mirror of
https://github.com/f4exb/sdrangel.git
synced 2026-06-07 08:24:43 -04:00
Sonar fixes
This commit is contained in:
f4exb
@@ -30,22 +30,13 @@ const int SSBModSource::m_ssbFftLen = 1024;
|
||||
const int SSBModSource::m_levelNbSamples = 480; // every 10ms
|
||||
|
||||
SSBModSource::SSBModSource() :
|
||||
m_channelSampleRate(48000),
|
||||
m_channelFrequencyOffset(0),
|
||||
m_spectrumSink(nullptr),
|
||||
m_audioSampleRate(48000),
|
||||
m_audioFifo(12000),
|
||||
m_feedbackAudioFifo(12000),
|
||||
m_levelCalcCount(0),
|
||||
m_peakLevel(0.0f),
|
||||
m_levelSum(0.0f),
|
||||
m_ifstream(nullptr),
|
||||
m_cwKeyer(nullptr)
|
||||
m_feedbackAudioFifo(12000)
|
||||
{
|
||||
m_audioFifo.setLabel("SSBModSource.m_audioFifo");
|
||||
m_feedbackAudioFifo.setLabel("SSBModSource.m_feedbackAudioFifo");
|
||||
m_SSBFilter = new fftfilt(m_settings.m_lowCutoff / m_audioSampleRate, m_settings.m_bandwidth / m_audioSampleRate, m_ssbFftLen);
|
||||
m_DSBFilter = new fftfilt((2.0f * m_settings.m_bandwidth) / m_audioSampleRate, 2 * m_ssbFftLen);
|
||||
m_SSBFilter = new fftfilt(m_settings.m_lowCutoff / (float) m_audioSampleRate, m_settings.m_bandwidth / (float) m_audioSampleRate, m_ssbFftLen);
|
||||
m_DSBFilter = new fftfilt((2.0f * m_settings.m_bandwidth) / (float) 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];
|
||||
std::fill(m_SSBFilterBuffer, m_SSBFilterBuffer+(m_ssbFftLen>>1), Complex{0,0});
|
||||
@@ -68,15 +59,15 @@ SSBModSource::SSBModSource() :
|
||||
m_sumCount = 0;
|
||||
|
||||
m_magsq = 0.0;
|
||||
m_toneNco.setFreq(1000.0, m_audioSampleRate);
|
||||
m_toneNco.setFreq(1000.0, (float) m_audioSampleRate);
|
||||
|
||||
m_audioCompressor.initSimple(
|
||||
m_audioSampleRate,
|
||||
m_settings.m_cmpPreGainDB, // pregain (dB)
|
||||
m_settings.m_cmpThresholdDB, // threshold (dB)
|
||||
(float) m_settings.m_cmpPreGainDB, // pregain (dB)
|
||||
(float) m_settings.m_cmpThresholdDB, // threshold (dB)
|
||||
20, // knee (dB)
|
||||
12, // ratio (dB)
|
||||
0.003, // attack (s)
|
||||
0.003f,// attack (s)
|
||||
0.25 // release (s)
|
||||
);
|
||||
|
||||
@@ -140,7 +131,7 @@ void SSBModSource::pullOne(Sample& sample)
|
||||
|
||||
void SSBModSource::prefetch(unsigned int nbSamples)
|
||||
{
|
||||
unsigned int nbSamplesAudio = nbSamples * ((Real) m_audioSampleRate / (Real) m_channelSampleRate);
|
||||
unsigned int nbSamplesAudio = (nbSamples * (unsigned int) ((Real) m_audioSampleRate / (Real) m_channelSampleRate));
|
||||
pullAudio(nbSamplesAudio);
|
||||
}
|
||||
|
||||
@@ -174,13 +165,16 @@ void SSBModSource::modulateSample()
|
||||
|
||||
if (m_settings.m_audioBinaural)
|
||||
{
|
||||
m_demodBuffer[m_demodBufferFill++] = m_modSample.real() * std::numeric_limits<int16_t>::max();
|
||||
m_demodBuffer[m_demodBufferFill++] = m_modSample.imag() * std::numeric_limits<int16_t>::max();
|
||||
m_demodBuffer[m_demodBufferFill] = (qint16) (m_modSample.real() * std::numeric_limits<int16_t>::max());
|
||||
m_demodBufferFill++;
|
||||
m_demodBuffer[m_demodBufferFill] = (qint16) (m_modSample.imag() * std::numeric_limits<int16_t>::max());
|
||||
m_demodBufferFill++;
|
||||
}
|
||||
else
|
||||
{
|
||||
// take projection on real axis
|
||||
m_demodBuffer[m_demodBufferFill++] = m_modSample.real() * std::numeric_limits<int16_t>::max();
|
||||
m_demodBuffer[m_demodBufferFill] = (qint16) (m_modSample.real() * std::numeric_limits<int16_t>::max());
|
||||
m_demodBufferFill++;
|
||||
}
|
||||
|
||||
if (m_demodBufferFill >= m_demodBuffer.size())
|
||||
@@ -188,13 +182,11 @@ void SSBModSource::modulateSample()
|
||||
QList<ObjectPipe*> dataPipes;
|
||||
MainCore::instance()->getDataPipes().getDataPipes(m_channel, "demod", dataPipes);
|
||||
|
||||
if (dataPipes.size() > 0)
|
||||
if (!dataPipes.empty())
|
||||
{
|
||||
QList<ObjectPipe*>::iterator it = dataPipes.begin();
|
||||
|
||||
for (; it != dataPipes.end(); ++it)
|
||||
for (auto& dataPipe : dataPipes)
|
||||
{
|
||||
DataFifo *fifo = qobject_cast<DataFifo*>((*it)->m_element);
|
||||
DataFifo *fifo = qobject_cast<DataFifo*>(dataPipe->m_element);
|
||||
|
||||
if (fifo)
|
||||
{
|
||||
@@ -225,42 +217,33 @@ void SSBModSource::pullAF(Complex& sample)
|
||||
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)
|
||||
auto decim_mask = (unsigned char) ((decim - 1) & 0xFF); // counter LSB bit mask for decimation by 2^(m_scaleLog2 - 1)
|
||||
|
||||
switch (m_settings.m_modAFInput)
|
||||
{
|
||||
case SSBModSettings::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();
|
||||
}
|
||||
}
|
||||
if (m_settings.m_dsb)
|
||||
{
|
||||
Real t = m_toneNco.next()/1.25f;
|
||||
sample.real(t);
|
||||
sample.imag(t);
|
||||
}
|
||||
else
|
||||
{
|
||||
if (m_settings.m_usb) {
|
||||
sample = m_toneNco.nextIQ();
|
||||
} else {
|
||||
sample = m_toneNco.nextQI();
|
||||
}
|
||||
}
|
||||
break;
|
||||
case SSBModSettings::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 && m_ifstream->is_open())
|
||||
{
|
||||
if (m_ifstream->eof())
|
||||
if (m_ifstream->eof() && m_settings.m_playLoop)
|
||||
{
|
||||
if (m_settings.m_playLoop)
|
||||
{
|
||||
m_ifstream->clear();
|
||||
m_ifstream->seekg(0, std::ios::beg);
|
||||
}
|
||||
m_ifstream->clear();
|
||||
m_ifstream->seekg(0, std::ios::beg);
|
||||
}
|
||||
|
||||
if (m_ifstream->eof())
|
||||
@@ -270,38 +253,38 @@ void SSBModSource::pullAF(Complex& sample)
|
||||
}
|
||||
else
|
||||
{
|
||||
if (m_settings.m_audioBinaural)
|
||||
{
|
||||
Complex c;
|
||||
m_ifstream->read(reinterpret_cast<char*>(&c), sizeof(Complex));
|
||||
if (m_settings.m_audioBinaural)
|
||||
{
|
||||
Complex c;
|
||||
m_ifstream->read(reinterpret_cast<char*>(&c), sizeof(Complex));
|
||||
|
||||
if (m_settings.m_audioFlipChannels)
|
||||
{
|
||||
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
|
||||
{
|
||||
}
|
||||
else
|
||||
{
|
||||
ci = c * m_settings.m_volumeFactor;
|
||||
}
|
||||
}
|
||||
else
|
||||
{
|
||||
Real real;
|
||||
m_ifstream->read(reinterpret_cast<char*>(&real), sizeof(Real));
|
||||
m_ifstream->read(reinterpret_cast<char*>(&real), sizeof(Real));
|
||||
|
||||
if (m_settings.m_agc)
|
||||
{
|
||||
if (m_settings.m_agc)
|
||||
{
|
||||
ci.real(clamp<float>(m_audioCompressor.compress(real), -1.0f, 1.0f));
|
||||
ci.imag(0.0f);
|
||||
ci *= m_settings.m_volumeFactor;
|
||||
}
|
||||
else
|
||||
{
|
||||
}
|
||||
else
|
||||
{
|
||||
ci.real(real * m_settings.m_volumeFactor);
|
||||
ci.imag(0.0f);
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
else
|
||||
@@ -312,24 +295,24 @@ void SSBModSource::pullAF(Complex& sample)
|
||||
break;
|
||||
case SSBModSettings::SSBModInputAudio:
|
||||
if (m_settings.m_audioBinaural)
|
||||
{
|
||||
if (m_settings.m_audioFlipChannels)
|
||||
{
|
||||
{
|
||||
if (m_settings.m_audioFlipChannels)
|
||||
{
|
||||
ci.real((m_audioBuffer[m_audioBufferFill].r / SDR_TX_SCALEF) * m_settings.m_volumeFactor);
|
||||
ci.imag((m_audioBuffer[m_audioBufferFill].l / SDR_TX_SCALEF) * m_settings.m_volumeFactor);
|
||||
}
|
||||
else
|
||||
{
|
||||
}
|
||||
else
|
||||
{
|
||||
ci.real((m_audioBuffer[m_audioBufferFill].l / SDR_TX_SCALEF) * m_settings.m_volumeFactor);
|
||||
ci.imag((m_audioBuffer[m_audioBufferFill].r / SDR_TX_SCALEF) * m_settings.m_volumeFactor);
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
else
|
||||
{
|
||||
if (m_settings.m_agc)
|
||||
{
|
||||
float sample = (m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) / 65536.0f;
|
||||
ci.real(clamp<float>(m_audioCompressor.compress(sample), -1.0f, 1.0f));
|
||||
float xsample = (m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) / 65536.0f;
|
||||
ci.real(clamp<float>(m_audioCompressor.compress(xsample), -1.0f, 1.0f));
|
||||
ci.imag(0.0f);
|
||||
ci *= m_settings.m_volumeFactor;
|
||||
}
|
||||
@@ -347,7 +330,7 @@ void SSBModSource::pullAF(Complex& sample)
|
||||
else
|
||||
{
|
||||
qDebug("SSBModSource::pullAF: starve audio samples: size: %lu", m_audioBuffer.size());
|
||||
m_audioBufferFill = m_audioBuffer.size() - 1;
|
||||
m_audioBufferFill = (unsigned int) (m_audioBuffer.size() - 1);
|
||||
}
|
||||
|
||||
break;
|
||||
@@ -356,56 +339,55 @@ void SSBModSource::pullAF(Complex& sample)
|
||||
break;
|
||||
}
|
||||
|
||||
Real fadeFactor;
|
||||
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;
|
||||
}
|
||||
}
|
||||
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
|
||||
{
|
||||
if (m_cwKeyer->getCWSmoother().getFadeSample(false, fadeFactor))
|
||||
{
|
||||
if (m_settings.m_dsb)
|
||||
{
|
||||
Real t = (m_toneNco.next() * fadeFactor)/1.25f;
|
||||
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 SSBModSettings::SSBModInputNone:
|
||||
default:
|
||||
sample.real(0.0f);
|
||||
sample.imag(0.0f);
|
||||
@@ -415,35 +397,35 @@ void SSBModSource::pullAF(Complex& sample)
|
||||
if ((m_settings.m_modAFInput == SSBModSettings::SSBModInputFile)
|
||||
|| (m_settings.m_modAFInput == SSBModSettings::SSBModInputAudio)) // real audio
|
||||
{
|
||||
if (m_settings.m_dsb)
|
||||
{
|
||||
n_out = m_DSBFilter->runDSB(ci, &filtered);
|
||||
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;
|
||||
}
|
||||
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);
|
||||
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;
|
||||
}
|
||||
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++;
|
||||
}
|
||||
sample = m_SSBFilterBuffer[m_SSBFilterBufferIndex];
|
||||
m_SSBFilterBufferIndex++;
|
||||
}
|
||||
|
||||
if (n_out > 0)
|
||||
{
|
||||
if (n_out > 0)
|
||||
{
|
||||
for (int i = 0; i < n_out; i++)
|
||||
{
|
||||
// Downsample by 2^(m_scaleLog2 - 1) for SSB band spectrum display
|
||||
@@ -453,23 +435,23 @@ void SSBModSource::pullAF(Complex& sample)
|
||||
|
||||
if (!(m_undersampleCount++ & decim_mask))
|
||||
{
|
||||
Real avgr = (m_sum.real() / decim) * 0.891235351562f * SDR_TX_SCALEF; //scaling at -1 dB to account for possible filter overshoot
|
||||
Real avgi = (m_sum.imag() / decim) * 0.891235351562f * SDR_TX_SCALEF;
|
||||
Real avgr = (m_sum.real() / (float) decim) * 0.891235351562f * SDR_TX_SCALEF; //scaling at -1 dB to account for possible filter overshoot
|
||||
Real avgi = (m_sum.imag() / (float) decim) * 0.891235351562f * SDR_TX_SCALEF;
|
||||
|
||||
if (!m_settings.m_dsb & !m_settings.m_usb)
|
||||
if (!m_settings.m_dsb && !m_settings.m_usb)
|
||||
{ // invert spectrum for LSB
|
||||
m_sampleBuffer.push_back(Sample(avgi, avgr));
|
||||
m_sampleBuffer.push_back(Sample((FixReal) avgi, (FixReal) avgr));
|
||||
}
|
||||
else
|
||||
{
|
||||
m_sampleBuffer.push_back(Sample(avgr, avgi));
|
||||
m_sampleBuffer.push_back(Sample((FixReal) avgr, (FixReal)avgi));
|
||||
}
|
||||
|
||||
m_sum.real(0.0);
|
||||
m_sum.imag(0.0);
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
} // Real audio
|
||||
else if ((m_settings.m_modAFInput == SSBModSettings::SSBModInputTone)
|
||||
|| (m_settings.m_modAFInput == SSBModSettings::SSBModInputCWTone)) // tone
|
||||
@@ -478,16 +460,16 @@ void SSBModSource::pullAF(Complex& sample)
|
||||
|
||||
if (!(m_undersampleCount++ & decim_mask))
|
||||
{
|
||||
Real avgr = (m_sum.real() / decim) * 0.891235351562f * SDR_TX_SCALEF; //scaling at -1 dB to account for possible filter overshoot
|
||||
Real avgi = (m_sum.imag() / decim) * 0.891235351562f * SDR_TX_SCALEF;
|
||||
Real avgr = (m_sum.real() / (float) decim) * 0.891235351562f * SDR_TX_SCALEF; //scaling at -1 dB to account for possible filter overshoot
|
||||
Real avgi = (m_sum.imag() / (float) decim) * 0.891235351562f * SDR_TX_SCALEF;
|
||||
|
||||
if (!m_settings.m_dsb & !m_settings.m_usb)
|
||||
if (!m_settings.m_dsb && !m_settings.m_usb)
|
||||
{ // invert spectrum for LSB
|
||||
m_sampleBuffer.push_back(Sample(avgi, avgr));
|
||||
m_sampleBuffer.push_back(Sample((FixReal) avgi, (FixReal) avgr));
|
||||
}
|
||||
else
|
||||
{
|
||||
m_sampleBuffer.push_back(Sample(avgr, avgi));
|
||||
m_sampleBuffer.push_back(Sample((FixReal) avgr, (FixReal) avgi));
|
||||
}
|
||||
|
||||
m_sum.real(0.0);
|
||||
@@ -538,18 +520,18 @@ void SSBModSource::pushFeedback(Complex c)
|
||||
}
|
||||
}
|
||||
|
||||
void SSBModSource::processOneSample(Complex& ci)
|
||||
void SSBModSource::processOneSample(const Complex& ci)
|
||||
{
|
||||
if (m_settings.m_modAFInput == SSBModSettings::SSBModInputCWTone) // minimize latency for CW
|
||||
{
|
||||
m_feedbackAudioBuffer[0].l = ci.real();
|
||||
m_feedbackAudioBuffer[0].r = ci.imag();
|
||||
m_feedbackAudioBuffer[0].l = (qint16) ci.real();
|
||||
m_feedbackAudioBuffer[0].r = (qint16) ci.imag();
|
||||
m_feedbackAudioFifo.writeOne((const quint8*) &m_feedbackAudioBuffer[0]);
|
||||
}
|
||||
else
|
||||
{
|
||||
m_feedbackAudioBuffer[m_feedbackAudioBufferFill].l = ci.real();
|
||||
m_feedbackAudioBuffer[m_feedbackAudioBufferFill].r = ci.imag();
|
||||
m_feedbackAudioBuffer[m_feedbackAudioBufferFill].l = (qint16) ci.real();
|
||||
m_feedbackAudioBuffer[m_feedbackAudioBufferFill].r = (qint16) ci.imag();
|
||||
++m_feedbackAudioBufferFill;
|
||||
|
||||
if (m_feedbackAudioBufferFill >= m_feedbackAudioBuffer.size())
|
||||
@@ -568,9 +550,9 @@ void SSBModSource::processOneSample(Complex& ci)
|
||||
}
|
||||
}
|
||||
|
||||
void SSBModSource::calculateLevel(Complex& sample)
|
||||
void SSBModSource::calculateLevel(const Complex& sample)
|
||||
{
|
||||
Real t = sample.real(); // TODO: possibly adjust depending on sample type
|
||||
Real t = sample.real();
|
||||
|
||||
if (m_levelCalcCount < m_levelNbSamples)
|
||||
{
|
||||
@@ -617,14 +599,14 @@ void SSBModSource::applyAudioSampleRate(int sampleRate)
|
||||
lowCutoff = band - 100.0f;
|
||||
}
|
||||
|
||||
m_SSBFilter->create_filter(lowCutoff / sampleRate, band / sampleRate);
|
||||
m_DSBFilter->create_dsb_filter((2.0f * band) / sampleRate);
|
||||
m_SSBFilter->create_filter(lowCutoff / (float) sampleRate, band / (float) sampleRate);
|
||||
m_DSBFilter->create_dsb_filter((2.0f * band) / (float) sampleRate);
|
||||
|
||||
m_settings.m_bandwidth = band;
|
||||
m_settings.m_lowCutoff = lowCutoff;
|
||||
m_settings.m_usb = usb;
|
||||
|
||||
m_toneNco.setFreq(m_settings.m_toneFrequency, sampleRate);
|
||||
m_toneNco.setFreq(m_settings.m_toneFrequency, (float) sampleRate);
|
||||
|
||||
if (m_cwKeyer)
|
||||
{
|
||||
@@ -632,7 +614,7 @@ void SSBModSource::applyAudioSampleRate(int sampleRate)
|
||||
m_cwKeyer->reset();
|
||||
}
|
||||
|
||||
m_audioCompressor.m_rate = sampleRate;
|
||||
m_audioCompressor.m_rate = (float) sampleRate;
|
||||
m_audioCompressor.initState();
|
||||
m_audioSampleRate = sampleRate;
|
||||
|
||||
@@ -641,7 +623,7 @@ void SSBModSource::applyAudioSampleRate(int sampleRate)
|
||||
QList<ObjectPipe*> pipes;
|
||||
MainCore::instance()->getMessagePipes().getMessagePipes(m_channel, "reportdemod", pipes);
|
||||
|
||||
if (pipes.size() > 0)
|
||||
if (!pipes.empty())
|
||||
{
|
||||
for (const auto& pipe : pipes)
|
||||
{
|
||||
@@ -665,7 +647,7 @@ void SSBModSource::applyFeedbackAudioSampleRate(int sampleRate)
|
||||
m_feedbackInterpolatorDistanceRemain = 0;
|
||||
m_feedbackInterpolatorConsumed = false;
|
||||
m_feedbackInterpolatorDistance = (Real) sampleRate / (Real) m_audioSampleRate;
|
||||
Real cutoff = std::min(sampleRate, m_audioSampleRate) / 2.2f;
|
||||
Real cutoff = (float) (std::min(sampleRate, m_audioSampleRate)) / 2.2f;
|
||||
m_feedbackInterpolator.create(48, sampleRate, cutoff, 3.0);
|
||||
m_feedbackAudioSampleRate = sampleRate;
|
||||
}
|
||||
@@ -693,12 +675,12 @@ void SSBModSource::applySettings(const SSBModSettings& settings, bool force)
|
||||
m_interpolatorConsumed = false;
|
||||
m_interpolatorDistance = (Real) m_audioSampleRate / (Real) m_channelSampleRate;
|
||||
m_interpolator.create(48, m_audioSampleRate, band, 3.0);
|
||||
m_SSBFilter->create_filter(lowCutoff / m_audioSampleRate, band / m_audioSampleRate);
|
||||
m_DSBFilter->create_dsb_filter((2.0f * band) / m_audioSampleRate);
|
||||
m_SSBFilter->create_filter(lowCutoff / (float) m_audioSampleRate, band / (float) m_audioSampleRate);
|
||||
m_DSBFilter->create_dsb_filter((2.0f * band) / (float) m_audioSampleRate);
|
||||
}
|
||||
|
||||
if ((settings.m_toneFrequency != m_settings.m_toneFrequency) || force) {
|
||||
m_toneNco.setFreq(settings.m_toneFrequency, m_audioSampleRate);
|
||||
m_toneNco.setFreq(settings.m_toneFrequency, (float) m_audioSampleRate);
|
||||
}
|
||||
|
||||
if ((settings.m_dsb != m_settings.m_dsb) || force)
|
||||
@@ -729,11 +711,11 @@ void SSBModSource::applySettings(const SSBModSettings& settings, bool force)
|
||||
{
|
||||
m_audioCompressor.initSimple(
|
||||
m_audioSampleRate,
|
||||
settings.m_cmpPreGainDB, // pregain (dB)
|
||||
settings.m_cmpThresholdDB, // threshold (dB)
|
||||
(float) settings.m_cmpPreGainDB, // pregain (dB)
|
||||
(float) settings.m_cmpThresholdDB, // threshold (dB)
|
||||
20, // knee (dB)
|
||||
12, // ratio (dB)
|
||||
0.003, // attack (s)
|
||||
0.003f,// attack (s)
|
||||
0.25 // release (s)
|
||||
);
|
||||
}
|
||||
@@ -751,8 +733,8 @@ void SSBModSource::applyChannelSettings(int channelSampleRate, int channelFreque
|
||||
<< " channelFrequencyOffset: " << channelFrequencyOffset;
|
||||
|
||||
if ((channelFrequencyOffset != m_channelFrequencyOffset)
|
||||
|| (channelSampleRate != m_channelSampleRate) || force) {
|
||||
m_carrierNco.setFreq(channelFrequencyOffset, channelSampleRate);
|
||||
|| (channelSampleRate != m_channelSampleRate) || force) {
|
||||
m_carrierNco.setFreq((float) channelFrequencyOffset, (float) channelSampleRate);
|
||||
}
|
||||
|
||||
if ((channelSampleRate != m_channelSampleRate) || force)
|
||||
@@ -769,7 +751,6 @@ void SSBModSource::applyChannelSettings(int channelSampleRate, int channelFreque
|
||||
|
||||
void SSBModSource::handleAudio()
|
||||
{
|
||||
QMutexLocker mlock(&m_mutex);
|
||||
unsigned int nbRead;
|
||||
|
||||
while ((nbRead = m_audioFifo.read(reinterpret_cast<quint8*>(&m_audioReadBuffer[m_audioReadBufferFill]), 4096)) != 0)
|
||||
|
||||
Reference in New Issue
Block a user