diff --git a/CMakeLists.txt b/CMakeLists.txt
index 7ed5aacbf..cc40ba3a1 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -79,6 +79,7 @@ set(sdrbase_SOURCES
sdrbase/dsp/fftengine.cpp
sdrbase/dsp/fftfilt.cxx
sdrbase/dsp/fftwindow.cpp
+ sdrbase/dsp/filterrc.cpp
sdrbase/dsp/filesink.cpp
sdrbase/dsp/interpolator.cpp
sdrbase/dsp/inthalfbandfilter.cpp
@@ -155,6 +156,7 @@ set(sdrbase_HEADERS
include/dsp/fftfilt.h
include/dsp/fftwengine.h
include/dsp/fftwindow.h
+ include/dsp/filterrc.h
include/dsp/filesink.h
include/dsp/gfft.h
include/dsp/interpolator.h
diff --git a/include/dsp/filterrc.h b/include/dsp/filterrc.h
new file mode 100644
index 000000000..7f1af7c89
--- /dev/null
+++ b/include/dsp/filterrc.h
@@ -0,0 +1,50 @@
+///////////////////////////////////////////////////////////////////////////////////
+// Copyright (C) 2015 F4EXB //
+// written by Edouard Griffiths //
+// //
+// This program is free software; you can redistribute it and/or modify //
+// it under the terms of the GNU General Public License as published by //
+// the Free Software Foundation as version 3 of the License, or //
+// //
+// This program is distributed in the hope that it will be useful, //
+// but WITHOUT ANY WARRANTY; without even the implied warranty of //
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
+// GNU General Public License V3 for more details. //
+// //
+// You should have received a copy of the GNU General Public License //
+// along with this program. If not, see . //
+///////////////////////////////////////////////////////////////////////////////////
+
+#ifndef INCLUDE_DSP_FILTERRC_H_
+#define INCLUDE_DSP_FILTERRC_H_
+
+#include "dsp/dsptypes.h"
+
+/** First order low-pass IIR filter for real-valued signals. */
+class LowPassFilterRC
+{
+public:
+
+ /**
+ * Construct 1st order low-pass IIR filter.
+ *
+ * timeconst :: RC time constant in seconds (1 / (2 * PI * cutoff_freq)
+ */
+ LowPassFilterRC(Real timeconst);
+
+ /**
+ * Reconfigure filter with new time constant
+ */
+ void configure(Real timeout);
+
+ /** Process samples. */
+ void process(const Real& sample_in, Real& sample_out);
+
+private:
+ Real m_timeconst;
+ Real m_y1;
+ Real m_a1;
+ Real m_b0;
+};
+
+#endif /* INCLUDE_DSP_FILTERRC_H_ */
diff --git a/plugins/channel/bfm/bfmdemod.cpp b/plugins/channel/bfm/bfmdemod.cpp
index 2fed4cb94..0541c33ac 100644
--- a/plugins/channel/bfm/bfmdemod.cpp
+++ b/plugins/channel/bfm/bfmdemod.cpp
@@ -32,7 +32,9 @@ BFMDemod::BFMDemod(SampleSink* sampleSink) :
m_sampleSink(sampleSink),
m_audioFifo(4, 250000),
m_settingsMutex(QMutex::Recursive),
- m_pilotPLL(19000/384000, 50/384000, 0.01)
+ m_pilotPLL(19000/384000, 50/384000, 0.01),
+ m_deemphasisFilterX(default_deemphasis * 48000 * 1.0e-6),
+ m_deemphasisFilterY(default_deemphasis * 48000 * 1.0e-6)
{
setObjectName("BFMDemod");
@@ -43,6 +45,8 @@ BFMDemod::BFMDemod(SampleSink* sampleSink) :
m_config.m_squelch = -60.0;
m_config.m_volume = 2.0;
m_config.m_audioSampleRate = DSPEngine::instance()->getAudioSampleRate(); // normally 48 kHz
+ m_deemphasisFilterX.configure(default_deemphasis * m_config.m_audioSampleRate * 1.0e-6);
+ m_deemphasisFilterY.configure(default_deemphasis * m_config.m_audioSampleRate * 1.0e-6);
m_rfFilter = new fftfilt(-50000.0 / 384000.0, 50000.0 / 384000.0, rfFilterFftLength);
apply();
@@ -120,7 +124,7 @@ void BFMDemod::feed(const SampleVector::const_iterator& begin, const SampleVecto
m_m1Sample = rf[i];
m_sampleBuffer.push_back(Sample(demod * (1<<15), 0.0));
- quint16 sampleStereo;
+ Real sampleStereo;
// Process stereo if stereo mode is selected
@@ -135,7 +139,7 @@ void BFMDemod::feed(const SampleVector::const_iterator& begin, const SampleVecto
if (m_interpolatorStereo.interpolate(&m_interpolatorStereoDistanceRemain, s, &cs))
{
- sampleStereo = (qint16)(cs.real() * 3000 * m_running.m_volume);
+ sampleStereo = cs.real();
}
}
@@ -143,15 +147,19 @@ void BFMDemod::feed(const SampleVector::const_iterator& begin, const SampleVecto
if (m_interpolator.interpolate(&m_interpolatorDistanceRemain, e, &ci))
{
- quint16 sample = (qint16)(ci.real() * 3000 * m_running.m_volume);
-
if (m_running.m_audioStereo)
{
- m_audioBuffer[m_audioBufferFill].l = sample + sampleStereo;
- m_audioBuffer[m_audioBufferFill].r = sample - sampleStereo;
+ Real deemph_l, deemph_r; // Pre-emphasis is applied on each channel before multiplexing
+ m_deemphasisFilterX.process(ci.real() + sampleStereo, deemph_l);
+ m_deemphasisFilterY.process(ci.real() - sampleStereo, deemph_r);
+ m_audioBuffer[m_audioBufferFill].l = (qint16)(deemph_l * 3000 * m_running.m_volume);
+ m_audioBuffer[m_audioBufferFill].r = (qint16)(deemph_r * 3000 * m_running.m_volume);
}
else
{
+ Real deemph;
+ m_deemphasisFilterX.process(ci.real() + sampleStereo, deemph);
+ quint16 sample = (qint16)(deemph * 3000 * m_running.m_volume);
m_audioBuffer[m_audioBufferFill].l = sample;
m_audioBuffer[m_audioBufferFill].r = sample;
}
@@ -319,6 +327,12 @@ void BFMDemod::apply()
m_squelchLevel *= m_squelchLevel;
}
+ if (m_config.m_audioSampleRate != m_running.m_audioSampleRate)
+ {
+ m_deemphasisFilterX.configure(default_deemphasis * m_config.m_audioSampleRate * 1.0e-6);
+ m_deemphasisFilterY.configure(default_deemphasis * m_config.m_audioSampleRate * 1.0e-6);
+ }
+
m_running.m_inputSampleRate = m_config.m_inputSampleRate;
m_running.m_inputFrequencyOffset = m_config.m_inputFrequencyOffset;
m_running.m_rfBandwidth = m_config.m_rfBandwidth;
diff --git a/plugins/channel/bfm/bfmdemod.h b/plugins/channel/bfm/bfmdemod.h
index 114aacd96..9dc2fcba3 100644
--- a/plugins/channel/bfm/bfmdemod.h
+++ b/plugins/channel/bfm/bfmdemod.h
@@ -27,6 +27,7 @@
#include "dsp/movingaverage.h"
#include "dsp/fftfilt.h"
#include "dsp/phaselock.h"
+#include "dsp/filterrc.h"
#include "audio/audiofifo.h"
#include "util/message.h"
@@ -147,6 +148,10 @@ private:
StereoPhaseLock m_pilotPLL;
Real m_pilotPLLSamples[2];
+ LowPassFilterRC m_deemphasisFilterX;
+ LowPassFilterRC m_deemphasisFilterY;
+ static const Real default_deemphasis = 50.0; // 50 us
+
void apply();
};
diff --git a/sdrbase/dsp/filterrc.cpp b/sdrbase/dsp/filterrc.cpp
new file mode 100644
index 000000000..c5a7be190
--- /dev/null
+++ b/sdrbase/dsp/filterrc.cpp
@@ -0,0 +1,58 @@
+///////////////////////////////////////////////////////////////////////////////////
+// Copyright (C) 2015 F4EXB //
+// written by Edouard Griffiths //
+// //
+// This program is free software; you can redistribute it and/or modify //
+// it under the terms of the GNU General Public License as published by //
+// the Free Software Foundation as version 3 of the License, or //
+// //
+// This program is distributed in the hope that it will be useful, //
+// but WITHOUT ANY WARRANTY; without even the implied warranty of //
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
+// GNU General Public License V3 for more details. //
+// //
+// You should have received a copy of the GNU General Public License //
+// along with this program. If not, see . //
+///////////////////////////////////////////////////////////////////////////////////
+
+#include
+#include "dsp/filterrc.h"
+
+// Construct 1st order low-pass IIR filter.
+LowPassFilterRC::LowPassFilterRC(Real timeconst) :
+ m_timeconst(timeconst),
+ m_y1(0)
+{
+ m_a1 = - exp(-1/m_timeconst);
+ m_b0 = 1 + m_a1;
+}
+
+// Reconfigure
+void LowPassFilterRC::configure(Real timeconst)
+{
+ m_timeconst = timeconst;
+ m_y1 = 0;
+ m_a1 = - exp(-1/m_timeconst);
+ m_b0 = 1 + m_a1;
+
+ qDebug() << "LowPassFilterRC::configure: t: " << m_timeconst
+ << " a1: " << m_a1
+ << " b0: " << m_b0;
+}
+
+// Process samples.
+void LowPassFilterRC::process(const Real& sample_in, Real& sample_out)
+{
+ /*
+ * Continuous domain:
+ * H(s) = 1 / (1 - s * timeconst)
+ *
+ * Discrete domain:
+ * H(z) = (1 - exp(-1/timeconst)) / (1 - exp(-1/timeconst) / z)
+ */
+
+ m_y1 = (sample_in * m_b0) - (m_y1 * m_a1);
+ sample_out = m_y1;
+}
+
+
diff --git a/sdrbase/dsp/phaselock.cpp b/sdrbase/dsp/phaselock.cpp
index 4840b9a60..f6464def7 100644
--- a/sdrbase/dsp/phaselock.cpp
+++ b/sdrbase/dsp/phaselock.cpp
@@ -81,6 +81,7 @@ PhaseLock::PhaseLock(Real freq, Real bandwidth, Real minsignal)
m_sample_cnt = 0;
}
+
void PhaseLock::configure(Real freq, Real bandwidth, Real minsignal)
{
qDebug("PhaseLock::configure: freq: %f bandwidth: %f minsignal: %f", freq, bandwidth, minsignal);
@@ -142,6 +143,7 @@ void PhaseLock::configure(Real freq, Real bandwidth, Real minsignal)
m_sample_cnt = 0;
}
+
// Process samples. Bufferized version
void PhaseLock::process(const std::vector& samples_in, std::vector& samples_out)
{
@@ -247,21 +249,6 @@ void PhaseLock::process(const std::vector& samples_in, std::vector&
m_sample_cnt += n;
}
-/*
-void PhaseLock::process(const Real& sample_in, Real& sample_out)
-{
- m_phase += m_freq;
-
- if (m_phase > 2.0 * M_PI) {
- m_phase -= 2.0 * M_PI;
- }
-
- Real psin = sin(m_phase);
- Real pcos = cos(m_phase);
-
- sample_out = 2 * psin * pcos;
-}*/
-
// Process samples. Multiple output
void PhaseLock::process(const Real& sample_in, Real *samples_out)