Initial commit of redesigned internal processing and audio management.

Note: Several of the decoders are currently non-functioning in this
commit.  If you want the latest functioning codebase, please checkout
the commit prior to this one.  All decoders will eventually be updated
with this new architecture.

Currently broken:
- Logging and recording
- Aux decoders (Fleetsync/MDC/LoJack/Tait)
- LTR/Passport decoders

Changes in this commit:
- Module architecture for decoders and components with central
provisioning by the DecoderFactory
- Transition from Complex sample stream processing to float buffer
processing for improved memory management
- Redesigned filter structures - to support buffers and streams
- Resolved several memory leak issues
- Universal sharing of the thread pool manager
- Added centralized audio management with call/user prioritization and
new gui with mute/volume controls
- Embedded audio metadata for eventual streaming support
- Redesigned ProcessingChain to support independent re-use
- Added new TrafficChannelManager to handle traffic channel allocations
- Traffic Channel pool size and Call timeout available in P25 and
MPT1327 configs
- Redesigned channel and decoder gui components

src/audio/output/MonoAudioOutput.java

@@ -0,0 +1,260 @@
+package audio.output;
+
+import java.nio.ByteBuffer;
+import java.nio.ByteOrder;
+import java.nio.ShortBuffer;
+import java.util.ArrayList;
+import java.util.List;
+import java.util.concurrent.Executors;
+import java.util.concurrent.LinkedTransferQueue;
+import java.util.concurrent.ScheduledExecutorService;
+import java.util.concurrent.TimeUnit;
+import java.util.concurrent.atomic.AtomicBoolean;
+
+import javax.sound.sampled.AudioSystem;
+import javax.sound.sampled.BooleanControl;
+import javax.sound.sampled.FloatControl;
+import javax.sound.sampled.LineUnavailableException;
+import javax.sound.sampled.SourceDataLine;
+
+import org.slf4j.Logger;
+import org.slf4j.LoggerFactory;
+
+import audio.AudioEvent;
+import audio.AudioFormats;
+import audio.AudioPacket;
+import audio.AudioPacket.Type;
+import controller.NamingThreadFactory;
+
+/**
+ * Audio output with automatic flow control based on the availability of 
+ * audio data packets.  
+ */
+public class MonoAudioOutput extends AudioOutput
+{
+	private final static Logger mLog = LoggerFactory.getLogger( MonoAudioOutput.class );
+
+	/* Output line buffer size is set at one second of audio at 48 kHz and
+	 * two bytes per sample */
+	private final static int BUFFER_SIZE = 48000;
+	
+	/* The queue processor will run every 20 milliseconds checking for inbound
+	 * audio and automatically starting or stopping audio playback.  We set the 
+	 * data line available space threshold to start when we're 20 ms away from 
+	 * being full, and the stop threshold when 20 ms away from being empty */
+	private final static int SAMPLE_SIZE_40_MS = 48000 / 25;
+	private final static int START_THRESHOLD = SAMPLE_SIZE_40_MS;
+	private final static int STOP_THRESHOLD = BUFFER_SIZE - SAMPLE_SIZE_40_MS;
+	
+	private LinkedTransferQueue<AudioPacket> mBuffer = new LinkedTransferQueue<>();
+
+	private ScheduledExecutorService mExecutorService;
+	
+	private SourceDataLine mOutput;
+	private FloatControl mGainControl;
+	private BooleanControl mMuteControl;
+	
+	private boolean mCanProcessAudio;
+	
+	private long mLastBufferReceived;
+	
+	private AudioEvent mAudioStartEvent;
+	private AudioEvent mAudioStopEvent;
+	
+	public MonoAudioOutput()
+	{
+		super();
+		
+		mAudioStartEvent = new AudioEvent( AudioEvent.Type.AUDIO_STARTED, getChannelName() );
+		mAudioStopEvent = new AudioEvent( AudioEvent.Type.AUDIO_STOPPED, getChannelName() );
+
+		try
+		{
+			mOutput = AudioSystem.getSourceDataLine( 
+						AudioFormats.PCM_SIGNED_48KHZ_16BITS );
+
+			if( mOutput != null )
+			{
+				mOutput.open( AudioFormats.PCM_SIGNED_48KHZ_16BITS, BUFFER_SIZE );
+				
+				mCanProcessAudio = true;
+				
+				mGainControl = (FloatControl)mOutput.getControl( 
+						FloatControl.Type.MASTER_GAIN );
+				
+				mMuteControl = (BooleanControl)mOutput.getControl( 
+						BooleanControl.Type.MUTE );
+				
+				mExecutorService = Executors.newSingleThreadScheduledExecutor( 
+						new NamingThreadFactory( "audio (mono) output" ) );
+
+				/* Run the queue processor task every 40 milliseconds */
+				mExecutorService.scheduleAtFixedRate( new QueueProcessor(), 
+						0, 40, TimeUnit.MILLISECONDS );
+			}
+		} 
+        catch ( LineUnavailableException e )
+		{
+        	mLog.error( "Couldn't obtain source data line for 48kHz PCM mono "
+        			+ "audio output" );
+		}
+	}
+
+	@Override
+	public String getChannelName()
+	{
+		return "Mono";
+	}
+
+	/**
+	 * Timestamp when most recent buffer was received
+	 */
+	public long getLastBufferReceived()
+	{
+		return mLastBufferReceived;
+	}
+
+	public void dispose()
+	{
+		super.dispose();
+		
+		mCanProcessAudio = false;
+
+		mBuffer.clear();
+		
+		if( mExecutorService != null )
+		{
+			mExecutorService.shutdown();
+		}
+		
+		if( mOutput != null )
+		{
+			mOutput.close();
+		}
+	}
+
+	@Override
+	public void receive( AudioPacket packet )
+	{
+		if( mCanProcessAudio )
+		{
+			mLastBufferReceived = System.currentTimeMillis();
+			
+			mBuffer.add( packet );
+		}
+	}
+	
+	
+	public class QueueProcessor implements Runnable
+	{
+		private AtomicBoolean mProcessing = new AtomicBoolean();
+		
+		public QueueProcessor()
+		{
+		}
+
+		@Override
+		public void run()
+		{
+			/* The mProcessing flag ensures that only one instance of the
+			 * processor can run at any given time */
+			if( mProcessing.compareAndSet( false, true ) )
+			{
+				List<AudioPacket> packets = new ArrayList<AudioPacket>();
+				
+				mBuffer.drainTo( packets );
+				
+				if( packets.size() > 0 )
+				{
+					for( AudioPacket packet: packets )
+					{
+						if( mCanProcessAudio )
+						{
+							if( isSquelched() )
+							{
+								checkStop();
+							}
+							else if( packet.getType() == Type.AUDIO )
+							{
+								float[] samples = packet.getAudioData();
+								
+								/* Little-endian byte buffer */
+								ByteBuffer buffer = 
+									ByteBuffer.allocate( samples.length * 2 )
+											.order( ByteOrder.LITTLE_ENDIAN );
+								
+								ShortBuffer shortBuffer = buffer.asShortBuffer();
+
+								for( float sample: samples )
+								{
+									shortBuffer.put( (short)( sample * Short.MAX_VALUE ) );
+								}
+								
+								mOutput.write( buffer.array(), 0, 
+										buffer.array().length );
+								
+								checkStart();
+								
+								broadcast( packet.getMetadata() );
+							}
+						}
+					}
+				}
+				else
+				{
+					checkStop();
+				}
+				
+				mProcessing.set( false );
+			}
+		}
+		
+		private void checkStart()
+		{
+			if( mCanProcessAudio && !isSquelched() && 
+				!mOutput.isRunning() && mOutput.available() <= START_THRESHOLD )
+			{
+				mOutput.start();
+				
+				mAudioEventBroadcaster.broadcast( mAudioStartEvent );
+			}
+		}
+		
+		private void checkStop()
+		{
+			if( mCanProcessAudio && mOutput.isRunning() &&
+				( mOutput.available() >= STOP_THRESHOLD || isSquelched() ) )
+			{
+				mOutput.drain();
+				mOutput.stop();
+
+				mAudioEventBroadcaster.broadcast( mAudioStopEvent );
+			}
+		}
+	}
+
+
+	@Override
+	public BooleanControl getMuteControl()
+	{
+		return mMuteControl;
+	}
+	
+	@Override
+	public boolean hasMuteControl()
+	{
+		return mMuteControl != null;
+	}
+
+	@Override
+	public FloatControl getGainControl()
+	{
+		return mGainControl;
+	}
+
+	@Override
+	public boolean hasGainControl()
+	{
+		return mGainControl != null;
+	}
+}