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https://github.com/f4exb/sdrangel.git
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238 lines
6.4 KiB
C++
238 lines
6.4 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany //
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// written by Christian Daniel //
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// (c) 2014 Modified by John Greb
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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 <QTime>
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#include <QDebug>
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#include <stdio.h>
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#include "ssbdemod.h"
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#include "audio/audiooutput.h"
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#include "dsp/dspengine.h"
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#include "dsp/channelizer.h"
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MESSAGE_CLASS_DEFINITION(SSBDemod::MsgConfigureSSBDemod, Message)
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SSBDemod::SSBDemod(SampleSink* sampleSink) :
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m_sampleSink(sampleSink),
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m_audioFifo(4, 24000),
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m_settingsMutex(QMutex::Recursive)
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{
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setObjectName("SSBDemod");
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m_Bandwidth = 5000;
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m_LowCutoff = 300;
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m_volume = 2.0;
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m_spanLog2 = 3;
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m_sampleRate = 96000;
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m_frequency = 0;
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m_nco.setFreq(m_frequency, m_sampleRate);
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m_audioSampleRate = DSPEngine::instance()->getAudioSampleRate();
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m_interpolator.create(16, m_sampleRate, 5000);
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m_sampleDistanceRemain = (Real) m_sampleRate / m_audioSampleRate;
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m_audioBuffer.resize(1<<9);
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m_audioBufferFill = 0;
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m_undersampleCount = 0;
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m_usb = true;
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SSBFilter = new fftfilt(m_LowCutoff / m_audioSampleRate, m_Bandwidth / m_audioSampleRate, ssbFftLen);
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DSPEngine::instance()->addAudioSink(&m_audioFifo);
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}
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SSBDemod::~SSBDemod()
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{
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if (SSBFilter)
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{
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delete SSBFilter;
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}
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DSPEngine::instance()->removeAudioSink(&m_audioFifo);
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}
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void SSBDemod::configure(MessageQueue* messageQueue, Real Bandwidth, Real LowCutoff, Real volume, int spanLog2)
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{
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Message* cmd = MsgConfigureSSBDemod::create(Bandwidth, LowCutoff, volume, spanLog2);
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messageQueue->push(cmd);
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}
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void SSBDemod::feed(const SampleVector::const_iterator& begin, const SampleVector::const_iterator& end, bool positiveOnly)
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{
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Complex ci;
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fftfilt::cmplx *sideband, sum;
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Real avg;
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int n_out;
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m_settingsMutex.lock();
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int decim = 1<<(m_spanLog2 - 1);
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unsigned char decim_mask = decim - 1; // counter LSB bit mask for decimation by 2^(m_scaleLog2 - 1)
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for(SampleVector::const_iterator it = begin; it < end; ++it)
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{
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Complex c(it->real() / 32768.0, it->imag() / 32768.0);
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c *= m_nco.nextIQ();
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if(m_interpolator.interpolate(&m_sampleDistanceRemain, c, &ci))
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{
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n_out = SSBFilter->runSSB(ci, &sideband, m_usb);
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m_sampleDistanceRemain += (Real)m_sampleRate / m_audioSampleRate;
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}
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else
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{
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n_out = 0;
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}
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for (int i = 0; i < n_out; i++)
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{
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Real demod = (sideband[i].real() + sideband[i].imag()) * 0.7 * 32768.0;
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// Downsample by 2^(m_scaleLog2 - 1) for SSB band spectrum display
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// smart decimation with bit gain using float arithmetic (23 bits significand)
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sum += sideband[i];
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if (!(m_undersampleCount++ & decim_mask))
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{
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avg = (sum.real() + sum.imag()) * 0.7 * 32768.0 / decim;
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m_sampleBuffer.push_back(Sample(avg, 0.0));
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sum.real() = 0.0;
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sum.imag() = 0.0;
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}
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qint16 sample = (qint16)(demod * m_volume * 10);
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m_audioBuffer[m_audioBufferFill].l = sample;
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m_audioBuffer[m_audioBufferFill].r = sample;
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++m_audioBufferFill;
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if (m_audioBufferFill >= m_audioBuffer.size())
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{
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uint res = m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 1);
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if (res != m_audioBufferFill)
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{
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qDebug("lost %u samples", m_audioBufferFill - res);
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}
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m_audioBufferFill = 0;
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}
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}
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}
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if (m_audioFifo.write((const quint8*)&m_audioBuffer[0], m_audioBufferFill, 0) != m_audioBufferFill)
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{
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qDebug("SSBDemod::feed: lost samples");
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}
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m_audioBufferFill = 0;
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if(m_sampleSink != 0)
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{
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m_sampleSink->feed(m_sampleBuffer.begin(), m_sampleBuffer.end(), true);
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}
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m_sampleBuffer.clear();
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m_settingsMutex.unlock();
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}
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void SSBDemod::start()
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{
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}
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void SSBDemod::stop()
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{
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}
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bool SSBDemod::handleMessage(const Message& cmd)
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{
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float band, lowCutoff;
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qDebug() << "SSBDemod::handleMessage";
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if (Channelizer::MsgChannelizerNotification::match(cmd))
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{
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Channelizer::MsgChannelizerNotification& notif = (Channelizer::MsgChannelizerNotification&) cmd;
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m_settingsMutex.lock();
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m_sampleRate = notif.getSampleRate();
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m_nco.setFreq(-notif.getFrequencyOffset(), m_sampleRate);
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m_interpolator.create(16, m_sampleRate, m_Bandwidth);
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m_sampleDistanceRemain = m_sampleRate / m_audioSampleRate;
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m_settingsMutex.unlock();
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qDebug() << "SSBDemod::handleMessage: MsgChannelizerNotification: m_sampleRate: " << m_sampleRate
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<< " frequencyOffset" << notif.getFrequencyOffset();
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return true;
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}
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else if (MsgConfigureSSBDemod::match(cmd))
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{
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MsgConfigureSSBDemod& cfg = (MsgConfigureSSBDemod&) cmd;
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m_settingsMutex.lock();
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band = cfg.getBandwidth();
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lowCutoff = cfg.getLoCutoff();
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if (band < 0) {
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band = -band;
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lowCutoff = -lowCutoff;
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m_usb = false;
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} else
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m_usb = true;
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if (band < 100.0f)
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{
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band = 100.0f;
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lowCutoff = 0;
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}
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m_Bandwidth = band;
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m_LowCutoff = lowCutoff;
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m_interpolator.create(16, m_sampleRate, band * 2.0f);
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SSBFilter->create_filter(m_LowCutoff / (float) m_audioSampleRate, m_Bandwidth / (float) m_audioSampleRate);
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m_volume = cfg.getVolume();
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m_volume *= m_volume * 0.1;
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m_spanLog2 = cfg.getSpanLog2();
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m_settingsMutex.unlock();
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qDebug() << " - MsgConfigureSSBDemod: m_Bandwidth: " << m_Bandwidth
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<< " m_LowCutoff: " << m_LowCutoff
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<< " m_volume: " << m_volume
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<< " m_spanLog2: " << m_spanLog2;
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return true;
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}
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else
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{
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if(m_sampleSink != 0)
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{
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return m_sampleSink->handleMessage(cmd);
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}
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else
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{
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return false;
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}
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}
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}
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