diff --git a/lib/filters.c b/lib/filters.c
index 124686f..1efdc70 100644
--- a/lib/filters.c
+++ b/lib/filters.c
@@ -24,35 +24,7 @@ void init_lpf(BiquadFilter* filter, float cutoffFreq, float qFactor, float sampl
     filter->z2 = 0.0f;
 }
 
-void init_hpf(BiquadFilter* filter, float cutoffFreq, float qFactor, float sampleRate) {
-    float cutoffNorm = cutoffFreq / sampleRate;
-    float K = tanf(M_PI * cutoffNorm);
-    float norm = 1.0f/(1.0f+K/qFactor+K*K);
-    filter->a0 = 1.0f*norm;
-    filter->a1 = -2.0f*norm;
-    filter->a2 = 1.0f*norm;
-    filter->b1 = 2.0f*(K*K-1.0f)*norm;
-    filter->b2 = (1.0f-K/qFactor+K*K)*norm;
-
-    filter->z1 = 0.0f;
-    filter->z2 = 0.0f;
-}
-
-void init_bpf(BiquadFilter* filter, float centerFreq, float qFactor, float sampleRate) {
-    float cutoffNorm = centerFreq / sampleRate;
-    float K = tanf(M_PI * cutoffNorm);
-    float norm = 1.0f/(1.0f+K/qFactor+K*K);
-    filter->a0 = K/qFactor*norm;
-    filter->a1 = 0.0f;
-    filter->a2 = -K/qFactor*norm;
-    filter->b1 = 2.0f*(K*K-1.0f)*norm;
-    filter->b2 = (1.0f-K/qFactor+K*K)*norm;
-
-    filter->z1 = 0.0f;
-    filter->z2 = 0.0f;
-}
-
-float apply_frequency_filter(BiquadFilter* filter, float input) {
+float apply_biquad(BiquadFilter* filter, float input) {
     float out = input*filter->a0+filter->z1;
     filter->z1 = input*filter->a1+filter->z2-filter->b1*out;
     filter->z2 = input*filter->a2-filter->b2*out;
diff --git a/lib/filters.h b/lib/filters.h
index 977d616..d56788d 100644
--- a/lib/filters.h
+++ b/lib/filters.h
@@ -21,9 +21,7 @@ typedef struct {
     float z1, z2;
 } BiquadFilter;
 void init_lpf(BiquadFilter* filter, float cutoffFreq, float qFactor, float sampleRate);
-void init_hpf(BiquadFilter* filter, float cutoffFreq, float qFactor, float sampleRate);
-void init_bpf(BiquadFilter* filter, float centerFreq, float qFactor, float sampleRate);
-float apply_frequency_filter(BiquadFilter* filter, float input);
+float apply_biquad(BiquadFilter* filter, float input);
 
 float hard_clip(float sample, float threshold);
 float voltage_db_to_voltage(float db);
diff --git a/lib/fm_modulator.c b/lib/fm_modulator.c
index 0ed4287..67fc5a5 100644
--- a/lib/fm_modulator.c
+++ b/lib/fm_modulator.c
@@ -3,12 +3,13 @@
 void init_fm_modulator(FMModulator *fm, float frequency, float deviation, float sample_rate) {
     fm->frequency = frequency;
     fm->deviation = deviation;
-    init_oscillator(&fm->osc, frequency, sample_rate);
+    fm->sample_rate = sample_rate;
+    fm->osc_phase = 0.0f;
 }
 
 float modulate_fm(FMModulator *fm, float sample) {
     float inst_freq = fm->frequency+(sample*fm->deviation);
     if (inst_freq < 0.0f) inst_freq = 0.0f;
-    change_oscillator_frequency(&fm->osc, inst_freq);
-    return get_oscillator_sin_sample(&fm->osc);
+    fm->osc_phase += fmodf(fm->osc_phase + ((M_2PI * inst_freq) / fm->sample_rate), M_2PI);
+    return sinf(fm->osc_phase);
 }
\ No newline at end of file
diff --git a/lib/fm_modulator.h b/lib/fm_modulator.h
index 7079a33..9391fed 100644
--- a/lib/fm_modulator.h
+++ b/lib/fm_modulator.h
@@ -6,7 +6,8 @@ typedef struct
 {
     float frequency;
     float deviation;
-    Oscillator osc;
+    float osc_phase;
+    float sample_rate;
 } FMModulator;
 
 void init_fm_modulator(FMModulator *fm, float frequency, float deviation, float sample_rate);
diff --git a/src/fm95.c b/src/fm95.c
index 54f9e95..acf1ac8 100644
--- a/src/fm95.c
+++ b/src/fm95.c
@@ -414,10 +414,10 @@ int main(int argc, char **argv) {
             float current_mpx_in = mpx_in[i];
             float current_sca_in = sca_in[i];
 
-            float ready_l = apply_frequency_filter(&lpf_l, l_in);
-            float ready_r = apply_frequency_filter(&lpf_r, r_in);
-            ready_l = apply_preemphasis(&preemp_l, ready_l)*4;
-            ready_r = apply_preemphasis(&preemp_r, ready_r)*4;
+            float ready_l = apply_biquad(&lpf_l, l_in);
+            float ready_r = apply_biquad(&lpf_r, r_in);
+            ready_l = apply_preemphasis(&preemp_l, ready_l)*2;
+            ready_r = apply_preemphasis(&preemp_r, ready_r)*2;
             ready_l = hard_clip(ready_l*audio_volume, clipper_threshold);
             ready_r = hard_clip(ready_r*audio_volume, clipper_threshold);