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sdrangel/plugins/channel/bfm/bfmdemod.cpp

285 lines
8.6 KiB
C++

///////////////////////////////////////////////////////////////////////////////////
// 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 <http://www.gnu.org/licenses/>. //
///////////////////////////////////////////////////////////////////////////////////
#include <QTime>
#include <QDebug>
#include <stdio.h>
#include <complex.h>
#include "audio/audiooutput.h"
#include "dsp/dspengine.h"
#include "dsp/channelizer.h"
#include "dsp/pidcontroller.h"
#include "bfmdemod.h"
MESSAGE_CLASS_DEFINITION(BFMDemod::MsgConfigureBFMDemod, Message)
BFMDemod::BFMDemod(SampleSink* sampleSink) :
m_sampleSink(sampleSink),
m_audioFifo(4, 250000),
m_settingsMutex(QMutex::Recursive)
{
setObjectName("BFMDemod");
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);
}
BFMDemod::~BFMDemod()
{
if (m_rfFilter)
{
delete m_rfFilter;
}
DSPEngine::instance()->removeAudioSink(&m_audioFifo);
}
void BFMDemod::configure(MessageQueue* messageQueue, Real rfBandwidth, Real afBandwidth, Real volume, Real squelch)
{
Message* cmd = MsgConfigureBFMDemod::create(rfBandwidth, afBandwidth, volume, squelch);
messageQueue->push(cmd);
}
void BFMDemod::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() / 32768.0f, it->imag() / 32768.0f);
c *= m_nco.nextIQ();
rf_out = m_rfFilter->runFilt(c, &rf); // filter RF before demod
for (int i =0 ; i <rf_out; i++)
{
msq = rf[i].real()*rf[i].real() + rf[i].imag()*rf[i].imag();
m_movingAverage.feed(msq);
if(m_movingAverage.average() >= 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("BFMDemod::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("BFMDemod::feed: %u/%u tail samples written", res, m_audioBufferFill);
}
m_audioBufferFill = 0;
}
if(m_sampleSink != 0)
{
m_sampleSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), false);
}
m_sampleBuffer.clear();
m_settingsMutex.unlock();
}
void BFMDemod::start()
{
m_squelchState = 0;
m_audioFifo.clear();
m_m1Sample = 0;
}
void BFMDemod::stop()
{
}
bool BFMDemod::handleMessage(const Message& cmd)
{
qDebug() << "BFMDemod::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() << "BFMDemod::handleMessage: MsgChannelizerNotification: m_inputSampleRate: " << m_config.m_inputSampleRate
<< " m_inputFrequencyOffset: " << m_config.m_inputFrequencyOffset;
return true;
}
else if (MsgConfigureBFMDemod::match(cmd))
{
MsgConfigureBFMDemod& cfg = (MsgConfigureBFMDemod&) 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() << "BFMDemod::handleMessage: MsgConfigureBFMDemod: 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 BFMDemod::apply()
{
if((m_config.m_inputFrequencyOffset != m_running.m_inputFrequencyOffset) ||
(m_config.m_inputSampleRate != m_running.m_inputSampleRate))
{
qDebug() << "BFMDemod::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() << "BFMDemod::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() << "BFMDemod::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() << "BFMDemod::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() << "BFMDemod::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;
}