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-rw-r--r--distrib/sdl-1.2.15/src/audio/dc/SDL_dcaudio.c246
-rw-r--r--distrib/sdl-1.2.15/src/audio/dc/SDL_dcaudio.h41
-rw-r--r--distrib/sdl-1.2.15/src/audio/dc/aica.c271
-rw-r--r--distrib/sdl-1.2.15/src/audio/dc/aica.h40
4 files changed, 598 insertions, 0 deletions
diff --git a/distrib/sdl-1.2.15/src/audio/dc/SDL_dcaudio.c b/distrib/sdl-1.2.15/src/audio/dc/SDL_dcaudio.c
new file mode 100644
index 0000000..88daa87
--- /dev/null
+++ b/distrib/sdl-1.2.15/src/audio/dc/SDL_dcaudio.c
@@ -0,0 +1,246 @@
+/*
+ SDL - Simple DirectMedia Layer
+ Copyright (C) 1997-2012 Sam Lantinga
+
+ This library is free software; you can redistribute it and/or
+ modify it under the terms of the GNU Lesser General Public
+ License as published by the Free Software Foundation; either
+ version 2.1 of the License, or (at your option) any later version.
+
+ This library is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ Lesser General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public
+ License along with this library; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+
+ Sam Lantinga
+ slouken@libsdl.org
+
+*/
+#include "SDL_config.h"
+
+/* Output dreamcast aica */
+
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "../SDL_audiomem.h"
+#include "../SDL_audio_c.h"
+#include "../SDL_audiodev_c.h"
+#include "SDL_dcaudio.h"
+
+#include "aica.h"
+#include <dc/spu.h>
+
+/* Audio driver functions */
+static int DCAUD_OpenAudio(_THIS, SDL_AudioSpec *spec);
+static void DCAUD_WaitAudio(_THIS);
+static void DCAUD_PlayAudio(_THIS);
+static Uint8 *DCAUD_GetAudioBuf(_THIS);
+static void DCAUD_CloseAudio(_THIS);
+
+/* Audio driver bootstrap functions */
+static int DCAUD_Available(void)
+{
+ return 1;
+}
+
+static void DCAUD_DeleteDevice(SDL_AudioDevice *device)
+{
+ SDL_free(device->hidden);
+ SDL_free(device);
+}
+
+static SDL_AudioDevice *DCAUD_CreateDevice(int devindex)
+{
+ SDL_AudioDevice *this;
+
+ /* Initialize all variables that we clean on shutdown */
+ this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
+ if ( this ) {
+ SDL_memset(this, 0, (sizeof *this));
+ this->hidden = (struct SDL_PrivateAudioData *)
+ SDL_malloc((sizeof *this->hidden));
+ }
+ if ( (this == NULL) || (this->hidden == NULL) ) {
+ SDL_OutOfMemory();
+ if ( this ) {
+ SDL_free(this);
+ }
+ return(0);
+ }
+ SDL_memset(this->hidden, 0, (sizeof *this->hidden));
+
+ /* Set the function pointers */
+ this->OpenAudio = DCAUD_OpenAudio;
+ this->WaitAudio = DCAUD_WaitAudio;
+ this->PlayAudio = DCAUD_PlayAudio;
+ this->GetAudioBuf = DCAUD_GetAudioBuf;
+ this->CloseAudio = DCAUD_CloseAudio;
+
+ this->free = DCAUD_DeleteDevice;
+
+ spu_init();
+
+ return this;
+}
+
+AudioBootStrap DCAUD_bootstrap = {
+ "dcaudio", "Dreamcast AICA audio",
+ DCAUD_Available, DCAUD_CreateDevice
+};
+
+/* This function waits until it is possible to write a full sound buffer */
+static void DCAUD_WaitAudio(_THIS)
+{
+ if (this->hidden->playing) {
+ /* wait */
+ while(aica_get_pos(0)/this->spec.samples == this->hidden->nextbuf) {
+ thd_pass();
+ }
+ }
+}
+
+#define SPU_RAM_BASE 0xa0800000
+
+static void spu_memload_stereo8(int leftpos,int rightpos,void *src0,size_t size)
+{
+ uint8 *src = src0;
+ uint32 *left = (uint32*)(leftpos +SPU_RAM_BASE);
+ uint32 *right = (uint32*)(rightpos+SPU_RAM_BASE);
+ size = (size+7)/8;
+ while(size--) {
+ unsigned lval,rval;
+ lval = *src++;
+ rval = *src++;
+ lval|= (*src++)<<8;
+ rval|= (*src++)<<8;
+ lval|= (*src++)<<16;
+ rval|= (*src++)<<16;
+ lval|= (*src++)<<24;
+ rval|= (*src++)<<24;
+ g2_write_32(left++,lval);
+ g2_write_32(right++,rval);
+ g2_fifo_wait();
+ }
+}
+
+static void spu_memload_stereo16(int leftpos,int rightpos,void *src0,size_t size)
+{
+ uint16 *src = src0;
+ uint32 *left = (uint32*)(leftpos +SPU_RAM_BASE);
+ uint32 *right = (uint32*)(rightpos+SPU_RAM_BASE);
+ size = (size+7)/8;
+ while(size--) {
+ unsigned lval,rval;
+ lval = *src++;
+ rval = *src++;
+ lval|= (*src++)<<16;
+ rval|= (*src++)<<16;
+ g2_write_32(left++,lval);
+ g2_write_32(right++,rval);
+ g2_fifo_wait();
+ }
+}
+
+static void DCAUD_PlayAudio(_THIS)
+{
+ SDL_AudioSpec *spec = &this->spec;
+ unsigned int offset;
+
+ if (this->hidden->playing) {
+ /* wait */
+ while(aica_get_pos(0)/spec->samples == this->hidden->nextbuf) {
+ thd_pass();
+ }
+ }
+
+ offset = this->hidden->nextbuf*spec->size;
+ this->hidden->nextbuf^=1;
+ /* Write the audio data, checking for EAGAIN on broken audio drivers */
+ if (spec->channels==1) {
+ spu_memload(this->hidden->leftpos+offset,this->hidden->mixbuf,this->hidden->mixlen);
+ } else {
+ offset/=2;
+ if ((this->spec.format&255)==8) {
+ spu_memload_stereo8(this->hidden->leftpos+offset,this->hidden->rightpos+offset,this->hidden->mixbuf,this->hidden->mixlen);
+ } else {
+ spu_memload_stereo16(this->hidden->leftpos+offset,this->hidden->rightpos+offset,this->hidden->mixbuf,this->hidden->mixlen);
+ }
+ }
+
+ if (!this->hidden->playing) {
+ int mode;
+ this->hidden->playing = 1;
+ mode = (spec->format==AUDIO_S8)?SM_8BIT:SM_16BIT;
+ if (spec->channels==1) {
+ aica_play(0,mode,this->hidden->leftpos,0,spec->samples*2,spec->freq,255,128,1);
+ } else {
+ aica_play(0,mode,this->hidden->leftpos ,0,spec->samples*2,spec->freq,255,0,1);
+ aica_play(1,mode,this->hidden->rightpos,0,spec->samples*2,spec->freq,255,255,1);
+ }
+ }
+}
+
+static Uint8 *DCAUD_GetAudioBuf(_THIS)
+{
+ return(this->hidden->mixbuf);
+}
+
+static void DCAUD_CloseAudio(_THIS)
+{
+ aica_stop(0);
+ if (this->spec.channels==2) aica_stop(1);
+ if ( this->hidden->mixbuf != NULL ) {
+ SDL_FreeAudioMem(this->hidden->mixbuf);
+ this->hidden->mixbuf = NULL;
+ }
+}
+
+static int DCAUD_OpenAudio(_THIS, SDL_AudioSpec *spec)
+{
+ Uint16 test_format = SDL_FirstAudioFormat(spec->format);
+ int valid_datatype = 0;
+ while ((!valid_datatype) && (test_format)) {
+ spec->format = test_format;
+ switch (test_format) {
+ /* only formats Dreamcast accepts... */
+ case AUDIO_S8:
+ case AUDIO_S16LSB:
+ valid_datatype = 1;
+ break;
+
+ default:
+ test_format = SDL_NextAudioFormat();
+ break;
+ }
+ }
+
+ if (!valid_datatype) { /* shouldn't happen, but just in case... */
+ SDL_SetError("Unsupported audio format");
+ return (-1);
+ }
+
+ if (spec->channels > 2)
+ spec->channels = 2; /* no more than stereo on the Dreamcast. */
+
+ /* Update the fragment size as size in bytes */
+ SDL_CalculateAudioSpec(spec);
+
+ /* Allocate mixing buffer */
+ this->hidden->mixlen = spec->size;
+ this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
+ if ( this->hidden->mixbuf == NULL ) {
+ return(-1);
+ }
+ SDL_memset(this->hidden->mixbuf, spec->silence, spec->size);
+ this->hidden->leftpos = 0x11000;
+ this->hidden->rightpos = 0x11000+spec->size;
+ this->hidden->playing = 0;
+ this->hidden->nextbuf = 0;
+
+ /* We're ready to rock and roll. :-) */
+ return(0);
+}
diff --git a/distrib/sdl-1.2.15/src/audio/dc/SDL_dcaudio.h b/distrib/sdl-1.2.15/src/audio/dc/SDL_dcaudio.h
new file mode 100644
index 0000000..fba95b3
--- /dev/null
+++ b/distrib/sdl-1.2.15/src/audio/dc/SDL_dcaudio.h
@@ -0,0 +1,41 @@
+/*
+ SDL - Simple DirectMedia Layer
+ Copyright (C) 1997-2012 Sam Lantinga
+
+ This library is free software; you can redistribute it and/or
+ modify it under the terms of the GNU Lesser General Public
+ License as published by the Free Software Foundation; either
+ version 2.1 of the License, or (at your option) any later version.
+
+ This library is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ Lesser General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public
+ License along with this library; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+
+ Sam Lantinga
+ slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_dcaudio_h
+#define _SDL_dcaudio_h
+
+#include "../SDL_sysaudio.h"
+
+/* Hidden "this" pointer for the video functions */
+#define _THIS SDL_AudioDevice *this
+
+struct SDL_PrivateAudioData {
+ /* The file descriptor for the audio device */
+ Uint8 *mixbuf;
+ Uint32 mixlen;
+ int playing;
+ int leftpos,rightpos;
+ int nextbuf;
+};
+
+#endif /* _SDL_dcaudio_h */
diff --git a/distrib/sdl-1.2.15/src/audio/dc/aica.c b/distrib/sdl-1.2.15/src/audio/dc/aica.c
new file mode 100644
index 0000000..b6a1c93
--- /dev/null
+++ b/distrib/sdl-1.2.15/src/audio/dc/aica.c
@@ -0,0 +1,271 @@
+/* This file is part of the Dreamcast function library.
+ * Please see libdream.c for further details.
+ *
+ * (c)2000 Dan Potter
+ * modify BERO
+ */
+#include "aica.h"
+
+#include <arch/irq.h>
+#include <dc/spu.h>
+
+/* #define dc_snd_base ((volatile unsigned char *)0x00800000) */ /* arm side */
+#define dc_snd_base ((volatile unsigned char *)0xa0700000) /* dc side */
+
+/* Some convienence macros */
+#define SNDREGADDR(x) (0xa0700000 + (x))
+#define CHNREGADDR(ch,x) SNDREGADDR(0x80*(ch)+(x))
+
+
+#define SNDREG32(x) (*(volatile unsigned long *)SNDREGADDR(x))
+#define SNDREG8(x) (*(volatile unsigned char *)SNDREGADDR(x))
+#define CHNREG32(ch, x) (*(volatile unsigned long *)CHNREGADDR(ch,x))
+#define CHNREG8(ch, x) (*(volatile unsigned long *)CHNREGADDR(ch,x))
+
+#define G2_LOCK(OLD) \
+ do { \
+ if (!irq_inside_int()) \
+ OLD = irq_disable(); \
+ /* suspend any G2 DMA here... */ \
+ while((*(volatile unsigned int *)0xa05f688c) & 0x20) \
+ ; \
+ } while(0)
+
+#define G2_UNLOCK(OLD) \
+ do { \
+ /* resume any G2 DMA here... */ \
+ if (!irq_inside_int()) \
+ irq_restore(OLD); \
+ } while(0)
+
+
+void aica_init() {
+ int i, j, old = 0;
+
+ /* Initialize AICA channels */
+ G2_LOCK(old);
+ SNDREG32(0x2800) = 0x0000;
+
+ for (i=0; i<64; i++) {
+ for (j=0; j<0x80; j+=4) {
+ if ((j&31)==0) g2_fifo_wait();
+ CHNREG32(i, j) = 0;
+ }
+ g2_fifo_wait();
+ CHNREG32(i,0) = 0x8000;
+ CHNREG32(i,20) = 0x1f;
+ }
+
+ SNDREG32(0x2800) = 0x000f;
+ g2_fifo_wait();
+ G2_UNLOCK(old);
+}
+
+/* Translates a volume from linear form to logarithmic form (required by
+ the AICA chip */
+/* int logs[] = {
+
+0, 40, 50, 58, 63, 68, 73, 77, 80, 83, 86, 89, 92, 94, 97, 99, 101, 103,
+105, 107, 109, 111, 112, 114, 116, 117, 119, 120, 122, 123, 125, 126, 127,
+129, 130, 131, 133, 134, 135, 136, 137, 139, 140, 141, 142, 143, 144, 145,
+146, 147, 148, 149, 150, 151, 152, 153, 154, 155, 156, 156, 157, 158, 159,
+160, 161, 162, 162, 163, 164, 165, 166, 166, 167, 168, 169, 170, 170, 171,
+172, 172, 173, 174, 175, 175, 176, 177, 177, 178, 179, 180, 180, 181, 182,
+182, 183, 183, 184, 185, 185, 186, 187, 187, 188, 188, 189, 190, 190, 191,
+191, 192, 193, 193, 194, 194, 195, 196, 196, 197, 197, 198, 198, 199, 199,
+200, 201, 201, 202, 202, 203, 203, 204, 204, 205, 205, 206, 206, 207, 207,
+208, 208, 209, 209, 210, 210, 211, 211, 212, 212, 213, 213, 214, 214, 215,
+215, 216, 216, 217, 217, 217, 218, 218, 219, 219, 220, 220, 221, 221, 222,
+222, 222, 223, 223, 224, 224, 225, 225, 225, 226, 226, 227, 227, 228, 228,
+228, 229, 229, 230, 230, 230, 231, 231, 232, 232, 232, 233, 233, 234, 234,
+234, 235, 235, 236, 236, 236, 237, 237, 238, 238, 238, 239, 239, 240, 240,
+240, 241, 241, 241, 242, 242, 243, 243, 243, 244, 244, 244, 245, 245, 245,
+246, 246, 247, 247, 247, 248, 248, 248, 249, 249, 249, 250, 250, 250, 251,
+251, 251, 252, 252, 252, 253, 253, 253, 254, 254, 254, 255
+
+}; */
+
+const static unsigned char logs[] = {
+ 0, 15, 22, 27, 31, 35, 39, 42, 45, 47, 50, 52, 55, 57, 59, 61,
+ 63, 65, 67, 69, 71, 73, 74, 76, 78, 79, 81, 82, 84, 85, 87, 88,
+ 90, 91, 92, 94, 95, 96, 98, 99, 100, 102, 103, 104, 105, 106,
+ 108, 109, 110, 111, 112, 113, 114, 116, 117, 118, 119, 120, 121,
+ 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 134,
+ 135, 136, 137, 138, 138, 139, 140, 141, 142, 143, 144, 145, 146,
+ 146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 156,
+ 157, 158, 159, 160, 160, 161, 162, 163, 164, 164, 165, 166, 167,
+ 167, 168, 169, 170, 170, 171, 172, 173, 173, 174, 175, 176, 176,
+ 177, 178, 178, 179, 180, 181, 181, 182, 183, 183, 184, 185, 185,
+ 186, 187, 187, 188, 189, 189, 190, 191, 191, 192, 193, 193, 194,
+ 195, 195, 196, 197, 197, 198, 199, 199, 200, 200, 201, 202, 202,
+ 203, 204, 204, 205, 205, 206, 207, 207, 208, 209, 209, 210, 210,
+ 211, 212, 212, 213, 213, 214, 215, 215, 216, 216, 217, 217, 218,
+ 219, 219, 220, 220, 221, 221, 222, 223, 223, 224, 224, 225, 225,
+ 226, 227, 227, 228, 228, 229, 229, 230, 230, 231, 232, 232, 233,
+ 233, 234, 234, 235, 235, 236, 236, 237, 237, 238, 239, 239, 240,
+ 240, 241, 241, 242, 242, 243, 243, 244, 244, 245, 245, 246, 246,
+ 247, 247, 248, 248, 249, 249, 250, 250, 251, 251, 252, 252, 253, 254, 255
+};
+
+/* For the moment this is going to have to suffice, until we really
+ figure out what these mean. */
+#define AICA_PAN(x) ((x)==0x80?(0):((x)<0x80?(0x1f):(0x0f)))
+#define AICA_VOL(x) (0xff - logs[128 + (((x) & 0xff) / 2)])
+//#define AICA_VOL(x) (0xff - logs[x&255])
+
+static inline unsigned AICA_FREQ(unsigned freq) {
+ unsigned long freq_lo, freq_base = 5644800;
+ int freq_hi = 7;
+
+ /* Need to convert frequency to floating point format
+ (freq_hi is exponent, freq_lo is mantissa)
+ Formula is ferq = 44100*2^freq_hi*(1+freq_lo/1024) */
+ while (freq < freq_base && freq_hi > -8) {
+ freq_base >>= 1;
+ --freq_hi;
+ }
+ while (freq < freq_base && freq_hi > -8) {
+ freq_base >>= 1;
+ freq_hi--;
+ }
+ freq_lo = (freq<<10) / freq_base;
+ return (freq_hi << 11) | (freq_lo & 1023);
+}
+
+/* Sets up a sound channel completely. This is generally good if you want
+ a quick and dirty way to play notes. If you want a more comprehensive
+ set of routines (more like PC wavetable cards) see below.
+
+ ch is the channel to play on (0 - 63)
+ smpptr is the pointer to the sound data; if you're running off the
+ SH4, then this ought to be (ptr - 0xa0800000); otherwise it's just
+ ptr. Basically, it's an offset into sound ram.
+ mode is one of the mode constants (16 bit, 8 bit, ADPCM)
+ nsamp is the number of samples to play (not number of bytes!)
+ freq is the sampling rate of the sound
+ vol is the volume, 0 to 0xff (0xff is louder)
+ pan is a panning constant -- 0 is left, 128 is center, 255 is right.
+
+ This routine (and the similar ones) owe a lot to Marcus' sound example --
+ I hadn't gotten quite this far into dissecting the individual regs yet. */
+void aica_play(int ch,int mode,unsigned long smpptr,int loopst,int loopend,int freq,int vol,int pan,int loopflag) {
+/* int i;
+*/
+ int val;
+ int old = 0;
+
+ /* Stop the channel (if it's already playing) */
+ aica_stop(ch);
+ /* doesn't seem to be needed, but it's here just in case */
+/*
+ for (i=0; i<256; i++) {
+ asm("nop");
+ asm("nop");
+ asm("nop");
+ asm("nop");
+ }
+*/
+ G2_LOCK(old);
+ /* Envelope setup. The first of these is the loop point,
+ e.g., where the sample starts over when it loops. The second
+ is the loop end. This is the full length of the sample when
+ you are not looping, or the loop end point when you are (though
+ storing more than that is a waste of memory if you're not doing
+ volume enveloping). */
+ CHNREG32(ch, 8) = loopst & 0xffff;
+ CHNREG32(ch, 12) = loopend & 0xffff;
+
+ /* Write resulting values */
+ CHNREG32(ch, 24) = AICA_FREQ(freq);
+
+ /* Set volume, pan, and some other things that we don't know what
+ they do =) */
+ CHNREG32(ch, 36) = AICA_PAN(pan) | (0xf<<8);
+ /* Convert the incoming volume and pan into hardware values */
+ /* Vol starts at zero so we can ramp */
+ vol = AICA_VOL(vol);
+ CHNREG32(ch, 40) = 0x24 | (vol<<8);
+ /* Convert the incoming volume and pan into hardware values */
+ /* Vol starts at zero so we can ramp */
+
+ /* If we supported volume envelopes (which we don't yet) then
+ this value would set that up. The top 4 bits determine the
+ envelope speed. f is the fastest, 1 is the slowest, and 0
+ seems to be an invalid value and does weird things). The
+ default (below) sets it into normal mode (play and terminate/loop).
+ CHNREG32(ch, 16) = 0xf010;
+ */
+ CHNREG32(ch, 16) = 0x1f; /* No volume envelope */
+
+
+ /* Set sample format, buffer address, and looping control. If
+ 0x0200 mask is set on reg 0, the sample loops infinitely. If
+ it's not set, the sample plays once and terminates. We'll
+ also set the bits to start playback here. */
+ CHNREG32(ch, 4) = smpptr & 0xffff;
+ val = 0xc000 | 0x0000 | (mode<<7) | (smpptr >> 16);
+ if (loopflag) val|=0x200;
+
+ CHNREG32(ch, 0) = val;
+
+ G2_UNLOCK(old);
+
+ /* Enable playback */
+ /* CHNREG32(ch, 0) |= 0xc000; */
+ g2_fifo_wait();
+
+#if 0
+ for (i=0xff; i>=vol; i--) {
+ if ((i&7)==0) g2_fifo_wait();
+ CHNREG32(ch, 40) = 0x24 | (i<<8);;
+ }
+
+ g2_fifo_wait();
+#endif
+}
+
+/* Stop the sound on a given channel */
+void aica_stop(int ch) {
+ g2_write_32(CHNREGADDR(ch, 0),(g2_read_32(CHNREGADDR(ch, 0)) & ~0x4000) | 0x8000);
+ g2_fifo_wait();
+}
+
+
+/* The rest of these routines can change the channel in mid-stride so you
+ can do things like vibrato and panning effects. */
+
+/* Set channel volume */
+void aica_vol(int ch,int vol) {
+// g2_write_8(CHNREGADDR(ch, 41),AICA_VOL(vol));
+ g2_write_32(CHNREGADDR(ch, 40),(g2_read_32(CHNREGADDR(ch, 40))&0xffff00ff)|(AICA_VOL(vol)<<8) );
+ g2_fifo_wait();
+}
+
+/* Set channel pan */
+void aica_pan(int ch,int pan) {
+// g2_write_8(CHNREGADDR(ch, 36),AICA_PAN(pan));
+ g2_write_32(CHNREGADDR(ch, 36),(g2_read_32(CHNREGADDR(ch, 36))&0xffffff00)|(AICA_PAN(pan)) );
+ g2_fifo_wait();
+}
+
+/* Set channel frequency */
+void aica_freq(int ch,int freq) {
+ g2_write_32(CHNREGADDR(ch, 24),AICA_FREQ(freq));
+ g2_fifo_wait();
+}
+
+/* Get channel position */
+int aica_get_pos(int ch) {
+#if 1
+ /* Observe channel ch */
+ g2_write_32(SNDREGADDR(0x280c),(g2_read_32(SNDREGADDR(0x280c))&0xffff00ff) | (ch<<8));
+ g2_fifo_wait();
+ /* Update position counters */
+ return g2_read_32(SNDREGADDR(0x2814)) & 0xffff;
+#else
+ /* Observe channel ch */
+ g2_write_8(SNDREGADDR(0x280d),ch);
+ /* Update position counters */
+ return g2_read_32(SNDREGADDR(0x2814)) & 0xffff;
+#endif
+}
diff --git a/distrib/sdl-1.2.15/src/audio/dc/aica.h b/distrib/sdl-1.2.15/src/audio/dc/aica.h
new file mode 100644
index 0000000..2721e42
--- /dev/null
+++ b/distrib/sdl-1.2.15/src/audio/dc/aica.h
@@ -0,0 +1,40 @@
+/*
+ SDL - Simple DirectMedia Layer
+ Copyright (C) 1997-2012 Sam Lantinga
+
+ This library is free software; you can redistribute it and/or
+ modify it under the terms of the GNU Lesser General Public
+ License as published by the Free Software Foundation; either
+ version 2.1 of the License, or (at your option) any later version.
+
+ This library is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ Lesser General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public
+ License along with this library; if not, write to the Free Software
+ Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+
+ Sam Lantinga
+ slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _AICA_H_
+#define _AICA_H_
+
+#define AICA_MEM 0xa0800000
+
+#define SM_8BIT 1
+#define SM_16BIT 0
+#define SM_ADPCM 2
+
+void aica_play(int ch,int mode,unsigned long smpptr,int looptst,int loopend,int freq,int vol,int pan,int loopflag);
+void aica_stop(int ch);
+void aica_vol(int ch,int vol);
+void aica_pan(int ch,int pan);
+void aica_freq(int ch,int freq);
+int aica_get_pos(int ch);
+
+#endif
generated by cgit v1.1 at 2024-05-28 00:55:11 +0000