mirror of
https://github.com/f4exb/sdrangel.git
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548 lines
17 KiB
C++
548 lines
17 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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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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// (c) 2014 Modified by John Greb
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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 <QDebug>
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#include <stdio.h>
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#include "audio/audiooutput.h"
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#include "audio/audionetsink.h"
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#include "dsp/dspengine.h"
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#include "dsp/downchannelizer.h"
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#include "dsp/threadedbasebandsamplesink.h"
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#include "dsp/dspcommands.h"
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#include "device/devicesourceapi.h"
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#include "util/db.h"
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#include "ssbdemod.h"
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MESSAGE_CLASS_DEFINITION(SSBDemod::MsgConfigureSSBDemod, Message)
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MESSAGE_CLASS_DEFINITION(SSBDemod::MsgConfigureSSBDemodPrivate, Message)
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MESSAGE_CLASS_DEFINITION(SSBDemod::MsgConfigureChannelizer, Message)
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const QString SSBDemod::m_channelIdURI = "de.maintech.sdrangelove.channel.ssb";
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const QString SSBDemod::m_channelId = "SSBDemod";
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const int SSBDemod::m_udpBlockSize = 512;
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SSBDemod::SSBDemod(DeviceSourceAPI *deviceAPI) :
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ChannelSinkAPI(m_channelIdURI),
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m_deviceAPI(deviceAPI),
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m_audioBinaual(false),
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m_audioFlipChannels(false),
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m_dsb(false),
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m_audioMute(false),
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m_agc(12000, agcTarget, 1e-2),
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m_agcActive(false),
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m_agcClamping(false),
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m_agcNbSamples(12000),
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m_agcPowerThreshold(1e-2),
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m_agcThresholdGate(0),
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m_audioActive(false),
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m_sampleSink(0),
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m_audioFifo(24000),
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m_settingsMutex(QMutex::Recursive)
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{
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setObjectName(m_channelId);
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m_Bandwidth = 5000;
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m_LowCutoff = 300;
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m_volume = 2.0;
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m_spanLog2 = 3;
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m_inputSampleRate = 48000;
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m_inputFrequencyOffset = 0;
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m_audioSampleRate = DSPEngine::instance()->getAudioSampleRate();
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m_audioBuffer.resize(1<<14);
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m_audioBufferFill = 0;
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m_undersampleCount = 0;
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m_sum = 0;
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m_usb = true;
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m_magsq = 0.0f;
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m_magsqSum = 0.0f;
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m_magsqPeak = 0.0f;
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m_magsqCount = 0;
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m_agc.setClampMax(SDR_RX_SCALED*SDR_RX_SCALED);
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m_agc.setClamping(m_agcClamping);
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SSBFilter = new fftfilt(m_LowCutoff / m_audioSampleRate, m_Bandwidth / m_audioSampleRate, ssbFftLen);
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DSBFilter = new fftfilt((2.0f * m_Bandwidth) / m_audioSampleRate, 2 * ssbFftLen);
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DSPEngine::instance()->addAudioSink(&m_audioFifo);
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m_audioNetSink = new AudioNetSink(0); // parent thread allocated dynamically
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m_audioNetSink->setDestination(m_settings.m_udpAddress, m_settings.m_udpPort);
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m_channelizer = new DownChannelizer(this);
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m_threadedChannelizer = new ThreadedBasebandSampleSink(m_channelizer, this);
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m_deviceAPI->addThreadedSink(m_threadedChannelizer);
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m_deviceAPI->addChannelAPI(this);
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applyChannelSettings(m_inputSampleRate, m_inputFrequencyOffset, true);
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applySettings(m_settings, true);
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}
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SSBDemod::~SSBDemod()
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{
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if (SSBFilter) delete SSBFilter;
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if (DSBFilter) delete DSBFilter;
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DSPEngine::instance()->removeAudioSink(&m_audioFifo);
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delete m_audioNetSink;
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m_deviceAPI->removeChannelAPI(this);
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m_deviceAPI->removeThreadedSink(m_threadedChannelizer);
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delete m_threadedChannelizer;
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delete m_channelizer;
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}
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bool SSBDemod::isAudioNetSinkRTPCapable() const
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{
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return m_audioNetSink && m_audioNetSink->isRTPCapable();
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}
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void SSBDemod::configure(MessageQueue* messageQueue,
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Real Bandwidth,
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Real LowCutoff,
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Real volume,
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int spanLog2,
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bool audioBinaural,
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bool audioFlipChannel,
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bool dsb,
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bool audioMute,
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bool agc,
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bool agcClamping,
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int agcTimeLog2,
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int agcPowerThreshold,
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int agcThresholdGate)
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{
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Message* cmd = MsgConfigureSSBDemodPrivate::create(
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Bandwidth,
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LowCutoff,
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volume,
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spanLog2,
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audioBinaural,
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audioFlipChannel,
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dsb,
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audioMute,
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agc,
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agcClamping,
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agcTimeLog2,
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agcPowerThreshold,
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agcThresholdGate);
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messageQueue->push(cmd);
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}
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void SSBDemod::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end, bool positiveOnly __attribute__((unused)))
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{
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Complex ci;
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fftfilt::cmplx *sideband;
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int n_out;
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m_settingsMutex.lock();
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int decim = 1<<(m_spanLog2 - 1);
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unsigned char decim_mask = decim - 1; // counter LSB bit mask for decimation by 2^(m_scaleLog2 - 1)
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for(SampleVector::const_iterator it = begin; it < end; ++it)
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{
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Complex c(it->real(), it->imag());
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c *= m_nco.nextIQ();
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if(m_interpolator.decimate(&m_interpolatorDistanceRemain, c, &ci))
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{
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if (m_dsb)
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{
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n_out = DSBFilter->runDSB(ci, &sideband);
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}
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else
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{
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n_out = SSBFilter->runSSB(ci, &sideband, m_usb);
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}
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m_interpolatorDistanceRemain += m_interpolatorDistance;
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}
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else
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{
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n_out = 0;
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}
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for (int i = 0; i < n_out; i++)
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{
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// Downsample by 2^(m_scaleLog2 - 1) for SSB band spectrum display
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// smart decimation with bit gain using float arithmetic (23 bits significand)
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m_sum += sideband[i];
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if (!(m_undersampleCount++ & decim_mask))
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{
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Real avgr = m_sum.real() / decim;
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Real avgi = m_sum.imag() / decim;
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m_magsq = (avgr * avgr + avgi * avgi) / (SDR_RX_SCALED*SDR_RX_SCALED);
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m_magsqSum += m_magsq;
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if (m_magsq > m_magsqPeak)
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{
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m_magsqPeak = m_magsq;
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}
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m_magsqCount++;
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if (!m_dsb & !m_usb)
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{ // invert spectrum for LSB
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m_sampleBuffer.push_back(Sample(avgi, avgr));
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}
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else
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{
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m_sampleBuffer.push_back(Sample(avgr, avgi));
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}
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m_sum.real(0.0);
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m_sum.imag(0.0);
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}
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double agcVal = m_agcActive ? m_agc.feedAndGetValue(sideband[i]) : 10.0; // 10.0 for 3276.8, 1.0 for 327.68
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m_audioActive = agcVal != 0.0;
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if (m_audioMute)
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{
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m_audioBuffer[m_audioBufferFill].r = 0;
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m_audioBuffer[m_audioBufferFill].l = 0;
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if (m_settings.m_copyAudioToUDP) { m_audioNetSink->write(0); }
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}
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else
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{
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if (m_audioBinaual)
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{
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if (m_audioFlipChannels)
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{
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m_audioBuffer[m_audioBufferFill].r = (qint16)(sideband[i].imag() * m_volume * agcVal);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(sideband[i].real() * m_volume * agcVal);
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}
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else
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{
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m_audioBuffer[m_audioBufferFill].r = (qint16)(sideband[i].real() * m_volume * agcVal);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(sideband[i].imag() * m_volume * agcVal);
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}
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if (m_settings.m_copyAudioToUDP) { m_audioNetSink->write(m_audioBuffer[m_audioBufferFill].r + m_audioBuffer[m_audioBufferFill].l); }
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}
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else
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{
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Real demod = (sideband[i].real() + sideband[i].imag()) * 0.7;
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qint16 sample = (qint16)(demod * m_volume * agcVal);
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m_audioBuffer[m_audioBufferFill].l = sample;
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m_audioBuffer[m_audioBufferFill].r = sample;
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if (m_settings.m_copyAudioToUDP) { m_audioNetSink->write(sample); }
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}
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}
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++m_audioBufferFill;
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if (m_audioBufferFill >= m_audioBuffer.size())
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{
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uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10);
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if (res != m_audioBufferFill)
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{
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qDebug("SSBDemod::feed: %u/%u samples written", res, m_audioBufferFill);
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}
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m_audioBufferFill = 0;
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}
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}
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}
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uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10);
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if (res != m_audioBufferFill)
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{
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qDebug("SSBDemod::feed: %u/%u tail samples written", res, m_audioBufferFill);
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}
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m_audioBufferFill = 0;
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if (m_sampleSink != 0)
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{
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m_sampleSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), !m_dsb);
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}
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m_sampleBuffer.clear();
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m_settingsMutex.unlock();
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}
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void SSBDemod::start()
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{
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applyChannelSettings(m_inputSampleRate, m_inputFrequencyOffset, true);
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}
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void SSBDemod::stop()
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{
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}
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bool SSBDemod::handleMessage(const Message& cmd)
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{
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if (DownChannelizer::MsgChannelizerNotification::match(cmd))
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{
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DownChannelizer::MsgChannelizerNotification& notif = (DownChannelizer::MsgChannelizerNotification&) cmd;
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qDebug("SSBDemod::handleMessage: MsgChannelizerNotification: m_sampleRate");
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applyChannelSettings(notif.getSampleRate(), notif.getFrequencyOffset());
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return true;
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}
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else if (MsgConfigureChannelizer::match(cmd))
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{
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MsgConfigureChannelizer& cfg = (MsgConfigureChannelizer&) cmd;
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qDebug() << "SSBDemod::handleMessage: MsgConfigureChannelizer: sampleRate: " << cfg.getSampleRate()
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<< " centerFrequency: " << cfg.getCenterFrequency();
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m_channelizer->configure(m_channelizer->getInputMessageQueue(),
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cfg.getSampleRate(),
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cfg.getCenterFrequency());
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return true;
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}
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else if (MsgConfigureSSBDemod::match(cmd))
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{
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MsgConfigureSSBDemod& cfg = (MsgConfigureSSBDemod&) cmd;
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qDebug("SSBDemod::handleMessage: MsgConfigureSSBDemod");
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applySettings(cfg.getSettings(), cfg.getForce());
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return true;
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}
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else if (BasebandSampleSink::MsgThreadedSink::match(cmd))
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{
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BasebandSampleSink::MsgThreadedSink& cfg = (BasebandSampleSink::MsgThreadedSink&) cmd;
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const QThread *thread = cfg.getThread();
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qDebug("SSBDemod::handleMessage: BasebandSampleSink::MsgThreadedSink: %p", thread);
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m_audioNetSink->moveToThread(const_cast<QThread*>(thread)); // use the thread for udp sinks
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return true;
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}
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else if (DSPSignalNotification::match(cmd))
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{
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return true;
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}
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else
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{
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if(m_sampleSink != 0)
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{
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return m_sampleSink->handleMessage(cmd);
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}
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else
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{
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return false;
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}
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}
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}
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void SSBDemod::applyChannelSettings(int inputSampleRate, int inputFrequencyOffset, bool force)
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{
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qDebug() << "SSBDemod::applyChannelSettings:"
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<< " inputSampleRate: " << inputSampleRate
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<< " inputFrequencyOffset: " << inputFrequencyOffset;
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if ((m_inputFrequencyOffset != inputFrequencyOffset) ||
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(m_inputSampleRate != inputSampleRate) || force)
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{
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m_nco.setFreq(-inputFrequencyOffset, inputSampleRate);
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}
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if ((m_inputSampleRate != inputSampleRate) || force)
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{
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m_settingsMutex.lock();
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m_interpolator.create(16, inputSampleRate, m_Bandwidth * 1.5f, 2.0f);
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m_interpolatorDistanceRemain = 0;
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m_interpolatorDistance = (Real) inputSampleRate / (Real) m_settings.m_audioSampleRate;
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m_settingsMutex.unlock();
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}
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m_inputSampleRate = inputSampleRate;
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m_inputFrequencyOffset = inputFrequencyOffset;
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}
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void SSBDemod::applySettings(const SSBDemodSettings& settings, bool force)
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{
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qDebug() << "SSBDemod::applySettings:"
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<< " m_inputFrequencyOffset: " << settings.m_inputFrequencyOffset
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<< " m_rfBandwidth: " << settings.m_rfBandwidth
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<< " m_lowCutoff: " << settings.m_lowCutoff
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<< " m_volume: " << settings.m_volume
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<< " m_spanLog2: " << settings.m_spanLog2
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<< " m_audioBinaual: " << settings.m_audioBinaural
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<< " m_audioFlipChannels: " << settings.m_audioFlipChannels
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<< " m_dsb: " << settings.m_dsb
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<< " m_audioMute: " << settings.m_audioMute
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<< " m_copyAudioToUDP: " << settings.m_copyAudioToUDP
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<< " m_copyAudioUseRTP: " << settings.m_copyAudioUseRTP
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<< " m_agcActive: " << settings.m_agc
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<< " m_agcClamping: " << settings.m_agcClamping
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<< " m_agcTimeLog2: " << settings.m_agcTimeLog2
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<< " agcPowerThreshold: " << settings.m_agcPowerThreshold
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<< " agcThresholdGate: " << settings.m_agcThresholdGate;
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if((m_settings.m_rfBandwidth != settings.m_rfBandwidth) ||
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(m_settings.m_lowCutoff != settings.m_lowCutoff) ||
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(m_settings.m_audioSampleRate != settings.m_audioSampleRate) || force)
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{
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float band, lowCutoff;
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band = settings.m_rfBandwidth;
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lowCutoff = settings.m_lowCutoff;
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m_audioSampleRate = settings.m_audioSampleRate;
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if (band < 0) {
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band = -band;
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lowCutoff = -lowCutoff;
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m_usb = false;
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} else {
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m_usb = true;
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}
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if (band < 100.0f)
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{
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band = 100.0f;
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lowCutoff = 0;
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}
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m_Bandwidth = band;
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m_LowCutoff = lowCutoff;
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m_settingsMutex.lock();
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m_interpolator.create(16, m_inputSampleRate, m_Bandwidth * 1.5f, 2.0f);
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m_interpolatorDistanceRemain = 0;
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m_interpolatorDistance = (Real) m_inputSampleRate / (Real) m_settings.m_audioSampleRate;
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SSBFilter->create_filter(m_LowCutoff / (float) m_audioSampleRate, m_Bandwidth / (float) m_audioSampleRate);
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DSBFilter->create_dsb_filter((2.0f * m_Bandwidth) / (float) m_audioSampleRate);
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m_settingsMutex.unlock();
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}
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if ((m_settings.m_volume != settings.m_volume) || force)
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{
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m_volume = settings.m_volume;
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m_volume /= 4.0; // for 3276.8
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}
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if ((m_settings.m_agcTimeLog2 != settings.m_agcTimeLog2) ||
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(m_settings.m_agcPowerThreshold != settings.m_agcPowerThreshold) ||
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(m_settings.m_agcThresholdGate != settings.m_agcThresholdGate) ||
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(m_settings.m_agcClamping != settings.m_agcClamping) || force)
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{
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int agcNbSamples = 48 * (1<<settings.m_agcTimeLog2);
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m_agc.setThresholdEnable(settings.m_agcPowerThreshold != -SSBDemodSettings::m_minPowerThresholdDB);
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double agcPowerThreshold = CalcDb::powerFromdB(settings.m_agcPowerThreshold) * (SDR_RX_SCALED*SDR_RX_SCALED);
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int agcThresholdGate = 48 * settings.m_agcThresholdGate; // ms
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bool agcClamping = settings.m_agcClamping;
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if (m_agcNbSamples != agcNbSamples)
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{
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m_settingsMutex.lock();
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m_agc.resize(agcNbSamples, agcTarget);
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m_agc.setStepDownDelay(agcNbSamples);
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m_agcNbSamples = agcNbSamples;
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m_settingsMutex.unlock();
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}
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if (m_agcPowerThreshold != agcPowerThreshold)
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{
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m_agc.setThreshold(agcPowerThreshold);
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m_agcPowerThreshold = agcPowerThreshold;
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}
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if (m_agcThresholdGate != agcThresholdGate)
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{
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m_agc.setGate(agcThresholdGate);
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m_agcThresholdGate = agcThresholdGate;
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}
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if (m_agcClamping != agcClamping)
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{
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m_agc.setClamping(agcClamping);
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m_agcClamping = agcClamping;
|
|
}
|
|
|
|
qDebug() << "SBDemod::applySettings: AGC:"
|
|
<< " agcNbSamples: " << agcNbSamples
|
|
<< " agcPowerThreshold: " << agcPowerThreshold
|
|
<< " agcThresholdGate: " << agcThresholdGate
|
|
<< " agcClamping: " << agcClamping;
|
|
}
|
|
|
|
if ((m_settings.m_udpAddress != settings.m_udpAddress)
|
|
|| (m_settings.m_udpPort != settings.m_udpPort) || force)
|
|
{
|
|
m_audioNetSink->setDestination(settings.m_udpAddress, settings.m_udpPort);
|
|
}
|
|
|
|
if ((settings.m_copyAudioUseRTP != m_settings.m_copyAudioUseRTP) || force)
|
|
{
|
|
if (settings.m_copyAudioUseRTP)
|
|
{
|
|
if (m_audioNetSink->selectType(AudioNetSink::SinkRTP)) {
|
|
qDebug("NFMDemod::applySettings: set audio sink to RTP mode");
|
|
} else {
|
|
qWarning("NFMDemod::applySettings: RTP support for audio sink not available. Fall back too UDP");
|
|
}
|
|
}
|
|
else
|
|
{
|
|
if (m_audioNetSink->selectType(AudioNetSink::SinkUDP)) {
|
|
qDebug("NFMDemod::applySettings: set audio sink to UDP mode");
|
|
} else {
|
|
qWarning("NFMDemod::applySettings: failed to set audio sink to UDP mode");
|
|
}
|
|
}
|
|
}
|
|
|
|
m_spanLog2 = settings.m_spanLog2;
|
|
m_audioBinaual = settings.m_audioBinaural;
|
|
m_audioFlipChannels = settings.m_audioFlipChannels;
|
|
m_dsb = settings.m_dsb;
|
|
m_audioMute = settings.m_audioMute;
|
|
m_agcActive = settings.m_agc;
|
|
|
|
m_settings = settings;
|
|
}
|
|
|
|
QByteArray SSBDemod::serialize() const
|
|
{
|
|
return m_settings.serialize();
|
|
}
|
|
|
|
bool SSBDemod::deserialize(const QByteArray& data)
|
|
{
|
|
if (m_settings.deserialize(data))
|
|
{
|
|
MsgConfigureSSBDemod *msg = MsgConfigureSSBDemod::create(m_settings, true);
|
|
m_inputMessageQueue.push(msg);
|
|
return true;
|
|
}
|
|
else
|
|
{
|
|
m_settings.resetToDefaults();
|
|
MsgConfigureSSBDemod *msg = MsgConfigureSSBDemod::create(m_settings, true);
|
|
m_inputMessageQueue.push(msg);
|
|
return false;
|
|
}
|
|
}
|