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321 lines
11 KiB
C++
321 lines
11 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 <QDebug>
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#include <complex.h>
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#include "audio/audiooutput.h"
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#include "dsp/fftfilt.h"
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#include "util/db.h"
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#include "util/stepfunctions.h"
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#include "amdemodsink.h"
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AMDemodSink::AMDemodSink() :
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m_channelSampleRate(48000),
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m_channelFrequencyOffset(0),
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m_squelchOpen(false),
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m_squelchDelayLine(9600),
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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_volumeAGC(0.003),
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m_syncAMAGC(12000, 0.1, 1e-2),
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m_audioFifo(48000)
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{
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m_audioBuffer.resize(1<<14);
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m_audioBufferFill = 0;
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m_magsq = 0.0;
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DSBFilter = new fftfilt((2.0f * m_settings.m_rfBandwidth) / m_audioSampleRate, 2 * 1024);
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SSBFilter = new fftfilt(0.0f, m_settings.m_rfBandwidth / m_audioSampleRate, 1024);
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m_syncAMAGC.setThresholdEnable(false);
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m_syncAMAGC.resize(12000, 6000, 0.1);
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m_pllFilt.create(101, m_audioSampleRate, 200.0);
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m_pll.computeCoefficients(0.05, 0.707, 1000);
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m_syncAMBuffIndex = 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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AMDemodSink::~AMDemodSink()
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{
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delete DSBFilter;
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delete SSBFilter;
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}
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void AMDemodSink::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end)
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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 // decimate
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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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if (m_audioBufferFill > 0)
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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("AMDemodSink::feed: %u/%u tail 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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void AMDemodSink::processOneSample(Complex &ci)
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{
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Real re = ci.real() / SDR_RX_SCALEF;
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Real im = ci.imag() / SDR_RX_SCALEF;
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Real magsq = re*re + im*im;
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m_movingAverage(magsq);
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m_magsq = m_movingAverage.asDouble();
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m_magsqSum += magsq;
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if (magsq > m_magsqPeak)
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{
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m_magsqPeak = magsq;
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}
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m_magsqCount++;
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m_squelchDelayLine.write(magsq);
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if (m_magsq < m_squelchLevel)
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{
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if (m_squelchCount > 0) {
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m_squelchCount--;
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}
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}
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else
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{
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if (m_squelchCount < m_audioSampleRate / 10) {
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m_squelchCount++;
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}
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}
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qint16 sample;
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m_squelchOpen = (m_squelchCount >= m_audioSampleRate / 20);
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if (m_squelchOpen && !m_settings.m_audioMute)
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{
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Real demod;
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if (m_settings.m_pll)
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{
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std::complex<float> s(re, im);
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s = m_pllFilt.filter(s);
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m_pll.feed(s.real(), s.imag());
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float yr = re * m_pll.getImag() - im * m_pll.getReal();
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float yi = re * m_pll.getReal() + im * m_pll.getImag();
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fftfilt::cmplx *sideband;
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std::complex<float> cs(yr, yi);
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int n_out;
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if (m_settings.m_syncAMOperation == AMDemodSettings::SyncAMDSB) {
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n_out = DSBFilter->runDSB(cs, &sideband, false);
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} else {
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n_out = SSBFilter->runSSB(cs, &sideband, m_settings.m_syncAMOperation == AMDemodSettings::SyncAMUSB, false);
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}
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for (int i = 0; i < n_out; i++)
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{
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float agcVal = m_syncAMAGC.feedAndGetValue(sideband[i]);
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fftfilt::cmplx z = sideband[i] * agcVal; // * m_syncAMAGC.getStepValue();
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if (m_settings.m_syncAMOperation == AMDemodSettings::SyncAMDSB) {
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m_syncAMBuff[i] = (z.real() + z.imag());
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} else if (m_settings.m_syncAMOperation == AMDemodSettings::SyncAMUSB) {
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m_syncAMBuff[i] = (z.real() + z.imag());
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} else {
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m_syncAMBuff[i] = (z.real() + z.imag());
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}
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m_syncAMBuffIndex = 0;
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}
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m_syncAMBuffIndex = m_syncAMBuffIndex < 2*1024 ? m_syncAMBuffIndex : 0;
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demod = m_syncAMBuff[m_syncAMBuffIndex++]*4.0f; // mos pifometrico
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}
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else
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{
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demod = sqrt(m_squelchDelayLine.readBack(m_audioSampleRate/20));
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m_volumeAGC.feed(demod);
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demod = (demod - m_volumeAGC.getValue()) / m_volumeAGC.getValue();
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}
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if (m_settings.m_bandpassEnable)
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{
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demod = m_bandpass.filter(demod);
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demod /= 301.0f;
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}
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else
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{
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demod = m_lowpass.filter(demod);
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}
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Real attack = (m_squelchCount - 0.05f * m_audioSampleRate) / (0.05f * m_audioSampleRate);
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sample = demod * StepFunctions::smootherstep(attack) * (m_audioSampleRate/24) * m_settings.m_volume;
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}
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else
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{
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sample = 0;
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}
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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_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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{
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qDebug("AMDemodSink::processOneSample: %u/%u 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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}
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void AMDemodSink::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
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{
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qDebug() << "AMDemodSink::applyChannelSettings:"
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<< " channelSampleRate: " << channelSampleRate
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<< " channelFrequencyOffset: " << channelFrequencyOffset
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<< " m_audioSampleRate: " << m_audioSampleRate;
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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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m_interpolator.create(16, channelSampleRate, m_settings.m_rfBandwidth / 2.2f);
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m_interpolatorDistanceRemain = 0;
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m_interpolatorDistance = (Real) channelSampleRate / (Real) m_audioSampleRate;
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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 AMDemodSink::applySettings(const AMDemodSettings& settings, bool force)
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{
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qDebug() << "AMDemodSink::applySettings:"
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<< " m_inputFrequencyOffset: " << settings.m_inputFrequencyOffset
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<< " m_rfBandwidth: " << settings.m_rfBandwidth
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<< " m_volume: " << settings.m_volume
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<< " m_squelch: " << settings.m_squelch
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<< " m_audioMute: " << settings.m_audioMute
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<< " m_bandpassEnable: " << settings.m_bandpassEnable
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<< " m_audioDeviceName: " << settings.m_audioDeviceName
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<< " m_pll: " << settings.m_pll
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<< " m_syncAMOperation: " << (int) settings.m_syncAMOperation
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<< " force: " << force;
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if((m_settings.m_rfBandwidth != settings.m_rfBandwidth) ||
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(m_settings.m_bandpassEnable != settings.m_bandpassEnable) || force)
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{
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m_interpolator.create(16, m_channelSampleRate, settings.m_rfBandwidth / 2.2f);
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m_interpolatorDistanceRemain = 0;
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m_interpolatorDistance = (Real) m_channelSampleRate / (Real) m_audioSampleRate;
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m_bandpass.create(301, m_audioSampleRate, 300.0, settings.m_rfBandwidth / 2.0f);
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m_lowpass.create(301, m_audioSampleRate, settings.m_rfBandwidth / 2.0f);
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DSBFilter->create_dsb_filter((2.0f * settings.m_rfBandwidth) / (float) m_audioSampleRate);
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}
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if ((m_settings.m_squelch != settings.m_squelch) || force) {
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m_squelchLevel = CalcDb::powerFromdB(settings.m_squelch);
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}
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if ((m_settings.m_pll != settings.m_pll) || force)
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{
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if (settings.m_pll)
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{
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m_volumeAGC.resizeNew(m_audioSampleRate/4, 0.003);
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m_syncAMBuffIndex = 0;
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}
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else
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{
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m_volumeAGC.resizeNew(m_audioSampleRate/10, 0.003);
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}
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}
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if ((m_settings.m_syncAMOperation != settings.m_syncAMOperation) || force) {
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m_syncAMBuffIndex = 0;
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}
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m_settings = settings;
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}
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void AMDemodSink::applyAudioSampleRate(int sampleRate)
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{
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qDebug("AMDemodSink::applyAudioSampleRate: sampleRate: %d m_channelSampleRate: %d", sampleRate, m_channelSampleRate);
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m_interpolator.create(16, m_channelSampleRate, m_settings.m_rfBandwidth / 2.2f);
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m_interpolatorDistanceRemain = 0;
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m_interpolatorDistance = (Real) m_channelSampleRate / (Real) sampleRate;
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m_bandpass.create(301, sampleRate, 300.0, m_settings.m_rfBandwidth / 2.0f);
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m_lowpass.create(301, sampleRate, m_settings.m_rfBandwidth / 2.0f);
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m_audioFifo.setSize(sampleRate);
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m_squelchDelayLine.resize(sampleRate/5);
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DSBFilter->create_dsb_filter((2.0f * m_settings.m_rfBandwidth) / (float) sampleRate);
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m_pllFilt.create(101, sampleRate, 200.0);
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if (m_settings.m_pll) {
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m_volumeAGC.resizeNew(sampleRate, 0.003);
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} else {
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m_volumeAGC.resizeNew(sampleRate/10, 0.003);
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}
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m_syncAMAGC.resize(sampleRate/4, sampleRate/8, 0.1);
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m_pll.setSampleRate(sampleRate);
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m_audioSampleRate = sampleRate;
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}
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