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568 lines
18 KiB
C++
568 lines
18 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2019-2020, 2022 Edouard Griffiths, F4EXB <f4exb06@gmail.com> //
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// Copyright (C) 2022 Jiří Pinkava <jiri.pinkava@rossum.ai> //
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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 <QDebug>
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#include "codec2/freedv_api.h"
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#include "dsp/basebandsamplesink.h"
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#include "freedvmodsource.h"
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const int FreeDVModSource::m_levelNbSamples = 80; // every 10ms
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const int FreeDVModSource::m_ssbFftLen = 1024;
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FreeDVModSource::FreeDVModSource() :
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m_channelSampleRate(48000),
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m_channelFrequencyOffset(0),
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m_modemSampleRate(48000), // // default 2400A mode
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m_lowCutoff(0.0),
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m_hiCutoff(6000.0),
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m_SSBFilter(nullptr),
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m_SSBFilterBuffer(nullptr),
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m_SSBFilterBufferIndex(0),
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m_audioSampleRate(48000),
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m_audioFifo(12000),
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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_freeDV(nullptr),
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m_nSpeechSamples(0),
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m_nNomModemSamples(0),
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m_iSpeech(0),
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m_iModem(0),
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m_speechIn(nullptr),
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m_modOut(nullptr),
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m_scaleFactor(SDR_TX_SCALEF)
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{
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m_audioFifo.setLabel("FreeDVModSource.m_audioFifo");
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m_SSBFilter = new fftfilt(m_lowCutoff / m_audioSampleRate, m_hiCutoff / m_audioSampleRate, 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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std::fill(m_SSBFilterBuffer, m_SSBFilterBuffer+(m_ssbFftLen>>1), Complex{0,0});
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m_audioBuffer.resize(24000);
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m_audioBufferFill = 0;
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m_audioReadBuffer.resize(24000);
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m_audioReadBufferFill = 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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applySettings(m_settings, true);
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applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
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}
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FreeDVModSource::~FreeDVModSource()
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{
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delete m_SSBFilter;
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delete[] m_SSBFilterBuffer;
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if (m_freeDV) {
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freedv_close(m_freeDV);
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}
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}
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void FreeDVModSource::pull(SampleVector::iterator begin, unsigned int nbSamples)
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{
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QMutexLocker mlock(&m_mutex);
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std::for_each(
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begin,
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begin + nbSamples,
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[this](Sample& s) {
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pullOne(s);
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}
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);
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}
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void FreeDVModSource::pullOne(Sample& sample)
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{
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Complex ci;
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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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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 FreeDVModSource::prefetch(unsigned int nbSamples)
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{
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unsigned int nbSamplesAudio = nbSamples * ((Real) m_audioSampleRate / (Real) m_channelSampleRate);
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pullAudio(nbSamplesAudio);
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}
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void FreeDVModSource::pullAudio(unsigned int nbSamples)
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{
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QMutexLocker mlock(&m_mutex);
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unsigned int nbSamplesAudio = nbSamples * ((Real) m_audioSampleRate / (Real) m_modemSampleRate);
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if (nbSamplesAudio > m_audioBuffer.size()) {
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m_audioBuffer.resize(nbSamplesAudio);
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}
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std::copy(&m_audioReadBuffer[0], &m_audioReadBuffer[nbSamplesAudio], &m_audioBuffer[0]);
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m_audioBufferFill = 0;
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if (m_audioReadBufferFill > nbSamplesAudio) // copy back remaining samples at the start of the read buffer
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{
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std::copy(&m_audioReadBuffer[nbSamplesAudio], &m_audioReadBuffer[m_audioReadBufferFill], &m_audioReadBuffer[0]);
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m_audioReadBufferFill = m_audioReadBufferFill - nbSamplesAudio; // adjust current read buffer fill pointer
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}
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}
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qint16 FreeDVModSource::getAudioSample()
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{
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qint16 sample;
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if (m_audioBufferFill < m_audioBuffer.size())
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{
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sample = (m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) * (m_settings.m_volumeFactor / 2.0f);
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m_audioBufferFill++;
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}
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else
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{
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unsigned int size = m_audioBuffer.size();
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qDebug("FreeDVModSource::getAudioSample: starve audio samples: size: %u", size);
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sample = (m_audioBuffer[size-1].l + m_audioBuffer[size-1].r) * (m_settings.m_volumeFactor / 2.0f);
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}
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return sample;
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}
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void FreeDVModSource::modulateSample()
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{
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pullAF(m_modSample);
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if (!m_settings.m_gaugeInputElseModem) {
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calculateLevel(m_modSample);
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}
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}
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void FreeDVModSource::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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if (m_iModem >= m_nNomModemSamples)
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{
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switch (m_settings.m_modAFInput)
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{
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case FreeDVModSettings::FreeDVModInputTone:
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for (int i = 0; i < m_nSpeechSamples; i++)
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{
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m_speechIn[i] = m_toneNco.next() * 32768.0f * m_settings.m_volumeFactor;
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if (m_settings.m_gaugeInputElseModem) {
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calculateLevel(m_speechIn[i]);
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}
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}
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freedv_tx(m_freeDV, m_modOut, m_speechIn);
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break;
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case FreeDVModSettings::FreeDVModInputFile:
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if (m_iModem >= m_nNomModemSamples)
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{
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if (m_ifstream && m_ifstream->is_open())
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{
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std::fill(m_speechIn, m_speechIn + m_nSpeechSamples, 0);
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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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std::fill(m_modOut, m_modOut + m_nNomModemSamples, 0);
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}
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else
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{
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m_ifstream->read(reinterpret_cast<char*>(m_speechIn), sizeof(int16_t) * m_nSpeechSamples);
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if ((m_settings.m_volumeFactor != 1.0) || m_settings.m_gaugeInputElseModem)
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{
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for (int i = 0; i < m_nSpeechSamples; i++)
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{
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if (m_settings.m_volumeFactor != 1.0) {
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m_speechIn[i] *= m_settings.m_volumeFactor;
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}
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if (m_settings.m_gaugeInputElseModem) {
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calculateLevel(m_speechIn[i]);
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}
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}
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}
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freedv_tx(m_freeDV, m_modOut, m_speechIn);
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}
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}
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else
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{
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std::fill(m_modOut, m_modOut + m_nNomModemSamples, 0);
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}
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}
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break;
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case FreeDVModSettings::FreeDVModInputAudio:
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for (int i = 0; i < m_nSpeechSamples; i++)
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{
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qint16 audioSample = getAudioSample();
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while (!m_audioResampler.downSample(audioSample, m_speechIn[i]))
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{
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audioSample = getAudioSample();
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}
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if (m_settings.m_gaugeInputElseModem) {
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calculateLevel(m_speechIn[i]);
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}
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}
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freedv_tx(m_freeDV, m_modOut, m_speechIn);
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break;
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case FreeDVModSettings::FreeDVModInputCWTone:
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for (int i = 0; i < m_nSpeechSamples; i++)
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{
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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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m_speechIn[i] = m_toneNco.next() * 32768.0f * fadeFactor * m_settings.m_volumeFactor;
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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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m_speechIn[i] = m_toneNco.next() * 32768.0f * fadeFactor * m_settings.m_volumeFactor;
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}
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else
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{
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m_speechIn[i] = 0;
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m_toneNco.setPhase(0);
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}
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}
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if (m_settings.m_gaugeInputElseModem) {
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calculateLevel(m_speechIn[i]);
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}
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}
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freedv_tx(m_freeDV, m_modOut, m_speechIn);
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break;
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case FreeDVModSettings::FreeDVModInputNone:
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default:
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std::fill(m_speechIn, m_speechIn + m_nSpeechSamples, 0);
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freedv_tx(m_freeDV, m_modOut, m_speechIn);
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break;
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}
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m_iModem = 0;
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}
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ci.real(m_modOut[m_iModem++] / m_scaleFactor);
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ci.imag(0.0f);
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n_out = m_SSBFilter->runSSB(ci, &filtered, true); // 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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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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m_sampleBuffer.push_back(Sample(avgr, avgi));
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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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if (m_spectrumSink) {
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m_spectrumSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), true); // SSB
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}
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m_sampleBuffer.clear();
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}
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sample = m_SSBFilterBuffer[m_SSBFilterBufferIndex++];
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}
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void FreeDVModSource::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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m_rmsLevel = sqrt(m_levelSum / m_levelNbSamples);
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m_peakLevelOut = m_peakLevel;
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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 FreeDVModSource::calculateLevel(qint16& sample)
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{
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Real t = sample / SDR_TX_SCALEF;
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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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m_rmsLevel = sqrt(m_levelSum / m_levelNbSamples);
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m_peakLevelOut = m_peakLevel;
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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 FreeDVModSource::applyAudioSampleRate(unsigned int sampleRate)
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{
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qDebug("FreeDVModSource::applyAudioSampleRate: %d", sampleRate);
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// TODO: put up simple IIR interpolator when sampleRate < m_modemSampleRate
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m_audioResampler.setDecimation(sampleRate / m_channelSampleRate);
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m_audioResampler.setAudioFilters(sampleRate, sampleRate, 250, 3300);
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m_audioSampleRate = sampleRate;
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}
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void FreeDVModSource::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
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{
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qDebug() << "FreeDVMod::applyChannelSettings:"
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<< " channelSampleRate: " << channelSampleRate
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<< " channelFrequencyOffset: " << channelFrequencyOffset;
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if ((channelFrequencyOffset != m_channelFrequencyOffset) ||
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(channelSampleRate != m_channelSampleRate) || force)
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{
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m_carrierNco.setFreq(channelFrequencyOffset, channelSampleRate);
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}
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if ((channelSampleRate != m_channelSampleRate) || force)
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{
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m_interpolatorDistanceRemain = 0;
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m_interpolatorConsumed = false;
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m_interpolatorDistance = (Real) m_modemSampleRate / (Real) channelSampleRate;
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m_interpolator.create(48, m_modemSampleRate, m_hiCutoff, 3.0);
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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 FreeDVModSource::applyFreeDVMode(FreeDVModSettings::FreeDVMode mode)
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{
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m_hiCutoff = FreeDVModSettings::getHiCutoff(mode);
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m_lowCutoff = FreeDVModSettings::getLowCutoff(mode);
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int modemSampleRate = FreeDVModSettings::getModSampleRate(mode);
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QMutexLocker mlock(&m_mutex);
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m_SSBFilter->create_filter(m_lowCutoff / modemSampleRate, m_hiCutoff / modemSampleRate);
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// baseband interpolator and filter
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if (modemSampleRate != m_modemSampleRate)
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{
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m_interpolatorDistanceRemain = 0;
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m_interpolatorConsumed = false;
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m_interpolatorDistance = (Real) modemSampleRate / (Real) m_channelSampleRate;
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m_interpolator.create(48, modemSampleRate, m_hiCutoff, 3.0);
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m_modemSampleRate = modemSampleRate;
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}
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// FreeDV object
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if (m_freeDV) {
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freedv_close(m_freeDV);
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}
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int fdv_mode = -1;
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switch(mode)
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{
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case FreeDVModSettings::FreeDVMode700C:
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fdv_mode = FREEDV_MODE_700C;
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m_scaleFactor = SDR_TX_SCALEF / 6.4f;
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break;
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case FreeDVModSettings::FreeDVMode700D:
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fdv_mode = FREEDV_MODE_700D;
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m_scaleFactor = SDR_TX_SCALEF / 3.2f;
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break;
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case FreeDVModSettings::FreeDVMode800XA:
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fdv_mode = FREEDV_MODE_800XA;
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m_scaleFactor = SDR_TX_SCALEF / 10.3f;
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break;
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case FreeDVModSettings::FreeDVMode1600:
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fdv_mode = FREEDV_MODE_1600;
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m_scaleFactor = SDR_TX_SCALEF / 4.0f;
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break;
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case FreeDVModSettings::FreeDVMode2400A:
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default:
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fdv_mode = FREEDV_MODE_2400A;
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m_scaleFactor = SDR_TX_SCALEF / 10.3f;
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break;
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}
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if (fdv_mode == FREEDV_MODE_700D)
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{
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struct freedv_advanced adv;
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adv.interleave_frames = 1;
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m_freeDV = freedv_open_advanced(fdv_mode, &adv);
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}
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else
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{
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m_freeDV = freedv_open(fdv_mode);
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}
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if (m_freeDV)
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{
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freedv_set_test_frames(m_freeDV, 0);
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freedv_set_snr_squelch_thresh(m_freeDV, -100.0);
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freedv_set_squelch_en(m_freeDV, 1);
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freedv_set_clip(m_freeDV, 0);
|
|
freedv_set_tx_bpf(m_freeDV, 1);
|
|
freedv_set_ext_vco(m_freeDV, 0);
|
|
|
|
int nSpeechSamples = freedv_get_n_speech_samples(m_freeDV);
|
|
int nNomModemSamples = freedv_get_n_nom_modem_samples(m_freeDV);
|
|
int Fs = freedv_get_modem_sample_rate(m_freeDV);
|
|
int Rs = freedv_get_modem_symbol_rate(m_freeDV);
|
|
|
|
if (nSpeechSamples != m_nSpeechSamples)
|
|
{
|
|
if (m_speechIn) {
|
|
delete[] m_speechIn;
|
|
}
|
|
|
|
m_speechIn = new int16_t[nSpeechSamples];
|
|
m_nSpeechSamples = nSpeechSamples;
|
|
}
|
|
|
|
if (nNomModemSamples != m_nNomModemSamples)
|
|
{
|
|
if (m_modOut) {
|
|
delete[] m_modOut;
|
|
}
|
|
|
|
m_modOut = new int16_t[nNomModemSamples];
|
|
m_nNomModemSamples = nNomModemSamples;
|
|
}
|
|
|
|
m_iSpeech = 0;
|
|
m_iModem = 0;
|
|
|
|
qDebug() << "FreeDVMod::applyFreeDVMode:"
|
|
<< " fdv_mode: " << fdv_mode
|
|
<< " m_modemSampleRate: " << m_modemSampleRate
|
|
<< " m_lowCutoff: " << m_lowCutoff
|
|
<< " m_hiCutoff: " << m_hiCutoff
|
|
<< " Fs: " << Fs
|
|
<< " Rs: " << Rs
|
|
<< " m_nSpeechSamples: " << m_nSpeechSamples
|
|
<< " m_nNomModemSamples: " << m_nNomModemSamples;
|
|
}
|
|
}
|
|
|
|
void FreeDVModSource::applySettings(const FreeDVModSettings& settings, bool force)
|
|
{
|
|
if ((settings.m_toneFrequency != m_settings.m_toneFrequency) || force) {
|
|
m_toneNco.setFreq(settings.m_toneFrequency, m_channelSampleRate);
|
|
}
|
|
|
|
if ((settings.m_modAFInput != m_settings.m_modAFInput) || force)
|
|
{
|
|
if (settings.m_modAFInput == FreeDVModSettings::FreeDVModInputAudio) {
|
|
connect(&m_audioFifo, SIGNAL(dataReady()), this, SLOT(handleAudio()));
|
|
} else {
|
|
disconnect(&m_audioFifo, SIGNAL(dataReady()), this, SLOT(handleAudio()));
|
|
}
|
|
}
|
|
|
|
m_settings = settings;
|
|
}
|
|
|
|
void FreeDVModSource::handleAudio()
|
|
{
|
|
unsigned int nbRead;
|
|
|
|
while ((nbRead = m_audioFifo.read(reinterpret_cast<quint8*>(&m_audioReadBuffer[m_audioReadBufferFill]), 4096)) != 0)
|
|
{
|
|
if (m_audioReadBufferFill + nbRead + 4096 < m_audioReadBuffer.size()) {
|
|
m_audioReadBufferFill += nbRead;
|
|
}
|
|
}
|
|
}
|