mirror of https://github.com/f4exb/sdrangel.git
SSB demod: implemented interpolator for audio
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@ -157,124 +157,133 @@ void SSBDemod::configure(MessageQueue* messageQueue,
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void SSBDemod::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end, bool positiveOnly)
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{
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(void) positiveOnly;
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Complex ci;
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fftfilt::cmplx *sideband;
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int n_out;
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Complex ci;
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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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if (m_interpolatorDistance < 1.0f) // interpolate
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{
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while (!m_interpolator.interpolate(&m_interpolatorDistanceRemain, c, &ci))
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{
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processOneSample(ci);
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m_interpolatorDistanceRemain += m_interpolatorDistance;
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}
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}
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else
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{
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if (m_interpolator.decimate(&m_interpolatorDistanceRemain, c, &ci))
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{
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processOneSample(ci);
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m_interpolatorDistanceRemain += m_interpolatorDistance;
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}
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}
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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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m_settingsMutex.unlock();
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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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void SSBDemod::processOneSample(Complex &ci)
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{
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fftfilt::cmplx *sideband;
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int n_out = 0;
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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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m_sum += sideband[i];
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if (m_dsb) {
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n_out = DSBFilter->runDSB(ci, &sideband);
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} else {
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n_out = SSBFilter->runSSB(ci, &sideband, m_usb);
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}
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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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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_magsqSum += m_magsq;
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m_sum += sideband[i];
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if (m_magsq > m_magsqPeak)
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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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float agcVal = m_agcActive ? m_agc.feedAndGetValue(sideband[i]) : 10.0; // 10.0 for 3276.8, 1.0 for 327.68
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fftfilt::cmplx& delayedSample = m_squelchDelayLine.readBack(m_agc.getStepDownDelay());
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m_audioActive = delayedSample.real() != 0.0;
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m_squelchDelayLine.write(sideband[i]*agcVal);
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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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}
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else
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{
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fftfilt::cmplx z = delayedSample * m_agc.getStepValue();
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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_magsqPeak = m_magsq;
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m_audioBuffer[m_audioBufferFill].r = (qint16)(z.imag() * m_volume);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(z.real() * m_volume);
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}
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else
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{
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m_audioBuffer[m_audioBufferFill].r = (qint16)(z.real() * m_volume);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(z.imag() * m_volume);
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}
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}
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else
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{
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Real demod = (z.real() + z.imag()) * 0.7;
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qint16 sample = (qint16)(demod * m_volume);
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m_audioBuffer[m_audioBufferFill].l = sample;
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m_audioBuffer[m_audioBufferFill].r = sample;
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}
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}
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m_magsqCount++;
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++m_audioBufferFill;
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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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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);
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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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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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float agcVal = m_agcActive ? m_agc.feedAndGetValue(sideband[i]) : 10.0; // 10.0 for 3276.8, 1.0 for 327.68
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fftfilt::cmplx& delayedSample = m_squelchDelayLine.readBack(m_agc.getStepDownDelay());
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m_audioActive = delayedSample.real() != 0.0;
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m_squelchDelayLine.write(sideband[i]*agcVal);
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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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}
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else
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{
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fftfilt::cmplx z = delayedSample * m_agc.getStepValue();
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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)(z.imag() * m_volume);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(z.real() * m_volume);
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}
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else
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{
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m_audioBuffer[m_audioBufferFill].r = (qint16)(z.real() * m_volume);
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m_audioBuffer[m_audioBufferFill].l = (qint16)(z.imag() * m_volume);
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}
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}
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else
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{
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Real demod = (z.real() + z.imag()) * 0.7;
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qint16 sample = (qint16)(demod * m_volume);
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m_audioBuffer[m_audioBufferFill].l = sample;
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m_audioBuffer[m_audioBufferFill].r = 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);
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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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m_audioBufferFill = 0;
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}
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}
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uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill);
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@ -292,7 +301,6 @@ void SSBDemod::feed(const SampleVector::const_iterator& begin, const SampleVecto
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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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@ -338,6 +338,8 @@ private:
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void webapiFormatChannelReport(SWGSDRangel::SWGChannelReport& response);
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void webapiReverseSendSettings(QList<QString>& channelSettingsKeys, const SSBDemodSettings& settings, bool force);
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void processOneSample(Complex &ci);
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private slots:
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void networkManagerFinished(QNetworkReply *reply);
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};
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