From daa80b5a180e0d26e425959e43100eaf7bb1ed44 Mon Sep 17 00:00:00 2001
From: lxpollitt <630494+lxpollitt@users.noreply.github.com>
Date: Mon, 8 Jun 2026 16:57:03 +0100
Subject: [PATCH] Add DC offset correction to desktop audio

---
 .../java/emu/joric/lwjgl3/AY38912PSG.java     | 42 +++++++++++++++----
 1 file changed, 35 insertions(+), 7 deletions(-)

diff --git a/lwjgl3/src/main/java/emu/joric/lwjgl3/AY38912PSG.java b/lwjgl3/src/main/java/emu/joric/lwjgl3/AY38912PSG.java
index 6b03d6f..1d18970 100644
--- a/lwjgl3/src/main/java/emu/joric/lwjgl3/AY38912PSG.java
+++ b/lwjgl3/src/main/java/emu/joric/lwjgl3/AY38912PSG.java
@@ -81,6 +81,15 @@ public class AY38912PSG implements AYPSG {
   private int sampleBufferOffset = 0;
   private float cyclesToNextSample;
   private SourceDataLine audioLine;
+
+  // One-pole DC-blocker state and coefficient used to model the AC coupling capacitor
+  // on Oric audio output to remove the DC offset that the audio chip's emulated unipolar
+  // signal would otherwise carry into the audio line output.
+  // R = 0.995 gives a -3 dB corner at ~17.5 Hz @ 22050 Hz sample rate, which should be
+  // below any expected normally audible Oric content.
+  private static final float DC_BLOCKER_R = 0.995f;
+  private float dcBlockerX1;
+  private float dcBlockerY1;
   
   /**
    * The AY-3-8912 in the Oric gets its data from the 6522 VIA chip.
@@ -147,8 +156,11 @@ public void init(Via via, Keyboard keyboard, Snapshot snapshot) {
     } catch (LineUnavailableException lue) {
       audioLine = null;
     }
-    
+
     cyclesToNextSample = CYCLES_PER_SAMPLE;
+
+    dcBlockerX1 = 0f;
+    dcBlockerY1 = 0f;
   }
   
   /**
@@ -473,12 +485,28 @@ public void writeSample() {
       }
     }
     
-    int sample =  (((((VOLUME_LEVELS[volumeA] * cnt[A]) >> 13) + 
-                     ((VOLUME_LEVELS[volumeB] * cnt[B]) >> 13) + 
-                     ((VOLUME_LEVELS[volumeC] * cnt[C]) >> 13)) & 0x7FFF));
-    
-    sampleBuffer[sampleBufferOffset + 0] = (byte)(sample & 0x00FF);
-    sampleBuffer[sampleBufferOffset + 1] = (byte)((sample & 0xFF00) >> 8);
+    int sample = Math.min(((VOLUME_LEVELS[volumeA] * cnt[A]) >> 13) +
+                  ((VOLUME_LEVELS[volumeB] * cnt[B]) >> 13) +
+                  ((VOLUME_LEVELS[volumeC] * cnt[C]) >> 13), 0x7FFF);
+
+    // Use a simple DC blocker to convert to -1.0 to 1.0, then scale to signed
+    // 16-bit PCM, which is what the audio line needs. The output clamp is folded
+    // into the same expression as the filter, so on the rare transient that hits
+    // the rail (e.g. a register write flipping the chip from full silence to full
+    // output in one sample), the clamped value is what gets fed back into the
+    // filter state. (Clamping the filter state is arguably less mathematically
+    // accurate, because the clamping behaviour is non-linear. But this approach
+    // brings us out of the clipping state and into more normal behaviour faster,
+    // and is likely a closer approximation of the original hardware circuit
+    // behaviour.)
+    float x = sample / 16384.0f;
+    float y = Math.max(-1f, Math.min(1f,
+                    x - dcBlockerX1 + DC_BLOCKER_R * dcBlockerY1));
+    dcBlockerX1 = x;
+    dcBlockerY1 = y;
+    int int16Sample = (int)(y * 32767f);
+    sampleBuffer[sampleBufferOffset + 0] = (byte)(int16Sample & 0x00FF);
+    sampleBuffer[sampleBufferOffset + 1] = (byte)((int16Sample & 0xFF00) >> 8);
     
     // If the sample buffer is full, write it out to the audio line.
     if ((sampleBufferOffset += 2) == sampleBuffer.length) {