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Changed directory structure to contain less symlinks so dependencies will get recalculated correctly by make

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pelya
2010-10-12 19:04:21 +03:00
parent a5ff12846e
commit 2e1a4992d3
1349 changed files with 33 additions and 48 deletions
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1217
project/jni/sdl-1.3/src/audio/SDL_audio.c Normal file
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project/jni/sdl-1.3/src/audio/SDL_audio_c.h Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* Functions and variables exported from SDL_audio.c for SDL_sysaudio.c */
/* Functions to get a list of "close" audio formats */
extern SDL_AudioFormat SDL_FirstAudioFormat(SDL_AudioFormat format);
extern SDL_AudioFormat SDL_NextAudioFormat(void);
/* Function to calculate the size and silence for a SDL_AudioSpec */
extern void SDL_CalculateAudioSpec(SDL_AudioSpec * spec);
/* The actual mixing thread function */
extern int SDLCALL SDL_RunAudio(void *audiop);
/* this is used internally to access some autogenerated code. */
typedef struct
{
SDL_AudioFormat src_fmt;
SDL_AudioFormat dst_fmt;
SDL_AudioFilter filter;
} SDL_AudioTypeFilters;
extern const SDL_AudioTypeFilters sdl_audio_type_filters[];
/* this is used internally to access some autogenerated code. */
typedef struct
{
SDL_AudioFormat fmt;
int channels;
int upsample;
int multiple;
SDL_AudioFilter filter;
} SDL_AudioRateFilters;
extern const SDL_AudioRateFilters sdl_audio_rate_filters[];
/* vi: set ts=4 sw=4 expandtab: */

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project/jni/sdl-1.3/src/audio/SDL_audiocvt.c Normal file
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project/jni/sdl-1.3/src/audio/SDL_audiodev.c Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* Get the name of the audio device we use for output */
#if SDL_AUDIO_DRIVER_BSD || SDL_AUDIO_DRIVER_OSS || SDL_AUDIO_DRIVER_SUNAUDIO
#include <fcntl.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <unistd.h> /* For close() */
#include "SDL_stdinc.h"
#include "SDL_audiodev_c.h"
#ifndef _PATH_DEV_DSP
#if defined(__NETBSD__) || defined(__OPENBSD__)
#define _PATH_DEV_DSP "/dev/audio"
#else
#define _PATH_DEV_DSP "/dev/dsp"
#endif
#endif
#ifndef _PATH_DEV_DSP24
#define _PATH_DEV_DSP24 "/dev/sound/dsp"
#endif
#ifndef _PATH_DEV_AUDIO
#define _PATH_DEV_AUDIO "/dev/audio"
#endif
static inline void
test_device(const char *fname, int flags, int (*test) (int fd),
char ***devices, int *devCount)
{
struct stat sb;
if ((stat(fname, &sb) == 0) && (S_ISCHR(sb.st_mode))) {
int audio_fd = open(fname, flags, 0);
if ((audio_fd >= 0) && (test(audio_fd))) {
void *p =
SDL_realloc(*devices, ((*devCount) + 1) * sizeof(char *));
if (p != NULL) {
size_t len = strlen(fname) + 1;
char *str = (char *) SDL_malloc(len);
*devices = (char **) p;
if (str != NULL) {
SDL_strlcpy(str, fname, len);
(*devices)[(*devCount)++] = str;
}
}
close(audio_fd);
}
}
}
void
SDL_FreeUnixAudioDevices(char ***devices, int *devCount)
{
int i = *devCount;
if ((i > 0) && (*devices != NULL)) {
while (i--) {
SDL_free((*devices)[i]);
}
}
if (*devices != NULL) {
SDL_free(*devices);
}
*devices = NULL;
*devCount = 0;
}
static int
test_stub(int fd)
{
return 1;
}
void
SDL_EnumUnixAudioDevices(int flags, int classic, int (*test) (int fd),
char ***devices, int *devCount)
{
const char *audiodev;
char audiopath[1024];
if (test == NULL)
test = test_stub;
/* Figure out what our audio device is */
if (((audiodev = SDL_getenv("SDL_PATH_DSP")) == NULL) &&
((audiodev = SDL_getenv("AUDIODEV")) == NULL)) {
if (classic) {
audiodev = _PATH_DEV_AUDIO;
} else {
struct stat sb;
/* Added support for /dev/sound/\* in Linux 2.4 */
if (((stat("/dev/sound", &sb) == 0) && S_ISDIR(sb.st_mode))
&& ((stat(_PATH_DEV_DSP24, &sb) == 0)
&& S_ISCHR(sb.st_mode))) {
audiodev = _PATH_DEV_DSP24;
} else {
audiodev = _PATH_DEV_DSP;
}
}
}
test_device(audiodev, flags, test, devices, devCount);
if (SDL_strlen(audiodev) < (sizeof(audiopath) - 3)) {
int instance = 0;
while (instance++ <= 64) {
SDL_snprintf(audiopath, SDL_arraysize(audiopath),
"%s%d", audiodev, instance);
test_device(audiopath, flags, test, devices, devCount);
}
}
}
#endif /* Audio driver selection */
/* vi: set ts=4 sw=4 expandtab: */

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project/jni/sdl-1.3/src/audio/SDL_audiodev_c.h Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
void SDL_EnumUnixAudioDevices(int flags, int classic, int (*test) (int fd),
char ***devs, int *count);
void SDL_FreeUnixAudioDevices(char ***devices, int *devCount);
/* vi: set ts=4 sw=4 expandtab: */

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#define SDL_AllocAudioMem SDL_malloc
#define SDL_FreeAudioMem SDL_free
/* vi: set ts=4 sw=4 expandtab: */

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project/jni/sdl-1.3/src/audio/SDL_audiotypecvt.c Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* This provides the default mixing callback for the SDL audio routines */
#include "SDL_cpuinfo.h"
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "SDL_sysaudio.h"
#include "SDL_mixer_MMX.h"
#include "SDL_mixer_MMX_VC.h"
#include "SDL_mixer_m68k.h"
/* 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_MixAudioFormat(Uint8 * dst, const Uint8 * src, SDL_AudioFormat format,
Uint32 len, int volume)
{
if (volume == 0) {
return;
}
switch (format) {
case AUDIO_U8:
{
#if defined(__GNUC__) && defined(__M68000__) && !defined(__mcoldfire__) && defined(SDL_ASSEMBLY_ROUTINES)
SDL_MixAudio_m68k_U8((char *) dst, (char *) src,
(unsigned long) len, (long) volume,
(char *) mix8);
#else
Uint8 src_sample;
while (len--) {
src_sample = *src;
ADJUST_VOLUME_U8(src_sample, volume);
*dst = mix8[*dst + src_sample];
++dst;
++src;
}
#endif
}
break;
case AUDIO_S8:
{
#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */
#if defined(__GNUC__) && defined(__i386__) && defined(SDL_ASSEMBLY_ROUTINES)
if (SDL_HasMMX()) {
SDL_MixAudio_MMX_S8((char *) dst, (char *) src,
(unsigned int) len, (int) volume);
} else
#elif ((defined(_MSC_VER) && defined(_M_IX86)) || defined(__WATCOMC__)) && defined(SDL_ASSEMBLY_ROUTINES)
if (SDL_HasMMX()) {
SDL_MixAudio_MMX_S8_VC((char *) dst, (char *) src,
(unsigned int) len, (int) volume);
} else
#endif
#endif
#if defined(__GNUC__) && defined(__M68000__) && !defined(__mcoldfire__) && defined(SDL_ASSEMBLY_ROUTINES)
SDL_MixAudio_m68k_S8((char *) dst, (char *) src,
(unsigned long) len, (long) volume);
#else
{
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;
}
}
#endif
}
break;
case AUDIO_S16LSB:
{
#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */
#if defined(__GNUC__) && defined(__i386__) && defined(SDL_ASSEMBLY_ROUTINES)
if (SDL_HasMMX()) {
SDL_MixAudio_MMX_S16((char *) dst, (char *) src,
(unsigned int) len, (int) volume);
} else
#elif ((defined(_MSC_VER) && defined(_M_IX86)) || defined(__WATCOMC__)) && defined(SDL_ASSEMBLY_ROUTINES)
if (SDL_HasMMX()) {
SDL_MixAudio_MMX_S16_VC((char *) dst, (char *) src,
(unsigned int) len, (int) volume);
} else
#endif
#endif
#if defined(__GNUC__) && defined(__M68000__) && !defined(__mcoldfire__) && defined(SDL_ASSEMBLY_ROUTINES)
SDL_MixAudio_m68k_S16LSB((short *) dst, (short *) src,
(unsigned long) len, (long) volume);
#else
{
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;
}
}
#endif
}
break;
case AUDIO_S16MSB:
{
#if defined(__GNUC__) && defined(__M68000__) && !defined(__mcoldfire__) && defined(SDL_ASSEMBLY_ROUTINES)
SDL_MixAudio_m68k_S16MSB((short *) dst, (short *) src,
(unsigned long) len, (long) volume);
#else
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;
}
#endif
}
break;
case AUDIO_S32LSB:
{
const Uint32 *src32 = (Uint32 *) src;
Uint32 *dst32 = (Uint32 *) dst;
Sint64 src1, src2;
Sint64 dst_sample;
const Sint64 max_audioval = ((((Sint64) 1) << (32 - 1)) - 1);
const Sint64 min_audioval = -(((Sint64) 1) << (32 - 1));
len /= 4;
while (len--) {
src1 = (Sint64) ((Sint32) SDL_SwapLE32(*src32));
src32++;
ADJUST_VOLUME(src1, volume);
src2 = (Sint64) ((Sint32) SDL_SwapLE32(*dst32));
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapLE32((Uint32) ((Sint32) dst_sample));
}
}
break;
case AUDIO_S32MSB:
{
const Uint32 *src32 = (Uint32 *) src;
Uint32 *dst32 = (Uint32 *) dst;
Sint64 src1, src2;
Sint64 dst_sample;
const Sint64 max_audioval = ((((Sint64) 1) << (32 - 1)) - 1);
const Sint64 min_audioval = -(((Sint64) 1) << (32 - 1));
len /= 4;
while (len--) {
src1 = (Sint64) ((Sint32) SDL_SwapBE32(*src32));
src32++;
ADJUST_VOLUME(src1, volume);
src2 = (Sint64) ((Sint32) SDL_SwapBE32(*dst32));
dst_sample = src1 + src2;
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapBE32((Uint32) ((Sint32) dst_sample));
}
}
break;
case AUDIO_F32LSB:
{
const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
const float fvolume = (float) volume;
const float *src32 = (float *) src;
float *dst32 = (float *) dst;
float src1, src2;
double dst_sample;
/* !!! FIXME: are these right? */
const double max_audioval = 3.402823466e+38F;
const double min_audioval = -3.402823466e+38F;
len /= 4;
while (len--) {
src1 = ((SDL_SwapFloatLE(*src32) * fvolume) * fmaxvolume);
src2 = SDL_SwapFloatLE(*dst32);
src32++;
dst_sample = ((double) src1) + ((double) src2);
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapFloatLE((float) dst_sample);
}
}
break;
case AUDIO_F32MSB:
{
const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
const float fvolume = (float) volume;
const float *src32 = (float *) src;
float *dst32 = (float *) dst;
float src1, src2;
double dst_sample;
/* !!! FIXME: are these right? */
const double max_audioval = 3.402823466e+38F;
const double min_audioval = -3.402823466e+38F;
len /= 4;
while (len--) {
src1 = ((SDL_SwapFloatBE(*src32) * fvolume) * fmaxvolume);
src2 = SDL_SwapFloatBE(*dst32);
src32++;
dst_sample = ((double) src1) + ((double) src2);
if (dst_sample > max_audioval) {
dst_sample = max_audioval;
} else if (dst_sample < min_audioval) {
dst_sample = min_audioval;
}
*(dst32++) = SDL_SwapFloatBE((float) dst_sample);
}
}
break;
default: /* If this happens... FIXME! */
SDL_SetError("SDL_MixAudio(): unknown audio format");
return;
}
}
/* vi: set ts=4 sw=4 expandtab: */

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/*
MMX assembler version of SDL_MixAudio for signed little endian 16 bit samples and signed 8 bit samples
Copyright 2002 Stephane Marchesin (stephane.marchesin@wanadoo.fr)
This code is licensed under the LGPL (see COPYING for details)
Assumes buffer size in bytes is a multiple of 16
Assumes SDL_MIX_MAXVOLUME = 128
*/
/***********************************************
* Mixing for 16 bit signed buffers
***********************************************/
#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */
#if defined(__GNUC__) && defined(__i386__) && defined(SDL_ASSEMBLY_ROUTINES)
void
SDL_MixAudio_MMX_S16(char *dst, char *src, unsigned int size, int volume)
{
__asm__ __volatile__(" movl %3,%%eax\n" /* eax = volume */
" movl %2,%%edx\n" /* edx = size */
" shrl $4,%%edx\n" /* process 16 bytes per iteration = 8 samples */
" jz .endS16\n" " pxor %%mm0,%%mm0\n" " movd %%eax,%%mm0\n" " movq %%mm0,%%mm1\n" " psllq $16,%%mm0\n" " por %%mm1,%%mm0\n" " psllq $16,%%mm0\n" " por %%mm1,%%mm0\n" " psllq $16,%%mm0\n" " por %%mm1,%%mm0\n" /* mm0 = vol|vol|vol|vol */
".align 8\n" " .mixloopS16:\n" " movq (%1),%%mm1\n" /* mm1 = a|b|c|d */
" movq %%mm1,%%mm2\n" /* mm2 = a|b|c|d */
" movq 8(%1),%%mm4\n" /* mm4 = e|f|g|h */
/* pré charger le buffer dst dans mm7 */
" movq (%0),%%mm7\n" /* mm7 = dst[0] */
/* multiplier par le volume */
" pmullw %%mm0,%%mm1\n" /* mm1 = l(a*v)|l(b*v)|l(c*v)|l(d*v) */
" pmulhw %%mm0,%%mm2\n" /* mm2 = h(a*v)|h(b*v)|h(c*v)|h(d*v) */
" movq %%mm4,%%mm5\n" /* mm5 = e|f|g|h */
" pmullw %%mm0,%%mm4\n" /* mm4 = l(e*v)|l(f*v)|l(g*v)|l(h*v) */
" pmulhw %%mm0,%%mm5\n" /* mm5 = h(e*v)|h(f*v)|h(g*v)|h(h*v) */
" movq %%mm1,%%mm3\n" /* mm3 = l(a*v)|l(b*v)|l(c*v)|l(d*v) */
" punpckhwd %%mm2,%%mm1\n" /* mm1 = a*v|b*v */
" movq %%mm4,%%mm6\n" /* mm6 = l(e*v)|l(f*v)|l(g*v)|l(h*v) */
" punpcklwd %%mm2,%%mm3\n" /* mm3 = c*v|d*v */
" punpckhwd %%mm5,%%mm4\n" /* mm4 = e*f|f*v */
" punpcklwd %%mm5,%%mm6\n" /* mm6 = g*v|h*v */
/* pré charger le buffer dst dans mm5 */
" movq 8(%0),%%mm5\n" /* mm5 = dst[1] */
/* diviser par 128 */
" psrad $7,%%mm1\n" /* mm1 = a*v/128|b*v/128 , 128 = SDL_MIX_MAXVOLUME */
" add $16,%1\n" " psrad $7,%%mm3\n" /* mm3 = c*v/128|d*v/128 */
" psrad $7,%%mm4\n" /* mm4 = e*v/128|f*v/128 */
/* mm1 = le sample avec le volume modifié */
" packssdw %%mm1,%%mm3\n" /* mm3 = s(a*v|b*v|c*v|d*v) */
" psrad $7,%%mm6\n" /* mm6= g*v/128|h*v/128 */
" paddsw %%mm7,%%mm3\n" /* mm3 = adjust_volume(src)+dst */
/* mm4 = le sample avec le volume modifié */
" packssdw %%mm4,%%mm6\n" /* mm6 = s(e*v|f*v|g*v|h*v) */
" movq %%mm3,(%0)\n" " paddsw %%mm5,%%mm6\n" /* mm6 = adjust_volume(src)+dst */
" movq %%mm6,8(%0)\n"
" add $16,%0\n"
" dec %%edx\n"
" jnz .mixloopS16\n"
" emms\n"
".endS16:\n"::"r"(dst), "r"(src),
"m"(size), "m"(volume):"eax", "edx", "memory");
}
/*////////////////////////////////////////////// */
/* Mixing for 8 bit signed buffers */
/*////////////////////////////////////////////// */
void
SDL_MixAudio_MMX_S8(char *dst, char *src, unsigned int size, int volume)
{
__asm__ __volatile__(" movl %3,%%eax\n" /* eax = volume */
" movd %%eax,%%mm0\n" " movq %%mm0,%%mm1\n" " psllq $16,%%mm0\n" " por %%mm1,%%mm0\n" " psllq $16,%%mm0\n" " por %%mm1,%%mm0\n" " psllq $16,%%mm0\n" " por %%mm1,%%mm0\n" " movl %2,%%edx\n" /* edx = size */
" shr $3,%%edx\n" /* process 8 bytes per iteration = 8 samples */
" cmp $0,%%edx\n" " je .endS8\n" ".align 8\n" " .mixloopS8:\n" " pxor %%mm2,%%mm2\n" /* mm2 = 0 */
" movq (%1),%%mm1\n" /* mm1 = a|b|c|d|e|f|g|h */
" movq %%mm1,%%mm3\n" /* mm3 = a|b|c|d|e|f|g|h */
/* on va faire le "sign extension" en faisant un cmp avec 0 qui retourne 1 si <0, 0 si >0 */
" pcmpgtb %%mm1,%%mm2\n" /* mm2 = 11111111|00000000|00000000.... */
" punpckhbw %%mm2,%%mm1\n" /* mm1 = 0|a|0|b|0|c|0|d */
" punpcklbw %%mm2,%%mm3\n" /* mm3 = 0|e|0|f|0|g|0|h */
" movq (%0),%%mm2\n" /* mm2 = destination */
" pmullw %%mm0,%%mm1\n" /* mm1 = v*a|v*b|v*c|v*d */
" add $8,%1\n" " pmullw %%mm0,%%mm3\n" /* mm3 = v*e|v*f|v*g|v*h */
" psraw $7,%%mm1\n" /* mm1 = v*a/128|v*b/128|v*c/128|v*d/128 */
" psraw $7,%%mm3\n" /* mm3 = v*e/128|v*f/128|v*g/128|v*h/128 */
" packsswb %%mm1,%%mm3\n" /* mm1 = v*a/128|v*b/128|v*c/128|v*d/128|v*e/128|v*f/128|v*g/128|v*h/128 */
" paddsb %%mm2,%%mm3\n" /* add to destination buffer */
" movq %%mm3,(%0)\n" /* store back to ram */
" add $8,%0\n"
" dec %%edx\n"
" jnz .mixloopS8\n"
".endS8:\n"
" emms\n"::"r"(dst), "r"(src), "m"(size),
"m"(volume):"eax", "edx", "memory");
}
#endif
#endif /* SDL_BUGGY_MMX_MIXERS */
/* vi: set ts=4 sw=4 expandtab: */

17
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/*
headers for MMX assembler version of SDL_MixAudio
Copyright 2002 Stephane Marchesin (stephane.marchesin@wanadoo.fr)
This code is licensed under the LGPL (see COPYING for details)
Assumes buffer size in bytes is a multiple of 16
Assumes SDL_MIX_MAXVOLUME = 128
*/
#include "SDL_config.h"
#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */
#if defined(__GNUC__) && defined(__i386__) && defined(SDL_ASSEMBLY_ROUTINES)
void SDL_MixAudio_MMX_S16(char *, char *, unsigned int, int);
void SDL_MixAudio_MMX_S8(char *, char *, unsigned int, int);
#endif
#endif /* SDL_BUGGY_MMX_MIXERS */
/* vi: set ts=4 sw=4 expandtab: */

190
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#include "SDL_mixer_MMX_VC.h"
#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */
#if ((defined(_MSC_VER) && defined(_M_IX86)) || defined(__WATCOMC__)) && defined(SDL_ASSEMBLY_ROUTINES)
// MMX assembler version of SDL_MixAudio for signed little endian 16 bit samples and signed 8 bit samples
// Copyright 2002 Stephane Marchesin (stephane.marchesin@wanadoo.fr)
// Converted to Intel ASM notation by Cth
// This code is licensed under the LGPL (see COPYING for details)
//
// Assumes buffer size in bytes is a multiple of 16
// Assumes SDL_MIX_MAXVOLUME = 128
////////////////////////////////////////////////
// Mixing for 16 bit signed buffers
////////////////////////////////////////////////
void
SDL_MixAudio_MMX_S16_VC(char *dst, char *src, unsigned int nSize, int volume)
{
/* *INDENT-OFF* */
__asm
{
push edi
push esi
push ebx
mov edi, dst // edi = dst
mov esi, src // esi = src
mov eax, volume // eax = volume
mov ebx, nSize // ebx = size
shr ebx, 4 // process 16 bytes per iteration = 8 samples
jz endS16
pxor mm0, mm0
movd mm0, eax //%%eax,%%mm0
movq mm1, mm0 //%%mm0,%%mm1
psllq mm0, 16 //$16,%%mm0
por mm0, mm1 //%%mm1,%%mm0
psllq mm0, 16 //$16,%%mm0
por mm0, mm1 //%%mm1,%%mm0
psllq mm0, 16 //$16,%%mm0
por mm0, mm1 //%%mm1,%%mm0 // mm0 = vol|vol|vol|vol
#ifndef __WATCOMC__
align 16
#endif
mixloopS16:
movq mm1, [esi] //(%%esi),%%mm1\n" // mm1 = a|b|c|d
movq mm2, mm1 //%%mm1,%%mm2\n" // mm2 = a|b|c|d
movq mm4, [esi + 8] //8(%%esi),%%mm4\n" // mm4 = e|f|g|h
// pre charger le buffer dst dans mm7
movq mm7, [edi] //(%%edi),%%mm7\n" // mm7 = dst[0]"
// multiplier par le volume
pmullw mm1, mm0 //%%mm0,%%mm1\n" // mm1 = l(a*v)|l(b*v)|l(c*v)|l(d*v)
pmulhw mm2, mm0 //%%mm0,%%mm2\n" // mm2 = h(a*v)|h(b*v)|h(c*v)|h(d*v)
movq mm5, mm4 //%%mm4,%%mm5\n" // mm5 = e|f|g|h
pmullw mm4, mm0 //%%mm0,%%mm4\n" // mm4 = l(e*v)|l(f*v)|l(g*v)|l(h*v)
pmulhw mm5, mm0 //%%mm0,%%mm5\n" // mm5 = h(e*v)|h(f*v)|h(g*v)|h(h*v)
movq mm3, mm1 //%%mm1,%%mm3\n" // mm3 = l(a*v)|l(b*v)|l(c*v)|l(d*v)
punpckhwd mm1, mm2 //%%mm2,%%mm1\n" // mm1 = a*v|b*v
movq mm6, mm4 //%%mm4,%%mm6\n" // mm6 = l(e*v)|l(f*v)|l(g*v)|l(h*v)
punpcklwd mm3, mm2 //%%mm2,%%mm3\n" // mm3 = c*v|d*v
punpckhwd mm4, mm5 //%%mm5,%%mm4\n" // mm4 = e*f|f*v
punpcklwd mm6, mm5 //%%mm5,%%mm6\n" // mm6 = g*v|h*v
// pre charger le buffer dst dans mm5
movq mm5, [edi + 8] //8(%%edi),%%mm5\n" // mm5 = dst[1]
// diviser par 128
psrad mm1, 7 //$7,%%mm1\n" // mm1 = a*v/128|b*v/128 , 128 = SDL_MIX_MAXVOLUME
add esi, 16 //$16,%%esi\n"
psrad mm3, 7 //$7,%%mm3\n" // mm3 = c*v/128|d*v/128
psrad mm4, 7 //$7,%%mm4\n" // mm4 = e*v/128|f*v/128
// mm1 = le sample avec le volume modifie
packssdw mm3, mm1 //%%mm1,%%mm3\n" // mm3 = s(a*v|b*v|c*v|d*v)
psrad mm6, 7 //$7,%%mm6\n" // mm6= g*v/128|h*v/128
paddsw mm3, mm7 //%%mm7,%%mm3\n" // mm3 = adjust_volume(src)+dst
// mm4 = le sample avec le volume modifie
packssdw mm6, mm4 //%%mm4,%%mm6\n" // mm6 = s(e*v|f*v|g*v|h*v)
movq [edi], mm3 //%%mm3,(%%edi)\n"
paddsw mm6, mm5 //%%mm5,%%mm6\n" // mm6 = adjust_volume(src)+dst
movq [edi + 8], mm6 //%%mm6,8(%%edi)\n"
add edi, 16 //$16,%%edi\n"
dec ebx //%%ebx\n"
jnz mixloopS16
endS16:
emms
pop ebx
pop esi
pop edi
}
/* *INDENT-ON* */
}
////////////////////////////////////////////////
// Mixing for 8 bit signed buffers
////////////////////////////////////////////////
void
SDL_MixAudio_MMX_S8_VC(char *dst, char *src, unsigned int nSize, int volume)
{
/* *INDENT-OFF* */
_asm
{
push edi
push esi
push ebx
mov edi, dst //movl %0,%%edi // edi = dst
mov esi, src //%1,%%esi // esi = src
mov eax, volume //%3,%%eax // eax = volume
movd mm0, eax //%%eax,%%mm0
movq mm1, mm0 //%%mm0,%%mm1
psllq mm0, 16 //$16,%%mm0
por mm0, mm1 //%%mm1,%%mm0
psllq mm0, 16 //$16,%%mm0
por mm0, mm1 //%%mm1,%%mm0
psllq mm0, 16 //$16,%%mm0
por mm0, mm1 //%%mm1,%%mm0
mov ebx, nSize //%2,%%ebx // ebx = size
shr ebx, 3 //$3,%%ebx // process 8 bytes per iteration = 8 samples
cmp ebx, 0 //$0,%%ebx
je endS8
#ifndef __WATCOMC__
align 16
#endif
mixloopS8:
pxor mm2, mm2 //%%mm2,%%mm2 // mm2 = 0
movq mm1, [esi] //(%%esi),%%mm1 // mm1 = a|b|c|d|e|f|g|h
movq mm3, mm1 //%%mm1,%%mm3 // mm3 = a|b|c|d|e|f|g|h
// on va faire le "sign extension" en faisant un cmp avec 0 qui retourne 1 si <0, 0 si >0
pcmpgtb mm2, mm1 //%%mm1,%%mm2 // mm2 = 11111111|00000000|00000000....
punpckhbw mm1, mm2 //%%mm2,%%mm1 // mm1 = 0|a|0|b|0|c|0|d
punpcklbw mm3, mm2 //%%mm2,%%mm3 // mm3 = 0|e|0|f|0|g|0|h
movq mm2, [edi] //(%%edi),%%mm2 // mm2 = destination
pmullw mm1, mm0 //%%mm0,%%mm1 // mm1 = v*a|v*b|v*c|v*d
add esi, 8 //$8,%%esi
pmullw mm3, mm0 //%%mm0,%%mm3 // mm3 = v*e|v*f|v*g|v*h
psraw mm1, 7 //$7,%%mm1 // mm1 = v*a/128|v*b/128|v*c/128|v*d/128
psraw mm3, 7 //$7,%%mm3 // mm3 = v*e/128|v*f/128|v*g/128|v*h/128
packsswb mm3, mm1 //%%mm1,%%mm3 // mm1 = v*a/128|v*b/128|v*c/128|v*d/128|v*e/128|v*f/128|v*g/128|v*h/128
paddsb mm3, mm2 //%%mm2,%%mm3 // add to destination buffer
movq [edi], mm3 //%%mm3,(%%edi) // store back to ram
add edi, 8 //$8,%%edi
dec ebx //%%ebx
jnz mixloopS8
endS8:
emms
pop ebx
pop esi
pop edi
}
/* *INDENT-ON* */
}
#endif /* SDL_ASSEMBLY_ROUTINES */
#endif /* SDL_BUGGY_MMX_MIXERS */
/* vi: set ts=4 sw=4 expandtab: */

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */
#if ((defined(_MSC_VER) && defined(_M_IX86)) || defined(__WATCOMC__)) && defined(SDL_ASSEMBLY_ROUTINES)
/* headers for MMX assembler version of SDL_MixAudio
Copyright 2002 Stephane Marchesin (stephane.marchesin@wanadoo.fr)
Converted to Intel ASM notation by Cth
This code is licensed under the LGPL (see COPYING for details)
Assumes buffer size in bytes is a multiple of 16
Assumes SDL_MIX_MAXVOLUME = 128
*/
void SDL_MixAudio_MMX_S16_VC(char *, char *, unsigned int, int);
void SDL_MixAudio_MMX_S8_VC(char *, char *, unsigned int, int);
#endif
#endif /* SDL_BUGGY_MMX_MIXERS */
/* vi: set ts=4 sw=4 expandtab: */

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/*
m68k assembly mix routines
Patrice Mandin
*/
#if defined(__M68000__) && !defined(__mcoldfire__) && defined(__GNUC__)
void
SDL_MixAudio_m68k_U8(char *dst, char *src, long len, long volume, char *mix8)
{
__asm__ __volatile__("tstl %2\n" " beqs stoploop_u8\n" "mixloop_u8:\n"
/* Mix a sample */
" moveq #0,%%d0\n" " moveq #0,%%d1\n" " moveb %1@+,%%d0\n" /* d0 = *src++ */
" sub #128,%%d0\n" /* d0 -= 128 */
" muls %3,%%d0\n" /* d0 *= volume (0<=volume<=128) */
" moveb %0@,%%d1\n" /* d1 = *dst */
" asr #7,%%d0\n" /* d0 /= 128 (SDL_MIX_MAXVOLUME) */
" add #128,%%d0\n" /* d0 += 128 */
" add %%d1,%%d0\n"
" moveb %4@(%%d0:w),%0@+\n"
/* Loop till done */
" subql #1,%2\n" " bhis mixloop_u8\n" "stoploop_u8:\n": /* no return value */
: /* input */
"a"(dst), "a"(src), "d"(len), "d"(volume), "a"(mix8): /* clobbered registers */
"d0", "d1", "cc", "memory");
}
void
SDL_MixAudio_m68k_S8(char *dst, char *src, long len, long volume)
{
__asm__ __volatile__("tstl %2\n"
" beqs stoploop_s8\n"
" moveq #-128,%%d2\n"
" moveq #127,%%d3\n" "mixloop_s8:\n"
/* Mix a sample */
" moveq #0,%%d0\n" " moveq #0,%%d1\n" " moveb %1@+,%%d0\n" /* d0 = *src++ */
" muls %3,%%d0\n" /* d0 *= volume (0<=volume<=128) */
" moveb %0@,%%d1\n" /* d1 = *dst */
" asr #7,%%d0\n" /* d0 /= 128 (SDL_MIX_MAXVOLUME) */
" add %%d1,%%d0\n"
" cmp %%d2,%%d0\n"
" bges lower_limit_s8\n"
" move %%d2,%%d0\n"
"lower_limit_s8:\n"
" cmp %%d3,%%d0\n"
" bles upper_limit_s8\n"
" move %%d3,%%d0\n"
"upper_limit_s8:\n" " moveb %%d0,%0@+\n"
/* Loop till done */
" subql #1,%2\n" " bhis mixloop_s8\n" "stoploop_s8:\n": /* no return value */
: /* input */
"a"(dst), "a"(src), "d"(len), "d"(volume): /* clobbered registers */
"d0", "d1", "d2", "d3", "cc", "memory");
}
void
SDL_MixAudio_m68k_S16MSB(short *dst, short *src, long len, long volume)
{
__asm__ __volatile__("tstl %2\n"
" beqs stoploop_s16msb\n"
" movel #-32768,%%d2\n"
" movel #32767,%%d3\n"
" lsrl #1,%2\n" "mixloop_s16msb:\n"
/* Mix a sample */
" move %1@+,%%d0\n" /* d0 = *src++ */
" muls %3,%%d0\n" /* d0 *= volume (0<=volume<=128) */
" move %0@,%%d1\n" /* d1 = *dst */
" extl %%d1\n" /* extend d1 to 32 bits */
" asrl #7,%%d0\n" /* d0 /= 128 (SDL_MIX_MAXVOLUME) */
" addl %%d1,%%d0\n"
" cmpl %%d2,%%d0\n"
" bges lower_limit_s16msb\n"
" move %%d2,%%d0\n"
"lower_limit_s16msb:\n"
" cmpl %%d3,%%d0\n"
" bles upper_limit_s16msb\n"
" move %%d3,%%d0\n"
"upper_limit_s16msb:\n" " move %%d0,%0@+\n"
/* Loop till done */
" subql #1,%2\n" " bhis mixloop_s16msb\n" "stoploop_s16msb:\n": /* no return value */
: /* input */
"a"(dst), "a"(src), "d"(len), "d"(volume): /* clobbered registers */
"d0", "d1", "d2", "d3", "cc", "memory");
}
void
SDL_MixAudio_m68k_S16LSB(short *dst, short *src, long len, long volume)
{
__asm__ __volatile__("tstl %2\n"
" beqs stoploop_s16lsb\n"
" movel #-32768,%%d2\n"
" movel #32767,%%d3\n"
" lsrl #1,%2\n" "mixloop_s16lsb:\n"
/* Mix a sample */
" move %1@+,%%d0\n" /* d0 = *src++ */
" rorw #8,%%d0\n" " muls %3,%%d0\n" /* d0 *= volume (0<=volume<=128) */
" move %0@,%%d1\n" /* d1 = *dst */
" rorw #8,%%d1\n" " extl %%d1\n" /* extend d1 to 32 bits */
" asrl #7,%%d0\n" /* d0 /= 128 (SDL_MIX_MAXVOLUME) */
" addl %%d1,%%d0\n"
" cmpl %%d2,%%d0\n"
" bges lower_limit_s16lsb\n"
" move %%d2,%%d0\n"
"lower_limit_s16lsb:\n"
" cmpl %%d3,%%d0\n"
" bles upper_limit_s16lsb\n"
" move %%d3,%%d0\n"
"upper_limit_s16lsb:\n"
" rorw #8,%%d0\n" " move %%d0,%0@+\n"
/* Loop till done */
" subql #1,%2\n" " bhis mixloop_s16lsb\n" "stoploop_s16lsb:\n": /* no return value */
: /* input */
"a"(dst), "a"(src), "d"(len), "d"(volume): /* clobbered registers */
"d0", "d1", "d2", "d3", "cc", "memory");
}
#endif
/* vi: set ts=4 sw=4 expandtab: */

38
project/jni/sdl-1.3/src/audio/SDL_mixer_m68k.h Normal file
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@@ -0,0 +1,38 @@
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/*
m68k assembly mix routines
Patrice Mandin
*/
#if defined(__M68000__) && defined(__GNUC__)
void SDL_MixAudio_m68k_U8(char *dst, char *src, long len, long volume,
char *mix8);
void SDL_MixAudio_m68k_S8(char *dst, char *src, long len, long volume);
void SDL_MixAudio_m68k_S16MSB(short *dst, short *src, long len, long volume);
void SDL_MixAudio_m68k_S16LSB(short *dst, short *src, long len, long volume);
#endif
/* vi: set ts=4 sw=4 expandtab: */

129
project/jni/sdl-1.3/src/audio/SDL_sysaudio.h Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is SDL_free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_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 _THIS SDL_AudioDevice *_this
typedef struct SDL_AudioDriverImpl
{
int (*DetectDevices) (int iscapture);
const char *(*GetDeviceName) (int index, int iscapture);
int (*OpenDevice) (_THIS, const char *devname, int iscapture);
void (*ThreadInit) (_THIS); /* Called by audio thread at start */
void (*WaitDevice) (_THIS);
void (*PlayDevice) (_THIS);
Uint8 *(*GetDeviceBuf) (_THIS);
void (*WaitDone) (_THIS);
void (*CloseDevice) (_THIS);
void (*LockDevice) (_THIS);
void (*UnlockDevice) (_THIS);
void (*Deinitialize) (void);
/* Some flags to push duplicate code into the core and reduce #ifdefs. */
int ProvidesOwnCallbackThread:1;
int SkipMixerLock:1;
int HasCaptureSupport:1;
int OnlyHasDefaultOutputDevice:1;
int OnlyHasDefaultInputDevice:1;
} SDL_AudioDriverImpl;
typedef struct SDL_AudioDriver
{
/* * * */
/* The name of this audio driver */
const char *name;
/* * * */
/* The description of this audio driver */
const char *desc;
SDL_AudioDriverImpl impl;
} SDL_AudioDriver;
/* Streamer */
typedef struct
{
Uint8 *buffer;
int max_len; /* the maximum length in bytes */
int read_pos, write_pos; /* the position of the write and read heads in bytes */
} SDL_AudioStreamer;
/* Define the SDL audio driver structure */
struct SDL_AudioDevice
{
/* * * */
/* 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;
/* The streamer, if sample rate conversion necessitates it */
int use_streamer;
SDL_AudioStreamer streamer;
/* Current state flags */
int iscapture;
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;
SDL_threadID threadid;
/* * * */
/* Data private to this driver */
struct SDL_PrivateAudioData *hidden;
};
#undef _THIS
typedef struct AudioBootStrap
{
const char *name;
const char *desc;
int (*init) (SDL_AudioDriverImpl * impl);
int demand_only:1; /* 1==request explicitly, or it won't be available. */
} AudioBootStrap;
#endif /* _SDL_sysaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

636
project/jni/sdl-1.3/src/audio/SDL_wave.c Normal file
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@@ -0,0 +1,636 @@
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* Microsoft WAVE file loading routines */
#include "SDL_audio.h"
#include "SDL_wave.h"
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 = (Uint16) delta;
state->iSamp2 = state->iSamp1;
state->iSamp1 = (Sint16) 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 *) SDL_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;
}
SDL_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;
unsigned int c, channels;
/* Check to make sure we have enough variables in the state array */
channels = IMA_ADPCM_state.wavefmt.channels;
if (channels > SDL_arraysize(IMA_ADPCM_state.state)) {
SDL_SetError("IMA ADPCM decoder can only handle %d channels",
SDL_arraysize(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 *) SDL_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] = (Uint8) (state[c].sample & 0xFF);
decoded[1] = (Uint8) (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;
}
SDL_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 IEEE_float_encoded, MS_ADPCM_encoded, IMA_ADPCM_encoded;
int samplesize;
/* WAV magic header */
Uint32 RIFFchunk;
Uint32 wavelen = 0;
Uint32 WAVEmagic;
Uint32 headerDiff = 0;
/* 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);
if (wavelen == WAVE) { /* The RIFFchunk has already been read */
WAVEmagic = wavelen;
wavelen = RIFFchunk;
RIFFchunk = RIFF;
} else {
WAVEmagic = SDL_ReadLE32(src);
}
if ((RIFFchunk != RIFF) || (WAVEmagic != WAVE)) {
SDL_SetError("Unrecognized file type (not WAVE)");
was_error = 1;
goto done;
}
headerDiff += sizeof(Uint32); /* for WAVE */
/* Read the audio data format chunk */
chunk.data = NULL;
do {
if (chunk.data != NULL) {
SDL_free(chunk.data);
chunk.data = NULL;
}
lenread = ReadChunk(src, &chunk);
if (lenread < 0) {
was_error = 1;
goto done;
}
/* 2 Uint32's for chunk header+len, plus the lenread */
headerDiff += lenread + 2 * sizeof(Uint32);
} 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;
}
IEEE_float_encoded = MS_ADPCM_encoded = IMA_ADPCM_encoded = 0;
switch (SDL_SwapLE16(format->encoding)) {
case PCM_CODE:
/* We can understand this */
break;
case IEEE_FLOAT_CODE:
IEEE_float_encoded = 1;
/* 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;
case MP3_CODE:
SDL_SetError("MPEG Layer 3 data not supported",
SDL_SwapLE16(format->encoding));
was_error = 1;
goto done;
default:
SDL_SetError("Unknown WAVE data format: 0x%.4x",
SDL_SwapLE16(format->encoding));
was_error = 1;
goto done;
}
SDL_memset(spec, 0, (sizeof *spec));
spec->freq = SDL_SwapLE32(format->frequency);
if (IEEE_float_encoded) {
if ((SDL_SwapLE16(format->bitspersample)) != 32) {
was_error = 1;
} else {
spec->format = AUDIO_F32;
}
} else {
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;
case 32:
spec->format = AUDIO_S32;
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) {
SDL_free(*audio_buf);
*audio_buf = NULL;
}
lenread = ReadChunk(src, &chunk);
if (lenread < 0) {
was_error = 1;
goto done;
}
*audio_len = lenread;
*audio_buf = chunk.data;
if (chunk.magic != DATA)
headerDiff += lenread + 2 * sizeof(Uint32);
} while (chunk.magic != DATA);
headerDiff += 2 * sizeof(Uint32); /* for the data chunk and len */
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 = ((SDL_AUDIO_BITSIZE(spec->format)) / 8) * spec->channels;
*audio_len &= ~(samplesize - 1);
done:
if (format != NULL) {
SDL_free(format);
}
if (src) {
if (freesrc) {
SDL_RWclose(src);
} else {
/* seek to the end of the file (given by the RIFF chunk) */
SDL_RWseek(src, wavelen - chunk.length - headerDiff, RW_SEEK_CUR);
}
}
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) {
SDL_free(audio_buf);
}
}
static int
ReadChunk(SDL_RWops * src, Chunk * chunk)
{
chunk->magic = SDL_ReadLE32(src);
chunk->length = SDL_ReadLE32(src);
chunk->data = (Uint8 *) SDL_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);
SDL_free(chunk->data);
chunk->data = NULL;
return (-1);
}
return (chunk->length);
}
/* vi: set ts=4 sw=4 expandtab: */

65
project/jni/sdl-1.3/src/audio/SDL_wave.h Normal file
View File

@@ -0,0 +1,65 @@
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is SDL_free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* 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 IEEE_FLOAT_CODE 0x0003
#define IMA_ADPCM_CODE 0x0011
#define MP3_CODE 0x0055
#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;
} Chunk;
/* vi: set ts=4 sw=4 expandtab: */

691
project/jni/sdl-1.3/src/audio/alsa/SDL_alsa_audio.c Normal file
View File

@@ -0,0 +1,691 @@
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* Allow access to a raw mixing buffer */
#include <sys/types.h>
#include <signal.h> /* For kill() */
#include <errno.h>
#include <string.h>
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_alsa_audio.h"
#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
#include "SDL_loadso.h"
#endif
/* The tag name used by ALSA audio */
#define DRIVER_NAME "alsa"
static int (*ALSA_snd_pcm_open)
(snd_pcm_t **, const char *, snd_pcm_stream_t, int);
static int (*ALSA_snd_pcm_close) (snd_pcm_t * pcm);
static snd_pcm_sframes_t(*ALSA_snd_pcm_writei)
(snd_pcm_t *, const void *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_recover) (snd_pcm_t *, int, int);
static int (*ALSA_snd_pcm_prepare) (snd_pcm_t *);
static int (*ALSA_snd_pcm_drain) (snd_pcm_t *);
static const char *(*ALSA_snd_strerror) (int);
static size_t(*ALSA_snd_pcm_hw_params_sizeof) (void);
static size_t(*ALSA_snd_pcm_sw_params_sizeof) (void);
static void (*ALSA_snd_pcm_hw_params_copy)
(snd_pcm_hw_params_t *, const snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_hw_params_any) (snd_pcm_t *, snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_hw_params_set_access)
(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_access_t);
static int (*ALSA_snd_pcm_hw_params_set_format)
(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_format_t);
static int (*ALSA_snd_pcm_hw_params_set_channels)
(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int);
static int (*ALSA_snd_pcm_hw_params_get_channels)
(const snd_pcm_hw_params_t *, unsigned int *);
static int (*ALSA_snd_pcm_hw_params_set_rate_near)
(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
static int (*ALSA_snd_pcm_hw_params_set_period_size_near)
(snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *);
static int (*ALSA_snd_pcm_hw_params_get_period_size)
(const snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *);
static int (*ALSA_snd_pcm_hw_params_set_periods_near)
(snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *);
static int (*ALSA_snd_pcm_hw_params_get_periods)
(const snd_pcm_hw_params_t *, unsigned int *, int *);
static int (*ALSA_snd_pcm_hw_params_set_buffer_size_near)
(snd_pcm_t *pcm, snd_pcm_hw_params_t *, snd_pcm_uframes_t *);
static int (*ALSA_snd_pcm_hw_params_get_buffer_size)
(const snd_pcm_hw_params_t *, snd_pcm_uframes_t *);
static int (*ALSA_snd_pcm_hw_params) (snd_pcm_t *, snd_pcm_hw_params_t *);
static int (*ALSA_snd_pcm_sw_params_current) (snd_pcm_t *,
snd_pcm_sw_params_t *);
static int (*ALSA_snd_pcm_sw_params_set_start_threshold)
(snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t);
static int (*ALSA_snd_pcm_sw_params) (snd_pcm_t *, snd_pcm_sw_params_t *);
static int (*ALSA_snd_pcm_nonblock) (snd_pcm_t *, int);
static int (*ALSA_snd_pcm_wait)(snd_pcm_t *, int);
#define snd_pcm_hw_params_sizeof ALSA_snd_pcm_hw_params_sizeof
#define snd_pcm_sw_params_sizeof ALSA_snd_pcm_sw_params_sizeof
#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC;
static void *alsa_handle = NULL;
static int
load_alsa_sym(const char *fn, void **addr)
{
*addr = SDL_LoadFunction(alsa_handle, fn);
if (*addr == NULL) {
/* Don't call SDL_SetError(): SDL_LoadFunction already did. */
return 0;
}
return 1;
}
/* cast funcs to char* first, to please GCC's strict aliasing rules. */
#define SDL_ALSA_SYM(x) \
if (!load_alsa_sym(#x, (void **) (char *) &ALSA_##x)) return -1
#else
#define SDL_ALSA_SYM(x) ALSA_##x = x
#endif
static int
load_alsa_syms(void)
{
SDL_ALSA_SYM(snd_pcm_open);
SDL_ALSA_SYM(snd_pcm_close);
SDL_ALSA_SYM(snd_pcm_writei);
SDL_ALSA_SYM(snd_pcm_recover);
SDL_ALSA_SYM(snd_pcm_prepare);
SDL_ALSA_SYM(snd_pcm_drain);
SDL_ALSA_SYM(snd_strerror);
SDL_ALSA_SYM(snd_pcm_hw_params_sizeof);
SDL_ALSA_SYM(snd_pcm_sw_params_sizeof);
SDL_ALSA_SYM(snd_pcm_hw_params_copy);
SDL_ALSA_SYM(snd_pcm_hw_params_any);
SDL_ALSA_SYM(snd_pcm_hw_params_set_access);
SDL_ALSA_SYM(snd_pcm_hw_params_set_format);
SDL_ALSA_SYM(snd_pcm_hw_params_set_channels);
SDL_ALSA_SYM(snd_pcm_hw_params_get_channels);
SDL_ALSA_SYM(snd_pcm_hw_params_set_rate_near);
SDL_ALSA_SYM(snd_pcm_hw_params_set_period_size_near);
SDL_ALSA_SYM(snd_pcm_hw_params_get_period_size);
SDL_ALSA_SYM(snd_pcm_hw_params_set_periods_near);
SDL_ALSA_SYM(snd_pcm_hw_params_get_periods);
SDL_ALSA_SYM(snd_pcm_hw_params_set_buffer_size_near);
SDL_ALSA_SYM(snd_pcm_hw_params_get_buffer_size);
SDL_ALSA_SYM(snd_pcm_hw_params);
SDL_ALSA_SYM(snd_pcm_sw_params_current);
SDL_ALSA_SYM(snd_pcm_sw_params_set_start_threshold);
SDL_ALSA_SYM(snd_pcm_sw_params);
SDL_ALSA_SYM(snd_pcm_nonblock);
SDL_ALSA_SYM(snd_pcm_wait);
return 0;
}
#undef SDL_ALSA_SYM
#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
static void
UnloadALSALibrary(void)
{
if (alsa_handle != NULL) {
SDL_UnloadObject(alsa_handle);
alsa_handle = NULL;
}
}
static int
LoadALSALibrary(void)
{
int retval = 0;
if (alsa_handle == NULL) {
alsa_handle = SDL_LoadObject(alsa_library);
if (alsa_handle == NULL) {
retval = -1;
/* Don't call SDL_SetError(): SDL_LoadObject already did. */
} else {
retval = load_alsa_syms();
if (retval < 0) {
UnloadALSALibrary();
}
}
}
return retval;
}
#else
static void
UnloadALSALibrary(void)
{
}
static int
LoadALSALibrary(void)
{
load_alsa_syms();
return 0;
}
#endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */
static const char *
get_audio_device(int channels)
{
const char *device;
device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */
if (device == NULL) {
switch (channels) {
case 6:
device = "plug:surround51";
break;
case 4:
device = "plug:surround40";
break;
default:
device = "default";
break;
}
}
return device;
}
/* This function waits until it is possible to write a full sound buffer */
static void
ALSA_WaitDevice(_THIS)
{
/* We're in blocking mode, so there's nothing to do here */
}
/* !!! FIXME: is there a channel swizzler in alsalib instead? */
/*
* http://bugzilla.libsdl.org/show_bug.cgi?id=110
* "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
* and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
*/
#define SWIZ6(T) \
T *ptr = (T *) this->hidden->mixbuf; \
Uint32 i; \
for (i = 0; i < this->spec.samples; i++, ptr += 6) { \
T tmp; \
tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \
tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \
}
static __inline__ void
swizzle_alsa_channels_6_64bit(_THIS)
{
SWIZ6(Uint64);
}
static __inline__ void
swizzle_alsa_channels_6_32bit(_THIS)
{
SWIZ6(Uint32);
}
static __inline__ void
swizzle_alsa_channels_6_16bit(_THIS)
{
SWIZ6(Uint16);
}
static __inline__ void
swizzle_alsa_channels_6_8bit(_THIS)
{
SWIZ6(Uint8);
}
#undef SWIZ6
/*
* Called right before feeding this->hidden->mixbuf to the hardware. Swizzle
* channels from Windows/Mac order to the format alsalib will want.
*/
static __inline__ void
swizzle_alsa_channels(_THIS)
{
if (this->spec.channels == 6) {
const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */
if (fmtsize == 16)
swizzle_alsa_channels_6_16bit(this);
else if (fmtsize == 8)
swizzle_alsa_channels_6_8bit(this);
else if (fmtsize == 32)
swizzle_alsa_channels_6_32bit(this);
else if (fmtsize == 64)
swizzle_alsa_channels_6_64bit(this);
}
/* !!! FIXME: update this for 7.1 if needed, later. */
}
static void
ALSA_PlayDevice(_THIS)
{
int status;
const Uint8 *sample_buf = (const Uint8 *) this->hidden->mixbuf;
const int frame_size = (((int) (this->spec.format & 0xFF)) / 8) *
this->spec.channels;
snd_pcm_uframes_t frames_left = ((snd_pcm_uframes_t) this->spec.samples);
swizzle_alsa_channels(this);
while ( frames_left > 0 && this->enabled ) {
/* !!! FIXME: This works, but needs more testing before going live */
/*ALSA_snd_pcm_wait(this->hidden->pcm_handle, -1);*/
status = ALSA_snd_pcm_writei(this->hidden->pcm_handle,
sample_buf, frames_left);
if (status < 0) {
if (status == -EAGAIN) {
/* Apparently snd_pcm_recover() doesn't handle this case -
does it assume snd_pcm_wait() above? */
SDL_Delay(1);
continue;
}
status = ALSA_snd_pcm_recover(this->hidden->pcm_handle, status, 0);
if (status < 0) {
/* Hmm, not much we can do - abort */
fprintf(stderr, "ALSA write failed (unrecoverable): %s\n",
ALSA_snd_strerror(status));
this->enabled = 0;
return;
}
continue;
}
sample_buf += status * frame_size;
frames_left -= status;
}
}
static Uint8 *
ALSA_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
}
static void
ALSA_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
if (this->hidden->pcm_handle) {
ALSA_snd_pcm_drain(this->hidden->pcm_handle);
ALSA_snd_pcm_close(this->hidden->pcm_handle);
this->hidden->pcm_handle = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
static int
ALSA_finalize_hardware(_THIS, snd_pcm_hw_params_t *hwparams, int override)
{
int status;
snd_pcm_uframes_t bufsize;
/* "set" the hardware with the desired parameters */
status = ALSA_snd_pcm_hw_params(this->hidden->pcm_handle, hwparams);
if ( status < 0 ) {
return(-1);
}
/* Get samples for the actual buffer size */
status = ALSA_snd_pcm_hw_params_get_buffer_size(hwparams, &bufsize);
if ( status < 0 ) {
return(-1);
}
if ( !override && bufsize != this->spec.samples * 2 ) {
return(-1);
}
/* !!! FIXME: Is this safe to do? */
this->spec.samples = bufsize / 2;
/* This is useful for debugging */
if ( SDL_getenv("SDL_AUDIO_ALSA_DEBUG") ) {
snd_pcm_uframes_t persize = 0;
unsigned int periods = 0;
ALSA_snd_pcm_hw_params_get_period_size(hwparams, &persize, NULL);
ALSA_snd_pcm_hw_params_get_periods(hwparams, &periods, NULL);
fprintf(stderr,
"ALSA: period size = %ld, periods = %u, buffer size = %lu\n",
persize, periods, bufsize);
}
return(0);
}
static int
ALSA_set_period_size(_THIS, snd_pcm_hw_params_t *params, int override)
{
const char *env;
int status;
snd_pcm_hw_params_t *hwparams;
snd_pcm_uframes_t frames;
unsigned int periods;
/* Copy the hardware parameters for this setup */
snd_pcm_hw_params_alloca(&hwparams);
ALSA_snd_pcm_hw_params_copy(hwparams, params);
if ( !override ) {
env = SDL_getenv("SDL_AUDIO_ALSA_SET_PERIOD_SIZE");
if ( env ) {
override = SDL_atoi(env);
if ( override == 0 ) {
return(-1);
}
}
}
frames = this->spec.samples;
status = ALSA_snd_pcm_hw_params_set_period_size_near(
this->hidden->pcm_handle, hwparams, &frames, NULL);
if ( status < 0 ) {
return(-1);
}
periods = 2;
status = ALSA_snd_pcm_hw_params_set_periods_near(
this->hidden->pcm_handle, hwparams, &periods, NULL);
if ( status < 0 ) {
return(-1);
}
return ALSA_finalize_hardware(this, hwparams, override);
}
static int
ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params, int override)
{
const char *env;
int status;
snd_pcm_hw_params_t *hwparams;
snd_pcm_uframes_t frames;
/* Copy the hardware parameters for this setup */
snd_pcm_hw_params_alloca(&hwparams);
ALSA_snd_pcm_hw_params_copy(hwparams, params);
if ( !override ) {
env = SDL_getenv("SDL_AUDIO_ALSA_SET_BUFFER_SIZE");
if ( env ) {
override = SDL_atoi(env);
if ( override == 0 ) {
return(-1);
}
}
}
frames = this->spec.samples * 2;
status = ALSA_snd_pcm_hw_params_set_buffer_size_near(
this->hidden->pcm_handle, hwparams, &frames);
if ( status < 0 ) {
return(-1);
}
return ALSA_finalize_hardware(this, hwparams, override);
}
static int
ALSA_OpenDevice(_THIS, const char *devname, int iscapture)
{
int status = 0;
snd_pcm_t *pcm_handle = NULL;
snd_pcm_hw_params_t *hwparams = NULL;
snd_pcm_sw_params_t *swparams = NULL;
snd_pcm_format_t format = 0;
SDL_AudioFormat test_format = 0;
unsigned int rate = 0;
unsigned int channels = 0;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Open the audio device */
/* Name of device should depend on # channels in spec */
status = ALSA_snd_pcm_open(&pcm_handle,
get_audio_device(this->spec.channels),
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if (status < 0) {
ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't open audio device: %s",
ALSA_snd_strerror(status));
return 0;
}
this->hidden->pcm_handle = pcm_handle;
/* Figure out what the hardware is capable of */
snd_pcm_hw_params_alloca(&hwparams);
status = ALSA_snd_pcm_hw_params_any(pcm_handle, hwparams);
if (status < 0) {
ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't get hardware config: %s",
ALSA_snd_strerror(status));
return 0;
}
/* SDL only uses interleaved sample output */
status = ALSA_snd_pcm_hw_params_set_access(pcm_handle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (status < 0) {
ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't set interleaved access: %s",
ALSA_snd_strerror(status));
return 0;
}
/* Try for a closest match on audio format */
status = -1;
for (test_format = SDL_FirstAudioFormat(this->spec.format);
test_format && (status < 0);) {
status = 0; /* if we can't support a format, it'll become -1. */
switch (test_format) {
case AUDIO_U8:
format = SND_PCM_FORMAT_U8;
break;
case AUDIO_S8:
format = SND_PCM_FORMAT_S8;
break;
case AUDIO_S16LSB:
format = SND_PCM_FORMAT_S16_LE;
break;
case AUDIO_S16MSB:
format = SND_PCM_FORMAT_S16_BE;
break;
case AUDIO_U16LSB:
format = SND_PCM_FORMAT_U16_LE;
break;
case AUDIO_U16MSB:
format = SND_PCM_FORMAT_U16_BE;
break;
case AUDIO_S32LSB:
format = SND_PCM_FORMAT_S32_LE;
break;
case AUDIO_S32MSB:
format = SND_PCM_FORMAT_S32_BE;
break;
case AUDIO_F32LSB:
format = SND_PCM_FORMAT_FLOAT_LE;
break;
case AUDIO_F32MSB:
format = SND_PCM_FORMAT_FLOAT_BE;
break;
default:
status = -1;
break;
}
if (status >= 0) {
status = ALSA_snd_pcm_hw_params_set_format(pcm_handle,
hwparams, format);
}
if (status < 0) {
test_format = SDL_NextAudioFormat();
}
}
if (status < 0) {
ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't find any hardware audio formats");
return 0;
}
this->spec.format = test_format;
/* Set the number of channels */
status = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams,
this->spec.channels);
channels = this->spec.channels;
if (status < 0) {
status = ALSA_snd_pcm_hw_params_get_channels(hwparams, &channels);
if (status < 0) {
ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't set audio channels");
return 0;
}
this->spec.channels = channels;
}
/* Set the audio rate */
rate = this->spec.freq;
status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams,
&rate, NULL);
if (status < 0) {
ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't set audio frequency: %s",
ALSA_snd_strerror(status));
return 0;
}
this->spec.freq = rate;
/* Set the buffer size, in samples */
if ( ALSA_set_period_size(this, hwparams, 0) < 0 &&
ALSA_set_buffer_size(this, hwparams, 0) < 0 ) {
/* Failed to set desired buffer size, do the best you can... */
if ( ALSA_set_period_size(this, hwparams, 1) < 0 ) {
ALSA_CloseDevice(this);
SDL_SetError("Couldn't set hardware audio parameters: %s", ALSA_snd_strerror(status));
return(-1);
}
}
/* Set the software parameters */
snd_pcm_sw_params_alloca(&swparams);
status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams);
if (status < 0) {
ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't get software config: %s",
ALSA_snd_strerror(status));
return 0;
}
status =
ALSA_snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 1);
if (status < 0) {
ALSA_CloseDevice(this);
SDL_SetError("ALSA: Couldn't set start threshold: %s",
ALSA_snd_strerror(status));
return 0;
}
status = ALSA_snd_pcm_sw_params(pcm_handle, swparams);
if (status < 0) {
ALSA_CloseDevice(this);
SDL_SetError("Couldn't set software audio parameters: %s",
ALSA_snd_strerror(status));
return 0;
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
ALSA_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/* Switch to blocking mode for playback */
ALSA_snd_pcm_nonblock(pcm_handle, 0);
/* We're ready to rock and roll. :-) */
return 1;
}
static void
ALSA_Deinitialize(void)
{
UnloadALSALibrary();
}
static int
ALSA_Init(SDL_AudioDriverImpl * impl)
{
if (LoadALSALibrary() < 0) {
return 0;
}
/* Set the function pointers */
impl->OpenDevice = ALSA_OpenDevice;
impl->WaitDevice = ALSA_WaitDevice;
impl->GetDeviceBuf = ALSA_GetDeviceBuf;
impl->PlayDevice = ALSA_PlayDevice;
impl->CloseDevice = ALSA_CloseDevice;
impl->Deinitialize = ALSA_Deinitialize;
impl->OnlyHasDefaultOutputDevice = 1; /* !!! FIXME: Add device enum! */
return 1; /* this audio target is available. */
}
AudioBootStrap ALSA_bootstrap = {
DRIVER_NAME, "ALSA PCM audio", ALSA_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

46
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@@ -0,0 +1,46 @@
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _ALSA_PCM_audio_h
#define _ALSA_PCM_audio_h
#include <alsa/asoundlib.h>
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
/* The audio device handle */
snd_pcm_t *pcm_handle;
/* Raw mixing buffer */
Uint8 *mixbuf;
int mixlen;
};
#endif /* _ALSA_PCM_audio_h */
/* vi: set ts=4 sw=4 expandtab: */

349
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2009 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
This file written by Ryan C. Gordon (icculus@icculus.org)
*/
#include "SDL_config.h"
#include "SDL_version.h"
#include "SDL_rwops.h"
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "../SDL_audiodev_c.h"
#include "SDL_androidaudio.h"
#include "SDL_mutex.h"
#include "SDL_thread.h"
#include <jni.h>
#include <android/log.h>
#include <string.h> // for memset()
#include <pthread.h>
#define _THIS SDL_AudioDevice *this
/* Audio driver functions */
static void ANDROIDAUD_WaitAudio(_THIS);
static void ANDROIDAUD_PlayAudio(_THIS);
static Uint8 *ANDROIDAUD_GetAudioBuf(_THIS);
static void ANDROIDAUD_CloseAudio(_THIS);
static void ANDROIDAUD_ThreadInit(_THIS);
static void ANDROIDAUD_ThreadDeinit(_THIS);
#if SDL_VERSION_ATLEAST(1,3,0)
static int ANDROIDAUD_OpenAudio(_THIS, const char *devname, int iscapture);
static void ANDROIDAUD_DeleteDevice()
{
}
static int ANDROIDAUD_CreateDevice(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->OpenDevice = ANDROIDAUD_OpenAudio;
impl->WaitDevice = ANDROIDAUD_WaitAudio;
impl->PlayDevice = ANDROIDAUD_PlayAudio;
impl->GetDeviceBuf = ANDROIDAUD_GetAudioBuf;
impl->CloseDevice = ANDROIDAUD_CloseAudio;
impl->ThreadInit = ANDROIDAUD_ThreadInit;
impl->WaitDone = ANDROIDAUD_ThreadDeinit;
impl->Deinitialize = ANDROIDAUD_DeleteDevice;
impl->OnlyHasDefaultOutputDevice = 1;
return 1;
}
AudioBootStrap ANDROIDAUD_bootstrap = {
"android", "SDL Android audio driver",
ANDROIDAUD_CreateDevice, 0
};
#else
static int ANDROIDAUD_OpenAudio(_THIS, SDL_AudioSpec *spec);
static int ANDROIDAUD_Available(void)
{
return(1);
}
static void ANDROIDAUD_DeleteDevice(SDL_AudioDevice *device)
{
SDL_free(device);
}
static SDL_AudioDevice *ANDROIDAUD_CreateDevice(int devindex)
{
SDL_AudioDevice *this;
/* Initialize all variables that we clean on shutdown */
this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
if ( this ) {
SDL_memset(this, 0, (sizeof *this));
this->hidden = NULL;
} else {
SDL_OutOfMemory();
return(0);
}
/* Set the function pointers */
this->OpenAudio = ANDROIDAUD_OpenAudio;
this->WaitAudio = ANDROIDAUD_WaitAudio;
this->PlayAudio = ANDROIDAUD_PlayAudio;
this->GetAudioBuf = ANDROIDAUD_GetAudioBuf;
this->CloseAudio = ANDROIDAUD_CloseAudio;
this->ThreadInit = ANDROIDAUD_ThreadInit;
this->WaitDone = ANDROIDAUD_ThreadDeinit;
this->free = ANDROIDAUD_DeleteDevice;
return this;
}
AudioBootStrap ANDROIDAUD_bootstrap = {
"android", "SDL Android audio driver",
ANDROIDAUD_Available, ANDROIDAUD_CreateDevice
};
#endif
static unsigned char * audioBuffer = NULL;
static size_t audioBufferSize = 0;
// Extremely wicked JNI environment to call Java functions from C code
static jbyteArray audioBufferJNI = NULL;
static JavaVM *jniVM = NULL;
static jobject JavaAudioThread = NULL;
static jmethodID JavaInitAudio = NULL;
static jmethodID JavaDeinitAudio = NULL;
static jmethodID JavaPauseAudioPlayback = NULL;
static jmethodID JavaResumeAudioPlayback = NULL;
static Uint8 *ANDROIDAUD_GetAudioBuf(_THIS)
{
return(audioBuffer);
}
#if SDL_VERSION_ATLEAST(1,3,0)
static int ANDROIDAUD_OpenAudio (_THIS, const char *devname, int iscapture)
{
SDL_AudioSpec *audioFormat = &this->spec;
#else
static int ANDROIDAUD_OpenAudio (_THIS, SDL_AudioSpec *spec)
{
SDL_AudioSpec *audioFormat = spec;
#endif
int bytesPerSample;
JNIEnv * jniEnv = NULL;
this->hidden = NULL;
if( ! (audioFormat->format == AUDIO_S8 || audioFormat->format == AUDIO_S16) )
{
__android_log_print(ANDROID_LOG_ERROR, "libSDL", "Application requested unsupported audio format - only S8 and S16 are supported");
return (-1); // TODO: enable format conversion? Don't know how to do that in SDL
}
bytesPerSample = (audioFormat->format & 0xFF) / 8;
audioFormat->format = ( bytesPerSample == 2 ) ? AUDIO_S16 : AUDIO_S8;
__android_log_print(ANDROID_LOG_INFO, "libSDL", "ANDROIDAUD_OpenAudio(): app requested audio bytespersample %d freq %d channels %d samples %d", bytesPerSample, audioFormat->freq, (int)audioFormat->channels, (int)audioFormat->samples);
if(audioFormat->samples <= 0)
audioFormat->samples = 128; // Some sane value
if( audioFormat->samples > 32768 ) // Why anyone need so huge audio buffer?
{
audioFormat->samples = 32768;
__android_log_print(ANDROID_LOG_INFO, "libSDL", "ANDROIDAUD_OpenAudio(): limiting samples size to ", (int)audioFormat->samples);
}
SDL_CalculateAudioSpec(audioFormat);
(*jniVM)->AttachCurrentThread(jniVM, &jniEnv, NULL);
if( !jniEnv )
{
__android_log_print(ANDROID_LOG_ERROR, "libSDL", "ANDROIDAUD_OpenAudio: Java VM AttachCurrentThread() failed");
return (-1); // TODO: enable format conversion? Don't know how to do that in SDL
}
// The returned audioBufferSize may be huge, up to 100 Kb for 44100 because user may have selected large audio buffer to get rid of choppy sound
audioBufferSize = (*jniEnv)->CallIntMethod( jniEnv, JavaAudioThread, JavaInitAudio,
(jint)audioFormat->freq, (jint)audioFormat->channels,
(jint)(( bytesPerSample == 2 ) ? 1 : 0), (jint)(audioFormat->size) );
if( audioBufferSize == 0 )
{
__android_log_print(ANDROID_LOG_INFO, "libSDL", "ANDROIDAUD_OpenAudio(): failed to get audio buffer from JNI");
ANDROIDAUD_CloseAudio(this);
return(-1);
}
/* We cannot call DetachCurrentThread() from main thread or we'll crash */
/* (*jniVM)->DetachCurrentThread(jniVM); */
audioFormat->samples = audioBufferSize / bytesPerSample / audioFormat->channels;
audioFormat->size = audioBufferSize;
__android_log_print(ANDROID_LOG_INFO, "libSDL", "ANDROIDAUD_OpenAudio(): app opened audio bytespersample %d freq %d channels %d bufsize %d", bytesPerSample, audioFormat->freq, (int)audioFormat->channels, audioBufferSize);
SDL_CalculateAudioSpec(audioFormat);
#if SDL_VERSION_ATLEAST(1,3,0)
return(1);
#else
return(0);
#endif
}
static void ANDROIDAUD_CloseAudio(_THIS)
{
//__android_log_print(ANDROID_LOG_INFO, "libSDL", "ANDROIDAUD_CloseAudio()");
JNIEnv * jniEnv = NULL;
(*jniVM)->AttachCurrentThread(jniVM, &jniEnv, NULL);
(*jniEnv)->DeleteGlobalRef(jniEnv, audioBufferJNI);
audioBufferJNI = NULL;
audioBuffer = NULL;
audioBufferSize = 0;
(*jniEnv)->CallIntMethod( jniEnv, JavaAudioThread, JavaDeinitAudio );
/* We cannot call DetachCurrentThread() from main thread or we'll crash */
/* (*jniVM)->DetachCurrentThread(jniVM); */
}
/* This function waits until it is possible to write a full sound buffer */
static void ANDROIDAUD_WaitAudio(_THIS)
{
/* We will block in PlayAudio(), do nothing here */
}
static JNIEnv * jniEnvPlaying = NULL;
static jmethodID JavaFillBuffer = NULL;
static void ANDROIDAUD_ThreadInit(_THIS)
{
jclass JavaAudioThreadClass = NULL;
jmethodID JavaInitThread = NULL;
jmethodID JavaGetBuffer = NULL;
jboolean isCopy = JNI_TRUE;
(*jniVM)->AttachCurrentThread(jniVM, &jniEnvPlaying, NULL);
JavaAudioThreadClass = (*jniEnvPlaying)->GetObjectClass(jniEnvPlaying, JavaAudioThread);
JavaFillBuffer = (*jniEnvPlaying)->GetMethodID(jniEnvPlaying, JavaAudioThreadClass, "fillBuffer", "()I");
/* HACK: raise our own thread priority to max to get rid of "W/AudioFlinger: write blocked for 54 msecs" errors */
JavaInitThread = (*jniEnvPlaying)->GetMethodID(jniEnvPlaying, JavaAudioThreadClass, "initAudioThread", "()I");
(*jniEnvPlaying)->CallIntMethod( jniEnvPlaying, JavaAudioThread, JavaInitThread );
JavaGetBuffer = (*jniEnvPlaying)->GetMethodID(jniEnvPlaying, JavaAudioThreadClass, "getBuffer", "()[B");
audioBufferJNI = (*jniEnvPlaying)->CallObjectMethod( jniEnvPlaying, JavaAudioThread, JavaGetBuffer );
audioBufferJNI = (*jniEnvPlaying)->NewGlobalRef(jniEnvPlaying, audioBufferJNI);
audioBuffer = (unsigned char *) (*jniEnvPlaying)->GetByteArrayElements(jniEnvPlaying, audioBufferJNI, &isCopy);
if( !audioBuffer )
{
__android_log_print(ANDROID_LOG_ERROR, "libSDL", "ANDROIDAUD_ThreadInit() JNI::GetByteArrayElements() failed! we will crash now");
return;
}
if( isCopy == JNI_TRUE )
__android_log_print(ANDROID_LOG_ERROR, "libSDL", "ANDROIDAUD_ThreadInit(): JNI returns a copy of byte array - no audio will be played");
//__android_log_print(ANDROID_LOG_INFO, "libSDL", "ANDROIDAUD_ThreadInit()");
SDL_memset(audioBuffer, this->spec.silence, this->spec.size);
};
static void ANDROIDAUD_ThreadDeinit(_THIS)
{
(*jniVM)->DetachCurrentThread(jniVM);
};
static void ANDROIDAUD_PlayAudio(_THIS)
{
//__android_log_print(ANDROID_LOG_INFO, "libSDL", "ANDROIDAUD_PlayAudio()");
jboolean isCopy = JNI_TRUE;
(*jniEnvPlaying)->ReleaseByteArrayElements(jniEnvPlaying, audioBufferJNI, (jbyte *)audioBuffer, 0);
audioBuffer = NULL;
(*jniEnvPlaying)->CallIntMethod( jniEnvPlaying, JavaAudioThread, JavaFillBuffer );
audioBuffer = (unsigned char *) (*jniEnvPlaying)->GetByteArrayElements(jniEnvPlaying, audioBufferJNI, &isCopy);
if( !audioBuffer )
__android_log_print(ANDROID_LOG_ERROR, "libSDL", "ANDROIDAUD_PlayAudio() JNI::GetByteArrayElements() failed! we will crash now");
if( isCopy == JNI_TRUE )
__android_log_print(ANDROID_LOG_INFO, "libSDL", "ANDROIDAUD_PlayAudio() JNI returns a copy of byte array - that's slow");
}
int SDL_ANDROID_PauseAudioPlayback(void)
{
JNIEnv * jniEnv = NULL;
(*jniVM)->AttachCurrentThread(jniVM, &jniEnv, NULL);
return (*jniEnv)->CallIntMethod( jniEnv, JavaAudioThread, JavaPauseAudioPlayback );
};
int SDL_ANDROID_ResumeAudioPlayback(void)
{
JNIEnv * jniEnv = NULL;
(*jniVM)->AttachCurrentThread(jniVM, &jniEnv, NULL);
return (*jniEnv)->CallIntMethod( jniEnv, JavaAudioThread, JavaResumeAudioPlayback );
};
#ifndef SDL_JAVA_PACKAGE_PATH
#error You have to define SDL_JAVA_PACKAGE_PATH to your package path with dots replaced with underscores, for example "com_example_SanAngeles"
#endif
#define JAVA_EXPORT_NAME2(name,package) Java_##package##_##name
#define JAVA_EXPORT_NAME1(name,package) JAVA_EXPORT_NAME2(name,package)
#define JAVA_EXPORT_NAME(name) JAVA_EXPORT_NAME1(name,SDL_JAVA_PACKAGE_PATH)
JNIEXPORT jint JNICALL JAVA_EXPORT_NAME(AudioThread_nativeAudioInitJavaCallbacks) (JNIEnv * jniEnv, jobject thiz)
{
jclass JavaAudioThreadClass = NULL;
JavaAudioThread = (*jniEnv)->NewGlobalRef(jniEnv, thiz);
JavaAudioThreadClass = (*jniEnv)->GetObjectClass(jniEnv, JavaAudioThread);
JavaInitAudio = (*jniEnv)->GetMethodID(jniEnv, JavaAudioThreadClass, "initAudio", "(IIII)I");
JavaDeinitAudio = (*jniEnv)->GetMethodID(jniEnv, JavaAudioThreadClass, "deinitAudio", "()I");
JavaPauseAudioPlayback = (*jniEnv)->GetMethodID(jniEnv, JavaAudioThreadClass, "pauseAudioPlayback", "()I");
JavaResumeAudioPlayback = (*jniEnv)->GetMethodID(jniEnv, JavaAudioThreadClass, "resumeAudioPlayback", "()I");
}
JNIEXPORT jint JNICALL JNI_OnLoad(JavaVM *vm, void *reserved)
{
jniVM = vm;
return JNI_VERSION_1_2;
};
JNIEXPORT void JNICALL JNI_OnUnload(JavaVM *vm, void *reserved)
{
/* TODO: free JavaAudioThread */
jniVM = vm;
};

32
project/jni/sdl-1.3/src/audio/android/SDL_androidaudio.h Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2009 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_androidaudio_h
#define _SDL_androidaudio_h
#include "../SDL_sysaudio.h"
struct SDL_PrivateAudioData {
};
#endif /* _SDL_androidaudio_h */

379
project/jni/sdl-1.3/src/audio/arts/SDL_artsaudio.c Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* Allow access to a raw mixing buffer */
#ifdef HAVE_SIGNAL_H
#include <signal.h>
#endif
#include <unistd.h>
#include <errno.h>
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_artsaudio.h"
#ifdef SDL_AUDIO_DRIVER_ARTS_DYNAMIC
#include "SDL_name.h"
#include "SDL_loadso.h"
#else
#define SDL_NAME(X) X
#endif
/* The tag name used by artsc audio */
#define ARTS_DRIVER_NAME "arts"
#ifdef SDL_AUDIO_DRIVER_ARTS_DYNAMIC
static const char *arts_library = SDL_AUDIO_DRIVER_ARTS_DYNAMIC;
static void *arts_handle = NULL;
/* !!! FIXME: I hate this SDL_NAME clutter...it makes everything so messy! */
static int (*SDL_NAME(arts_init)) (void);
static void (*SDL_NAME(arts_free)) (void);
static arts_stream_t(*SDL_NAME(arts_play_stream)) (int rate, int bits,
int channels,
const char *name);
static int (*SDL_NAME(arts_stream_set)) (arts_stream_t s,
arts_parameter_t param, int value);
static int (*SDL_NAME(arts_stream_get)) (arts_stream_t s,
arts_parameter_t param);
static int (*SDL_NAME(arts_write)) (arts_stream_t s, const void *buffer,
int count);
static void (*SDL_NAME(arts_close_stream)) (arts_stream_t s);
static int (*SDL_NAME(arts_suspended)) (void);
static const char *(*SDL_NAME(arts_error_text)) (int errorcode);
#define SDL_ARTS_SYM(x) { #x, (void **) (char *) &SDL_NAME(x) }
static struct
{
const char *name;
void **func;
} arts_functions[] = {
/* *INDENT-OFF* */
SDL_ARTS_SYM(arts_init),
SDL_ARTS_SYM(arts_free),
SDL_ARTS_SYM(arts_play_stream),
SDL_ARTS_SYM(arts_stream_set),
SDL_ARTS_SYM(arts_stream_get),
SDL_ARTS_SYM(arts_write),
SDL_ARTS_SYM(arts_close_stream),
SDL_ARTS_SYM(arts_suspended),
SDL_ARTS_SYM(arts_error_text),
/* *INDENT-ON* */
};
#undef SDL_ARTS_SYM
static void
UnloadARTSLibrary()
{
if (arts_handle != NULL) {
SDL_UnloadObject(arts_handle);
arts_handle = NULL;
}
}
static int
LoadARTSLibrary(void)
{
int i, retval = -1;
if (arts_handle == NULL) {
arts_handle = SDL_LoadObject(arts_library);
if (arts_handle != NULL) {
retval = 0;
for (i = 0; i < SDL_arraysize(arts_functions); ++i) {
*arts_functions[i].func =
SDL_LoadFunction(arts_handle, arts_functions[i].name);
if (!*arts_functions[i].func) {
retval = -1;
UnloadARTSLibrary();
break;
}
}
}
}
return retval;
}
#else
static void
UnloadARTSLibrary()
{
return;
}
static int
LoadARTSLibrary(void)
{
return 0;
}
#endif /* SDL_AUDIO_DRIVER_ARTS_DYNAMIC */
/* This function waits until it is possible to write a full sound buffer */
static void
ARTS_WaitDevice(_THIS)
{
Sint32 ticks;
/* Check to see if the thread-parent process is still alive */
{
static int cnt = 0;
/* Note that this only works with thread implementations
that use a different process id for each thread.
*/
/* Check every 10 loops */
if (this->hidden->parent && (((++cnt) % 10) == 0)) {
if (kill(this->hidden->parent, 0) < 0 && errno == ESRCH) {
this->enabled = 0;
}
}
}
/* Use timer for general audio synchronization */
ticks =
((Sint32) (this->hidden->next_frame - SDL_GetTicks())) - FUDGE_TICKS;
if (ticks > 0) {
SDL_Delay(ticks);
}
}
static void
ARTS_PlayDevice(_THIS)
{
/* Write the audio data */
int written = SDL_NAME(arts_write) (this->hidden->stream,
this->hidden->mixbuf,
this->hidden->mixlen);
/* If timer synchronization is enabled, set the next write frame */
if (this->hidden->frame_ticks) {
this->hidden->next_frame += this->hidden->frame_ticks;
}
/* If we couldn't write, assume fatal error for now */
if (written < 0) {
this->enabled = 0;
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", written);
#endif
}
static void
ARTS_WaitDone(_THIS)
{
/* !!! FIXME: camp here until buffer drains... SDL_Delay(???); */
}
static Uint8 *
ARTS_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
}
static void
ARTS_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
if (this->hidden->stream) {
SDL_NAME(arts_close_stream) (this->hidden->stream);
this->hidden->stream = 0;
}
SDL_NAME(arts_free) ();
SDL_free(this->hidden);
this->hidden = NULL;
}
}
static int
ARTS_OpenDevice(_THIS, const char *devname, int iscapture)
{
int rc = 0;
int bits = 0, frag_spec = 0;
SDL_AudioFormat test_format = 0, format = 0;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Try for a closest match on audio format */
for (test_format = SDL_FirstAudioFormat(this->spec.format);
!format && test_format;) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
switch (test_format) {
case AUDIO_U8:
bits = 8;
format = 1;
break;
case AUDIO_S16LSB:
bits = 16;
format = 1;
break;
default:
format = 0;
break;
}
if (!format) {
test_format = SDL_NextAudioFormat();
}
}
if (format == 0) {
ARTS_CloseDevice(this);
SDL_SetError("Couldn't find any hardware audio formats");
return 0;
}
this->spec.format = test_format;
if ((rc = SDL_NAME(arts_init) ()) != 0) {
ARTS_CloseDevice(this);
SDL_SetError("Unable to initialize ARTS: %s",
SDL_NAME(arts_error_text) (rc));
return 0;
}
if (!SDL_NAME(arts_suspended) ()) {
ARTS_CloseDevice(this);
SDL_SetError("ARTS can not open audio device");
return 0;
}
this->hidden->stream = SDL_NAME(arts_play_stream) (this->spec.freq,
bits,
this->spec.channels,
"SDL");
/* Play nothing so we have at least one write (server bug workaround). */
SDL_NAME(arts_write) (this->hidden->stream, "", 0);
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
/* Determine the power of two of the fragment size */
for (frag_spec = 0; (0x01 << frag_spec) < this->spec.size; ++frag_spec);
if ((0x01 << frag_spec) != this->spec.size) {
ARTS_CloseDevice(this);
SDL_SetError("Fragment size must be a power of two");
return 0;
}
frag_spec |= 0x00020000; /* two fragments, for low latency */
#ifdef ARTS_P_PACKET_SETTINGS
SDL_NAME(arts_stream_set) (this->hidden->stream,
ARTS_P_PACKET_SETTINGS, frag_spec);
#else
SDL_NAME(arts_stream_set) (this->hidden->stream, ARTS_P_PACKET_SIZE,
frag_spec & 0xffff);
SDL_NAME(arts_stream_set) (this->hidden->stream, ARTS_P_PACKET_COUNT,
frag_spec >> 16);
#endif
this->spec.size = SDL_NAME(arts_stream_get) (this->hidden->stream,
ARTS_P_PACKET_SIZE);
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
ARTS_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/* Get the parent process id (we're the parent of the audio thread) */
this->hidden->parent = getpid();
/* We're ready to rock and roll. :-) */
return 1;
}
static void
ARTS_Deinitialize(void)
{
UnloadARTSLibrary();
}
static int
ARTS_Init(SDL_AudioDriverImpl * impl)
{
if (LoadARTSLibrary() < 0) {
return 0;
} else {
if (SDL_NAME(arts_init) () != 0) {
UnloadARTSLibrary();
SDL_SetError("ARTS: arts_init failed (no audio server?)");
return 0;
}
/* Play a stream so aRts doesn't crash */
if (SDL_NAME(arts_suspended) ()) {
arts_stream_t stream;
stream = SDL_NAME(arts_play_stream) (44100, 16, 2, "SDL");
SDL_NAME(arts_write) (stream, "", 0);
SDL_NAME(arts_close_stream) (stream);
}
SDL_NAME(arts_free) ();
}
/* Set the function pointers */
impl->OpenDevice = ARTS_OpenDevice;
impl->PlayDevice = ARTS_PlayDevice;
impl->WaitDevice = ARTS_WaitDevice;
impl->GetDeviceBuf = ARTS_GetDeviceBuf;
impl->CloseDevice = ARTS_CloseDevice;
impl->WaitDone = ARTS_WaitDone;
impl->Deinitialize = ARTS_Deinitialize;
impl->OnlyHasDefaultOutputDevice = 1;
return 1; /* this audio target is available. */
}
AudioBootStrap ARTS_bootstrap = {
ARTS_DRIVER_NAME, "Analog RealTime Synthesizer", ARTS_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

53
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@@ -0,0 +1,53 @@
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_artscaudio_h
#define _SDL_artscaudio_h
#include <artsc.h>
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
/* The stream descriptor for the audio device */
arts_stream_t stream;
/* The parent process id, to detect when application quits */
pid_t parent;
/* Raw mixing buffer */
Uint8 *mixbuf;
int mixlen;
/* Support for audio timing using a timer, in addition to select() */
float frame_ticks;
float next_frame;
};
#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
#endif /* _SDL_artscaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

218
project/jni/sdl-1.3/src/audio/baudio/SDL_beaudio.cc Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2009 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* Allow access to the audio stream on BeOS */
#include <SoundPlayer.h>
#include "../../main/beos/SDL_BeApp.h"
extern "C"
{
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "../SDL_sysaudio.h"
#include "../../thread/beos/SDL_systhread_c.h"
#include "SDL_beaudio.h"
}
/* !!! FIXME: have the callback call the higher level to avoid code dupe. */
/* The BeOS callback for handling the audio buffer */
static void
FillSound(void *device, void *stream, size_t len,
const media_raw_audio_format & format)
{
SDL_AudioDevice *audio = (SDL_AudioDevice *) device;
/* Silence the buffer, since it's ours */
SDL_memset(stream, audio->spec.silence, len);
/* Only do soemthing if audio is enabled */
if (!audio->enabled)
return;
if (!audio->paused) {
if (audio->convert.needed) {
SDL_mutexP(audio->mixer_lock);
(*audio->spec.callback) (audio->spec.userdata,
(Uint8 *) audio->convert.buf,
audio->convert.len);
SDL_mutexV(audio->mixer_lock);
SDL_ConvertAudio(&audio->convert);
SDL_memcpy(stream, audio->convert.buf, audio->convert.len_cvt);
} else {
SDL_mutexP(audio->mixer_lock);
(*audio->spec.callback) (audio->spec.userdata,
(Uint8 *) stream, len);
SDL_mutexV(audio->mixer_lock);
}
}
}
static void
BEOSAUDIO_CloseDevice(_THIS)
{
if (_this->hidden != NULL) {
if (_this->hidden->audio_obj) {
_this->hidden->audio_obj->Stop();
delete _this->hidden->audio_obj;
_this->hidden->audio_obj = NULL;
}
delete _this->hidden;
_this->hidden = NULL;
}
}
static int
BEOSAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
{
int valid_datatype = 0;
media_raw_audio_format format;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(_this->spec.format);
/* Initialize all variables that we clean on shutdown */
_this->hidden = new SDL_PrivateAudioData;
if (_this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(_this->hidden, 0, (sizeof *_this->hidden));
/* Parse the audio format and fill the Be raw audio format */
SDL_memset(&format, '\0', sizeof(media_raw_audio_format));
format.byte_order = B_MEDIA_LITTLE_ENDIAN;
format.frame_rate = (float) _this->spec.freq;
format.channel_count = _this->spec.channels; /* !!! FIXME: support > 2? */
while ((!valid_datatype) && (test_format)) {
valid_datatype = 1;
_this->spec.format = test_format;
switch (test_format) {
case AUDIO_S8:
format.format = media_raw_audio_format::B_AUDIO_CHAR;
break;
case AUDIO_U8:
format.format = media_raw_audio_format::B_AUDIO_UCHAR;
break;
case AUDIO_S16LSB:
format.format = media_raw_audio_format::B_AUDIO_SHORT;
break;
case AUDIO_S16MSB:
format.format = media_raw_audio_format::B_AUDIO_SHORT;
format.byte_order = B_MEDIA_BIG_ENDIAN;
break;
case AUDIO_S32LSB:
format.format = media_raw_audio_format::B_AUDIO_INT;
break;
case AUDIO_S32MSB:
format.format = media_raw_audio_format::B_AUDIO_INT;
format.byte_order = B_MEDIA_BIG_ENDIAN;
break;
case AUDIO_F32LSB:
format.format = media_raw_audio_format::B_AUDIO_FLOAT;
break;
case AUDIO_F32MSB:
format.format = media_raw_audio_format::B_AUDIO_FLOAT;
format.byte_order = B_MEDIA_BIG_ENDIAN;
break;
default:
valid_datatype = 0;
test_format = SDL_NextAudioFormat();
break;
}
}
format.buffer_size = _this->spec.samples;
if (!valid_datatype) { /* shouldn't happen, but just in case... */
BEOSAUDIO_CloseDevice(_this);
SDL_SetError("Unsupported audio format");
return 0;
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&_this->spec);
/* Subscribe to the audio stream (creates a new thread) */
sigset_t omask;
SDL_MaskSignals(&omask);
_this->hidden->audio_obj = new BSoundPlayer(&format, "SDL Audio",
FillSound, NULL, _this);
SDL_UnmaskSignals(&omask);
if (_this->hidden->audio_obj->Start() == B_NO_ERROR) {
_this->hidden->audio_obj->SetHasData(true);
} else {
BEOSAUDIO_CloseDevice(_this);
SDL_SetError("Unable to start Be audio");
return 0;
}
/* We're running! */
return 1;
}
static void
BEOSAUDIO_Deinitialize(void)
{
SDL_QuitBeApp();
}
static int
BEOSAUDIO_Init(SDL_AudioDriverImpl * impl)
{
/* Initialize the Be Application, if it's not already started */
if (SDL_InitBeApp() < 0) {
return 0;
}
/* Set the function pointers */
impl->OpenDevice = BEOSAUDIO_OpenDevice;
impl->CloseDevice = BEOSAUDIO_CloseDevice;
impl->Deinitialize = BEOSAUDIO_Deinitialize;
impl->ProvidesOwnCallbackThread = 1;
impl->OnlyHasDefaultOutputDevice = 1;
return 1; /* this audio target is available. */
}
extern "C"
{
extern AudioBootStrap BEOSAUDIO_bootstrap;
}
AudioBootStrap BEOSAUDIO_bootstrap = {
"baudio", "BeOS BSoundPlayer", BEOSAUDIO_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

39
project/jni/sdl-1.3/src/audio/baudio/SDL_beaudio.h Normal file
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@@ -0,0 +1,39 @@
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_beaudio_h
#define _SDL_beaudio_h
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *_this
struct SDL_PrivateAudioData
{
BSoundPlayer *audio_obj;
};
#endif /* _SDL_beaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

456
project/jni/sdl-1.3/src/audio/bsd/SDL_bsdaudio.c Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/*
* Driver for native OpenBSD/NetBSD audio(4).
* vedge@vedge.com.ar.
*/
#include <errno.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/time.h>
#include <sys/ioctl.h>
#include <sys/stat.h>
#include <sys/types.h>
#include <sys/audioio.h>
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "../SDL_audiodev_c.h"
#include "SDL_bsdaudio.h"
/* The tag name used by NetBSD/OpenBSD audio */
#ifdef __NetBSD__
#define BSD_AUDIO_DRIVER_NAME "netbsd"
#define BSD_AUDIO_DRIVER_DESC "Native NetBSD audio"
#else
#define BSD_AUDIO_DRIVER_NAME "openbsd"
#define BSD_AUDIO_DRIVER_DESC "Native OpenBSD audio"
#endif
/* Open the audio device for playback, and don't block if busy */
/* #define USE_BLOCKING_WRITES */
/* Use timer for synchronization */
/* #define USE_TIMER_SYNC */
/* #define DEBUG_AUDIO */
/* #define DEBUG_AUDIO_STREAM */
#ifdef USE_BLOCKING_WRITES
#define OPEN_FLAGS_OUTPUT O_WRONLY
#define OPEN_FLAGS_INPUT O_RDONLY
#else
#define OPEN_FLAGS_OUTPUT (O_WRONLY|O_NONBLOCK)
#define OPEN_FLAGS_INPUT (O_RDONLY|O_NONBLOCK)
#endif
/* !!! FIXME: so much cut and paste with dsp/dma drivers... */
static char **outputDevices = NULL;
static int outputDeviceCount = 0;
static char **inputDevices = NULL;
static int inputDeviceCount = 0;
static inline void
free_device_list(char ***devs, int *count)
{
SDL_FreeUnixAudioDevices(devs, count);
}
static inline void
build_device_list(int iscapture, char ***devs, int *count)
{
const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
free_device_list(devs, count);
SDL_EnumUnixAudioDevices(flags, 0, NULL, devs, count);
}
static inline void
build_device_lists(void)
{
build_device_list(0, &outputDevices, &outputDeviceCount);
build_device_list(1, &inputDevices, &inputDeviceCount);
}
static inline void
free_device_lists(void)
{
free_device_list(&outputDevices, &outputDeviceCount);
free_device_list(&inputDevices, &inputDeviceCount);
}
static void
BSDAUDIO_Deinitialize(void)
{
free_device_lists();
}
static int
BSDAUDIO_DetectDevices(int iscapture)
{
if (iscapture) {
build_device_list(1, &inputDevices, &inputDeviceCount);
return inputDeviceCount;
} else {
build_device_list(0, &outputDevices, &outputDeviceCount);
return outputDeviceCount;
}
return 0; /* shouldn't ever hit this. */
}
static const char *
BSDAUDIO_GetDeviceName(int index, int iscapture)
{
if ((iscapture) && (index < inputDeviceCount)) {
return inputDevices[index];
} else if ((!iscapture) && (index < outputDeviceCount)) {
return outputDevices[index];
}
SDL_SetError("No such device");
return NULL;
}
static void
BSDAUDIO_Status(_THIS)
{
#ifdef DEBUG_AUDIO
/* *INDENT-OFF* */
audio_info_t info;
if (ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info) < 0) {
fprintf(stderr, "AUDIO_GETINFO failed.\n");
return;
}
fprintf(stderr, "\n"
"[play/record info]\n"
"buffer size : %d bytes\n"
"sample rate : %i Hz\n"
"channels : %i\n"
"precision : %i-bit\n"
"encoding : 0x%x\n"
"seek : %i\n"
"sample count : %i\n"
"EOF count : %i\n"
"paused : %s\n"
"error occured : %s\n"
"waiting : %s\n"
"active : %s\n"
"",
info.play.buffer_size,
info.play.sample_rate,
info.play.channels,
info.play.precision,
info.play.encoding,
info.play.seek,
info.play.samples,
info.play.eof,
info.play.pause ? "yes" : "no",
info.play.error ? "yes" : "no",
info.play.waiting ? "yes" : "no",
info.play.active ? "yes" : "no");
fprintf(stderr, "\n"
"[audio info]\n"
"monitor_gain : %i\n"
"hw block size : %d bytes\n"
"hi watermark : %i\n"
"lo watermark : %i\n"
"audio mode : %s\n"
"",
info.monitor_gain,
info.blocksize,
info.hiwat, info.lowat,
(info.mode == AUMODE_PLAY) ? "PLAY"
: (info.mode = AUMODE_RECORD) ? "RECORD"
: (info.mode == AUMODE_PLAY_ALL ? "PLAY_ALL" : "?"));
/* *INDENT-ON* */
#endif /* DEBUG_AUDIO */
}
/* This function waits until it is possible to write a full sound buffer */
static void
BSDAUDIO_WaitDevice(_THIS)
{
#ifndef USE_BLOCKING_WRITES /* Not necessary when using blocking writes */
/* See if we need to use timed audio synchronization */
if (this->hidden->frame_ticks) {
/* Use timer for general audio synchronization */
Sint32 ticks;
ticks =
((Sint32) (this->hidden->next_frame - SDL_GetTicks())) -
FUDGE_TICKS;
if (ticks > 0) {
SDL_Delay(ticks);
}
} else {
/* Use select() for audio synchronization */
fd_set fdset;
struct timeval timeout;
FD_ZERO(&fdset);
FD_SET(this->hidden->audio_fd, &fdset);
timeout.tv_sec = 10;
timeout.tv_usec = 0;
#ifdef DEBUG_AUDIO
fprintf(stderr, "Waiting for audio to get ready\n");
#endif
if (select(this->hidden->audio_fd + 1, NULL, &fdset, NULL, &timeout)
<= 0) {
const char *message =
"Audio timeout - buggy audio driver? (disabled)";
/* In general we should never print to the screen,
but in this case we have no other way of letting
the user know what happened.
*/
fprintf(stderr, "SDL: %s\n", message);
this->enabled = 0;
/* Don't try to close - may hang */
this->hidden->audio_fd = -1;
#ifdef DEBUG_AUDIO
fprintf(stderr, "Done disabling audio\n");
#endif
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Ready!\n");
#endif
}
#endif /* !USE_BLOCKING_WRITES */
}
static void
BSDAUDIO_PlayDevice(_THIS)
{
int written, p = 0;
/* Write the audio data, checking for EAGAIN on broken audio drivers */
do {
written = write(this->hidden->audio_fd,
&this->hidden->mixbuf[p], this->hidden->mixlen - p);
if (written > 0)
p += written;
if (written == -1 && errno != 0 && errno != EAGAIN && errno != EINTR) {
/* Non recoverable error has occurred. It should be reported!!! */
perror("audio");
break;
}
if (p < written
|| ((written < 0) && ((errno == 0) || (errno == EAGAIN)))) {
SDL_Delay(1); /* Let a little CPU time go by */
}
} while (p < written);
/* If timer synchronization is enabled, set the next write frame */
if (this->hidden->frame_ticks) {
this->hidden->next_frame += this->hidden->frame_ticks;
}
/* If we couldn't write, assume fatal error for now */
if (written < 0) {
this->enabled = 0;
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", written);
#endif
}
static Uint8 *
BSDAUDIO_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
}
static void
BSDAUDIO_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
if (this->hidden->audio_fd >= 0) {
close(this->hidden->audio_fd);
this->hidden->audio_fd = -1;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
static int
BSDAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
{
const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
SDL_AudioFormat format = 0;
audio_info_t info;
/* We don't care what the devname is...we'll try to open anything. */
/* ...but default to first name in the list... */
if (devname == NULL) {
if (((iscapture) && (inputDeviceCount == 0)) ||
((!iscapture) && (outputDeviceCount == 0))) {
SDL_SetError("No such audio device");
return 0;
}
devname = ((iscapture) ? inputDevices[0] : outputDevices[0]);
}
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Open the audio device */
this->hidden->audio_fd = open(devname, flags, 0);
if (this->hidden->audio_fd < 0) {
SDL_SetError("Couldn't open %s: %s", devname, strerror(errno));
return 0;
}
AUDIO_INITINFO(&info);
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
/* Set to play mode */
info.mode = AUMODE_PLAY;
if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) < 0) {
BSDAUDIO_CloseDevice(this);
SDL_SetError("Couldn't put device into play mode");
return 0;
}
AUDIO_INITINFO(&info);
for (format = SDL_FirstAudioFormat(this->spec.format);
format; format = SDL_NextAudioFormat()) {
switch (format) {
case AUDIO_U8:
info.play.encoding = AUDIO_ENCODING_ULINEAR;
info.play.precision = 8;
break;
case AUDIO_S8:
info.play.encoding = AUDIO_ENCODING_SLINEAR;
info.play.precision = 8;
break;
case AUDIO_S16LSB:
info.play.encoding = AUDIO_ENCODING_SLINEAR_LE;
info.play.precision = 16;
break;
case AUDIO_S16MSB:
info.play.encoding = AUDIO_ENCODING_SLINEAR_BE;
info.play.precision = 16;
break;
case AUDIO_U16LSB:
info.play.encoding = AUDIO_ENCODING_ULINEAR_LE;
info.play.precision = 16;
break;
case AUDIO_U16MSB:
info.play.encoding = AUDIO_ENCODING_ULINEAR_BE;
info.play.precision = 16;
break;
default:
continue;
}
if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) == 0) {
break;
}
}
if (!format) {
BSDAUDIO_CloseDevice(this);
SDL_SetError("No supported encoding for 0x%x", this->spec.format);
return 0;
}
this->spec.format = format;
AUDIO_INITINFO(&info);
info.play.channels = this->spec.channels;
if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) == -1) {
this->spec.channels = 1;
}
AUDIO_INITINFO(&info);
info.play.sample_rate = this->spec.freq;
info.blocksize = this->spec.size;
info.hiwat = 5;
info.lowat = 3;
(void) ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info);
(void) ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info);
this->spec.freq = info.play.sample_rate;
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
BSDAUDIO_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
BSDAUDIO_Status(this);
/* We're ready to rock and roll. :-) */
return (0);
}
static int
BSDAUDIO_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->DetectDevices = BSDAUDIO_DetectDevices;
impl->GetDeviceName = BSDAUDIO_GetDeviceName;
impl->OpenDevice = BSDAUDIO_OpenDevice;
impl->PlayDevice = BSDAUDIO_PlayDevice;
impl->WaitDevice = BSDAUDIO_WaitDevice;
impl->GetDeviceBuf = BSDAUDIO_GetDeviceBuf;
impl->CloseDevice = BSDAUDIO_CloseDevice;
impl->Deinitialize = BSDAUDIO_Deinitialize;
build_device_lists();
return 1; /* this audio target is available. */
}
AudioBootStrap BSD_AUDIO_bootstrap = {
BSD_AUDIO_DRIVER_NAME, BSD_AUDIO_DRIVER_DESC, BSDAUDIO_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

52
project/jni/sdl-1.3/src/audio/bsd/SDL_bsdaudio.h Normal file
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@@ -0,0 +1,52 @@
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_bsdaudio_h
#define _SDL_bsdaudio_h
#include "../SDL_sysaudio.h"
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
/* The file descriptor for the audio device */
int audio_fd;
/* The parent process id, to detect when application quits */
pid_t parent;
/* Raw mixing buffer */
Uint8 *mixbuf;
int mixlen;
/* Support for audio timing using a timer, in addition to select() */
float frame_ticks;
float next_frame;
};
#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
#endif /* _SDL_bsdaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

168
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 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
This file written by Ryan C. Gordon (icculus@icculus.org)
*/
#include "SDL_config.h"
/* Output raw audio data to a file. */
#if HAVE_STDIO_H
#include <stdio.h>
#endif
#include "SDL_rwops.h"
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_diskaudio.h"
/* The tag name used by DISK audio */
#define DISKAUD_DRIVER_NAME "disk"
/* environment variables and defaults. */
#define DISKENVR_OUTFILE "SDL_DISKAUDIOFILE"
#define DISKDEFAULT_OUTFILE "sdlaudio.raw"
#define DISKENVR_WRITEDELAY "SDL_DISKAUDIODELAY"
#define DISKDEFAULT_WRITEDELAY 150
static const char *
DISKAUD_GetOutputFilename(const char *devname)
{
if (devname == NULL) {
devname = SDL_getenv(DISKENVR_OUTFILE);
if (devname == NULL) {
devname = DISKDEFAULT_OUTFILE;
}
}
return devname;
}
/* This function waits until it is possible to write a full sound buffer */
static void
DISKAUD_WaitDevice(_THIS)
{
SDL_Delay(this->hidden->write_delay);
}
static void
DISKAUD_PlayDevice(_THIS)
{
size_t written;
/* Write the audio data */
written = SDL_RWwrite(this->hidden->output,
this->hidden->mixbuf, 1, this->hidden->mixlen);
/* If we couldn't write, assume fatal error for now */
if (written != this->hidden->mixlen) {
this->enabled = 0;
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", written);
#endif
}
static Uint8 *
DISKAUD_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
}
static void
DISKAUD_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
if (this->hidden->output != NULL) {
SDL_RWclose(this->hidden->output);
this->hidden->output = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
static int
DISKAUD_OpenDevice(_THIS, const char *devname, int iscapture)
{
const char *envr = SDL_getenv(DISKENVR_WRITEDELAY);
const char *fname = DISKAUD_GetOutputFilename(devname);
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc(sizeof(*this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, sizeof(*this->hidden));
/* Open the audio device */
this->hidden->output = SDL_RWFromFile(fname, "wb");
if (this->hidden->output == NULL) {
DISKAUD_CloseDevice(this);
return 0;
}
/* Allocate mixing buffer */
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
DISKAUD_CloseDevice(this);
return 0;
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
this->hidden->mixlen = this->spec.size;
this->hidden->write_delay =
(envr) ? SDL_atoi(envr) : DISKDEFAULT_WRITEDELAY;
#if HAVE_STDIO_H
fprintf(stderr,
"WARNING: You are using the SDL disk writer audio driver!\n"
" Writing to file [%s].\n", fname);
#endif
/* We're ready to rock and roll. :-) */
return 1;
}
static int
DISKAUD_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->OpenDevice = DISKAUD_OpenDevice;
impl->WaitDevice = DISKAUD_WaitDevice;
impl->PlayDevice = DISKAUD_PlayDevice;
impl->GetDeviceBuf = DISKAUD_GetDeviceBuf;
impl->CloseDevice = DISKAUD_CloseDevice;
return 1; /* this audio target is available. */
}
AudioBootStrap DISKAUD_bootstrap = {
DISKAUD_DRIVER_NAME, "direct-to-disk audio", DISKAUD_Init, 1
};
/* vi: set ts=4 sw=4 expandtab: */

43
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_diskaudio_h
#define _SDL_diskaudio_h
#include "SDL_rwops.h"
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
/* The file descriptor for the audio device */
SDL_RWops *output;
Uint8 *mixbuf;
Uint32 mixlen;
Uint32 write_delay;
};
#endif /* _SDL_diskaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

535
project/jni/sdl-1.3/src/audio/dma/SDL_dmaaudio.c Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* !!! FIXME: merge this driver with "dsp". */
/* Allow access to a raw mixing buffer */
#include <stdio.h>
#include <string.h> /* For strerror() */
#include <errno.h>
#include <unistd.h>
#include <fcntl.h>
#include <signal.h>
#include <sys/types.h>
#include <sys/time.h>
#include <sys/ioctl.h>
#include <sys/stat.h>
#include <sys/mman.h>
#if SDL_AUDIO_DRIVER_OSS_SOUNDCARD_H
/* This is installed on some systems */
#include <soundcard.h>
#else
/* This is recommended by OSS */
#include <sys/soundcard.h>
#endif
#ifndef MAP_FAILED
#define MAP_FAILED ((Uint8 *)-1)
#endif
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "../SDL_audiodev_c.h"
#include "SDL_dmaaudio.h"
/* The tag name used by DMA audio */
#define DMA_DRIVER_NAME "dma"
/* Open the audio device for playback, and don't block if busy */
#define OPEN_FLAGS_INPUT (O_RDWR|O_NONBLOCK)
#define OPEN_FLAGS_OUTPUT (O_RDWR|O_NONBLOCK)
static char **outputDevices = NULL;
static int outputDeviceCount = 0;
static char **inputDevices = NULL;
static int inputDeviceCount = 0;
static int
test_for_mmap(int fd)
{
int caps = 0;
struct audio_buf_info info;
if ((ioctl(fd, SNDCTL_DSP_GETCAPS, &caps) == 0) &&
(caps & DSP_CAP_TRIGGER) && (caps & DSP_CAP_MMAP) &&
(ioctl(fd, SNDCTL_DSP_GETOSPACE, &info) == 0)) {
size_t len = info.fragstotal * info.fragsize;
Uint8 *buf = (Uint8 *) mmap(NULL, len, PROT_WRITE, MAP_SHARED, fd, 0);
if (buf != MAP_FAILED) {
munmap(buf, len);
return 1;
}
}
return 0;
}
static inline void
free_device_list(char ***devs, int *count)
{
SDL_FreeUnixAudioDevices(devs, count);
}
static inline void
build_device_list(int iscapture, char ***devs, int *count)
{
const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
free_device_list(devs, count);
SDL_EnumUnixAudioDevices(flags, 0, test_for_mmap, devs, count);
}
static inline void
build_device_lists(void)
{
build_device_list(0, &outputDevices, &outputDeviceCount);
build_device_list(1, &inputDevices, &inputDeviceCount);
}
static inline void
free_device_lists(void)
{
free_device_list(&outputDevices, &outputDeviceCount);
free_device_list(&inputDevices, &inputDeviceCount);
}
static void
DMA_Deinitialize(void)
{
free_device_lists();
}
static int
DMA_DetectDevices(int iscapture)
{
if (iscapture) {
build_device_list(1, &inputDevices, &inputDeviceCount);
return inputDeviceCount;
} else {
build_device_list(0, &outputDevices, &outputDeviceCount);
return outputDeviceCount;
}
return 0; /* shouldn't ever hit this. */
}
static const char *
DMA_GetDeviceName(int index, int iscapture)
{
if ((iscapture) && (index < inputDeviceCount)) {
return inputDevices[index];
} else if ((!iscapture) && (index < outputDeviceCount)) {
return outputDevices[index];
}
SDL_SetError("No such device");
return NULL;
}
static int
DMA_ReopenAudio(_THIS, const char *audiodev, int format, int stereo)
{
int frag_spec;
int value;
/* Close and then reopen the audio device */
close(audio_fd);
audio_fd = open(audiodev, O_RDWR, 0);
if (audio_fd < 0) {
SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
return (-1);
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
/* Determine the power of two of the fragment size */
for (frag_spec = 0; (0x01 << frag_spec) < this->spec.size; ++frag_spec);
if ((0x01 << frag_spec) != this->spec.size) {
SDL_SetError("Fragment size must be a power of two");
return (-1);
}
/* Set the audio buffering parameters */
if (ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_spec) < 0) {
SDL_SetError("Couldn't set audio fragment spec");
return (-1);
}
/* Set the audio format */
value = format;
if ((ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) || (value != format)) {
SDL_SetError("Couldn't set audio format");
return (-1);
}
/* Set mono or stereo audio */
value = (this->spec.channels > 1);
if ((ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) < 0) ||
(value != stereo)) {
SDL_SetError("Couldn't set audio channels");
return (-1);
}
/* Set the DSP frequency */
value = this->spec.freq;
if (ioctl(audio_fd, SNDCTL_DSP_SPEED, &value) < 0) {
SDL_SetError("Couldn't set audio frequency");
return (-1);
}
this->spec.freq = value;
/* We successfully re-opened the audio */
return (0);
}
static void
DMA_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (dma_buf != NULL) {
munmap(dma_buf, dma_len);
dma_buf = NULL;
}
if (audio_fd >= 0) {
close(audio_fd);
audio_fd = -1;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
static int
DMA_OpenDevice(_THIS, const char *devname, int iscapture)
{
const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
int format;
int stereo;
int value;
SDL_AudioFormat test_format;
struct audio_buf_info info;
/* We don't care what the devname is...we'll try to open anything. */
/* ...but default to first name in the list... */
if (devname == NULL) {
if (((iscapture) && (inputDeviceCount == 0)) ||
((!iscapture) && (outputDeviceCount == 0))) {
SDL_SetError("No such audio device");
return 0;
}
devname = ((iscapture) ? inputDevices[0] : outputDevices[0]);
}
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Open the audio device */
audio_fd = open(devname, flags, 0);
if (audio_fd < 0) {
DMA_CloseDevice(this);
SDL_SetError("Couldn't open %s: %s", devname, strerror(errno));
return 0;
}
dma_buf = NULL;
ioctl(audio_fd, SNDCTL_DSP_RESET, 0);
/* Get a list of supported hardware formats */
if (ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &value) < 0) {
DMA_CloseDevice(this);
SDL_SetError("Couldn't get audio format list");
return 0;
}
/* Try for a closest match on audio format */
format = 0;
for (test_format = SDL_FirstAudioFormat(this->spec.format);
!format && test_format;) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
switch (test_format) {
case AUDIO_U8:
if (value & AFMT_U8) {
format = AFMT_U8;
}
break;
case AUDIO_S8:
if (value & AFMT_S8) {
format = AFMT_S8;
}
break;
case AUDIO_S16LSB:
if (value & AFMT_S16_LE) {
format = AFMT_S16_LE;
}
break;
case AUDIO_S16MSB:
if (value & AFMT_S16_BE) {
format = AFMT_S16_BE;
}
break;
case AUDIO_U16LSB:
if (value & AFMT_U16_LE) {
format = AFMT_U16_LE;
}
break;
case AUDIO_U16MSB:
if (value & AFMT_U16_BE) {
format = AFMT_U16_BE;
}
break;
default:
format = 0;
break;
}
if (!format) {
test_format = SDL_NextAudioFormat();
}
}
if (format == 0) {
DMA_CloseDevice(this);
SDL_SetError("Couldn't find any hardware audio formats");
return 0;
}
this->spec.format = test_format;
/* Set the audio format */
value = format;
if ((ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) || (value != format)) {
DMA_CloseDevice(this);
SDL_SetError("Couldn't set audio format");
return 0;
}
/* Set mono or stereo audio (currently only two channels supported) */
stereo = (this->spec.channels > 1);
ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
if (stereo) {
this->spec.channels = 2;
} else {
this->spec.channels = 1;
}
/* Because some drivers don't allow setting the buffer size
after setting the format, we must re-open the audio device
once we know what format and channels are supported
*/
if (DMA_ReopenAudio(this, devname, format, stereo) < 0) {
DMA_CloseDevice(this);
/* Error is set by DMA_ReopenAudio() */
return 0;
}
/* Memory map the audio buffer */
if (ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &info) < 0) {
DMA_CloseDevice(this);
SDL_SetError("Couldn't get OSPACE parameters");
return 0;
}
this->spec.size = info.fragsize;
this->spec.samples = this->spec.size / ((this->spec.format & 0xFF) / 8);
this->spec.samples /= this->spec.channels;
num_buffers = info.fragstotal;
dma_len = num_buffers * this->spec.size;
dma_buf = (Uint8 *) mmap(NULL, dma_len, PROT_WRITE, MAP_SHARED,
audio_fd, 0);
if (dma_buf == MAP_FAILED) {
DMA_CloseDevice(this);
SDL_SetError("DMA memory map failed");
dma_buf = NULL;
return 0;
}
SDL_memset(dma_buf, this->spec.silence, dma_len);
/* Check to see if we need to use select() workaround */
{
char *workaround;
workaround = SDL_getenv("SDL_DSP_NOSELECT");
if (workaround) {
frame_ticks =
(float) (this->spec.samples * 1000) / this->spec.freq;
next_frame = SDL_GetTicks() + frame_ticks;
}
}
/* Trigger audio playback */
value = 0;
ioctl(audio_fd, SNDCTL_DSP_SETTRIGGER, &value);
value = PCM_ENABLE_OUTPUT;
if (ioctl(audio_fd, SNDCTL_DSP_SETTRIGGER, &value) < 0) {
DMA_CloseDevice(this);
SDL_SetError("Couldn't trigger audio output");
return 0;
}
/* Get the parent process id (we're the parent of the audio thread) */
parent = getpid();
/* We're ready to rock and roll. :-) */
return 1;
}
/* This function waits until it is possible to write a full sound buffer */
static void
DMA_WaitDevice(_THIS)
{
fd_set fdset;
/* Check to see if the thread-parent process is still alive */
{
static int cnt = 0;
/* Note that this only works with thread implementations
that use a different process id for each thread.
*/
if (parent && (((++cnt) % 10) == 0)) { /* Check every 10 loops */
if (kill(parent, 0) < 0 && errno == ESRCH) {
this->enabled = 0;
}
}
}
/* See if we need to use timed audio synchronization */
if (frame_ticks) {
/* Use timer for general audio synchronization */
Sint32 ticks;
ticks = ((Sint32) (next_frame - SDL_GetTicks())) - FUDGE_TICKS;
if (ticks > 0) {
SDL_Delay(ticks);
}
} else {
/* Use select() for audio synchronization */
struct timeval timeout;
FD_ZERO(&fdset);
FD_SET(audio_fd, &fdset);
timeout.tv_sec = 10;
timeout.tv_usec = 0;
#ifdef DEBUG_AUDIO
fprintf(stderr, "Waiting for audio to get ready\n");
#endif
if (select(audio_fd + 1, NULL, &fdset, NULL, &timeout) <= 0) {
const char *message =
#ifdef AUDIO_OSPACE_HACK
"Audio timeout - buggy audio driver? (trying ospace)";
#else
"Audio timeout - buggy audio driver? (disabled)";
#endif
/* In general we should never print to the screen,
but in this case we have no other way of letting
the user know what happened.
*/
fprintf(stderr, "SDL: %s\n", message);
#ifdef AUDIO_OSPACE_HACK
/* We may be able to use GET_OSPACE trick */
frame_ticks = (float) (this->spec.samples * 1000) /
this->spec.freq;
next_frame = SDL_GetTicks() + frame_ticks;
#else
this->enabled = 0;
/* Don't try to close - may hang */
audio_fd = -1;
#ifdef DEBUG_AUDIO
fprintf(stderr, "Done disabling audio\n");
#endif
#endif /* AUDIO_OSPACE_HACK */
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Ready!\n");
#endif
}
}
static void
DMA_PlayDevice(_THIS)
{
/* If timer synchronization is enabled, set the next write frame */
if (frame_ticks) {
next_frame += frame_ticks;
}
return;
}
static Uint8 *
DMA_GetDeviceBuf(_THIS)
{
count_info info;
int playing;
int filling;
/* Get number of blocks, looping if we're not using select() */
do {
if (ioctl(audio_fd, SNDCTL_DSP_GETOPTR, &info) < 0) {
/* Uh oh... */
this->enabled = 0;
return (NULL);
}
} while (frame_ticks && (info.blocks < 1));
#ifdef DEBUG_AUDIO
if (info.blocks > 1) {
printf("Warning: audio underflow (%d frags)\n", info.blocks - 1);
}
#endif
playing = info.ptr / this->spec.size;
filling = (playing + 1) % num_buffers;
return (dma_buf + (filling * this->spec.size));
}
static int
DMA_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->DetectDevices = DMA_DetectDevices;
impl->GetDeviceName = DMA_GetDeviceName;
impl->OpenDevice = DMA_OpenDevice;
impl->WaitDevice = DMA_WaitDevice;
impl->PlayDevice = DMA_PlayDevice;
impl->GetDeviceBuf = DMA_GetDeviceBuf;
impl->CloseDevice = DMA_CloseDevice;
impl->Deinitialize = DMA_Deinitialize;
build_device_lists();
return 1; /* this audio target is available. */
}
AudioBootStrap DMA_bootstrap = {
DMA_DRIVER_NAME, "OSS /dev/dsp DMA audio", DMA_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_dmaaudio_h
#define _SDL_dmaaudio_h
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
/* The file descriptor for the audio device */
int audio_fd;
/* The parent process id, to detect when application quits */
pid_t parent;
/* Raw mixing buffer */
Uint8 *dma_buf;
int dma_len;
int num_buffers;
/* Support for audio timing using a timer, in addition to select() */
float frame_ticks;
float next_frame;
};
#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
/* Old variable names */
/* !!! FIXME: remove these. */
#define audio_fd (this->hidden->audio_fd)
#define parent (this->hidden->parent)
#define dma_buf (this->hidden->dma_buf)
#define dma_len (this->hidden->dma_len)
#define num_buffers (this->hidden->num_buffers)
#define frame_ticks (this->hidden->frame_ticks)
#define next_frame (this->hidden->next_frame)
#endif /* _SDL_dmaaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

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project/jni/sdl-1.3/src/audio/dmedia/SDL_irixaudio.c Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include <errno.h>
#include "SDL_config.h"
/* Allow access to a raw mixing buffer (For IRIX 6.5 and higher) */
/* patch for IRIX 5 by Georg Schwarz 18/07/2004 */
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_irixaudio.h"
#ifndef AL_RESOURCE /* as a test whether we use the old IRIX audio libraries */
#define OLD_IRIX_AUDIO
#define alClosePort(x) ALcloseport(x)
#define alFreeConfig(x) ALfreeconfig(x)
#define alGetFillable(x) ALgetfillable(x)
#define alNewConfig() ALnewconfig()
#define alOpenPort(x,y,z) ALopenport(x,y,z)
#define alSetChannels(x,y) ALsetchannels(x,y)
#define alSetQueueSize(x,y) ALsetqueuesize(x,y)
#define alSetSampFmt(x,y) ALsetsampfmt(x,y)
#define alSetWidth(x,y) ALsetwidth(x,y)
#endif
void static
IRIXAUDIO_WaitDevice(_THIS)
{
Sint32 timeleft;
timeleft = this->spec.samples - alGetFillable(this->hidden->audio_port);
if (timeleft > 0) {
timeleft /= (this->spec.freq / 1000);
SDL_Delay((Uint32) timeleft);
}
}
static void
IRIXAUDIO_PlayDevice(_THIS)
{
/* Write the audio data out */
ALport port = this->hidden->audio_port;
Uint8 *mixbuf = this->hidden->mixbuf;
if (alWriteFrames(port, mixbuf, this->spec.samples) < 0) {
/* Assume fatal error, for now */
this->enabled = 0;
}
}
static Uint8 *
IRIXAUDIO_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
}
static void
IRIXAUDIO_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
if (this->hidden->audio_port != NULL) {
alClosePort(this->hidden->audio_port);
this->hidden->audio_port = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
static int
IRIXAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
{
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
long width = 0;
long fmt = 0;
int valid = 0;
/* !!! FIXME: Handle multiple devices and capture? */
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
#ifdef OLD_IRIX_AUDIO
{
long audio_param[2];
audio_param[0] = AL_OUTPUT_RATE;
audio_param[1] = this->spec.freq;
valid = (ALsetparams(AL_DEFAULT_DEVICE, audio_param, 2) < 0);
}
#else
{
ALpv audio_param;
audio_param.param = AL_RATE;
audio_param.value.i = this->spec.freq;
valid = (alSetParams(AL_DEFAULT_OUTPUT, &audio_param, 1) < 0);
}
#endif
while ((!valid) && (test_format)) {
valid = 1;
this->spec.format = test_format;
switch (test_format) {
case AUDIO_S8:
width = AL_SAMPLE_8;
fmt = AL_SAMPFMT_TWOSCOMP;
break;
case AUDIO_S16SYS:
width = AL_SAMPLE_16;
fmt = AL_SAMPFMT_TWOSCOMP;
break;
case AUDIO_F32SYS:
width = 0; /* not used here... */
fmt = AL_SAMPFMT_FLOAT;
break;
/* Docs say there is int24, but not int32.... */
default:
valid = 0;
test_format = SDL_NextAudioFormat();
break;
}
if (valid) {
ALconfig audio_config = alNewConfig();
valid = 0;
if (audio_config) {
if (alSetChannels(audio_config, this->spec.channels) < 0) {
if (this->spec.channels > 2) { /* can't handle > stereo? */
this->spec.channels = 2; /* try again below. */
}
}
if ((alSetSampFmt(audio_config, fmt) >= 0) &&
((!width) || (alSetWidth(audio_config, width) >= 0)) &&
(alSetQueueSize(audio_config, this->spec.samples * 2) >=
0)
&& (alSetChannels(audio_config, this->spec.channels) >=
0)) {
this->hidden->audio_port = alOpenPort("SDL audio", "w",
audio_config);
if (this->hidden->audio_port == NULL) {
/* docs say AL_BAD_CHANNELS happens here, too. */
int err = oserror();
if (err == AL_BAD_CHANNELS) {
this->spec.channels = 2;
alSetChannels(audio_config, this->spec.channels);
this->hidden->audio_port =
alOpenPort("SDL audio", "w", audio_config);
}
}
if (this->hidden->audio_port != NULL) {
valid = 1;
}
}
alFreeConfig(audio_config);
}
}
}
if (!valid) {
IRIXAUDIO_CloseDevice(this);
SDL_SetError("Unsupported audio format");
return 0;
}
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* Allocate mixing buffer */
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->spec.size);
if (this->hidden->mixbuf == NULL) {
IRIXAUDIO_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/* We're ready to rock and roll. :-) */
return 1;
}
static int
IRIXAUDIO_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->OpenDevice = DSP_OpenDevice;
impl->PlayDevice = DSP_PlayDevice;
impl->WaitDevice = DSP_WaitDevice;
impl->GetDeviceBuf = DSP_GetDeviceBuf;
impl->CloseDevice = DSP_CloseDevice;
impl->OnlyHasDefaultOutputDevice = 1; /* !!! FIXME: not true, I think. */
return 1; /* this audio target is available. */
}
AudioBootStrap IRIXAUDIO_bootstrap = {
"AL", "IRIX DMedia audio", IRIXAUDIO_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_irixaudio_h
#define _SDL_irixaudio_h
#include <dmedia/audio.h>
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
ALport audio_port; /* The handle for the audio device */
Uint8 *mixbuf; /* The app mixing buffer */
};
#endif /* _SDL_irixaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 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
Modified in Oct 2004 by Hannu Savolainen
hannu@opensound.com
*/
#include "SDL_config.h"
/* Allow access to a raw mixing buffer */
#include <stdio.h> /* For perror() */
#include <string.h> /* For strerror() */
#include <errno.h>
#include <unistd.h>
#include <fcntl.h>
#include <signal.h>
#include <sys/time.h>
#include <sys/ioctl.h>
#include <sys/stat.h>
#if SDL_AUDIO_DRIVER_OSS_SOUNDCARD_H
/* This is installed on some systems */
#include <soundcard.h>
#else
/* This is recommended by OSS */
#include <sys/soundcard.h>
#endif
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "../SDL_audiodev_c.h"
#include "SDL_dspaudio.h"
/* The tag name used by DSP audio */
#define DSP_DRIVER_NAME "dsp"
/* Open the audio device for playback, and don't block if busy */
#define OPEN_FLAGS_OUTPUT (O_WRONLY|O_NONBLOCK)
#define OPEN_FLAGS_INPUT (O_RDONLY|O_NONBLOCK)
static char **outputDevices = NULL;
static int outputDeviceCount = 0;
static char **inputDevices = NULL;
static int inputDeviceCount = 0;
static inline void
free_device_list(char ***devs, int *count)
{
SDL_FreeUnixAudioDevices(devs, count);
}
static inline void
build_device_list(int iscapture, char ***devs, int *count)
{
const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
free_device_list(devs, count);
SDL_EnumUnixAudioDevices(flags, 0, NULL, devs, count);
}
static inline void
build_device_lists(void)
{
build_device_list(0, &outputDevices, &outputDeviceCount);
build_device_list(1, &inputDevices, &inputDeviceCount);
}
static inline void
free_device_lists(void)
{
free_device_list(&outputDevices, &outputDeviceCount);
free_device_list(&inputDevices, &inputDeviceCount);
}
static void
DSP_Deinitialize(void)
{
free_device_lists();
}
static int
DSP_DetectDevices(int iscapture)
{
if (iscapture) {
build_device_list(1, &inputDevices, &inputDeviceCount);
return inputDeviceCount;
} else {
build_device_list(0, &outputDevices, &outputDeviceCount);
return outputDeviceCount;
}
return 0; /* shouldn't ever hit this. */
}
static const char *
DSP_GetDeviceName(int index, int iscapture)
{
if ((iscapture) && (index < inputDeviceCount)) {
return inputDevices[index];
} else if ((!iscapture) && (index < outputDeviceCount)) {
return outputDevices[index];
}
SDL_SetError("No such device");
return NULL;
}
static void
DSP_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
if (this->hidden->audio_fd >= 0) {
close(this->hidden->audio_fd);
this->hidden->audio_fd = -1;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
static int
DSP_OpenDevice(_THIS, const char *devname, int iscapture)
{
const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
int format;
int value;
int frag_spec;
SDL_AudioFormat test_format;
/* We don't care what the devname is...we'll try to open anything. */
/* ...but default to first name in the list... */
if (devname == NULL) {
if (((iscapture) && (inputDeviceCount == 0)) ||
((!iscapture) && (outputDeviceCount == 0))) {
SDL_SetError("No such audio device");
return 0;
}
devname = ((iscapture) ? inputDevices[0] : outputDevices[0]);
}
/* Make sure fragment size stays a power of 2, or OSS fails. */
/* I don't know which of these are actually legal values, though... */
if (this->spec.channels > 8)
this->spec.channels = 8;
else if (this->spec.channels > 4)
this->spec.channels = 4;
else if (this->spec.channels > 2)
this->spec.channels = 2;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Open the audio device */
this->hidden->audio_fd = open(devname, flags, 0);
if (this->hidden->audio_fd < 0) {
DSP_CloseDevice(this);
SDL_SetError("Couldn't open %s: %s", devname, strerror(errno));
return 0;
}
this->hidden->mixbuf = NULL;
/* Make the file descriptor use blocking writes with fcntl() */
{
long ctlflags;
ctlflags = fcntl(this->hidden->audio_fd, F_GETFL);
ctlflags &= ~O_NONBLOCK;
if (fcntl(this->hidden->audio_fd, F_SETFL, ctlflags) < 0) {
DSP_CloseDevice(this);
SDL_SetError("Couldn't set audio blocking mode");
return 0;
}
}
/* Get a list of supported hardware formats */
if (ioctl(this->hidden->audio_fd, SNDCTL_DSP_GETFMTS, &value) < 0) {
perror("SNDCTL_DSP_GETFMTS");
DSP_CloseDevice(this);
SDL_SetError("Couldn't get audio format list");
return 0;
}
/* Try for a closest match on audio format */
format = 0;
for (test_format = SDL_FirstAudioFormat(this->spec.format);
!format && test_format;) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
switch (test_format) {
case AUDIO_U8:
if (value & AFMT_U8) {
format = AFMT_U8;
}
break;
case AUDIO_S16LSB:
if (value & AFMT_S16_LE) {
format = AFMT_S16_LE;
}
break;
case AUDIO_S16MSB:
if (value & AFMT_S16_BE) {
format = AFMT_S16_BE;
}
break;
#if 0
/*
* These formats are not used by any real life systems so they are not
* needed here.
*/
case AUDIO_S8:
if (value & AFMT_S8) {
format = AFMT_S8;
}
break;
case AUDIO_U16LSB:
if (value & AFMT_U16_LE) {
format = AFMT_U16_LE;
}
break;
case AUDIO_U16MSB:
if (value & AFMT_U16_BE) {
format = AFMT_U16_BE;
}
break;
#endif
default:
format = 0;
break;
}
if (!format) {
test_format = SDL_NextAudioFormat();
}
}
if (format == 0) {
DSP_CloseDevice(this);
SDL_SetError("Couldn't find any hardware audio formats");
return 0;
}
this->spec.format = test_format;
/* Set the audio format */
value = format;
if ((ioctl(this->hidden->audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) ||
(value != format)) {
perror("SNDCTL_DSP_SETFMT");
DSP_CloseDevice(this);
SDL_SetError("Couldn't set audio format");
return 0;
}
/* Set the number of channels of output */
value = this->spec.channels;
if (ioctl(this->hidden->audio_fd, SNDCTL_DSP_CHANNELS, &value) < 0) {
perror("SNDCTL_DSP_CHANNELS");
DSP_CloseDevice(this);
SDL_SetError("Cannot set the number of channels");
return 0;
}
this->spec.channels = value;
/* Set the DSP frequency */
value = this->spec.freq;
if (ioctl(this->hidden->audio_fd, SNDCTL_DSP_SPEED, &value) < 0) {
perror("SNDCTL_DSP_SPEED");
DSP_CloseDevice(this);
SDL_SetError("Couldn't set audio frequency");
return 0;
}
this->spec.freq = value;
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
/* Determine the power of two of the fragment size */
for (frag_spec = 0; (0x01U << frag_spec) < this->spec.size; ++frag_spec);
if ((0x01U << frag_spec) != this->spec.size) {
DSP_CloseDevice(this);
SDL_SetError("Fragment size must be a power of two");
return 0;
}
frag_spec |= 0x00020000; /* two fragments, for low latency */
/* Set the audio buffering parameters */
#ifdef DEBUG_AUDIO
fprintf(stderr, "Requesting %d fragments of size %d\n",
(frag_spec >> 16), 1 << (frag_spec & 0xFFFF));
#endif
if (ioctl(this->hidden->audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_spec) < 0) {
perror("SNDCTL_DSP_SETFRAGMENT");
}
#ifdef DEBUG_AUDIO
{
audio_buf_info info;
ioctl(this->hidden->audio_fd, SNDCTL_DSP_GETOSPACE, &info);
fprintf(stderr, "fragments = %d\n", info.fragments);
fprintf(stderr, "fragstotal = %d\n", info.fragstotal);
fprintf(stderr, "fragsize = %d\n", info.fragsize);
fprintf(stderr, "bytes = %d\n", info.bytes);
}
#endif
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
DSP_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/* We're ready to rock and roll. :-) */
return 1;
}
static void
DSP_PlayDevice(_THIS)
{
const Uint8 *mixbuf = this->hidden->mixbuf;
const int mixlen = this->hidden->mixlen;
if (write(this->hidden->audio_fd, mixbuf, mixlen) == -1) {
perror("Audio write");
this->enabled = 0;
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", mixlen);
#endif
}
static Uint8 *
DSP_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
}
static int
DSP_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->DetectDevices = DSP_DetectDevices;
impl->GetDeviceName = DSP_GetDeviceName;
impl->OpenDevice = DSP_OpenDevice;
impl->PlayDevice = DSP_PlayDevice;
impl->GetDeviceBuf = DSP_GetDeviceBuf;
impl->CloseDevice = DSP_CloseDevice;
impl->Deinitialize = DSP_Deinitialize;
build_device_lists();
return 1; /* this audio target is available. */
}
AudioBootStrap DSP_bootstrap = {
DSP_DRIVER_NAME, "OSS /dev/dsp standard audio", DSP_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_dspaudio_h
#define _SDL_dspaudio_h
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
/* The file descriptor for the audio device */
int audio_fd;
/* Raw mixing buffer */
Uint8 *mixbuf;
int mixlen;
};
#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
#endif /* _SDL_dspaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 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
This file written by Ryan C. Gordon (icculus@icculus.org)
*/
#include "SDL_config.h"
/* Output audio to nowhere... */
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "SDL_dummyaudio.h"
static int
DUMMYAUD_OpenDevice(_THIS, const char *devname, int iscapture)
{
return 1; /* always succeeds. */
}
static int
DUMMYAUD_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->OpenDevice = DUMMYAUD_OpenDevice;
impl->OnlyHasDefaultOutputDevice = 1;
return 1; /* this audio target is available. */
}
AudioBootStrap DUMMYAUD_bootstrap = {
"dummy", "SDL dummy audio driver", DUMMYAUD_Init, 1
};
/* vi: set ts=4 sw=4 expandtab: */

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_dummyaudio_h
#define _SDL_dummyaudio_h
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
/* The file descriptor for the audio device */
Uint8 *mixbuf;
Uint32 mixlen;
Uint32 write_delay;
Uint32 initial_calls;
};
#endif /* _SDL_dummyaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* Allow access to an ESD network stream mixing buffer */
#include <sys/types.h>
#include <unistd.h>
#include <signal.h>
#include <errno.h>
#include <esd.h>
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_esdaudio.h"
#ifdef SDL_AUDIO_DRIVER_ESD_DYNAMIC
#include "SDL_name.h"
#include "SDL_loadso.h"
#else
#define SDL_NAME(X) X
#endif
/* The tag name used by ESD audio */
#define ESD_DRIVER_NAME "esd"
#ifdef SDL_AUDIO_DRIVER_ESD_DYNAMIC
static const char *esd_library = SDL_AUDIO_DRIVER_ESD_DYNAMIC;
static void *esd_handle = NULL;
static int (*SDL_NAME(esd_open_sound)) (const char *host);
static int (*SDL_NAME(esd_close)) (int esd);
static int (*SDL_NAME(esd_play_stream)) (esd_format_t format, int rate,
const char *host, const char *name);
#define SDL_ESD_SYM(x) { #x, (void **) (char *) &SDL_NAME(x) }
static struct
{
const char *name;
void **func;
} const esd_functions[] = {
SDL_ESD_SYM(esd_open_sound),
SDL_ESD_SYM(esd_close), SDL_ESD_SYM(esd_play_stream),
};
#undef SDL_ESD_SYM
static void
UnloadESDLibrary()
{
if (esd_handle != NULL) {
SDL_UnloadObject(esd_handle);
esd_handle = NULL;
}
}
static int
LoadESDLibrary(void)
{
int i, retval = -1;
if (esd_handle == NULL) {
esd_handle = SDL_LoadObject(esd_library);
if (esd_handle) {
retval = 0;
for (i = 0; i < SDL_arraysize(esd_functions); ++i) {
*esd_functions[i].func =
SDL_LoadFunction(esd_handle, esd_functions[i].name);
if (!*esd_functions[i].func) {
retval = -1;
UnloadESDLibrary();
break;
}
}
}
}
return retval;
}
#else
static void
UnloadESDLibrary()
{
return;
}
static int
LoadESDLibrary(void)
{
return 0;
}
#endif /* SDL_AUDIO_DRIVER_ESD_DYNAMIC */
/* This function waits until it is possible to write a full sound buffer */
static void
ESD_WaitDevice(_THIS)
{
Sint32 ticks;
/* Check to see if the thread-parent process is still alive */
{
static int cnt = 0;
/* Note that this only works with thread implementations
that use a different process id for each thread.
*/
/* Check every 10 loops */
if (this->hidden->parent && (((++cnt) % 10) == 0)) {
if (kill(this->hidden->parent, 0) < 0 && errno == ESRCH) {
this->enabled = 0;
}
}
}
/* Use timer for general audio synchronization */
ticks =
((Sint32) (this->hidden->next_frame - SDL_GetTicks())) - FUDGE_TICKS;
if (ticks > 0) {
SDL_Delay(ticks);
}
}
static void
ESD_PlayDevice(_THIS)
{
int written = 0;
/* Write the audio data, checking for EAGAIN on broken audio drivers */
do {
written = write(this->hidden->audio_fd,
this->hidden->mixbuf, this->hidden->mixlen);
if ((written < 0) && ((errno == 0) || (errno == EAGAIN))) {
SDL_Delay(1); /* Let a little CPU time go by */
}
} while ((written < 0) &&
((errno == 0) || (errno == EAGAIN) || (errno == EINTR)));
/* Set the next write frame */
this->hidden->next_frame += this->hidden->frame_ticks;
/* If we couldn't write, assume fatal error for now */
if (written < 0) {
this->enabled = 0;
}
}
static Uint8 *
ESD_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
}
static void
ESD_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
if (this->hidden->audio_fd >= 0) {
SDL_NAME(esd_close) (this->hidden->audio_fd);
this->hidden->audio_fd = -1;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
/* Try to get the name of the program */
static char *
get_progname(void)
{
char *progname = NULL;
#ifdef __LINUX__
FILE *fp;
static char temp[BUFSIZ];
SDL_snprintf(temp, SDL_arraysize(temp), "/proc/%d/cmdline", getpid());
fp = fopen(temp, "r");
if (fp != NULL) {
if (fgets(temp, sizeof(temp) - 1, fp)) {
progname = SDL_strrchr(temp, '/');
if (progname == NULL) {
progname = temp;
} else {
progname = progname + 1;
}
}
fclose(fp);
}
#endif
return (progname);
}
static int
ESD_OpenDevice(_THIS, const char *devname, int iscapture)
{
esd_format_t format = (ESD_STREAM | ESD_PLAY);
SDL_AudioFormat test_format = 0;
int found = 0;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
this->hidden->audio_fd = -1;
/* Convert audio spec to the ESD audio format */
/* Try for a closest match on audio format */
for (test_format = SDL_FirstAudioFormat(this->spec.format);
!found && test_format; test_format = SDL_NextAudioFormat()) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
found = 1;
switch (test_format) {
case AUDIO_U8:
format |= ESD_BITS8;
break;
case AUDIO_S16SYS:
format |= ESD_BITS16;
break;
default:
found = 0;
break;
}
}
if (!found) {
ESD_CloseDevice(this);
SDL_SetError("Couldn't find any hardware audio formats");
return 0;
}
if (this->spec.channels == 1) {
format |= ESD_MONO;
} else {
format |= ESD_STEREO;
}
#if 0
this->spec.samples = ESD_BUF_SIZE; /* Darn, no way to change this yet */
#endif
/* Open a connection to the ESD audio server */
this->hidden->audio_fd =
SDL_NAME(esd_play_stream) (format, this->spec.freq, NULL,
get_progname());
if (this->hidden->audio_fd < 0) {
ESD_CloseDevice(this);
SDL_SetError("Couldn't open ESD connection");
return 0;
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
this->hidden->frame_ticks =
(float) (this->spec.samples * 1000) / this->spec.freq;
this->hidden->next_frame = SDL_GetTicks() + this->hidden->frame_ticks;
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
ESD_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/* Get the parent process id (we're the parent of the audio thread) */
this->hidden->parent = getpid();
/* We're ready to rock and roll. :-) */
return 1;
}
static void
ESD_Deinitialize(void)
{
UnloadESDLibrary();
}
static int
ESD_Init(SDL_AudioDriverImpl * impl)
{
if (LoadESDLibrary() < 0) {
return 0;
} else {
int connection = 0;
/* Don't start ESD if it's not running */
SDL_setenv("ESD_NO_SPAWN", "1", 0);
connection = SDL_NAME(esd_open_sound) (NULL);
if (connection < 0) {
UnloadESDLibrary();
SDL_SetError("ESD: esd_open_sound failed (no audio server?)");
return 0;
}
SDL_NAME(esd_close) (connection);
}
/* Set the function pointers */
impl->OpenDevice = ESD_OpenDevice;
impl->PlayDevice = ESD_PlayDevice;
impl->WaitDevice = ESD_WaitDevice;
impl->GetDeviceBuf = ESD_GetDeviceBuf;
impl->CloseDevice = ESD_CloseDevice;
impl->Deinitialize = ESD_Deinitialize;
impl->OnlyHasDefaultOutputDevice = 1;
return 1; /* this audio target is available. */
}
AudioBootStrap ESD_bootstrap = {
ESD_DRIVER_NAME, "Enlightened Sound Daemon", ESD_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

51
project/jni/sdl-1.3/src/audio/esd/SDL_esdaudio.h Normal file
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@@ -0,0 +1,51 @@
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_esdaudio_h
#define _SDL_esdaudio_h
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
/* The file descriptor for the audio device */
int audio_fd;
/* The parent process id, to detect when application quits */
pid_t parent;
/* Raw mixing buffer */
Uint8 *mixbuf;
int mixlen;
/* Support for audio timing using a timer */
float frame_ticks;
float next_frame;
};
#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
#endif /* _SDL_esdaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

351
project/jni/sdl-1.3/src/audio/fusionsound/SDL_fsaudio.c Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* Allow access to a raw mixing buffer */
#ifdef HAVE_SIGNAL_H
#include <signal.h>
#endif
#include <unistd.h>
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_fsaudio.h"
#include <fusionsound/fusionsound_version.h>
//#define SDL_AUDIO_DRIVER_FUSIONSOUND_DYNAMIC "libfusionsound.so"
#ifdef SDL_AUDIO_DRIVER_FUSIONSOUND_DYNAMIC
#include "SDL_name.h"
#include "SDL_loadso.h"
#else
#define SDL_NAME(X) X
#endif
#if (FUSIONSOUND_MAJOR_VERSION == 1) && (FUSIONSOUND_MINOR_VERSION < 1)
typedef DFBResult DirectResult;
#endif
/* The tag name used by fusionsoundc audio */
#define SDL_FS_DRIVER_NAME "fusionsound"
/* Buffers to use - more than 2 gives a lot of latency */
#define FUSION_BUFFERS (2)
#ifdef SDL_AUDIO_DRIVER_FUSIONSOUND_DYNAMIC
static const char *fs_library = SDL_AUDIO_DRIVER_FUSIONSOUND_DYNAMIC;
static void *fs_handle = NULL;
static DirectResult (*SDL_NAME(FusionSoundInit)) (int *argc, char *(*argv[]));
static DirectResult (*SDL_NAME(FusionSoundCreate)) (IFusionSound **
ret_interface);
#define SDL_FS_SYM(x) { #x, (void **) (char *) &SDL_NAME(x) }
static struct
{
const char *name;
void **func;
} fs_functions[] = {
/* *INDENT-OFF* */
SDL_FS_SYM(FusionSoundInit),
SDL_FS_SYM(FusionSoundCreate),
/* *INDENT-ON* */
};
#undef SDL_FS_SYM
static void
UnloadFusionSoundLibrary()
{
if (fs_handle != NULL) {
SDL_UnloadObject(fs_handle);
fs_handle = NULL;
}
}
static int
LoadFusionSoundLibrary(void)
{
int i, retval = -1;
if (fs_handle == NULL) {
fs_handle = SDL_LoadObject(fs_library);
if (fs_handle != NULL) {
retval = 0;
for (i = 0; i < SDL_arraysize(fs_functions); ++i) {
*fs_functions[i].func =
SDL_LoadFunction(fs_handle, fs_functions[i].name);
if (!*fs_functions[i].func) {
retval = -1;
UnloadFusionSoundLibrary();
break;
}
}
}
}
return retval;
}
#else
static void
UnloadFusionSoundLibrary()
{
return;
}
static int
LoadFusionSoundLibrary(void)
{
return 0;
}
#endif /* SDL_AUDIO_DRIVER_FUSIONSOUND_DYNAMIC */
/* This function waits until it is possible to write a full sound buffer */
static void
SDL_FS_WaitDevice(_THIS)
{
this->hidden->stream->Wait(this->hidden->stream,
this->hidden->mixsamples);
}
static void
SDL_FS_PlayDevice(_THIS)
{
DirectResult ret;
ret = this->hidden->stream->Write(this->hidden->stream,
this->hidden->mixbuf,
this->hidden->mixsamples);
/* If we couldn't write, assume fatal error for now */
if (ret) {
this->enabled = 0;
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", this->hidden->mixlen);
#endif
}
static void
SDL_FS_WaitDone(_THIS)
{
this->hidden->stream->Wait(this->hidden->stream,
this->hidden->mixsamples * FUSION_BUFFERS);
}
static Uint8 *
SDL_FS_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
}
static void
SDL_FS_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
if (this->hidden->stream) {
this->hidden->stream->Release(this->hidden->stream);
this->hidden->stream = NULL;
}
if (this->hidden->fs) {
this->hidden->fs->Release(this->hidden->fs);
this->hidden->fs = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
static int
SDL_FS_OpenDevice(_THIS, const char *devname, int iscapture)
{
int bytes;
SDL_AudioFormat test_format = 0, format = 0;
FSSampleFormat fs_format;
FSStreamDescription desc;
DirectResult ret;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Try for a closest match on audio format */
for (test_format = SDL_FirstAudioFormat(this->spec.format);
!format && test_format;) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
switch (test_format) {
case AUDIO_U8:
fs_format = FSSF_U8;
bytes = 1;
format = 1;
break;
case AUDIO_S16SYS:
fs_format = FSSF_S16;
bytes = 2;
format = 1;
break;
case AUDIO_S32SYS:
fs_format = FSSF_S32;
bytes = 4;
format = 1;
break;
case AUDIO_F32SYS:
fs_format = FSSF_FLOAT;
bytes = 4;
format = 1;
break;
default:
format = 0;
break;
}
if (!format) {
test_format = SDL_NextAudioFormat();
}
}
if (format == 0) {
SDL_FS_CloseDevice(this);
SDL_SetError("Couldn't find any hardware audio formats");
return 0;
}
this->spec.format = test_format;
/* Retrieve the main sound interface. */
ret = SDL_NAME(FusionSoundCreate) (&this->hidden->fs);
if (ret) {
SDL_FS_CloseDevice(this);
SDL_SetError("Unable to initialize FusionSound: %d", ret);
return 0;
}
this->hidden->mixsamples = this->spec.size / bytes / this->spec.channels;
/* Fill stream description. */
desc.flags = FSSDF_SAMPLERATE | FSSDF_BUFFERSIZE |
FSSDF_CHANNELS | FSSDF_SAMPLEFORMAT | FSSDF_PREBUFFER;
desc.samplerate = this->spec.freq;
desc.buffersize = this->spec.size * FUSION_BUFFERS;
desc.channels = this->spec.channels;
desc.prebuffer = 10;
desc.sampleformat = fs_format;
ret =
this->hidden->fs->CreateStream(this->hidden->fs, &desc,
&this->hidden->stream);
if (ret) {
SDL_FS_CloseDevice(this);
SDL_SetError("Unable to create FusionSoundStream: %d", ret);
return 0;
}
/* See what we got */
desc.flags = FSSDF_SAMPLERATE | FSSDF_BUFFERSIZE |
FSSDF_CHANNELS | FSSDF_SAMPLEFORMAT;
ret = this->hidden->stream->GetDescription(this->hidden->stream, &desc);
this->spec.freq = desc.samplerate;
this->spec.size =
desc.buffersize / FUSION_BUFFERS * bytes * desc.channels;
this->spec.channels = desc.channels;
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
SDL_FS_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/* We're ready to rock and roll. :-) */
return 1;
}
static void
SDL_FS_Deinitialize(void)
{
UnloadFusionSoundLibrary();
}
static int
SDL_FS_Init(SDL_AudioDriverImpl * impl)
{
if (LoadFusionSoundLibrary() < 0) {
return 0;
} else {
DirectResult ret;
ret = SDL_NAME(FusionSoundInit) (NULL, NULL);
if (ret) {
UnloadFusionSoundLibrary();
SDL_SetError
("FusionSound: SDL_FS_init failed (FusionSoundInit: %d)",
ret);
return 0;
}
}
/* Set the function pointers */
impl->OpenDevice = SDL_FS_OpenDevice;
impl->PlayDevice = SDL_FS_PlayDevice;
impl->WaitDevice = SDL_FS_WaitDevice;
impl->GetDeviceBuf = SDL_FS_GetDeviceBuf;
impl->CloseDevice = SDL_FS_CloseDevice;
impl->WaitDone = SDL_FS_WaitDone;
impl->Deinitialize = SDL_FS_Deinitialize;
impl->OnlyHasDefaultOutputDevice = 1;
return 1; /* this audio target is available. */
}
AudioBootStrap FUSIONSOUND_bootstrap = {
SDL_FS_DRIVER_NAME, "FusionSound", SDL_FS_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

50
project/jni/sdl-1.3/src/audio/fusionsound/SDL_fsaudio.h Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_fsaudio_h
#define _SDL_fsaudio_h
#include <fusionsound/fusionsound.h>
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
/* Interface */
IFusionSound *fs;
/* The stream interface for the audio device */
IFusionSoundStream *stream;
/* Raw mixing buffer */
Uint8 *mixbuf;
int mixlen;
int mixsamples;
};
#endif /* _SDL_fsaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

340
project/jni/sdl-1.3/src/audio/iphoneos/SDL_coreaudio_iphone.c Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#include <AudioUnit/AudioUnit.h>
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "../SDL_sysaudio.h"
#include "SDL_coreaudio_iphone.h"
#define DEBUG_COREAUDIO 0
static void
COREAUDIO_Deinitialize(void)
{
}
/* The CoreAudio callback */
static OSStatus
outputCallback(void *inRefCon,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp,
UInt32 inBusNumber, UInt32 inNumberFrames,
AudioBufferList * ioDataList)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) inRefCon;
AudioBuffer *ioData = &ioDataList->mBuffers[0];
UInt32 remaining, len;
void *ptr;
/* Is there ever more than one buffer, and what do you do with it? */
if (ioDataList->mNumberBuffers != 1) {
return noErr;
}
/* Only do anything if audio is enabled and not paused */
if (!this->enabled || this->paused) {
SDL_memset(ioData->mData, this->spec.silence, ioData->mDataByteSize);
return 0;
}
/* No SDL conversion should be needed here, ever, since we accept
any input format in OpenAudio, and leave the conversion to CoreAudio.
*/
/*
assert(!this->convert.needed);
assert(this->spec.channels == ioData->mNumberChannels);
*/
remaining = ioData->mDataByteSize;
ptr = ioData->mData;
while (remaining > 0) {
if (this->hidden->bufferOffset >= this->hidden->bufferSize) {
/* Generate the data */
SDL_memset(this->hidden->buffer, this->spec.silence,
this->hidden->bufferSize);
SDL_mutexP(this->mixer_lock);
(*this->spec.callback) (this->spec.userdata, this->hidden->buffer,
this->hidden->bufferSize);
SDL_mutexV(this->mixer_lock);
this->hidden->bufferOffset = 0;
}
len = this->hidden->bufferSize - this->hidden->bufferOffset;
if (len > remaining)
len = remaining;
SDL_memcpy(ptr,
(char *) this->hidden->buffer + this->hidden->bufferOffset,
len);
ptr = (char *) ptr + len;
remaining -= len;
this->hidden->bufferOffset += len;
}
return 0;
}
static OSStatus
inputCallback(void *inRefCon,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp,
UInt32 inBusNumber, UInt32 inNumberFrames,
AudioBufferList * ioData)
{
//err = AudioUnitRender(afr->fAudioUnit, ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, afr->fAudioBuffer);
// !!! FIXME: write me!
return noErr;
}
static void
COREAUDIO_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->audioUnitOpened) {
OSStatus result = noErr;
AURenderCallbackStruct callback;
const AudioUnitElement output_bus = 0;
const AudioUnitElement input_bus = 1;
const int iscapture = this->iscapture;
const AudioUnitElement bus =
((iscapture) ? input_bus : output_bus);
const AudioUnitScope scope =
((iscapture) ? kAudioUnitScope_Output :
kAudioUnitScope_Input);
/* stop processing the audio unit */
result = AudioOutputUnitStop(this->hidden->audioUnit);
/* Remove the input callback */
SDL_memset(&callback, '\0', sizeof(AURenderCallbackStruct));
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioUnitProperty_SetRenderCallback,
scope, bus, &callback,
sizeof(callback));
//CloseComponent(this->hidden->audioUnit);
this->hidden->audioUnitOpened = 0;
}
SDL_free(this->hidden->buffer);
SDL_free(this->hidden);
this->hidden = NULL;
}
}
#define CHECK_RESULT(msg) \
if (result != noErr) { \
COREAUDIO_CloseDevice(this); \
SDL_SetError("CoreAudio error (%s): %d", msg, result); \
return 0; \
}
static int
prepare_audiounit(_THIS, const char *devname, int iscapture,
const AudioStreamBasicDescription * strdesc)
{
OSStatus result = noErr;
AURenderCallbackStruct callback;
AudioComponentDescription desc;
AudioComponent comp = NULL;
UInt32 enableIO = 0;
const AudioUnitElement output_bus = 0;
const AudioUnitElement input_bus = 1;
const AudioUnitElement bus = ((iscapture) ? input_bus : output_bus);
const AudioUnitScope scope = ((iscapture) ? kAudioUnitScope_Output :
kAudioUnitScope_Input);
SDL_memset(&desc, '\0', sizeof(AudioComponentDescription));
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_RemoteIO;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
comp = AudioComponentFindNext(NULL, &desc);
if (comp == NULL) {
SDL_SetError("Couldn't find requested CoreAudio component");
return 0;
}
/* Open & initialize the audio unit */
/*
AudioComponentInstanceNew only available on iPhone OS 2.0 and Mac OS X 10.6
We can't use OpenAComponent on iPhone because it is not present
*/
result = AudioComponentInstanceNew(comp, &this->hidden->audioUnit);
CHECK_RESULT("AudioComponentInstanceNew");
this->hidden->audioUnitOpened = 1;
// !!! FIXME: this is wrong?
enableIO = ((iscapture) ? 1 : 0);
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input, input_bus,
&enableIO, sizeof(enableIO));
CHECK_RESULT("AudioUnitSetProperty (kAudioUnitProperty_EnableIO input)");
// !!! FIXME: this is wrong?
enableIO = ((iscapture) ? 0 : 1);
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output, output_bus,
&enableIO, sizeof(enableIO));
CHECK_RESULT("AudioUnitSetProperty (kAudioUnitProperty_EnableIO output)");
/*result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioOutputUnitProperty_CurrentDevice,
kAudioUnitScope_Global, 0,
&this->hidden->deviceID,
sizeof(AudioDeviceID));
CHECK_RESULT("AudioUnitSetProperty (kAudioOutputUnitProperty_CurrentDevice)"); */
/* Set the data format of the audio unit. */
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioUnitProperty_StreamFormat,
scope, bus, strdesc, sizeof(*strdesc));
CHECK_RESULT("AudioUnitSetProperty (kAudioUnitProperty_StreamFormat)");
/* Set the audio callback */
SDL_memset(&callback, '\0', sizeof(AURenderCallbackStruct));
callback.inputProc = ((iscapture) ? inputCallback : outputCallback);
callback.inputProcRefCon = this;
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioUnitProperty_SetRenderCallback,
scope, bus, &callback, sizeof(callback));
CHECK_RESULT
("AudioUnitSetProperty (kAudioUnitProperty_SetInputCallback)");
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
/* Allocate a sample buffer */
this->hidden->bufferOffset = this->hidden->bufferSize = this->spec.size;
this->hidden->buffer = SDL_malloc(this->hidden->bufferSize);
result = AudioUnitInitialize(this->hidden->audioUnit);
CHECK_RESULT("AudioUnitInitialize");
/* Finally, start processing of the audio unit */
result = AudioOutputUnitStart(this->hidden->audioUnit);
CHECK_RESULT("AudioOutputUnitStart");
/* We're running! */
return 1;
}
static int
COREAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
{
AudioStreamBasicDescription strdesc;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
int valid_datatype = 0;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return (0);
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Setup a AudioStreamBasicDescription with the requested format */
SDL_memset(&strdesc, '\0', sizeof(AudioStreamBasicDescription));
strdesc.mFormatID = kAudioFormatLinearPCM;
strdesc.mFormatFlags = kLinearPCMFormatFlagIsPacked;
strdesc.mChannelsPerFrame = this->spec.channels;
strdesc.mSampleRate = this->spec.freq;
strdesc.mFramesPerPacket = 1;
while ((!valid_datatype) && (test_format)) {
this->spec.format = test_format;
/* Just a list of valid SDL formats, so people don't pass junk here. */
switch (test_format) {
case AUDIO_U8:
case AUDIO_S8:
case AUDIO_U16LSB:
case AUDIO_S16LSB:
case AUDIO_U16MSB:
case AUDIO_S16MSB:
case AUDIO_S32LSB:
case AUDIO_S32MSB:
case AUDIO_F32LSB:
case AUDIO_F32MSB:
valid_datatype = 1;
strdesc.mBitsPerChannel = SDL_AUDIO_BITSIZE(this->spec.format);
if (SDL_AUDIO_ISBIGENDIAN(this->spec.format))
strdesc.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
if (SDL_AUDIO_ISFLOAT(this->spec.format))
strdesc.mFormatFlags |= kLinearPCMFormatFlagIsFloat;
else if (SDL_AUDIO_ISSIGNED(this->spec.format))
strdesc.mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
break;
}
}
if (!valid_datatype) { /* shouldn't happen, but just in case... */
COREAUDIO_CloseDevice(this);
SDL_SetError("Unsupported audio format");
return 0;
}
strdesc.mBytesPerFrame =
strdesc.mBitsPerChannel * strdesc.mChannelsPerFrame / 8;
strdesc.mBytesPerPacket =
strdesc.mBytesPerFrame * strdesc.mFramesPerPacket;
if (!prepare_audiounit(this, devname, iscapture, &strdesc)) {
COREAUDIO_CloseDevice(this);
return 0; /* prepare_audiounit() will call SDL_SetError()... */
}
return 1; /* good to go. */
}
static int
COREAUDIO_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->OpenDevice = COREAUDIO_OpenDevice;
impl->CloseDevice = COREAUDIO_CloseDevice;
impl->Deinitialize = COREAUDIO_Deinitialize;
impl->ProvidesOwnCallbackThread = 1;
/* added for iPhone */
impl->OnlyHasDefaultInputDevice = 1;
impl->OnlyHasDefaultOutputDevice = 1;
impl->HasCaptureSupport = 0; /* still needs to be written */
return 1; /* this audio target is available. */
}
AudioBootStrap COREAUDIOIPHONE_bootstrap = {
"coreaudio-iphoneos", "SDL CoreAudio (iPhone OS) audio driver",
COREAUDIO_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

43
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_coreaudio_h
#define _SDL_coreaudio_h
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
AudioUnit audioUnit;
int audioUnitOpened;
void *buffer;
UInt32 bufferOffset;
UInt32 bufferSize;
// AudioDeviceID deviceID;
};
#endif /* _SDL_coreaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#include <CoreAudio/CoreAudio.h>
#include <CoreServices/CoreServices.h>
#include <AudioUnit/AudioUnit.h>
#if MAC_OS_X_VERSION_MAX_ALLOWED <= 1050
#include <AudioUnit/AUNTComponent.h>
#endif
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "../SDL_sysaudio.h"
#include "SDL_coreaudio.h"
#define DEBUG_COREAUDIO 0
typedef struct COREAUDIO_DeviceList
{
AudioDeviceID id;
const char *name;
} COREAUDIO_DeviceList;
static COREAUDIO_DeviceList *inputDevices = NULL;
static int inputDeviceCount = 0;
static COREAUDIO_DeviceList *outputDevices = NULL;
static int outputDeviceCount = 0;
static void
free_device_list(COREAUDIO_DeviceList ** devices, int *devCount)
{
if (*devices) {
int i = *devCount;
while (i--)
SDL_free((void *) (*devices)[i].name);
SDL_free(*devices);
*devices = NULL;
}
*devCount = 0;
}
static void
build_device_list(int iscapture, COREAUDIO_DeviceList ** devices,
int *devCount)
{
Boolean outWritable = 0;
OSStatus result = noErr;
UInt32 size = 0;
AudioDeviceID *devs = NULL;
UInt32 i = 0;
UInt32 max = 0;
free_device_list(devices, devCount);
result = AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices,
&size, &outWritable);
if (result != kAudioHardwareNoError)
return;
devs = (AudioDeviceID *) alloca(size);
if (devs == NULL)
return;
max = size / sizeof(AudioDeviceID);
*devices = (COREAUDIO_DeviceList *) SDL_malloc(max * sizeof(**devices));
if (*devices == NULL)
return;
result = AudioHardwareGetProperty(kAudioHardwarePropertyDevices,
&size, devs);
if (result != kAudioHardwareNoError)
return;
for (i = 0; i < max; i++) {
CFStringRef cfstr = NULL;
char *ptr = NULL;
AudioDeviceID dev = devs[i];
AudioBufferList *buflist = NULL;
int usable = 0;
CFIndex len = 0;
result = AudioDeviceGetPropertyInfo(dev, 0, iscapture,
kAudioDevicePropertyStreamConfiguration,
&size, &outWritable);
if (result != noErr)
continue;
buflist = (AudioBufferList *) SDL_malloc(size);
if (buflist == NULL)
continue;
result = AudioDeviceGetProperty(dev, 0, iscapture,
kAudioDevicePropertyStreamConfiguration,
&size, buflist);
if (result == noErr) {
UInt32 j;
for (j = 0; j < buflist->mNumberBuffers; j++) {
if (buflist->mBuffers[j].mNumberChannels > 0) {
usable = 1;
break;
}
}
}
SDL_free(buflist);
if (!usable)
continue;
size = sizeof(CFStringRef);
result = AudioDeviceGetProperty(dev, 0, iscapture,
kAudioDevicePropertyDeviceNameCFString,
&size, &cfstr);
if (result != kAudioHardwareNoError)
continue;
len = CFStringGetMaximumSizeForEncoding(CFStringGetLength(cfstr),
kCFStringEncodingUTF8);
ptr = (char *) SDL_malloc(len + 1);
usable = ((ptr != NULL) &&
(CFStringGetCString
(cfstr, ptr, len + 1, kCFStringEncodingUTF8)));
CFRelease(cfstr);
if (usable) {
len = strlen(ptr);
/* Some devices have whitespace at the end...trim it. */
while ((len > 0) && (ptr[len - 1] == ' ')) {
len--;
}
usable = (len > 0);
}
if (!usable) {
SDL_free(ptr);
} else {
ptr[len] = '\0';
#if DEBUG_COREAUDIO
printf("COREAUDIO: Found %s device #%d: '%s' (devid %d)\n",
((iscapture) ? "capture" : "output"),
(int) *devCount, ptr, (int) dev);
#endif
(*devices)[*devCount].id = dev;
(*devices)[*devCount].name = ptr;
(*devCount)++;
}
}
}
static inline void
build_device_lists(void)
{
build_device_list(0, &outputDevices, &outputDeviceCount);
build_device_list(1, &inputDevices, &inputDeviceCount);
}
static inline void
free_device_lists(void)
{
free_device_list(&outputDevices, &outputDeviceCount);
free_device_list(&inputDevices, &inputDeviceCount);
}
static int
find_device_id(const char *devname, int iscapture, AudioDeviceID * id)
{
int i = ((iscapture) ? inputDeviceCount : outputDeviceCount);
COREAUDIO_DeviceList *devs = ((iscapture) ? inputDevices : outputDevices);
while (i--) {
if (SDL_strcmp(devname, devs->name) == 0) {
*id = devs->id;
return 1;
}
devs++;
}
return 0;
}
static int
COREAUDIO_DetectDevices(int iscapture)
{
if (iscapture) {
build_device_list(1, &inputDevices, &inputDeviceCount);
return inputDeviceCount;
} else {
build_device_list(0, &outputDevices, &outputDeviceCount);
return outputDeviceCount;
}
return 0; /* shouldn't ever hit this. */
}
static const char *
COREAUDIO_GetDeviceName(int index, int iscapture)
{
if ((iscapture) && (index < inputDeviceCount)) {
return inputDevices[index].name;
} else if ((!iscapture) && (index < outputDeviceCount)) {
return outputDevices[index].name;
}
SDL_SetError("No such device");
return NULL;
}
static void
COREAUDIO_Deinitialize(void)
{
free_device_lists();
}
/* The CoreAudio callback */
static OSStatus
outputCallback(void *inRefCon,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp,
UInt32 inBusNumber, UInt32 inNumberFrames,
AudioBufferList * ioData)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) inRefCon;
AudioBuffer *abuf;
UInt32 remaining, len;
void *ptr;
UInt32 i;
/* Only do anything if audio is enabled and not paused */
if (!this->enabled || this->paused) {
for (i = 0; i < ioData->mNumberBuffers; i++) {
abuf = &ioData->mBuffers[i];
SDL_memset(abuf->mData, this->spec.silence, abuf->mDataByteSize);
}
return 0;
}
/* No SDL conversion should be needed here, ever, since we accept
any input format in OpenAudio, and leave the conversion to CoreAudio.
*/
/*
assert(!this->convert.needed);
assert(this->spec.channels == ioData->mNumberChannels);
*/
for (i = 0; i < ioData->mNumberBuffers; i++) {
abuf = &ioData->mBuffers[i];
remaining = abuf->mDataByteSize;
ptr = abuf->mData;
while (remaining > 0) {
if (this->hidden->bufferOffset >= this->hidden->bufferSize) {
/* Generate the data */
SDL_memset(this->hidden->buffer, this->spec.silence,
this->hidden->bufferSize);
SDL_mutexP(this->mixer_lock);
(*this->spec.callback)(this->spec.userdata,
this->hidden->buffer, this->hidden->bufferSize);
SDL_mutexV(this->mixer_lock);
this->hidden->bufferOffset = 0;
}
len = this->hidden->bufferSize - this->hidden->bufferOffset;
if (len > remaining)
len = remaining;
SDL_memcpy(ptr, (char *)this->hidden->buffer +
this->hidden->bufferOffset, len);
ptr = (char *)ptr + len;
remaining -= len;
this->hidden->bufferOffset += len;
}
}
return 0;
}
static OSStatus
inputCallback(void *inRefCon,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp,
UInt32 inBusNumber, UInt32 inNumberFrames,
AudioBufferList * ioData)
{
//err = AudioUnitRender(afr->fAudioUnit, ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, afr->fAudioBuffer);
// !!! FIXME: write me!
return noErr;
}
static void
COREAUDIO_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->audioUnitOpened) {
OSStatus result = noErr;
AURenderCallbackStruct callback;
const AudioUnitElement output_bus = 0;
const AudioUnitElement input_bus = 1;
const int iscapture = this->iscapture;
const AudioUnitElement bus =
((iscapture) ? input_bus : output_bus);
const AudioUnitScope scope =
((iscapture) ? kAudioUnitScope_Output :
kAudioUnitScope_Input);
/* stop processing the audio unit */
result = AudioOutputUnitStop(this->hidden->audioUnit);
/* Remove the input callback */
SDL_memset(&callback, '\0', sizeof(AURenderCallbackStruct));
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioUnitProperty_SetRenderCallback,
scope, bus, &callback,
sizeof(callback));
CloseComponent(this->hidden->audioUnit);
this->hidden->audioUnitOpened = 0;
}
SDL_free(this->hidden->buffer);
SDL_free(this->hidden);
this->hidden = NULL;
}
}
#define CHECK_RESULT(msg) \
if (result != noErr) { \
COREAUDIO_CloseDevice(this); \
SDL_SetError("CoreAudio error (%s): %d", msg, (int) result); \
return 0; \
}
static int
find_device_by_name(_THIS, const char *devname, int iscapture)
{
AudioDeviceID devid = 0;
OSStatus result = noErr;
UInt32 size = 0;
UInt32 alive = 0;
pid_t pid = 0;
if (devname == NULL) {
size = sizeof(AudioDeviceID);
const AudioHardwarePropertyID propid =
((iscapture) ? kAudioHardwarePropertyDefaultInputDevice :
kAudioHardwarePropertyDefaultOutputDevice);
result = AudioHardwareGetProperty(propid, &size, &devid);
CHECK_RESULT("AudioHardwareGetProperty (default device)");
} else {
if (!find_device_id(devname, iscapture, &devid)) {
SDL_SetError("CoreAudio: No such audio device.");
return 0;
}
}
size = sizeof(alive);
result = AudioDeviceGetProperty(devid, 0, iscapture,
kAudioDevicePropertyDeviceIsAlive,
&size, &alive);
CHECK_RESULT
("AudioDeviceGetProperty (kAudioDevicePropertyDeviceIsAlive)");
if (!alive) {
SDL_SetError("CoreAudio: requested device exists, but isn't alive.");
return 0;
}
size = sizeof(pid);
result = AudioDeviceGetProperty(devid, 0, iscapture,
kAudioDevicePropertyHogMode, &size, &pid);
/* some devices don't support this property, so errors are fine here. */
if ((result == noErr) && (pid != -1)) {
SDL_SetError("CoreAudio: requested device is being hogged.");
return 0;
}
this->hidden->deviceID = devid;
return 1;
}
static int
prepare_audiounit(_THIS, const char *devname, int iscapture,
const AudioStreamBasicDescription * strdesc)
{
OSStatus result = noErr;
AURenderCallbackStruct callback;
ComponentDescription desc;
Component comp = NULL;
const AudioUnitElement output_bus = 0;
const AudioUnitElement input_bus = 1;
const AudioUnitElement bus = ((iscapture) ? input_bus : output_bus);
const AudioUnitScope scope = ((iscapture) ? kAudioUnitScope_Output :
kAudioUnitScope_Input);
if (!find_device_by_name(this, devname, iscapture)) {
SDL_SetError("Couldn't find requested CoreAudio device");
return 0;
}
SDL_memset(&desc, '\0', sizeof(ComponentDescription));
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_DefaultOutput;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
comp = FindNextComponent(NULL, &desc);
if (comp == NULL) {
SDL_SetError("Couldn't find requested CoreAudio component");
return 0;
}
/* Open & initialize the audio unit */
result = OpenAComponent(comp, &this->hidden->audioUnit);
CHECK_RESULT("OpenAComponent");
this->hidden->audioUnitOpened = 1;
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioOutputUnitProperty_CurrentDevice,
kAudioUnitScope_Global, 0,
&this->hidden->deviceID,
sizeof(AudioDeviceID));
CHECK_RESULT
("AudioUnitSetProperty (kAudioOutputUnitProperty_CurrentDevice)");
/* Set the data format of the audio unit. */
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioUnitProperty_StreamFormat,
scope, bus, strdesc, sizeof(*strdesc));
CHECK_RESULT("AudioUnitSetProperty (kAudioUnitProperty_StreamFormat)");
/* Set the audio callback */
SDL_memset(&callback, '\0', sizeof(AURenderCallbackStruct));
callback.inputProc = ((iscapture) ? inputCallback : outputCallback);
callback.inputProcRefCon = this;
result = AudioUnitSetProperty(this->hidden->audioUnit,
kAudioUnitProperty_SetRenderCallback,
scope, bus, &callback, sizeof(callback));
CHECK_RESULT
("AudioUnitSetProperty (kAudioUnitProperty_SetRenderCallback)");
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
/* Allocate a sample buffer */
this->hidden->bufferOffset = this->hidden->bufferSize = this->spec.size;
this->hidden->buffer = SDL_malloc(this->hidden->bufferSize);
result = AudioUnitInitialize(this->hidden->audioUnit);
CHECK_RESULT("AudioUnitInitialize");
/* Finally, start processing of the audio unit */
result = AudioOutputUnitStart(this->hidden->audioUnit);
CHECK_RESULT("AudioOutputUnitStart");
/* We're running! */
return 1;
}
static int
COREAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
{
AudioStreamBasicDescription strdesc;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
int valid_datatype = 0;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return (0);
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Setup a AudioStreamBasicDescription with the requested format */
SDL_memset(&strdesc, '\0', sizeof(AudioStreamBasicDescription));
strdesc.mFormatID = kAudioFormatLinearPCM;
strdesc.mFormatFlags = kLinearPCMFormatFlagIsPacked;
strdesc.mChannelsPerFrame = this->spec.channels;
strdesc.mSampleRate = this->spec.freq;
strdesc.mFramesPerPacket = 1;
while ((!valid_datatype) && (test_format)) {
this->spec.format = test_format;
/* Just a list of valid SDL formats, so people don't pass junk here. */
switch (test_format) {
case AUDIO_U8:
case AUDIO_S8:
case AUDIO_U16LSB:
case AUDIO_S16LSB:
case AUDIO_U16MSB:
case AUDIO_S16MSB:
case AUDIO_S32LSB:
case AUDIO_S32MSB:
case AUDIO_F32LSB:
case AUDIO_F32MSB:
valid_datatype = 1;
strdesc.mBitsPerChannel = SDL_AUDIO_BITSIZE(this->spec.format);
if (SDL_AUDIO_ISBIGENDIAN(this->spec.format))
strdesc.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
if (SDL_AUDIO_ISFLOAT(this->spec.format))
strdesc.mFormatFlags |= kLinearPCMFormatFlagIsFloat;
else if (SDL_AUDIO_ISSIGNED(this->spec.format))
strdesc.mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
break;
}
}
if (!valid_datatype) { /* shouldn't happen, but just in case... */
COREAUDIO_CloseDevice(this);
SDL_SetError("Unsupported audio format");
return 0;
}
strdesc.mBytesPerFrame =
strdesc.mBitsPerChannel * strdesc.mChannelsPerFrame / 8;
strdesc.mBytesPerPacket =
strdesc.mBytesPerFrame * strdesc.mFramesPerPacket;
if (!prepare_audiounit(this, devname, iscapture, &strdesc)) {
COREAUDIO_CloseDevice(this);
return 0; /* prepare_audiounit() will call SDL_SetError()... */
}
return 1; /* good to go. */
}
static int
COREAUDIO_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->DetectDevices = COREAUDIO_DetectDevices;
impl->GetDeviceName = COREAUDIO_GetDeviceName;
impl->OpenDevice = COREAUDIO_OpenDevice;
impl->CloseDevice = COREAUDIO_CloseDevice;
impl->Deinitialize = COREAUDIO_Deinitialize;
impl->ProvidesOwnCallbackThread = 1;
build_device_lists(); /* do an initial check for devices... */
return 1; /* this audio target is available. */
}
AudioBootStrap COREAUDIO_bootstrap = {
"coreaudio", "Mac OS X CoreAudio", COREAUDIO_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

43
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_coreaudio_h
#define _SDL_coreaudio_h
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
AudioUnit audioUnit;
int audioUnitOpened;
void *buffer;
UInt32 bufferOffset;
UInt32 bufferSize;
AudioDeviceID deviceID;
};
#endif /* _SDL_coreaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* Tru64 UNIX MME support */
#include <mme_api.h>
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "SDL_mmeaudio.h"
static BOOL inUse[NUM_BUFFERS];
static void
SetMMerror(char *function, MMRESULT code)
{
int len;
char errbuf[MAXERRORLENGTH];
SDL_snprintf(errbuf, SDL_arraysize(errbuf), "%s: ", function);
len = SDL_strlen(errbuf);
waveOutGetErrorText(code, errbuf + len, MAXERRORLENGTH - len);
SDL_SetError("%s", errbuf);
}
static void CALLBACK
MME_Callback(HWAVEOUT hwo,
UINT uMsg, DWORD dwInstance, LPARAM dwParam1, LPARAM dwParam2)
{
WAVEHDR *wp = (WAVEHDR *) dwParam1;
if (uMsg == WOM_DONE)
inUse[wp->dwUser] = FALSE;
}
static int
MME_OpenDevice(_THIS, const char *devname, int iscapture)
{
int valid_format = 0;
MMRESULT result;
Uint8 *mixbuf = NULL;
int i;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Set basic WAVE format parameters */
this->hidden->shm = mmeAllocMem(sizeof(*this->hidden->shm));
if (this->hidden->shm == NULL) {
MME_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden->shm, '\0', sizeof(*this->hidden->shm));
this->hidden->shm->sound = 0;
this->hidden->shm->wFmt.wf.wFormatTag = WAVE_FORMAT_PCM;
/* Determine the audio parameters from the AudioSpec */
/* Try for a closest match on audio format */
for (test_format = SDL_FirstAudioFormat(this->spec.format);
!valid_format && test_format;) {
valid_format = 1;
switch (test_format) {
case AUDIO_U8:
case AUDIO_S16:
case AUDIO_S32:
break;
default:
valid_format = 0;
test_format = SDL_NextAudioFormat();
}
}
if (!valid_format) {
MME_CloseDevice(this);
SDL_SetError("Unsupported audio format");
return 0;
}
this->spec.format = test_format;
this->hidden->shm->wFmt.wBitsPerSample = SDL_AUDIO_BITSIZE(test_format);
/* !!! FIXME: Can this handle more than stereo? */
this->hidden->shm->wFmt.wf.nChannels = this->spec.channels;
this->hidden->shm->wFmt.wf.nSamplesPerSec = this->spec.freq;
this->hidden->shm->wFmt.wf.nBlockAlign =
this->hidden->shm->wFmt.wf.nChannels *
this->hidden->shm->wFmt.wBitsPerSample / 8;
this->hidden->shm->wFmt.wf.nAvgBytesPerSec =
this->hidden->shm->wFmt.wf.nSamplesPerSec *
this->hidden->shm->wFmt.wf.nBlockAlign;
/* Check the buffer size -- minimum of 1/4 second (word aligned) */
if (this->spec.samples < (this->spec.freq / 4))
this->spec.samples = ((this->spec.freq / 4) + 3) & ~3;
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* Open the audio device */
result = waveOutOpen(&(this->hidden->shm->sound),
WAVE_MAPPER,
&(this->hidden->shm->wFmt.wf),
MME_Callback,
NULL, (CALLBACK_FUNCTION | WAVE_OPEN_SHAREABLE));
if (result != MMSYSERR_NOERROR) {
MME_CloseDevice(this);
SetMMerror("waveOutOpen()", result);
return 0;
}
/* Create the sound buffers */
mixbuf = (Uint8 *) mmeAllocBuffer(NUM_BUFFERS * (this->spec.size));
if (mixbuf == NULL) {
MME_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
this->hidden->mixbuf = mixbuf;
for (i = 0; i < NUM_BUFFERS; i++) {
this->hidden->shm->wHdr[i].lpData = &mixbuf[i * (this->spec.size)];
this->hidden->shm->wHdr[i].dwBufferLength = this->spec.size;
this->hidden->shm->wHdr[i].dwFlags = 0;
this->hidden->shm->wHdr[i].dwUser = i;
this->hidden->shm->wHdr[i].dwLoops = 0; /* loop control counter */
this->hidden->shm->wHdr[i].lpNext = NULL; /* reserved for driver */
this->hidden->shm->wHdr[i].reserved = 0;
inUse[i] = FALSE;
}
this->hidden->next_buffer = 0;
return 1;
}
static void
MME_WaitDevice(_THIS)
{
while (inUse[this->hidden->next_buffer]) {
mmeWaitForCallbacks();
mmeProcessCallbacks();
}
}
static Uint8 *
MME_GetDeviceBuf(_THIS)
{
void *retval = this->hidden->shm->wHdr[this->hidden->next_buffer].lpData;
inUse[this->hidden->next_buffer] = TRUE;
return (Uint8 *) retval;
}
static void
MME_PlayDevice(_THIS)
{
/* Queue it up */
waveOutWrite(this->hidden->shm->sound,
&(this->hidden->shm->wHdr[this->hidden->next_buffer]),
sizeof(WAVEHDR));
this->hidden->next_buffer = (this->hidden->next_buffer + 1) % NUM_BUFFERS;
}
static void
MME_WaitDone(_THIS)
{
MMRESULT result;
int i;
if (this->hidden->shm->sound) {
for (i = 0; i < NUM_BUFFERS; i++)
while (inUse[i]) {
mmeWaitForCallbacks();
mmeProcessCallbacks();
}
result = waveOutReset(this->hidden->shm->sound);
if (result != MMSYSERR_NOERROR)
SetMMerror("waveOutReset()", result);
mmeProcessCallbacks();
}
}
static void
MME_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
MMRESULT result;
if (this->hidden->mixbuf) {
result = mmeFreeBuffer(this->hidden->mixbuf);
if (result != MMSYSERR_NOERROR)
SetMMerror("mmeFreeBuffer", result);
this->hidden->mixbuf = NULL;
}
if (this->hidden->shm) {
if (this->hidden->shm->sound) {
result = waveOutClose(this->hidden->shm->sound);
if (result != MMSYSERR_NOERROR)
SetMMerror("waveOutClose()", result);
mmeProcessCallbacks();
}
result = mmeFreeMem(this->hidden->shm);
if (result != MMSYSERR_NOERROR)
SetMMerror("mmeFreeMem()", result);
this->hidden->shm = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
static int
MME_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->OpenDevice = MME_OpenDevice;
impl->WaitDevice = MME_WaitDevice;
impl->WaitDone = MME_WaitDone;
impl->PlayDevice = MME_PlayDevice;
impl->GetDeviceBuf = MME_GetDeviceBuf;
impl->CloseDevice = MME_CloseDevice;
impl->OnlyHasDefaultOutputDevice = 1;
return 1; /* this audio target is available. */
}
/* !!! FIXME: Windows "windib" driver is called waveout, too */
AudioBootStrap MMEAUDIO_bootstrap = {
"waveout", "Tru64 MME WaveOut", MME_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

51
project/jni/sdl-1.3/src/audio/mme/SDL_mmeaudio.h Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* Allow access to a raw mixing buffer */
#ifndef _SDL_mmeaudio_h
#define _SDL_mmeaudio_h
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define NUM_BUFFERS 2
struct SharedMem
{
HWAVEOUT sound;
WAVEHDR wHdr[NUM_BUFFERS];
PCMWAVEFORMAT wFmt;
};
struct SDL_PrivateAudioData
{
Uint8 *mixbuf; /* The raw allocated mixing buffer */
struct SharedMem *shm;
int next_buffer;
};
#endif /* _SDL_mmeaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

408
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 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
This driver was written by:
Erik Inge Bolsø
knan@mo.himolde.no
*/
#include "SDL_config.h"
/* Allow access to a raw mixing buffer */
#include <signal.h>
#include <unistd.h>
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "SDL_loadso.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_nasaudio.h"
/* The tag name used by nas audio */
#define NAS_DRIVER_NAME "nas"
static struct SDL_PrivateAudioData *this2 = NULL;
static void (*NAS_AuCloseServer) (AuServer *);
static void (*NAS_AuNextEvent) (AuServer *, AuBool, AuEvent *);
static AuBool(*NAS_AuDispatchEvent) (AuServer *, AuEvent *);
static AuFlowID(*NAS_AuCreateFlow) (AuServer *, AuStatus *);
static void (*NAS_AuStartFlow) (AuServer *, AuFlowID, AuStatus *);
static void (*NAS_AuSetElements)
(AuServer *, AuFlowID, AuBool, int, AuElement *, AuStatus *);
static void (*NAS_AuWriteElement)
(AuServer *, AuFlowID, int, AuUint32, AuPointer, AuBool, AuStatus *);
static AuServer *(*NAS_AuOpenServer)
(_AuConst char *, int, _AuConst char *, int, _AuConst char *, char **);
static AuEventHandlerRec *(*NAS_AuRegisterEventHandler)
(AuServer *, AuMask, int, AuID, AuEventHandlerCallback, AuPointer);
#ifdef SDL_AUDIO_DRIVER_NAS_DYNAMIC
static const char *nas_library = SDL_AUDIO_DRIVER_NAS_DYNAMIC;
static void *nas_handle = NULL;
static int
load_nas_sym(const char *fn, void **addr)
{
*addr = SDL_LoadFunction(nas_handle, fn);
if (*addr == NULL) {
return 0;
}
return 1;
}
/* cast funcs to char* first, to please GCC's strict aliasing rules. */
#define SDL_NAS_SYM(x) \
if (!load_nas_sym(#x, (void **) (char *) &NAS_##x)) return -1
#else
#define SDL_NAS_SYM(x) NAS_##x = x
#endif
static int
load_nas_syms(void)
{
SDL_NAS_SYM(AuCloseServer);
SDL_NAS_SYM(AuNextEvent);
SDL_NAS_SYM(AuDispatchEvent);
SDL_NAS_SYM(AuCreateFlow);
SDL_NAS_SYM(AuStartFlow);
SDL_NAS_SYM(AuSetElements);
SDL_NAS_SYM(AuWriteElement);
SDL_NAS_SYM(AuOpenServer);
SDL_NAS_SYM(AuRegisterEventHandler);
return 0;
}
#undef SDL_NAS_SYM
#ifdef SDL_AUDIO_DRIVER_NAS_DYNAMIC
static void
UnloadNASLibrary(void)
{
if (nas_handle != NULL) {
SDL_UnloadObject(nas_handle);
nas_handle = NULL;
}
}
static int
LoadNASLibrary(void)
{
int retval = 0;
if (nas_handle == NULL) {
nas_handle = SDL_LoadObject(nas_library);
if (nas_handle == NULL) {
/* Copy error string so we can use it in a new SDL_SetError(). */
char *origerr = SDL_GetError();
size_t len = SDL_strlen(origerr) + 1;
char *err = (char *) alloca(len);
SDL_strlcpy(err, origerr, len);
retval = -1;
SDL_SetError("NAS: SDL_LoadObject('%s') failed: %s\n",
nas_library, err);
} else {
retval = load_nas_syms();
if (retval < 0) {
UnloadNASLibrary();
}
}
}
return retval;
}
#else
static void
UnloadNASLibrary(void)
{
}
static int
LoadNASLibrary(void)
{
load_nas_syms();
return 0;
}
#endif /* SDL_AUDIO_DRIVER_NAS_DYNAMIC */
/* This function waits until it is possible to write a full sound buffer */
static void
NAS_WaitDevice(_THIS)
{
while (this->hidden->buf_free < this->hidden->mixlen) {
AuEvent ev;
NAS_AuNextEvent(this->hidden->aud, AuTrue, &ev);
NAS_AuDispatchEvent(this->hidden->aud, &ev);
}
}
static void
NAS_PlayDevice(_THIS)
{
while (this->hidden->mixlen > this->hidden->buf_free) {
/*
* We think the buffer is full? Yikes! Ask the server for events,
* in the hope that some of them is LowWater events telling us more
* of the buffer is free now than what we think.
*/
AuEvent ev;
NAS_AuNextEvent(this->hidden->aud, AuTrue, &ev);
NAS_AuDispatchEvent(this->hidden->aud, &ev);
}
this->hidden->buf_free -= this->hidden->mixlen;
/* Write the audio data */
NAS_AuWriteElement(this->hidden->aud, this->hidden->flow, 0,
this->hidden->mixlen, this->hidden->mixbuf, AuFalse,
NULL);
this->hidden->written += this->hidden->mixlen;
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", this->hidden->mixlen);
#endif
}
static Uint8 *
NAS_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
}
static void
NAS_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
if (this->hidden->aud) {
NAS_AuCloseServer(this->hidden->aud);
this->hidden->aud = 0;
}
SDL_free(this->hidden);
this2 = this->hidden = NULL;
}
}
static unsigned char
sdlformat_to_auformat(unsigned int fmt)
{
switch (fmt) {
case AUDIO_U8:
return AuFormatLinearUnsigned8;
case AUDIO_S8:
return AuFormatLinearSigned8;
case AUDIO_U16LSB:
return AuFormatLinearUnsigned16LSB;
case AUDIO_U16MSB:
return AuFormatLinearUnsigned16MSB;
case AUDIO_S16LSB:
return AuFormatLinearSigned16LSB;
case AUDIO_S16MSB:
return AuFormatLinearSigned16MSB;
}
return AuNone;
}
static AuBool
event_handler(AuServer * aud, AuEvent * ev, AuEventHandlerRec * hnd)
{
switch (ev->type) {
case AuEventTypeElementNotify:
{
AuElementNotifyEvent *event = (AuElementNotifyEvent *) ev;
switch (event->kind) {
case AuElementNotifyKindLowWater:
if (this2->buf_free >= 0) {
this2->really += event->num_bytes;
gettimeofday(&this2->last_tv, 0);
this2->buf_free += event->num_bytes;
} else {
this2->buf_free = event->num_bytes;
}
break;
case AuElementNotifyKindState:
switch (event->cur_state) {
case AuStatePause:
if (event->reason != AuReasonUser) {
if (this2->buf_free >= 0) {
this2->really += event->num_bytes;
gettimeofday(&this2->last_tv, 0);
this2->buf_free += event->num_bytes;
} else {
this2->buf_free = event->num_bytes;
}
}
break;
}
}
}
}
return AuTrue;
}
static AuDeviceID
find_device(_THIS, int nch)
{
/* These "Au" things are all macros, not functions... */
int i;
for (i = 0; i < AuServerNumDevices(this->hidden->aud); i++) {
if ((AuDeviceKind(AuServerDevice(this->hidden->aud, i)) ==
AuComponentKindPhysicalOutput) &&
AuDeviceNumTracks(AuServerDevice(this->hidden->aud, i)) == nch) {
return AuDeviceIdentifier(AuServerDevice(this->hidden->aud, i));
}
}
return AuNone;
}
static int
NAS_OpenDevice(_THIS, const char *devname, int iscapture)
{
AuElement elms[3];
int buffer_size;
SDL_AudioFormat test_format, format;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Try for a closest match on audio format */
format = 0;
for (test_format = SDL_FirstAudioFormat(this->spec.format);
!format && test_format;) {
format = sdlformat_to_auformat(test_format);
if (format == AuNone) {
test_format = SDL_NextAudioFormat();
}
}
if (format == 0) {
NAS_CloseDevice(this);
SDL_SetError("NAS: Couldn't find any hardware audio formats");
return 0;
}
this->spec.format = test_format;
this->hidden->aud = NAS_AuOpenServer("", 0, NULL, 0, NULL, NULL);
if (this->hidden->aud == 0) {
NAS_CloseDevice(this);
SDL_SetError("NAS: Couldn't open connection to NAS server");
return 0;
}
this->hidden->dev = find_device(this, this->spec.channels);
if ((this->hidden->dev == AuNone)
|| (!(this->hidden->flow = NAS_AuCreateFlow(this->hidden->aud, 0)))) {
NAS_CloseDevice(this);
SDL_SetError("NAS: Couldn't find a fitting device on NAS server");
return 0;
}
buffer_size = this->spec.freq;
if (buffer_size < 4096)
buffer_size = 4096;
if (buffer_size > 32768)
buffer_size = 32768; /* So that the buffer won't get unmanageably big. */
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
this2 = this->hidden;
AuMakeElementImportClient(elms, this->spec.freq, format,
this->spec.channels, AuTrue, buffer_size,
buffer_size / 4, 0, NULL);
AuMakeElementExportDevice(elms + 1, 0, this->hidden->dev, this->spec.freq,
AuUnlimitedSamples, 0, NULL);
NAS_AuSetElements(this->hidden->aud, this->hidden->flow, AuTrue, 2, elms,
NULL);
NAS_AuRegisterEventHandler(this->hidden->aud, AuEventHandlerIDMask, 0,
this->hidden->flow, event_handler,
(AuPointer) NULL);
NAS_AuStartFlow(this->hidden->aud, this->hidden->flow, NULL);
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
NAS_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/* We're ready to rock and roll. :-) */
return 1;
}
static void
NAS_Deinitialize(void)
{
UnloadNASLibrary();
}
static int
NAS_Init(SDL_AudioDriverImpl * impl)
{
if (LoadNASLibrary() < 0) {
return 0;
} else {
AuServer *aud = NAS_AuOpenServer("", 0, NULL, 0, NULL, NULL);
if (aud == NULL) {
SDL_SetError("NAS: AuOpenServer() failed (no audio server?)");
return 0;
}
NAS_AuCloseServer(aud);
}
/* Set the function pointers */
impl->OpenDevice = NAS_OpenDevice;
impl->PlayDevice = NAS_PlayDevice;
impl->WaitDevice = NAS_WaitDevice;
impl->GetDeviceBuf = NAS_GetDeviceBuf;
impl->CloseDevice = NAS_CloseDevice;
impl->Deinitialize = NAS_Deinitialize;
impl->OnlyHasDefaultOutputDevice = 1; /* !!! FIXME: is this true? */
return 1; /* this audio target is available. */
}
AudioBootStrap NAS_bootstrap = {
NAS_DRIVER_NAME, "Network Audio System", NAS_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 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
This driver was written by:
Erik Inge Bolsø
knan@mo.himolde.no
*/
#include "SDL_config.h"
#ifndef _SDL_nasaudio_h
#define _SDL_nasaudio_h
#ifdef __sgi
#include <nas/audiolib.h>
#else
#include <audio/audiolib.h>
#endif
#include <sys/time.h>
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
AuServer *aud;
AuFlowID flow;
AuDeviceID dev;
/* Raw mixing buffer */
Uint8 *mixbuf;
int mixlen;
int written;
int really;
int bps;
struct timeval last_tv;
int buf_free;
};
#endif /* _SDL_nasaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 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
This file written by Ryan C. Gordon (icculus@icculus.org)
*/
#include "SDL_config.h"
/* Output audio to NDS */
#include <nds.h>
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "SDL_ndsaudio.h"
static int
NDSAUD_OpenDevice(_THIS, const char *devname, int iscapture)
{
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
int valid_datatype = 0;
this->hidden = SDL_malloc(sizeof(*(this->hidden)));
if (!this->hidden) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
while ((!valid_datatype) && (test_format)) {
this->spec.format = test_format;
switch (test_format) {
case AUDIO_S8:
/*case AUDIO_S16LSB: */
valid_datatype = 1;
break;
default:
test_format = SDL_NextAudioFormat();
break;
}
}
/* set the generic sound parameters */
setGenericSound(22050, /* sample rate */
127, /* volume */
64, /* panning/balance */
0); /* sound format */
return 1;
}
static void
NDSAUD_PlayDevice(_THIS)
{
TransferSoundData *sound = SDL_malloc(sizeof(TransferSoundData));
if (!sound) {
SDL_OutOfMemory();
}
playGenericSound(this->hidden->mixbuf, this->hidden->mixlen);
#if 0
// sound->data = this->hidden->mixbuf;/* pointer to raw audio data */
// sound->len = this->hidden->mixlen; /* size of raw data pointed to above */
// sound->rate = 22050; /* sample rate = 22050Hz */
// sound->vol = 127; /* volume [0..127] for [min..max] */
// sound->pan = 64; /* balance [0..127] for [left..right] */
// sound->format = 0; /* 0 for 16-bit, 1 for 8-bit */
// playSound(sound);
#endif
}
static Uint8 *
NDSAUD_GetDeviceBuf(_THIS)
{
return this->hidden->mixbuf; /* is this right? */
}
static void
NDSAUD_WaitDevice(_THIS)
{
/* stub */
}
static void
NDSAUD_CloseDevice(_THIS)
{
/* stub */
}
static int
NDSAUD_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->OpenDevice = NDSAUD_OpenDevice;
impl->PlayDevice = NDSAUD_PlayDevice;
impl->WaitDevice = NDSAUD_WaitDevice;
impl->GetDeviceBuf = NDSAUD_GetDeviceBuf;
impl->CloseDevice = NDSAUD_CloseDevice;
/* and the capabilities */
impl->HasCaptureSupport = 1;
impl->OnlyHasDefaultOutputDevice = 1;
impl->OnlyHasDefaultInputDevice = 1;
return 1; /* this audio target is available. */
}
AudioBootStrap NDSAUD_bootstrap = {
"nds", "SDL NDS audio driver", NDSAUD_Init, 0 /*1? */
};
/* vi: set ts=4 sw=4 expandtab: */

44
project/jni/sdl-1.3/src/audio/nds/SDL_ndsaudio.h Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_ndsaudio_h
#define _SDL_ndsaudio_h
#include "../SDL_sysaudio.h"
#include <nds/arm9/sound.h>
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
TransferSoundData *sound;
/* The file descriptor for the audio device */
Uint8 *mixbuf;
Uint32 mixlen;
Uint32 write_delay;
Uint32 initial_calls;
};
#endif /* _SDL_ndsaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

554
project/jni/sdl-1.3/src/audio/paudio/SDL_paudio.c Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 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
Carsten Griwodz
griff@kom.tu-darmstadt.de
based on linux/SDL_dspaudio.c by Sam Lantinga
*/
#include "SDL_config.h"
/* Allow access to a raw mixing buffer */
#include <errno.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/time.h>
#include <sys/ioctl.h>
#include <sys/types.h>
#include <sys/stat.h>
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "SDL_stdinc.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_paudio.h"
#define DEBUG_AUDIO 0
/* A conflict within AIX 4.3.3 <sys/> headers and probably others as well.
* I guess nobody ever uses audio... Shame over AIX header files. */
#include <sys/machine.h>
#undef BIG_ENDIAN
#include <sys/audio.h>
/* The tag name used by paud audio */
#define PAUDIO_DRIVER_NAME "paud"
/* Open the audio device for playback, and don't block if busy */
/* #define OPEN_FLAGS (O_WRONLY|O_NONBLOCK) */
#define OPEN_FLAGS O_WRONLY
/* Get the name of the audio device we use for output */
#ifndef _PATH_DEV_DSP
#define _PATH_DEV_DSP "/dev/%caud%c/%c"
#endif
static char devsettings[][3] = {
{'p', '0', '1'}, {'p', '0', '2'}, {'p', '0', '3'}, {'p', '0', '4'},
{'p', '1', '1'}, {'p', '1', '2'}, {'p', '1', '3'}, {'p', '1', '4'},
{'p', '2', '1'}, {'p', '2', '2'}, {'p', '2', '3'}, {'p', '2', '4'},
{'p', '3', '1'}, {'p', '3', '2'}, {'p', '3', '3'}, {'p', '3', '4'},
{'b', '0', '1'}, {'b', '0', '2'}, {'b', '0', '3'}, {'b', '0', '4'},
{'b', '1', '1'}, {'b', '1', '2'}, {'b', '1', '3'}, {'b', '1', '4'},
{'b', '2', '1'}, {'b', '2', '2'}, {'b', '2', '3'}, {'b', '2', '4'},
{'b', '3', '1'}, {'b', '3', '2'}, {'b', '3', '3'}, {'b', '3', '4'},
{'\0', '\0', '\0'}
};
static int
OpenUserDefinedDevice(char *path, int maxlen, int flags)
{
const char *audiodev;
int fd;
/* Figure out what our audio device is */
if ((audiodev = SDL_getenv("SDL_PATH_DSP")) == NULL) {
audiodev = SDL_getenv("AUDIODEV");
}
if (audiodev == NULL) {
return -1;
}
fd = open(audiodev, flags, 0);
if (path != NULL) {
SDL_strlcpy(path, audiodev, maxlen);
path[maxlen - 1] = '\0';
}
return fd;
}
static int
OpenAudioPath(char *path, int maxlen, int flags, int classic)
{
struct stat sb;
int cycle = 0;
int fd = OpenUserDefinedDevice(path, maxlen, flags);
if (fd != -1) {
return fd;
}
/* !!! FIXME: do we really need a table here? */
while (devsettings[cycle][0] != '\0') {
char audiopath[1024];
SDL_snprintf(audiopath, SDL_arraysize(audiopath),
_PATH_DEV_DSP,
devsettings[cycle][0],
devsettings[cycle][1], devsettings[cycle][2]);
if (stat(audiopath, &sb) == 0) {
fd = open(audiopath, flags, 0);
if (fd > 0) {
if (path != NULL) {
SDL_strlcpy(path, audiopath, maxlen);
}
return fd;
}
}
}
return -1;
}
/* This function waits until it is possible to write a full sound buffer */
static void
PAUDIO_WaitDevice(_THIS)
{
fd_set fdset;
/* See if we need to use timed audio synchronization */
if (this->hidden->frame_ticks) {
/* Use timer for general audio synchronization */
Sint32 ticks;
ticks =
((Sint32) (this->hidden->next_frame - SDL_GetTicks())) -
FUDGE_TICKS;
if (ticks > 0) {
SDL_Delay(ticks);
}
} else {
audio_buffer paud_bufinfo;
/* Use select() for audio synchronization */
struct timeval timeout;
FD_ZERO(&fdset);
FD_SET(this->hidden->audio_fd, &fdset);
if (ioctl(this->hidden->audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Couldn't get audio buffer information\n");
#endif
timeout.tv_sec = 10;
timeout.tv_usec = 0;
} else {
long ms_in_buf = paud_bufinfo.write_buf_time;
timeout.tv_sec = ms_in_buf / 1000;
ms_in_buf = ms_in_buf - timeout.tv_sec * 1000;
timeout.tv_usec = ms_in_buf * 1000;
#ifdef DEBUG_AUDIO
fprintf(stderr,
"Waiting for write_buf_time=%ld,%ld\n",
timeout.tv_sec, timeout.tv_usec);
#endif
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Waiting for audio to get ready\n");
#endif
if (select(this->hidden->audio_fd + 1, NULL, &fdset, NULL, &timeout)
<= 0) {
const char *message =
"Audio timeout - buggy audio driver? (disabled)";
/*
* In general we should never print to the screen,
* but in this case we have no other way of letting
* the user know what happened.
*/
fprintf(stderr, "SDL: %s - %s\n", strerror(errno), message);
this->enabled = 0;
/* Don't try to close - may hang */
this->hidden->audio_fd = -1;
#ifdef DEBUG_AUDIO
fprintf(stderr, "Done disabling audio\n");
#endif
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Ready!\n");
#endif
}
}
static void
PAUDIO_PlayDevice(_THIS)
{
int written = 0;
const Uint8 *mixbuf = this->hidden->mixbuf;
const size_t mixlen = this->hidden->mixlen;
/* Write the audio data, checking for EAGAIN on broken audio drivers */
do {
written = write(this->hidden->audio_fd, mixbuf, mixlen);
if ((written < 0) && ((errno == 0) || (errno == EAGAIN))) {
SDL_Delay(1); /* Let a little CPU time go by */
}
} while ((written < 0) &&
((errno == 0) || (errno == EAGAIN) || (errno == EINTR)));
/* If timer synchronization is enabled, set the next write frame */
if (this->hidden->frame_ticks) {
this->hidden->next_frame += this->hidden->frame_ticks;
}
/* If we couldn't write, assume fatal error for now */
if (written < 0) {
this->enabled = 0;
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", written);
#endif
}
static Uint8 *
PAUDIO_GetDeviceBuf(_THIS)
{
return this->hidden->mixbuf;
}
static void
PAUDIO_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
if (this->hidden->audio_fd >= 0) {
close(this->hidden->audio_fd);
this->hidden->audio_fd = -1;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
static int
PAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
{
const char *workaround = SDL_getenv("SDL_DSP_NOSELECT");
char audiodev[1024];
const char *err = NULL;
int format;
int bytes_per_sample;
SDL_AudioFormat test_format;
audio_init paud_init;
audio_buffer paud_bufinfo;
audio_status paud_status;
audio_control paud_control;
audio_change paud_change;
int fd = -1;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Open the audio device */
fd = OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
this->hidden->audio_fd = fd;
if (fd < 0) {
PAUDIO_CloseDevice(this);
SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
return 0;
}
/*
* We can't set the buffer size - just ask the device for the maximum
* that we can have.
*/
if (ioctl(fd, AUDIO_BUFFER, &paud_bufinfo) < 0) {
PAUDIO_CloseDevice(this);
SDL_SetError("Couldn't get audio buffer information");
return 0;
}
if (this->spec.channels > 1)
this->spec.channels = 2;
else
this->spec.channels = 1;
/*
* Fields in the audio_init structure:
*
* Ignored by us:
*
* paud.loadpath[LOAD_PATH]; * DSP code to load, MWave chip only?
* paud.slot_number; * slot number of the adapter
* paud.device_id; * adapter identification number
*
* Input:
*
* paud.srate; * the sampling rate in Hz
* paud.bits_per_sample; * 8, 16, 32, ...
* paud.bsize; * block size for this rate
* paud.mode; * ADPCM, PCM, MU_LAW, A_LAW, SOURCE_MIX
* paud.channels; * 1=mono, 2=stereo
* paud.flags; * FIXED - fixed length data
* * LEFT_ALIGNED, RIGHT_ALIGNED (var len only)
* * TWOS_COMPLEMENT - 2's complement data
* * SIGNED - signed? comment seems wrong in sys/audio.h
* * BIG_ENDIAN
* paud.operation; * PLAY, RECORD
*
* Output:
*
* paud.flags; * PITCH - pitch is supported
* * INPUT - input is supported
* * OUTPUT - output is supported
* * MONITOR - monitor is supported
* * VOLUME - volume is supported
* * VOLUME_DELAY - volume delay is supported
* * BALANCE - balance is supported
* * BALANCE_DELAY - balance delay is supported
* * TREBLE - treble control is supported
* * BASS - bass control is supported
* * BESTFIT_PROVIDED - best fit returned
* * LOAD_CODE - DSP load needed
* paud.rc; * NO_PLAY - DSP code can't do play requests
* * NO_RECORD - DSP code can't do record requests
* * INVALID_REQUEST - request was invalid
* * CONFLICT - conflict with open's flags
* * OVERLOADED - out of DSP MIPS or memory
* paud.position_resolution; * smallest increment for position
*/
paud_init.srate = this->spec.freq;
paud_init.mode = PCM;
paud_init.operation = PLAY;
paud_init.channels = this->spec.channels;
/* Try for a closest match on audio format */
format = 0;
for (test_format = SDL_FirstAudioFormat(this->spec.format);
!format && test_format;) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
switch (test_format) {
case AUDIO_U8:
bytes_per_sample = 1;
paud_init.bits_per_sample = 8;
paud_init.flags = TWOS_COMPLEMENT | FIXED;
format = 1;
break;
case AUDIO_S8:
bytes_per_sample = 1;
paud_init.bits_per_sample = 8;
paud_init.flags = SIGNED | TWOS_COMPLEMENT | FIXED;
format = 1;
break;
case AUDIO_S16LSB:
bytes_per_sample = 2;
paud_init.bits_per_sample = 16;
paud_init.flags = SIGNED | TWOS_COMPLEMENT | FIXED;
format = 1;
break;
case AUDIO_S16MSB:
bytes_per_sample = 2;
paud_init.bits_per_sample = 16;
paud_init.flags = BIG_ENDIAN | SIGNED | TWOS_COMPLEMENT | FIXED;
format = 1;
break;
case AUDIO_U16LSB:
bytes_per_sample = 2;
paud_init.bits_per_sample = 16;
paud_init.flags = TWOS_COMPLEMENT | FIXED;
format = 1;
break;
case AUDIO_U16MSB:
bytes_per_sample = 2;
paud_init.bits_per_sample = 16;
paud_init.flags = BIG_ENDIAN | TWOS_COMPLEMENT | FIXED;
format = 1;
break;
default:
break;
}
if (!format) {
test_format = SDL_NextAudioFormat();
}
}
if (format == 0) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Couldn't find any hardware audio formats\n");
#endif
PAUDIO_CloseDevice(this);
SDL_SetError("Couldn't find any hardware audio formats");
return 0;
}
this->spec.format = test_format;
/*
* We know the buffer size and the max number of subsequent writes
* that can be pending. If more than one can pend, allow the application
* to do something like double buffering between our write buffer and
* the device's own buffer that we are filling with write() anyway.
*
* We calculate this->spec.samples like this because
* SDL_CalculateAudioSpec() will give put paud_bufinfo.write_buf_cap
* (or paud_bufinfo.write_buf_cap/2) into this->spec.size in return.
*/
if (paud_bufinfo.request_buf_cap == 1) {
this->spec.samples = paud_bufinfo.write_buf_cap
/ bytes_per_sample / this->spec.channels;
} else {
this->spec.samples = paud_bufinfo.write_buf_cap
/ bytes_per_sample / this->spec.channels / 2;
}
paud_init.bsize = bytes_per_sample * this->spec.channels;
SDL_CalculateAudioSpec(&this->spec);
/*
* The AIX paud device init can't modify the values of the audio_init
* structure that we pass to it. So we don't need any recalculation
* of this stuff and no reinit call as in linux dsp and dma code.
*
* /dev/paud supports all of the encoding formats, so we don't need
* to do anything like reopening the device, either.
*/
if (ioctl(fd, AUDIO_INIT, &paud_init) < 0) {
switch (paud_init.rc) {
case 1:
err = "Couldn't set audio format: DSP can't do play requests";
break;
case 2:
err = "Couldn't set audio format: DSP can't do record requests";
break;
case 4:
err = "Couldn't set audio format: request was invalid";
break;
case 5:
err = "Couldn't set audio format: conflict with open's flags";
break;
case 6:
err = "Couldn't set audio format: out of DSP MIPS or memory";
break;
default:
err = "Couldn't set audio format: not documented in sys/audio.h";
break;
}
}
if (err != NULL) {
PAUDIO_CloseDevice(this);
SDL_SetError("Paudio: %s", err);
return 0;
}
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
PAUDIO_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/*
* Set some paramters: full volume, first speaker that we can find.
* Ignore the other settings for now.
*/
paud_change.input = AUDIO_IGNORE; /* the new input source */
paud_change.output = OUTPUT_1; /* EXTERNAL_SPEAKER,INTERNAL_SPEAKER,OUTPUT_1 */
paud_change.monitor = AUDIO_IGNORE; /* the new monitor state */
paud_change.volume = 0x7fffffff; /* volume level [0-0x7fffffff] */
paud_change.volume_delay = AUDIO_IGNORE; /* the new volume delay */
paud_change.balance = 0x3fffffff; /* the new balance */
paud_change.balance_delay = AUDIO_IGNORE; /* the new balance delay */
paud_change.treble = AUDIO_IGNORE; /* the new treble state */
paud_change.bass = AUDIO_IGNORE; /* the new bass state */
paud_change.pitch = AUDIO_IGNORE; /* the new pitch state */
paud_control.ioctl_request = AUDIO_CHANGE;
paud_control.request_info = (char *) &paud_change;
if (ioctl(fd, AUDIO_CONTROL, &paud_control) < 0) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Can't change audio display settings\n");
#endif
}
/*
* Tell the device to expect data. Actual start will wait for
* the first write() call.
*/
paud_control.ioctl_request = AUDIO_START;
paud_control.position = 0;
if (ioctl(fd, AUDIO_CONTROL, &paud_control) < 0) {
PAUDIO_CloseDevice(this);
#ifdef DEBUG_AUDIO
fprintf(stderr, "Can't start audio play\n");
#endif
SDL_SetError("Can't start audio play");
return 0;
}
/* Check to see if we need to use select() workaround */
if (workaround != NULL) {
this->hidden->frame_ticks = (float) (this->spec.samples * 1000) /
this->spec.freq;
this->hidden->next_frame = SDL_GetTicks() + this->hidden->frame_ticks;
}
/* We're ready to rock and roll. :-) */
return 1;
}
static int
PAUDIO_Init(SDL_AudioDriverImpl * impl)
{
/* !!! FIXME: not right for device enum? */
int fd = OpenAudioPath(NULL, 0, OPEN_FLAGS, 0);
if (fd < 0) {
SDL_SetError("PAUDIO: Couldn't open audio device");
return 0;
}
close(fd);
/* Set the function pointers */
impl->OpenDevice = DSP_OpenDevice;
impl->PlayDevice = DSP_PlayDevice;
impl->PlayDevice = DSP_WaitDevice;
impl->GetDeviceBuf = DSP_GetDeviceBuf;
impl->CloseDevice = DSP_CloseDevice;
impl->OnlyHasDefaultOutputDevice = 1; /* !!! FIXME: add device enum! */
return 1; /* this audio target is available. */
}
AudioBootStrap PAUDIO_bootstrap = {
PAUDIO_DRIVER_NAME, "AIX Paudio", PAUDIO_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

48
project/jni/sdl-1.3/src/audio/paudio/SDL_paudio.h Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_paudaudio_h
#define _SDL_paudaudio_h
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
/* The file descriptor for the audio device */
int audio_fd;
/* Raw mixing buffer */
Uint8 *mixbuf;
int mixlen;
/* Support for audio timing using a timer, in addition to select() */
float frame_ticks;
float next_frame;
};
#define FUDGE_TICKS 10 /* The scheduler overhead ticks per frame */
#endif /* _SDL_paudaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

540
project/jni/sdl-1.3/src/audio/pulseaudio/SDL_pulseaudio.c Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 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
*/
/*
The PulseAudio target for SDL 1.3 is based on the 1.3 arts target, with
the appropriate parts replaced with the 1.2 PulseAudio target code. This
was the cleanest way to move it to 1.3. The 1.2 target was written by
Stéphan Kochen: stephan .a.t. kochen.nl
*/
#include "SDL_config.h"
/* Allow access to a raw mixing buffer */
#ifdef HAVE_SIGNAL_H
#include <signal.h>
#endif
#include <unistd.h>
#include <sys/types.h>
#include <errno.h>
#include <pulse/pulseaudio.h>
#include <pulse/simple.h>
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_pulseaudio.h"
#include "SDL_loadso.h"
/* The tag name used by pulse audio */
#define PULSEAUDIO_DRIVER_NAME "pulseaudio"
#if (PA_API_VERSION < 12)
/** Return non-zero if the passed state is one of the connected states */
static inline int PA_CONTEXT_IS_GOOD(pa_context_state_t x) {
return
x == PA_CONTEXT_CONNECTING ||
x == PA_CONTEXT_AUTHORIZING ||
x == PA_CONTEXT_SETTING_NAME ||
x == PA_CONTEXT_READY;
}
/** Return non-zero if the passed state is one of the connected states */
static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x) {
return
x == PA_STREAM_CREATING ||
x == PA_STREAM_READY;
}
#endif /* pulseaudio <= 0.9.10 */
static pa_simple *(*PULSEAUDIO_pa_simple_new) (const char *, const char *,
pa_stream_direction_t, const char *, const char *, const pa_sample_spec *,
const pa_channel_map *, const pa_buffer_attr *, int *);
static void (*PULSEAUDIO_pa_simple_free) (pa_simple *);
static pa_channel_map *(*PULSEAUDIO_pa_channel_map_init_auto) (
pa_channel_map *, unsigned, pa_channel_map_def_t);
static const char * (*PULSEAUDIO_pa_strerror) (int);
static pa_mainloop * (*PULSEAUDIO_pa_mainloop_new) (void);
static pa_mainloop_api * (*PULSEAUDIO_pa_mainloop_get_api) (pa_mainloop *);
static int (*PULSEAUDIO_pa_mainloop_iterate) (pa_mainloop *, int, int *);
static void (*PULSEAUDIO_pa_mainloop_free) (pa_mainloop *);
static pa_operation_state_t (*PULSEAUDIO_pa_operation_get_state) (
pa_operation *);
static void (*PULSEAUDIO_pa_operation_cancel) (pa_operation *);
static void (*PULSEAUDIO_pa_operation_unref) (pa_operation *);
static pa_context * (*PULSEAUDIO_pa_context_new) (pa_mainloop_api *,
const char *);
static int (*PULSEAUDIO_pa_context_connect) (pa_context *, const char *,
pa_context_flags_t, const pa_spawn_api *);
static pa_context_state_t (*PULSEAUDIO_pa_context_get_state) (pa_context *);
static void (*PULSEAUDIO_pa_context_disconnect) (pa_context *);
static void (*PULSEAUDIO_pa_context_unref) (pa_context *);
static pa_stream * (*PULSEAUDIO_pa_stream_new) (pa_context *, const char *,
const pa_sample_spec *, const pa_channel_map *);
static int (*PULSEAUDIO_pa_stream_connect_playback) (pa_stream *, const char *,
const pa_buffer_attr *, pa_stream_flags_t, pa_cvolume *, pa_stream *);
static pa_stream_state_t (*PULSEAUDIO_pa_stream_get_state) (pa_stream *);
static size_t (*PULSEAUDIO_pa_stream_writable_size) (pa_stream *);
static int (*PULSEAUDIO_pa_stream_write) (pa_stream *, const void *, size_t,
pa_free_cb_t, int64_t, pa_seek_mode_t);
static pa_operation * (*PULSEAUDIO_pa_stream_drain) (pa_stream *,
pa_stream_success_cb_t, void *);
static int (*PULSEAUDIO_pa_stream_disconnect) (pa_stream *);
static void (*PULSEAUDIO_pa_stream_unref) (pa_stream *);
static int load_pulseaudio_syms(void);
#ifdef SDL_AUDIO_DRIVER_PULSEAUDIO_DYNAMIC
static const char *pulseaudio_library = SDL_AUDIO_DRIVER_PULSEAUDIO_DYNAMIC;
static void *pulseaudio_handle = NULL;
static int
load_pulseaudio_sym(const char *fn, void **addr)
{
*addr = SDL_LoadFunction(pulseaudio_handle, fn);
if (*addr == NULL) {
/* Don't call SDL_SetError(): SDL_LoadFunction already did. */
return 0;
}
return 1;
}
/* cast funcs to char* first, to please GCC's strict aliasing rules. */
#define SDL_PULSEAUDIO_SYM(x) \
if (!load_pulseaudio_sym(#x, (void **) (char *) &PULSEAUDIO_##x)) return -1
static void
UnloadPulseAudioLibrary(void)
{
if (pulseaudio_handle != NULL) {
SDL_UnloadObject(pulseaudio_handle);
pulseaudio_handle = NULL;
}
}
static int
LoadPulseAudioLibrary(void)
{
int retval = 0;
if (pulseaudio_handle == NULL) {
pulseaudio_handle = SDL_LoadObject(pulseaudio_library);
if (pulseaudio_handle == NULL) {
retval = -1;
/* Don't call SDL_SetError(): SDL_LoadObject already did. */
} else {
retval = load_pulseaudio_syms();
if (retval < 0) {
UnloadPulseAudioLibrary();
}
}
}
return retval;
}
#else
#define SDL_PULSEAUDIO_SYM(x) PULSEAUDIO_##x = x
static void
UnloadPulseAudioLibrary(void)
{
}
static int
LoadPulseAudioLibrary(void)
{
load_pulseaudio_syms();
return 0;
}
#endif /* SDL_AUDIO_DRIVER_PULSEAUDIO_DYNAMIC */
static int
load_pulseaudio_syms(void)
{
SDL_PULSEAUDIO_SYM(pa_simple_new);
SDL_PULSEAUDIO_SYM(pa_simple_free);
SDL_PULSEAUDIO_SYM(pa_mainloop_new);
SDL_PULSEAUDIO_SYM(pa_mainloop_get_api);
SDL_PULSEAUDIO_SYM(pa_mainloop_iterate);
SDL_PULSEAUDIO_SYM(pa_mainloop_free);
SDL_PULSEAUDIO_SYM(pa_operation_get_state);
SDL_PULSEAUDIO_SYM(pa_operation_cancel);
SDL_PULSEAUDIO_SYM(pa_operation_unref);
SDL_PULSEAUDIO_SYM(pa_context_new);
SDL_PULSEAUDIO_SYM(pa_context_connect);
SDL_PULSEAUDIO_SYM(pa_context_get_state);
SDL_PULSEAUDIO_SYM(pa_context_disconnect);
SDL_PULSEAUDIO_SYM(pa_context_unref);
SDL_PULSEAUDIO_SYM(pa_stream_new);
SDL_PULSEAUDIO_SYM(pa_stream_connect_playback);
SDL_PULSEAUDIO_SYM(pa_stream_get_state);
SDL_PULSEAUDIO_SYM(pa_stream_writable_size);
SDL_PULSEAUDIO_SYM(pa_stream_write);
SDL_PULSEAUDIO_SYM(pa_stream_drain);
SDL_PULSEAUDIO_SYM(pa_stream_disconnect);
SDL_PULSEAUDIO_SYM(pa_stream_unref);
SDL_PULSEAUDIO_SYM(pa_channel_map_init_auto);
SDL_PULSEAUDIO_SYM(pa_strerror);
return 0;
}
/* This function waits until it is possible to write a full sound buffer */
static void
PULSEAUDIO_WaitDevice(_THIS)
{
struct SDL_PrivateAudioData *h = this->hidden;
while(1) {
if (PULSEAUDIO_pa_context_get_state(h->context) != PA_CONTEXT_READY ||
PULSEAUDIO_pa_stream_get_state(h->stream) != PA_STREAM_READY ||
PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
this->enabled = 0;
return;
}
if (PULSEAUDIO_pa_stream_writable_size(h->stream) >= h->mixlen) {
return;
}
}
}
static void
PULSEAUDIO_PlayDevice(_THIS)
{
/* Write the audio data */
struct SDL_PrivateAudioData *h = this->hidden;
if (PULSEAUDIO_pa_stream_write(h->stream, h->mixbuf, h->mixlen, NULL, 0LL,
PA_SEEK_RELATIVE) < 0) {
this->enabled = 0;
}
}
static void
stream_drain_complete(pa_stream *s, int success, void *userdata)
{
/* no-op for pa_stream_drain() to use for callback. */
}
static void
PULSEAUDIO_WaitDone(_THIS)
{
struct SDL_PrivateAudioData *h = this->hidden;
pa_operation *o;
o = PULSEAUDIO_pa_stream_drain(h->stream, stream_drain_complete, NULL);
if (!o) {
return;
}
while (PULSEAUDIO_pa_operation_get_state(o) != PA_OPERATION_DONE) {
if (PULSEAUDIO_pa_context_get_state(h->context) != PA_CONTEXT_READY ||
PULSEAUDIO_pa_stream_get_state(h->stream) != PA_STREAM_READY ||
PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
PULSEAUDIO_pa_operation_cancel(o);
break;
}
}
PULSEAUDIO_pa_operation_unref(o);
}
static Uint8 *
PULSEAUDIO_GetDeviceBuf(_THIS)
{
return (this->hidden->mixbuf);
}
static void
PULSEAUDIO_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
if (this->hidden->stream) {
PULSEAUDIO_pa_stream_disconnect(this->hidden->stream);
PULSEAUDIO_pa_stream_unref(this->hidden->stream);
this->hidden->stream = NULL;
}
if (this->hidden->context != NULL) {
PULSEAUDIO_pa_context_disconnect(this->hidden->context);
PULSEAUDIO_pa_context_unref(this->hidden->context);
this->hidden->context = NULL;
}
if (this->hidden->mainloop != NULL) {
PULSEAUDIO_pa_mainloop_free(this->hidden->mainloop);
this->hidden->mainloop = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
/* !!! FIXME: this could probably be expanded. */
/* Try to get the name of the program */
static char *
get_progname(void)
{
#ifdef __LINUX__
char *progname = NULL;
FILE *fp;
static char temp[BUFSIZ];
SDL_snprintf(temp, SDL_arraysize(temp), "/proc/%d/cmdline", getpid());
fp = fopen(temp, "r");
if (fp != NULL) {
if (fgets(temp, sizeof(temp) - 1, fp)) {
progname = SDL_strrchr(temp, '/');
if (progname == NULL) {
progname = temp;
} else {
progname = progname + 1;
}
}
fclose(fp);
}
return(progname);
#elif defined(__NetBSD__)
return getprogname();
#else
return("unknown");
#endif
}
static int
PULSEAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
{
struct SDL_PrivateAudioData *h = NULL;
Uint16 test_format = 0;
pa_sample_spec paspec;
pa_buffer_attr paattr;
pa_channel_map pacmap;
pa_stream_flags_t flags = 0;
int state = 0;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
h = this->hidden;
paspec.format = PA_SAMPLE_INVALID;
/* Try for a closest match on audio format */
for (test_format = SDL_FirstAudioFormat(this->spec.format);
(paspec.format == PA_SAMPLE_INVALID) && test_format;) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
switch (test_format) {
case AUDIO_U8:
paspec.format = PA_SAMPLE_U8;
break;
case AUDIO_S16LSB:
paspec.format = PA_SAMPLE_S16LE;
break;
case AUDIO_S16MSB:
paspec.format = PA_SAMPLE_S16BE;
break;
default:
paspec.format = PA_SAMPLE_INVALID;
break;
}
if (paspec.format == PA_SAMPLE_INVALID) {
test_format = SDL_NextAudioFormat();
}
}
if (paspec.format == PA_SAMPLE_INVALID) {
PULSEAUDIO_CloseDevice(this);
SDL_SetError("Couldn't find any hardware audio formats");
return 0;
}
this->spec.format = test_format;
/* Calculate the final parameters for this audio specification */
#ifdef PA_STREAM_ADJUST_LATENCY
this->spec.samples /= 2; /* Mix in smaller chunck to avoid underruns */
#endif
SDL_CalculateAudioSpec(&this->spec);
/* Allocate mixing buffer */
h->mixlen = this->spec.size;
h->mixbuf = (Uint8 *) SDL_AllocAudioMem(h->mixlen);
if (h->mixbuf == NULL) {
PULSEAUDIO_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(h->mixbuf, this->spec.silence, this->spec.size);
paspec.channels = this->spec.channels;
paspec.rate = this->spec.freq;
/* Reduced prebuffering compared to the defaults. */
#ifdef PA_STREAM_ADJUST_LATENCY
/* 2x original requested bufsize */
paattr.tlength = h->mixlen * 4;
paattr.prebuf = -1;
paattr.maxlength = -1;
/* -1 can lead to pa_stream_writable_size() >= mixlen never being true */
paattr.minreq = h->mixlen;
flags = PA_STREAM_ADJUST_LATENCY;
#else
paattr.tlength = h->mixlen*2;
paattr.prebuf = h->mixlen*2;
paattr.maxlength = h->mixlen*2;
paattr.minreq = h->mixlen;
#endif
/* The SDL ALSA output hints us that we use Windows' channel mapping */
/* http://bugzilla.libsdl.org/show_bug.cgi?id=110 */
PULSEAUDIO_pa_channel_map_init_auto(&pacmap, this->spec.channels,
PA_CHANNEL_MAP_WAVEEX);
/* Set up a new main loop */
if (!(h->mainloop = PULSEAUDIO_pa_mainloop_new())) {
PULSEAUDIO_CloseDevice(this);
SDL_SetError("pa_mainloop_new() failed");
return 0;
}
h->mainloop_api = PULSEAUDIO_pa_mainloop_get_api(h->mainloop);
h->context = PULSEAUDIO_pa_context_new(h->mainloop_api, get_progname());
if (!h->context) {
PULSEAUDIO_CloseDevice(this);
SDL_SetError("pa_context_new() failed");
return 0;
}
/* Connect to the PulseAudio server */
if (PULSEAUDIO_pa_context_connect(h->context, NULL, 0, NULL) < 0) {
PULSEAUDIO_CloseDevice(this);
SDL_SetError("Could not setup connection to PulseAudio");
return 0;
}
do {
if (PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
PULSEAUDIO_CloseDevice(this);
SDL_SetError("pa_mainloop_iterate() failed");
return 0;
}
state = PULSEAUDIO_pa_context_get_state(h->context);
if (!PA_CONTEXT_IS_GOOD(state)) {
PULSEAUDIO_CloseDevice(this);
SDL_SetError("Could not connect to PulseAudio");
return 0;
}
} while (state != PA_CONTEXT_READY);
h->stream = PULSEAUDIO_pa_stream_new(
h->context,
"Simple DirectMedia Layer", /* stream description */
&paspec, /* sample format spec */
&pacmap /* channel map */
);
if (h->stream == NULL) {
PULSEAUDIO_CloseDevice(this);
SDL_SetError("Could not set up PulseAudio stream");
return 0;
}
if (PULSEAUDIO_pa_stream_connect_playback(h->stream, NULL, &paattr, flags,
NULL, NULL) < 0) {
PULSEAUDIO_CloseDevice(this);
SDL_SetError("Could not connect PulseAudio stream");
return 0;
}
do {
if (PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
PULSEAUDIO_CloseDevice(this);
SDL_SetError("pa_mainloop_iterate() failed");
return 0;
}
state = PULSEAUDIO_pa_stream_get_state(h->stream);
if (!PA_STREAM_IS_GOOD(state)) {
PULSEAUDIO_CloseDevice(this);
SDL_SetError("Could not create to PulseAudio stream");
return 0;
}
} while (state != PA_STREAM_READY);
/* We're ready to rock and roll. :-) */
return 1;
}
static void
PULSEAUDIO_Deinitialize(void)
{
UnloadPulseAudioLibrary();
}
static int
PULSEAUDIO_Init(SDL_AudioDriverImpl * impl)
{
if (LoadPulseAudioLibrary() < 0) {
return 0;
}
/* Set the function pointers */
impl->OpenDevice = PULSEAUDIO_OpenDevice;
impl->PlayDevice = PULSEAUDIO_PlayDevice;
impl->WaitDevice = PULSEAUDIO_WaitDevice;
impl->GetDeviceBuf = PULSEAUDIO_GetDeviceBuf;
impl->CloseDevice = PULSEAUDIO_CloseDevice;
impl->WaitDone = PULSEAUDIO_WaitDone;
impl->Deinitialize = PULSEAUDIO_Deinitialize;
impl->OnlyHasDefaultOutputDevice = 1;
return 1; /* this audio target is available. */
}
AudioBootStrap PULSEAUDIO_bootstrap = {
PULSEAUDIO_DRIVER_NAME, "PulseAudio", PULSEAUDIO_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

49
project/jni/sdl-1.3/src/audio/pulseaudio/SDL_pulseaudio.h Normal file
View File

@@ -0,0 +1,49 @@
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_pulseaudio_h
#define _SDL_pulseaudio_h
#include <pulse/simple.h>
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
/* pulseaudio structures */
pa_mainloop *mainloop;
pa_mainloop_api *mainloop_api;
pa_context *context;
pa_stream *stream;
/* Raw mixing buffer */
Uint8 *mixbuf;
int mixlen;
};
#endif /* _SDL_pulseaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

893
project/jni/sdl-1.3/src/audio/qsa/SDL_qsa_audio.c Normal file
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@@ -0,0 +1,893 @@
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#include <errno.h>
#include <unistd.h>
#include <fcntl.h>
#include <signal.h>
#include <sys/types.h>
#include <sys/time.h>
#include <sched.h>
#include <sys/select.h>
#include <sys/neutrino.h>
#include <sys/asoundlib.h>
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_qsa_audio.h"
/* The tag name used by QSA audio framework */
#define DRIVER_NAME "qsa"
/* default channel communication parameters */
#define DEFAULT_CPARAMS_RATE 44100
#define DEFAULT_CPARAMS_VOICES 1
#define DEFAULT_CPARAMS_FRAG_SIZE 4096
#define DEFAULT_CPARAMS_FRAGS_MIN 1
#define DEFAULT_CPARAMS_FRAGS_MAX 1
#define QSA_NO_WORKAROUNDS 0x00000000
#define QSA_MMAP_WORKAROUND 0x00000001
struct BuggyCards
{
char *cardname;
unsigned long bugtype;
};
#define QSA_WA_CARDS 3
#define QSA_MAX_CARD_NAME_LENGTH 33
struct BuggyCards buggycards[QSA_WA_CARDS] = {
{"Sound Blaster Live!", QSA_MMAP_WORKAROUND},
{"Vortex 8820", QSA_MMAP_WORKAROUND},
{"Vortex 8830", QSA_MMAP_WORKAROUND},
};
/* List of found devices */
#define QSA_MAX_DEVICES 32
#define QSA_MAX_NAME_LENGTH 81+16 /* Hardcoded in QSA, can't be changed */
typedef struct _QSA_Device
{
char name[QSA_MAX_NAME_LENGTH]; /* Long audio device name for SDL */
int cardno;
int deviceno;
} QSA_Device;
QSA_Device qsa_playback_device[QSA_MAX_DEVICES];
uint32_t qsa_playback_devices;
QSA_Device qsa_capture_device[QSA_MAX_DEVICES];
uint32_t qsa_capture_devices;
static inline void
QSA_SetError(const char *fn, int status)
{
SDL_SetError("QSA: %s() failed: %s", fn, snd_strerror(status));
}
/* card names check to apply the workarounds */
static int
QSA_CheckBuggyCards(_THIS, unsigned long checkfor)
{
char scardname[QSA_MAX_CARD_NAME_LENGTH];
int it;
if (snd_card_get_name
(this->hidden->cardno, scardname, QSA_MAX_CARD_NAME_LENGTH - 1) < 0) {
return 0;
}
for (it = 0; it < QSA_WA_CARDS; it++) {
if (SDL_strcmp(buggycards[it].cardname, scardname) == 0) {
if (buggycards[it].bugtype == checkfor) {
return 1;
}
}
}
return 0;
}
static void
QSA_ThreadInit(_THIS)
{
struct sched_param param;
int status;
/* Increase default 10 priority to 25 to avoid jerky sound */
status = SchedGet(0, 0, &param);
param.sched_priority = param.sched_curpriority + 15;
status = SchedSet(0, 0, SCHED_NOCHANGE, &param);
}
/* PCM channel parameters initialize function */
static void
QSA_InitAudioParams(snd_pcm_channel_params_t * cpars)
{
SDL_memset(cpars, 0, sizeof(snd_pcm_channel_params_t));
cpars->channel = SND_PCM_CHANNEL_PLAYBACK;
cpars->mode = SND_PCM_MODE_BLOCK;
cpars->start_mode = SND_PCM_START_DATA;
cpars->stop_mode = SND_PCM_STOP_STOP;
cpars->format.format = SND_PCM_SFMT_S16_LE;
cpars->format.interleave = 1;
cpars->format.rate = DEFAULT_CPARAMS_RATE;
cpars->format.voices = DEFAULT_CPARAMS_VOICES;
cpars->buf.block.frag_size = DEFAULT_CPARAMS_FRAG_SIZE;
cpars->buf.block.frags_min = DEFAULT_CPARAMS_FRAGS_MIN;
cpars->buf.block.frags_max = DEFAULT_CPARAMS_FRAGS_MAX;
}
/* This function waits until it is possible to write a full sound buffer */
static void
QSA_WaitDevice(_THIS)
{
fd_set wfds;
fd_set rfds;
int selectret;
struct timeval timeout;
if (!this->hidden->iscapture) {
FD_ZERO(&wfds);
FD_SET(this->hidden->audio_fd, &wfds);
} else {
FD_ZERO(&rfds);
FD_SET(this->hidden->audio_fd, &rfds);
}
do {
/* Setup timeout for playing one fragment equal to 2 seconds */
/* If timeout occured than something wrong with hardware or driver */
/* For example, Vortex 8820 audio driver stucks on second DAC because */
/* it doesn't exist ! */
timeout.tv_sec = 2;
timeout.tv_usec = 0;
this->hidden->timeout_on_wait = 0;
if (!this->hidden->iscapture) {
selectret =
select(this->hidden->audio_fd + 1, NULL, &wfds, NULL,
&timeout);
} else {
selectret =
select(this->hidden->audio_fd + 1, &rfds, NULL, NULL,
&timeout);
}
switch (selectret) {
case -1:
{
SDL_SetError("QSA: select() failed: %s\n", strerror(errno));
return;
}
break;
case 0:
{
SDL_SetError("QSA: timeout on buffer waiting occured\n");
this->hidden->timeout_on_wait = 1;
return;
}
break;
default:
{
if (!this->hidden->iscapture) {
if (FD_ISSET(this->hidden->audio_fd, &wfds)) {
return;
}
} else {
if (FD_ISSET(this->hidden->audio_fd, &rfds)) {
return;
}
}
}
break;
}
} while (1);
}
static void
QSA_PlayDevice(_THIS)
{
snd_pcm_channel_status_t cstatus;
int written;
int status;
int towrite;
void *pcmbuffer;
if ((!this->enabled) || (!this->hidden)) {
return;
}
towrite = this->spec.size;
pcmbuffer = this->hidden->pcm_buf;
/* Write the audio data, checking for EAGAIN (buffer full) and underrun */
do {
written =
snd_pcm_plugin_write(this->hidden->audio_handle, pcmbuffer,
towrite);
if (written != towrite) {
/* Check if samples playback got stuck somewhere in hardware or in */
/* the audio device driver */
if ((errno == EAGAIN) && (written == 0)) {
if (this->hidden->timeout_on_wait != 0) {
SDL_SetError("QSA: buffer playback timeout\n");
return;
}
}
/* Check for errors or conditions */
if ((errno == EAGAIN) || (errno == EWOULDBLOCK)) {
/* Let a little CPU time go by and try to write again */
SDL_Delay(1);
/* if we wrote some data */
towrite -= written;
pcmbuffer += written * this->spec.channels;
continue;
} else {
if ((errno == EINVAL) || (errno == EIO)) {
SDL_memset(&cstatus, 0, sizeof(cstatus));
if (!this->hidden->iscapture) {
cstatus.channel = SND_PCM_CHANNEL_PLAYBACK;
} else {
cstatus.channel = SND_PCM_CHANNEL_CAPTURE;
}
status =
snd_pcm_plugin_status(this->hidden->audio_handle,
&cstatus);
if (status < 0) {
QSA_SetError("snd_pcm_plugin_status", status);
return;
}
if ((cstatus.status == SND_PCM_STATUS_UNDERRUN) ||
(cstatus.status == SND_PCM_STATUS_READY)) {
if (!this->hidden->iscapture) {
status =
snd_pcm_plugin_prepare(this->hidden->
audio_handle,
SND_PCM_CHANNEL_PLAYBACK);
} else {
status =
snd_pcm_plugin_prepare(this->hidden->
audio_handle,
SND_PCM_CHANNEL_CAPTURE);
}
if (status < 0) {
QSA_SetError("snd_pcm_plugin_prepare", status);
return;
}
}
continue;
} else {
return;
}
}
} else {
/* we wrote all remaining data */
towrite -= written;
pcmbuffer += written * this->spec.channels;
}
} while ((towrite > 0) && (this->enabled));
/* If we couldn't write, assume fatal error for now */
if (towrite != 0) {
this->enabled = 0;
}
}
static Uint8 *
QSA_GetDeviceBuf(_THIS)
{
return this->hidden->pcm_buf;
}
static void
QSA_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->audio_handle != NULL) {
if (!this->hidden->iscapture) {
/* Finish playing available samples */
snd_pcm_plugin_flush(this->hidden->audio_handle,
SND_PCM_CHANNEL_PLAYBACK);
} else {
/* Cancel unread samples during capture */
snd_pcm_plugin_flush(this->hidden->audio_handle,
SND_PCM_CHANNEL_CAPTURE);
}
snd_pcm_close(this->hidden->audio_handle);
this->hidden->audio_handle = NULL;
}
if (this->hidden->pcm_buf != NULL) {
SDL_FreeAudioMem(this->hidden->pcm_buf);
this->hidden->pcm_buf = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
static int
QSA_OpenDevice(_THIS, const char *devname, int iscapture)
{
int status = 0;
int format = 0;
SDL_AudioFormat test_format = 0;
int found = 0;
snd_pcm_channel_setup_t csetup;
snd_pcm_channel_params_t cparams;
/* Initialize all variables that we clean on shutdown */
this->hidden =
(struct SDL_PrivateAudioData *) SDL_calloc(1,
(sizeof
(struct
SDL_PrivateAudioData)));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, sizeof(struct SDL_PrivateAudioData));
/* Initialize channel transfer parameters to default */
QSA_InitAudioParams(&cparams);
/* Initialize channel direction: capture or playback */
this->hidden->iscapture = iscapture;
/* Find deviceid and cardid by device name for playback */
if ((!this->hidden->iscapture) && (devname != NULL)) {
uint32_t device;
int32_t status;
/* Search in the playback devices */
device = 0;
do {
status = SDL_strcmp(qsa_playback_device[device].name, devname);
if (status == 0) {
/* Found requested device */
this->hidden->deviceno = qsa_playback_device[device].deviceno;
this->hidden->cardno = qsa_playback_device[device].cardno;
break;
}
device++;
if (device >= qsa_playback_devices) {
QSA_CloseDevice(this);
SDL_SetError("No such playback device");
return 0;
}
} while (1);
}
/* Find deviceid and cardid by device name for capture */
if ((this->hidden->iscapture) && (devname != NULL)) {
/* Search in the capture devices */
uint32_t device;
int32_t status;
/* Searching in the playback devices */
device = 0;
do {
status = SDL_strcmp(qsa_capture_device[device].name, devname);
if (status == 0) {
/* Found requested device */
this->hidden->deviceno = qsa_capture_device[device].deviceno;
this->hidden->cardno = qsa_capture_device[device].cardno;
break;
}
device++;
if (device >= qsa_capture_devices) {
QSA_CloseDevice(this);
SDL_SetError("No such capture device");
return 0;
}
} while (1);
}
/* Check if SDL requested default audio device */
if (devname == NULL) {
/* Open system default audio device */
if (!this->hidden->iscapture) {
status = snd_pcm_open_preferred(&this->hidden->audio_handle,
&this->hidden->cardno,
&this->hidden->deviceno,
SND_PCM_OPEN_PLAYBACK);
} else {
status = snd_pcm_open_preferred(&this->hidden->audio_handle,
&this->hidden->cardno,
&this->hidden->deviceno,
SND_PCM_OPEN_CAPTURE);
}
} else {
/* Open requested audio device */
if (!this->hidden->iscapture) {
status =
snd_pcm_open(&this->hidden->audio_handle,
this->hidden->cardno, this->hidden->deviceno,
SND_PCM_OPEN_PLAYBACK);
} else {
status =
snd_pcm_open(&this->hidden->audio_handle,
this->hidden->cardno, this->hidden->deviceno,
SND_PCM_OPEN_CAPTURE);
}
}
/* Check if requested device is opened */
if (status < 0) {
this->hidden->audio_handle = NULL;
QSA_CloseDevice(this);
QSA_SetError("snd_pcm_open", status);
return 0;
}
if (!QSA_CheckBuggyCards(this, QSA_MMAP_WORKAROUND)) {
/* Disable QSA MMAP plugin for buggy audio drivers */
status =
snd_pcm_plugin_set_disable(this->hidden->audio_handle,
PLUGIN_DISABLE_MMAP);
if (status < 0) {
QSA_CloseDevice(this);
QSA_SetError("snd_pcm_plugin_set_disable", status);
return 0;
}
}
/* Try for a closest match on audio format */
format = 0;
/* can't use format as SND_PCM_SFMT_U8 = 0 in qsa */
found = 0;
for (test_format = SDL_FirstAudioFormat(this->spec.format); !found;) {
/* if match found set format to equivalent QSA format */
switch (test_format) {
case AUDIO_U8:
{
format = SND_PCM_SFMT_U8;
found = 1;
}
break;
case AUDIO_S8:
{
format = SND_PCM_SFMT_S8;
found = 1;
}
break;
case AUDIO_S16LSB:
{
format = SND_PCM_SFMT_S16_LE;
found = 1;
}
break;
case AUDIO_S16MSB:
{
format = SND_PCM_SFMT_S16_BE;
found = 1;
}
break;
case AUDIO_U16LSB:
{
format = SND_PCM_SFMT_U16_LE;
found = 1;
}
break;
case AUDIO_U16MSB:
{
format = SND_PCM_SFMT_U16_BE;
found = 1;
}
break;
case AUDIO_S32LSB:
{
format = SND_PCM_SFMT_S32_LE;
found = 1;
}
break;
case AUDIO_S32MSB:
{
format = SND_PCM_SFMT_S32_BE;
found = 1;
}
break;
case AUDIO_F32LSB:
{
format = SND_PCM_SFMT_FLOAT_LE;
found = 1;
}
break;
case AUDIO_F32MSB:
{
format = SND_PCM_SFMT_FLOAT_BE;
found = 1;
}
break;
default:
{
break;
}
}
if (!found) {
test_format = SDL_NextAudioFormat();
}
}
/* assumes test_format not 0 on success */
if (test_format == 0) {
QSA_CloseDevice(this);
SDL_SetError("QSA: Couldn't find any hardware audio formats");
return 0;
}
this->spec.format = test_format;
/* Set the audio format */
cparams.format.format = format;
/* Set mono/stereo/4ch/6ch/8ch audio */
cparams.format.voices = this->spec.channels;
/* Set rate */
cparams.format.rate = this->spec.freq;
/* Setup the transfer parameters according to cparams */
status = snd_pcm_plugin_params(this->hidden->audio_handle, &cparams);
if (status < 0) {
QSA_CloseDevice(this);
QSA_SetError("snd_pcm_channel_params", status);
return 0;
}
/* Make sure channel is setup right one last time */
SDL_memset(&csetup, '\0', sizeof(csetup));
if (!this->hidden->iscapture) {
csetup.channel = SND_PCM_CHANNEL_PLAYBACK;
} else {
csetup.channel = SND_PCM_CHANNEL_CAPTURE;
}
/* Setup an audio channel */
if (snd_pcm_plugin_setup(this->hidden->audio_handle, &csetup) < 0) {
QSA_CloseDevice(this);
SDL_SetError("QSA: Unable to setup channel\n");
return 0;
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
this->hidden->pcm_len = this->spec.size;
if (this->hidden->pcm_len == 0) {
this->hidden->pcm_len =
csetup.buf.block.frag_size * this->spec.channels *
(snd_pcm_format_width(format) / 8);
}
/*
* Allocate memory to the audio buffer and initialize with silence
* (Note that buffer size must be a multiple of fragment size, so find
* closest multiple)
*/
this->hidden->pcm_buf =
(Uint8 *) SDL_AllocAudioMem(this->hidden->pcm_len);
if (this->hidden->pcm_buf == NULL) {
QSA_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden->pcm_buf, this->spec.silence,
this->hidden->pcm_len);
/* get the file descriptor */
if (!this->hidden->iscapture) {
this->hidden->audio_fd =
snd_pcm_file_descriptor(this->hidden->audio_handle,
SND_PCM_CHANNEL_PLAYBACK);
} else {
this->hidden->audio_fd =
snd_pcm_file_descriptor(this->hidden->audio_handle,
SND_PCM_CHANNEL_CAPTURE);
}
if (this->hidden->audio_fd < 0) {
QSA_CloseDevice(this);
QSA_SetError("snd_pcm_file_descriptor", status);
return 0;
}
/* Prepare an audio channel */
if (!this->hidden->iscapture) {
/* Prepare audio playback */
status =
snd_pcm_plugin_prepare(this->hidden->audio_handle,
SND_PCM_CHANNEL_PLAYBACK);
} else {
/* Prepare audio capture */
status =
snd_pcm_plugin_prepare(this->hidden->audio_handle,
SND_PCM_CHANNEL_CAPTURE);
}
if (status < 0) {
QSA_CloseDevice(this);
QSA_SetError("snd_pcm_plugin_prepare", status);
return 0;
}
/* We're really ready to rock and roll. :-) */
return 1;
}
int
QSA_DetectDevices(int iscapture)
{
uint32_t it;
uint32_t cards;
uint32_t devices;
int32_t status;
/* Detect amount of available devices */
/* this value can be changed in the runtime */
cards = snd_cards();
/* If io-audio manager is not running we will get 0 as number */
/* of available audio devices */
if (cards == 0) {
/* We have no any available audio devices */
return 0;
}
/* Find requested devices by type */
if (!iscapture) {
/* Playback devices enumeration requested */
for (it = 0; it < cards; it++) {
devices = 0;
do {
status =
snd_card_get_longname(it,
qsa_playback_device
[qsa_playback_devices].name,
QSA_MAX_NAME_LENGTH);
if (status == EOK) {
snd_pcm_t *handle;
/* Add device number to device name */
sprintf(qsa_playback_device[qsa_playback_devices].name +
SDL_strlen(qsa_playback_device
[qsa_playback_devices].name), " d%d",
devices);
/* Store associated card number id */
qsa_playback_device[qsa_playback_devices].cardno = it;
/* Check if this device id could play anything */
status =
snd_pcm_open(&handle, it, devices,
SND_PCM_OPEN_PLAYBACK);
if (status == EOK) {
qsa_playback_device[qsa_playback_devices].deviceno =
devices;
status = snd_pcm_close(handle);
if (status == EOK) {
qsa_playback_devices++;
}
} else {
/* Check if we got end of devices list */
if (status == -ENOENT) {
break;
}
}
} else {
break;
}
/* Check if we reached maximum devices count */
if (qsa_playback_devices >= QSA_MAX_DEVICES) {
break;
}
devices++;
} while (1);
/* Check if we reached maximum devices count */
if (qsa_playback_devices >= QSA_MAX_DEVICES) {
break;
}
}
} else {
/* Capture devices enumeration requested */
for (it = 0; it < cards; it++) {
devices = 0;
do {
status =
snd_card_get_longname(it,
qsa_capture_device
[qsa_capture_devices].name,
QSA_MAX_NAME_LENGTH);
if (status == EOK) {
snd_pcm_t *handle;
/* Add device number to device name */
sprintf(qsa_capture_device[qsa_capture_devices].name +
SDL_strlen(qsa_capture_device
[qsa_capture_devices].name), " d%d",
devices);
/* Store associated card number id */
qsa_capture_device[qsa_capture_devices].cardno = it;
/* Check if this device id could play anything */
status =
snd_pcm_open(&handle, it, devices,
SND_PCM_OPEN_CAPTURE);
if (status == EOK) {
qsa_capture_device[qsa_capture_devices].deviceno =
devices;
status = snd_pcm_close(handle);
if (status == EOK) {
qsa_capture_devices++;
}
} else {
/* Check if we got end of devices list */
if (status == -ENOENT) {
break;
}
}
/* Check if we reached maximum devices count */
if (qsa_capture_devices >= QSA_MAX_DEVICES) {
break;
}
} else {
break;
}
devices++;
} while (1);
/* Check if we reached maximum devices count */
if (qsa_capture_devices >= QSA_MAX_DEVICES) {
break;
}
}
}
/* Return amount of available playback or capture devices */
if (!iscapture) {
return qsa_playback_devices;
} else {
return qsa_capture_devices;
}
}
const char *
QSA_GetDeviceName(int index, int iscapture)
{
if (!iscapture) {
if (index >= qsa_playback_devices) {
return "No such playback device";
}
return qsa_playback_device[index].name;
} else {
if (index >= qsa_capture_devices) {
return "No such capture device";
}
return qsa_capture_device[index].name;
}
}
void
QSA_WaitDone(_THIS)
{
if (!this->hidden->iscapture) {
if (this->hidden->audio_handle != NULL) {
/* Wait till last fragment is played and stop channel */
snd_pcm_plugin_flush(this->hidden->audio_handle,
SND_PCM_CHANNEL_PLAYBACK);
}
} else {
if (this->hidden->audio_handle != NULL) {
/* Discard all unread data and stop channel */
snd_pcm_plugin_flush(this->hidden->audio_handle,
SND_PCM_CHANNEL_CAPTURE);
}
}
}
void
QSA_Deinitialize(void)
{
/* Clear devices array on shutdown */
SDL_memset(qsa_playback_device, 0x00,
sizeof(QSA_Device) * QSA_MAX_DEVICES);
SDL_memset(qsa_capture_device, 0x00,
sizeof(QSA_Device) * QSA_MAX_DEVICES);
qsa_playback_devices = 0;
qsa_capture_devices = 0;
}
static int
QSA_Init(SDL_AudioDriverImpl * impl)
{
snd_pcm_t *handle = NULL;
int32_t status = 0;
/* Clear devices array */
SDL_memset(qsa_playback_device, 0x00,
sizeof(QSA_Device) * QSA_MAX_DEVICES);
SDL_memset(qsa_capture_device, 0x00,
sizeof(QSA_Device) * QSA_MAX_DEVICES);
qsa_playback_devices = 0;
qsa_capture_devices = 0;
/* Set function pointers */
/* DeviceLock and DeviceUnlock functions are used default, */
/* provided by SDL, which uses pthread_mutex for lock/unlock */
impl->DetectDevices = QSA_DetectDevices;
impl->GetDeviceName = QSA_GetDeviceName;
impl->OpenDevice = QSA_OpenDevice;
impl->ThreadInit = QSA_ThreadInit;
impl->WaitDevice = QSA_WaitDevice;
impl->PlayDevice = QSA_PlayDevice;
impl->GetDeviceBuf = QSA_GetDeviceBuf;
impl->CloseDevice = QSA_CloseDevice;
impl->WaitDone = QSA_WaitDone;
impl->Deinitialize = QSA_Deinitialize;
impl->LockDevice = NULL;
impl->UnlockDevice = NULL;
impl->OnlyHasDefaultOutputDevice = 0;
impl->ProvidesOwnCallbackThread = 0;
impl->SkipMixerLock = 0;
impl->HasCaptureSupport = 1;
impl->OnlyHasDefaultOutputDevice = 0;
impl->OnlyHasDefaultInputDevice = 0;
/* Check if io-audio manager is running or not */
status = snd_cards();
if (status == 0) {
/* if no, return immediately */
return 1;
}
return 1; /* this audio target is available. */
}
AudioBootStrap QSAAUDIO_bootstrap = {
DRIVER_NAME, "QNX QSA Audio", QSA_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

58
project/jni/sdl-1.3/src/audio/qsa/SDL_qsa_audio.h Normal file
View File

@@ -0,0 +1,58 @@
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef __SDL_QSA_AUDIO_H__
#define __SDL_QSA_AUDIO_H__
#include <sys/asoundlib.h>
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice* this
struct SDL_PrivateAudioData
{
/* SDL capture state */
int iscapture;
/* The audio device handle */
int cardno;
int deviceno;
snd_pcm_t *audio_handle;
/* The audio file descriptor */
int audio_fd;
/* Select timeout status */
uint32_t timeout_on_wait;
/* Raw mixing buffer */
Uint8 *pcm_buf;
Uint32 pcm_len;
};
#endif /* __SDL_QSA_AUDIO_H__ */
/* vi: set ts=4 sw=4 expandtab: */

762
project/jni/sdl-1.3/src/audio/sdlgenaudiocvt.pl Executable file
View File

@@ -0,0 +1,762 @@
#!/usr/bin/perl -w
use warnings;
use strict;
my @audiotypes = qw(
U8
S8
U16LSB
S16LSB
U16MSB
S16MSB
S32LSB
S32MSB
F32LSB
F32MSB
);
my @channels = ( 1, 2, 4, 6, 8 );
my %funcs;
my $custom_converters = 0;
sub getTypeConvertHashId {
my ($from, $to) = @_;
return "TYPECONVERTER $from/$to";
}
sub getResamplerHashId {
my ($from, $channels, $upsample, $multiple) = @_;
return "RESAMPLER $from/$channels/$upsample/$multiple";
}
sub outputHeader {
print <<EOF;
/* DO NOT EDIT! This file is generated by sdlgenaudiocvt.pl */
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken\@libsdl.org
*/
#include "SDL_config.h"
#include "SDL_audio.h"
#include "SDL_audio_c.h"
#ifndef DEBUG_CONVERT
#define DEBUG_CONVERT 0
#endif
/* If you can guarantee your data and need space, you can eliminate code... */
/* Just build the arbitrary resamplers if you're saving code space. */
#ifndef LESS_RESAMPLERS
#define LESS_RESAMPLERS 0
#endif
/* Don't build any resamplers if you're REALLY saving code space. */
#ifndef NO_RESAMPLERS
#define NO_RESAMPLERS 0
#endif
/* Don't build any type converters if you're saving code space. */
#ifndef NO_CONVERTERS
#define NO_CONVERTERS 0
#endif
/* *INDENT-OFF* */
EOF
my @vals = ( 127, 32767, 2147483647 );
foreach (@vals) {
my $val = $_;
my $fval = 1.0 / $val;
print("#define DIVBY${val} ${fval}f\n");
}
print("\n");
}
sub outputFooter {
print <<EOF;
/* $custom_converters converters generated. */
/* *INDENT-ON* */
/* vi: set ts=4 sw=4 expandtab: */
EOF
}
sub splittype {
my $t = shift;
my ($signed, $size, $endian) = $t =~ /([USF])(\d+)([LM]SB|)/;
my $float = ($signed eq 'F') ? 1 : 0;
$signed = (($float) or ($signed eq 'S')) ? 1 : 0;
$endian = 'NONE' if ($endian eq '');
my $ctype = '';
if ($float) {
$ctype = (($size == 32) ? 'float' : 'double');
} else {
$ctype = (($signed) ? 'S' : 'U') . "int${size}";
}
return ($signed, $float, $size, $endian, $ctype);
}
sub getSwapFunc {
my ($size, $signed, $float, $endian, $val) = @_;
my $BEorLE = (($endian eq 'MSB') ? 'BE' : 'LE');
my $code = '';
if ($float) {
$code = "SDL_SwapFloat${BEorLE}($val)";
} else {
if ($size > 8) {
$code = "SDL_Swap${BEorLE}${size}($val)";
} else {
$code = $val;
}
if (($signed) and (!$float)) {
$code = "((Sint${size}) $code)";
}
}
return "${code}";
}
sub maxIntVal {
my $size = shift;
if ($size == 8) {
return 0x7F;
} elsif ($size == 16) {
return 0x7FFF;
} elsif ($size == 32) {
return 0x7FFFFFFF;
}
die("bug in script.\n");
}
sub getFloatToIntMult {
my $size = shift;
my $val = maxIntVal($size) . '.0';
$val .= 'f' if ($size < 32);
return $val;
}
sub getIntToFloatDivBy {
my $size = shift;
return 'DIVBY' . maxIntVal($size);
}
sub getSignFlipVal {
my $size = shift;
if ($size == 8) {
return '0x80';
} elsif ($size == 16) {
return '0x8000';
} elsif ($size == 32) {
return '0x80000000';
}
die("bug in script.\n");
}
sub buildCvtFunc {
my ($from, $to) = @_;
my ($fsigned, $ffloat, $fsize, $fendian, $fctype) = splittype($from);
my ($tsigned, $tfloat, $tsize, $tendian, $tctype) = splittype($to);
my $diffs = 0;
$diffs++ if ($fsize != $tsize);
$diffs++ if ($fsigned != $tsigned);
$diffs++ if ($ffloat != $tfloat);
$diffs++ if ($fendian ne $tendian);
return if ($diffs == 0);
my $hashid = getTypeConvertHashId($from, $to);
if (1) { # !!! FIXME: if ($diffs > 1) {
my $sym = "SDL_Convert_${from}_to_${to}";
$funcs{$hashid} = $sym;
$custom_converters++;
# Always unsigned for ints, for possible byteswaps.
my $srctype = (($ffloat) ? 'float' : "Uint${fsize}");
print <<EOF;
static void SDLCALL
${sym}(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
int i;
const $srctype *src;
$tctype *dst;
#if DEBUG_CONVERT
fprintf(stderr, "Converting AUDIO_${from} to AUDIO_${to}.\\n");
#endif
EOF
if ($fsize < $tsize) {
my $mult = $tsize / $fsize;
print <<EOF;
src = ((const $srctype *) (cvt->buf + cvt->len_cvt)) - 1;
dst = (($tctype *) (cvt->buf + cvt->len_cvt * $mult)) - 1;
for (i = cvt->len_cvt / sizeof ($srctype); i; --i, --src, --dst) {
EOF
} else {
print <<EOF;
src = (const $srctype *) cvt->buf;
dst = ($tctype *) cvt->buf;
for (i = cvt->len_cvt / sizeof ($srctype); i; --i, ++src, ++dst) {
EOF
}
# Have to convert to/from float/int.
# !!! FIXME: cast through double for int32<->float?
my $code = getSwapFunc($fsize, $fsigned, $ffloat, $fendian, '*src');
if ($ffloat != $tfloat) {
if ($ffloat) {
my $mult = getFloatToIntMult($tsize);
if (!$tsigned) { # bump from -1.0f/1.0f to 0.0f/2.0f
$code = "($code + 1.0f)";
}
$code = "(($tctype) ($code * $mult))";
} else {
# $divby will be the reciprocal, to avoid pipeline stalls
# from floating point division...so multiply it.
my $divby = getIntToFloatDivBy($fsize);
$code = "(((float) $code) * $divby)";
if (!$fsigned) { # bump from 0.0f/2.0f to -1.0f/1.0f.
$code = "($code - 1.0f)";
}
}
} else {
# All integer conversions here.
if ($fsigned != $tsigned) {
my $signflipval = getSignFlipVal($fsize);
$code = "(($code) ^ $signflipval)";
}
my $shiftval = abs($fsize - $tsize);
if ($fsize < $tsize) {
$code = "((($tctype) $code) << $shiftval)";
} elsif ($fsize > $tsize) {
$code = "(($tctype) ($code >> $shiftval))";
}
}
my $swap = getSwapFunc($tsize, $tsigned, $tfloat, $tendian, 'val');
print <<EOF;
const $tctype val = $code;
*dst = ${swap};
}
EOF
if ($fsize > $tsize) {
my $divby = $fsize / $tsize;
print(" cvt->len_cvt /= $divby;\n");
} elsif ($fsize < $tsize) {
my $mult = $tsize / $fsize;
print(" cvt->len_cvt *= $mult;\n");
}
print <<EOF;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, AUDIO_$to);
}
}
EOF
} else {
if ($fsigned != $tsigned) {
$funcs{$hashid} = 'SDL_ConvertSigned';
} elsif ($ffloat != $tfloat) {
$funcs{$hashid} = 'SDL_ConvertFloat';
} elsif ($fsize != $tsize) {
$funcs{$hashid} = 'SDL_ConvertSize';
} elsif ($fendian ne $tendian) {
$funcs{$hashid} = 'SDL_ConvertEndian';
} else {
die("error in script.\n");
}
}
}
sub buildTypeConverters {
print "#if !NO_CONVERTERS\n\n";
foreach (@audiotypes) {
my $from = $_;
foreach (@audiotypes) {
my $to = $_;
buildCvtFunc($from, $to);
}
}
print "#endif /* !NO_CONVERTERS */\n\n\n";
print "const SDL_AudioTypeFilters sdl_audio_type_filters[] =\n{\n";
print "#if !NO_CONVERTERS\n";
foreach (@audiotypes) {
my $from = $_;
foreach (@audiotypes) {
my $to = $_;
if ($from ne $to) {
my $hashid = getTypeConvertHashId($from, $to);
my $sym = $funcs{$hashid};
print(" { AUDIO_$from, AUDIO_$to, $sym },\n");
}
}
}
print "#endif /* !NO_CONVERTERS */\n";
print(" { 0, 0, NULL }\n");
print "};\n\n\n";
}
sub getBiggerCtype {
my ($isfloat, $size) = @_;
if ($isfloat) {
if ($size == 32) {
return 'double';
}
die("bug in script.\n");
}
if ($size == 8) {
return 'Sint16';
} elsif ($size == 16) {
return 'Sint32'
} elsif ($size == 32) {
return 'Sint64'
}
die("bug in script.\n");
}
# These handle arbitrary resamples...44100Hz to 48000Hz, for example.
# Man, this code is skanky.
sub buildArbitraryResampleFunc {
# !!! FIXME: we do a lot of unnecessary and ugly casting in here, due to getSwapFunc().
my ($from, $channels, $upsample) = @_;
my ($fsigned, $ffloat, $fsize, $fendian, $fctype) = splittype($from);
my $bigger = getBiggerCtype($ffloat, $fsize);
my $interp = ($ffloat) ? '* 0.5' : '>> 1';
my $resample = ($upsample) ? 'Upsample' : 'Downsample';
my $hashid = getResamplerHashId($from, $channels, $upsample, 0);
my $sym = "SDL_${resample}_${from}_${channels}c";
$funcs{$hashid} = $sym;
$custom_converters++;
my $fudge = $fsize * $channels * 2; # !!! FIXME
my $eps_adjust = ($upsample) ? 'dstsize' : 'srcsize';
my $incr = '';
my $incr2 = '';
# !!! FIXME: DEBUG_CONVERT should report frequencies.
print <<EOF;
static void SDLCALL
${sym}(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
#if DEBUG_CONVERT
fprintf(stderr, "$resample arbitrary (x%f) AUDIO_${from}, ${channels} channels.\\n", cvt->rate_incr);
#endif
const int srcsize = cvt->len_cvt - $fudge;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
register int eps = 0;
EOF
my $endcomparison = '!=';
# Upsampling (growing the buffer) needs to work backwards, since we
# overwrite the buffer as we go.
if ($upsample) {
$endcomparison = '>'; # dst > target
print <<EOF;
$fctype *dst = (($fctype *) (cvt->buf + dstsize)) - $channels;
const $fctype *src = (($fctype *) (cvt->buf + cvt->len_cvt)) - $channels;
const $fctype *target = ((const $fctype *) cvt->buf) - $channels;
EOF
} else {
$endcomparison = '<'; # dst < target
print <<EOF;
$fctype *dst = ($fctype *) cvt->buf;
const $fctype *src = ($fctype *) cvt->buf;
const $fctype *target = (const $fctype *) (cvt->buf + dstsize);
EOF
}
for (my $i = 0; $i < $channels; $i++) {
my $idx = ($upsample) ? (($channels - $i) - 1) : $i;
my $val = getSwapFunc($fsize, $fsigned, $ffloat, $fendian, "src[$idx]");
print <<EOF;
$fctype sample${idx} = $val;
EOF
}
for (my $i = 0; $i < $channels; $i++) {
my $idx = ($upsample) ? (($channels - $i) - 1) : $i;
print <<EOF;
$fctype last_sample${idx} = sample${idx};
EOF
}
print <<EOF;
while (dst $endcomparison target) {
EOF
if ($upsample) {
for (my $i = 0; $i < $channels; $i++) {
# !!! FIXME: don't do this swap every write, just when the samples change.
my $idx = (($channels - $i) - 1);
my $val = getSwapFunc($fsize, $fsigned, $ffloat, $fendian, "sample${idx}");
print <<EOF;
dst[$idx] = $val;
EOF
}
$incr = ($channels == 1) ? 'dst--' : "dst -= $channels";
$incr2 = ($channels == 1) ? 'src--' : "src -= $channels";
print <<EOF;
$incr;
eps += srcsize;
if ((eps << 1) >= dstsize) {
$incr2;
EOF
} else { # downsample.
$incr = ($channels == 1) ? 'src++' : "src += $channels";
print <<EOF;
$incr;
eps += dstsize;
if ((eps << 1) >= srcsize) {
EOF
for (my $i = 0; $i < $channels; $i++) {
my $val = getSwapFunc($fsize, $fsigned, $ffloat, $fendian, "sample${i}");
print <<EOF;
dst[$i] = $val;
EOF
}
$incr = ($channels == 1) ? 'dst++' : "dst += $channels";
print <<EOF;
$incr;
EOF
}
for (my $i = 0; $i < $channels; $i++) {
my $idx = ($upsample) ? (($channels - $i) - 1) : $i;
my $swapped = getSwapFunc($fsize, $fsigned, $ffloat, $fendian, "src[$idx]");
print <<EOF;
sample${idx} = ($fctype) (((($bigger) $swapped) + (($bigger) last_sample${idx})) $interp);
EOF
}
for (my $i = 0; $i < $channels; $i++) {
my $idx = ($upsample) ? (($channels - $i) - 1) : $i;
print <<EOF;
last_sample${idx} = sample${idx};
EOF
}
print <<EOF;
eps -= $eps_adjust;
}
}
EOF
print <<EOF;
cvt->len_cvt = dstsize;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
EOF
}
# These handle clean resamples...doubling and quadrupling the sample rate, etc.
sub buildMultipleResampleFunc {
# !!! FIXME: we do a lot of unnecessary and ugly casting in here, due to getSwapFunc().
my ($from, $channels, $upsample, $multiple) = @_;
my ($fsigned, $ffloat, $fsize, $fendian, $fctype) = splittype($from);
my $bigger = getBiggerCtype($ffloat, $fsize);
my $interp = ($ffloat) ? '* 0.5' : '>> 1';
my $interp2 = ($ffloat) ? '* 0.25' : '>> 2';
my $mult3 = ($ffloat) ? '3.0' : '3';
my $lencvtop = ($upsample) ? '*' : '/';
my $resample = ($upsample) ? 'Upsample' : 'Downsample';
my $hashid = getResamplerHashId($from, $channels, $upsample, $multiple);
my $sym = "SDL_${resample}_${from}_${channels}c_x${multiple}";
$funcs{$hashid} = $sym;
$custom_converters++;
# !!! FIXME: DEBUG_CONVERT should report frequencies.
print <<EOF;
static void SDLCALL
${sym}(SDL_AudioCVT * cvt, SDL_AudioFormat format)
{
#if DEBUG_CONVERT
fprintf(stderr, "$resample (x${multiple}) AUDIO_${from}, ${channels} channels.\\n");
#endif
const int srcsize = cvt->len_cvt;
const int dstsize = cvt->len_cvt $lencvtop $multiple;
EOF
my $endcomparison = '!=';
# Upsampling (growing the buffer) needs to work backwards, since we
# overwrite the buffer as we go.
if ($upsample) {
$endcomparison = '>'; # dst > target
print <<EOF;
$fctype *dst = (($fctype *) (cvt->buf + dstsize)) - $channels;
const $fctype *src = (($fctype *) (cvt->buf + cvt->len_cvt)) - $channels;
const $fctype *target = ((const $fctype *) cvt->buf) - $channels;
EOF
} else {
$endcomparison = '<'; # dst < target
print <<EOF;
$fctype *dst = ($fctype *) cvt->buf;
const $fctype *src = ($fctype *) cvt->buf;
const $fctype *target = (const $fctype *) (cvt->buf + dstsize);
EOF
}
for (my $i = 0; $i < $channels; $i++) {
my $idx = ($upsample) ? (($channels - $i) - 1) : $i;
my $val = getSwapFunc($fsize, $fsigned, $ffloat, $fendian, "src[$idx]");
print <<EOF;
$bigger last_sample${idx} = ($bigger) $val;
EOF
}
print <<EOF;
while (dst $endcomparison target) {
EOF
for (my $i = 0; $i < $channels; $i++) {
my $idx = ($upsample) ? (($channels - $i) - 1) : $i;
my $val = getSwapFunc($fsize, $fsigned, $ffloat, $fendian, "src[$idx]");
print <<EOF;
const $bigger sample${idx} = ($bigger) $val;
EOF
}
my $incr = '';
if ($upsample) {
$incr = ($channels == 1) ? 'src--' : "src -= $channels";
} else {
my $amount = $channels * $multiple;
$incr = "src += $amount"; # can't ever be 1, so no "++" version.
}
print <<EOF;
$incr;
EOF
# !!! FIXME: This really begs for some Altivec or SSE, etc.
if ($upsample) {
if ($multiple == 2) {
for (my $i = $channels-1; $i >= 0; $i--) {
my $dsti = $i + $channels;
print <<EOF;
dst[$dsti] = ($fctype) ((sample${i} + last_sample${i}) $interp);
EOF
}
for (my $i = $channels-1; $i >= 0; $i--) {
my $dsti = $i;
print <<EOF;
dst[$dsti] = ($fctype) sample${i};
EOF
}
} elsif ($multiple == 4) {
for (my $i = $channels-1; $i >= 0; $i--) {
my $dsti = $i + ($channels * 3);
print <<EOF;
dst[$dsti] = ($fctype) sample${i};
EOF
}
for (my $i = $channels-1; $i >= 0; $i--) {
my $dsti = $i + ($channels * 2);
print <<EOF;
dst[$dsti] = ($fctype) ((($mult3 * sample${i}) + last_sample${i}) $interp2);
EOF
}
for (my $i = $channels-1; $i >= 0; $i--) {
my $dsti = $i + ($channels * 1);
print <<EOF;
dst[$dsti] = ($fctype) ((sample${i} + last_sample${i}) $interp);
EOF
}
for (my $i = $channels-1; $i >= 0; $i--) {
my $dsti = $i + ($channels * 0);
print <<EOF;
dst[$dsti] = ($fctype) ((sample${i} + ($mult3 * last_sample${i})) $interp2);
EOF
}
} else {
die('bug in program.'); # we only handle x2 and x4.
}
} else { # downsample.
if ($multiple == 2) {
for (my $i = 0; $i < $channels; $i++) {
print <<EOF;
dst[$i] = ($fctype) ((sample${i} + last_sample${i}) $interp);
EOF
}
} elsif ($multiple == 4) {
# !!! FIXME: interpolate all 4 samples?
for (my $i = 0; $i < $channels; $i++) {
print <<EOF;
dst[$i] = ($fctype) ((sample${i} + last_sample${i}) $interp);
EOF
}
} else {
die('bug in program.'); # we only handle x2 and x4.
}
}
for (my $i = 0; $i < $channels; $i++) {
my $idx = ($upsample) ? (($channels - $i) - 1) : $i;
print <<EOF;
last_sample${idx} = sample${idx};
EOF
}
if ($upsample) {
my $amount = $channels * $multiple;
$incr = "dst -= $amount"; # can't ever be 1, so no "--" version.
} else {
$incr = ($channels == 1) ? 'dst++' : "dst += $channels";
}
print <<EOF;
$incr;
}
cvt->len_cvt = dstsize;
if (cvt->filters[++cvt->filter_index]) {
cvt->filters[cvt->filter_index] (cvt, format);
}
}
EOF
}
sub buildResamplers {
print "#if !NO_RESAMPLERS\n\n";
foreach (@audiotypes) {
my $from = $_;
foreach (@channels) {
my $channel = $_;
buildArbitraryResampleFunc($from, $channel, 1);
buildArbitraryResampleFunc($from, $channel, 0);
}
}
print "\n#if !LESS_RESAMPLERS\n\n";
foreach (@audiotypes) {
my $from = $_;
foreach (@channels) {
my $channel = $_;
for (my $multiple = 2; $multiple <= 4; $multiple += 2) {
buildMultipleResampleFunc($from, $channel, 1, $multiple);
buildMultipleResampleFunc($from, $channel, 0, $multiple);
}
}
}
print "#endif /* !LESS_RESAMPLERS */\n";
print "#endif /* !NO_RESAMPLERS */\n\n\n";
print "const SDL_AudioRateFilters sdl_audio_rate_filters[] =\n{\n";
print "#if !NO_RESAMPLERS\n";
foreach (@audiotypes) {
my $from = $_;
foreach (@channels) {
my $channel = $_;
for (my $upsample = 0; $upsample <= 1; $upsample++) {
my $hashid = getResamplerHashId($from, $channel, $upsample, 0);
my $sym = $funcs{$hashid};
print(" { AUDIO_$from, $channel, $upsample, 0, $sym },\n");
}
}
}
print "#if !LESS_RESAMPLERS\n";
foreach (@audiotypes) {
my $from = $_;
foreach (@channels) {
my $channel = $_;
for (my $multiple = 2; $multiple <= 4; $multiple += 2) {
for (my $upsample = 0; $upsample <= 1; $upsample++) {
my $hashid = getResamplerHashId($from, $channel, $upsample, $multiple);
my $sym = $funcs{$hashid};
print(" { AUDIO_$from, $channel, $upsample, $multiple, $sym },\n");
}
}
}
}
print "#endif /* !LESS_RESAMPLERS */\n";
print "#endif /* !NO_RESAMPLERS */\n";
print(" { 0, 0, 0, 0, NULL }\n");
print "};\n\n";
}
# mainline ...
outputHeader();
buildTypeConverters();
buildResamplers();
outputFooter();
exit 0;
# end of sdlgenaudiocvt.pl ...

454
project/jni/sdl-1.3/src/audio/sun/SDL_sunaudio.c Normal file
View File

@@ -0,0 +1,454 @@
/* I'm gambling no one uses this audio backend...we'll see who emails. :) */
#error this code has not been updated for SDL 1.3.
#error if no one emails icculus at icculus.org and tells him that this
#error code is needed, this audio backend will eventually be removed from SDL.
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* Allow access to a raw mixing buffer */
#include <fcntl.h>
#include <errno.h>
#ifdef __NETBSD__
#include <sys/ioctl.h>
#include <sys/audioio.h>
#endif
#ifdef __SVR4
#include <sys/audioio.h>
#else
#include <sys/time.h>
#include <sys/types.h>
#endif
#include <unistd.h>
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "../SDL_audiodev_c.h"
#include "SDL_sunaudio.h"
/* Open the audio device for playback, and don't block if busy */
#define OPEN_FLAGS (O_WRONLY|O_NONBLOCK)
/* Audio driver functions */
static int DSP_OpenAudio(_THIS, SDL_AudioSpec * spec);
static void DSP_WaitAudio(_THIS);
static void DSP_PlayAudio(_THIS);
static Uint8 *DSP_GetAudioBuf(_THIS);
static void DSP_CloseAudio(_THIS);
static Uint8 snd2au(int sample);
/* Audio driver bootstrap functions */
static int
Audio_Available(void)
{
int fd;
int available;
available = 0;
fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 1);
if (fd >= 0) {
available = 1;
close(fd);
}
return (available);
}
static void
Audio_DeleteDevice(SDL_AudioDevice * device)
{
SDL_free(device->hidden);
SDL_free(device);
}
static SDL_AudioDevice *
Audio_CreateDevice(int devindex)
{
SDL_AudioDevice *this;
/* Initialize all variables that we clean on shutdown */
this = (SDL_AudioDevice *) SDL_malloc(sizeof(SDL_AudioDevice));
if (this) {
SDL_memset(this, 0, (sizeof *this));
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
}
if ((this == NULL) || (this->hidden == NULL)) {
SDL_OutOfMemory();
if (this) {
SDL_free(this);
}
return (0);
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
audio_fd = -1;
/* Set the function pointers */
this->OpenAudio = DSP_OpenAudio;
this->WaitAudio = DSP_WaitAudio;
this->PlayAudio = DSP_PlayAudio;
this->GetAudioBuf = DSP_GetAudioBuf;
this->CloseAudio = DSP_CloseAudio;
this->free = Audio_DeleteDevice;
return this;
}
AudioBootStrap SUNAUDIO_bootstrap = {
"audio", "UNIX /dev/audio interface",
Audio_Available, Audio_CreateDevice, 0
};
#ifdef DEBUG_AUDIO
void
CheckUnderflow(_THIS)
{
#ifdef AUDIO_GETINFO
audio_info_t info;
int left;
ioctl(audio_fd, AUDIO_GETINFO, &info);
left = (written - info.play.samples);
if (written && (left == 0)) {
fprintf(stderr, "audio underflow!\n");
}
#endif
}
#endif
void
DSP_WaitAudio(_THIS)
{
#ifdef AUDIO_GETINFO
#define SLEEP_FUDGE 10 /* 10 ms scheduling fudge factor */
audio_info_t info;
Sint32 left;
ioctl(audio_fd, AUDIO_GETINFO, &info);
left = (written - info.play.samples);
if (left > fragsize) {
Sint32 sleepy;
sleepy = ((left - fragsize) / frequency);
sleepy -= SLEEP_FUDGE;
if (sleepy > 0) {
SDL_Delay(sleepy);
}
}
#else
fd_set fdset;
FD_ZERO(&fdset);
FD_SET(audio_fd, &fdset);
select(audio_fd + 1, NULL, &fdset, NULL, NULL);
#endif
}
void
DSP_PlayAudio(_THIS)
{
/* Write the audio data */
if (ulaw_only) {
/* Assuming that this->spec.freq >= 8000 Hz */
int accum, incr, pos;
Uint8 *aubuf;
accum = 0;
incr = this->spec.freq / 8;
aubuf = ulaw_buf;
switch (audio_fmt & 0xFF) {
case 8:
{
Uint8 *sndbuf;
sndbuf = mixbuf;
for (pos = 0; pos < fragsize; ++pos) {
*aubuf = snd2au((0x80 - *sndbuf) * 64);
accum += incr;
while (accum > 0) {
accum -= 1000;
sndbuf += 1;
}
aubuf += 1;
}
}
break;
case 16:
{
Sint16 *sndbuf;
sndbuf = (Sint16 *) mixbuf;
for (pos = 0; pos < fragsize; ++pos) {
*aubuf = snd2au(*sndbuf / 4);
accum += incr;
while (accum > 0) {
accum -= 1000;
sndbuf += 1;
}
aubuf += 1;
}
}
break;
}
#ifdef DEBUG_AUDIO
CheckUnderflow(this);
#endif
if (write(audio_fd, ulaw_buf, fragsize) < 0) {
/* Assume fatal error, for now */
this->enabled = 0;
}
written += fragsize;
} else {
#ifdef DEBUG_AUDIO
CheckUnderflow(this);
#endif
if (write(audio_fd, mixbuf, this->spec.size) < 0) {
/* Assume fatal error, for now */
this->enabled = 0;
}
written += fragsize;
}
}
Uint8 *
DSP_GetAudioBuf(_THIS)
{
return (mixbuf);
}
void
DSP_CloseAudio(_THIS)
{
if (mixbuf != NULL) {
SDL_FreeAudioMem(mixbuf);
mixbuf = NULL;
}
if (ulaw_buf != NULL) {
SDL_free(ulaw_buf);
ulaw_buf = NULL;
}
close(audio_fd);
}
int
DSP_OpenAudio(_THIS, SDL_AudioSpec * spec)
{
char audiodev[1024];
#ifdef AUDIO_SETINFO
int enc;
#endif
int desired_freq = spec->freq;
/* Initialize our freeable variables, in case we fail */
audio_fd = -1;
mixbuf = NULL;
ulaw_buf = NULL;
/* Determine the audio parameters from the AudioSpec */
switch (SDL_AUDIO_BITSIZE(spec->format)) {
case 8:
{ /* Unsigned 8 bit audio data */
spec->format = AUDIO_U8;
#ifdef AUDIO_SETINFO
enc = AUDIO_ENCODING_LINEAR8;
#endif
}
break;
case 16:
{ /* Signed 16 bit audio data */
spec->format = AUDIO_S16SYS;
#ifdef AUDIO_SETINFO
enc = AUDIO_ENCODING_LINEAR;
#endif
}
break;
default:
{
/* !!! FIXME: fallback to conversion on unsupported types! */
SDL_SetError("Unsupported audio format");
return (-1);
}
}
audio_fmt = spec->format;
/* Open the audio device */
audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 1);
if (audio_fd < 0) {
SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
return (-1);
}
ulaw_only = 0; /* modern Suns do support linear audio */
#ifdef AUDIO_SETINFO
for (;;) {
audio_info_t info;
AUDIO_INITINFO(&info); /* init all fields to "no change" */
/* Try to set the requested settings */
info.play.sample_rate = spec->freq;
info.play.channels = spec->channels;
info.play.precision = (enc == AUDIO_ENCODING_ULAW)
? 8 : spec->format & 0xff;
info.play.encoding = enc;
if (ioctl(audio_fd, AUDIO_SETINFO, &info) == 0) {
/* Check to be sure we got what we wanted */
if (ioctl(audio_fd, AUDIO_GETINFO, &info) < 0) {
SDL_SetError("Error getting audio parameters: %s",
strerror(errno));
return -1;
}
if (info.play.encoding == enc
&& info.play.precision == (spec->format & 0xff)
&& info.play.channels == spec->channels) {
/* Yow! All seems to be well! */
spec->freq = info.play.sample_rate;
break;
}
}
switch (enc) {
case AUDIO_ENCODING_LINEAR8:
/* unsigned 8bit apparently not supported here */
enc = AUDIO_ENCODING_LINEAR;
spec->format = AUDIO_S16SYS;
break; /* try again */
case AUDIO_ENCODING_LINEAR:
/* linear 16bit didn't work either, resort to µ-law */
enc = AUDIO_ENCODING_ULAW;
spec->channels = 1;
spec->freq = 8000;
spec->format = AUDIO_U8;
ulaw_only = 1;
break;
default:
/* oh well... */
SDL_SetError("Error setting audio parameters: %s",
strerror(errno));
return -1;
}
}
#endif /* AUDIO_SETINFO */
written = 0;
/* We can actually convert on-the-fly to U-Law */
if (ulaw_only) {
spec->freq = desired_freq;
fragsize = (spec->samples * 1000) / (spec->freq / 8);
frequency = 8;
ulaw_buf = (Uint8 *) SDL_malloc(fragsize);
if (ulaw_buf == NULL) {
SDL_OutOfMemory();
return (-1);
}
spec->channels = 1;
} else {
fragsize = spec->samples;
frequency = spec->freq / 1000;
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Audio device %s U-Law only\n",
ulaw_only ? "is" : "is not");
fprintf(stderr, "format=0x%x chan=%d freq=%d\n",
spec->format, spec->channels, spec->freq);
#endif
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(spec);
/* Allocate mixing buffer */
mixbuf = (Uint8 *) SDL_AllocAudioMem(spec->size);
if (mixbuf == NULL) {
SDL_OutOfMemory();
return (-1);
}
SDL_memset(mixbuf, spec->silence, spec->size);
/* We're ready to rock and roll. :-) */
return (0);
}
/************************************************************************/
/* This function (snd2au()) copyrighted: */
/************************************************************************/
/* Copyright 1989 by Rich Gopstein and Harris Corporation */
/* */
/* Permission to use, copy, modify, and distribute this software */
/* and its documentation for any purpose and without fee is */
/* hereby granted, provided that the above copyright notice */
/* appears in all copies and that both that copyright notice and */
/* this permission notice appear in supporting documentation, and */
/* that the name of Rich Gopstein and Harris Corporation not be */
/* used in advertising or publicity pertaining to distribution */
/* of the software without specific, written prior permission. */
/* Rich Gopstein and Harris Corporation make no representations */
/* about the suitability of this software for any purpose. It */
/* provided "as is" without express or implied warranty. */
/************************************************************************/
static Uint8
snd2au(int sample)
{
int mask;
if (sample < 0) {
sample = -sample;
mask = 0x7f;
} else {
mask = 0xff;
}
if (sample < 32) {
sample = 0xF0 | (15 - sample / 2);
} else if (sample < 96) {
sample = 0xE0 | (15 - (sample - 32) / 4);
} else if (sample < 224) {
sample = 0xD0 | (15 - (sample - 96) / 8);
} else if (sample < 480) {
sample = 0xC0 | (15 - (sample - 224) / 16);
} else if (sample < 992) {
sample = 0xB0 | (15 - (sample - 480) / 32);
} else if (sample < 2016) {
sample = 0xA0 | (15 - (sample - 992) / 64);
} else if (sample < 4064) {
sample = 0x90 | (15 - (sample - 2016) / 128);
} else if (sample < 8160) {
sample = 0x80 | (15 - (sample - 4064) / 256);
} else {
sample = 0x80;
}
return (mask & sample);
}
/* vi: set ts=4 sw=4 expandtab: */

58
project/jni/sdl-1.3/src/audio/sun/SDL_sunaudio.h Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_sunaudio_h
#define _SDL_sunaudio_h
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
/* The file descriptor for the audio device */
int audio_fd;
SDL_AudioFormat audio_fmt; /* The app audio format */
Uint8 *mixbuf; /* The app mixing buffer */
int ulaw_only; /* Flag -- does hardware only output U-law? */
Uint8 *ulaw_buf; /* The U-law mixing buffer */
Sint32 written; /* The number of samples written */
int fragsize; /* The audio fragment size in samples */
int frequency; /* The audio frequency in KHz */
};
/* Old variable names */
#define audio_fd (this->hidden->audio_fd)
#define audio_fmt (this->hidden->audio_fmt)
#define mixbuf (this->hidden->mixbuf)
#define ulaw_only (this->hidden->ulaw_only)
#define ulaw_buf (this->hidden->ulaw_buf)
#define written (this->hidden->written)
#define fragsize (this->hidden->fragsize)
#define frequency (this->hidden->frequency)
#endif /* _SDL_sunaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

563
project/jni/sdl-1.3/src/audio/ums/SDL_umsaudio.c Normal file
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/* I'm gambling no one uses this audio backend...we'll see who emails. :) */
#error this code has not been updated for SDL 1.3.
#error if no one emails icculus at icculus.org and tells him that this
#error code is needed, this audio backend will eventually be removed from SDL.
/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 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
Carsten Griwodz
griff@kom.tu-darmstadt.de
based on linux/SDL_dspaudio.c by Sam Lantinga
*/
#include "SDL_config.h"
/* Allow access to a raw mixing buffer */
#include <errno.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/types.h>
#include <sys/time.h>
#include <sys/ioctl.h>
#include <sys/stat.h>
#include <sys/mman.h>
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "SDL_umsaudio.h"
/* The tag name used by UMS audio */
#define UMS_DRIVER_NAME "ums"
#define DEBUG_AUDIO 1
/* Audio driver functions */
static int UMS_OpenAudio(_THIS, SDL_AudioSpec * spec);
static void UMS_PlayAudio(_THIS);
static Uint8 *UMS_GetAudioBuf(_THIS);
static void UMS_CloseAudio(_THIS);
static UMSAudioDevice_ReturnCode UADOpen(_THIS, string device, string mode,
long flags);
static UMSAudioDevice_ReturnCode UADClose(_THIS);
static UMSAudioDevice_ReturnCode UADGetBitsPerSample(_THIS, long *bits);
static UMSAudioDevice_ReturnCode UADSetBitsPerSample(_THIS, long bits);
static UMSAudioDevice_ReturnCode UADSetSampleRate(_THIS, long rate,
long *set_rate);
static UMSAudioDevice_ReturnCode UADSetByteOrder(_THIS, string byte_order);
static UMSAudioDevice_ReturnCode UADSetAudioFormatType(_THIS, string fmt);
static UMSAudioDevice_ReturnCode UADSetNumberFormat(_THIS, string fmt);
static UMSAudioDevice_ReturnCode UADInitialize(_THIS);
static UMSAudioDevice_ReturnCode UADStart(_THIS);
static UMSAudioDevice_ReturnCode UADStop(_THIS);
static UMSAudioDevice_ReturnCode UADSetTimeFormat(_THIS,
UMSAudioTypes_TimeFormat
fmt);
static UMSAudioDevice_ReturnCode UADWriteBuffSize(_THIS, long *buff_size);
static UMSAudioDevice_ReturnCode UADWriteBuffRemain(_THIS, long *buff_size);
static UMSAudioDevice_ReturnCode UADWriteBuffUsed(_THIS, long *buff_size);
static UMSAudioDevice_ReturnCode UADSetDMABufferSize(_THIS, long bytes,
long *bytes_ret);
static UMSAudioDevice_ReturnCode UADSetVolume(_THIS, long volume);
static UMSAudioDevice_ReturnCode UADSetBalance(_THIS, long balance);
static UMSAudioDevice_ReturnCode UADSetChannels(_THIS, long channels);
static UMSAudioDevice_ReturnCode UADPlayRemainingData(_THIS, boolean block);
static UMSAudioDevice_ReturnCode UADEnableOutput(_THIS, string output,
long *left_gain,
long *right_gain);
static UMSAudioDevice_ReturnCode UADWrite(_THIS, UMSAudioTypes_Buffer * buff,
long samples,
long *samples_written);
/* Audio driver bootstrap functions */
static int
Audio_Available(void)
{
return 1;
}
static void
Audio_DeleteDevice(_THIS)
{
if (this->hidden->playbuf._buffer)
SDL_free(this->hidden->playbuf._buffer);
if (this->hidden->fillbuf._buffer)
SDL_free(this->hidden->fillbuf._buffer);
_somFree(this->hidden->umsdev);
SDL_free(this->hidden);
SDL_free(this);
}
static SDL_AudioDevice *
Audio_CreateDevice(int devindex)
{
SDL_AudioDevice *this;
/*
* Allocate and initialize management storage and private management
* storage for this SDL-using library.
*/
this = (SDL_AudioDevice *) SDL_malloc(sizeof(SDL_AudioDevice));
if (this) {
SDL_memset(this, 0, (sizeof *this));
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
}
if ((this == NULL) || (this->hidden == NULL)) {
SDL_OutOfMemory();
if (this) {
SDL_free(this);
}
return (0);
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
#ifdef DEBUG_AUDIO
fprintf(stderr, "Creating UMS Audio device\n");
#endif
/*
* Calls for UMS env initialization and audio object construction.
*/
this->hidden->ev = somGetGlobalEnvironment();
this->hidden->umsdev = UMSAudioDeviceNew();
/*
* Set the function pointers.
*/
this->OpenAudio = UMS_OpenAudio;
this->WaitAudio = NULL; /* we do blocking output */
this->PlayAudio = UMS_PlayAudio;
this->GetAudioBuf = UMS_GetAudioBuf;
this->CloseAudio = UMS_CloseAudio;
this->free = Audio_DeleteDevice;
#ifdef DEBUG_AUDIO
fprintf(stderr, "done\n");
#endif
return this;
}
AudioBootStrap UMS_bootstrap = {
UMS_DRIVER_NAME, "AIX UMS audio",
Audio_Available, Audio_CreateDevice, 0
};
static Uint8 *
UMS_GetAudioBuf(_THIS)
{
#ifdef DEBUG_AUDIO
fprintf(stderr, "enter UMS_GetAudioBuf\n");
#endif
return this->hidden->fillbuf._buffer;
/*
long bufSize;
UMSAudioDevice_ReturnCode rc;
rc = UADSetTimeFormat(this, UMSAudioTypes_Bytes );
rc = UADWriteBuffSize(this, bufSize );
*/
}
static void
UMS_CloseAudio(_THIS)
{
UMSAudioDevice_ReturnCode rc;
#ifdef DEBUG_AUDIO
fprintf(stderr, "enter UMS_CloseAudio\n");
#endif
rc = UADPlayRemainingData(this, TRUE);
rc = UADStop(this);
rc = UADClose(this);
}
static void
UMS_PlayAudio(_THIS)
{
UMSAudioDevice_ReturnCode rc;
long samplesToWrite;
long samplesWritten;
UMSAudioTypes_Buffer swpbuf;
#ifdef DEBUG_AUDIO
fprintf(stderr, "enter UMS_PlayAudio\n");
#endif
samplesToWrite =
this->hidden->playbuf._length / this->hidden->bytesPerSample;
do {
rc = UADWrite(this, &this->hidden->playbuf,
samplesToWrite, &samplesWritten);
samplesToWrite -= samplesWritten;
/* rc values: UMSAudioDevice_Success
* UMSAudioDevice_Failure
* UMSAudioDevice_Preempted
* UMSAudioDevice_Interrupted
* UMSAudioDevice_DeviceError
*/
if (rc == UMSAudioDevice_DeviceError) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Returning from PlayAudio with devices error\n");
#endif
return;
}
} while (samplesToWrite > 0);
SDL_LockAudio();
SDL_memcpy(&swpbuf, &this->hidden->playbuf, sizeof(UMSAudioTypes_Buffer));
SDL_memcpy(&this->hidden->playbuf, &this->hidden->fillbuf,
sizeof(UMSAudioTypes_Buffer));
SDL_memcpy(&this->hidden->fillbuf, &swpbuf, sizeof(UMSAudioTypes_Buffer));
SDL_UnlockAudio();
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote audio data and swapped buffer\n");
#endif
}
#if 0
// /* Set the DSP frequency */
// value = spec->freq;
// if ( ioctl(this->hidden->audio_fd, SOUND_PCM_WRITE_RATE, &value) < 0 ) {
// SDL_SetError("Couldn't set audio frequency");
// return(-1);
// }
// spec->freq = value;
#endif
static int
UMS_OpenAudio(_THIS, SDL_AudioSpec * spec)
{
char *audiodev = "/dev/paud0";
long lgain;
long rgain;
long outRate;
long outBufSize;
long bitsPerSample;
long samplesPerSec;
long success;
SDL_AudioFormat test_format;
int frag_spec;
UMSAudioDevice_ReturnCode rc;
#ifdef DEBUG_AUDIO
fprintf(stderr, "enter UMS_OpenAudio\n");
#endif
rc = UADOpen(this, audiodev, "PLAY", UMSAudioDevice_BlockingIO);
if (rc != UMSAudioDevice_Success) {
SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
return -1;
}
rc = UADSetAudioFormatType(this, "PCM");
success = 0;
test_format = SDL_FirstAudioFormat(spec->format);
do {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
switch (test_format) {
case AUDIO_U8:
/* from the mac code: better ? */
/* sample_bits = spec->size / spec->samples / spec->channels * 8; */
success = 1;
bitsPerSample = 8;
rc = UADSetSampleRate(this, spec->freq << 16, &outRate);
rc = UADSetByteOrder(this, "MSB"); /* irrelevant */
rc = UADSetNumberFormat(this, "UNSIGNED");
break;
case AUDIO_S8:
success = 1;
bitsPerSample = 8;
rc = UADSetSampleRate(this, spec->freq << 16, &outRate);
rc = UADSetByteOrder(this, "MSB"); /* irrelevant */
rc = UADSetNumberFormat(this, "SIGNED");
break;
case AUDIO_S16LSB:
success = 1;
bitsPerSample = 16;
rc = UADSetSampleRate(this, spec->freq << 16, &outRate);
rc = UADSetByteOrder(this, "LSB");
rc = UADSetNumberFormat(this, "SIGNED");
break;
case AUDIO_S16MSB:
success = 1;
bitsPerSample = 16;
rc = UADSetSampleRate(this, spec->freq << 16, &outRate);
rc = UADSetByteOrder(this, "MSB");
rc = UADSetNumberFormat(this, "SIGNED");
break;
case AUDIO_U16LSB:
success = 1;
bitsPerSample = 16;
rc = UADSetSampleRate(this, spec->freq << 16, &outRate);
rc = UADSetByteOrder(this, "LSB");
rc = UADSetNumberFormat(this, "UNSIGNED");
break;
case AUDIO_U16MSB:
success = 1;
bitsPerSample = 16;
rc = UADSetSampleRate(this, spec->freq << 16, &outRate);
rc = UADSetByteOrder(this, "MSB");
rc = UADSetNumberFormat(this, "UNSIGNED");
break;
default:
break;
}
if (!success) {
test_format = SDL_NextAudioFormat();
}
} while (!success && test_format);
if (success == 0) {
SDL_SetError("Couldn't find any hardware audio formats");
return -1;
}
spec->format = test_format;
for (frag_spec = 0; (0x01 << frag_spec) < spec->size; ++frag_spec);
if ((0x01 << frag_spec) != spec->size) {
SDL_SetError("Fragment size must be a power of two");
return -1;
}
if (frag_spec > 2048)
frag_spec = 2048;
this->hidden->bytesPerSample = (bitsPerSample / 8) * spec->channels;
samplesPerSec = this->hidden->bytesPerSample * outRate;
this->hidden->playbuf._length = 0;
this->hidden->playbuf._maximum = spec->size;
this->hidden->playbuf._buffer = (unsigned char *) SDL_malloc(spec->size);
this->hidden->fillbuf._length = 0;
this->hidden->fillbuf._maximum = spec->size;
this->hidden->fillbuf._buffer = (unsigned char *) SDL_malloc(spec->size);
rc = UADSetBitsPerSample(this, bitsPerSample);
rc = UADSetDMABufferSize(this, frag_spec, &outBufSize);
rc = UADSetChannels(this, spec->channels); /* functions reduces to mono or stereo */
lgain = 100; /*maximum left input gain */
rgain = 100; /*maimum right input gain */
rc = UADEnableOutput(this, "LINE_OUT", &lgain, &rgain);
rc = UADInitialize(this);
rc = UADStart(this);
rc = UADSetVolume(this, 100);
rc = UADSetBalance(this, 0);
/* We're ready to rock and roll. :-) */
return 0;
}
static UMSAudioDevice_ReturnCode
UADGetBitsPerSample(_THIS, long *bits)
{
return UMSAudioDevice_get_bits_per_sample(this->hidden->umsdev,
this->hidden->ev, bits);
}
static UMSAudioDevice_ReturnCode
UADSetBitsPerSample(_THIS, long bits)
{
return UMSAudioDevice_set_bits_per_sample(this->hidden->umsdev,
this->hidden->ev, bits);
}
static UMSAudioDevice_ReturnCode
UADSetSampleRate(_THIS, long rate, long *set_rate)
{
/* from the mac code: sample rate = spec->freq << 16; */
return UMSAudioDevice_set_sample_rate(this->hidden->umsdev,
this->hidden->ev, rate, set_rate);
}
static UMSAudioDevice_ReturnCode
UADSetByteOrder(_THIS, string byte_order)
{
return UMSAudioDevice_set_byte_order(this->hidden->umsdev,
this->hidden->ev, byte_order);
}
static UMSAudioDevice_ReturnCode
UADSetAudioFormatType(_THIS, string fmt)
{
/* possible PCM, A_LAW or MU_LAW */
return UMSAudioDevice_set_audio_format_type(this->hidden->umsdev,
this->hidden->ev, fmt);
}
static UMSAudioDevice_ReturnCode
UADSetNumberFormat(_THIS, string fmt)
{
/* possible SIGNED, UNSIGNED, or TWOS_COMPLEMENT */
return UMSAudioDevice_set_number_format(this->hidden->umsdev,
this->hidden->ev, fmt);
}
static UMSAudioDevice_ReturnCode
UADInitialize(_THIS)
{
return UMSAudioDevice_initialize(this->hidden->umsdev, this->hidden->ev);
}
static UMSAudioDevice_ReturnCode
UADStart(_THIS)
{
return UMSAudioDevice_start(this->hidden->umsdev, this->hidden->ev);
}
static UMSAudioDevice_ReturnCode
UADSetTimeFormat(_THIS, UMSAudioTypes_TimeFormat fmt)
{
/*
* Switches the time format to the new format, immediately.
* possible UMSAudioTypes_Msecs, UMSAudioTypes_Bytes or UMSAudioTypes_Samples
*/
return UMSAudioDevice_set_time_format(this->hidden->umsdev,
this->hidden->ev, fmt);
}
static UMSAudioDevice_ReturnCode
UADWriteBuffSize(_THIS, long *buff_size)
{
/*
* returns write buffer size in the current time format
*/
return UMSAudioDevice_write_buff_size(this->hidden->umsdev,
this->hidden->ev, buff_size);
}
static UMSAudioDevice_ReturnCode
UADWriteBuffRemain(_THIS, long *buff_size)
{
/*
* returns amount of available space in the write buffer
* in the current time format
*/
return UMSAudioDevice_write_buff_remain(this->hidden->umsdev,
this->hidden->ev, buff_size);
}
static UMSAudioDevice_ReturnCode
UADWriteBuffUsed(_THIS, long *buff_size)
{
/*
* returns amount of filled space in the write buffer
* in the current time format
*/
return UMSAudioDevice_write_buff_used(this->hidden->umsdev,
this->hidden->ev, buff_size);
}
static UMSAudioDevice_ReturnCode
UADSetDMABufferSize(_THIS, long bytes, long *bytes_ret)
{
/*
* Request a new DMA buffer size, maximum requested size 2048.
* Takes effect with next initialize() call.
* Devices may or may not support DMA.
*/
return UMSAudioDevice_set_DMA_buffer_size(this->hidden->umsdev,
this->hidden->ev,
bytes, bytes_ret);
}
static UMSAudioDevice_ReturnCode
UADSetVolume(_THIS, long volume)
{
/*
* Set the volume.
* Takes effect immediately.
*/
return UMSAudioDevice_set_volume(this->hidden->umsdev,
this->hidden->ev, volume);
}
static UMSAudioDevice_ReturnCode
UADSetBalance(_THIS, long balance)
{
/*
* Set the balance.
* Takes effect immediately.
*/
return UMSAudioDevice_set_balance(this->hidden->umsdev,
this->hidden->ev, balance);
}
static UMSAudioDevice_ReturnCode
UADSetChannels(_THIS, long channels)
{
/*
* Set mono or stereo.
* Takes effect with next initialize() call.
*/
if (channels != 1)
channels = 2;
return UMSAudioDevice_set_number_of_channels(this->hidden->umsdev,
this->hidden->ev, channels);
}
static UMSAudioDevice_ReturnCode
UADOpen(_THIS, string device, string mode, long flags)
{
return UMSAudioDevice_open(this->hidden->umsdev,
this->hidden->ev, device, mode, flags);
}
static UMSAudioDevice_ReturnCode
UADWrite(_THIS, UMSAudioTypes_Buffer * buff,
long samples, long *samples_written)
{
return UMSAudioDevice_write(this->hidden->umsdev,
this->hidden->ev,
buff, samples, samples_written);
}
static UMSAudioDevice_ReturnCode
UADPlayRemainingData(_THIS, boolean block)
{
return UMSAudioDevice_play_remaining_data(this->hidden->umsdev,
this->hidden->ev, block);
}
static UMSAudioDevice_ReturnCode
UADStop(_THIS)
{
return UMSAudioDevice_stop(this->hidden->umsdev, this->hidden->ev);
}
static UMSAudioDevice_ReturnCode
UADClose(_THIS)
{
return UMSAudioDevice_close(this->hidden->umsdev, this->hidden->ev);
}
static UMSAudioDevice_ReturnCode
UADEnableOutput(_THIS, string output, long *left_gain, long *right_gain)
{
return UMSAudioDevice_enable_output(this->hidden->umsdev,
this->hidden->ev,
output, left_gain, right_gain);
}
/* vi: set ts=4 sw=4 expandtab: */

50
project/jni/sdl-1.3/src/audio/ums/SDL_umsaudio.h Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 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
Carsten Griwodz
griff@kom.tu-darmstadt.de
based on linux/SDL_dspaudio.h by Sam Lantinga
*/
#include "SDL_config.h"
#ifndef _SDL_UMSaudio_h
#define _SDL_UMSaudio_h
#include <UMS/UMSAudioDevice.h>
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
/* Pointer to the (open) UMS audio device */
Environment *ev;
UMSAudioDevice umsdev;
/* Raw mixing buffer */
UMSAudioTypes_Buffer playbuf;
UMSAudioTypes_Buffer fillbuf;
long bytesPerSample;
};
#endif /* _SDL_UMSaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

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project/jni/sdl-1.3/src/audio/windib/SDL_dibaudio.c Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* Allow access to a raw mixing buffer */
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#include <mmsystem.h>
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "SDL_dibaudio.h"
#if defined(_WIN32_WCE) && (_WIN32_WCE < 300)
#include "win_ce_semaphore.h"
#endif
#if defined(_WIN32_WCE)
#define WINDOWS_OS_NAME "Windows CE/PocketPC"
#elif defined(WIN64)
#define WINDOWS_OS_NAME "Win64"
#else
#define WINDOWS_OS_NAME "Win32"
#endif
/* The Win32 callback for filling the WAVE device */
static void CALLBACK
FillSound(HWAVEOUT hwo, UINT uMsg, DWORD_PTR dwInstance,
DWORD dwParam1, DWORD dwParam2)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) dwInstance;
/* Only service "buffer done playing" messages */
if (uMsg != WOM_DONE)
return;
/* Signal that we are done playing a buffer */
#if defined(_WIN32_WCE) && (_WIN32_WCE < 300)
ReleaseSemaphoreCE(this->hidden->audio_sem, 1, NULL);
#else
ReleaseSemaphore(this->hidden->audio_sem, 1, NULL);
#endif
}
static void
SetMMerror(char *function, MMRESULT code)
{
size_t len;
char errbuf[MAXERRORLENGTH];
#ifdef _WIN32_WCE
wchar_t werrbuf[MAXERRORLENGTH];
#endif
SDL_snprintf(errbuf, SDL_arraysize(errbuf), "%s: ", function);
len = SDL_strlen(errbuf);
#ifdef _WIN32_WCE
/* UNICODE version */
waveOutGetErrorText(code, werrbuf, MAXERRORLENGTH - len);
WideCharToMultiByte(CP_ACP, 0, werrbuf, -1, errbuf + len,
MAXERRORLENGTH - len, NULL, NULL);
#else
waveOutGetErrorText(code, errbuf + len, (UINT) (MAXERRORLENGTH - len));
#endif
SDL_SetError("%s", errbuf);
}
/* Set high priority for the audio thread */
static void
WINWAVEOUT_ThreadInit(_THIS)
{
SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_HIGHEST);
}
void
WINWAVEOUT_WaitDevice(_THIS)
{
/* Wait for an audio chunk to finish */
#if defined(_WIN32_WCE) && (_WIN32_WCE < 300)
WaitForSemaphoreCE(this->hidden->audio_sem, INFINITE);
#else
WaitForSingleObject(this->hidden->audio_sem, INFINITE);
#endif
}
Uint8 *
WINWAVEOUT_GetDeviceBuf(_THIS)
{
return (Uint8 *) (this->hidden->
wavebuf[this->hidden->next_buffer].lpData);
}
void
WINWAVEOUT_PlayDevice(_THIS)
{
/* Queue it up */
waveOutWrite(this->hidden->sound,
&this->hidden->wavebuf[this->hidden->next_buffer],
sizeof(this->hidden->wavebuf[0]));
this->hidden->next_buffer = (this->hidden->next_buffer + 1) % NUM_BUFFERS;
}
void
WINWAVEOUT_WaitDone(_THIS)
{
int i, left;
do {
left = NUM_BUFFERS;
for (i = 0; i < NUM_BUFFERS; ++i) {
if (this->hidden->wavebuf[i].dwFlags & WHDR_DONE) {
--left;
}
}
if (left > 0) {
SDL_Delay(100);
}
} while (left > 0);
}
void
WINWAVEOUT_CloseDevice(_THIS)
{
/* Close up audio */
if (this->hidden != NULL) {
int i;
if (this->hidden->audio_sem) {
#if defined(_WIN32_WCE) && (_WIN32_WCE < 300)
CloseSynchHandle(this->hidden->audio_sem);
#else
CloseHandle(this->hidden->audio_sem);
#endif
this->hidden->audio_sem = 0;
}
if (this->hidden->sound) {
waveOutClose(this->hidden->sound);
this->hidden->sound = 0;
}
/* Clean up mixing buffers */
for (i = 0; i < NUM_BUFFERS; ++i) {
if (this->hidden->wavebuf[i].dwUser != 0xFFFF) {
waveOutUnprepareHeader(this->hidden->sound,
&this->hidden->wavebuf[i],
sizeof(this->hidden->wavebuf[i]));
this->hidden->wavebuf[i].dwUser = 0xFFFF;
}
}
if (this->hidden->mixbuf != NULL) {
/* Free raw mixing buffer */
SDL_free(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
int
WINWAVEOUT_OpenDevice(_THIS, const char *devname, int iscapture)
{
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
int valid_datatype = 0;
MMRESULT result;
WAVEFORMATEX waveformat;
int i;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Initialize the wavebuf structures for closing */
for (i = 0; i < NUM_BUFFERS; ++i)
this->hidden->wavebuf[i].dwUser = 0xFFFF;
while ((!valid_datatype) && (test_format)) {
valid_datatype = 1;
this->spec.format = test_format;
switch (test_format) {
case AUDIO_U8:
case AUDIO_S16:
case AUDIO_S32:
break; /* valid. */
default:
valid_datatype = 0;
test_format = SDL_NextAudioFormat();
break;
}
}
if (!valid_datatype) {
WINWAVEOUT_CloseDevice(this);
SDL_SetError("Unsupported audio format");
return 0;
}
/* Set basic WAVE format parameters */
SDL_memset(&waveformat, '\0', sizeof(waveformat));
waveformat.wFormatTag = WAVE_FORMAT_PCM;
waveformat.wBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
if (this->spec.channels > 2)
this->spec.channels = 2; /* !!! FIXME: is this right? */
waveformat.nChannels = this->spec.channels;
waveformat.nSamplesPerSec = this->spec.freq;
waveformat.nBlockAlign =
waveformat.nChannels * (waveformat.wBitsPerSample / 8);
waveformat.nAvgBytesPerSec =
waveformat.nSamplesPerSec * waveformat.nBlockAlign;
/* Check the buffer size -- minimum of 1/4 second (word aligned) */
if (this->spec.samples < (this->spec.freq / 4))
this->spec.samples = ((this->spec.freq / 4) + 3) & ~3;
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* Open the audio device */
result = waveOutOpen(&this->hidden->sound, WAVE_MAPPER, &waveformat,
(DWORD_PTR) FillSound, (DWORD_PTR) this,
CALLBACK_FUNCTION);
if (result != MMSYSERR_NOERROR) {
WINWAVEOUT_CloseDevice(this);
SetMMerror("waveOutOpen()", result);
return 0;
}
#ifdef SOUND_DEBUG
/* Check the sound device we retrieved */
{
WAVEOUTCAPS caps;
result = waveOutGetDevCaps((UINT) this->hidden->sound,
&caps, sizeof(caps));
if (result != MMSYSERR_NOERROR) {
WINWAVEOUT_CloseDevice(this);
SetMMerror("waveOutGetDevCaps()", result);
return 0;
}
printf("Audio device: %s\n", caps.szPname);
}
#endif
/* Create the audio buffer semaphore */
this->hidden->audio_sem =
#if defined(_WIN32_WCE) && (_WIN32_WCE < 300)
CreateSemaphoreCE(NULL, NUM_BUFFERS - 1, NUM_BUFFERS, NULL);
#else
CreateSemaphore(NULL, NUM_BUFFERS - 1, NUM_BUFFERS, NULL);
#endif
if (this->hidden->audio_sem == NULL) {
WINWAVEOUT_CloseDevice(this);
SDL_SetError("Couldn't create semaphore");
return 0;
}
/* Create the sound buffers */
this->hidden->mixbuf =
(Uint8 *) SDL_malloc(NUM_BUFFERS * this->spec.size);
if (this->hidden->mixbuf == NULL) {
WINWAVEOUT_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
for (i = 0; i < NUM_BUFFERS; ++i) {
SDL_memset(&this->hidden->wavebuf[i], '\0',
sizeof(this->hidden->wavebuf[i]));
this->hidden->wavebuf[i].dwBufferLength = this->spec.size;
this->hidden->wavebuf[i].dwFlags = WHDR_DONE;
this->hidden->wavebuf[i].lpData =
(LPSTR) & this->hidden->mixbuf[i * this->spec.size];
result = waveOutPrepareHeader(this->hidden->sound,
&this->hidden->wavebuf[i],
sizeof(this->hidden->wavebuf[i]));
if (result != MMSYSERR_NOERROR) {
WINWAVEOUT_CloseDevice(this);
SetMMerror("waveOutPrepareHeader()", result);
return 0;
}
}
return 1; /* Ready to go! */
}
static int
WINWAVEOUT_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->OpenDevice = WINWAVEOUT_OpenDevice;
impl->ThreadInit = WINWAVEOUT_ThreadInit;
impl->PlayDevice = WINWAVEOUT_PlayDevice;
impl->WaitDevice = WINWAVEOUT_WaitDevice;
impl->WaitDone = WINWAVEOUT_WaitDone;
impl->GetDeviceBuf = WINWAVEOUT_GetDeviceBuf;
impl->CloseDevice = WINWAVEOUT_CloseDevice;
impl->OnlyHasDefaultOutputDevice = 1; /* !!! FIXME: Is this true? */
return 1; /* this audio target is available. */
}
AudioBootStrap WINWAVEOUT_bootstrap = {
"waveout", WINDOWS_OS_NAME " WaveOut", WINWAVEOUT_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

45
project/jni/sdl-1.3/src/audio/windib/SDL_dibaudio.h Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_dibaudio_h
#define _SDL_dibaudio_h
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define NUM_BUFFERS 2 /* -- Don't lower this! */
struct SDL_PrivateAudioData
{
HWAVEOUT sound;
HANDLE audio_sem;
Uint8 *mixbuf; /* The raw allocated mixing buffer */
WAVEHDR wavebuf[NUM_BUFFERS]; /* Wave audio fragments */
int next_buffer;
};
#endif /* _SDL_dibaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

518
project/jni/sdl-1.3/src/audio/windx5/SDL_dx5audio.c Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* Allow access to a raw mixing buffer */
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "SDL_dx5audio.h"
/* !!! FIXME: move this somewhere that other drivers can use it... */
#if defined(_WIN32_WCE)
#define WINDOWS_OS_NAME "Windows CE/PocketPC"
#elif defined(WIN64)
#define WINDOWS_OS_NAME "Win64"
#else
#define WINDOWS_OS_NAME "Win32"
#endif
/* DirectX function pointers for audio */
static HINSTANCE DSoundDLL = NULL;
static HRESULT(WINAPI * DSoundCreate) (LPGUID, LPDIRECTSOUND *, LPUNKNOWN) =
NULL;
static void
DSOUND_Unload(void)
{
if (DSoundDLL != NULL) {
FreeLibrary(DSoundDLL);
}
DSoundCreate = NULL;
DSoundDLL = NULL;
}
static int
DSOUND_Load(void)
{
int loaded = 0;
DSOUND_Unload();
DSoundDLL = LoadLibrary(TEXT("DSOUND.DLL"));
if (DSoundDLL == NULL) {
SDL_SetError("DirectSound: failed to load DSOUND.DLL");
} else {
/* Now make sure we have DirectX 5 or better... */
/* (DirectSoundCaptureCreate was added in DX5) */
if (!GetProcAddress(DSoundDLL, TEXT("DirectSoundCaptureCreate"))) {
SDL_SetError("DirectSound: System doesn't appear to have DX5.");
} else {
DSoundCreate = (void *) GetProcAddress(DSoundDLL,
TEXT("DirectSoundCreate"));
}
if (!DSoundCreate) {
SDL_SetError("DirectSound: Failed to find DirectSoundCreate");
} else {
loaded = 1;
}
}
if (!loaded) {
DSOUND_Unload();
}
return loaded;
}
static void
SetDSerror(const char *function, int code)
{
static const char *error;
static char errbuf[1024];
errbuf[0] = 0;
switch (code) {
case E_NOINTERFACE:
error = "Unsupported interface -- Is DirectX 5.0 or later installed?";
break;
case DSERR_ALLOCATED:
error = "Audio device in use";
break;
case DSERR_BADFORMAT:
error = "Unsupported audio format";
break;
case DSERR_BUFFERLOST:
error = "Mixing buffer was lost";
break;
case DSERR_CONTROLUNAVAIL:
error = "Control requested is not available";
break;
case DSERR_INVALIDCALL:
error = "Invalid call for the current state";
break;
case DSERR_INVALIDPARAM:
error = "Invalid parameter";
break;
case DSERR_NODRIVER:
error = "No audio device found";
break;
case DSERR_OUTOFMEMORY:
error = "Out of memory";
break;
case DSERR_PRIOLEVELNEEDED:
error = "Caller doesn't have priority";
break;
case DSERR_UNSUPPORTED:
error = "Function not supported";
break;
default:
SDL_snprintf(errbuf, SDL_arraysize(errbuf),
"%s: Unknown DirectSound error: 0x%x", function, code);
break;
}
if (!errbuf[0]) {
SDL_snprintf(errbuf, SDL_arraysize(errbuf), "%s: %s", function,
error);
}
SDL_SetError("%s", errbuf);
return;
}
/* DirectSound needs to be associated with a window */
static HWND mainwin = NULL;
/* */
void
DSOUND_SoundFocus(HWND hwnd)
{
/* !!! FIXME: probably broken with multi-window support in SDL 1.3 ... */
mainwin = hwnd;
}
static void
DSOUND_ThreadInit(_THIS)
{
SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_HIGHEST);
}
static void
DSOUND_WaitDevice(_THIS)
{
DWORD status = 0;
DWORD cursor = 0;
DWORD junk = 0;
HRESULT result = DS_OK;
/* Semi-busy wait, since we have no way of getting play notification
on a primary mixing buffer located in hardware (DirectX 5.0)
*/
result = IDirectSoundBuffer_GetCurrentPosition(this->hidden->mixbuf,
&junk, &cursor);
if (result != DS_OK) {
if (result == DSERR_BUFFERLOST) {
IDirectSoundBuffer_Restore(this->hidden->mixbuf);
}
#ifdef DEBUG_SOUND
SetDSerror("DirectSound GetCurrentPosition", result);
#endif
return;
}
while ((cursor / this->hidden->mixlen) == this->hidden->lastchunk) {
/* FIXME: find out how much time is left and sleep that long */
SDL_Delay(1);
/* Try to restore a lost sound buffer */
IDirectSoundBuffer_GetStatus(this->hidden->mixbuf, &status);
if ((status & DSBSTATUS_BUFFERLOST)) {
IDirectSoundBuffer_Restore(this->hidden->mixbuf);
IDirectSoundBuffer_GetStatus(this->hidden->mixbuf, &status);
if ((status & DSBSTATUS_BUFFERLOST)) {
break;
}
}
if (!(status & DSBSTATUS_PLAYING)) {
result = IDirectSoundBuffer_Play(this->hidden->mixbuf, 0, 0,
DSBPLAY_LOOPING);
if (result == DS_OK) {
continue;
}
#ifdef DEBUG_SOUND
SetDSerror("DirectSound Play", result);
#endif
return;
}
/* Find out where we are playing */
result = IDirectSoundBuffer_GetCurrentPosition(this->hidden->mixbuf,
&junk, &cursor);
if (result != DS_OK) {
SetDSerror("DirectSound GetCurrentPosition", result);
return;
}
}
}
static void
DSOUND_PlayDevice(_THIS)
{
/* Unlock the buffer, allowing it to play */
if (this->hidden->locked_buf) {
IDirectSoundBuffer_Unlock(this->hidden->mixbuf,
this->hidden->locked_buf,
this->hidden->mixlen, NULL, 0);
}
}
static Uint8 *
DSOUND_GetDeviceBuf(_THIS)
{
DWORD cursor = 0;
DWORD junk = 0;
HRESULT result = DS_OK;
DWORD rawlen = 0;
/* Figure out which blocks to fill next */
this->hidden->locked_buf = NULL;
result = IDirectSoundBuffer_GetCurrentPosition(this->hidden->mixbuf,
&junk, &cursor);
if (result == DSERR_BUFFERLOST) {
IDirectSoundBuffer_Restore(this->hidden->mixbuf);
result = IDirectSoundBuffer_GetCurrentPosition(this->hidden->mixbuf,
&junk, &cursor);
}
if (result != DS_OK) {
SetDSerror("DirectSound GetCurrentPosition", result);
return (NULL);
}
cursor /= this->hidden->mixlen;
#ifdef DEBUG_SOUND
/* Detect audio dropouts */
{
DWORD spot = cursor;
if (spot < this->hidden->lastchunk) {
spot += this->hidden->num_buffers;
}
if (spot > this->hidden->lastchunk + 1) {
fprintf(stderr, "Audio dropout, missed %d fragments\n",
(spot - (this->hidden->lastchunk + 1)));
}
}
#endif
this->hidden->lastchunk = cursor;
cursor = (cursor + 1) % this->hidden->num_buffers;
cursor *= this->hidden->mixlen;
/* Lock the audio buffer */
result = IDirectSoundBuffer_Lock(this->hidden->mixbuf, cursor,
this->hidden->mixlen,
(LPVOID *) & this->hidden->locked_buf,
&rawlen, NULL, &junk, 0);
if (result == DSERR_BUFFERLOST) {
IDirectSoundBuffer_Restore(this->hidden->mixbuf);
result = IDirectSoundBuffer_Lock(this->hidden->mixbuf, cursor,
this->hidden->mixlen,
(LPVOID *) & this->
hidden->locked_buf, &rawlen, NULL,
&junk, 0);
}
if (result != DS_OK) {
SetDSerror("DirectSound Lock", result);
return (NULL);
}
return (this->hidden->locked_buf);
}
static void
DSOUND_WaitDone(_THIS)
{
Uint8 *stream = DSOUND_GetDeviceBuf(this);
/* Wait for the playing chunk to finish */
if (stream != NULL) {
SDL_memset(stream, this->spec.silence, this->hidden->mixlen);
DSOUND_PlayDevice(this);
}
DSOUND_WaitDevice(this);
/* Stop the looping sound buffer */
IDirectSoundBuffer_Stop(this->hidden->mixbuf);
}
static void
DSOUND_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->sound != NULL) {
if (this->hidden->mixbuf != NULL) {
/* Clean up the audio buffer */
IDirectSoundBuffer_Release(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
IDirectSound_Release(this->hidden->sound);
this->hidden->sound = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
/* This function tries to create a secondary audio buffer, and returns the
number of audio chunks available in the created buffer.
*/
static int
CreateSecondary(_THIS, HWND focus, WAVEFORMATEX * wavefmt)
{
LPDIRECTSOUND sndObj = this->hidden->sound;
LPDIRECTSOUNDBUFFER *sndbuf = &this->hidden->mixbuf;
Uint32 chunksize = this->spec.size;
const int numchunks = 8;
HRESULT result = DS_OK;
DSBUFFERDESC format;
LPVOID pvAudioPtr1, pvAudioPtr2;
DWORD dwAudioBytes1, dwAudioBytes2;
/* Try to set primary mixing privileges */
if (focus) {
result = IDirectSound_SetCooperativeLevel(sndObj,
focus, DSSCL_PRIORITY);
} else {
result = IDirectSound_SetCooperativeLevel(sndObj,
GetDesktopWindow(),
DSSCL_NORMAL);
}
if (result != DS_OK) {
SetDSerror("DirectSound SetCooperativeLevel", result);
return (-1);
}
/* Try to create the secondary buffer */
SDL_memset(&format, 0, sizeof(format));
format.dwSize = sizeof(format);
format.dwFlags = DSBCAPS_GETCURRENTPOSITION2;
if (!focus) {
format.dwFlags |= DSBCAPS_GLOBALFOCUS;
} else {
format.dwFlags |= DSBCAPS_STICKYFOCUS;
}
format.dwBufferBytes = numchunks * chunksize;
if ((format.dwBufferBytes < DSBSIZE_MIN) ||
(format.dwBufferBytes > DSBSIZE_MAX)) {
SDL_SetError("Sound buffer size must be between %d and %d",
DSBSIZE_MIN / numchunks, DSBSIZE_MAX / numchunks);
return (-1);
}
format.dwReserved = 0;
format.lpwfxFormat = wavefmt;
result = IDirectSound_CreateSoundBuffer(sndObj, &format, sndbuf, NULL);
if (result != DS_OK) {
SetDSerror("DirectSound CreateSoundBuffer", result);
return (-1);
}
IDirectSoundBuffer_SetFormat(*sndbuf, wavefmt);
/* Silence the initial audio buffer */
result = IDirectSoundBuffer_Lock(*sndbuf, 0, format.dwBufferBytes,
(LPVOID *) & pvAudioPtr1, &dwAudioBytes1,
(LPVOID *) & pvAudioPtr2, &dwAudioBytes2,
DSBLOCK_ENTIREBUFFER);
if (result == DS_OK) {
SDL_memset(pvAudioPtr1, this->spec.silence, dwAudioBytes1);
IDirectSoundBuffer_Unlock(*sndbuf,
(LPVOID) pvAudioPtr1, dwAudioBytes1,
(LPVOID) pvAudioPtr2, dwAudioBytes2);
}
/* We're ready to go */
return (numchunks);
}
static int
DSOUND_OpenDevice(_THIS, const char *devname, int iscapture)
{
HRESULT result;
WAVEFORMATEX waveformat;
int valid_format = 0;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
/* !!! FIXME: handle devname */
/* !!! FIXME: handle iscapture */
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
while ((!valid_format) && (test_format)) {
switch (test_format) {
case AUDIO_U8:
case AUDIO_S16:
case AUDIO_S32:
this->spec.format = test_format;
valid_format = 1;
break;
}
test_format = SDL_NextAudioFormat();
}
if (!valid_format) {
DSOUND_CloseDevice(this);
SDL_SetError("DirectSound: Unsupported audio format");
return 0;
}
SDL_memset(&waveformat, 0, sizeof(waveformat));
waveformat.wFormatTag = WAVE_FORMAT_PCM;
waveformat.wBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
waveformat.nChannels = this->spec.channels;
waveformat.nSamplesPerSec = this->spec.freq;
waveformat.nBlockAlign =
waveformat.nChannels * (waveformat.wBitsPerSample / 8);
waveformat.nAvgBytesPerSec =
waveformat.nSamplesPerSec * waveformat.nBlockAlign;
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* Open the audio device */
result = DSoundCreate(NULL, &this->hidden->sound, NULL);
if (result != DS_OK) {
DSOUND_CloseDevice(this);
SetDSerror("DirectSoundCreate", result);
return 0;
}
/* Create the audio buffer to which we write */
this->hidden->num_buffers = CreateSecondary(this, mainwin, &waveformat);
if (this->hidden->num_buffers < 0) {
DSOUND_CloseDevice(this);
return 0;
}
/* The buffer will auto-start playing in DSOUND_WaitDevice() */
this->hidden->mixlen = this->spec.size;
return 1; /* good to go. */
}
static void
DSOUND_Deinitialize(void)
{
DSOUND_Unload();
}
static int
DSOUND_Init(SDL_AudioDriverImpl * impl)
{
OSVERSIONINFO ver;
/*
* Unfortunately, the sound drivers on NT have higher latencies than the
* audio buffers used by many SDL applications, so there are gaps in the
* audio - it sounds terrible. Punt for now.
*/
SDL_memset(&ver, '\0', sizeof(OSVERSIONINFO));
ver.dwOSVersionInfoSize = sizeof(OSVERSIONINFO);
GetVersionEx(&ver);
if (ver.dwPlatformId == VER_PLATFORM_WIN32_NT) {
if (ver.dwMajorVersion <= 4) {
return 0; /* NT4.0 or earlier. Disable dsound support. */
}
}
if (!DSOUND_Load()) {
return 0;
}
/* Set the function pointers */
impl->OpenDevice = DSOUND_OpenDevice;
impl->PlayDevice = DSOUND_PlayDevice;
impl->WaitDevice = DSOUND_WaitDevice;
impl->WaitDone = DSOUND_WaitDone;
impl->ThreadInit = DSOUND_ThreadInit;
impl->GetDeviceBuf = DSOUND_GetDeviceBuf;
impl->CloseDevice = DSOUND_CloseDevice;
impl->Deinitialize = DSOUND_Deinitialize;
impl->OnlyHasDefaultOutputDevice = 1; /* !!! FIXME */
return 1; /* this audio target is available. */
}
AudioBootStrap DSOUND_bootstrap = {
"dsound", WINDOWS_OS_NAME "DirectSound", DSOUND_Init, 0
};
/* vi: set ts=4 sw=4 expandtab: */

47
project/jni/sdl-1.3/src/audio/windx5/SDL_dx5audio.h Normal file
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2010 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
#ifndef _SDL_dx5audio_h
#define _SDL_dx5audio_h
#include "directx.h"
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
/* The DirectSound objects */
struct SDL_PrivateAudioData
{
LPDIRECTSOUND sound;
LPDIRECTSOUNDBUFFER mixbuf;
int num_buffers;
int mixlen;
DWORD lastchunk;
Uint8 *locked_buf;
};
#endif /* _SDL_dx5audio_h */
/* vi: set ts=4 sw=4 expandtab: */

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project/jni/sdl-1.3/src/audio/windx5/directx.h Normal file
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#ifndef _directx_h
#define _directx_h
/* Include all of the DirectX 5.0 headers and adds any necessary tweaks */
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#include <mmsystem.h>
#ifndef WIN32
#define WIN32
#endif
#undef WINNT
/* Far pointers don't exist in 32-bit code */
#ifndef FAR
#define FAR
#endif
/* Error codes not yet included in Win32 API header files */
#ifndef MAKE_HRESULT
#define MAKE_HRESULT(sev,fac,code) \
((HRESULT)(((unsigned long)(sev)<<31) | ((unsigned long)(fac)<<16) | ((unsigned long)(code))))
#endif
#ifndef S_OK
#define S_OK (HRESULT)0x00000000L
#endif
#ifndef SUCCEEDED
#define SUCCEEDED(x) ((HRESULT)(x) >= 0)
#endif
#ifndef FAILED
#define FAILED(x) ((HRESULT)(x)<0)
#endif
#ifndef E_FAIL
#define E_FAIL (HRESULT)0x80000008L
#endif
#ifndef E_NOINTERFACE
#define E_NOINTERFACE (HRESULT)0x80004002L
#endif
#ifndef E_OUTOFMEMORY
#define E_OUTOFMEMORY (HRESULT)0x8007000EL
#endif
#ifndef E_INVALIDARG
#define E_INVALIDARG (HRESULT)0x80070057L
#endif
#ifndef E_NOTIMPL
#define E_NOTIMPL (HRESULT)0x80004001L
#endif
#ifndef REGDB_E_CLASSNOTREG
#define REGDB_E_CLASSNOTREG (HRESULT)0x80040154L
#endif
/* Severity codes */
#ifndef SEVERITY_ERROR
#define SEVERITY_ERROR 1
#endif
/* Error facility codes */
#ifndef FACILITY_WIN32
#define FACILITY_WIN32 7
#endif
#ifndef FIELD_OFFSET
#define FIELD_OFFSET(type, field) ((LONG)&(((type *)0)->field))
#endif
/* DirectX headers (if it isn't included, I haven't tested it yet)
*/
/* We need these defines to mark what version of DirectX API we use */
#define DIRECTDRAW_VERSION 0x0700
#define DIRECTSOUND_VERSION 0x0500
#define DIRECTINPUT_VERSION 0x0500
#include <ddraw.h>
#include <dsound.h>
#include <dinput.h>
#endif /* _directx_h */
/* vi: set ts=4 sw=4 expandtab: */