/////////////////////////////////////////////////////////////////////////////////// // Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany // // written by Christian Daniel // // (c) 2014 Modified by John Greb // // // 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 #include #include #include "ssbdemod.h" #include "audio/audiooutput.h" #include "dsp/dspengine.h" #include "dsp/channelizer.h" MESSAGE_CLASS_DEFINITION(SSBDemod::MsgConfigureSSBDemod, Message) SSBDemod::SSBDemod(SampleSink* sampleSink) : m_sampleSink(sampleSink), m_audioFifo(4, 24000), m_settingsMutex(QMutex::Recursive) { setObjectName("SSBDemod"); m_Bandwidth = 5000; m_LowCutoff = 300; m_volume = 2.0; m_spanLog2 = 3; m_sampleRate = 96000; m_frequency = 0; m_nco.setFreq(m_frequency, m_sampleRate); m_audioSampleRate = DSPEngine::instance()->getAudioSampleRate(); m_interpolator.create(16, m_sampleRate, 5000); m_sampleDistanceRemain = (Real) m_sampleRate / m_audioSampleRate; m_audioBuffer.resize(1<<9); m_audioBufferFill = 0; m_undersampleCount = 0; m_usb = true; SSBFilter = new fftfilt(m_LowCutoff / m_audioSampleRate, m_Bandwidth / m_audioSampleRate, ssbFftLen); DSPEngine::instance()->addAudioSink(&m_audioFifo); } SSBDemod::~SSBDemod() { if (SSBFilter) { delete SSBFilter; } DSPEngine::instance()->removeAudioSink(&m_audioFifo); } void SSBDemod::configure(MessageQueue* messageQueue, Real Bandwidth, Real LowCutoff, Real volume, int spanLog2) { Message* cmd = MsgConfigureSSBDemod::create(Bandwidth, LowCutoff, volume, spanLog2); messageQueue->push(cmd); } void SSBDemod::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end, bool positiveOnly) { Complex ci; fftfilt::cmplx *sideband, sum; Real avg; int n_out; m_settingsMutex.lock(); int decim = 1<<(m_spanLog2 - 1); unsigned char decim_mask = decim - 1; // counter LSB bit mask for decimation by 2^(m_scaleLog2 - 1) for(SampleVector::const_iterator it = begin; it < end; ++it) { Complex c(it->real() / 32768.0, it->imag() / 32768.0); c *= m_nco.nextIQ(); if(m_interpolator.interpolate(&m_sampleDistanceRemain, c, &ci)) { n_out = SSBFilter->runSSB(ci, &sideband, m_usb); m_sampleDistanceRemain += (Real)m_sampleRate / m_audioSampleRate; } else { n_out = 0; } for (int i = 0; i < n_out; i++) { Real demod = (sideband[i].real() + sideband[i].imag()) * 0.7 * 32768.0; // Downsample by 2^(m_scaleLog2 - 1) for SSB band spectrum display // smart decimation with bit gain using float arithmetic (23 bits significand) sum += sideband[i]; if (!(m_undersampleCount++ & decim_mask)) { avg = (sum.real() + sum.imag()) * 0.7 * 32768.0 / decim; m_sampleBuffer.push_back(Sample(avg, 0.0)); sum.real() = 0.0; sum.imag() = 0.0; } qint16 sample = (qint16)(demod * m_volume * 10); m_audioBuffer[m_audioBufferFill].l = sample; m_audioBuffer[m_audioBufferFill].r = sample; ++m_audioBufferFill; if (m_audioBufferFill >= m_audioBuffer.size()) { uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 1); if (res != m_audioBufferFill) { qDebug("lost %u samples", m_audioBufferFill - res); } m_audioBufferFill = 0; } } } if (m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 0) != m_audioBufferFill) { qDebug("SSBDemod::feed: lost samples"); } m_audioBufferFill = 0; if(m_sampleSink != 0) { m_sampleSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), true); } m_sampleBuffer.clear(); m_settingsMutex.unlock(); } void SSBDemod::start() { } void SSBDemod::stop() { } bool SSBDemod::handleMessage(const Message& cmd) { float band, lowCutoff; qDebug() << "SSBDemod::handleMessage"; if (Channelizer::MsgChannelizerNotification::match(cmd)) { Channelizer::MsgChannelizerNotification& notif = (Channelizer::MsgChannelizerNotification&) cmd; m_settingsMutex.lock(); m_sampleRate = notif.getSampleRate(); m_nco.setFreq(-notif.getFrequencyOffset(), m_sampleRate); m_interpolator.create(16, m_sampleRate, m_Bandwidth); m_sampleDistanceRemain = m_sampleRate / m_audioSampleRate; m_settingsMutex.unlock(); qDebug() << "SSBDemod::handleMessage: MsgChannelizerNotification: m_sampleRate: " << m_sampleRate << " frequencyOffset" << notif.getFrequencyOffset(); return true; } else if (MsgConfigureSSBDemod::match(cmd)) { MsgConfigureSSBDemod& cfg = (MsgConfigureSSBDemod&) cmd; m_settingsMutex.lock(); band = cfg.getBandwidth(); lowCutoff = cfg.getLoCutoff(); if (band < 0) { band = -band; lowCutoff = -lowCutoff; m_usb = false; } else m_usb = true; if (band < 100.0f) { band = 100.0f; lowCutoff = 0; } m_Bandwidth = band; m_LowCutoff = lowCutoff; m_interpolator.create(16, m_sampleRate, band * 2.0f); SSBFilter->create_filter(m_LowCutoff / (float) m_audioSampleRate, m_Bandwidth / (float) m_audioSampleRate); m_volume = cfg.getVolume(); m_volume *= m_volume * 0.1; m_spanLog2 = cfg.getSpanLog2(); m_settingsMutex.unlock(); qDebug() << " - MsgConfigureSSBDemod: m_Bandwidth: " << m_Bandwidth << " m_LowCutoff: " << m_LowCutoff << " m_volume: " << m_volume << " m_spanLog2: " << m_spanLog2; return true; } else { if(m_sampleSink != 0) { return m_sampleSink->handleMessage(cmd); } else { return false; } } }