///////////////////////////////////////////////////////////////////////////////////
// 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 "nfmdemod.h"
#include "audio/audiooutput.h"
#include "dsp/dspcommands.h"
#include "dsp/pidcontroller.h"
MESSAGE_CLASS_DEFINITION(NFMDemod::MsgConfigureNFMDemod, Message)
NFMDemod::NFMDemod(AudioFifo* audioFifo, SampleSink* sampleSink) :
m_sampleSink(sampleSink),
m_audioFifo(audioFifo)
{
m_config.m_inputSampleRate = 96000;
m_config.m_inputFrequencyOffset = 0;
m_config.m_rfBandwidth = 12500;
m_config.m_afBandwidth = 3000;
m_config.m_squelch = -40.0;
m_config.m_volume = 2.0;
m_config.m_audioSampleRate = 48000;
apply();
m_audioBuffer.resize(16384);
m_audioBufferFill = 0;
m_movingAverage.resize(16, 0);
m_agcLevel = 0.003;
m_AGC.resize(4096, m_agcLevel, 0, 0.1*m_agcLevel);
}
NFMDemod::~NFMDemod()
{
}
void NFMDemod::configure(MessageQueue* messageQueue, Real rfBandwidth, Real afBandwidth, Real volume, Real squelch)
{
Message* cmd = MsgConfigureNFMDemod::create(rfBandwidth, afBandwidth, volume, squelch);
cmd->submit(messageQueue, this);
}
float arctan2(Real y, Real x)
{
Real coeff_1 = M_PI / 4;
Real coeff_2 = 3 * coeff_1;
Real abs_y = fabs(y) + 1e-10; // kludge to prevent 0/0 condition
Real angle;
if( x>= 0) {
Real r = (x - abs_y) / (x + abs_y);
angle = coeff_1 - coeff_1 * r;
} else {
Real r = (x + abs_y) / (abs_y - x);
angle = coeff_2 - coeff_1 * r;
}
if(y < 0)
return(-angle);
else return(angle);
}
Real angleDist(Real a, Real b)
{
Real dist = b - a;
while(dist <= M_PI)
dist += 2 * M_PI;
while(dist >= M_PI)
dist -= 2 * M_PI;
return dist;
}
void NFMDemod::feed(SampleVector::const_iterator begin, SampleVector::const_iterator end, bool firstOfBurst)
{
Complex ci;
if(m_audioFifo->size() <= 0)
return;
for(SampleVector::const_iterator it = begin; it != end; ++it) {
Complex c(it->real() / 32768.0, it->imag() / 32768.0);
c *= m_nco.nextIQ();
{
if(m_interpolator.interpolate(&m_interpolatorDistanceRemain, c, &ci)) {
m_sampleBuffer.push_back(Sample(ci.real() * 32767.0, ci.imag() * 32767.0));
m_movingAverage.feed(ci.real() * ci.real() + ci.imag() * ci.imag());
if(m_movingAverage.average() >= m_squelchLevel)
m_squelchState = m_running.m_audioSampleRate/ 20;
qint16 sample;
if(m_squelchState > 0) {
m_squelchState--;
m_AGC.feed(abs(ci));
ci *= (m_agcLevel / m_AGC.getValue());
// demod
/*
Real argument = arg(ci);
Real demod = argument - m_lastArgument;
m_lastArgument = argument;
*/
/*
// Original NFM
Complex d = conj(m_m1Sample) * ci;
Real demod = atan2(d.imag(), d.real());
demod /= M_PI;
*/
/*
Real argument1 = arg(ci);//atan2(ci.imag(), ci.real());
Real argument2 = m_lastSample.real();
Real demod = angleDist(argument2, argument1);
m_lastSample = Complex(argument1, 0);
*/
// Alternative without atan - needs AGC
// http://www.embedded.com/design/configurable-systems/4212086/DSP-Tricks--Frequency-demodulation-algorithms-
Real ip = ci.real() - m_m2Sample.real();
Real qp = ci.imag() - m_m2Sample.imag();
Real h1 = m_m1Sample.real() * qp;
Real h2 = m_m1Sample.imag() * ip;
Real demod = (h1 - h2) * 10000;
m_m2Sample = m_m1Sample;
m_m1Sample = ci;
// AF processing
demod = m_lowpass.filter(demod);
/*
if(demod < -1)
demod = -1;
else if(demod > 1)
demod = 1;
*/
demod *= m_running.m_volume;
sample = demod * 32700;
} else {
m_AGC.close();
sample = 0;
}
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("lost %u audio samples", m_audioBufferFill - res);
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("lost %u samples", m_audioBufferFill - res);
m_audioBufferFill = 0;
}
if(m_sampleSink != NULL)
m_sampleSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), false);
m_sampleBuffer.clear();
}
void NFMDemod::start()
{
m_squelchState = 0;
m_audioFifo->clear();
m_interpolatorRegulation = 0.9999;
m_interpolatorDistance = 1.0;
m_interpolatorDistanceRemain = 0.0;
m_m1Sample = 0;
}
void NFMDemod::stop()
{
}
bool NFMDemod::handleMessage(Message* cmd)
{
if(DSPSignalNotification::match(cmd)) {
DSPSignalNotification* signal = (DSPSignalNotification*)cmd;
m_config.m_inputSampleRate = signal->getSampleRate();
m_config.m_inputFrequencyOffset = signal->getFrequencyOffset();
apply();
cmd->completed();
return true;
} else if(MsgConfigureNFMDemod::match(cmd)) {
MsgConfigureNFMDemod* cfg = (MsgConfigureNFMDemod*)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();
return true;
} else {
if(m_sampleSink != NULL)
return m_sampleSink->handleMessage(cmd);
else return false;
}
}
void NFMDemod::apply()
{
if((m_config.m_inputFrequencyOffset != m_running.m_inputFrequencyOffset) ||
(m_config.m_inputSampleRate != m_running.m_inputSampleRate)) {
m_nco.setFreq(-m_config.m_inputFrequencyOffset, m_config.m_inputSampleRate);
}
if((m_config.m_inputSampleRate != m_running.m_inputSampleRate) ||
(m_config.m_rfBandwidth != m_running.m_rfBandwidth)) {
m_interpolator.create(16, m_config.m_inputSampleRate, m_config.m_rfBandwidth / 2.2);
m_interpolatorDistanceRemain = 0;
m_interpolatorDistance = m_config.m_inputSampleRate / m_config.m_audioSampleRate;
}
if((m_config.m_afBandwidth != m_running.m_afBandwidth) ||
(m_config.m_audioSampleRate != m_running.m_audioSampleRate)) {
m_lowpass.create(21, m_config.m_audioSampleRate, m_config.m_afBandwidth);
}
if(m_config.m_squelch != m_running.m_squelch) {
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_squelch = m_config.m_squelch;
m_running.m_volume = m_config.m_volume;
m_running.m_audioSampleRate = m_config.m_audioSampleRate;
}