Updated Highly Quixotic to include new, more accurate sound emulation.

Frameworks/HighlyQuixotic/HighlyQuixotic.xcodeproj/project.pbxproj

@@ -7,12 +7,12 @@
 	objects = {
 
 /* Begin PBXBuildFile section */
+		8334C69520B40367001DA8DC /* qsound_ctr.c in Sources */ = {isa = PBXBuildFile; fileRef = 8334C69320B40367001DA8DC /* qsound_ctr.c */; };
+		8334C69620B40367001DA8DC /* qsound_ctr.h in Headers */ = {isa = PBXBuildFile; fileRef = 8334C69420B40367001DA8DC /* qsound_ctr.h */; };
 		834378EC17F96E2600584396 /* InfoPlist.strings in Resources */ = {isa = PBXBuildFile; fileRef = 834378EA17F96E2600584396 /* InfoPlist.strings */; };
 		8343791E17F96EA600584396 /* emuconfig.h in Headers */ = {isa = PBXBuildFile; fileRef = 8343791517F96EA600584396 /* emuconfig.h */; settings = {ATTRIBUTES = (Public, ); }; };
 		8343791F17F96EA600584396 /* kabuki.c in Sources */ = {isa = PBXBuildFile; fileRef = 8343791617F96EA600584396 /* kabuki.c */; };
 		8343792017F96EA600584396 /* kabuki.h in Headers */ = {isa = PBXBuildFile; fileRef = 8343791717F96EA600584396 /* kabuki.h */; };
-		8343792117F96EA600584396 /* qmix.c in Sources */ = {isa = PBXBuildFile; fileRef = 8343791817F96EA600584396 /* qmix.c */; };
-		8343792217F96EA600584396 /* qmix.h in Headers */ = {isa = PBXBuildFile; fileRef = 8343791917F96EA600584396 /* qmix.h */; };
 		8343792317F96EA600584396 /* qsound.c in Sources */ = {isa = PBXBuildFile; fileRef = 8343791A17F96EA600584396 /* qsound.c */; };
 		8343792417F96EA600584396 /* qsound.h in Headers */ = {isa = PBXBuildFile; fileRef = 8343791B17F96EA600584396 /* qsound.h */; settings = {ATTRIBUTES = (Public, ); }; };
 		8343792517F96EA600584396 /* z80.c in Sources */ = {isa = PBXBuildFile; fileRef = 8343791C17F96EA600584396 /* z80.c */; };
@@ -20,6 +20,8 @@
 /* End PBXBuildFile section */
 
 /* Begin PBXFileReference section */
+		8334C69320B40367001DA8DC /* qsound_ctr.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = qsound_ctr.c; path = Core/qsound_ctr.c; sourceTree = "<group>"; };
+		8334C69420B40367001DA8DC /* qsound_ctr.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = qsound_ctr.h; path = Core/qsound_ctr.h; sourceTree = "<group>"; };
 		833F682E1CDBCAB100AFB9F0 /* es */ = {isa = PBXFileReference; lastKnownFileType = text.plist.strings; name = es; path = es.lproj/InfoPlist.strings; sourceTree = "<group>"; };
 		834378DE17F96E2600584396 /* HighlyQuixotic.framework */ = {isa = PBXFileReference; explicitFileType = wrapper.framework; includeInIndex = 0; path = HighlyQuixotic.framework; sourceTree = BUILT_PRODUCTS_DIR; };
 		834378E917F96E2600584396 /* HighlyQuixotic-Info.plist */ = {isa = PBXFileReference; lastKnownFileType = text.plist.xml; path = "HighlyQuixotic-Info.plist"; sourceTree = "<group>"; };
@@ -27,8 +29,6 @@
 		8343791517F96EA600584396 /* emuconfig.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = emuconfig.h; path = Core/emuconfig.h; sourceTree = "<group>"; };
 		8343791617F96EA600584396 /* kabuki.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = kabuki.c; path = Core/kabuki.c; sourceTree = "<group>"; };
 		8343791717F96EA600584396 /* kabuki.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = kabuki.h; path = Core/kabuki.h; sourceTree = "<group>"; };
-		8343791817F96EA600584396 /* qmix.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = qmix.c; path = Core/qmix.c; sourceTree = "<group>"; };
-		8343791917F96EA600584396 /* qmix.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = qmix.h; path = Core/qmix.h; sourceTree = "<group>"; };
 		8343791A17F96EA600584396 /* qsound.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = qsound.c; path = Core/qsound.c; sourceTree = "<group>"; };
 		8343791B17F96EA600584396 /* qsound.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = qsound.h; path = Core/qsound.h; sourceTree = "<group>"; };
 		8343791C17F96EA600584396 /* z80.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = z80.c; path = Core/z80.c; sourceTree = "<group>"; };
@@ -84,10 +84,10 @@
 				8343791517F96EA600584396 /* emuconfig.h */,
 				8343791617F96EA600584396 /* kabuki.c */,
 				8343791717F96EA600584396 /* kabuki.h */,
-				8343791817F96EA600584396 /* qmix.c */,
-				8343791917F96EA600584396 /* qmix.h */,
 				8343791A17F96EA600584396 /* qsound.c */,
 				8343791B17F96EA600584396 /* qsound.h */,
+				8334C69320B40367001DA8DC /* qsound_ctr.c */,
+				8334C69420B40367001DA8DC /* qsound_ctr.h */,
 				8343791C17F96EA600584396 /* z80.c */,
 				8343791D17F96EA600584396 /* z80.h */,
 				834378E817F96E2600584396 /* Supporting Files */,
@@ -113,8 +113,8 @@
 			files = (
 				8343792417F96EA600584396 /* qsound.h in Headers */,
 				8343791E17F96EA600584396 /* emuconfig.h in Headers */,
-				8343792217F96EA600584396 /* qmix.h in Headers */,
 				8343792017F96EA600584396 /* kabuki.h in Headers */,
+				8334C69620B40367001DA8DC /* qsound_ctr.h in Headers */,
 				8343792617F96EA600584396 /* z80.h in Headers */,
 			);
 			runOnlyForDeploymentPostprocessing = 0;
@@ -191,8 +191,8 @@
 			files = (
 				8343792517F96EA600584396 /* z80.c in Sources */,
 				8343792317F96EA600584396 /* qsound.c in Sources */,
-				8343792117F96EA600584396 /* qmix.c in Sources */,
 				8343791F17F96EA600584396 /* kabuki.c in Sources */,
+				8334C69520B40367001DA8DC /* qsound_ctr.c in Sources */,
 			);
 			runOnlyForDeploymentPostprocessing = 0;
 		};

Frameworks/HighlyQuixotic/HighlyQuixotic/Core/qsound.c

@@ -12,7 +12,7 @@
 
 #include "z80.h"
 #include "kabuki.h"
-#include "qmix.h"
+#include "qsound_ctr.h"
 
 /////////////////////////////////////////////////////////////////////////////
 //
@@ -77,7 +77,6 @@ sint32 EMU_CALL qsound_init(void) {
   qsound_endian_check();
   qsound_size_check();
   r = z80_init(); if(r) return r;
-  r = qmix_init(); if(r) return r;
   library_was_initialized = 1;
   return 0;
 }
@@ -148,7 +147,7 @@ uint32 EMU_CALL qsound_get_state_size(void) {
   size += sizeof(struct Z80_MEMORY_MAP) * qsound_map_read_entries;
   size += sizeof(struct Z80_MEMORY_MAP) * qsound_map_write_entries;
   size += z80_get_state_size();
-  size += qmix_get_state_size();
+  size += device_get_qsound_ctr_state_size();
   return size;
 }
 
@@ -166,7 +165,6 @@ void EMU_CALL qsound_set_rates(
 ) {
   QSOUNDSTATE->cycles_per_irq    = (z80 + (irq   /2)) / irq   ;
   QSOUNDSTATE->cycles_per_sample = (z80 + (sample/2)) / sample;
-  qmix_set_sample_rate(QMIXSTATE, sample);
 }
 
 void EMU_CALL qsound_clear_state(void *state) {
@@ -179,7 +177,7 @@ void EMU_CALL qsound_clear_state(void *state) {
   QSOUNDSTATE->map_read   = (void*)(((char*)state)+offset); offset += sizeof(struct Z80_MEMORY_MAP) * qsound_map_read_entries;
   QSOUNDSTATE->map_write  = (void*)(((char*)state)+offset); offset += sizeof(struct Z80_MEMORY_MAP) * qsound_map_write_entries;
   QSOUNDSTATE->z80_state  = (void*)(((char*)state)+offset); offset += z80_get_state_size();
-  QSOUNDSTATE->qmix_state = (void*)(((char*)state)+offset); offset += qmix_get_state_size();
+  QSOUNDSTATE->qmix_state = (void*)(((char*)state)+offset); offset += device_get_qsound_ctr_state_size();
   //
   // Set other variables
   //
@@ -200,10 +198,10 @@ void EMU_CALL qsound_clear_state(void *state) {
     qsound_map_out
   );
 
-  qmix_clear_state(QMIXSTATE);
+  device_start_qsound_ctr(60000000, QMIXSTATE);
 
   // Set rates
-  qsound_set_rates(QSOUNDSTATE, 8000000, 250, 44100);
+  qsound_set_rates(QSOUNDSTATE, 8000000, 250, 24038);
   QSOUNDSTATE->cycles_until_irq = QSOUNDSTATE->cycles_per_irq;
 
   // Done
@@ -291,7 +289,7 @@ static void EMU_CALL flush_sound(struct QSOUND_STATE *state) {
   }
   if(!samples_needed) return;
 
-  qmix_render(QMIXSTATE, state->sound_buffer, samples_needed);
+  qsoundc_update(QMIXSTATE, samples_needed, state->sound_buffer);
 
   if(state->sound_buffer) state->sound_buffer += 2 * samples_needed;
   state->sound_buffer_samples_free -= samples_needed;
@@ -302,24 +300,21 @@ static void EMU_CALL flush_sound(struct QSOUND_STATE *state) {
 // Register reads/writes
 //
 static uint8 EMU_CALL qsound_status_r(void *state, uint16 a) {
-  return 0x80;
+  flush_sound(QSOUNDSTATE);
+  return qsoundc_r(QMIXSTATE, a);
 }
 
 static void EMU_CALL qsound_data_h_w(void *state, uint16 a, uint8 d) {
-  QSOUNDSTATE->qsound_data_latch =
+  qsoundc_w(QMIXSTATE, 0, d);
-    (QSOUNDSTATE->qsound_data_latch & 0x00FF) |
-    (((uint32)d) << 8);
 }
 
 static void EMU_CALL qsound_data_l_w(void *state, uint16 a, uint8 d) {
-  QSOUNDSTATE->qsound_data_latch =
+  qsoundc_w(QMIXSTATE, 1, d);
-    (QSOUNDSTATE->qsound_data_latch & 0xFF00) |
-    ((uint32)d);
 }
 
 static void EMU_CALL qsound_cmd_w(void *state, uint16 a, uint8 d) {
   flush_sound(QSOUNDSTATE);
-  qmix_command(QMIXSTATE, d, QSOUNDSTATE->qsound_data_latch);
+  qsoundc_w(QMIXSTATE, 2, d);
 }
 
 /////////////////////////////////////////////////////////////////////////////
@@ -625,7 +620,8 @@ void EMU_CALL qsound_set_kabuki_key(void *state,
 }
 
 void EMU_CALL qsound_set_sample_rom(void *state, void *rom, uint32 size) {
-  qmix_set_sample_rom(QMIXSTATE, rom, size);
+  qsoundc_alloc_rom(QMIXSTATE, size);
+  qsoundc_write_rom(QMIXSTATE, 0, size, (const UINT8 *)rom);
 }
 
 /////////////////////////////////////////////////////////////////////////////

Frameworks/HighlyQuixotic/HighlyQuixotic/Core/qsound_ctr.c

@@ -0,0 +1,888 @@
+/*
+
+	Capcom DL-1425 QSound emulator
+	==============================
+	
+	by superctr (Ian Karlsson)
+	with thanks to Valley Bell
+
+	2018-05-12 - 2018-05-15
+
+*/
+
+#include <stdlib.h>
+#include <string.h>	// for memset
+#include <math.h>
+#include <stdint.h>
+#include "qsound_ctr.h"
+
+#ifdef _MSC_VER
+#define INLINE __inline
+#else
+#define INLINE inline
+#endif
+
+#define CLAMP(x, low, high)  (((x) > (high)) ? (high) : (((x) < (low)) ? (low) : (x)))
+
+INLINE UINT32 pow2_mask(UINT32 v)
+{
+	if (v == 0)
+		return 0;
+	v--;
+	v |= (v >> 1);
+	v |= (v >> 2);
+	v |= (v >> 4);
+	v |= (v >> 8);
+	v |= (v >> 16);
+	return v;
+}
+
+struct qsound_voice {
+	UINT16 bank;
+	INT16 addr; // top word is the sample address
+	UINT16 phase;
+	UINT16 rate;
+	INT16 loop_len;
+	INT16 end_addr;
+	INT16 volume;
+	INT16 echo;
+};
+
+struct qsound_adpcm {
+	UINT16 start_addr;
+	UINT16 end_addr;
+	UINT16 bank;
+	INT16 volume;
+	UINT16 flag;
+	INT16 cur_vol;
+	INT16 step_size;
+	UINT16 cur_addr;
+};
+
+// Q1 Filter
+struct qsound_fir {
+	int tap_count;	// usually 95
+	int delay_pos;
+	INT16 table_pos;
+	INT16 taps[95];
+	INT16 delay_line[95];
+};
+
+// Delay line
+struct qsound_delay {
+	INT16 delay;
+	INT16 volume;
+	INT16 write_pos;
+	INT16 read_pos;
+	INT16 delay_line[51];
+};
+
+struct qsound_echo {
+	UINT16 end_pos;
+	
+	INT16 feedback;
+	INT16 length;
+	INT16 last_sample;
+	INT16 delay_line[1024];
+	INT16 delay_pos;
+};
+
+struct qsound_chip {
+
+	UINT8* romData;
+	UINT32 romSize;
+	UINT32 romMask;
+	UINT32 muteMask;
+	
+	// ==================================================== //
+
+	UINT16 data_latch;
+	INT16 out[2];
+
+	INT16 pan_tables[2][2][98];
+
+	struct qsound_voice voice[16];
+	struct qsound_adpcm adpcm[3];
+
+	UINT16 voice_pan[16+3];
+	INT16 voice_output[16+3];
+
+	struct qsound_echo echo;
+
+	struct qsound_fir filter[2];
+	struct qsound_fir alt_filter[2];
+
+	struct qsound_delay wet[2];
+	struct qsound_delay dry[2];
+
+	UINT16 state;
+	UINT16 next_state;
+
+	UINT16 delay_update;
+
+	int state_counter;
+	UINT8 ready_flag;
+
+	UINT16 *register_map[256];
+};
+
+static void init_pan_tables(struct qsound_chip *chip);
+static void init_register_map(struct qsound_chip *chip);
+static void update_sample(struct qsound_chip *chip);
+
+static void state_init(struct qsound_chip *chip);
+static void state_refresh_filter_1(struct qsound_chip *chip);
+static void state_refresh_filter_2(struct qsound_chip *chip);
+static void state_normal_update(struct qsound_chip *chip);
+
+INLINE INT16 get_sample(struct qsound_chip *chip, UINT16 bank,UINT16 address);
+INLINE const INT16* get_filter_table(struct qsound_chip *chip, UINT16 offset);
+INLINE INT16 pcm_update(struct qsound_chip *chip, int voice_no, INT32 *echo_out);
+INLINE void adpcm_update(struct qsound_chip *chip, int voice_no, int nibble);
+INLINE INT16 echo(struct qsound_echo *r,INT32 input);
+INLINE INT32 fir(struct qsound_fir *f, INT16 input);
+INLINE INT32 delay(struct qsound_delay *d, INT32 input);
+INLINE void delay_update(struct qsound_delay *d);
+
+// ****************************************************************************
+
+UINT32 device_get_qsound_ctr_state_size()
+{
+	return sizeof(struct qsound_chip);
+}
+
+UINT32 device_start_qsound_ctr(UINT32 clock, void* retDevInf)
+{
+	struct qsound_chip* chip = (struct qsound_chip *) retDevInf;
+
+	memset(chip, 0, sizeof(struct qsound_chip));
+	
+	chip->romData = NULL;
+	chip->romSize = 0x00;
+	chip->romMask = 0x00;
+	
+	qsoundc_set_mute_mask(chip, 0x00000);
+	
+	init_pan_tables(chip);
+	init_register_map(chip);
+
+	return clock / 2 / 1248;
+}
+
+void device_stop_qsound_ctr(void* info)
+{
+	struct qsound_chip* chip = (struct qsound_chip*)info;
+	
+	free(chip->romData);
+	free(chip);
+	
+	return;
+}
+
+void device_reset_qsound_ctr(void* info)
+{
+	struct qsound_chip* chip = (struct qsound_chip*)info;
+	
+	chip->ready_flag = 0;
+	chip->out[0] = chip->out[1] = 0;
+	chip->state = 0;
+	chip->state_counter = 0;
+	
+	return;
+}
+
+UINT8 qsoundc_r(void* info, UINT8 offset)
+{
+	struct qsound_chip* chip = (struct qsound_chip*)info;
+	
+	return chip->ready_flag;
+}
+
+void qsoundc_w(void* info, UINT8 offset, UINT8 data)
+{
+	struct qsound_chip* chip = (struct qsound_chip*)info;
+	
+	switch (offset)
+	{
+		case 0:
+			chip->data_latch = (chip->data_latch & 0x00ff) | (data << 8);
+			break;
+		case 1:
+			chip->data_latch = (chip->data_latch & 0xff00) | data;
+			break;
+		case 2:
+			qsoundc_write_data(chip, data, chip->data_latch);
+			break;
+		default:
+			break;
+	}
+	
+	return;
+}
+
+void qsoundc_write_data(void* info, UINT8 address, UINT16 data)
+{
+	struct qsound_chip* chip = (struct qsound_chip*)info;
+	
+	UINT16 *destination = chip->register_map[address];
+	if(destination)
+		*destination = data;
+	chip->ready_flag = 0;
+
+	return;
+}
+
+void qsoundc_update(void* param, UINT32 samples, INT16* output)
+{
+	struct qsound_chip* chip = (struct qsound_chip*)param;
+	UINT32 curSmpl;
+
+	if (output)
+	{
+		memset(output, 0, samples * 2 * sizeof(*output));
+
+		for (curSmpl = 0; curSmpl < samples; curSmpl++)
+		{
+			update_sample(chip);
+			output[curSmpl * 2 + 0] = chip->out[0];
+			output[curSmpl * 2 + 1] = chip->out[1];
+		}
+	}
+	else
+	{
+		for (curSmpl = 0; curSmpl < samples; curSmpl++)
+		{
+			update_sample(chip);
+		}
+	}
+	
+	return;
+}
+
+void qsoundc_alloc_rom(void* info, UINT32 memsize)
+{
+	struct qsound_chip* chip = (struct qsound_chip*)info;
+	
+	if (chip->romSize == memsize)
+		return;
+	
+	chip->romData = (UINT8*)realloc(chip->romData, memsize);
+	chip->romSize = memsize;
+	chip->romMask = pow2_mask(memsize);
+	memset(chip->romData, 0xFF, memsize);
+	
+	return;
+}
+
+void qsoundc_write_rom(void* info, UINT32 offset, UINT32 length, const UINT8* data)
+{
+	struct qsound_chip* chip = (struct qsound_chip*)info;
+	
+	if (offset > chip->romSize)
+		return;
+	if (offset + length > chip->romSize)
+		length = chip->romSize - offset;
+	
+	memcpy(chip->romData + offset, data, length);
+	
+	return;
+}
+
+void qsoundc_set_mute_mask(void* info, UINT32 MuteMask)
+{
+	struct qsound_chip* chip = (struct qsound_chip*)info;
+	
+	chip->muteMask = MuteMask;
+	
+	return;
+}
+
+// ============================================================================
+
+static const INT16 qsound_dry_mix_table[33] = {
+	-16384,-16384,-16384,-16384,-16384,-16384,-16384,-16384,
+	-16384,-16384,-16384,-16384,-16384,-16384,-16384,-16384,
+	-16384,-14746,-13107,-11633,-10486,-9175,-8520,-7209,
+	-6226,-5226,-4588,-3768,-3277,-2703,-2130,-1802,
+	0
+};
+
+static const INT16 qsound_wet_mix_table[33] = {
+	0,-1638,-1966,-2458,-2949,-3441,-4096,-4669,
+	-4915,-5120,-5489,-6144,-7537,-8831,-9339,-9830,
+	-10240,-10322,-10486,-10568,-10650,-11796,-12288,-12288,
+	-12534,-12648,-12780,-12829,-12943,-13107,-13418,-14090,
+	-16384
+};
+
+static const INT16 qsound_linear_mix_table[33] = {
+	-16379,-16338,-16257,-16135,-15973,-15772,-15531,-15251,
+	-14934,-14580,-14189,-13763,-13303,-12810,-12284,-11729,
+	-11729,-11144,-10531,-9893,-9229,-8543,-7836,-7109,
+	-6364,-5604,-4829,-4043,-3246,-2442,-1631,-817,
+	0
+};
+
+static const INT16 qsound_filter_data[5][95] = {
+	{	// d53 - 0
+		0,0,0,6,44,-24,-53,-10,59,-40,-27,1,39,-27,56,127,174,36,-13,49,
+		212,142,143,-73,-20,66,-108,-117,-399,-265,-392,-569,-473,-71,95,-319,-218,-230,331,638,
+		449,477,-180,532,1107,750,9899,3828,-2418,1071,-176,191,-431,64,117,-150,-274,-97,-238,165,
+		166,250,-19,4,37,204,186,-6,140,-77,-1,1,18,-10,-151,-149,-103,-9,55,23,
+		-102,-97,-11,13,-48,-27,5,18,-61,-30,64,72,0,0,0,
+	},
+	{	// db2 - 1 - default left filter
+		0,0,0,85,24,-76,-123,-86,-29,-14,-20,-7,6,-28,-87,-89,-5,100,154,160,
+		150,118,41,-48,-78,-23,59,83,-2,-176,-333,-344,-203,-66,-39,2,224,495,495,280,
+		432,1340,2483,5377,1905,658,0,97,347,285,35,-95,-78,-82,-151,-192,-171,-149,-147,-113,
+		-22,71,118,129,127,110,71,31,20,36,46,23,-27,-63,-53,-21,-19,-60,-92,-69,
+		-12,25,29,30,40,41,29,30,46,39,-15,-74,0,0,0,
+	},
+	{	// e11 - 2 - default right filter
+		0,0,0,23,42,47,29,10,2,-14,-54,-92,-93,-70,-64,-77,-57,18,94,113,
+		87,69,67,50,25,29,58,62,24,-39,-131,-256,-325,-234,-45,58,78,223,485,496,
+		127,6,857,2283,2683,4928,1328,132,79,314,189,-80,-90,35,-21,-186,-195,-99,-136,-258,
+		-189,82,257,185,53,41,84,68,38,63,77,14,-60,-71,-71,-120,-151,-84,14,29,
+		-8,7,66,69,12,-3,54,92,52,-6,-15,-2,0,0,0,
+	},
+	{	// e70 - 3
+		0,0,0,2,-28,-37,-17,0,-9,-22,-3,35,52,39,20,7,-6,2,55,121,
+		129,67,8,1,9,-6,-16,16,66,96,118,130,75,-47,-92,43,223,239,151,219,
+		440,475,226,206,940,2100,2663,4980,865,49,-33,186,231,103,42,114,191,184,116,29,
+		-47,-72,-21,60,96,68,31,32,63,87,76,39,7,14,55,85,67,18,-12,-3,
+		21,34,29,6,-27,-49,-37,-2,16,0,-21,-16,0,0,0,
+	},
+	{	// ecf - 4
+		0,0,0,48,7,-22,-29,-10,24,54,59,29,-36,-117,-185,-213,-185,-99,13,90,
+		83,24,-5,23,53,47,38,56,67,57,75,107,16,-242,-440,-355,-120,-33,-47,152,
+		501,472,-57,-292,544,1937,2277,6145,1240,153,47,200,152,36,64,134,74,-82,-208,-266,
+		-268,-188,-42,65,74,56,89,133,114,44,-3,-1,17,29,29,-2,-76,-156,-187,-151,
+		-85,-31,-5,7,20,32,24,-5,-20,6,48,62,0,0,0,
+	}
+};
+
+static const INT16 qsound_filter_data2[209] = {
+	// f2e - following 95 values used for "disable output" filter
+	0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
+	0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
+	0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
+	0,0,0,0,0,0,0,0,0,
+	
+	// f73 - following 45 values used for "mode 2" filter (overlaps with f2e)
+	0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
+	0,0,0,0,0,0,
+	-371,-196,-268,-512,-303,-315,-184,-76,276,-256,298,196,990,236,1114,-126,4377,6549,791,
+	
+	// fa0 - filtering disabled (for 95-taps) (use fa3 or fa4 for mode2 filters)
+	0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
+	0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
+	0,0,0,0,0,0,0,-16384,0,0,0,0,0,0,0,0,0,0,0,0,
+	0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
+	0,0,0,0,0,0,0,0,0,0,0,0,0,0,0
+};
+
+static const INT16 adpcm_step_table[16] = {
+	154, 154, 128, 102, 77, 58, 58, 58,
+	58, 58, 58, 58, 77, 102, 128, 154
+};
+
+// DSP states
+enum {
+	STATE_INIT1		= 0x288,
+	STATE_INIT2		= 0x61a,
+	STATE_REFRESH1	= 0x039,
+	STATE_REFRESH2	= 0x04f,
+	STATE_NORMAL1	= 0x314,
+	STATE_NORMAL2 	= 0x6b2,
+};
+
+enum {
+	PANTBL_LEFT		= 0,
+	PANTBL_RIGHT	= 1,
+	PANTBL_DRY		= 0,
+	PANTBL_WET		= 1,
+};
+
+static void init_pan_tables(struct qsound_chip *chip)
+{
+	int i;
+	for(i=0;i<33;i++)
+	{
+		// dry mixing levels
+		chip->pan_tables[PANTBL_LEFT][PANTBL_DRY][i] = qsound_dry_mix_table[i]; 
+		chip->pan_tables[PANTBL_RIGHT][PANTBL_DRY][i] = qsound_dry_mix_table[32-i];
+		// wet mixing levels
+		chip->pan_tables[PANTBL_LEFT][PANTBL_WET][i] = qsound_wet_mix_table[i]; 
+		chip->pan_tables[PANTBL_RIGHT][PANTBL_WET][i] = qsound_wet_mix_table[32-i];
+		// linear panning, only for dry component. wet component is muted.
+		chip->pan_tables[PANTBL_LEFT][PANTBL_DRY][i+0x30] = qsound_linear_mix_table[i];
+		chip->pan_tables[PANTBL_RIGHT][PANTBL_DRY][i+0x30] = qsound_linear_mix_table[32-i]; 
+	}
+}
+
+static void init_register_map(struct qsound_chip *chip)
+{
+	int i;
+	
+	// unused registers
+	for(i=0;i<256;i++)
+		chip->register_map[i] = NULL;
+	
+	// PCM registers
+	for(i=0;i<16;i++) // PCM voices
+	{
+		chip->register_map[(i<<3)+0] = (UINT16*)&chip->voice[(i+1)%16].bank; // Bank applies to the next channel
+		chip->register_map[(i<<3)+1] = (UINT16*)&chip->voice[i].addr; // Current sample position and start position.
+		chip->register_map[(i<<3)+2] = (UINT16*)&chip->voice[i].rate; // 4.12 fixed point decimal.
+		chip->register_map[(i<<3)+3] = (UINT16*)&chip->voice[i].phase;
+		chip->register_map[(i<<3)+4] = (UINT16*)&chip->voice[i].loop_len;
+		chip->register_map[(i<<3)+5] = (UINT16*)&chip->voice[i].end_addr;
+		chip->register_map[(i<<3)+6] = (UINT16*)&chip->voice[i].volume;
+		chip->register_map[(i<<3)+7] = NULL;	// unused
+		chip->register_map[i+0x80] = (UINT16*)&chip->voice_pan[i];
+		chip->register_map[i+0xba] = (UINT16*)&chip->voice[i].echo;
+	}
+	
+	// ADPCM registers
+	for(i=0;i<3;i++) // ADPCM voices
+	{
+		// ADPCM sample rate is fixed to 8khz. (one channel is updated every third sample)
+		chip->register_map[(i<<2)+0xca] = (UINT16*)&chip->adpcm[i].start_addr;
+		chip->register_map[(i<<2)+0xcb] = (UINT16*)&chip->adpcm[i].end_addr;
+		chip->register_map[(i<<2)+0xcc] = (UINT16*)&chip->adpcm[i].bank;
+		chip->register_map[(i<<2)+0xcd] = (UINT16*)&chip->adpcm[i].volume;
+		chip->register_map[i+0xd6] = (UINT16*)&chip->adpcm[i].flag; // non-zero to start ADPCM playback
+		chip->register_map[i+0x90] = (UINT16*)&chip->voice_pan[16+i];
+	}
+	
+	// QSound registers
+	chip->register_map[0x93] = (UINT16*)&chip->echo.feedback;
+	chip->register_map[0xd9] = (UINT16*)&chip->echo.end_pos;
+	chip->register_map[0xe2] = (UINT16*)&chip->delay_update; // non-zero to update delays
+	chip->register_map[0xe3] = (UINT16*)&chip->next_state;
+	for(i=0;i<2;i++)  // left, right
+	{
+		// Wet
+		chip->register_map[(i<<1)+0xda] = (UINT16*)&chip->filter[i].table_pos;
+		chip->register_map[(i<<1)+0xde] = (UINT16*)&chip->wet[i].delay;
+		chip->register_map[(i<<1)+0xe4] = (UINT16*)&chip->wet[i].volume;
+		// Dry
+		chip->register_map[(i<<1)+0xdb] = (UINT16*)&chip->alt_filter[i].table_pos;
+		chip->register_map[(i<<1)+0xdf] = (UINT16*)&chip->dry[i].delay;
+		chip->register_map[(i<<1)+0xe5] = (UINT16*)&chip->dry[i].volume;
+	}
+}
+
+INLINE INT16 get_sample(struct qsound_chip *chip, UINT16 bank,UINT16 address)
+{
+	UINT32 rom_addr;
+	UINT8 sample_data;
+
+	if (! chip->romMask)
+		return 0;	// no ROM loaded
+	if (! (bank & 0x8000))
+		return 0;	// ignore attempts to read from DSP program ROM
+
+	bank &= 0x7FFF;
+	rom_addr = (bank << 16) | (address << 0);
+	
+	sample_data = chip->romData[rom_addr];
+	
+	return (INT16)((sample_data << 8) | (sample_data << 0));	// MAME currently expands the 8 bit ROM data to 16 bits this way.
+}
+
+INLINE const INT16* get_filter_table(struct qsound_chip *chip, UINT16 offset)
+{
+	int index;
+	
+	if (offset >= 0xf2e && offset < 0xfff)
+		return &qsound_filter_data2[offset-0xf2e];	// overlapping filter data
+	
+	index = (offset-0xd53)/95;
+	if(index >= 0 && index < 5)
+		return qsound_filter_data[index];	// normal tables
+	
+	return NULL;	// no filter found.
+}
+
+/********************************************************************/
+
+// updates one DSP sample
+static void update_sample(struct qsound_chip *chip)
+{
+	switch(chip->state)
+	{
+		default:
+		case STATE_INIT1:
+		case STATE_INIT2:
+			state_init(chip);
+			return;
+		case STATE_REFRESH1:
+			state_refresh_filter_1(chip);
+			return;
+		case STATE_REFRESH2:
+			state_refresh_filter_2(chip);
+			return;
+		case STATE_NORMAL1:
+		case STATE_NORMAL2:
+			state_normal_update(chip);
+			return;
+	}
+}
+
+// Initialization routine
+static void state_init(struct qsound_chip *chip)
+{
+	int mode = (chip->state == STATE_INIT2) ? 1 : 0;
+	int i;
+	
+	// we're busy for 4 samples, including the filter refresh.
+	if(chip->state_counter >= 2)
+	{
+		chip->state_counter = 0;
+		chip->state = chip->next_state;
+		return;
+	}
+	else if(chip->state_counter == 1)
+	{
+		chip->state_counter++;
+		return;
+	}
+	
+	memset(chip->voice, 0, sizeof(chip->voice));
+	memset(chip->adpcm, 0, sizeof(chip->adpcm));
+	memset(chip->filter, 0, sizeof(chip->filter));
+	memset(chip->alt_filter, 0, sizeof(chip->alt_filter));
+	memset(chip->wet, 0, sizeof(chip->wet));
+	memset(chip->dry, 0, sizeof(chip->dry));
+	memset(&chip->echo, 0, sizeof(chip->echo));
+	
+	for(i=0;i<19;i++)
+	{
+		chip->voice_pan[i] = 0x120;
+		chip->voice_output[i] = 0;
+	}
+
+	for(i=0;i<16;i++)
+		chip->voice[i].bank = 0x8000;
+	for(i=0;i<3;i++)
+		chip->adpcm[i].bank = 0x8000;
+	
+	if(mode == 0)
+	{
+		// mode 1
+		chip->wet[0].delay = 0;
+		chip->dry[0].delay = 46;
+		chip->wet[1].delay = 0;
+		chip->dry[1].delay = 48;
+		chip->filter[0].table_pos = 0xdb2;
+		chip->filter[1].table_pos = 0xe11;
+		chip->echo.end_pos = 0x554 + 6;
+		chip->next_state = STATE_REFRESH1;
+	}
+	else
+	{
+		// mode 2
+		chip->wet[0].delay = 1;
+		chip->dry[0].delay = 0;
+		chip->wet[1].delay = 0;
+		chip->dry[1].delay = 0;
+		chip->filter[0].table_pos = 0xf73;
+		chip->filter[1].table_pos = 0xfa4;
+		chip->alt_filter[0].table_pos = 0xf73;
+		chip->alt_filter[1].table_pos = 0xfa4;
+		chip->echo.end_pos = 0x53c + 6;
+		chip->next_state = STATE_REFRESH2;
+	}
+
+	chip->wet[0].volume = 0x3fff;
+	chip->dry[0].volume = 0x3fff;
+	chip->wet[1].volume = 0x3fff;
+	chip->dry[1].volume = 0x3fff;
+	
+	chip->delay_update = 1;
+	chip->ready_flag = 0;
+	chip->state_counter = 1;
+}
+
+// Updates filter parameters for mode 1
+static void state_refresh_filter_1(struct qsound_chip *chip)
+{
+	const INT16 *table;
+	int ch;
+	
+	for(ch=0; ch<2; ch++)
+	{
+		chip->filter[ch].delay_pos = 0;
+		chip->filter[ch].tap_count = 95;
+	
+		table = get_filter_table(chip,chip->filter[ch].table_pos);
+		if (table != NULL)
+			memcpy(chip->filter[ch].taps, table, 95 * sizeof(INT16));
+	}
+	
+	chip->state = chip->next_state = STATE_NORMAL1;
+}
+
+// Updates filter parameters for mode 2
+static void state_refresh_filter_2(struct qsound_chip *chip)
+{
+	const INT16 *table;
+	int ch;
+	
+	for(ch=0; ch<2; ch++)
+	{
+		chip->filter[ch].delay_pos = 0;
+		chip->filter[ch].tap_count = 45;
+	
+		table = get_filter_table(chip,chip->filter[ch].table_pos);
+		if (table != NULL)
+			memcpy(chip->filter[ch].taps, table, 45 * sizeof(INT16));
+		
+		chip->alt_filter[ch].delay_pos = 0;
+		chip->alt_filter[ch].tap_count = 44;
+	
+		table = get_filter_table(chip,chip->filter[ch].table_pos);
+		if (table != NULL)
+			memcpy(chip->alt_filter[ch].taps, table, 44 * sizeof(INT16));
+	}
+	
+	chip->state = chip->next_state = STATE_NORMAL2;
+}
+
+// Updates a PCM voice. There are 16 voices, each are updated every sample
+// with full rate and volume control.
+INLINE INT16 pcm_update(struct qsound_chip *chip, int voice_no, INT32 *echo_out)
+{
+	struct qsound_voice *v = &chip->voice[voice_no];
+	INT32 new_phase;
+	INT16 output;
+	
+	if (chip->muteMask & (1<<voice_no))
+		return 0;
+	
+	// Read sample from rom and apply volume
+	output = (v->volume * get_sample(chip, v->bank, v->addr))>>14;
+	
+	*echo_out += (output * v->echo)<<2;
+	
+	// Add delta to the phase and loop back if required
+	new_phase = v->rate + ((v->addr<<12) | (v->phase>>4));
+	
+	if((new_phase>>12) >= v->end_addr)
+		new_phase -= (v->loop_len<<12);
+	
+	new_phase = CLAMP(new_phase, -0x8000000, 0x7FFFFFF);
+	v->addr = new_phase>>12;
+	v->phase = (new_phase<<4)&0xffff;
+	
+	return output;
+}
+
+// Updates an ADPCM voice. There are 3 voices, one is updated every sample
+// (effectively making the ADPCM rate 1/3 of the master sample rate), and
+// volume is set when starting samples only.
+// The ADPCM algorithm is supposedly similar to Yamaha ADPCM. It also seems
+// like Capcom never used it, so this was not emulated in the earlier QSound
+// emulators.
+INLINE void adpcm_update(struct qsound_chip *chip, int voice_no, int nibble)
+{
+	struct qsound_adpcm *v = &chip->adpcm[voice_no];
+
+	INT32 delta;
+	INT8 step;
+
+	if (chip->muteMask & (1<<(16+voice_no)))
+	{
+		chip->voice_output[16+voice_no] = 0;
+		return;
+	}
+
+	if(!nibble)
+	{
+		// Mute voice when it reaches the end address.
+		if(v->start_addr == v->end_addr)
+			v->cur_vol = 0;
+		
+		// Playback start flag
+		if(v->flag)
+		{
+			chip->voice_output[16+voice_no] = 0;
+			v->flag = 0;
+			v->step_size = 10;
+			v->cur_vol = v->volume;
+			v->cur_addr = v->start_addr;
+		}
+		
+		// get top nibble
+		step = get_sample(chip, v->bank, v->cur_addr) >> 8;
+	}
+	else
+	{
+		// get bottom nibble
+		step = get_sample(chip, v->bank, v->cur_addr++) >> 4;
+	}
+
+	// shift with sign extend
+	step >>= 4;
+	
+	// delta = (0.5 + abs(v->step)) * v->step_size
+	delta = ((1+abs(step<<1)) * v->step_size)>>1;
+	if(step <= 0)
+		delta = -delta;
+	delta += chip->voice_output[16+voice_no];
+	delta = CLAMP(delta,-32768,32767);
+	
+	chip->voice_output[16+voice_no] = (delta * v->cur_vol)>>16;
+	
+	v->step_size = (adpcm_step_table[8+step] * v->step_size) >> 6;
+	v->step_size = CLAMP(v->step_size, 1, 2000);
+}
+
+// The echo effect is pretty simple. A moving average filter is used on
+// the output from the delay line to smooth samples over time. 
+INLINE INT16 echo(struct qsound_echo *r,INT32 input)
+{
+	// get average of last 2 samples from the delay line
+	INT32 new_sample;
+	INT32 old_sample = r->delay_line[r->delay_pos];
+	INT32 last_sample = r->last_sample;
+	
+	r->last_sample = old_sample;
+	old_sample = (old_sample+last_sample) >> 1;
+	
+	// add current sample to the delay line
+	new_sample = input + ((old_sample * r->feedback)<<2);
+	r->delay_line[r->delay_pos++] = new_sample>>16;
+	
+	if(r->delay_pos >= r->length)
+		r->delay_pos = 0;
+	
+	return old_sample;
+}
+
+// Process a sample update
+static void state_normal_update(struct qsound_chip *chip)
+{
+	int v, ch;
+	INT32 echo_input = 0;
+	INT16 echo_output;
+	
+	chip->ready_flag = 0x80;
+
+	// recalculate echo length
+	if(chip->state == STATE_NORMAL2)
+		chip->echo.length = chip->echo.end_pos - 0x53c;
+	else
+		chip->echo.length = chip->echo.end_pos - 0x554;
+	
+	chip->echo.length = CLAMP(chip->echo.length, 0, 1024);
+	
+	// update PCM voices
+	for(v=0; v<16; v++)
+		chip->voice_output[v] = pcm_update(chip, v, &echo_input);
+
+	// update ADPCM voices (one every third sample)
+	adpcm_update(chip, chip->state_counter % 3, chip->state_counter / 3);
+	
+	echo_output = echo(&chip->echo,echo_input);
+	
+	// now, we do the magic stuff
+	for(ch=0; ch<2; ch++)
+	{
+		// Echo is output on the unfiltered component of the left channel and
+		// the filtered component of the right channel.
+		INT32 wet = (ch == 1) ? echo_output<<16 : 0;
+		INT32 dry = (ch == 0) ? echo_output<<16 : 0;
+		INT32 output = 0;
+		
+		for(v=0; v<19; v++)
+		{
+			UINT16 pan_index = chip->voice_pan[v]-0x110;
+			if(pan_index > 97)
+				pan_index = 97;
+			
+			// Apply different volume tables on the dry and wet inputs.
+			dry -= (chip->voice_output[v] * chip->pan_tables[ch][PANTBL_DRY][pan_index])<<2;
+			wet -= (chip->voice_output[v] * chip->pan_tables[ch][PANTBL_WET][pan_index])<<2;
+		}
+		
+		// Apply FIR filter on 'wet' input
+		wet = fir(&chip->filter[ch], wet >> 16);
+		
+		// in mode 2, we do this on the 'dry' input too
+		if(chip->state == STATE_NORMAL2)
+			dry = fir(&chip->alt_filter[ch], dry >> 16);
+		
+		// output goes through a delay line and attenuation
+		output = (delay(&chip->wet[ch], wet) + delay(&chip->dry[ch], dry))<<2;
+		
+		// DSP round function
+		output = (output + 0x8000) & ~0xffff;
+		chip->out[ch] = output >> 16;
+		
+		if(chip->delay_update)
+		{
+			delay_update(&chip->wet[ch]);
+			delay_update(&chip->dry[ch]);
+		}
+	}
+	
+	chip->delay_update = 0;
+	
+	// after 6 samples, the next state is executed.
+	chip->state_counter++;
+	if(chip->state_counter > 5)
+	{
+		chip->state_counter = 0;
+		chip->state = chip->next_state;
+	}
+}
+
+// Apply the FIR filter used as the Q1 transfer function
+INLINE INT32 fir(struct qsound_fir *f, INT16 input)
+{
+	INT32 output = 0, tap = 0;
+	
+	for(; tap < (f->tap_count-1); tap++)
+	{
+		output -= (f->taps[tap] * f->delay_line[f->delay_pos++])<<2;
+		
+		if(f->delay_pos >= f->tap_count-1)
+			f->delay_pos = 0;
+	}
+	
+	output -= (f->taps[tap] * input)<<2;
+	
+	f->delay_line[f->delay_pos++] = input;
+	if(f->delay_pos >= f->tap_count-1)
+		f->delay_pos = 0;
+	
+	return output;
+}
+
+// Apply delay line and component volume
+INLINE INT32 delay(struct qsound_delay *d, INT32 input)
+{
+	INT32 output;
+	
+	d->delay_line[d->write_pos++] = input>>16;
+	if(d->write_pos >= 51)
+		d->write_pos = 0;
+
+	output = d->delay_line[d->read_pos++]*d->volume;
+	if(d->read_pos >= 51)
+		d->read_pos = 0;
+
+	return output;
+}
+
+// Update the delay read position to match new delay length
+INLINE void delay_update(struct qsound_delay *d)
+{
+	INT16 new_read_pos = (d->write_pos - d->delay) % 51;
+	if(new_read_pos < 0)
+		new_read_pos += 51;
+	
+	d->read_pos = new_read_pos;
+}

Frameworks/HighlyQuixotic/HighlyQuixotic/Core/qsound_ctr.h

@@ -0,0 +1,27 @@
+#ifndef __QSOUND_CTR_H__
+#define __QSOUND_CTR_H__
+
+#include <stdint.h>
+
+typedef int8_t INT8;
+typedef int16_t INT16;
+typedef int32_t INT32;
+typedef uint8_t UINT8;
+typedef uint16_t UINT16;
+typedef uint32_t UINT32;
+
+void qsoundc_update(void* param, UINT32 samples, INT16* output);
+UINT32 device_get_qsound_ctr_state_size();
+UINT32 device_start_qsound_ctr(UINT32 clock, void* retDevInf);
+void device_stop_qsound_ctr(void* info);
+void device_reset_qsound_ctr(void* info);
+
+UINT8 qsoundc_r(void* info, UINT8 offset);
+void qsoundc_w(void* info, UINT8 offset, UINT8 data);
+void qsoundc_write_data(void* info, UINT8 address, UINT16 data);
+
+void qsoundc_alloc_rom(void* info, UINT32 memsize);
+void qsoundc_write_rom(void* info, UINT32 offset, UINT32 length, const UINT8* data);
+void qsoundc_set_mute_mask(void* info, UINT32 MuteMask);
+
+#endif	// __QSOUND_CTR_H__

Plugins/HighlyComplete/HighlyComplete/HCDecoder.mm

@@ -1289,6 +1289,8 @@ static int usf_info(void * context, const char * name, const char * value)
 
     if ( type == 2 )
         sampleRate = 48000;
+    else if ( type == 0x41 )
+        sampleRate = 24038;
     
     tagLengthMs = info.tag_length_ms;
     tagFadeMs = info.tag_fade_ms;