1
0
mirror of https://github.com/f4exb/sdrangel.git synced 2024-11-07 17:16:02 -05:00
sdrangel/sdrbase/dsp/channelizer.cpp

329 lines
9.7 KiB
C++

#include "dsp/channelizer.h"
#include "dsp/inthalfbandfilter.h"
#include "dsp/dspcommands.h"
Channelizer::Channelizer(SampleSink* sampleSink) :
m_sampleSink(sampleSink),
m_inputSampleRate(100000),
m_requestedOutputSampleRate(100000),
m_requestedCenterFrequency(0),
m_currentOutputSampleRate(100000),
m_currentCenterFrequency(0)
{
}
Channelizer::~Channelizer()
{
freeFilterChain();
}
void Channelizer::configure(MessageQueue* messageQueue, int sampleRate, int centerFrequency)
{
Message* cmd = DSPConfigureChannelizer::create(sampleRate, centerFrequency);
cmd->submit(messageQueue, this);
}
void Channelizer::feed(SampleVector::const_iterator begin, SampleVector::const_iterator end, bool firstOfBurst)
{
for(SampleVector::const_iterator sample = begin; sample != end; ++sample) {
Sample s(*sample);
FilterStages::iterator stage = m_filterStages.begin();
while(stage != m_filterStages.end()) {
if(!(*stage)->work(&s))
break;
++stage;
}
if(stage == m_filterStages.end())
m_sampleBuffer.push_back(s);
}
if(m_sampleSink != NULL)
m_sampleSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), firstOfBurst);
m_sampleBuffer.clear();
}
void Channelizer::start()
{
if(m_sampleSink != NULL)
m_sampleSink->start();
}
void Channelizer::stop()
{
if(m_sampleSink != NULL)
m_sampleSink->stop();
}
bool Channelizer::handleMessage(Message* cmd)
{
if(DSPSignalNotification::match(cmd)) {
DSPSignalNotification* signal = (DSPSignalNotification*)cmd;
m_inputSampleRate = signal->getSampleRate();
applyConfiguration();
cmd->completed();
if(m_sampleSink != NULL) {
signal = DSPSignalNotification::create(m_currentOutputSampleRate, m_currentCenterFrequency);
if(!m_sampleSink->handleMessage(signal))
signal->completed();
}
return true;
} else if(DSPConfigureChannelizer::match(cmd)) {
DSPConfigureChannelizer* chan = (DSPConfigureChannelizer*)cmd;
m_requestedOutputSampleRate = chan->getSampleRate();
m_requestedCenterFrequency = chan->getCenterFrequency();
applyConfiguration();
cmd->completed();
if(m_sampleSink != NULL) {
DSPSignalNotification* signal = DSPSignalNotification::create(m_currentOutputSampleRate, m_currentCenterFrequency);
if(!m_sampleSink->handleMessage(signal))
signal->completed();
}
return true;
} else {
if(m_sampleSink != NULL)
return m_sampleSink->handleMessage(cmd);
else return false;
}
}
void Channelizer::applyConfiguration()
{
freeFilterChain();
m_currentCenterFrequency = createFilterChain(
m_inputSampleRate / -2, m_inputSampleRate / 2,
m_requestedCenterFrequency - m_requestedOutputSampleRate / 2, m_requestedCenterFrequency + m_requestedOutputSampleRate / 2);
m_currentOutputSampleRate = m_inputSampleRate / (1 << m_filterStages.size());
}
Channelizer::FilterStage::FilterStage(Mode mode) :
m_filter(new IntHalfbandFilter),
m_workFunction(NULL)
{
switch(mode) {
case ModeCenter:
m_workFunction = &IntHalfbandFilter::workDecimateCenter;
break;
case ModeLowerHalf:
m_workFunction = &IntHalfbandFilter::workDecimateLowerHalf;
break;
case ModeUpperHalf:
m_workFunction = &IntHalfbandFilter::workDecimateUpperHalf;
break;
}
}
Channelizer::FilterStage::~FilterStage()
{
delete m_filter;
}
bool Channelizer::signalContainsChannel(Real sigStart, Real sigEnd, Real chanStart, Real chanEnd) const
{
//qDebug(" testing signal [%f, %f], channel [%f, %f]", sigStart, sigEnd, chanStart, chanEnd);
if(sigEnd <= sigStart)
return false;
if(chanEnd <= chanStart)
return false;
return (sigStart <= chanStart) && (sigEnd >= chanEnd);
}
Real Channelizer::createFilterChain(Real sigStart, Real sigEnd, Real chanStart, Real chanEnd)
{
Real sigBw = sigEnd - sigStart;
Real safetyMargin = sigBw / 20;
Real rot = sigBw / 4;
safetyMargin = 0;
//qDebug("Signal [%f, %f] (BW %f), Channel [%f, %f], Rot %f, Safety %f", sigStart, sigEnd, sigBw, chanStart, chanEnd, rot, safetyMargin);
#if 1
// check if it fits into the left half
if(signalContainsChannel(sigStart + safetyMargin, sigStart + sigBw / 2.0 - safetyMargin, chanStart, chanEnd)) {
//qDebug("-> take left half (rotate by +1/4 and decimate by 2)");
m_filterStages.push_back(new FilterStage(FilterStage::ModeLowerHalf));
return createFilterChain(sigStart, sigStart + sigBw / 2.0, chanStart, chanEnd);
}
// check if it fits into the right half
if(signalContainsChannel(sigEnd - sigBw / 2.0f + safetyMargin, sigEnd - safetyMargin, chanStart, chanEnd)) {
//qDebug("-> take right half (rotate by -1/4 and decimate by 2)");
m_filterStages.push_back(new FilterStage(FilterStage::ModeUpperHalf));
return createFilterChain(sigEnd - sigBw / 2.0f, sigEnd, chanStart, chanEnd);
}
// check if it fits into the center
if(signalContainsChannel(sigStart + rot + safetyMargin, sigStart + rot + sigBw / 2.0f - safetyMargin, chanStart, chanEnd)) {
//qDebug("-> take center half (decimate by 2)");
m_filterStages.push_back(new FilterStage(FilterStage::ModeCenter));
return createFilterChain(sigStart + rot, sigStart + sigBw / 2.0f + rot, chanStart, chanEnd);
}
#endif
Real ofs = ((chanEnd - chanStart) / 2.0 + chanStart) - ((sigEnd - sigStart) / 2.0 + sigStart);
qDebug("-> complete (final BW %f, frequency offset %f)", sigBw, ofs);
return ofs;
}
void Channelizer::freeFilterChain()
{
for(FilterStages::iterator it = m_filterStages.begin(); it != m_filterStages.end(); ++it)
delete *it;
m_filterStages.clear();
}
#if 0
///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany //
// written by Christian Daniel //
// //
// 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 //
// //
// 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 <QTime>
#include <stdio.h>
#include "channelizer.h"
#include "hardware/audiooutput.h"
Channelizer::Channelizer()
{
#if 0
m_spectrum.configure(128 , 25, FFTWindow::Bartlett);
m_buffer.resize(2048);
m_bufferFill = 0;
m_nco.setFreq(-100000, 500000);
m_interpolator.create(51, 32, 32 * 500000, 150000 / 2);
m_distance = 500000.0 / 176400.0;
m_interpolator2.create(19, 8, 8 * 176400, 15000 / 2);
m_distance2 = 4;
m_audioFifo.setSize(4, 44100 / 2 * 4);
m_audioOutput = new AudioOutput;
m_audioOutput->start(0, 44100, &m_audioFifo);
m_resampler = 1.0;
m_resamplerCtrl.setup(0.00001, 0, -0.00001);
#endif
}
Channelizer::~Channelizer()
{
#if 0
m_audioOutput->stop();
delete m_audioOutput;
#endif
}
#if 0
void Channelizer::setGLSpectrum(GLSpectrum* glSpectrum)
{
m_spectrum.setGLSpectrum(glSpectrum);
}
#endif
size_t Channelizer::workUnitSize()
{
#if 0
return m_buffer.size();
#endif
return 0;
}
size_t Channelizer::work(SampleVector::const_iterator begin, SampleVector::const_iterator end)
{
#if 0
int buffered = m_audioOutput->bufferedSamples();
if(m_audioFifo.fill() < (m_audioFifo.size() / 6)) {
while(m_audioFifo.fill() < (m_audioFifo.size() / 2)) {
quint32 d = 0;
m_audioFifo.write((quint8*)&d, 4);
}
qDebug("underflow - fill %d (vs %d)", m_audioFifo.fill(), m_audioFifo.size() / 4 / 2);
}
buffered = m_audioOutput->bufferedSamples();
int fill = m_audioFifo.fill() / 4 + buffered;
float err = (float)fill / ((m_audioFifo.size() / 4) / 2);
float ctrl = m_resamplerCtrl.feed(err);
//float resamplerRate = (ctrl / 1.0);
float resamplerRate = err;
if(resamplerRate < 0.9999)
resamplerRate = 0.9999;
else if(resamplerRate > 1.0001)
resamplerRate = 1.0001;
m_resampler = m_resampler * 0.99 + resamplerRate * 0.01;
//m_resampler = resamplerRate;
if(m_resampler < 0.995)
m_resampler = 0.995;
else if(m_resampler > 1.005)
m_resampler = 1.005;
//qDebug("%lld %5d %f %f %f", QDateTime::currentMSecsSinceEpoch(), fill, ctrl, m_resampler, err);
struct AudioSample {
qint16 l;
qint16 r;
};
size_t count = end - begin;
Complex ci;
bool consumed;
bool consumed2;
for(SampleVector::const_iterator it = begin; it < end; it++) {
Complex c(it->real() / 32768.0, it->imag() / 32768.0);
c *= m_nco.nextIQ();
consumed = false;
if(m_interpolator.interpolate(&m_distance, c, &consumed, &ci)) {
Complex d = ci * conj(m_lastSample);
m_lastSample = ci;
//Complex demod(atan2(d.imag(), d.real()) * 0.5, 0);
Real demod = atan2(d.imag(), d.real()) / M_PI;
consumed2 = false;
c = Complex(demod, 0);
while(!consumed2) {
if(m_interpolator2.interpolate(&m_distance2, c, &consumed2, &ci)) {
m_buffer[m_bufferFill++] = Sample(ci.real() * 32767.0, 0.0);
AudioSample s;
s.l = ci.real() * 32767.0;
s.r = s.l;
m_audioFifo.write((quint8*)&s, 4, 1);
if(m_bufferFill >= m_buffer.size()) {
m_spectrum.feed(m_buffer.begin(), m_buffer.end());
m_bufferFill = 0;
}
m_distance2 += 4.0 * m_resampler;
}
}
m_distance += 500000 / 176400.0;
}
}
return count;
#endif
}
#endif