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FreeDV mod: reworked input audio. Implements #495

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This commit is contained in:
f4exb
2020-11-22 18:47:01 +01:00
parent f19431ac5c
commit 49cceb40cc
2 changed files with 70 additions and 13 deletions
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plugins/channeltx/modfreedv/freedvmodsource.cpp
+58 -11
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@@ -35,7 +35,7 @@ FreeDVModSource::FreeDVModSource() :
m_SSBFilterBuffer(0),
m_SSBFilterBufferIndex(0),
m_audioSampleRate(48000),
m_audioFifo(4800),
m_audioFifo(12000),
m_levelCalcCount(0),
m_peakLevel(0.0f),
m_levelSum(0.0f),
@@ -46,14 +46,17 @@ FreeDVModSource::FreeDVModSource() :
m_iModem(0),
m_speechIn(nullptr),
m_modOut(0),
m_scaleFactor(SDR_TX_SCALEF)
m_scaleFactor(SDR_TX_SCALEF),
m_mutex(QMutex::Recursive)
{
m_SSBFilter = new fftfilt(m_lowCutoff / m_audioSampleRate, m_hiCutoff / m_audioSampleRate, m_ssbFftLen);
m_SSBFilterBuffer = new Complex[m_ssbFftLen>>1]; // filter returns data exactly half of its size
std::fill(m_SSBFilterBuffer, m_SSBFilterBuffer+(m_ssbFftLen>>1), Complex{0,0});
m_audioBuffer.resize(1<<14);
m_audioBuffer.resize(24000);
m_audioBufferFill = 0;
m_audioReadBuffer.resize(24000);
m_audioReadBufferFill = 0;
m_sum.real(0.0f);
m_sum.imag(0.0f);
@@ -131,15 +134,40 @@ void FreeDVModSource::prefetch(unsigned int nbSamples)
void FreeDVModSource::pullAudio(unsigned int nbSamples)
{
QMutexLocker mlock(&m_mutex);
unsigned int nbSamplesAudio = nbSamples * ((Real) m_audioSampleRate / (Real) m_modemSampleRate);
if (nbSamplesAudio > m_audioBuffer.size())
{
if (nbSamplesAudio > m_audioBuffer.size()) {
m_audioBuffer.resize(nbSamplesAudio);
}
m_audioFifo.read(reinterpret_cast<quint8*>(&m_audioBuffer[0]), nbSamplesAudio);
std::copy(&m_audioReadBuffer[0], &m_audioReadBuffer[nbSamplesAudio], &m_audioBuffer[0]);
m_audioBufferFill = 0;
if (m_audioReadBufferFill > nbSamplesAudio) // copy back remaining samples at the start of the read buffer
{
std::copy(&m_audioReadBuffer[nbSamplesAudio], &m_audioReadBuffer[m_audioReadBufferFill], &m_audioReadBuffer[0]);
m_audioReadBufferFill = m_audioReadBufferFill - nbSamplesAudio; // adjust current read buffer fill pointer
}
}
qint16 FreeDVModSource::getAudioSample()
{
qint16 sample;
if (m_audioBufferFill < m_audioBuffer.size())
{
sample = (m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) * (m_settings.m_volumeFactor / 2.0f);
m_audioBufferFill++;
}
else
{
unsigned int size = m_audioBuffer.size();
qDebug("FreeDVModSource::getAudioSample: starve audio samples: size: %u", size);
sample = (m_audioBuffer[size-1].l + m_audioBuffer[size-1].r) * (m_settings.m_volumeFactor / 2.0f);
}
return sample;
}
void FreeDVModSource::modulateSample()
@@ -148,7 +176,6 @@ void FreeDVModSource::modulateSample()
if (!m_settings.m_gaugeInputElseModem) {
calculateLevel(m_modSample);
}
m_audioBufferFill++;
}
void FreeDVModSource::pullAF(Complex& sample)
@@ -231,13 +258,11 @@ void FreeDVModSource::pullAF(Complex& sample)
case FreeDVModSettings::FreeDVModInputAudio:
for (int i = 0; i < m_nSpeechSamples; i++)
{
qint16 audioSample = (m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) * (m_settings.m_volumeFactor / 2.0f);
m_audioBufferFill++;
qint16 audioSample = getAudioSample();
while (!m_audioResampler.downSample(audioSample, m_speechIn[i]))
{
audioSample = (m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) * (m_settings.m_volumeFactor / 2.0f);
m_audioBufferFill++;
audioSample = getAudioSample();
}
if (m_settings.m_gaugeInputElseModem) {
@@ -518,5 +543,27 @@ void FreeDVModSource::applySettings(const FreeDVModSettings& settings, bool forc
m_toneNco.setFreq(settings.m_toneFrequency, m_channelSampleRate);
}
if ((settings.m_modAFInput != m_settings.m_modAFInput) || force)
{
if (settings.m_modAFInput == FreeDVModSettings::FreeDVModInputAudio) {
connect(&m_audioFifo, SIGNAL(dataReady()), this, SLOT(handleAudio()));
} else {
disconnect(&m_audioFifo, SIGNAL(dataReady()), this, SLOT(handleAudio()));
}
}
m_settings = settings;
}
void FreeDVModSource::handleAudio()
{
QMutexLocker mlock(&m_mutex);
unsigned int nbRead;
while ((nbRead = m_audioFifo.read(reinterpret_cast<quint8*>(&m_audioReadBuffer[m_audioReadBufferFill]), 4096)) != 0)
{
if (m_audioReadBufferFill + nbRead + 4096 < m_audioReadBuffer.size()) {
m_audioReadBufferFill += nbRead;
}
}
}