/////////////////////////////////////////////////////////////////////////////////// // Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany // // written by Christian Daniel // // // // This program is free software; you can redistribute it and/or modify // // it under the terms of the GNU General Public License as published by // // the Free Software Foundation as version 3 of the License, or // // // // This program is distributed in the hope that it will be useful, // // but WITHOUT ANY WARRANTY; without even the implied warranty of // // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the // // GNU General Public License V3 for more details. // // // // You should have received a copy of the GNU General Public License // // along with this program. If not, see . // /////////////////////////////////////////////////////////////////////////////////// #include #include #include #include #include "audio/audiooutput.h" #include "dsp/dspengine.h" #include "dsp/channelizer.h" #include "dsp/pidcontroller.h" #include "wfmdemod.h" MESSAGE_CLASS_DEFINITION(WFMDemod::MsgConfigureWFMDemod, Message) WFMDemod::WFMDemod(SampleSink* sampleSink) : m_sampleSink(sampleSink), m_audioFifo(4, 250000), m_settingsMutex(QMutex::Recursive) { setObjectName("WFMDemod"); m_config.m_inputSampleRate = 384000; m_config.m_inputFrequencyOffset = 0; m_config.m_rfBandwidth = 180000; m_config.m_afBandwidth = 15000; m_config.m_squelch = -60.0; m_config.m_volume = 2.0; m_config.m_audioSampleRate = DSPEngine::instance()->getAudioSampleRate(); m_rfFilter = new fftfilt(-50000.0 / 384000.0, 50000.0 / 384000.0, rfFilterFftLength); apply(); m_audioBuffer.resize(16384); m_audioBufferFill = 0; m_movingAverage.resize(16, 0); DSPEngine::instance()->addAudioSink(&m_audioFifo); } WFMDemod::~WFMDemod() { if (m_rfFilter) { delete m_rfFilter; } DSPEngine::instance()->removeAudioSink(&m_audioFifo); } void WFMDemod::configure(MessageQueue* messageQueue, Real rfBandwidth, Real afBandwidth, Real volume, Real squelch) { Message* cmd = MsgConfigureWFMDemod::create(rfBandwidth, afBandwidth, volume, squelch); messageQueue->push(cmd); } void WFMDemod::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end, bool firstOfBurst) { Complex ci; fftfilt::cmplx *rf; int rf_out; Real msq, demod; m_settingsMutex.lock(); for (SampleVector::const_iterator it = begin; it != end; ++it) { Complex c(it->real() / 32768.0, it->imag() / 32768.0); c *= m_nco.nextIQ(); rf_out = m_rfFilter->runFilt(c, &rf); // filter RF before demod for (int i =0 ; i = m_squelchLevel) m_squelchState = m_running.m_rfBandwidth / 20; // decay rate if(m_squelchState > 0) { m_squelchState--; // Alternative without atan // http://www.embedded.com/design/configurable-systems/4212086/DSP-Tricks--Frequency-demodulation-algorithms- // in addition it needs scaling by instantaneous magnitude squared and volume (0..10) adjustment factor Real ip = rf[i].real() - m_m2Sample.real(); Real qp = rf[i].imag() - m_m2Sample.imag(); Real h1 = m_m1Sample.real() * qp; Real h2 = m_m1Sample.imag() * ip; demod = (h1 - h2) / (msq * 10.0); } else { demod = 0; } m_m2Sample = m_m1Sample; m_m1Sample = rf[i]; Complex e(demod, 0); if(m_interpolator.interpolate(&m_interpolatorDistanceRemain, e, &ci)) { quint16 sample = (qint16)(ci.real() * 3000 * m_running.m_volume); m_sampleBuffer.push_back(Sample(sample, sample)); m_audioBuffer[m_audioBufferFill].l = sample; m_audioBuffer[m_audioBufferFill].r = sample; ++m_audioBufferFill; if(m_audioBufferFill >= m_audioBuffer.size()) { uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 1); if(res != m_audioBufferFill) { qDebug("WFMDemod::feed: %u/%u audio samples written", res, m_audioBufferFill); } m_audioBufferFill = 0; } m_interpolatorDistanceRemain += m_interpolatorDistance; } } } if(m_audioBufferFill > 0) { uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 1); if(res != m_audioBufferFill) { qDebug("WFMDemod::feed: %u/%u tail samples written", res, m_audioBufferFill); } m_audioBufferFill = 0; } if(m_sampleSink != NULL) { m_sampleSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), false); } m_sampleBuffer.clear(); m_settingsMutex.unlock(); } void WFMDemod::start() { m_squelchState = 0; m_audioFifo.clear(); m_m1Sample = 0; } void WFMDemod::stop() { } bool WFMDemod::handleMessage(const Message& cmd) { qDebug() << "WFMDemod::handleMessage"; if (Channelizer::MsgChannelizerNotification::match(cmd)) { Channelizer::MsgChannelizerNotification& notif = (Channelizer::MsgChannelizerNotification&) cmd; m_config.m_inputSampleRate = notif.getSampleRate(); m_config.m_inputFrequencyOffset = notif.getFrequencyOffset(); apply(); qDebug() << "WFMDemod::handleMessage: MsgChannelizerNotification: m_inputSampleRate: " << m_config.m_inputSampleRate << " m_inputFrequencyOffset: " << m_config.m_inputFrequencyOffset; return true; } else if (MsgConfigureWFMDemod::match(cmd)) { MsgConfigureWFMDemod& cfg = (MsgConfigureWFMDemod&) cmd; m_config.m_rfBandwidth = cfg.getRFBandwidth(); m_config.m_afBandwidth = cfg.getAFBandwidth(); m_config.m_volume = cfg.getVolume(); m_config.m_squelch = cfg.getSquelch(); apply(); qDebug() << "WFMDemod::handleMessage: MsgConfigureWFMDemod: m_rfBandwidth: " << m_config.m_rfBandwidth << " m_afBandwidth: " << m_config.m_afBandwidth << " m_volume: " << m_config.m_volume << " m_squelch: " << m_config.m_squelch; return true; } else { if (m_sampleSink != 0) { return m_sampleSink->handleMessage(cmd); } else { return false; } } } void WFMDemod::apply() { if((m_config.m_inputFrequencyOffset != m_running.m_inputFrequencyOffset) || (m_config.m_inputSampleRate != m_running.m_inputSampleRate)) { qDebug() << "WFMDemod::handleMessage: m_nco.setFreq"; m_nco.setFreq(-m_config.m_inputFrequencyOffset, m_config.m_inputSampleRate); } if((m_config.m_inputSampleRate != m_running.m_inputSampleRate) || (m_config.m_afBandwidth != m_running.m_afBandwidth)) { m_settingsMutex.lock(); qDebug() << "WFMDemod::handleMessage: m_interpolator.create"; m_interpolator.create(16, m_config.m_inputSampleRate, m_config.m_afBandwidth); m_interpolatorDistanceRemain = (Real) m_config.m_inputSampleRate / m_config.m_audioSampleRate; m_interpolatorDistance = (Real) m_config.m_inputSampleRate / (Real) m_config.m_audioSampleRate; m_settingsMutex.unlock(); } if((m_config.m_inputSampleRate != m_running.m_inputSampleRate) || (m_config.m_rfBandwidth != m_running.m_rfBandwidth) || (m_config.m_inputFrequencyOffset != m_running.m_inputFrequencyOffset)) { m_settingsMutex.lock(); qDebug() << "WFMDemod::handleMessage: m_rfFilter->create_filter"; Real lowCut = -(m_config.m_rfBandwidth / 2.0) / m_config.m_inputSampleRate; Real hiCut = (m_config.m_rfBandwidth / 2.0) / m_config.m_inputSampleRate; m_rfFilter->create_filter(lowCut, hiCut); m_settingsMutex.unlock(); } if((m_config.m_afBandwidth != m_running.m_afBandwidth) || (m_config.m_audioSampleRate != m_running.m_audioSampleRate)) { m_settingsMutex.lock(); qDebug() << "WFMDemod::handleMessage: m_lowpass.create"; m_lowpass.create(21, m_config.m_audioSampleRate, m_config.m_afBandwidth); m_settingsMutex.unlock(); } if(m_config.m_squelch != m_running.m_squelch) { qDebug() << "WFMDemod::handleMessage: set m_squelchLevel"; m_squelchLevel = pow(10.0, m_config.m_squelch / 20.0); m_squelchLevel *= m_squelchLevel; } m_running.m_inputSampleRate = m_config.m_inputSampleRate; m_running.m_inputFrequencyOffset = m_config.m_inputFrequencyOffset; m_running.m_rfBandwidth = m_config.m_rfBandwidth; m_running.m_afBandwidth = m_config.m_afBandwidth; m_running.m_squelch = m_config.m_squelch; m_running.m_volume = m_config.m_volume; m_running.m_audioSampleRate = m_config.m_audioSampleRate; }