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[WIP] Updates the LTR-Standard and LTR-Net decoders to use a new LTR

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decoder with a zero-crossing symbol timing error detector.  The new
decoder is optimized for 8 kHz FM demodulated audio produced by the
AudioModule.
This commit is contained in:
Denny
2018-06-02 08:47:48 -04:00
parent c13315f731
commit a898be265f
64 changed files with 2660 additions and 1285 deletions
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143
src/main/java/io/github/dsheirer/audio/AudioModule.java
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@@ -1,19 +1,16 @@
/*******************************************************************************
* sdrtrunk
* Copyright (C) 2014-2017 Dennis Sheirer
* sdr-trunk
* Copyright (C) 2014-2018 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 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.
* 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/>
* 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;
@@ -23,90 +20,76 @@ import io.github.dsheirer.audio.squelch.SquelchState;
import io.github.dsheirer.channel.metadata.Metadata;
import io.github.dsheirer.dsp.filter.design.FilterDesignException;
import io.github.dsheirer.dsp.filter.fir.FIRFilterSpecification;
import io.github.dsheirer.dsp.filter.fir.real.RealFIRFilter2;
import io.github.dsheirer.dsp.filter.fir.remez.RemezFIRFilterDesigner;
import io.github.dsheirer.dsp.filter.polyphase.PolyphaseFIRDecimatingFilter_RB;
import io.github.dsheirer.module.Module;
import io.github.dsheirer.sample.Listener;
import io.github.dsheirer.sample.real.IFilteredRealBufferListener;
import io.github.dsheirer.sample.real.RealBuffer;
import io.github.dsheirer.sample.buffer.IReusableBufferListener;
import io.github.dsheirer.sample.buffer.ReusableBuffer;
import org.slf4j.Logger;
import org.slf4j.LoggerFactory;
/**
* Provides packaging of demodulated audio sample buffers into audio packets for
* broadcast to registered audio packet listeners. Includes audio packet
* metadata in constructed audio packets.
* Provides packaging of demodulated audio sample buffers into audio packets for broadcast to registered audio packet
* listeners. Includes audio packet metadata in constructed audio packets.
*
* Incorporates audio squelch state listener to control if audio packets are
* broadcast or ignored.
* Incorporates audio squelch state listener to control if audio packets are broadcast or ignored.
*
* This class is designed to support 8 kHz sample rate demodulated audio.
*/
public class AudioModule extends Module implements IAudioPacketProvider, IFilteredRealBufferListener,
ISquelchStateListener, Listener<RealBuffer>
public class AudioModule extends Module implements IAudioPacketProvider, IReusableBufferListener,
ISquelchStateListener, Listener<ReusableBuffer>
{
protected static final Logger mLog = LoggerFactory.getLogger(AudioModule.class);
private static float[] mDecimationCoefficients;
private static float[] sHighPassFilterCoefficients;
static
{
//Band-pass filter to both decimate (48kHz to 8kHz) and block 0 - 300 Hz (LTR signalling).
FIRFilterSpecification specification = FIRFilterSpecification.bandPassBuilder()
.sampleRate(48000)
.stopFrequency1(150)
.passFrequencyBegin(300)
.passFrequencyEnd(3500)
.stopFrequency2(4000)
.stopRipple(0.0004)
.passRipple(0.008)
FIRFilterSpecification specification = FIRFilterSpecification.highPassBuilder()
.sampleRate(8000)
.stopBandCutoff(200)
.stopBandAmplitude(0.0)
.stopBandRipple(0.025)
.passBandStart(300)
.passBandAmplitude(1.0)
.passBandRipple(0.01)
.build();
RemezFIRFilterDesigner designer = new RemezFIRFilterDesigner(specification);
try
{
if(!designer.isValid())
{
throw new FilterDesignException("Couldn't design the audio decimation filter");
}
RemezFIRFilterDesigner designer = new RemezFIRFilterDesigner(specification);
mDecimationCoefficients = designer.getImpulseResponse();
if(designer.isValid())
{
sHighPassFilterCoefficients = designer.getImpulseResponse();
}
}
catch(FilterDesignException e)
catch(FilterDesignException fde)
{
mLog.debug("Error designing filter", e);
mLog.error("Filter design error", fde);
}
}
private RealFIRFilter2 mHighPassFilter = new RealFIRFilter2(sHighPassFilterCoefficients);
private SquelchStateListener mSquelchStateListener = new SquelchStateListener();
private SquelchState mSquelchState = SquelchState.SQUELCH;
private Listener<AudioPacket> mAudioPacketListener;
private PolyphaseFIRDecimatingFilter_RB mAudioDecimationFilter;
private Metadata mMetadata;
/**
* Creates an Audio Module.
* @param metadata to use for audio packets produced by this audio module.
*/
public AudioModule(Metadata metadata)
{
mMetadata = metadata;
mAudioDecimationFilter = new PolyphaseFIRDecimatingFilter_RB(mDecimationCoefficients, 6, 2.0f);
mAudioDecimationFilter.setListener(new Listener<RealBuffer>()
{
@Override
public void receive(RealBuffer realBuffer)
{
if(mAudioPacketListener != null)
{
AudioPacket packet = new AudioPacket(realBuffer.getSamples(), mMetadata.copyOf());
mAudioPacketListener.receive(packet);
}
}
});
}
@Override
public void dispose()
{
mSquelchStateListener = null;
mAudioPacketListener = null;
mSquelchStateListener = null;
}
@Override
@@ -123,32 +106,13 @@ public class AudioModule extends Module implements IAudioPacketProvider, IFilter
@Override
public void stop()
{
/* Issue an end audio packet in case a recorder is still rolling */
/* Issue an end-audio packet in case a recorder is still rolling */
if(mAudioPacketListener != null)
{
mAudioPacketListener.receive(new AudioPacket(AudioPacket.Type.END, mMetadata.copyOf()));
}
}
/**
* Processes demodulated audio samples into audio packets with current audio
* metadata and sends to the registered listener
*/
@Override
public void receive(RealBuffer buffer)
{
if(mAudioPacketListener != null && mSquelchState == SquelchState.UNSQUELCH)
{
mAudioDecimationFilter.receive(buffer);
}
}
@Override
public Listener<RealBuffer> getFilteredRealBufferListener()
{
return this;
}
@Override
public void setAudioPacketListener(Listener<AudioPacket> listener)
{
@@ -167,6 +131,29 @@ public class AudioModule extends Module implements IAudioPacketProvider, IFilter
return mSquelchStateListener;
}
@Override
public void receive(ReusableBuffer reusableBuffer)
{
if(mAudioPacketListener != null && mSquelchState == SquelchState.UNSQUELCH)
{
ReusableBuffer highPassFiltered = mHighPassFilter.filter(reusableBuffer);
AudioPacket packet = new AudioPacket(highPassFiltered.getSamplesCopy(), mMetadata.copyOf());
mAudioPacketListener.receive(packet);
highPassFiltered.decrementUserCount();
}
else
{
reusableBuffer.decrementUserCount();
}
}
@Override
public Listener getReusableBufferListener()
{
//Redirect received reusable buffers to the receive(buffer) method
return this;
}
/**
* Wrapper for squelch state listener
*/