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sdrangel/plugins/channeltx/modssb/ssbmod.cpp

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///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2016 Edouard Griffiths, F4EXB //
// //
// 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 "ssbmod.h"
#include <QTime>
#include <QDebug>
#include <QMutexLocker>
#include <stdio.h>
#include <complex.h>
#include <dsp/upchannelizer.h>
#include "dsp/dspengine.h"
#include "dsp/pidcontroller.h"
#include "dsp/threadedbasebandsamplesource.h"
#include "device/devicesinkapi.h"
#include "util/db.h"
MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureSSBMod, Message)
MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureChannelizer, Message)
MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureFileSourceName, Message)
MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureFileSourceSeek, Message)
MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureAFInput, Message)
MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureFileSourceStreamTiming, Message)
MESSAGE_CLASS_DEFINITION(SSBMod::MsgReportFileSourceStreamData, Message)
MESSAGE_CLASS_DEFINITION(SSBMod::MsgReportFileSourceStreamTiming, Message)
const int SSBMod::m_levelNbSamples = 480; // every 10ms
const int SSBMod::m_ssbFftLen = 1024;
SSBMod::SSBMod(DeviceSinkAPI *deviceAPI, BasebandSampleSink* sampleSink) :
m_deviceAPI(deviceAPI),
m_SSBFilter(0),
m_DSBFilter(0),
m_SSBFilterBuffer(0),
m_DSBFilterBuffer(0),
m_SSBFilterBufferIndex(0),
m_DSBFilterBufferIndex(0),
m_sampleSink(sampleSink),
m_movingAverage(40, 0),
m_audioFifo(4800),
m_settingsMutex(QMutex::Recursive),
m_fileSize(0),
m_recordLength(0),
m_sampleRate(48000),
m_afInput(SSBModInputNone),
m_levelCalcCount(0),
m_peakLevel(0.0f),
m_levelSum(0.0f),
m_inAGC(9600, 0.2, 1e-4)
{
setObjectName("SSBMod");
m_SSBFilter = new fftfilt(m_settings.m_lowCutoff / m_settings.m_audioSampleRate, m_settings.m_bandwidth / m_settings.m_audioSampleRate, m_ssbFftLen);
m_DSBFilter = new fftfilt((2.0f * m_settings.m_bandwidth) / m_settings.m_audioSampleRate, 2 * m_ssbFftLen);
m_SSBFilterBuffer = new Complex[m_ssbFftLen>>1]; // filter returns data exactly half of its size
m_DSBFilterBuffer = new Complex[m_ssbFftLen];
memset(m_SSBFilterBuffer, 0, sizeof(Complex)*(m_ssbFftLen>>1));
memset(m_DSBFilterBuffer, 0, sizeof(Complex)*(m_ssbFftLen));
m_audioBuffer.resize(1<<14);
m_audioBufferFill = 0;
m_sum.real(0.0f);
m_sum.imag(0.0f);
m_undersampleCount = 0;
m_sumCount = 0;
m_movingAverage.resize(16, 0);
m_magsq = 0.0;
m_toneNco.setFreq(1000.0, m_settings.m_audioSampleRate);
DSPEngine::instance()->addAudioSource(&m_audioFifo);
// CW keyer
m_cwKeyer.setSampleRate(m_settings.m_audioSampleRate);
m_cwKeyer.setWPM(13);
m_cwKeyer.setMode(CWKeyer::CWNone);
m_inAGC.setGate(m_settings.m_agcThresholdGate);
m_inAGC.setStepDownDelay(m_settings.m_agcThresholdDelay);
m_inAGC.setClamping(true);
m_channelizer = new UpChannelizer(this);
m_threadedChannelizer = new ThreadedBasebandSampleSource(m_channelizer, this);
m_deviceAPI->addThreadedSource(m_threadedChannelizer);
applySettings(m_settings, true);
}
SSBMod::~SSBMod()
{
if (m_SSBFilter) {
delete m_SSBFilter;
}
if (m_DSBFilter) {
delete m_DSBFilter;
}
if (m_SSBFilterBuffer) {
delete m_SSBFilterBuffer;
}
if (m_DSBFilterBuffer) {
delete m_DSBFilterBuffer;
}
DSPEngine::instance()->removeAudioSource(&m_audioFifo);
m_deviceAPI->removeThreadedSource(m_threadedChannelizer);
delete m_threadedChannelizer;
delete m_channelizer;
}
void SSBMod::pull(Sample& sample)
{
Complex ci;
m_settingsMutex.lock();
if (m_interpolatorDistance > 1.0f) // decimate
{
modulateSample();
while (!m_interpolator.decimate(&m_interpolatorDistanceRemain, m_modSample, &ci))
{
modulateSample();
}
}
else
{
if (m_interpolator.interpolate(&m_interpolatorDistanceRemain, m_modSample, &ci))
{
modulateSample();
}
}
m_interpolatorDistanceRemain += m_interpolatorDistance;
ci *= m_carrierNco.nextIQ(); // shift to carrier frequency
ci *= 29204.0f; //scaling at -1 dB to account for possible filter overshoot
m_settingsMutex.unlock();
Real magsq = ci.real() * ci.real() + ci.imag() * ci.imag();
magsq /= (1<<30);
m_movingAverage.feed(magsq);
m_magsq = m_movingAverage.average();
sample.m_real = (FixReal) ci.real();
sample.m_imag = (FixReal) ci.imag();
}
void SSBMod::pullAudio(int nbSamples)
{
unsigned int nbSamplesAudio = nbSamples * ((Real) m_settings.m_audioSampleRate / (Real) m_settings.m_basebandSampleRate);
if (nbSamplesAudio > m_audioBuffer.size())
{
m_audioBuffer.resize(nbSamplesAudio);
}
m_audioFifo.read(reinterpret_cast<quint8*>(&m_audioBuffer[0]), nbSamplesAudio, 10);
m_audioBufferFill = 0;
}
void SSBMod::modulateSample()
{
pullAF(m_modSample);
calculateLevel(m_modSample);
m_audioBufferFill++;
}
void SSBMod::pullAF(Complex& sample)
{
if (m_settings.m_audioMute)
{
sample.real(0.0f);
sample.imag(0.0f);
return;
}
Complex ci;
fftfilt::cmplx *filtered;
int n_out = 0;
int decim = 1<<(m_settings.m_spanLog2 - 1);
unsigned char decim_mask = decim - 1; // counter LSB bit mask for decimation by 2^(m_scaleLog2 - 1)
switch (m_afInput)
{
case SSBModInputTone:
if (m_settings.m_dsb)
{
Real t = m_toneNco.next()/1.25;
sample.real(t);
sample.imag(t);
}
else
{
if (m_settings.m_usb) {
sample = m_toneNco.nextIQ();
} else {
sample = m_toneNco.nextQI();
}
}
break;
case SSBModInputFile:
// Monaural (mono):
// sox f4exb_call.wav --encoding float --endian little f4exb_call.raw
// ffplay -f f32le -ar 48k -ac 1 f4exb_call.raw
// Binaural (stereo):
// sox f4exb_call.wav --encoding float --endian little f4exb_call.raw
// ffplay -f f32le -ar 48k -ac 2 f4exb_call.raw
if (m_ifstream.is_open())
{
if (m_ifstream.eof())
{
if (m_settings.m_playLoop)
{
m_ifstream.clear();
m_ifstream.seekg(0, std::ios::beg);
}
}
if (m_ifstream.eof())
{
ci.real(0.0f);
ci.imag(0.0f);
}
else
{
if (m_settings.m_audioBinaural)
{
Complex c;
m_ifstream.read(reinterpret_cast<char*>(&c), sizeof(Complex));
if (m_settings.m_audioFlipChannels)
{
ci.real(c.imag() * m_settings.m_volumeFactor);
ci.imag(c.real() * m_settings.m_volumeFactor);
}
else
{
ci = c * m_settings.m_volumeFactor;
}
}
else
{
Real real;
m_ifstream.read(reinterpret_cast<char*>(&real), sizeof(Real));
if (m_settings.m_agc)
{
ci.real(real);
ci.imag(0.0f);
m_inAGC.feed(ci);
ci *= m_settings.m_volumeFactor;
}
else
{
ci.real(real * m_settings.m_volumeFactor);
ci.imag(0.0f);
}
}
}
}
else
{
ci.real(0.0f);
ci.imag(0.0f);
}
break;
case SSBModInputAudio:
if (m_settings.m_audioBinaural)
{
if (m_settings.m_audioFlipChannels)
{
ci.real((m_audioBuffer[m_audioBufferFill].r / 32768.0f) * m_settings.m_volumeFactor);
ci.imag((m_audioBuffer[m_audioBufferFill].l / 32768.0f) * m_settings.m_volumeFactor);
}
else
{
ci.real((m_audioBuffer[m_audioBufferFill].l / 32768.0f) * m_settings.m_volumeFactor);
ci.imag((m_audioBuffer[m_audioBufferFill].r / 32768.0f) * m_settings.m_volumeFactor);
}
}
else
{
if (m_settings.m_agc)
{
ci.real(((m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) / 65536.0f));
ci.imag(0.0f);
m_inAGC.feed(ci);
ci *= m_settings.m_volumeFactor;
}
else
{
ci.real(((m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) / 65536.0f) * m_settings.m_volumeFactor);
ci.imag(0.0f);
}
}
break;
case SSBModInputCWTone:
Real fadeFactor;
if (m_cwKeyer.getSample())
{
m_cwKeyer.getCWSmoother().getFadeSample(true, fadeFactor);
if (m_settings.m_dsb)
{
Real t = m_toneNco.next() * fadeFactor;
sample.real(t);
sample.imag(t);
}
else
{
if (m_settings.m_usb) {
sample = m_toneNco.nextIQ() * fadeFactor;
} else {
sample = m_toneNco.nextQI() * fadeFactor;
}
}
}
else
{
if (m_cwKeyer.getCWSmoother().getFadeSample(false, fadeFactor))
{
if (m_settings.m_dsb)
{
Real t = (m_toneNco.next() * fadeFactor)/1.25;
sample.real(t);
sample.imag(t);
}
else
{
if (m_settings.m_usb) {
sample = m_toneNco.nextIQ() * fadeFactor;
} else {
sample = m_toneNco.nextQI() * fadeFactor;
}
}
}
else
{
sample.real(0.0f);
sample.imag(0.0f);
m_toneNco.setPhase(0);
}
}
break;
case SSBModInputNone:
default:
break;
}
if ((m_afInput == SSBModInputFile) || (m_afInput == SSBModInputAudio)) // real audio
{
if (m_settings.m_dsb)
{
n_out = m_DSBFilter->runDSB(ci, &filtered);
if (n_out > 0)
{
memcpy((void *) m_DSBFilterBuffer, (const void *) filtered, n_out*sizeof(Complex));
m_DSBFilterBufferIndex = 0;
}
sample = m_DSBFilterBuffer[m_DSBFilterBufferIndex];
m_DSBFilterBufferIndex++;
}
else
{
n_out = m_SSBFilter->runSSB(ci, &filtered, m_settings.m_usb);
if (n_out > 0)
{
memcpy((void *) m_SSBFilterBuffer, (const void *) filtered, n_out*sizeof(Complex));
m_SSBFilterBufferIndex = 0;
}
sample = m_SSBFilterBuffer[m_SSBFilterBufferIndex];
m_SSBFilterBufferIndex++;
}
if (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 += filtered[i];
if (!(m_undersampleCount++ & decim_mask))
{
Real avgr = (m_sum.real() / decim) * 29204.0f; //scaling at -1 dB to account for possible filter overshoot
Real avgi = (m_sum.imag() / decim) * 29204.0f;
// m_magsqSpectrum = (avgr * avgr + avgi * avgi) / (1<<30);
//
// m_magsqSum += m_magsqSpectrum;
//
// if (m_magsqSpectrum > m_magsqPeak)
// {
// m_magsqPeak = m_magsqSpectrum;
// }
//
// m_magsqCount++;
if (!m_settings.m_dsb & !m_settings.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);
}
}
}
} // Real audio
else if ((m_afInput == SSBModInputTone) || (m_afInput == SSBModInputCWTone)) // tone
{
m_sum += sample;
if (!(m_undersampleCount++ & decim_mask))
{
Real avgr = (m_sum.real() / decim) * 29204.0f; //scaling at -1 dB to account for possible filter overshoot
Real avgi = (m_sum.imag() / decim) * 29204.0f;
// m_magsqSpectrum = (avgr * avgr + avgi * avgi) / (1<<30);
//
// m_magsqSum += m_magsqSpectrum;
//
// if (m_magsqSpectrum > m_magsqPeak)
// {
// m_magsqPeak = m_magsqSpectrum;
// }
//
// m_magsqCount++;
if (!m_settings.m_dsb & !m_settings.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_sumCount < (m_settings.m_dsb ? m_ssbFftLen : m_ssbFftLen>>1))
{
n_out = 0;
m_sumCount++;
}
else
{
n_out = m_sumCount;
m_sumCount = 0;
}
}
if (n_out > 0)
{
if (m_sampleSink != 0)
{
m_sampleSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), !m_settings.m_dsb);
}
m_sampleBuffer.clear();
}
}
void SSBMod::calculateLevel(Complex& sample)
{
Real t = sample.real(); // TODO: possibly adjust depending on sample type
if (m_levelCalcCount < m_levelNbSamples)
{
m_peakLevel = std::max(std::fabs(m_peakLevel), t);
m_levelSum += t * t;
m_levelCalcCount++;
}
else
{
qreal rmsLevel = sqrt(m_levelSum / m_levelNbSamples);
//qDebug("NFMMod::calculateLevel: %f %f", rmsLevel, m_peakLevel);
emit levelChanged(rmsLevel, m_peakLevel, m_levelNbSamples);
m_peakLevel = 0.0f;
m_levelSum = 0.0f;
m_levelCalcCount = 0;
}
}
void SSBMod::start()
{
qDebug() << "SSBMod::start: m_outputSampleRate: " << m_settings.m_outputSampleRate
<< " m_inputFrequencyOffset: " << m_settings.m_inputFrequencyOffset;
m_audioFifo.clear();
}
void SSBMod::stop()
{
}
bool SSBMod::handleMessage(const Message& cmd)
{
if (UpChannelizer::MsgChannelizerNotification::match(cmd))
{
UpChannelizer::MsgChannelizerNotification& notif = (UpChannelizer::MsgChannelizerNotification&) cmd;
SSBModSettings settings = m_settings;
settings.m_basebandSampleRate = notif.getBasebandSampleRate();
settings.m_outputSampleRate = notif.getSampleRate();
settings.m_inputFrequencyOffset = notif.getFrequencyOffset();
applySettings(settings);
qDebug() << "SSBMod::handleMessage: MsgChannelizerNotification:"
<< " m_basebandSampleRate: " << settings.m_basebandSampleRate
<< " m_outputSampleRate: " << settings.m_outputSampleRate
<< " m_inputFrequencyOffset: " << settings.m_inputFrequencyOffset;
return true;
}
else if (MsgConfigureChannelizer::match(cmd))
{
MsgConfigureChannelizer& cfg = (MsgConfigureChannelizer&) cmd;
m_channelizer->configure(m_channelizer->getInputMessageQueue(),
cfg.getSampleRate(),
cfg.getCenterFrequency());
qDebug() << "SSBMod::handleMessage: MsgConfigureChannelizer: sampleRate: " << cfg.getSampleRate()
<< " centerFrequency: " << cfg.getCenterFrequency();
return true;
}
else if (MsgConfigureSSBMod::match(cmd))
{
float band, lowCutoff;
MsgConfigureSSBMod& cfg = (MsgConfigureSSBMod&) cmd;
SSBModSettings settings = cfg.getSettings();
// These settings are set with UpChannelizer::MsgChannelizerNotification
settings.m_basebandSampleRate = m_settings.m_basebandSampleRate;
settings.m_outputSampleRate = m_settings.m_outputSampleRate;
settings.m_inputFrequencyOffset = m_settings.m_inputFrequencyOffset;
band = settings.m_bandwidth;
lowCutoff = settings.m_lowCutoff;
if (band < 0) // negative means LSB
{
band = -band; // turn to positive
lowCutoff = -lowCutoff;
settings.m_usb = false; // and take note of side band
}
else
{
settings.m_usb = true;
}
if (band < 100.0f) // at least 100 Hz
{
band = 100.0f;
lowCutoff = 0;
}
if (band - lowCutoff < 100.0f) {
lowCutoff = band - 100.0f;
}
settings.m_bandwidth = band;
settings.m_lowCutoff = lowCutoff;
applySettings(settings, cfg.getForce());
qDebug() << "SSBMod::handleMessage: MsgConfigureSSBMod:"
<< " m_bandwidth: " << settings.m_bandwidth
<< " m_lowCutoff: " << settings.m_lowCutoff
<< " m_toneFrequency: " << settings.m_toneFrequency
<< " m_volumeFactor: " << settings.m_volumeFactor
<< " m_spanLog2: " << settings.m_spanLog2
<< " m_outputSampleRate: " << m_settings.m_outputSampleRate
<< " m_audioSampleRate: " << settings.m_audioSampleRate
<< " m_audioBinaural: " << settings.m_audioBinaural
<< " m_audioFlipChannels: " << settings.m_audioFlipChannels
<< " m_dsb: " << settings.m_dsb
<< " m_audioMute: " << settings.m_audioMute
<< " m_playLoop: " << settings.m_playLoop
<< " m_agc: " << settings.m_agc
<< " m_agcTime: " << settings.m_agcTime
<< " m_agcOrder: " << settings.m_agcOrder
<< " m_agcThresholdEnable: " << settings.m_agcThresholdEnable
<< " m_agcThreshold: " << settings.m_agcThreshold
<< " m_agcThresholdGate: " << settings.m_agcThresholdGate
<< " m_agcThresholdDelay: " << settings.m_agcThresholdDelay
<< " force: " << cfg.getForce();
return true;
}
else if (MsgConfigureFileSourceName::match(cmd))
{
MsgConfigureFileSourceName& conf = (MsgConfigureFileSourceName&) cmd;
m_fileName = conf.getFileName();
openFileStream();
return true;
}
else if (MsgConfigureFileSourceSeek::match(cmd))
{
MsgConfigureFileSourceSeek& conf = (MsgConfigureFileSourceSeek&) cmd;
int seekPercentage = conf.getPercentage();
seekFileStream(seekPercentage);
return true;
}
else if (MsgConfigureAFInput::match(cmd))
{
MsgConfigureAFInput& conf = (MsgConfigureAFInput&) cmd;
m_afInput = conf.getAFInput();
return true;
}
else if (MsgConfigureFileSourceStreamTiming::match(cmd))
{
std::size_t samplesCount;
if (m_ifstream.eof()) {
samplesCount = m_fileSize / sizeof(Real);
} else {
samplesCount = m_ifstream.tellg() / sizeof(Real);
}
MsgReportFileSourceStreamTiming *report;
report = MsgReportFileSourceStreamTiming::create(samplesCount);
getMessageQueueToGUI()->push(report);
return true;
}
else
{
return false;
}
}
void SSBMod::openFileStream()
{
if (m_ifstream.is_open()) {
m_ifstream.close();
}
m_ifstream.open(m_fileName.toStdString().c_str(), std::ios::binary | std::ios::ate);
m_fileSize = m_ifstream.tellg();
m_ifstream.seekg(0,std::ios_base::beg);
m_sampleRate = 48000; // fixed rate
m_recordLength = m_fileSize / (sizeof(Real) * m_sampleRate);
qDebug() << "SSBMod::openFileStream: " << m_fileName.toStdString().c_str()
<< " fileSize: " << m_fileSize << "bytes"
<< " length: " << m_recordLength << " seconds";
MsgReportFileSourceStreamData *report;
report = MsgReportFileSourceStreamData::create(m_sampleRate, m_recordLength);
getMessageQueueToGUI()->push(report);
}
void SSBMod::seekFileStream(int seekPercentage)
{
QMutexLocker mutexLocker(&m_settingsMutex);
if (m_ifstream.is_open())
{
int seekPoint = ((m_recordLength * seekPercentage) / 100) * m_sampleRate;
seekPoint *= sizeof(Real);
m_ifstream.clear();
m_ifstream.seekg(seekPoint, std::ios::beg);
}
}
void SSBMod::applySettings(const SSBModSettings& settings, bool force)
{
if ((settings.m_bandwidth != m_settings.m_bandwidth) ||
(settings.m_lowCutoff != m_settings.m_lowCutoff) ||
(settings.m_audioSampleRate != m_settings.m_audioSampleRate) || force)
{
m_settingsMutex.lock();
m_SSBFilter->create_filter(settings.m_lowCutoff / settings.m_audioSampleRate, settings.m_bandwidth / settings.m_audioSampleRate);
m_DSBFilter->create_dsb_filter((2.0f * settings.m_bandwidth) / settings.m_audioSampleRate);
m_settingsMutex.unlock();
}
if ((settings.m_inputFrequencyOffset != m_settings.m_inputFrequencyOffset) ||
(settings.m_outputSampleRate != m_settings.m_outputSampleRate) || force)
{
m_settingsMutex.lock();
m_carrierNco.setFreq(settings.m_inputFrequencyOffset, settings.m_outputSampleRate);
m_settingsMutex.unlock();
}
if((settings.m_outputSampleRate != m_settings.m_outputSampleRate) ||
(settings.m_bandwidth != m_settings.m_bandwidth) ||
(settings.m_audioSampleRate != m_settings.m_audioSampleRate) || force)
{
m_settingsMutex.lock();
m_interpolatorDistanceRemain = 0;
m_interpolatorConsumed = false;
m_interpolatorDistance = (Real) settings.m_audioSampleRate / (Real) settings.m_outputSampleRate;
m_interpolator.create(48, settings.m_audioSampleRate, settings.m_bandwidth, 3.0);
m_settingsMutex.unlock();
}
if ((settings.m_toneFrequency != m_settings.m_toneFrequency) ||
(settings.m_audioSampleRate != m_settings.m_audioSampleRate) || force)
{
m_settingsMutex.lock();
m_toneNco.setFreq(settings.m_toneFrequency, settings.m_audioSampleRate);
m_settingsMutex.unlock();
}
if ((settings.m_audioSampleRate != m_settings.m_audioSampleRate) || force)
{
m_settingsMutex.lock();
m_cwKeyer.setSampleRate(settings.m_audioSampleRate);
m_settingsMutex.unlock();
}
if ((settings.m_dsb != m_settings.m_dsb) || force)
{
if (settings.m_dsb)
{
memset(m_DSBFilterBuffer, 0, sizeof(Complex)*(m_ssbFftLen));
m_DSBFilterBufferIndex = 0;
}
else
{
memset(m_SSBFilterBuffer, 0, sizeof(Complex)*(m_ssbFftLen>>1));
m_SSBFilterBufferIndex = 0;
}
}
if ((settings.m_agcTime != m_settings.m_agcTime) ||
(settings.m_agcOrder != m_settings.m_agcOrder) || force)
{
m_inAGC.resize(settings.m_agcTime, settings.m_agcOrder);
}
if ((settings.m_agcThresholdEnable != m_settings.m_agcThresholdEnable) || force)
{
m_inAGC.setThresholdEnable(settings.m_agcThresholdEnable);
}
if ((settings.m_agcThreshold != m_settings.m_agcThreshold) || force)
{
m_inAGC.setThreshold(settings.m_agcThreshold);
}
if ((settings.m_agcThresholdGate != m_settings.m_agcThresholdGate) || force)
{
m_inAGC.setGate(settings.m_agcThresholdGate);
}
if ((settings.m_agcThresholdDelay != m_settings.m_agcThresholdDelay) || force)
{
m_inAGC.setStepDownDelay(settings.m_agcThresholdDelay);
}
m_settings = settings;
}
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