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Multiple audio support: SSB modulator

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This commit is contained in:
f4exb
2018-03-29 16:57:42 +02:00
parent eb57c1aca6
commit 0d7f73f595
7 changed files with 154 additions and 34 deletions
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plugins/channeltx/modssb/ssbmod.cpp
+117 -30
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@@ -70,8 +70,11 @@ SSBMod::SSBMod(DeviceSinkAPI *deviceAPI) :
{
setObjectName(m_channelId);
m_SSBFilter = new fftfilt(m_settings.m_lowCutoff / m_settings.m_audioSampleRate, m_settings.m_bandwidth / m_settings.m_audioSampleRate, m_ssbFftLen);
m_DSBFilter = new fftfilt((2.0f * m_settings.m_bandwidth) / m_settings.m_audioSampleRate, 2 * m_ssbFftLen);
DSPEngine::instance()->getAudioDeviceManager()->addAudioSource(&m_audioFifo, getInputMessageQueue());
m_audioSampleRate = DSPEngine::instance()->getAudioDeviceManager()->getInputSampleRate();
m_SSBFilter = new fftfilt(m_settings.m_lowCutoff / m_audioSampleRate, m_settings.m_bandwidth / m_audioSampleRate, m_ssbFftLen);
m_DSBFilter = new fftfilt((2.0f * m_settings.m_bandwidth) / m_audioSampleRate, 2 * m_ssbFftLen);
m_SSBFilterBuffer = new Complex[m_ssbFftLen>>1]; // filter returns data exactly half of its size
m_DSBFilterBuffer = new Complex[m_ssbFftLen];
memset(m_SSBFilterBuffer, 0, sizeof(Complex)*(m_ssbFftLen>>1));
@@ -87,11 +90,10 @@ SSBMod::SSBMod(DeviceSinkAPI *deviceAPI) :
m_magsq = 0.0;
m_toneNco.setFreq(1000.0, m_settings.m_audioSampleRate);
DSPEngine::instance()->getAudioDeviceManager()->addAudioSource(&m_audioFifo, getInputMessageQueue());
m_toneNco.setFreq(1000.0, m_audioSampleRate);
// CW keyer
m_cwKeyer.setSampleRate(m_settings.m_audioSampleRate);
m_cwKeyer.setSampleRate(48000);
m_cwKeyer.setWPM(13);
m_cwKeyer.setMode(CWKeyerSettings::CWNone);
@@ -175,7 +177,7 @@ void SSBMod::pull(Sample& sample)
void SSBMod::pullAudio(int nbSamples)
{
unsigned int nbSamplesAudio = nbSamples * ((Real) m_settings.m_audioSampleRate / (Real) m_basebandSampleRate);
unsigned int nbSamplesAudio = nbSamples * ((Real) m_audioSampleRate / (Real) m_basebandSampleRate);
if (nbSamplesAudio > m_audioBuffer.size())
{
@@ -580,9 +582,26 @@ bool SSBMod::handleMessage(const Message& cmd)
samplesCount = m_ifstream.tellg() / sizeof(Real);
}
MsgReportFileSourceStreamTiming *report;
report = MsgReportFileSourceStreamTiming::create(samplesCount);
getMessageQueueToGUI()->push(report);
if (getMessageQueueToGUI())
{
MsgReportFileSourceStreamTiming *report;
report = MsgReportFileSourceStreamTiming::create(samplesCount);
getMessageQueueToGUI()->push(report);
}
return true;
}
else if (DSPConfigureAudio::match(cmd))
{
DSPConfigureAudio& cfg = (DSPConfigureAudio&) cmd;
uint32_t sampleRate = cfg.getSampleRate();
qDebug() << "SSBMod::handleMessage: DSPConfigureAudio:"
<< " sampleRate: " << sampleRate;
if (sampleRate != m_audioSampleRate) {
applyAudioSampleRate(sampleRate);
}
return true;
}
@@ -613,9 +632,12 @@ void SSBMod::openFileStream()
<< " fileSize: " << m_fileSize << "bytes"
<< " length: " << m_recordLength << " seconds";
MsgReportFileSourceStreamData *report;
report = MsgReportFileSourceStreamData::create(m_sampleRate, m_recordLength);
getMessageQueueToGUI()->push(report);
if (getMessageQueueToGUI())
{
MsgReportFileSourceStreamData *report;
report = MsgReportFileSourceStreamData::create(m_sampleRate, m_recordLength);
getMessageQueueToGUI()->push(report);
}
}
void SSBMod::seekFileStream(int seekPercentage)
@@ -631,6 +653,68 @@ void SSBMod::seekFileStream(int seekPercentage)
}
}
void SSBMod::applyAudioSampleRate(int sampleRate)
{
qDebug("SSBMod::applyAudioSampleRate: %d", sampleRate);
MsgConfigureChannelizer* channelConfigMsg = MsgConfigureChannelizer::create(
sampleRate, m_settings.m_inputFrequencyOffset);
m_inputMessageQueue.push(channelConfigMsg);
m_settingsMutex.lock();
m_interpolatorDistanceRemain = 0;
m_interpolatorConsumed = false;
m_interpolatorDistance = (Real) sampleRate / (Real) m_outputSampleRate;
m_interpolator.create(48, sampleRate, m_settings.m_bandwidth, 3.0);
float band = m_settings.m_bandwidth;
float lowCutoff = m_settings.m_lowCutoff;
bool usb = m_settings.m_usb;
if (band < 0) // negative means LSB
{
band = -band; // turn to positive
lowCutoff = -lowCutoff;
usb = false; // and take note of side band
}
else
{
usb = true;
}
if (band < 100.0f) // at least 100 Hz
{
band = 100.0f;
lowCutoff = 0;
}
if (band - lowCutoff < 100.0f) {
lowCutoff = band - 100.0f;
}
m_SSBFilter->create_filter(lowCutoff / sampleRate, band / sampleRate);
m_DSBFilter->create_dsb_filter((2.0f * band) / sampleRate);
m_settings.m_bandwidth = band;
m_settings.m_lowCutoff = lowCutoff;
m_settings.m_usb = usb;
m_toneNco.setFreq(m_settings.m_toneFrequency, sampleRate);
m_cwKeyer.setSampleRate(sampleRate);
m_settingsMutex.unlock();
m_audioSampleRate = sampleRate;
if (getMessageQueueToGUI())
{
DSPConfigureAudio *cfg = new DSPConfigureAudio(m_audioSampleRate);
getMessageQueueToGUI()->push(cfg);
}
}
void SSBMod::applyChannelSettings(int basebandSampleRate, int outputSampleRate, int inputFrequencyOffset, bool force)
{
qDebug() << "SSBMod::applyChannelSettings:"
@@ -651,8 +735,8 @@ void SSBMod::applyChannelSettings(int basebandSampleRate, int outputSampleRate,
m_settingsMutex.lock();
m_interpolatorDistanceRemain = 0;
m_interpolatorConsumed = false;
m_interpolatorDistance = (Real) m_settings.m_audioSampleRate / (Real) outputSampleRate;
m_interpolator.create(48, m_settings.m_audioSampleRate, m_settings.m_bandwidth, 3.0);
m_interpolatorDistance = (Real) m_audioSampleRate / (Real) outputSampleRate;
m_interpolator.create(48, m_audioSampleRate, m_settings.m_bandwidth, 3.0);
m_settingsMutex.unlock();
}
@@ -668,8 +752,7 @@ void SSBMod::applySettings(const SSBModSettings& settings, bool force)
bool usb = settings.m_usb;
if ((settings.m_bandwidth != m_settings.m_bandwidth) ||
(settings.m_lowCutoff != m_settings.m_lowCutoff) ||
(settings.m_audioSampleRate != m_settings.m_audioSampleRate) || force)
(settings.m_lowCutoff != m_settings.m_lowCutoff) || force)
{
if (band < 0) // negative means LSB
{
@@ -695,25 +778,17 @@ void SSBMod::applySettings(const SSBModSettings& settings, bool force)
m_settingsMutex.lock();
m_interpolatorDistanceRemain = 0;
m_interpolatorConsumed = false;
m_interpolatorDistance = (Real) settings.m_audioSampleRate / (Real) m_outputSampleRate;
m_interpolator.create(48, settings.m_audioSampleRate, band, 3.0);
m_SSBFilter->create_filter(lowCutoff / settings.m_audioSampleRate, band / settings.m_audioSampleRate);
m_DSBFilter->create_dsb_filter((2.0f * band) / settings.m_audioSampleRate);
m_interpolatorDistance = (Real) m_audioSampleRate / (Real) m_outputSampleRate;
m_interpolator.create(48, m_audioSampleRate, band, 3.0);
m_SSBFilter->create_filter(lowCutoff / m_audioSampleRate, band / m_audioSampleRate);
m_DSBFilter->create_dsb_filter((2.0f * band) / m_audioSampleRate);
m_settingsMutex.unlock();
}
if ((settings.m_toneFrequency != m_settings.m_toneFrequency) ||
(settings.m_audioSampleRate != m_settings.m_audioSampleRate) || force)
if ((settings.m_toneFrequency != m_settings.m_toneFrequency) || force)
{
m_settingsMutex.lock();
m_toneNco.setFreq(settings.m_toneFrequency, settings.m_audioSampleRate);
m_settingsMutex.unlock();
}
if ((settings.m_audioSampleRate != m_settings.m_audioSampleRate) || force)
{
m_settingsMutex.lock();
m_cwKeyer.setSampleRate(settings.m_audioSampleRate);
m_toneNco.setFreq(settings.m_toneFrequency, m_audioSampleRate);
m_settingsMutex.unlock();
}
@@ -759,6 +834,18 @@ void SSBMod::applySettings(const SSBModSettings& settings, bool force)
m_inAGC.setStepDownDelay(settings.m_agcThresholdDelay);
}
if ((settings.m_audioDeviceName != m_settings.m_audioDeviceName) || force)
{
AudioDeviceManager *audioDeviceManager = DSPEngine::instance()->getAudioDeviceManager();
int audioDeviceIndex = audioDeviceManager->getInputDeviceIndex(settings.m_audioDeviceName);
audioDeviceManager->addAudioSource(&m_audioFifo, getInputMessageQueue(), audioDeviceIndex);
uint32_t audioSampleRate = audioDeviceManager->getInputSampleRate(audioDeviceIndex);
if (m_audioSampleRate != audioSampleRate) {
applyAudioSampleRate(audioSampleRate);
}
}
m_settings = settings;
m_settings.m_bandwidth = band;
m_settings.m_lowCutoff = lowCutoff;