diff --git a/ctru-rs/examples/audio_filters.rs b/ctru-rs/examples/audio_filters.rs
index 5625d8f..7ef4967 100644
--- a/ctru-rs/examples/audio_filters.rs
+++ b/ctru-rs/examples/audio_filters.rs
@@ -11,29 +11,34 @@ use ctru::services::ndsp::{
 
 const SAMPLE_RATE: usize = 22050;
 const SAMPLES_PER_BUF: usize = SAMPLE_RATE / 30; // 735
-const BYTES_PER_SAMPLE: usize = 2;
+const BYTES_PER_SAMPLE: usize = 4;
 const AUDIO_WAVE_LENGTH: usize = SAMPLES_PER_BUF * BYTES_PER_SAMPLE;
 
 // Note Frequencies
 const NOTEFREQ: [f32; 7] = [220., 440., 880., 1760., 3520., 7040., 14080.];
 
-// audioBuffer is stereo PCM16
-fn fill_buffer(audio_data: &mut [u8], frequency: f32) {
-    let formatted_data = audio_data.chunks_exact_mut(2);
+// audioBuffer is Stereo PCM16
+// As such, a sample is made up of 2 "Mono" samples (2 * i16 = u32), one for each channel (left and right)
+fn fill_buffer(audio_data: &mut [u8], frequency: f32, offset: &mut usize) {
+    let formatted_data = audio_data.chunks_exact_mut(4);
 
-    let mut i = 0.;
+    let mut i: usize = 0;
     for chunk in formatted_data {
         // This is a simple sine wave, with a frequency of `frequency` Hz, and an amplitude 30% of maximum.
-        let sample: f32 = (frequency / 30. * (i / SAMPLES_PER_BUF as f32) * 2. * PI).sin();
+        let sample: f32 = (frequency * ((i + *offset) as f32 / SAMPLE_RATE as f32) * 2. * PI).sin();
         let amplitude = 0.3 * i16::MAX as f32;
 
-        // This operation is safe, since we are writing to a slice of exactly 16 bits
-        let chunk_ptr: &mut i16 = unsafe { std::mem::transmute(chunk.as_mut_ptr()) };
+        // This operation is safe, since we are writing to a slice of exactly 32 bits
+        let chunk_ptr: &mut [i16; 2] = unsafe { std::mem::transmute(chunk.as_mut_ptr()) };
         // Stereo samples are interleaved: left and right channels.
-        *chunk_ptr = (sample * amplitude) as i16;
+        chunk_ptr[0] = (sample * amplitude) as i16;
+        chunk_ptr[1] = (sample * amplitude) as i16;
 
-        i += 1.;
+        i += 1;
     }
+
+    // Adds the SAMPLES_PER_BUF length back to the offset
+    *offset += i;
 }
 
 fn main() {
@@ -77,9 +82,13 @@ fn main() {
 
     // We set up two wave buffers and alternate between the two,
     // effectively streaming an infinitely long sine wave.
+    let mut offset: usize = 0;
+
     let mut audio_data1 = Box::new_in([0u8; AUDIO_WAVE_LENGTH], LinearAllocator);
-    fill_buffer(&mut audio_data1[..], NOTEFREQ[4]);
-    let mut audio_data2 = audio_data1.clone();
+    fill_buffer(&mut audio_data1[..], NOTEFREQ[4], &mut offset);
+
+    let mut audio_data2 = Box::new_in([0u8; AUDIO_WAVE_LENGTH], LinearAllocator);
+    fill_buffer(&mut audio_data2[..], NOTEFREQ[4], &mut offset);
 
     let mut audio_buffer1 =
         WaveBuffer::new(audio_data1, AudioFormat::PCM16Stereo).expect("Couldn't sync DSP cache");
@@ -158,7 +167,11 @@ fn main() {
 
         let status = current.get_status();
         if let WaveStatus::Done = status {
-            fill_buffer(current.get_mut_wavebuffer().get_mut_data(), NOTEFREQ[note]);
+            fill_buffer(
+                current.get_mut_wavebuffer().get_mut_data(),
+                NOTEFREQ[note],
+                &mut offset,
+            );
             channel_zero.queue_wave(current);
 
             altern = !altern;
diff --git a/ctru-rs/src/services/ndsp/mod.rs b/ctru-rs/src/services/ndsp/mod.rs
index eb2ff80..4c63f47 100644
--- a/ctru-rs/src/services/ndsp/mod.rs
+++ b/ctru-rs/src/services/ndsp/mod.rs
@@ -220,11 +220,16 @@ impl AudioFormat {
     /// 16 bit formats return 2 (bytes)
     pub fn sample_size(self) -> u8 {
         match self {
-            AudioFormat::PCM16Mono | AudioFormat::PCM16Stereo => 2,
+            AudioFormat::PCM8Mono | AudioFormat::ADPCMMono => 1,
+            AudioFormat::PCM16Mono | AudioFormat::PCM8Stereo => 2,
+            AudioFormat::PCM16Stereo => 4,
             AudioFormat::SurroundPreprocessed => {
-                panic!("Can't find size for Sourround Preprocessed audio: format is under research")
+                panic!("Can't find sample size for Sourround Preprocessed audio: format is under research")
+            }
+            // TODO: Understand what this is.
+            AudioFormat::FrontBypass => {
+                panic!("Can't find sample size for FrontBypass audio")
             }
-            _ => 1,
         }
     }
 }