mirror of
https://github.com/f4exb/sdrangel.git
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400 lines
14 KiB
C++
400 lines
14 KiB
C++
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///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2019 Edouard Griffiths, F4EXB //
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// //
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// This program is free software; you can redistribute it and/or modify //
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// it under the terms of the GNU General Public License as published by //
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// the Free Software Foundation as version 3 of the License, or //
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// (at your option) any later version. //
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// //
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// This program is distributed in the hope that it will be useful, //
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// but WITHOUT ANY WARRANTY; without even the implied warranty of //
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
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// GNU General Public License V3 for more details. //
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// //
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// You should have received a copy of the GNU General Public License //
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// along with this program. If not, see <http://www.gnu.org/licenses/>. //
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///////////////////////////////////////////////////////////////////////////////////
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#include <stdio.h>
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#include <QTime>
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#include <QDebug>
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#include "audio/audiooutput.h"
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#include "dsp/dspengine.h"
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#include "dsp/dspcommands.h"
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#include "dsp/devicesamplemimo.h"
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#include "dsp/basebandsamplesink.h"
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#include "device/deviceapi.h"
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#include "util/db.h"
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#include "ssbdemodsink.h"
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const int SSBDemodSink::m_ssbFftLen = 1024;
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const int SSBDemodSink::m_agcTarget = 3276.8; // -10 dB amplitude => -20 dB power: center of normal signal
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SSBDemodSink::SSBDemodSink() :
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m_audioBinaual(false),
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m_audioFlipChannels(false),
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m_dsb(false),
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m_audioMute(false),
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m_agc(12000, m_agcTarget, 1e-2),
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m_agcActive(false),
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m_agcClamping(false),
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m_agcNbSamples(12000),
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m_agcPowerThreshold(1e-2),
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m_agcThresholdGate(0),
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m_squelchDelayLine(2*48000),
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m_audioActive(false),
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m_spectrumSink(nullptr),
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m_audioFifo(24000)
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{
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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_channelSampleRate = 48000;
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m_channelFrequencyOffset = 0;
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m_audioBuffer.resize(1<<14);
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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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m_agc.setClampMax(SDR_RX_SCALED/100.0);
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m_agc.setClamping(m_agcClamping);
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SSBFilter = new fftfilt(m_LowCutoff / m_audioSampleRate, m_Bandwidth / m_audioSampleRate, m_ssbFftLen);
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DSBFilter = new fftfilt((2.0f * m_Bandwidth) / m_audioSampleRate, 2 * m_ssbFftLen);
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applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
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applySettings(m_settings, true);
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}
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SSBDemodSink::~SSBDemodSink()
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{
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delete SSBFilter;
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delete DSBFilter;
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}
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void SSBDemodSink::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end)
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{
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Complex ci;
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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(), it->imag());
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c *= m_nco.nextIQ();
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if (m_interpolatorDistance < 1.0f) // interpolate
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{
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while (!m_interpolator.interpolate(&m_interpolatorDistanceRemain, c, &ci))
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{
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processOneSample(ci);
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m_interpolatorDistanceRemain += m_interpolatorDistance;
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}
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}
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else
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{
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if (m_interpolator.decimate(&m_interpolatorDistanceRemain, c, &ci))
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{
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processOneSample(ci);
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m_interpolatorDistanceRemain += m_interpolatorDistance;
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}
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}
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}
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}
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void SSBDemodSink::processOneSample(Complex &ci)
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{
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fftfilt::cmplx *sideband;
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int n_out = 0;
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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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if (m_dsb) {
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n_out = DSBFilter->runDSB(ci, &sideband);
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} else {
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n_out = SSBFilter->runSSB(ci, &sideband, m_usb);
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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) / (SDR_RX_SCALED*SDR_RX_SCALED);
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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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float agcVal = m_agcActive ? m_agc.feedAndGetValue(sideband[i]) : 0.1;
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fftfilt::cmplx& delayedSample = m_squelchDelayLine.readBack(m_agc.getStepDownDelay());
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m_audioActive = delayedSample.real() != 0.0;
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m_squelchDelayLine.write(sideband[i]*agcVal);
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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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fftfilt::cmplx z = m_agcActive ? delayedSample * m_agc.getStepValue() : delayedSample;
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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)(z.imag() * m_volume);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(z.real() * m_volume);
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}
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else
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{
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m_audioBuffer[m_audioBufferFill].r = (qint16)(z.real() * m_volume);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(z.imag() * m_volume);
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}
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}
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else
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{
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Real demod = (z.real() + z.imag()) * 0.7;
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qint16 sample = (qint16)(demod * m_volume);
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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);
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if (res != m_audioBufferFill) {
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qDebug("SSBDemodSink::feed: %u/%u samples written", res, m_audioBufferFill);
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}
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m_audioBufferFill = 0;
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}
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}
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uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill);
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if (res != m_audioBufferFill) {
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qDebug("SSBDemodSink::feed: %u/%u tail samples written", res, m_audioBufferFill);
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}
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m_audioBufferFill = 0;
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if (m_spectrumSink != 0) {
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m_spectrumSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), !m_dsb);
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}
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m_sampleBuffer.clear();
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}
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void SSBDemodSink::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
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{
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qDebug() << "SSBDemodSink::applyChannelSettings:"
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<< " channelSampleRate: " << channelSampleRate
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<< " channelFrequencyOffset: " << channelFrequencyOffset;
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if ((m_channelFrequencyOffset != channelFrequencyOffset) ||
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(m_channelSampleRate != channelSampleRate) || force)
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{
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m_nco.setFreq(-channelFrequencyOffset, channelSampleRate);
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}
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if ((m_channelSampleRate != channelSampleRate) || force)
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{
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Real interpolatorBandwidth = (m_Bandwidth * 1.5f) > channelSampleRate ? channelSampleRate : (m_Bandwidth * 1.5f);
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m_interpolator.create(16, channelSampleRate, interpolatorBandwidth, 2.0f);
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m_interpolatorDistanceRemain = 0;
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m_interpolatorDistance = (Real) channelSampleRate / (Real) m_audioSampleRate;
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}
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m_channelSampleRate = channelSampleRate;
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m_channelFrequencyOffset = channelFrequencyOffset;
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}
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void SSBDemodSink::applyAudioSampleRate(int sampleRate)
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{
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qDebug("SSBDemodSink::applyAudioSampleRate: %d", sampleRate);
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Real interpolatorBandwidth = (m_Bandwidth * 1.5f) > m_channelSampleRate ? m_channelSampleRate : (m_Bandwidth * 1.5f);
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m_interpolator.create(16, m_channelSampleRate, interpolatorBandwidth, 2.0f);
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m_interpolatorDistanceRemain = 0;
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m_interpolatorDistance = (Real) m_channelSampleRate / (Real) sampleRate;
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SSBFilter->create_filter(m_LowCutoff / (float) sampleRate, m_Bandwidth / (float) sampleRate);
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DSBFilter->create_dsb_filter((2.0f * m_Bandwidth) / (float) sampleRate);
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int agcNbSamples = (sampleRate / 1000) * (1<<m_settings.m_agcTimeLog2);
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int agcThresholdGate = (sampleRate / 1000) * m_settings.m_agcThresholdGate; // ms
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if (m_agcNbSamples != agcNbSamples)
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{
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m_agc.resize(agcNbSamples, agcNbSamples/2, m_agcTarget);
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m_agc.setStepDownDelay(agcNbSamples);
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m_agcNbSamples = agcNbSamples;
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}
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if (m_agcThresholdGate != agcThresholdGate)
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{
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m_agc.setGate(agcThresholdGate);
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m_agcThresholdGate = agcThresholdGate;
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}
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m_audioFifo.setSize(sampleRate);
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m_audioSampleRate = sampleRate;
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}
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void SSBDemodSink::applySettings(const SSBDemodSettings& settings, bool force)
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{
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qDebug() << "SSBDemodSink::applySettings:"
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<< " m_inputFrequencyOffset: " << settings.m_inputFrequencyOffset
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<< " m_rfBandwidth: " << settings.m_rfBandwidth
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<< " m_lowCutoff: " << settings.m_lowCutoff
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<< " m_volume: " << settings.m_volume
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<< " m_spanLog2: " << settings.m_spanLog2
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<< " m_audioBinaual: " << settings.m_audioBinaural
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<< " m_audioFlipChannels: " << settings.m_audioFlipChannels
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<< " m_dsb: " << settings.m_dsb
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<< " m_audioMute: " << settings.m_audioMute
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<< " m_agcActive: " << settings.m_agc
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<< " m_agcClamping: " << settings.m_agcClamping
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<< " m_agcTimeLog2: " << settings.m_agcTimeLog2
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<< " agcPowerThreshold: " << settings.m_agcPowerThreshold
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<< " agcThresholdGate: " << settings.m_agcThresholdGate
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<< " m_audioDeviceName: " << settings.m_audioDeviceName
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<< " m_streamIndex: " << settings.m_streamIndex
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<< " m_useReverseAPI: " << settings.m_useReverseAPI
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<< " m_reverseAPIAddress: " << settings.m_reverseAPIAddress
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<< " m_reverseAPIPort: " << settings.m_reverseAPIPort
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<< " m_reverseAPIDeviceIndex: " << settings.m_reverseAPIDeviceIndex
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<< " m_reverseAPIChannelIndex: " << settings.m_reverseAPIChannelIndex
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<< " force: " << force;
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if((m_settings.m_rfBandwidth != settings.m_rfBandwidth) ||
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(m_settings.m_lowCutoff != settings.m_lowCutoff) || force)
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{
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float band, lowCutoff;
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band = settings.m_rfBandwidth;
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lowCutoff = settings.m_lowCutoff;
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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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}
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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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Real interpolatorBandwidth = (m_Bandwidth * 1.5f) > m_channelSampleRate ? m_channelSampleRate : (m_Bandwidth * 1.5f);
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m_interpolator.create(16, m_channelSampleRate, interpolatorBandwidth, 2.0f);
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m_interpolatorDistanceRemain = 0;
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m_interpolatorDistance = (Real) m_channelSampleRate / (Real) m_audioSampleRate;
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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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}
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if ((m_settings.m_volume != settings.m_volume) || force)
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{
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m_volume = settings.m_volume;
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m_volume /= 4.0; // for 3276.8
|
||
|
|
}
|
||
|
|
|
||
|
|
if ((m_settings.m_agcTimeLog2 != settings.m_agcTimeLog2) ||
|
||
|
|
(m_settings.m_agcPowerThreshold != settings.m_agcPowerThreshold) ||
|
||
|
|
(m_settings.m_agcThresholdGate != settings.m_agcThresholdGate) ||
|
||
|
|
(m_settings.m_agcClamping != settings.m_agcClamping) || force)
|
||
|
|
{
|
||
|
|
int agcNbSamples = (m_audioSampleRate / 1000) * (1<<settings.m_agcTimeLog2);
|
||
|
|
m_agc.setThresholdEnable(settings.m_agcPowerThreshold != -SSBDemodSettings::m_minPowerThresholdDB);
|
||
|
|
double agcPowerThreshold = CalcDb::powerFromdB(settings.m_agcPowerThreshold) * (SDR_RX_SCALED*SDR_RX_SCALED);
|
||
|
|
int agcThresholdGate = (m_audioSampleRate / 1000) * settings.m_agcThresholdGate; // ms
|
||
|
|
bool agcClamping = settings.m_agcClamping;
|
||
|
|
|
||
|
|
if (m_agcNbSamples != agcNbSamples)
|
||
|
|
{
|
||
|
|
m_agc.resize(agcNbSamples, agcNbSamples/2, m_agcTarget);
|
||
|
|
m_agc.setStepDownDelay(agcNbSamples);
|
||
|
|
m_agcNbSamples = agcNbSamples;
|
||
|
|
}
|
||
|
|
|
||
|
|
if (m_agcPowerThreshold != agcPowerThreshold)
|
||
|
|
{
|
||
|
|
m_agc.setThreshold(agcPowerThreshold);
|
||
|
|
m_agcPowerThreshold = agcPowerThreshold;
|
||
|
|
}
|
||
|
|
|
||
|
|
if (m_agcThresholdGate != agcThresholdGate)
|
||
|
|
{
|
||
|
|
m_agc.setGate(agcThresholdGate);
|
||
|
|
m_agcThresholdGate = agcThresholdGate;
|
||
|
|
}
|
||
|
|
|
||
|
|
if (m_agcClamping != agcClamping)
|
||
|
|
{
|
||
|
|
m_agc.setClamping(agcClamping);
|
||
|
|
m_agcClamping = agcClamping;
|
||
|
|
}
|
||
|
|
|
||
|
|
qDebug() << "SBDemodSink::applySettings: AGC:"
|
||
|
|
<< " agcNbSamples: " << agcNbSamples
|
||
|
|
<< " agcPowerThreshold: " << agcPowerThreshold
|
||
|
|
<< " agcThresholdGate: " << agcThresholdGate
|
||
|
|
<< " agcClamping: " << agcClamping;
|
||
|
|
}
|
||
|
|
|
||
|
|
m_spanLog2 = settings.m_spanLog2;
|
||
|
|
m_audioBinaual = settings.m_audioBinaural;
|
||
|
|
m_audioFlipChannels = settings.m_audioFlipChannels;
|
||
|
|
m_dsb = settings.m_dsb;
|
||
|
|
m_audioMute = settings.m_audioMute;
|
||
|
|
m_agcActive = settings.m_agc;
|
||
|
|
m_settings = settings;
|
||
|
|
}
|
||
|
|
|