mirror of
https://github.com/f4exb/sdrangel.git
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876 lines
26 KiB
C++
876 lines
26 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2016 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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// //
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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 "ssbmod.h"
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#include <QTime>
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#include <QDebug>
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#include <QMutexLocker>
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#include <stdio.h>
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#include <complex.h>
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#include "dsp/upchannelizer.h"
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#include "dsp/dspengine.h"
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#include "dsp/threadedbasebandsamplesource.h"
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#include "dsp/dspcommands.h"
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#include "device/devicesinkapi.h"
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#include "util/db.h"
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MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureSSBMod, Message)
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MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureChannelizer, Message)
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MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureFileSourceName, Message)
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MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureFileSourceSeek, Message)
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MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureAFInput, Message)
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MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureFileSourceStreamTiming, Message)
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MESSAGE_CLASS_DEFINITION(SSBMod::MsgReportFileSourceStreamData, Message)
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MESSAGE_CLASS_DEFINITION(SSBMod::MsgReportFileSourceStreamTiming, Message)
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const QString SSBMod::m_channelIdURI = "sdrangel.channeltx.modssb";
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const QString SSBMod::m_channelId = "SSBMod";
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const int SSBMod::m_levelNbSamples = 480; // every 10ms
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const int SSBMod::m_ssbFftLen = 1024;
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SSBMod::SSBMod(DeviceSinkAPI *deviceAPI) :
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ChannelSourceAPI(m_channelIdURI),
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m_deviceAPI(deviceAPI),
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m_basebandSampleRate(48000),
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m_outputSampleRate(48000),
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m_inputFrequencyOffset(0),
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m_SSBFilter(0),
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m_DSBFilter(0),
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m_SSBFilterBuffer(0),
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m_DSBFilterBuffer(0),
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m_SSBFilterBufferIndex(0),
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m_DSBFilterBufferIndex(0),
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m_sampleSink(0),
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m_audioFifo(4800),
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m_settingsMutex(QMutex::Recursive),
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m_fileSize(0),
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m_recordLength(0),
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m_sampleRate(48000),
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m_afInput(SSBModInputNone),
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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_inAGC(9600, 0.2, 1e-4)
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{
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setObjectName(m_channelId);
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DSPEngine::instance()->getAudioDeviceManager()->addAudioSource(&m_audioFifo, getInputMessageQueue());
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m_audioSampleRate = DSPEngine::instance()->getAudioDeviceManager()->getInputSampleRate();
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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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memset(m_SSBFilterBuffer, 0, sizeof(Complex)*(m_ssbFftLen>>1));
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memset(m_DSBFilterBuffer, 0, sizeof(Complex)*(m_ssbFftLen));
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m_audioBuffer.resize(1<<14);
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m_audioBufferFill = 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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// CW keyer
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m_cwKeyer.setSampleRate(48000);
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m_cwKeyer.setWPM(13);
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m_cwKeyer.setMode(CWKeyerSettings::CWNone);
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m_inAGC.setGate(m_settings.m_agcThresholdGate);
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m_inAGC.setStepDownDelay(m_settings.m_agcThresholdDelay);
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m_inAGC.setClamping(true);
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applyChannelSettings(m_basebandSampleRate, m_outputSampleRate, m_inputFrequencyOffset, true);
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applySettings(m_settings, true);
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m_channelizer = new UpChannelizer(this);
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m_threadedChannelizer = new ThreadedBasebandSampleSource(m_channelizer, this);
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m_deviceAPI->addThreadedSource(m_threadedChannelizer);
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m_deviceAPI->addChannelAPI(this);
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}
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SSBMod::~SSBMod()
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{
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if (m_SSBFilter) {
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delete m_SSBFilter;
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}
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if (m_DSBFilter) {
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delete m_DSBFilter;
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}
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if (m_SSBFilterBuffer) {
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delete m_SSBFilterBuffer;
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}
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if (m_DSBFilterBuffer) {
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delete m_DSBFilterBuffer;
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}
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DSPEngine::instance()->getAudioDeviceManager()->removeAudioSource(&m_audioFifo);
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m_deviceAPI->removeChannelAPI(this);
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m_deviceAPI->removeThreadedSource(m_threadedChannelizer);
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delete m_threadedChannelizer;
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delete m_channelizer;
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}
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void SSBMod::pull(Sample& sample)
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{
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Complex ci;
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m_settingsMutex.lock();
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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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m_settingsMutex.unlock();
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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 SSBMod::pullAudio(int nbSamples)
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{
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unsigned int nbSamplesAudio = nbSamples * ((Real) m_audioSampleRate / (Real) m_basebandSampleRate);
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if (nbSamplesAudio > m_audioBuffer.size())
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{
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m_audioBuffer.resize(nbSamplesAudio);
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}
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m_audioFifo.read(reinterpret_cast<quint8*>(&m_audioBuffer[0]), nbSamplesAudio, 10);
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m_audioBufferFill = 0;
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}
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void SSBMod::modulateSample()
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{
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pullAF(m_modSample);
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calculateLevel(m_modSample);
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m_audioBufferFill++;
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}
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void SSBMod::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_afInput)
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{
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case 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 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.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(real);
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ci.imag(0.0f);
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m_inAGC.feed(ci);
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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 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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ci.real(((m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) / 65536.0f));
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ci.imag(0.0f);
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m_inAGC.feed(ci);
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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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break;
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case 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 SSBModInputNone:
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default:
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break;
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}
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if ((m_afInput == SSBModInputFile) || (m_afInput == 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_afInput == SSBModInputTone) || (m_afInput == 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_sampleSink != 0)
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{
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m_sampleSink->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 SSBMod::calculateLevel(Complex& sample)
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{
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Real t = sample.real(); // TODO: possibly adjust depending on sample type
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if (m_levelCalcCount < m_levelNbSamples)
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{
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m_peakLevel = std::max(std::fabs(m_peakLevel), t);
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m_levelSum += t * t;
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m_levelCalcCount++;
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}
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else
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{
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qreal rmsLevel = sqrt(m_levelSum / m_levelNbSamples);
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//qDebug("NFMMod::calculateLevel: %f %f", rmsLevel, m_peakLevel);
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emit levelChanged(rmsLevel, m_peakLevel, m_levelNbSamples);
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m_peakLevel = 0.0f;
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m_levelSum = 0.0f;
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m_levelCalcCount = 0;
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}
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}
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void SSBMod::start()
|
|
{
|
|
qDebug() << "SSBMod::start: m_outputSampleRate: " << m_outputSampleRate
|
|
<< " m_inputFrequencyOffset: " << m_settings.m_inputFrequencyOffset;
|
|
|
|
m_audioFifo.clear();
|
|
applyChannelSettings(m_basebandSampleRate, m_outputSampleRate, m_inputFrequencyOffset, true);
|
|
}
|
|
|
|
void SSBMod::stop()
|
|
{
|
|
}
|
|
|
|
bool SSBMod::handleMessage(const Message& cmd)
|
|
{
|
|
if (UpChannelizer::MsgChannelizerNotification::match(cmd))
|
|
{
|
|
UpChannelizer::MsgChannelizerNotification& notif = (UpChannelizer::MsgChannelizerNotification&) cmd;
|
|
qDebug() << "SSBMod::handleMessage: MsgChannelizerNotification";
|
|
|
|
applyChannelSettings(notif.getBasebandSampleRate(), notif.getSampleRate(), notif.getFrequencyOffset());
|
|
|
|
return true;
|
|
}
|
|
else if (MsgConfigureChannelizer::match(cmd))
|
|
{
|
|
MsgConfigureChannelizer& cfg = (MsgConfigureChannelizer&) cmd;
|
|
qDebug() << "SSBMod::handleMessage: MsgConfigureChannelizer: sampleRate: " << cfg.getSampleRate()
|
|
<< " centerFrequency: " << cfg.getCenterFrequency();
|
|
|
|
m_channelizer->configure(m_channelizer->getInputMessageQueue(),
|
|
cfg.getSampleRate(),
|
|
cfg.getCenterFrequency());
|
|
|
|
return true;
|
|
}
|
|
else if (MsgConfigureSSBMod::match(cmd))
|
|
{
|
|
MsgConfigureSSBMod& cfg = (MsgConfigureSSBMod&) cmd;
|
|
qDebug() << "SSBMod::handleMessage: MsgConfigureSSBMod";
|
|
|
|
applySettings(cfg.getSettings(), 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);
|
|
}
|
|
|
|
if (getMessageQueueToGUI())
|
|
{
|
|
MsgReportFileSourceStreamTiming *report;
|
|
report = MsgReportFileSourceStreamTiming::create(samplesCount);
|
|
getMessageQueueToGUI()->push(report);
|
|
}
|
|
|
|
return true;
|
|
}
|
|
else if (DSPConfigureAudio::match(cmd))
|
|
{
|
|
DSPConfigureAudio& cfg = (DSPConfigureAudio&) cmd;
|
|
uint32_t sampleRate = cfg.getSampleRate();
|
|
|
|
qDebug() << "SSBMod::handleMessage: DSPConfigureAudio:"
|
|
<< " sampleRate: " << sampleRate;
|
|
|
|
if (sampleRate != m_audioSampleRate) {
|
|
applyAudioSampleRate(sampleRate);
|
|
}
|
|
|
|
return true;
|
|
}
|
|
else if (DSPSignalNotification::match(cmd))
|
|
{
|
|
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";
|
|
|
|
if (getMessageQueueToGUI())
|
|
{
|
|
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::applyAudioSampleRate(int sampleRate)
|
|
{
|
|
qDebug("SSBMod::applyAudioSampleRate: %d", sampleRate);
|
|
|
|
|
|
MsgConfigureChannelizer* channelConfigMsg = MsgConfigureChannelizer::create(
|
|
sampleRate, m_settings.m_inputFrequencyOffset);
|
|
m_inputMessageQueue.push(channelConfigMsg);
|
|
|
|
m_settingsMutex.lock();
|
|
|
|
m_interpolatorDistanceRemain = 0;
|
|
m_interpolatorConsumed = false;
|
|
m_interpolatorDistance = (Real) sampleRate / (Real) m_outputSampleRate;
|
|
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 < 0) // negative means LSB
|
|
{
|
|
band = -band; // turn to positive
|
|
lowCutoff = -lowCutoff;
|
|
usb = false; // and take note of side band
|
|
}
|
|
else
|
|
{
|
|
usb = true;
|
|
}
|
|
|
|
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_settingsMutex.unlock();
|
|
|
|
m_audioSampleRate = sampleRate;
|
|
|
|
if (getMessageQueueToGUI())
|
|
{
|
|
DSPConfigureAudio *cfg = new DSPConfigureAudio(m_audioSampleRate);
|
|
getMessageQueueToGUI()->push(cfg);
|
|
}
|
|
}
|
|
|
|
void SSBMod::applyChannelSettings(int basebandSampleRate, int outputSampleRate, int inputFrequencyOffset, bool force)
|
|
{
|
|
qDebug() << "SSBMod::applyChannelSettings:"
|
|
<< " basebandSampleRate: " << basebandSampleRate
|
|
<< " outputSampleRate: " << outputSampleRate
|
|
<< " inputFrequencyOffset: " << inputFrequencyOffset;
|
|
|
|
if ((inputFrequencyOffset != m_inputFrequencyOffset) ||
|
|
(outputSampleRate != m_outputSampleRate) || force)
|
|
{
|
|
m_settingsMutex.lock();
|
|
m_carrierNco.setFreq(inputFrequencyOffset, outputSampleRate);
|
|
m_settingsMutex.unlock();
|
|
}
|
|
|
|
if ((outputSampleRate != m_outputSampleRate) || force)
|
|
{
|
|
m_settingsMutex.lock();
|
|
m_interpolatorDistanceRemain = 0;
|
|
m_interpolatorConsumed = false;
|
|
m_interpolatorDistance = (Real) m_audioSampleRate / (Real) outputSampleRate;
|
|
m_interpolator.create(48, m_audioSampleRate, m_settings.m_bandwidth, 3.0);
|
|
m_settingsMutex.unlock();
|
|
}
|
|
|
|
m_basebandSampleRate = basebandSampleRate;
|
|
m_outputSampleRate = outputSampleRate;
|
|
m_inputFrequencyOffset = inputFrequencyOffset;
|
|
}
|
|
|
|
void SSBMod::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 < 0) // negative means LSB
|
|
{
|
|
band = -band; // turn to positive
|
|
lowCutoff = -lowCutoff;
|
|
usb = false; // and take note of side band
|
|
}
|
|
else
|
|
{
|
|
usb = true;
|
|
}
|
|
|
|
if (band < 100.0f) // at least 100 Hz
|
|
{
|
|
band = 100.0f;
|
|
lowCutoff = 0;
|
|
}
|
|
|
|
if (band - lowCutoff < 100.0f) {
|
|
lowCutoff = band - 100.0f;
|
|
}
|
|
|
|
m_settingsMutex.lock();
|
|
m_interpolatorDistanceRemain = 0;
|
|
m_interpolatorConsumed = false;
|
|
m_interpolatorDistance = (Real) m_audioSampleRate / (Real) m_outputSampleRate;
|
|
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_settingsMutex.unlock();
|
|
}
|
|
|
|
if ((settings.m_toneFrequency != m_settings.m_toneFrequency) || force)
|
|
{
|
|
m_settingsMutex.lock();
|
|
m_toneNco.setFreq(settings.m_toneFrequency, 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_settingsMutex.lock();
|
|
m_inAGC.resize(settings.m_agcTime, settings.m_agcOrder);
|
|
m_settingsMutex.unlock();
|
|
}
|
|
|
|
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);
|
|
}
|
|
|
|
if ((settings.m_audioDeviceName != m_settings.m_audioDeviceName) || force)
|
|
{
|
|
AudioDeviceManager *audioDeviceManager = DSPEngine::instance()->getAudioDeviceManager();
|
|
int audioDeviceIndex = audioDeviceManager->getInputDeviceIndex(settings.m_audioDeviceName);
|
|
audioDeviceManager->addAudioSource(&m_audioFifo, getInputMessageQueue(), audioDeviceIndex);
|
|
uint32_t audioSampleRate = audioDeviceManager->getInputSampleRate(audioDeviceIndex);
|
|
|
|
if (m_audioSampleRate != audioSampleRate) {
|
|
applyAudioSampleRate(audioSampleRate);
|
|
}
|
|
}
|
|
|
|
m_settings = settings;
|
|
m_settings.m_bandwidth = band;
|
|
m_settings.m_lowCutoff = lowCutoff;
|
|
m_settings.m_usb = usb;
|
|
}
|
|
|
|
QByteArray SSBMod::serialize() const
|
|
{
|
|
return m_settings.serialize();
|
|
}
|
|
|
|
bool SSBMod::deserialize(const QByteArray& data)
|
|
{
|
|
if (m_settings.deserialize(data))
|
|
{
|
|
MsgConfigureSSBMod *msg = MsgConfigureSSBMod::create(m_settings, true);
|
|
m_inputMessageQueue.push(msg);
|
|
return true;
|
|
}
|
|
else
|
|
{
|
|
m_settings.resetToDefaults();
|
|
MsgConfigureSSBMod *msg = MsgConfigureSSBMod::create(m_settings, true);
|
|
m_inputMessageQueue.push(msg);
|
|
return false;
|
|
}
|
|
}
|