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284 lines
8.6 KiB
C++
284 lines
8.6 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany //
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// written by Christian Daniel //
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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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// //
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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 <QTime>
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#include <QDebug>
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#include <stdio.h>
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#include <complex.h>
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#include "audio/audiooutput.h"
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#include "dsp/dspengine.h"
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#include "dsp/channelizer.h"
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#include "dsp/pidcontroller.h"
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#include "wfmdemod.h"
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MESSAGE_CLASS_DEFINITION(WFMDemod::MsgConfigureWFMDemod, Message)
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WFMDemod::WFMDemod(SampleSink* sampleSink) :
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m_sampleSink(sampleSink),
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m_audioFifo(4, 250000),
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m_settingsMutex(QMutex::Recursive)
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{
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setObjectName("WFMDemod");
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m_config.m_inputSampleRate = 384000;
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m_config.m_inputFrequencyOffset = 0;
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m_config.m_rfBandwidth = 180000;
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m_config.m_afBandwidth = 15000;
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m_config.m_squelch = -60.0;
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m_config.m_volume = 2.0;
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m_config.m_audioSampleRate = DSPEngine::instance()->getAudioSampleRate();
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m_rfFilter = new fftfilt(-50000.0 / 384000.0, 50000.0 / 384000.0, rfFilterFftLength);
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apply();
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m_audioBuffer.resize(16384);
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m_audioBufferFill = 0;
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m_movingAverage.resize(16, 0);
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DSPEngine::instance()->addAudioSink(&m_audioFifo);
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}
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WFMDemod::~WFMDemod()
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{
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if (m_rfFilter)
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{
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delete m_rfFilter;
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}
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DSPEngine::instance()->removeAudioSink(&m_audioFifo);
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}
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void WFMDemod::configure(MessageQueue* messageQueue, Real rfBandwidth, Real afBandwidth, Real volume, Real squelch)
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{
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Message* cmd = MsgConfigureWFMDemod::create(rfBandwidth, afBandwidth, volume, squelch);
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messageQueue->push(cmd);
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}
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void WFMDemod::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end, bool firstOfBurst)
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{
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Complex ci;
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fftfilt::cmplx *rf;
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int rf_out;
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Real msq, demod;
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m_settingsMutex.lock();
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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() / 32768.0f, it->imag() / 32768.0f);
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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_movingAverage.feed(msq);
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if(m_movingAverage.average() >= m_squelchLevel)
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m_squelchState = m_running.m_rfBandwidth / 20; // decay rate
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if(m_squelchState > 0)
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{
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m_squelchState--;
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// Alternative without atan
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// http://www.embedded.com/design/configurable-systems/4212086/DSP-Tricks--Frequency-demodulation-algorithms-
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// in addition it needs scaling by instantaneous magnitude squared and volume (0..10) adjustment factor
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Real ip = rf[i].real() - m_m2Sample.real();
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Real qp = rf[i].imag() - m_m2Sample.imag();
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Real h1 = m_m1Sample.real() * qp;
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Real h2 = m_m1Sample.imag() * ip;
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demod = (h1 - h2) / (msq * 10.0);
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}
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else
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{
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demod = 0;
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}
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m_m2Sample = m_m1Sample;
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m_m1Sample = rf[i];
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Complex e(demod, 0);
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if(m_interpolator.interpolate(&m_interpolatorDistanceRemain, e, &ci))
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{
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quint16 sample = (qint16)(ci.real() * 3000 * m_running.m_volume);
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m_sampleBuffer.push_back(Sample(sample, sample));
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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, 1);
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if(res != m_audioBufferFill)
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{
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qDebug("WFMDemod::feed: %u/%u audio samples written", res, m_audioBufferFill);
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}
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m_audioBufferFill = 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_audioBufferFill > 0)
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{
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uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 1);
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if(res != m_audioBufferFill)
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{
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qDebug("WFMDemod::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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if(m_sampleSink != NULL)
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{
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m_sampleSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), false);
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}
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m_sampleBuffer.clear();
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m_settingsMutex.unlock();
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}
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void WFMDemod::start()
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{
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m_squelchState = 0;
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m_audioFifo.clear();
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m_m1Sample = 0;
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}
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void WFMDemod::stop()
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{
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}
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bool WFMDemod::handleMessage(const Message& cmd)
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{
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qDebug() << "WFMDemod::handleMessage";
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if (Channelizer::MsgChannelizerNotification::match(cmd))
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{
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Channelizer::MsgChannelizerNotification& notif = (Channelizer::MsgChannelizerNotification&) cmd;
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m_config.m_inputSampleRate = notif.getSampleRate();
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m_config.m_inputFrequencyOffset = notif.getFrequencyOffset();
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apply();
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qDebug() << "WFMDemod::handleMessage: MsgChannelizerNotification: m_inputSampleRate: " << m_config.m_inputSampleRate
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<< " m_inputFrequencyOffset: " << m_config.m_inputFrequencyOffset;
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return true;
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}
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else if (MsgConfigureWFMDemod::match(cmd))
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{
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MsgConfigureWFMDemod& cfg = (MsgConfigureWFMDemod&) cmd;
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m_config.m_rfBandwidth = cfg.getRFBandwidth();
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m_config.m_afBandwidth = cfg.getAFBandwidth();
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m_config.m_volume = cfg.getVolume();
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m_config.m_squelch = cfg.getSquelch();
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apply();
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qDebug() << "WFMDemod::handleMessage: MsgConfigureWFMDemod: m_rfBandwidth: " << m_config.m_rfBandwidth
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<< " m_afBandwidth: " << m_config.m_afBandwidth
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<< " m_volume: " << m_config.m_volume
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<< " m_squelch: " << m_config.m_squelch;
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return true;
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}
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else
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{
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if (m_sampleSink != 0)
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{
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return m_sampleSink->handleMessage(cmd);
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}
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else
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{
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return false;
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}
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}
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}
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void WFMDemod::apply()
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{
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if((m_config.m_inputFrequencyOffset != m_running.m_inputFrequencyOffset) ||
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(m_config.m_inputSampleRate != m_running.m_inputSampleRate))
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{
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qDebug() << "WFMDemod::handleMessage: m_nco.setFreq";
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m_nco.setFreq(-m_config.m_inputFrequencyOffset, m_config.m_inputSampleRate);
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}
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if((m_config.m_inputSampleRate != m_running.m_inputSampleRate) ||
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(m_config.m_afBandwidth != m_running.m_afBandwidth))
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{
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m_settingsMutex.lock();
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qDebug() << "WFMDemod::handleMessage: m_interpolator.create";
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m_interpolator.create(16, m_config.m_inputSampleRate, m_config.m_afBandwidth);
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m_interpolatorDistanceRemain = (Real) m_config.m_inputSampleRate / m_config.m_audioSampleRate;
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m_interpolatorDistance = (Real) m_config.m_inputSampleRate / (Real) m_config.m_audioSampleRate;
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m_settingsMutex.unlock();
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}
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if((m_config.m_inputSampleRate != m_running.m_inputSampleRate) ||
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(m_config.m_rfBandwidth != m_running.m_rfBandwidth) ||
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(m_config.m_inputFrequencyOffset != m_running.m_inputFrequencyOffset))
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{
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m_settingsMutex.lock();
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qDebug() << "WFMDemod::handleMessage: m_rfFilter->create_filter";
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Real lowCut = -(m_config.m_rfBandwidth / 2.0) / m_config.m_inputSampleRate;
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Real hiCut = (m_config.m_rfBandwidth / 2.0) / m_config.m_inputSampleRate;
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m_rfFilter->create_filter(lowCut, hiCut);
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m_settingsMutex.unlock();
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}
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if((m_config.m_afBandwidth != m_running.m_afBandwidth) ||
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(m_config.m_audioSampleRate != m_running.m_audioSampleRate))
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{
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m_settingsMutex.lock();
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qDebug() << "WFMDemod::handleMessage: m_lowpass.create";
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m_lowpass.create(21, m_config.m_audioSampleRate, m_config.m_afBandwidth);
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m_settingsMutex.unlock();
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}
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if(m_config.m_squelch != m_running.m_squelch) {
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qDebug() << "WFMDemod::handleMessage: set m_squelchLevel";
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m_squelchLevel = pow(10.0, m_config.m_squelch / 20.0);
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m_squelchLevel *= m_squelchLevel;
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}
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m_running.m_inputSampleRate = m_config.m_inputSampleRate;
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m_running.m_inputFrequencyOffset = m_config.m_inputFrequencyOffset;
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m_running.m_rfBandwidth = m_config.m_rfBandwidth;
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m_running.m_afBandwidth = m_config.m_afBandwidth;
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m_running.m_squelch = m_config.m_squelch;
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m_running.m_volume = m_config.m_volume;
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m_running.m_audioSampleRate = m_config.m_audioSampleRate;
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}
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