kopia lustrzana https://github.com/f4exb/sdrangel
421 wiersze
13 KiB
C++
421 wiersze
13 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 "ammodsource.h"
|
|
|
|
const int AMModSource::m_levelNbSamples = 480; // every 10ms
|
|
|
|
AMModSource::AMModSource() :
|
|
m_channelSampleRate(48000),
|
|
m_channelFrequencyOffset(0),
|
|
m_audioSampleRate(48000),
|
|
m_audioFifo(12000),
|
|
m_feedbackAudioFifo(48000),
|
|
m_levelCalcCount(0),
|
|
m_peakLevel(0.0f),
|
|
m_levelSum(0.0f),
|
|
m_ifstream(nullptr),
|
|
m_mutex(QMutex::Recursive)
|
|
{
|
|
m_audioFifo.setLabel("AMModSource.m_audioFifo");
|
|
m_feedbackAudioFifo.setLabel("AMModSource.m_feedbackAudioFifo");
|
|
m_audioBuffer.resize(24000);
|
|
m_audioBufferFill = 0;
|
|
m_audioReadBuffer.resize(24000);
|
|
m_audioReadBufferFill = 0;
|
|
m_feedbackAudioBuffer.resize(1<<14);
|
|
m_feedbackAudioBufferFill = 0;
|
|
m_demodBuffer.resize(1<<12);
|
|
m_demodBufferFill = 0;
|
|
m_demodBufferEnabled = false;
|
|
|
|
m_magsq = 0.0;
|
|
|
|
applySettings(m_settings, true);
|
|
applyChannelSettings(m_channelSampleRate, m_channelFrequencyOffset, true);
|
|
}
|
|
|
|
AMModSource::~AMModSource()
|
|
{
|
|
}
|
|
|
|
void AMModSource::pull(SampleVector::iterator begin, unsigned int nbSamples)
|
|
{
|
|
std::for_each(
|
|
begin,
|
|
begin + nbSamples,
|
|
[this](Sample& s) {
|
|
pullOne(s);
|
|
}
|
|
);
|
|
}
|
|
|
|
void AMModSource::pullOne(Sample& sample)
|
|
{
|
|
if (m_settings.m_channelMute)
|
|
{
|
|
sample.m_real = 0.0f;
|
|
sample.m_imag = 0.0f;
|
|
return;
|
|
}
|
|
|
|
Complex ci;
|
|
|
|
if (m_interpolatorDistance > 1.0f) // decimate
|
|
{
|
|
modulateSample();
|
|
|
|
while (!m_interpolator.decimate(&m_interpolatorDistanceRemain, m_modSample, &ci))
|
|
{
|
|
modulateSample();
|
|
}
|
|
}
|
|
else
|
|
{
|
|
if (m_interpolator.interpolate(&m_interpolatorDistanceRemain, m_modSample, &ci))
|
|
{
|
|
modulateSample();
|
|
}
|
|
}
|
|
|
|
m_interpolatorDistanceRemain += m_interpolatorDistance;
|
|
|
|
ci *= m_carrierNco.nextIQ(); // shift to carrier frequency
|
|
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] = ci.real() + ci.imag();
|
|
++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 AMModSource::prefetch(unsigned int nbSamples)
|
|
{
|
|
unsigned int nbSamplesAudio = nbSamples * ((Real) m_audioSampleRate / (Real) m_channelSampleRate);
|
|
pullAudio(nbSamplesAudio);
|
|
}
|
|
|
|
void AMModSource::pullAudio(unsigned int nbSamples)
|
|
{
|
|
QMutexLocker mlock(&m_mutex);
|
|
|
|
if (nbSamples > m_audioBuffer.size()) {
|
|
m_audioBuffer.resize(nbSamples);
|
|
}
|
|
|
|
std::copy(&m_audioReadBuffer[0], &m_audioReadBuffer[nbSamples], &m_audioBuffer[0]);
|
|
m_audioBufferFill = 0;
|
|
|
|
if (m_audioReadBufferFill > nbSamples) // copy back remaining samples at the start of the read buffer
|
|
{
|
|
std::copy(&m_audioReadBuffer[nbSamples], &m_audioReadBuffer[m_audioReadBufferFill], &m_audioReadBuffer[0]);
|
|
m_audioReadBufferFill = m_audioReadBufferFill - nbSamples; // adjust current read buffer fill pointer
|
|
}
|
|
}
|
|
|
|
void AMModSource::modulateSample()
|
|
{
|
|
Real t;
|
|
|
|
pullAF(t);
|
|
|
|
if (m_settings.m_feedbackAudioEnable) {
|
|
pushFeedback(t * m_settings.m_feedbackVolumeFactor * 16384.0f);
|
|
}
|
|
|
|
calculateLevel(t);
|
|
m_audioBufferFill++;
|
|
|
|
m_modSample.real((t*m_settings.m_modFactor + 1.0f) * 16384.0f); // modulate and scale zero frequency carrier
|
|
m_modSample.imag(0.0f);
|
|
}
|
|
|
|
void AMModSource::pullAF(Real& sample)
|
|
{
|
|
switch (m_settings.m_modAFInput)
|
|
{
|
|
case AMModSettings::AMModInputTone:
|
|
sample = m_toneNco.next();
|
|
break;
|
|
case AMModSettings::AMModInputFile:
|
|
// 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
|
|
{
|
|
m_ifstream->read(reinterpret_cast<char*>(&sample), sizeof(Real));
|
|
sample *= m_settings.m_volumeFactor;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
sample = 0.0f;
|
|
}
|
|
break;
|
|
case AMModSettings::AMModInputAudio:
|
|
sample = ((m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) / 65536.0f) * m_settings.m_volumeFactor;
|
|
break;
|
|
case AMModSettings::AMModInputCWTone:
|
|
Real fadeFactor;
|
|
|
|
if (m_cwKeyer.getSample())
|
|
{
|
|
m_cwKeyer.getCWSmoother().getFadeSample(true, fadeFactor);
|
|
sample = m_toneNco.next() * fadeFactor;
|
|
}
|
|
else
|
|
{
|
|
if (m_cwKeyer.getCWSmoother().getFadeSample(false, fadeFactor))
|
|
{
|
|
sample = m_toneNco.next() * fadeFactor;
|
|
}
|
|
else
|
|
{
|
|
sample = 0.0f;
|
|
m_toneNco.setPhase(0);
|
|
}
|
|
}
|
|
break;
|
|
case AMModSettings::AMModInputNone:
|
|
default:
|
|
sample = 0.0f;
|
|
break;
|
|
}
|
|
}
|
|
|
|
void AMModSource::pushFeedback(Real sample)
|
|
{
|
|
Complex c(sample, sample);
|
|
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 AMModSource::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())
|
|
{
|
|
uint res = m_feedbackAudioFifo.write((const quint8*)&m_feedbackAudioBuffer[0], m_feedbackAudioBufferFill);
|
|
|
|
if (res != m_feedbackAudioBufferFill)
|
|
{
|
|
qDebug("AMModChannelSource::pushFeedback: %u/%u audio samples written m_feedbackInterpolatorDistance: %f",
|
|
res, m_feedbackAudioBufferFill, m_feedbackInterpolatorDistance);
|
|
m_feedbackAudioFifo.clear();
|
|
}
|
|
|
|
m_feedbackAudioBufferFill = 0;
|
|
}
|
|
}
|
|
|
|
void AMModSource::calculateLevel(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 AMModSource::applyAudioSampleRate(int sampleRate)
|
|
{
|
|
if (sampleRate < 0)
|
|
{
|
|
qWarning("AMModSource::applyAudioSampleRate: invalid sample rate %d", sampleRate);
|
|
return;
|
|
}
|
|
|
|
qDebug("AMModSource::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_toneNco.setFreq(m_settings.m_toneFrequency, sampleRate);
|
|
m_cwKeyer.setSampleRate(sampleRate);
|
|
m_cwKeyer.reset();
|
|
|
|
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);
|
|
}
|
|
}
|
|
|
|
m_audioSampleRate = sampleRate;
|
|
applyFeedbackAudioSampleRate(m_feedbackAudioSampleRate);
|
|
}
|
|
|
|
void AMModSource::applyFeedbackAudioSampleRate(int sampleRate)
|
|
{
|
|
if (sampleRate < 0)
|
|
{
|
|
qWarning("AMModSource::applyFeedbackAudioSampleRate: invalid sample rate %d", sampleRate);
|
|
return;
|
|
}
|
|
|
|
qDebug("AMModSource::applyFeedbackAudioSampleRate: %u", sampleRate);
|
|
|
|
m_feedbackInterpolatorDistanceRemain = 0;
|
|
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 AMModSource::applySettings(const AMModSettings& settings, bool force)
|
|
{
|
|
if ((settings.m_rfBandwidth != m_settings.m_rfBandwidth) || force)
|
|
{
|
|
m_settings.m_rfBandwidth = settings.m_rfBandwidth;
|
|
applyAudioSampleRate(m_audioSampleRate);
|
|
}
|
|
|
|
if ((settings.m_toneFrequency != m_settings.m_toneFrequency) || force)
|
|
{
|
|
m_toneNco.setFreq(settings.m_toneFrequency, m_audioSampleRate);
|
|
}
|
|
|
|
if ((settings.m_modAFInput != m_settings.m_modAFInput) || force)
|
|
{
|
|
if (settings.m_modAFInput == AMModSettings::AMModInputAudio) {
|
|
connect(&m_audioFifo, SIGNAL(dataReady()), this, SLOT(handleAudio()));
|
|
} else {
|
|
disconnect(&m_audioFifo, SIGNAL(dataReady()), this, SLOT(handleAudio()));
|
|
}
|
|
}
|
|
|
|
m_settings = settings;
|
|
}
|
|
|
|
void AMModSource::applyChannelSettings(int channelSampleRate, int channelFrequencyOffset, bool force)
|
|
{
|
|
qDebug() << "AMModSource::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_rfBandwidth / 2.2, 3.0);
|
|
}
|
|
|
|
m_channelSampleRate = channelSampleRate;
|
|
m_channelFrequencyOffset = channelFrequencyOffset;
|
|
}
|
|
|
|
void AMModSource::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;
|
|
}
|
|
}
|
|
}
|