mirror of
https://github.com/f4exb/sdrangel.git
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758 lines
23 KiB
C++
758 lines
23 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2019 Edouard Griffiths, F4EXB //
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// //
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// This program is free software; you can redistribute it and/or modify //
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// it under the terms of the GNU General Public License as published by //
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// the Free Software Foundation as version 3 of the License, or //
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// (at your option) any later version. //
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// //
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// This program is distributed in the hope that it will be useful, //
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// but WITHOUT ANY WARRANTY; without even the implied warranty of //
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
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// GNU General Public License V3 for more details. //
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// //
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// You should have received a copy of the GNU General Public License //
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// along with this program. If not, see <http://www.gnu.org/licenses/>. //
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///////////////////////////////////////////////////////////////////////////////////
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#include <QDebug>
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#include "dsp/basebandsamplesink.h"
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#include "dsp/misc.h"
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#include "dsp/datafifo.h"
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#include "util/messagequeue.h"
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#include "maincore.h"
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#include "ssbmodsource.h"
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const int SSBModSource::m_ssbFftLen = 1024;
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const int SSBModSource::m_levelNbSamples = 480; // every 10ms
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SSBModSource::SSBModSource() :
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m_channelSampleRate(48000),
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m_channelFrequencyOffset(0),
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m_audioSampleRate(48000),
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m_audioFifo(12000),
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m_feedbackAudioFifo(48000),
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m_levelCalcCount(0),
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m_peakLevel(0.0f),
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m_levelSum(0.0f),
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m_ifstream(nullptr),
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m_mutex(QMutex::Recursive)
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{
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m_SSBFilter = new fftfilt(m_settings.m_lowCutoff / m_audioSampleRate, m_settings.m_bandwidth / m_audioSampleRate, m_ssbFftLen);
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m_DSBFilter = new fftfilt((2.0f * m_settings.m_bandwidth) / m_audioSampleRate, 2 * m_ssbFftLen);
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m_SSBFilterBuffer = new Complex[m_ssbFftLen>>1]; // filter returns data exactly half of its size
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m_DSBFilterBuffer = new Complex[m_ssbFftLen];
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std::fill(m_SSBFilterBuffer, m_SSBFilterBuffer+(m_ssbFftLen>>1), Complex{0,0});
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std::fill(m_DSBFilterBuffer, m_DSBFilterBuffer+m_ssbFftLen, Complex{0,0});
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m_audioBuffer.resize(24000);
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m_audioBufferFill = 0;
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m_audioReadBuffer.resize(24000);
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m_audioReadBufferFill = 0;
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m_feedbackAudioBuffer.resize(1<<14);
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m_feedbackAudioBufferFill = 0;
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m_demodBuffer.resize(1<<12);
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m_demodBufferFill = 0;
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m_sum.real(0.0f);
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m_sum.imag(0.0f);
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m_undersampleCount = 0;
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m_sumCount = 0;
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m_magsq = 0.0;
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m_toneNco.setFreq(1000.0, m_audioSampleRate);
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m_cwKeyer.setSampleRate(m_audioSampleRate);
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m_cwKeyer.reset();
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m_audioCompressor.initSimple(
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m_audioSampleRate,
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m_settings.m_cmpPreGainDB, // pregain (dB)
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m_settings.m_cmpThresholdDB, // threshold (dB)
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20, // knee (dB)
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12, // ratio (dB)
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0.003, // attack (s)
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0.25 // release (s)
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);
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applySettings(m_settings, true);
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applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
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}
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SSBModSource::~SSBModSource()
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{
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delete m_SSBFilter;
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delete m_DSBFilter;
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delete[] m_SSBFilterBuffer;
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delete[] m_DSBFilterBuffer;
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}
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void SSBModSource::pull(SampleVector::iterator begin, unsigned int nbSamples)
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{
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std::for_each(
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begin,
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begin + nbSamples,
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[this](Sample& s) {
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pullOne(s);
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}
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);
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}
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void SSBModSource::pullOne(Sample& sample)
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{
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Complex ci;
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if (m_interpolatorDistance > 1.0f) // decimate
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{
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modulateSample();
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while (!m_interpolator.decimate(&m_interpolatorDistanceRemain, m_modSample, &ci))
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{
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modulateSample();
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}
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}
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else
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{
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if (m_interpolator.interpolate(&m_interpolatorDistanceRemain, m_modSample, &ci))
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{
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modulateSample();
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}
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}
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m_interpolatorDistanceRemain += m_interpolatorDistance;
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ci *= m_carrierNco.nextIQ(); // shift to carrier frequency
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ci *= 0.891235351562f * SDR_TX_SCALEF; //scaling at -1 dB to account for possible filter overshoot
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double magsq = ci.real() * ci.real() + ci.imag() * ci.imag();
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magsq /= (SDR_TX_SCALED*SDR_TX_SCALED);
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m_movingAverage(magsq);
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m_magsq = m_movingAverage.asDouble();
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sample.m_real = (FixReal) ci.real();
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sample.m_imag = (FixReal) ci.imag();
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}
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void SSBModSource::prefetch(unsigned int nbSamples)
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{
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unsigned int nbSamplesAudio = nbSamples * ((Real) m_audioSampleRate / (Real) m_channelSampleRate);
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pullAudio(nbSamplesAudio);
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}
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void SSBModSource::pullAudio(unsigned int nbSamplesAudio)
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{
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QMutexLocker mlock(&m_mutex);
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if (nbSamplesAudio > m_audioBuffer.size()) {
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m_audioBuffer.resize(nbSamplesAudio);
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}
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std::copy(&m_audioReadBuffer[0], &m_audioReadBuffer[nbSamplesAudio], &m_audioBuffer[0]);
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m_audioBufferFill = 0;
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if (m_audioReadBufferFill > nbSamplesAudio) // copy back remaining samples at the start of the read buffer
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{
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std::copy(&m_audioReadBuffer[nbSamplesAudio], &m_audioReadBuffer[m_audioReadBufferFill], &m_audioReadBuffer[0]);
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m_audioReadBufferFill = m_audioReadBufferFill - nbSamplesAudio; // adjust current read buffer fill pointer
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}
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}
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void SSBModSource::modulateSample()
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{
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pullAF(m_modSample);
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if (m_settings.m_feedbackAudioEnable) {
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pushFeedback(m_modSample * m_settings.m_feedbackVolumeFactor * 16384.0f);
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}
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calculateLevel(m_modSample);
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if (m_settings.m_audioBinaural)
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{
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m_demodBuffer[m_demodBufferFill++] = m_modSample.real() * std::numeric_limits<int16_t>::max();
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m_demodBuffer[m_demodBufferFill++] = m_modSample.imag() * std::numeric_limits<int16_t>::max();
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}
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else
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{
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// take projection on real axis
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m_demodBuffer[m_demodBufferFill++] = m_modSample.real() * std::numeric_limits<int16_t>::max();
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}
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if (m_demodBufferFill >= m_demodBuffer.size())
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{
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QList<DataFifo*> *dataFifos = MainCore::instance()->getDataPipes().getFifos(m_channel, "demod");
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if (dataFifos)
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{
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QList<DataFifo*>::iterator it = dataFifos->begin();
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for (; it != dataFifos->end(); ++it)
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{
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(*it)->write(
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(quint8*) &m_demodBuffer[0],
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m_demodBuffer.size() * sizeof(qint16),
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m_settings.m_audioBinaural ? DataFifo::DataTypeCI16 : DataFifo::DataTypeI16
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);
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}
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}
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m_demodBufferFill = 0;
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}
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}
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void SSBModSource::pullAF(Complex& sample)
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{
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if (m_settings.m_audioMute)
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{
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sample.real(0.0f);
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sample.imag(0.0f);
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return;
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}
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Complex ci;
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fftfilt::cmplx *filtered;
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int n_out = 0;
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int decim = 1<<(m_settings.m_spanLog2 - 1);
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unsigned char decim_mask = decim - 1; // counter LSB bit mask for decimation by 2^(m_scaleLog2 - 1)
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switch (m_settings.m_modAFInput)
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{
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case SSBModSettings::SSBModInputTone:
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if (m_settings.m_dsb)
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{
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Real t = m_toneNco.next()/1.25;
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sample.real(t);
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sample.imag(t);
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}
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else
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{
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if (m_settings.m_usb) {
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sample = m_toneNco.nextIQ();
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} else {
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sample = m_toneNco.nextQI();
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}
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}
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break;
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case SSBModSettings::SSBModInputFile:
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// Monaural (mono):
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// sox f4exb_call.wav --encoding float --endian little f4exb_call.raw
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// ffplay -f f32le -ar 48k -ac 1 f4exb_call.raw
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// Binaural (stereo):
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// sox f4exb_call.wav --encoding float --endian little f4exb_call.raw
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// ffplay -f f32le -ar 48k -ac 2 f4exb_call.raw
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if (m_ifstream && m_ifstream->is_open())
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{
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if (m_ifstream->eof())
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{
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if (m_settings.m_playLoop)
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{
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m_ifstream->clear();
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m_ifstream->seekg(0, std::ios::beg);
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}
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}
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if (m_ifstream->eof())
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{
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ci.real(0.0f);
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ci.imag(0.0f);
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}
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else
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{
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if (m_settings.m_audioBinaural)
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{
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Complex c;
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m_ifstream->read(reinterpret_cast<char*>(&c), sizeof(Complex));
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if (m_settings.m_audioFlipChannels)
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{
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ci.real(c.imag() * m_settings.m_volumeFactor);
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ci.imag(c.real() * m_settings.m_volumeFactor);
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}
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else
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{
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ci = c * m_settings.m_volumeFactor;
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}
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}
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else
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{
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Real real;
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m_ifstream->read(reinterpret_cast<char*>(&real), sizeof(Real));
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if (m_settings.m_agc)
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{
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ci.real(clamp<float>(m_audioCompressor.compress(real), -1.0f, 1.0f));
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ci.imag(0.0f);
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ci *= m_settings.m_volumeFactor;
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}
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else
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{
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ci.real(real * m_settings.m_volumeFactor);
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ci.imag(0.0f);
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}
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}
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}
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}
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else
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{
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ci.real(0.0f);
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ci.imag(0.0f);
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}
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break;
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case SSBModSettings::SSBModInputAudio:
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if (m_settings.m_audioBinaural)
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{
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if (m_settings.m_audioFlipChannels)
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{
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ci.real((m_audioBuffer[m_audioBufferFill].r / SDR_TX_SCALEF) * m_settings.m_volumeFactor);
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ci.imag((m_audioBuffer[m_audioBufferFill].l / SDR_TX_SCALEF) * m_settings.m_volumeFactor);
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}
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else
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{
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ci.real((m_audioBuffer[m_audioBufferFill].l / SDR_TX_SCALEF) * m_settings.m_volumeFactor);
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ci.imag((m_audioBuffer[m_audioBufferFill].r / SDR_TX_SCALEF) * m_settings.m_volumeFactor);
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}
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}
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else
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{
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if (m_settings.m_agc)
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{
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float sample = (m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) / 65536.0f;
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ci.real(clamp<float>(m_audioCompressor.compress(sample), -1.0f, 1.0f));
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ci.imag(0.0f);
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ci *= m_settings.m_volumeFactor;
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}
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else
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{
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ci.real(((m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) / 65536.0f) * m_settings.m_volumeFactor);
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ci.imag(0.0f);
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}
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}
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if (m_audioBufferFill < m_audioBuffer.size() - 1)
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{
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m_audioBufferFill++;
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}
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else
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{
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qDebug("SSBModSource::pullAF: starve audio samples: size: %lu", m_audioBuffer.size());
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m_audioBufferFill = m_audioBuffer.size() - 1;
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}
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break;
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case SSBModSettings::SSBModInputCWTone:
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Real fadeFactor;
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if (m_cwKeyer.getSample())
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{
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m_cwKeyer.getCWSmoother().getFadeSample(true, fadeFactor);
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if (m_settings.m_dsb)
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{
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Real t = m_toneNco.next() * fadeFactor;
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sample.real(t);
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sample.imag(t);
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}
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else
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{
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if (m_settings.m_usb) {
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sample = m_toneNco.nextIQ() * fadeFactor;
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} else {
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sample = m_toneNco.nextQI() * fadeFactor;
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}
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}
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}
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else
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{
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if (m_cwKeyer.getCWSmoother().getFadeSample(false, fadeFactor))
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{
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if (m_settings.m_dsb)
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{
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Real t = (m_toneNco.next() * fadeFactor)/1.25;
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sample.real(t);
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sample.imag(t);
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}
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else
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{
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if (m_settings.m_usb) {
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sample = m_toneNco.nextIQ() * fadeFactor;
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} else {
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sample = m_toneNco.nextQI() * fadeFactor;
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}
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}
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}
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else
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{
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sample.real(0.0f);
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sample.imag(0.0f);
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m_toneNco.setPhase(0);
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}
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}
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break;
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case SSBModSettings::SSBModInputNone:
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default:
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sample.real(0.0f);
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sample.imag(0.0f);
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break;
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}
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if ((m_settings.m_modAFInput == SSBModSettings::SSBModInputFile)
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|| (m_settings.m_modAFInput == SSBModSettings::SSBModInputAudio)) // real audio
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{
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if (m_settings.m_dsb)
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{
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n_out = m_DSBFilter->runDSB(ci, &filtered);
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if (n_out > 0)
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{
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memcpy((void *) m_DSBFilterBuffer, (const void *) filtered, n_out*sizeof(Complex));
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m_DSBFilterBufferIndex = 0;
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}
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sample = m_DSBFilterBuffer[m_DSBFilterBufferIndex];
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m_DSBFilterBufferIndex++;
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}
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else
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{
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n_out = m_SSBFilter->runSSB(ci, &filtered, m_settings.m_usb);
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if (n_out > 0)
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{
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memcpy((void *) m_SSBFilterBuffer, (const void *) filtered, n_out*sizeof(Complex));
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m_SSBFilterBufferIndex = 0;
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}
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sample = m_SSBFilterBuffer[m_SSBFilterBufferIndex];
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m_SSBFilterBufferIndex++;
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}
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if (n_out > 0)
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{
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for (int i = 0; i < n_out; i++)
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{
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// Downsample by 2^(m_scaleLog2 - 1) for SSB band spectrum display
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// smart decimation with bit gain using float arithmetic (23 bits significand)
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m_sum += filtered[i];
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if (!(m_undersampleCount++ & decim_mask))
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{
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Real avgr = (m_sum.real() / decim) * 0.891235351562f * SDR_TX_SCALEF; //scaling at -1 dB to account for possible filter overshoot
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Real avgi = (m_sum.imag() / decim) * 0.891235351562f * SDR_TX_SCALEF;
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if (!m_settings.m_dsb & !m_settings.m_usb)
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{ // invert spectrum for LSB
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m_sampleBuffer.push_back(Sample(avgi, avgr));
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}
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else
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{
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m_sampleBuffer.push_back(Sample(avgr, avgi));
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}
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m_sum.real(0.0);
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m_sum.imag(0.0);
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}
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}
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}
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} // Real audio
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else if ((m_settings.m_modAFInput == SSBModSettings::SSBModInputTone)
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|| (m_settings.m_modAFInput == SSBModSettings::SSBModInputCWTone)) // tone
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{
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m_sum += sample;
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if (!(m_undersampleCount++ & decim_mask))
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{
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Real avgr = (m_sum.real() / decim) * 0.891235351562f * SDR_TX_SCALEF; //scaling at -1 dB to account for possible filter overshoot
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Real avgi = (m_sum.imag() / decim) * 0.891235351562f * SDR_TX_SCALEF;
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if (!m_settings.m_dsb & !m_settings.m_usb)
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{ // invert spectrum for LSB
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m_sampleBuffer.push_back(Sample(avgi, avgr));
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}
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else
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{
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m_sampleBuffer.push_back(Sample(avgr, avgi));
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}
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m_sum.real(0.0);
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m_sum.imag(0.0);
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}
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if (m_sumCount < (m_settings.m_dsb ? m_ssbFftLen : m_ssbFftLen>>1))
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{
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n_out = 0;
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m_sumCount++;
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}
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else
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{
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n_out = m_sumCount;
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m_sumCount = 0;
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}
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}
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if (n_out > 0)
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{
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if (m_spectrumSink) {
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m_spectrumSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), !m_settings.m_dsb);
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}
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m_sampleBuffer.clear();
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}
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}
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void SSBModSource::pushFeedback(Complex c)
|
|
{
|
|
Complex ci;
|
|
|
|
if (m_feedbackInterpolatorDistance < 1.0f) // interpolate
|
|
{
|
|
while (!m_feedbackInterpolator.interpolate(&m_feedbackInterpolatorDistanceRemain, c, &ci))
|
|
{
|
|
processOneSample(ci);
|
|
m_feedbackInterpolatorDistanceRemain += m_feedbackInterpolatorDistance;
|
|
}
|
|
}
|
|
else // decimate
|
|
{
|
|
if (m_feedbackInterpolator.decimate(&m_feedbackInterpolatorDistanceRemain, c, &ci))
|
|
{
|
|
processOneSample(ci);
|
|
m_feedbackInterpolatorDistanceRemain += m_feedbackInterpolatorDistance;
|
|
}
|
|
}
|
|
}
|
|
|
|
void SSBModSource::processOneSample(Complex& ci)
|
|
{
|
|
m_feedbackAudioBuffer[m_feedbackAudioBufferFill].l = ci.real();
|
|
m_feedbackAudioBuffer[m_feedbackAudioBufferFill].r = ci.imag();
|
|
++m_feedbackAudioBufferFill;
|
|
|
|
if (m_feedbackAudioBufferFill >= m_feedbackAudioBuffer.size())
|
|
{
|
|
uint res = m_feedbackAudioFifo.write((const quint8*)&m_feedbackAudioBuffer[0], m_feedbackAudioBufferFill);
|
|
|
|
if (res != m_feedbackAudioBufferFill)
|
|
{
|
|
qDebug("SSBModSource::pushFeedback: %u/%u audio samples written m_feedbackInterpolatorDistance: %f",
|
|
res, m_feedbackAudioBufferFill, m_feedbackInterpolatorDistance);
|
|
m_feedbackAudioFifo.clear();
|
|
}
|
|
|
|
m_feedbackAudioBufferFill = 0;
|
|
}
|
|
}
|
|
|
|
void SSBModSource::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
|
|
{
|
|
m_rmsLevel = sqrt(m_levelSum / m_levelNbSamples);
|
|
m_peakLevelOut = m_peakLevel;
|
|
m_peakLevel = 0.0f;
|
|
m_levelSum = 0.0f;
|
|
m_levelCalcCount = 0;
|
|
}
|
|
}
|
|
|
|
void SSBModSource::applyAudioSampleRate(int sampleRate)
|
|
{
|
|
if (sampleRate < 0)
|
|
{
|
|
qWarning("SSBModSource::applyAudioSampleRate: invalid sample rate %d", sampleRate);
|
|
return;
|
|
}
|
|
|
|
qDebug("SSBModSource::applyAudioSampleRate: %d", sampleRate);
|
|
|
|
m_interpolatorDistanceRemain = 0;
|
|
m_interpolatorConsumed = false;
|
|
m_interpolatorDistance = (Real) sampleRate / (Real) m_channelSampleRate;
|
|
m_interpolator.create(48, sampleRate, m_settings.m_bandwidth, 3.0);
|
|
|
|
float band = m_settings.m_bandwidth;
|
|
float lowCutoff = m_settings.m_lowCutoff;
|
|
bool usb = m_settings.m_usb;
|
|
|
|
if (band < 100.0f) // at least 100 Hz
|
|
{
|
|
band = 100.0f;
|
|
lowCutoff = 0;
|
|
}
|
|
|
|
if (band - lowCutoff < 100.0f) {
|
|
lowCutoff = band - 100.0f;
|
|
}
|
|
|
|
m_SSBFilter->create_filter(lowCutoff / sampleRate, band / sampleRate);
|
|
m_DSBFilter->create_dsb_filter((2.0f * band) / 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_cwKeyer.setSampleRate(sampleRate);
|
|
m_cwKeyer.reset();
|
|
|
|
m_audioCompressor.m_rate = sampleRate;
|
|
m_audioCompressor.initState();
|
|
m_audioSampleRate = sampleRate;
|
|
|
|
applyFeedbackAudioSampleRate(m_feedbackAudioSampleRate);
|
|
|
|
QList<MessageQueue*> *messageQueues = MainCore::instance()->getMessagePipes().getMessageQueues(m_channel, "reportdemod");
|
|
|
|
if (messageQueues)
|
|
{
|
|
QList<MessageQueue*>::iterator it = messageQueues->begin();
|
|
|
|
for (; it != messageQueues->end(); ++it)
|
|
{
|
|
MainCore::MsgChannelDemodReport *msg = MainCore::MsgChannelDemodReport::create(m_channel, sampleRate);
|
|
(*it)->push(msg);
|
|
}
|
|
}
|
|
}
|
|
|
|
void SSBModSource::applyFeedbackAudioSampleRate(int sampleRate)
|
|
{
|
|
if (sampleRate < 0)
|
|
{
|
|
qWarning("SSBModSource::applyFeedbackAudioSampleRate: invalid sample rate %d", sampleRate);
|
|
return;
|
|
}
|
|
|
|
qDebug("SSBModSource::applyFeedbackAudioSampleRate: %d", sampleRate);
|
|
|
|
m_feedbackInterpolatorDistanceRemain = 0;
|
|
m_feedbackInterpolatorConsumed = false;
|
|
m_feedbackInterpolatorDistance = (Real) sampleRate / (Real) m_audioSampleRate;
|
|
Real cutoff = std::min(sampleRate, m_audioSampleRate) / 2.2f;
|
|
m_feedbackInterpolator.create(48, sampleRate, cutoff, 3.0);
|
|
m_feedbackAudioSampleRate = sampleRate;
|
|
}
|
|
|
|
void SSBModSource::applySettings(const SSBModSettings& settings, bool force)
|
|
{
|
|
float band = settings.m_bandwidth;
|
|
float lowCutoff = settings.m_lowCutoff;
|
|
bool usb = settings.m_usb;
|
|
|
|
if ((settings.m_bandwidth != m_settings.m_bandwidth) ||
|
|
(settings.m_lowCutoff != m_settings.m_lowCutoff) || force)
|
|
{
|
|
if (band < 100.0f) // at least 100 Hz
|
|
{
|
|
band = 100.0f;
|
|
lowCutoff = 0;
|
|
}
|
|
|
|
if (band - lowCutoff < 100.0f) {
|
|
lowCutoff = band - 100.0f;
|
|
}
|
|
|
|
m_interpolatorDistanceRemain = 0;
|
|
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);
|
|
}
|
|
|
|
if ((settings.m_toneFrequency != m_settings.m_toneFrequency) || force) {
|
|
m_toneNco.setFreq(settings.m_toneFrequency, m_audioSampleRate);
|
|
}
|
|
|
|
if ((settings.m_dsb != m_settings.m_dsb) || force)
|
|
{
|
|
if (settings.m_dsb)
|
|
{
|
|
std::fill(m_DSBFilterBuffer, m_DSBFilterBuffer+m_ssbFftLen, Complex{0,0});
|
|
m_DSBFilterBufferIndex = 0;
|
|
}
|
|
else
|
|
{
|
|
std::fill(m_SSBFilterBuffer, m_SSBFilterBuffer+(m_ssbFftLen>>1), Complex{0,0});
|
|
m_SSBFilterBufferIndex = 0;
|
|
}
|
|
}
|
|
|
|
if ((settings.m_modAFInput != m_settings.m_modAFInput) || force)
|
|
{
|
|
if (settings.m_modAFInput == SSBModSettings::SSBModInputAudio) {
|
|
connect(&m_audioFifo, SIGNAL(dataReady()), this, SLOT(handleAudio()));
|
|
} else {
|
|
disconnect(&m_audioFifo, SIGNAL(dataReady()), this, SLOT(handleAudio()));
|
|
}
|
|
}
|
|
|
|
if ((settings.m_cmpThresholdDB != m_settings.m_cmpThresholdDB) ||
|
|
(settings.m_cmpPreGainDB != m_settings.m_cmpPreGainDB) || force)
|
|
{
|
|
m_audioCompressor.initSimple(
|
|
m_audioSampleRate,
|
|
settings.m_cmpPreGainDB, // pregain (dB)
|
|
settings.m_cmpThresholdDB, // threshold (dB)
|
|
20, // knee (dB)
|
|
12, // ratio (dB)
|
|
0.003, // attack (s)
|
|
0.25 // release (s)
|
|
);
|
|
}
|
|
|
|
m_settings = settings;
|
|
m_settings.m_bandwidth = band;
|
|
m_settings.m_lowCutoff = lowCutoff;
|
|
m_settings.m_usb = usb;
|
|
}
|
|
|
|
void SSBModSource::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
|
|
{
|
|
qDebug() << "SSBModSource::applyChannelSettings:"
|
|
<< " channelSampleRate: " << channelSampleRate
|
|
<< " channelFrequencyOffset: " << channelFrequencyOffset;
|
|
|
|
if ((channelFrequencyOffset != m_channelFrequencyOffset)
|
|
|| (channelSampleRate != m_channelSampleRate) || force) {
|
|
m_carrierNco.setFreq(channelFrequencyOffset, channelSampleRate);
|
|
}
|
|
|
|
if ((channelSampleRate != m_channelSampleRate) || force)
|
|
{
|
|
m_interpolatorDistanceRemain = 0;
|
|
m_interpolatorConsumed = false;
|
|
m_interpolatorDistance = (Real) m_audioSampleRate / (Real) channelSampleRate;
|
|
m_interpolator.create(48, m_audioSampleRate, m_settings.m_bandwidth, 3.0);
|
|
}
|
|
|
|
m_channelSampleRate = channelSampleRate;
|
|
m_channelFrequencyOffset = channelFrequencyOffset;
|
|
}
|
|
|
|
void SSBModSource::handleAudio()
|
|
{
|
|
QMutexLocker mlock(&m_mutex);
|
|
unsigned int nbRead;
|
|
|
|
while ((nbRead = m_audioFifo.read(reinterpret_cast<quint8*>(&m_audioReadBuffer[m_audioReadBufferFill]), 4096)) != 0)
|
|
{
|
|
if (m_audioReadBufferFill + nbRead + 4096 < m_audioReadBuffer.size()) {
|
|
m_audioReadBufferFill += nbRead;
|
|
}
|
|
}
|
|
}
|