sdrangel/plugins/channelrx/demodam/amdemod.h

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8.2 KiB
C++

///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2015 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 //
// //
// 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/>. //
///////////////////////////////////////////////////////////////////////////////////
#ifndef INCLUDE_AMDEMOD_H
#define INCLUDE_AMDEMOD_H
#include <QMutex>
#include <vector>
#include "dsp/basebandsamplesink.h"
#include "channel/channelsinkapi.h"
#include "dsp/nco.h"
#include "dsp/interpolator.h"
#include "util/movingaverage.h"
#include "dsp/agc.h"
#include "dsp/bandpass.h"
#include "audio/audiofifo.h"
#include "util/message.h"
#include "amdemodsettings.h"
class DeviceSourceAPI;
class DownChannelizer;
class ThreadedBasebandSampleSink;
class AMDemod : public BasebandSampleSink, public ChannelSinkAPI {
Q_OBJECT
public:
class MsgConfigureAMDemod : public Message {
MESSAGE_CLASS_DECLARATION
public:
const AMDemodSettings& getSettings() const { return m_settings; }
bool getForce() const { return m_force; }
static MsgConfigureAMDemod* create(const AMDemodSettings& settings, bool force)
{
return new MsgConfigureAMDemod(settings, force);
}
private:
AMDemodSettings m_settings;
bool m_force;
MsgConfigureAMDemod(const AMDemodSettings& settings, bool force) :
Message(),
m_settings(settings),
m_force(force)
{ }
};
class MsgConfigureChannelizer : public Message {
MESSAGE_CLASS_DECLARATION
public:
int getSampleRate() const { return m_sampleRate; }
int getCenterFrequency() const { return m_centerFrequency; }
static MsgConfigureChannelizer* create(int sampleRate, int centerFrequency)
{
return new MsgConfigureChannelizer(sampleRate, centerFrequency);
}
private:
int m_sampleRate;
int m_centerFrequency;
MsgConfigureChannelizer(int sampleRate, int centerFrequency) :
Message(),
m_sampleRate(sampleRate),
m_centerFrequency(centerFrequency)
{ }
};
AMDemod(DeviceSourceAPI *deviceAPI);
~AMDemod();
virtual void destroy() { delete this; }
virtual void feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end, bool po);
virtual void start();
virtual void stop();
virtual bool handleMessage(const Message& cmd);
virtual void getIdentifier(QString& id) { id = objectName(); }
virtual void getTitle(QString& title) { title = m_settings.m_title; }
virtual qint64 getCenterFrequency() const { return m_settings.m_inputFrequencyOffset; }
virtual QByteArray serialize() const;
virtual bool deserialize(const QByteArray& data);
virtual int webapiSettingsGet(
SWGSDRangel::SWGChannelSettings& response,
QString& errorMessage);
virtual int webapiSettingsPutPatch(
bool force,
const QStringList& channelSettingsKeys,
SWGSDRangel::SWGChannelSettings& response,
QString& errorMessage);
virtual int webapiReportGet(
SWGSDRangel::SWGChannelReport& response,
QString& errorMessage);
uint32_t getAudioSampleRate() const { return m_audioSampleRate; }
double getMagSq() const { return m_magsq; }
bool getSquelchOpen() const { return m_squelchOpen; }
void getMagSqLevels(double& avg, double& peak, int& nbSamples)
{
avg = m_magsqCount == 0 ? 1e-10 : m_magsqSum / m_magsqCount;
peak = m_magsqPeak == 0.0 ? 1e-10 : m_magsqPeak;
nbSamples = m_magsqCount == 0 ? 1 : m_magsqCount;
m_magsqSum = 0.0f;
m_magsqPeak = 0.0f;
m_magsqCount = 0;
}
static const QString m_channelIdURI;
static const QString m_channelId;
private:
enum RateState {
RSInitialFill,
RSRunning
};
DeviceSourceAPI *m_deviceAPI;
ThreadedBasebandSampleSink* m_threadedChannelizer;
DownChannelizer* m_channelizer;
int m_inputSampleRate;
int m_inputFrequencyOffset;
AMDemodSettings m_settings;
uint32_t m_audioSampleRate;
bool m_running;
NCO m_nco;
Interpolator m_interpolator;
Real m_interpolatorDistance;
Real m_interpolatorDistanceRemain;
Real m_squelchLevel;
uint32_t m_squelchCount;
bool m_squelchOpen;
double m_magsq;
double m_magsqSum;
double m_magsqPeak;
int m_magsqCount;
MovingAverageUtil<Real, double, 16> m_movingAverage;
SimpleAGC<4096> m_volumeAGC;
Bandpass<Real> m_bandpass;
AudioVector m_audioBuffer;
uint32_t m_audioBufferFill;
AudioFifo m_audioFifo;
static const int m_udpBlockSize;
QMutex m_settingsMutex;
void applyChannelSettings(int inputSampleRate, int inputFrequencyOffset, bool force = false);
void applySettings(const AMDemodSettings& settings, bool force = false);
void applyAudioSampleRate(int sampleRate);
void webapiFormatChannelSettings(SWGSDRangel::SWGChannelSettings& response, const AMDemodSettings& settings);
void webapiFormatChannelReport(SWGSDRangel::SWGChannelReport& response);
void processOneSample(Complex &ci)
{
Real re = ci.real() / SDR_RX_SCALED;
Real im = ci.imag() / SDR_RX_SCALED;
Real magsq = re*re + im*im;
m_movingAverage(magsq);
m_magsq = m_movingAverage.asDouble();
m_magsqSum += magsq;
if (magsq > m_magsqPeak)
{
m_magsqPeak = magsq;
}
m_magsqCount++;
if (m_magsq >= m_squelchLevel)
{
if (m_squelchCount <= m_audioSampleRate / 10)
{
m_squelchCount++;
}
}
else
{
if (m_squelchCount > 1)
{
m_squelchCount -= 2;
}
}
qint16 sample;
if ((m_squelchCount >= m_audioSampleRate / 20) && !m_settings.m_audioMute)
{
Real demod = sqrt(magsq);
m_volumeAGC.feed(demod);
demod = (demod - m_volumeAGC.getValue()) / m_volumeAGC.getValue();
if (m_settings.m_bandpassEnable)
{
demod = m_bandpass.filter(demod);
demod /= 301.0f;
}
Real attack = (m_squelchCount - 0.05f * m_audioSampleRate) / (0.05f * m_audioSampleRate);
sample = demod * attack * (m_audioSampleRate/24) * m_settings.m_volume;
m_squelchOpen = true;
}
else
{
sample = 0;
m_squelchOpen = false;
}
m_audioBuffer[m_audioBufferFill].l = sample;
m_audioBuffer[m_audioBufferFill].r = sample;
++m_audioBufferFill;
if (m_audioBufferFill >= m_audioBuffer.size())
{
uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 10);
if (res != m_audioBufferFill)
{
qDebug("AMDemod::processOneSample: %u/%u audio samples written", res, m_audioBufferFill);
m_audioFifo.clear();
}
m_audioBufferFill = 0;
}
}
};
#endif // INCLUDE_AMDEMOD_H