Browse Source

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

master
parent
commit
be53a3dcd8
5 changed files with 938 additions and 25 deletions
  1. +8
    -8
      Frameworks/HighlyQuixotic/HighlyQuixotic.xcodeproj/project.pbxproj
  2. +13
    -17
      Frameworks/HighlyQuixotic/HighlyQuixotic/Core/qsound.c
  3. +888
    -0
      Frameworks/HighlyQuixotic/HighlyQuixotic/Core/qsound_ctr.c
  4. +27
    -0
      Frameworks/HighlyQuixotic/HighlyQuixotic/Core/qsound_ctr.h
  5. +2
    -0
      Plugins/HighlyComplete/HighlyComplete/HCDecoder.mm

+ 8
- 8
Frameworks/HighlyQuixotic/HighlyQuixotic.xcodeproj/project.pbxproj View File

@@ -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;
};


+ 13
- 17
Frameworks/HighlyQuixotic/HighlyQuixotic/Core/qsound.c View File

@@ -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 =
(QSOUNDSTATE->qsound_data_latch & 0x00FF) |
(((uint32)d) << 8);
qsoundc_w(QMIXSTATE, 0, d);
}
static void EMU_CALL qsound_data_l_w(void *state, uint16 a, uint8 d) {
QSOUNDSTATE->qsound_data_latch =
(QSOUNDSTATE->qsound_data_latch & 0xFF00) |
((uint32)d);
qsoundc_w(QMIXSTATE, 1, 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);
}
/////////////////////////////////////////////////////////////////////////////

+ 888
- 0
Frameworks/HighlyQuixotic/HighlyQuixotic/Core/qsound_ctr.c View File

@@ -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;
}

+ 27
- 0
Frameworks/HighlyQuixotic/HighlyQuixotic/Core/qsound_ctr.h View File

@@ -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__

+ 2
- 0
Plugins/HighlyComplete/HighlyComplete/HCDecoder.mm View File

@@ -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;


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