mirror of
https://github.com/f4exb/sdrangel.git
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316 lines
8.4 KiB
C++
316 lines
8.4 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany //
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// written by Christian Daniel //
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// (c) 2014 Modified by John Greb
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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 "../../channelrx/demodssb/ssbdemod.h"
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#include <dsp/downchannelizer.h>
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#include <QTime>
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#include <QDebug>
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#include <stdio.h>
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#include "audio/audiooutput.h"
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#include "dsp/dspengine.h"
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MESSAGE_CLASS_DEFINITION(SSBDemod::MsgConfigureSSBDemod, Message)
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SSBDemod::SSBDemod(BasebandSampleSink* sampleSink) :
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m_sampleSink(sampleSink),
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m_audioFifo(4, 24000),
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m_settingsMutex(QMutex::Recursive),
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m_audioBinaual(false),
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m_audioFlipChannels(false)
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{
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setObjectName("SSBDemod");
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m_Bandwidth = 5000;
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m_LowCutoff = 300;
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m_volume = 2.0;
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m_spanLog2 = 3;
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m_sampleRate = 96000;
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m_frequency = 0;
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m_nco.setFreq(m_frequency, m_sampleRate);
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m_audioSampleRate = DSPEngine::instance()->getAudioSampleRate();
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m_interpolator.create(16, m_sampleRate, 5000);
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m_sampleDistanceRemain = (Real) m_sampleRate / m_audioSampleRate;
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m_audioBuffer.resize(1<<9);
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m_audioBufferFill = 0;
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m_undersampleCount = 0;
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m_sum = 0;
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m_usb = true;
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m_magsq = 0.0f;
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m_magsqSum = 0.0f;
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m_magsqPeak = 0.0f;
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m_magsqCount = 0;
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SSBFilter = new fftfilt(m_LowCutoff / m_audioSampleRate, m_Bandwidth / m_audioSampleRate, ssbFftLen);
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DSBFilter = new fftfilt((2.0f * m_Bandwidth) / m_audioSampleRate, 2 * ssbFftLen);
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DSPEngine::instance()->addAudioSink(&m_audioFifo);
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}
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SSBDemod::~SSBDemod()
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{
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if (SSBFilter) delete SSBFilter;
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if (DSBFilter) delete DSBFilter;
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DSPEngine::instance()->removeAudioSink(&m_audioFifo);
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}
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void SSBDemod::configure(MessageQueue* messageQueue,
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Real Bandwidth,
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Real LowCutoff,
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Real volume,
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int spanLog2,
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bool audioBinaural,
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bool audioFlipChannel,
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bool dsb,
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bool audioMute)
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{
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Message* cmd = MsgConfigureSSBDemod::create(Bandwidth, LowCutoff, volume, spanLog2, audioBinaural, audioFlipChannel, dsb, audioMute);
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messageQueue->push(cmd);
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}
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void SSBDemod::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end, bool positiveOnly)
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{
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Complex ci;
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fftfilt::cmplx *sideband;
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Real avg;
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int n_out;
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m_settingsMutex.lock();
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int decim = 1<<(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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for(SampleVector::const_iterator it = begin; it < end; ++it)
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{
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//Complex c(it->real() / 32768.0, it->imag() / 32768.0);
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Complex c(it->real(), it->imag());
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c *= m_nco.nextIQ();
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if(m_interpolator.decimate(&m_sampleDistanceRemain, c, &ci))
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{
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if (m_dsb)
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{
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n_out = DSBFilter->runDSB(ci, &sideband);
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}
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else
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{
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n_out = SSBFilter->runSSB(ci, &sideband, m_usb);
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}
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m_sampleDistanceRemain += (Real)m_sampleRate / m_audioSampleRate;
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}
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else
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{
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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 += sideband[i];
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if (!(m_undersampleCount++ & decim_mask))
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{
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Real avgr = m_sum.real() / decim;
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Real avgi = m_sum.imag() / decim;
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m_magsq = (avgr * avgr + avgi * avgi) / (1<<30);
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m_magsqSum += m_magsq;
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if (m_magsq > m_magsqPeak)
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{
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m_magsqPeak = m_magsq;
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}
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m_magsqCount++;
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if (!m_dsb & !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_audioMute)
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{
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m_audioBuffer[m_audioBufferFill].r = 0;
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m_audioBuffer[m_audioBufferFill].l = 0;
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}
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else
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{
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if (m_audioBinaual)
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{
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if (m_audioFlipChannels)
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{
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m_audioBuffer[m_audioBufferFill].r = (qint16)(sideband[i].imag() * m_volume * 100);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(sideband[i].real() * m_volume * 100);
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}
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else
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{
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m_audioBuffer[m_audioBufferFill].r = (qint16)(sideband[i].real() * m_volume * 100);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(sideband[i].imag() * m_volume * 100);
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}
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}
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else
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{
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Real demod = (sideband[i].real() + sideband[i].imag()) * 0.7;
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qint16 sample = (qint16)(demod * m_volume * 100);
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m_audioBuffer[m_audioBufferFill].l = sample;
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m_audioBuffer[m_audioBufferFill].r = sample;
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}
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}
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++m_audioBufferFill;
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if (m_audioBufferFill >= m_audioBuffer.size())
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{
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uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 1);
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if (res != m_audioBufferFill)
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{
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qDebug("lost %u samples", m_audioBufferFill - res);
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}
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m_audioBufferFill = 0;
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}
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}
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}
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if (m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 0) != m_audioBufferFill)
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{
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qDebug("SSBDemod::feed: lost samples");
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}
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m_audioBufferFill = 0;
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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_dsb);
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}
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m_sampleBuffer.clear();
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m_settingsMutex.unlock();
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}
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void SSBDemod::start()
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{
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}
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void SSBDemod::stop()
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{
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}
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bool SSBDemod::handleMessage(const Message& cmd)
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{
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float band, lowCutoff;
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qDebug() << "SSBDemod::handleMessage";
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if (DownChannelizer::MsgChannelizerNotification::match(cmd))
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{
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DownChannelizer::MsgChannelizerNotification& notif = (DownChannelizer::MsgChannelizerNotification&) cmd;
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m_settingsMutex.lock();
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m_sampleRate = notif.getSampleRate();
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m_nco.setFreq(-notif.getFrequencyOffset(), m_sampleRate);
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m_interpolator.create(16, m_sampleRate, m_Bandwidth);
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m_sampleDistanceRemain = m_sampleRate / m_audioSampleRate;
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m_settingsMutex.unlock();
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qDebug() << "SSBDemod::handleMessage: MsgChannelizerNotification: m_sampleRate: " << m_sampleRate
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<< " frequencyOffset" << notif.getFrequencyOffset();
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return true;
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}
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else if (MsgConfigureSSBDemod::match(cmd))
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{
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MsgConfigureSSBDemod& cfg = (MsgConfigureSSBDemod&) cmd;
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m_settingsMutex.lock();
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band = cfg.getBandwidth();
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lowCutoff = cfg.getLoCutoff();
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if (band < 0) {
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band = -band;
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lowCutoff = -lowCutoff;
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m_usb = false;
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} else
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m_usb = true;
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if (band < 100.0f)
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{
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band = 100.0f;
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lowCutoff = 0;
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}
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m_Bandwidth = band;
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m_LowCutoff = lowCutoff;
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m_interpolator.create(16, m_sampleRate, band * 2.0f);
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SSBFilter->create_filter(m_LowCutoff / (float) m_audioSampleRate, m_Bandwidth / (float) m_audioSampleRate);
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DSBFilter->create_dsb_filter((2.0f * m_Bandwidth) / (float) m_audioSampleRate);
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m_volume = cfg.getVolume();
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m_volume *= m_volume * 0.1;
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m_spanLog2 = cfg.getSpanLog2();
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m_audioBinaual = cfg.getAudioBinaural();
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m_audioFlipChannels = cfg.getAudioFlipChannels();
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m_dsb = cfg.getDSB();
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m_audioMute = cfg.getAudioMute();
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m_settingsMutex.unlock();
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qDebug() << "SBDemod::handleMessage: MsgConfigureSSBDemod: m_Bandwidth: " << m_Bandwidth
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<< " m_LowCutoff: " << m_LowCutoff
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<< " m_volume: " << m_volume
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<< " m_spanLog2: " << m_spanLog2
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<< " m_audioBinaual: " << m_audioBinaual
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<< " m_audioFlipChannels: " << m_audioFlipChannels
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<< " m_dsb: " << m_dsb
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<< "m_audioMute: " << m_audioMute;
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return true;
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}
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else
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{
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if(m_sampleSink != 0)
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{
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return m_sampleSink->handleMessage(cmd);
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}
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else
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{
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return false;
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}
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}
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}
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