mirror of
https://github.com/f4exb/sdrangel.git
synced 2024-11-22 16:08:39 -05:00
201 lines
6.6 KiB
C++
201 lines
6.6 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 "aptdemod.h"
|
|
#include "aptdemodsink.h"
|
|
|
|
APTDemodSink::APTDemodSink() :
|
|
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)
|
|
{
|
|
// 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;
|
|
}
|