diff --git a/plugins/channelrx/demodfreedv/freedvdemod.cpp b/plugins/channelrx/demodfreedv/freedvdemod.cpp
index e3c3191bb..8b22fa3ac 100644
--- a/plugins/channelrx/demodfreedv/freedvdemod.cpp
+++ b/plugins/channelrx/demodfreedv/freedvdemod.cpp
@@ -49,8 +49,8 @@ const QString FreeDVDemod::m_channelId = "FreeDVDemod";
 FreeDVDemod::FreeDVDemod(DeviceSourceAPI *deviceAPI) :
         ChannelSinkAPI(m_channelIdURI),
         m_deviceAPI(deviceAPI),
-        m_hiCutoff(5000),
-        m_lowCutoff(300),
+        m_hiCutoff(6000),
+        m_lowCutoff(0),
         m_volume(2),
         m_spanLog2(3),
         m_sum(0),
@@ -97,7 +97,7 @@ FreeDVDemod::FreeDVDemod(DeviceSourceAPI *deviceAPI) :
 	m_agc.setClampMax(SDR_RX_SCALED/100.0);
 	m_agc.setClamping(m_agcClamping);
 
-	SSBFilter = new fftfilt(m_lowCutoff / m_audioSampleRate, m_hiCutoff / m_audioSampleRate, ssbFftLen);
+	SSBFilter = new fftfilt(m_lowCutoff / m_modemSampleRate, m_hiCutoff / m_modemSampleRate, ssbFftLen);
 
     applyChannelSettings(m_inputSampleRate, m_inputFrequencyOffset, true);
 	applySettings(m_settings, true);
@@ -320,7 +320,7 @@ bool FreeDVDemod::handleMessage(const Message& cmd)
 
 void FreeDVDemod::pushSampleToDV(int16_t sample)
 {
-    qint16 speechSample, audioSample;
+    qint16 audioSample;
 
     if (m_iModem == m_nin)
     {
@@ -328,9 +328,7 @@ void FreeDVDemod::pushSampleToDV(int16_t sample)
 
         for (int i = 0; i < nout; i++)
         {
-            speechSample = (qint16)(m_speechOut[i] * m_volume);
-
-            while (!m_audioResampler.upSample(speechSample, audioSample)) {
+            while (!m_audioResampler.upSample(m_speechOut[i], audioSample)) {
                 pushSampleToAudio(audioSample);
             }
 
@@ -346,8 +344,8 @@ void FreeDVDemod::pushSampleToDV(int16_t sample)
 
 void FreeDVDemod::pushSampleToAudio(int16_t sample)
 {
-    m_audioBuffer[m_audioBufferFill].l = sample;
-    m_audioBuffer[m_audioBufferFill].r = sample;
+    m_audioBuffer[m_audioBufferFill].l = sample * m_volume;
+    m_audioBuffer[m_audioBufferFill].r = sample * m_volume;
     ++m_audioBufferFill;
 
     if (m_audioBufferFill >= m_audioBuffer.size())
@@ -379,7 +377,7 @@ void FreeDVDemod::applyChannelSettings(int inputSampleRate, int inputFrequencyOf
         m_settingsMutex.lock();
         m_interpolator.create(16, inputSampleRate, m_hiCutoff * 1.5f, 2.0f);
         m_interpolatorDistanceRemain = 0;
-        m_interpolatorDistance = (Real) inputSampleRate / (Real) m_audioSampleRate;
+        m_interpolatorDistance = (Real) inputSampleRate / (Real) m_modemSampleRate;
         m_settingsMutex.unlock();
     }
 
@@ -394,7 +392,7 @@ void FreeDVDemod::applyAudioSampleRate(int sampleRate)
     m_settingsMutex.lock();
     m_audioFifo.setSize(sampleRate);
     m_audioResampler.setDecimation(sampleRate / m_speechSampleRate);
-    m_audioResampler.setAudioFilters(sampleRate, m_speechSampleRate, 250, 3300);
+    m_audioResampler.setAudioFilters(sampleRate, sampleRate, 250, 3300, 4.0f);
     m_settingsMutex.unlock();
 
     m_audioSampleRate = sampleRate;
@@ -415,15 +413,17 @@ void FreeDVDemod::applyFreeDVMode(FreeDVDemodSettings::FreeDVMode mode)
                 modemSampleRate, m_settings.m_inputFrequencyOffset);
         m_inputMessageQueue.push(channelConfigMsg);
 
-        m_interpolator.create(16, m_inputSampleRate, m_hiCutoff * 1.5f, 2.0f);
         m_interpolatorDistanceRemain = 0;
+        //m_interpolatorConsumed = false;
         m_interpolatorDistance = (Real) m_inputSampleRate / (Real) modemSampleRate;
-
+        m_interpolator.create(16, m_inputSampleRate, m_hiCutoff * 1.5f, 2.0f);
         SSBFilter->create_filter(m_lowCutoff / (float) modemSampleRate, m_hiCutoff / (float) modemSampleRate);
 
         int agcNbSamples = (modemSampleRate / 1000) * (1<<m_settings.m_agcTimeLog2);
         int agcThresholdGate = (modemSampleRate / 1000) * m_settings.m_agcThresholdGate; // ms
 
+        m_modemSampleRate = modemSampleRate;
+
         if (m_agcNbSamples != agcNbSamples)
         {
             m_agc.resize(agcNbSamples, agcNbSamples/2, agcTarget);
@@ -437,8 +437,6 @@ void FreeDVDemod::applyFreeDVMode(FreeDVDemodSettings::FreeDVMode mode)
             m_agcThresholdGate = agcThresholdGate;
         }
 
-        m_modemSampleRate = modemSampleRate;
-
         if (getMessageQueueToGUI())
         {
             DSPConfigureAudio *cfg = new DSPConfigureAudio(m_modemSampleRate);
@@ -583,10 +581,10 @@ void FreeDVDemod::applySettings(const FreeDVDemodSettings& settings, bool force)
         (m_settings.m_agcThresholdGate != settings.m_agcThresholdGate) ||
         (m_settings.m_agcClamping != settings.m_agcClamping) || force)
     {
-        int agcNbSamples = (m_audioSampleRate / 1000) * (1<<settings.m_agcTimeLog2);
+        int agcNbSamples = (m_modemSampleRate / 1000) * (1<<settings.m_agcTimeLog2);
         m_agc.setThresholdEnable(settings.m_agcPowerThreshold != -FreeDVDemodSettings::m_minPowerThresholdDB);
         double agcPowerThreshold = CalcDb::powerFromdB(settings.m_agcPowerThreshold) * (SDR_RX_SCALED*SDR_RX_SCALED);
-        int agcThresholdGate = (m_audioSampleRate / 1000) * settings.m_agcThresholdGate; // ms
+        int agcThresholdGate = (m_modemSampleRate / 1000) * settings.m_agcThresholdGate; // ms
         bool agcClamping = settings.m_agcClamping;
 
         if (m_agcNbSamples != agcNbSamples)
@@ -752,7 +750,7 @@ int FreeDVDemod::webapiSettingsPutPatch(
     if (frequencyOffsetChanged)
     {
         MsgConfigureChannelizer* channelConfigMsg = MsgConfigureChannelizer::create(
-                m_audioSampleRate, settings.m_inputFrequencyOffset);
+                m_modemSampleRate, settings.m_inputFrequencyOffset);
         m_inputMessageQueue.push(channelConfigMsg);
     }
 
diff --git a/sdrbase/audio/audiofilter.cpp b/sdrbase/audio/audiofilter.cpp
index 1f61569eb..53c1e6e76 100644
--- a/sdrbase/audio/audiofilter.cpp
+++ b/sdrbase/audio/audiofilter.cpp
@@ -38,23 +38,23 @@ AudioFilter::~AudioFilter()
 {}
 
 
-void AudioFilter::setDecimFilters(int srHigh, int srLow, float fcHigh, float fcLow)
+void AudioFilter::setDecimFilters(int srHigh, int srLow, float fcHigh, float fcLow, float fgain)
 {
     double fcNormHigh = fcHigh / srHigh;
     double fcNormLow = fcLow / srLow;
 
-    calculate2(false, fcNormHigh, m_lpva, m_lpvb);
-    calculate2(true, fcNormLow, m_hpva, m_hpvb);
+    calculate2(false, fcNormHigh, m_lpva, m_lpvb, fgain);
+    calculate2(true, fcNormLow, m_hpva, m_hpvb, fgain);
 
     m_filterLP.setCoeffs(m_lpva, m_lpvb);
     m_filterHP.setCoeffs(m_hpva, m_hpvb);
 }
 
-void AudioFilter::calculate2(bool highPass, double fc, float *va, float *vb)
+void AudioFilter::calculate2(bool highPass, double fc, float *va, float *vb, float fgain)
 {
     double a[22], b[22];
 
-    cheby(highPass, fc, 0.5, 2, a, b); // low-pass, 0.5% ripple, 2 pole filter
+    cheby(highPass, fc, 0.5, 2, a, b, fgain); // low-pass, 0.5% ripple, 2 pole filter
 
     // Copy to the 2-pole filter coefficients
     for (int i=0; i<3; i++) {
@@ -79,7 +79,7 @@ void AudioFilter::calculate2(bool highPass, double fc, float *va, float *vb)
  * Adapted from BASIC program in table 20-4 of
  * https://www.analog.com/media/en/technical-documentation/dsp-book/dsp_book_Ch20.pdf
  */
-void AudioFilter::cheby(bool highPass, double fc, float pr, int np, double *a, double *b)
+void AudioFilter::cheby(bool highPass, double fc, float pr, int np, double *a, double *b, float fgain)
 {
     double a0, a1, a2, b1, b2;
     double ta[22], tb[22];
@@ -135,6 +135,7 @@ void AudioFilter::cheby(bool highPass, double fc, float pr, int np, double *a, d
     }
 
     double gain = sa/(1.0 -sb);
+    gain /= fgain;
 
     for (int i=0; i<20; i++) {
         a[i] /= gain;
diff --git a/sdrbase/audio/audiofilter.h b/sdrbase/audio/audiofilter.h
index 935351b31..91989d4a2 100644
--- a/sdrbase/audio/audiofilter.h
+++ b/sdrbase/audio/audiofilter.h
@@ -47,14 +47,14 @@ public:
 
     void useHP(bool useHP) { m_useHP = useHP; }
     bool usesHP() const { return m_useHP; }
-    void setDecimFilters(int srHigh, int srLow, float fcHigh, float fcLow);
+    void setDecimFilters(int srHigh, int srLow, float fcHigh, float fcLow, float gain = 1.0f);
     float run(const float& sample);
     float runHP(const float& sample);
     float runLP(const float& sample);
 
 private:
-    void calculate2(bool highPass, double fc, float *a, float *b); // two pole Chebyshev calculation
-    void cheby(bool highPass, double fc, float pr, int np, double *a, double *b);
+    void calculate2(bool highPass, double fc, float *a, float *b, float fgain); // two pole Chebyshev calculation
+    void cheby(bool highPass, double fc, float pr, int np, double *a, double *b, float fgain);
     void cheby_sub(bool highPass, double fc, float pr, int np, int stage,
             double& a0, double& a1, double& a2, double& b1, double& b2);
 
diff --git a/sdrbase/audio/audioresampler.cpp b/sdrbase/audio/audioresampler.cpp
index 7a38105b6..48eae8662 100644
--- a/sdrbase/audio/audioresampler.cpp
+++ b/sdrbase/audio/audioresampler.cpp
@@ -30,7 +30,7 @@ void AudioResampler::setDecimation(uint32_t decimation)
     m_decimation = decimation == 0 ? 1 : decimation;
 }
 
-void AudioResampler::setAudioFilters(int srHigh, int srLow, int fcLow, int fcHigh)
+void AudioResampler::setAudioFilters(int srHigh, int srLow, int fcLow, int fcHigh, float gain)
 {
     srHigh = (srHigh <= 100 ? 100 : srHigh);
     srLow = (srLow <= 0 ? 1 : srLow);
@@ -40,7 +40,7 @@ void AudioResampler::setAudioFilters(int srHigh, int srLow, int fcLow, int fcHig
     fcHigh = fcHigh < 100 ? 100 : fcHigh;
     fcLow = fcLow > fcHigh - 100 ? fcHigh - 100 : fcLow;
 
-    m_audioFilter.setDecimFilters(srHigh, srLow, fcHigh, fcLow);
+    m_audioFilter.setDecimFilters(srHigh, srLow, fcHigh, fcLow, gain);
 }
 
 bool AudioResampler::downSample(qint16 sampleIn, qint16& sampleOut)
diff --git a/sdrbase/audio/audioresampler.h b/sdrbase/audio/audioresampler.h
index 55273bc82..577f85049 100644
--- a/sdrbase/audio/audioresampler.h
+++ b/sdrbase/audio/audioresampler.h
@@ -29,7 +29,7 @@ public:
 
     void setDecimation(uint32_t decimation);
     uint32_t getDecimation() const { return m_decimation; }
-    void setAudioFilters(int srHigh, int srLow, int fcLow, int fcHigh);
+    void setAudioFilters(int srHigh, int srLow, int fcLow, int fcHigh, float gain=1.0f);
     bool downSample(qint16 sampleIn, qint16& sampleOut);
     bool upSample(qint16 sampleIn, qint16& sampleOut);