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mirror of https://github.com/f4exb/sdrangel.git synced 2024-11-25 17:28:50 -05:00

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

This commit is contained in:
f4exb 2018-06-25 00:01:25 +02:00
parent 175e4ca98a
commit 12380d4e51
5 changed files with 131 additions and 1 deletions

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@ -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

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@ -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];
}

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@ -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_ */

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@ -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;

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@ -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_ */