kopia lustrzana https://github.com/f4exb/sdrangel
96 wiersze
3.1 KiB
C++
96 wiersze
3.1 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2019 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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// (at your option) any later version. //
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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 "audioresampler.h"
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AudioResampler::AudioResampler() :
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m_decimation(1),
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m_decimationCount(0)
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{}
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AudioResampler::~AudioResampler()
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{}
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void AudioResampler::setDecimation(uint32_t decimation)
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{
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m_decimation = decimation == 0 ? 1 : decimation;
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}
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void AudioResampler::setAudioFilters(int srHigh, int srLow, int fcLow, int fcHigh, float gain)
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{
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srHigh = (srHigh <= 100 ? 100 : srHigh);
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srLow = (srLow <= 0 ? 1 : srLow);
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srLow = srLow > srHigh - 50 ? srHigh - 50 : srLow;
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fcLow = fcLow < 0 ? 0 : fcLow;
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fcHigh = fcHigh < 100 ? 100 : fcHigh;
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fcLow = fcLow > fcHigh - 100 ? fcHigh - 100 : fcLow;
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m_audioFilter.setDecimFilters(srHigh, srLow, fcHigh, fcLow, gain);
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}
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bool AudioResampler::downSample(qint16 sampleIn, qint16& sampleOut)
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{
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if (m_decimation == 1)
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{
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sampleOut = sampleIn;
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return true;
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}
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if (m_decimationCount >= m_decimation - 1)
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{
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float lpSample = m_audioFilter.run(sampleIn / 32768.0f);
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sampleOut = lpSample * 32768.0f;
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m_decimationCount = 0;
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return true;
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}
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else
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{
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m_decimationCount++;
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return false;
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}
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}
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bool AudioResampler::upSample(qint16 sampleIn, qint16& sampleOut)
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{
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float lpSample;
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if (m_decimation == 1)
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{
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sampleOut = sampleIn;
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return true;
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}
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if (m_decimationCount >= m_decimation - 1)
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{
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m_decimationCount = 0;
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lpSample = m_audioFilter.run(sampleIn / 32768.0f);
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sampleOut = lpSample * 32768.0f;
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return true;
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}
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else
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{
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m_decimationCount++;
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lpSample = m_audioFilter.run(0.0f);
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sampleOut = lpSample * 32768.0f;
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return false;
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}
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}
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