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sdrangel/plugins/channelrx/demodapt/aptdemodsink.cpp

210 lines
6.8 KiB
C++

///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2019 Edouard Griffiths, F4EXB //
// Copyright (C) 2021 Jon Beniston, M7RCE //
// //
// 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/dspengine.h"
#include "dsp/dspengine.h"
#include "util/db.h"
#include "util/stepfunctions.h"
#include "pipes/pipeendpoint.h"
#include "maincore.h"
#include "aptdemod.h"
#include "aptdemodsink.h"
APTDemodSink::APTDemodSink(APTDemod *packetDemod) :
m_aptDemod(packetDemod),
m_channelSampleRate(APTDEMOD_AUDIO_SAMPLE_RATE),
m_channelFrequencyOffset(0),
m_magsqSum(0.0f),
m_magsqPeak(0.0f),
m_magsqCount(0),
m_imageWorkerMessageQueue(nullptr),
m_samples(nullptr)
{
m_magsq = 0.0;
applySettings(m_settings, true);
applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
m_samplesLength = APTDEMOD_AUDIO_SAMPLE_RATE * APT_MAX_HEIGHT / 2; // APT broadcasts at 2 lines per second
m_samples = new float[m_samplesLength];
resetDecoder();
}
void APTDemodSink::resetDecoder()
{
m_sampleCount = 0;
m_writeIdx = 0;
m_readIdx = 0;
apt_init(APTDEMOD_AUDIO_SAMPLE_RATE);
m_row = 0;
m_zenith = 0;
}
APTDemodSink::~APTDemodSink()
{
delete[] m_samples;
}
// callback from APT library to get audio samples
static int getsamples(void *context, float *samples, int count)
{
APTDemodSink *sink = (APTDemodSink *)context;
return sink->getSamples(samples, count);
}
int APTDemodSink::getSamples(float *samples, int count)
{
for (int i = 0; i < count; i++)
{
if ((m_sampleCount > 0) && (m_readIdx < m_samplesLength))
{
*samples++ = m_samples[m_readIdx++];
m_sampleCount--;
}
else
return i;
}
return count;
}
void APTDemodSink::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;
}
}
}
// Have we enough samples to decode one line?
// 2 lines per second
if (m_sampleCount >= APTDEMOD_AUDIO_SAMPLE_RATE)
{
if (getImageWorkerMessageQueue())
{
float *pixels = new float[APT_PROW_WIDTH];
apt_getpixelrow(pixels, m_row, &m_zenith, m_row == 0, getsamples, this);
getImageWorkerMessageQueue()->push(APTDemod::MsgPixels::create(pixels, m_zenith));
}
m_row++;
}
}
void APTDemodSink::processOneSample(Complex &ci)
{
Complex ca;
// FM demodulation
double magsqRaw;
Real deviation;
Real fmDemod = m_phaseDiscri.phaseDiscriminatorDelta(ci, magsqRaw, deviation);
// Add to sample buffer, if there's space and decoding is enabled
if ((m_writeIdx < m_samplesLength) && m_settings.m_decodeEnabled)
{
m_samples[m_writeIdx++] = fmDemod;
m_sampleCount++;
}
// Calculate average and peak levels for level meter
Real magsq = magsqRaw / (SDR_RX_SCALED*SDR_RX_SCALED);
m_movingAverage(magsq);
m_magsq = m_movingAverage.asDouble();
m_magsqSum += magsq;
if (magsq > m_magsqPeak)
{
m_magsqPeak = magsq;
}
m_magsqCount++;
}
void APTDemodSink::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
{
qDebug() << "APTDemodSink::applyChannelSettings:"
<< " channelSampleRate: " << channelSampleRate
<< " channelFrequencyOffset: " << channelFrequencyOffset;
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, 2.2);
m_interpolatorDistance = (Real) channelSampleRate / (Real) APTDEMOD_AUDIO_SAMPLE_RATE;
m_interpolatorDistanceRemain = m_interpolatorDistance;
}
m_channelSampleRate = channelSampleRate;
m_channelFrequencyOffset = channelFrequencyOffset;
}
void APTDemodSink::applySettings(const APTDemodSettings& settings, bool force)
{
qDebug() << "APTDemodSink::applySettings:"
<< " m_rfBandwidth: " << settings.m_rfBandwidth
<< " m_fmDeviation: " << settings.m_fmDeviation
<< " m_decodeEnabled: " << settings.m_decodeEnabled
<< " force: " << force;
if ((settings.m_rfBandwidth != m_settings.m_rfBandwidth) || force)
{
m_interpolator.create(16, m_channelSampleRate, settings.m_rfBandwidth, 2.2);
m_interpolatorDistance = (Real) m_channelSampleRate / (Real) APTDEMOD_AUDIO_SAMPLE_RATE;
m_interpolatorDistanceRemain = m_interpolatorDistance;
}
if ((settings.m_fmDeviation != m_settings.m_fmDeviation) || force)
{
m_phaseDiscri.setFMScaling(APTDEMOD_AUDIO_SAMPLE_RATE / (2.0f * settings.m_fmDeviation));
}
m_settings = settings;
}