mirror of
https://github.com/f4exb/sdrangel.git
synced 2025-11-03 13:11:20 -05:00
503 lines
19 KiB
C++
503 lines
19 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
|
|
// Copyright (C) 2019-2023 Edouard Griffiths, F4EXB <f4exb06@gmail.com> //
|
|
// //
|
|
// 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 //
|
|
// (at your option) any later version. //
|
|
// //
|
|
// 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 <stdio.h>
|
|
|
|
#include <QTime>
|
|
#include <QDebug>
|
|
|
|
#include "dsp/spectrumvis.h"
|
|
#include "dsp/datafifo.h"
|
|
#include "util/db.h"
|
|
#include "util/messagequeue.h"
|
|
#include "maincore.h"
|
|
|
|
#include "ssbdemodsink.h"
|
|
|
|
const int SSBDemodSink::m_ssbFftLen = 2048;
|
|
const int SSBDemodSink::m_agcTarget = 3276; // 32768/10 -10 dB amplitude => -20 dB power: center of normal signal
|
|
|
|
SSBDemodSink::SSBDemodSink() :
|
|
m_audioBinaual(false),
|
|
m_audioFlipChannels(false),
|
|
m_dsb(false),
|
|
m_audioMute(false),
|
|
m_agc(12000, m_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_spectrumSink(nullptr),
|
|
m_audioFifo(24000),
|
|
m_audioSampleRate(48000)
|
|
{
|
|
m_Bandwidth = 5000;
|
|
m_LowCutoff = 300;
|
|
m_volume = 2.0;
|
|
m_spanLog2 = 3;
|
|
m_channelSampleRate = 48000;
|
|
m_channelFrequencyOffset = 0;
|
|
|
|
m_audioBuffer.resize(m_audioSampleRate / 10);
|
|
m_audioBufferFill = 0;
|
|
m_undersampleCount = 0;
|
|
m_sum = 0;
|
|
|
|
m_demodBuffer.resize(1<<12);
|
|
m_demodBufferFill = 0;
|
|
|
|
m_usb = true;
|
|
m_magsq = 0.0;
|
|
m_magsqSum = 0.0;
|
|
m_magsqPeak = 0.0;
|
|
m_magsqCount = 0;
|
|
|
|
SSBFilter = new fftfilt(m_LowCutoff / m_audioSampleRate, m_Bandwidth / m_audioSampleRate, m_ssbFftLen);
|
|
DSBFilter = new fftfilt((2.0f * m_Bandwidth) / m_audioSampleRate, 2 * m_ssbFftLen);
|
|
|
|
m_lowpassI.create(101, m_audioSampleRate, m_Bandwidth * 1.2);
|
|
m_lowpassQ.create(101, m_audioSampleRate, m_Bandwidth * 1.2);
|
|
|
|
applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
|
|
applySettings(m_settings, true);
|
|
}
|
|
|
|
SSBDemodSink::~SSBDemodSink()
|
|
{
|
|
delete SSBFilter;
|
|
delete DSBFilter;
|
|
}
|
|
|
|
void SSBDemodSink::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end)
|
|
{
|
|
if (m_channelSampleRate == 0) {
|
|
return;
|
|
}
|
|
|
|
Complex ci;
|
|
|
|
for(SampleVector::const_iterator it = begin; it < end; ++it)
|
|
{
|
|
Complex c(it->real(), it->imag());
|
|
c *= m_nco.nextIQ();
|
|
|
|
if (m_interpolatorDistance < 1.0f) // interpolate
|
|
{
|
|
while (!m_interpolator.interpolate(&m_interpolatorDistanceRemain, c, &ci))
|
|
{
|
|
processOneSample(ci);
|
|
m_interpolatorDistanceRemain += m_interpolatorDistance;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
if (m_interpolator.decimate(&m_interpolatorDistanceRemain, c, &ci))
|
|
{
|
|
processOneSample(ci);
|
|
m_interpolatorDistanceRemain += m_interpolatorDistance;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void SSBDemodSink::processOneSample(Complex &ci)
|
|
{
|
|
fftfilt::cmplx *sideband;
|
|
int n_out = 0;
|
|
int decim = 1<<(m_spanLog2 - 1);
|
|
unsigned char decim_mask = decim - 1; // counter LSB bit mask for decimation by 2^(m_scaleLog2 - 1)
|
|
|
|
if (m_dsb) {
|
|
n_out = DSBFilter->runDSB(ci, &sideband);
|
|
} else {
|
|
n_out = SSBFilter->runSSB(ci, &sideband, m_usb);
|
|
}
|
|
|
|
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]) : 1.0;
|
|
fftfilt::cmplx& delayedSample = m_squelchDelayLine.readBack(m_agc.getStepDownDelay());
|
|
m_audioActive = delayedSample.real() != 0.0;
|
|
|
|
// Prevent overload based on squared magnitude variation
|
|
// Only if AGC is active
|
|
if (m_agcActive && m_agcClamping && (agcVal > 100.0 || agcVal == 0.0))
|
|
{
|
|
// qDebug("SSBDemodSink::processOneSample: %f", agcVal);
|
|
m_agc.reset(m_agcTarget*m_agcTarget);
|
|
m_squelchDelayLine.write(fftfilt::cmplx{0.0, 0.0});
|
|
}
|
|
else
|
|
{
|
|
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 = (m_agcActive && m_agcClamping) ?
|
|
fftfilt::cmplx{m_lowpassI.filter(delayedSample.real()), m_lowpassQ.filter(delayedSample.imag())}
|
|
: delayedSample;
|
|
|
|
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);
|
|
}
|
|
|
|
m_demodBuffer[m_demodBufferFill++] = z.real();
|
|
m_demodBuffer[m_demodBufferFill++] = z.imag();
|
|
}
|
|
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_demodBuffer[m_demodBufferFill++] = (z.real() + z.imag()) * 0.7;
|
|
}
|
|
|
|
if (m_demodBufferFill >= m_demodBuffer.size())
|
|
{
|
|
QList<ObjectPipe*> dataPipes;
|
|
MainCore::instance()->getDataPipes().getDataPipes(m_channel, "demod", dataPipes);
|
|
|
|
if (dataPipes.size() > 0)
|
|
{
|
|
QList<ObjectPipe*>::iterator it = dataPipes.begin();
|
|
|
|
for (; it != dataPipes.end(); ++it)
|
|
{
|
|
DataFifo *fifo = qobject_cast<DataFifo*>((*it)->m_element);
|
|
|
|
if (fifo)
|
|
{
|
|
fifo->write(
|
|
(quint8*) &m_demodBuffer[0],
|
|
m_demodBuffer.size() * sizeof(qint16),
|
|
m_audioBinaual ? DataFifo::DataTypeCI16 : DataFifo::DataTypeI16
|
|
);
|
|
}
|
|
}
|
|
}
|
|
|
|
m_demodBufferFill = 0;
|
|
}
|
|
}
|
|
|
|
++m_audioBufferFill;
|
|
|
|
if (m_audioBufferFill >= m_audioBuffer.size())
|
|
{
|
|
std::size_t res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], std::min(m_audioBufferFill, m_audioBuffer.size()));
|
|
|
|
if (res != m_audioBufferFill) {
|
|
qDebug("SSBDemodSink::processOneSample: %lu/%lu samples written", res, m_audioBufferFill);
|
|
}
|
|
|
|
m_audioBufferFill = 0;
|
|
}
|
|
}
|
|
|
|
if (m_spectrumSink && (m_sampleBuffer.size() != 0))
|
|
{
|
|
m_spectrumSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), !m_dsb);
|
|
m_sampleBuffer.clear();
|
|
}
|
|
}
|
|
|
|
void SSBDemodSink::setDNR(bool dnr)
|
|
{
|
|
SSBFilter->setDNR(dnr);
|
|
}
|
|
|
|
void SSBDemodSink::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
|
|
{
|
|
qDebug() << "SSBDemodSink::applyChannelSettings:"
|
|
<< " channelSampleRate: " << channelSampleRate
|
|
<< " channelFrequencyOffset: " << channelFrequencyOffset;
|
|
|
|
if ((m_channelFrequencyOffset != channelFrequencyOffset) ||
|
|
(m_channelSampleRate != channelSampleRate) || force)
|
|
{
|
|
m_nco.setFreq(-channelFrequencyOffset, channelSampleRate);
|
|
}
|
|
|
|
if ((m_channelSampleRate != channelSampleRate) || force)
|
|
{
|
|
Real interpolatorBandwidth = (m_Bandwidth * 1.5f) > channelSampleRate ? channelSampleRate : (m_Bandwidth * 1.5f);
|
|
m_interpolator.create(16, channelSampleRate, interpolatorBandwidth, 2.0f);
|
|
m_interpolatorDistanceRemain = 0;
|
|
m_interpolatorDistance = (Real) channelSampleRate / (Real) m_audioSampleRate;
|
|
}
|
|
|
|
m_channelSampleRate = channelSampleRate;
|
|
m_channelFrequencyOffset = channelFrequencyOffset;
|
|
}
|
|
|
|
void SSBDemodSink::applyAudioSampleRate(int sampleRate)
|
|
{
|
|
qDebug("SSBDemodSink::applyAudioSampleRate: %d", sampleRate);
|
|
|
|
Real interpolatorBandwidth = (m_Bandwidth * 1.5f) > m_channelSampleRate ? m_channelSampleRate : (m_Bandwidth * 1.5f);
|
|
m_interpolator.create(16, m_channelSampleRate, interpolatorBandwidth, 2.0f);
|
|
m_interpolatorDistanceRemain = 0;
|
|
m_interpolatorDistance = (Real) m_channelSampleRate / (Real) sampleRate;
|
|
|
|
SSBFilter->create_filter(m_LowCutoff / (float) sampleRate, m_Bandwidth / (float) sampleRate, m_settings.m_filterBank[m_settings.m_filterIndex].m_fftWindow);
|
|
DSBFilter->create_dsb_filter(m_Bandwidth / (float) sampleRate, m_settings.m_filterBank[m_settings.m_filterIndex].m_fftWindow);
|
|
|
|
m_lowpassI.create(101, sampleRate, m_Bandwidth * 1.2);
|
|
m_lowpassQ.create(101, sampleRate, m_Bandwidth * 1.2);
|
|
|
|
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, m_agcTarget);
|
|
m_agc.setStepDownDelay(agcNbSamples);
|
|
m_agcNbSamples = agcNbSamples;
|
|
}
|
|
|
|
if (m_agcThresholdGate != agcThresholdGate)
|
|
{
|
|
m_agc.setGate(agcThresholdGate);
|
|
m_agcThresholdGate = agcThresholdGate;
|
|
}
|
|
|
|
m_audioFifo.setSize(sampleRate);
|
|
m_audioSampleRate = sampleRate;
|
|
m_audioBuffer.resize(sampleRate / 10);
|
|
m_audioBufferFill = 0;
|
|
|
|
QList<ObjectPipe*> pipes;
|
|
MainCore::instance()->getMessagePipes().getMessagePipes(m_channel, "reportdemod", pipes);
|
|
|
|
if (pipes.size() > 0)
|
|
{
|
|
for (const auto& pipe : pipes)
|
|
{
|
|
MessageQueue* messageQueue = qobject_cast<MessageQueue*>(pipe->m_element);
|
|
|
|
if (messageQueue)
|
|
{
|
|
MainCore::MsgChannelDemodReport *msg = MainCore::MsgChannelDemodReport::create(m_channel, sampleRate);
|
|
messageQueue->push(msg);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void SSBDemodSink::applySettings(const SSBDemodSettings& settings, bool force)
|
|
{
|
|
qDebug() << "SSBDemodSink::applySettings:"
|
|
<< " m_inputFrequencyOffset: " << settings.m_inputFrequencyOffset
|
|
<< " m_filterIndex: " << settings.m_filterIndex
|
|
<< " [m_spanLog2: " << settings.m_filterBank[settings.m_filterIndex].m_spanLog2
|
|
<< " m_rfBandwidth: " << settings.m_filterBank[settings.m_filterIndex].m_rfBandwidth
|
|
<< " m_lowCutoff: " << settings.m_filterBank[settings.m_filterIndex].m_lowCutoff
|
|
<< " m_fftWindow: " << settings.m_filterBank[settings.m_filterIndex].m_fftWindow << "]"
|
|
<< " m_volume: " << settings.m_volume
|
|
<< " 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_dnr: " << settings.m_dnr
|
|
<< " m_dnrScheme: " << settings.m_dnrScheme
|
|
<< " m_dnrAboveAvgFactor: " << settings.m_dnrAboveAvgFactor
|
|
<< " m_dnrSigmaFactor: " << settings.m_dnrSigmaFactor
|
|
<< " m_dnrNbPeaks: " << settings.m_dnrNbPeaks
|
|
<< " m_dnrAlpha: " << settings.m_dnrAlpha
|
|
<< " m_audioDeviceName: " << settings.m_audioDeviceName
|
|
<< " m_streamIndex: " << settings.m_streamIndex
|
|
<< " m_useReverseAPI: " << settings.m_useReverseAPI
|
|
<< " m_reverseAPIAddress: " << settings.m_reverseAPIAddress
|
|
<< " m_reverseAPIPort: " << settings.m_reverseAPIPort
|
|
<< " m_reverseAPIDeviceIndex: " << settings.m_reverseAPIDeviceIndex
|
|
<< " m_reverseAPIChannelIndex: " << settings.m_reverseAPIChannelIndex
|
|
<< " force: " << force;
|
|
|
|
if((m_settings.m_filterBank[m_settings.m_filterIndex].m_rfBandwidth != settings.m_filterBank[settings.m_filterIndex].m_rfBandwidth) ||
|
|
(m_settings.m_filterBank[m_settings.m_filterIndex].m_lowCutoff != settings.m_filterBank[settings.m_filterIndex].m_lowCutoff) ||
|
|
(m_settings.m_filterBank[m_settings.m_filterIndex].m_fftWindow != settings.m_filterBank[settings.m_filterIndex].m_fftWindow) || force)
|
|
{
|
|
float band, lowCutoff;
|
|
|
|
band = settings.m_filterBank[settings.m_filterIndex].m_rfBandwidth;
|
|
lowCutoff = settings.m_filterBank[settings.m_filterIndex].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;
|
|
|
|
Real interpolatorBandwidth = (m_Bandwidth * 1.5f) > m_channelSampleRate ? m_channelSampleRate : (m_Bandwidth * 1.5f);
|
|
m_interpolator.create(16, m_channelSampleRate, interpolatorBandwidth, 2.0f);
|
|
m_interpolatorDistanceRemain = 0;
|
|
m_interpolatorDistance = (Real) m_channelSampleRate / (Real) m_audioSampleRate;
|
|
SSBFilter->create_filter(m_LowCutoff / (float) m_audioSampleRate, m_Bandwidth / (float) m_audioSampleRate, settings.m_filterBank[settings.m_filterIndex].m_fftWindow);
|
|
DSBFilter->create_dsb_filter(m_Bandwidth / (float) m_audioSampleRate, settings.m_filterBank[settings.m_filterIndex].m_fftWindow);
|
|
m_lowpassI.create(101, m_audioSampleRate, m_Bandwidth * 1.2);
|
|
m_lowpassQ.create(101, m_audioSampleRate, m_Bandwidth * 1.2);
|
|
}
|
|
|
|
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_agc.resize(agcNbSamples, agcNbSamples/2, m_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_agcClamping = agcClamping;
|
|
}
|
|
|
|
qDebug() << "SBDemodSink::applySettings: AGC:"
|
|
<< " agcNbSamples: " << agcNbSamples
|
|
<< " agcPowerThreshold: " << agcPowerThreshold
|
|
<< " agcThresholdGate: " << agcThresholdGate
|
|
<< " agcClamping: " << agcClamping;
|
|
}
|
|
|
|
if ((m_settings.m_dnr != settings.m_dnr) || force) {
|
|
setDNR(settings.m_dnr);
|
|
}
|
|
|
|
if ((m_settings.m_dnrScheme != settings.m_dnrScheme) || force) {
|
|
SSBFilter->setDNRScheme((FFTNoiseReduction::Scheme) settings.m_dnrScheme);
|
|
}
|
|
|
|
if ((m_settings.m_dnrAboveAvgFactor != settings.m_dnrAboveAvgFactor) || force) {
|
|
SSBFilter->setDNRAboveAvgFactor(settings.m_dnrAboveAvgFactor);
|
|
}
|
|
|
|
if ((m_settings.m_dnrSigmaFactor != settings.m_dnrSigmaFactor) || force) {
|
|
SSBFilter->setDNRSigmaFactor(settings.m_dnrSigmaFactor);
|
|
}
|
|
|
|
if ((m_settings.m_dnrNbPeaks != settings.m_dnrNbPeaks) || force) {
|
|
SSBFilter->setDNRNbPeaks(settings.m_dnrNbPeaks);
|
|
}
|
|
|
|
if ((m_settings.m_dnrAlpha != settings.m_dnrAlpha) || force) {
|
|
SSBFilter->setDNRAlpha(settings.m_dnrAlpha);
|
|
}
|
|
|
|
m_spanLog2 = settings.m_filterBank[settings.m_filterIndex].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;
|
|
}
|