--- uae/src/audio.c	2018/04/24 16:43:19	1.1.1.3
+++ uae/src/audio.c	2018/04/24 16:48:32	1.1.1.6
@@ -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 +108,139 @@ void sample16_handler (void)
     check_sound_buffers ();
 }
 
+void sample16i_rh_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 sample16i_crux_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;
+
+    {    
+	struct audio_channel_data *cdp;
+	int ratio, ratio1;
+#define INTERVAL (sample_evtime * 3)
+	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;
+    data0 += data1;
+    {
+	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 +308,153 @@ void sample16s_handler (void)
     check_sound_buffers ();
 }
 
+void sample16si_crux_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;
+
+    {    
+	struct audio_channel_data *cdp;
+	int ratio, ratio1;
+#define INTERVAL (sample_evtime * 3)
+	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;
+    {
+	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 sample16si_rh_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;
+    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;
+    {
+	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
@@ -220,6 +499,14 @@ void sample16s_handler (void)
 {
     sample16_handler();
 }
+void sample16si_crux_handler (void)
+{
+    sample16i_crux_handler();
+}
+void sample16si_rh_handler (void)
+{
+    sample16i_rh_handler();
+}
 #endif
 
 static uae_u8 int2ulaw (int ch)
@@ -301,6 +588,7 @@ 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 +606,7 @@ 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 +642,7 @@ 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 +657,7 @@ 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)
@@ -423,47 +714,59 @@ void audio_reset (void)
     next_sample_evtime = sample_evtime;
 }
 
-static __inline__ int sound_prefs_changed (void)
+STATIC_INLINE int sound_prefs_changed (void)
 {
     return (changed_prefs.produce_sound != currprefs.produce_sound
 	    || changed_prefs.stereo != currprefs.stereo
+	    || changed_prefs.sound_freq != currprefs.sound_freq
 	    || changed_prefs.sound_bits != currprefs.sound_bits);
 }
 
 void check_prefs_changed_audio (void)
 {
-    if (! sound_available || ! sound_prefs_changed ())
-	return;
-
-    close_sound ();
-
-    currprefs.produce_sound = changed_prefs.produce_sound;
-    currprefs.stereo = changed_prefs.stereo;
-    currprefs.sound_bits = changed_prefs.sound_bits;
+    if (sound_available && sound_prefs_changed ()) {
+	close_sound ();
 
-    if (currprefs.produce_sound < 2)
-	return;
-
-    if (init_sound ()) {
-	last_cycles = cycles - 1;
-	next_sample_evtime = sample_evtime;
-	return;
-    }
-    if (! sound_available) {
-	fprintf (stderr, "Sound is not supported.\n");
-    } else {
-	fprintf (stderr, "Sorry, can't initialize sound.\n");
-	currprefs.produce_sound = 0;
-	/* So we don't do this every frame */
-	changed_prefs.produce_sound = 0;
+	currprefs.produce_sound = changed_prefs.produce_sound;
+	currprefs.stereo = changed_prefs.stereo;
+	currprefs.sound_bits = changed_prefs.sound_bits;
+	currprefs.sound_freq = changed_prefs.sound_freq;
+
+	if (currprefs.produce_sound >= 2) {
+	    if (init_sound ()) {
+		last_cycles = cycles - 1;
+		next_sample_evtime = sample_evtime;
+	    } else
+		if (! sound_available) {
+		    fprintf (stderr, "Sound is not supported.\n");
+		} else {
+		    fprintf (stderr, "Sorry, can't initialize sound.\n");
+		    currprefs.produce_sound = 0;
+		    /* So we don't do this every frame */
+		    changed_prefs.produce_sound = 0;
+		}
+	}
     }
+    /* Select the right interpolation method.  */
+    if (sample_handler == sample16_handler
+	|| sample_handler == sample16i_crux_handler
+	|| sample_handler == sample16i_rh_handler)
+	sample_handler = (currprefs.sound_interpol == 0 ? sample16_handler
+			  : currprefs.sound_interpol == 1 ? sample16i_rh_handler
+			  : sample16i_crux_handler);
+    else if (sample_handler == sample16s_handler
+	     || sample_handler == sample16si_crux_handler
+	     || sample_handler == sample16si_rh_handler)
+	sample_handler = (currprefs.sound_interpol == 0 ? sample16s_handler
+			  : currprefs.sound_interpol == 1 ? sample16si_rh_handler
+			  : sample16si_crux_handler);
 }
 
 void update_audio (void)
 {
     unsigned long int n_cycles;
 
-    if (currprefs.produce_sound == 0)
+    if (currprefs.produce_sound < 2)
 	return;
 
     n_cycles = cycles - last_cycles;