mirror of
https://github.com/f4exb/sdrangel.git
synced 2024-11-23 16:38:37 -05:00
363 lines
12 KiB
C++
363 lines
12 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
|
|
// Copyright (C) 2019-2023 Edouard Griffiths, F4EXB <f4exb06@gmail.com> //
|
|
// Copyright (C) 2023 Jon Beniston, M7RCE <jon@beniston.com> //
|
|
// Copyright (C) 2023 Daniele Forsi <iu5hkx@gmail.com> //
|
|
// //
|
|
// 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 <QDebug>
|
|
|
|
#include <complex.h>
|
|
|
|
#include "dsp/fftfilt.h"
|
|
#include "dsp/datafifo.h"
|
|
#include "util/db.h"
|
|
#include "util/stepfunctions.h"
|
|
#include "util/messagequeue.h"
|
|
#include "maincore.h"
|
|
|
|
#include "amdemodsink.h"
|
|
|
|
AMDemodSink::AMDemodSink() :
|
|
m_channelSampleRate(48000),
|
|
m_channelFrequencyOffset(0),
|
|
m_audioSampleRate(48000),
|
|
m_squelchCount(0),
|
|
m_squelchOpen(false),
|
|
m_squelchDelayLine(9600),
|
|
m_magsqSum(0.0f),
|
|
m_magsqPeak(0.0f),
|
|
m_magsqCount(0),
|
|
m_volumeAGC(0.003),
|
|
m_syncAMAGC(12000, 0.1, 1e-2),
|
|
m_audioFifo(48000)
|
|
{
|
|
m_audioBuffer.resize(1<<14);
|
|
m_audioBufferFill = 0;
|
|
m_demodBuffer.resize(1<<12);
|
|
m_demodBufferFill = 0;
|
|
|
|
m_magsq = 0.0;
|
|
|
|
DSBFilter = new fftfilt((2.0f * m_settings.m_rfBandwidth) / m_audioSampleRate, 2 * 1024);
|
|
SSBFilter = new fftfilt(0.0f, m_settings.m_rfBandwidth / m_audioSampleRate, 1024);
|
|
m_syncAMAGC.setThresholdEnable(false);
|
|
m_syncAMAGC.resize(12000, 6000, 0.1);
|
|
|
|
m_pllFilt.create(101, m_audioSampleRate, 200.0);
|
|
m_pll.computeCoefficients(0.05, 0.707, 1000);
|
|
m_syncAMBuffIndex = 0;
|
|
|
|
applySettings(m_settings, true);
|
|
applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
|
|
}
|
|
|
|
AMDemodSink::~AMDemodSink()
|
|
{
|
|
delete DSBFilter;
|
|
delete SSBFilter;
|
|
}
|
|
|
|
void AMDemodSink::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end)
|
|
{
|
|
Complex ci;
|
|
|
|
for (SampleVector::const_iterator it = begin; it != end; ++it)
|
|
{
|
|
Complex c(it->real(), it->imag());
|
|
c *= m_nco.nextIQ();
|
|
|
|
if (m_interpolatorDistance < 1.0f) // interpolate
|
|
{
|
|
while (!m_interpolator.interpolate(&m_interpolatorDistanceRemain, c, &ci))
|
|
{
|
|
processOneSample(ci);
|
|
m_interpolatorDistanceRemain += m_interpolatorDistance;
|
|
}
|
|
}
|
|
else // decimate
|
|
{
|
|
if (m_interpolator.decimate(&m_interpolatorDistanceRemain, c, &ci))
|
|
{
|
|
processOneSample(ci);
|
|
m_interpolatorDistanceRemain += m_interpolatorDistance;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void AMDemodSink::processOneSample(Complex &ci)
|
|
{
|
|
Real re = ci.real() / SDR_RX_SCALEF;
|
|
Real im = ci.imag() / SDR_RX_SCALEF;
|
|
Real magsq = re*re + im*im;
|
|
m_movingAverage(magsq);
|
|
m_magsq = m_movingAverage.asDouble();
|
|
m_magsqSum += magsq;
|
|
|
|
if (magsq > m_magsqPeak)
|
|
{
|
|
m_magsqPeak = magsq;
|
|
}
|
|
|
|
m_magsqCount++;
|
|
|
|
m_squelchDelayLine.write(magsq);
|
|
|
|
if (m_magsq < m_squelchLevel)
|
|
{
|
|
if (m_squelchCount > 0) {
|
|
m_squelchCount--;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
if (m_squelchCount < m_audioSampleRate / 10) {
|
|
m_squelchCount++;
|
|
}
|
|
}
|
|
|
|
qint16 sample;
|
|
|
|
m_squelchOpen = (m_squelchCount >= m_audioSampleRate / 20);
|
|
|
|
if (m_squelchOpen && !m_settings.m_audioMute)
|
|
{
|
|
Real demod;
|
|
|
|
if (m_settings.m_pll)
|
|
{
|
|
std::complex<float> s(re, im);
|
|
s = m_pllFilt.filter(s);
|
|
m_pll.feed(s.real(), s.imag());
|
|
float yr = re * m_pll.getImag() - im * m_pll.getReal();
|
|
float yi = re * m_pll.getReal() + im * m_pll.getImag();
|
|
|
|
fftfilt::cmplx *sideband;
|
|
std::complex<float> cs(yr, yi);
|
|
int n_out;
|
|
|
|
if (m_settings.m_syncAMOperation == AMDemodSettings::SyncAMDSB) {
|
|
n_out = DSBFilter->runDSB(cs, &sideband, false);
|
|
} else {
|
|
n_out = SSBFilter->runSSB(cs, &sideband, m_settings.m_syncAMOperation == AMDemodSettings::SyncAMUSB, false);
|
|
}
|
|
|
|
for (int i = 0; i < n_out; i++)
|
|
{
|
|
float agcVal = m_syncAMAGC.feedAndGetValue(sideband[i]);
|
|
fftfilt::cmplx z = sideband[i] * agcVal; // * m_syncAMAGC.getStepValue();
|
|
|
|
if (m_settings.m_syncAMOperation == AMDemodSettings::SyncAMDSB) {
|
|
m_syncAMBuff[i] = (z.real() + z.imag());
|
|
} else if (m_settings.m_syncAMOperation == AMDemodSettings::SyncAMUSB) {
|
|
m_syncAMBuff[i] = (z.real() + z.imag());
|
|
} else {
|
|
m_syncAMBuff[i] = (z.real() + z.imag());
|
|
}
|
|
|
|
m_syncAMBuffIndex = 0;
|
|
}
|
|
|
|
m_syncAMBuffIndex = m_syncAMBuffIndex < 2*1024 ? m_syncAMBuffIndex : 0;
|
|
demod = m_syncAMBuff[m_syncAMBuffIndex++]*4.0f; // mos pifometrico
|
|
}
|
|
else
|
|
{
|
|
demod = sqrt(m_squelchDelayLine.readBack(m_audioSampleRate/20));
|
|
m_volumeAGC.feed(demod);
|
|
demod = (demod - m_volumeAGC.getValue()) / m_volumeAGC.getValue();
|
|
}
|
|
|
|
if (m_settings.m_bandpassEnable)
|
|
{
|
|
demod = m_bandpass.filter(demod);
|
|
}
|
|
else
|
|
{
|
|
demod = m_lowpass.filter(demod);
|
|
}
|
|
|
|
Real attack = (m_squelchCount - 0.05f * m_audioSampleRate) / (0.05f * m_audioSampleRate);
|
|
sample = demod * StepFunctions::smootherstep(attack) * (m_audioSampleRate/24) * m_settings.m_volume;
|
|
}
|
|
else
|
|
{
|
|
sample = 0;
|
|
}
|
|
|
|
m_audioBuffer[m_audioBufferFill].l = sample;
|
|
m_audioBuffer[m_audioBufferFill].r = sample;
|
|
++m_audioBufferFill;
|
|
|
|
if (m_audioBufferFill >= m_audioBuffer.size())
|
|
{
|
|
std::size_t res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], std::min(m_audioBufferFill, m_audioBuffer.size()));
|
|
|
|
if (res != m_audioBufferFill)
|
|
{
|
|
qDebug("AMDemodSink::processOneSample: %lu/%lu audio samples written", res, m_audioBufferFill);
|
|
m_audioFifo.clear();
|
|
}
|
|
|
|
m_audioBufferFill = 0;
|
|
}
|
|
|
|
m_demodBuffer[m_demodBufferFill] = sample;
|
|
++m_demodBufferFill;
|
|
|
|
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::DataTypeI16);
|
|
}
|
|
}
|
|
}
|
|
|
|
m_demodBufferFill = 0;
|
|
}
|
|
}
|
|
|
|
void AMDemodSink::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
|
|
{
|
|
qDebug() << "AMDemodSink::applyChannelSettings:"
|
|
<< " channelSampleRate: " << channelSampleRate
|
|
<< " channelFrequencyOffset: " << channelFrequencyOffset
|
|
<< " m_audioSampleRate: " << m_audioSampleRate;
|
|
|
|
if ((m_channelFrequencyOffset != channelFrequencyOffset) ||
|
|
(m_channelSampleRate != channelSampleRate) || force)
|
|
{
|
|
m_nco.setFreq(-channelFrequencyOffset, channelSampleRate);
|
|
}
|
|
|
|
if ((m_channelSampleRate != channelSampleRate) || force)
|
|
{
|
|
m_interpolator.create(16, channelSampleRate, m_settings.m_rfBandwidth / 2.2f);
|
|
m_interpolatorDistanceRemain = 0;
|
|
m_interpolatorDistance = (Real) channelSampleRate / (Real) m_audioSampleRate;
|
|
}
|
|
|
|
m_channelSampleRate = channelSampleRate;
|
|
m_channelFrequencyOffset = channelFrequencyOffset;
|
|
}
|
|
|
|
void AMDemodSink::applySettings(const AMDemodSettings& settings, bool force)
|
|
{
|
|
qDebug() << "AMDemodSink::applySettings:"
|
|
<< " m_inputFrequencyOffset: " << settings.m_inputFrequencyOffset
|
|
<< " m_rfBandwidth: " << settings.m_rfBandwidth
|
|
<< " m_volume: " << settings.m_volume
|
|
<< " m_squelch: " << settings.m_squelch
|
|
<< " m_audioMute: " << settings.m_audioMute
|
|
<< " m_bandpassEnable: " << settings.m_bandpassEnable
|
|
<< " m_afBandwidth: " << settings.m_afBandwidth
|
|
<< " m_audioDeviceName: " << settings.m_audioDeviceName
|
|
<< " m_pll: " << settings.m_pll
|
|
<< " m_syncAMOperation: " << (int) settings.m_syncAMOperation
|
|
<< " force: " << force;
|
|
|
|
if((m_settings.m_rfBandwidth != settings.m_rfBandwidth) ||
|
|
(m_settings.m_bandpassEnable != settings.m_bandpassEnable) ||
|
|
(m_settings.m_afBandwidth != settings.m_afBandwidth) || force)
|
|
{
|
|
m_interpolator.create(16, m_channelSampleRate, settings.m_rfBandwidth / 2.2f);
|
|
m_interpolatorDistanceRemain = 0;
|
|
m_interpolatorDistance = (Real) m_channelSampleRate / (Real) m_audioSampleRate;
|
|
m_bandpass.create(301, m_audioSampleRate, 300.0, settings.m_afBandwidth / 2.0f);
|
|
m_lowpass.create(301, m_audioSampleRate, settings.m_afBandwidth / 2.0f);
|
|
DSBFilter->create_dsb_filter((2.0f * settings.m_rfBandwidth) / (float) m_audioSampleRate);
|
|
}
|
|
|
|
if ((m_settings.m_squelch != settings.m_squelch) || force) {
|
|
m_squelchLevel = CalcDb::powerFromdB(settings.m_squelch);
|
|
}
|
|
|
|
if ((m_settings.m_pll != settings.m_pll) || force)
|
|
{
|
|
if (settings.m_pll)
|
|
{
|
|
m_volumeAGC.resizeNew(m_audioSampleRate/4, 0.003);
|
|
m_syncAMBuffIndex = 0;
|
|
}
|
|
else
|
|
{
|
|
m_volumeAGC.resizeNew(m_audioSampleRate/10, 0.003);
|
|
}
|
|
}
|
|
|
|
if ((m_settings.m_syncAMOperation != settings.m_syncAMOperation) || force) {
|
|
m_syncAMBuffIndex = 0;
|
|
}
|
|
|
|
m_settings = settings;
|
|
}
|
|
|
|
void AMDemodSink::applyAudioSampleRate(int sampleRate)
|
|
{
|
|
if (sampleRate < 0)
|
|
{
|
|
qWarning("AMDemodSink::applyAudioSampleRate: invalid sample rate: %d", sampleRate);
|
|
return;
|
|
}
|
|
|
|
qDebug("AMDemodSink::applyAudioSampleRate: sampleRate: %d m_channelSampleRate: %d", sampleRate, m_channelSampleRate);
|
|
|
|
m_interpolator.create(16, m_channelSampleRate, m_settings.m_rfBandwidth / 2.2f);
|
|
m_interpolatorDistanceRemain = 0;
|
|
m_interpolatorDistance = (Real) m_channelSampleRate / (Real) sampleRate;
|
|
m_bandpass.create(301, sampleRate, 300.0, m_settings.m_afBandwidth / 2.0f);
|
|
m_lowpass.create(301, sampleRate, m_settings.m_afBandwidth / 2.0f);
|
|
m_audioFifo.setSize(sampleRate);
|
|
m_squelchDelayLine.resize(sampleRate/5);
|
|
DSBFilter->create_dsb_filter((2.0f * m_settings.m_rfBandwidth) / (float) sampleRate);
|
|
m_pllFilt.create(101, sampleRate, 200.0);
|
|
|
|
if (m_settings.m_pll) {
|
|
m_volumeAGC.resizeNew(sampleRate, 0.003);
|
|
} else {
|
|
m_volumeAGC.resizeNew(sampleRate/10, 0.003);
|
|
}
|
|
|
|
m_syncAMAGC.resize(sampleRate/4, sampleRate/8, 0.1);
|
|
m_pll.setSampleRate(sampleRate);
|
|
|
|
QList<ObjectPipe*> pipes;
|
|
MainCore::instance()->getMessagePipes().getMessagePipes(m_channel, "reportdemod", pipes);
|
|
|
|
if (pipes.size() > 0)
|
|
{
|
|
for (const auto& pipe : pipes)
|
|
{
|
|
MessageQueue *messageQueue = qobject_cast<MessageQueue*>(pipe->m_element);
|
|
MainCore::MsgChannelDemodReport *msg = MainCore::MsgChannelDemodReport::create(m_channel, sampleRate);
|
|
messageQueue->push(msg);
|
|
}
|
|
}
|
|
|
|
m_audioSampleRate = sampleRate;
|
|
}
|