mirror of
https://github.com/f4exb/sdrangel.git
synced 2024-11-30 03:38:55 -05:00
387 lines
10 KiB
C++
387 lines
10 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 "ssbdemod.h"
|
|
|
|
#include <dsp/downchannelizer.h>
|
|
#include <QTime>
|
|
#include <QDebug>
|
|
#include <stdio.h>
|
|
#include "audio/audiooutput.h"
|
|
#include "dsp/dspengine.h"
|
|
#include "util/db.h"
|
|
|
|
MESSAGE_CLASS_DEFINITION(SSBDemod::MsgConfigureSSBDemod, Message)
|
|
|
|
SSBDemod::SSBDemod(BasebandSampleSink* sampleSink) :
|
|
m_audioBinaual(false),
|
|
m_audioFlipChannels(false),
|
|
m_dsb(false),
|
|
m_audioMute(false),
|
|
m_agc(12000, agcTarget, 1e-2),
|
|
m_agcActive(false),
|
|
m_agcClamping(false),
|
|
m_agcNbSamples(12000),
|
|
m_agcPowerThreshold(1e-2),
|
|
m_agcThresholdGate(0),
|
|
m_audioActive(false),
|
|
m_sampleSink(sampleSink),
|
|
m_audioFifo(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_sum = 0;
|
|
|
|
m_usb = true;
|
|
m_magsq = 0.0f;
|
|
m_magsqSum = 0.0f;
|
|
m_magsqPeak = 0.0f;
|
|
m_magsqCount = 0;
|
|
|
|
m_agc.setClampMax(32768.0*32768.0);
|
|
m_agc.setClamping(m_agcClamping);
|
|
|
|
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,
|
|
bool agc,
|
|
bool agcClamping,
|
|
int agcTimeLog2,
|
|
int agcPowerThreshold,
|
|
int agcThresholdGate)
|
|
{
|
|
Message* cmd = MsgConfigureSSBDemod::create(
|
|
Bandwidth,
|
|
LowCutoff,
|
|
volume,
|
|
spanLog2,
|
|
audioBinaural,
|
|
audioFlipChannel,
|
|
dsb,
|
|
audioMute,
|
|
agc,
|
|
agcClamping,
|
|
agcTimeLog2,
|
|
agcPowerThreshold,
|
|
agcThresholdGate);
|
|
messageQueue->push(cmd);
|
|
}
|
|
|
|
void SSBDemod::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end, bool positiveOnly __attribute__((unused)))
|
|
{
|
|
Complex ci;
|
|
fftfilt::cmplx *sideband;
|
|
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);
|
|
}
|
|
|
|
double agcVal = m_agcActive ? m_agc.feedAndGetValue(sideband[i]) : 10.0; // 10.0 for 3276.8, 1.0 for 327.68
|
|
m_audioActive = agcVal != 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 * agcVal);
|
|
m_audioBuffer[m_audioBufferFill].l = (qint16)(sideband[i].real() * m_volume * agcVal);
|
|
}
|
|
else
|
|
{
|
|
m_audioBuffer[m_audioBufferFill].r = (qint16)(sideband[i].real() * m_volume * agcVal);
|
|
m_audioBuffer[m_audioBufferFill].l = (qint16)(sideband[i].imag() * m_volume * agcVal);
|
|
}
|
|
}
|
|
else
|
|
{
|
|
Real demod = (sideband[i].real() + sideband[i].imag()) * 0.7;
|
|
qint16 sample = (qint16)(demod * m_volume * agcVal);
|
|
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 *= 2.0; // for 327.68
|
|
m_volume /= 4.0; // for 3276.8
|
|
|
|
m_spanLog2 = cfg.getSpanLog2();
|
|
m_audioBinaual = cfg.getAudioBinaural();
|
|
m_audioFlipChannels = cfg.getAudioFlipChannels();
|
|
m_dsb = cfg.getDSB();
|
|
m_audioMute = cfg.getAudioMute();
|
|
m_agcActive = cfg.getAGC();
|
|
|
|
int agcNbSamples = 48 * (1<<cfg.getAGCTimeLog2());
|
|
m_agc.setThresholdEnable(cfg.getAGCPowerThershold() != -99);
|
|
double agcPowerThreshold = CalcDb::powerFromdB(cfg.getAGCPowerThershold()) * (1<<30);
|
|
int agcThresholdGate = 48 * cfg.getAGCThersholdGate(); // ms
|
|
bool agcClamping = cfg.getAGCClamping();
|
|
|
|
if (m_agcNbSamples != agcNbSamples)
|
|
{
|
|
m_agc.resize(agcNbSamples, agcTarget);
|
|
m_agc.setStepDownDelay(agcNbSamples);
|
|
m_agcNbSamples = agcNbSamples;
|
|
}
|
|
|
|
if (m_agcPowerThreshold != agcPowerThreshold)
|
|
{
|
|
m_agc.setThreshold(agcPowerThreshold);
|
|
m_agcPowerThreshold = agcPowerThreshold;
|
|
}
|
|
|
|
if (m_agcThresholdGate != agcThresholdGate)
|
|
{
|
|
m_agc.setGate(agcThresholdGate);
|
|
m_agcThresholdGate = agcThresholdGate;
|
|
}
|
|
|
|
if (m_agcClamping != agcClamping)
|
|
{
|
|
m_agc.setClamping(agcClamping);
|
|
m_agcClamping = agcClamping;
|
|
}
|
|
|
|
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
|
|
<< " m_agcActive: " << m_agcActive
|
|
<< " m_agcClamping: " << m_agcClamping
|
|
<< " agcNbSamples: " << agcNbSamples
|
|
<< " agcPowerThreshold: " << agcPowerThreshold
|
|
<< " agcThresholdGate: " << agcThresholdGate;
|
|
|
|
return true;
|
|
}
|
|
else
|
|
{
|
|
if(m_sampleSink != 0)
|
|
{
|
|
return m_sampleSink->handleMessage(cmd);
|
|
}
|
|
else
|
|
{
|
|
return false;
|
|
}
|
|
}
|
|
}
|