/////////////////////////////////////////////////////////////////////////////////// // Copyright (C) 2016 Edouard Griffiths, F4EXB // // // // This program is free software; you can redistribute it and/or modify // // it under the terms of the GNU General Public License as published by // // the Free Software Foundation as version 3 of the License, or // // // // This program is distributed in the hope that it will be useful, // // but WITHOUT ANY WARRANTY; without even the implied warranty of // // MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the // // GNU General Public License V3 for more details. // // // // You should have received a copy of the GNU General Public License // // along with this program. If not, see . // /////////////////////////////////////////////////////////////////////////////////// #include "ssbmod.h" #include #include #include #include #include #include #include "dsp/dspengine.h" #include "dsp/pidcontroller.h" #include "dsp/threadedbasebandsamplesource.h" #include "device/devicesinkapi.h" #include "util/db.h" MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureSSBMod, Message) MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureChannelizer, Message) MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureFileSourceName, Message) MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureFileSourceSeek, Message) MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureAFInput, Message) MESSAGE_CLASS_DEFINITION(SSBMod::MsgConfigureFileSourceStreamTiming, Message) MESSAGE_CLASS_DEFINITION(SSBMod::MsgReportFileSourceStreamData, Message) MESSAGE_CLASS_DEFINITION(SSBMod::MsgReportFileSourceStreamTiming, Message) const QString SSBMod::m_channelIdURI = "sdrangel.channeltx.modssb"; const QString SSBMod::m_channelId = "SSBMod"; const int SSBMod::m_levelNbSamples = 480; // every 10ms const int SSBMod::m_ssbFftLen = 1024; SSBMod::SSBMod(DeviceSinkAPI *deviceAPI) : m_deviceAPI(deviceAPI), m_absoluteFrequencyOffset(0), m_SSBFilter(0), m_DSBFilter(0), m_SSBFilterBuffer(0), m_DSBFilterBuffer(0), m_SSBFilterBufferIndex(0), m_DSBFilterBufferIndex(0), m_sampleSink(0), m_movingAverage(40, 0), m_audioFifo(4800), m_settingsMutex(QMutex::Recursive), m_fileSize(0), m_recordLength(0), m_sampleRate(48000), m_afInput(SSBModInputNone), m_levelCalcCount(0), m_peakLevel(0.0f), m_levelSum(0.0f), m_inAGC(9600, 0.2, 1e-4) { setObjectName(m_channelId); m_SSBFilter = new fftfilt(m_settings.m_lowCutoff / m_settings.m_audioSampleRate, m_settings.m_bandwidth / m_settings.m_audioSampleRate, m_ssbFftLen); m_DSBFilter = new fftfilt((2.0f * m_settings.m_bandwidth) / m_settings.m_audioSampleRate, 2 * m_ssbFftLen); m_SSBFilterBuffer = new Complex[m_ssbFftLen>>1]; // filter returns data exactly half of its size m_DSBFilterBuffer = new Complex[m_ssbFftLen]; memset(m_SSBFilterBuffer, 0, sizeof(Complex)*(m_ssbFftLen>>1)); memset(m_DSBFilterBuffer, 0, sizeof(Complex)*(m_ssbFftLen)); m_audioBuffer.resize(1<<14); m_audioBufferFill = 0; m_sum.real(0.0f); m_sum.imag(0.0f); m_undersampleCount = 0; m_sumCount = 0; m_movingAverage.resize(16, 0); m_magsq = 0.0; m_toneNco.setFreq(1000.0, m_settings.m_audioSampleRate); DSPEngine::instance()->addAudioSource(&m_audioFifo); // CW keyer m_cwKeyer.setSampleRate(m_settings.m_audioSampleRate); m_cwKeyer.setWPM(13); m_cwKeyer.setMode(CWKeyerSettings::CWNone); m_inAGC.setGate(m_settings.m_agcThresholdGate); m_inAGC.setStepDownDelay(m_settings.m_agcThresholdDelay); m_inAGC.setClamping(true); m_channelizer = new UpChannelizer(this); m_threadedChannelizer = new ThreadedBasebandSampleSource(m_channelizer, this); m_deviceAPI->addThreadedSource(m_threadedChannelizer); m_deviceAPI->addChannelAPI(this); applySettings(m_settings, true); } SSBMod::~SSBMod() { if (m_SSBFilter) { delete m_SSBFilter; } if (m_DSBFilter) { delete m_DSBFilter; } if (m_SSBFilterBuffer) { delete m_SSBFilterBuffer; } if (m_DSBFilterBuffer) { delete m_DSBFilterBuffer; } DSPEngine::instance()->removeAudioSource(&m_audioFifo); m_deviceAPI->removeChannelAPI(this); m_deviceAPI->removeThreadedSource(m_threadedChannelizer); delete m_threadedChannelizer; delete m_channelizer; } void SSBMod::pull(Sample& sample) { Complex ci; m_settingsMutex.lock(); if (m_interpolatorDistance > 1.0f) // decimate { modulateSample(); while (!m_interpolator.decimate(&m_interpolatorDistanceRemain, m_modSample, &ci)) { modulateSample(); } } else { if (m_interpolator.interpolate(&m_interpolatorDistanceRemain, m_modSample, &ci)) { modulateSample(); } } m_interpolatorDistanceRemain += m_interpolatorDistance; ci *= m_carrierNco.nextIQ(); // shift to carrier frequency ci *= 29204.0f; //scaling at -1 dB to account for possible filter overshoot m_settingsMutex.unlock(); Real magsq = ci.real() * ci.real() + ci.imag() * ci.imag(); magsq /= (1<<30); m_movingAverage.feed(magsq); m_magsq = m_movingAverage.average(); sample.m_real = (FixReal) ci.real(); sample.m_imag = (FixReal) ci.imag(); } void SSBMod::pullAudio(int nbSamples) { unsigned int nbSamplesAudio = nbSamples * ((Real) m_settings.m_audioSampleRate / (Real) m_settings.m_basebandSampleRate); if (nbSamplesAudio > m_audioBuffer.size()) { m_audioBuffer.resize(nbSamplesAudio); } m_audioFifo.read(reinterpret_cast(&m_audioBuffer[0]), nbSamplesAudio, 10); m_audioBufferFill = 0; } void SSBMod::modulateSample() { pullAF(m_modSample); calculateLevel(m_modSample); m_audioBufferFill++; } void SSBMod::pullAF(Complex& sample) { if (m_settings.m_audioMute) { sample.real(0.0f); sample.imag(0.0f); return; } Complex ci; fftfilt::cmplx *filtered; int n_out = 0; int decim = 1<<(m_settings.m_spanLog2 - 1); unsigned char decim_mask = decim - 1; // counter LSB bit mask for decimation by 2^(m_scaleLog2 - 1) switch (m_afInput) { case SSBModInputTone: if (m_settings.m_dsb) { Real t = m_toneNco.next()/1.25; sample.real(t); sample.imag(t); } else { if (m_settings.m_usb) { sample = m_toneNco.nextIQ(); } else { sample = m_toneNco.nextQI(); } } break; case SSBModInputFile: // Monaural (mono): // sox f4exb_call.wav --encoding float --endian little f4exb_call.raw // ffplay -f f32le -ar 48k -ac 1 f4exb_call.raw // Binaural (stereo): // sox f4exb_call.wav --encoding float --endian little f4exb_call.raw // ffplay -f f32le -ar 48k -ac 2 f4exb_call.raw if (m_ifstream.is_open()) { if (m_ifstream.eof()) { if (m_settings.m_playLoop) { m_ifstream.clear(); m_ifstream.seekg(0, std::ios::beg); } } if (m_ifstream.eof()) { ci.real(0.0f); ci.imag(0.0f); } else { if (m_settings.m_audioBinaural) { Complex c; m_ifstream.read(reinterpret_cast(&c), sizeof(Complex)); if (m_settings.m_audioFlipChannels) { ci.real(c.imag() * m_settings.m_volumeFactor); ci.imag(c.real() * m_settings.m_volumeFactor); } else { ci = c * m_settings.m_volumeFactor; } } else { Real real; m_ifstream.read(reinterpret_cast(&real), sizeof(Real)); if (m_settings.m_agc) { ci.real(real); ci.imag(0.0f); m_inAGC.feed(ci); ci *= m_settings.m_volumeFactor; } else { ci.real(real * m_settings.m_volumeFactor); ci.imag(0.0f); } } } } else { ci.real(0.0f); ci.imag(0.0f); } break; case SSBModInputAudio: if (m_settings.m_audioBinaural) { if (m_settings.m_audioFlipChannels) { ci.real((m_audioBuffer[m_audioBufferFill].r / 32768.0f) * m_settings.m_volumeFactor); ci.imag((m_audioBuffer[m_audioBufferFill].l / 32768.0f) * m_settings.m_volumeFactor); } else { ci.real((m_audioBuffer[m_audioBufferFill].l / 32768.0f) * m_settings.m_volumeFactor); ci.imag((m_audioBuffer[m_audioBufferFill].r / 32768.0f) * m_settings.m_volumeFactor); } } else { if (m_settings.m_agc) { ci.real(((m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) / 65536.0f)); ci.imag(0.0f); m_inAGC.feed(ci); ci *= m_settings.m_volumeFactor; } else { ci.real(((m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) / 65536.0f) * m_settings.m_volumeFactor); ci.imag(0.0f); } } break; case SSBModInputCWTone: Real fadeFactor; if (m_cwKeyer.getSample()) { m_cwKeyer.getCWSmoother().getFadeSample(true, fadeFactor); if (m_settings.m_dsb) { Real t = m_toneNco.next() * fadeFactor; sample.real(t); sample.imag(t); } else { if (m_settings.m_usb) { sample = m_toneNco.nextIQ() * fadeFactor; } else { sample = m_toneNco.nextQI() * fadeFactor; } } } else { if (m_cwKeyer.getCWSmoother().getFadeSample(false, fadeFactor)) { if (m_settings.m_dsb) { Real t = (m_toneNco.next() * fadeFactor)/1.25; sample.real(t); sample.imag(t); } else { if (m_settings.m_usb) { sample = m_toneNco.nextIQ() * fadeFactor; } else { sample = m_toneNco.nextQI() * fadeFactor; } } } else { sample.real(0.0f); sample.imag(0.0f); m_toneNco.setPhase(0); } } break; case SSBModInputNone: default: break; } if ((m_afInput == SSBModInputFile) || (m_afInput == SSBModInputAudio)) // real audio { if (m_settings.m_dsb) { n_out = m_DSBFilter->runDSB(ci, &filtered); if (n_out > 0) { memcpy((void *) m_DSBFilterBuffer, (const void *) filtered, n_out*sizeof(Complex)); m_DSBFilterBufferIndex = 0; } sample = m_DSBFilterBuffer[m_DSBFilterBufferIndex]; m_DSBFilterBufferIndex++; } else { n_out = m_SSBFilter->runSSB(ci, &filtered, m_settings.m_usb); if (n_out > 0) { memcpy((void *) m_SSBFilterBuffer, (const void *) filtered, n_out*sizeof(Complex)); m_SSBFilterBufferIndex = 0; } sample = m_SSBFilterBuffer[m_SSBFilterBufferIndex]; m_SSBFilterBufferIndex++; } if (n_out > 0) { for (int i = 0; i < n_out; i++) { // Downsample by 2^(m_scaleLog2 - 1) for SSB band spectrum display // smart decimation with bit gain using float arithmetic (23 bits significand) m_sum += filtered[i]; if (!(m_undersampleCount++ & decim_mask)) { Real avgr = (m_sum.real() / decim) * 29204.0f; //scaling at -1 dB to account for possible filter overshoot Real avgi = (m_sum.imag() / decim) * 29204.0f; // m_magsqSpectrum = (avgr * avgr + avgi * avgi) / (1<<30); // // m_magsqSum += m_magsqSpectrum; // // if (m_magsqSpectrum > m_magsqPeak) // { // m_magsqPeak = m_magsqSpectrum; // } // // m_magsqCount++; if (!m_settings.m_dsb & !m_settings.m_usb) { // invert spectrum for LSB m_sampleBuffer.push_back(Sample(avgi, avgr)); } else { m_sampleBuffer.push_back(Sample(avgr, avgi)); } m_sum.real(0.0); m_sum.imag(0.0); } } } } // Real audio else if ((m_afInput == SSBModInputTone) || (m_afInput == SSBModInputCWTone)) // tone { m_sum += sample; if (!(m_undersampleCount++ & decim_mask)) { Real avgr = (m_sum.real() / decim) * 29204.0f; //scaling at -1 dB to account for possible filter overshoot Real avgi = (m_sum.imag() / decim) * 29204.0f; // m_magsqSpectrum = (avgr * avgr + avgi * avgi) / (1<<30); // // m_magsqSum += m_magsqSpectrum; // // if (m_magsqSpectrum > m_magsqPeak) // { // m_magsqPeak = m_magsqSpectrum; // } // // m_magsqCount++; if (!m_settings.m_dsb & !m_settings.m_usb) { // invert spectrum for LSB m_sampleBuffer.push_back(Sample(avgi, avgr)); } else { m_sampleBuffer.push_back(Sample(avgr, avgi)); } m_sum.real(0.0); m_sum.imag(0.0); } if (m_sumCount < (m_settings.m_dsb ? m_ssbFftLen : m_ssbFftLen>>1)) { n_out = 0; m_sumCount++; } else { n_out = m_sumCount; m_sumCount = 0; } } if (n_out > 0) { if (m_sampleSink != 0) { m_sampleSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), !m_settings.m_dsb); } m_sampleBuffer.clear(); } } void SSBMod::calculateLevel(Complex& sample) { Real t = sample.real(); // TODO: possibly adjust depending on sample type if (m_levelCalcCount < m_levelNbSamples) { m_peakLevel = std::max(std::fabs(m_peakLevel), t); m_levelSum += t * t; m_levelCalcCount++; } else { qreal rmsLevel = sqrt(m_levelSum / m_levelNbSamples); //qDebug("NFMMod::calculateLevel: %f %f", rmsLevel, m_peakLevel); emit levelChanged(rmsLevel, m_peakLevel, m_levelNbSamples); m_peakLevel = 0.0f; m_levelSum = 0.0f; m_levelCalcCount = 0; } } void SSBMod::start() { qDebug() << "SSBMod::start: m_outputSampleRate: " << m_settings.m_outputSampleRate << " m_inputFrequencyOffset: " << m_settings.m_inputFrequencyOffset; m_audioFifo.clear(); } void SSBMod::stop() { } bool SSBMod::handleMessage(const Message& cmd) { if (UpChannelizer::MsgChannelizerNotification::match(cmd)) { UpChannelizer::MsgChannelizerNotification& notif = (UpChannelizer::MsgChannelizerNotification&) cmd; SSBModSettings settings = m_settings; settings.m_basebandSampleRate = notif.getBasebandSampleRate(); settings.m_outputSampleRate = notif.getSampleRate(); settings.m_inputFrequencyOffset = notif.getFrequencyOffset(); applySettings(settings); qDebug() << "SSBMod::handleMessage: MsgChannelizerNotification:" << " m_basebandSampleRate: " << settings.m_basebandSampleRate << " m_outputSampleRate: " << settings.m_outputSampleRate << " m_inputFrequencyOffset: " << settings.m_inputFrequencyOffset; return true; } else if (MsgConfigureChannelizer::match(cmd)) { MsgConfigureChannelizer& cfg = (MsgConfigureChannelizer&) cmd; m_channelizer->configure(m_channelizer->getInputMessageQueue(), cfg.getSampleRate(), cfg.getCenterFrequency()); qDebug() << "SSBMod::handleMessage: MsgConfigureChannelizer: sampleRate: " << cfg.getSampleRate() << " centerFrequency: " << cfg.getCenterFrequency(); return true; } else if (MsgConfigureSSBMod::match(cmd)) { float band, lowCutoff; MsgConfigureSSBMod& cfg = (MsgConfigureSSBMod&) cmd; SSBModSettings settings = cfg.getSettings(); // These settings are set with UpChannelizer::MsgChannelizerNotification m_absoluteFrequencyOffset = settings.m_inputFrequencyOffset; settings.m_basebandSampleRate = m_settings.m_basebandSampleRate; settings.m_outputSampleRate = m_settings.m_outputSampleRate; settings.m_inputFrequencyOffset = m_settings.m_inputFrequencyOffset; band = settings.m_bandwidth; lowCutoff = settings.m_lowCutoff; if (band < 0) // negative means LSB { band = -band; // turn to positive lowCutoff = -lowCutoff; settings.m_usb = false; // and take note of side band } else { settings.m_usb = true; } if (band < 100.0f) // at least 100 Hz { band = 100.0f; lowCutoff = 0; } if (band - lowCutoff < 100.0f) { lowCutoff = band - 100.0f; } settings.m_bandwidth = band; settings.m_lowCutoff = lowCutoff; applySettings(settings, cfg.getForce()); qDebug() << "SSBMod::handleMessage: MsgConfigureSSBMod:" << " m_bandwidth: " << settings.m_bandwidth << " m_lowCutoff: " << settings.m_lowCutoff << " m_toneFrequency: " << settings.m_toneFrequency << " m_volumeFactor: " << settings.m_volumeFactor << " m_spanLog2: " << settings.m_spanLog2 << " m_outputSampleRate: " << m_settings.m_outputSampleRate << " m_audioSampleRate: " << settings.m_audioSampleRate << " m_audioBinaural: " << settings.m_audioBinaural << " m_audioFlipChannels: " << settings.m_audioFlipChannels << " m_dsb: " << settings.m_dsb << " m_audioMute: " << settings.m_audioMute << " m_playLoop: " << settings.m_playLoop << " m_agc: " << settings.m_agc << " m_agcTime: " << settings.m_agcTime << " m_agcOrder: " << settings.m_agcOrder << " m_agcThresholdEnable: " << settings.m_agcThresholdEnable << " m_agcThreshold: " << settings.m_agcThreshold << " m_agcThresholdGate: " << settings.m_agcThresholdGate << " m_agcThresholdDelay: " << settings.m_agcThresholdDelay << " force: " << cfg.getForce(); return true; } else if (MsgConfigureFileSourceName::match(cmd)) { MsgConfigureFileSourceName& conf = (MsgConfigureFileSourceName&) cmd; m_fileName = conf.getFileName(); openFileStream(); return true; } else if (MsgConfigureFileSourceSeek::match(cmd)) { MsgConfigureFileSourceSeek& conf = (MsgConfigureFileSourceSeek&) cmd; int seekPercentage = conf.getPercentage(); seekFileStream(seekPercentage); return true; } else if (MsgConfigureAFInput::match(cmd)) { MsgConfigureAFInput& conf = (MsgConfigureAFInput&) cmd; m_afInput = conf.getAFInput(); return true; } else if (MsgConfigureFileSourceStreamTiming::match(cmd)) { std::size_t samplesCount; if (m_ifstream.eof()) { samplesCount = m_fileSize / sizeof(Real); } else { samplesCount = m_ifstream.tellg() / sizeof(Real); } MsgReportFileSourceStreamTiming *report; report = MsgReportFileSourceStreamTiming::create(samplesCount); getMessageQueueToGUI()->push(report); return true; } else { return false; } } void SSBMod::openFileStream() { if (m_ifstream.is_open()) { m_ifstream.close(); } m_ifstream.open(m_fileName.toStdString().c_str(), std::ios::binary | std::ios::ate); m_fileSize = m_ifstream.tellg(); m_ifstream.seekg(0,std::ios_base::beg); m_sampleRate = 48000; // fixed rate m_recordLength = m_fileSize / (sizeof(Real) * m_sampleRate); qDebug() << "SSBMod::openFileStream: " << m_fileName.toStdString().c_str() << " fileSize: " << m_fileSize << "bytes" << " length: " << m_recordLength << " seconds"; MsgReportFileSourceStreamData *report; report = MsgReportFileSourceStreamData::create(m_sampleRate, m_recordLength); getMessageQueueToGUI()->push(report); } void SSBMod::seekFileStream(int seekPercentage) { QMutexLocker mutexLocker(&m_settingsMutex); if (m_ifstream.is_open()) { int seekPoint = ((m_recordLength * seekPercentage) / 100) * m_sampleRate; seekPoint *= sizeof(Real); m_ifstream.clear(); m_ifstream.seekg(seekPoint, std::ios::beg); } } void SSBMod::applySettings(const SSBModSettings& settings, bool force) { if ((settings.m_bandwidth != m_settings.m_bandwidth) || (settings.m_lowCutoff != m_settings.m_lowCutoff) || (settings.m_audioSampleRate != m_settings.m_audioSampleRate) || force) { m_settingsMutex.lock(); m_SSBFilter->create_filter(settings.m_lowCutoff / settings.m_audioSampleRate, settings.m_bandwidth / settings.m_audioSampleRate); m_DSBFilter->create_dsb_filter((2.0f * settings.m_bandwidth) / settings.m_audioSampleRate); m_settingsMutex.unlock(); } if ((settings.m_inputFrequencyOffset != m_settings.m_inputFrequencyOffset) || (settings.m_outputSampleRate != m_settings.m_outputSampleRate) || force) { m_settingsMutex.lock(); m_carrierNco.setFreq(settings.m_inputFrequencyOffset, settings.m_outputSampleRate); m_settingsMutex.unlock(); } if((settings.m_outputSampleRate != m_settings.m_outputSampleRate) || (settings.m_bandwidth != m_settings.m_bandwidth) || (settings.m_audioSampleRate != m_settings.m_audioSampleRate) || force) { m_settingsMutex.lock(); m_interpolatorDistanceRemain = 0; m_interpolatorConsumed = false; m_interpolatorDistance = (Real) settings.m_audioSampleRate / (Real) settings.m_outputSampleRate; m_interpolator.create(48, settings.m_audioSampleRate, settings.m_bandwidth, 3.0); m_settingsMutex.unlock(); } if ((settings.m_toneFrequency != m_settings.m_toneFrequency) || (settings.m_audioSampleRate != m_settings.m_audioSampleRate) || force) { m_settingsMutex.lock(); m_toneNco.setFreq(settings.m_toneFrequency, settings.m_audioSampleRate); m_settingsMutex.unlock(); } if ((settings.m_audioSampleRate != m_settings.m_audioSampleRate) || force) { m_settingsMutex.lock(); m_cwKeyer.setSampleRate(settings.m_audioSampleRate); m_settingsMutex.unlock(); } if ((settings.m_dsb != m_settings.m_dsb) || force) { if (settings.m_dsb) { memset(m_DSBFilterBuffer, 0, sizeof(Complex)*(m_ssbFftLen)); m_DSBFilterBufferIndex = 0; } else { memset(m_SSBFilterBuffer, 0, sizeof(Complex)*(m_ssbFftLen>>1)); m_SSBFilterBufferIndex = 0; } } if ((settings.m_agcTime != m_settings.m_agcTime) || (settings.m_agcOrder != m_settings.m_agcOrder) || force) { m_inAGC.resize(settings.m_agcTime, settings.m_agcOrder); } if ((settings.m_agcThresholdEnable != m_settings.m_agcThresholdEnable) || force) { m_inAGC.setThresholdEnable(settings.m_agcThresholdEnable); } if ((settings.m_agcThreshold != m_settings.m_agcThreshold) || force) { m_inAGC.setThreshold(settings.m_agcThreshold); } if ((settings.m_agcThresholdGate != m_settings.m_agcThresholdGate) || force) { m_inAGC.setGate(settings.m_agcThresholdGate); } if ((settings.m_agcThresholdDelay != m_settings.m_agcThresholdDelay) || force) { m_inAGC.setStepDownDelay(settings.m_agcThresholdDelay); } m_settings = settings; }