mirror of
https://github.com/f4exb/sdrangel.git
synced 2024-11-04 16:01:14 -05:00
389 lines
14 KiB
C++
389 lines
14 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
|
|
// Copyright (C) 2019 Edouard Griffiths, F4EXB //
|
|
// //
|
|
// 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 //
|
|
// (at your option) any later version. //
|
|
// //
|
|
// 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 "boost/format.hpp"
|
|
#include <stdio.h>
|
|
#include <complex.h>
|
|
|
|
#include <QTime>
|
|
#include <QDebug>
|
|
|
|
#include "audio/audiooutputdevice.h"
|
|
#include "dsp/dspengine.h"
|
|
#include "dsp/dspcommands.h"
|
|
#include "dsp/devicesamplemimo.h"
|
|
#include "dsp/basebandsamplesink.h"
|
|
#include "dsp/datafifo.h"
|
|
#include "pipes/datapipes.h"
|
|
#include "util/db.h"
|
|
#include "maincore.h"
|
|
|
|
#include "rdsparser.h"
|
|
#include "bfmdemodsink.h"
|
|
|
|
const Real BFMDemodSink::default_deemphasis = 50.0; // 50 us
|
|
const int BFMDemodSink::default_excursion = 750000; // +/- 75 kHz
|
|
|
|
BFMDemodSink::BFMDemodSink() :
|
|
m_channel(nullptr),
|
|
m_channelSampleRate(48000),
|
|
m_channelFrequencyOffset(0),
|
|
m_audioSampleRate(48000),
|
|
m_audioBufferFill(0),
|
|
m_audioFifo(48000),
|
|
m_pilotPLL(19000/384000, 50/384000, 0.01),
|
|
m_deemphasisFilterX(default_deemphasis * 48000 * 1.0e-6),
|
|
m_deemphasisFilterY(default_deemphasis * 48000 * 1.0e-6),
|
|
m_fmExcursion(default_excursion)
|
|
{
|
|
m_magsq = 0.0f;
|
|
m_magsqSum = 0.0f;
|
|
m_magsqPeak = 0.0f;
|
|
m_magsqCount = 0;
|
|
|
|
m_squelchLevel = 0;
|
|
m_squelchState = 0;
|
|
|
|
m_interpolatorDistance = 0.0f;
|
|
m_interpolatorDistanceRemain = 0.0f;
|
|
|
|
m_interpolatorRDSDistance = 0.0f;
|
|
m_interpolatorRDSDistanceRemain = 0.0f;
|
|
|
|
m_interpolatorStereoDistance = 0.0f;
|
|
m_interpolatorStereoDistanceRemain = 0.0f;
|
|
|
|
m_spectrumSink = nullptr;
|
|
m_m1Arg = 0;
|
|
|
|
m_rfFilter = new fftfilt(-50000.0 / 384000.0, 50000.0 / 384000.0, filtFftLen);
|
|
|
|
m_deemphasisFilterX.configure(default_deemphasis * m_audioSampleRate * 1.0e-6);
|
|
m_deemphasisFilterY.configure(default_deemphasis * m_audioSampleRate * 1.0e-6);
|
|
m_phaseDiscri.setFMScaling(384000/m_fmExcursion);
|
|
|
|
m_audioBuffer.resize(1<<14);
|
|
m_audioBufferFill = 0;
|
|
|
|
m_demodBuffer.resize(1<<13);
|
|
m_demodBufferFill = 0;
|
|
|
|
applySettings(m_settings, true);
|
|
applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
|
|
}
|
|
|
|
BFMDemodSink::~BFMDemodSink()
|
|
{
|
|
delete m_rfFilter;
|
|
}
|
|
|
|
void BFMDemodSink::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end)
|
|
{
|
|
Complex ci, cs, cr;
|
|
fftfilt::cmplx *rf;
|
|
int rf_out;
|
|
double msq;
|
|
Real demod;
|
|
|
|
m_sampleBuffer.clear();
|
|
|
|
for (SampleVector::const_iterator it = begin; it != end; ++it)
|
|
{
|
|
Complex c(it->real() / SDR_RX_SCALEF, it->imag() / SDR_RX_SCALEF);
|
|
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_magsqSum += msq;
|
|
|
|
if (msq > m_magsqPeak) {
|
|
m_magsqPeak = msq;
|
|
}
|
|
|
|
m_magsqCount++;
|
|
|
|
if (msq >= m_squelchLevel)
|
|
{
|
|
if (m_squelchState < m_settings.m_rfBandwidth / 10) { // twice attack and decay rate
|
|
m_squelchState++;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
if (m_squelchState > 0) {
|
|
m_squelchState--;
|
|
}
|
|
}
|
|
|
|
if (m_squelchState > m_settings.m_rfBandwidth / 20) { // squelch open
|
|
demod = m_phaseDiscri.phaseDiscriminator(rf[i]);
|
|
} else {
|
|
demod = 0;
|
|
}
|
|
|
|
if (!m_settings.m_showPilot) {
|
|
m_sampleBuffer.push_back(Sample(demod * SDR_RX_SCALEF, 0.0));
|
|
}
|
|
|
|
if (m_settings.m_rdsActive)
|
|
{
|
|
//Complex r(demod * 2.0 * std::cos(3.0 * m_pilotPLLSamples[3]), 0.0);
|
|
Complex r(demod * 2.0 * std::cos(3.0 * m_pilotPLLSamples[3]), 0.0);
|
|
|
|
if (m_interpolatorRDS.decimate(&m_interpolatorRDSDistanceRemain, r, &cr))
|
|
{
|
|
bool bit;
|
|
|
|
if (m_rdsDemod.process(cr.real(), bit))
|
|
{
|
|
if (m_rdsDecoder.frameSync(bit)) {
|
|
m_rdsParser.parseGroup(m_rdsDecoder.getGroup());
|
|
}
|
|
}
|
|
|
|
m_interpolatorRDSDistanceRemain += m_interpolatorRDSDistance;
|
|
}
|
|
}
|
|
|
|
Real sampleStereo = 0.0f;
|
|
|
|
// Process stereo if stereo mode is selected
|
|
|
|
if (m_settings.m_audioStereo)
|
|
{
|
|
m_pilotPLL.process(demod, m_pilotPLLSamples);
|
|
|
|
if (m_settings.m_showPilot) {
|
|
m_sampleBuffer.push_back(Sample(m_pilotPLLSamples[1] * SDR_RX_SCALEF, 0.0)); // debug 38 kHz pilot
|
|
}
|
|
|
|
if (m_settings.m_lsbStereo)
|
|
{
|
|
// 1.17 * 0.7 = 0.819
|
|
Complex s(demod * m_pilotPLLSamples[1], demod * m_pilotPLLSamples[2]);
|
|
|
|
if (m_interpolatorStereo.decimate(&m_interpolatorStereoDistanceRemain, s, &cs))
|
|
{
|
|
sampleStereo = cs.real() + cs.imag();
|
|
m_interpolatorStereoDistanceRemain += m_interpolatorStereoDistance;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
Complex s(demod * 1.17 * m_pilotPLLSamples[1], 0);
|
|
|
|
if (m_interpolatorStereo.decimate(&m_interpolatorStereoDistanceRemain, s, &cs))
|
|
{
|
|
sampleStereo = cs.real();
|
|
m_interpolatorStereoDistanceRemain += m_interpolatorStereoDistance;
|
|
}
|
|
}
|
|
}
|
|
|
|
Complex e(demod, 0);
|
|
|
|
if (m_interpolator.decimate(&m_interpolatorDistanceRemain, e, &ci))
|
|
{
|
|
if (m_settings.m_audioStereo)
|
|
{
|
|
Real deemph_l, deemph_r; // Pre-emphasis is applied on each channel before multiplexing
|
|
m_deemphasisFilterX.process(ci.real() + sampleStereo, deemph_l);
|
|
m_deemphasisFilterY.process(ci.real() - sampleStereo, deemph_r);
|
|
m_audioBuffer[m_audioBufferFill].l = (qint16)(deemph_l * (1<<12) * m_settings.m_volume);
|
|
m_audioBuffer[m_audioBufferFill].r = (qint16)(deemph_r * (1<<12) * m_settings.m_volume);
|
|
}
|
|
else
|
|
{
|
|
Real deemph;
|
|
m_deemphasisFilterX.process(ci.real(), deemph);
|
|
quint16 sample = (qint16)(deemph * (1<<12) * m_settings.m_volume);
|
|
m_audioBuffer[m_audioBufferFill].l = sample;
|
|
m_audioBuffer[m_audioBufferFill].r = sample;
|
|
}
|
|
|
|
m_demodBuffer[m_demodBufferFill++] = m_audioBuffer[m_audioBufferFill].l;
|
|
m_demodBuffer[m_demodBufferFill++] = m_audioBuffer[m_audioBufferFill].r;
|
|
|
|
++m_audioBufferFill;
|
|
|
|
if (m_audioBufferFill >= m_audioBuffer.size())
|
|
{
|
|
if (m_audioBufferFill > m_audioBuffer.size()) {
|
|
qDebug("BFMDemodSink::processOneSample: dropping %d samples", (int) m_audioBufferFill - (int) m_audioBuffer.size());
|
|
}
|
|
|
|
uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBuffer.size());
|
|
|
|
if(res != m_audioBuffer.size()) {
|
|
qDebug("BFMDemodSink::feed: %u/%lu audio samples written", res, m_audioBuffer.size());
|
|
}
|
|
|
|
m_audioBufferFill = 0;
|
|
}
|
|
|
|
if (m_demodBufferFill >= m_demodBuffer.size())
|
|
{
|
|
QList<ObjectPipe*> dataPipes;
|
|
MainCore::instance()->getDataPipes().getDataPipes(m_channel, "demod", dataPipes);
|
|
|
|
if (dataPipes.size() > 0)
|
|
{
|
|
QList<ObjectPipe*>::iterator it = dataPipes.begin();
|
|
|
|
for (; it != dataPipes.end(); ++it)
|
|
{
|
|
DataFifo *fifo = qobject_cast<DataFifo*>((*it)->m_element);
|
|
|
|
if (fifo) {
|
|
fifo->write((quint8*) &m_demodBuffer[0], m_demodBuffer.size() * sizeof(qint16), DataFifo::DataTypeCI16);
|
|
}
|
|
}
|
|
}
|
|
|
|
m_demodBufferFill = 0;
|
|
}
|
|
|
|
m_interpolatorDistanceRemain += m_interpolatorDistance;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (m_spectrumSink && (m_sampleBuffer.size() != 0))
|
|
{
|
|
m_spectrumSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), true);
|
|
m_sampleBuffer.clear();
|
|
}
|
|
}
|
|
|
|
void BFMDemodSink::applyAudioSampleRate(int sampleRate)
|
|
{
|
|
if (sampleRate < 0)
|
|
{
|
|
qWarning("BFMDemodSink::applyAudioSampleRate: invalid sample rate: %d", sampleRate);
|
|
return;
|
|
}
|
|
|
|
qDebug("BFMDemodSink::applyAudioSampleRate: %u", sampleRate);
|
|
|
|
m_interpolator.create(16, m_channelSampleRate, m_settings.m_afBandwidth);
|
|
m_interpolatorDistanceRemain = (Real) m_channelSampleRate / sampleRate;
|
|
m_interpolatorDistance = (Real) m_channelSampleRate / (Real) sampleRate;
|
|
|
|
m_interpolatorStereo.create(16, m_channelSampleRate, m_settings.m_afBandwidth);
|
|
m_interpolatorStereoDistanceRemain = (Real) m_channelSampleRate / sampleRate;
|
|
m_interpolatorStereoDistance = (Real) m_channelSampleRate / (Real) sampleRate;
|
|
|
|
m_deemphasisFilterX.configure(default_deemphasis * sampleRate * 1.0e-6);
|
|
m_deemphasisFilterY.configure(default_deemphasis * sampleRate * 1.0e-6);
|
|
|
|
m_audioSampleRate = sampleRate;
|
|
}
|
|
|
|
void BFMDemodSink::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
|
|
{
|
|
qDebug() << "BFMDemodSink::applyChannelSettings:"
|
|
<< " channelSampleRate: " << channelSampleRate
|
|
<< " channelFrequencyOffset: " << channelFrequencyOffset;
|
|
|
|
if((channelFrequencyOffset != m_channelFrequencyOffset) ||
|
|
(channelSampleRate != m_channelSampleRate) || force)
|
|
{
|
|
m_nco.setFreq(-channelFrequencyOffset, channelSampleRate);
|
|
}
|
|
|
|
if ((channelSampleRate != m_channelSampleRate) || force)
|
|
{
|
|
m_pilotPLL.configure(19000.0/channelSampleRate, 50.0/channelSampleRate, 0.01);
|
|
|
|
m_interpolator.create(16, channelSampleRate, m_settings.m_afBandwidth);
|
|
m_interpolatorDistanceRemain = (Real) channelSampleRate / m_audioSampleRate;
|
|
m_interpolatorDistance = (Real) channelSampleRate / (Real) m_audioSampleRate;
|
|
|
|
m_interpolatorStereo.create(16, channelSampleRate, m_settings.m_afBandwidth);
|
|
m_interpolatorStereoDistanceRemain = (Real) channelSampleRate / m_audioSampleRate;
|
|
m_interpolatorStereoDistance = (Real) channelSampleRate / (Real) m_audioSampleRate;
|
|
|
|
m_interpolatorRDS.create(4, channelSampleRate, 600.0);
|
|
m_interpolatorRDSDistanceRemain = (Real) channelSampleRate / 250000.0;
|
|
m_interpolatorRDSDistance = (Real) channelSampleRate / 250000.0;
|
|
|
|
Real lowCut = -(m_settings.m_rfBandwidth / 2.0) / channelSampleRate;
|
|
Real hiCut = (m_settings.m_rfBandwidth / 2.0) / channelSampleRate;
|
|
m_rfFilter->create_filter(lowCut, hiCut);
|
|
m_phaseDiscri.setFMScaling(channelSampleRate / m_fmExcursion);
|
|
}
|
|
|
|
m_channelSampleRate = channelSampleRate;
|
|
m_channelFrequencyOffset = channelFrequencyOffset;
|
|
}
|
|
|
|
void BFMDemodSink::applySettings(const BFMDemodSettings& settings, bool force)
|
|
{
|
|
qDebug() << "BFMDemodSink::applySettings: MsgConfigureBFMDemod:"
|
|
<< " m_inputFrequencyOffset: " << settings.m_inputFrequencyOffset
|
|
<< " m_rfBandwidth: " << settings.m_rfBandwidth
|
|
<< " m_afBandwidth: " << settings.m_afBandwidth
|
|
<< " m_volume: " << settings.m_volume
|
|
<< " m_squelch: " << settings.m_squelch
|
|
<< " m_audioStereo: " << settings.m_audioStereo
|
|
<< " m_lsbStereo: " << settings.m_lsbStereo
|
|
<< " m_showPilot: " << settings.m_showPilot
|
|
<< " m_rdsActive: " << settings.m_rdsActive
|
|
<< " m_audioDeviceName: " << settings.m_audioDeviceName
|
|
<< " m_streamIndex: " << settings.m_streamIndex
|
|
<< " m_useReverseAPI: " << settings.m_useReverseAPI
|
|
<< " force: " << force;
|
|
|
|
if ((settings.m_audioStereo && (settings.m_audioStereo != m_settings.m_audioStereo)) || force) {
|
|
m_pilotPLL.configure(19000.0/m_channelSampleRate, 50.0/m_channelSampleRate, 0.01);
|
|
}
|
|
|
|
if ((settings.m_afBandwidth != m_settings.m_afBandwidth) || force)
|
|
{
|
|
m_interpolator.create(16, m_channelSampleRate, settings.m_afBandwidth);
|
|
m_interpolatorDistanceRemain = (Real) m_channelSampleRate / m_audioSampleRate;
|
|
m_interpolatorDistance = (Real) m_channelSampleRate / (Real) m_audioSampleRate;
|
|
|
|
m_interpolatorStereo.create(16, m_channelSampleRate, settings.m_afBandwidth);
|
|
m_interpolatorStereoDistanceRemain = (Real) m_channelSampleRate / m_audioSampleRate;
|
|
m_interpolatorStereoDistance = (Real) m_channelSampleRate / (Real) m_audioSampleRate;
|
|
|
|
m_interpolatorRDS.create(4, m_channelSampleRate, 600.0);
|
|
m_interpolatorRDSDistanceRemain = (Real) m_channelSampleRate / 250000.0;
|
|
m_interpolatorRDSDistance = (Real) m_channelSampleRate / 250000.0;
|
|
|
|
m_lowpass.create(21, m_audioSampleRate, settings.m_afBandwidth);
|
|
}
|
|
|
|
if ((settings.m_rfBandwidth != m_settings.m_rfBandwidth) || force)
|
|
{
|
|
Real lowCut = -(settings.m_rfBandwidth / 2.0) / m_channelSampleRate;
|
|
Real hiCut = (settings.m_rfBandwidth / 2.0) / m_channelSampleRate;
|
|
m_rfFilter->create_filter(lowCut, hiCut);
|
|
m_phaseDiscri.setFMScaling(m_channelSampleRate / m_fmExcursion);
|
|
}
|
|
|
|
if ((settings.m_squelch != m_settings.m_squelch) || force) {
|
|
m_squelchLevel = std::pow(10.0, settings.m_squelch / 10.0);
|
|
}
|
|
|
|
m_settings = settings;
|
|
}
|