mirror of
https://github.com/DSheirer/sdrtrunk.git
synced 2026-11-12 08:37:25 -05:00
f4d8ea281d
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
78 lines
2.3 KiB
Java
78 lines
2.3 KiB
Java
/*******************************************************************************
|
|
* SDR Trunk
|
|
* Copyright (C) 2014,2015 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 dsp.fm;
|
|
|
|
public abstract class FMDemodulator
|
|
{
|
|
private float mPreviousI = 0.0f;
|
|
private float mPreviousQ = 0.0f;
|
|
protected float mGain;
|
|
|
|
public FMDemodulator( float gain )
|
|
{
|
|
mGain = gain;
|
|
}
|
|
|
|
public float demodulate( float currentI, float currentQ )
|
|
{
|
|
/**
|
|
* Multiply the current sample against the complex conjugate of the
|
|
* previous sample to derive the phase delta between the two samples
|
|
*
|
|
* Negating the previous sample quadrature produces the conjugate
|
|
*/
|
|
double inphase = ( currentI * mPreviousI ) - ( currentQ * -mPreviousQ );
|
|
double quadrature = ( currentQ * mPreviousI ) + ( currentI * -mPreviousQ );
|
|
|
|
double angle = 0.0f;
|
|
|
|
//Check for divide by zero
|
|
if( inphase != 0 )
|
|
{
|
|
/**
|
|
* Use the arc-tangent of quadrature divided by inphase to
|
|
* get the phase angle (+/-) which was directly manipulated by the
|
|
* original message waveform during the modulation. This value now
|
|
* serves as the instantaneous amplitude of the demodulated signal
|
|
*/
|
|
double denominator = 1.0d / inphase;
|
|
angle = Math.atan( (double)quadrature * denominator );
|
|
}
|
|
|
|
/**
|
|
* Store the current sample to use during the next iteration
|
|
*/
|
|
mPreviousI = currentI;
|
|
mPreviousQ = currentQ;
|
|
|
|
return (float)( angle * mGain );
|
|
}
|
|
|
|
public abstract void dispose();
|
|
|
|
public void reset()
|
|
{
|
|
mPreviousI = 0.0f;
|
|
mPreviousQ = 0.0f;
|
|
}
|
|
|
|
public void setGain( float gain )
|
|
{
|
|
mGain = gain;
|
|
}
|
|
} |