diff --git a/src/3.c b/src/3.c
index 4f3d907..ca974cf 100644
--- a/src/3.c
+++ b/src/3.c
@@ -471,11 +471,13 @@ void init_audio_filters (dsd_state * state)
   HPFilter_Init(&state->HRCFilterR, 960, (float)1/(float)48000);
 
 	//PBFilter_Init(PBFilter *filter, float HPF_cutoffFreqHz, float LPF_cutoffFreqHz, float sampleTimeS);
-	// void NOTCHFilter_Init(NOTCHFilter *filter, float centerFreqHz, float notchWidthHz, float sampleTimeS);
+	//NOTCHFilter_Init(NOTCHFilter *filter, float centerFreqHz, float notchWidthHz, float sampleTimeS);
 
-	//unclear what we should set up for these, output is super quiet after using them
-	PBFilter_Init(&state->PBF, 48000, 12000, (float)1/(float)48000);
-	NOTCHFilter_Init(&state->NF, 48000, 12000, (float)1/(float)48000);
+	//TODO: Make PBFilter initiazation values based on rtl bandwidth values?
+
+	//passband filter working (seems to be), notch filter unsure which values to use
+	PBFilter_Init(&state->PBF, 8000, 12000, (float)1/(float)1536000); //RTL Sampling at 1536000 S/s.
+	NOTCHFilter_Init(&state->NF, 1000, 4000, (float)1/(float)1536000); //this one probably isn't set up correctly
 	
 }
 
@@ -524,7 +526,8 @@ 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]);
+		// 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]);
 	}
 }
diff --git a/src/dsd_symbol.c b/src/dsd_symbol.c
index 7c74ec5..3e7523d 100644
--- a/src/dsd_symbol.c
+++ b/src/dsd_symbol.c
@@ -279,9 +279,17 @@ getSymbol (dsd_opts * opts, dsd_state * state, int have_sync)
           }
 
           // analog audio filtering
-          // lpf (state, state->analog_out, 960);
+          // 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);
+
+          //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
+          //   nf(state, state->analog_out, 960);
+
           //seems to be working now, but RMS values are lower on actual analog signal than on no signal but noise
           if ( (opts->rtl_rms > opts->rtl_squelch_level) && opts->monitor_input_audio == 1 && state->carrier == 0 ) //added carrier check here in lieu of disabling it above
           {
diff --git a/src/edacs-fme.c b/src/edacs-fme.c
index 0190a30..ae3364a 100644
--- a/src/edacs-fme.c
+++ b/src/edacs-fme.c
@@ -211,6 +211,14 @@ 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);
+    }
+
     //reconfigured to use seperate audio out stream that is always 48k short
     if (opts->audio_out_type == 0 && opts->slot1_on == 1)
     {
diff --git a/src/m17.c b/src/m17.c
index fdf017b..6b601f0 100644
--- a/src/m17.c
+++ b/src/m17.c
@@ -1857,6 +1857,8 @@ void encodeM17STR(dsd_opts * opts, dsd_state * state)
     {
       hpf (state, voice1, 160);
       hpf (state, voice2, 160);
+      pbf (state, voice1, 160);
+      pbf (state, voice2, 160);
     }
 
     //convert out audio input into CODEC2 (3200bps) 8 byte data stream