/////////////////////////////////////////////////////////////////////////////////// // 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 = (Real) m_config.m_inputSampleRate / (Real) 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; }