()) * 0.7;
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qint16 sample = (qint16)(demod * 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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m_demodBuffer[m_demodBufferFill++] = (z.real() + z.imag()) * 0.7;
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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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{
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fifo->write(
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(quint8*) &m_demodBuffer[0],
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m_demodBuffer.size() * sizeof(qint16),
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DataFifo::DataTypeI16
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);
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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_audioBufferFill;
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if (m_audioBufferFill >= m_audioBuffer.size())
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{
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uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill);
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if (res != m_audioBufferFill) {
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qDebug("FT8DemodSink::processOneSample: %u/%u samples written", res, m_audioBufferFill);
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}
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m_audioBufferFill = 0;
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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 FT8DemodSink::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
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{
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qDebug() << "FT8DemodSink::applyChannelSettings:"
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<< " channelSampleRate: " << channelSampleRate
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<< " channelFrequencyOffset: " << channelFrequencyOffset;
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if ((m_channelFrequencyOffset != channelFrequencyOffset) ||
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(m_channelSampleRate != channelSampleRate) || force)
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{
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m_nco.setFreq(-channelFrequencyOffset, channelSampleRate);
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}
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if ((m_channelSampleRate != channelSampleRate) || force)
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{
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Real interpolatorBandwidth = (m_Bandwidth * 1.5f) > channelSampleRate ? channelSampleRate : (m_Bandwidth * 1.5f);
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m_interpolator.create(16, channelSampleRate, interpolatorBandwidth, 2.0f);
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m_interpolatorDistanceRemain = 0;
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m_interpolatorDistance = (Real) channelSampleRate / (Real) m_ft8SampleRate;
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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 FT8DemodSink::applyFT8SampleRate(int sampleRate)
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{
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qDebug("FT8DemodSink::applyFT8SampleRate: %d", sampleRate);
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Real interpolatorBandwidth = (m_Bandwidth * 1.5f) > m_channelSampleRate ? m_channelSampleRate : (m_Bandwidth * 1.5f);
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m_interpolator.create(16, m_channelSampleRate, interpolatorBandwidth, 2.0f);
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m_interpolatorDistanceRemain = 0;
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m_interpolatorDistance = (Real) m_channelSampleRate / (Real) sampleRate;
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SSBFilter->create_filter(m_LowCutoff / (float) sampleRate, m_Bandwidth / (float) sampleRate, m_settings.m_filterBank[m_settings.m_filterIndex].m_fftWindow);
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int agcNbSamples = (sampleRate / 1000) * (1<<m_settings.m_agcTimeLog2);
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int agcThresholdGate = (sampleRate / 1000) * m_settings.m_agcThresholdGate; // ms
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if (m_agcNbSamples != agcNbSamples)
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{
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m_agc.resize(agcNbSamples, agcNbSamples/2, m_agcTarget);
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m_agc.setStepDownDelay(agcNbSamples);
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m_agcNbSamples = agcNbSamples;
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}
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if (m_agcThresholdGate != agcThresholdGate)
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{
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m_agc.setGate(agcThresholdGate);
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m_agcThresholdGate = agcThresholdGate;
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}
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m_audioFifo.setSize(sampleRate);
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m_ft8SampleRate = sampleRate;
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QList<ObjectPipe*> pipes;
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MainCore::instance()->getMessagePipes().getMessagePipes(m_channel, "reportdemod", pipes);
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if (pipes.size() > 0)
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{
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for (const auto& pipe : pipes)
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{
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MessageQueue* messageQueue = qobject_cast<MessageQueue*>(pipe->m_element);
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if (messageQueue)
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{
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MainCore::MsgChannelDemodReport *msg = MainCore::MsgChannelDemodReport::create(m_channel, sampleRate);
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messageQueue->push(msg);
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}
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}
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}
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}
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void FT8DemodSink::applySettings(const FT8DemodSettings& settings, bool force)
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{
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qDebug() << "FT8DemodSink::applySettings:"
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<< " m_inputFrequencyOffset: " << settings.m_inputFrequencyOffset
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<< " m_filterIndex: " << settings.m_filterIndex
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<< " [m_spanLog2: " << settings.m_filterBank[settings.m_filterIndex].m_spanLog2
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<< " m_rfBandwidth: " << settings.m_filterBank[settings.m_filterIndex].m_rfBandwidth
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<< " m_lowCutoff: " << settings.m_filterBank[settings.m_filterIndex].m_lowCutoff
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<< " m_fftWindow: " << settings.m_filterBank[settings.m_filterIndex].m_fftWindow << "]"
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<< " m_volume: " << settings.m_volume
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<< " m_agcActive: " << settings.m_agc
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<< " m_agcClamping: " << settings.m_agcClamping
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<< " m_agcTimeLog2: " << settings.m_agcTimeLog2
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<< " agcPowerThreshold: " << settings.m_agcPowerThreshold
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<< " agcThresholdGate: " << settings.m_agcThresholdGate
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<< " m_ft8SampleRate: " << settings.m_ft8SampleRate
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<< " m_streamIndex: " << settings.m_streamIndex
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<< " m_useReverseAPI: " << settings.m_useReverseAPI
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<< " m_reverseAPIAddress: " << settings.m_reverseAPIAddress
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<< " m_reverseAPIPort: " << settings.m_reverseAPIPort
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<< " m_reverseAPIDeviceIndex: " << settings.m_reverseAPIDeviceIndex
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<< " m_reverseAPIChannelIndex: " << settings.m_reverseAPIChannelIndex
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<< " force: " << force;
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if((m_settings.m_filterBank[m_settings.m_filterIndex].m_rfBandwidth != settings.m_filterBank[settings.m_filterIndex].m_rfBandwidth) ||
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(m_settings.m_filterBank[m_settings.m_filterIndex].m_lowCutoff != settings.m_filterBank[settings.m_filterIndex].m_lowCutoff) ||
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(m_settings.m_filterBank[m_settings.m_filterIndex].m_fftWindow != settings.m_filterBank[settings.m_filterIndex].m_fftWindow) || force)
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{
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float band, lowCutoff;
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band = settings.m_filterBank[settings.m_filterIndex].m_rfBandwidth;
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lowCutoff = settings.m_filterBank[settings.m_filterIndex].m_lowCutoff;
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if (band < 0) {
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band = -band;
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lowCutoff = -lowCutoff;
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m_usb = false;
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} else {
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m_usb = true;
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}
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if (band < 100.0f)
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{
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band = 100.0f;
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lowCutoff = 0;
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}
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m_Bandwidth = band;
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m_LowCutoff = lowCutoff;
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Real interpolatorBandwidth = (m_Bandwidth * 1.5f) > m_channelSampleRate ? m_channelSampleRate : (m_Bandwidth * 1.5f);
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m_interpolator.create(16, m_channelSampleRate, interpolatorBandwidth, 2.0f);
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m_interpolatorDistanceRemain = 0;
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m_interpolatorDistance = (Real) m_channelSampleRate / (Real) m_ft8SampleRate;
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SSBFilter->create_filter(m_LowCutoff / (float) m_ft8SampleRate, m_Bandwidth / (float) m_ft8SampleRate, settings.m_filterBank[settings.m_filterIndex].m_fftWindow);
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}
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if ((m_settings.m_volume != settings.m_volume) || force)
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{
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m_volume = settings.m_volume;
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m_volume /= 4.0; // for 3276.8
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}
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if ((m_settings.m_agcTimeLog2 != settings.m_agcTimeLog2) ||
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(m_settings.m_agcPowerThreshold != settings.m_agcPowerThreshold) ||
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(m_settings.m_agcThresholdGate != settings.m_agcThresholdGate) ||
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(m_settings.m_agcClamping != settings.m_agcClamping) || force)
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{
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int agcNbSamples = (m_ft8SampleRate / 1000) * (1<<settings.m_agcTimeLog2);
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m_agc.setThresholdEnable(settings.m_agcPowerThreshold != -FT8DemodSettings::m_minPowerThresholdDB);
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double agcPowerThreshold = CalcDb::powerFromdB(settings.m_agcPowerThreshold) * (SDR_RX_SCALED*SDR_RX_SCALED);
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int agcThresholdGate = (m_ft8SampleRate / 1000) * settings.m_agcThresholdGate; // ms
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bool agcClamping = settings.m_agcClamping;
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if (m_agcNbSamples != agcNbSamples)
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{
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m_agc.resize(agcNbSamples, agcNbSamples/2, m_agcTarget);
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m_agc.setStepDownDelay(agcNbSamples);
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m_agcNbSamples = agcNbSamples;
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}
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if (m_agcPowerThreshold != agcPowerThreshold)
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{
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m_agc.setThreshold(agcPowerThreshold);
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m_agcPowerThreshold = agcPowerThreshold;
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}
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if (m_agcThresholdGate != agcThresholdGate)
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{
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m_agc.setGate(agcThresholdGate);
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m_agcThresholdGate = agcThresholdGate;
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}
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if (m_agcClamping != agcClamping)
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{
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m_agc.setClamping(agcClamping);
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m_agcClamping = agcClamping;
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}
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qDebug() << "FT8DemodSink::applySettings: AGC:"
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<< " agcNbSamples: " << agcNbSamples
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<< " agcPowerThreshold: " << agcPowerThreshold
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<< " agcThresholdGate: " << agcThresholdGate
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<< " agcClamping: " << agcClamping;
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}
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m_spanLog2 = settings.m_filterBank[settings.m_filterIndex].m_spanLog2;
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m_agcActive = settings.m_agc;
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m_settings = settings;
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}
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