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mirror of https://github.com/f4exb/sdrangel.git synced 2024-12-22 17:45:48 -05:00

AM Modulator: audio input (mic) basic support

This commit is contained in:
f4exb 2016-11-29 01:09:06 +01:00
parent 4c3db01409
commit c5f1acdcf1
20 changed files with 191 additions and 67 deletions

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@ -34,15 +34,14 @@ From version 2 SDRangel can integrate more than one hardware device running conc
Transmission or signal generation support for eligible devices (BladeRF and HackRF) is progressively introduced with the following roadmap:
- Phase 1: version 2.2.0: generation to file (File Sink) with AM modulator with simple sine modulation. Fixed sample rate of 48 kS/s (no effective interpolation)
- Phase 2: version 2.2.x: full baseband interpolation chain: in AM modulator and Up Channelizer.
- Phase 1: version 2.2.0: generation to file (File Sink) with AM modulator with simple sine modulation. Fixed sample rate of 48 kS/s (no effective interpolation)
- Phase 2: version 2.2.2: full baseband interpolation chain: in AM modulator and Up Channelizer.
- 2.3.0: SDRplay came into play ...
- Phase 3a: version 2.3.1: Improve AM modulator with audio file input
- Phase 3b: version 2.3.x: Improve AM modulator with audio input (Mic) support
- Phase 3: version 2.3.1: Improve AM modulator with audio file input and audio input (Mic) support
- Phase 4a: version 2.4.0: FM modulator
- Phase 4b: version 2.4.x: WFM modulator
- Phase 4c: version 2.4.x: SSB modulator
- phase 5: version 3.0.0: BladeRF and HackRF support including final interpolation stage.
- phase 5: version 3.0.0: BladeRF and HackRF support including final interpolation stage.
<h2>Airspy</h2>

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@ -35,6 +35,7 @@ MESSAGE_CLASS_DEFINITION(AMMod::MsgReportFileSourceStreamTiming, Message)
AMMod::AMMod() :
m_audioFifo(4, 48000),
m_settingsMutex(QMutex::Recursive),
m_fileSize(0),
m_recordLength(0),
@ -52,28 +53,31 @@ AMMod::AMMod() :
apply();
m_audioBuffer.resize(1<<14);
m_audioBufferFill = 0;
//m_audioBuffer.resize(1<<14);
//m_audioBufferFill = 0;
m_movingAverage.resize(16, 0);
m_volumeAGC.resize(4096, 0.003, 0);
m_magsq = 0.0;
m_toneNco.setFreq(1000.0, m_config.m_audioSampleRate);
DSPEngine::instance()->addAudioSource(&m_audioFifo);
}
AMMod::~AMMod()
{
DSPEngine::instance()->removeAudioSource(&m_audioFifo);
}
void AMMod::configure(MessageQueue* messageQueue,
Real rfBandwidth,
Real afBandwidth,
float modFactor,
int volumeTenths,
bool audioMute,
bool playLoop)
{
Message* cmd = MsgConfigureAMMod::create(rfBandwidth, afBandwidth, modFactor, audioMute, playLoop);
Message* cmd = MsgConfigureAMMod::create(rfBandwidth, afBandwidth, modFactor, volumeTenths, audioMute, playLoop);
messageQueue->push(cmd);
}
@ -124,6 +128,8 @@ void AMMod::pull(Sample& sample)
void AMMod::pullAF(Real& sample)
{
int16_t audioSample[2];
switch (m_afInput)
{
case AMModInputTone:
@ -158,7 +164,8 @@ void AMMod::pullAF(Real& sample)
}
break;
case AMModInputAudio:
sample = 0.0f; // TODO
m_audioFifo.read(reinterpret_cast<quint8*>(audioSample), 1, 10);
sample = ((audioSample[0] + audioSample[1]) * m_running.m_volumeFactor) / 6553600.0f;
break;
case AMModInputNone:
default:
@ -205,6 +212,7 @@ bool AMMod::handleMessage(const Message& cmd)
m_config.m_rfBandwidth = cfg.getRFBandwidth();
m_config.m_afBandwidth = cfg.getAFBandwidth();
m_config.m_modFactor = cfg.getModFactor();
m_config.m_volumeFactor = cfg.getVolumeFactor();
m_config.m_audioMute = cfg.getAudioMute();
m_config.m_playLoop = cfg.getPlayLoop();
@ -214,6 +222,7 @@ bool AMMod::handleMessage(const Message& cmd)
<< " m_rfBandwidth: " << m_config.m_rfBandwidth
<< " m_afBandwidth: " << m_config.m_afBandwidth
<< " m_modFactor: " << m_config.m_modFactor
<< " m_volumeFactor: " << m_config.m_volumeFactor
<< " m_audioMute: " << m_config.m_audioMute
<< " m_playLoop: " << m_config.m_playLoop;
@ -298,6 +307,7 @@ void AMMod::apply()
m_running.m_rfBandwidth = m_config.m_rfBandwidth;
m_running.m_afBandwidth = m_config.m_afBandwidth;
m_running.m_modFactor = m_config.m_modFactor;
m_running.m_volumeFactor = m_config.m_volumeFactor;
m_running.m_audioSampleRate = m_config.m_audioSampleRate;
m_running.m_audioMute = m_config.m_audioMute;
m_running.m_playLoop = m_config.m_playLoop;

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@ -174,7 +174,7 @@ public:
AMMod();
~AMMod();
void configure(MessageQueue* messageQueue, Real rfBandwidth, Real afBandwidth, float modFactor, bool audioMute, bool playLoop);
void configure(MessageQueue* messageQueue, Real rfBandwidth, Real afBandwidth, float modFactor, int volumeFactor, bool audioMute, bool playLoop);
virtual void pull(Sample& sample);
virtual void start();
@ -192,26 +192,29 @@ private:
Real getRFBandwidth() const { return m_rfBandwidth; }
Real getAFBandwidth() const { return m_afBandwidth; }
float getModFactor() const { return m_modFactor; }
int getVolumeFactor() const { return m_volumeFactor; }
bool getAudioMute() const { return m_audioMute; }
bool getPlayLoop() const { return m_playLoop; }
static MsgConfigureAMMod* create(Real rfBandwidth, Real afBandwidth, float modFactor, bool audioMute, bool playLoop)
static MsgConfigureAMMod* create(Real rfBandwidth, Real afBandwidth, float modFactor, int volumeFactor, bool audioMute, bool playLoop)
{
return new MsgConfigureAMMod(rfBandwidth, afBandwidth, modFactor, audioMute, playLoop);
return new MsgConfigureAMMod(rfBandwidth, afBandwidth, modFactor, volumeFactor, audioMute, playLoop);
}
private:
Real m_rfBandwidth;
Real m_afBandwidth;
float m_modFactor;
int m_volumeFactor;
bool m_audioMute;
bool m_playLoop;
MsgConfigureAMMod(Real rfBandwidth, Real afBandwidth, float modFactor, bool audioMute, bool playLoop) :
MsgConfigureAMMod(Real rfBandwidth, Real afBandwidth, float modFactor, int volumeFactor, bool audioMute, bool playLoop) :
Message(),
m_rfBandwidth(rfBandwidth),
m_afBandwidth(afBandwidth),
m_modFactor(modFactor),
m_volumeFactor(volumeFactor),
m_audioMute(audioMute),
m_playLoop(playLoop)
{ }
@ -236,6 +239,7 @@ private:
Real m_rfBandwidth;
Real m_afBandwidth;
float m_modFactor;
int m_volumeFactor;
quint32 m_audioSampleRate;
bool m_audioMute;
bool m_playLoop;
@ -246,6 +250,7 @@ private:
m_rfBandwidth(-1),
m_afBandwidth(-1),
m_modFactor(0.2f),
m_volumeFactor(20),
m_audioSampleRate(0),
m_audioMute(false),
m_playLoop(false)
@ -270,8 +275,8 @@ private:
MovingAverage<Real> m_movingAverage;
SimpleAGC m_volumeAGC;
AudioVector m_audioBuffer;
uint m_audioBufferFill;
//AudioVector m_audioBuffer;
//uint m_audioBufferFill;
AudioFifo m_audioFifo;
SampleVector m_sampleBuffer;

View File

@ -77,6 +77,7 @@ void AMModGUI::resetToDefaults()
ui->rfBW->setValue(6);
ui->afBW->setValue(3);
ui->modPercent->setValue(20);
ui->micVolume->setValue(50);
ui->deltaFrequency->setValue(0);
blockApplySettings(false);
@ -219,6 +220,12 @@ void AMModGUI::on_modPercent_valueChanged(int value)
applySettings();
}
void AMModGUI::on_micVolume_valueChanged(int value)
{
ui->micVolumeText->setText(QString("%1").arg(value));
applySettings();
}
void AMModGUI::on_audioMute_toggled(bool checked)
{
applySettings();
@ -391,6 +398,7 @@ void AMModGUI::applySettings()
m_rfBW[ui->rfBW->value()],
ui->afBW->value() * 1000.0,
ui->modPercent->value() / 100.0f,
ui->micVolume->value(),
ui->audioMute->isChecked(),
ui->playLoop->isChecked());
}

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@ -63,6 +63,7 @@ private slots:
void on_rfBW_valueChanged(int value);
void on_afBW_valueChanged(int value);
void on_modPercent_valueChanged(int value);
void on_micVolume_valueChanged(int value);
void on_audioMute_toggled(bool checked);
void on_tone_toggled(bool checked);
void on_mic_toggled(bool checked);

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@ -6,8 +6,8 @@
<rect>
<x>0</x>
<y>0</y>
<width>261</width>
<height>180</height>
<width>266</width>
<height>190</height>
</rect>
</property>
<property name="font">
@ -27,7 +27,7 @@
<rect>
<x>10</x>
<y>10</y>
<width>241</width>
<width>251</width>
<height>161</height>
</rect>
</property>
@ -38,16 +38,7 @@
<property name="spacing">
<number>3</number>
</property>
<property name="leftMargin">
<number>2</number>
</property>
<property name="topMargin">
<number>2</number>
</property>
<property name="rightMargin">
<number>2</number>
</property>
<property name="bottomMargin">
<property name="margin">
<number>2</number>
</property>
<item>
@ -276,6 +267,18 @@
</item>
<item>
<widget class="QSlider" name="modPercent">
<property name="sizePolicy">
<sizepolicy hsizetype="Expanding" vsizetype="Minimum">
<horstretch>0</horstretch>
<verstretch>0</verstretch>
</sizepolicy>
</property>
<property name="minimumSize">
<size>
<width>0</width>
<height>0</height>
</size>
</property>
<property name="toolTip">
<string>Modulation percentage</string>
</property>
@ -297,7 +300,7 @@
<widget class="QLabel" name="modPercentText">
<property name="minimumSize">
<size>
<width>50</width>
<width>30</width>
<height>0</height>
</size>
</property>
@ -309,6 +312,57 @@
</property>
</widget>
</item>
<item>
<widget class="Line" name="line">
<property name="orientation">
<enum>Qt::Vertical</enum>
</property>
</widget>
</item>
<item>
<widget class="QDial" name="micVolume">
<property name="maximumSize">
<size>
<width>24</width>
<height>24</height>
</size>
</property>
<property name="toolTip">
<string>Audio input volume</string>
</property>
<property name="maximum">
<number>100</number>
</property>
<property name="pageStep">
<number>1</number>
</property>
<property name="value">
<number>50</number>
</property>
</widget>
</item>
<item>
<widget class="QLabel" name="micVolumeText">
<property name="minimumSize">
<size>
<width>30</width>
<height>0</height>
</size>
</property>
<property name="toolTip">
<string>Audio input volume level</string>
</property>
<property name="statusTip">
<string/>
</property>
<property name="text">
<string>50</string>
</property>
<property name="alignment">
<set>Qt::AlignRight|Qt::AlignTrailing|Qt::AlignVCenter</set>
</property>
</widget>
</item>
</layout>
</item>
<item>

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@ -266,13 +266,14 @@ void FileSinkGui::on_startStop_toggled(bool checked)
{
qDebug("FileSinkGui::on_startStop_toggled: device start failed");
}
// DSPEngine::instance()->startAudio(); // TODO: activate when audio input is available
DSPEngine::instance()->startAudioInput();
}
}
else
{
m_deviceAPI->stopGeneration();
// DSPEngine::instance()->stopAudio(); // TODO: activate when audio input is available
DSPEngine::instance()->stopAudioInput();
}
}

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@ -351,13 +351,13 @@ void AirspyGui::on_startStop_toggled(bool checked)
if (m_deviceAPI->initAcquisition())
{
m_deviceAPI->startAcquisition();
DSPEngine::instance()->startAudio();
DSPEngine::instance()->startAudioOutput();
}
}
else
{
m_deviceAPI->stopAcquisition();
DSPEngine::instance()->stopAudio();
DSPEngine::instance()->stopAudioOutput();
}
}

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@ -358,13 +358,13 @@ void BladerfGui::on_startStop_toggled(bool checked)
if (m_deviceAPI->initAcquisition())
{
m_deviceAPI->startAcquisition();
DSPEngine::instance()->startAudio();
DSPEngine::instance()->startAudioOutput();
}
}
else
{
m_deviceAPI->stopAcquisition();
DSPEngine::instance()->stopAudio();
DSPEngine::instance()->stopAudioOutput();
}
}

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@ -428,13 +428,13 @@ void FCDProGui::on_startStop_toggled(bool checked)
if (m_deviceAPI->initAcquisition())
{
m_deviceAPI->startAcquisition();
DSPEngine::instance()->startAudio();
DSPEngine::instance()->startAudioOutput();
}
}
else
{
m_deviceAPI->stopAcquisition();
DSPEngine::instance()->stopAudio();
DSPEngine::instance()->stopAudioOutput();
}
}

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@ -292,13 +292,13 @@ void FCDProPlusGui::on_startStop_toggled(bool checked)
if (m_deviceAPI->initAcquisition())
{
m_deviceAPI->startAcquisition();
DSPEngine::instance()->startAudio();
DSPEngine::instance()->startAudioOutput();
}
}
else
{
m_deviceAPI->stopAcquisition();
DSPEngine::instance()->stopAudio();
DSPEngine::instance()->stopAudioOutput();
}
}

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@ -221,13 +221,13 @@ void FileSourceGui::on_startStop_toggled(bool checked)
if (m_deviceAPI->initAcquisition())
{
m_deviceAPI->startAcquisition();
DSPEngine::instance()->startAudio();
DSPEngine::instance()->startAudioOutput();
}
}
else
{
m_deviceAPI->stopAcquisition();
DSPEngine::instance()->stopAudio();
DSPEngine::instance()->stopAudioOutput();
}
}

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@ -334,13 +334,13 @@ void HackRFGui::on_startStop_toggled(bool checked)
if (m_deviceAPI->initAcquisition())
{
m_deviceAPI->startAcquisition();
DSPEngine::instance()->startAudio();
DSPEngine::instance()->startAudioOutput();
}
}
else
{
m_deviceAPI->stopAcquisition();
DSPEngine::instance()->stopAudio();
DSPEngine::instance()->stopAudioOutput();
}
}

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@ -318,13 +318,13 @@ void RTLSDRGui::on_startStop_toggled(bool checked)
if (m_deviceAPI->initAcquisition())
{
m_deviceAPI->startAcquisition();
DSPEngine::instance()->startAudio();
DSPEngine::instance()->startAudioOutput();
}
}
else
{
m_deviceAPI->stopAcquisition();
DSPEngine::instance()->stopAudio();
DSPEngine::instance()->stopAudioOutput();
}
}

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@ -589,13 +589,13 @@ void SDRdaemonGui::on_startStop_toggled(bool checked)
if (m_deviceAPI->initAcquisition())
{
m_deviceAPI->startAcquisition();
DSPEngine::instance()->startAudio();
DSPEngine::instance()->startAudioOutput();
}
}
else
{
m_deviceAPI->stopAcquisition();
DSPEngine::instance()->stopAudio();
DSPEngine::instance()->stopAudioOutput();
}
}

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@ -547,13 +547,13 @@ void SDRdaemonFECGui::on_startStop_toggled(bool checked)
if (m_deviceAPI->initAcquisition())
{
m_deviceAPI->startAcquisition();
DSPEngine::instance()->startAudio();
DSPEngine::instance()->startAudioOutput();
}
}
else
{
m_deviceAPI->stopAcquisition();
DSPEngine::instance()->stopAudio();
DSPEngine::instance()->stopAudioOutput();
}
}

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@ -453,13 +453,13 @@ void SDRPlayGui::on_startStop_toggled(bool checked)
if (m_deviceAPI->initAcquisition())
{
m_deviceAPI->startAcquisition();
DSPEngine::instance()->startAudio();
DSPEngine::instance()->startAudioOutput();
}
}
else
{
m_deviceAPI->stopAcquisition();
DSPEngine::instance()->stopAudio();
DSPEngine::instance()->stopAudioOutput();
}
}

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@ -96,7 +96,7 @@ bool AudioInput::start(int device, int rate)
m_audioInput = new QAudioInput(devInfo, m_audioFormat);
QIODevice::open(QIODevice::ReadOnly);
QIODevice::open(QIODevice::ReadWrite);
m_audioInput->start(this);

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@ -26,7 +26,8 @@
DSPEngine::DSPEngine() :
m_deviceSourceEnginesUIDSequence(0),
m_deviceSinkEnginesUIDSequence(0),
m_audioSampleRate(48000) // Use default output device at 48 kHz
m_audioOutputSampleRate(48000), // Use default output device at 48 kHz
m_audioInputSampleRate(48000) // Use default input device at 48 kHz
{
m_dvSerialSupport = false;
}
@ -86,28 +87,50 @@ void DSPEngine::removeLastDeviceSinkEngine()
}
}
void DSPEngine::startAudio()
void DSPEngine::startAudioOutput()
{
m_audioOutput.start(-1, m_audioSampleRate);
m_audioSampleRate = m_audioOutput.getRate(); // update with actual rate
m_audioOutput.start(-1, m_audioOutputSampleRate);
m_audioOutputSampleRate = m_audioOutput.getRate(); // update with actual rate
}
void DSPEngine::stopAudio()
void DSPEngine::stopAudioOutput()
{
m_audioOutput.stop();
}
void DSPEngine::startAudioImmediate()
void DSPEngine::startAudioOutputImmediate()
{
m_audioOutput.start(-1, m_audioSampleRate);
m_audioSampleRate = m_audioOutput.getRate(); // update with actual rate
m_audioOutput.start(-1, m_audioOutputSampleRate);
m_audioOutputSampleRate = m_audioOutput.getRate(); // update with actual rate
}
void DSPEngine::stopAudioImmediate()
void DSPEngine::stopAudioOutputImmediate()
{
m_audioOutput.stop();
}
void DSPEngine::startAudioInput()
{
m_audioInput.start(-1, m_audioInputSampleRate);
m_audioInputSampleRate = m_audioInput.getRate(); // update with actual rate
}
void DSPEngine::stopAudioInput()
{
m_audioInput.stop();
}
void DSPEngine::startAudioInputImmediate()
{
m_audioInput.start(-1, m_audioInputSampleRate);
m_audioInputSampleRate = m_audioInput.getRate(); // update with actual rate
}
void DSPEngine::stopAudioInputImmediate()
{
m_audioInput.stop();
}
void DSPEngine::addAudioSink(AudioFifo* audioFifo)
{
qDebug("DSPEngine::addAudioSink");
@ -120,6 +143,18 @@ void DSPEngine::removeAudioSink(AudioFifo* audioFifo)
m_audioOutput.removeFifo(audioFifo);
}
void DSPEngine::addAudioSource(AudioFifo* audioFifo)
{
qDebug("DSPEngine::addAudioSource");
m_audioInput.addFifo(audioFifo);
}
void DSPEngine::removeAudioSource(AudioFifo* audioFifo)
{
qDebug("DSPEngine::removeAudioSource");
m_audioInput.removeFifo(audioFifo);
}
DSPDeviceSourceEngine *DSPEngine::getDeviceSourceEngineByUID(uint uid)
{
std::vector<DSPDeviceSourceEngine*>::iterator it = m_deviceSourceEngines.begin();

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@ -21,6 +21,7 @@
#include <QObject>
#include <vector>
#include "audio/audiooutput.h"
#include "audio/audioinput.h"
#include "util/export.h"
#ifdef DSD_USE_SERIALDV
#include "dsp/dvserialengine.h"
@ -37,7 +38,7 @@ public:
static DSPEngine *instance();
uint getAudioSampleRate() const { return m_audioSampleRate; }
uint getAudioSampleRate() const { return m_audioOutputSampleRate; }
DSPDeviceSourceEngine *addDeviceSourceEngine();
void removeLastDeviceSourceEngine();
@ -45,10 +46,15 @@ public:
DSPDeviceSinkEngine *addDeviceSinkEngine();
void removeLastDeviceSinkEngine();
void startAudio();
void stopAudio();
void startAudioImmediate();
void stopAudioImmediate();
void startAudioOutput();
void stopAudioOutput();
void startAudioOutputImmediate();
void stopAudioOutputImmediate();
void startAudioInput();
void stopAudioInput();
void startAudioInputImmediate();
void stopAudioInputImmediate();
DSPDeviceSourceEngine *getDeviceSourceEngineByIndex(uint deviceIndex) { return m_deviceSourceEngines[deviceIndex]; }
DSPDeviceSourceEngine *getDeviceSourceEngineByUID(uint uid);
@ -59,6 +65,9 @@ public:
void addAudioSink(AudioFifo* audioFifo); //!< Add the audio sink
void removeAudioSink(AudioFifo* audioFifo); //!< Remove the audio sink
void addAudioSource(AudioFifo* audioFifo); //!< Add an audio source
void removeAudioSource(AudioFifo* audioFifo); //!< Remove an audio source
// Serial DV methods:
bool hasDVSerialSupport()
@ -92,7 +101,9 @@ private:
std::vector<DSPDeviceSinkEngine*> m_deviceSinkEngines;
uint m_deviceSinkEnginesUIDSequence;
AudioOutput m_audioOutput;
uint m_audioSampleRate;
AudioInput m_audioInput;
uint m_audioOutputSampleRate;
uint m_audioInputSampleRate;
bool m_dvSerialSupport;
#ifdef DSD_USE_SERIALDV
DVSerialEngine m_dvSerialEngine;