Automatic Gain On Analog WIP; Manual Gain Control;

include/dsd.h

@@ -983,6 +983,8 @@ void playSynthesizedVoiceFS3 (dsd_opts * opts, dsd_state * state); //float stere
 void playSynthesizedVoiceFS4 (dsd_opts * opts, dsd_state * state); //float stereo mix 4v2 P25p2
 void playSynthesizedVoiceFM (dsd_opts * opts, dsd_state * state);  //float mono
 void agf (dsd_opts * opts, dsd_state * state, float samp[160], int slot); //float gain control
+void agsm (dsd_opts * opts, dsd_state * state, short * input, int len); //short gain control for analog things
+void analog_gain (dsd_opts * opts, dsd_state * state, short * input, int len); //manual gain handling for analong things
 //new short stuff
 void playSynthesizedVoiceSS (dsd_opts * opts, dsd_state * state);   //short stereo mix
 void playSynthesizedVoiceSS3 (dsd_opts * opts, dsd_state * state);  //short stereo mix 3v2 DMR

src/3.c

@@ -526,9 +526,7 @@ void pbf(dsd_state * state, short * input, int len)
   for (i = 0; i < len; i++)
 	{
 		// fprintf (stderr, "\n in: %05d", input[i]);
-		// input[i] = PBFilter_Update(&state->PBF, input[i]); //filter only
-		input[i] = PBFilter_Update(&state->PBF, input[i]) * 1.5f; //add a little extra gain back
-		// fprintf (stderr, "\n out: %05d", input[i]);
+		input[i] = PBFilter_Update(&state->PBF, input[i]);
 	}
 }
 

src/dsd_audio2.c

@@ -1409,4 +1409,87 @@ void agf (dsd_opts * opts, dsd_state * state, float samp[160], int slot)
   
 //   AGF_END: ; //do nothing
 
-// }
+// }
+
+//automatic gain short mono for analog audio and some digital mono (WIP)
+void agsm (dsd_opts * opts, dsd_state * state, short * input, int len)
+{
+  int i, j;
+
+  UNUSED(state);
+
+  float avg = 0.0f;        //will float be suffient?
+  float coeff = 0.0f;     //gain coeffiecient
+  float max = 0.0f;      //the highest sample value he wave
+  float den = 1500.0f;  //denominator value
+  float gain = (25.0f / opts->audio_gain); UNUSED(gain);
+  float samp[960]; memset (samp, 0.0f, 960*sizeof(float));
+
+  //assign internal float from short input
+  for (i = 0; i < len; i++)
+    samp[i] = (float)input[i];
+
+  // gain = 2.5f;
+  // for (i = 0; i < len; i++)
+  //   samp[i] *= gain;
+
+  for (j = 0; j < len/20; j++)
+  {
+
+    for (i = 0; i < 20; i++)
+    {
+      if ( fabsf (samp[(j*20)+i]) > max)
+      {
+        max = fabsf (samp[(j*20)+i]);
+      }
+    }
+
+    for (i = 0; i < 20; i++)
+      avg += (float)samp[(j*20)+i];
+
+    avg /= 20;
+    
+    coeff = fabsf (den / max);
+
+
+    //apply the coefficient to bring the max value to our desired maximum value
+    for (i = 0; i < 20; i++)
+    {
+      if ( fabsf (samp[(j*20)+i]) > 110.0f) //try not to apply to near zero values
+        samp[(j*20)+i] *= coeff;
+    }
+    
+    //debug
+    // fprintf (stderr, "\n M: %f; C: %f; A: %f; ", max, coeff, avg);
+
+    max = 0; //reset
+    avg = 0; //reset
+
+  } //end j loop
+
+  // debug
+  // for (i = 0; i < len; i++)
+  // {
+  //   fprintf (stderr, " in: %d", input[i]);
+  //   fprintf (stderr, " out: %f", samp[i]);
+  // }
+    
+  //return new smaple values post agc
+  for (i = 0; i < len; i++)
+    input[i] = (short)samp[i];
+
+}
+
+//until analog agc is fixed, going to use a manual gain control on this
+//that scales a bit differently on the 1-50 index values
+void analog_gain (dsd_opts * opts, dsd_state * state, short * input, int len)
+{
+
+  int i;
+  UNUSED(state);
+
+  float gain = 1.0f + (opts->audio_gain / 25.0f); //range is 1 - 3?
+
+  for (i = 0; i < len; i++)
+    input[i] *= gain;
+}

src/dsd_symbol.c

@@ -278,13 +278,15 @@ getSymbol (dsd_opts * opts, dsd_state * state, int have_sync)
             sf_write_sync (opts->wav_out_raw);
           }
 
+          //manual gain control -- seems to work better if applied before filtering
+          analog_gain (opts, state, state->analog_out, 960);
+
           // analog audio filtering
           // lpf(state, state->analog_out, 960);
           hpf (state, state->analog_out, 960);
 
-          //analog sounds really good now, still a tad bit quiet though
-          if (opts->audio_in_type == 3) //passband filter if using rtl_fm
-            pbf(state, state->analog_out, 960);
+          //analog sounds really good now
+          pbf(state, state->analog_out, 960);
 
           //unsure if this one helps anymore or not, probably not be setup correctly
           // if (opts->audio_in_type == 3) //notch filter if using rtl_fm

src/edacs-fme.c

@@ -202,6 +202,11 @@ void edacs_analog(dsd_opts * opts, dsd_state * state, int afs, unsigned char lcn
       sr += digitize (opts, state, (int)analog1[i]);
     }
 
+    //manual gain control
+    analog_gain (opts, state, analog1, 960);
+    analog_gain (opts, state, analog2, 960);
+    analog_gain (opts, state, analog3, 960);
+
     // analog audio filtering
     // lpf (state, analog1, 960);
     // lpf (state, analog2, 960);
@@ -211,13 +216,10 @@ void edacs_analog(dsd_opts * opts, dsd_state * state, int afs, unsigned char lcn
     hpf (state, analog2, 960);
     hpf (state, analog3, 960);
 
-    //TODO: Test this somehow, the EDACS analog remote only ATM (Should be alright though)
-    if (opts->audio_in_type == 3)
-    {
-      pbf (state, analog1, 960);
-      pbf (state, analog2, 960);
-      pbf (state, analog3, 960);
-    }
+    pbf (state, analog1, 960);
+    pbf (state, analog2, 960);
+    pbf (state, analog3, 960);
+
 
     //reconfigured to use seperate audio out stream that is always 48k short
     if (opts->audio_out_type == 0 && opts->slot1_on == 1)