DSD demod: use audio compressor when processing voice with serial DV

sdrbase/CMakeLists.txt

@@ -3,6 +3,7 @@ project (sdrbase)
 #set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -fsanitize=undefined")
 
 set(sdrbase_SOURCES
+    audio/audiocompressor.cpp
     audio/audiodevicemanager.cpp
     audio/audiofifo.cpp
     audio/audiooutput.cpp
@@ -90,6 +91,7 @@ set(sdrbase_SOURCES
 )
 
 set(sdrbase_HEADERS
+	audio/audiocompressor.h
     audio/audiodevicemanager.h
     audio/audiofifo.h
     audio/audiooutput.h

sdrbase/audio/audiocompressor.cpp

@@ -0,0 +1,87 @@
+///////////////////////////////////////////////////////////////////////////////////
+// Copyright (C) 2018 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 <http://www.gnu.org/licenses/>.          //
+///////////////////////////////////////////////////////////////////////////////////
+
+#include "audiocompressor.h"
+
+AudioCompressor::AudioCompressor()
+{
+    fillLUT2();
+}
+
+AudioCompressor::~AudioCompressor()
+{}
+
+void AudioCompressor::fillLUT()
+{
+    for (int i=0; i<8192; i++) {
+        m_lut[i] = (24576/8192)*i;
+    }
+
+    for (int i=8192; i<2*8192; i++) {
+        m_lut[i] = 24576 + 0.5f*(i-8192);
+    }
+
+    for (int i=2*8192; i<3*8192; i++) {
+        m_lut[i] = 24576 + 4096 + 0.25f*(i-2*8192);
+    }
+
+    for (int i=3*8192; i<4*8192; i++) {
+        m_lut[i] = 24576 + 4096 + 2048 + 0.125f*(i-3*8192);
+    }
+}
+
+void AudioCompressor::fillLUT2()
+{
+    for (int i=0; i<4096; i++) {
+        m_lut[i] = (24576/4096)*i;
+    }
+
+    for (int i=4096; i<2*4096; i++) {
+        m_lut[i] = 24576 + 0.5f*(i-4096);
+    }
+
+    for (int i=2*4096; i<3*4096; i++) {
+        m_lut[i] = 24576 + 2048 + 0.25f*(i-2*4096);
+    }
+
+    for (int i=3*4096; i<4*4096; i++) {
+        m_lut[i] = 24576 + 2048 + 1024 + 0.125f*(i-3*4096);
+    }
+
+    for (int i=4*4096; i<5*4096; i++) {
+        m_lut[i] = 24576 + 2048 + 1024 + 512 + 0.0625f*(i-4*4096);
+    }
+
+    for (int i=5*4096; i<6*4096; i++) {
+        m_lut[i] = 24576 + 2048 + 1024 + 512 + 256 + 0.03125f*(i-5*4096);
+    }
+
+    for (int i=6*4096; i<7*4096; i++) {
+        m_lut[i] = 24576 + 2048 + 1024 + 512 + 256 + 128 + 0.015625f*(i-6*4096);
+    }
+
+    for (int i=7*4096; i<8*4096; i++) {
+        m_lut[i] = 24576 + 2048 + 1024 + 512 + 256 + 128 + 64 + 0.0078125f*(i-7*4096);
+    }
+}
+
+int16_t AudioCompressor::compress(int16_t sample)
+{
+    int16_t sign = sample < 0 ? -1 : 1;
+    int16_t abs = sample < 0 ? -sample : sample;
+    return sign * m_lut[abs];
+}

sdrbase/audio/audiocompressor.h

@@ -0,0 +1,39 @@
+///////////////////////////////////////////////////////////////////////////////////
+// Copyright (C) 2018 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 <http://www.gnu.org/licenses/>.          //
+///////////////////////////////////////////////////////////////////////////////////
+
+#ifndef SDRBASE_AUDIO_AUDIOCOMPRESSOR_H_
+#define SDRBASE_AUDIO_AUDIOCOMPRESSOR_H_
+
+#include <stdint.h>
+#include "export.h"
+
+class SDRBASE_API AudioCompressor
+{
+public:
+    AudioCompressor();
+    ~AudioCompressor();
+    void fillLUT();   //!< 4 bands
+    void fillLUT2();  //!< 8 bands (default)
+    int16_t compress(int16_t sample);
+
+private:
+    int16_t m_lut[32768];
+};
+
+
+
+#endif /* SDRBASE_AUDIO_AUDIOCOMPRESSOR_H_ */

sdrbase/dsp/dvserialworker.cpp

@@ -159,7 +159,7 @@ void DVSerialWorker::upsample(int upsampling, short *in, int nbSamplesIn, unsign
 {
     for (int i = 0; i < nbSamplesIn; i++)
     {
-        float cur = m_upsampleFilter.usesHP() ? m_upsampleFilter.runHP((float) in[i]) : (float) in[i];
+        float cur = m_upsampleFilter.usesHP() ? m_upsampleFilter.runHP((float) m_compressor.compress(in[i])) : (float) m_compressor.compress(in[i]);
         float prev = m_upsamplerLastValue;
         qint16 upsample;
 

sdrbase/dsp/dvserialworker.h

@@ -32,6 +32,7 @@
 #include "export.h"
 #include "dsp/filtermbe.h"
 #include "dsp/dsptypes.h"
+#include "audio/audiocompressor.h"
 
 class AudioFifo;
 
@@ -153,6 +154,7 @@ private:
     int m_upsampling;
     float m_volume;
     float m_upsamplingFactors[7];
+    AudioCompressor m_compressor;
 };
 
 #endif /* SDRBASE_DSP_DVSERIALWORKER_H_ */