diff --git a/.vscode/c_cpp_properties.json b/.vscode/c_cpp_properties.json
index 0211f69..79ddb0c 100644
--- a/.vscode/c_cpp_properties.json
+++ b/.vscode/c_cpp_properties.json
@@ -3,8 +3,7 @@
         {
             "name": "Win32",
             "includePath": [
-                "${workspaceFolder}/**",
-                "C:\\lua\\*"
+                "${workspaceFolder}/**"
             ],
             "defines": [
                 "_DEBUG",
diff --git a/lib/stereo_encoder.c b/lib/stereo_encoder.c
index 38898b9..16bbcbc 100644
--- a/lib/stereo_encoder.c
+++ b/lib/stereo_encoder.c
@@ -16,31 +16,28 @@ void init_stereo_encoder(StereoEncoder* st, uint8_t stereo_ssb, uint8_t multipli
 float stereo_encode(StereoEncoder* st, uint8_t enabled, float left, float right, float *audio) {
     float mid = (left+right) * 0.5f;
     if(!enabled) {
-        *audio = mid * 2.0f * st->audio_volume;
+        *audio = mid * st->audio_volume * 2.0f;
         return 0.0f;
     }
 
-    if(st->stereo_hilbert) mid = delay_line(&st->delay, mid);
-
     float side = (left-right) * 0.5f;
-
-    float complex stereo_hilbert = 0+0*I;
-    if(st->stereo_hilbert) firhilbf_r2c_execute(st->stereo_hilbert, side, &stereo_hilbert);
-    
     float signalx1 = get_oscillator_sin_multiplier_ni(st->osc, st->multiplier);
-
-    float signalx2cos = 0.0f;
-    if(st->stereo_hilbert) {
-        signalx1 = delay_line(&st->delay_pilot, signalx1);
-        signalx2cos = get_oscillator_cos_multiplier_ni(st->osc, st->multiplier * 2.0f);
-    }
     float signalx2 = get_oscillator_sin_multiplier_ni(st->osc, st->multiplier * 2.0f);
-
-    *audio = mid * st->audio_volume;
     if(st->stereo_hilbert) {
-        float stereo = (crealf(stereo_hilbert) * signalx2) - (cimagf(stereo_hilbert) * signalx2cos);
-        *audio += stereo * st->audio_volume;
-    } else *audio += (side*signalx2) * st->audio_volume;
+        float complex stereo_hilbert = 0+0*I;
+        float signalx2cos = get_oscillator_cos_multiplier_ni(st->osc, st->multiplier * 2.0f);
+
+        mid = delay_line(&st->delay, mid);
+        *audio = mid * st->audio_volume;
+        signalx1 = delay_line(&st->delay_pilot, signalx1);
+
+        firhilbf_r2c_execute(st->stereo_hilbert, side, &stereo_hilbert);
+
+        *audio += ((crealf(stereo_hilbert) * signalx2) - (cimagf(stereo_hilbert) * signalx2cos)) * st->audio_volume;
+        return signalx1 * st->pilot_volume;
+    }
+
+    *audio = mid * st->audio_volume + (side*signalx2) * st->audio_volume;
     return signalx1 * st->pilot_volume;
 }
 
diff --git a/src/fm95.c b/src/fm95.c
index c1e842b..6ef6907 100644
--- a/src/fm95.c
+++ b/src/fm95.c
@@ -164,8 +164,6 @@ if((pulse_error = write_PulseOutputDevice(&runtime->output_device, output, sizeo
 
 int run_fm95(FM95_Config* config, FM95_Runtime* runtime, FM95_RunResult* result) {
 	float output[BUFFER_SIZE];
-	FM95_RunResult temp_result;
-	memset(&temp_result, 0, sizeof(FM95_RunResult));
 
 	int pulse_error;
 
@@ -214,14 +212,14 @@ int run_fm95(FM95_Config* config, FM95_Runtime* runtime, FM95_RunResult* result)
 			float r = audio_stereo_input[2*i+1]*config->audio_preamp;
 
 			float mono = 0.5f * (fabsf(l) + fabsf(r));
-			temp_result.input_level = mono;
+			result->input_level = mono;
 
 			if(config->agc_max != 0.0) {
 				float agc_gain = process_agc(&runtime->agc, mono);
 				l *= agc_gain;
 				r *= agc_gain;
-				temp_result.agc_gain = agc_gain;
-			} else temp_result.agc_gain = 0.0f;
+				result->agc_gain = agc_gain;
+			} else result->agc_gain = 0.0f;
 
 			float mod_l, mod_r;
 
@@ -238,7 +236,7 @@ int run_fm95(FM95_Config* config, FM95_Runtime* runtime, FM95_RunResult* result)
 			mod_l = tanhf(mod_l * config->volumes.drive) * softclip_norm;
 			mod_r = tanhf(mod_r * config->volumes.drive) * softclip_norm;
 
-			if(do_result) temp_result.audio_level = (mod_l + mod_r) * 0.5f;
+			if(do_result) result->audio_level = (mod_l + mod_r) * 0.5f;
 
 			mpx = stereo_encode(&runtime->stencode, config->stereo, mod_l, mod_r, &audio);
 
@@ -266,12 +264,11 @@ int run_fm95(FM95_Config* config, FM95_Runtime* runtime, FM95_RunResult* result)
 				}
 			}
 
-			mpx = bs412_compress(&runtime->bs412, audio, mpx+mpx_in[i], &temp_result.mpx_power);
-			temp_result.bs412_gain = runtime->bs412.gain;
+			mpx = bs412_compress(&runtime->bs412, audio, mpx+mpx_in[i], &result->mpx_power);
+			result->bs412_gain = runtime->bs412.gain;
 
 			output[i] = tanhf(mpx)*config->master_volume; // Ensure peak deviation of 75 khz (or the set deviation), assuming we're calibrated correctly
-		} memcpy(result, &temp_result, sizeof(FM95_RunResult));
-		_pulse_output;
+		} _pulse_output;
 	}
 
 	return 0;
@@ -338,13 +335,8 @@ static int config_handler(void* user, const char* section, const char* name, con
 	else if(MATCH("advanced", "lpf_order")) pconfig->lpf_order = atoi(value);
 	else if(MATCH("advanced", "stereo_ssb")) pconfig->stereo_ssb = atoi(value);
 	else if(MATCH("advanced", "preemp_unity")) pconfig->preemp_unity_freq = strtof(value, NULL);
-	else if(MATCH("advanced", "sample_rate")) {
-		pconfig->sample_rate = atoi(value);
-		if(pconfig->sample_rate < 0) {
-			fprintf(stderr, "Negative sample rate\n");
-			return 0;
-		}
-	} else if(MATCH("advanced", "lpf_cutoff")) {
+	else if(MATCH("advanced", "sample_rate")) pconfig->sample_rate = atoi(value);
+	else if(MATCH("advanced", "lpf_cutoff")) {
 		pconfig->lpf_cutoff = strtof(value, NULL);
 		if(pconfig->lpf_cutoff > (pconfig->sample_rate * 0.5)) {
 			pconfig->lpf_cutoff = (pconfig->sample_rate * 0.5);
@@ -465,8 +457,8 @@ static void *handle_client(ipc_client_arg_t *arg) {
 	FM95_Data* data = arg->user_data;
     free(arg);
 
-    char buf[BUF_SIZE];
-	char reply = 0;
+    uint8_t buf[BUF_SIZE];
+	uint8_t reply = 0;
     ssize_t n;
 	float val;
 	uint8_t bval;