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Regard whitespace Rev 9560 → Rev 9561

/contrib/sdk/sources/SDL-1.2.2_newlib/include/SDL.h
42,6 → 42,7
#include "SDL_events.h"
#include "SDL_video.h"
#include "SDL_byteorder.h"
#include "SDL_stdinc.h"
#include "SDL_version.h"
 
#include "begin_code.h"
/contrib/sdk/sources/SDL-1.2.2_newlib/include/SDL_rwops.h
96,6 → 96,10
extern DECLSPEC SDL_RWops * SDL_AllocRW(void);
extern DECLSPEC void SDL_FreeRW(SDL_RWops *area);
 
#define RW_SEEK_SET SEEK_SET /**< Seek from the beginning of data */
#define RW_SEEK_CUR SEEK_CUR /**< Seek relative to current read point */
#define RW_SEEK_END SEEK_END /**< Seek relative to the end of data */
 
/* Macros to easily read and write from an SDL_RWops structure */
#define SDL_RWseek(ctx, offset, whence) (ctx)->seek(ctx, offset, whence)
#define SDL_RWtell(ctx) (ctx)->seek(ctx, 0, SEEK_CUR)
/contrib/sdk/sources/SDL-1.2.2_newlib/include/SDL_stdinc.h
0,0 → 1,112
/*
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
*/
 
/** @file SDL_stdinc.h
* This is a general header that includes C language support
*/
 
#ifndef _SDL_stdinc_h
#define _SDL_stdinc_h
 
 
#include <sys/types.h>
#include <stdio.h>
#include <stdlib.h>
#include <stddef.h>
#include <stdarg.h>
#include <strings.h>
#include <inttypes.h>
#include <stdint.h>
#include <ctype.h>
 
/** The number of elements in an array */
#define SDL_arraysize(array) (sizeof(array)/sizeof(array[0]))
 
/* Use proper C++ casts when compiled as C++ to be compatible with the option
-Wold-style-cast of GCC (and -Werror=old-style-cast in GCC 4.2 and above. */
#ifdef __cplusplus
#define SDL_reinterpret_cast(type, expression) reinterpret_cast<type>(expression)
#define SDL_static_cast(type, expression) static_cast<type>(expression)
#else
#define SDL_reinterpret_cast(type, expression) ((type)(expression))
#define SDL_static_cast(type, expression) ((type)(expression))
#endif
 
typedef enum {
DUMMY_ENUM_VALUE
} SDL_DUMMY_ENUM;
 
#include "begin_code.h"
/* Set up for C function definitions, even when using C++ */
#ifdef __cplusplus
extern "C" {
#endif
 
#define SDL_malloc malloc
#define SDL_calloc calloc
#define SDL_realloc realloc
#define SDL_free free
#define SDL_stack_alloc(type, count) (type*)SDL_malloc(sizeof(type)*(count))
#define SDL_stack_free(data) SDL_free(data)
#define SDL_qsort qsort
#define SDL_abs abs
#define SDL_min(x, y) (((x) < (y)) ? (x) : (y))
#define SDL_max(x, y) (((x) > (y)) ? (x) : (y))
#define SDL_isdigit(X) isdigit(X)
#define SDL_isspace(X) isspace(X)
#define SDL_toupper(X) toupper(X)
#define SDL_tolower(X) tolower(X)
#define SDL_memset memset
#define SDL_memmove memmove
#define SDL_memcmp memcmp
#define SDL_strlen strlen
#define SDL_strlcpy strlcpy
#define SDL_strlcat strlcat
#define SDL_strdup strdup
#define SDL_strrev _strrev
#define SDL_strupr _strupr
#define SDL_strlwr _strlwr
#define SDL_strchr strchr
#define SDL_strrchr strrchr
#define SDL_strstr strstr
#define SDL_itoa itoa
#define SDL_ltoa _ltoa
#define SDL_uitoa _uitoa
#define SDL_ultoa _ultoa
#define SDL_strtol strtol
#define SDL_strtoul strtoul
#define SDL_strtod strtod
#define SDL_atoi atoi
#define SDL_atof atof
#define SDL_strcmp strcmp
#define SDL_strncmp strncmp
#define SDL_sscanf sscanf
#define SDL_snprintf snprintf
#define SDL_vsnprintf vsnprintf
 
/* Ends C function definitions when using C++ */
#ifdef __cplusplus
}
#endif
#include "close_code.h"
 
#endif /* _SDL_stdinc_h */
/contrib/sdk/sources/SDL-1.2.2_newlib/include/SDL_video.h
168,6 → 168,8
Uint32 UnusedBits3 :16;
Uint32 video_mem; /* The total amount of video memory (in K) */
SDL_PixelFormat *vfmt; /* Value: The format of the video surface */
Uint32 current_h; /* Current screen width */
Uint32 current_w; /* Current screen height */
} SDL_VideoInfo;
 
 
/contrib/sdk/sources/SDL-1.2.2_newlib/include/begin_code.h
48,6 → 48,29
# endif
#endif
 
/**
* @def SDLCALL
* By default SDL uses the C calling convention
*/
#ifndef SDLCALL
# if defined(__WIN32__) && !defined(__GNUC__)
# define SDLCALL __cdecl
# elif defined(__OS2__)
# if defined (__GNUC__) && __GNUC__ < 4
# /* Added support for GCC-EMX <v4.x */
# /* this is needed for XFree86/OS2 developement */
# /* F. Ambacher(anakor@snafu.de) 05.2008 */
# define SDLCALL _cdecl
# else
# /* On other compilers on OS/2, we use the _System calling convention */
# /* to be compatible with every compiler */
# define SDLCALL _System
# endif
# else
# define SDLCALL
# endif
#endif /* SDLCALL */
 
/* Force structure packing at 4 byte alignment.
This is necessary if the header is included in code which has structure
packing set to an alternate value, say for loading structures from disk.
/contrib/sdk/sources/SDL-1.2.2_newlib/src/libSDL.def
File deleted
/contrib/sdk/sources/SDL-1.2.2_newlib/src/Makefile
8,7 → 8,7
SDK_DIR:= $(abspath ../../..)
 
LDFLAGS+= -shared -s -T dll.lds --entry _DllStartup --image-base=0 --out-implib $(LIBNAME).dll.a
LDFLAGS+= -L/home/max/autobuild/tools/win32/mingw32/lib
LDFLAGS+= -L../../../lib
 
endian_OBJS = endian/SDL_endian.o
file_OBJS = file/SDL_rwops.o
29,7 → 29,7
video/SDL_video.o video/SDL_yuv.o video/SDL_yuv_mmx.o \
video/SDL_yuv_sw.o video/menuetos/SDL_menuetevents.o \
video/menuetos/SDL_menuetvideo.o
audio_OBJS = audio/SDL_kolibri_audio.o
audio_OBJS = audio/SDL_kolibri_audio.o audio/SDL_audiocvt.o audio/SDL_mixer.o audio/SDL_wave.o
 
curr_OBJS = SDL.o SDL_error.o SDL_fatal.o SDL_getenv.o
36,8 → 36,11
OBJS = $(endian_OBJS) $(file_OBJS) $(hermes_OBJS) $(thread_OBJS) \
$(timer_OBJS) $(event_OBJS) $(video_OBJS) $(curr_OBJS) $(audio_OBJS)
CFLAGS = -c -O2 -D_REENTRANT -I../include -I SYSCALL/include -I. -DPACKAGE=\"SDL\" -DVERSION=\"1.2.2\" \
-fexpensive-optimizations -Wall -DENABLE_AUDIO -UDISABLE_AUDIO -DDISABLE_JOYSTICK \
CFLAGS = -c -O2 -mpreferred-stack-boundary=2 -fno-ident -fomit-frame-pointer -fno-stack-check \
-fno-stack-protector -mno-stack-arg-probe -fno-exceptions -fno-asynchronous-unwind-tables \
-ffast-math -mno-ms-bitfields -march=pentium-mmx -fexpensive-optimizations \
-D_REENTRANT -I../include -I SYSCALL/include -I. -DPACKAGE=\"SDL\" -DVERSION=\"1.2.2\" \
-Wall -DENABLE_AUDIO -UDISABLE_AUDIO -DDISABLE_JOYSTICK \
-DDISABLE_CDROM -DDISABLE_THREADS -DENABLE_TIMERS \
-DUSE_ASMBLIT -Ihermes -Iaudio -Ivideo -Ievents \
-Ijoystick -Icdrom -Ithread -Itimer -Iendian -Ifile -DENABLE_KOLIBRIOS \
45,7 → 48,7
-D__KOLIBRIOS__ -DDEBUG_VIDEO -UWIN32 -U_Win32 -U_WIN32 -U__MINGW32__ \
-I../../newlib/libc/include/
 
all: $(LIBNAME).dll $(LIBNAME).a
all: $(LIBNAME).a $(LIBNAME).dll
 
install: $(LIBNAME)
mv -f $(LIBNAME) $(SDK_DIR)/lib
52,15 → 55,15
$(LIBNAME).a: $(OBJS)
$(MAKE) -C SYSCALL/src
$(AR) -crs $(LIBNAME).a $(OBJS) SYSCALL/src/os.o
$(AR) -crs ../../../lib/$(LIBNAME).a $(OBJS) SYSCALL/src/os.o
 
$(LIBNAME).dll: libSDL.def $(OBJS)
$(LD) $(LDFLAGS) -o $@ libSDL.def $(OBJS) SYSCALL/src/os.o $(LIBS) -ldll -lsound -lc.dll
$(STRIP) $@
$(LIBNAME).dll: $(OBJS)
$(LD) $(LDFLAGS) -o $@ $(OBJS) SYSCALL/src/os.o $(LIBS) -ldll -lsound -lc.dll
$(STRIP) -S $@
 
%.o : %.asm Makefile
nasm -f coff $<
nasm -Ihermes -f coff $<
%.o : %.c Makefile
$(CC) $(CFLAGS) -o $@ $<
/contrib/sdk/sources/SDL-1.2.2_newlib/src/SDL_error_c.h
19,7 → 19,7
Sam Lantinga
slouken@devolution.com
*/
 
////////////////////
#ifdef SAVE_RCSID
static char rcsid =
"@(#) $Id: SDL_error_c.h,v 1.2 2001/04/26 16:50:17 hercules Exp $";
/contrib/sdk/sources/SDL-1.2.2_newlib/src/audio/SDL_audiocvt.c
0,0 → 1,642
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997, 1998, 1999, 2000, 2001 Sam Lantinga
 
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Library General Public
License as published by the Free Software Foundation; either
version 2 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
Library General Public License for more details.
 
You should have received a copy of the GNU Library General Public
License along with this library; if not, write to the Free
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
 
Sam Lantinga
slouken@devolution.com
*/
 
#ifdef SAVE_RCSID
static char rcsid =
"@(#) $Id: SDL_audiocvt.c,v 1.2 2001/04/26 16:50:17 hercules Exp $";
#endif
 
/* Functions for audio drivers to perform runtime conversion of audio format */
 
#include <stdio.h>
 
#include "SDL_error.h"
#include "SDL_audio.h"
 
 
/* Effectively mix right and left channels into a single channel */
void SDL_ConvertMono(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Sint32 sample;
 
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting to mono\n");
#endif
switch (format&0x8018) {
 
case AUDIO_U8: {
Uint8 *src, *dst;
 
src = cvt->buf;
dst = cvt->buf;
for ( i=cvt->len_cvt/2; i; --i ) {
sample = src[0] + src[1];
if ( sample > 255 ) {
*dst = 255;
} else {
*dst = sample;
}
src += 2;
dst += 1;
}
}
break;
 
case AUDIO_S8: {
Sint8 *src, *dst;
 
src = (Sint8 *)cvt->buf;
dst = (Sint8 *)cvt->buf;
for ( i=cvt->len_cvt/2; i; --i ) {
sample = src[0] + src[1];
if ( sample > 127 ) {
*dst = 127;
} else
if ( sample < -128 ) {
*dst = -128;
} else {
*dst = sample;
}
src += 2;
dst += 1;
}
}
break;
 
case AUDIO_U16: {
Uint8 *src, *dst;
 
src = cvt->buf;
dst = cvt->buf;
if ( (format & 0x1000) == 0x1000 ) {
for ( i=cvt->len_cvt/4; i; --i ) {
sample = (Uint16)((src[0]<<8)|src[1])+
(Uint16)((src[2]<<8)|src[3]);
if ( sample > 65535 ) {
dst[0] = 0xFF;
dst[1] = 0xFF;
} else {
dst[1] = (sample&0xFF);
sample >>= 8;
dst[0] = (sample&0xFF);
}
src += 4;
dst += 2;
}
} else {
for ( i=cvt->len_cvt/4; i; --i ) {
sample = (Uint16)((src[1]<<8)|src[0])+
(Uint16)((src[3]<<8)|src[2]);
if ( sample > 65535 ) {
dst[0] = 0xFF;
dst[1] = 0xFF;
} else {
dst[0] = (sample&0xFF);
sample >>= 8;
dst[1] = (sample&0xFF);
}
src += 4;
dst += 2;
}
}
}
break;
 
case AUDIO_S16: {
Uint8 *src, *dst;
 
src = cvt->buf;
dst = cvt->buf;
if ( (format & 0x1000) == 0x1000 ) {
for ( i=cvt->len_cvt/4; i; --i ) {
sample = (Sint16)((src[0]<<8)|src[1])+
(Sint16)((src[2]<<8)|src[3]);
if ( sample > 32767 ) {
dst[0] = 0x7F;
dst[1] = 0xFF;
} else
if ( sample < -32768 ) {
dst[0] = 0x80;
dst[1] = 0x00;
} else {
dst[1] = (sample&0xFF);
sample >>= 8;
dst[0] = (sample&0xFF);
}
src += 4;
dst += 2;
}
} else {
for ( i=cvt->len_cvt/4; i; --i ) {
sample = (Sint16)((src[1]<<8)|src[0])+
(Sint16)((src[3]<<8)|src[2]);
if ( sample > 32767 ) {
dst[1] = 0x7F;
dst[0] = 0xFF;
} else
if ( sample < -32768 ) {
dst[1] = 0x80;
dst[0] = 0x00;
} else {
dst[0] = (sample&0xFF);
sample >>= 8;
dst[1] = (sample&0xFF);
}
src += 4;
dst += 2;
}
}
}
break;
}
cvt->len_cvt /= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
 
 
/* Duplicate a mono channel to both stereo channels */
void SDL_ConvertStereo(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
 
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting to stereo\n");
#endif
if ( (format & 0xFF) == 16 ) {
Uint16 *src, *dst;
 
src = (Uint16 *)(cvt->buf+cvt->len_cvt);
dst = (Uint16 *)(cvt->buf+cvt->len_cvt*2);
for ( i=cvt->len_cvt/2; i; --i ) {
dst -= 2;
src -= 1;
dst[0] = src[0];
dst[1] = src[0];
}
} else {
Uint8 *src, *dst;
 
src = cvt->buf+cvt->len_cvt;
dst = cvt->buf+cvt->len_cvt*2;
for ( i=cvt->len_cvt; i; --i ) {
dst -= 2;
src -= 1;
dst[0] = src[0];
dst[1] = src[0];
}
}
cvt->len_cvt *= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
 
/* Convert 8-bit to 16-bit - LSB */
void SDL_Convert16LSB(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
 
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting to 16-bit LSB\n");
#endif
src = cvt->buf+cvt->len_cvt;
dst = cvt->buf+cvt->len_cvt*2;
for ( i=cvt->len_cvt; i; --i ) {
src -= 1;
dst -= 2;
dst[1] = *src;
dst[0] = 0;
}
format = ((format & ~0x0008) | AUDIO_U16LSB);
cvt->len_cvt *= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
/* Convert 8-bit to 16-bit - MSB */
void SDL_Convert16MSB(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
 
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting to 16-bit MSB\n");
#endif
src = cvt->buf+cvt->len_cvt;
dst = cvt->buf+cvt->len_cvt*2;
for ( i=cvt->len_cvt; i; --i ) {
src -= 1;
dst -= 2;
dst[0] = *src;
dst[1] = 0;
}
format = ((format & ~0x0008) | AUDIO_U16MSB);
cvt->len_cvt *= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
 
/* Convert 16-bit to 8-bit */
void SDL_Convert8(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
 
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting to 8-bit\n");
#endif
src = cvt->buf;
dst = cvt->buf;
if ( (format & 0x1000) != 0x1000 ) { /* Little endian */
++src;
}
for ( i=cvt->len_cvt/2; i; --i ) {
*dst = *src;
src += 2;
dst += 1;
}
format = ((format & ~0x9010) | AUDIO_U8);
cvt->len_cvt /= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
 
/* Toggle signed/unsigned */
void SDL_ConvertSign(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *data;
 
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio signedness\n");
#endif
data = cvt->buf;
if ( (format & 0xFF) == 16 ) {
if ( (format & 0x1000) != 0x1000 ) { /* Little endian */
++data;
}
for ( i=cvt->len_cvt/2; i; --i ) {
*data ^= 0x80;
data += 2;
}
} else {
for ( i=cvt->len_cvt; i; --i ) {
*data++ ^= 0x80;
}
}
format = (format ^ 0x8000);
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
 
/* Toggle endianness */
void SDL_ConvertEndian(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *data, tmp;
 
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio endianness\n");
#endif
data = cvt->buf;
for ( i=cvt->len_cvt/2; i; --i ) {
tmp = data[0];
data[0] = data[1];
data[1] = tmp;
data += 2;
}
format = (format ^ 0x1000);
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
 
/* Convert rate up by multiple of 2 */
void SDL_RateMUL2(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
 
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio rate * 2\n");
#endif
src = cvt->buf+cvt->len_cvt;
dst = cvt->buf+cvt->len_cvt*2;
switch (format & 0xFF) {
case 8:
for ( i=cvt->len_cvt; i; --i ) {
src -= 1;
dst -= 2;
dst[0] = src[0];
dst[1] = src[0];
}
break;
case 16:
for ( i=cvt->len_cvt/2; i; --i ) {
src -= 2;
dst -= 4;
dst[0] = src[0];
dst[1] = src[1];
dst[2] = src[0];
dst[3] = src[1];
}
break;
}
cvt->len_cvt *= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
 
/* Convert rate down by multiple of 2 */
void SDL_RateDIV2(SDL_AudioCVT *cvt, Uint16 format)
{
int i;
Uint8 *src, *dst;
 
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio rate / 2\n");
#endif
src = cvt->buf;
dst = cvt->buf;
switch (format & 0xFF) {
case 8:
for ( i=cvt->len_cvt/2; i; --i ) {
dst[0] = src[0];
src += 2;
dst += 1;
}
break;
case 16:
for ( i=cvt->len_cvt/4; i; --i ) {
dst[0] = src[0];
dst[1] = src[1];
src += 4;
dst += 2;
}
break;
}
cvt->len_cvt /= 2;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
 
/* Very slow rate conversion routine */
void SDL_RateSLOW(SDL_AudioCVT *cvt, Uint16 format)
{
double ipos;
int i, clen;
 
#ifdef DEBUG_CONVERT
fprintf(stderr, "Converting audio rate * %4.4f\n", 1.0/cvt->rate_incr);
#endif
clen = (int)((double)cvt->len_cvt / cvt->rate_incr);
if ( cvt->rate_incr > 1.0 ) {
switch (format & 0xFF) {
case 8: {
Uint8 *output;
 
output = cvt->buf;
ipos = 0.0;
for ( i=clen; i; --i ) {
*output = cvt->buf[(int)ipos];
ipos += cvt->rate_incr;
output += 1;
}
}
break;
 
case 16: {
Uint16 *output;
 
clen &= ~1;
output = (Uint16 *)cvt->buf;
ipos = 0.0;
for ( i=clen/2; i; --i ) {
*output=((Uint16 *)cvt->buf)[(int)ipos];
ipos += cvt->rate_incr;
output += 1;
}
}
break;
}
} else {
switch (format & 0xFF) {
case 8: {
Uint8 *output;
 
output = cvt->buf+clen;
ipos = (double)cvt->len_cvt;
for ( i=clen; i; --i ) {
ipos -= cvt->rate_incr;
output -= 1;
*output = cvt->buf[(int)ipos];
}
}
break;
 
case 16: {
Uint16 *output;
 
clen &= ~1;
output = (Uint16 *)(cvt->buf+clen);
ipos = (double)cvt->len_cvt/2;
for ( i=clen/2; i; --i ) {
ipos -= cvt->rate_incr;
output -= 1;
*output=((Uint16 *)cvt->buf)[(int)ipos];
}
}
break;
}
}
cvt->len_cvt = clen;
if ( cvt->filters[++cvt->filter_index] ) {
cvt->filters[cvt->filter_index](cvt, format);
}
}
 
int SDL_ConvertAudio(SDL_AudioCVT *cvt)
{
/* Make sure there's data to convert */
if ( cvt->buf == NULL ) {
SDL_SetError("No buffer allocated for conversion");
return(-1);
}
/* Return okay if no conversion is necessary */
cvt->len_cvt = cvt->len;
if ( cvt->filters[0] == NULL ) {
return(0);
}
 
/* Set up the conversion and go! */
cvt->filter_index = 0;
cvt->filters[0](cvt, cvt->src_format);
return(0);
}
 
/* Creates a set of audio filters to convert from one format to another.
Returns -1 if the format conversion is not supported, or 1 if the
audio filter is set up.
*/
int SDL_BuildAudioCVT(SDL_AudioCVT *cvt,
Uint16 src_format, Uint8 src_channels, int src_rate,
Uint16 dst_format, Uint8 dst_channels, int dst_rate)
{
/* Start off with no conversion necessary */
cvt->needed = 0;
cvt->filter_index = 0;
cvt->filters[0] = NULL;
cvt->len_mult = 1;
cvt->len_ratio = 1.0;
 
/* First filter: Endian conversion from src to dst */
if ( (src_format & 0x1000) != (dst_format & 0x1000)
&& ((src_format & 0xff) != 8) ) {
cvt->filters[cvt->filter_index++] = SDL_ConvertEndian;
}
/* Second filter: Sign conversion -- signed/unsigned */
if ( (src_format & 0x8000) != (dst_format & 0x8000) ) {
cvt->filters[cvt->filter_index++] = SDL_ConvertSign;
}
 
/* Next filter: Convert 16 bit <--> 8 bit PCM */
if ( (src_format & 0xFF) != (dst_format & 0xFF) ) {
switch (dst_format&0x10FF) {
case AUDIO_U8:
cvt->filters[cvt->filter_index++] =
SDL_Convert8;
cvt->len_ratio /= 2;
break;
case AUDIO_U16LSB:
cvt->filters[cvt->filter_index++] =
SDL_Convert16LSB;
cvt->len_mult *= 2;
cvt->len_ratio *= 2;
break;
case AUDIO_U16MSB:
cvt->filters[cvt->filter_index++] =
SDL_Convert16MSB;
cvt->len_mult *= 2;
cvt->len_ratio *= 2;
break;
}
}
 
/* Last filter: Mono/Stereo conversion */
if ( src_channels != dst_channels ) {
while ( (src_channels*2) <= dst_channels ) {
cvt->filters[cvt->filter_index++] =
SDL_ConvertStereo;
cvt->len_mult *= 2;
src_channels *= 2;
cvt->len_ratio *= 2;
}
/* This assumes that 4 channel audio is in the format:
Left {front/back} + Right {front/back}
so converting to L/R stereo works properly.
*/
while ( ((src_channels%2) == 0) &&
((src_channels/2) >= dst_channels) ) {
cvt->filters[cvt->filter_index++] =
SDL_ConvertMono;
src_channels /= 2;
cvt->len_ratio /= 2;
}
if ( src_channels != dst_channels ) {
/* Uh oh.. */;
}
}
 
/* Do rate conversion */
cvt->rate_incr = 0.0;
if ( (src_rate/100) != (dst_rate/100) ) {
Uint32 hi_rate, lo_rate;
int len_mult;
double len_ratio;
void (*rate_cvt)(SDL_AudioCVT *cvt, Uint16 format);
 
if ( src_rate > dst_rate ) {
hi_rate = src_rate;
lo_rate = dst_rate;
rate_cvt = SDL_RateDIV2;
len_mult = 1;
len_ratio = 0.5;
} else {
hi_rate = dst_rate;
lo_rate = src_rate;
rate_cvt = SDL_RateMUL2;
len_mult = 2;
len_ratio = 2.0;
}
/* If hi_rate = lo_rate*2^x then conversion is easy */
while ( ((lo_rate*2)/100) <= (hi_rate/100) ) {
cvt->filters[cvt->filter_index++] = rate_cvt;
cvt->len_mult *= len_mult;
lo_rate *= 2;
cvt->len_ratio *= len_ratio;
}
/* We may need a slow conversion here to finish up */
if ( (lo_rate/100) != (hi_rate/100) ) {
#if 1
/* The problem with this is that if the input buffer is
say 1K, and the conversion rate is say 1.1, then the
output buffer is 1.1K, which may not be an acceptable
buffer size for the audio driver (not a power of 2)
*/
/* For now, punt and hope the rate distortion isn't great.
*/
#else
if ( src_rate < dst_rate ) {
cvt->rate_incr = (double)lo_rate/hi_rate;
cvt->len_mult *= 2;
cvt->len_ratio /= cvt->rate_incr;
} else {
cvt->rate_incr = (double)hi_rate/lo_rate;
cvt->len_ratio *= cvt->rate_incr;
}
cvt->filters[cvt->filter_index++] = SDL_RateSLOW;
#endif
}
}
 
/* Set up the filter information */
if ( cvt->filter_index != 0 ) {
cvt->needed = 1;
cvt->src_format = src_format;
cvt->dst_format = dst_format;
cvt->len = 0;
cvt->buf = NULL;
cvt->filters[cvt->filter_index] = NULL;
}
return(cvt->needed);
}
/contrib/sdk/sources/SDL-1.2.2_newlib/src/audio/SDL_mixer.c
0,0 → 1,218
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997, 1998, 1999, 2000, 2001 Sam Lantinga
 
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Library General Public
License as published by the Free Software Foundation; either
version 2 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
Library General Public License for more details.
 
You should have received a copy of the GNU Library General Public
License along with this library; if not, write to the Free
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
 
Sam Lantinga
slouken@devolution.com
*/
 
#ifdef SAVE_RCSID
static char rcsid =
"@(#) $Id: SDL_mixer.c,v 1.2 2001/04/26 16:50:17 hercules Exp $";
#endif
 
/* This provides the default mixing callback for the SDL audio routines */
 
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
 
#include "SDL_audio.h"
#include "SDL_mutex.h"
#include "SDL_timer.h"
#include "SDL_sysaudio.h"
 
SDL_AudioDevice *current_audio = NULL;
 
/* This table is used to add two sound values together and pin
* the value to avoid overflow. (used with permission from ARDI)
* Changed to use 0xFE instead of 0xFF for better sound quality.
*/
static const Uint8 mix8[] =
{
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x02, 0x03,
0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E,
0x0F, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0x17, 0x18, 0x19,
0x1A, 0x1B, 0x1C, 0x1D, 0x1E, 0x1F, 0x20, 0x21, 0x22, 0x23, 0x24,
0x25, 0x26, 0x27, 0x28, 0x29, 0x2A, 0x2B, 0x2C, 0x2D, 0x2E, 0x2F,
0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37, 0x38, 0x39, 0x3A,
0x3B, 0x3C, 0x3D, 0x3E, 0x3F, 0x40, 0x41, 0x42, 0x43, 0x44, 0x45,
0x46, 0x47, 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, 0x50,
0x51, 0x52, 0x53, 0x54, 0x55, 0x56, 0x57, 0x58, 0x59, 0x5A, 0x5B,
0x5C, 0x5D, 0x5E, 0x5F, 0x60, 0x61, 0x62, 0x63, 0x64, 0x65, 0x66,
0x67, 0x68, 0x69, 0x6A, 0x6B, 0x6C, 0x6D, 0x6E, 0x6F, 0x70, 0x71,
0x72, 0x73, 0x74, 0x75, 0x76, 0x77, 0x78, 0x79, 0x7A, 0x7B, 0x7C,
0x7D, 0x7E, 0x7F, 0x80, 0x81, 0x82, 0x83, 0x84, 0x85, 0x86, 0x87,
0x88, 0x89, 0x8A, 0x8B, 0x8C, 0x8D, 0x8E, 0x8F, 0x90, 0x91, 0x92,
0x93, 0x94, 0x95, 0x96, 0x97, 0x98, 0x99, 0x9A, 0x9B, 0x9C, 0x9D,
0x9E, 0x9F, 0xA0, 0xA1, 0xA2, 0xA3, 0xA4, 0xA5, 0xA6, 0xA7, 0xA8,
0xA9, 0xAA, 0xAB, 0xAC, 0xAD, 0xAE, 0xAF, 0xB0, 0xB1, 0xB2, 0xB3,
0xB4, 0xB5, 0xB6, 0xB7, 0xB8, 0xB9, 0xBA, 0xBB, 0xBC, 0xBD, 0xBE,
0xBF, 0xC0, 0xC1, 0xC2, 0xC3, 0xC4, 0xC5, 0xC6, 0xC7, 0xC8, 0xC9,
0xCA, 0xCB, 0xCC, 0xCD, 0xCE, 0xCF, 0xD0, 0xD1, 0xD2, 0xD3, 0xD4,
0xD5, 0xD6, 0xD7, 0xD8, 0xD9, 0xDA, 0xDB, 0xDC, 0xDD, 0xDE, 0xDF,
0xE0, 0xE1, 0xE2, 0xE3, 0xE4, 0xE5, 0xE6, 0xE7, 0xE8, 0xE9, 0xEA,
0xEB, 0xEC, 0xED, 0xEE, 0xEF, 0xF0, 0xF1, 0xF2, 0xF3, 0xF4, 0xF5,
0xF6, 0xF7, 0xF8, 0xF9, 0xFA, 0xFB, 0xFC, 0xFD, 0xFE, 0xFE, 0xFE,
0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE
};
 
/* The volume ranges from 0 - 128 */
#define ADJUST_VOLUME(s, v) (s = (s*v)/SDL_MIX_MAXVOLUME)
#define ADJUST_VOLUME_U8(s, v) (s = (((s-128)*v)/SDL_MIX_MAXVOLUME)+128)
 
void SDL_MixAudio (Uint8 *dst, const Uint8 *src, Uint32 len, int volume)
{
Uint16 format;
 
if ( volume == 0 ) {
return;
}
/* Mix the user-level audio format */
if ( current_audio ) {
if ( current_audio->convert.needed ) {
format = current_audio->convert.src_format;
} else {
format = current_audio->spec.format;
}
} else {
format = AUDIO_S16;
}
format = AUDIO_S16;
switch (format) {
 
case AUDIO_U8: {
Uint8 src_sample;
 
while ( len-- ) {
src_sample = *src;
ADJUST_VOLUME_U8(src_sample, volume);
*dst = mix8[*dst+src_sample];
++dst;
++src;
}
}
break;
 
case AUDIO_S8: {
Sint8 *dst8, *src8;
Sint8 src_sample;
int dst_sample;
const int max_audioval = ((1<<(8-1))-1);
const int min_audioval = -(1<<(8-1));
 
src8 = (Sint8 *)src;
dst8 = (Sint8 *)dst;
while ( len-- ) {
src_sample = *src8;
ADJUST_VOLUME(src_sample, volume);
dst_sample = *dst8 + src_sample;
if ( dst_sample > max_audioval ) {
*dst8 = max_audioval;
} else
if ( dst_sample < min_audioval ) {
*dst8 = min_audioval;
} else {
*dst8 = dst_sample;
}
++dst8;
++src8;
}
}
break;
 
case AUDIO_S16LSB: {
Sint16 src1, src2;
int dst_sample;
const int max_audioval = ((1<<(16-1))-1);
const int min_audioval = -(1<<(16-1));
 
len /= 2;
while ( len-- ) {
src1 = ((src[1])<<8|src[0]);
ADJUST_VOLUME(src1, volume);
src2 = ((dst[1])<<8|dst[0]);
src += 2;
dst_sample = src1+src2;
if ( dst_sample > max_audioval ) {
dst_sample = max_audioval;
} else
if ( dst_sample < min_audioval ) {
dst_sample = min_audioval;
}
dst[0] = dst_sample&0xFF;
dst_sample >>= 8;
dst[1] = dst_sample&0xFF;
dst += 2;
}
}
break;
 
case AUDIO_S16MSB: {
Sint16 src1, src2;
int dst_sample;
const int max_audioval = ((1<<(16-1))-1);
const int min_audioval = -(1<<(16-1));
 
len /= 2;
while ( len-- ) {
src1 = ((src[0])<<8|src[1]);
ADJUST_VOLUME(src1, volume);
src2 = ((dst[0])<<8|dst[1]);
src += 2;
dst_sample = src1+src2;
if ( dst_sample > max_audioval ) {
dst_sample = max_audioval;
} else
if ( dst_sample < min_audioval ) {
dst_sample = min_audioval;
}
dst[1] = dst_sample&0xFF;
dst_sample >>= 8;
dst[0] = dst_sample&0xFF;
dst += 2;
}
}
break;
 
default: /* If this happens... FIXME! */
SDL_SetError("SDL_MixAudio(): unknown audio format");
return;
}
}
/contrib/sdk/sources/SDL-1.2.2_newlib/src/audio/SDL_sysaudio.h
0,0 → 1,150
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997, 1998, 1999, 2000, 2001 Sam Lantinga
 
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Library General Public
License as published by the Free Software Foundation; either
version 2 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
Library General Public License for more details.
 
You should have received a copy of the GNU Library General Public
License along with this library; if not, write to the Free
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
 
Sam Lantinga
slouken@devolution.com
*/
 
#ifdef SAVE_RCSID
static char rcsid =
"@(#) $Id: SDL_sysaudio.h,v 1.8 2001/07/23 02:58:42 slouken Exp $";
#endif
 
#ifndef _SDL_sysaudio_h
#define _SDL_sysaudio_h
 
#include "SDL_mutex.h"
#include "SDL_thread.h"
 
/* The SDL audio driver */
typedef struct SDL_AudioDevice SDL_AudioDevice;
 
/* Define the SDL audio driver structure */
#define _THIS SDL_AudioDevice *_this
#ifndef _STATUS
#define _STATUS SDL_status *status
#endif
struct SDL_AudioDevice {
/* * * */
/* The name of this audio driver */
const char *name;
 
/* * * */
/* The description of this audio driver */
const char *desc;
 
/* * * */
/* Public driver functions */
int (*OpenAudio)(_THIS, SDL_AudioSpec *spec);
void (*ThreadInit)(_THIS); /* Called by audio thread at start */
void (*WaitAudio)(_THIS);
void (*PlayAudio)(_THIS);
Uint8 *(*GetAudioBuf)(_THIS);
void (*WaitDone)(_THIS);
void (*CloseAudio)(_THIS);
 
/* * * */
/* Data common to all devices */
 
/* The current audio specification (shared with audio thread) */
SDL_AudioSpec spec;
 
/* An audio conversion block for audio format emulation */
SDL_AudioCVT convert;
 
/* Current state flags */
int enabled;
int paused;
int opened;
 
/* Fake audio buffer for when the audio hardware is busy */
Uint8 *fake_stream;
 
/* A semaphore for locking the mixing buffers */
SDL_mutex *mixer_lock;
 
/* A thread to feed the audio device */
SDL_Thread *thread;
Uint32 threadid;
 
/* * * */
/* Data private to this driver */
struct SDL_PrivateAudioData *hidden;
 
/* * * */
/* The function used to dispose of this structure */
void (*free)(_THIS);
};
#undef _THIS
 
typedef struct AudioBootStrap {
const char *name;
const char *desc;
int (*available)(void);
SDL_AudioDevice *(*create)(int devindex);
} AudioBootStrap;
 
#ifdef OPENBSD_AUDIO_SUPPORT
extern AudioBootStrap OPENBSD_AUDIO_bootstrap;
#endif
#ifdef OSS_SUPPORT
extern AudioBootStrap DSP_bootstrap;
extern AudioBootStrap DMA_bootstrap;
#endif
#ifdef ALSA_SUPPORT
extern AudioBootStrap ALSA_bootstrap;
#endif
#if (defined(unix) && !defined(__CYGWIN32__)) && \
!defined(OSS_SUPPORT) && !defined(ALSA_SUPPORT)
extern AudioBootStrap AUDIO_bootstrap;
#endif
#ifdef ARTSC_SUPPORT
extern AudioBootStrap ARTSC_bootstrap;
#endif
#ifdef ESD_SUPPORT
extern AudioBootStrap ESD_bootstrap;
#endif
#ifdef NAS_SUPPORT
extern AudioBootStrap NAS_bootstrap;
#endif
#ifdef ENABLE_DIRECTX
extern AudioBootStrap DSOUND_bootstrap;
#endif
#ifdef ENABLE_WINDIB
extern AudioBootStrap WAVEOUT_bootstrap;
#endif
#ifdef _AIX
extern AudioBootStrap Paud_bootstrap;
#endif
#ifdef __BEOS__
extern AudioBootStrap BAUDIO_bootstrap;
#endif
#if defined(macintosh) || TARGET_API_MAC_CARBON
extern AudioBootStrap SNDMGR_bootstrap;
#endif
#ifdef ENABLE_AHI
extern AudioBootStrap AHI_bootstrap;
#endif
#ifdef DISKAUD_SUPPORT
extern AudioBootStrap DISKAUD_bootstrap;
#endif
 
/* This is the current audio device */
extern SDL_AudioDevice *current_audio;
 
#endif /* _SDL_sysaudio_h */
/contrib/sdk/sources/SDL-1.2.2_newlib/src/audio/SDL_wave.c
0,0 → 1,591
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997, 1998, 1999, 2000, 2001 Sam Lantinga
 
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Library General Public
License as published by the Free Software Foundation; either
version 2 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
Library General Public License for more details.
 
You should have received a copy of the GNU Library General Public
License along with this library; if not, write to the Free
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
 
Sam Lantinga
slouken@devolution.com
*/
 
#ifdef SAVE_RCSID
static char rcsid =
"@(#) $Id: SDL_wave.c,v 1.2 2001/04/26 16:50:17 hercules Exp $";
#endif
 
#ifndef DISABLE_FILE
 
/* Microsoft WAVE file loading routines */
 
#include <stdlib.h>
#include <string.h>
 
#include "SDL_error.h"
#include "SDL_audio.h"
#include "SDL_wave.h"
#include "SDL_endian.h"
 
#ifndef NELEMS
#define NELEMS(array) ((sizeof array)/(sizeof array[0]))
#endif
 
static int ReadChunk(SDL_RWops *src, Chunk *chunk);
 
struct MS_ADPCM_decodestate {
Uint8 hPredictor;
Uint16 iDelta;
Sint16 iSamp1;
Sint16 iSamp2;
};
static struct MS_ADPCM_decoder {
WaveFMT wavefmt;
Uint16 wSamplesPerBlock;
Uint16 wNumCoef;
Sint16 aCoeff[7][2];
/* * * */
struct MS_ADPCM_decodestate state[2];
} MS_ADPCM_state;
 
static int InitMS_ADPCM(WaveFMT *format)
{
Uint8 *rogue_feel;
Uint16 extra_info;
int i;
 
/* Set the rogue pointer to the MS_ADPCM specific data */
MS_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding);
MS_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels);
MS_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency);
MS_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate);
MS_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign);
MS_ADPCM_state.wavefmt.bitspersample =
SDL_SwapLE16(format->bitspersample);
rogue_feel = (Uint8 *)format+sizeof(*format);
if ( sizeof(*format) == 16 ) {
extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]);
rogue_feel += sizeof(Uint16);
}
MS_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]);
rogue_feel += sizeof(Uint16);
MS_ADPCM_state.wNumCoef = ((rogue_feel[1]<<8)|rogue_feel[0]);
rogue_feel += sizeof(Uint16);
if ( MS_ADPCM_state.wNumCoef != 7 ) {
SDL_SetError("Unknown set of MS_ADPCM coefficients");
return(-1);
}
for ( i=0; i<MS_ADPCM_state.wNumCoef; ++i ) {
MS_ADPCM_state.aCoeff[i][0] = ((rogue_feel[1]<<8)|rogue_feel[0]);
rogue_feel += sizeof(Uint16);
MS_ADPCM_state.aCoeff[i][1] = ((rogue_feel[1]<<8)|rogue_feel[0]);
rogue_feel += sizeof(Uint16);
}
return(0);
}
 
static Sint32 MS_ADPCM_nibble(struct MS_ADPCM_decodestate *state,
Uint8 nybble, Sint16 *coeff)
{
const Sint32 max_audioval = ((1<<(16-1))-1);
const Sint32 min_audioval = -(1<<(16-1));
const Sint32 adaptive[] = {
230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};
Sint32 new_sample, delta;
 
new_sample = ((state->iSamp1 * coeff[0]) +
(state->iSamp2 * coeff[1]))/256;
if ( nybble & 0x08 ) {
new_sample += state->iDelta * (nybble-0x10);
} else {
new_sample += state->iDelta * nybble;
}
if ( new_sample < min_audioval ) {
new_sample = min_audioval;
} else
if ( new_sample > max_audioval ) {
new_sample = max_audioval;
}
delta = ((Sint32)state->iDelta * adaptive[nybble])/256;
if ( delta < 16 ) {
delta = 16;
}
state->iDelta = delta;
state->iSamp2 = state->iSamp1;
state->iSamp1 = new_sample;
return(new_sample);
}
 
static int MS_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len)
{
struct MS_ADPCM_decodestate *state[2];
Uint8 *freeable, *encoded, *decoded;
Sint32 encoded_len, samplesleft;
Sint8 nybble, stereo;
Sint16 *coeff[2];
Sint32 new_sample;
 
/* Allocate the proper sized output buffer */
encoded_len = *audio_len;
encoded = *audio_buf;
freeable = *audio_buf;
*audio_len = (encoded_len/MS_ADPCM_state.wavefmt.blockalign) *
MS_ADPCM_state.wSamplesPerBlock*
MS_ADPCM_state.wavefmt.channels*sizeof(Sint16);
*audio_buf = (Uint8 *)malloc(*audio_len);
if ( *audio_buf == NULL ) {
SDL_Error(SDL_ENOMEM);
return(-1);
}
decoded = *audio_buf;
 
/* Get ready... Go! */
stereo = (MS_ADPCM_state.wavefmt.channels == 2);
state[0] = &MS_ADPCM_state.state[0];
state[1] = &MS_ADPCM_state.state[stereo];
while ( encoded_len >= MS_ADPCM_state.wavefmt.blockalign ) {
/* Grab the initial information for this block */
state[0]->hPredictor = *encoded++;
if ( stereo ) {
state[1]->hPredictor = *encoded++;
}
state[0]->iDelta = ((encoded[1]<<8)|encoded[0]);
encoded += sizeof(Sint16);
if ( stereo ) {
state[1]->iDelta = ((encoded[1]<<8)|encoded[0]);
encoded += sizeof(Sint16);
}
state[0]->iSamp1 = ((encoded[1]<<8)|encoded[0]);
encoded += sizeof(Sint16);
if ( stereo ) {
state[1]->iSamp1 = ((encoded[1]<<8)|encoded[0]);
encoded += sizeof(Sint16);
}
state[0]->iSamp2 = ((encoded[1]<<8)|encoded[0]);
encoded += sizeof(Sint16);
if ( stereo ) {
state[1]->iSamp2 = ((encoded[1]<<8)|encoded[0]);
encoded += sizeof(Sint16);
}
coeff[0] = MS_ADPCM_state.aCoeff[state[0]->hPredictor];
coeff[1] = MS_ADPCM_state.aCoeff[state[1]->hPredictor];
 
/* Store the two initial samples we start with */
decoded[0] = state[0]->iSamp2&0xFF;
decoded[1] = state[0]->iSamp2>>8;
decoded += 2;
if ( stereo ) {
decoded[0] = state[1]->iSamp2&0xFF;
decoded[1] = state[1]->iSamp2>>8;
decoded += 2;
}
decoded[0] = state[0]->iSamp1&0xFF;
decoded[1] = state[0]->iSamp1>>8;
decoded += 2;
if ( stereo ) {
decoded[0] = state[1]->iSamp1&0xFF;
decoded[1] = state[1]->iSamp1>>8;
decoded += 2;
}
 
/* Decode and store the other samples in this block */
samplesleft = (MS_ADPCM_state.wSamplesPerBlock-2)*
MS_ADPCM_state.wavefmt.channels;
while ( samplesleft > 0 ) {
nybble = (*encoded)>>4;
new_sample = MS_ADPCM_nibble(state[0],nybble,coeff[0]);
decoded[0] = new_sample&0xFF;
new_sample >>= 8;
decoded[1] = new_sample&0xFF;
decoded += 2;
 
nybble = (*encoded)&0x0F;
new_sample = MS_ADPCM_nibble(state[1],nybble,coeff[1]);
decoded[0] = new_sample&0xFF;
new_sample >>= 8;
decoded[1] = new_sample&0xFF;
decoded += 2;
 
++encoded;
samplesleft -= 2;
}
encoded_len -= MS_ADPCM_state.wavefmt.blockalign;
}
free(freeable);
return(0);
}
 
struct IMA_ADPCM_decodestate {
Sint32 sample;
Sint8 index;
};
static struct IMA_ADPCM_decoder {
WaveFMT wavefmt;
Uint16 wSamplesPerBlock;
/* * * */
struct IMA_ADPCM_decodestate state[2];
} IMA_ADPCM_state;
 
static int InitIMA_ADPCM(WaveFMT *format)
{
Uint8 *rogue_feel;
Uint16 extra_info;
 
/* Set the rogue pointer to the IMA_ADPCM specific data */
IMA_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding);
IMA_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels);
IMA_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency);
IMA_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate);
IMA_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign);
IMA_ADPCM_state.wavefmt.bitspersample =
SDL_SwapLE16(format->bitspersample);
rogue_feel = (Uint8 *)format+sizeof(*format);
if ( sizeof(*format) == 16 ) {
extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]);
rogue_feel += sizeof(Uint16);
}
IMA_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]);
return(0);
}
 
static Sint32 IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state,Uint8 nybble)
{
const Sint32 max_audioval = ((1<<(16-1))-1);
const Sint32 min_audioval = -(1<<(16-1));
const int index_table[16] = {
-1, -1, -1, -1,
2, 4, 6, 8,
-1, -1, -1, -1,
2, 4, 6, 8
};
const Sint32 step_table[89] = {
7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31,
34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130,
143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408,
449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282,
1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327,
3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630,
9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350,
22385, 24623, 27086, 29794, 32767
};
Sint32 delta, step;
 
/* Compute difference and new sample value */
step = step_table[state->index];
delta = step >> 3;
if ( nybble & 0x04 ) delta += step;
if ( nybble & 0x02 ) delta += (step >> 1);
if ( nybble & 0x01 ) delta += (step >> 2);
if ( nybble & 0x08 ) delta = -delta;
state->sample += delta;
 
/* Update index value */
state->index += index_table[nybble];
if ( state->index > 88 ) {
state->index = 88;
} else
if ( state->index < 0 ) {
state->index = 0;
}
 
/* Clamp output sample */
if ( state->sample > max_audioval ) {
state->sample = max_audioval;
} else
if ( state->sample < min_audioval ) {
state->sample = min_audioval;
}
return(state->sample);
}
 
/* Fill the decode buffer with a channel block of data (8 samples) */
static void Fill_IMA_ADPCM_block(Uint8 *decoded, Uint8 *encoded,
int channel, int numchannels, struct IMA_ADPCM_decodestate *state)
{
int i;
Sint8 nybble;
Sint32 new_sample;
 
decoded += (channel * 2);
for ( i=0; i<4; ++i ) {
nybble = (*encoded)&0x0F;
new_sample = IMA_ADPCM_nibble(state, nybble);
decoded[0] = new_sample&0xFF;
new_sample >>= 8;
decoded[1] = new_sample&0xFF;
decoded += 2 * numchannels;
 
nybble = (*encoded)>>4;
new_sample = IMA_ADPCM_nibble(state, nybble);
decoded[0] = new_sample&0xFF;
new_sample >>= 8;
decoded[1] = new_sample&0xFF;
decoded += 2 * numchannels;
 
++encoded;
}
}
 
static int IMA_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len)
{
struct IMA_ADPCM_decodestate *state;
Uint8 *freeable, *encoded, *decoded;
Sint32 encoded_len, samplesleft;
int c, channels;
 
/* Check to make sure we have enough variables in the state array */
channels = IMA_ADPCM_state.wavefmt.channels;
if ( channels > NELEMS(IMA_ADPCM_state.state) ) {
SDL_SetError("IMA ADPCM decoder can only handle %d channels",
NELEMS(IMA_ADPCM_state.state));
return(-1);
}
state = IMA_ADPCM_state.state;
 
/* Allocate the proper sized output buffer */
encoded_len = *audio_len;
encoded = *audio_buf;
freeable = *audio_buf;
*audio_len = (encoded_len/IMA_ADPCM_state.wavefmt.blockalign) *
IMA_ADPCM_state.wSamplesPerBlock*
IMA_ADPCM_state.wavefmt.channels*sizeof(Sint16);
*audio_buf = (Uint8 *)malloc(*audio_len);
if ( *audio_buf == NULL ) {
SDL_Error(SDL_ENOMEM);
return(-1);
}
decoded = *audio_buf;
 
/* Get ready... Go! */
while ( encoded_len >= IMA_ADPCM_state.wavefmt.blockalign ) {
/* Grab the initial information for this block */
for ( c=0; c<channels; ++c ) {
/* Fill the state information for this block */
state[c].sample = ((encoded[1]<<8)|encoded[0]);
encoded += 2;
if ( state[c].sample & 0x8000 ) {
state[c].sample -= 0x10000;
}
state[c].index = *encoded++;
/* Reserved byte in buffer header, should be 0 */
if ( *encoded++ != 0 ) {
/* Uh oh, corrupt data? Buggy code? */;
}
 
/* Store the initial sample we start with */
decoded[0] = state[c].sample&0xFF;
decoded[1] = state[c].sample>>8;
decoded += 2;
}
 
/* Decode and store the other samples in this block */
samplesleft = (IMA_ADPCM_state.wSamplesPerBlock-1)*channels;
while ( samplesleft > 0 ) {
for ( c=0; c<channels; ++c ) {
Fill_IMA_ADPCM_block(decoded, encoded,
c, channels, &state[c]);
encoded += 4;
samplesleft -= 8;
}
decoded += (channels * 8 * 2);
}
encoded_len -= IMA_ADPCM_state.wavefmt.blockalign;
}
free(freeable);
return(0);
}
 
SDL_AudioSpec * SDL_LoadWAV_RW (SDL_RWops *src, int freesrc,
SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len)
{
int was_error;
Chunk chunk;
int lenread;
int MS_ADPCM_encoded, IMA_ADPCM_encoded;
int samplesize;
 
/* WAV magic header */
Uint32 RIFFchunk;
Uint32 wavelen;
Uint32 WAVEmagic;
 
/* FMT chunk */
WaveFMT *format = NULL;
 
/* Make sure we are passed a valid data source */
was_error = 0;
if ( src == NULL ) {
was_error = 1;
goto done;
}
/* Check the magic header */
RIFFchunk = SDL_ReadLE32(src);
wavelen = SDL_ReadLE32(src);
WAVEmagic = SDL_ReadLE32(src);
if ( (RIFFchunk != RIFF) || (WAVEmagic != WAVE) ) {
SDL_SetError("Unrecognized file type (not WAVE)");
was_error = 1;
goto done;
}
 
/* Read the audio data format chunk */
chunk.data = NULL;
do {
if ( chunk.data != NULL ) {
free(chunk.data);
}
lenread = ReadChunk(src, &chunk);
if ( lenread < 0 ) {
was_error = 1;
goto done;
}
} while ( (chunk.magic == FACT) || (chunk.magic == LIST) );
 
/* Decode the audio data format */
format = (WaveFMT *)chunk.data;
if ( chunk.magic != FMT ) {
SDL_SetError("Complex WAVE files not supported");
was_error = 1;
goto done;
}
MS_ADPCM_encoded = IMA_ADPCM_encoded = 0;
switch (SDL_SwapLE16(format->encoding)) {
case PCM_CODE:
/* We can understand this */
break;
case MS_ADPCM_CODE:
/* Try to understand this */
if ( InitMS_ADPCM(format) < 0 ) {
was_error = 1;
goto done;
}
MS_ADPCM_encoded = 1;
break;
case IMA_ADPCM_CODE:
/* Try to understand this */
if ( InitIMA_ADPCM(format) < 0 ) {
was_error = 1;
goto done;
}
IMA_ADPCM_encoded = 1;
break;
default:
SDL_SetError("Unknown WAVE data format: 0x%.4x",
SDL_SwapLE16(format->encoding));
was_error = 1;
goto done;
}
memset(spec, 0, (sizeof *spec));
spec->freq = SDL_SwapLE32(format->frequency);
switch (SDL_SwapLE16(format->bitspersample)) {
case 4:
if ( MS_ADPCM_encoded || IMA_ADPCM_encoded ) {
spec->format = AUDIO_S16;
} else {
was_error = 1;
}
break;
case 8:
spec->format = AUDIO_U8;
break;
case 16:
spec->format = AUDIO_S16;
break;
default:
was_error = 1;
break;
}
if ( was_error ) {
SDL_SetError("Unknown %d-bit PCM data format",
SDL_SwapLE16(format->bitspersample));
goto done;
}
spec->channels = (Uint8)SDL_SwapLE16(format->channels);
spec->samples = 4096; /* Good default buffer size */
 
/* Read the audio data chunk */
*audio_buf = NULL;
do {
if ( *audio_buf != NULL ) {
free(*audio_buf);
}
lenread = ReadChunk(src, &chunk);
if ( lenread < 0 ) {
was_error = 1;
goto done;
}
*audio_len = lenread;
*audio_buf = chunk.data;
} while ( chunk.magic != DATA );
 
if ( MS_ADPCM_encoded ) {
if ( MS_ADPCM_decode(audio_buf, audio_len) < 0 ) {
was_error = 1;
goto done;
}
}
if ( IMA_ADPCM_encoded ) {
if ( IMA_ADPCM_decode(audio_buf, audio_len) < 0 ) {
was_error = 1;
goto done;
}
}
 
/* Don't return a buffer that isn't a multiple of samplesize */
samplesize = ((spec->format & 0xFF)/8)*spec->channels;
*audio_len &= ~(samplesize-1);
 
done:
if ( format != NULL ) {
free(format);
}
if ( freesrc && src ) {
SDL_RWclose(src);
}
if ( was_error ) {
spec = NULL;
}
return(spec);
}
 
/* Since the WAV memory is allocated in the shared library, it must also
be freed here. (Necessary under Win32, VC++)
*/
void SDL_FreeWAV(Uint8 *audio_buf)
{
if ( audio_buf != NULL ) {
free(audio_buf);
}
}
 
static int ReadChunk(SDL_RWops *src, Chunk *chunk)
{
chunk->magic = SDL_ReadLE32(src);
chunk->length = SDL_ReadLE32(src);
chunk->data = (Uint8 *)malloc(chunk->length);
if ( chunk->data == NULL ) {
SDL_Error(SDL_ENOMEM);
return(-1);
}
if ( SDL_RWread(src, chunk->data, chunk->length, 1) != 1 ) {
SDL_Error(SDL_EFREAD);
free(chunk->data);
return(-1);
}
return(chunk->length);
}
 
#endif /* ENABLE_FILE */
/contrib/sdk/sources/SDL-1.2.2_newlib/src/audio/SDL_wave.h
0,0 → 1,65
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997, 1998, 1999, 2000, 2001 Sam Lantinga
 
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Library General Public
License as published by the Free Software Foundation; either
version 2 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
Library General Public License for more details.
 
You should have received a copy of the GNU Library General Public
License along with this library; if not, write to the Free
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
 
Sam Lantinga
slouken@devolution.com
*/
 
#ifdef SAVE_RCSID
static char rcsid =
"@(#) $Id: SDL_wave.h,v 1.2 2001/04/26 16:50:17 hercules Exp $";
#endif
 
/* WAVE files are little-endian */
 
/*******************************************/
/* Define values for Microsoft WAVE format */
/*******************************************/
#define RIFF 0x46464952 /* "RIFF" */
#define WAVE 0x45564157 /* "WAVE" */
#define FACT 0x74636166 /* "fact" */
#define LIST 0x5453494c /* "LIST" */
#define FMT 0x20746D66 /* "fmt " */
#define DATA 0x61746164 /* "data" */
#define PCM_CODE 0x0001
#define MS_ADPCM_CODE 0x0002
#define IMA_ADPCM_CODE 0x0011
#define WAVE_MONO 1
#define WAVE_STEREO 2
 
/* Normally, these three chunks come consecutively in a WAVE file */
typedef struct WaveFMT {
/* Not saved in the chunk we read:
Uint32 FMTchunk;
Uint32 fmtlen;
*/
Uint16 encoding;
Uint16 channels; /* 1 = mono, 2 = stereo */
Uint32 frequency; /* One of 11025, 22050, or 44100 Hz */
Uint32 byterate; /* Average bytes per second */
Uint16 blockalign; /* Bytes per sample block */
Uint16 bitspersample; /* One of 8, 12, 16, or 4 for ADPCM */
} WaveFMT;
 
/* The general chunk found in the WAVE file */
typedef struct Chunk {
Uint32 magic;
Uint32 length;
Uint8 *data; /* Data includes magic and length */
} Chunk;
 
/contrib/sdk/sources/SDL-1.2.2_newlib/src/video/SDL_video.c
114,6 → 114,31
#endif
 
 
typedef union{
unsigned val;
struct{
short x;
short y;
};
}ksys_pos_t;
 
static inline
ksys_pos_t _ksys_screen_size()
{
ksys_pos_t size;
ksys_pos_t size_tmp;
__asm__ __volatile__(
"int $0x40"
:"=a"(size_tmp)
:"a"(14)
:"memory"
);
size.x = size_tmp.y;
size.y = size_tmp.x;
return size;
}
 
 
/*
* Initialize the video and event subsystems -- determine native pixel format
*/
235,6 → 260,10
#endif
video->info.vfmt = SDL_VideoSurface->format;
 
ksys_pos_t screen_s = _ksys_screen_size();
video->info.current_h = screen_s.y;
video->info.current_w = screen_s.x;
 
/* Start the event loop */
if ( SDL_StartEventLoop(flags) < 0 ) {
SDL_VideoQuit();