Fixed handling a null input source for some functions, in order to work with the new squelch.
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4230198d91
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42ce9d4669
2 changed files with 21 additions and 23 deletions
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@ -741,7 +741,7 @@ complexf fmdemod_quadri_cf(complexf* input, float* output, int input_size, float
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}
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for (int i=0; i<input_size; i++) //@fmdemod_quadri_cf: output division
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{
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output[i]=fmdemod_quadri_K*output[i]/temp[i];
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output[i]=(temp[i])?fmdemod_quadri_K*output[i]/temp[i]:0;
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}
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return input[input_size-1];
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@ -47,7 +47,7 @@ float shift_addition_cc(complexf *input, complexf* output, int input_size, shift
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}
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starting_phase+=d.rate*PI*input_size;
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while(starting_phase>PI) starting_phase-=2*PI; //@shift_addition_cc: normalize starting_phase
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while(starting_phase<-PI) starting_phase+=2*PI;
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while(starting_phase<-PI) starting_phase+=2*PI;
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return starting_phase;
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}
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@ -82,12 +82,12 @@ void shift_addition_cc_test(shift_addition_data_t d)
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sinphi=sinphi_last*d.cosdelta+cosphi_last*d.sindelta;
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phi+=d.rate*PI;
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while(phi>2*PI) phi-=2*PI; //@shift_addition_cc: normalize phase
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if(i%SACCTEST_STEP==0)
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if(i%SACCTEST_STEP==0)
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{
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avg_counter=avg_size;
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avg=0;
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}
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if(avg_counter)
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if(avg_counter)
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{
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avg+=fabs(cosphi-cos(phi));
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if(!--avg_counter) printf("%g ", avg/avg_size);
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@ -128,7 +128,7 @@ decimating_shift_addition_status_t decimating_shift_addition_cc(complexf *input,
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s.starting_phase+=d.rate*PI*k;
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s.output_size=k;
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while(s.starting_phase>PI) s.starting_phase-=2*PI; //@shift_addition_cc: normalize starting_phase
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while(s.starting_phase<-PI) s.starting_phase+=2*PI;
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while(s.starting_phase<-PI) s.starting_phase+=2*PI;
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return s;
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}
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@ -142,10 +142,10 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
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hang_time = (hang_time_ms / 1000) * sample_rate
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hang_time is given in samples, and should be about 4ms.
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hang_time can be switched off by setting it to zero (not recommended).
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max_gain = pow(2, adc_bits)
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max_gain should be no more than the dynamic range of your A/D converter.
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gain_filter_alpha = 1 / ((fs/(2*PI*fc))+1)
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max_gain should be no more than the dynamic range of your A/D converter.
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gain_filter_alpha = 1 / ((fs/(2*PI*fc))+1)
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>>> 1 / ((48000./(2*3.141592654*100))+1)
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0.012920836043344543
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@ -153,7 +153,7 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
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0.0013072857061786625
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Literature:
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Literature:
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ww.qsl.net/va3iul/Files/Automatic_Gain_Control.pdf
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page 7 of http://www.arrl.org/files/file/Technology/tis/info/pdf/021112qex027.pdf
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@ -170,10 +170,10 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
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output[0]=last_gain*input[0]; //we skip this one sample, because it is easier this way
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for(int i=1;i<input_size;i++) //@agc_ff
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{
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//The error is the difference between the required gain at the actual sample, and the previous gain value.
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//The error is the difference between the required gain at the actual sample, and the previous gain value.
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//We actually use an envelope detector.
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input_abs=fabs(input[i]);
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error=reference/input_abs-gain;
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error=reference/input_abs-gain;
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if(input[i]!=0) //We skip samples containing 0, as the gain would be infinity for those to keep up with the reference.
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{
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@ -184,12 +184,12 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
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//However, attack_rate should be higher than the decay_rate as we want to avoid clipping signals.
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//that had a sudden increase in their amplitude.
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//It's also important to note that this algorithm has an exponential gain ramp.
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if(error<0) //INCREASE IN SIGNAL LEVEL
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{
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if(last_peak<input_abs)
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if(last_peak<input_abs)
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{
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attack_wait_counter=attack_wait_time;
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last_peak=input_abs;
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}
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@ -201,11 +201,11 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
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}
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else
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{
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//If the signal level increases, we decrease the gain quite fast.
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dgain=error*attack_rate;
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//If the signal level increases, we decrease the gain quite fast.
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dgain=error*attack_rate;
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//Before starting to increase the gain next time, we will be waiting until hang_time for sure.
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hang_counter=hang_time;
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}
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}
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else //DECREASE IN SIGNAL LEVEL
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@ -216,13 +216,12 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
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dgain=0; //..until then, AGC is inactive and gain doesn't change.
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}
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else dgain=error*decay_rate; //If the signal level decreases, we increase the gain quite slowly.
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}
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}
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gain=gain+dgain;
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//fprintf(stderr,"g=%f dg=%f\n",gain,dgain);
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if(gain>max_gain) gain=max_gain; //We also have to limit our gain, it can't be infinity.
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if(gain<0) gain=0;
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}
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if(gain>max_gain) gain=max_gain; //We also have to limit our gain, it can't be infinity.
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if(gain<0) gain=0;
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//output[i]=gain*input[i]; //Here we do the actual scaling of the samples.
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//Here we do the actual scaling of the samples, but we run an IIR filter on the gain values:
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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]
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@ -230,8 +229,7 @@ float agc_ff(float* input, float* output, int input_size, float reference, float
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last_gain=gain;
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}
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return gain; //this will be the last_gain next time
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return gain; //this will be the last_gain next time
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}
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#endif
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