mirror of
https://github.com/f4exb/sdrangel.git
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210 lines
6.8 KiB
C++
210 lines
6.8 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2019 Edouard Griffiths, F4EXB //
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// Copyright (C) 2021 Jon Beniston, M7RCE //
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// //
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// This program is free software; you can redistribute it and/or modify //
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// it under the terms of the GNU General Public License as published by //
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// the Free Software Foundation as version 3 of the License, or //
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// (at your option) any later version. //
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// //
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// This program is distributed in the hope that it will be useful, //
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// but WITHOUT ANY WARRANTY; without even the implied warranty of //
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
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// GNU General Public License V3 for more details. //
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// //
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// You should have received a copy of the GNU General Public License //
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// along with this program. If not, see <http://www.gnu.org/licenses/>. //
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///////////////////////////////////////////////////////////////////////////////////
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#include <QDebug>
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#include <complex.h>
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#include "dsp/dspengine.h"
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#include "dsp/dspengine.h"
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#include "util/db.h"
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#include "util/stepfunctions.h"
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#include "pipes/pipeendpoint.h"
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#include "maincore.h"
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#include "aptdemod.h"
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#include "aptdemodsink.h"
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APTDemodSink::APTDemodSink(APTDemod *packetDemod) :
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m_aptDemod(packetDemod),
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m_channelSampleRate(APTDEMOD_AUDIO_SAMPLE_RATE),
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m_channelFrequencyOffset(0),
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m_magsqSum(0.0f),
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m_magsqPeak(0.0f),
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m_magsqCount(0),
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m_imageWorkerMessageQueue(nullptr),
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m_samples(nullptr)
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{
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m_magsq = 0.0;
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applySettings(m_settings, true);
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applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
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m_samplesLength = APTDEMOD_AUDIO_SAMPLE_RATE * APT_MAX_HEIGHT / 2; // APT broadcasts at 2 lines per second
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m_samples = new float[m_samplesLength];
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resetDecoder();
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}
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void APTDemodSink::resetDecoder()
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{
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m_sampleCount = 0;
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m_writeIdx = 0;
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m_readIdx = 0;
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apt_init(APTDEMOD_AUDIO_SAMPLE_RATE);
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m_row = 0;
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m_zenith = 0;
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}
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APTDemodSink::~APTDemodSink()
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{
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delete[] m_samples;
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}
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// callback from APT library to get audio samples
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static int getsamples(void *context, float *samples, int count)
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{
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APTDemodSink *sink = (APTDemodSink *)context;
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return sink->getSamples(samples, count);
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}
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int APTDemodSink::getSamples(float *samples, int count)
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{
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for (int i = 0; i < count; i++)
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{
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if ((m_sampleCount > 0) && (m_readIdx < m_samplesLength))
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{
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*samples++ = m_samples[m_readIdx++];
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m_sampleCount--;
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}
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else
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return i;
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}
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return count;
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}
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void APTDemodSink::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end)
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{
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Complex ci;
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for (SampleVector::const_iterator it = begin; it != end; ++it)
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{
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Complex c(it->real(), it->imag());
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c *= m_nco.nextIQ();
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if (m_interpolatorDistance < 1.0f) // interpolate
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{
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while (!m_interpolator.interpolate(&m_interpolatorDistanceRemain, c, &ci))
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{
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processOneSample(ci);
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m_interpolatorDistanceRemain += m_interpolatorDistance;
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}
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}
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else // decimate
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{
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if (m_interpolator.decimate(&m_interpolatorDistanceRemain, c, &ci))
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{
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processOneSample(ci);
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m_interpolatorDistanceRemain += m_interpolatorDistance;
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}
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}
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}
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// Have we enough samples to decode one line?
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// 2 lines per second
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if (m_sampleCount >= APTDEMOD_AUDIO_SAMPLE_RATE)
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{
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if (getImageWorkerMessageQueue())
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{
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float *pixels = new float[APT_PROW_WIDTH];
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apt_getpixelrow(pixels, m_row, &m_zenith, m_row == 0, getsamples, this);
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getImageWorkerMessageQueue()->push(APTDemod::MsgPixels::create(pixels, m_zenith));
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}
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m_row++;
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}
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}
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void APTDemodSink::processOneSample(Complex &ci)
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{
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Complex ca;
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// FM demodulation
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double magsqRaw;
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Real deviation;
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Real fmDemod = m_phaseDiscri.phaseDiscriminatorDelta(ci, magsqRaw, deviation);
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// Add to sample buffer, if there's space and decoding is enabled
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if ((m_writeIdx < m_samplesLength) && m_settings.m_decodeEnabled)
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{
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m_samples[m_writeIdx++] = fmDemod;
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m_sampleCount++;
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}
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// Calculate average and peak levels for level meter
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Real magsq = magsqRaw / (SDR_RX_SCALED*SDR_RX_SCALED);
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m_movingAverage(magsq);
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m_magsq = m_movingAverage.asDouble();
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m_magsqSum += magsq;
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if (magsq > m_magsqPeak)
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{
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m_magsqPeak = magsq;
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}
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m_magsqCount++;
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}
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void APTDemodSink::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
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{
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qDebug() << "APTDemodSink::applyChannelSettings:"
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<< " channelSampleRate: " << channelSampleRate
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<< " channelFrequencyOffset: " << channelFrequencyOffset;
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if ((m_channelFrequencyOffset != channelFrequencyOffset) ||
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(m_channelSampleRate != channelSampleRate) || force)
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{
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m_nco.setFreq(-channelFrequencyOffset, channelSampleRate);
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}
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if ((m_channelSampleRate != channelSampleRate) || force)
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{
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m_interpolator.create(16, channelSampleRate, m_settings.m_rfBandwidth, 2.2);
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m_interpolatorDistance = (Real) channelSampleRate / (Real) APTDEMOD_AUDIO_SAMPLE_RATE;
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m_interpolatorDistanceRemain = m_interpolatorDistance;
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}
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m_channelSampleRate = channelSampleRate;
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m_channelFrequencyOffset = channelFrequencyOffset;
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}
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void APTDemodSink::applySettings(const APTDemodSettings& settings, bool force)
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{
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qDebug() << "APTDemodSink::applySettings:"
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<< " m_rfBandwidth: " << settings.m_rfBandwidth
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<< " m_fmDeviation: " << settings.m_fmDeviation
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<< " m_decodeEnabled: " << settings.m_decodeEnabled
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<< " force: " << force;
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if ((settings.m_rfBandwidth != m_settings.m_rfBandwidth) || force)
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{
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m_interpolator.create(16, m_channelSampleRate, settings.m_rfBandwidth, 2.2);
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m_interpolatorDistance = (Real) m_channelSampleRate / (Real) APTDEMOD_AUDIO_SAMPLE_RATE;
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m_interpolatorDistanceRemain = m_interpolatorDistance;
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}
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if ((settings.m_fmDeviation != m_settings.m_fmDeviation) || force)
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{
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m_phaseDiscri.setFMScaling(APTDEMOD_AUDIO_SAMPLE_RATE / (2.0f * settings.m_fmDeviation));
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}
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m_settings = settings;
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}
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