mirror of
https://github.com/f4exb/sdrangel.git
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329 lines
9.7 KiB
C++
329 lines
9.7 KiB
C++
#include "dsp/channelizer.h"
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#include "dsp/inthalfbandfilter.h"
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#include "dsp/dspcommands.h"
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Channelizer::Channelizer(SampleSink* sampleSink) :
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m_sampleSink(sampleSink),
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m_inputSampleRate(100000),
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m_requestedOutputSampleRate(100000),
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m_requestedCenterFrequency(0),
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m_currentOutputSampleRate(100000),
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m_currentCenterFrequency(0)
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{
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}
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Channelizer::~Channelizer()
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{
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freeFilterChain();
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}
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void Channelizer::configure(MessageQueue* messageQueue, int sampleRate, int centerFrequency)
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{
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Message* cmd = DSPConfigureChannelizer::create(sampleRate, centerFrequency);
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cmd->submit(messageQueue, this);
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}
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void Channelizer::feed(SampleVector::const_iterator begin, SampleVector::const_iterator end, bool firstOfBurst)
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{
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for(SampleVector::const_iterator sample = begin; sample != end; ++sample) {
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Sample s(*sample);
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FilterStages::iterator stage = m_filterStages.begin();
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while(stage != m_filterStages.end()) {
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if(!(*stage)->work(&s))
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break;
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++stage;
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}
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if(stage == m_filterStages.end())
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m_sampleBuffer.push_back(s);
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}
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if(m_sampleSink != NULL)
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m_sampleSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), firstOfBurst);
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m_sampleBuffer.clear();
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}
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void Channelizer::start()
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{
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if(m_sampleSink != NULL)
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m_sampleSink->start();
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}
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void Channelizer::stop()
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{
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if(m_sampleSink != NULL)
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m_sampleSink->stop();
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}
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bool Channelizer::handleMessage(Message* cmd)
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{
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if(DSPSignalNotification::match(cmd)) {
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DSPSignalNotification* signal = (DSPSignalNotification*)cmd;
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m_inputSampleRate = signal->getSampleRate();
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applyConfiguration();
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cmd->completed();
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if(m_sampleSink != NULL) {
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signal = DSPSignalNotification::create(m_currentOutputSampleRate, m_currentCenterFrequency);
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if(!m_sampleSink->handleMessage(signal))
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signal->completed();
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}
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return true;
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} else if(DSPConfigureChannelizer::match(cmd)) {
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DSPConfigureChannelizer* chan = (DSPConfigureChannelizer*)cmd;
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m_requestedOutputSampleRate = chan->getSampleRate();
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m_requestedCenterFrequency = chan->getCenterFrequency();
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applyConfiguration();
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cmd->completed();
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if(m_sampleSink != NULL) {
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DSPSignalNotification* signal = DSPSignalNotification::create(m_currentOutputSampleRate, m_currentCenterFrequency);
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if(!m_sampleSink->handleMessage(signal))
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signal->completed();
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}
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return true;
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} else {
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if(m_sampleSink != NULL)
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return m_sampleSink->handleMessage(cmd);
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else return false;
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}
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}
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void Channelizer::applyConfiguration()
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{
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freeFilterChain();
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m_currentCenterFrequency = createFilterChain(
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m_inputSampleRate / -2, m_inputSampleRate / 2,
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m_requestedCenterFrequency - m_requestedOutputSampleRate / 2, m_requestedCenterFrequency + m_requestedOutputSampleRate / 2);
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m_currentOutputSampleRate = m_inputSampleRate / (1 << m_filterStages.size());
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}
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Channelizer::FilterStage::FilterStage(Mode mode) :
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m_filter(new IntHalfbandFilter),
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m_workFunction(NULL)
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{
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switch(mode) {
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case ModeCenter:
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m_workFunction = &IntHalfbandFilter::workDecimateCenter;
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break;
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case ModeLowerHalf:
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m_workFunction = &IntHalfbandFilter::workDecimateLowerHalf;
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break;
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case ModeUpperHalf:
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m_workFunction = &IntHalfbandFilter::workDecimateUpperHalf;
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break;
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}
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}
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Channelizer::FilterStage::~FilterStage()
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{
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delete m_filter;
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}
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bool Channelizer::signalContainsChannel(Real sigStart, Real sigEnd, Real chanStart, Real chanEnd) const
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{
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//qDebug(" testing signal [%f, %f], channel [%f, %f]", sigStart, sigEnd, chanStart, chanEnd);
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if(sigEnd <= sigStart)
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return false;
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if(chanEnd <= chanStart)
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return false;
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return (sigStart <= chanStart) && (sigEnd >= chanEnd);
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}
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Real Channelizer::createFilterChain(Real sigStart, Real sigEnd, Real chanStart, Real chanEnd)
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{
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Real sigBw = sigEnd - sigStart;
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Real safetyMargin = sigBw / 20;
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Real rot = sigBw / 4;
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safetyMargin = 0;
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//qDebug("Signal [%f, %f] (BW %f), Channel [%f, %f], Rot %f, Safety %f", sigStart, sigEnd, sigBw, chanStart, chanEnd, rot, safetyMargin);
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#if 1
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// check if it fits into the left half
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if(signalContainsChannel(sigStart + safetyMargin, sigStart + sigBw / 2.0 - safetyMargin, chanStart, chanEnd)) {
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//qDebug("-> take left half (rotate by +1/4 and decimate by 2)");
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m_filterStages.push_back(new FilterStage(FilterStage::ModeLowerHalf));
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return createFilterChain(sigStart, sigStart + sigBw / 2.0, chanStart, chanEnd);
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}
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// check if it fits into the right half
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if(signalContainsChannel(sigEnd - sigBw / 2.0f + safetyMargin, sigEnd - safetyMargin, chanStart, chanEnd)) {
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//qDebug("-> take right half (rotate by -1/4 and decimate by 2)");
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m_filterStages.push_back(new FilterStage(FilterStage::ModeUpperHalf));
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return createFilterChain(sigEnd - sigBw / 2.0f, sigEnd, chanStart, chanEnd);
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}
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// check if it fits into the center
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if(signalContainsChannel(sigStart + rot + safetyMargin, sigStart + rot + sigBw / 2.0f - safetyMargin, chanStart, chanEnd)) {
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//qDebug("-> take center half (decimate by 2)");
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m_filterStages.push_back(new FilterStage(FilterStage::ModeCenter));
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return createFilterChain(sigStart + rot, sigStart + sigBw / 2.0f + rot, chanStart, chanEnd);
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}
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#endif
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Real ofs = ((chanEnd - chanStart) / 2.0 + chanStart) - ((sigEnd - sigStart) / 2.0 + sigStart);
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qDebug("-> complete (final BW %f, frequency offset %f)", sigBw, ofs);
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return ofs;
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}
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void Channelizer::freeFilterChain()
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{
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for(FilterStages::iterator it = m_filterStages.begin(); it != m_filterStages.end(); ++it)
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delete *it;
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m_filterStages.clear();
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}
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#if 0
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///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany //
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// written by Christian Daniel //
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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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// //
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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 <QTime>
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#include <stdio.h>
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#include "channelizer.h"
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#include "hardware/audiooutput.h"
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Channelizer::Channelizer()
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{
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#if 0
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m_spectrum.configure(128 , 25, FFTWindow::Bartlett);
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m_buffer.resize(2048);
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m_bufferFill = 0;
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m_nco.setFreq(-100000, 500000);
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m_interpolator.create(51, 32, 32 * 500000, 150000 / 2);
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m_distance = 500000.0 / 176400.0;
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m_interpolator2.create(19, 8, 8 * 176400, 15000 / 2);
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m_distance2 = 4;
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m_audioFifo.setSize(4, 44100 / 2 * 4);
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m_audioOutput = new AudioOutput;
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m_audioOutput->start(0, 44100, &m_audioFifo);
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m_resampler = 1.0;
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m_resamplerCtrl.setup(0.00001, 0, -0.00001);
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#endif
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}
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Channelizer::~Channelizer()
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{
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#if 0
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m_audioOutput->stop();
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delete m_audioOutput;
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#endif
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}
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#if 0
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void Channelizer::setGLSpectrum(GLSpectrum* glSpectrum)
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{
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m_spectrum.setGLSpectrum(glSpectrum);
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}
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#endif
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size_t Channelizer::workUnitSize()
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{
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#if 0
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return m_buffer.size();
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#endif
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return 0;
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}
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size_t Channelizer::work(SampleVector::const_iterator begin, SampleVector::const_iterator end)
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{
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#if 0
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int buffered = m_audioOutput->bufferedSamples();
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if(m_audioFifo.fill() < (m_audioFifo.size() / 6)) {
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while(m_audioFifo.fill() < (m_audioFifo.size() / 2)) {
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quint32 d = 0;
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m_audioFifo.write((quint8*)&d, 4);
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}
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qDebug("underflow - fill %d (vs %d)", m_audioFifo.fill(), m_audioFifo.size() / 4 / 2);
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}
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buffered = m_audioOutput->bufferedSamples();
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int fill = m_audioFifo.fill() / 4 + buffered;
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float err = (float)fill / ((m_audioFifo.size() / 4) / 2);
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float ctrl = m_resamplerCtrl.feed(err);
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//float resamplerRate = (ctrl / 1.0);
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float resamplerRate = err;
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if(resamplerRate < 0.9999)
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resamplerRate = 0.9999;
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else if(resamplerRate > 1.0001)
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resamplerRate = 1.0001;
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m_resampler = m_resampler * 0.99 + resamplerRate * 0.01;
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//m_resampler = resamplerRate;
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if(m_resampler < 0.995)
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m_resampler = 0.995;
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else if(m_resampler > 1.005)
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m_resampler = 1.005;
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//qDebug("%lld %5d %f %f %f", QDateTime::currentMSecsSinceEpoch(), fill, ctrl, m_resampler, err);
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struct AudioSample {
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qint16 l;
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qint16 r;
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};
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size_t count = end - begin;
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Complex ci;
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bool consumed;
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bool consumed2;
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for(SampleVector::const_iterator it = begin; it < end; it++) {
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Complex c(it->real() / 32768.0, it->imag() / 32768.0);
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c *= m_nco.nextIQ();
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consumed = false;
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if(m_interpolator.interpolate(&m_distance, c, &consumed, &ci)) {
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Complex d = ci * conj(m_lastSample);
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m_lastSample = ci;
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//Complex demod(atan2(d.imag(), d.real()) * 0.5, 0);
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Real demod = atan2(d.imag(), d.real()) / M_PI;
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consumed2 = false;
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c = Complex(demod, 0);
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while(!consumed2) {
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if(m_interpolator2.interpolate(&m_distance2, c, &consumed2, &ci)) {
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m_buffer[m_bufferFill++] = Sample(ci.real() * 32767.0, 0.0);
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AudioSample s;
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s.l = ci.real() * 32767.0;
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s.r = s.l;
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m_audioFifo.write((quint8*)&s, 4, 1);
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if(m_bufferFill >= m_buffer.size()) {
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m_spectrum.feed(m_buffer.begin(), m_buffer.end());
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m_bufferFill = 0;
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}
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m_distance2 += 4.0 * m_resampler;
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}
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}
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m_distance += 500000 / 176400.0;
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}
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}
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return count;
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#endif
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}
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#endif
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