--- uae/src/audio.c	2018/04/24 17:18:12	1.1.1.17
+++ uae/src/audio.c	2018/04/24 17:21:38	1.1.1.19
@@ -6,6 +6,7 @@
   * Copyright 1995, 1996, 1997 Bernd Schmidt
   * Copyright 1996 Marcus Sundberg
   * Copyright 1996 Manfred Thole
+  * Copyright 2005 Heikki Orsila
   * Copyright 2006 Toni Wilen
   *
   * new filter algorithm and anti&sinc interpolators by Antti S. Lankila
@@ -55,7 +56,6 @@ struct audio_channel_data {
     unsigned int wlen;
     int current_sample, last_sample;
     int vol;
-    int *voltbl;
     uae_u16 dat, nextdat, len;
     int sample_accum, sample_accum_time;
     int sinc_output_state;
@@ -80,20 +80,13 @@ void init_sound_table16 (void)
 
     for (i = 0; i < 256; i++)
 	for (j = 0; j < 64; j++)
-	    sound_table[j][i] = j * (uae_s8)i * (get_audio_ismono () ? 1 : 2);
+	    sound_table[j][i] = j * (uae_s8)i * 2;
 }
 
-#ifdef MULTIPLICATION_PROFITABLE
 typedef uae_s8 sample8_t;
 #define DO_CHANNEL_1(v, c) do { (v) *= audio_channel[c].vol; } while (0)
 #define SBASEVAL16(logn) ((logn) == 1 ? SOUND16_BASE_VAL >> 1 : SOUND16_BASE_VAL)
 #define FINISH_DATA(data, b, logn) do { if (14 - (b) + (logn) > 0) (data) >>= 14 - (b) + (logn); else (data) <<= (b) - 14 - (logn); } while (0);
-#else
-typedef uae_u8 sample8_t;
-#define DO_CHANNEL_1(v, c) do { (v) = audio_channel[c].voltbl[(v)]; } while (0)
-#define SBASEVAL16(logn) SOUND16_BASE_VAL
-#define FINISH_DATA(data, b, logn)
-#endif
 
 static uae_u32 right_word_saved[SOUND_MAX_DELAY_BUFFER];
 static uae_u32 left_word_saved[SOUND_MAX_DELAY_BUFFER];
@@ -333,188 +326,6 @@ STATIC_INLINE void samplexx_sinc_handler
     }
 }
 
-static void sample16i_sinc_handler (void)
-{
-    int datas[4], data1;
-
-    samplexx_sinc_handler (datas);
-    data1 = datas[0] + datas[3] + datas[1] + datas[2];
-    FINISH_DATA (data1, 16, 2);
-    PUT_SOUND_WORD (data1);
-    check_sound_buffers ();
-}
-
-void sample16_handler (void)
-{
-    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;
-    DO_CHANNEL_1 (data0, 0);
-    DO_CHANNEL_1 (data1, 1);
-    DO_CHANNEL_1 (data2, 2);
-    DO_CHANNEL_1 (data3, 3);
-    data0 &= audio_channel[0].adk_mask;
-    data1 &= audio_channel[1].adk_mask;
-    data2 &= audio_channel[2].adk_mask;
-    data3 &= audio_channel[3].adk_mask;
-    data0 += data1;
-    data0 += data2;
-    data0 += data3;
-    {
-	uae_u32 data = SBASEVAL16(2) + data0;
-	FINISH_DATA (data, 16, 2);
-	if (sound_use_filter)
-	    data = filter (data, &sound_filter_state[0]);
-	PUT_SOUND_WORD (data);
-    }
-    check_sound_buffers ();
-}
-
-/* This interpolator examines sample points when Paula switches the output
- * voltage and computes the average of Paula's output */
-static void sample16i_anti_handler (void)
-{
-    int datas[4], data1;
-
-    samplexx_anti_handler (datas);
-    data1 = datas[0] + datas[3] + datas[1] + datas[2];
-    FINISH_DATA (data1, 16, 2);
-    if (sound_use_filter)
-	data1 = filter (data1, &sound_filter_state[0]);
-    PUT_SOUND_WORD (data1);
-    check_sound_buffers ();
-}
-
-static void sample16i_rh_handler (void)
-{
-    unsigned long delta, ratio;
-
-    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 (data, 16, 2);
-	if (sound_use_filter)
-	    data = filter (data, &sound_filter_state[0]);
-	PUT_SOUND_WORD (data);
-    }
-
-    check_sound_buffers ();
-}
-
-static void sample16i_crux_handler (void)
-{
-    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;
-	unsigned long ratio, ratio1;
-#define INTERVAL (scaled_sample_evtime * 3)
-	cdp = audio_channel + 0;
-	ratio1 = cdp->per - cdp->evtime;
-	ratio = (ratio1 << 12) / INTERVAL;
-	if (cdp->evtime < scaled_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 < scaled_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 < scaled_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 < scaled_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 (data, 16, 2);
-	if (sound_use_filter)
-	    data = filter (data, &sound_filter_state[0]);
-	PUT_SOUND_WORD (data);
-    }
-    check_sound_buffers ();
-}
-
-#ifdef HAVE_STEREO_SUPPORT
 static void sample16si_anti_handler (void)
 {
     int datas[4], data1, data2;
@@ -584,181 +395,17 @@ void sample16s_handler (void)
     check_sound_buffers ();
 }
 
-static void sample16si_crux_handler (void)
-{
-    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;
-	unsigned long ratio, ratio1;
-#define INTERVAL (scaled_sample_evtime * 3)
-	cdp = audio_channel + 0;
-	ratio1 = cdp->per - cdp->evtime;
-	ratio = (ratio1 << 12) / INTERVAL;
-	if (cdp->evtime < scaled_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 < scaled_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 < scaled_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 < scaled_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 (data, 16, 1);
-	if (sound_use_filter)
-	    data = filter (data, &sound_filter_state[0]);
-	put_sound_word_right (data);
-    }
-
-    {
-	uae_u32 data = SBASEVAL16 (1) + data1;
-	FINISH_DATA (data, 16, 1);
-	if (sound_use_filter)
-	    data = filter (data, &sound_filter_state[1]);
-	put_sound_word_left (data);
-    }
-    check_sound_buffers ();
-}
-
-static void sample16si_rh_handler (void)
-{
-    unsigned long delta, ratio;
-
-    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 (data, 16, 1);
-	if (sound_use_filter)
-	    data = filter (data, &sound_filter_state[0]);
-	put_sound_word_right (data);
-    }
-
-    {
-	uae_u32 data = SBASEVAL16 (1) + data1;
-	FINISH_DATA (data, 16, 1);
-	if (sound_use_filter)
-	    data = filter (data, &sound_filter_state[1]);
-	put_sound_word_left (data);
-    }
-    check_sound_buffers ();
-}
-
-#else
-void sample16s_handler (void)
-{
-    sample16_handler ();
-}
-static void sample16si_crux_handler (void)
-{
-    sample16i_crux_handler ();
-}
-static void sample16si_rh_handler (void)
-{
-    sample16i_rh_handler ();
-}
-#endif
-
 void switch_audio_interpol (void)
 {
     if (currprefs.sound_interpol == 0) {
 	changed_prefs.sound_interpol = 1;
-	write_log ("Interpol on: rh\n");
+	write_log ("Resampler on: sinc\n");
     } else if (currprefs.sound_interpol == 1) {
 	changed_prefs.sound_interpol = 2;
-	write_log ("Interpol on: crux\n");
-    } else if (currprefs.sound_interpol == 2) {
-	changed_prefs.sound_interpol = 3;
-	write_log ("Interpol on: sinc\n");
-    } else if (currprefs.sound_interpol == 3) {
-	changed_prefs.sound_interpol = 4;
-	write_log ("Interpol on: anti\n");
+	write_log ("Resampler on: anti\n");
     } else {
 	changed_prefs.sound_interpol = 0;
-	write_log ("Interpol off\n");
+	write_log ("Resampler off\n");
     }
     return;
 }
@@ -905,12 +552,8 @@ static void audio_handler (unsigned int
 
 	    /* Volume attachment? */
 	    if (audav) {
-		if (nr < 3) {
+		if (nr < 3)
 		    (cdp+1)->vol = cdp->dat;
-#ifndef MULTIPLICATION_PROFITABLE
-		    (cdp+1)->voltbl = sound_table[cdp->dat];
-#endif
-		}
 	    }
 	}
 	break;
@@ -961,11 +604,6 @@ void audio_reset (void)
 	for (i = 0; i < 4; i++)
 	    audio_channel[i].dmaen = (dmacon & 0x200) && (dmacon & (1 << i));
 
-#ifndef MULTIPLICATION_PROFITABLE
-    for (i = 0; i < 4; i++)
-	audio_channel[i].voltbl = sound_table[audio_channel[i].vol];
-#endif
-
     last_cycles = get_cycles ();
     next_sample_evtime = scaled_sample_evtime;
     schedule_audio ();
@@ -1050,33 +688,20 @@ void check_prefs_changed_audio (void)
     led_filter_audio();
 
     /* Select the right interpolation method.  */
-    if (sample_handler == sample16_handler
-	|| sample_handler == sample16i_crux_handler
-	|| sample_handler == sample16i_rh_handler
-	|| sample_handler == sample16i_sinc_handler
-	|| sample_handler == sample16i_anti_handler)
+    if (sample_handler == sample16s_handler
+	|| sample_handler == sample16si_sinc_handler
+	|| sample_handler == sample16si_anti_handler)
     {
-	sample_handler = (currprefs.sound_interpol == 0 ? sample16_handler
-			  : currprefs.sound_interpol == 1 ? sample16i_rh_handler
-			  : currprefs.sound_interpol == 2 ? sample16i_crux_handler
-			  : currprefs.sound_interpol == 3 ? sample16i_sinc_handler
-			  : sample16i_anti_handler);
-    } else if (sample_handler == sample16s_handler
-	       || sample_handler == sample16si_crux_handler
-	       || sample_handler == sample16si_rh_handler
-	       || sample_handler == sample16si_sinc_handler
-	       || sample_handler == sample16si_anti_handler)
 	sample_handler = (currprefs.sound_interpol == 0 ? sample16s_handler
-			  : currprefs.sound_interpol == 1 ? sample16si_rh_handler
-			  : currprefs.sound_interpol == 2 ? sample16si_crux_handler
-			  : currprefs.sound_interpol == 3 ? sample16si_sinc_handler
+			  : currprefs.sound_interpol == 1 ? sample16si_sinc_handler
 			  : sample16si_anti_handler);
+    }
     sample_prehandler = NULL;
-    if (currprefs.sound_interpol == 3) {
+    if (currprefs.sound_interpol == 1) {
 	sound_use_filter_sinc = sound_use_filter;
 	sound_use_filter = 0;
 	sample_prehandler = sinc_prehandler;
-    } else if (currprefs.sound_interpol == 4) {
+    } else if (currprefs.sound_interpol == 2) {
 	sample_prehandler = anti_prehandler;
     }
 }
@@ -1265,9 +890,6 @@ void AUDxVOL (int nr, uae_u16 v)
     update_audio ();
 
     audio_channel[nr].vol = v2;
-#ifndef MULTIPLICATION_PROFITABLE
-    audio_channel[nr].voltbl = sound_table[v2];
-#endif
 }
 
 void update_adkmasks (void)