NBFM Channel level CTCSS/DCS and audio features (#2425)

NBFM decoder enhancements.  CTCSS and DCS squelch control.  Adds deemphasis, high pass filter control, and auto-level control

src/main/java/io/github/dsheirer/audio/AbstractAudioFilter.java

@@ -0,0 +1,17 @@
+package io.github.dsheirer.audio;
+
+/**
+ * Used for defining an array list of filter methods to be implemented in AudioModule
+ * The filters will be applied in the order given in the list.
+ * Each audio filter mush implement the filter method.
+ */
+
+public abstract class AbstractAudioFilter
+{
+    /**
+     * To be overridden by each audio filter
+     * @param audio a buffer of 8 KHz audio samples
+     * @return float[] filtered buffer of 8 KHz audio samples
+     */
+    public abstract float[] filter(float[] audio);
+}

src/main/java/io/github/dsheirer/audio/AudioGainFilter.java

@@ -0,0 +1,44 @@
+/*
+ * *****************************************************************************
+ * Copyright (C) 2014-2026 Dennis Sheirer
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>
+ * ****************************************************************************
+ */
+
+package io.github.dsheirer.audio;
+
+import io.github.dsheirer.dsp.gain.AudioGainAndDcFilter;
+
+/**
+ * This is a wrapper for the AudioGainAndDcFilter class so it can be used with the AudioModule filter list.
+ * Usually instantiated in the DecoderFactory
+ */
+public class AudioGainFilter extends AbstractAudioFilter
+{
+    AudioGainAndDcFilter mAudioGain;
+
+    public AudioGainFilter(float minGain, float maxGain, float objectiveAmplitude)
+    {
+        mAudioGain = new AudioGainAndDcFilter(minGain, maxGain, objectiveAmplitude);
+        mAudioGain.setDecayRate(2);     // to remove pumping effect
+        mAudioGain.reset();
+    }
+
+    @Override
+    public float[] filter(float[] audio)
+    {
+        return mAudioGain.process(audio);
+    }
+}

src/main/java/io/github/dsheirer/audio/AudioModule.java

@@ -32,6 +32,8 @@ import io.github.dsheirer.sample.real.IRealBufferListener;
 import org.slf4j.Logger;
 import org.slf4j.LoggerFactory;
 
+import java.util.List;
+
 /**
  * Provides packaging of demodulated audio sample buffers into audio segments for broadcast to registered listeners.
  * Includes audio packet metadata in constructed audio segments.
@@ -42,36 +44,7 @@ public class AudioModule extends AbstractAudioModule implements ISquelchStateLis
     Listener<float[]>
 {
     private static final Logger mLog = LoggerFactory.getLogger(AudioModule.class);
-    private static float[] sHighPassFilterCoefficients;
-    private final boolean mAudioFilterEnable;
-
-    static
-    {
-        FIRFilterSpecification specification = FIRFilterSpecification.highPassBuilder()
-            .sampleRate(8000)
-            .stopBandCutoff(200)
-            .stopBandAmplitude(0.0)
-            .stopBandRipple(0.025)
-            .passBandStart(300)
-            .passBandAmplitude(1.0)
-            .passBandRipple(0.01)
-            .build();
-        try
-        {
-            RemezFIRFilterDesigner designer = new RemezFIRFilterDesigner(specification);
-
-            if(designer.isValid())
-            {
-                sHighPassFilterCoefficients = designer.getImpulseResponse();
-            }
-        }
-        catch(FilterDesignException fde)
-        {
-            mLog.error("Filter design error", fde);
-        }
-    }
-
-    private final IRealFilter mHighPassFilter = FilterFactory.getRealFilter(sHighPassFilterCoefficients);
+    private List<AbstractAudioFilter>mAudioFilters;
     private final SquelchStateListener mSquelchStateListener = new SquelchStateListener();
     private SquelchState mSquelchState = SquelchState.SQUELCH;
 
@@ -81,23 +54,23 @@ public class AudioModule extends AbstractAudioModule implements ISquelchStateLis
      * @param aliasList for aliasing identifiers
      * @param timeslot for this audio module
      * @param maxAudioSegmentLength in milliseconds
-     * @param audioFilterEnable to enable or disable high-pass audio filter
+     * @param audioFilters a list of audio filters to apply to the incoming audio samples
      */
-    public AudioModule(AliasList aliasList, int timeslot, long maxAudioSegmentLength, boolean audioFilterEnable)
+    public AudioModule(AliasList aliasList, int timeslot, long maxAudioSegmentLength, List<AbstractAudioFilter> audioFilters)
     {
         super(aliasList, timeslot, maxAudioSegmentLength);
-        mAudioFilterEnable = audioFilterEnable;
+        mAudioFilters = audioFilters;
     }
 
     /**
      * Creates an Audio Module.
      * @param aliasList for aliasing identifiers
-     * @param audioFilterEnable to enable or disable high-pass audio filter
+     * @param audioFilters a list of audio filters to apply to the incoming audio samples
      */
-    public AudioModule(AliasList aliasList, boolean audioFilterEnable)
+    public AudioModule(AliasList aliasList,  List<AbstractAudioFilter> audioFilters)
     {
         super(aliasList);
-        mAudioFilterEnable = audioFilterEnable;
+        mAudioFilters = audioFilters;
     }
 
     @Override
@@ -128,9 +101,13 @@ public class AudioModule extends AbstractAudioModule implements ISquelchStateLis
     {
         if(mSquelchState == SquelchState.UNSQUELCH)
         {
-            if(mAudioFilterEnable)
+            // apply audio filters if any (i.e. deemphasis, high pass, etc.)
+            if(!mAudioFilters.isEmpty())
             {
-                audioBuffer = mHighPassFilter.filter(audioBuffer);
+                for(AbstractAudioFilter filter : mAudioFilters)
+                {
+                    audioBuffer = filter.filter(audioBuffer);
+                }
             }
 
             addAudio(audioBuffer);

src/main/java/io/github/dsheirer/audio/HighPassAudioFilter.java

@@ -0,0 +1,50 @@
+package io.github.dsheirer.audio;
+
+import io.github.dsheirer.dsp.filter.FilterFactory;
+import io.github.dsheirer.dsp.filter.design.FilterDesignException;
+import io.github.dsheirer.dsp.filter.fir.FIRFilterSpecification;
+import io.github.dsheirer.dsp.filter.fir.real.IRealFilter;
+import io.github.dsheirer.dsp.filter.fir.remez.RemezFIRFilterDesigner;
+import org.slf4j.Logger;
+import org.slf4j.LoggerFactory;
+
+public class HighPassAudioFilter extends AbstractAudioFilter
+{
+    private static final Logger mLog = LoggerFactory.getLogger(HighPassAudioFilter.class);
+    private static float[] sHighPassFilterCoefficients;
+
+    static
+    {
+        FIRFilterSpecification specification = FIRFilterSpecification.highPassBuilder()
+                .sampleRate(8000)
+                .stopBandCutoff(200)
+                .stopBandAmplitude(0.0)
+                .stopBandRipple(0.025)
+                .passBandStart(300)
+                .passBandAmplitude(1.0)
+                .passBandRipple(0.01)
+                .build();
+        try
+        {
+            RemezFIRFilterDesigner designer = new RemezFIRFilterDesigner(specification);
+
+            if(designer.isValid())
+            {
+                sHighPassFilterCoefficients = designer.getImpulseResponse();
+            }
+        }
+        catch(FilterDesignException fde)
+        {
+            mLog.error("Filter design error", fde);
+        }
+    }
+
+    private final IRealFilter mHighPassFilter = FilterFactory.getRealFilter(sHighPassFilterCoefficients);
+
+    @Override
+    public float[] filter(float[] audio)
+    {
+        audio = mHighPassFilter.filter(audio);
+        return audio;
+    }
+}