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Updates SDL to version 2.0.4, which makes it compatible with VS2015.

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This commit is contained in:
Areloch 2016-04-07 00:40:06 -05:00
parent 7b52fed504
commit b63ef177f4
924 changed files with 79241 additions and 39934 deletions
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930
Engine/lib/sdl/src/audio/SDL_audio.c
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File diff suppressed because it is too large Load diff

2
Engine/lib/sdl/src/audio/SDL_audio_c.h
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@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

2
Engine/lib/sdl/src/audio/SDL_audiocvt.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

36
Engine/lib/sdl/src/audio/SDL_audiodev.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -46,18 +46,21 @@
#define _PATH_DEV_AUDIO "/dev/audio"
#endif
static SDL_INLINE void
test_device(const char *fname, int flags, int (*test) (int fd),
SDL_AddAudioDevice addfn)
static void
test_device(const int iscapture, const char *fname, int flags, int (*test) (int fd))
{
struct stat sb;
if ((stat(fname, &sb) == 0) && (S_ISCHR(sb.st_mode))) {
const int audio_fd = open(fname, flags, 0);
if (audio_fd >= 0) {
if (test(audio_fd)) {
addfn(fname);
}
const int okay = test(audio_fd);
close(audio_fd);
if (okay) {
static size_t dummyhandle = 0;
dummyhandle++;
SDL_assert(dummyhandle != 0);
SDL_AddAudioDevice(iscapture, fname, (void *) dummyhandle);
}
}
}
}
@ -68,11 +71,10 @@ test_stub(int fd)
return 1;
}
void
SDL_EnumUnixAudioDevices(int iscapture, int classic, int (*test)(int fd),
SDL_AddAudioDevice addfn)
static void
SDL_EnumUnixAudioDevices_Internal(const int iscapture, const int classic, int (*test)(int))
{
const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
const int flags = iscapture ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT;
const char *audiodev;
char audiopath[1024];
@ -97,17 +99,25 @@ SDL_EnumUnixAudioDevices(int iscapture, int classic, int (*test)(int fd),
}
}
}
test_device(audiodev, flags, test, addfn);
test_device(iscapture, audiodev, flags, test);
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, addfn);
test_device(iscapture, audiopath, flags, test);
}
}
}
void
SDL_EnumUnixAudioDevices(const int classic, int (*test)(int))
{
SDL_EnumUnixAudioDevices_Internal(SDL_TRUE, classic, test);
SDL_EnumUnixAudioDevices_Internal(SDL_FALSE, classic, test);
}
#endif /* Audio driver selection */
/* vi: set ts=4 sw=4 expandtab: */

5
Engine/lib/sdl/src/audio/SDL_audiodev_c.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -33,7 +33,6 @@
#define OPEN_FLAGS_INPUT (O_RDONLY|O_NONBLOCK)
#endif
void SDL_EnumUnixAudioDevices(int iscapture, int classic,
int (*test) (int fd), SDL_AddAudioDevice addfn);
extern void SDL_EnumUnixAudioDevices(const int classic, int (*test)(int));
/* vi: set ts=4 sw=4 expandtab: */

2
Engine/lib/sdl/src/audio/SDL_audiomem.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

202
Engine/lib/sdl/src/audio/SDL_audiotypecvt.c
View file

@ -1,7 +1,7 @@
/* DO NOT EDIT! This file is generated by sdlgenaudiocvt.pl */
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -2300,7 +2300,7 @@ SDL_Upsample_U8_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 16;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/1)) * cvt->rate_incr) * 1;
register int eps = 0;
Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 1;
const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 1;
@ -2332,7 +2332,7 @@ SDL_Downsample_U8_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 16;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/1)) * cvt->rate_incr) * 1;
register int eps = 0;
Uint8 *dst = (Uint8 *) cvt->buf;
const Uint8 *src = (Uint8 *) cvt->buf;
@ -2364,7 +2364,7 @@ SDL_Upsample_U8_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 2;
const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 2;
@ -2401,7 +2401,7 @@ SDL_Downsample_U8_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Uint8 *dst = (Uint8 *) cvt->buf;
const Uint8 *src = (Uint8 *) cvt->buf;
@ -2438,7 +2438,7 @@ SDL_Upsample_U8_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 4;
const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 4;
@ -2485,7 +2485,7 @@ SDL_Downsample_U8_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Uint8 *dst = (Uint8 *) cvt->buf;
const Uint8 *src = (Uint8 *) cvt->buf;
@ -2532,7 +2532,7 @@ SDL_Upsample_U8_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 96;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/6)) * cvt->rate_incr) * 6;
register int eps = 0;
Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 6;
const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 6;
@ -2589,7 +2589,7 @@ SDL_Downsample_U8_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 96;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/6)) * cvt->rate_incr) * 6;
register int eps = 0;
Uint8 *dst = (Uint8 *) cvt->buf;
const Uint8 *src = (Uint8 *) cvt->buf;
@ -2646,7 +2646,7 @@ SDL_Upsample_U8_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Uint8 *dst = ((Uint8 *) (cvt->buf + dstsize)) - 8;
const Uint8 *src = ((Uint8 *) (cvt->buf + cvt->len_cvt)) - 8;
@ -2713,7 +2713,7 @@ SDL_Downsample_U8_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Uint8 *dst = (Uint8 *) cvt->buf;
const Uint8 *src = (Uint8 *) cvt->buf;
@ -2780,7 +2780,7 @@ SDL_Upsample_S8_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 16;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/1)) * cvt->rate_incr) * 1;
register int eps = 0;
Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 1;
const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 1;
@ -2812,7 +2812,7 @@ SDL_Downsample_S8_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 16;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/1)) * cvt->rate_incr) * 1;
register int eps = 0;
Sint8 *dst = (Sint8 *) cvt->buf;
const Sint8 *src = (Sint8 *) cvt->buf;
@ -2844,7 +2844,7 @@ SDL_Upsample_S8_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 2;
const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 2;
@ -2881,7 +2881,7 @@ SDL_Downsample_S8_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Sint8 *dst = (Sint8 *) cvt->buf;
const Sint8 *src = (Sint8 *) cvt->buf;
@ -2918,7 +2918,7 @@ SDL_Upsample_S8_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 4;
const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 4;
@ -2965,7 +2965,7 @@ SDL_Downsample_S8_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint8 *dst = (Sint8 *) cvt->buf;
const Sint8 *src = (Sint8 *) cvt->buf;
@ -3012,7 +3012,7 @@ SDL_Upsample_S8_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 96;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/6)) * cvt->rate_incr) * 6;
register int eps = 0;
Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 6;
const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 6;
@ -3069,7 +3069,7 @@ SDL_Downsample_S8_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 96;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/6)) * cvt->rate_incr) * 6;
register int eps = 0;
Sint8 *dst = (Sint8 *) cvt->buf;
const Sint8 *src = (Sint8 *) cvt->buf;
@ -3126,7 +3126,7 @@ SDL_Upsample_S8_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint8 *dst = ((Sint8 *) (cvt->buf + dstsize)) - 8;
const Sint8 *src = ((Sint8 *) (cvt->buf + cvt->len_cvt)) - 8;
@ -3193,7 +3193,7 @@ SDL_Downsample_S8_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint8 *dst = (Sint8 *) cvt->buf;
const Sint8 *src = (Sint8 *) cvt->buf;
@ -3260,7 +3260,7 @@ SDL_Upsample_U16LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 1;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 1;
@ -3292,7 +3292,7 @@ SDL_Downsample_U16LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@ -3324,7 +3324,7 @@ SDL_Upsample_U16LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 2;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 2;
@ -3361,7 +3361,7 @@ SDL_Downsample_U16LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@ -3398,7 +3398,7 @@ SDL_Upsample_U16LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 4;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 4;
@ -3445,7 +3445,7 @@ SDL_Downsample_U16LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@ -3492,7 +3492,7 @@ SDL_Upsample_U16LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 6;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 6;
@ -3549,7 +3549,7 @@ SDL_Downsample_U16LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@ -3606,7 +3606,7 @@ SDL_Upsample_U16LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 8;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 8;
@ -3673,7 +3673,7 @@ SDL_Downsample_U16LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@ -3740,7 +3740,7 @@ SDL_Upsample_S16LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 1;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 1;
@ -3772,7 +3772,7 @@ SDL_Downsample_S16LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@ -3804,7 +3804,7 @@ SDL_Upsample_S16LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 2;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 2;
@ -3841,7 +3841,7 @@ SDL_Downsample_S16LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@ -3878,7 +3878,7 @@ SDL_Upsample_S16LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 4;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 4;
@ -3925,7 +3925,7 @@ SDL_Downsample_S16LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@ -3972,7 +3972,7 @@ SDL_Upsample_S16LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 6;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 6;
@ -4029,7 +4029,7 @@ SDL_Downsample_S16LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@ -4086,7 +4086,7 @@ SDL_Upsample_S16LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 8;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 8;
@ -4153,7 +4153,7 @@ SDL_Downsample_S16LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@ -4220,7 +4220,7 @@ SDL_Upsample_U16MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 1;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 1;
@ -4252,7 +4252,7 @@ SDL_Downsample_U16MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@ -4284,7 +4284,7 @@ SDL_Upsample_U16MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 2;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 2;
@ -4321,7 +4321,7 @@ SDL_Downsample_U16MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@ -4358,7 +4358,7 @@ SDL_Upsample_U16MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 4;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 4;
@ -4405,7 +4405,7 @@ SDL_Downsample_U16MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@ -4452,7 +4452,7 @@ SDL_Upsample_U16MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 6;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 6;
@ -4509,7 +4509,7 @@ SDL_Downsample_U16MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@ -4566,7 +4566,7 @@ SDL_Upsample_U16MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Uint16 *dst = ((Uint16 *) (cvt->buf + dstsize)) - 8;
const Uint16 *src = ((Uint16 *) (cvt->buf + cvt->len_cvt)) - 8;
@ -4633,7 +4633,7 @@ SDL_Downsample_U16MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Uint16 *dst = (Uint16 *) cvt->buf;
const Uint16 *src = (Uint16 *) cvt->buf;
@ -4700,7 +4700,7 @@ SDL_Upsample_S16MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 1;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 1;
@ -4732,7 +4732,7 @@ SDL_Downsample_S16MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 32;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/2)) * cvt->rate_incr) * 2;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@ -4764,7 +4764,7 @@ SDL_Upsample_S16MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 2;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 2;
@ -4801,7 +4801,7 @@ SDL_Downsample_S16MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@ -4838,7 +4838,7 @@ SDL_Upsample_S16MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 4;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 4;
@ -4885,7 +4885,7 @@ SDL_Downsample_S16MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@ -4932,7 +4932,7 @@ SDL_Upsample_S16MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 6;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 6;
@ -4989,7 +4989,7 @@ SDL_Downsample_S16MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 192;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/12)) * cvt->rate_incr) * 12;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@ -5046,7 +5046,7 @@ SDL_Upsample_S16MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint16 *dst = ((Sint16 *) (cvt->buf + dstsize)) - 8;
const Sint16 *src = ((Sint16 *) (cvt->buf + cvt->len_cvt)) - 8;
@ -5113,7 +5113,7 @@ SDL_Downsample_S16MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint16 *dst = (Sint16 *) cvt->buf;
const Sint16 *src = (Sint16 *) cvt->buf;
@ -5180,7 +5180,7 @@ SDL_Upsample_S32LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 1;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 1;
@ -5212,7 +5212,7 @@ SDL_Downsample_S32LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@ -5244,7 +5244,7 @@ SDL_Upsample_S32LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 2;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 2;
@ -5281,7 +5281,7 @@ SDL_Downsample_S32LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@ -5318,7 +5318,7 @@ SDL_Upsample_S32LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 4;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 4;
@ -5365,7 +5365,7 @@ SDL_Downsample_S32LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@ -5412,7 +5412,7 @@ SDL_Upsample_S32LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 6;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 6;
@ -5469,7 +5469,7 @@ SDL_Downsample_S32LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@ -5526,7 +5526,7 @@ SDL_Upsample_S32LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 8;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 8;
@ -5593,7 +5593,7 @@ SDL_Downsample_S32LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@ -5660,7 +5660,7 @@ SDL_Upsample_S32MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 1;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 1;
@ -5692,7 +5692,7 @@ SDL_Downsample_S32MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@ -5724,7 +5724,7 @@ SDL_Upsample_S32MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 2;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 2;
@ -5761,7 +5761,7 @@ SDL_Downsample_S32MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@ -5798,7 +5798,7 @@ SDL_Upsample_S32MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 4;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 4;
@ -5845,7 +5845,7 @@ SDL_Downsample_S32MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@ -5892,7 +5892,7 @@ SDL_Upsample_S32MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 6;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 6;
@ -5949,7 +5949,7 @@ SDL_Downsample_S32MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@ -6006,7 +6006,7 @@ SDL_Upsample_S32MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
Sint32 *dst = ((Sint32 *) (cvt->buf + dstsize)) - 8;
const Sint32 *src = ((Sint32 *) (cvt->buf + cvt->len_cvt)) - 8;
@ -6073,7 +6073,7 @@ SDL_Downsample_S32MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
Sint32 *dst = (Sint32 *) cvt->buf;
const Sint32 *src = (Sint32 *) cvt->buf;
@ -6140,7 +6140,7 @@ SDL_Upsample_F32LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 1;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 1;
@ -6172,7 +6172,7 @@ SDL_Downsample_F32LSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@ -6204,7 +6204,7 @@ SDL_Upsample_F32LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 2;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 2;
@ -6241,7 +6241,7 @@ SDL_Downsample_F32LSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@ -6278,7 +6278,7 @@ SDL_Upsample_F32LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 4;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 4;
@ -6325,7 +6325,7 @@ SDL_Downsample_F32LSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@ -6372,7 +6372,7 @@ SDL_Upsample_F32LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 6;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 6;
@ -6429,7 +6429,7 @@ SDL_Downsample_F32LSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@ -6486,7 +6486,7 @@ SDL_Upsample_F32LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 8;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 8;
@ -6553,7 +6553,7 @@ SDL_Downsample_F32LSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@ -6620,7 +6620,7 @@ SDL_Upsample_F32MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 1;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 1;
@ -6652,7 +6652,7 @@ SDL_Downsample_F32MSB_1c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 64;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/4)) * cvt->rate_incr) * 4;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@ -6684,7 +6684,7 @@ SDL_Upsample_F32MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 2;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 2;
@ -6721,7 +6721,7 @@ SDL_Downsample_F32MSB_2c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 128;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/8)) * cvt->rate_incr) * 8;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@ -6758,7 +6758,7 @@ SDL_Upsample_F32MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 4;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 4;
@ -6805,7 +6805,7 @@ SDL_Downsample_F32MSB_4c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 256;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/16)) * cvt->rate_incr) * 16;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@ -6852,7 +6852,7 @@ SDL_Upsample_F32MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 6;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 6;
@ -6909,7 +6909,7 @@ SDL_Downsample_F32MSB_6c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 384;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/24)) * cvt->rate_incr) * 24;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;
@ -6966,7 +6966,7 @@ SDL_Upsample_F32MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
float *dst = ((float *) (cvt->buf + dstsize)) - 8;
const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - 8;
@ -7033,7 +7033,7 @@ SDL_Downsample_F32MSB_8c(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - 512;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/32)) * cvt->rate_incr) * 32;
register int eps = 0;
float *dst = (float *) cvt->buf;
const float *src = (float *) cvt->buf;

2
Engine/lib/sdl/src/audio/SDL_mixer.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

81
Engine/lib/sdl/src/audio/SDL_sysaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -30,34 +30,83 @@
typedef struct SDL_AudioDevice SDL_AudioDevice;
#define _THIS SDL_AudioDevice *_this
/* Used by audio targets during DetectDevices() */
typedef void (*SDL_AddAudioDevice)(const char *name);
/* Audio targets should call this as devices are added to the system (such as
a USB headset being plugged in), and should also be called for
for every device found during DetectDevices(). */
extern void SDL_AddAudioDevice(const int iscapture, const char *name, void *handle);
/* Audio targets should call this as devices are removed, so SDL can update
its list of available devices. */
extern void SDL_RemoveAudioDevice(const int iscapture, void *handle);
/* Audio targets should call this if an opened audio device is lost while
being used. This can happen due to i/o errors, or a device being unplugged,
etc. If the device is totally gone, please also call SDL_RemoveAudioDevice()
as appropriate so SDL's list of devices is accurate. */
extern void SDL_OpenedAudioDeviceDisconnected(SDL_AudioDevice *device);
/* This is the size of a packet when using SDL_QueueAudio(). We allocate
these as necessary and pool them, under the assumption that we'll
eventually end up with a handful that keep recycling, meeting whatever
the app needs. We keep packing data tightly as more arrives to avoid
wasting space, and if we get a giant block of data, we'll split them
into multiple packets behind the scenes. My expectation is that most
apps will have 2-3 of these in the pool. 8k should cover most needs, but
if this is crippling for some embedded system, we can #ifdef this.
The system preallocates enough packets for 2 callbacks' worth of data. */
#define SDL_AUDIOBUFFERQUEUE_PACKETLEN (8 * 1024)
/* Used by apps that queue audio instead of using the callback. */
typedef struct SDL_AudioBufferQueue
{
Uint8 data[SDL_AUDIOBUFFERQUEUE_PACKETLEN]; /* packet data. */
Uint32 datalen; /* bytes currently in use in this packet. */
Uint32 startpos; /* bytes currently consumed in this packet. */
struct SDL_AudioBufferQueue *next; /* next item in linked list. */
} SDL_AudioBufferQueue;
typedef struct SDL_AudioDriverImpl
{
void (*DetectDevices) (int iscapture, SDL_AddAudioDevice addfn);
int (*OpenDevice) (_THIS, const char *devname, int iscapture);
void (*DetectDevices) (void);
int (*OpenDevice) (_THIS, void *handle, const char *devname, int iscapture);
void (*ThreadInit) (_THIS); /* Called by audio thread at start */
void (*WaitDevice) (_THIS);
void (*PlayDevice) (_THIS);
int (*GetPendingBytes) (_THIS);
Uint8 *(*GetDeviceBuf) (_THIS);
void (*WaitDone) (_THIS);
void (*CloseDevice) (_THIS);
void (*LockDevice) (_THIS);
void (*UnlockDevice) (_THIS);
void (*FreeDeviceHandle) (void *handle); /**< SDL is done with handle from SDL_AddAudioDevice() */
void (*Deinitialize) (void);
/* !!! FIXME: add pause(), so we can optimize instead of mixing silence. */
/* Some flags to push duplicate code into the core and reduce #ifdefs. */
/* !!! FIXME: these should be SDL_bool */
int ProvidesOwnCallbackThread;
int SkipMixerLock; /* !!! FIXME: do we need this anymore? */
int HasCaptureSupport;
int OnlyHasDefaultOutputDevice;
int OnlyHasDefaultInputDevice;
int AllowsArbitraryDeviceNames;
} SDL_AudioDriverImpl;
typedef struct SDL_AudioDeviceItem
{
void *handle;
struct SDL_AudioDeviceItem *next;
#if (defined(__GNUC__) && (__GNUC__ <= 2))
char name[1]; /* actually variable length. */
#else
char name[];
#endif
} SDL_AudioDeviceItem;
typedef struct SDL_AudioDriver
{
/* * * */
@ -70,11 +119,14 @@ typedef struct SDL_AudioDriver
SDL_AudioDriverImpl impl;
char **outputDevices;
/* A mutex for device detection */
SDL_mutex *detectionLock;
SDL_bool captureDevicesRemoved;
SDL_bool outputDevicesRemoved;
int outputDeviceCount;
char **inputDevices;
int inputDeviceCount;
SDL_AudioDeviceItem *outputDevices;
SDL_AudioDeviceItem *inputDevices;
} SDL_AudioDriver;
@ -92,6 +144,7 @@ struct SDL_AudioDevice
{
/* * * */
/* Data common to all devices */
SDL_AudioDeviceID id;
/* The current audio specification (shared with audio thread) */
SDL_AudioSpec spec;
@ -104,21 +157,29 @@ struct SDL_AudioDevice
SDL_AudioStreamer streamer;
/* Current state flags */
/* !!! FIXME: should be SDL_bool */
int iscapture;
int enabled;
int enabled; /* true if device is functioning and connected. */
int shutdown; /* true if we are signaling the play thread to end. */
int paused;
int opened;
/* Fake audio buffer for when the audio hardware is busy */
Uint8 *fake_stream;
/* A semaphore for locking the mixing buffers */
/* A mutex for locking the mixing buffers */
SDL_mutex *mixer_lock;
/* A thread to feed the audio device */
SDL_Thread *thread;
SDL_threadID threadid;
/* Queued buffers (if app not using callback). */
SDL_AudioBufferQueue *buffer_queue_head; /* device fed from here. */
SDL_AudioBufferQueue *buffer_queue_tail; /* queue fills to here. */
SDL_AudioBufferQueue *buffer_queue_pool; /* these are unused packets. */
Uint32 queued_bytes; /* number of bytes of audio data in the queue. */
/* * * */
/* Data private to this driver */
struct SDL_PrivateAudioData *hidden;

17
Engine/lib/sdl/src/audio/SDL_wave.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -121,7 +121,8 @@ 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;
Sint8 nybble;
Uint8 stereo;
Sint16 *coeff[2];
Sint32 new_sample;
@ -278,7 +279,8 @@ IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state, Uint8 nybble)
} else if (state->index < 0) {
state->index = 0;
}
step = step_table[state->index];
/* explicit cast to avoid gcc warning about using 'char' as array index */
step = step_table[(int)state->index];
delta = step >> 3;
if (nybble & 0x04)
delta += step;
@ -343,8 +345,8 @@ IMA_ADPCM_decode(Uint8 ** audio_buf, Uint32 * audio_len)
/* 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));
SDL_SetError("IMA ADPCM decoder can only handle %u channels",
(unsigned int)SDL_arraysize(IMA_ADPCM_state.state));
return (-1);
}
state = IMA_ADPCM_state.state;
@ -458,7 +460,7 @@ SDL_LoadWAV_RW(SDL_RWops * src, int freesrc,
}
/* 2 Uint32's for chunk header+len, plus the lenread */
headerDiff += lenread + 2 * sizeof(Uint32);
} while ((chunk.magic == FACT) || (chunk.magic == LIST));
} while ((chunk.magic == FACT) || (chunk.magic == LIST) || (chunk.magic == BEXT) || (chunk.magic == JUNK));
/* Decode the audio data format */
format = (WaveFMT *) chunk.data;
@ -493,8 +495,7 @@ SDL_LoadWAV_RW(SDL_RWops * src, int freesrc,
IMA_ADPCM_encoded = 1;
break;
case MP3_CODE:
SDL_SetError("MPEG Layer 3 data not supported",
SDL_SwapLE16(format->encoding));
SDL_SetError("MPEG Layer 3 data not supported");
was_error = 1;
goto done;
default:

4
Engine/lib/sdl/src/audio/SDL_wave.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -29,6 +29,8 @@
#define WAVE 0x45564157 /* "WAVE" */
#define FACT 0x74636166 /* "fact" */
#define LIST 0x5453494c /* "LIST" */
#define BEXT 0x74786562 /* "bext" */
#define JUNK 0x4B4E554A /* "JUNK" */
#define FMT 0x20746D66 /* "fmt " */
#define DATA 0x61746164 /* "data" */
#define PCM_CODE 0x0001

6
Engine/lib/sdl/src/audio/alsa/SDL_alsa_audio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -320,7 +320,7 @@ ALSA_PlayDevice(_THIS)
/* Hmm, not much we can do - abort */
fprintf(stderr, "ALSA write failed (unrecoverable): %s\n",
ALSA_snd_strerror(status));
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
return;
}
continue;
@ -465,7 +465,7 @@ ALSA_set_buffer_size(_THIS, snd_pcm_hw_params_t *params, int override)
}
static int
ALSA_OpenDevice(_THIS, const char *devname, int iscapture)
ALSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
int status = 0;
snd_pcm_t *pcm_handle = NULL;

2
Engine/lib/sdl/src/audio/alsa/SDL_alsa_audio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

50
Engine/lib/sdl/src/audio/android/SDL_androidaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -32,10 +32,10 @@
#include <android/log.h>
static void * audioDevice;
static SDL_AudioDevice* audioDevice = NULL;
static int
AndroidAUD_OpenDevice(_THIS, const char *devname, int iscapture)
AndroidAUD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
SDL_AudioFormat test_format;
@ -49,6 +49,11 @@ AndroidAUD_OpenDevice(_THIS, const char *devname, int iscapture)
}
audioDevice = this;
this->hidden = (struct SDL_PrivateAudioData *) SDL_calloc(1, (sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
test_format = SDL_FirstAudioFormat(this->spec.format);
while (test_format != 0) { /* no "UNKNOWN" constant */
@ -110,6 +115,10 @@ AndroidAUD_CloseDevice(_THIS)
Android_JNI_CloseAudioDevice();
if (audioDevice == this) {
if (audioDevice->hidden != NULL) {
SDL_free(this->hidden);
this->hidden = NULL;
}
audioDevice = NULL;
}
}
@ -135,6 +144,41 @@ AudioBootStrap ANDROIDAUD_bootstrap = {
"android", "SDL Android audio driver", AndroidAUD_Init, 0
};
/* Pause (block) all non already paused audio devices by taking their mixer lock */
void AndroidAUD_PauseDevices(void)
{
/* TODO: Handle multiple devices? */
struct SDL_PrivateAudioData *private;
if(audioDevice != NULL && audioDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) audioDevice->hidden;
if (audioDevice->paused) {
/* The device is already paused, leave it alone */
private->resume = SDL_FALSE;
}
else {
SDL_LockMutex(audioDevice->mixer_lock);
audioDevice->paused = SDL_TRUE;
private->resume = SDL_TRUE;
}
}
}
/* Resume (unblock) all non already paused audio devices by releasing their mixer lock */
void AndroidAUD_ResumeDevices(void)
{
/* TODO: Handle multiple devices? */
struct SDL_PrivateAudioData *private;
if(audioDevice != NULL && audioDevice->hidden != NULL) {
private = (struct SDL_PrivateAudioData *) audioDevice->hidden;
if (private->resume) {
audioDevice->paused = SDL_FALSE;
private->resume = SDL_FALSE;
SDL_UnlockMutex(audioDevice->mixer_lock);
}
}
}
#endif /* SDL_AUDIO_DRIVER_ANDROID */
/* vi: set ts=4 sw=4 expandtab: */

4
Engine/lib/sdl/src/audio/android/SDL_androidaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -30,6 +30,8 @@
struct SDL_PrivateAudioData
{
/* Resume device if it was paused automatically */
int resume;
};
static void AndroidAUD_CloseDevice(_THIS);

8
Engine/lib/sdl/src/audio/arts/SDL_artsaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -151,7 +151,7 @@ ARTS_WaitDevice(_THIS)
/* Check every 10 loops */
if (this->hidden->parent && (((++cnt) % 10) == 0)) {
if (kill(this->hidden->parent, 0) < 0 && errno == ESRCH) {
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
}
}
}
@ -179,7 +179,7 @@ ARTS_PlayDevice(_THIS)
/* If we couldn't write, assume fatal error for now */
if (written < 0) {
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", written);
@ -229,7 +229,7 @@ ARTS_Suspend(void)
}
static int
ARTS_OpenDevice(_THIS, const char *devname, int iscapture)
ARTS_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
int rc = 0;
int bits = 0, frag_spec = 0;

2
Engine/lib/sdl/src/audio/arts/SDL_artsaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

14
Engine/lib/sdl/src/audio/bsd/SDL_bsdaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -51,9 +51,9 @@
static void
BSDAUDIO_DetectDevices(int iscapture, SDL_AddAudioDevice addfn)
BSDAUDIO_DetectDevices(void)
{
SDL_EnumUnixAudioDevices(iscapture, 0, NULL, addfn);
SDL_EnumUnixAudioDevices(0, NULL);
}
@ -150,7 +150,7 @@ BSDAUDIO_WaitDevice(_THIS)
the user know what happened.
*/
fprintf(stderr, "SDL: %s\n", message);
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
/* Don't try to close - may hang */
this->hidden->audio_fd = -1;
#ifdef DEBUG_AUDIO
@ -195,7 +195,7 @@ BSDAUDIO_PlayDevice(_THIS)
/* If we couldn't write, assume fatal error for now */
if (written < 0) {
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", written);
@ -224,7 +224,7 @@ BSDAUDIO_CloseDevice(_THIS)
}
static int
BSDAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
BSDAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
SDL_AudioFormat format = 0;
@ -348,6 +348,8 @@ BSDAUDIO_Init(SDL_AudioDriverImpl * impl)
impl->GetDeviceBuf = BSDAUDIO_GetDeviceBuf;
impl->CloseDevice = BSDAUDIO_CloseDevice;
impl->AllowsArbitraryDeviceNames = 1;
return 1; /* this audio target is available. */
}

2
Engine/lib/sdl/src/audio/bsd/SDL_bsdaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

337
Engine/lib/sdl/src/audio/coreaudio/SDL_coreaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -19,6 +19,9 @@
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#if SDL_AUDIO_DRIVER_COREAUDIO
#include "SDL_audio.h"
#include "../SDL_audio_c.h"
#include "../SDL_sysaudio.h"
@ -37,31 +40,48 @@ static void COREAUDIO_CloseDevice(_THIS);
}
#if MACOSX_COREAUDIO
typedef void (*addDevFn)(const char *name, AudioDeviceID devId, void *data);
static const AudioObjectPropertyAddress devlist_address = {
kAudioHardwarePropertyDevices,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster
};
static void
addToDevList(const char *name, AudioDeviceID devId, void *data)
typedef void (*addDevFn)(const char *name, const int iscapture, AudioDeviceID devId, void *data);
typedef struct AudioDeviceList
{
SDL_AddAudioDevice addfn = (SDL_AddAudioDevice) data;
addfn(name);
AudioDeviceID devid;
SDL_bool alive;
struct AudioDeviceList *next;
} AudioDeviceList;
static AudioDeviceList *output_devs = NULL;
static AudioDeviceList *capture_devs = NULL;
static SDL_bool
add_to_internal_dev_list(const int iscapture, AudioDeviceID devId)
{
AudioDeviceList *item = (AudioDeviceList *) SDL_malloc(sizeof (AudioDeviceList));
if (item == NULL) {
return SDL_FALSE;
}
item->devid = devId;
item->alive = SDL_TRUE;
item->next = iscapture ? capture_devs : output_devs;
if (iscapture) {
capture_devs = item;
} else {
output_devs = item;
}
return SDL_TRUE;
}
typedef struct
{
const char *findname;
AudioDeviceID devId;
int found;
} FindDevIdData;
static void
findDevId(const char *name, AudioDeviceID devId, void *_data)
addToDevList(const char *name, const int iscapture, AudioDeviceID devId, void *data)
{
FindDevIdData *data = (FindDevIdData *) _data;
if (!data->found) {
if (SDL_strcmp(name, data->findname) == 0) {
data->found = 1;
data->devId = devId;
}
if (add_to_internal_dev_list(iscapture, devId)) {
SDL_AddAudioDevice(iscapture, name, (void *) ((size_t) devId));
}
}
@ -74,14 +94,8 @@ build_device_list(int iscapture, addDevFn addfn, void *addfndata)
UInt32 i = 0;
UInt32 max = 0;
AudioObjectPropertyAddress addr = {
kAudioHardwarePropertyDevices,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster
};
result = AudioObjectGetPropertyDataSize(kAudioObjectSystemObject, &addr,
0, NULL, &size);
result = AudioObjectGetPropertyDataSize(kAudioObjectSystemObject,
&devlist_address, 0, NULL, &size);
if (result != kAudioHardwareNoError)
return;
@ -89,8 +103,8 @@ build_device_list(int iscapture, addDevFn addfn, void *addfndata)
if (devs == NULL)
return;
result = AudioObjectGetPropertyData(kAudioObjectSystemObject, &addr,
0, NULL, &size, devs);
result = AudioObjectGetPropertyData(kAudioObjectSystemObject,
&devlist_address, 0, NULL, &size, devs);
if (result != kAudioHardwareNoError)
return;
@ -102,10 +116,17 @@ build_device_list(int iscapture, addDevFn addfn, void *addfndata)
AudioBufferList *buflist = NULL;
int usable = 0;
CFIndex len = 0;
const AudioObjectPropertyAddress addr = {
kAudioDevicePropertyStreamConfiguration,
iscapture ? kAudioDevicePropertyScopeInput : kAudioDevicePropertyScopeOutput,
kAudioObjectPropertyElementMaster
};
addr.mScope = iscapture ? kAudioDevicePropertyScopeInput :
kAudioDevicePropertyScopeOutput;
addr.mSelector = kAudioDevicePropertyStreamConfiguration;
const AudioObjectPropertyAddress nameaddr = {
kAudioObjectPropertyName,
iscapture ? kAudioDevicePropertyScopeInput : kAudioDevicePropertyScopeOutput,
kAudioObjectPropertyElementMaster
};
result = AudioObjectGetPropertyDataSize(dev, &addr, 0, NULL, &size);
if (result != noErr)
@ -133,9 +154,9 @@ build_device_list(int iscapture, addDevFn addfn, void *addfndata)
if (!usable)
continue;
addr.mSelector = kAudioObjectPropertyName;
size = sizeof (CFStringRef);
result = AudioObjectGetPropertyData(dev, &addr, 0, NULL, &size, &cfstr);
result = AudioObjectGetPropertyData(dev, &nameaddr, 0, NULL, &size, &cfstr);
if (result != kAudioHardwareNoError)
continue;
@ -166,79 +187,84 @@ build_device_list(int iscapture, addDevFn addfn, void *addfndata)
((iscapture) ? "capture" : "output"),
(int) *devCount, ptr, (int) dev);
#endif
addfn(ptr, dev, addfndata);
addfn(ptr, iscapture, dev, addfndata);
}
SDL_free(ptr); /* addfn() would have copied the string. */
}
}
static void
COREAUDIO_DetectDevices(int iscapture, SDL_AddAudioDevice addfn)
free_audio_device_list(AudioDeviceList **list)
{
build_device_list(iscapture, addToDevList, addfn);
AudioDeviceList *item = *list;
while (item) {
AudioDeviceList *next = item->next;
SDL_free(item);
item = next;
}
*list = NULL;
}
static int
find_device_by_name(_THIS, const char *devname, int iscapture)
static void
COREAUDIO_DetectDevices(void)
{
AudioDeviceID devid = 0;
OSStatus result = noErr;
UInt32 size = 0;
UInt32 alive = 0;
pid_t pid = 0;
build_device_list(SDL_TRUE, addToDevList, NULL);
build_device_list(SDL_FALSE, addToDevList, NULL);
}
AudioObjectPropertyAddress addr = {
0,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster
};
if (devname == NULL) {
size = sizeof (AudioDeviceID);
addr.mSelector =
((iscapture) ? kAudioHardwarePropertyDefaultInputDevice :
kAudioHardwarePropertyDefaultOutputDevice);
result = AudioObjectGetPropertyData(kAudioObjectSystemObject, &addr,
0, NULL, &size, &devid);
CHECK_RESULT("AudioHardwareGetProperty (default device)");
} else {
FindDevIdData data;
SDL_zero(data);
data.findname = devname;
build_device_list(iscapture, findDevId, &data);
if (!data.found) {
SDL_SetError("CoreAudio: No such audio device.");
return 0;
static void
build_device_change_list(const char *name, const int iscapture, AudioDeviceID devId, void *data)
{
AudioDeviceList **list = (AudioDeviceList **) data;
AudioDeviceList *item;
for (item = *list; item != NULL; item = item->next) {
if (item->devid == devId) {
item->alive = SDL_TRUE;
return;
}
devid = data.devId;
}
addr.mSelector = kAudioDevicePropertyDeviceIsAlive;
addr.mScope = iscapture ? kAudioDevicePropertyScopeInput :
kAudioDevicePropertyScopeOutput;
add_to_internal_dev_list(iscapture, devId); /* new device, add it. */
SDL_AddAudioDevice(iscapture, name, (void *) ((size_t) devId));
}
size = sizeof (alive);
result = AudioObjectGetPropertyData(devid, &addr, 0, NULL, &size, &alive);
CHECK_RESULT
("AudioDeviceGetProperty (kAudioDevicePropertyDeviceIsAlive)");
if (!alive) {
SDL_SetError("CoreAudio: requested device exists, but isn't alive.");
return 0;
static void
reprocess_device_list(const int iscapture, AudioDeviceList **list)
{
AudioDeviceList *item;
AudioDeviceList *prev = NULL;
for (item = *list; item != NULL; item = item->next) {
item->alive = SDL_FALSE;
}
addr.mSelector = kAudioDevicePropertyHogMode;
size = sizeof (pid);
result = AudioObjectGetPropertyData(devid, &addr, 0, NULL, &size, &pid);
build_device_list(iscapture, build_device_change_list, list);
/* 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;
/* free items in the list that aren't still alive. */
item = *list;
while (item != NULL) {
AudioDeviceList *next = item->next;
if (item->alive) {
prev = item;
} else {
SDL_RemoveAudioDevice(iscapture, (void *) ((size_t) item->devid));
if (prev) {
prev->next = item->next;
} else {
*list = item->next;
}
SDL_free(item);
}
item = next;
}
}
this->hidden->deviceID = devid;
return 1;
/* this is called when the system's list of available audio devices changes. */
static OSStatus
device_list_changed(AudioObjectID systemObj, UInt32 num_addr, const AudioObjectPropertyAddress *addrs, void *data)
{
reprocess_device_list(SDL_TRUE, &capture_devs);
reprocess_device_list(SDL_FALSE, &output_devs);
return 0;
}
#endif
@ -314,12 +340,54 @@ inputCallback(void *inRefCon,
}
#if MACOSX_COREAUDIO
static const AudioObjectPropertyAddress alive_address =
{
kAudioDevicePropertyDeviceIsAlive,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster
};
static OSStatus
device_unplugged(AudioObjectID devid, UInt32 num_addr, const AudioObjectPropertyAddress *addrs, void *data)
{
SDL_AudioDevice *this = (SDL_AudioDevice *) data;
SDL_bool dead = SDL_FALSE;
UInt32 isAlive = 1;
UInt32 size = sizeof (isAlive);
OSStatus error;
if (!this->enabled) {
return 0; /* already known to be dead. */
}
error = AudioObjectGetPropertyData(this->hidden->deviceID, &alive_address,
0, NULL, &size, &isAlive);
if (error == kAudioHardwareBadDeviceError) {
dead = SDL_TRUE; /* device was unplugged. */
} else if ((error == kAudioHardwareNoError) && (!isAlive)) {
dead = SDL_TRUE; /* device died in some other way. */
}
if (dead) {
SDL_OpenedAudioDeviceDisconnected(this);
}
return 0;
}
#endif
static void
COREAUDIO_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->audioUnitOpened) {
OSStatus result = noErr;
#if MACOSX_COREAUDIO
/* Unregister our disconnect callback. */
AudioObjectRemovePropertyListener(this->hidden->deviceID, &alive_address, device_unplugged, this);
#endif
AURenderCallbackStruct callback;
const AudioUnitElement output_bus = 0;
const AudioUnitElement input_bus = 1;
@ -331,14 +399,13 @@ COREAUDIO_CloseDevice(_THIS)
kAudioUnitScope_Input);
/* stop processing the audio unit */
result = AudioOutputUnitStop(this->hidden->audioUnit);
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));
AudioUnitSetProperty(this->hidden->audioUnit,
kAudioUnitProperty_SetRenderCallback,
scope, bus, &callback, sizeof(callback));
#if MACOSX_COREAUDIO
CloseComponent(this->hidden->audioUnit);
@ -354,9 +421,63 @@ COREAUDIO_CloseDevice(_THIS)
}
}
#if MACOSX_COREAUDIO
static int
prepare_device(_THIS, void *handle, int iscapture)
{
AudioDeviceID devid = (AudioDeviceID) ((size_t) handle);
OSStatus result = noErr;
UInt32 size = 0;
UInt32 alive = 0;
pid_t pid = 0;
AudioObjectPropertyAddress addr = {
0,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster
};
if (handle == NULL) {
size = sizeof (AudioDeviceID);
addr.mSelector =
((iscapture) ? kAudioHardwarePropertyDefaultInputDevice :
kAudioHardwarePropertyDefaultOutputDevice);
result = AudioObjectGetPropertyData(kAudioObjectSystemObject, &addr,
0, NULL, &size, &devid);
CHECK_RESULT("AudioHardwareGetProperty (default device)");
}
addr.mSelector = kAudioDevicePropertyDeviceIsAlive;
addr.mScope = iscapture ? kAudioDevicePropertyScopeInput :
kAudioDevicePropertyScopeOutput;
size = sizeof (alive);
result = AudioObjectGetPropertyData(devid, &addr, 0, NULL, &size, &alive);
CHECK_RESULT
("AudioDeviceGetProperty (kAudioDevicePropertyDeviceIsAlive)");
if (!alive) {
SDL_SetError("CoreAudio: requested device exists, but isn't alive.");
return 0;
}
addr.mSelector = kAudioDevicePropertyHogMode;
size = sizeof (pid);
result = AudioObjectGetPropertyData(devid, &addr, 0, NULL, &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;
}
#endif
static int
prepare_audiounit(_THIS, const char *devname, int iscapture,
prepare_audiounit(_THIS, void *handle, int iscapture,
const AudioStreamBasicDescription * strdesc)
{
OSStatus result = noErr;
@ -375,8 +496,7 @@ prepare_audiounit(_THIS, const char *devname, int iscapture,
kAudioUnitScope_Input);
#if MACOSX_COREAUDIO
if (!find_device_by_name(this, devname, iscapture)) {
SDL_SetError("Couldn't find requested CoreAudio device");
if (!prepare_device(this, handle, iscapture)) {
return 0;
}
#endif
@ -453,13 +573,18 @@ prepare_audiounit(_THIS, const char *devname, int iscapture,
result = AudioOutputUnitStart(this->hidden->audioUnit);
CHECK_RESULT("AudioOutputUnitStart");
#if MACOSX_COREAUDIO
/* Fire a callback if the device stops being "alive" (disconnected, etc). */
AudioObjectAddPropertyListener(this->hidden->deviceID, &alive_address, device_unplugged, this);
#endif
/* We're running! */
return 1;
}
static int
COREAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
COREAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
AudioStreamBasicDescription strdesc;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
@ -518,7 +643,7 @@ COREAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
strdesc.mBytesPerPacket =
strdesc.mBytesPerFrame * strdesc.mFramesPerPacket;
if (!prepare_audiounit(this, devname, iscapture, &strdesc)) {
if (!prepare_audiounit(this, handle, iscapture, &strdesc)) {
COREAUDIO_CloseDevice(this);
return -1; /* prepare_audiounit() will call SDL_SetError()... */
}
@ -526,15 +651,27 @@ COREAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
return 0; /* good to go. */
}
static void
COREAUDIO_Deinitialize(void)
{
#if MACOSX_COREAUDIO
AudioObjectRemovePropertyListener(kAudioObjectSystemObject, &devlist_address, device_list_changed, NULL);
free_audio_device_list(&capture_devs);
free_audio_device_list(&output_devs);
#endif
}
static int
COREAUDIO_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->OpenDevice = COREAUDIO_OpenDevice;
impl->CloseDevice = COREAUDIO_CloseDevice;
impl->Deinitialize = COREAUDIO_Deinitialize;
#if MACOSX_COREAUDIO
impl->DetectDevices = COREAUDIO_DetectDevices;
AudioObjectAddPropertyListener(kAudioObjectSystemObject, &devlist_address, device_list_changed, NULL);
#else
impl->OnlyHasDefaultOutputDevice = 1;
@ -556,4 +693,6 @@ AudioBootStrap COREAUDIO_bootstrap = {
"coreaudio", "CoreAudio", COREAUDIO_Init, 0
};
#endif /* SDL_AUDIO_DRIVER_COREAUDIO */
/* vi: set ts=4 sw=4 expandtab: */

2
Engine/lib/sdl/src/audio/coreaudio/SDL_coreaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

67
Engine/lib/sdl/src/audio/directsound/SDL_directsound.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -144,15 +144,22 @@ SetDSerror(const char *function, int code)
return SDL_SetError("%s", errbuf);
}
static void
DSOUND_FreeDeviceHandle(void *handle)
{
SDL_free(handle);
}
static BOOL CALLBACK
FindAllDevs(LPGUID guid, LPCWSTR desc, LPCWSTR module, LPVOID data)
{
SDL_AddAudioDevice addfn = (SDL_AddAudioDevice) data;
const int iscapture = (int) ((size_t) data);
if (guid != NULL) { /* skip default device */
char *str = WIN_StringToUTF8(desc);
if (str != NULL) {
addfn(str);
LPGUID cpyguid = (LPGUID) SDL_malloc(sizeof (GUID));
SDL_memcpy(cpyguid, guid, sizeof (GUID));
SDL_AddAudioDevice(iscapture, str, cpyguid);
SDL_free(str); /* addfn() makes a copy of this string. */
}
}
@ -160,13 +167,10 @@ FindAllDevs(LPGUID guid, LPCWSTR desc, LPCWSTR module, LPVOID data)
}
static void
DSOUND_DetectDevices(int iscapture, SDL_AddAudioDevice addfn)
DSOUND_DetectDevices(void)
{
if (iscapture) {
pDirectSoundCaptureEnumerateW(FindAllDevs, addfn);
} else {
pDirectSoundEnumerateW(FindAllDevs, addfn);
}
pDirectSoundCaptureEnumerateW(FindAllDevs, (void *) ((size_t) 1));
pDirectSoundEnumerateW(FindAllDevs, (void *) ((size_t) 0));
}
@ -419,53 +423,14 @@ CreateSecondary(_THIS, HWND focus)
return (numchunks);
}
typedef struct FindDevGUIDData
{
const char *devname;
GUID guid;
int found;
} FindDevGUIDData;
static BOOL CALLBACK
FindDevGUID(LPGUID guid, LPCWSTR desc, LPCWSTR module, LPVOID _data)
{
if (guid != NULL) { /* skip the default device. */
FindDevGUIDData *data = (FindDevGUIDData *) _data;
char *str = WIN_StringToUTF8(desc);
const int match = (SDL_strcmp(str, data->devname) == 0);
SDL_free(str);
if (match) {
data->found = 1;
SDL_memcpy(&data->guid, guid, sizeof (data->guid));
return FALSE; /* found it! stop enumerating. */
}
}
return TRUE; /* keep enumerating. */
}
static int
DSOUND_OpenDevice(_THIS, const char *devname, int iscapture)
DSOUND_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
HRESULT result;
SDL_bool valid_format = SDL_FALSE;
SDL_bool tried_format = SDL_FALSE;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
FindDevGUIDData devguid;
LPGUID guid = NULL;
if (devname != NULL) {
devguid.found = 0;
devguid.devname = devname;
if (iscapture)
pDirectSoundCaptureEnumerateW(FindDevGUID, &devguid);
else
pDirectSoundEnumerateW(FindDevGUID, &devguid);
if (!devguid.found) {
return SDL_SetError("DirectSound: Requested device not found");
}
guid = &devguid.guid;
}
LPGUID guid = (LPGUID) handle;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
@ -536,6 +501,8 @@ DSOUND_Init(SDL_AudioDriverImpl * impl)
impl->WaitDone = DSOUND_WaitDone;
impl->GetDeviceBuf = DSOUND_GetDeviceBuf;
impl->CloseDevice = DSOUND_CloseDevice;
impl->FreeDeviceHandle = DSOUND_FreeDeviceHandle;
impl->Deinitialize = DSOUND_Deinitialize;
return 1; /* this audio target is available. */

4
Engine/lib/sdl/src/audio/directsound/SDL_directsound.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -23,7 +23,7 @@
#ifndef _SDL_directsound_h
#define _SDL_directsound_h
#include "directx.h"
#include "../../core/windows/SDL_directx.h"
#include "../SDL_sysaudio.h"

102
Engine/lib/sdl/src/audio/directsound/directx.h
View file

@ -1,102 +0,0 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#ifndef _directx_h
#define _directx_h
/* Include all of the DirectX 8.0 headers and adds any necessary tweaks */
#include "../../core/windows/SDL_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 0x0800
#define DIRECTINPUT_VERSION 0x0500
#include <ddraw.h>
#include <dsound.h>
#include <dinput.h>
#endif /* _directx_h */
/* vi: set ts=4 sw=4 expandtab: */

19
Engine/lib/sdl/src/audio/disk/SDL_diskaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -71,7 +71,7 @@ DISKAUD_PlayDevice(_THIS)
/* If we couldn't write, assume fatal error for now */
if (written != this->hidden->mixlen) {
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", written);
@ -100,10 +100,11 @@ DISKAUD_CloseDevice(_THIS)
}
static int
DISKAUD_OpenDevice(_THIS, const char *devname, int iscapture)
DISKAUD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
/* handle != NULL means "user specified the placeholder name on the fake detected device list" */
const char *fname = DISKAUD_GetOutputFilename(handle ? NULL : devname);
const char *envr = SDL_getenv(DISKENVR_WRITEDELAY);
const char *fname = DISKAUD_GetOutputFilename(devname);
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc(sizeof(*this->hidden));
@ -141,6 +142,13 @@ DISKAUD_OpenDevice(_THIS, const char *devname, int iscapture)
return 0;
}
static void
DISKAUD_DetectDevices(void)
{
/* !!! FIXME: stole this literal string from DEFAULT_OUTPUT_DEVNAME in SDL_audio.c */
SDL_AddAudioDevice(SDL_FALSE, "System audio output device", (void *) 0x1);
}
static int
DISKAUD_Init(SDL_AudioDriverImpl * impl)
{
@ -150,6 +158,9 @@ DISKAUD_Init(SDL_AudioDriverImpl * impl)
impl->PlayDevice = DISKAUD_PlayDevice;
impl->GetDeviceBuf = DISKAUD_GetDeviceBuf;
impl->CloseDevice = DISKAUD_CloseDevice;
impl->DetectDevices = DISKAUD_DetectDevices;
impl->AllowsArbitraryDeviceNames = 1;
return 1; /* this audio target is available. */
}

2
Engine/lib/sdl/src/audio/disk/SDL_diskaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

12
Engine/lib/sdl/src/audio/dsp/SDL_dspaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -51,9 +51,9 @@
static void
DSP_DetectDevices(int iscapture, SDL_AddAudioDevice addfn)
DSP_DetectDevices(void)
{
SDL_EnumUnixAudioDevices(iscapture, 0, NULL, addfn);
SDL_EnumUnixAudioDevices(0, NULL);
}
@ -74,7 +74,7 @@ DSP_CloseDevice(_THIS)
static int
DSP_OpenDevice(_THIS, const char *devname, int iscapture)
DSP_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
int format;
@ -270,7 +270,7 @@ DSP_PlayDevice(_THIS)
const int mixlen = this->hidden->mixlen;
if (write(this->hidden->audio_fd, mixbuf, mixlen) == -1) {
perror("Audio write");
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", mixlen);
@ -293,6 +293,8 @@ DSP_Init(SDL_AudioDriverImpl * impl)
impl->GetDeviceBuf = DSP_GetDeviceBuf;
impl->CloseDevice = DSP_CloseDevice;
impl->AllowsArbitraryDeviceNames = 1;
return 1; /* this audio target is available. */
}

2
Engine/lib/sdl/src/audio/dsp/SDL_dspaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

4
Engine/lib/sdl/src/audio/dummy/SDL_dummyaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -27,7 +27,7 @@
#include "SDL_dummyaudio.h"
static int
DUMMYAUD_OpenDevice(_THIS, const char *devname, int iscapture)
DUMMYAUD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
return 0; /* always succeeds. */
}

2
Engine/lib/sdl/src/audio/dummy/SDL_dummyaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

279
Engine/lib/sdl/src/audio/emscripten/SDL_emscriptenaudio.c Normal file
View file

@ -0,0 +1,279 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#if SDL_AUDIO_DRIVER_EMSCRIPTEN
#include "SDL_audio.h"
#include "SDL_log.h"
#include "../SDL_audio_c.h"
#include "SDL_emscriptenaudio.h"
#include <emscripten/emscripten.h>
static int
copyData(_THIS)
{
int byte_len;
if (this->hidden->write_off + this->convert.len_cvt > this->hidden->mixlen) {
if (this->hidden->write_off > this->hidden->read_off) {
SDL_memmove(this->hidden->mixbuf,
this->hidden->mixbuf + this->hidden->read_off,
this->hidden->mixlen - this->hidden->read_off);
this->hidden->write_off = this->hidden->write_off - this->hidden->read_off;
} else {
this->hidden->write_off = 0;
}
this->hidden->read_off = 0;
}
SDL_memcpy(this->hidden->mixbuf + this->hidden->write_off,
this->convert.buf,
this->convert.len_cvt);
this->hidden->write_off += this->convert.len_cvt;
byte_len = this->hidden->write_off - this->hidden->read_off;
return byte_len;
}
static void
HandleAudioProcess(_THIS)
{
Uint8 *buf = NULL;
int byte_len = 0;
int bytes = SDL_AUDIO_BITSIZE(this->spec.format) / 8;
int bytes_in = SDL_AUDIO_BITSIZE(this->convert.src_format) / 8;
/* Only do soemthing if audio is enabled */
if (!this->enabled)
return;
if (this->paused)
return;
if (this->convert.needed) {
if (this->hidden->conv_in_len != 0) {
this->convert.len = this->hidden->conv_in_len * bytes_in * this->spec.channels;
}
(*this->spec.callback) (this->spec.userdata,
this->convert.buf,
this->convert.len);
SDL_ConvertAudio(&this->convert);
buf = this->convert.buf;
byte_len = this->convert.len_cvt;
/* size mismatch*/
if (byte_len != this->spec.size) {
if (!this->hidden->mixbuf) {
this->hidden->mixlen = this->spec.size > byte_len ? this->spec.size * 2 : byte_len * 2;
this->hidden->mixbuf = SDL_malloc(this->hidden->mixlen);
}
/* copy existing data */
byte_len = copyData(this);
/* read more data*/
while (byte_len < this->spec.size) {
(*this->spec.callback) (this->spec.userdata,
this->convert.buf,
this->convert.len);
SDL_ConvertAudio(&this->convert);
byte_len = copyData(this);
}
byte_len = this->spec.size;
buf = this->hidden->mixbuf + this->hidden->read_off;
this->hidden->read_off += byte_len;
}
} else {
if (!this->hidden->mixbuf) {
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = SDL_malloc(this->hidden->mixlen);
}
(*this->spec.callback) (this->spec.userdata,
this->hidden->mixbuf,
this->hidden->mixlen);
buf = this->hidden->mixbuf;
byte_len = this->hidden->mixlen;
}
if (buf) {
EM_ASM_ARGS({
var numChannels = SDL2.audio.currentOutputBuffer['numberOfChannels'];
for (var c = 0; c < numChannels; ++c) {
var channelData = SDL2.audio.currentOutputBuffer['getChannelData'](c);
if (channelData.length != $1) {
throw 'Web Audio output buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
}
for (var j = 0; j < $1; ++j) {
channelData[j] = getValue($0 + (j*numChannels + c)*4, 'float');
}
}
}, buf, byte_len / bytes / this->spec.channels);
}
}
static void
Emscripten_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
/* Clean up the audio buffer */
SDL_free(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
static int
Emscripten_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
SDL_bool valid_format = SDL_FALSE;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
int i;
float f;
int result;
while ((!valid_format) && (test_format)) {
switch (test_format) {
case AUDIO_F32: /* web audio only supports floats */
this->spec.format = test_format;
valid_format = SDL_TRUE;
break;
}
test_format = SDL_NextAudioFormat();
}
if (!valid_format) {
/* Didn't find a compatible format :( */
return SDL_SetError("No compatible audio format!");
}
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* based on parts of library_sdl.js */
/* create context (TODO: this puts stuff in the global namespace...)*/
result = EM_ASM_INT_V({
if(typeof(SDL2) === 'undefined')
SDL2 = {};
if(typeof(SDL2.audio) === 'undefined')
SDL2.audio = {};
if (!SDL2.audioContext) {
if (typeof(AudioContext) !== 'undefined') {
SDL2.audioContext = new AudioContext();
} else if (typeof(webkitAudioContext) !== 'undefined') {
SDL2.audioContext = new webkitAudioContext();
} else {
return -1;
}
}
return 0;
});
if (result < 0) {
return SDL_SetError("Web Audio API is not available!");
}
/* limit to native freq */
int sampleRate = EM_ASM_INT_V({
return SDL2.audioContext['sampleRate'];
});
if(this->spec.freq != sampleRate) {
for (i = this->spec.samples; i > 0; i--) {
f = (float)i / (float)sampleRate * (float)this->spec.freq;
if (SDL_floor(f) == f) {
this->hidden->conv_in_len = SDL_floor(f);
break;
}
}
this->spec.freq = sampleRate;
}
SDL_CalculateAudioSpec(&this->spec);
/* setup a ScriptProcessorNode */
EM_ASM_ARGS({
SDL2.audio.scriptProcessorNode = SDL2.audioContext['createScriptProcessor']($1, 0, $0);
SDL2.audio.scriptProcessorNode['onaudioprocess'] = function (e) {
SDL2.audio.currentOutputBuffer = e['outputBuffer'];
Runtime.dynCall('vi', $2, [$3]);
};
SDL2.audio.scriptProcessorNode['connect'](SDL2.audioContext['destination']);
}, this->spec.channels, this->spec.samples, HandleAudioProcess, this);
return 0;
}
static int
Emscripten_Init(SDL_AudioDriverImpl * impl)
{
/* Set the function pointers */
impl->OpenDevice = Emscripten_OpenDevice;
impl->CloseDevice = Emscripten_CloseDevice;
/* only one output */
impl->OnlyHasDefaultOutputDevice = 1;
/* no threads here */
impl->SkipMixerLock = 1;
impl->ProvidesOwnCallbackThread = 1;
/* check availability */
int available = EM_ASM_INT_V({
if (typeof(AudioContext) !== 'undefined') {
return 1;
} else if (typeof(webkitAudioContext) !== 'undefined') {
return 1;
}
return 0;
});
if (!available) {
SDL_SetError("No audio context available");
}
return available;
}
AudioBootStrap EmscriptenAudio_bootstrap = {
"emscripten", "SDL emscripten audio driver", Emscripten_Init, 0
};
#endif /* SDL_AUDIO_DRIVER_EMSCRIPTEN */
/* vi: set ts=4 sw=4 expandtab: */

42
Engine/lib/sdl/src/audio/emscripten/SDL_emscriptenaudio.h Normal file
View file

@ -0,0 +1,42 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#ifndef _SDL_emscriptenaudio_h
#define _SDL_emscriptenaudio_h
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
struct SDL_PrivateAudioData
{
Uint8 *mixbuf;
Uint32 mixlen;
Uint32 conv_in_len;
Uint32 write_off, read_off;
};
#endif /* _SDL_emscriptenaudio_h */
/* vi: set ts=4 sw=4 expandtab: */

8
Engine/lib/sdl/src/audio/esd/SDL_esdaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -129,7 +129,7 @@ ESD_WaitDevice(_THIS)
/* Check every 10 loops */
if (this->hidden->parent && (((++cnt) % 10) == 0)) {
if (kill(this->hidden->parent, 0) < 0 && errno == ESRCH) {
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
}
}
}
@ -161,7 +161,7 @@ ESD_PlayDevice(_THIS)
/* If we couldn't write, assume fatal error for now */
if (written < 0) {
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
}
}
@ -215,7 +215,7 @@ get_progname(void)
static int
ESD_OpenDevice(_THIS, const char *devname, int iscapture)
ESD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
esd_format_t format = (ESD_STREAM | ESD_PLAY);
SDL_AudioFormat test_format = 0;

2
Engine/lib/sdl/src/audio/esd/SDL_esdaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

6
Engine/lib/sdl/src/audio/fusionsound/SDL_fsaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -143,7 +143,7 @@ SDL_FS_PlayDevice(_THIS)
this->hidden->mixsamples);
/* If we couldn't write, assume fatal error for now */
if (ret) {
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", this->hidden->mixlen);
@ -186,7 +186,7 @@ SDL_FS_CloseDevice(_THIS)
static int
SDL_FS_OpenDevice(_THIS, const char *devname, int iscapture)
SDL_FS_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
int bytes;
SDL_AudioFormat test_format = 0, format = 0;

2
Engine/lib/sdl/src/audio/fusionsound/SDL_fsaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

4
Engine/lib/sdl/src/audio/haiku/SDL_haikuaudio.cc
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -111,7 +111,7 @@ UnmaskSignals(sigset_t * omask)
static int
HAIKUAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
HAIKUAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
int valid_datatype = 0;
media_raw_audio_format format;

2
Engine/lib/sdl/src/audio/haiku/SDL_haikuaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

152
Engine/lib/sdl/src/audio/nacl/SDL_naclaudio.c Normal file
View file

@ -0,0 +1,152 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#if SDL_AUDIO_DRIVER_NACL
#include "SDL_naclaudio.h"
#include "SDL_audio.h"
#include "SDL_mutex.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "../SDL_audiodev_c.h"
#include "ppapi/c/pp_errors.h"
#include "ppapi/c/pp_instance.h"
#include "ppapi_simple/ps.h"
#include "ppapi_simple/ps_interface.h"
#include "ppapi_simple/ps_event.h"
/* The tag name used by NACL audio */
#define NACLAUD_DRIVER_NAME "nacl"
#define SAMPLE_FRAME_COUNT 4096
/* Audio driver functions */
static int NACLAUD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture);
static void NACLAUD_CloseDevice(_THIS);
static void nacl_audio_callback(void* samples, uint32_t buffer_size, PP_TimeDelta latency, void* data);
/* FIXME: Make use of latency if needed */
static void nacl_audio_callback(void* samples, uint32_t buffer_size, PP_TimeDelta latency, void* data) {
SDL_AudioDevice* _this = (SDL_AudioDevice*) data;
SDL_LockMutex(private->mutex);
if (_this->enabled && !_this->paused) {
if (_this->convert.needed) {
SDL_LockMutex(_this->mixer_lock);
(*_this->spec.callback) (_this->spec.userdata,
(Uint8 *) _this->convert.buf,
_this->convert.len);
SDL_UnlockMutex(_this->mixer_lock);
SDL_ConvertAudio(&_this->convert);
SDL_memcpy(samples, _this->convert.buf, _this->convert.len_cvt);
} else {
SDL_LockMutex(_this->mixer_lock);
(*_this->spec.callback) (_this->spec.userdata, (Uint8 *) samples, buffer_size);
SDL_UnlockMutex(_this->mixer_lock);
}
} else {
SDL_memset(samples, 0, buffer_size);
}
return;
}
static void NACLAUD_CloseDevice(SDL_AudioDevice *device) {
const PPB_Core *core = PSInterfaceCore();
const PPB_Audio *ppb_audio = PSInterfaceAudio();
SDL_PrivateAudioData *hidden = (SDL_PrivateAudioData *) device->hidden;
ppb_audio->StopPlayback(hidden->audio);
SDL_DestroyMutex(hidden->mutex);
core->ReleaseResource(hidden->audio);
}
static int
NACLAUD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture) {
PP_Instance instance = PSGetInstanceId();
const PPB_Audio *ppb_audio = PSInterfaceAudio();
const PPB_AudioConfig *ppb_audiocfg = PSInterfaceAudioConfig();
private = (SDL_PrivateAudioData *) SDL_calloc(1, (sizeof *private));
if (private == NULL) {
SDL_OutOfMemory();
return 0;
}
private->mutex = SDL_CreateMutex();
_this->spec.freq = 44100;
_this->spec.format = AUDIO_S16LSB;
_this->spec.channels = 2;
_this->spec.samples = ppb_audiocfg->RecommendSampleFrameCount(
instance,
PP_AUDIOSAMPLERATE_44100,
SAMPLE_FRAME_COUNT);
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&_this->spec);
private->audio = ppb_audio->Create(
instance,
ppb_audiocfg->CreateStereo16Bit(instance, PP_AUDIOSAMPLERATE_44100, _this->spec.samples),
nacl_audio_callback,
_this);
/* Start audio playback while we are still on the main thread. */
ppb_audio->StartPlayback(private->audio);
return 1;
}
static int
NACLAUD_Init(SDL_AudioDriverImpl * impl)
{
if (PSGetInstanceId() == 0) {
return 0;
}
/* Set the function pointers */
impl->OpenDevice = NACLAUD_OpenDevice;
impl->CloseDevice = NACLAUD_CloseDevice;
impl->OnlyHasDefaultOutputDevice = 1;
impl->ProvidesOwnCallbackThread = 1;
/*
* impl->WaitDevice = NACLAUD_WaitDevice;
* impl->GetDeviceBuf = NACLAUD_GetDeviceBuf;
* impl->PlayDevice = NACLAUD_PlayDevice;
* impl->Deinitialize = NACLAUD_Deinitialize;
*/
return 1;
}
AudioBootStrap NACLAUD_bootstrap = {
NACLAUD_DRIVER_NAME, "SDL NaCl Audio Driver",
NACLAUD_Init, 0
};
#endif /* SDL_AUDIO_DRIVER_NACL */
/* vi: set ts=4 sw=4 expandtab: */

41
Engine/lib/sdl/src/audio/nacl/SDL_naclaudio.h Normal file
View file

@ -0,0 +1,41 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#ifndef _SDL_naclaudio_h
#define _SDL_naclaudio_h
#include "SDL_audio.h"
#include "../SDL_sysaudio.h"
#include "SDL_mutex.h"
#include "ppapi/c/ppb_audio.h"
#define _THIS SDL_AudioDevice *_this
#define private _this->hidden
typedef struct SDL_PrivateAudioData {
SDL_mutex* mutex;
PP_Resource audio;
} SDL_PrivateAudioData;
#endif /* _SDL_naclaudio_h */

4
Engine/lib/sdl/src/audio/nas/SDL_nasaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -276,7 +276,7 @@ find_device(_THIS, int nch)
}
static int
NAS_OpenDevice(_THIS, const char *devname, int iscapture)
NAS_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
AuElement elms[3];
int buffer_size;

2
Engine/lib/sdl/src/audio/nas/SDL_nasaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

10
Engine/lib/sdl/src/audio/paudio/SDL_paudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -111,7 +111,7 @@ OpenAudioPath(char *path, int maxlen, int flags, int classic)
if (stat(audiopath, &sb) == 0) {
fd = open(audiopath, flags, 0);
if (fd > 0) {
if (fd >= 0) {
if (path != NULL) {
SDL_strlcpy(path, audiopath, maxlen);
}
@ -176,7 +176,7 @@ PAUDIO_WaitDevice(_THIS)
* the user know what happened.
*/
fprintf(stderr, "SDL: %s - %s\n", strerror(errno), message);
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
/* Don't try to close - may hang */
this->hidden->audio_fd = -1;
#ifdef DEBUG_AUDIO
@ -212,7 +212,7 @@ PAUDIO_PlayDevice(_THIS)
/* If we couldn't write, assume fatal error for now */
if (written < 0) {
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", written);
@ -241,7 +241,7 @@ PAUDIO_CloseDevice(_THIS)
}
static int
PAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
PAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
const char *workaround = SDL_getenv("SDL_DSP_NOSELECT");
char audiodev[1024];

2
Engine/lib/sdl/src/audio/paudio/SDL_paudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

10
Engine/lib/sdl/src/audio/psp/SDL_pspaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -18,6 +18,9 @@
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#include "../../SDL_internal.h"
#if SDL_AUDIO_DRIVER_PSP
#include <stdio.h>
#include <string.h>
@ -40,7 +43,7 @@
#define PSPAUD_DRIVER_NAME "psp"
static int
PSPAUD_OpenDevice(_THIS, const char *devname, int iscapture)
PSPAUD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
int format, mixlen, i;
this->hidden = (struct SDL_PrivateAudioData *)
@ -191,5 +194,6 @@ AudioBootStrap PSPAUD_bootstrap = {
/* SDL_AUDI */
#endif /* SDL_AUDIO_DRIVER_PSP */
/* vi: set ts=4 sw=4 expandtab: */

4
Engine/lib/sdl/src/audio/psp/SDL_pspaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -24,7 +24,7 @@
#include "../SDL_sysaudio.h"
/* Hidden "this" pointer for the video functions */
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
#define NUM_BUFFERS 2

467
Engine/lib/sdl/src/audio/pulseaudio/SDL_pulseaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -26,6 +26,7 @@
Stéphan Kochen: stephan .a.t. kochen.nl
*/
#include "../../SDL_internal.h"
#include "SDL_assert.h"
#if SDL_AUDIO_DRIVER_PULSEAUDIO
@ -38,7 +39,6 @@
#include <sys/types.h>
#include <errno.h>
#include <pulse/pulseaudio.h>
#include <pulse/simple.h>
#include "SDL_timer.h"
#include "SDL_audio.h"
@ -66,16 +66,14 @@ static SDL_INLINE int PA_STREAM_IS_GOOD(pa_stream_state_t x) {
static const char *(*PULSEAUDIO_pa_get_library_version) (void);
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 int (*PULSEAUDIO_pa_mainloop_run) (pa_mainloop *, int *);
static void (*PULSEAUDIO_pa_mainloop_quit) (pa_mainloop *, int);
static void (*PULSEAUDIO_pa_mainloop_free) (pa_mainloop *);
static pa_operation_state_t (*PULSEAUDIO_pa_operation_get_state) (
@ -87,7 +85,13 @@ 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_operation * (*PULSEAUDIO_pa_context_get_sink_info_list) (pa_context *, pa_sink_info_cb_t, void *);
static pa_operation * (*PULSEAUDIO_pa_context_get_source_info_list) (pa_context *, pa_source_info_cb_t, void *);
static pa_operation * (*PULSEAUDIO_pa_context_get_sink_info_by_index) (pa_context *, uint32_t, pa_sink_info_cb_t, void *);
static pa_operation * (*PULSEAUDIO_pa_context_get_source_info_by_index) (pa_context *, uint32_t, pa_source_info_cb_t, void *);
static pa_context_state_t (*PULSEAUDIO_pa_context_get_state) (pa_context *);
static pa_operation * (*PULSEAUDIO_pa_context_subscribe) (pa_context *, pa_subscription_mask_t, pa_context_success_cb_t, void *);
static void (*PULSEAUDIO_pa_context_set_subscribe_callback) (pa_context *, pa_context_subscribe_cb_t, void *);
static void (*PULSEAUDIO_pa_context_disconnect) (pa_context *);
static void (*PULSEAUDIO_pa_context_unref) (pa_context *);
@ -179,18 +183,24 @@ static int
load_pulseaudio_syms(void)
{
SDL_PULSEAUDIO_SYM(pa_get_library_version);
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_run);
SDL_PULSEAUDIO_SYM(pa_mainloop_quit);
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_sink_info_list);
SDL_PULSEAUDIO_SYM(pa_context_get_source_info_list);
SDL_PULSEAUDIO_SYM(pa_context_get_sink_info_by_index);
SDL_PULSEAUDIO_SYM(pa_context_get_source_info_by_index);
SDL_PULSEAUDIO_SYM(pa_context_get_state);
SDL_PULSEAUDIO_SYM(pa_context_subscribe);
SDL_PULSEAUDIO_SYM(pa_context_set_subscribe_callback);
SDL_PULSEAUDIO_SYM(pa_context_disconnect);
SDL_PULSEAUDIO_SYM(pa_context_unref);
SDL_PULSEAUDIO_SYM(pa_stream_new);
@ -206,122 +216,6 @@ load_pulseaudio_syms(void)
return 0;
}
/* Check to see if we can connect to PulseAudio */
static SDL_bool
CheckPulseAudioAvailable()
{
pa_simple *s;
pa_sample_spec ss;
ss.format = PA_SAMPLE_S16NE;
ss.channels = 1;
ss.rate = 22050;
s = PULSEAUDIO_pa_simple_new(NULL, "SDL", PA_STREAM_PLAYBACK, NULL,
"Test", &ss, NULL, NULL, NULL);
if (s) {
PULSEAUDIO_pa_simple_free(s);
return SDL_TRUE;
} else {
return SDL_FALSE;
}
}
/* 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) {
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;
}
}
static SDL_INLINE int
squashVersion(const int major, const int minor, const int patch)
{
@ -344,8 +238,193 @@ getAppName(void)
return "SDL Application"; /* oh well. */
}
static void
WaitForPulseOperation(pa_mainloop *mainloop, pa_operation *o)
{
/* This checks for NO errors currently. Either fix that, check results elsewhere, or do things you don't care about. */
if (mainloop && o) {
SDL_bool okay = SDL_TRUE;
while (okay && (PULSEAUDIO_pa_operation_get_state(o) == PA_OPERATION_RUNNING)) {
okay = (PULSEAUDIO_pa_mainloop_iterate(mainloop, 1, NULL) >= 0);
}
PULSEAUDIO_pa_operation_unref(o);
}
}
static void
DisconnectFromPulseServer(pa_mainloop *mainloop, pa_context *context)
{
if (context) {
PULSEAUDIO_pa_context_disconnect(context);
PULSEAUDIO_pa_context_unref(context);
}
if (mainloop != NULL) {
PULSEAUDIO_pa_mainloop_free(mainloop);
}
}
static int
PULSEAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
ConnectToPulseServer_Internal(pa_mainloop **_mainloop, pa_context **_context)
{
pa_mainloop *mainloop = NULL;
pa_context *context = NULL;
pa_mainloop_api *mainloop_api = NULL;
int state = 0;
*_mainloop = NULL;
*_context = NULL;
/* Set up a new main loop */
if (!(mainloop = PULSEAUDIO_pa_mainloop_new())) {
return SDL_SetError("pa_mainloop_new() failed");
}
*_mainloop = mainloop;
mainloop_api = PULSEAUDIO_pa_mainloop_get_api(mainloop);
SDL_assert(mainloop_api); /* this never fails, right? */
context = PULSEAUDIO_pa_context_new(mainloop_api, getAppName());
if (!context) {
return SDL_SetError("pa_context_new() failed");
}
*_context = context;
/* Connect to the PulseAudio server */
if (PULSEAUDIO_pa_context_connect(context, NULL, 0, NULL) < 0) {
return SDL_SetError("Could not setup connection to PulseAudio");
}
do {
if (PULSEAUDIO_pa_mainloop_iterate(mainloop, 1, NULL) < 0) {
return SDL_SetError("pa_mainloop_iterate() failed");
}
state = PULSEAUDIO_pa_context_get_state(context);
if (!PA_CONTEXT_IS_GOOD(state)) {
return SDL_SetError("Could not connect to PulseAudio");
}
} while (state != PA_CONTEXT_READY);
return 0; /* connected and ready! */
}
static int
ConnectToPulseServer(pa_mainloop **_mainloop, pa_context **_context)
{
const int retval = ConnectToPulseServer_Internal(_mainloop, _context);
if (retval < 0) {
DisconnectFromPulseServer(*_mainloop, *_context);
}
return retval;
}
/* 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 (this->enabled) {
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) {
SDL_OpenedAudioDeviceDisconnected(this);
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 (this->enabled) {
if (PULSEAUDIO_pa_stream_write(h->stream, h->mixbuf, h->mixlen, NULL, 0LL, PA_SEEK_RELATIVE) < 0) {
SDL_OpenedAudioDeviceDisconnected(this);
}
}
}
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)
{
if (this->enabled) {
struct SDL_PrivateAudioData *h = this->hidden;
pa_operation *o = PULSEAUDIO_pa_stream_drain(h->stream, stream_drain_complete, NULL);
if (o) {
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) {
SDL_FreeAudioMem(this->hidden->mixbuf);
SDL_free(this->hidden->device_name);
if (this->hidden->stream) {
PULSEAUDIO_pa_stream_disconnect(this->hidden->stream);
PULSEAUDIO_pa_stream_unref(this->hidden->stream);
}
DisconnectFromPulseServer(this->hidden->mainloop, this->hidden->context);
SDL_free(this->hidden);
this->hidden = NULL;
}
}
static void
DeviceNameCallback(pa_context *c, const pa_sink_info *i, int is_last, void *data)
{
if (i) {
char **devname = (char **) data;
*devname = SDL_strdup(i->name);
}
}
static SDL_bool
FindDeviceName(struct SDL_PrivateAudioData *h, void *handle)
{
const uint32_t idx = ((uint32_t) ((size_t) handle)) - 1;
if (handle == NULL) { /* NULL == default device. */
return SDL_TRUE;
}
WaitForPulseOperation(h->mainloop, PULSEAUDIO_pa_context_get_sink_info_by_index(h->context, idx, DeviceNameCallback, &h->device_name));
return (h->device_name != NULL);
}
static int
PULSEAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
struct SDL_PrivateAudioData *h = NULL;
Uint16 test_format = 0;
@ -442,42 +521,21 @@ PULSEAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
paattr.minreq = h->mixlen;
#endif
if (ConnectToPulseServer(&h->mainloop, &h->context) < 0) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Could not connect to PulseAudio server");
}
if (!FindDeviceName(h, handle)) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Requested PulseAudio sink missing?");
}
/* 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);
return SDL_SetError("pa_mainloop_new() failed");
}
h->mainloop_api = PULSEAUDIO_pa_mainloop_get_api(h->mainloop);
h->context = PULSEAUDIO_pa_context_new(h->mainloop_api, getAppName());
if (!h->context) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("pa_context_new() failed");
}
/* Connect to the PulseAudio server */
if (PULSEAUDIO_pa_context_connect(h->context, NULL, 0, NULL) < 0) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Could not setup connection to PulseAudio");
}
do {
if (PULSEAUDIO_pa_mainloop_iterate(h->mainloop, 1, NULL) < 0) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("pa_mainloop_iterate() failed");
}
state = PULSEAUDIO_pa_context_get_state(h->context);
if (!PA_CONTEXT_IS_GOOD(state)) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Could not connect to PulseAudio");
}
} while (state != PA_CONTEXT_READY);
h->stream = PULSEAUDIO_pa_stream_new(
h->context,
"Simple DirectMedia Layer", /* stream description */
@ -490,7 +548,13 @@ PULSEAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
return SDL_SetError("Could not set up PulseAudio stream");
}
if (PULSEAUDIO_pa_stream_connect_playback(h->stream, NULL, &paattr, flags,
/* now that we have multi-device support, don't move a stream from
a device that was unplugged to something else, unless we're default. */
if (h->device_name != NULL) {
flags |= PA_STREAM_DONT_MOVE;
}
if (PULSEAUDIO_pa_stream_connect_playback(h->stream, h->device_name, &paattr, flags,
NULL, NULL) < 0) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Could not connect PulseAudio stream");
@ -504,7 +568,7 @@ PULSEAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
state = PULSEAUDIO_pa_stream_get_state(h->stream);
if (!PA_STREAM_IS_GOOD(state)) {
PULSEAUDIO_CloseDevice(this);
return SDL_SetError("Could not create to PulseAudio stream");
return SDL_SetError("Could not connect PulseAudio stream");
}
} while (state != PA_STREAM_READY);
@ -512,10 +576,92 @@ PULSEAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
return 0;
}
static pa_mainloop *hotplug_mainloop = NULL;
static pa_context *hotplug_context = NULL;
static SDL_Thread *hotplug_thread = NULL;
/* device handles are device index + 1, cast to void*, so we never pass a NULL. */
/* This is called when PulseAudio adds an output ("sink") device. */
static void
SinkInfoCallback(pa_context *c, const pa_sink_info *i, int is_last, void *data)
{
if (i) {
SDL_AddAudioDevice(SDL_FALSE, i->description, (void *) ((size_t) i->index+1));
}
}
/* This is called when PulseAudio adds a capture ("source") device. */
static void
SourceInfoCallback(pa_context *c, const pa_source_info *i, int is_last, void *data)
{
if (i) {
/* Skip "monitor" sources. These are just output from other sinks. */
if (i->monitor_of_sink == PA_INVALID_INDEX) {
SDL_AddAudioDevice(SDL_TRUE, i->description, (void *) ((size_t) i->index+1));
}
}
}
/* This is called when PulseAudio has a device connected/removed/changed. */
static void
HotplugCallback(pa_context *c, pa_subscription_event_type_t t, uint32_t idx, void *data)
{
const SDL_bool added = ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_NEW);
const SDL_bool removed = ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) == PA_SUBSCRIPTION_EVENT_REMOVE);
if (added || removed) { /* we only care about add/remove events. */
const SDL_bool sink = ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SINK);
const SDL_bool source = ((t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK) == PA_SUBSCRIPTION_EVENT_SOURCE);
/* adds need sink details from the PulseAudio server. Another callback... */
if (added && sink) {
PULSEAUDIO_pa_context_get_sink_info_by_index(hotplug_context, idx, SinkInfoCallback, NULL);
} else if (added && source) {
PULSEAUDIO_pa_context_get_source_info_by_index(hotplug_context, idx, SourceInfoCallback, NULL);
} else if (removed && (sink || source)) {
/* removes we can handle just with the device index. */
SDL_RemoveAudioDevice(source != 0, (void *) ((size_t) idx+1));
}
}
}
/* this runs as a thread while the Pulse target is initialized to catch hotplug events. */
static int SDLCALL
HotplugThread(void *data)
{
pa_operation *o;
SDL_SetThreadPriority(SDL_THREAD_PRIORITY_LOW);
PULSEAUDIO_pa_context_set_subscribe_callback(hotplug_context, HotplugCallback, NULL);
o = PULSEAUDIO_pa_context_subscribe(hotplug_context, PA_SUBSCRIPTION_MASK_SINK | PA_SUBSCRIPTION_MASK_SOURCE, NULL, NULL);
PULSEAUDIO_pa_operation_unref(o); /* don't wait for it, just do our thing. */
PULSEAUDIO_pa_mainloop_run(hotplug_mainloop, NULL);
return 0;
}
static void
PULSEAUDIO_DetectDevices()
{
WaitForPulseOperation(hotplug_mainloop, PULSEAUDIO_pa_context_get_sink_info_list(hotplug_context, SinkInfoCallback, NULL));
WaitForPulseOperation(hotplug_mainloop, PULSEAUDIO_pa_context_get_source_info_list(hotplug_context, SourceInfoCallback, NULL));
/* ok, we have a sane list, let's set up hotplug notifications now... */
hotplug_thread = SDL_CreateThread(HotplugThread, "PulseHotplug", NULL);
}
static void
PULSEAUDIO_Deinitialize(void)
{
if (hotplug_thread) {
PULSEAUDIO_pa_mainloop_quit(hotplug_mainloop, 0);
SDL_WaitThread(hotplug_thread, NULL);
hotplug_thread = NULL;
}
DisconnectFromPulseServer(hotplug_mainloop, hotplug_context);
hotplug_mainloop = NULL;
hotplug_context = NULL;
UnloadPulseAudioLibrary();
}
@ -526,12 +672,13 @@ PULSEAUDIO_Init(SDL_AudioDriverImpl * impl)
return 0;
}
if (!CheckPulseAudioAvailable()) {
if (ConnectToPulseServer(&hotplug_mainloop, &hotplug_context) < 0) {
UnloadPulseAudioLibrary();
return 0;
}
/* Set the function pointers */
impl->DetectDevices = PULSEAUDIO_DetectDevices;
impl->OpenDevice = PULSEAUDIO_OpenDevice;
impl->PlayDevice = PULSEAUDIO_PlayDevice;
impl->WaitDevice = PULSEAUDIO_WaitDevice;
@ -539,12 +686,10 @@ PULSEAUDIO_Init(SDL_AudioDriverImpl * impl)
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", "PulseAudio", PULSEAUDIO_Init, 0
};

5
Engine/lib/sdl/src/audio/pulseaudio/SDL_pulseaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -32,9 +32,10 @@
struct SDL_PrivateAudioData
{
char *device_name;
/* pulseaudio structures */
pa_mainloop *mainloop;
pa_mainloop_api *mainloop_api;
pa_context *context;
pa_stream *stream;

114
Engine/lib/sdl/src/audio/qsa/SDL_qsa_audio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -19,6 +19,15 @@
3. This notice may not be removed or altered from any source distribution.
*/
/*
* !!! FIXME: streamline this a little by removing all the
* !!! FIXME: if (capture) {} else {} sections that are identical
* !!! FIXME: except for one flag.
*/
/* !!! FIXME: can this target support hotplugging? */
/* !!! FIXME: ...does SDL2 even support QNX? */
#include "../../SDL_internal.h"
#if SDL_AUDIO_DRIVER_QSA
@ -300,7 +309,7 @@ QSA_PlayDevice(_THIS)
/* If we couldn't write, assume fatal error for now */
if (towrite != 0) {
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
}
}
@ -337,8 +346,9 @@ QSA_CloseDevice(_THIS)
}
static int
QSA_OpenDevice(_THIS, const char *devname, int iscapture)
QSA_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
const QSA_Device *device = (const QSA_Device *) handle;
int status = 0;
int format = 0;
SDL_AudioFormat test_format = 0;
@ -363,80 +373,19 @@ QSA_OpenDevice(_THIS, const char *devname, int iscapture)
/* 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);
return SDL_SetError("No such playback device");
}
} 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);
return SDL_SetError("No such capture device");
}
} 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 {
if (device != NULL) {
/* 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);
}
this->hidden->deviceno = device->deviceno;
this->hidden->cardno = device->cardno;
status = snd_pcm_open(&this->hidden->audio_handle,
device->cardno, device->deviceno,
iscapture ? SND_PCM_OPEN_PLAYBACK : SND_PCM_OPEN_CAPTURE);
} else {
/* Open system default audio device */
status = snd_pcm_open_preferred(&this->hidden->audio_handle,
&this->hidden->cardno,
&this->hidden->deviceno,
iscapture ? SND_PCM_OPEN_PLAYBACK : SND_PCM_OPEN_CAPTURE);
}
/* Check if requested device is opened */
@ -638,7 +587,7 @@ QSA_OpenDevice(_THIS, const char *devname, int iscapture)
}
static void
QSA_DetectDevices(int iscapture, SDL_AddAudioDevice addfn)
QSA_DetectDevices(void)
{
uint32_t it;
uint32_t cards;
@ -656,8 +605,9 @@ QSA_DetectDevices(int iscapture, SDL_AddAudioDevice addfn)
return;
}
/* !!! FIXME: code duplication */
/* Find requested devices by type */
if (!iscapture) {
{ /* output devices */
/* Playback devices enumeration requested */
for (it = 0; it < cards; it++) {
devices = 0;
@ -688,7 +638,7 @@ QSA_DetectDevices(int iscapture, SDL_AddAudioDevice addfn)
devices;
status = snd_pcm_close(handle);
if (status == EOK) {
addfn(qsa_playback_device[qsa_playback_devices].name);
SDL_AddAudioDevice(SDL_FALSE, qsa_playback_device[qsa_playback_devices].name, &qsa_playback_device[qsa_playback_devices]);
qsa_playback_devices++;
}
} else {
@ -713,7 +663,9 @@ QSA_DetectDevices(int iscapture, SDL_AddAudioDevice addfn)
break;
}
}
} else {
}
{ /* capture devices */
/* Capture devices enumeration requested */
for (it = 0; it < cards; it++) {
devices = 0;
@ -744,7 +696,7 @@ QSA_DetectDevices(int iscapture, SDL_AddAudioDevice addfn)
devices;
status = snd_pcm_close(handle);
if (status == EOK) {
addfn(qsa_capture_device[qsa_capture_devices].name);
SDL_AddAudioDevice(SDL_TRUE, qsa_capture_device[qsa_capture_devices].name, &qsa_capture_device[qsa_capture_devices]);
qsa_capture_devices++;
}
} else {

2
Engine/lib/sdl/src/audio/qsa/SDL_qsa_audio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

5
Engine/lib/sdl/src/audio/sdlgenaudiocvt.pl
View file

@ -38,7 +38,7 @@ sub outputHeader {
/* DO NOT EDIT! This file is generated by sdlgenaudiocvt.pl */
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken\@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken\@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -383,6 +383,7 @@ sub buildArbitraryResampleFunc {
my $eps_adjust = ($upsample) ? 'dstsize' : 'srcsize';
my $incr = '';
my $incr2 = '';
my $block_align = $channels * $fsize/8;
# !!! FIXME: DEBUG_CONVERT should report frequencies.
@ -395,7 +396,7 @@ ${sym}(SDL_AudioCVT * cvt, SDL_AudioFormat format)
#endif
const int srcsize = cvt->len_cvt - $fudge;
const int dstsize = (int) (((double)cvt->len_cvt) * cvt->rate_incr);
const int dstsize = (int) (((double)(cvt->len_cvt/${block_align})) * cvt->rate_incr) * ${block_align};
register int eps = 0;
EOF

41
Engine/lib/sdl/src/audio/sndio/SDL_sndioaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -158,7 +158,7 @@ SNDIO_PlayDevice(_THIS)
/* If we couldn't write, assume fatal error for now */
if ( written == 0 ) {
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Wrote %d bytes of audio data\n", written);
@ -193,7 +193,7 @@ SNDIO_CloseDevice(_THIS)
}
static int
SNDIO_OpenDevice(_THIS, const char *devname, int iscapture)
SNDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
struct sio_par par;
@ -209,7 +209,7 @@ SNDIO_OpenDevice(_THIS, const char *devname, int iscapture)
this->hidden->mixlen = this->spec.size;
/* !!! FIXME: SIO_DEVANY can be a specific device... */
if ((this->hidden->dev = SNDIO_sio_open(NULL, SIO_PLAY, 0)) == NULL) {
if ((this->hidden->dev = SNDIO_sio_open(SIO_DEVANY, SIO_PLAY, 0)) == NULL) {
SNDIO_CloseDevice(this);
return SDL_SetError("sio_open() failed");
}
@ -229,7 +229,17 @@ SNDIO_OpenDevice(_THIS, const char *devname, int iscapture)
par.sig = SDL_AUDIO_ISSIGNED(test_format) ? 1 : 0;
par.bits = SDL_AUDIO_BITSIZE(test_format);
if (SNDIO_sio_setpar(this->hidden->dev, &par) == 1) {
if (SNDIO_sio_setpar(this->hidden->dev, &par) == 0) {
continue;
}
if (SNDIO_sio_getpar(this->hidden->dev, &par) == 0) {
SNDIO_CloseDevice(this);
return SDL_SetError("sio_getpar() failed");
}
if (par.bps != SIO_BPS(par.bits)) {
continue;
}
if ((par.bits == 8 * par.bps) || (par.msb)) {
status = 0;
break;
}
@ -242,26 +252,21 @@ SNDIO_OpenDevice(_THIS, const char *devname, int iscapture)
return SDL_SetError("sndio: Couldn't find any hardware audio formats");
}
if (SNDIO_sio_getpar(this->hidden->dev, &par) == 0) {
SNDIO_CloseDevice(this);
return SDL_SetError("sio_getpar() failed");
}
if ((par.bits == 32) && (par.sig) && (par.le))
if ((par.bps == 4) && (par.sig) && (par.le))
this->spec.format = AUDIO_S32LSB;
else if ((par.bits == 32) && (par.sig) && (!par.le))
else if ((par.bps == 4) && (par.sig) && (!par.le))
this->spec.format = AUDIO_S32MSB;
else if ((par.bits == 16) && (par.sig) && (par.le))
else if ((par.bps == 2) && (par.sig) && (par.le))
this->spec.format = AUDIO_S16LSB;
else if ((par.bits == 16) && (par.sig) && (!par.le))
else if ((par.bps == 2) && (par.sig) && (!par.le))
this->spec.format = AUDIO_S16MSB;
else if ((par.bits == 16) && (!par.sig) && (par.le))
else if ((par.bps == 2) && (!par.sig) && (par.le))
this->spec.format = AUDIO_U16LSB;
else if ((par.bits == 16) && (!par.sig) && (!par.le))
else if ((par.bps == 2) && (!par.sig) && (!par.le))
this->spec.format = AUDIO_U16MSB;
else if ((par.bits == 8) && (par.sig))
else if ((par.bps == 1) && (par.sig))
this->spec.format = AUDIO_S8;
else if ((par.bits == 8) && (!par.sig))
else if ((par.bps == 1) && (!par.sig))
this->spec.format = AUDIO_U8;
else {
SNDIO_CloseDevice(this);

2
Engine/lib/sdl/src/audio/sndio/SDL_sndioaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

14
Engine/lib/sdl/src/audio/sun/SDL_sunaudio.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -56,9 +56,9 @@ static Uint8 snd2au(int sample);
/* Audio driver bootstrap functions */
static void
SUNAUDIO_DetectDevices(int iscapture, SDL_AddAudioDevice addfn)
SUNAUDIO_DetectDevices(void)
{
SDL_EnumUnixAudioDevices(iscapture, 1, (int (*)(int fd)) NULL, addfn);
SDL_EnumUnixAudioDevices(1, (int (*)(int)) NULL);
}
#ifdef DEBUG_AUDIO
@ -158,7 +158,7 @@ SUNAUDIO_PlayDevice(_THIS)
if (write(this->hidden->audio_fd, this->hidden->ulaw_buf,
this->hidden->fragsize) < 0) {
/* Assume fatal error, for now */
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
}
this->hidden->written += this->hidden->fragsize;
} else {
@ -168,7 +168,7 @@ SUNAUDIO_PlayDevice(_THIS)
if (write(this->hidden->audio_fd, this->hidden->mixbuf,
this->spec.size) < 0) {
/* Assume fatal error, for now */
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
}
this->hidden->written += this->hidden->fragsize;
}
@ -198,7 +198,7 @@ SUNAUDIO_CloseDevice(_THIS)
}
static int
SUNAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
SUNAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
SDL_AudioFormat format = 0;
@ -414,6 +414,8 @@ SUNAUDIO_Init(SDL_AudioDriverImpl * impl)
impl->GetDeviceBuf = SUNAUDIO_GetDeviceBuf;
impl->CloseDevice = SUNAUDIO_CloseDevice;
impl->AllowsArbitraryDeviceNames = 1;
return 1; /* this audio target is available. */
}

2
Engine/lib/sdl/src/audio/sun/SDL_sunaudio.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

65
Engine/lib/sdl/src/audio/winmm/SDL_winmm.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -36,8 +36,9 @@
#define WAVE_FORMAT_IEEE_FLOAT 0x0003
#endif
#define DETECT_DEV_IMPL(typ, capstyp) \
static void DetectWave##typ##Devs(SDL_AddAudioDevice addfn) { \
#define DETECT_DEV_IMPL(iscap, typ, capstyp) \
static void DetectWave##typ##Devs(void) { \
const UINT iscapture = iscap ? 1 : 0; \
const UINT devcount = wave##typ##GetNumDevs(); \
capstyp caps; \
UINT i; \
@ -45,24 +46,21 @@ static void DetectWave##typ##Devs(SDL_AddAudioDevice addfn) { \
if (wave##typ##GetDevCaps(i,&caps,sizeof(caps))==MMSYSERR_NOERROR) { \
char *name = WIN_StringToUTF8(caps.szPname); \
if (name != NULL) { \
addfn(name); \
SDL_AddAudioDevice((int) iscapture, name, (void *) ((size_t) i+1)); \
SDL_free(name); \
} \
} \
} \
}
DETECT_DEV_IMPL(Out, WAVEOUTCAPS)
DETECT_DEV_IMPL(In, WAVEINCAPS)
DETECT_DEV_IMPL(SDL_FALSE, Out, WAVEOUTCAPS)
DETECT_DEV_IMPL(SDL_TRUE, In, WAVEINCAPS)
static void
WINMM_DetectDevices(int iscapture, SDL_AddAudioDevice addfn)
WINMM_DetectDevices(void)
{
if (iscapture) {
DetectWaveInDevs(addfn);
} else {
DetectWaveOutDevs(addfn);
}
DetectWaveInDevs();
DetectWaveOutDevs();
}
static void CALLBACK
@ -220,48 +218,19 @@ PrepWaveFormat(_THIS, UINT devId, WAVEFORMATEX *pfmt, const int iscapture)
}
static int
WINMM_OpenDevice(_THIS, const char *devname, int iscapture)
WINMM_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
int valid_datatype = 0;
MMRESULT result;
WAVEFORMATEX waveformat;
UINT devId = WAVE_MAPPER; /* WAVE_MAPPER == choose system's default */
char *utf8 = NULL;
UINT i;
if (devname != NULL) { /* specific device requested? */
if (iscapture) {
const UINT devcount = waveInGetNumDevs();
WAVEINCAPS caps;
for (i = 0; (i < devcount) && (devId == WAVE_MAPPER); i++) {
result = waveInGetDevCaps(i, &caps, sizeof (caps));
if (result != MMSYSERR_NOERROR)
continue;
else if ((utf8 = WIN_StringToUTF8(caps.szPname)) == NULL)
continue;
else if (SDL_strcmp(devname, utf8) == 0)
devId = i;
SDL_free(utf8);
}
} else {
const UINT devcount = waveOutGetNumDevs();
WAVEOUTCAPS caps;
for (i = 0; (i < devcount) && (devId == WAVE_MAPPER); i++) {
result = waveOutGetDevCaps(i, &caps, sizeof (caps));
if (result != MMSYSERR_NOERROR)
continue;
else if ((utf8 = WIN_StringToUTF8(caps.szPname)) == NULL)
continue;
else if (SDL_strcmp(devname, utf8) == 0)
devId = i;
SDL_free(utf8);
}
}
if (devId == WAVE_MAPPER) {
return SDL_SetError("Requested device not found");
}
if (handle != NULL) { /* specific device requested? */
/* -1 because we increment the original value to avoid NULL. */
const size_t val = ((size_t) handle) - 1;
devId = (UINT) val;
}
/* Initialize all variables that we clean on shutdown */
@ -279,10 +248,6 @@ WINMM_OpenDevice(_THIS, const char *devname, int iscapture)
if (this->spec.channels > 2)
this->spec.channels = 2; /* !!! FIXME: is this right? */
/* 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;
while ((!valid_datatype) && (test_format)) {
switch (test_format) {
case AUDIO_U8:

2
Engine/lib/sdl/src/audio/winmm/SDL_winmm.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

107
Engine/lib/sdl/src/audio/xaudio2/SDL_xaudio2.c
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
@ -58,13 +58,20 @@
/* The configure script already did any necessary checking */
# define SDL_XAUDIO2_HAS_SDK 1
#elif defined(__WINRT__)
/* WinRT always has access to the .the XAudio 2 SDK */
/* WinRT always has access to the XAudio 2 SDK (albeit with a header file
that doesn't compile as C code).
*/
# define SDL_XAUDIO2_HAS_SDK
#include "SDL_xaudio2.h" /* ... compiles as C code, in contrast to XAudio2 headers
in the Windows SDK, v.10.0.10240.0 (Win 10's initial SDK)
*/
#else
/* XAudio2 exists as of the March 2008 DirectX SDK
The XAudio2 implementation available in the Windows 8 SDK targets Windows 8 and newer.
If you want to build SDL with XAudio2 support you should install the DirectX SDK.
/* XAudio2 exists in the last DirectX SDK as well as the latest Windows SDK.
To enable XAudio2 support, you will need to add the location of your DirectX SDK headers to
the SDL projects additional include directories and then set SDL_XAUDIO2_HAS_SDK=1 as a
preprocessor define
*/
#if 0 /* See comment above */
#include <dxsdkver.h>
#if (!defined(_DXSDK_BUILD_MAJOR) || (_DXSDK_BUILD_MAJOR < 1284))
# pragma message("Your DirectX SDK is too old. Disabling XAudio2 support.")
@ -72,6 +79,7 @@
# define SDL_XAUDIO2_HAS_SDK 1
#endif
#endif
#endif /* 0 */
#ifdef SDL_XAUDIO2_HAS_SDK
@ -82,17 +90,10 @@
#endif
#endif
/* The XAudio header file, when #include'd on WinRT, will only compile in C++
files, but not C. A few preprocessor-based hacks are defined below in order
to get xaudio2.h to compile in the C/non-C++ file, SDL_xaudio2.c.
*/
#ifdef __WINRT__
#define uuid(x)
#define DX_BUILD
#endif
#if !defined(_SDL_XAUDIO2_H)
#define INITGUID 1
#include <xaudio2.h>
#endif
/* Hidden "this" pointer for the audio functions */
#define _THIS SDL_AudioDevice *this
@ -126,16 +127,13 @@ struct SDL_PrivateAudioData
static void
XAUDIO2_DetectDevices(int iscapture, SDL_AddAudioDevice addfn)
XAUDIO2_DetectDevices(void)
{
IXAudio2 *ixa2 = NULL;
UINT32 devcount = 0;
UINT32 i = 0;
if (iscapture) {
SDL_SetError("XAudio2: capture devices unsupported.");
return;
} else if (XAudio2Create(&ixa2, 0, XAUDIO2_DEFAULT_PROCESSOR) != S_OK) {
if (XAudio2Create(&ixa2, 0, XAUDIO2_DEFAULT_PROCESSOR) != S_OK) {
SDL_SetError("XAudio2: XAudio2Create() failed at detection.");
return;
} else if (IXAudio2_GetDeviceCount(ixa2, &devcount) != S_OK) {
@ -149,8 +147,8 @@ XAUDIO2_DetectDevices(int iscapture, SDL_AddAudioDevice addfn)
if (IXAudio2_GetDeviceDetails(ixa2, i, &details) == S_OK) {
char *str = WIN_StringToUTF8(details.DisplayName);
if (str != NULL) {
addfn(str);
SDL_free(str); /* addfn() made a copy of the string. */
SDL_AddAudioDevice(SDL_FALSE, str, (void *) ((size_t) i+1));
SDL_free(str); /* SDL_AddAudioDevice made a copy of the string. */
}
}
}
@ -169,8 +167,8 @@ VoiceCBOnBufferEnd(THIS_ void *data)
static void STDMETHODCALLTYPE
VoiceCBOnVoiceError(THIS_ void *data, HRESULT Error)
{
/* !!! FIXME: attempt to recover, or mark device disconnected. */
SDL_assert(0 && "write me!");
SDL_AudioDevice *this = (SDL_AudioDevice *) data;
SDL_OpenedAudioDeviceDisconnected(this);
}
/* no-op callbacks... */
@ -221,7 +219,7 @@ XAUDIO2_PlayDevice(_THIS)
if (result != S_OK) { /* uhoh, panic! */
IXAudio2SourceVoice_FlushSourceBuffers(source);
this->enabled = 0;
SDL_OpenedAudioDeviceDisconnected(this);
}
}
@ -241,14 +239,14 @@ XAUDIO2_WaitDone(_THIS)
SDL_assert(!this->enabled); /* flag that stops playing. */
IXAudio2SourceVoice_Discontinuity(source);
#if SDL_XAUDIO2_WIN8
IXAudio2SourceVoice_GetState(source, &state, 0);
IXAudio2SourceVoice_GetState(source, &state, XAUDIO2_VOICE_NOSAMPLESPLAYED);
#else
IXAudio2SourceVoice_GetState(source, &state);
#endif
while (state.BuffersQueued > 0) {
SDL_SemWait(this->hidden->semaphore);
#if SDL_XAUDIO2_WIN8
IXAudio2SourceVoice_GetState(source, &state, 0);
IXAudio2SourceVoice_GetState(source, &state, XAUDIO2_VOICE_NOSAMPLESPLAYED);
#else
IXAudio2SourceVoice_GetState(source, &state);
#endif
@ -289,7 +287,7 @@ XAUDIO2_CloseDevice(_THIS)
}
static int
XAUDIO2_OpenDevice(_THIS, const char *devname, int iscapture)
XAUDIO2_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
HRESULT result = S_OK;
WAVEFORMATEX waveformat;
@ -315,9 +313,17 @@ XAUDIO2_OpenDevice(_THIS, const char *devname, int iscapture)
static IXAudio2VoiceCallback callbacks = { &callbacks_vtable };
if (iscapture) {
return SDL_SetError("XAudio2: capture devices unsupported.");
} else if (XAudio2Create(&ixa2, 0, XAUDIO2_DEFAULT_PROCESSOR) != S_OK) {
#if defined(SDL_XAUDIO2_WIN8)
/* !!! FIXME: hook up hotplugging. */
#else
if (handle != NULL) { /* specific device requested? */
/* -1 because we increment the original value to avoid NULL. */
const size_t val = ((size_t) handle) - 1;
devId = (UINT32) val;
}
#endif
if (XAudio2Create(&ixa2, 0, XAUDIO2_DEFAULT_PROCESSOR) != S_OK) {
return SDL_SetError("XAudio2: XAudio2Create() failed at open.");
}
@ -332,37 +338,6 @@ XAUDIO2_OpenDevice(_THIS, const char *devname, int iscapture)
ixa2->SetDebugConfiguration(&debugConfig);
*/
#if ! defined(__WINRT__)
if (devname != NULL) {
UINT32 devcount = 0;
UINT32 i = 0;
if (IXAudio2_GetDeviceCount(ixa2, &devcount) != S_OK) {
IXAudio2_Release(ixa2);
return SDL_SetError("XAudio2: IXAudio2_GetDeviceCount() failed.");
}
for (i = 0; i < devcount; i++) {
XAUDIO2_DEVICE_DETAILS details;
if (IXAudio2_GetDeviceDetails(ixa2, i, &details) == S_OK) {
char *str = WIN_StringToUTF8(details.DisplayName);
if (str != NULL) {
const int match = (SDL_strcmp(str, devname) == 0);
SDL_free(str);
if (match) {
devId = i;
break;
}
}
}
}
if (i == devcount) {
IXAudio2_Release(ixa2);
return SDL_SetError("XAudio2: Requested device not found.");
}
}
#endif
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
@ -529,6 +504,16 @@ XAUDIO2_Init(SDL_AudioDriverImpl * impl)
impl->CloseDevice = XAUDIO2_CloseDevice;
impl->Deinitialize = XAUDIO2_Deinitialize;
/* !!! FIXME: We can apparently use a C++ interface on Windows 8
* !!! FIXME: (Windows::Devices::Enumeration::DeviceInformation) for device
* !!! FIXME: detection, but it's not implemented here yet.
* !!! FIXME: see http://blogs.msdn.com/b/chuckw/archive/2012/04/02/xaudio2-and-windows-8-consumer-preview.aspx
* !!! FIXME: for now, force the default device.
*/
#if defined(SDL_XAUDIO2_WIN8) || defined(__WINRT__)
impl->OnlyHasDefaultOutputDevice = 1;
#endif
return 1; /* this audio target is available. */
#endif
}

386
Engine/lib/sdl/src/audio/xaudio2/SDL_xaudio2.h Normal file
View file

@ -0,0 +1,386 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
#ifndef _SDL_XAUDIO2_H
#define _SDL_XAUDIO2_H
#include <windows.h>
#include <mmreg.h>
#include <objbase.h>
/* XAudio2 packs its structure members together as tightly as possible.
This pragma is needed to ensure compatibility with XAudio2 on 64-bit
platforms.
*/
#pragma pack(push, 1)
typedef interface IXAudio2 IXAudio2;
typedef interface IXAudio2SourceVoice IXAudio2SourceVoice;
typedef interface IXAudio2MasteringVoice IXAudio2MasteringVoice;
typedef interface IXAudio2EngineCallback IXAudio2EngineCallback;
typedef interface IXAudio2VoiceCallback IXAudio2VoiceCallback;
typedef interface IXAudio2Voice IXAudio2Voice;
typedef interface IXAudio2SubmixVoice IXAudio2SubmixVoice;
typedef enum _AUDIO_STREAM_CATEGORY {
AudioCategory_Other = 0,
AudioCategory_ForegroundOnlyMedia,
AudioCategory_BackgroundCapableMedia,
AudioCategory_Communications,
AudioCategory_Alerts,
AudioCategory_SoundEffects,
AudioCategory_GameEffects,
AudioCategory_GameMedia,
AudioCategory_GameChat,
AudioCategory_Movie,
AudioCategory_Media
} AUDIO_STREAM_CATEGORY;
typedef struct XAUDIO2_BUFFER {
UINT32 Flags;
UINT32 AudioBytes;
const BYTE *pAudioData;
UINT32 PlayBegin;
UINT32 PlayLength;
UINT32 LoopBegin;
UINT32 LoopLength;
UINT32 LoopCount;
void *pContext;
} XAUDIO2_BUFFER;
typedef struct XAUDIO2_BUFFER_WMA {
const UINT32 *pDecodedPacketCumulativeBytes;
UINT32 PacketCount;
} XAUDIO2_BUFFER_WMA;
typedef struct XAUDIO2_SEND_DESCRIPTOR {
UINT32 Flags;
IXAudio2Voice *pOutputVoice;
} XAUDIO2_SEND_DESCRIPTOR;
typedef struct XAUDIO2_VOICE_SENDS {
UINT32 SendCount;
XAUDIO2_SEND_DESCRIPTOR *pSends;
} XAUDIO2_VOICE_SENDS;
typedef struct XAUDIO2_EFFECT_DESCRIPTOR {
IUnknown *pEffect;
BOOL InitialState;
UINT32 OutputChannels;
} XAUDIO2_EFFECT_DESCRIPTOR;
typedef struct XAUDIO2_EFFECT_CHAIN {
UINT32 EffectCount;
XAUDIO2_EFFECT_DESCRIPTOR *pEffectDescriptors;
} XAUDIO2_EFFECT_CHAIN;
typedef struct XAUDIO2_PERFORMANCE_DATA {
UINT64 AudioCyclesSinceLastQuery;
UINT64 TotalCyclesSinceLastQuery;
UINT32 MinimumCyclesPerQuantum;
UINT32 MaximumCyclesPerQuantum;
UINT32 MemoryUsageInBytes;
UINT32 CurrentLatencyInSamples;
UINT32 GlitchesSinceEngineStarted;
UINT32 ActiveSourceVoiceCount;
UINT32 TotalSourceVoiceCount;
UINT32 ActiveSubmixVoiceCount;
UINT32 ActiveResamplerCount;
UINT32 ActiveMatrixMixCount;
UINT32 ActiveXmaSourceVoices;
UINT32 ActiveXmaStreams;
} XAUDIO2_PERFORMANCE_DATA;
typedef struct XAUDIO2_DEBUG_CONFIGURATION {
UINT32 TraceMask;
UINT32 BreakMask;
BOOL LogThreadID;
BOOL LogFileline;
BOOL LogFunctionName;
BOOL LogTiming;
} XAUDIO2_DEBUG_CONFIGURATION;
typedef struct XAUDIO2_VOICE_DETAILS {
UINT32 CreationFlags;
UINT32 ActiveFlags;
UINT32 InputChannels;
UINT32 InputSampleRate;
} XAUDIO2_VOICE_DETAILS;
typedef enum XAUDIO2_FILTER_TYPE {
LowPassFilter = 0,
BandPassFilter = 1,
HighPassFilter = 2,
NotchFilter = 3,
LowPassOnePoleFilter = 4,
HighPassOnePoleFilter = 5
} XAUDIO2_FILTER_TYPE;
typedef struct XAUDIO2_FILTER_PARAMETERS {
XAUDIO2_FILTER_TYPE Type;
float Frequency;
float OneOverQ;
} XAUDIO2_FILTER_PARAMETERS;
typedef struct XAUDIO2_VOICE_STATE {
void *pCurrentBufferContext;
UINT32 BuffersQueued;
UINT64 SamplesPlayed;
} XAUDIO2_VOICE_STATE;
typedef UINT32 XAUDIO2_PROCESSOR;
#define Processor1 0x00000001
#define XAUDIO2_DEFAULT_PROCESSOR Processor1
#define XAUDIO2_E_DEVICE_INVALIDATED 0x88960004
#define XAUDIO2_COMMIT_NOW 0
#define XAUDIO2_VOICE_NOSAMPLESPLAYED 0x0100
#define XAUDIO2_DEFAULT_CHANNELS 0
extern HRESULT __stdcall XAudio2Create(
_Out_ IXAudio2 **ppXAudio2,
_In_ UINT32 Flags,
_In_ XAUDIO2_PROCESSOR XAudio2Processor
);
#undef INTERFACE
#define INTERFACE IXAudio2
typedef interface IXAudio2 {
const struct IXAudio2Vtbl FAR* lpVtbl;
} IXAudio2;
typedef const struct IXAudio2Vtbl IXAudio2Vtbl;
const struct IXAudio2Vtbl
{
/* IUnknown */
STDMETHOD(QueryInterface)(THIS_ REFIID iid, LPVOID *ppv) PURE;
STDMETHOD_(ULONG, AddRef)(THIS) PURE;
STDMETHOD_(ULONG, Release)(THIS) PURE;
/* IXAudio2 */
STDMETHOD_(HRESULT, RegisterForCallbacks)(THIS, IXAudio2EngineCallback *pCallback) PURE;
STDMETHOD_(VOID, UnregisterForCallbacks)(THIS, IXAudio2EngineCallback *pCallback) PURE;
STDMETHOD_(HRESULT, CreateSourceVoice)(THIS, IXAudio2SourceVoice **ppSourceVoice,
const WAVEFORMATEX *pSourceFormat,
UINT32 Flags,
float MaxFrequencyRatio,
IXAudio2VoiceCallback *pCallback,
const XAUDIO2_VOICE_SENDS *pSendList,
const XAUDIO2_EFFECT_CHAIN *pEffectChain) PURE;
STDMETHOD_(HRESULT, CreateSubmixVoice)(THIS, IXAudio2SubmixVoice **ppSubmixVoice,
UINT32 InputChannels,
UINT32 InputSampleRate,
UINT32 Flags,
UINT32 ProcessingStage,
const XAUDIO2_VOICE_SENDS *pSendList,
const XAUDIO2_EFFECT_CHAIN *pEffectChain) PURE;
STDMETHOD_(HRESULT, CreateMasteringVoice)(THIS, IXAudio2MasteringVoice **ppMasteringVoice,
UINT32 InputChannels,
UINT32 InputSampleRate,
UINT32 Flags,
LPCWSTR szDeviceId,
const XAUDIO2_EFFECT_CHAIN *pEffectChain,
AUDIO_STREAM_CATEGORY StreamCategory) PURE;
STDMETHOD_(HRESULT, StartEngine)(THIS) PURE;
STDMETHOD_(VOID, StopEngine)(THIS) PURE;
STDMETHOD_(HRESULT, CommitChanges)(THIS, UINT32 OperationSet) PURE;
STDMETHOD_(HRESULT, GetPerformanceData)(THIS, XAUDIO2_PERFORMANCE_DATA *pPerfData) PURE;
STDMETHOD_(HRESULT, SetDebugConfiguration)(THIS, XAUDIO2_DEBUG_CONFIGURATION *pDebugConfiguration,
VOID *pReserved) PURE;
};
#define IXAudio2_Release(A) ((A)->lpVtbl->Release(A))
#define IXAudio2_CreateSourceVoice(A,B,C,D,E,F,G,H) ((A)->lpVtbl->CreateSourceVoice(A,B,C,D,E,F,G,H))
#define IXAudio2_CreateMasteringVoice(A,B,C,D,E,F,G,H) ((A)->lpVtbl->CreateMasteringVoice(A,B,C,D,E,F,G,H))
#define IXAudio2_StartEngine(A) ((A)->lpVtbl->StartEngine(A))
#define IXAudio2_StopEngine(A) ((A)->lpVtbl->StopEngine(A))
#undef INTERFACE
#define INTERFACE IXAudio2SourceVoice
typedef interface IXAudio2SourceVoice {
const struct IXAudio2SourceVoiceVtbl FAR* lpVtbl;
} IXAudio2SourceVoice;
typedef const struct IXAudio2SourceVoiceVtbl IXAudio2SourceVoiceVtbl;
const struct IXAudio2SourceVoiceVtbl
{
/* MSDN says that IXAudio2Voice inherits from IXAudio2, but MSVC's debugger
* says otherwise, and that IXAudio2Voice doesn't inherit from any other
* interface!
*/
/* IXAudio2Voice */
STDMETHOD_(VOID, GetVoiceDetails)(THIS, XAUDIO2_VOICE_DETAILS *pVoiceDetails) PURE;
STDMETHOD_(HRESULT, SetOutputVoices)(THIS, const XAUDIO2_VOICE_SENDS *pSendList) PURE;
STDMETHOD_(HRESULT, SetEffectChain)(THIS, const XAUDIO2_EFFECT_CHAIN *pEffectChain) PURE;
STDMETHOD_(HRESULT, EnableEffect)(THIS, UINT32 EffectIndex, UINT32 OperationSet) PURE;
STDMETHOD_(HRESULT, DisableEffect)(THIS, UINT32 EffectIndex, UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetEffectState)(THIS, UINT32 EffectIndex, BOOL *pEnabled) PURE;
STDMETHOD_(HRESULT, SetEffectParameters)(THIS, UINT32 EffectIndex,
const void *pParameters,
UINT32 ParametersByteSize,
UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetEffectParameters)(THIS, UINT32 EffectIndex,
void *pParameters,
UINT32 ParametersByteSize) PURE;
STDMETHOD_(HRESULT, SetFilterParameters)(THIS, const XAUDIO2_FILTER_PARAMETERS *pParameters,
UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetFilterParameters)(THIS, XAUDIO2_FILTER_PARAMETERS *pParameters) PURE;
STDMETHOD_(HRESULT, SetOutputFilterParameters)(THIS, IXAudio2Voice *pDestinationVoice,
XAUDIO2_FILTER_PARAMETERS *pParameters,
UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetOutputFilterParameters)(THIS, IXAudio2Voice *pDestinationVoice,
XAUDIO2_FILTER_PARAMETERS *pParameters) PURE;
STDMETHOD_(HRESULT, SetVolume)(THIS, float Volume,
UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetVolume)(THIS, float *pVolume) PURE;
STDMETHOD_(HRESULT, SetChannelVolumes)(THIS, UINT32 Channels,
const float *pVolumes,
UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetChannelVolumes)(THIS, UINT32 Channels,
float *pVolumes) PURE;
STDMETHOD_(HRESULT, SetOutputMatrix)(THIS, IXAudio2Voice *pDestinationVoice,
UINT32 SourceChannels,
UINT32 DestinationChannels,
const float *pLevelMatrix,
UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetOutputMatrix)(THIS, IXAudio2Voice *pDestinationVoice,
UINT32 SourceChannels,
UINT32 DestinationChannels,
float *pLevelMatrix) PURE;
STDMETHOD_(VOID, DestroyVoice)(THIS) PURE;
/* IXAudio2SourceVoice */
STDMETHOD_(HRESULT, Start)(THIS, UINT32 Flags,
UINT32 OperationSet) PURE;
STDMETHOD_(HRESULT, Stop)(THIS, UINT32 Flags,
UINT32 OperationSet) PURE;
STDMETHOD_(HRESULT, SubmitSourceBuffer)(THIS, const XAUDIO2_BUFFER *pBuffer,
const XAUDIO2_BUFFER_WMA *pBufferWMA) PURE;
STDMETHOD_(HRESULT, FlushSourceBuffers)(THIS) PURE;
STDMETHOD_(HRESULT, Discontinuity)(THIS) PURE;
STDMETHOD_(HRESULT, ExitLoop)(THIS, UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetState)(THIS, XAUDIO2_VOICE_STATE *pVoiceState,
UINT32 Flags) PURE;
STDMETHOD_(HRESULT, SetFrequencyRatio)(THIS, float Ratio,
UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetFrequencyRatio)(THIS, float *pRatio) PURE;
STDMETHOD_(HRESULT, SetSourceSampleRate)(THIS, UINT32 NewSourceSampleRate) PURE;
};
#define IXAudio2SourceVoice_DestroyVoice(A) ((A)->lpVtbl->DestroyVoice(A))
#define IXAudio2SourceVoice_Start(A,B,C) ((A)->lpVtbl->Start(A,B,C))
#define IXAudio2SourceVoice_Stop(A,B,C) ((A)->lpVtbl->Stop(A,B,C))
#define IXAudio2SourceVoice_SubmitSourceBuffer(A,B,C) ((A)->lpVtbl->SubmitSourceBuffer(A,B,C))
#define IXAudio2SourceVoice_FlushSourceBuffers(A) ((A)->lpVtbl->FlushSourceBuffers(A))
#define IXAudio2SourceVoice_Discontinuity(A) ((A)->lpVtbl->Discontinuity(A))
#define IXAudio2SourceVoice_GetState(A,B,C) ((A)->lpVtbl->GetState(A,B,C))
#undef INTERFACE
#define INTERFACE IXAudio2MasteringVoice
typedef interface IXAudio2MasteringVoice {
const struct IXAudio2MasteringVoiceVtbl FAR* lpVtbl;
} IXAudio2MasteringVoice;
typedef const struct IXAudio2MasteringVoiceVtbl IXAudio2MasteringVoiceVtbl;
const struct IXAudio2MasteringVoiceVtbl
{
/* MSDN says that IXAudio2Voice inherits from IXAudio2, but MSVC's debugger
* says otherwise, and that IXAudio2Voice doesn't inherit from any other
* interface!
*/
/* IXAudio2Voice */
STDMETHOD_(VOID, GetVoiceDetails)(THIS, XAUDIO2_VOICE_DETAILS *pVoiceDetails) PURE;
STDMETHOD_(HRESULT, SetOutputVoices)(THIS, const XAUDIO2_VOICE_SENDS *pSendList) PURE;
STDMETHOD_(HRESULT, SetEffectChain)(THIS, const XAUDIO2_EFFECT_CHAIN *pEffectChain) PURE;
STDMETHOD_(HRESULT, EnableEffect)(THIS, UINT32 EffectIndex, UINT32 OperationSet) PURE;
STDMETHOD_(HRESULT, DisableEffect)(THIS, UINT32 EffectIndex, UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetEffectState)(THIS, UINT32 EffectIndex, BOOL *pEnabled) PURE;
STDMETHOD_(HRESULT, SetEffectParameters)(THIS, UINT32 EffectIndex,
const void *pParameters,
UINT32 ParametersByteSize,
UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetEffectParameters)(THIS, UINT32 EffectIndex,
void *pParameters,
UINT32 ParametersByteSize) PURE;
STDMETHOD_(HRESULT, SetFilterParameters)(THIS, const XAUDIO2_FILTER_PARAMETERS *pParameters,
UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetFilterParameters)(THIS, XAUDIO2_FILTER_PARAMETERS *pParameters) PURE;
STDMETHOD_(HRESULT, SetOutputFilterParameters)(THIS, IXAudio2Voice *pDestinationVoice,
XAUDIO2_FILTER_PARAMETERS *pParameters,
UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetOutputFilterParameters)(THIS, IXAudio2Voice *pDestinationVoice,
XAUDIO2_FILTER_PARAMETERS *pParameters) PURE;
STDMETHOD_(HRESULT, SetVolume)(THIS, float Volume,
UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetVolume)(THIS, float *pVolume) PURE;
STDMETHOD_(HRESULT, SetChannelVolumes)(THIS, UINT32 Channels,
const float *pVolumes,
UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetChannelVolumes)(THIS, UINT32 Channels,
float *pVolumes) PURE;
STDMETHOD_(HRESULT, SetOutputMatrix)(THIS, IXAudio2Voice *pDestinationVoice,
UINT32 SourceChannels,
UINT32 DestinationChannels,
const float *pLevelMatrix,
UINT32 OperationSet) PURE;
STDMETHOD_(VOID, GetOutputMatrix)(THIS, IXAudio2Voice *pDestinationVoice,
UINT32 SourceChannels,
UINT32 DestinationChannels,
float *pLevelMatrix) PURE;
STDMETHOD_(VOID, DestroyVoice)(THIS) PURE;
/* IXAudio2SourceVoice */
STDMETHOD_(VOID, GetChannelMask)(THIS, DWORD *pChannelMask) PURE;
};
#define IXAudio2MasteringVoice_DestroyVoice(A) ((A)->lpVtbl->DestroyVoice(A))
#undef INTERFACE
#define INTERFACE IXAudio2VoiceCallback
typedef interface IXAudio2VoiceCallback {
const struct IXAudio2VoiceCallbackVtbl FAR* lpVtbl;
} IXAudio2VoiceCallback;
typedef const struct IXAudio2VoiceCallbackVtbl IXAudio2VoiceCallbackVtbl;
const struct IXAudio2VoiceCallbackVtbl
{
/* MSDN says that IXAudio2VoiceCallback inherits from IXAudio2, but SDL's
* own code says otherwise, and that IXAudio2VoiceCallback doesn't inherit
* from any other interface!
*/
/* IXAudio2VoiceCallback */
STDMETHOD_(VOID, OnVoiceProcessingPassStart)(THIS, UINT32 BytesRequired) PURE;
STDMETHOD_(VOID, OnVoiceProcessingPassEnd)(THIS) PURE;
STDMETHOD_(VOID, OnStreamEnd)(THIS) PURE;
STDMETHOD_(VOID, OnBufferStart)(THIS, void *pBufferContext) PURE;
STDMETHOD_(VOID, OnBufferEnd)(THIS, void *pBufferContext) PURE;
STDMETHOD_(VOID, OnLoopEnd)(THIS, void *pBufferContext) PURE;
STDMETHOD_(VOID, OnVoiceError)(THIS, void *pBufferContext, HRESULT Error) PURE;
};
#pragma pack(pop) /* Undo pragma push */
#endif /* _SDL_XAUDIO2_H */
/* vi: set ts=4 sw=4 expandtab: */

2
Engine/lib/sdl/src/audio/xaudio2/SDL_xaudio2_winrthelpers.cpp
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages

2
Engine/lib/sdl/src/audio/xaudio2/SDL_xaudio2_winrthelpers.h
View file

@ -1,6 +1,6 @@
/*
Simple DirectMedia Layer
Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org>
Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages