mirror of
https://github.com/f4exb/sdrangel.git
synced 2024-11-14 12:22:00 -05:00
580 lines
18 KiB
C++
580 lines
18 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 "audio/audiooutput.h"
|
|
#include "dsp/dspengine.h"
|
|
#include "dsp/downchannelizer.h"
|
|
#include "dsp/threadedbasebandsamplesink.h"
|
|
#include "dsp/dspcommands.h"
|
|
#include "device/devicesourceapi.h"
|
|
#include "util/db.h"
|
|
|
|
#include "ssbdemod.h"
|
|
|
|
MESSAGE_CLASS_DEFINITION(SSBDemod::MsgConfigureSSBDemod, Message)
|
|
MESSAGE_CLASS_DEFINITION(SSBDemod::MsgConfigureSSBDemodPrivate, Message)
|
|
MESSAGE_CLASS_DEFINITION(SSBDemod::MsgConfigureChannelizer, Message)
|
|
|
|
const QString SSBDemod::m_channelIdURI = "de.maintech.sdrangelove.channel.ssb";
|
|
const QString SSBDemod::m_channelId = "SSBDemod";
|
|
|
|
SSBDemod::SSBDemod(DeviceSourceAPI *deviceAPI) :
|
|
ChannelSinkAPI(m_channelIdURI),
|
|
m_deviceAPI(deviceAPI),
|
|
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_squelchDelayLine(2*48000),
|
|
m_audioActive(false),
|
|
m_sampleSink(0),
|
|
m_audioFifo(24000),
|
|
m_settingsMutex(QMutex::Recursive)
|
|
{
|
|
setObjectName(m_channelId);
|
|
|
|
m_Bandwidth = 5000;
|
|
m_LowCutoff = 300;
|
|
m_volume = 2.0;
|
|
m_spanLog2 = 3;
|
|
m_inputSampleRate = 48000;
|
|
m_inputFrequencyOffset = 0;
|
|
|
|
DSPEngine::instance()->getAudioDeviceManager()->addAudioSink(&m_audioFifo, getInputMessageQueue());
|
|
m_audioSampleRate = DSPEngine::instance()->getAudioDeviceManager()->getOutputSampleRate();
|
|
|
|
m_audioBuffer.resize(1<<14);
|
|
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(SDR_RX_SCALED*SDR_RX_SCALED);
|
|
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);
|
|
|
|
applyChannelSettings(m_inputSampleRate, m_inputFrequencyOffset, true);
|
|
applySettings(m_settings, true);
|
|
|
|
m_channelizer = new DownChannelizer(this);
|
|
m_threadedChannelizer = new ThreadedBasebandSampleSink(m_channelizer, this);
|
|
m_deviceAPI->addThreadedSink(m_threadedChannelizer);
|
|
m_deviceAPI->addChannelAPI(this);
|
|
}
|
|
|
|
SSBDemod::~SSBDemod()
|
|
{
|
|
DSPEngine::instance()->getAudioDeviceManager()->removeAudioSink(&m_audioFifo);
|
|
|
|
m_deviceAPI->removeChannelAPI(this);
|
|
m_deviceAPI->removeThreadedSink(m_threadedChannelizer);
|
|
delete m_threadedChannelizer;
|
|
delete m_channelizer;
|
|
delete SSBFilter;
|
|
delete DSBFilter;
|
|
}
|
|
|
|
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 = MsgConfigureSSBDemodPrivate::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(), it->imag());
|
|
c *= m_nco.nextIQ();
|
|
|
|
if(m_interpolator.decimate(&m_interpolatorDistanceRemain, c, &ci))
|
|
{
|
|
if (m_dsb)
|
|
{
|
|
n_out = DSBFilter->runDSB(ci, &sideband);
|
|
}
|
|
else
|
|
{
|
|
n_out = SSBFilter->runSSB(ci, &sideband, m_usb);
|
|
}
|
|
|
|
m_interpolatorDistanceRemain += m_interpolatorDistance;
|
|
}
|
|
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) / (SDR_RX_SCALED*SDR_RX_SCALED);
|
|
|
|
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);
|
|
}
|
|
|
|
float agcVal = m_agcActive ? m_agc.feedAndGetValue(sideband[i]) : 10.0; // 10.0 for 3276.8, 1.0 for 327.68
|
|
fftfilt::cmplx& delayedSample = m_squelchDelayLine.readBack(m_agc.getStepDownDelay());
|
|
m_audioActive = delayedSample.real() != 0.0;
|
|
m_squelchDelayLine.write(sideband[i]*agcVal);
|
|
|
|
if (m_audioMute)
|
|
{
|
|
m_audioBuffer[m_audioBufferFill].r = 0;
|
|
m_audioBuffer[m_audioBufferFill].l = 0;
|
|
}
|
|
else
|
|
{
|
|
fftfilt::cmplx z = delayedSample * m_agc.getStepValue();
|
|
|
|
if (m_audioBinaual)
|
|
{
|
|
if (m_audioFlipChannels)
|
|
{
|
|
m_audioBuffer[m_audioBufferFill].r = (qint16)(z.imag() * m_volume);
|
|
m_audioBuffer[m_audioBufferFill].l = (qint16)(z.real() * m_volume);
|
|
}
|
|
else
|
|
{
|
|
m_audioBuffer[m_audioBufferFill].r = (qint16)(z.real() * m_volume);
|
|
m_audioBuffer[m_audioBufferFill].l = (qint16)(z.imag() * m_volume);
|
|
}
|
|
}
|
|
else
|
|
{
|
|
Real demod = (z.real() + z.imag()) * 0.7;
|
|
qint16 sample = (qint16)(demod * m_volume);
|
|
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, 10);
|
|
|
|
if (res != m_audioBufferFill)
|
|
{
|
|
qDebug("SSBDemod::feed: %u/%u samples written", res, m_audioBufferFill);
|
|
}
|
|
|
|
m_audioBufferFill = 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10);
|
|
|
|
if (res != m_audioBufferFill)
|
|
{
|
|
qDebug("SSBDemod::feed: %u/%u tail samples written", res, m_audioBufferFill);
|
|
}
|
|
|
|
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()
|
|
{
|
|
applyChannelSettings(m_inputSampleRate, m_inputFrequencyOffset, true);
|
|
}
|
|
|
|
void SSBDemod::stop()
|
|
{
|
|
}
|
|
|
|
bool SSBDemod::handleMessage(const Message& cmd)
|
|
{
|
|
if (DownChannelizer::MsgChannelizerNotification::match(cmd))
|
|
{
|
|
DownChannelizer::MsgChannelizerNotification& notif = (DownChannelizer::MsgChannelizerNotification&) cmd;
|
|
qDebug("SSBDemod::handleMessage: MsgChannelizerNotification: m_sampleRate");
|
|
|
|
applyChannelSettings(notif.getSampleRate(), notif.getFrequencyOffset());
|
|
|
|
return true;
|
|
}
|
|
else if (MsgConfigureChannelizer::match(cmd))
|
|
{
|
|
MsgConfigureChannelizer& cfg = (MsgConfigureChannelizer&) cmd;
|
|
qDebug() << "SSBDemod::handleMessage: MsgConfigureChannelizer: sampleRate: " << cfg.getSampleRate()
|
|
<< " centerFrequency: " << cfg.getCenterFrequency();
|
|
|
|
m_channelizer->configure(m_channelizer->getInputMessageQueue(),
|
|
cfg.getSampleRate(),
|
|
cfg.getCenterFrequency());
|
|
|
|
return true;
|
|
}
|
|
else if (MsgConfigureSSBDemod::match(cmd))
|
|
{
|
|
MsgConfigureSSBDemod& cfg = (MsgConfigureSSBDemod&) cmd;
|
|
qDebug("SSBDemod::handleMessage: MsgConfigureSSBDemod");
|
|
|
|
applySettings(cfg.getSettings(), cfg.getForce());
|
|
|
|
return true;
|
|
}
|
|
else if (BasebandSampleSink::MsgThreadedSink::match(cmd))
|
|
{
|
|
BasebandSampleSink::MsgThreadedSink& cfg = (BasebandSampleSink::MsgThreadedSink&) cmd;
|
|
const QThread *thread = cfg.getThread();
|
|
qDebug("SSBDemod::handleMessage: BasebandSampleSink::MsgThreadedSink: %p", thread);
|
|
return true;
|
|
}
|
|
else if (DSPConfigureAudio::match(cmd))
|
|
{
|
|
DSPConfigureAudio& cfg = (DSPConfigureAudio&) cmd;
|
|
uint32_t sampleRate = cfg.getSampleRate();
|
|
|
|
qDebug() << "SSBDemod::handleMessage: DSPConfigureAudio:"
|
|
<< " sampleRate: " << sampleRate;
|
|
|
|
if (sampleRate != m_audioSampleRate) {
|
|
applyAudioSampleRate(sampleRate);
|
|
}
|
|
|
|
return true;
|
|
}
|
|
else if (DSPSignalNotification::match(cmd))
|
|
{
|
|
return true;
|
|
}
|
|
else
|
|
{
|
|
if(m_sampleSink != 0)
|
|
{
|
|
return m_sampleSink->handleMessage(cmd);
|
|
}
|
|
else
|
|
{
|
|
return false;
|
|
}
|
|
}
|
|
}
|
|
|
|
void SSBDemod::applyChannelSettings(int inputSampleRate, int inputFrequencyOffset, bool force)
|
|
{
|
|
qDebug() << "SSBDemod::applyChannelSettings:"
|
|
<< " inputSampleRate: " << inputSampleRate
|
|
<< " inputFrequencyOffset: " << inputFrequencyOffset;
|
|
|
|
if ((m_inputFrequencyOffset != inputFrequencyOffset) ||
|
|
(m_inputSampleRate != inputSampleRate) || force)
|
|
{
|
|
m_nco.setFreq(-inputFrequencyOffset, inputSampleRate);
|
|
}
|
|
|
|
if ((m_inputSampleRate != inputSampleRate) || force)
|
|
{
|
|
m_settingsMutex.lock();
|
|
m_interpolator.create(16, inputSampleRate, m_Bandwidth * 1.5f, 2.0f);
|
|
m_interpolatorDistanceRemain = 0;
|
|
m_interpolatorDistance = (Real) inputSampleRate / (Real) m_audioSampleRate;
|
|
m_settingsMutex.unlock();
|
|
}
|
|
|
|
m_inputSampleRate = inputSampleRate;
|
|
m_inputFrequencyOffset = inputFrequencyOffset;
|
|
}
|
|
|
|
void SSBDemod::applyAudioSampleRate(int sampleRate)
|
|
{
|
|
qDebug("SSBDemod::applyAudioSampleRate: %d", sampleRate);
|
|
|
|
MsgConfigureChannelizer* channelConfigMsg = MsgConfigureChannelizer::create(
|
|
sampleRate, m_settings.m_inputFrequencyOffset);
|
|
m_inputMessageQueue.push(channelConfigMsg);
|
|
|
|
m_settingsMutex.lock();
|
|
|
|
m_interpolator.create(16, m_inputSampleRate, m_Bandwidth * 1.5f, 2.0f);
|
|
m_interpolatorDistanceRemain = 0;
|
|
m_interpolatorDistance = (Real) m_inputSampleRate / (Real) sampleRate;
|
|
|
|
SSBFilter->create_filter(m_LowCutoff / (float) sampleRate, m_Bandwidth / (float) sampleRate);
|
|
DSBFilter->create_dsb_filter((2.0f * m_Bandwidth) / (float) sampleRate);
|
|
|
|
int agcNbSamples = (sampleRate / 1000) * (1<<m_settings.m_agcTimeLog2);
|
|
int agcThresholdGate = (sampleRate / 1000) * m_settings.m_agcThresholdGate; // ms
|
|
|
|
if (m_agcNbSamples != agcNbSamples)
|
|
{
|
|
m_agc.resize(agcNbSamples, agcNbSamples/2, agcTarget);
|
|
m_agc.setStepDownDelay(agcNbSamples);
|
|
m_agcNbSamples = agcNbSamples;
|
|
}
|
|
|
|
if (m_agcThresholdGate != agcThresholdGate)
|
|
{
|
|
m_agc.setGate(agcThresholdGate);
|
|
m_agcThresholdGate = agcThresholdGate;
|
|
}
|
|
|
|
m_audioFifo.setSize(sampleRate);
|
|
|
|
m_settingsMutex.unlock();
|
|
|
|
m_audioSampleRate = sampleRate;
|
|
|
|
if (m_guiMessageQueue) // forward to GUI if any
|
|
{
|
|
DSPConfigureAudio *cfg = new DSPConfigureAudio(m_audioSampleRate);
|
|
m_guiMessageQueue->push(cfg);
|
|
}
|
|
}
|
|
|
|
void SSBDemod::applySettings(const SSBDemodSettings& settings, bool force)
|
|
{
|
|
qDebug() << "SSBDemod::applySettings:"
|
|
<< " m_inputFrequencyOffset: " << settings.m_inputFrequencyOffset
|
|
<< " m_rfBandwidth: " << settings.m_rfBandwidth
|
|
<< " m_lowCutoff: " << settings.m_lowCutoff
|
|
<< " m_volume: " << settings.m_volume
|
|
<< " m_spanLog2: " << settings.m_spanLog2
|
|
<< " m_audioBinaual: " << settings.m_audioBinaural
|
|
<< " m_audioFlipChannels: " << settings.m_audioFlipChannels
|
|
<< " m_dsb: " << settings.m_dsb
|
|
<< " m_audioMute: " << settings.m_audioMute
|
|
<< " m_agcActive: " << settings.m_agc
|
|
<< " m_agcClamping: " << settings.m_agcClamping
|
|
<< " m_agcTimeLog2: " << settings.m_agcTimeLog2
|
|
<< " agcPowerThreshold: " << settings.m_agcPowerThreshold
|
|
<< " agcThresholdGate: " << settings.m_agcThresholdGate
|
|
<< " m_audioDeviceName: " << settings.m_audioDeviceName
|
|
<< " force: " << force;
|
|
|
|
if((m_settings.m_rfBandwidth != settings.m_rfBandwidth) ||
|
|
(m_settings.m_lowCutoff != settings.m_lowCutoff) || force)
|
|
{
|
|
float band, lowCutoff;
|
|
|
|
band = settings.m_rfBandwidth;
|
|
lowCutoff = settings.m_lowCutoff;
|
|
|
|
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_settingsMutex.lock();
|
|
m_interpolator.create(16, m_inputSampleRate, m_Bandwidth * 1.5f, 2.0f);
|
|
m_interpolatorDistanceRemain = 0;
|
|
m_interpolatorDistance = (Real) m_inputSampleRate / (Real) m_audioSampleRate;
|
|
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_settingsMutex.unlock();
|
|
}
|
|
|
|
if ((m_settings.m_volume != settings.m_volume) || force)
|
|
{
|
|
m_volume = settings.m_volume;
|
|
m_volume /= 4.0; // for 3276.8
|
|
}
|
|
|
|
if ((m_settings.m_agcTimeLog2 != settings.m_agcTimeLog2) ||
|
|
(m_settings.m_agcPowerThreshold != settings.m_agcPowerThreshold) ||
|
|
(m_settings.m_agcThresholdGate != settings.m_agcThresholdGate) ||
|
|
(m_settings.m_agcClamping != settings.m_agcClamping) || force)
|
|
{
|
|
int agcNbSamples = (m_audioSampleRate / 1000) * (1<<settings.m_agcTimeLog2);
|
|
m_agc.setThresholdEnable(settings.m_agcPowerThreshold != -SSBDemodSettings::m_minPowerThresholdDB);
|
|
double agcPowerThreshold = CalcDb::powerFromdB(settings.m_agcPowerThreshold) * (SDR_RX_SCALED*SDR_RX_SCALED);
|
|
int agcThresholdGate = (m_audioSampleRate / 1000) * settings.m_agcThresholdGate; // ms
|
|
bool agcClamping = settings.m_agcClamping;
|
|
|
|
if (m_agcNbSamples != agcNbSamples)
|
|
{
|
|
m_settingsMutex.lock();
|
|
m_agc.resize(agcNbSamples, agcNbSamples/2, agcTarget);
|
|
m_agc.setStepDownDelay(agcNbSamples);
|
|
m_agcNbSamples = agcNbSamples;
|
|
m_settingsMutex.unlock();
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
qDebug() << "SBDemod::applySettings: AGC:"
|
|
<< " agcNbSamples: " << agcNbSamples
|
|
<< " agcPowerThreshold: " << agcPowerThreshold
|
|
<< " agcThresholdGate: " << agcThresholdGate
|
|
<< " agcClamping: " << agcClamping;
|
|
}
|
|
|
|
if ((settings.m_audioDeviceName != m_settings.m_audioDeviceName) || force)
|
|
{
|
|
AudioDeviceManager *audioDeviceManager = DSPEngine::instance()->getAudioDeviceManager();
|
|
int audioDeviceIndex = audioDeviceManager->getOutputDeviceIndex(settings.m_audioDeviceName);
|
|
audioDeviceManager->addAudioSink(&m_audioFifo, getInputMessageQueue(), audioDeviceIndex);
|
|
uint32_t audioSampleRate = audioDeviceManager->getOutputSampleRate(audioDeviceIndex);
|
|
|
|
if (m_audioSampleRate != audioSampleRate) {
|
|
applyAudioSampleRate(audioSampleRate);
|
|
}
|
|
}
|
|
|
|
m_spanLog2 = settings.m_spanLog2;
|
|
m_audioBinaual = settings.m_audioBinaural;
|
|
m_audioFlipChannels = settings.m_audioFlipChannels;
|
|
m_dsb = settings.m_dsb;
|
|
m_audioMute = settings.m_audioMute;
|
|
m_agcActive = settings.m_agc;
|
|
|
|
m_settings = settings;
|
|
}
|
|
|
|
QByteArray SSBDemod::serialize() const
|
|
{
|
|
return m_settings.serialize();
|
|
}
|
|
|
|
bool SSBDemod::deserialize(const QByteArray& data)
|
|
{
|
|
if (m_settings.deserialize(data))
|
|
{
|
|
MsgConfigureSSBDemod *msg = MsgConfigureSSBDemod::create(m_settings, true);
|
|
m_inputMessageQueue.push(msg);
|
|
return true;
|
|
}
|
|
else
|
|
{
|
|
m_settings.resetToDefaults();
|
|
MsgConfigureSSBDemod *msg = MsgConfigureSSBDemod::create(m_settings, true);
|
|
m_inputMessageQueue.push(msg);
|
|
return false;
|
|
}
|
|
}
|