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413 lines
12 KiB
C++
413 lines
12 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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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 <QDebug>
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#include "dsp/datafifo.h"
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#include "util/messagequeue.h"
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#include "maincore.h"
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#include "ammodsource.h"
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const int AMModSource::m_levelNbSamples = 480; // every 10ms
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AMModSource::AMModSource() :
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m_channelSampleRate(48000),
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m_channelFrequencyOffset(0),
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m_audioSampleRate(48000),
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m_audioFifo(12000),
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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_mutex(QMutex::Recursive)
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{
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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_feedbackAudioBuffer.resize(1<<14);
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m_feedbackAudioBufferFill = 0;
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m_demodBuffer.resize(1<<12);
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m_demodBufferFill = 0;
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m_demodBufferEnabled = false;
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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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AMModSource::~AMModSource()
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{
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}
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void AMModSource::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 AMModSource::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;
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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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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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m_demodBuffer[m_demodBufferFill] = ci.real() + ci.imag();
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++m_demodBufferFill;
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if (m_demodBufferFill >= m_demodBuffer.size())
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{
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QList<DataFifo*> *dataFifos = MainCore::instance()->getDataPipes().getFifos(m_channel, "demod");
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if (dataFifos)
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{
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QList<DataFifo*>::iterator it = dataFifos->begin();
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for (; it != dataFifos->end(); ++it) {
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(*it)->write((quint8*) &m_demodBuffer[0], m_demodBuffer.size() * sizeof(qint16));
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}
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}
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m_demodBufferFill = 0;
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}
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}
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void AMModSource::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 AMModSource::pullAudio(unsigned int nbSamples)
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{
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QMutexLocker mlock(&m_mutex);
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if (nbSamples > m_audioBuffer.size()) {
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m_audioBuffer.resize(nbSamples);
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}
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std::copy(&m_audioReadBuffer[0], &m_audioReadBuffer[nbSamples], &m_audioBuffer[0]);
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m_audioBufferFill = 0;
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if (m_audioReadBufferFill > nbSamples) // copy back remaining samples at the start of the read buffer
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{
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std::copy(&m_audioReadBuffer[nbSamples], &m_audioReadBuffer[m_audioReadBufferFill], &m_audioReadBuffer[0]);
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m_audioReadBufferFill = m_audioReadBufferFill - nbSamples; // adjust current read buffer fill pointer
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}
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}
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void AMModSource::modulateSample()
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{
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Real t;
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pullAF(t);
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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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calculateLevel(t);
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m_audioBufferFill++;
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m_modSample.real((t*m_settings.m_modFactor + 1.0f) * 16384.0f); // modulate and scale zero frequency carrier
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m_modSample.imag(0.0f);
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}
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void AMModSource::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 AMModSettings::AMModInputTone:
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sample = m_toneNco.next();
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break;
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case AMModSettings::AMModInputFile:
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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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m_ifstream->read(reinterpret_cast<char*>(&sample), sizeof(Real));
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sample *= 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 AMModSettings::AMModInputAudio:
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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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break;
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case AMModSettings::AMModInputCWTone:
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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_toneNco.next() * 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_toneNco.next() * 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_toneNco.setPhase(0);
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}
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}
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break;
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case AMModSettings::AMModInputNone:
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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 AMModSource::pushFeedback(Real sample)
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{
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Complex c(sample, sample);
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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 AMModSource::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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uint 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("AMModChannelSource::pushFeedback: %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 AMModSource::calculateLevel(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 AMModSource::applyAudioSampleRate(int sampleRate)
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{
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if (sampleRate < 0)
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{
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qWarning("AMModSource::applyAudioSampleRate: invalid sample rate %d", sampleRate);
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return;
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}
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qDebug("AMModSource::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_toneNco.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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QList<MessageQueue*> *messageQueues = MainCore::instance()->getMessagePipes().getMessageQueues(m_channel, "reportdemod");
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if (messageQueues)
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{
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QList<MessageQueue*>::iterator it = messageQueues->begin();
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for (; it != messageQueues->end(); ++it)
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{
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MainCore::MsgChannelDemodReport *msg = MainCore::MsgChannelDemodReport::create(m_channel, sampleRate);
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(*it)->push(msg);
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}
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}
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m_audioSampleRate = sampleRate;
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applyFeedbackAudioSampleRate(m_feedbackAudioSampleRate);
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}
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void AMModSource::applyFeedbackAudioSampleRate(int sampleRate)
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{
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if (sampleRate < 0)
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{
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qWarning("AMModSource::applyFeedbackAudioSampleRate: invalid sample rate %d", sampleRate);
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return;
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}
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qDebug("AMModSource::applyFeedbackAudioSampleRate: %u", sampleRate);
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m_feedbackInterpolatorDistanceRemain = 0;
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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 AMModSource::applySettings(const AMModSettings& settings, bool force)
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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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m_settings.m_rfBandwidth = settings.m_rfBandwidth;
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applyAudioSampleRate(m_audioSampleRate);
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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_audioSampleRate);
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}
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if ((settings.m_modAFInput != m_settings.m_modAFInput) || force)
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{
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if (settings.m_modAFInput == AMModSettings::AMModInputAudio) {
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connect(&m_audioFifo, SIGNAL(dataReady()), this, SLOT(handleAudio()));
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} else {
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disconnect(&m_audioFifo, SIGNAL(dataReady()), this, SLOT(handleAudio()));
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}
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}
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m_settings = settings;
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}
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void AMModSource::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
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{
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qDebug() << "AMModSource::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_audioSampleRate / (Real) channelSampleRate;
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m_interpolator.create(48, m_audioSampleRate, m_settings.m_rfBandwidth / 2.2, 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 AMModSource::handleAudio()
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{
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QMutexLocker mlock(&m_mutex);
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unsigned int nbRead;
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while ((nbRead = m_audioFifo.read(reinterpret_cast<quint8*>(&m_audioReadBuffer[m_audioReadBufferFill]), 4096)) != 0)
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{
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if (m_audioReadBufferFill + nbRead + 4096 < m_audioReadBuffer.size()) {
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m_audioReadBufferFill += nbRead;
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}
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}
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}
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