csdr/libcsdr_gpl.c

/*
This file is part of libcsdr.

	Copyright (c) Andras Retzler, HA7ILM <randras@sdr.hu>
	Copyright (c) Warren Pratt, NR0V <warren@wpratt.com>
	Copyright 2006,2010,2012 Free Software Foundation, Inc.

    libcsdr 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.

    libcsdr 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 libcsdr.  If not, see <http://www.gnu.org/licenses/>.

*/

#include "libcsdr_gpl.h"

#ifdef LIBCSDR_GPL

float shift_addition_cc(complexf *input, complexf* output, int input_size, shift_addition_data_t d, float starting_phase)
{
	//The original idea was taken from wdsp:
	//http://svn.tapr.org/repos_sdr_hpsdr/trunk/W5WC/PowerSDR_HPSDR_mRX_PS/Source/wdsp/shift.c

	//However, this method introduces noise (from floating point rounding errors), which increases until the end of the buffer.
	//fprintf(stderr, "cosd=%g sind=%g\n", d.cosdelta, d.sindelta);
	float cosphi=cos(starting_phase);
	float sinphi=sin(starting_phase);
	float cosphi_last, sinphi_last;
	for(int i=0;i<input_size;i++) //@shift_addition_cc: work
	{
		iof(output,i)=cosphi*iof(input,i)-sinphi*qof(input,i);
		qof(output,i)=sinphi*iof(input,i)+cosphi*qof(input,i);
		//using the trigonometric addition formulas
		//cos(phi+delta)=cos(phi)cos(delta)-sin(phi)*sin(delta)
		cosphi_last=cosphi;
		sinphi_last=sinphi;
		cosphi=cosphi_last*d.cosdelta-sinphi_last*d.sindelta;
		sinphi=sinphi_last*d.cosdelta+cosphi_last*d.sindelta;
	}
	starting_phase+=d.rate*PI*input_size;
	while(starting_phase>PI) starting_phase-=2*PI; //@shift_addition_cc: normalize starting_phase
	while(starting_phase<-PI) starting_phase+=2*PI;
	return starting_phase;
}

shift_addition_data_t shift_addition_init(float rate)
{
	rate*=2;
	shift_addition_data_t out;
	out.sindelta=sin(rate*PI);
	out.cosdelta=cos(rate*PI);
	out.rate=rate;
	return out;
}

#define SACCTEST_LOOPS 50
#define SACCTEST_STEP 10000

void shift_addition_cc_test(shift_addition_data_t d)
{
	float phi=0;
	float cosphi=cos(phi);
	float sinphi=sin(phi);
	float cosphi_last, sinphi_last;
	int avg_size=(int)(2.0/d.rate+1.0); //average one period of sine
	int avg_counter=0;
	float avg=0;
	printf("error_vector=[");
	for(unsigned i=0;i<SACCTEST_STEP*SACCTEST_LOOPS;i++) //@shift_addition_cc: work
	{
		cosphi_last=cosphi;
		sinphi_last=sinphi;
		cosphi=cosphi_last*d.cosdelta-sinphi_last*d.sindelta;
		sinphi=sinphi_last*d.cosdelta+cosphi_last*d.sindelta;
		phi+=d.rate*PI;
		while(phi>2*PI) phi-=2*PI; //@shift_addition_cc: normalize phase
		if(i%SACCTEST_STEP==0)
		{
			avg_counter=avg_size;
			avg=0;
		}
		if(avg_counter)
		{
			avg+=fabs(cosphi-cos(phi));
			if(!--avg_counter) printf("%g ", avg/avg_size);
		}
	}
	printf("]; error_vector_db=20*log10(error_vector); plot(error_vector_db);\n");
}

shift_addition_data_t decimating_shift_addition_init(float rate, int decimation)
{
	return shift_addition_init(rate*decimation);
}

decimating_shift_addition_status_t decimating_shift_addition_cc(complexf *input, complexf* output, int input_size, shift_addition_data_t d, int decimation, decimating_shift_addition_status_t s)
{
	//The original idea was taken from wdsp:
	//http://svn.tapr.org/repos_sdr_hpsdr/trunk/W5WC/PowerSDR_HPSDR_mRX_PS/Source/wdsp/shift.c
	//However, this method introduces noise (from floating point rounding errors), which increases until the end of the buffer.
	//fprintf(stderr, "cosd=%g sind=%g\n", d.cosdelta, d.sindelta);
	float cosphi=cos(s.starting_phase);
	float sinphi=sin(s.starting_phase);
	float cosphi_last, sinphi_last;
	int i;
	int k=0;
	for(i=s.decimation_remain;i<input_size;i+=decimation) //@shift_addition_cc: work
	{
		iof(output,k)=cosphi*iof(input,i)-sinphi*qof(input,i);
		qof(output,k)=sinphi*iof(input,i)+cosphi*qof(input,i);
		k++;
		//using the trigonometric addition formulas
		//cos(phi+delta)=cos(phi)cos(delta)-sin(phi)*sin(delta)
		cosphi_last=cosphi;
		sinphi_last=sinphi;
		cosphi=cosphi_last*d.cosdelta-sinphi_last*d.sindelta;
		sinphi=sinphi_last*d.cosdelta+cosphi_last*d.sindelta;
	}
	s.decimation_remain=i-input_size;
	s.starting_phase+=d.rate*PI*k;
	s.output_size=k;
	while(s.starting_phase>PI) s.starting_phase-=2*PI; //@shift_addition_cc: normalize starting_phase
	while(s.starting_phase<-PI) s.starting_phase+=2*PI;
	return s;
}


float agc_ff(float* input, float* output, int input_size, float reference, float attack_rate, float decay_rate, float max_gain, short hang_time, short attack_wait_time, float gain_filter_alpha, float last_gain)
{
	/*
		Notes on parameters (with some default values):
			attack_rate = 0.01
			decay_rate = 0.001
			hang_time = (hang_time_ms / 1000) * sample_rate
				hang_time is given in samples, and should be about 4ms.
				hang_time can be switched off by setting it to zero (not recommended).

			max_gain = pow(2, adc_bits)
				max_gain should be no more than the dynamic range of your A/D converter.
			gain_filter_alpha = 1 / ((fs/(2*PI*fc))+1)

			>>> 1 / ((48000./(2*3.141592654*100))+1)
			0.012920836043344543
			>>> 1 / ((48000./(2*3.141592654*10))+1)
			0.0013072857061786625


		Literature:
			ww.qsl.net/va3iul/Files/Automatic_Gain_Control.pdf
			page 7 of http://www.arrl.org/files/file/Technology/tis/info/pdf/021112qex027.pdf

		Examples:
			http://svn.tapr.org/repos_sdr_hpsdr/trunk/W5WC/PowerSDR_HPSDR_mRX_PS/Source/wdsp/wcpAGC.c
			GNU Radio's agc,agc2,agc3 have quite good ideas about this.
	*/
	register short hang_counter=0;
	register short attack_wait_counter=0;
	float gain=last_gain;
	float last_peak=reference/last_gain; //approx.
	float input_abs;
	float error, dgain;
	output[0]=last_gain*input[0]; //we skip this one sample, because it is easier this way
	for(int i=1;i<input_size;i++) //@agc_ff
	{
		//The error is the difference between the required gain at the actual sample, and the previous gain value.
		//We actually use an envelope detector.
		input_abs=fabs(input[i]);
		error=reference/input_abs-gain;

		if(input[i]!=0) //We skip samples containing 0, as the gain would be infinity for those to keep up with the reference.
		{
			//An AGC is something nonlinear that's easier to implement in software:
			//if the amplitude decreases, we increase the gain by minimizing the gain error by attack_rate.
			//We also have a decay_rate that comes into consideration when the amplitude increases.
			//The higher these rates are, the faster is the response of the AGC to amplitude changes.
			//However, attack_rate should be higher than the decay_rate as we want to avoid clipping signals.
			//that had a sudden increase in their amplitude.
			//It's also important to note that this algorithm has an exponential gain ramp.

			if(error<0) //INCREASE IN SIGNAL LEVEL
			{
				if(last_peak<input_abs)
				{

					attack_wait_counter=attack_wait_time;
					last_peak=input_abs;
				}
				if(attack_wait_counter>0)
				{
					attack_wait_counter--;
					//fprintf(stderr,"A");
					dgain=0;
				}
				else
				{
					//If the signal level increases, we decrease the gain quite fast.
					dgain=error*attack_rate;
					//Before starting to increase the gain next time, we will be waiting until hang_time for sure.
					hang_counter=hang_time;

				}
			}
			else //DECREASE IN SIGNAL LEVEL
			{
				if(hang_counter>0) //Before starting to increase the gain, we will be waiting until hang_time.
				{
					hang_counter--;
					dgain=0; //..until then, AGC is inactive and gain doesn't change.
				}
				else dgain=error*decay_rate; //If the signal level decreases, we increase the gain quite slowly.
			}
			gain=gain+dgain;
			//fprintf(stderr,"g=%f dg=%f\n",gain,dgain);
		}
		if(gain>max_gain) gain=max_gain; //We also have to limit our gain, it can't be infinity.
		if(gain<0) gain=0;
		//output[i]=gain*input[i]; //Here we do the actual scaling of the samples.
		//Here we do the actual scaling of the samples, but we run an IIR filter on the gain values:
		output[i]=(gain=gain+last_gain-gain_filter_alpha*last_gain)*input[i]; //dc-pass-filter: freqz([1 -1],[1 -0.99]) y[i]=x[i]+y[i-1]-alpha*x[i-1]
		//output[i]=input[i]*(last_gain+gain_filter_alpha*(gain-last_gain)); //LPF

		last_gain=gain;
	}
	return gain; //this will be the last_gain next time
}

#endif