diff --git a/src/include/86box/snd_ym7128.h b/src/include/86box/snd_ym7128.h
index a0796b1fa..f0657425f 100644
--- a/src/include/86box/snd_ym7128.h
+++ b/src/include/86box/snd_ym7128.h
@@ -19,18 +19,18 @@ typedef struct ym7128_t {
     int c1;
     int t[9];
 
-    int16_t filter_dat;
-    int16_t prev_l;
-    int16_t prev_r;
+    int16_t filter_dat[2];
+    int16_t prev_l[2];
+    int16_t prev_r[2];
 
-    int16_t delay_buffer[2400];
-    int     delay_pos;
+    int16_t delay_buffer[2][2400];
+    int     delay_pos[2];
 
     int16_t last_samp;
 } ym7128_t;
 
 void ym7128_init(ym7128_t *ym7128);
 void ym7128_write(ym7128_t *ym7128, uint8_t val);
-void ym7128_apply(ym7128_t *ym7128, int16_t *buffer, int len);
+void ym7128_apply(ym7128_t *ym7128, int16_t *buffer, int i, int len);
 
 #endif /*SOUND_YM7128_H*/
diff --git a/src/sound/snd_adlibgold.c b/src/sound/snd_adlibgold.c
index f708b6967..0181db72e 100644
--- a/src/sound/snd_adlibgold.c
+++ b/src/sound/snd_adlibgold.c
@@ -82,6 +82,7 @@ typedef struct adgold_t {
     int treble;
     int bass;
 
+    int16_t samp_buffer[SOUNDBUFLEN * 2];
     int16_t opl_buffer[MUSICBUFLEN * 2];
     int16_t mma_buffer[2][SOUNDBUFLEN];
 
@@ -788,25 +789,25 @@ adgold_get_buffer(int32_t *buffer, int len, void *priv)
     adgold_update(adgold);
 
     for (c = 0; c < len * 2; c += 2) {
-        adgold->opl_buffer[c] = ((adgold->mma_buffer[0][c >> 1] * adgold->samp_vol_l) >> 7) / 4;
-        adgold->opl_buffer[c + 1] = ((adgold->mma_buffer[1][c >> 1] * adgold->samp_vol_r) >> 7) / 4;
+        adgold->samp_buffer[c] = ((adgold->mma_buffer[0][c >> 1] * adgold->samp_vol_l) >> 7) / 4;
+        adgold->samp_buffer[c + 1] = ((adgold->mma_buffer[1][c >> 1] * adgold->samp_vol_r) >> 7) / 4;
     }
 
     if (adgold->surround_enabled)
-        ym7128_apply(&adgold->ym7128, adgold->opl_buffer, len);
+        ym7128_apply(&adgold->ym7128, adgold->samp_buffer, 0, len);
 
     switch (adgold->adgold_38x_regs[0x8] & 6) {
         case 0:
             for (c = 0; c < len * 2; c++)
-                adgold->opl_buffer[c] = 0;
+                adgold->samp_buffer[c] = 0;
             break;
         case 2: /*Left channel only*/
             for (c = 0; c < len * 2; c += 2)
-                adgold->opl_buffer[c + 1] = adgold->opl_buffer[c];
+                adgold->samp_buffer[c + 1] = adgold->samp_buffer[c];
             break;
         case 4: /*Right channel only*/
             for (c = 0; c < len * 2; c += 2)
-                adgold->opl_buffer[c] = adgold->opl_buffer[c + 1];
+                adgold->samp_buffer[c] = adgold->samp_buffer[c + 1];
             break;
         case 6: /*Left and right channels*/
             break;
@@ -818,7 +819,7 @@ adgold_get_buffer(int32_t *buffer, int len, void *priv)
     switch (adgold->adgold_38x_regs[0x8] & 0x18) {
         case 0x00: /*Forced mono*/
             for (c = 0; c < len * 2; c += 2)
-                adgold->opl_buffer[c] = adgold->opl_buffer[c + 1] = ((int32_t) adgold->opl_buffer[c] + (int32_t) adgold->opl_buffer[c + 1]) / 2;
+                adgold->samp_buffer[c] = adgold->samp_buffer[c + 1] = ((int32_t) adgold->samp_buffer[c] + (int32_t) adgold->samp_buffer[c + 1]) / 2;
             break;
         case 0x08: /*Linear stereo*/
             break;
@@ -826,17 +827,17 @@ adgold_get_buffer(int32_t *buffer, int len, void *priv)
             /*Filter left channel, leave right channel unchanged*/
             /*Filter cutoff is largely a guess*/
             for (c = 0; c < len * 2; c += 2)
-                adgold->opl_buffer[c] += adgold_pseudo_stereo_iir(0, adgold->opl_buffer[c]);
+                adgold->samp_buffer[c] += adgold_pseudo_stereo_iir(0, adgold->samp_buffer[c]);
             break;
         case 0x18: /*Spatial stereo*/
             /*Quite probably wrong, I only have the diagram in the TDA8425 datasheet
               and a very vague understanding of how op-amps work to go on*/
             for (c = 0; c < len * 2; c += 2) {
-                int16_t l = adgold->opl_buffer[c];
-                int16_t r = adgold->opl_buffer[c + 1];
+                int16_t l = adgold->samp_buffer[c];
+                int16_t r = adgold->samp_buffer[c + 1];
 
-                adgold->opl_buffer[c] += (r / 3) + ((l * 2) / 3);
-                adgold->opl_buffer[c + 1] += (l / 3) + ((r * 2) / 3);
+                adgold->samp_buffer[c] += (r / 3) + ((l * 2) / 3);
+                adgold->samp_buffer[c + 1] += (l / 3) + ((r * 2) / 3);
             }
             break;
 
@@ -850,7 +851,7 @@ adgold_get_buffer(int32_t *buffer, int len, void *priv)
         int32_t highpass;
 
         /*Output is deliberately halved to avoid clipping*/
-        temp     = ((int32_t) adgold->opl_buffer[c] * adgold->vol_l) >> 18;
+        temp     = ((int32_t) adgold->samp_buffer[c] * adgold->vol_l) >> 17;
         lowpass  = adgold_lowpass_iir(0, 0, temp);
         highpass = adgold_highpass_iir(0, 0, temp);
         if (adgold->bass > 6)
@@ -867,7 +868,7 @@ adgold_get_buffer(int32_t *buffer, int len, void *priv)
             temp = 32767;
         buffer[c] += temp;
 
-        temp     = ((int32_t) adgold->opl_buffer[c + 1] * adgold->vol_r) >> 18;
+        temp     = ((int32_t) adgold->samp_buffer[c + 1] * adgold->vol_r) >> 17;
         lowpass  = adgold_lowpass_iir(0, 1, temp);
         highpass = adgold_highpass_iir(0, 1, temp);
         if (adgold->bass > 6)
@@ -902,7 +903,7 @@ adgold_get_music_buffer(int32_t *buffer, int len, void *priv)
     }
 
     if (adgold->surround_enabled)
-        ym7128_apply(&adgold->ym7128, adgold->opl_buffer, len);
+        ym7128_apply(&adgold->ym7128, adgold->opl_buffer, 1, len);
 
     switch (adgold->adgold_38x_regs[0x8] & 6) {
         case 0:
@@ -959,7 +960,7 @@ adgold_get_music_buffer(int32_t *buffer, int len, void *priv)
         int32_t highpass;
 
         /*Output is deliberately halved to avoid clipping*/
-        temp     = ((int32_t) adgold->opl_buffer[c] * adgold->vol_l) >> 18;
+        temp     = ((int32_t) adgold->opl_buffer[c] * adgold->vol_l) >> 17;
         lowpass  = adgold_lowpass_iir(1, 0, temp);
         highpass = adgold_highpass_iir(1, 0, temp);
         if (adgold->bass > 6)
@@ -976,7 +977,7 @@ adgold_get_music_buffer(int32_t *buffer, int len, void *priv)
             temp = 32767;
         buffer[c] += temp;
 
-        temp     = ((int32_t) adgold->opl_buffer[c + 1] * adgold->vol_r) >> 18;
+        temp     = ((int32_t) adgold->opl_buffer[c + 1] * adgold->vol_r) >> 17;
         lowpass  = adgold_lowpass_iir(1, 1, temp);
         highpass = adgold_highpass_iir(1, 1, temp);
         if (adgold->bass > 6)
diff --git a/src/sound/snd_ym7128.c b/src/sound/snd_ym7128.c
index 59e5691e9..08b33f414 100644
--- a/src/sound/snd_ym7128.c
+++ b/src/sound/snd_ym7128.c
@@ -111,10 +111,10 @@ ym7128_write(ym7128_t *ym7128, uint8_t val)
     ym7128->a0  = new_a0;
 }
 
-#define GET_DELAY_SAMPLE(ym7128, offset) (((ym7128->delay_pos - offset) < 0) ? ym7128->delay_buffer[(ym7128->delay_pos - offset) + 2400] : ym7128->delay_buffer[ym7128->delay_pos - offset])
+#define GET_DELAY_SAMPLE(ym7128, offset) (((ym7128->delay_pos[i] - offset) < 0) ? ym7128->delay_buffer[i][(ym7128->delay_pos[i] - offset) + 2400] : ym7128->delay_buffer[i][ym7128->delay_pos[i] - offset])
 
 void
-ym7128_apply(ym7128_t *ym7128, int16_t *buffer, int len)
+ym7128_apply(ym7128_t *ym7128, int16_t *buffer, int i, int len)
 {
     for (int c = 0; c < len * 2; c += 4) {
         /*YM7128 samples a mono stream at ~24 kHz, so downsample*/
@@ -125,13 +125,13 @@ ym7128_apply(ym7128_t *ym7128, int16_t *buffer, int len)
         int32_t samp_r = 0;
 
         filter_temp = GET_DELAY_SAMPLE(ym7128, ym7128->t[0]);
-        filter_out  = ((filter_temp * ym7128->c0) >> 11) + ((ym7128->filter_dat * ym7128->c1) >> 11);
+        filter_out  = ((filter_temp * ym7128->c0) >> 11) + ((ym7128->filter_dat[i] * ym7128->c1) >> 11);
         filter_out  = (filter_out * ym7128->vc) >> 16;
 
         samp = (samp * ym7128->vm) >> 16;
         samp += filter_out;
 
-        ym7128->delay_buffer[ym7128->delay_pos] = samp;
+        ym7128->delay_buffer[i][ym7128->delay_pos[i]] = samp;
 
         for (uint8_t d = 0; d < 8; d++) {
             samp_l += (GET_DELAY_SAMPLE(ym7128, ym7128->t[d + 1]) * ym7128->gl[d]) >> 16;
@@ -141,17 +141,17 @@ ym7128_apply(ym7128_t *ym7128, int16_t *buffer, int len)
         samp_l = (samp_l * ym7128->vl * 2) >> 16;
         samp_r = (samp_r * ym7128->vr * 2) >> 16;
 
-        buffer[c] += (samp_l + (int32_t) ym7128->prev_l) / 2;
-        buffer[c + 1] += (samp_r + (int32_t) ym7128->prev_r) / 2;
+        buffer[c] += (samp_l + (int32_t) ym7128->prev_l[i]) / 2;
+        buffer[c + 1] += (samp_r + (int32_t) ym7128->prev_r[i]) / 2;
         buffer[c + 2] += samp_l;
         buffer[c + 3] += samp_r;
 
-        ym7128->delay_pos++;
-        if (ym7128->delay_pos >= 2400)
-            ym7128->delay_pos = 0;
+        ym7128->delay_pos[i]++;
+        if (ym7128->delay_pos[i] >= 2400)
+            ym7128->delay_pos[i] = 0;
 
-        ym7128->filter_dat = filter_temp;
-        ym7128->prev_l     = samp_l;
-        ym7128->prev_r     = samp_r;
+        ym7128->filter_dat[i] = filter_temp;
+        ym7128->prev_l[i]     = samp_l;
+        ym7128->prev_r[i]     = samp_r;
     }
 }