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Fixed handling a null input source for some functions, in order to work with the new squelch.

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ha7ilm 2016-03-20 23:30:21 +01:00
parent 4230198d91
commit 42ce9d4669
2 changed files with 21 additions and 23 deletions
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2
libcsdr.c
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@ -741,7 +741,7 @@ complexf fmdemod_quadri_cf(complexf* input, float* output, int input_size, float
} }
for (int i=0; i<input_size; i++) //@fmdemod_quadri_cf: output division for (int i=0; i<input_size; i++) //@fmdemod_quadri_cf: output division
{ {
output[i]=fmdemod_quadri_K*output[i]/temp[i]; output[i]=(temp[i])?fmdemod_quadri_K*output[i]/temp[i]:0;
} }
return input[input_size-1]; return input[input_size-1];

42
libcsdr_gpl.c
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@ -47,7 +47,7 @@ float shift_addition_cc(complexf *input, complexf* output, int input_size, shift
} }
starting_phase+=d.rate*PI*input_size; 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; //@shift_addition_cc: normalize starting_phase
while(starting_phase<-PI) starting_phase+=2*PI; while(starting_phase<-PI) starting_phase+=2*PI;
return starting_phase; return starting_phase;
} }
@ -82,12 +82,12 @@ void shift_addition_cc_test(shift_addition_data_t d)
sinphi=sinphi_last*d.cosdelta+cosphi_last*d.sindelta; sinphi=sinphi_last*d.cosdelta+cosphi_last*d.sindelta;
phi+=d.rate*PI; phi+=d.rate*PI;
while(phi>2*PI) phi-=2*PI; //@shift_addition_cc: normalize phase while(phi>2*PI) phi-=2*PI; //@shift_addition_cc: normalize phase
if(i%SACCTEST_STEP==0) if(i%SACCTEST_STEP==0)
{ {
avg_counter=avg_size; avg_counter=avg_size;
avg=0; avg=0;
} }
if(avg_counter) if(avg_counter)
{ {
avg+=fabs(cosphi-cos(phi)); avg+=fabs(cosphi-cos(phi));
if(!--avg_counter) printf("%g ", avg/avg_size); if(!--avg_counter) printf("%g ", avg/avg_size);
@ -128,7 +128,7 @@ decimating_shift_addition_status_t decimating_shift_addition_cc(complexf *input,
s.starting_phase+=d.rate*PI*k; s.starting_phase+=d.rate*PI*k;
s.output_size=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; //@shift_addition_cc: normalize starting_phase
while(s.starting_phase<-PI) s.starting_phase+=2*PI; while(s.starting_phase<-PI) s.starting_phase+=2*PI;
return s; return s;
} }
@ -142,10 +142,10 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
hang_time = (hang_time_ms / 1000) * sample_rate hang_time = (hang_time_ms / 1000) * sample_rate
hang_time is given in samples, and should be about 4ms. hang_time is given in samples, and should be about 4ms.
hang_time can be switched off by setting it to zero (not recommended). hang_time can be switched off by setting it to zero (not recommended).
max_gain = pow(2, adc_bits) max_gain = pow(2, adc_bits)
max_gain should be no more than the dynamic range of your A/D converter. max_gain should be no more than the dynamic range of your A/D converter.
gain_filter_alpha = 1 / ((fs/(2*PI*fc))+1) gain_filter_alpha = 1 / ((fs/(2*PI*fc))+1)
>>> 1 / ((48000./(2*3.141592654*100))+1) >>> 1 / ((48000./(2*3.141592654*100))+1)
0.012920836043344543 0.012920836043344543
@ -153,7 +153,7 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
0.0013072857061786625 0.0013072857061786625
Literature: Literature:
ww.qsl.net/va3iul/Files/Automatic_Gain_Control.pdf ww.qsl.net/va3iul/Files/Automatic_Gain_Control.pdf
page 7 of http://www.arrl.org/files/file/Technology/tis/info/pdf/021112qex027.pdf page 7 of http://www.arrl.org/files/file/Technology/tis/info/pdf/021112qex027.pdf
@ -170,10 +170,10 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
output[0]=last_gain*input[0]; //we skip this one sample, because it is easier this way 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 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. //The error is the difference between the required gain at the actual sample, and the previous gain value.
//We actually use an envelope detector. //We actually use an envelope detector.
input_abs=fabs(input[i]); input_abs=fabs(input[i]);
error=reference/input_abs-gain; 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. if(input[i]!=0) //We skip samples containing 0, as the gain would be infinity for those to keep up with the reference.
{ {
@ -184,12 +184,12 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
//However, attack_rate should be higher than the decay_rate as we want to avoid clipping signals. //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. //that had a sudden increase in their amplitude.
//It's also important to note that this algorithm has an exponential gain ramp. //It's also important to note that this algorithm has an exponential gain ramp.
if(error<0) //INCREASE IN SIGNAL LEVEL if(error<0) //INCREASE IN SIGNAL LEVEL
{ {
if(last_peak<input_abs) if(last_peak<input_abs)
{ {
attack_wait_counter=attack_wait_time; attack_wait_counter=attack_wait_time;
last_peak=input_abs; last_peak=input_abs;
} }
@ -201,11 +201,11 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
} }
else else
{ {
//If the signal level increases, we decrease the gain quite fast. //If the signal level increases, we decrease the gain quite fast.
dgain=error*attack_rate; dgain=error*attack_rate;
//Before starting to increase the gain next time, we will be waiting until hang_time for sure. //Before starting to increase the gain next time, we will be waiting until hang_time for sure.
hang_counter=hang_time; hang_counter=hang_time;
} }
} }
else //DECREASE IN SIGNAL LEVEL else //DECREASE IN SIGNAL LEVEL
@ -216,13 +216,12 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
dgain=0; //..until then, AGC is inactive and gain doesn't change. 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. else dgain=error*decay_rate; //If the signal level decreases, we increase the gain quite slowly.
} }
gain=gain+dgain; gain=gain+dgain;
//fprintf(stderr,"g=%f dg=%f\n",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;
} }
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. //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: //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]=(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]
@ -230,8 +229,7 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
last_gain=gain; last_gain=gain;
} }
return gain; //this will be the last_gain next time return gain; //this will be the last_gain next time
} }
#endif #endif