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wolf3d:

Browse Source 
- Now used standart SDL timer functions;
 - Removed old files and useless   functions.

git-svn-id: svn://kolibrios.org@9790 a494cfbc-eb01-0410-851d-a64ba20cac60
This commit is contained in:
turbocat 2022-04-26 08:58:22 +00:00
parent 2f26d486e6
commit 9ba1adb152
38 changed files with 65 additions and 2845 deletions
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22
contrib/games/wolf3d/Makefile
View File

@ -1,14 +1,18 @@
CC = kos32-gcc
LD = kos32-ld
OBJCOPY = kos32-objcopy
KPACK = kpack
SDK_DIR = $(abspath ../../sdk)
CFLAGS = -c -fno-ident -O2 -fomit-frame-pointer -fno-ident -U__WIN32__ -U_Win32 -U_WIN32 -U__MINGW32__ -UWIN32 -D_KOLIBRI
LDFLAGS = -static -S -nostdlib -T $(SDK_DIR)/sources/newlib/app.lds --image-base 0
LDFLAGS = -static -S -nostdlib -T $(SDK_DIR)/sources/newlib/app.lds --image-base 0 --subsystem native
INCLUDES = -I$(SDK_DIR)/sources/newlib/libc/include -I$(SDK_DIR)/sources/SDL-1.2.2_newlib/include -I. -I SDL_mixer
LIBPATH = -L $(SDK_DIR)/lib -L /home/autobuild/tools/win32/mingw32/lib -L $(SDK_DIR)/lib
TARGET = bin/wolf3d
OBJECTS += wl_cloudsky.o
OBJECTS += wl_debug.o
OBJECTS += id_sd.o
@ -40,10 +44,6 @@ OBJECTS += joystick_stub.o
OBJECTS += kolibri.o
OBJECTS += mame/fmopl.o
SDL_OBJ += SDL/SDL_wave.o
SDL_OBJ += SDL/SDL_audiocvt.o
SDL_OBJ += SDL/SDL_mixer.o
SDL_OBJ += SDL/uSDL.o
SDL_MIX_OBJ += SDL_mixer/mixer.o
SDL_MIX_OBJ += SDL_mixer/music.o
@ -52,10 +52,12 @@ SDL_MIX_OBJ += SDL_mixer/load_voc.o
SDL_MIX_OBJ += SDL_mixer/effects_internal.o
SDL_MIX_OBJ += SDL_mixer/effect_position.o
default: $(OBJECTS) $(SDL_MIX_OBJ) $(SDL_OBJ)
kos32-ld $(LDFLAGS) $(LIBPATH) --subsystem native -o bin/wolf3d $(OBJECTS) $(SDL_MIX_OBJ) $(SDL_OBJ) -lSDLn -lsound -lstdc++ -lsupc++ -lgcc -lc.dll
objcopy bin/wolf3d -O binary
kpack --nologo bin/wolf3d
LIBS = -lSDLn -lsound -lgcc -lc.dll
$(TARGET): $(OBJECTS) $(SDL_MIX_OBJ) $(SDL_OBJ)
$(LD) $(LDFLAGS) $(LIBPATH) -o $(TARGET) $(OBJECTS) $(SDL_MIX_OBJ) $(SDL_OBJ) $(LIBS)
$(OBJCOPY) $(TARGET) -O binary
$(KPACK) --nologo $(TARGET)
%.o : %.cpp
$(CC) $(CFLAGS) $(INCLUDES) -o $@ $<
@ -64,4 +66,4 @@ default: $(OBJECTS) $(SDL_MIX_OBJ) $(SDL_OBJ)
$(CC) $(CFLAGS) $(INCLUDES) -o $@ $<
clean:
rm *.o SDL_mixer/*.o mame/*.o SDL/*.o *.d SDL_mixer/*.d mame/*.d SDL/*.d
rm -f $(OBJECTS)

642
contrib/games/wolf3d/SDL/SDL_audiocvt.c
View File

@ -1,642 +0,0 @@
/*
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);
}

218
contrib/games/wolf3d/SDL/SDL_mixer.c
View File

@ -1,218 +0,0 @@
/*
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;
}
}

150
contrib/games/wolf3d/SDL/SDL_sysaudio.h
View File

@ -1,150 +0,0 @@
/*
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 */

591
contrib/games/wolf3d/SDL/SDL_wave.c
View File

@ -1,591 +0,0 @@
/*
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 */

65
contrib/games/wolf3d/SDL/SDL_wave.h
View File

@ -1,65 +0,0 @@
/*
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;

28
contrib/games/wolf3d/SDL/uSDL.c
View File

@ -1,28 +0,0 @@
static unsigned __starttime;
void uSDL_StartTicks(void){
__asm__ __volatile__ (
"int $0x40"
:"=a"(__starttime)
:"a"(26),"b"(9)
:"memory"
);
}
unsigned uSDL_GetTicks(void){
unsigned __curtime;
__asm__ __volatile__(
"int $0x40"
:"=a"(__curtime)
:"a"(26),"b"(9)
:"memory"
);
return (__curtime-__starttime)*10;
}
void uSDL_Delay(unsigned ms){
unsigned start = uSDL_GetTicks();
do{
__asm__ __volatile__("int $0x40" :: "a"(5),"b"(1));
}while (uSDL_GetTicks()-start < ms);
}

66
contrib/games/wolf3d/SDL_mixer/dynamic_flac.h
View File

@ -1,66 +0,0 @@
/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
The following file defines all of the functions/objects used to dynamically
link to the libFLAC library.
~ Austen Dicken (admin@cvpcs.org)
*/
#ifdef FLAC_MUSIC
#include <FLAC/stream_decoder.h>
typedef struct {
int loaded;
void *handle;
FLAC__StreamDecoder *(*FLAC__stream_decoder_new)();
void (*FLAC__stream_decoder_delete)(FLAC__StreamDecoder *decoder);
FLAC__StreamDecoderInitStatus (*FLAC__stream_decoder_init_stream)(
FLAC__StreamDecoder *decoder,
FLAC__StreamDecoderReadCallback read_callback,
FLAC__StreamDecoderSeekCallback seek_callback,
FLAC__StreamDecoderTellCallback tell_callback,
FLAC__StreamDecoderLengthCallback length_callback,
FLAC__StreamDecoderEofCallback eof_callback,
FLAC__StreamDecoderWriteCallback write_callback,
FLAC__StreamDecoderMetadataCallback metadata_callback,
FLAC__StreamDecoderErrorCallback error_callback,
void *client_data);
FLAC__bool (*FLAC__stream_decoder_finish)(FLAC__StreamDecoder *decoder);
FLAC__bool (*FLAC__stream_decoder_flush)(FLAC__StreamDecoder *decoder);
FLAC__bool (*FLAC__stream_decoder_process_single)(
FLAC__StreamDecoder *decoder);
FLAC__bool (*FLAC__stream_decoder_process_until_end_of_metadata)(
FLAC__StreamDecoder *decoder);
FLAC__bool (*FLAC__stream_decoder_process_until_end_of_stream)(
FLAC__StreamDecoder *decoder);
FLAC__bool (*FLAC__stream_decoder_seek_absolute)(
FLAC__StreamDecoder *decoder,
FLAC__uint64 sample);
FLAC__StreamDecoderState (*FLAC__stream_decoder_get_state)(
const FLAC__StreamDecoder *decoder);
} flac_loader;
extern flac_loader flac;
#endif /* FLAC_MUSIC */
extern int Mix_InitFLAC();
extern void Mix_QuitFLAC();

57
contrib/games/wolf3d/SDL_mixer/dynamic_fluidsynth.h
View File

@ -1,57 +0,0 @@
/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
James Le Cuirot
chewi@aura-online.co.uk
*/
#ifdef USE_FLUIDSYNTH_MIDI
#include <fluidsynth.h>
typedef struct {
int loaded;
void *handle;
int (*delete_fluid_player)(fluid_player_t*);
void (*delete_fluid_settings)(fluid_settings_t*);
int (*delete_fluid_synth)(fluid_synth_t*);
int (*fluid_player_add)(fluid_player_t*, const char*);
int (*fluid_player_add_mem)(fluid_player_t*, const void*, size_t);
int (*fluid_player_get_status)(fluid_player_t*);
int (*fluid_player_play)(fluid_player_t*);
int (*fluid_player_set_loop)(fluid_player_t*, int);
int (*fluid_player_stop)(fluid_player_t*);
int (*fluid_settings_setnum)(fluid_settings_t*, const char*, double);
fluid_settings_t* (*fluid_synth_get_settings)(fluid_synth_t*);
void (*fluid_synth_set_gain)(fluid_synth_t*, float);
int (*fluid_synth_sfload)(fluid_synth_t*, const char*, int);
int (*fluid_synth_write_s16)(fluid_synth_t*, int, void*, int, int, void*, int, int);
fluid_player_t* (*new_fluid_player)(fluid_synth_t*);
fluid_settings_t* (*new_fluid_settings)(void);
fluid_synth_t* (*new_fluid_synth)(fluid_settings_t*);
} fluidsynth_loader;
extern fluidsynth_loader fluidsynth;
#endif /* USE_FLUIDSYNTH_MIDI */
extern int Mix_InitFluidSynth();
extern void Mix_QuitFluidSynth();

62
contrib/games/wolf3d/SDL_mixer/dynamic_mod.h
View File

@ -1,62 +0,0 @@
/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#ifdef MOD_MUSIC
#include "mikmod.h"
typedef struct {
int loaded;
void *handle;
void (*MikMod_Exit)(void);
CHAR* (*MikMod_InfoDriver)(void);
CHAR* (*MikMod_InfoLoader)(void);
BOOL (*MikMod_Init)(CHAR*);
void (*MikMod_RegisterAllLoaders)(void);
void (*MikMod_RegisterDriver)(struct MDRIVER*);
int* MikMod_errno;
char* (*MikMod_strerror)(int);
BOOL (*Player_Active)(void);
void (*Player_Free)(MODULE*);
MODULE* (*Player_LoadGeneric)(MREADER*,int,BOOL);
void (*Player_SetPosition)(UWORD);
void (*Player_SetVolume)(SWORD);
void (*Player_Start)(MODULE*);
void (*Player_Stop)(void);
ULONG (*VC_WriteBytes)(SBYTE*,ULONG);
struct MDRIVER* drv_nos;
UWORD* md_device;
UWORD* md_mixfreq;
UWORD* md_mode;
UBYTE* md_musicvolume;
UBYTE* md_pansep;
UBYTE* md_reverb;
UBYTE* md_sndfxvolume;
UBYTE* md_volume;
} mikmod_loader;
extern mikmod_loader mikmod;
#endif /* MOD_MUSIC */
extern int Mix_InitMOD();
extern void Mix_QuitMOD();

47
contrib/games/wolf3d/SDL_mixer/dynamic_mp3.h
View File

@ -1,47 +0,0 @@
/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#ifdef MP3_MUSIC
#include "smpeg.h"
typedef struct {
int loaded;
void *handle;
void (*SMPEG_actualSpec)( SMPEG *mpeg, SDL_AudioSpec *spec );
void (*SMPEG_delete)( SMPEG* mpeg );
void (*SMPEG_enableaudio)( SMPEG* mpeg, int enable );
void (*SMPEG_enablevideo)( SMPEG* mpeg, int enable );
SMPEG* (*SMPEG_new_rwops)(SDL_RWops *src, SMPEG_Info* info, int sdl_audio);
void (*SMPEG_play)( SMPEG* mpeg );
int (*SMPEG_playAudio)( SMPEG *mpeg, Uint8 *stream, int len );
void (*SMPEG_rewind)( SMPEG* mpeg );
void (*SMPEG_setvolume)( SMPEG* mpeg, int volume );
void (*SMPEG_skip)( SMPEG* mpeg, float seconds );
SMPEGstatus (*SMPEG_status)( SMPEG* mpeg );
void (*SMPEG_stop)( SMPEG* mpeg );
} smpeg_loader;
extern smpeg_loader smpeg;
#endif /* MUSIC_MP3 */
extern int Mix_InitMP3();
extern void Mix_QuitMP3();

53
contrib/games/wolf3d/SDL_mixer/dynamic_ogg.h
View File

@ -1,53 +0,0 @@
/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#ifdef OGG_MUSIC
#ifdef OGG_USE_TREMOR
#include <tremor/ivorbisfile.h>
#else
#include <vorbis/vorbisfile.h>
#endif
typedef struct {
int loaded;
void *handle;
int (*ov_clear)(OggVorbis_File *vf);
vorbis_info *(*ov_info)(OggVorbis_File *vf,int link);
int (*ov_open_callbacks)(void *datasource, OggVorbis_File *vf, char *initial, long ibytes, ov_callbacks callbacks);
ogg_int64_t (*ov_pcm_total)(OggVorbis_File *vf,int i);
#ifdef OGG_USE_TREMOR
long (*ov_read)(OggVorbis_File *vf,char *buffer,int length, int *bitstream);
#else
long (*ov_read)(OggVorbis_File *vf,char *buffer,int length, int bigendianp,int word,int sgned,int *bitstream);
#endif
#ifdef OGG_USE_TREMOR
int (*ov_time_seek)(OggVorbis_File *vf,ogg_int64_t pos);
#else
int (*ov_time_seek)(OggVorbis_File *vf,double pos);
#endif
} vorbis_loader;
extern vorbis_loader vorbis;
#endif /* OGG_MUSIC */
extern int Mix_InitOgg();
extern void Mix_QuitOgg();

51
contrib/games/wolf3d/SDL_mixer/fluidsynth.h
View File

@ -1,51 +0,0 @@
/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
James Le Cuirot
chewi@aura-online.co.uk
*/
#ifndef _FLUIDSYNTH_H_
#define _FLUIDSYNTH_H_
#ifdef USE_FLUIDSYNTH_MIDI
#include "dynamic_fluidsynth.h"
#include <SDL_rwops.h>
#include <SDL_audio.h>
typedef struct {
SDL_AudioCVT convert;
fluid_synth_t *synth;
fluid_player_t* player;
} FluidSynthMidiSong;
int fluidsynth_init(SDL_AudioSpec *mixer);
FluidSynthMidiSong *fluidsynth_loadsong_RW(SDL_RWops *rw, int freerw);
void fluidsynth_freesong(FluidSynthMidiSong *song);
void fluidsynth_start(FluidSynthMidiSong *song);
void fluidsynth_stop(FluidSynthMidiSong *song);
int fluidsynth_active(FluidSynthMidiSong *song);
void fluidsynth_setvolume(FluidSynthMidiSong *song, int volume);
int fluidsynth_playsome(FluidSynthMidiSong *song, void *stream, int len);
#endif /* USE_FLUIDSYNTH_MIDI */
#endif /* _FLUIDSYNTH_H_ */

31
contrib/games/wolf3d/SDL_mixer/load_flac.h
View File

@ -1,31 +0,0 @@
/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
This is the source needed to decode a FLAC into a waveform.
~ Austen Dicken (admin@cvpcs.org).
*/
/* $Id: $ */
#ifdef FLAC_MUSIC
/* Don't call this directly; use Mix_LoadWAV_RW() for now. */
SDL_AudioSpec *Mix_LoadFLAC_RW (SDL_RWops *src, int freesrc,
SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len);
#endif

31
contrib/games/wolf3d/SDL_mixer/load_ogg.h
View File

@ -1,31 +0,0 @@
/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
This is the source needed to decode an Ogg Vorbis into a waveform.
This file by Vaclav Slavik (vaclav.slavik@matfyz.cz).
*/
/* $Id$ */
#ifdef OGG_MUSIC
/* Don't call this directly; use Mix_LoadWAV_RW() for now. */
SDL_AudioSpec *Mix_LoadOGG_RW (SDL_RWops *src, int freesrc,
SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len);
#endif

31
contrib/games/wolf3d/SDL_mixer/mixer.c
View File

@ -32,19 +32,16 @@
#include "SDL_mixer.h"
#include "load_aiff.h"
#include "load_voc.h"
#include "load_ogg.h"
#include "load_flac.h"
#include "dynamic_flac.h"
#include "dynamic_mod.h"
#include "dynamic_mp3.h"
#include "dynamic_ogg.h"
//#include "load_ogg.h"
//#include "load_flac.h"
//#include "dynamic_flac.h"
//#include "dynamic_mod.h"
//#include "dynamic_mp3.h"
//#include "dynamic_ogg.h"
#define __MIX_INTERNAL_EFFECT__
#include "effects_internal.h"
#define uSDL_Delay SDL_Delay
#define uSDL_GetTicks SDL_GetTicks
/* Magic numbers for various audio file formats */
#define RIFF 0x46464952 /* "RIFF" */
#define WAVE 0x45564157 /* "WAVE" */
@ -303,7 +300,7 @@ static void mix_channels(void *udata, Uint8 *stream, int len)
}
/* Mix any playing channels... */
sdl_ticks = uSDL_GetTicks();
sdl_ticks = SDL_GetTicks();
for ( i=0; i<num_channels; ++i ) {
if( ! mix_channel[i].paused ) {
if ( mix_channel[i].expire > 0 && mix_channel[i].expire < sdl_ticks ) {
@ -860,7 +857,7 @@ int Mix_PlayChannelTimed(int which, Mix_Chunk *chunk, int loops, int ticks)
/* Queue up the audio data for this channel */
if ( which >= 0 && which < num_channels ) {
Uint32 sdl_ticks = uSDL_GetTicks();
Uint32 sdl_ticks = SDL_GetTicks();
if (Mix_Playing(which))
_Mix_channel_done_playing(which);
mix_channel[which].samples = chunk->abuf;
@ -891,7 +888,7 @@ int Mix_ExpireChannel(int which, int ticks)
}
} else if ( which < num_channels ) {
SDL_LockAudio();
mix_channel[which].expire = (ticks>0) ? ( uSDL_GetTicks() + ticks) : 0;
mix_channel[which].expire = (ticks>0) ? ( SDL_GetTicks() + ticks) : 0;
SDL_UnlockAudio();
++ status;
}
@ -930,7 +927,7 @@ int Mix_FadeInChannelTimed(int which, Mix_Chunk *chunk, int loops, int ms, int t
/* Queue up the audio data for this channel */
if ( which >= 0 && which < num_channels ) {
Uint32 sdl_ticks = uSDL_GetTicks();
Uint32 sdl_ticks = SDL_GetTicks();
if (Mix_Playing(which))
_Mix_channel_done_playing(which);
mix_channel[which].samples = chunk->abuf;
@ -1049,7 +1046,7 @@ int Mix_FadeOutChannel(int which, int ms)
mix_channel[which].fade_volume = mix_channel[which].volume;
mix_channel[which].fading = MIX_FADING_OUT;
mix_channel[which].fade_length = ms;
mix_channel[which].ticks_fade = uSDL_GetTicks();
mix_channel[which].ticks_fade = SDL_GetTicks();
/* only change fade_volume_reset if we're not fading. */
if (mix_channel[which].fading == MIX_NO_FADING) {
@ -1154,7 +1151,7 @@ void Mix_CloseAudio(void)
/* Pause a particular channel (or all) */
void Mix_Pause(int which)
{
Uint32 sdl_ticks = uSDL_GetTicks();
Uint32 sdl_ticks = SDL_GetTicks();
if ( which == -1 ) {
int i;
@ -1173,7 +1170,7 @@ void Mix_Pause(int which)
/* Resume a paused channel */
void Mix_Resume(int which)
{
Uint32 sdl_ticks = uSDL_GetTicks();
Uint32 sdl_ticks = SDL_GetTicks();
SDL_LockAudio();
if ( which == -1 ) {
@ -1263,7 +1260,7 @@ int Mix_GroupCount(int tag)
int Mix_GroupOldest(int tag)
{
int chan = -1;
Uint32 mintime = uSDL_GetTicks();
Uint32 mintime = SDL_GetTicks();
int i;
for( i=0; i < num_channels; i ++ ) {
if ( (mix_channel[i].tag==tag || tag==-1) && mix_channel[i].playing > 0

9
contrib/games/wolf3d/SDL_mixer/music.c
View File

@ -70,10 +70,7 @@
#if defined(MP3_MUSIC) || defined(MP3_MAD_MUSIC)
static SDL_AudioSpec used_mixer;
#endif
unsigned uSDL_GetTicks();
#define uSDL_Delay SDL_Delay
#define uSDL_GetTicks SDL_GetTicks
unsigned SDL_GetTicks();
int volatile music_active = 1;
static int volatile music_stopped = 0;
@ -764,7 +761,7 @@ void Mix_FreeMusic(Mix_Music *music)
/* Wait for any fade out to finish */
while ( music->fading == MIX_FADING_OUT ) {
SDL_UnlockAudio();
uSDL_Delay(100);
SDL_Delay(100);
SDL_LockAudio();
}
if ( music == music_playing ) {
@ -1016,7 +1013,7 @@ int Mix_FadeInMusicPos(Mix_Music *music, int loops, int ms, double position)
/* If the current music is fading out, wait for the fade to complete */
while ( music_playing && (music_playing->fading == MIX_FADING_OUT) ) {
SDL_UnlockAudio();
uSDL_Delay(100);
SDL_Delay(100);
SDL_LockAudio();
}
music_active = 1;

62
contrib/games/wolf3d/SDL_mixer/music_cmd.h
View File

@ -1,62 +0,0 @@
/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
/* This file supports an external command for playing music */
#ifdef CMD_MUSIC
#include <sys/types.h>
#include <limits.h>
#include <stdio.h>
#if defined(__linux__) && defined(__arm__)
# include <linux/limits.h>
#endif
typedef struct {
char file[PATH_MAX];
char cmd[PATH_MAX];
pid_t pid;
} MusicCMD;
/* Unimplemented */
extern void MusicCMD_SetVolume(int volume);
/* Load a music stream from the given file */
extern MusicCMD *MusicCMD_LoadSong(const char *cmd, const char *file);
/* Start playback of a given music stream */
extern void MusicCMD_Start(MusicCMD *music);
/* Stop playback of a stream previously started with MusicCMD_Start() */
extern void MusicCMD_Stop(MusicCMD *music);
/* Pause playback of a given music stream */
extern void MusicCMD_Pause(MusicCMD *music);
/* Resume playback of a given music stream */
extern void MusicCMD_Resume(MusicCMD *music);
/* Close the given music stream */
extern void MusicCMD_FreeSong(MusicCMD *music);
/* Return non-zero if a stream is currently playing */
extern int MusicCMD_Active(MusicCMD *music);
#endif /* CMD_MUSIC */

90
contrib/games/wolf3d/SDL_mixer/music_flac.h
View File

@ -1,90 +0,0 @@
/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
Header to handle loading FLAC music files in SDL.
~ Austen Dicken (admin@cvpcs.org)
*/
/* $Id: $ */
#ifdef FLAC_MUSIC
#include <FLAC/stream_decoder.h>
typedef struct {
FLAC__uint64 sample_size;
unsigned sample_rate;
unsigned channels;
unsigned bits_per_sample;
FLAC__uint64 total_samples;
// the following are used to handle the callback nature of the writer
int max_to_read;
char *data; // pointer to beginning of data array
int data_len; // size of data array
int data_read; // amount of data array used
char *overflow; // pointer to beginning of overflow array
int overflow_len; // size of overflow array
int overflow_read; // amount of overflow array used
} FLAC_Data;
typedef struct {
int playing;
int volume;
int section;
FLAC__StreamDecoder *flac_decoder;
FLAC_Data flac_data;
SDL_RWops *rwops;
int freerw;
SDL_AudioCVT cvt;
int len_available;
Uint8 *snd_available;
} FLAC_music;
/* Initialize the FLAC player, with the given mixer settings
This function returns 0, or -1 if there was an error.
*/
extern int FLAC_init(SDL_AudioSpec *mixer);
/* Set the volume for a FLAC stream */
extern void FLAC_setvolume(FLAC_music *music, int volume);
/* Load an FLAC stream from an SDL_RWops object */
extern FLAC_music *FLAC_new_RW(SDL_RWops *rw, int freerw);
/* Start playback of a given FLAC stream */
extern void FLAC_play(FLAC_music *music);
/* Return non-zero if a stream is currently playing */
extern int FLAC_playing(FLAC_music *music);
/* Play some of a stream previously started with FLAC_play() */
extern int FLAC_playAudio(FLAC_music *music, Uint8 *stream, int len);
/* Stop playback of a stream previously started with FLAC_play() */
extern void FLAC_stop(FLAC_music *music);
/* Close the given FLAC stream */
extern void FLAC_delete(FLAC_music *music);
/* Jump (seek) to a given position (time is in seconds) */
extern void FLAC_jump_to_time(FLAC_music *music, double time);
#endif /* FLAC_MUSIC */

72
contrib/games/wolf3d/SDL_mixer/music_mad.h
View File

@ -1,72 +0,0 @@
/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#ifdef MP3_MAD_MUSIC
#include "mad.h"
#include "SDL_rwops.h"
#include "SDL_audio.h"
#include "SDL_mixer.h"
#define MAD_INPUT_BUFFER_SIZE (5*8192)
#define MAD_OUTPUT_BUFFER_SIZE 8192
enum {
MS_input_eof = 0x0001,
MS_input_error = 0x0001,
MS_decode_eof = 0x0002,
MS_decode_error = 0x0004,
MS_error_flags = 0x000f,
MS_playing = 0x0100,
MS_cvt_decoded = 0x0200,
};
typedef struct {
SDL_RWops *rw;
int freerw;
struct mad_stream stream;
struct mad_frame frame;
struct mad_synth synth;
int frames_read;
mad_timer_t next_frame_start;
int volume;
int status;
int output_begin, output_end;
SDL_AudioSpec mixer;
SDL_AudioCVT cvt;
unsigned char input_buffer[MAD_INPUT_BUFFER_SIZE + MAD_BUFFER_GUARD];
unsigned char output_buffer[MAD_OUTPUT_BUFFER_SIZE];
} mad_data;
mad_data *mad_openFileRW(SDL_RWops *rw, SDL_AudioSpec *mixer, int freerw);
void mad_closeFile(mad_data *mp3_mad);
void mad_start(mad_data *mp3_mad);
void mad_stop(mad_data *mp3_mad);
int mad_isPlaying(mad_data *mp3_mad);
int mad_getSamples(mad_data *mp3_mad, Uint8 *stream, int len);
void mad_seek(mad_data *mp3_mad, double position);
void mad_setVolume(mad_data *mp3_mad, int volume);
#endif

62
contrib/games/wolf3d/SDL_mixer/music_mod.h
View File

@ -1,62 +0,0 @@
/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
/* $Id: music_mod.h 4211 2008-12-08 00:27:32Z slouken $ */
#ifdef MOD_MUSIC
/* This file supports MOD tracker music streams */
struct MODULE;
/* Initialize the Ogg Vorbis player, with the given mixer settings
This function returns 0, or -1 if there was an error.
*/
extern int MOD_init(SDL_AudioSpec *mixer);
/* Uninitialize the music players */
extern void MOD_exit(void);
/* Set the volume for a MOD stream */
extern void MOD_setvolume(struct MODULE *music, int volume);
/* Load a MOD stream from an SDL_RWops object */
extern struct MODULE *MOD_new_RW(SDL_RWops *rw, int freerw);
/* Start playback of a given MOD stream */
extern void MOD_play(struct MODULE *music);
/* Return non-zero if a stream is currently playing */
extern int MOD_playing(struct MODULE *music);
/* Play some of a stream previously started with MOD_play() */
extern int MOD_playAudio(struct MODULE *music, Uint8 *stream, int len);
/* Stop playback of a stream previously started with MOD_play() */
extern void MOD_stop(struct MODULE *music);
/* Close the given MOD stream */
extern void MOD_delete(struct MODULE *music);
/* Jump (seek) to a given position (time is in seconds) */
extern void MOD_jump_to_time(struct MODULE *music, double time);
#endif /* MOD_MUSIC */

42
contrib/games/wolf3d/SDL_mixer/music_modplug.h
View File

@ -1,42 +0,0 @@
#ifdef MODPLUG_MUSIC
#include "modplug.h"
#include "SDL_rwops.h"
#include "SDL_audio.h"
#include "SDL_mixer.h"
typedef struct {
ModPlugFile *file;
int playing;
} modplug_data;
int modplug_init(SDL_AudioSpec *mixer);
/* Uninitialize the music players */
void modplug_exit(void);
/* Set the volume for a modplug stream */
void modplug_setvolume(modplug_data *music, int volume);
/* Load a modplug stream from an SDL_RWops object */
modplug_data *modplug_new_RW(SDL_RWops *rw, int freerw);
/* Start playback of a given modplug stream */
void modplug_play(modplug_data *music);
/* Return non-zero if a stream is currently playing */
int modplug_playing(modplug_data *music);
/* Play some of a stream previously started with modplug_play() */
int modplug_playAudio(modplug_data *music, Uint8 *stream, int len);
/* Stop playback of a stream previously started with modplug_play() */
void modplug_stop(modplug_data *music);
/* Close the given modplug stream */
void modplug_delete(modplug_data *music);
/* Jump (seek) to a given position (time is in seconds) */
void modplug_jump_to_time(modplug_data *music, double time);
#endif

75
contrib/games/wolf3d/SDL_mixer/music_ogg.h
View File

@ -1,75 +0,0 @@
/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
/* $Id$ */
#ifdef OGG_MUSIC
/* This file supports Ogg Vorbis music streams */
#ifdef OGG_USE_TREMOR
#include <tremor/ivorbisfile.h>
#else
#include <vorbis/vorbisfile.h>
#endif
typedef struct {
SDL_RWops *rw;
int freerw;
int playing;
int volume;
OggVorbis_File vf;
int section;
SDL_AudioCVT cvt;
int len_available;
Uint8 *snd_available;
} OGG_music;
/* Initialize the Ogg Vorbis player, with the given mixer settings
This function returns 0, or -1 if there was an error.
*/
extern int OGG_init(SDL_AudioSpec *mixer);
/* Set the volume for an OGG stream */
extern void OGG_setvolume(OGG_music *music, int volume);
/* Load an OGG stream from an SDL_RWops object */
extern OGG_music *OGG_new_RW(SDL_RWops *rw, int freerw);
/* Start playback of a given OGG stream */
extern void OGG_play(OGG_music *music);
/* Return non-zero if a stream is currently playing */
extern int OGG_playing(OGG_music *music);
/* Play some of a stream previously started with OGG_play() */
extern int OGG_playAudio(OGG_music *music, Uint8 *stream, int len);
/* Stop playback of a stream previously started with OGG_play() */
extern void OGG_stop(OGG_music *music);
/* Close the given OGG stream */
extern void OGG_delete(OGG_music *music);
/* Jump (seek) to a given position (time is in seconds) */
extern void OGG_jump_to_time(OGG_music *music, double time);
#endif /* OGG_MUSIC */

60
contrib/games/wolf3d/SDL_mixer/wavestream.h
View File

@ -1,60 +0,0 @@
/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
/* $Id$ */
/* This file supports streaming WAV files, without volume adjustment */
#include <stdio.h>
typedef struct {
SDL_RWops *rw;
SDL_bool freerw;
long start;
long stop;
SDL_AudioCVT cvt;
} WAVStream;
/* Initialize the WAVStream player, with the given mixer settings
This function returns 0, or -1 if there was an error.
*/
extern int WAVStream_Init(SDL_AudioSpec *mixer);
/* Unimplemented */
extern void WAVStream_SetVolume(int volume);
/* Load a WAV stream from an SDL_RWops object */
extern WAVStream *WAVStream_LoadSong_RW(SDL_RWops *rw, const char *magic, int freerw);
/* Start playback of a given WAV stream */
extern void WAVStream_Start(WAVStream *wave);
/* Play some of a stream previously started with WAVStream_Start() */
extern int WAVStream_PlaySome(Uint8 *stream, int len);
/* Stop playback of a stream previously started with WAVStream_Start() */
extern void WAVStream_Stop(void);
/* Close the given WAV stream */
extern void WAVStream_FreeSong(WAVStream *wave);
/* Return non-zero if a stream is currently playing */
extern int WAVStream_Active(void);

5
contrib/games/wolf3d/Tupfile.lua
View File

@ -20,10 +20,9 @@ compile_gcc{
"joystick_stub.cpp", "kolibri.cpp", "mame/fmopl.cpp",
}
-- SDL and SDL_mixer --
-- SDL_mixer stubs --
compile_gcc{
"SDL/SDL_wave.c", "SDL/SDL_audiocvt.c", "SDL/SDL_mixer.c", "SDL_mixer/mixer.c", "SDL_mixer/music.c",
"SDL_mixer/load_aiff.c", "SDL_mixer/load_voc.c",
"SDL_mixer/mixer.c", "SDL_mixer/music.c", "SDL_mixer/load_aiff.c", "SDL_mixer/load_voc.c",
"SDL_mixer/effects_internal.c", "SDL_mixer/effect_position.c",
}

2
contrib/games/wolf3d/id_in.cpp
View File

@ -650,7 +650,7 @@ boolean IN_UserInput(longword delay)
IN_ProcessEvents();
if (IN_CheckAck())
return true;
uSDL_Delay(5);
SDL_Delay(5);
} while (GetTimeCount() - lasttime < delay);
return(false);
}

4
contrib/games/wolf3d/id_sd.cpp
View File

@ -935,7 +935,7 @@ SD_SetMusicMode(SMMode mode)
SD_FadeOutMusic();
while (SD_MusicPlaying())
uSDL_Delay(5);
SDL_Delay(5);
switch (mode)
{
@ -1284,7 +1284,7 @@ void
SD_WaitSoundDone(void)
{
while (SD_SoundPlaying())
uSDL_Delay(5);
SDL_Delay(5);
}
///////////////////////////////////////////////////////////////////////////

10
contrib/games/wolf3d/id_sd.h
View File

@ -119,17 +119,11 @@ extern SMMode MusicMode;
extern int DigiMap[];
extern int DigiChannel[];
#ifdef _KOLIBRI
extern void uSDL_Delay(unsigned time);
#else
#define uSDL_Delay SDL_Delay
#endif
#define GetTimeCount() (( uSDL_GetTicks()*7)/100)
#define GetTimeCount() ((SDL_GetTicks()*7)/100)
inline void Delay(int wolfticks)
{
if(wolfticks>0) uSDL_Delay(wolfticks * 100/ 7);
if(wolfticks>0) SDL_Delay(wolfticks * 100/ 7);
}
// Function prototypes

4
contrib/games/wolf3d/id_us_1.cpp
View File

@ -741,7 +741,7 @@ US_LineInput(int x,int y,char *buf,const char *def,boolean escok,
cursorvis ^= true;
}
else uSDL_Delay(5);
else SDL_Delay(5);
if (cursorvis)
USL_XORICursor(x,y,s,cursor);
@ -772,7 +772,7 @@ US_LineInput(int x,int y,char *buf,const char *def,boolean escok,
void US_InitRndT(int randomize)
{
if(randomize)
rndindex = ( uSDL_GetTicks() >> 4) & 0xff;
rndindex = ( SDL_GetTicks() >> 4) & 0xff;
else
rndindex = 0;
}

2
contrib/games/wolf3d/id_vl.h
View File

@ -28,7 +28,7 @@ extern SDL_Color gamepal[256];
// VGA hardware routines
//
#define VL_WaitVBL(a) uSDL_Delay((a)*8)
#define VL_WaitVBL(a) SDL_Delay((a)*8)
void VL_SetVGAPlaneMode (void);
void VL_SetTextMode (void);

210
contrib/games/wolf3d/kolibri.cpp
View File

@ -1,215 +1,7 @@
#include <stdlib.h>
#include <sys/stat.h>
#include <sys/ksys.h>
#include <string.h>
#define asm_inline __asm__ __volatile__
#pragma pack(push,1)
typedef union{
unsigned val;
struct{
short x;
short y;
};
}ksys_pos_t;
typedef union ksys_oskey_t{
unsigned val;
struct{
unsigned char state;
unsigned char code;
unsigned char ctrl_key;
};
}ksys_oskey_t;
typedef struct{
unsigned handle;
unsigned io_code;
unsigned *input;
int inp_size;
void *output;
int out_size;
}ksys_ioctl_t;
typedef struct{
void *data;
size_t size;
}ksys_ufile_t;
typedef struct{
unsigned p00;
union{
uint64_t p04;
struct {
unsigned p04dw;
unsigned p08dw;
};
};
unsigned p12;
union {
unsigned p16;
const char *new_name;
void *bdfe;
void *buf16;
const void *cbuf16;
};
char p20;
const char *p21;
}ksys70_t;
typedef struct {
int cpu_usage; //+0
int window_pos_info; //+4
short int reserved1; //+8
char name[12]; //+10
int memstart; //+22
int memused; //+26
int pid; //+30
int winx_start; //+34
int winy_start; //+38
int winx_size; //+42
int winy_size; //+46
short int slot_info; //+50
short int reserved2; //+52
int clientx; //+54
int clienty; //+58
int clientwidth; //+62
int clientheight; //+66
unsigned char window_state;//+70
char reserved3[1024-71]; //+71
}ksys_proc_table_t;
#pragma pack(pop)
static inline
int _ksys_process_info(ksys_proc_table_t* table, int pid)
{
int val;
asm_inline(
"int $0x40"
:"=a"(val)
:"a"(9), "b"(table), "c"(pid)
:"memory"
);
return val;
}
static inline
void _ksys_change_window(int new_x, int new_y, int new_w, int new_h)
{
asm_inline(
"int $0x40"
::"a"(67), "b"(new_x), "c"(new_y), "d"(new_w),"S"(new_h)
);
}
static inline
ksys_pos_t _ksys_screen_size()
{
ksys_pos_t size;
ksys_pos_t size_tmp;
asm_inline(
"int $0x40"
:"=a"(size_tmp)
:"a"(14)
:"memory"
);
size.x = size_tmp.y;
size.y = size_tmp.x;
return size;
}
void *memrchr(const void *m, int c, size_t n)
{
const unsigned char *s = (const unsigned char*)m;
c = (unsigned char)c;
while (n--) if (s[n]==c) return (void *)(s+n);
return 0;
}
void kolibri_set_win_center()
{
ksys_proc_table_t *info = (ksys_proc_table_t*)malloc(sizeof(ksys_proc_table_t));
_ksys_process_info(info, -1);
ksys_pos_t screen_size= _ksys_screen_size();
int new_x = screen_size.x/2-info->winx_size/2;
int new_y = screen_size.y/2-info->winy_size/2;
_ksys_change_window(new_x, new_y, -1, -1);
free(info);
}
int mkdir(const char *path, unsigned v)
{
int status;
ksys70_t dir_opt;
dir_opt.p00 = 9;
dir_opt.p21 = path;
asm_inline(
"int $0x40"
:"=a"(status)
:"a"(70), "b"(&dir_opt)
:"memory"
);
return status;
}
char *dirname (char *path)
{
static const char dot[] = ".";
char *last_slash;
/* Find last '/'. */
last_slash = path != NULL ? strrchr (path, '/') : NULL;
if (last_slash != NULL && last_slash != path && last_slash[1] == '\0')
{
/* Determine whether all remaining characters are slashes. */
char *runp;
for (runp = last_slash; runp != path; --runp)
if (runp[-1] != '/')
break;
/* The '/' is the last character, we have to look further. */
if (runp != path)
last_slash = (char*)memrchr((void*)path, '/', runp - path);
}
if (last_slash != NULL)
{
/* Determine whether all remaining characters are slashes. */
char *runp;
for (runp = last_slash; runp != path; --runp)
if (runp[-1] != '/')
break;
/* Terminate the path. */
if (runp == path)
{
/* The last slash is the first character in the string. We have to
return "/". As a special case we have to return "//" if there
are exactly two slashes at the beginning of the string. See
XBD 4.10 Path Name Resolution for more information. */
if (last_slash == path + 1)
++last_slash;
else
last_slash = path + 1;
}
else
last_slash = runp;
last_slash[0] = '\0';
}
else
/* This assignment is ill-designed but the XPG specs require to
return a string containing "." in any case no directory part is
found and so a static and constant string is required. */
path = (char *) dot;
return path;
}
void setcwd(char* path){
asm_inline(
"int $0x40"
::"a"(30), "b"(1), "c"(path)
:"memory"
);
}
extern unsigned screenWidth;
extern unsigned screenHeight;

6
contrib/games/wolf3d/wl_def.h
View File

@ -24,10 +24,6 @@
# define O_BINARY 0
#endif
#define uSDL_Delay SDL_Delay
#define uSDL_GetTicks SDL_GetTicks
#pragma pack(1)
#if defined(_arch_dreamcast)
@ -1395,7 +1391,7 @@ static inline fixed FixedMul(fixed a, fixed b)
#endif
#define DEMOCOND_SDL (!DEMOCOND_ORIG)
#define GetTicks() (( uSDL_GetTicks()*7)/100)
#define GetTicks() (( SDL_GetTicks()*7)/100)
#define ISPOINTER(x) ((((uintptr_t)(x)) & ~0xffff) != 0)

4
contrib/games/wolf3d/wl_draw.cpp
View File

@ -1074,12 +1074,12 @@ void CalcTics (void)
if (lasttimecount > (int32_t) GetTimeCount())
lasttimecount = GetTimeCount(); // if the game was paused a LONG time
uint32_t curtime = uSDL_GetTicks();
uint32_t curtime = SDL_GetTicks();
tics = (curtime * 7) / 100 - lasttimecount;
if(!tics)
{
// wait until end of current tic
uSDL_Delay(((lasttimecount + 1) * 100) / 7 - curtime);
SDL_Delay(((lasttimecount + 1) * 100) / 7 - curtime);
tics = 1;
}

2
contrib/games/wolf3d/wl_inter.cpp
View File

@ -403,7 +403,7 @@ BJ_Breathe (void)
static int which = 0, max = 10;
int pics[2] = { L_GUYPIC, L_GUY2PIC };
uSDL_Delay(5);
SDL_Delay(5);
if ((int32_t) GetTimeCount () - lastBreathTime > max)
{

16
contrib/games/wolf3d/wl_main.cpp
View File

@ -11,6 +11,11 @@
#include "wl_atmos.h"
#include <SDL_syswm.h>
#ifdef _KOLIBRI
#include <sys/ksys.h>
#include <libgen.h>
#endif
/*
=============================================================================
@ -24,10 +29,6 @@
*/
extern byte signon[];
extern void kolibri_set_win_center();
extern char* dirname(char* path);
extern void setcwd(char* path);
extern boolean SD_Started;
/*
=============================================================================
@ -1128,7 +1129,7 @@ void DoJukebox(void)
#ifndef SPEAR
#ifndef UPLOAD
start = (( uSDL_GetTicks()/10)%3)*6;
start = (( SDL_GetTicks()/10)%3)*6;
#else
start = 0;
#endif
@ -1239,9 +1240,6 @@ static void InitGame()
#endif
SignonScreen ();
#ifdef _KOLIBRI
kolibri_set_win_center();
#endif
#if defined _WIN32
if(!fullscreen)
@ -1978,7 +1976,7 @@ extern void kolibri_set_win_max(void);
int main (int argc, char *argv[])
{
#ifdef _KOLIBRI
setcwd(dirname(argv[0]));
_ksys_setcwd(dirname(argv[0]));
kolibri_set_win_max();
#endif

20
contrib/games/wolf3d/wl_menu.cpp
View File

@ -1956,7 +1956,7 @@ MouseSensitivity (int)
DrawMouseSens ();
do
{
uSDL_Delay(5);
SDL_Delay(5);
ReadAnyControl (&ci);
switch (ci.dir)
{
@ -2228,7 +2228,7 @@ EnterCtrlData (int index, CustomCtrls * cust, void (*DrawRtn) (int), void (*Prin
redraw = 0;
}
uSDL_Delay(5);
SDL_Delay(5);
ReadAnyControl (&ci);
if (type == MOUSE || type == JOYSTICK)
@ -2274,7 +2274,7 @@ EnterCtrlData (int index, CustomCtrls * cust, void (*DrawRtn) (int), void (*Prin
lastFlashTime = GetTimeCount();
VW_UpdateScreen ();
}
else uSDL_Delay(5);
else SDL_Delay(5);
//
// WHICH TYPE OF INPUT DO WE PROCESS?
@ -2397,7 +2397,7 @@ EnterCtrlData (int index, CustomCtrls * cust, void (*DrawRtn) (int), void (*Prin
while (!cust->allowed[which]);
redraw = 1;
SD_PlaySound (MOVEGUN1SND);
while (ReadAnyControl (&ci), ci.dir != dir_None) uSDL_Delay(5);
while (ReadAnyControl (&ci), ci.dir != dir_None) SDL_Delay(5);
IN_ClearKeysDown ();
break;
@ -2411,7 +2411,7 @@ EnterCtrlData (int index, CustomCtrls * cust, void (*DrawRtn) (int), void (*Prin
while (!cust->allowed[which]);
redraw = 1;
SD_PlaySound (MOVEGUN1SND);
while (ReadAnyControl (&ci), ci.dir != dir_None) uSDL_Delay(5);
while (ReadAnyControl (&ci), ci.dir != dir_None) SDL_Delay(5);
IN_ClearKeysDown ();
break;
case dir_North:
@ -2837,7 +2837,7 @@ CP_ChangeView (int)
do
{
CheckPause ();
uSDL_Delay(5);
SDL_Delay(5);
ReadAnyControl (&ci);
switch (ci.dir)
{
@ -3284,7 +3284,7 @@ HandleMenu (CP_iteminfo * item_i, CP_itemtype * items, void (*routine) (int w))
routine (which);
VW_UpdateScreen ();
}
else uSDL_Delay(5);
else SDL_Delay(5);
CheckPause ();
@ -3484,7 +3484,7 @@ DrawHalfStep (int x, int y)
VWB_DrawPic (x, y, C_CURSOR1PIC);
VW_UpdateScreen ();
SD_PlaySound (MOVEGUN1SND);
uSDL_Delay(1); //Fixed too long delay in the menu
SDL_Delay(1); //Fixed too long delay in the menu
}
@ -3526,7 +3526,7 @@ TicDelay (int count)
int32_t startTime = GetTimeCount ();
do
{
uSDL_Delay(5);
SDL_Delay(5);
ReadAnyControl (&ci);
}
while ((int32_t) GetTimeCount () - startTime < count && ci.dir != dir_None);
@ -3732,7 +3732,7 @@ Confirm (const char *string)
tick ^= 1;
lastBlinkTime = GetTimeCount();
}
else uSDL_Delay(5);
else SDL_Delay(5);
#ifdef SPANISH
}

4
contrib/games/wolf3d/wl_play.cpp
View File

@ -406,11 +406,11 @@ void PollControls (void)
if (demoplayback || demorecord) // demo recording and playback needs to be constant
{
// wait up to DEMOTICS Wolf tics
uint32_t curtime = uSDL_GetTicks();
uint32_t curtime = SDL_GetTicks();
lasttimecount += DEMOTICS;
int32_t timediff = (lasttimecount * 100) / 7 - curtime;
if(timediff > 0)
uSDL_Delay(timediff);
SDL_Delay(timediff);
if(timediff < -2 * DEMOTICS) // more than 2-times DEMOTICS behind?
lasttimecount = (curtime * 7) / 100; // yes, set to current timecount

2
contrib/games/wolf3d/wl_text.cpp
View File

@ -669,7 +669,7 @@ void ShowArticle (char *article)
firstpage = false;
}
}
uSDL_Delay(5);
SDL_Delay(5);
LastScan = 0;
ReadAnyControl(&ci);