--- uae/src/audio.c	2018/04/24 16:44:39	1.1.1.4
+++ uae/src/audio.c	2018/04/24 16:47:17	1.1.1.5
@@ -79,7 +79,6 @@ typedef uae_u8 sample8_t;
 
 #define DO_CHANNEL(v, c) do { (v) &= audio_channel[c].adk_mask; data += v; } while (0);
 
-/* Templates! I want templates! */
 void sample16_handler (void)
 {
     BEGIN_BENCH
@@ -109,6 +108,62 @@ void sample16_handler (void)
     check_sound_buffers ();
 }
 
+void sample16i_handler (void)
+{
+    unsigned long delta, ratio;
+
+    BEGIN_BENCH
+
+    uae_u32 data0 = audio_channel[0].current_sample;
+    uae_u32 data1 = audio_channel[1].current_sample;
+    uae_u32 data2 = audio_channel[2].current_sample;
+    uae_u32 data3 = audio_channel[3].current_sample;
+    uae_u32 data0p = audio_channel[0].last_sample;
+    uae_u32 data1p = audio_channel[1].last_sample;
+    uae_u32 data2p = audio_channel[2].last_sample;
+    uae_u32 data3p = audio_channel[3].last_sample;
+    DO_CHANNEL_1 (data0, 0);
+    DO_CHANNEL_1 (data1, 1);
+    DO_CHANNEL_1 (data2, 2);
+    DO_CHANNEL_1 (data3, 3);
+    DO_CHANNEL_1 (data0p, 0);
+    DO_CHANNEL_1 (data1p, 1);
+    DO_CHANNEL_1 (data2p, 2);
+    DO_CHANNEL_1 (data3p, 3);
+
+    data0 &= audio_channel[0].adk_mask;
+    data0p &= audio_channel[0].adk_mask;
+    data1 &= audio_channel[1].adk_mask;
+    data1p &= audio_channel[1].adk_mask;
+    data2 &= audio_channel[2].adk_mask;
+    data2p &= audio_channel[2].adk_mask;
+    data3 &= audio_channel[3].adk_mask;
+    data3p &= audio_channel[3].adk_mask;
+
+    /* linear interpolation and summing up... */
+    delta = audio_channel[0].per;
+    ratio = ((audio_channel[0].evtime % delta) << 8) / delta;
+    data0 = (data0 * (256 - ratio) + data0p * ratio) >> 8;
+    delta = audio_channel[1].per;
+    ratio = ((audio_channel[1].evtime % delta) << 8) / delta;
+    data0 += (data1 * (256 - ratio) + data1p * ratio) >> 8;
+    delta = audio_channel[2].per;
+    ratio = ((audio_channel[2].evtime % delta) << 8) / delta;
+    data0 += (data2 * (256 - ratio) + data2p * ratio) >> 8;
+    delta = audio_channel[3].per;
+    ratio = ((audio_channel[3].evtime % delta) << 8) / delta;
+    data0 += (data3 * (256 - ratio) + data3p * ratio) >> 8;
+
+    {
+	uae_u32 data = SBASEVAL16(2) + data0;
+	FINISH_DATA(16, 2);
+	PUT_SOUND_WORD (data);
+    }
+    END_BENCH
+
+    check_sound_buffers ();
+}
+
 void sample8_handler (void)
 {
     BEGIN_BENCH
@@ -176,6 +231,106 @@ void sample16s_handler (void)
     check_sound_buffers ();
 }
 
+void sample16si_handler (void)
+{
+    unsigned long delta, ratio;
+
+    BEGIN_BENCH
+
+    uae_u32 data0 = audio_channel[0].current_sample;
+    uae_u32 data1 = audio_channel[1].current_sample;
+    uae_u32 data2 = audio_channel[2].current_sample;
+    uae_u32 data3 = audio_channel[3].current_sample;
+    uae_u32 data0p = audio_channel[0].last_sample;
+    uae_u32 data1p = audio_channel[1].last_sample;
+    uae_u32 data2p = audio_channel[2].last_sample;
+    uae_u32 data3p = audio_channel[3].last_sample;
+
+    DO_CHANNEL_1 (data0, 0);
+    DO_CHANNEL_1 (data1, 1);
+    DO_CHANNEL_1 (data2, 2);
+    DO_CHANNEL_1 (data3, 3);
+    DO_CHANNEL_1 (data0p, 0);
+    DO_CHANNEL_1 (data1p, 1);
+    DO_CHANNEL_1 (data2p, 2);
+    DO_CHANNEL_1 (data3p, 3);
+
+    data0 &= audio_channel[0].adk_mask;
+    data0p &= audio_channel[0].adk_mask;
+    data1 &= audio_channel[1].adk_mask;
+    data1p &= audio_channel[1].adk_mask;
+    data2 &= audio_channel[2].adk_mask;
+    data2p &= audio_channel[2].adk_mask;
+    data3 &= audio_channel[3].adk_mask;
+    data3p &= audio_channel[3].adk_mask;
+
+# if 0
+    /* linear interpolation and summing up... */
+    delta = audio_channel[0].per;
+    ratio = ((audio_channel[0].evtime % delta) << 8) / delta;
+    data0 = (data0 * (256 - ratio) + data0p * ratio) >> 8;
+    delta = audio_channel[1].per;
+    ratio = ((audio_channel[1].evtime % delta) << 8) / delta;
+    data1 = (data1 * (256 - ratio) + data1p * ratio) >> 8;
+    delta = audio_channel[2].per;
+    ratio = ((audio_channel[2].evtime % delta) << 8) / delta;
+    data1 += (data2 * (256 - ratio) + data2p * ratio) >> 8;
+    delta = audio_channel[3].per;
+    ratio = ((audio_channel[3].evtime % delta) << 8) / delta;
+    data0 += (data3 * (256 - ratio) + data3p * ratio) >> 8;
+#else
+    {    
+	struct audio_channel_data *cdp;
+	int ratio, ratio1;
+#define INTERVAL (sample_evtime)
+	cdp = audio_channel + 0;
+	ratio1 = cdp->per - cdp->evtime;
+	ratio = (ratio1 << 12) / INTERVAL;
+	if (cdp->evtime < sample_evtime || ratio1 >= INTERVAL)
+	    ratio = 4096;
+	data0 = (data0 * ratio + data0p * (4096 - ratio)) >> 12;
+
+	cdp = audio_channel + 1;
+	ratio1 = cdp->per - cdp->evtime;
+	ratio = (ratio1 << 12) / INTERVAL;
+	if (cdp->evtime < sample_evtime || ratio1 >= INTERVAL)
+	    ratio = 4096;
+	data1 = (data1 * ratio + data1p * (4096 - ratio)) >> 12;
+
+	cdp = audio_channel + 2;
+	ratio1 = cdp->per - cdp->evtime;
+	ratio = (ratio1 << 12) / INTERVAL;
+	if (cdp->evtime < sample_evtime || ratio1 >= INTERVAL)
+	    ratio = 4096;
+	data2 = (data2 * ratio + data2p * (4096 - ratio)) >> 12;
+
+	cdp = audio_channel + 3;
+	ratio1 = cdp->per - cdp->evtime;
+	ratio = (ratio1 << 12) / INTERVAL;
+	if (cdp->evtime < sample_evtime || ratio1 >= INTERVAL)
+	    ratio = 4096;
+	data3 = (data3 * ratio + data3p * (4096 - ratio)) >> 12;
+    }
+    data1 += data2;
+    data0 += data3;
+#endif
+    {
+	uae_u32 data = SBASEVAL16 (1) + data0;
+	FINISH_DATA (16, 1);
+	PUT_SOUND_WORD_RIGHT (data);
+    }
+
+    {
+	uae_u32 data = SBASEVAL16 (1) + data1;
+	FINISH_DATA (16, 1);
+	PUT_SOUND_WORD_LEFT (data);
+    }
+    
+    END_BENCH
+    
+    check_sound_buffers ();
+}
+
 void sample8s_handler (void)
 {
     BEGIN_BENCH
@@ -301,6 +456,7 @@ static void audio_handler (int nr)
 
 	cdp->evtime = cdp->per;
 	cdp->dat = cdp->nextdat;
+	cdp->last_sample = cdp->current_sample;
 	cdp->current_sample = (sample8_t)(cdp->dat >> 8);
 
 	cdp->state = 2;
@@ -318,6 +474,7 @@ static void audio_handler (int nr)
 	if (currprefs.produce_sound == 0)
 	    cdp->per = 65535;
 
+	cdp->last_sample = cdp->current_sample;
 	cdp->current_sample = (sample8_t)(cdp->dat & 0xFF);
 	cdp->evtime = cdp->per;
 
@@ -353,6 +510,7 @@ static void audio_handler (int nr)
 
 	if ((INTREQR() & (0x80 << nr)) && !cdp->dmaen) {
 	    cdp->state = 0;
+	    cdp->last_sample = 0;
 	    cdp->current_sample = 0;
 	    break;
 	} else {
@@ -367,6 +525,7 @@ static void audio_handler (int nr)
 	    cdp->intreq2 = 0;
 
 	    cdp->dat = cdp->nextdat;
+	    cdp->last_sample = cdp->current_sample;
 	    cdp->current_sample = (sample8_t)(cdp->dat >> 8);
 
 	    if (cdp->dmaen && napnav)