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AM demod: fixed delayed squelch
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debian/changelog
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3
debian/changelog
vendored
@ -4,8 +4,9 @@ sdrangel (3.14.5-1) unstable; urgency=medium
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* Added a benchmark program testing decimators
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* Added a benchmark program testing decimators
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* Optimization of decimators using even/odd technique
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* Optimization of decimators using even/odd technique
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* SSB mod: fixed channel unregistration
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* SSB mod: fixed channel unregistration
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* AM demod: fixed delayed squelch
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-- Edouard Griffiths, F4EXB <f4exb06@gmail.com> Fri, 04 May 2018 20:14:18 +0200
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-- Edouard Griffiths, F4EXB <f4exb06@gmail.com> Sun, 06 May 2018 20:14:18 +0200
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sdrangel (3.14.4-1) unstable; urgency=medium
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sdrangel (3.14.4-1) unstable; urgency=medium
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@ -49,7 +49,7 @@ AMDemod::AMDemod(DeviceSourceAPI *deviceAPI) :
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m_inputFrequencyOffset(0),
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m_inputFrequencyOffset(0),
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m_running(false),
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m_running(false),
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m_squelchOpen(false),
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m_squelchOpen(false),
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m_squelchDelayLine(12000),
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m_squelchDelayLine(9600),
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m_magsqSum(0.0f),
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m_magsqSum(0.0f),
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m_magsqPeak(0.0f),
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m_magsqPeak(0.0f),
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m_magsqCount(0),
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m_magsqCount(0),
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@ -232,7 +232,7 @@ void AMDemod::applyAudioSampleRate(int sampleRate)
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m_interpolatorDistance = (Real) m_inputSampleRate / (Real) sampleRate;
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m_interpolatorDistance = (Real) m_inputSampleRate / (Real) sampleRate;
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m_bandpass.create(301, sampleRate, 300.0, m_settings.m_rfBandwidth / 2.0f);
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m_bandpass.create(301, sampleRate, 300.0, m_settings.m_rfBandwidth / 2.0f);
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m_audioFifo.setSize(sampleRate);
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m_audioFifo.setSize(sampleRate);
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m_squelchDelayLine.resize(sampleRate/4);
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m_squelchDelayLine.resize(sampleRate/5);
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m_settingsMutex.unlock();
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m_settingsMutex.unlock();
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m_audioSampleRate = sampleRate;
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m_audioSampleRate = sampleRate;
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@ -30,7 +30,6 @@
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#include "audio/audiofifo.h"
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#include "audio/audiofifo.h"
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#include "util/message.h"
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#include "util/message.h"
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#include "util/doublebufferfifo.h"
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#include "util/doublebufferfifo.h"
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#include "amdemodsettings.h"
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#include "amdemodsettings.h"
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class DeviceSourceAPI;
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class DeviceSourceAPI;
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@ -183,8 +182,8 @@ private:
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void processOneSample(Complex &ci)
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void processOneSample(Complex &ci)
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{
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{
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Real re = ci.real() / SDR_RX_SCALED;
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Real re = ci.real() / SDR_RX_SCALEF;
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Real im = ci.imag() / SDR_RX_SCALED;
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Real im = ci.imag() / SDR_RX_SCALEF;
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Real magsq = re*re + im*im;
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Real magsq = re*re + im*im;
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m_movingAverage(magsq);
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m_movingAverage(magsq);
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m_magsq = m_movingAverage.asDouble();
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m_magsq = m_movingAverage.asDouble();
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@ -197,43 +196,28 @@ private:
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m_magsqCount++;
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m_magsqCount++;
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// if (m_magsq >= m_squelchLevel)
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m_squelchDelayLine.write(magsq);
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// {
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// if (m_squelchCount <= m_audioSampleRate / 10)
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// {
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// m_squelchCount++;
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// }
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// }
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// else
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// {
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// if (m_squelchCount > 1)
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// {
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// m_squelchCount -= 2;
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// }
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// }
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if (m_magsq >= m_squelchLevel)
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if (m_magsq < m_squelchLevel)
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{
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if (m_squelchCount < m_audioSampleRate / 10) {
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m_squelchCount++;
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}
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m_squelchDelayLine.write(magsq);
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}
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else
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{
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{
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if (m_squelchCount > 0) {
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if (m_squelchCount > 0) {
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m_squelchCount--;
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m_squelchCount--;
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}
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}
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}
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m_squelchDelayLine.write(0);
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else
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{
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if (m_squelchCount < m_audioSampleRate / 10) {
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m_squelchCount++;
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}
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}
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}
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qint16 sample;
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qint16 sample;
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if ((m_squelchCount >= m_audioSampleRate / 20) && !m_settings.m_audioMute)
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m_squelchOpen = (m_squelchCount >= m_audioSampleRate / 20);
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if (m_squelchOpen && !m_settings.m_audioMute)
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{
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{
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Real demod = sqrt(m_squelchDelayLine.readBack(m_audioSampleRate / 20));
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Real demod = sqrt(m_squelchDelayLine.readBack(m_audioSampleRate/20));
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m_volumeAGC.feed(demod);
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m_volumeAGC.feed(demod);
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demod = (demod - m_volumeAGC.getValue()) / m_volumeAGC.getValue();
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demod = (demod - m_volumeAGC.getValue()) / m_volumeAGC.getValue();
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@ -245,12 +229,10 @@ private:
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Real attack = (m_squelchCount - 0.05f * m_audioSampleRate) / (0.05f * m_audioSampleRate);
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Real attack = (m_squelchCount - 0.05f * m_audioSampleRate) / (0.05f * m_audioSampleRate);
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sample = demod * attack * (m_audioSampleRate/24) * m_settings.m_volume;
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sample = demod * attack * (m_audioSampleRate/24) * m_settings.m_volume;
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m_squelchOpen = true;
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}
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}
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else
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else
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{
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{
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sample = 0;
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sample = 0;
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m_squelchOpen = false;
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}
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}
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m_audioBuffer[m_audioBufferFill].l = sample;
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m_audioBuffer[m_audioBufferFill].l = sample;
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