kopia lustrzana https://github.com/f4exb/sdrangel
DSD demod: use audio compressor when processing voice with serial DV
f4exb
rodzic
175e4ca98a
commit
12380d4e51
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@ -3,6 +3,7 @@ project (sdrbase)
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#set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -fsanitize=undefined")
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set(sdrbase_SOURCES
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audio/audiocompressor.cpp
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audio/audiodevicemanager.cpp
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audio/audiofifo.cpp
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audio/audiooutput.cpp
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@ -90,6 +91,7 @@ set(sdrbase_SOURCES
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)
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set(sdrbase_HEADERS
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audio/audiocompressor.h
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audio/audiodevicemanager.h
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audio/audiofifo.h
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audio/audiooutput.h
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@ -0,0 +1,87 @@
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///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2018 F4EXB //
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// written by Edouard Griffiths //
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// //
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// This program is free software; you can redistribute it and/or modify //
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// it under the terms of the GNU General Public License as published by //
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// the Free Software Foundation as version 3 of the License, or //
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// //
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// This program is distributed in the hope that it will be useful, //
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// but WITHOUT ANY WARRANTY; without even the implied warranty of //
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
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// GNU General Public License V3 for more details. //
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// //
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// You should have received a copy of the GNU General Public License //
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// along with this program. If not, see <http://www.gnu.org/licenses/>. //
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///////////////////////////////////////////////////////////////////////////////////
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#include "audiocompressor.h"
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AudioCompressor::AudioCompressor()
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{
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fillLUT2();
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}
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AudioCompressor::~AudioCompressor()
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{}
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void AudioCompressor::fillLUT()
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{
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for (int i=0; i<8192; i++) {
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m_lut[i] = (24576/8192)*i;
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}
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for (int i=8192; i<2*8192; i++) {
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m_lut[i] = 24576 + 0.5f*(i-8192);
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}
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for (int i=2*8192; i<3*8192; i++) {
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m_lut[i] = 24576 + 4096 + 0.25f*(i-2*8192);
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}
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for (int i=3*8192; i<4*8192; i++) {
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m_lut[i] = 24576 + 4096 + 2048 + 0.125f*(i-3*8192);
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}
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}
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void AudioCompressor::fillLUT2()
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{
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for (int i=0; i<4096; i++) {
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m_lut[i] = (24576/4096)*i;
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}
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for (int i=4096; i<2*4096; i++) {
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m_lut[i] = 24576 + 0.5f*(i-4096);
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}
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for (int i=2*4096; i<3*4096; i++) {
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m_lut[i] = 24576 + 2048 + 0.25f*(i-2*4096);
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}
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for (int i=3*4096; i<4*4096; i++) {
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m_lut[i] = 24576 + 2048 + 1024 + 0.125f*(i-3*4096);
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}
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for (int i=4*4096; i<5*4096; i++) {
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m_lut[i] = 24576 + 2048 + 1024 + 512 + 0.0625f*(i-4*4096);
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}
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for (int i=5*4096; i<6*4096; i++) {
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m_lut[i] = 24576 + 2048 + 1024 + 512 + 256 + 0.03125f*(i-5*4096);
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}
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for (int i=6*4096; i<7*4096; i++) {
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m_lut[i] = 24576 + 2048 + 1024 + 512 + 256 + 128 + 0.015625f*(i-6*4096);
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}
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for (int i=7*4096; i<8*4096; i++) {
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m_lut[i] = 24576 + 2048 + 1024 + 512 + 256 + 128 + 64 + 0.0078125f*(i-7*4096);
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}
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}
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int16_t AudioCompressor::compress(int16_t sample)
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{
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int16_t sign = sample < 0 ? -1 : 1;
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int16_t abs = sample < 0 ? -sample : sample;
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return sign * m_lut[abs];
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}
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@ -0,0 +1,39 @@
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///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2018 F4EXB //
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// written by Edouard Griffiths //
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// //
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// This program is free software; you can redistribute it and/or modify //
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// it under the terms of the GNU General Public License as published by //
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// the Free Software Foundation as version 3 of the License, or //
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// //
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// This program is distributed in the hope that it will be useful, //
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// but WITHOUT ANY WARRANTY; without even the implied warranty of //
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
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// GNU General Public License V3 for more details. //
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// //
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// You should have received a copy of the GNU General Public License //
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// along with this program. If not, see <http://www.gnu.org/licenses/>. //
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///////////////////////////////////////////////////////////////////////////////////
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#ifndef SDRBASE_AUDIO_AUDIOCOMPRESSOR_H_
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#define SDRBASE_AUDIO_AUDIOCOMPRESSOR_H_
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#include <stdint.h>
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#include "export.h"
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class SDRBASE_API AudioCompressor
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{
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public:
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AudioCompressor();
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~AudioCompressor();
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void fillLUT(); //!< 4 bands
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void fillLUT2(); //!< 8 bands (default)
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int16_t compress(int16_t sample);
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private:
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int16_t m_lut[32768];
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};
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#endif /* SDRBASE_AUDIO_AUDIOCOMPRESSOR_H_ */
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@ -159,7 +159,7 @@ void DVSerialWorker::upsample(int upsampling, short *in, int nbSamplesIn, unsign
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{
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for (int i = 0; i < nbSamplesIn; i++)
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{
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float cur = m_upsampleFilter.usesHP() ? m_upsampleFilter.runHP((float) in[i]) : (float) in[i];
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float cur = m_upsampleFilter.usesHP() ? m_upsampleFilter.runHP((float) m_compressor.compress(in[i])) : (float) m_compressor.compress(in[i]);
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float prev = m_upsamplerLastValue;
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qint16 upsample;
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@ -32,6 +32,7 @@
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#include "export.h"
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#include "dsp/filtermbe.h"
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#include "dsp/dsptypes.h"
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#include "audio/audiocompressor.h"
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class AudioFifo;
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@ -153,6 +154,7 @@ private:
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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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AudioCompressor m_compressor;
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};
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#endif /* SDRBASE_DSP_DVSERIALWORKER_H_ */
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