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sdrangel/plugins/channeltx/modwfm/wfmmodsource.cpp

482 lines
15 KiB
C++

// Copyright (C) 2019 Edouard Griffiths, F4EXB //
// //
// This program is free software; you can redistribute it and/or modify //
// it under the terms of the GNU General Public License as published by //
// the Free Software Foundation as version 3 of the License, or //
// (at your option) any later version. //
// //
// This program is distributed in the hope that it will be useful, //
// but WITHOUT ANY WARRANTY; without even the implied warranty of //
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
// GNU General Public License V3 for more details. //
// //
// You should have received a copy of the GNU General Public License //
// along with this program. If not, see <http://www.gnu.org/licenses/>. //
///////////////////////////////////////////////////////////////////////////////////
#include <QDebug>
#include "dsp/datafifo.h"
#include "util/messagequeue.h"
#include "maincore.h"
#include "wfmmodsource.h"
const int WFMModSource::m_rfFilterFFTLength = 1024;
const int WFMModSource::m_levelNbSamples = 480; // every 10ms
WFMModSource::WFMModSource() :
m_channelSampleRate(384000),
m_channelFrequencyOffset(0),
m_modPhasor(0.0f),
m_audioSampleRate(48000),
m_audioFifo(12000),
m_feedbackAudioSampleRate(48000),
m_feedbackAudioFifo(48000),
m_levelCalcCount(0),
m_peakLevel(0.0f),
m_levelSum(0.0f),
m_ifstream(nullptr)
{
m_audioFifo.setLabel("WFMModSource.m_audioFifo");
m_feedbackAudioFifo.setLabel("WFMModSource.m_feedbackAudioFifo");
m_rfFilter = new fftfilt(-62500.0 / 384000.0, 62500.0 / 384000.0, m_rfFilterFFTLength);
m_rfFilterBuffer = new Complex[m_rfFilterFFTLength];
std::fill(m_rfFilterBuffer, m_rfFilterBuffer+m_rfFilterFFTLength, Complex{0,0});
m_rfFilterBufferIndex = 0;
m_audioBuffer.resize(24000);
m_audioBufferFill = 0;
m_audioReadBuffer.resize(24000);
m_audioReadBufferFill = 0;
m_magsq = 0.0;
m_feedbackAudioBuffer.resize(1<<14);
m_feedbackAudioBufferFill = 0;
m_demodBuffer.resize(1<<12);
m_demodBufferFill = 0;
applySettings(m_settings, true);
applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
}
WFMModSource::~WFMModSource()
{
delete m_rfFilter;
delete[] m_rfFilterBuffer;
}
void WFMModSource::pull(SampleVector::iterator begin, unsigned int nbSamples)
{
std::for_each(
begin,
begin + nbSamples,
[this](Sample& s) {
pullOne(s);
}
);
}
void WFMModSource::pullOne(Sample& sample)
{
if (m_settings.m_channelMute)
{
sample.m_real = 0.0f;
sample.m_imag = 0.0f;
return;
}
Complex ci, ri;
fftfilt::cmplx *rf;
int rf_out;
Real t;
if ((m_settings.m_modAFInput == WFMModSettings::WFMModInputFile)
|| (m_settings.m_modAFInput == WFMModSettings::WFMModInputAudio)
|| (m_settings.m_modAFInput == WFMModSettings::WFMModInputCWTone))
{
if (m_interpolatorDistance > 1.0f) // decimate
{
modulateAudio();
while (!m_interpolator.decimate(&m_interpolatorDistanceRemain, m_modSample, &ri)) {
modulateAudio();
}
}
else // interpolate
{
if (m_interpolator.interpolate(&m_interpolatorDistanceRemain, m_modSample, &ri)) {
modulateAudio();
}
}
t = ri.real();
m_interpolatorDistanceRemain += m_interpolatorDistance;
}
else
{
pullAF(t);
calculateLevel(t);
}
m_modPhasor += (m_settings.m_fmDeviation / (float) m_channelSampleRate) * t * M_PI * 2.0f;
// limit phasor range to ]-pi,pi]
if (m_modPhasor > M_PI) {
m_modPhasor -= (2.0f * M_PI);
}
ci.real(cos(m_modPhasor) * 0.891235351562f * SDR_TX_SCALEF); // -1 dB
ci.imag(sin(m_modPhasor) * 0.891235351562f * SDR_TX_SCALEF);
// RF filtering
rf_out = m_rfFilter->runFilt(ci, &rf);
if (rf_out > 0)
{
memcpy((void *) m_rfFilterBuffer, (const void *) rf, rf_out*sizeof(Complex));
m_rfFilterBufferIndex = 0;
}
ci = m_rfFilterBuffer[m_rfFilterBufferIndex] * m_carrierNco.nextIQ(); // shift to carrier frequency
m_rfFilterBufferIndex++;
double magsq = ci.real() * ci.real() + ci.imag() * ci.imag();
magsq /= (SDR_TX_SCALED*SDR_TX_SCALED);
m_movingAverage(magsq);
m_magsq = m_movingAverage.asDouble();
sample.m_real = (FixReal) ci.real();
sample.m_imag = (FixReal) ci.imag();
m_demodBuffer[m_demodBufferFill] = t * std::numeric_limits<int16_t>::max();
++m_demodBufferFill;
if (m_demodBufferFill >= m_demodBuffer.size())
{
QList<ObjectPipe*> dataPipes;
MainCore::instance()->getDataPipes().getDataPipes(m_channel, "demod", dataPipes);
if (dataPipes.size() > 0)
{
QList<ObjectPipe*>::iterator it = dataPipes.begin();
for (; it != dataPipes.end(); ++it)
{
DataFifo *fifo = qobject_cast<DataFifo*>((*it)->m_element);
if (fifo) {
fifo->write((quint8*) &m_demodBuffer[0], m_demodBuffer.size() * sizeof(qint16), DataFifo::DataTypeI16);
}
}
}
m_demodBufferFill = 0;
}
}
void WFMModSource::modulateAudio()
{
Real t;
pullAF(t);
calculateLevel(t);
m_modSample.real(t);
m_modSample.imag(0.0f);
if (m_settings.m_feedbackAudioEnable) {
pushFeedback(t * m_settings.m_feedbackVolumeFactor * 16384.0f);
}
}
void WFMModSource::prefetch(unsigned int nbSamples)
{
unsigned int nbSamplesAudio = nbSamples * ((Real) m_audioSampleRate / (Real) m_channelSampleRate);
pullAudio(nbSamplesAudio);
}
void WFMModSource::pullAudio(unsigned int nbSamplesAudio)
{
QMutexLocker mlock(&m_mutex);
if (nbSamplesAudio > m_audioBuffer.size()) {
m_audioBuffer.resize(nbSamplesAudio);
}
std::copy(&m_audioReadBuffer[0], &m_audioReadBuffer[nbSamplesAudio], &m_audioBuffer[0]);
m_audioBufferFill = 0;
if (m_audioReadBufferFill > nbSamplesAudio) // copy back remaining samples at the start of the read buffer
{
std::copy(&m_audioReadBuffer[nbSamplesAudio], &m_audioReadBuffer[m_audioReadBufferFill], &m_audioReadBuffer[0]);
m_audioReadBufferFill = m_audioReadBufferFill - nbSamplesAudio; // adjust current read buffer fill pointer
}
}
void WFMModSource::pullAF(Real& sample)
{
switch (m_settings.m_modAFInput)
{
case WFMModSettings::WFMModInputTone:
sample = m_toneNco.next() * m_settings.m_volumeFactor;
break;
case WFMModSettings::WFMModInputFile:
// sox f4exb_call.wav --encoding float --endian little f4exb_call.raw
// ffplay -f f32le -ar 48k -ac 1 f4exb_call.raw
if (m_ifstream && m_ifstream->is_open())
{
if (m_ifstream->eof())
{
if (m_settings.m_playLoop)
{
m_ifstream->clear();
m_ifstream->seekg(0, std::ios::beg);
}
}
if (m_ifstream->eof())
{
sample = 0.0f;
}
else
{
Real s;
m_ifstream->read(reinterpret_cast<char*>(&s), sizeof(Real));
sample = s * m_settings.m_volumeFactor;
}
}
else
{
sample = 0.0f;
}
break;
case WFMModSettings::WFMModInputAudio:
{
if (m_audioBufferFill < m_audioBuffer.size())
{
sample = ((m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) / 65536.0f) * m_settings.m_volumeFactor;
m_audioBufferFill++;
}
else
{
unsigned int size = m_audioBuffer.size();
qDebug("WFMModSource::pullAF: starve audio samples: size: %u", size);
sample = ((m_audioBuffer[size-1].l + m_audioBuffer[size-1].r) / 65536.0f) * m_settings.m_volumeFactor;
}
}
break;
case WFMModSettings::WFMModInputCWTone:
Real fadeFactor;
if (m_cwKeyer.getSample())
{
m_cwKeyer.getCWSmoother().getFadeSample(true, fadeFactor);
sample = m_cwToneNco.next() * m_settings.m_volumeFactor * fadeFactor * 0.99f;
}
else
{
if (m_cwKeyer.getCWSmoother().getFadeSample(false, fadeFactor))
{
sample = m_cwToneNco.next() * m_settings.m_volumeFactor * fadeFactor * 0.99f;
}
else
{
sample = 0.0f;
m_cwToneNco.setPhase(0);
}
}
break;
case WFMModSettings::WFMModInputNone:
default:
sample = 0.0f;
break;
}
}
void WFMModSource::pushFeedback(Complex c)
{
Complex ci;
if (m_feedbackInterpolatorDistance < 1.0f) // interpolate
{
while (!m_feedbackInterpolator.interpolate(&m_feedbackInterpolatorDistanceRemain, c, &ci))
{
processOneSample(ci);
m_feedbackInterpolatorDistanceRemain += m_feedbackInterpolatorDistance;
}
}
else // decimate
{
if (m_feedbackInterpolator.decimate(&m_feedbackInterpolatorDistanceRemain, c, &ci))
{
processOneSample(ci);
m_feedbackInterpolatorDistanceRemain += m_feedbackInterpolatorDistance;
}
}
}
void WFMModSource::processOneSample(Complex& ci)
{
m_feedbackAudioBuffer[m_feedbackAudioBufferFill].l = ci.real();
m_feedbackAudioBuffer[m_feedbackAudioBufferFill].r = ci.imag();
++m_feedbackAudioBufferFill;
if (m_feedbackAudioBufferFill >= m_feedbackAudioBuffer.size())
{
unsigned int res = m_feedbackAudioFifo.write((const quint8*)&m_feedbackAudioBuffer[0], m_feedbackAudioBufferFill);
if (res != m_feedbackAudioBufferFill)
{
qDebug("WFMModSource::processOneSample: %u/%u audio samples written m_feedbackInterpolatorDistance: %f",
res, m_feedbackAudioBufferFill, m_feedbackInterpolatorDistance);
m_feedbackAudioFifo.clear();
}
m_feedbackAudioBufferFill = 0;
}
}
void WFMModSource::calculateLevel(const Real& sample)
{
if (m_levelCalcCount < m_levelNbSamples)
{
m_peakLevel = std::max(std::fabs(m_peakLevel), sample);
m_levelSum += sample * sample;
m_levelCalcCount++;
}
else
{
m_rmsLevel = sqrt(m_levelSum / m_levelNbSamples);
m_peakLevelOut = m_peakLevel;
m_peakLevel = 0.0f;
m_levelSum = 0.0f;
m_levelCalcCount = 0;
}
}
void WFMModSource::applyAudioSampleRate(int sampleRate)
{
if (sampleRate < 0)
{
qWarning("WFMModSource::applyAudioSampleRate: %d", sampleRate);
return;
}
qDebug("WFMModSource::applyAudioSampleRate: %d", sampleRate);
m_interpolatorDistanceRemain = 0;
m_interpolatorConsumed = false;
m_interpolatorDistance = (Real) sampleRate / (Real) m_channelSampleRate;
m_interpolator.create(48, sampleRate, m_settings.m_rfBandwidth / 2.2, 3.0);
m_cwToneNco.setFreq(m_settings.m_toneFrequency, sampleRate);
m_cwKeyer.setSampleRate(sampleRate);
m_cwKeyer.reset();
m_audioSampleRate = sampleRate;
applyFeedbackAudioSampleRate(m_feedbackAudioSampleRate);
QList<ObjectPipe*> pipes;
MainCore::instance()->getMessagePipes().getMessagePipes(m_channel, "reportdemod", pipes);
if (pipes.size() > 0)
{
for (const auto& pipe : pipes)
{
MessageQueue* messageQueue = qobject_cast<MessageQueue*>(pipe->m_element);
MainCore::MsgChannelDemodReport *msg = MainCore::MsgChannelDemodReport::create(m_channel, sampleRate);
messageQueue->push(msg);
}
}
}
void WFMModSource::applyFeedbackAudioSampleRate(int sampleRate)
{
if (sampleRate < 0)
{
qWarning("WFMModSource::applyFeedbackAudioSampleRate: invalid sample rate %d", sampleRate);
return;
}
qDebug("WFMModSource::applyFeedbackAudioSampleRate: %d", sampleRate);
m_feedbackInterpolatorDistanceRemain = 0;
m_feedbackInterpolatorConsumed = false;
m_feedbackInterpolatorDistance = (Real) sampleRate / (Real) m_audioSampleRate;
Real cutoff = std::min(sampleRate, m_audioSampleRate) / 2.2f;
m_feedbackInterpolator.create(48, sampleRate, cutoff, 3.0);
m_feedbackAudioSampleRate = sampleRate;
}
void WFMModSource::applySettings(const WFMModSettings& settings, bool force)
{
if ((settings.m_afBandwidth != m_settings.m_afBandwidth) || force)
{
m_interpolatorDistanceRemain = 0;
m_interpolatorConsumed = false;
m_interpolatorDistance = (Real) m_audioSampleRate / (Real) m_channelSampleRate;
m_interpolator.create(48, m_audioSampleRate, settings.m_afBandwidth / 2.2, 3.0);
}
if ((settings.m_rfBandwidth != m_settings.m_rfBandwidth) || force)
{
Real lowCut = -(settings.m_rfBandwidth / 2.2) / m_channelSampleRate;
Real hiCut = (settings.m_rfBandwidth / 2.2) / m_channelSampleRate;
m_rfFilter->create_filter(lowCut, hiCut);
}
if ((settings.m_toneFrequency != m_settings.m_toneFrequency) || force)
{
m_toneNco.setFreq(settings.m_toneFrequency, m_channelSampleRate);
m_cwToneNco.setFreq(settings.m_toneFrequency, m_audioSampleRate);
}
if ((settings.m_modAFInput != m_settings.m_modAFInput) || force)
{
if (settings.m_modAFInput == WFMModSettings::WFMModInputAudio) {
connect(&m_audioFifo, SIGNAL(dataReady()), this, SLOT(handleAudio()));
} else {
disconnect(&m_audioFifo, SIGNAL(dataReady()), this, SLOT(handleAudio()));
}
}
m_settings = settings;
}
void WFMModSource::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
{
qDebug() << "WFMModSource::applyChannelSettings:"
<< " channelSampleRate: " << channelSampleRate
<< " channelFrequencyOffset: " << channelFrequencyOffset;
if ((channelFrequencyOffset != m_channelFrequencyOffset)
|| (channelSampleRate != m_channelSampleRate) || force) {
m_carrierNco.setFreq(channelFrequencyOffset, channelSampleRate);
}
if ((channelSampleRate != m_channelSampleRate) || force)
{
m_interpolatorDistanceRemain = 0;
m_interpolatorConsumed = false;
m_interpolatorDistance = (Real) m_audioSampleRate / (Real) channelSampleRate;
m_interpolator.create(48, m_audioSampleRate, m_settings.m_afBandwidth / 2.2, 3.0);
Real lowCut = -(m_settings.m_rfBandwidth / 2.0) / channelSampleRate;
Real hiCut = (m_settings.m_rfBandwidth / 2.0) / channelSampleRate;
m_rfFilter->create_filter(lowCut, hiCut);
m_toneNco.setFreq(m_settings.m_toneFrequency, channelSampleRate);
}
m_channelSampleRate = channelSampleRate;
m_channelFrequencyOffset = channelFrequencyOffset;
}
void WFMModSource::handleAudio()
{
QMutexLocker mlock(&m_mutex);
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;
}
}
}