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sdrangel/plugins/channelrx/demodssb/ssbdemod.cpp
2016-12-06 01:58:23 +01:00

316 lines
8.4 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 "../../channelrx/demodssb/ssbdemod.h"
#include <dsp/downchannelizer.h>
#include <QTime>
#include <QDebug>
#include <stdio.h>
#include "audio/audiooutput.h"
#include "dsp/dspengine.h"
MESSAGE_CLASS_DEFINITION(SSBDemod::MsgConfigureSSBDemod, Message)
SSBDemod::SSBDemod(BasebandSampleSink* sampleSink) :
m_sampleSink(sampleSink),
m_audioFifo(4, 24000),
m_settingsMutex(QMutex::Recursive),
m_audioBinaual(false),
m_audioFlipChannels(false)
{
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_sum = 0;
m_usb = true;
m_magsq = 0.0f;
m_magsqSum = 0.0f;
m_magsqPeak = 0.0f;
m_magsqCount = 0;
SSBFilter = new fftfilt(m_LowCutoff / m_audioSampleRate, m_Bandwidth / m_audioSampleRate, ssbFftLen);
DSBFilter = new fftfilt((2.0f * m_Bandwidth) / m_audioSampleRate, 2 * ssbFftLen);
DSPEngine::instance()->addAudioSink(&m_audioFifo);
}
SSBDemod::~SSBDemod()
{
if (SSBFilter) delete SSBFilter;
if (DSBFilter) delete DSBFilter;
DSPEngine::instance()->removeAudioSink(&m_audioFifo);
}
void SSBDemod::configure(MessageQueue* messageQueue,
Real Bandwidth,
Real LowCutoff,
Real volume,
int spanLog2,
bool audioBinaural,
bool audioFlipChannel,
bool dsb,
bool audioMute)
{
Message* cmd = MsgConfigureSSBDemod::create(Bandwidth, LowCutoff, volume, spanLog2, audioBinaural, audioFlipChannel, dsb, audioMute);
messageQueue->push(cmd);
}
void SSBDemod::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end, bool positiveOnly)
{
Complex ci;
fftfilt::cmplx *sideband;
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);
Complex c(it->real(), it->imag());
c *= m_nco.nextIQ();
if(m_interpolator.decimate(&m_sampleDistanceRemain, c, &ci))
{
if (m_dsb)
{
n_out = DSBFilter->runDSB(ci, &sideband);
}
else
{
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++)
{
// Downsample by 2^(m_scaleLog2 - 1) for SSB band spectrum display
// smart decimation with bit gain using float arithmetic (23 bits significand)
m_sum += sideband[i];
if (!(m_undersampleCount++ & decim_mask))
{
Real avgr = m_sum.real() / decim;
Real avgi = m_sum.imag() / decim;
m_magsq = (avgr * avgr + avgi * avgi) / (1<<30);
m_magsqSum += m_magsq;
if (m_magsq > m_magsqPeak)
{
m_magsqPeak = m_magsq;
}
m_magsqCount++;
if (!m_dsb & !m_usb)
{ // invert spectrum for LSB
m_sampleBuffer.push_back(Sample(avgi, avgr));
}
else
{
m_sampleBuffer.push_back(Sample(avgr, avgi));
}
m_sum.real(0.0);
m_sum.imag(0.0);
}
if (m_audioMute)
{
m_audioBuffer[m_audioBufferFill].r = 0;
m_audioBuffer[m_audioBufferFill].l = 0;
}
else
{
if (m_audioBinaual)
{
if (m_audioFlipChannels)
{
m_audioBuffer[m_audioBufferFill].r = (qint16)(sideband[i].imag() * m_volume * 100);
m_audioBuffer[m_audioBufferFill].l = (qint16)(sideband[i].real() * m_volume * 100);
}
else
{
m_audioBuffer[m_audioBufferFill].r = (qint16)(sideband[i].real() * m_volume * 100);
m_audioBuffer[m_audioBufferFill].l = (qint16)(sideband[i].imag() * m_volume * 100);
}
}
else
{
Real demod = (sideband[i].real() + sideband[i].imag()) * 0.7;
qint16 sample = (qint16)(demod * m_volume * 100);
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(), !m_dsb);
}
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 (DownChannelizer::MsgChannelizerNotification::match(cmd))
{
DownChannelizer::MsgChannelizerNotification& notif = (DownChannelizer::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);
DSBFilter->create_dsb_filter((2.0f * m_Bandwidth) / (float) m_audioSampleRate);
m_volume = cfg.getVolume();
m_volume *= m_volume * 0.1;
m_spanLog2 = cfg.getSpanLog2();
m_audioBinaual = cfg.getAudioBinaural();
m_audioFlipChannels = cfg.getAudioFlipChannels();
m_dsb = cfg.getDSB();
m_audioMute = cfg.getAudioMute();
m_settingsMutex.unlock();
qDebug() << "SBDemod::handleMessage: MsgConfigureSSBDemod: m_Bandwidth: " << m_Bandwidth
<< " m_LowCutoff: " << m_LowCutoff
<< " m_volume: " << m_volume
<< " m_spanLog2: " << m_spanLog2
<< " m_audioBinaual: " << m_audioBinaual
<< " m_audioFlipChannels: " << m_audioFlipChannels
<< " m_dsb: " << m_dsb
<< "m_audioMute: " << m_audioMute;
return true;
}
else
{
if(m_sampleSink != 0)
{
return m_sampleSink->handleMessage(cmd);
}
else
{
return false;
}
}
}