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384 lines
14 KiB
C++
384 lines
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 <cstdio>
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#include <complex.h>
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#include <QTime>
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#include <QDebug>
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#include "util/stepfunctions.h"
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#include "util/db.h"
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#include "util/messagequeue.h"
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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/misc.h"
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#include "dsp/datafifo.h"
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#include "device/deviceapi.h"
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#include "maincore.h"
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#include "nfmdemodreport.h"
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#include "nfmdemodsink.h"
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const double NFMDemodSink::afSqTones[] = {1000.0, 6000.0}; // {1200.0, 8000.0};
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const double NFMDemodSink::afSqTones_lowrate[] = {1000.0, 3500.0};
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const unsigned NFMDemodSink::FFT_FILTER_LENGTH = 1024;
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const unsigned NFMDemodSink::CTCSS_DETECTOR_RATE = 6000;
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NFMDemodSink::NFMDemodSink() :
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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_rfFilter(FFT_FILTER_LENGTH),
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m_ctcssIndex(0),
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m_sampleCount(0),
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m_squelchCount(0),
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m_squelchGate(4800),
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m_filterTaps((48000 / 48) | 1),
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m_squelchLevel(-990),
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m_squelchOpen(false),
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m_afSquelchOpen(false),
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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_afSquelch(),
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m_squelchDelayLine(24000),
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m_messageQueueToGUI(nullptr)
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{
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m_audioBuffer.resize(1<<16);
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m_demodBuffer.resize(1<<12);
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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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void NFMDemodSink::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end)
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{
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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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Complex ci;
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fftfilt::cmplx *rf;
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int 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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if (m_interpolatorDistance == 1.0f)
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{
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processOneSample(rf[i]);
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}
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else if (m_interpolatorDistance < 1.0f) // interpolate
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{
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while (!m_interpolator.interpolate(&m_interpolatorDistanceRemain, rf[i], &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, rf[i], &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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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("NFMDemodSink::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 NFMDemodSink::processOneSample(Complex &ci)
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{
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qint16 sample = 0;
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double magsqRaw; // = ci.real()*ci.real() + c.imag()*c.imag();
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Real deviation;
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Real demod = m_phaseDiscri.phaseDiscriminatorDelta(ci, magsqRaw, deviation);
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Real magsq = magsqRaw / (SDR_RX_SCALED*SDR_RX_SCALED);
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m_movingAverage(magsq);
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m_magsqSum += magsq;
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m_magsqPeak = std::max<double>(magsq, m_magsqPeak);
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m_magsqCount++;
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m_sampleCount++;
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bool squelchOpen = m_afSquelchOpen && m_settings.m_deltaSquelch;
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if (m_settings.m_deltaSquelch)
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{
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if (m_afSquelch.analyze(demod))
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{
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m_afSquelchOpen = squelchOpen = m_afSquelch.evaluate();
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if (!squelchOpen) {
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m_squelchDelayLine.zeroBack(m_audioSampleRate/10); // zero out evaluation period
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}
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}
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}
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else
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{
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squelchOpen = m_movingAverage >= m_squelchLevel;
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}
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if (squelchOpen)
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{
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m_squelchDelayLine.write(demod);
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if (m_squelchCount < 2*m_squelchGate) {
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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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m_squelchDelayLine.write(0);
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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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m_squelchOpen = m_squelchCount > m_squelchGate;
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int ctcssIndex = m_squelchOpen && m_settings.m_ctcssOn ? m_ctcssIndex : 0;
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if (m_squelchOpen)
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{
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if (m_settings.m_ctcssOn)
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{
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int factor = (m_audioSampleRate / CTCSS_DETECTOR_RATE) - 1; // decimate -> 6k
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if ((m_sampleCount & factor) == factor)
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{
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Real ctcssSample = m_ctcssLowpass.filter(demod);
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if (m_ctcssDetector.analyze(&ctcssSample))
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{
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int maxToneIndex;
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ctcssIndex = m_ctcssDetector.getDetectedTone(maxToneIndex) ? maxToneIndex + 1 : 0;
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}
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}
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}
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if (!m_settings.m_audioMute && (!m_settings.m_ctcssOn || m_ctcssIndexSelected == ctcssIndex || m_ctcssIndexSelected == 0))
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{
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Real audioSample = m_squelchDelayLine.readBack(m_squelchGate);
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audioSample = m_settings.m_highPass ? m_bandpass.filter(audioSample) : m_lowpass.filter(audioSample);
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audioSample *= m_settings.m_volume * std::numeric_limits<int16_t>::max();
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sample = clamp<float>(std::rint(audioSample), std::numeric_limits<int16_t>::lowest(), std::numeric_limits<int16_t>::max());
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}
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}
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if (ctcssIndex != m_ctcssIndex)
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{
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auto *guiQueue = getMessageQueueToGUI();
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if (guiQueue)
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{
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guiQueue->push(NFMDemodReport::MsgReportCTCSSFreq::create(
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ctcssIndex ? m_ctcssDetector.getToneSet()[ctcssIndex - 1] : 0));
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}
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m_ctcssIndex = ctcssIndex;
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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("NFMDemodSink::feed: %u/%u audio samples written", res, m_audioBufferFill);
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qDebug("NFMDemodSink::feed: m_audioSampleRate: %u m_channelSampleRate: %d", m_audioSampleRate, m_channelSampleRate);
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}
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m_audioBufferFill = 0;
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}
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m_demodBuffer[m_demodBufferFill] = sample;
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++m_demodBufferFill;
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if (m_demodBufferFill >= m_demodBuffer.size())
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{
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QList<DataFifo*> *dataFifos = MainCore::instance()->getDataPipes().getFifos(m_channel, "demod");
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if (dataFifos)
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{
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QList<DataFifo*>::iterator it = dataFifos->begin();
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for (; it != dataFifos->end(); ++it) {
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(*it)->write((quint8*) &m_demodBuffer[0], m_demodBuffer.size() * sizeof(qint16));
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}
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}
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m_demodBufferFill = 0;
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}
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}
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void NFMDemodSink::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
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{
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qDebug() << "NFMDemodSink::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_interpolator.create(16, channelSampleRate, m_settings.m_rfBandwidth / 2.2);
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m_interpolatorDistance = Real(channelSampleRate) / Real(m_audioSampleRate);
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m_interpolatorDistanceRemain = m_interpolatorDistance;
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Real lowCut = -Real(m_settings.m_fmDeviation) / channelSampleRate;
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Real hiCut = Real(m_settings.m_fmDeviation) / channelSampleRate;
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m_rfFilter.create_filter(lowCut, hiCut);
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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 NFMDemodSink::applySettings(const NFMDemodSettings& settings, bool force)
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{
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qDebug() << "NFMDemodSink::applySettings:"
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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_fmDeviation: " << settings.m_fmDeviation
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<< " m_volume: " << settings.m_volume
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<< " m_squelchGate: " << settings.m_squelchGate
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<< " m_deltaSquelch: " << settings.m_deltaSquelch
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<< " m_squelch: " << settings.m_squelch
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<< " m_ctcssIndex: " << settings.m_ctcssIndex
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<< " m_ctcssOn: " << settings.m_ctcssOn
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<< " m_highPass: " << settings.m_highPass
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<< " m_audioMute: " << settings.m_audioMute
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<< " m_audioDeviceName: " << settings.m_audioDeviceName
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<< " force: " << force;
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if ((settings.m_rfBandwidth != m_settings.m_rfBandwidth) || force)
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{
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m_interpolator.create(16, m_channelSampleRate, settings.m_rfBandwidth / 2.2);
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m_interpolatorDistance = Real(m_channelSampleRate) / Real(m_audioSampleRate);
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m_interpolatorDistanceRemain = m_interpolatorDistance;
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}
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if ((settings.m_fmDeviation != m_settings.m_fmDeviation) || force) {
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Real lowCut = -Real(settings.m_fmDeviation) / m_channelSampleRate;
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Real hiCut = Real(settings.m_fmDeviation) / m_channelSampleRate;
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m_rfFilter.create_filter(lowCut, hiCut);
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m_phaseDiscri.setFMScaling(Real(m_audioSampleRate) / (2.0f * settings.m_fmDeviation));
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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_bandpass.create(m_filterTaps, m_audioSampleRate, 300.0, settings.m_afBandwidth);
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m_lowpass.create(m_filterTaps, m_audioSampleRate, settings.m_afBandwidth);
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}
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if ((settings.m_squelchGate != m_settings.m_squelchGate) || force)
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{
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m_squelchGate = (m_audioSampleRate / 100) * settings.m_squelchGate; // gate is given in 10s of ms at 48000 Hz audio sample rate
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m_squelchCount = 0; // reset squelch open counter
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}
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if ((settings.m_squelch != m_settings.m_squelch) ||
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(settings.m_deltaSquelch != m_settings.m_deltaSquelch) || force)
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{
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if (settings.m_deltaSquelch)
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{ // input is a value in negative centis
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m_squelchLevel = (- settings.m_squelch) / 100.0;
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m_afSquelch.setThreshold(m_squelchLevel);
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m_afSquelch.reset();
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}
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else
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{ // input is a value in deci-Bels
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m_squelchLevel = std::pow(10.0, settings.m_squelch / 10.0);
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m_movingAverage.reset();
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}
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m_squelchCount = 0; // reset squelch open counter
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}
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if ((settings.m_ctcssIndex != m_settings.m_ctcssIndex) || force) {
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setSelectedCtcssIndex(settings.m_ctcssIndex);
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}
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m_settings = settings;
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}
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void NFMDemodSink::applyAudioSampleRate(unsigned int sampleRate)
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{
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qDebug("NFMDemodSink::applyAudioSampleRate: %u m_channelSampleRate: %d", sampleRate, m_channelSampleRate);
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m_filterTaps = (sampleRate / 48) | 1;
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m_ctcssLowpass.create((CTCSS_DETECTOR_RATE / 48) | 1, CTCSS_DETECTOR_RATE, 250.0);
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m_bandpass.create(m_filterTaps, sampleRate, 300.0, m_settings.m_afBandwidth);
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m_lowpass.create(m_filterTaps, sampleRate, m_settings.m_afBandwidth);
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m_squelchGate = (sampleRate / 100) * m_settings.m_squelchGate; // gate is given in 10s of ms at 48000 Hz audio sample rate
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m_squelchCount = 0; // reset squelch open counter
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m_ctcssDetector.setCoefficients(sampleRate/16, CTCSS_DETECTOR_RATE); // 0.5s / 2 Hz resolution
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if (sampleRate < 16000) {
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m_afSquelch.setCoefficients(sampleRate/2000, 600, sampleRate, 200, 0, afSqTones_lowrate); // 0.5ms test period, 300ms average span, audio SR, 100ms attack, no decay
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} else {
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m_afSquelch.setCoefficients(sampleRate/2000, 600, sampleRate, 200, 0, afSqTones); // 0.5ms test period, 300ms average span, audio SR, 100ms attack, no decay
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}
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m_afSquelch.setThreshold(m_squelchLevel);
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m_phaseDiscri.setFMScaling(Real(sampleRate) / (2.0f * m_settings.m_fmDeviation));
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m_audioFifo.setSize(sampleRate);
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m_squelchDelayLine.resize(sampleRate/2);
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m_interpolatorDistance = Real(m_channelSampleRate) / Real(sampleRate);
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m_interpolatorDistanceRemain = m_interpolatorDistance;
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m_audioSampleRate = sampleRate;
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QList<MessageQueue*> *messageQueues = MainCore::instance()->getMessagePipes().getMessageQueues(m_channel, "reportdemod");
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if (messageQueues)
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{
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QList<MessageQueue*>::iterator it = messageQueues->begin();
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for (; it != messageQueues->end(); ++it)
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{
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MainCore::MsgChannelDemodReport *msg = MainCore::MsgChannelDemodReport::create(m_channel, sampleRate);
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(*it)->push(msg);
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}
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}
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}
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