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sdrangel/plugins/channel/ssb/ssbdemod.cpp

238 lines
6.4 KiB
C++

///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany //
// written by Christian Daniel //
// (c) 2014 Modified by John Greb
// //
// 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 //
// //
// 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 <QTime>
#include <QDebug>
#include <stdio.h>
#include "ssbdemod.h"
#include "audio/audiooutput.h"
#include "dsp/dspengine.h"
#include "dsp/channelizer.h"
MESSAGE_CLASS_DEFINITION(SSBDemod::MsgConfigureSSBDemod, Message)
SSBDemod::SSBDemod(SampleSink* sampleSink) :
m_sampleSink(sampleSink),
m_audioFifo(4, 24000),
m_settingsMutex(QMutex::Recursive)
{
setObjectName("SSBDemod");
m_Bandwidth = 5000;
m_LowCutoff = 300;
m_volume = 2.0;
m_spanLog2 = 3;
m_sampleRate = 96000;
m_frequency = 0;
m_nco.setFreq(m_frequency, m_sampleRate);
m_audioSampleRate = DSPEngine::instance()->getAudioSampleRate();
m_interpolator.create(16, m_sampleRate, 5000);
m_sampleDistanceRemain = (Real) m_sampleRate / m_audioSampleRate;
m_audioBuffer.resize(1<<9);
m_audioBufferFill = 0;
m_undersampleCount = 0;
m_usb = true;
SSBFilter = new fftfilt(m_LowCutoff / m_audioSampleRate, m_Bandwidth / m_audioSampleRate, ssbFftLen);
DSPEngine::instance()->addAudioSink(&m_audioFifo);
}
SSBDemod::~SSBDemod()
{
if (SSBFilter)
{
delete SSBFilter;
}
DSPEngine::instance()->removeAudioSink(&m_audioFifo);
}
void SSBDemod::configure(MessageQueue* messageQueue, Real Bandwidth, Real LowCutoff, Real volume, int spanLog2)
{
Message* cmd = MsgConfigureSSBDemod::create(Bandwidth, LowCutoff, volume, spanLog2);
messageQueue->push(cmd);
}
void SSBDemod::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end, bool positiveOnly)
{
Complex ci;
fftfilt::cmplx *sideband, sum;
Real avg;
int n_out;
m_settingsMutex.lock();
int decim = 1<<(m_spanLog2 - 1);
unsigned char decim_mask = decim - 1; // counter LSB bit mask for decimation by 2^(m_scaleLog2 - 1)
for(SampleVector::const_iterator it = begin; it < end; ++it)
{
Complex c(it->real() / 32768.0, it->imag() / 32768.0);
c *= m_nco.nextIQ();
if(m_interpolator.interpolate(&m_sampleDistanceRemain, c, &ci))
{
n_out = SSBFilter->runSSB(ci, &sideband, m_usb);
m_sampleDistanceRemain += (Real)m_sampleRate / m_audioSampleRate;
}
else
{
n_out = 0;
}
for (int i = 0; i < n_out; i++)
{
Real demod = (sideband[i].real() + sideband[i].imag()) * 0.7 * 32768.0;
// Downsample by 2^(m_scaleLog2 - 1) for SSB band spectrum display
// smart decimation with bit gain using float arithmetic (23 bits significand)
sum += sideband[i];
if (!(m_undersampleCount++ & decim_mask))
{
avg = (sum.real() + sum.imag()) * 0.7 * 32768.0 / decim;
m_sampleBuffer.push_back(Sample(avg, 0.0));
sum.real() = 0.0;
sum.imag() = 0.0;
}
qint16 sample = (qint16)(demod * m_volume * 10);
m_audioBuffer[m_audioBufferFill].l = sample;
m_audioBuffer[m_audioBufferFill].r = sample;
++m_audioBufferFill;
if (m_audioBufferFill >= m_audioBuffer.size())
{
uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 1);
if (res != m_audioBufferFill)
{
qDebug("lost %u samples", m_audioBufferFill - res);
}
m_audioBufferFill = 0;
}
}
}
if (m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 0) != m_audioBufferFill)
{
qDebug("SSBDemod::feed: lost samples");
}
m_audioBufferFill = 0;
if(m_sampleSink != 0)
{
m_sampleSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), true);
}
m_sampleBuffer.clear();
m_settingsMutex.unlock();
}
void SSBDemod::start()
{
}
void SSBDemod::stop()
{
}
bool SSBDemod::handleMessage(const Message& cmd)
{
float band, lowCutoff;
qDebug() << "SSBDemod::handleMessage";
if (Channelizer::MsgChannelizerNotification::match(cmd))
{
Channelizer::MsgChannelizerNotification& notif = (Channelizer::MsgChannelizerNotification&) cmd;
m_settingsMutex.lock();
m_sampleRate = notif.getSampleRate();
m_nco.setFreq(-notif.getFrequencyOffset(), m_sampleRate);
m_interpolator.create(16, m_sampleRate, m_Bandwidth);
m_sampleDistanceRemain = m_sampleRate / m_audioSampleRate;
m_settingsMutex.unlock();
qDebug() << "SSBDemod::handleMessage: MsgChannelizerNotification: m_sampleRate: " << m_sampleRate
<< " frequencyOffset" << notif.getFrequencyOffset();
return true;
}
else if (MsgConfigureSSBDemod::match(cmd))
{
MsgConfigureSSBDemod& cfg = (MsgConfigureSSBDemod&) cmd;
m_settingsMutex.lock();
band = cfg.getBandwidth();
lowCutoff = cfg.getLoCutoff();
if (band < 0) {
band = -band;
lowCutoff = -lowCutoff;
m_usb = false;
} else
m_usb = true;
if (band < 100.0f)
{
band = 100.0f;
lowCutoff = 0;
}
m_Bandwidth = band;
m_LowCutoff = lowCutoff;
m_interpolator.create(16, m_sampleRate, band * 2.0f);
SSBFilter->create_filter(m_LowCutoff / (float) m_audioSampleRate, m_Bandwidth / (float) m_audioSampleRate);
m_volume = cfg.getVolume();
m_volume *= m_volume * 0.1;
m_spanLog2 = cfg.getSpanLog2();
m_settingsMutex.unlock();
qDebug() << " - MsgConfigureSSBDemod: m_Bandwidth: " << m_Bandwidth
<< " m_LowCutoff: " << m_LowCutoff
<< " m_volume: " << m_volume
<< " m_spanLog2: " << m_spanLog2;
return true;
}
else
{
if(m_sampleSink != 0)
{
return m_sampleSink->handleMessage(cmd);
}
else
{
return false;
}
}
}