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https://github.com/f4exb/sdrangel.git
synced 2024-11-22 16:08:39 -05:00
DV serial: use HP filter before interpolation and LP filter. Set HP -3dB corner at 300 Hz (for 8 kHz sampling rate)
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39bb65a198
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175e4ca98a
@ -27,13 +27,16 @@ DVSerialWorker::DVSerialWorker() :
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m_running(false),
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m_currentGainIn(0),
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m_currentGainOut(0),
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m_upsamplerLastValue(0),
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m_phase(0)
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m_upsamplerLastValue(0.0f),
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m_phase(0),
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m_upsampling(1),
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m_volume(1.0f)
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{
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m_audioBuffer.resize(48000);
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m_audioBufferFill = 0;
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m_audioFifo = 0;
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memset(m_dvAudioSamples, 0, SerialDV::MBE_AUDIO_BLOCK_SIZE*sizeof(short));
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setVolumeFactors();
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}
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DVSerialWorker::~DVSerialWorker()
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@ -81,14 +84,21 @@ void DVSerialWorker::handleInputMessages()
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{
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MsgMbeDecode *decodeMsg = (MsgMbeDecode *) message;
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int dBVolume = (decodeMsg->getVolumeIndex() - 30) / 2;
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m_upsampleFilter.useHP(decodeMsg->getUseHP());
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if (m_dvController.decode(m_dvAudioSamples, decodeMsg->getMbeFrame(), decodeMsg->getMbeRate(), dBVolume))
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{
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float volume = pow(10.0, dBVolume / 10.0f);
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int upsampling = decodeMsg->getUpsampling();
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upsampling = upsampling > 6 ? 6 : upsampling < 1 ? 1 : upsampling;
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if ((volume != m_volume) || (upsampling != m_upsampling))
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{
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m_volume = volume;
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m_upsampling = upsampling;
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setVolumeFactors();
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}
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m_upsampleFilter.useHP(decodeMsg->getUseHP());
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if (m_dvController.decode(m_dvAudioSamples, decodeMsg->getMbeFrame(), decodeMsg->getMbeRate()))
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{
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if (upsampling > 1) {
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upsample(upsampling, m_dvAudioSamples, SerialDV::MBE_AUDIO_BLOCK_SIZE, decodeMsg->getChannels());
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} else {
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@ -145,45 +155,17 @@ bool DVSerialWorker::hasFifo(AudioFifo *audioFifo)
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return m_audioFifo == audioFifo;
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}
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void DVSerialWorker::upsample6(short *in, int nbSamplesIn, unsigned char channels)
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{
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for (int i = 0; i < nbSamplesIn; i++)
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{
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int cur = (int) in[i];
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int prev = (int) m_upsamplerLastValue;
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qint16 upsample;
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for (int j = 1; j < 7; j++)
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{
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upsample = m_upsampleFilter.run((qint16) ((cur*j + prev*(6-j)) / 6));
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m_audioBuffer[m_audioBufferFill].l = channels & 1 ? upsample : 0;
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m_audioBuffer[m_audioBufferFill].r = (channels>>1) & 1 ? upsample : 0;
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if (m_audioBufferFill < m_audioBuffer.size() - 1)
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{
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++m_audioBufferFill;
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}
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else
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{
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qDebug("DVSerialWorker::upsample6: audio buffer is full check its size");
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}
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}
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m_upsamplerLastValue = in[i];
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}
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}
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void DVSerialWorker::upsample(int upsampling, short *in, int nbSamplesIn, unsigned char channels)
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{
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for (int i = 0; i < nbSamplesIn; i++)
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{
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int cur = (int) in[i];
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int prev = (int) m_upsamplerLastValue;
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float cur = m_upsampleFilter.usesHP() ? m_upsampleFilter.runHP((float) in[i]) : (float) in[i];
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float prev = m_upsamplerLastValue;
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qint16 upsample;
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for (int j = 1; j <= upsampling; j++)
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{
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upsample = m_upsampleFilter.run((qint16) ((cur*j + prev*(upsampling-j)) / upsampling));
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upsample = (qint16) m_upsampleFilter.runLP(cur*m_upsamplingFactors[j] + prev*m_upsamplingFactors[upsampling-j]);
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m_audioBuffer[m_audioBufferFill].l = channels & 1 ? upsample : 0;
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m_audioBuffer[m_audioBufferFill].r = (channels>>1) & 1 ? upsample : 0;
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@ -197,7 +179,7 @@ void DVSerialWorker::upsample(int upsampling, short *in, int nbSamplesIn, unsign
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}
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}
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m_upsamplerLastValue = in[i];
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m_upsamplerLastValue = cur;
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}
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}
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@ -205,8 +187,9 @@ void DVSerialWorker::noUpsample(short *in, int nbSamplesIn, unsigned char channe
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{
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for (int i = 0; i < nbSamplesIn; i++)
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{
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m_audioBuffer[m_audioBufferFill].l = channels & 1 ? in[i] : 0;
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m_audioBuffer[m_audioBufferFill].r = (channels>>1) & 1 ? in[i] : 0;
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float cur = m_upsampleFilter.usesHP() ? m_upsampleFilter.runHP((float) in[i]) : (float) in[i];
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m_audioBuffer[m_audioBufferFill].l = channels & 1 ? cur*m_upsamplingFactors[0] : 0;
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m_audioBuffer[m_audioBufferFill].r = (channels>>1) & 1 ? cur*m_upsamplingFactors[0] : 0;
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if (m_audioBufferFill < m_audioBuffer.size() - 1)
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{
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@ -219,6 +202,15 @@ void DVSerialWorker::noUpsample(short *in, int nbSamplesIn, unsigned char channe
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}
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}
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void DVSerialWorker::setVolumeFactors()
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{
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m_upsamplingFactors[0] = m_volume;
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for (int i = 1; i <= m_upsampling; i++) {
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m_upsamplingFactors[i] = (i*m_volume) / (float) m_upsampling;
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}
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}
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//void DVSerialWorker::upsample6(short *in, short *out, int nbSamplesIn)
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//{
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// for (int i = 0; i < nbSamplesIn; i++)
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@ -135,9 +135,9 @@ public slots:
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private:
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//void upsample6(short *in, short *out, int nbSamplesIn);
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void upsample6(short *in, int nbSamplesIn, unsigned char channels);
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void upsample(int upsampling, short *in, int nbSamplesIn, unsigned char channels);
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void noUpsample(short *in, int nbSamplesIn, unsigned char channels);
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void setVolumeFactors();
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SerialDV::DVController m_dvController;
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volatile bool m_running;
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@ -147,9 +147,12 @@ private:
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//short m_audioSamples[SerialDV::MBE_AUDIO_BLOCK_SIZE * 6 * 2]; // upsample to 48k and duplicate channel
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AudioVector m_audioBuffer;
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uint m_audioBufferFill;
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short m_upsamplerLastValue;
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float m_upsamplerLastValue;
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float m_phase;
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MBEAudioInterpolatorFilter m_upsampleFilter;
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int m_upsampling;
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float m_volume;
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float m_upsamplingFactors[7];
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};
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#endif /* SDRBASE_DSP_DVSERIALWORKER_H_ */
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@ -20,8 +20,8 @@
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const float MBEAudioInterpolatorFilter::m_lpa[3] = {1.0, 1.392667E+00, -5.474446E-01};
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const float MBEAudioInterpolatorFilter::m_lpb[3] = {3.869430E-02, 7.738860E-02, 3.869430E-02};
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const float MBEAudioInterpolatorFilter::m_hpa[3] = {1.000000e+00, 1.955578e+00, -9.565437e-01};
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const float MBEAudioInterpolatorFilter::m_hpb[3] = {9.780305e-01, -1.956061e+00, 9.780305e-01};
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const float MBEAudioInterpolatorFilter::m_hpa[3] = {1.000000e+00, 1.667871e+00, -7.156964e-01};
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const float MBEAudioInterpolatorFilter::m_hpb[3] = {8.459039e-01, -1.691760e+00, 8.459039e-01};
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MBEAudioInterpolatorFilter::MBEAudioInterpolatorFilter() :
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m_filterLP(m_lpa, m_lpb),
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@ -37,3 +37,13 @@ float MBEAudioInterpolatorFilter::run(const float& sample)
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{
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return m_useHP ? m_filterLP.run(m_filterHP.run(sample)) : m_filterLP.run(sample);
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}
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float MBEAudioInterpolatorFilter::runHP(const float& sample)
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{
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return m_filterHP.run(sample);
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}
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float MBEAudioInterpolatorFilter::runLP(const float& sample)
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{
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return m_filterLP.run(sample);
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}
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@ -67,7 +67,10 @@ public:
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~MBEAudioInterpolatorFilter();
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void useHP(bool useHP) { m_useHP = useHP; }
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bool usesHP() const { return m_useHP; }
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float run(const float& sample);
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float runHP(const float& sample);
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float runLP(const float& sample);
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private:
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IIRFilter<float, 2> m_filterLP;
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