VE7UWU/sdrangel
1
0
Fork 0
mirror of https://github.com/f4exb/sdrangel.git synced 2024-11-26 01:39:05 -05:00
Code Issues Projects Releases Wiki Activity

DV serial: use HP filter before interpolation and LP filter. Set HP -3dB corner at 300 Hz (for 8 kHz sampling rate)

Browse Source
This commit is contained in:
f4exb 2018-06-24 02:29:54 +02:00
parent 39bb65a198
commit 175e4ca98a
4 changed files with 52 additions and 44 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

74
sdrbase/dsp/dvserialworker.cpp
View File

@ -27,13 +27,16 @@ DVSerialWorker::DVSerialWorker() :
m_running(false),
m_currentGainIn(0),
m_currentGainOut(0),
m_upsamplerLastValue(0),
m_phase(0)
m_upsamplerLastValue(0.0f),
m_phase(0),
m_upsampling(1),
m_volume(1.0f)
{
m_audioBuffer.resize(48000);
m_audioBufferFill = 0;
m_audioFifo = 0;
memset(m_dvAudioSamples, 0, SerialDV::MBE_AUDIO_BLOCK_SIZE*sizeof(short));
setVolumeFactors();
}
DVSerialWorker::~DVSerialWorker()
@ -81,14 +84,21 @@ void DVSerialWorker::handleInputMessages()
{
MsgMbeDecode *decodeMsg = (MsgMbeDecode *) message;
int dBVolume = (decodeMsg->getVolumeIndex() - 30) / 2;
m_upsampleFilter.useHP(decodeMsg->getUseHP());
if (m_dvController.decode(m_dvAudioSamples, decodeMsg->getMbeFrame(), decodeMsg->getMbeRate(), dBVolume))
{
float volume = pow(10.0, dBVolume / 10.0f);
int upsampling = decodeMsg->getUpsampling();
upsampling = upsampling > 6 ? 6 : upsampling < 1 ? 1 : upsampling;
if ((volume != m_volume) || (upsampling != m_upsampling))
{
m_volume = volume;
m_upsampling = upsampling;
setVolumeFactors();
}
m_upsampleFilter.useHP(decodeMsg->getUseHP());
if (m_dvController.decode(m_dvAudioSamples, decodeMsg->getMbeFrame(), decodeMsg->getMbeRate()))
{
if (upsampling > 1) {
upsample(upsampling, m_dvAudioSamples, SerialDV::MBE_AUDIO_BLOCK_SIZE, decodeMsg->getChannels());
} else {
@ -145,45 +155,17 @@ bool DVSerialWorker::hasFifo(AudioFifo *audioFifo)
return m_audioFifo == audioFifo;
}
void DVSerialWorker::upsample6(short *in, int nbSamplesIn, unsigned char channels)
{
for (int i = 0; i < nbSamplesIn; i++)
{
int cur = (int) in[i];
int prev = (int) m_upsamplerLastValue;
qint16 upsample;
for (int j = 1; j < 7; j++)
{
upsample = m_upsampleFilter.run((qint16) ((cur*j + prev*(6-j)) / 6));
m_audioBuffer[m_audioBufferFill].l = channels & 1 ? upsample : 0;
m_audioBuffer[m_audioBufferFill].r = (channels>>1) & 1 ? upsample : 0;
if (m_audioBufferFill < m_audioBuffer.size() - 1)
{
++m_audioBufferFill;
}
else
{
qDebug("DVSerialWorker::upsample6: audio buffer is full check its size");
}
}
m_upsamplerLastValue = in[i];
}
}
void DVSerialWorker::upsample(int upsampling, short *in, int nbSamplesIn, unsigned char channels)
{
for (int i = 0; i < nbSamplesIn; i++)
{
int cur = (int) in[i];
int prev = (int) m_upsamplerLastValue;
float cur = m_upsampleFilter.usesHP() ? m_upsampleFilter.runHP((float) in[i]) : (float) in[i];
float prev = m_upsamplerLastValue;
qint16 upsample;
for (int j = 1; j <= upsampling; j++)
{
upsample = m_upsampleFilter.run((qint16) ((cur*j + prev*(upsampling-j)) / upsampling));
upsample = (qint16) m_upsampleFilter.runLP(cur*m_upsamplingFactors[j] + prev*m_upsamplingFactors[upsampling-j]);
m_audioBuffer[m_audioBufferFill].l = channels & 1 ? upsample : 0;
m_audioBuffer[m_audioBufferFill].r = (channels>>1) & 1 ? upsample : 0;
@ -197,7 +179,7 @@ void DVSerialWorker::upsample(int upsampling, short *in, int nbSamplesIn, unsign
}
}
m_upsamplerLastValue = in[i];
m_upsamplerLastValue = cur;
}
}
@ -205,8 +187,9 @@ void DVSerialWorker::noUpsample(short *in, int nbSamplesIn, unsigned char channe
{
for (int i = 0; i < nbSamplesIn; i++)
{
m_audioBuffer[m_audioBufferFill].l = channels & 1 ? in[i] : 0;
m_audioBuffer[m_audioBufferFill].r = (channels>>1) & 1 ? in[i] : 0;
float cur = m_upsampleFilter.usesHP() ? m_upsampleFilter.runHP((float) in[i]) : (float) in[i];
m_audioBuffer[m_audioBufferFill].l = channels & 1 ? cur*m_upsamplingFactors[0] : 0;
m_audioBuffer[m_audioBufferFill].r = (channels>>1) & 1 ? cur*m_upsamplingFactors[0] : 0;
if (m_audioBufferFill < m_audioBuffer.size() - 1)
{
@ -219,6 +202,15 @@ void DVSerialWorker::noUpsample(short *in, int nbSamplesIn, unsigned char channe
}
}
void DVSerialWorker::setVolumeFactors()
{
m_upsamplingFactors[0] = m_volume;
for (int i = 1; i <= m_upsampling; i++) {
m_upsamplingFactors[i] = (i*m_volume) / (float) m_upsampling;
}
}
//void DVSerialWorker::upsample6(short *in, short *out, int nbSamplesIn)
//{
// for (int i = 0; i < nbSamplesIn; i++)

7
sdrbase/dsp/dvserialworker.h
View File

@ -135,9 +135,9 @@ public slots:
private:
//void upsample6(short *in, short *out, int nbSamplesIn);
void upsample6(short *in, int nbSamplesIn, unsigned char channels);
void upsample(int upsampling, short *in, int nbSamplesIn, unsigned char channels);
void noUpsample(short *in, int nbSamplesIn, unsigned char channels);
void setVolumeFactors();
SerialDV::DVController m_dvController;
volatile bool m_running;
@ -147,9 +147,12 @@ private:
//short m_audioSamples[SerialDV::MBE_AUDIO_BLOCK_SIZE * 6 * 2]; // upsample to 48k and duplicate channel
AudioVector m_audioBuffer;
uint m_audioBufferFill;
short m_upsamplerLastValue;
float m_upsamplerLastValue;
float m_phase;
MBEAudioInterpolatorFilter m_upsampleFilter;
int m_upsampling;
float m_volume;
float m_upsamplingFactors[7];
};
#endif /* SDRBASE_DSP_DVSERIALWORKER_H_ */

14
sdrbase/dsp/filtermbe.cpp
View File

@ -20,8 +20,8 @@
const float MBEAudioInterpolatorFilter::m_lpa[3] = {1.0, 1.392667E+00, -5.474446E-01};
const float MBEAudioInterpolatorFilter::m_lpb[3] = {3.869430E-02, 7.738860E-02, 3.869430E-02};
const float MBEAudioInterpolatorFilter::m_hpa[3] = {1.000000e+00, 1.955578e+00, -9.565437e-01};
const float MBEAudioInterpolatorFilter::m_hpb[3] = {9.780305e-01, -1.956061e+00, 9.780305e-01};
const float MBEAudioInterpolatorFilter::m_hpa[3] = {1.000000e+00, 1.667871e+00, -7.156964e-01};
const float MBEAudioInterpolatorFilter::m_hpb[3] = {8.459039e-01, -1.691760e+00, 8.459039e-01};
MBEAudioInterpolatorFilter::MBEAudioInterpolatorFilter() :
m_filterLP(m_lpa, m_lpb),
@ -37,3 +37,13 @@ float MBEAudioInterpolatorFilter::run(const float& sample)
{
return m_useHP ? m_filterLP.run(m_filterHP.run(sample)) : m_filterLP.run(sample);
}
float MBEAudioInterpolatorFilter::runHP(const float& sample)
{
return m_filterHP.run(sample);
}
float MBEAudioInterpolatorFilter::runLP(const float& sample)
{
return m_filterLP.run(sample);
}

3
sdrbase/dsp/filtermbe.h
View File

@ -67,7 +67,10 @@ public:
~MBEAudioInterpolatorFilter();
void useHP(bool useHP) { m_useHP = useHP; }
bool usesHP() const { return m_useHP; }
float run(const float& sample);
float runHP(const float& sample);
float runLP(const float& sample);
private:
IIRFilter<float, 2> m_filterLP;