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403 lines
13 KiB
C++
403 lines
13 KiB
C++
// 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 <QDebug>
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#include "wfmmodsource.h"
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const int WFMModSource::m_rfFilterFFTLength = 1024;
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const int WFMModSource::m_levelNbSamples = 480; // every 10ms
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WFMModSource::WFMModSource() :
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m_channelSampleRate(384000),
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m_channelFrequencyOffset(0),
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m_modPhasor(0.0f),
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m_audioFifo(4800),
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m_feedbackAudioFifo(48000),
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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_ifstream(nullptr),
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m_audioSampleRate(48000),
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m_feedbackAudioSampleRate(48000)
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{
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m_rfFilter = new fftfilt(-62500.0 / 384000.0, 62500.0 / 384000.0, m_rfFilterFFTLength);
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m_rfFilterBuffer = new Complex[m_rfFilterFFTLength];
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std::fill(m_rfFilterBuffer, m_rfFilterBuffer+m_rfFilterFFTLength, Complex{0,0});
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m_rfFilterBufferIndex = 0;
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m_audioBuffer.resize(1<<14);
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m_audioBufferFill = 0;
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m_magsq = 0.0;
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m_feedbackAudioBuffer.resize(1<<14);
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m_feedbackAudioBufferFill = 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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WFMModSource::~WFMModSource()
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{
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delete m_rfFilter;
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delete[] m_rfFilterBuffer;
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}
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void WFMModSource::pull(SampleVector::iterator begin, unsigned int nbSamples)
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{
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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 WFMModSource::pullOne(Sample& sample)
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{
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if (m_settings.m_channelMute)
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{
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sample.m_real = 0.0f;
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sample.m_imag = 0.0f;
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return;
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}
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Complex ci, ri;
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fftfilt::cmplx *rf;
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int rf_out;
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Real t;
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if ((m_settings.m_modAFInput == WFMModSettings::WFMModInputFile)
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|| (m_settings.m_modAFInput == WFMModSettings::WFMModInputAudio)
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|| (m_settings.m_modAFInput == WFMModSettings::WFMModInputCWTone))
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{
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if (m_interpolatorDistance > 1.0f) // decimate
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{
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modulateAudio();
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while (!m_interpolator.decimate(&m_interpolatorDistanceRemain, m_modSample, &ri)) {
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modulateAudio();
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}
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}
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else // interpolate
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{
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if (m_interpolator.interpolate(&m_interpolatorDistanceRemain, m_modSample, &ri)) {
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modulateAudio();
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}
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}
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t = ri.real();
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m_interpolatorDistanceRemain += m_interpolatorDistance;
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}
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else
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{
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pullAF(t);
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calculateLevel(t);
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}
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m_modPhasor += (m_settings.m_fmDeviation / (float) m_channelSampleRate) * t * M_PI * 2.0f;
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// limit phasor range to ]-pi,pi]
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if (m_modPhasor > M_PI) {
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m_modPhasor -= (2.0f * M_PI);
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}
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ci.real(cos(m_modPhasor) * 0.891235351562f * SDR_TX_SCALEF); // -1 dB
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ci.imag(sin(m_modPhasor) * 0.891235351562f * SDR_TX_SCALEF);
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// RF filtering
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rf_out = m_rfFilter->runFilt(ci, &rf);
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if (rf_out > 0)
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{
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memcpy((void *) m_rfFilterBuffer, (const void *) rf, rf_out*sizeof(Complex));
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m_rfFilterBufferIndex = 0;
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}
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ci = m_rfFilterBuffer[m_rfFilterBufferIndex] * m_carrierNco.nextIQ(); // shift to carrier frequency
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m_rfFilterBufferIndex++;
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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 WFMModSource::modulateAudio()
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{
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Real t;
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pullAF(t);
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calculateLevel(t);
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m_modSample.real(t);
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m_modSample.imag(0.0f);
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if (m_settings.m_feedbackAudioEnable) {
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pushFeedback(t * m_settings.m_feedbackVolumeFactor * 16384.0f);
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}
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}
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void WFMModSource::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 WFMModSource::pullAudio(unsigned int nbSamplesAudio)
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{
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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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m_audioFifo.read(reinterpret_cast<quint8*>(&m_audioBuffer[0]), nbSamplesAudio);
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m_audioBufferFill = 0;
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}
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void WFMModSource::pullAF(Real& sample)
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{
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switch (m_settings.m_modAFInput)
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{
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case WFMModSettings::WFMModInputTone:
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sample = m_toneNco.next() * m_settings.m_volumeFactor;
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break;
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case WFMModSettings::WFMModInputFile:
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// sox f4exb_call.wav --encoding float --endian little f4exb_call.raw
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// ffplay -f f32le -ar 48k -ac 1 f4exb_call.raw
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if (m_ifstream && m_ifstream->is_open())
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{
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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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sample = 0.0f;
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}
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else
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{
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Real s;
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m_ifstream->read(reinterpret_cast<char*>(&s), sizeof(Real));
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sample = s * m_settings.m_volumeFactor;
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}
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}
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else
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{
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sample = 0.0f;
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}
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break;
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case WFMModSettings::WFMModInputAudio:
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{
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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) / 65536.0f) * m_settings.m_volumeFactor;
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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("WFMModSource::pullAF: starve audio samples: size: %u", size);
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sample = ((m_audioBuffer[size-1].l + m_audioBuffer[size-1].r) / 65536.0f) * m_settings.m_volumeFactor;
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}
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}
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break;
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case WFMModSettings::WFMModInputCWTone:
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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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sample = m_cwToneNco.next() * m_settings.m_volumeFactor * fadeFactor;
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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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sample = m_cwToneNco.next() * m_settings.m_volumeFactor * fadeFactor;
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}
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else
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{
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sample = 0.0f;
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m_cwToneNco.setPhase(0);
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}
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}
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break;
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case WFMModSettings::WFMModInputNone:
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default:
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sample = 0.0f;
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break;
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}
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}
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void WFMModSource::pushFeedback(Complex c)
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{
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Complex ci;
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if (m_feedbackInterpolatorDistance < 1.0f) // interpolate
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{
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while (!m_feedbackInterpolator.interpolate(&m_feedbackInterpolatorDistanceRemain, c, &ci))
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{
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processOneSample(ci);
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m_feedbackInterpolatorDistanceRemain += m_feedbackInterpolatorDistance;
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}
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}
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else // decimate
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{
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if (m_feedbackInterpolator.decimate(&m_feedbackInterpolatorDistanceRemain, c, &ci))
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{
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processOneSample(ci);
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m_feedbackInterpolatorDistanceRemain += m_feedbackInterpolatorDistance;
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}
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}
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}
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void WFMModSource::processOneSample(Complex& ci)
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{
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m_feedbackAudioBuffer[m_feedbackAudioBufferFill].l = ci.real();
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m_feedbackAudioBuffer[m_feedbackAudioBufferFill].r = ci.imag();
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++m_feedbackAudioBufferFill;
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if (m_feedbackAudioBufferFill >= m_feedbackAudioBuffer.size())
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{
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unsigned int res = m_feedbackAudioFifo.write((const quint8*)&m_feedbackAudioBuffer[0], m_feedbackAudioBufferFill);
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if (res != m_feedbackAudioBufferFill)
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{
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qDebug("WFMModSource::processOneSample: %u/%u audio samples written m_feedbackInterpolatorDistance: %f",
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res, m_feedbackAudioBufferFill, m_feedbackInterpolatorDistance);
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m_feedbackAudioFifo.clear();
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}
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m_feedbackAudioBufferFill = 0;
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}
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}
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void WFMModSource::calculateLevel(const Real& sample)
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{
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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), sample);
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m_levelSum += sample * sample;
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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 WFMModSource::applyAudioSampleRate(int sampleRate)
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{
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if (sampleRate < 0)
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{
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qWarning("WFMModSource::applyAudioSampleRate: %d", sampleRate);
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return;
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}
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qDebug("WFMModSource::applyAudioSampleRate: %d", sampleRate);
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m_interpolatorDistanceRemain = 0;
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m_interpolatorConsumed = false;
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m_interpolatorDistance = (Real) sampleRate / (Real) m_channelSampleRate;
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m_interpolator.create(48, sampleRate, m_settings.m_rfBandwidth / 2.2, 3.0);
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m_cwToneNco.setFreq(m_settings.m_toneFrequency, sampleRate);
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m_cwKeyer.setSampleRate(sampleRate);
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m_cwKeyer.reset();
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m_audioSampleRate = sampleRate;
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applyFeedbackAudioSampleRate(m_feedbackAudioSampleRate);
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}
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void WFMModSource::applyFeedbackAudioSampleRate(int sampleRate)
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{
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if (sampleRate < 0)
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{
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qWarning("WFMModSource::applyFeedbackAudioSampleRate: invalid sample rate %d", sampleRate);
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return;
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}
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qDebug("WFMModSource::applyFeedbackAudioSampleRate: %d", sampleRate);
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m_feedbackInterpolatorDistanceRemain = 0;
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m_feedbackInterpolatorConsumed = false;
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m_feedbackInterpolatorDistance = (Real) sampleRate / (Real) m_audioSampleRate;
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Real cutoff = std::min(sampleRate, m_audioSampleRate) / 2.2f;
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m_feedbackInterpolator.create(48, sampleRate, cutoff, 3.0);
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m_feedbackAudioSampleRate = sampleRate;
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}
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void WFMModSource::applySettings(const WFMModSettings& settings, bool force)
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{
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if ((settings.m_afBandwidth != m_settings.m_afBandwidth) || 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_audioSampleRate / (Real) m_channelSampleRate;
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m_interpolator.create(48, m_audioSampleRate, settings.m_afBandwidth / 2.2, 3.0);
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}
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if ((settings.m_rfBandwidth != m_settings.m_rfBandwidth) || force)
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{
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Real lowCut = -(settings.m_rfBandwidth / 2.2) / m_channelSampleRate;
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Real hiCut = (settings.m_rfBandwidth / 2.2) / m_channelSampleRate;
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m_rfFilter->create_filter(lowCut, hiCut);
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}
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if ((settings.m_toneFrequency != m_settings.m_toneFrequency) || force)
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{
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m_toneNco.setFreq(settings.m_toneFrequency, m_channelSampleRate);
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m_cwToneNco.setFreq(settings.m_toneFrequency, m_audioSampleRate);
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}
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m_settings = settings;
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}
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void WFMModSource::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
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{
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qDebug() << "WFMModSource::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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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_audioSampleRate / (Real) channelSampleRate;
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m_interpolator.create(48, m_audioSampleRate, m_settings.m_afBandwidth / 2.2, 3.0);
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Real lowCut = -(m_settings.m_rfBandwidth / 2.0) / channelSampleRate;
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Real hiCut = (m_settings.m_rfBandwidth / 2.0) / channelSampleRate;
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m_rfFilter->create_filter(lowCut, hiCut);
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m_toneNco.setFreq(m_settings.m_toneFrequency, channelSampleRate);
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}
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m_channelSampleRate = channelSampleRate;
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m_channelFrequencyOffset = channelFrequencyOffset;
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} |