sdl: remove non-rockbox drivers

We never use any of these other drivers, so having them around just takes
up space.

Change-Id: Iced812162df1fef3fd55522b7e700acb6c3bcd41

530 files changed, 0 insertions, 138647 deletions

diff --git a/apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.c b/apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.c
deleted file mode 100644
index f10733e432..0000000000
--- a/apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.c
+++ /dev/null
@@ -1,619 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/* Allow access to a raw mixing buffer */
-
-#include <sys/types.h>
-#include <signal.h>	/* For kill() */
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "SDL_alsa_audio.h"
-
-#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
-#include "SDL_name.h"
-#include "SDL_loadso.h"
-#else
-#define SDL_NAME(X)	X
-#endif
-
-
-/* The tag name used by ALSA audio */
-#define DRIVER_NAME         "alsa"
-
-/* Audio driver functions */
-static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void ALSA_WaitAudio(_THIS);
-static void ALSA_PlayAudio(_THIS);
-static Uint8 *ALSA_GetAudioBuf(_THIS);
-static void ALSA_CloseAudio(_THIS);
-
-#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
-
-static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC;
-static void *alsa_handle = NULL;
-static int alsa_loaded = 0;
-
-static int (*SDL_NAME(snd_pcm_open))(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode);
-static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm);
-static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_writei))(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size);
-static int (*SDL_NAME(snd_pcm_recover))(snd_pcm_t *pcm, int err, int silent);
-static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm);
-static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm);
-static const char *(*SDL_NAME(snd_strerror))(int errnum);
-static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void);
-static size_t (*SDL_NAME(snd_pcm_sw_params_sizeof))(void);
-static void (*SDL_NAME(snd_pcm_hw_params_copy))(snd_pcm_hw_params_t *dst, const snd_pcm_hw_params_t *src);
-static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
-static int (*SDL_NAME(snd_pcm_hw_params_set_access))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t access);
-static int (*SDL_NAME(snd_pcm_hw_params_set_format))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val);
-static int (*SDL_NAME(snd_pcm_hw_params_set_channels))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val);
-static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params, unsigned int *val);
-static int (*SDL_NAME(snd_pcm_hw_params_set_rate_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
-static int (*SDL_NAME(snd_pcm_hw_params_set_period_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val, int *dir);
-static int (*SDL_NAME(snd_pcm_hw_params_get_period_size))(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *frames, int *dir);
-static int (*SDL_NAME(snd_pcm_hw_params_set_periods_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
-static int (*SDL_NAME(snd_pcm_hw_params_get_periods))(const snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
-static int (*SDL_NAME(snd_pcm_hw_params_set_buffer_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val);
-static int (*SDL_NAME(snd_pcm_hw_params_get_buffer_size))(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val);
-static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
-/*
-*/
-static int (*SDL_NAME(snd_pcm_sw_params_set_avail_min))(snd_pcm_t *pcm, snd_pcm_sw_params_t *swparams, snd_pcm_uframes_t val);
-static int (*SDL_NAME(snd_pcm_sw_params_current))(snd_pcm_t *pcm, snd_pcm_sw_params_t *swparams);
-static int (*SDL_NAME(snd_pcm_sw_params_set_start_threshold))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);
-static int (*SDL_NAME(snd_pcm_sw_params))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params);
-static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock);
-static int (*SDL_NAME(snd_pcm_wait))(snd_pcm_t *pcm, int timeout);
-#define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof)
-#define snd_pcm_sw_params_sizeof SDL_NAME(snd_pcm_sw_params_sizeof)
-
-/* cast funcs to char* first, to please GCC's strict aliasing rules. */
-static struct {
-	const char *name;
-	void **func;
-} alsa_functions[] = {
-	{ "snd_pcm_open",	(void**)(char*)&SDL_NAME(snd_pcm_open)		},
-	{ "snd_pcm_close",	(void**)(char*)&SDL_NAME(snd_pcm_close)	},
-	{ "snd_pcm_writei",	(void**)(char*)&SDL_NAME(snd_pcm_writei)	},
-	{ "snd_pcm_recover",	(void**)(char*)&SDL_NAME(snd_pcm_recover)	},
-	{ "snd_pcm_prepare",	(void**)(char*)&SDL_NAME(snd_pcm_prepare)	},
-	{ "snd_pcm_drain",	(void**)(char*)&SDL_NAME(snd_pcm_drain)	},
-	{ "snd_strerror",	(void**)(char*)&SDL_NAME(snd_strerror)		},
-	{ "snd_pcm_hw_params_sizeof",		(void**)(char*)&SDL_NAME(snd_pcm_hw_params_sizeof)		},
-	{ "snd_pcm_sw_params_sizeof",		(void**)(char*)&SDL_NAME(snd_pcm_sw_params_sizeof)		},
-	{ "snd_pcm_hw_params_copy",		(void**)(char*)&SDL_NAME(snd_pcm_hw_params_copy)		},
-	{ "snd_pcm_hw_params_any",		(void**)(char*)&SDL_NAME(snd_pcm_hw_params_any)		},
-	{ "snd_pcm_hw_params_set_access",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_access)		},
-	{ "snd_pcm_hw_params_set_format",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_format)		},
-	{ "snd_pcm_hw_params_set_channels",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_channels)	},
-	{ "snd_pcm_hw_params_get_channels",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_channels)	},
-	{ "snd_pcm_hw_params_set_rate_near",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_rate_near)	},
-	{ "snd_pcm_hw_params_set_period_size_near",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_period_size_near)	},
-	{ "snd_pcm_hw_params_get_period_size",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_period_size)	},
-	{ "snd_pcm_hw_params_set_periods_near",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_periods_near)	},
-	{ "snd_pcm_hw_params_get_periods",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_periods)	},
-	{ "snd_pcm_hw_params_set_buffer_size_near",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_buffer_size_near) },
-	{ "snd_pcm_hw_params_get_buffer_size",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_buffer_size) },
-	{ "snd_pcm_hw_params",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params)	},
-	{ "snd_pcm_sw_params_set_avail_min",	(void**)(char*)&SDL_NAME(snd_pcm_sw_params_set_avail_min) },
-	{ "snd_pcm_sw_params_current",	(void**)(char*)&SDL_NAME(snd_pcm_sw_params_current)	},
-	{ "snd_pcm_sw_params_set_start_threshold",	(void**)(char*)&SDL_NAME(snd_pcm_sw_params_set_start_threshold)	},
-	{ "snd_pcm_sw_params",	(void**)(char*)&SDL_NAME(snd_pcm_sw_params)	},
-	{ "snd_pcm_nonblock",	(void**)(char*)&SDL_NAME(snd_pcm_nonblock)	},
-	{ "snd_pcm_wait",	(void**)(char*)&SDL_NAME(snd_pcm_wait)	},
-};
-
-static void UnloadALSALibrary(void) {
-	if (alsa_loaded) {
-		SDL_UnloadObject(alsa_handle);
-		alsa_handle = NULL;
-		alsa_loaded = 0;
-	}
-}
-
-static int LoadALSALibrary(void) {
-	int i, retval = -1;
-
-	alsa_handle = SDL_LoadObject(alsa_library);
-	if (alsa_handle) {
-		alsa_loaded = 1;
-		retval = 0;
-		for (i = 0; i < SDL_arraysize(alsa_functions); i++) {
-			*alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);
-			if (!*alsa_functions[i].func) {
-				retval = -1;
-				UnloadALSALibrary();
-				break;
-			}
-		}
-	}
-	return retval;
-}
-
-#else
-
-static void UnloadALSALibrary(void) {
-	return;
-}
-
-static int LoadALSALibrary(void) {
-	return 0;
-}
-
-#endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */
-
-static const char *get_audio_device(int channels)
-{
-	const char *device;
-	
-	device = SDL_getenv("AUDIODEV");	/* Is there a standard variable name? */
-	if ( device == NULL ) {
-		switch (channels) {
-		case 6:
-			device = "plug:surround51";
-			break;
-		case 4:
-			device = "plug:surround40";
-			break;
-		default:
-			device = "default";
-			break;
-		}
-	}
-	return device;
-}
-
-/* Audio driver bootstrap functions */
-
-static int Audio_Available(void)
-{
-	int available;
-	int status;
-	snd_pcm_t *handle;
-
-	available = 0;
-	if (LoadALSALibrary() < 0) {
-		return available;
-	}
-	status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(2), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
-	if ( status >= 0 ) {
-		available = 1;
-        	SDL_NAME(snd_pcm_close)(handle);
-	}
-	UnloadALSALibrary();
-	return(available);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-	SDL_free(device->hidden);
-	SDL_free(device);
-	UnloadALSALibrary();
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	LoadALSALibrary();
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-	if ( this ) {
-		SDL_memset(this, 0, (sizeof *this));
-		this->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *this->hidden));
-	}
-	if ( (this == NULL) || (this->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( this ) {
-			SDL_free(this);
-		}
-		return(0);
-	}
-	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-	/* Set the function pointers */
-	this->OpenAudio = ALSA_OpenAudio;
-	this->WaitAudio = ALSA_WaitAudio;
-	this->PlayAudio = ALSA_PlayAudio;
-	this->GetAudioBuf = ALSA_GetAudioBuf;
-	this->CloseAudio = ALSA_CloseAudio;
-
-	this->free = Audio_DeleteDevice;
-
-	return this;
-}
-
-AudioBootStrap ALSA_bootstrap = {
-	DRIVER_NAME, "ALSA PCM audio",
-	Audio_Available, Audio_CreateDevice
-};
-
-/* This function waits until it is possible to write a full sound buffer */
-static void ALSA_WaitAudio(_THIS)
-{
-	/* We're in blocking mode, so there's nothing to do here */
-}
-
-
-/*
- * http://bugzilla.libsdl.org/show_bug.cgi?id=110
- * "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
- *  and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
- */
-#define SWIZ6(T) \
-    T *ptr = (T *) mixbuf; \
-    Uint32 i; \
-    for (i = 0; i < this->spec.samples; i++, ptr += 6) { \
-        T tmp; \
-        tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \
-        tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \
-    }
-
-static __inline__ void swizzle_alsa_channels_6_64bit(_THIS) { SWIZ6(Uint64); }
-static __inline__ void swizzle_alsa_channels_6_32bit(_THIS) { SWIZ6(Uint32); }
-static __inline__ void swizzle_alsa_channels_6_16bit(_THIS) { SWIZ6(Uint16); }
-static __inline__ void swizzle_alsa_channels_6_8bit(_THIS) { SWIZ6(Uint8); }
-
-#undef SWIZ6
-
-
-/*
- * Called right before feeding this->mixbuf to the hardware. Swizzle channels
- *  from Windows/Mac order to the format alsalib will want.
- */
-static __inline__ void swizzle_alsa_channels(_THIS)
-{
-    if (this->spec.channels == 6) {
-        const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */
-        if (fmtsize == 16)
-            swizzle_alsa_channels_6_16bit(this);
-        else if (fmtsize == 8)
-            swizzle_alsa_channels_6_8bit(this);
-        else if (fmtsize == 32)
-            swizzle_alsa_channels_6_32bit(this);
-        else if (fmtsize == 64)
-            swizzle_alsa_channels_6_64bit(this);
-    }
-
-    /* !!! FIXME: update this for 7.1 if needed, later. */
-}
-
-
-static void ALSA_PlayAudio(_THIS)
-{
-	int status;
-	snd_pcm_uframes_t frames_left;
-	const Uint8 *sample_buf = (const Uint8 *) mixbuf;
-	const int frame_size = (((int) (this->spec.format & 0xFF)) / 8) * this->spec.channels;
-
-	swizzle_alsa_channels(this);
-
-	frames_left = ((snd_pcm_uframes_t) this->spec.samples);
-
-	while ( frames_left > 0 && this->enabled ) {
-		/* This works, but needs more testing before going live */
-		/*SDL_NAME(snd_pcm_wait)(pcm_handle, -1);*/
-
-		status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, frames_left);
-		if ( status < 0 ) {
-			if ( status == -EAGAIN ) {
-				/* Apparently snd_pcm_recover() doesn't handle this case - does it assume snd_pcm_wait() above? */
-				SDL_Delay(1);
-				continue;
-			}
-			status = SDL_NAME(snd_pcm_recover)(pcm_handle, status, 0);
-			if ( status < 0 ) {
-				/* Hmm, not much we can do - abort */
-				fprintf(stderr, "ALSA write failed (unrecoverable): %s\n", SDL_NAME(snd_strerror)(status));
-				this->enabled = 0;
-				return;
-			}
-			continue;
-		}
-		sample_buf += status * frame_size;
-		frames_left -= status;
-	}
-}
-
-static Uint8 *ALSA_GetAudioBuf(_THIS)
-{
-	return(mixbuf);
-}
-
-static void ALSA_CloseAudio(_THIS)
-{
-	if ( mixbuf != NULL ) {
-		SDL_FreeAudioMem(mixbuf);
-		mixbuf = NULL;
-	}
-	if ( pcm_handle ) {
-		SDL_NAME(snd_pcm_drain)(pcm_handle);
-		SDL_NAME(snd_pcm_close)(pcm_handle);
-		pcm_handle = NULL;
-	}
-}
-
-static int ALSA_finalize_hardware(_THIS, SDL_AudioSpec *spec, snd_pcm_hw_params_t *hwparams, int override)
-{
-	int status;
-	snd_pcm_uframes_t bufsize;
-
-	/* "set" the hardware with the desired parameters */
-	status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, hwparams);
-	if ( status < 0 ) {
-		return(-1);
-	}
-
-	/* Get samples for the actual buffer size */
-	status = SDL_NAME(snd_pcm_hw_params_get_buffer_size)(hwparams, &bufsize);
-	if ( status < 0 ) {
-		return(-1);
-	}
-	if ( !override && bufsize != spec->samples * 2 ) {
-		return(-1);
-	}
-
-	/* FIXME: Is this safe to do? */
-	spec->samples = bufsize / 2;
-
-	/* This is useful for debugging */
-	if ( getenv("SDL_AUDIO_ALSA_DEBUG") ) {
-		snd_pcm_uframes_t persize = 0;
-		unsigned int periods = 0;
-
-		SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams, &persize, NULL);
-		SDL_NAME(snd_pcm_hw_params_get_periods)(hwparams, &periods, NULL);
-
-		fprintf(stderr, "ALSA: period size = %ld, periods = %u, buffer size = %lu\n", persize, periods, bufsize);
-	}
-	return(0);
-}
-
-static int ALSA_set_period_size(_THIS, SDL_AudioSpec *spec, snd_pcm_hw_params_t *params, int override)
-{
-	const char *env;
-	int status;
-	snd_pcm_hw_params_t *hwparams;
-	snd_pcm_uframes_t frames;
-	unsigned int periods;
-
-	/* Copy the hardware parameters for this setup */
-	snd_pcm_hw_params_alloca(&hwparams);
-	SDL_NAME(snd_pcm_hw_params_copy)(hwparams, params);
-
-	if ( !override ) {
-		env = getenv("SDL_AUDIO_ALSA_SET_PERIOD_SIZE");
-		if ( env ) {
-			override = SDL_atoi(env);
-			if ( override == 0 ) {
-				return(-1);
-			}
-		}
-	}
-
-	frames = spec->samples;
-	status = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, &frames, NULL);
-	if ( status < 0 ) {
-		return(-1);
-	}
-
-	periods = 2;
-	status = SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, &periods, NULL);
-	if ( status < 0 ) {
-		return(-1);
-	}
-
-	return ALSA_finalize_hardware(this, spec, hwparams, override);
-}
-
-static int ALSA_set_buffer_size(_THIS, SDL_AudioSpec *spec, snd_pcm_hw_params_t *params, int override)
-{
-	const char *env;
-	int status;
-	snd_pcm_hw_params_t *hwparams;
-	snd_pcm_uframes_t frames;
-
-	/* Copy the hardware parameters for this setup */
-	snd_pcm_hw_params_alloca(&hwparams);
-	SDL_NAME(snd_pcm_hw_params_copy)(hwparams, params);
-
-	if ( !override ) {
-		env = getenv("SDL_AUDIO_ALSA_SET_BUFFER_SIZE");
-		if ( env ) {
-			override = SDL_atoi(env);
-			if ( override == 0 ) {
-				return(-1);
-			}
-		}
-	}
-
-	frames = spec->samples * 2;
-	status = SDL_NAME(snd_pcm_hw_params_set_buffer_size_near)(pcm_handle, hwparams, &frames);
-	if ( status < 0 ) {
-		return(-1);
-	}
-
-	return ALSA_finalize_hardware(this, spec, hwparams, override);
-}
-
-static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	int                  status;
-	snd_pcm_hw_params_t *hwparams;
-	snd_pcm_sw_params_t *swparams;
-	snd_pcm_format_t     format;
-	unsigned int         rate;
-	unsigned int 	     channels;
-	Uint16               test_format;
-
-	/* Open the audio device */
-	/* Name of device should depend on # channels in spec */
-	status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
-
-	if ( status < 0 ) {
-		SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status));
-		return(-1);
-	}
-
-	/* Figure out what the hardware is capable of */
-	snd_pcm_hw_params_alloca(&hwparams);
-	status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, hwparams);
-	if ( status < 0 ) {
-		SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status));
-		ALSA_CloseAudio(this);
-		return(-1);
-	}
-
-	/* SDL only uses interleaved sample output */
-	status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
-	if ( status < 0 ) {
-		SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status));
-		ALSA_CloseAudio(this);
-		return(-1);
-	}
-
-	/* Try for a closest match on audio format */
-	status = -1;
-	for ( test_format = SDL_FirstAudioFormat(spec->format);
-	      test_format && (status < 0); ) {
-		switch ( test_format ) {
-			case AUDIO_U8:
-				format = SND_PCM_FORMAT_U8;
-				break;
-			case AUDIO_S8:
-				format = SND_PCM_FORMAT_S8;
-				break;
-			case AUDIO_S16LSB:
-				format = SND_PCM_FORMAT_S16_LE;
-				break;
-			case AUDIO_S16MSB:
-				format = SND_PCM_FORMAT_S16_BE;
-				break;
-			case AUDIO_U16LSB:
-				format = SND_PCM_FORMAT_U16_LE;
-				break;
-			case AUDIO_U16MSB:
-				format = SND_PCM_FORMAT_U16_BE;
-				break;
-			default:
-				format = 0;
-				break;
-		}
-		if ( format != 0 ) {
-			status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, hwparams, format);
-		}
-		if ( status < 0 ) {
-			test_format = SDL_NextAudioFormat();
-		}
-	}
-	if ( status < 0 ) {
-		SDL_SetError("Couldn't find any hardware audio formats");
-		ALSA_CloseAudio(this);
-		return(-1);
-	}
-	spec->format = test_format;
-
-	/* Set the number of channels */
-	status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, hwparams, spec->channels);
-	channels = spec->channels;
-	if ( status < 0 ) {
-		status = SDL_NAME(snd_pcm_hw_params_get_channels)(hwparams, &channels);
-		if ( status < 0 ) {
-			SDL_SetError("Couldn't set audio channels");
-			ALSA_CloseAudio(this);
-			return(-1);
-		}
-		spec->channels = channels;
-	}
-
-	/* Set the audio rate */
-	rate = spec->freq;
-
-	status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, &rate, NULL);
-	if ( status < 0 ) {
-		SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status));
-		ALSA_CloseAudio(this);
-		return(-1);
-	}
-	spec->freq = rate;
-
-	/* Set the buffer size, in samples */
-	if ( ALSA_set_period_size(this, spec, hwparams, 0) < 0 &&
-	     ALSA_set_buffer_size(this, spec, hwparams, 0) < 0 ) {
-		/* Failed to set desired buffer size, do the best you can... */
-		if ( ALSA_set_period_size(this, spec, hwparams, 1) < 0 ) {
-			SDL_SetError("Couldn't set hardware audio parameters: %s", SDL_NAME(snd_strerror)(status));
-			ALSA_CloseAudio(this);
-			return(-1);
-		}
-	}
-
-	/* Set the software parameters */
-	snd_pcm_sw_params_alloca(&swparams);
-	status = SDL_NAME(snd_pcm_sw_params_current)(pcm_handle, swparams);
-	if ( status < 0 ) {
-		SDL_SetError("Couldn't get software config: %s", SDL_NAME(snd_strerror)(status));
-		ALSA_CloseAudio(this);
-		return(-1);
-	}
-	status = SDL_NAME(snd_pcm_sw_params_set_avail_min)(pcm_handle, swparams, spec->samples);
-	if ( status < 0 ) {
-		SDL_SetError("Couldn't set minimum available samples: %s", SDL_NAME(snd_strerror)(status));
-		ALSA_CloseAudio(this);
-		return(-1);
-	}
-	status = SDL_NAME(snd_pcm_sw_params_set_start_threshold)(pcm_handle, swparams, 1);
-	if ( status < 0 ) {
-		SDL_SetError("Couldn't set start threshold: %s", SDL_NAME(snd_strerror)(status));
-		ALSA_CloseAudio(this);
-		return(-1);
-	}
-	status = SDL_NAME(snd_pcm_sw_params)(pcm_handle, swparams);
-	if ( status < 0 ) {
-		SDL_SetError("Couldn't set software audio parameters: %s", SDL_NAME(snd_strerror)(status));
-		ALSA_CloseAudio(this);
-		return(-1);
-	}
-
-	/* Calculate the final parameters for this audio specification */
-	SDL_CalculateAudioSpec(spec);
-
-	/* Allocate mixing buffer */
-	mixlen = spec->size;
-	mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
-	if ( mixbuf == NULL ) {
-		ALSA_CloseAudio(this);
-		return(-1);
-	}
-	SDL_memset(mixbuf, spec->silence, spec->size);
-
-	/* Switch to blocking mode for playback */
-	SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0);
-
-	/* We're ready to rock and roll. :-) */
-	return(0);
-}
diff --git a/apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.h b/apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.h
deleted file mode 100644
index 55ae87b8ac..0000000000
--- a/apps/plugins/sdl/src/audio/alsa/SDL_alsa_audio.h
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _ALSA_PCM_audio_h
-#define _ALSA_PCM_audio_h
-
-#include <alsa/asoundlib.h>
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData {
-	/* The audio device handle */
-	snd_pcm_t *pcm_handle;
-
-	/* Raw mixing buffer */
-	Uint8 *mixbuf;
-	int    mixlen;
-};
-
-/* Old variable names */
-#define pcm_handle		(this->hidden->pcm_handle)
-#define mixbuf			(this->hidden->mixbuf)
-#define mixlen			(this->hidden->mixlen)
-
-#endif /* _ALSA_PCM_audio_h */
diff --git a/apps/plugins/sdl/src/audio/arts/SDL_artsaudio.c b/apps/plugins/sdl/src/audio/arts/SDL_artsaudio.c
deleted file mode 100644
index 373f8c1677..0000000000
--- a/apps/plugins/sdl/src/audio/arts/SDL_artsaudio.c
+++ /dev/null
@@ -1,362 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/* Allow access to a raw mixing buffer */
-
-#ifdef HAVE_SIGNAL_H
-#include <signal.h>
-#endif
-#include <unistd.h>
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_audiodev_c.h"
-#include "SDL_artsaudio.h"
-
-#ifdef SDL_AUDIO_DRIVER_ARTS_DYNAMIC
-#include "SDL_name.h"
-#include "SDL_loadso.h"
-#else
-#define SDL_NAME(X)	X
-#endif
-
-/* The tag name used by artsc audio */
-#define ARTS_DRIVER_NAME         "arts"
-
-/* Audio driver functions */
-static int ARTS_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void ARTS_WaitAudio(_THIS);
-static void ARTS_PlayAudio(_THIS);
-static Uint8 *ARTS_GetAudioBuf(_THIS);
-static void ARTS_CloseAudio(_THIS);
-
-#ifdef SDL_AUDIO_DRIVER_ARTS_DYNAMIC
-
-static const char *arts_library = SDL_AUDIO_DRIVER_ARTS_DYNAMIC;
-static void *arts_handle = NULL;
-static int arts_loaded = 0;
-
-static int (*SDL_NAME(arts_init))(void);
-static void (*SDL_NAME(arts_free))(void);
-static arts_stream_t (*SDL_NAME(arts_play_stream))(int rate, int bits, int channels, const char *name);
-static int (*SDL_NAME(arts_stream_set))(arts_stream_t s, arts_parameter_t param, int value);
-static int (*SDL_NAME(arts_stream_get))(arts_stream_t s, arts_parameter_t param);
-static int (*SDL_NAME(arts_write))(arts_stream_t s, const void *buffer, int count);
-static void (*SDL_NAME(arts_close_stream))(arts_stream_t s);
-static int (*SDL_NAME(arts_suspend))(void);
-static int (*SDL_NAME(arts_suspended))(void);
-static const char *(*SDL_NAME(arts_error_text))(int errorcode);
-
-static struct {
-	const char *name;
-	void **func;
-} arts_functions[] = {
-	{ "arts_init",		(void **)&SDL_NAME(arts_init)		},
-	{ "arts_free",		(void **)&SDL_NAME(arts_free)		},
-	{ "arts_play_stream",	(void **)&SDL_NAME(arts_play_stream)	},
-	{ "arts_stream_set",	(void **)&SDL_NAME(arts_stream_set)	},
-	{ "arts_stream_get",	(void **)&SDL_NAME(arts_stream_get)	},
-	{ "arts_write",		(void **)&SDL_NAME(arts_write)		},
-	{ "arts_close_stream",	(void **)&SDL_NAME(arts_close_stream)	},
-	{ "arts_suspend",	(void **)&SDL_NAME(arts_suspend)	},
-	{ "arts_suspended",	(void **)&SDL_NAME(arts_suspended)	},
-	{ "arts_error_text",	(void **)&SDL_NAME(arts_error_text)	},
-};
-
-static void UnloadARTSLibrary()
-{
-	if ( arts_loaded ) {
-		SDL_UnloadObject(arts_handle);
-		arts_handle = NULL;
-		arts_loaded = 0;
-	}
-}
-
-static int LoadARTSLibrary(void)
-{
-	int i, retval = -1;
-
-	arts_handle = SDL_LoadObject(arts_library);
-	if ( arts_handle ) {
-		arts_loaded = 1;
-		retval = 0;
-		for ( i=0; i<SDL_arraysize(arts_functions); ++i ) {
-			*arts_functions[i].func = SDL_LoadFunction(arts_handle, arts_functions[i].name);
-			if ( !*arts_functions[i].func ) {
-				retval = -1;
-				UnloadARTSLibrary();
-				break;
-			}
-		}
-	}
-	return retval;
-}
-
-#else
-
-static void UnloadARTSLibrary()
-{
-	return;
-}
-
-static int LoadARTSLibrary(void)
-{
-	return 0;
-}
-
-#endif /* SDL_AUDIO_DRIVER_ARTS_DYNAMIC */
-
-/* Audio driver bootstrap functions */
-
-static int ARTS_Suspend(void)
-{
-	const Uint32 abortms = SDL_GetTicks() + 3000; /* give up after 3 secs */
-	while ( (!SDL_NAME(arts_suspended)()) && (SDL_GetTicks() < abortms) ) {
-		if ( SDL_NAME(arts_suspend)() ) {
-			break;
-		}
-	}
-
-	return SDL_NAME(arts_suspended)();
-}
-
-static int Audio_Available(void)
-{
-	int available = 0;
-
-	if ( LoadARTSLibrary() < 0 ) {
-		return available;
-	}
-	if ( SDL_NAME(arts_init)() == 0 ) {
-		if ( ARTS_Suspend() ) {
-			/* Play a stream so aRts doesn't crash */
-			arts_stream_t stream2;
-			stream2=SDL_NAME(arts_play_stream)(44100, 16, 2, "SDL");
-			SDL_NAME(arts_write)(stream2, "", 0);
-			SDL_NAME(arts_close_stream)(stream2);
-			available = 1;
-		}
-		SDL_NAME(arts_free)();
-	}
-	UnloadARTSLibrary();
-
-	return available;
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-	SDL_free(device->hidden);
-	SDL_free(device);
-	UnloadARTSLibrary();
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	LoadARTSLibrary();
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-	if ( this ) {
-		SDL_memset(this, 0, (sizeof *this));
-		this->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *this->hidden));
-	}
-	if ( (this == NULL) || (this->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( this ) {
-			SDL_free(this);
-		}
-		return(0);
-	}
-	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-	stream = 0;
-
-	/* Set the function pointers */
-	this->OpenAudio = ARTS_OpenAudio;
-	this->WaitAudio = ARTS_WaitAudio;
-	this->PlayAudio = ARTS_PlayAudio;
-	this->GetAudioBuf = ARTS_GetAudioBuf;
-	this->CloseAudio = ARTS_CloseAudio;
-
-	this->free = Audio_DeleteDevice;
-
-	return this;
-}
-
-AudioBootStrap ARTS_bootstrap = {
-	ARTS_DRIVER_NAME, "Analog Realtime Synthesizer",
-	Audio_Available, Audio_CreateDevice
-};
-
-/* This function waits until it is possible to write a full sound buffer */
-static void ARTS_WaitAudio(_THIS)
-{
-	Sint32 ticks;
-
-	/* Check to see if the thread-parent process is still alive */
-	{ static int cnt = 0;
-		/* Note that this only works with thread implementations 
-		   that use a different process id for each thread.
-		*/
-		if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */
-			if ( kill(parent, 0) < 0 ) {
-				this->enabled = 0;
-			}
-		}
-	}
-
-	/* Use timer for general audio synchronization */
-	ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS;
-	if ( ticks > 0 ) {
-		SDL_Delay(ticks);
-	}
-}
-
-static void ARTS_PlayAudio(_THIS)
-{
-	int written;
-
-	/* Write the audio data */
-	written = SDL_NAME(arts_write)(stream, mixbuf, mixlen);
-	
-	/* If timer synchronization is enabled, set the next write frame */
-	if ( frame_ticks ) {
-		next_frame += frame_ticks;
-	}
-
-	/* If we couldn't write, assume fatal error for now */
-	if ( written < 0 ) {
-		this->enabled = 0;
-	}
-#ifdef DEBUG_AUDIO
-	fprintf(stderr, "Wrote %d bytes of audio data\n", written);
-#endif
-}
-
-static Uint8 *ARTS_GetAudioBuf(_THIS)
-{
-	return(mixbuf);
-}
-
-static void ARTS_CloseAudio(_THIS)
-{
-	if ( mixbuf != NULL ) {
-		SDL_FreeAudioMem(mixbuf);
-		mixbuf = NULL;
-	}
-	if ( stream ) {
-		SDL_NAME(arts_close_stream)(stream);
-		stream = 0;
-	}
-	SDL_NAME(arts_free)();
-}
-
-static int ARTS_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	int bits, frag_spec;
-	Uint16 test_format, format;
-	int error_code;
-
-	/* Reset the timer synchronization flag */
-	frame_ticks = 0.0;
-
-	mixbuf = NULL;
-
-	/* Try for a closest match on audio format */
-	format = 0;
-	bits = 0;
-	for ( test_format = SDL_FirstAudioFormat(spec->format);
-						! format && test_format; ) {
-#ifdef DEBUG_AUDIO
-		fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
-#endif
-		switch ( test_format ) {
-			case AUDIO_U8:
-				bits = 8;
-				format = 1;
-				break;
-			case AUDIO_S16LSB:
-				bits = 16;
-				format = 1;
-				break;
-			default:
-				format = 0;
-				break;
-		}
-		if ( ! format ) {
-			test_format = SDL_NextAudioFormat();
-		}
-	}
-	if ( format == 0 ) {
-		SDL_SetError("Couldn't find any hardware audio formats");
-		return(-1);
-	}
-	spec->format = test_format;
-
-	error_code = SDL_NAME(arts_init)();
-	if ( error_code != 0 ) {
-		SDL_SetError("Unable to initialize ARTS: %s", SDL_NAME(arts_error_text)(error_code));
-		return(-1);
-	}
-	if ( ! ARTS_Suspend() ) {
-		SDL_SetError("ARTS can not open audio device");
-		return(-1);
-	}
-	stream = SDL_NAME(arts_play_stream)(spec->freq, bits, spec->channels, "SDL");
-
-	/* Calculate the final parameters for this audio specification */
-	SDL_CalculateAudioSpec(spec);
-
-	/* Determine the power of two of the fragment size */
-	for ( frag_spec = 0; (0x01<<frag_spec) < spec->size; ++frag_spec );
-	if ( (0x01<<frag_spec) != spec->size ) {
-		SDL_SetError("Fragment size must be a power of two");
-		return(-1);
-	}
-	frag_spec |= 0x00020000;	/* two fragments, for low latency */
-
-#ifdef ARTS_P_PACKET_SETTINGS
-	SDL_NAME(arts_stream_set)(stream, ARTS_P_PACKET_SETTINGS, frag_spec);
-#else
-	SDL_NAME(arts_stream_set)(stream, ARTS_P_PACKET_SIZE, frag_spec&0xffff);
-	SDL_NAME(arts_stream_set)(stream, ARTS_P_PACKET_COUNT, frag_spec>>16);
-#endif
-	spec->size = SDL_NAME(arts_stream_get)(stream, ARTS_P_PACKET_SIZE);
-
-	/* Allocate mixing buffer */
-	mixlen = spec->size;
-	mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
-	if ( mixbuf == NULL ) {
-		return(-1);
-	}
-	SDL_memset(mixbuf, spec->silence, spec->size);
-
-	/* Get the parent process id (we're the parent of the audio thread) */
-	parent = getpid();
-
-	/* We're ready to rock and roll. :-) */
-	return(0);
-}
diff --git a/apps/plugins/sdl/src/audio/arts/SDL_artsaudio.h b/apps/plugins/sdl/src/audio/arts/SDL_artsaudio.h
deleted file mode 100644
index de3b22822c..0000000000
--- a/apps/plugins/sdl/src/audio/arts/SDL_artsaudio.h
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_artscaudio_h
-#define _SDL_artscaudio_h
-
-#include <artsc.h>
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData {
-	/* The stream descriptor for the audio device */
-	arts_stream_t stream;
-
-	/* The parent process id, to detect when application quits */
-	pid_t parent;
-
-	/* Raw mixing buffer */
-	Uint8 *mixbuf;
-	int    mixlen;
-
-	/* Support for audio timing using a timer, in addition to select() */
-	float frame_ticks;
-	float next_frame;
-};
-#define FUDGE_TICKS	10	/* The scheduler overhead ticks per frame */
-
-/* Old variable names */
-#define stream			(this->hidden->stream)
-#define parent			(this->hidden->parent)
-#define mixbuf			(this->hidden->mixbuf)
-#define mixlen			(this->hidden->mixlen)
-#define frame_ticks		(this->hidden->frame_ticks)
-#define next_frame		(this->hidden->next_frame)
-
-#endif /* _SDL_artscaudio_h */
-
diff --git a/apps/plugins/sdl/src/audio/baudio/SDL_beaudio.cc b/apps/plugins/sdl/src/audio/baudio/SDL_beaudio.cc
deleted file mode 100644
index de635f8bad..0000000000
--- a/apps/plugins/sdl/src/audio/baudio/SDL_beaudio.cc
+++ /dev/null
@@ -1,225 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/* Allow access to the audio stream on BeOS */
-
-#include <SoundPlayer.h>
-
-#include "../../main/beos/SDL_BeApp.h"
-
-extern "C" {
-
-#include "SDL_audio.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_sysaudio.h"
-#include "../../thread/beos/SDL_systhread_c.h"
-#include "SDL_beaudio.h"
-
-
-/* Audio driver functions */
-static int BE_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void BE_WaitAudio(_THIS);
-static void BE_PlayAudio(_THIS);
-static Uint8 *BE_GetAudioBuf(_THIS);
-static void BE_CloseAudio(_THIS);
-
-/* Audio driver bootstrap functions */
-
-static int Audio_Available(void)
-{
-	return(1);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-	SDL_free(device->hidden);
-	SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *device;
-
-	/* Initialize all variables that we clean on shutdown */
-	device = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-	if ( device ) {
-		SDL_memset(device, 0, (sizeof *device));
-		device->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *device->hidden));
-	}
-	if ( (device == NULL) || (device->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( device ) {
-			SDL_free(device);
-		}
-		return(0);
-	}
-	SDL_memset(device->hidden, 0, (sizeof *device->hidden));
-
-	/* Set the function pointers */
-	device->OpenAudio = BE_OpenAudio;
-	device->WaitAudio = BE_WaitAudio;
-	device->PlayAudio = BE_PlayAudio;
-	device->GetAudioBuf = BE_GetAudioBuf;
-	device->CloseAudio = BE_CloseAudio;
-
-	device->free = Audio_DeleteDevice;
-
-	return device;
-}
-
-AudioBootStrap BAUDIO_bootstrap = {
-	"baudio", "BeOS BSoundPlayer",
-	Audio_Available, Audio_CreateDevice
-};
-
-/* The BeOS callback for handling the audio buffer */
-static void FillSound(void *device, void *stream, size_t len, 
-					const media_raw_audio_format &format)
-{
-	SDL_AudioDevice *audio = (SDL_AudioDevice *)device;
-
-	/* Silence the buffer, since it's ours */
-	SDL_memset(stream, audio->spec.silence, len);
-
-	/* Only do soemthing if audio is enabled */
-	if ( ! audio->enabled )
-		return;
-
-	if ( ! audio->paused ) {
-		if ( audio->convert.needed ) {
-			SDL_mutexP(audio->mixer_lock);
-			(*audio->spec.callback)(audio->spec.userdata,
-				(Uint8 *)audio->convert.buf,audio->convert.len);
-			SDL_mutexV(audio->mixer_lock);
-			SDL_ConvertAudio(&audio->convert);
-			SDL_memcpy(stream,audio->convert.buf,audio->convert.len_cvt);
-		} else {
-			SDL_mutexP(audio->mixer_lock);
-			(*audio->spec.callback)(audio->spec.userdata,
-						(Uint8 *)stream, len);
-			SDL_mutexV(audio->mixer_lock);
-		}
-	}
-	return;
-}
-
-/* Dummy functions -- we don't use thread-based audio */
-void BE_WaitAudio(_THIS)
-{
-	return;
-}
-void BE_PlayAudio(_THIS)
-{
-	return;
-}
-Uint8 *BE_GetAudioBuf(_THIS)
-{
-	return(NULL);
-}
-
-void BE_CloseAudio(_THIS)
-{
-	if ( audio_obj ) {
-		audio_obj->Stop();
-		delete audio_obj;
-		audio_obj = NULL;
-	}
-
-	/* Quit the Be Application, if there's nothing left to do */
-	SDL_QuitBeApp();
-}
-
-int BE_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-    int valid_datatype = 0;
-    media_raw_audio_format format;
-    Uint16 test_format = SDL_FirstAudioFormat(spec->format);
-
-    /* Parse the audio format and fill the Be raw audio format */
-    memset(&format, '\0', sizeof (media_raw_audio_format));
-    format.byte_order = B_MEDIA_LITTLE_ENDIAN;
-    format.frame_rate = (float) spec->freq;
-    format.channel_count = spec->channels;  /* !!! FIXME: support > 2? */
-    while ((!valid_datatype) && (test_format)) {
-        valid_datatype = 1;
-        spec->format = test_format;
-        switch (test_format) {
-            case AUDIO_S8:
-                format.format = media_raw_audio_format::B_AUDIO_CHAR;
-                break;
-
-            case AUDIO_U8:
-                format.format = media_raw_audio_format::B_AUDIO_UCHAR;
-                break;
-
-            case AUDIO_S16LSB:
-                format.format = media_raw_audio_format::B_AUDIO_SHORT;
-                break;
-
-            case AUDIO_S16MSB:
-                format.format = media_raw_audio_format::B_AUDIO_SHORT;
-                format.byte_order = B_MEDIA_BIG_ENDIAN;
-                break;
-
-            default:
-                valid_datatype = 0;
-                test_format = SDL_NextAudioFormat();
-                break;
-        }
-    }
-
-    if (!valid_datatype) { /* shouldn't happen, but just in case... */
-        SDL_SetError("Unsupported audio format");
-        return (-1);
-    }
-
-    /* Initialize the Be Application, if it's not already started */
-    if (SDL_InitBeApp() < 0) {
-        return (-1);
-    }
-
-    format.buffer_size = spec->samples;
-
-	/* Calculate the final parameters for this audio specification */
-	SDL_CalculateAudioSpec(spec);
-
-	/* Subscribe to the audio stream (creates a new thread) */
-	{ sigset_t omask;
-		SDL_MaskSignals(&omask);
-		audio_obj = new BSoundPlayer(&format, "SDL Audio", FillSound,
-		                                                 NULL, _this);
-		SDL_UnmaskSignals(&omask);
-	}
-	if ( audio_obj->Start() == B_NO_ERROR ) {
-		audio_obj->SetHasData(true);
-	} else {
-		SDL_SetError("Unable to start Be audio");
-		return(-1);
-	}
-
-	/* We're running! */
-	return(1);
-}
-
-};	/* Extern C */
diff --git a/apps/plugins/sdl/src/audio/baudio/SDL_beaudio.h b/apps/plugins/sdl/src/audio/baudio/SDL_beaudio.h
deleted file mode 100644
index adaf1dee5d..0000000000
--- a/apps/plugins/sdl/src/audio/baudio/SDL_beaudio.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_lowaudio_h
-#define _SDL_lowaudio_h
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *_this
-
-struct SDL_PrivateAudioData {
-	BSoundPlayer *audio_obj;
-};
-
-/* Old variable names */
-#define audio_obj	(_this->hidden->audio_obj)
-
-#endif /* _SDL_lowaudio_h */
diff --git a/apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.c b/apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.c
deleted file mode 100644
index e5e0d9480a..0000000000
--- a/apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.c
+++ /dev/null
@@ -1,404 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/*
- * Driver for native OpenBSD/NetBSD audio(4).
- * vedge@vedge.com.ar.
- */
-
-#include <errno.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include <sys/time.h>
-#include <sys/ioctl.h>
-#include <sys/stat.h>
-#include <sys/types.h>
-#include <sys/audioio.h>
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_audiodev_c.h"
-#include "SDL_bsdaudio.h"
-
-/* The tag name used by NetBSD/OpenBSD audio */
-#ifdef __NetBSD__
-#define BSD_AUDIO_DRIVER_NAME         "netbsd"
-#define BSD_AUDIO_DRIVER_DESC         "Native NetBSD audio"
-#else
-#define BSD_AUDIO_DRIVER_NAME         "openbsd"
-#define BSD_AUDIO_DRIVER_DESC         "Native OpenBSD audio"
-#endif
-
-/* Open the audio device for playback, and don't block if busy */
-/* #define USE_BLOCKING_WRITES */
-
-/* Use timer for synchronization */
-/* #define USE_TIMER_SYNC */
-
-/* #define DEBUG_AUDIO */
-/* #define DEBUG_AUDIO_STREAM */
-
-#ifdef USE_BLOCKING_WRITES
-#define OPEN_FLAGS	O_WRONLY
-#else
-#define OPEN_FLAGS	(O_WRONLY|O_NONBLOCK)
-#endif
-
-/* Audio driver functions */
-static void OBSD_WaitAudio(_THIS);
-static int OBSD_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void OBSD_PlayAudio(_THIS);
-static Uint8 *OBSD_GetAudioBuf(_THIS);
-static void OBSD_CloseAudio(_THIS);
-
-#ifdef DEBUG_AUDIO
-static void OBSD_Status(_THIS);
-#endif
-
-/* Audio driver bootstrap functions */
-
-static int
-Audio_Available(void)
-{
-    int fd;
-    int available;
-
-    available = 0;
-    fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0);
-    if(fd >= 0) {
-	available = 1;
-	close(fd);
-    }
-    return(available);
-}
-
-static void
-Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-    SDL_free(device->hidden);
-    SDL_free(device);
-}
-
-static SDL_AudioDevice
-*Audio_CreateDevice(int devindex)
-{
-    SDL_AudioDevice *this;
-
-    /* Initialize all variables that we clean on shutdown */
-    this = (SDL_AudioDevice*)SDL_malloc(sizeof(SDL_AudioDevice));
-    if(this) {
-	SDL_memset(this, 0, (sizeof *this));
-	this->hidden =
-	    (struct SDL_PrivateAudioData*)SDL_malloc((sizeof *this->hidden));
-    }
-    if((this == NULL) || (this->hidden == NULL)) {
-	SDL_OutOfMemory();
-	if(this) SDL_free(this);
-	return(0);
-    }
-    SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-    audio_fd = -1;
-
-    /* Set the function pointers */
-    this->OpenAudio = OBSD_OpenAudio;
-    this->WaitAudio = OBSD_WaitAudio;
-    this->PlayAudio = OBSD_PlayAudio;
-    this->GetAudioBuf = OBSD_GetAudioBuf;
-    this->CloseAudio = OBSD_CloseAudio;
-
-    this->free = Audio_DeleteDevice;
-    
-    return this;
-}
-
-AudioBootStrap BSD_AUDIO_bootstrap = {
-	BSD_AUDIO_DRIVER_NAME, BSD_AUDIO_DRIVER_DESC,
-	Audio_Available, Audio_CreateDevice
-};
-
-/* This function waits until it is possible to write a full sound buffer */
-static void
-OBSD_WaitAudio(_THIS)
-{
-#ifndef USE_BLOCKING_WRITES /* Not necessary when using blocking writes */
-	/* See if we need to use timed audio synchronization */
-	if ( frame_ticks ) {
-		/* Use timer for general audio synchronization */
-		Sint32 ticks;
-
-		ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS;
-		if ( ticks > 0 ) {
-			SDL_Delay(ticks);
-		}
-	} else {
-		/* Use select() for audio synchronization */
-		fd_set fdset;
-		struct timeval timeout;
-
-		FD_ZERO(&fdset);
-		FD_SET(audio_fd, &fdset);
-		timeout.tv_sec = 10;
-		timeout.tv_usec = 0;
-#ifdef DEBUG_AUDIO
-		fprintf(stderr, "Waiting for audio to get ready\n");
-#endif
-		if ( select(audio_fd+1, NULL, &fdset, NULL, &timeout) <= 0 ) {
-			const char *message =
-			"Audio timeout - buggy audio driver? (disabled)";
-			/* In general we should never print to the screen,
-			   but in this case we have no other way of letting
-			   the user know what happened.
-			*/
-			fprintf(stderr, "SDL: %s\n", message);
-			this->enabled = 0;
-			/* Don't try to close - may hang */
-			audio_fd = -1;
-#ifdef DEBUG_AUDIO
-			fprintf(stderr, "Done disabling audio\n");
-#endif
-		}
-#ifdef DEBUG_AUDIO
-		fprintf(stderr, "Ready!\n");
-#endif
-	}
-#endif /* !USE_BLOCKING_WRITES */
-}
-
-static void
-OBSD_PlayAudio(_THIS)
-{
-	int written, p=0;
-
-	/* Write the audio data, checking for EAGAIN on broken audio drivers */
-	do {
-		written = write(audio_fd, &mixbuf[p], mixlen-p);
-		if (written>0)
-		   p += written;
-		if (written == -1 && errno != 0 && errno != EAGAIN && errno != EINTR)
-		{
-		   /* Non recoverable error has occurred. It should be reported!!! */
-		   perror("audio");
-		   break;
-		}
-
-		if ( p < written || ((written < 0) && ((errno == 0) || (errno == EAGAIN))) ) {
-			SDL_Delay(1);	/* Let a little CPU time go by */
-		}
-	} while ( p < written );
-
-	/* If timer synchronization is enabled, set the next write frame */
-	if ( frame_ticks ) {
-		next_frame += frame_ticks;
-	}
-
-	/* If we couldn't write, assume fatal error for now */
-	if ( written < 0 ) {
-		this->enabled = 0;
-	}
-#ifdef DEBUG_AUDIO
-	fprintf(stderr, "Wrote %d bytes of audio data\n", written);
-#endif
-}
-
-static Uint8
-*OBSD_GetAudioBuf(_THIS)
-{
-    return(mixbuf);
-}
-
-static void
-OBSD_CloseAudio(_THIS)
-{
-    if(mixbuf != NULL) {
-	SDL_FreeAudioMem(mixbuf);
-	mixbuf = NULL;
-    }
-    if(audio_fd >= 0) {
-	close(audio_fd);
-	audio_fd = -1;
-    }
-}
-
-#ifdef DEBUG_AUDIO
-void
-OBSD_Status(_THIS)
-{
-    audio_info_t info;
-
-    if(ioctl(audio_fd, AUDIO_GETINFO, &info) < 0) {
-	fprintf(stderr,"AUDIO_GETINFO failed.\n");
-	return;
-    }
-
-    fprintf(stderr,"\n"
-"[play/record info]\n"
-"buffer size	:   %d bytes\n"
-"sample rate	:   %i Hz\n"
-"channels	:   %i\n"
-"precision	:   %i-bit\n"
-"encoding	:   0x%x\n"
-"seek		:   %i\n"
-"sample count	:   %i\n"
-"EOF count	:   %i\n"
-"paused		:   %s\n"
-"error occured	:   %s\n"
-"waiting		:   %s\n"
-"active		:   %s\n"
-"",
-    info.play.buffer_size,
-    info.play.sample_rate,
-    info.play.channels,
-    info.play.precision,
-    info.play.encoding,
-    info.play.seek,
-    info.play.samples,
-    info.play.eof,
-    info.play.pause ? "yes" : "no",
-    info.play.error ? "yes" : "no",
-    info.play.waiting ? "yes" : "no",
-    info.play.active ? "yes": "no");
-
-    fprintf(stderr,"\n"
-"[audio info]\n"
-"monitor_gain	:   %i\n"
-"hw block size	:   %d bytes\n"
-"hi watermark	:   %i\n"
-"lo watermark	:   %i\n"
-"audio mode	:   %s\n"
-"",  
-    info.monitor_gain,
-    info.blocksize,
-    info.hiwat, info.lowat,
-    (info.mode == AUMODE_PLAY) ? "PLAY"
-    : (info.mode = AUMODE_RECORD) ? "RECORD"
-    : (info.mode == AUMODE_PLAY_ALL ? "PLAY_ALL"
-    : "?"));
-}
-#endif /* DEBUG_AUDIO */
-
-static int
-OBSD_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-    char audiodev[64];
-    Uint16 format;
-    audio_info_t info;
-
-    AUDIO_INITINFO(&info);
-    
-    /* Calculate the final parameters for this audio specification */
-    SDL_CalculateAudioSpec(spec);
-
-#ifdef USE_TIMER_SYNC
-    frame_ticks = 0.0;
-#endif
-
-    /* Open the audio device */
-    audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
-    if(audio_fd < 0) {
-	SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
-	return(-1);
-    }
-    
-    /* Set to play mode */
-    info.mode = AUMODE_PLAY;
-    if(ioctl(audio_fd, AUDIO_SETINFO, &info) < 0) {
-	SDL_SetError("Couldn't put device into play mode");
-	return(-1);
-    }
-    
-    mixbuf = NULL;
-    AUDIO_INITINFO(&info);
-    for (format = SDL_FirstAudioFormat(spec->format); 
-    	format; format = SDL_NextAudioFormat())
-    {
-	switch(format) {
-	case AUDIO_U8:
-	    info.play.encoding = AUDIO_ENCODING_ULINEAR;
-	    info.play.precision = 8;
-	    break;
-	case AUDIO_S8:
-	    info.play.encoding = AUDIO_ENCODING_SLINEAR;
-	    info.play.precision = 8;
-	    break;
-	case AUDIO_S16LSB:
-	    info.play.encoding = AUDIO_ENCODING_SLINEAR_LE;
-	    info.play.precision = 16;
-	    break;
-	case AUDIO_S16MSB:
-	    info.play.encoding = AUDIO_ENCODING_SLINEAR_BE;
-	    info.play.precision = 16;
-	    break;
-	case AUDIO_U16LSB:
-	    info.play.encoding = AUDIO_ENCODING_ULINEAR_LE;
-	    info.play.precision = 16;
-	    break;
-	case AUDIO_U16MSB:
-	    info.play.encoding = AUDIO_ENCODING_ULINEAR_BE;
-	    info.play.precision = 16;
-	    break;
-	default:
-	    continue;
-	}
-	if (ioctl(audio_fd, AUDIO_SETINFO, &info) == 0)
-	    break;
-    }
-
-    if(!format) {
-	SDL_SetError("No supported encoding for 0x%x", spec->format);
-	return(-1);
-    }
-
-    spec->format = format;
-
-    AUDIO_INITINFO(&info);
-    info.play.channels = spec->channels;
-    if (ioctl(audio_fd, AUDIO_SETINFO, &info) == -1)
-    	spec->channels = 1;
-    AUDIO_INITINFO(&info);
-    info.play.sample_rate = spec->freq;
-    info.blocksize = spec->size;
-    info.hiwat = 5;
-    info.lowat = 3;
-    (void)ioctl(audio_fd, AUDIO_SETINFO, &info);
-    (void)ioctl(audio_fd, AUDIO_GETINFO, &info);
-    spec->freq  = info.play.sample_rate;
-    /* Allocate mixing buffer */
-    mixlen = spec->size;
-    mixbuf = (Uint8*)SDL_AllocAudioMem(mixlen);
-    if(mixbuf == NULL) {
-	return(-1);
-    }
-    SDL_memset(mixbuf, spec->silence, spec->size);
-    
-    /* Get the parent process id (we're the parent of the audio thread) */
-    parent = getpid();
-
-#ifdef DEBUG_AUDIO
-    OBSD_Status(this);
-#endif
-
-    /* We're ready to rock and roll. :-) */
-    return(0);
-}
diff --git a/apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.h b/apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.h
deleted file mode 100644
index c9f69cf544..0000000000
--- a/apps/plugins/sdl/src/audio/bsd/SDL_bsdaudio.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_openbsdaudio_h
-#define _SDL_openbsdaudio_h
-
-#include "../SDL_sysaudio.h"
-
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData
-{
-    /* The file descriptor for the audio device */
-    int audio_fd;
-
-    /* The parent process id, to detect when application quits */
-    pid_t parent;
-
-    /* Raw mixing buffer */
-    Uint8 *mixbuf;
-    int    mixlen;
-
-    /* Support for audio timing using a timer, in addition to select() */
-    float frame_ticks;
-    float next_frame;
-};
-
-#define FUDGE_TICKS	10	/* The scheduler overhead ticks per frame */
-
-/* Old variable names */
-#define audio_fd		(this->hidden->audio_fd)
-#define parent			(this->hidden->parent)
-#define mixbuf			(this->hidden->mixbuf)
-#define mixlen			(this->hidden->mixlen)
-#define frame_ticks		(this->hidden->frame_ticks)
-#define next_frame		(this->hidden->next_frame)
-
-#endif /* _SDL_openbsdaudio_h */
diff --git a/apps/plugins/sdl/src/audio/dart/SDL_dart.c b/apps/plugins/sdl/src/audio/dart/SDL_dart.c
deleted file mode 100644
index 77e530db51..0000000000
--- a/apps/plugins/sdl/src/audio/dart/SDL_dart.c
+++ /dev/null
@@ -1,441 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/* Allow access to a raw mixing buffer */
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audio_c.h"
-#include "SDL_dart.h"
-
-// Buffer states:
-#define BUFFER_EMPTY       0
-#define BUFFER_USED        1
-
-typedef struct _tMixBufferDesc {
-  int              iBufferUsage;      // BUFFER_EMPTY or BUFFER_USED
-  SDL_AudioDevice *pSDLAudioDevice;
-} tMixBufferDesc, *pMixBufferDesc;
-
-
-//---------------------------------------------------------------------
-// DARTEventFunc
-//
-// This function is called by DART, when an event occures, like end of 
-// playback of a buffer, etc...
-//---------------------------------------------------------------------
-LONG APIENTRY DARTEventFunc(ULONG ulStatus,
-			    PMCI_MIX_BUFFER pBuffer,
-			    ULONG ulFlags)
-{
-  if (ulFlags && MIX_WRITE_COMPLETE)
-  { // Playback of buffer completed!
-
-    // Get pointer to buffer description
-    pMixBufferDesc pBufDesc;
-
-    if (pBuffer)
-    {
-      pBufDesc = (pMixBufferDesc) (*pBuffer).ulUserParm;
-
-      if (pBufDesc)
-      {
-        SDL_AudioDevice *pSDLAudioDevice = pBufDesc->pSDLAudioDevice;
-        // Set the buffer to be empty
-        pBufDesc->iBufferUsage = BUFFER_EMPTY;
-        // And notify DART feeder thread that it will have to work a bit.
-        if (pSDLAudioDevice)
-        DosPostEventSem(pSDLAudioDevice->hidden->hevAudioBufferPlayed);
-      }
-    }
-  }
-  return TRUE;
-}
-
-
-int DART_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-  Uint16 test_format = SDL_FirstAudioFormat(spec->format);
-  int valid_datatype = 0;
-  MCI_AMP_OPEN_PARMS AmpOpenParms;
-  MCI_GENERIC_PARMS GenericParms;
-  int iDeviceOrd = 0; // Default device to be used
-  int bOpenShared = 1; // Try opening it shared
-  int iBits = 16; // Default is 16 bits signed
-  int iFreq = 44100; // Default is 44KHz
-  int iChannels = 2; // Default is 2 channels (Stereo)
-  int iNumBufs = 2;  // Number of audio buffers: 2
-  int iBufSize;
-  int iOpenMode;
-  int iSilence;
-  int rc;
-
-  // First thing is to try to open a given DART device!
-  SDL_memset(&AmpOpenParms, 0, sizeof(MCI_AMP_OPEN_PARMS));
-  // pszDeviceType should contain the device type in low word, and device ordinal in high word!
-  AmpOpenParms.pszDeviceType = (PSZ) (MCI_DEVTYPE_AUDIO_AMPMIX | (iDeviceOrd << 16));
-
-  iOpenMode = MCI_WAIT | MCI_OPEN_TYPE_ID;
-  if (bOpenShared) iOpenMode |= MCI_OPEN_SHAREABLE;
-
-  rc = mciSendCommand( 0, MCI_OPEN,
-                       iOpenMode,
-		       (PVOID) &AmpOpenParms, 0);
-  if (rc!=MCIERR_SUCCESS) // No audio available??
-    return (-1);
-  // Save the device ID we got from DART!
-  // We will use this in the next calls!
-  iDeviceOrd = AmpOpenParms.usDeviceID;
-
-  // Determine the audio parameters from the AudioSpec
-  if (spec->channels > 2)
-    spec->channels = 2;  // !!! FIXME: more than stereo support in OS/2?
-
-  while ((!valid_datatype) && (test_format)) {
-    spec->format = test_format;
-    valid_datatype = 1;
-    switch (test_format) {
-      case AUDIO_U8:
-        // Unsigned 8 bit audio data
-        iSilence = 0x80;
-        iBits = 8;
-        break;
-
-      case AUDIO_S16LSB:
-        // Signed 16 bit audio data
-        iSilence = 0x00;
-        iBits = 16;
-        break;
-
-      default:
-        valid_datatype = 0;
-        test_format = SDL_NextAudioFormat();
-        break;
-    }
-  }
-
-  if (!valid_datatype) { // shouldn't happen, but just in case...
-    // Close DART, and exit with error code!
-    mciSendCommand(iDeviceOrd, MCI_CLOSE, MCI_WAIT, &GenericParms, 0);
-    SDL_SetError("Unsupported audio format");
-    return (-1);
-  }
-
-  iFreq = spec->freq;
-  iChannels = spec->channels;
-  /* Update the fragment size as size in bytes */
-  SDL_CalculateAudioSpec(spec);
-  iBufSize = spec->size;
-
-  // Now query this device if it supports the given freq/bits/channels!
-  SDL_memset(&(_this->hidden->MixSetupParms), 0, sizeof(MCI_MIXSETUP_PARMS));
-  _this->hidden->MixSetupParms.ulBitsPerSample = iBits;
-  _this->hidden->MixSetupParms.ulFormatTag = MCI_WAVE_FORMAT_PCM;
-  _this->hidden->MixSetupParms.ulSamplesPerSec = iFreq;
-  _this->hidden->MixSetupParms.ulChannels = iChannels;
-  _this->hidden->MixSetupParms.ulFormatMode = MCI_PLAY;
-  _this->hidden->MixSetupParms.ulDeviceType = MCI_DEVTYPE_WAVEFORM_AUDIO;
-  _this->hidden->MixSetupParms.pmixEvent = DARTEventFunc;
-  rc = mciSendCommand (iDeviceOrd, MCI_MIXSETUP,
-                       MCI_WAIT | MCI_MIXSETUP_QUERYMODE,
-                       &(_this->hidden->MixSetupParms), 0);
-  if (rc!=MCIERR_SUCCESS)
-  { // The device cannot handle this format!
-    // Close DART, and exit with error code!
-    mciSendCommand(iDeviceOrd, MCI_CLOSE, MCI_WAIT, &GenericParms, 0);
-    SDL_SetError("Audio device doesn't support requested audio format");
-    return(-1);
-  }
-  // The device can handle this format, so initialize!
-  rc = mciSendCommand(iDeviceOrd, MCI_MIXSETUP,
-                      MCI_WAIT | MCI_MIXSETUP_INIT,
-                      &(_this->hidden->MixSetupParms), 0);
-  if (rc!=MCIERR_SUCCESS)
-  { // The device could not be opened!
-    // Close DART, and exit with error code!
-    mciSendCommand(iDeviceOrd, MCI_CLOSE, MCI_WAIT, &GenericParms, 0);
-    SDL_SetError("Audio device could not be set up");
-    return(-1);
-  }
-  // Ok, the device is initialized.
-  // Now we should allocate buffers. For this, we need a place where
-  // the buffer descriptors will be:
-  _this->hidden->pMixBuffers = (MCI_MIX_BUFFER *) SDL_malloc(sizeof(MCI_MIX_BUFFER)*iNumBufs);
-  if (!(_this->hidden->pMixBuffers))
-  { // Not enough memory!
-    // Close DART, and exit with error code!
-    mciSendCommand(iDeviceOrd, MCI_CLOSE, MCI_WAIT, &GenericParms, 0);
-    SDL_SetError("Not enough memory for audio buffer descriptors");
-    return(-1);
-  }
-  // Now that we have the place for buffer list, we can ask DART for the
-  // buffers!
-  _this->hidden->BufferParms.ulNumBuffers = iNumBufs;               // Number of buffers
-  _this->hidden->BufferParms.ulBufferSize = iBufSize;               // each with this size
-  _this->hidden->BufferParms.pBufList = _this->hidden->pMixBuffers; // getting descriptorts into this list
-  // Allocate buffers!
-  rc = mciSendCommand(iDeviceOrd, MCI_BUFFER,
-                      MCI_WAIT | MCI_ALLOCATE_MEMORY,
-                      &(_this->hidden->BufferParms), 0);
-  if ((rc!=MCIERR_SUCCESS) || (iNumBufs != _this->hidden->BufferParms.ulNumBuffers) || (_this->hidden->BufferParms.ulBufferSize==0))
-  { // Could not allocate memory!
-    // Close DART, and exit with error code!
-    SDL_free(_this->hidden->pMixBuffers); _this->hidden->pMixBuffers = NULL;
-    mciSendCommand(iDeviceOrd, MCI_CLOSE, MCI_WAIT, &GenericParms, 0);
-    SDL_SetError("DART could not allocate buffers");
-    return(-1);
-  }
-  // Ok, we have all the buffers allocated, let's mark them!
-  {
-    int i;
-    for (i=0; i<iNumBufs; i++)
-    {
-      pMixBufferDesc pBufferDesc = (pMixBufferDesc) SDL_malloc(sizeof(tMixBufferDesc));;
-      // Check if this buffer was really allocated by DART
-      if ((!(_this->hidden->pMixBuffers[i].pBuffer)) || (!pBufferDesc))
-      { // Wrong buffer!
-        // Close DART, and exit with error code!
-        // Free buffer descriptions
-        { int j;
-          for (j=0; j<i; j++) SDL_free((void *)(_this->hidden->pMixBuffers[j].ulUserParm));
-        }
-        // and cleanup
-        mciSendCommand(iDeviceOrd, MCI_BUFFER, MCI_WAIT | MCI_DEALLOCATE_MEMORY, &(_this->hidden->BufferParms), 0);
-        SDL_free(_this->hidden->pMixBuffers); _this->hidden->pMixBuffers = NULL;
-        mciSendCommand(iDeviceOrd, MCI_CLOSE, MCI_WAIT, &GenericParms, 0);
-        SDL_SetError("Error at internal buffer check");
-        return(-1);
-      }
-      pBufferDesc->iBufferUsage = BUFFER_EMPTY;
-      pBufferDesc->pSDLAudioDevice = _this;
-
-      _this->hidden->pMixBuffers[i].ulBufferLength = _this->hidden->BufferParms.ulBufferSize;
-      _this->hidden->pMixBuffers[i].ulUserParm = (ULONG) pBufferDesc; // User parameter: Description of buffer
-      _this->hidden->pMixBuffers[i].ulFlags = 0; // Some stuff should be flagged here for DART, like end of
-                                            // audio data, but as we will continously send
-                                            // audio data, there will be no end.:)
-      SDL_memset(_this->hidden->pMixBuffers[i].pBuffer, iSilence, iBufSize);
-    }
-  }
-  _this->hidden->iNextFreeBuffer = 0;
-  _this->hidden->iLastPlayedBuf = -1;
-  // Create event semaphore
-  if (DosCreateEventSem(NULL, &(_this->hidden->hevAudioBufferPlayed), 0, FALSE)!=NO_ERROR)
-  {
-    // Could not create event semaphore!
-    {
-      int i;
-      for (i=0; i<iNumBufs; i++) SDL_free((void *)(_this->hidden->pMixBuffers[i].ulUserParm));
-    }
-    mciSendCommand(iDeviceOrd, MCI_BUFFER, MCI_WAIT | MCI_DEALLOCATE_MEMORY, &(_this->hidden->BufferParms), 0);
-    SDL_free(_this->hidden->pMixBuffers); _this->hidden->pMixBuffers = NULL;
-    mciSendCommand(iDeviceOrd, MCI_CLOSE, MCI_WAIT, &GenericParms, 0);
-    SDL_SetError("Could not create event semaphore");
-    return(-1);
-  }
-
-  // Store the new settings in global variables
-  _this->hidden->iCurrDeviceOrd = iDeviceOrd;
-  _this->hidden->iCurrFreq = iFreq;
-  _this->hidden->iCurrBits = iBits;
-  _this->hidden->iCurrChannels = iChannels;
-  _this->hidden->iCurrNumBufs = iNumBufs;
-  _this->hidden->iCurrBufSize = iBufSize;
-
-  return (0);
-}
-
-
-
-void DART_ThreadInit(_THIS)
-{
-  return;
-}
-
-/* This function waits until it is possible to write a full sound buffer */
-void DART_WaitAudio(_THIS)
-{
-  int i;
-  pMixBufferDesc pBufDesc;
-  ULONG ulPostCount;
-
-  DosResetEventSem(_this->hidden->hevAudioBufferPlayed, &ulPostCount);
-  // If there is already an empty buffer, then return now!
-  for (i=0; i<_this->hidden->iCurrNumBufs; i++)
-  {
-    pBufDesc = (pMixBufferDesc) _this->hidden->pMixBuffers[i].ulUserParm;
-    if (pBufDesc->iBufferUsage == BUFFER_EMPTY)
-      return;
-  }
-  // If there is no empty buffer, wait for one to be empty!
-  DosWaitEventSem(_this->hidden->hevAudioBufferPlayed, 1000); // Wait max 1 sec!!! Important!
-  return;
-}
-
-void DART_PlayAudio(_THIS)
-{
-  int iFreeBuf = _this->hidden->iNextFreeBuffer;
-  pMixBufferDesc pBufDesc;
-
-  pBufDesc = (pMixBufferDesc) _this->hidden->pMixBuffers[iFreeBuf].ulUserParm;
-  pBufDesc->iBufferUsage = BUFFER_USED;
-  // Send it to DART to be queued
-  _this->hidden->MixSetupParms.pmixWrite(_this->hidden->MixSetupParms.ulMixHandle,
-                                        &(_this->hidden->pMixBuffers[iFreeBuf]), 1);
-
-  _this->hidden->iLastPlayedBuf = iFreeBuf;
-  iFreeBuf = (iFreeBuf+1) % _this->hidden->iCurrNumBufs;
-  _this->hidden->iNextFreeBuffer = iFreeBuf;
-}
-
-Uint8 *DART_GetAudioBuf(_THIS)
-{
-  int iFreeBuf;
-  Uint8 *pResult;
-  pMixBufferDesc pBufDesc;
-
-  if (_this)
-  {
-    if (_this->hidden)
-    {
-      iFreeBuf = _this->hidden->iNextFreeBuffer;
-      pBufDesc = (pMixBufferDesc) _this->hidden->pMixBuffers[iFreeBuf].ulUserParm;
-      
-      if (pBufDesc)
-      {
-        if (pBufDesc->iBufferUsage == BUFFER_EMPTY)
-        {
-          pResult = _this->hidden->pMixBuffers[iFreeBuf].pBuffer;
-          return pResult; 
-        }
-      } else
-        printf("[DART_GetAudioBuf] : ERROR! pBufDesc = %p\n", pBufDesc);
-    } else
-      printf("[DART_GetAudioBuf] : ERROR! _this->hidden = %p\n", _this->hidden);
-  } else
-    printf("[DART_GetAudioBuf] : ERROR! _this = %p\n", _this);
-  return NULL;
-}
-
-void DART_WaitDone(_THIS)
-{
-  pMixBufferDesc pBufDesc;
-  ULONG ulPostCount;
-  APIRET rc;
-
-  pBufDesc = (pMixBufferDesc) _this->hidden->pMixBuffers[_this->hidden->iLastPlayedBuf].ulUserParm;
-  rc = NO_ERROR;
-  while ((pBufDesc->iBufferUsage != BUFFER_EMPTY) && (rc==NO_ERROR))
-  {
-    DosResetEventSem(_this->hidden->hevAudioBufferPlayed, &ulPostCount);
-    rc = DosWaitEventSem(_this->hidden->hevAudioBufferPlayed, 1000); // 1 sec timeout! Important!
-  }
-}
-
-void DART_CloseAudio(_THIS)
-{
-  MCI_GENERIC_PARMS GenericParms;
-  int rc;
-
-  // Stop DART playback
-  rc = mciSendCommand(_this->hidden->iCurrDeviceOrd, MCI_STOP, MCI_WAIT, &GenericParms, 0);
-  if (rc!=MCIERR_SUCCESS)
-  {
-#ifdef SFX_DEBUG_BUILD
-    printf("Could not stop DART playback!\n");
-    fflush(stdout);
-#endif
-  }
-
-  // Close event semaphore
-  DosCloseEventSem(_this->hidden->hevAudioBufferPlayed);
-
-  // Free memory of buffer descriptions
-  {
-    int i;
-    for (i=0; i<_this->hidden->iCurrNumBufs; i++) SDL_free((void *)(_this->hidden->pMixBuffers[i].ulUserParm));
-  }
-
-  // Deallocate buffers
-  rc = mciSendCommand(_this->hidden->iCurrDeviceOrd, MCI_BUFFER, MCI_WAIT | MCI_DEALLOCATE_MEMORY, &(_this->hidden->BufferParms), 0);
-
-  // Free bufferlist
-  SDL_free(_this->hidden->pMixBuffers); _this->hidden->pMixBuffers = NULL;
-
-  // Close dart
-  rc = mciSendCommand(_this->hidden->iCurrDeviceOrd, MCI_CLOSE, MCI_WAIT, &(GenericParms), 0);
-}
-
-/* Audio driver bootstrap functions */
-
-int Audio_Available(void)
-{
-  return(1);
-}
-
-void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-  SDL_free(device->hidden);
-  SDL_free(device);
-}
-
-SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-  SDL_AudioDevice *this;
-
-  /* Initialize all variables that we clean on shutdown */
-  this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-  if ( this )
-  {
-    SDL_memset(this, 0, (sizeof *this));
-    this->hidden = (struct SDL_PrivateAudioData *)
-      SDL_malloc((sizeof *this->hidden));
-  }
-  if ( (this == NULL) || (this->hidden == NULL) )
-  {
-    SDL_OutOfMemory();
-    if ( this )
-      SDL_free(this);
-    return(0);
-  }
-  SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-  /* Set the function pointers */
-  this->OpenAudio = DART_OpenAudio;
-  this->ThreadInit = DART_ThreadInit;
-  this->WaitAudio = DART_WaitAudio;
-  this->PlayAudio = DART_PlayAudio;
-  this->GetAudioBuf = DART_GetAudioBuf;
-  this->WaitDone = DART_WaitDone;
-  this->CloseAudio = DART_CloseAudio;
-
-  this->free = Audio_DeleteDevice;
-
-  return this;
-}
-
-AudioBootStrap DART_bootstrap = {
-	"dart", "OS/2 Direct Audio RouTines (DART)",
-	Audio_Available, Audio_CreateDevice
-};
-
diff --git a/apps/plugins/sdl/src/audio/dart/SDL_dart.h b/apps/plugins/sdl/src/audio/dart/SDL_dart.h
deleted file mode 100644
index 68c27bd9d3..0000000000
--- a/apps/plugins/sdl/src/audio/dart/SDL_dart.h
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_lowaudio_h
-#define _SDL_lowaudio_h
-
-#define INCL_TYPES
-#define INCL_DOSSEMAPHORES
-#define INCL_DOSRESOURCES
-#define INCL_DOSMISC
-#define INCL_DOSERRORS
-
-#define INCL_OS2MM
-#define INCL_MMIOOS2
-#define INCL_MCIOS2
-#include <os2.h>
-#include <os2me.h>     // DART stuff and MMIO stuff
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the audio functions */
-#define _THIS	SDL_AudioDevice *_this
-
-/* The DirectSound objects */
-struct SDL_PrivateAudioData
-{
-  int iCurrDeviceOrd;
-  int iCurrFreq;
-  int iCurrBits;
-  int iCurrChannels;
-  int iCurrNumBufs;
-  int iCurrBufSize;
-
-  int iLastPlayedBuf;
-  int iNextFreeBuffer;
-
-  MCI_BUFFER_PARMS BufferParms;     // Sound buffer parameters
-  MCI_MIX_BUFFER *pMixBuffers;      // Sound buffers
-  MCI_MIXSETUP_PARMS MixSetupParms; // Mixer setup parameters
-  HEV hevAudioBufferPlayed;         // Event semaphore to indicate that an audio buffer has been played by DART
-};
-
-#endif /* _SDL_lowaudio_h */
diff --git a/apps/plugins/sdl/src/audio/dc/SDL_dcaudio.c b/apps/plugins/sdl/src/audio/dc/SDL_dcaudio.c
deleted file mode 100644
index 88daa8723a..0000000000
--- a/apps/plugins/sdl/src/audio/dc/SDL_dcaudio.c
+++ /dev/null
@@ -1,246 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-
-*/
-#include "SDL_config.h"
-
-/* Output dreamcast aica */
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_audiodev_c.h"
-#include "SDL_dcaudio.h"
-
-#include "aica.h"
-#include <dc/spu.h>
-
-/* Audio driver functions */
-static int DCAUD_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void DCAUD_WaitAudio(_THIS);
-static void DCAUD_PlayAudio(_THIS);
-static Uint8 *DCAUD_GetAudioBuf(_THIS);
-static void DCAUD_CloseAudio(_THIS);
-
-/* Audio driver bootstrap functions */
-static int DCAUD_Available(void)
-{
-	return 1;
-}
-
-static void DCAUD_DeleteDevice(SDL_AudioDevice *device)
-{
-	SDL_free(device->hidden);
-	SDL_free(device);
-}
-
-static SDL_AudioDevice *DCAUD_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-	if ( this ) {
-		SDL_memset(this, 0, (sizeof *this));
-		this->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *this->hidden));
-	}
-	if ( (this == NULL) || (this->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( this ) {
-			SDL_free(this);
-		}
-		return(0);
-	}
-	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-	/* Set the function pointers */
-	this->OpenAudio = DCAUD_OpenAudio;
-	this->WaitAudio = DCAUD_WaitAudio;
-	this->PlayAudio = DCAUD_PlayAudio;
-	this->GetAudioBuf = DCAUD_GetAudioBuf;
-	this->CloseAudio = DCAUD_CloseAudio;
-
-	this->free = DCAUD_DeleteDevice;
-
-	spu_init();
-
-	return this;
-}
-
-AudioBootStrap DCAUD_bootstrap = {
-	"dcaudio", "Dreamcast AICA audio",
-	DCAUD_Available, DCAUD_CreateDevice
-};
-
-/* This function waits until it is possible to write a full sound buffer */
-static void DCAUD_WaitAudio(_THIS)
-{
-	if (this->hidden->playing) {
-		/* wait */
-		while(aica_get_pos(0)/this->spec.samples == this->hidden->nextbuf) {
-			thd_pass();
-		}
-	}
-}
-
-#define	SPU_RAM_BASE	0xa0800000
-
-static void spu_memload_stereo8(int leftpos,int rightpos,void *src0,size_t size)
-{
-	uint8 *src = src0;
-	uint32 *left  = (uint32*)(leftpos +SPU_RAM_BASE);
-	uint32 *right = (uint32*)(rightpos+SPU_RAM_BASE);
-	size = (size+7)/8;
-	while(size--) {
-		unsigned lval,rval;
-		lval = *src++;
-		rval = *src++;
-		lval|= (*src++)<<8;
-		rval|= (*src++)<<8;
-		lval|= (*src++)<<16;
-		rval|= (*src++)<<16;
-		lval|= (*src++)<<24;
-		rval|= (*src++)<<24;
-		g2_write_32(left++,lval);
-		g2_write_32(right++,rval);
-		g2_fifo_wait();
-	}
-}
-
-static void spu_memload_stereo16(int leftpos,int rightpos,void *src0,size_t size)
-{
-	uint16 *src = src0;
-	uint32 *left  = (uint32*)(leftpos +SPU_RAM_BASE);
-	uint32 *right = (uint32*)(rightpos+SPU_RAM_BASE);
-	size = (size+7)/8;
-	while(size--) {
-		unsigned lval,rval;
-		lval = *src++;
-		rval = *src++;
-		lval|= (*src++)<<16;
-		rval|= (*src++)<<16;
-		g2_write_32(left++,lval);
-		g2_write_32(right++,rval);
-		g2_fifo_wait();
-	}
-}
-
-static void DCAUD_PlayAudio(_THIS)
-{
-	SDL_AudioSpec *spec = &this->spec;
-	unsigned int offset;
-
-	if (this->hidden->playing) {
-		/* wait */
-		while(aica_get_pos(0)/spec->samples == this->hidden->nextbuf) {
-			thd_pass();
-		}
-	}
-
-	offset = this->hidden->nextbuf*spec->size;
-	this->hidden->nextbuf^=1;
-	/* Write the audio data, checking for EAGAIN on broken audio drivers */
-	if (spec->channels==1) {
-		spu_memload(this->hidden->leftpos+offset,this->hidden->mixbuf,this->hidden->mixlen);
-	} else {
-		offset/=2;
-		if ((this->spec.format&255)==8) {
-			spu_memload_stereo8(this->hidden->leftpos+offset,this->hidden->rightpos+offset,this->hidden->mixbuf,this->hidden->mixlen);
-		} else {
-			spu_memload_stereo16(this->hidden->leftpos+offset,this->hidden->rightpos+offset,this->hidden->mixbuf,this->hidden->mixlen);
-		}
-	}
-
-	if (!this->hidden->playing) {
-		int mode;
-		this->hidden->playing = 1;
-		mode = (spec->format==AUDIO_S8)?SM_8BIT:SM_16BIT;
-		if (spec->channels==1) {
-			aica_play(0,mode,this->hidden->leftpos,0,spec->samples*2,spec->freq,255,128,1);
-		} else {
-			aica_play(0,mode,this->hidden->leftpos ,0,spec->samples*2,spec->freq,255,0,1);
-			aica_play(1,mode,this->hidden->rightpos,0,spec->samples*2,spec->freq,255,255,1);
-		}
-	}
-}
-
-static Uint8 *DCAUD_GetAudioBuf(_THIS)
-{
-	return(this->hidden->mixbuf);
-}
-
-static void DCAUD_CloseAudio(_THIS)
-{
-	aica_stop(0);
-	if (this->spec.channels==2) aica_stop(1);
-	if ( this->hidden->mixbuf != NULL ) {
-		SDL_FreeAudioMem(this->hidden->mixbuf);
-		this->hidden->mixbuf = NULL;
-	}
-}
-
-static int DCAUD_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-    Uint16 test_format = SDL_FirstAudioFormat(spec->format);
-    int valid_datatype = 0;
-    while ((!valid_datatype) && (test_format)) {
-        spec->format = test_format;
-        switch (test_format) {
-            /* only formats Dreamcast accepts... */
-            case AUDIO_S8:
-            case AUDIO_S16LSB:
-                valid_datatype = 1;
-                break;
-
-            default:
-                test_format = SDL_NextAudioFormat();
-                break;
-        }
-    }
-
-    if (!valid_datatype) {  /* shouldn't happen, but just in case... */
-        SDL_SetError("Unsupported audio format");
-        return (-1);
-    }
-
-    if (spec->channels > 2)
-        spec->channels = 2;  /* no more than stereo on the Dreamcast. */
-
-	/* Update the fragment size as size in bytes */
-	SDL_CalculateAudioSpec(spec);
-
-	/* Allocate mixing buffer */
-	this->hidden->mixlen = spec->size;
-	this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
-	if ( this->hidden->mixbuf == NULL ) {
-		return(-1);
-	}
-	SDL_memset(this->hidden->mixbuf, spec->silence, spec->size);
-	this->hidden->leftpos = 0x11000;
-	this->hidden->rightpos = 0x11000+spec->size;
-	this->hidden->playing = 0;
-	this->hidden->nextbuf = 0;
-
-	/* We're ready to rock and roll. :-) */
-	return(0);
-}
diff --git a/apps/plugins/sdl/src/audio/dc/SDL_dcaudio.h b/apps/plugins/sdl/src/audio/dc/SDL_dcaudio.h
deleted file mode 100644
index fba95b3eda..0000000000
--- a/apps/plugins/sdl/src/audio/dc/SDL_dcaudio.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_dcaudio_h
-#define _SDL_dcaudio_h
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData {
-	/* The file descriptor for the audio device */
-	Uint8 *mixbuf;
-	Uint32 mixlen;
-	int playing;
-	int leftpos,rightpos;
-	int nextbuf;
-};
-
-#endif /* _SDL_dcaudio_h */
diff --git a/apps/plugins/sdl/src/audio/dc/aica.c b/apps/plugins/sdl/src/audio/dc/aica.c
deleted file mode 100644
index b6a1c93644..0000000000
--- a/apps/plugins/sdl/src/audio/dc/aica.c
+++ /dev/null
@@ -1,271 +0,0 @@
-/* This file is part of the Dreamcast function library.
- * Please see libdream.c for further details.
- *
- * (c)2000 Dan Potter
- * modify BERO
- */
-#include "aica.h"
-
-#include <arch/irq.h>
-#include <dc/spu.h>
-
-/* #define dc_snd_base ((volatile unsigned char *)0x00800000) */ /* arm side */
-#define dc_snd_base ((volatile unsigned char *)0xa0700000) /* dc side */
-
-/* Some convienence macros */
-#define	SNDREGADDR(x)	(0xa0700000 + (x))
-#define	CHNREGADDR(ch,x)	SNDREGADDR(0x80*(ch)+(x))
-
-
-#define SNDREG32(x)	(*(volatile unsigned long *)SNDREGADDR(x))
-#define SNDREG8(x)	(*(volatile unsigned char *)SNDREGADDR(x))
-#define CHNREG32(ch, x) (*(volatile unsigned long *)CHNREGADDR(ch,x))
-#define CHNREG8(ch, x)	(*(volatile unsigned long *)CHNREGADDR(ch,x))
-
-#define G2_LOCK(OLD) \
-	do { \
-		if (!irq_inside_int()) \
-			OLD = irq_disable(); \
-		/* suspend any G2 DMA here... */ \
-		while((*(volatile unsigned int *)0xa05f688c) & 0x20) \
-			; \
-	} while(0)
-
-#define G2_UNLOCK(OLD) \
-	do { \
-		/* resume any G2 DMA here... */ \
-		if (!irq_inside_int()) \
-			irq_restore(OLD); \
-	} while(0)
-
-
-void aica_init() {
-	int i, j, old = 0;
-	
-	/* Initialize AICA channels */	
-	G2_LOCK(old);
-	SNDREG32(0x2800) = 0x0000;
-	
-	for (i=0; i<64; i++) {
-		for (j=0; j<0x80; j+=4) {
-			if ((j&31)==0) g2_fifo_wait();
-			CHNREG32(i, j) = 0;
-		}
-		g2_fifo_wait();
-		CHNREG32(i,0) = 0x8000;
-		CHNREG32(i,20) = 0x1f;
-	}
-
-	SNDREG32(0x2800) = 0x000f;
-	g2_fifo_wait();
-	G2_UNLOCK(old);
-}
-
-/* Translates a volume from linear form to logarithmic form (required by
-   the AICA chip */
-/* int logs[] = {
-
-0, 40, 50, 58, 63, 68, 73, 77, 80, 83, 86, 89, 92, 94, 97, 99, 101, 103,
-105, 107, 109, 111, 112, 114, 116, 117, 119, 120, 122, 123, 125, 126, 127,
-129, 130, 131, 133, 134, 135, 136, 137, 139, 140, 141, 142, 143, 144, 145,
-146, 147, 148, 149, 150, 151, 152, 153, 154, 155, 156, 156, 157, 158, 159,
-160, 161, 162, 162, 163, 164, 165, 166, 166, 167, 168, 169, 170, 170, 171,
-172, 172, 173, 174, 175, 175, 176, 177, 177, 178, 179, 180, 180, 181, 182,
-182, 183, 183, 184, 185, 185, 186, 187, 187, 188, 188, 189, 190, 190, 191,
-191, 192, 193, 193, 194, 194, 195, 196, 196, 197, 197, 198, 198, 199, 199,
-200, 201, 201, 202, 202, 203, 203, 204, 204, 205, 205, 206, 206, 207, 207,
-208, 208, 209, 209, 210, 210, 211, 211, 212, 212, 213, 213, 214, 214, 215,
-215, 216, 216, 217, 217, 217, 218, 218, 219, 219, 220, 220, 221, 221, 222,
-222, 222, 223, 223, 224, 224, 225, 225, 225, 226, 226, 227, 227, 228, 228,
-228, 229, 229, 230, 230, 230, 231, 231, 232, 232, 232, 233, 233, 234, 234,
-234, 235, 235, 236, 236, 236, 237, 237, 238, 238, 238, 239, 239, 240, 240,
-240, 241, 241, 241, 242, 242, 243, 243, 243, 244, 244, 244, 245, 245, 245,
-246, 246, 247, 247, 247, 248, 248, 248, 249, 249, 249, 250, 250, 250, 251,
-251, 251, 252, 252, 252, 253, 253, 253, 254, 254, 254, 255
-
-}; */
-
-const static unsigned char logs[] = {
-	0, 15, 22, 27, 31, 35, 39, 42, 45, 47, 50, 52, 55, 57, 59, 61,
-	63, 65, 67, 69, 71, 73, 74, 76, 78, 79, 81, 82, 84, 85, 87, 88,
-	90, 91, 92, 94, 95, 96, 98, 99, 100, 102, 103, 104, 105, 106,
-	108, 109, 110, 111, 112, 113, 114, 116, 117, 118, 119, 120, 121,
-	122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 134,
-	135, 136, 137, 138, 138, 139, 140, 141, 142, 143, 144, 145, 146,
-	146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 156,
-	157, 158, 159, 160, 160, 161, 162, 163, 164, 164, 165, 166, 167,
-	167, 168, 169, 170, 170, 171, 172, 173, 173, 174, 175, 176, 176,
-	177, 178, 178, 179, 180, 181, 181, 182, 183, 183, 184, 185, 185,
-	186, 187, 187, 188, 189, 189, 190, 191, 191, 192, 193, 193, 194,
-	195, 195, 196, 197, 197, 198, 199, 199, 200, 200, 201, 202, 202,
-	203, 204, 204, 205, 205, 206, 207, 207, 208, 209, 209, 210, 210,
-	211, 212, 212, 213, 213, 214, 215, 215, 216, 216, 217, 217, 218,
-	219, 219, 220, 220, 221, 221, 222, 223, 223, 224, 224, 225, 225,
-	226, 227, 227, 228, 228, 229, 229, 230, 230, 231, 232, 232, 233,
-	233, 234, 234, 235, 235, 236, 236, 237, 237, 238, 239, 239, 240,
-	240, 241, 241, 242, 242, 243, 243, 244, 244, 245, 245, 246, 246,
-	247, 247, 248, 248, 249, 249, 250, 250, 251, 251, 252, 252, 253, 254, 255
-};
-
-/* For the moment this is going to have to suffice, until we really
-   figure out what these mean. */
-#define AICA_PAN(x) ((x)==0x80?(0):((x)<0x80?(0x1f):(0x0f)))
-#define AICA_VOL(x) (0xff - logs[128 + (((x) & 0xff) / 2)])
-//#define AICA_VOL(x) (0xff - logs[x&255])
-
-static inline unsigned  AICA_FREQ(unsigned freq)	{
-	unsigned long freq_lo, freq_base = 5644800;
-	int freq_hi = 7;
-
-	/* Need to convert frequency to floating point format
-	   (freq_hi is exponent, freq_lo is mantissa)
-	   Formula is ferq = 44100*2^freq_hi*(1+freq_lo/1024) */
-	while (freq < freq_base && freq_hi > -8) {
-		freq_base >>= 1;
-		--freq_hi;
-	}
-	while (freq < freq_base && freq_hi > -8) {
-		freq_base >>= 1;
-		freq_hi--;
-	}
-	freq_lo = (freq<<10) / freq_base;
-	return (freq_hi << 11) | (freq_lo & 1023);
-}
-
-/* Sets up a sound channel completely. This is generally good if you want
-   a quick and dirty way to play notes. If you want a more comprehensive
-   set of routines (more like PC wavetable cards) see below.
-   
-   ch is the channel to play on (0 - 63)
-   smpptr is the pointer to the sound data; if you're running off the
-     SH4, then this ought to be (ptr - 0xa0800000); otherwise it's just
-     ptr. Basically, it's an offset into sound ram.
-   mode is one of the mode constants (16 bit, 8 bit, ADPCM)
-   nsamp is the number of samples to play (not number of bytes!)
-   freq is the sampling rate of the sound
-   vol is the volume, 0 to 0xff (0xff is louder)
-   pan is a panning constant -- 0 is left, 128 is center, 255 is right.
-
-   This routine (and the similar ones) owe a lot to Marcus' sound example -- 
-   I hadn't gotten quite this far into dissecting the individual regs yet. */
-void aica_play(int ch,int mode,unsigned long smpptr,int loopst,int loopend,int freq,int vol,int pan,int loopflag) {
-/*	int i;
-*/
-	int val;
-	int old = 0;
-
-	/* Stop the channel (if it's already playing) */
-	aica_stop(ch);
-	/* doesn't seem to be needed, but it's here just in case */
-/*
-	for (i=0; i<256; i++) {
-		asm("nop");
-		asm("nop");
-		asm("nop");
-		asm("nop");
-	}
-*/
-	G2_LOCK(old);
-	/* Envelope setup. The first of these is the loop point,
-	   e.g., where the sample starts over when it loops. The second
-	   is the loop end. This is the full length of the sample when
-	   you are not looping, or the loop end point when you are (though
-	   storing more than that is a waste of memory if you're not doing
-	   volume enveloping). */
-	CHNREG32(ch, 8) = loopst & 0xffff;
-	CHNREG32(ch, 12) = loopend & 0xffff;
-	
-	/* Write resulting values */
-	CHNREG32(ch, 24) = AICA_FREQ(freq);
-	
-	/* Set volume, pan, and some other things that we don't know what
-	   they do =) */
-	CHNREG32(ch, 36) = AICA_PAN(pan) | (0xf<<8);
-	/* Convert the incoming volume and pan into hardware values */
-	/* Vol starts at zero so we can ramp */
-	vol = AICA_VOL(vol);
-	CHNREG32(ch, 40) = 0x24 | (vol<<8);
-	/* Convert the incoming volume and pan into hardware values */
-	/* Vol starts at zero so we can ramp */
-
-	/* If we supported volume envelopes (which we don't yet) then
-	   this value would set that up. The top 4 bits determine the
-	   envelope speed. f is the fastest, 1 is the slowest, and 0
-	   seems to be an invalid value and does weird things). The
-	   default (below) sets it into normal mode (play and terminate/loop).
-	CHNREG32(ch, 16) = 0xf010;
-	*/
-	CHNREG32(ch, 16) = 0x1f;	/* No volume envelope */
-	
-	
-	/* Set sample format, buffer address, and looping control. If
-	   0x0200 mask is set on reg 0, the sample loops infinitely. If
-	   it's not set, the sample plays once and terminates. We'll
-	   also set the bits to start playback here. */
-	CHNREG32(ch, 4) = smpptr & 0xffff;
-	val = 0xc000 | 0x0000 | (mode<<7) | (smpptr >> 16);
-	if (loopflag) val|=0x200;
-	
-	CHNREG32(ch, 0) = val;
-	
-	G2_UNLOCK(old);
-
-	/* Enable playback */
-	/* CHNREG32(ch, 0) |= 0xc000; */
-	g2_fifo_wait();
-
-#if 0
-	for (i=0xff; i>=vol; i--) {
-		if ((i&7)==0) g2_fifo_wait();
-		CHNREG32(ch, 40) =  0x24 | (i<<8);;
-	}
-
-	g2_fifo_wait();
-#endif
-}
-
-/* Stop the sound on a given channel */
-void aica_stop(int ch) {
-	g2_write_32(CHNREGADDR(ch, 0),(g2_read_32(CHNREGADDR(ch, 0)) & ~0x4000) | 0x8000);
-	g2_fifo_wait();
-}
-
-
-/* The rest of these routines can change the channel in mid-stride so you
-   can do things like vibrato and panning effects. */
-   
-/* Set channel volume */
-void aica_vol(int ch,int vol) {
-//	g2_write_8(CHNREGADDR(ch, 41),AICA_VOL(vol));
-	g2_write_32(CHNREGADDR(ch, 40),(g2_read_32(CHNREGADDR(ch, 40))&0xffff00ff)|(AICA_VOL(vol)<<8) );
-	g2_fifo_wait();
-}
-
-/* Set channel pan */
-void aica_pan(int ch,int pan) {
-//	g2_write_8(CHNREGADDR(ch, 36),AICA_PAN(pan));
-	g2_write_32(CHNREGADDR(ch, 36),(g2_read_32(CHNREGADDR(ch, 36))&0xffffff00)|(AICA_PAN(pan)) );
-	g2_fifo_wait();
-}
-
-/* Set channel frequency */
-void aica_freq(int ch,int freq) {
-	g2_write_32(CHNREGADDR(ch, 24),AICA_FREQ(freq));
-	g2_fifo_wait();
-}
-
-/* Get channel position */
-int aica_get_pos(int ch) {
-#if 1
-	/* Observe channel ch */
-	g2_write_32(SNDREGADDR(0x280c),(g2_read_32(SNDREGADDR(0x280c))&0xffff00ff) | (ch<<8));
-	g2_fifo_wait();
-	/* Update position counters */
-	return g2_read_32(SNDREGADDR(0x2814)) & 0xffff;
-#else
-	/* Observe channel ch */
-	g2_write_8(SNDREGADDR(0x280d),ch);
-	/* Update position counters */
-	return g2_read_32(SNDREGADDR(0x2814)) & 0xffff;
-#endif
-}
diff --git a/apps/plugins/sdl/src/audio/dc/aica.h b/apps/plugins/sdl/src/audio/dc/aica.h
deleted file mode 100644
index 2721e42821..0000000000
--- a/apps/plugins/sdl/src/audio/dc/aica.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _AICA_H_
-#define _AICA_H_
-
-#define	AICA_MEM	0xa0800000
-
-#define SM_8BIT		1
-#define SM_16BIT	0
-#define SM_ADPCM	2
-
-void aica_play(int ch,int mode,unsigned long smpptr,int looptst,int loopend,int freq,int vol,int pan,int loopflag);
-void aica_stop(int ch);
-void aica_vol(int ch,int vol);
-void aica_pan(int ch,int pan);
-void aica_freq(int ch,int freq);
-int aica_get_pos(int ch);
-
-#endif
diff --git a/apps/plugins/sdl/src/audio/disk/SDL_diskaudio.c b/apps/plugins/sdl/src/audio/disk/SDL_diskaudio.c
deleted file mode 100644
index c45d3f8c05..0000000000
--- a/apps/plugins/sdl/src/audio/disk/SDL_diskaudio.c
+++ /dev/null
@@ -1,186 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-
-    This file written by Ryan C. Gordon (icculus@icculus.org)
-*/
-#include "SDL_config.h"
-
-/* Output raw audio data to a file. */
-
-#if HAVE_STDIO_H
-#include <stdio.h>
-#endif
-
-#include "SDL_rwops.h"
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_audiodev_c.h"
-#include "SDL_diskaudio.h"
-
-/* The tag name used by DISK audio */
-#define DISKAUD_DRIVER_NAME         "disk"
-
-/* environment variables and defaults. */
-#define DISKENVR_OUTFILE         "SDL_DISKAUDIOFILE"
-#define DISKDEFAULT_OUTFILE      "/sdlaudio.raw"
-#define DISKENVR_WRITEDELAY      "SDL_DISKAUDIODELAY"
-#define DISKDEFAULT_WRITEDELAY   150
-
-/* Audio driver functions */
-static int DISKAUD_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void DISKAUD_WaitAudio(_THIS);
-static void DISKAUD_PlayAudio(_THIS);
-static Uint8 *DISKAUD_GetAudioBuf(_THIS);
-static void DISKAUD_CloseAudio(_THIS);
-
-static const char *DISKAUD_GetOutputFilename(void)
-{
-	const char *envr = SDL_getenv(DISKENVR_OUTFILE);
-	return((envr != NULL) ? envr : DISKDEFAULT_OUTFILE);
-}
-
-/* Audio driver bootstrap functions */
-static int DISKAUD_Available(void)
-{
-    //const char *envr = SDL_getenv("SDL_AUDIODRIVER");
-//	if (envr && (SDL_strcmp(envr, DISKAUD_DRIVER_NAME) == 0)) {
-		return(1);
-//	}
-//	return(0);
-}
-
-static void DISKAUD_DeleteDevice(SDL_AudioDevice *device)
-{
-	SDL_free(device->hidden);
-	SDL_free(device);
-}
-
-static SDL_AudioDevice *DISKAUD_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-	const char *envr;
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-	if ( this ) {
-		SDL_memset(this, 0, (sizeof *this));
-		this->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *this->hidden));
-	}
-	if ( (this == NULL) || (this->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( this ) {
-			SDL_free(this);
-		}
-		return(0);
-	}
-	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-	envr = SDL_getenv(DISKENVR_WRITEDELAY);
-	this->hidden->write_delay = (envr) ? SDL_atoi(envr) : DISKDEFAULT_WRITEDELAY;
-
-	/* Set the function pointers */
-	this->OpenAudio = DISKAUD_OpenAudio;
-	this->WaitAudio = DISKAUD_WaitAudio;
-	this->PlayAudio = DISKAUD_PlayAudio;
-	this->GetAudioBuf = DISKAUD_GetAudioBuf;
-	this->CloseAudio = DISKAUD_CloseAudio;
-
-	this->free = DISKAUD_DeleteDevice;
-
-	return this;
-}
-
-AudioBootStrap DISKAUD_bootstrap = {
-	DISKAUD_DRIVER_NAME, "direct-to-disk audio",
-	DISKAUD_Available, DISKAUD_CreateDevice
-};
-
-/* This function waits until it is possible to write a full sound buffer */
-static void DISKAUD_WaitAudio(_THIS)
-{
-	SDL_Delay(this->hidden->write_delay);
-}
-
-static void DISKAUD_PlayAudio(_THIS)
-{
-	int written;
-
-	/* Write the audio data */
-	written = SDL_RWwrite(this->hidden->output,
-                        this->hidden->mixbuf, 1,
-                        this->hidden->mixlen);
-
-	/* If we couldn't write, assume fatal error for now */
-	if ( (Uint32)written != this->hidden->mixlen ) {
-		this->enabled = 0;
-	}
-#ifdef DEBUG_AUDIO
-	fprintf(stderr, "Wrote %d bytes of audio data\n", written);
-#endif
-}
-
-static Uint8 *DISKAUD_GetAudioBuf(_THIS)
-{
-	return(this->hidden->mixbuf);
-}
-
-static void DISKAUD_CloseAudio(_THIS)
-{
-	if ( this->hidden->mixbuf != NULL ) {
-		SDL_FreeAudioMem(this->hidden->mixbuf);
-		this->hidden->mixbuf = NULL;
-	}
-	if ( this->hidden->output != NULL ) {
-		SDL_RWclose(this->hidden->output);
-		this->hidden->output = NULL;
-	}
-}
-
-static int DISKAUD_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	const char *fname = DISKAUD_GetOutputFilename();
-
-	/* Open the audio device */
-	this->hidden->output = SDL_RWFromFile(fname, "wb");
-	if ( this->hidden->output == NULL ) {
-		return(-1);
-	}
-
-#if HAVE_STDIO_H
-	fprintf(stderr, "WARNING: You are using the SDL disk writer"
-                    " audio driver!\n Writing to file [%s].\n", fname);
-#endif
-
-	/* Allocate mixing buffer */
-	this->hidden->mixlen = spec->size;
-	this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
-	if ( this->hidden->mixbuf == NULL ) {
-		return(-1);
-	}
-	SDL_memset(this->hidden->mixbuf, spec->silence, spec->size);
-
-	/* We're ready to rock and roll. :-) */
-	return(0);
-}
-
diff --git a/apps/plugins/sdl/src/audio/disk/SDL_diskaudio.h b/apps/plugins/sdl/src/audio/disk/SDL_diskaudio.h
deleted file mode 100644
index 24d7c9e34d..0000000000
--- a/apps/plugins/sdl/src/audio/disk/SDL_diskaudio.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_diskaudio_h
-#define _SDL_diskaudio_h
-
-#include "SDL_rwops.h"
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData {
-	/* The file descriptor for the audio device */
-	SDL_RWops *output;
-	Uint8 *mixbuf;
-	Uint32 mixlen;
-	Uint32 write_delay;
-};
-
-#endif /* _SDL_diskaudio_h */
diff --git a/apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.c b/apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.c
deleted file mode 100644
index 39f81d90ce..0000000000
--- a/apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.c
+++ /dev/null
@@ -1,455 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/* Allow access to a raw mixing buffer */
-
-#include <stdio.h>
-#include <string.h>	/* For strerror() */
-#include <errno.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include <signal.h>
-#include <sys/types.h>
-#include <sys/time.h>
-#include <sys/ioctl.h>
-#include <sys/stat.h>
-#include <sys/mman.h>
-
-#if SDL_AUDIO_DRIVER_OSS_SOUNDCARD_H
-/* This is installed on some systems */
-#include <soundcard.h>
-#else
-/* This is recommended by OSS */
-#include <sys/soundcard.h>
-#endif
-
-#ifndef MAP_FAILED
-#define MAP_FAILED	((Uint8 *)-1)
-#endif
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_audiodev_c.h"
-#include "SDL_dmaaudio.h"
-
-/* The tag name used by DMA audio */
-#define DMA_DRIVER_NAME         "dma"
-
-/* Open the audio device for playback, and don't block if busy */
-#define OPEN_FLAGS	(O_RDWR|O_NONBLOCK)
-
-/* Audio driver functions */
-static int DMA_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void DMA_WaitAudio(_THIS);
-static void DMA_PlayAudio(_THIS);
-static Uint8 *DMA_GetAudioBuf(_THIS);
-static void DMA_CloseAudio(_THIS);
-
-/* Audio driver bootstrap functions */
-
-static int Audio_Available(void)
-{
-	int available;
-	int fd;
-
-	available = 0;
-
-	fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0);
-	if ( fd >= 0 ) {
-		int caps;
-		struct audio_buf_info info;
-
-		if ( (ioctl(fd, SNDCTL_DSP_GETCAPS, &caps) == 0) &&
-	             (caps & DSP_CAP_TRIGGER) && (caps & DSP_CAP_MMAP) &&
-		     (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info) == 0) ) {
-			available = 1;
-		}
-		close(fd);
-	}
-	return(available);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-	SDL_free(device->hidden);
-	SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-	if ( this ) {
-		SDL_memset(this, 0, (sizeof *this));
-		this->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *this->hidden));
-	}
-	if ( (this == NULL) || (this->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( this ) {
-			SDL_free(this);
-		}
-		return(0);
-	}
-	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-	audio_fd = -1;
-
-	/* Set the function pointers */
-	this->OpenAudio = DMA_OpenAudio;
-	this->WaitAudio = DMA_WaitAudio;
-	this->PlayAudio = DMA_PlayAudio;
-	this->GetAudioBuf = DMA_GetAudioBuf;
-	this->CloseAudio = DMA_CloseAudio;
-
-	this->free = Audio_DeleteDevice;
-
-	return this;
-}
-
-AudioBootStrap DMA_bootstrap = {
-	DMA_DRIVER_NAME, "OSS /dev/dsp DMA audio",
-	Audio_Available, Audio_CreateDevice
-};
-
-/* This function waits until it is possible to write a full sound buffer */
-static void DMA_WaitAudio(_THIS)
-{
-	fd_set fdset;
-
-	/* Check to see if the thread-parent process is still alive */
-	{ static int cnt = 0;
-		/* Note that this only works with thread implementations 
-		   that use a different process id for each thread.
-		*/
-		if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */
-			if ( kill(parent, 0) < 0 ) {
-				this->enabled = 0;
-			}
-		}
-	}
-
-	/* See if we need to use timed audio synchronization */
-	if ( frame_ticks ) {
-		/* Use timer for general audio synchronization */
-		Sint32 ticks;
-
-		ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS;
-		if ( ticks > 0 ) {
-			SDL_Delay(ticks);
-		}
-	} else {
-		/* Use select() for audio synchronization */
-		struct timeval timeout;
-		FD_ZERO(&fdset);
-		FD_SET(audio_fd, &fdset);
-		timeout.tv_sec = 10;
-		timeout.tv_usec = 0;
-#ifdef DEBUG_AUDIO
-		fprintf(stderr, "Waiting for audio to get ready\n");
-#endif
-		if ( select(audio_fd+1, NULL, &fdset, NULL, &timeout) <= 0 ) {
-			const char *message =
-#ifdef AUDIO_OSPACE_HACK
-			"Audio timeout - buggy audio driver? (trying ospace)";
-#else
-			"Audio timeout - buggy audio driver? (disabled)";
-#endif
-			/* In general we should never print to the screen,
-			   but in this case we have no other way of letting
-			   the user know what happened.
-			*/
-			fprintf(stderr, "SDL: %s\n", message);
-#ifdef AUDIO_OSPACE_HACK
-			/* We may be able to use GET_OSPACE trick */
-			frame_ticks = (float)(this->spec->samples*1000) /
-			                      this->spec->freq;
-			next_frame = SDL_GetTicks()+frame_ticks;
-#else
-			this->enabled = 0;
-			/* Don't try to close - may hang */
-			audio_fd = -1;
-#ifdef DEBUG_AUDIO
-			fprintf(stderr, "Done disabling audio\n");
-#endif
-#endif /* AUDIO_OSPACE_HACK */
-		}
-#ifdef DEBUG_AUDIO
-		fprintf(stderr, "Ready!\n");
-#endif
-	}
-}
-
-static void DMA_PlayAudio(_THIS)
-{
-	/* If timer synchronization is enabled, set the next write frame */
-	if ( frame_ticks ) {
-		next_frame += frame_ticks;
-	}
-	return;
-}
-
-static Uint8 *DMA_GetAudioBuf(_THIS)
-{
-	count_info info;
-	int playing;
-	int filling;
-
-	/* Get number of blocks, looping if we're not using select() */
-	do {
-		if ( ioctl(audio_fd, SNDCTL_DSP_GETOPTR, &info) < 0 ) {
-			/* Uh oh... */
-			this->enabled = 0;
-			return(NULL);
-		}
-	} while ( frame_ticks && (info.blocks < 1) );
-#ifdef DEBUG_AUDIO
-	if ( info.blocks > 1 ) {
-		printf("Warning: audio underflow (%d frags)\n", info.blocks-1);
-	}
-#endif
-	playing = info.ptr / this->spec.size;
-	filling = (playing + 1)%num_buffers;
-	return (dma_buf + (filling * this->spec.size));
-}
-
-static void DMA_CloseAudio(_THIS)
-{
-	if ( dma_buf != NULL ) {
-		munmap(dma_buf, dma_len);
-		dma_buf = NULL;
-	}
-	if ( audio_fd >= 0 ) {
-		close(audio_fd);
-		audio_fd = -1;
-	}
-}
-
-static int DMA_ReopenAudio(_THIS, const char *audiodev, int format, int stereo,
-							SDL_AudioSpec *spec)
-{
-	int frag_spec;
-	int value;
-
-	/* Close and then reopen the audio device */
-	close(audio_fd);
-	audio_fd = open(audiodev, O_RDWR, 0);
-	if ( audio_fd < 0 ) {
-		SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
-		return(-1);
-	}
-
-	/* Calculate the final parameters for this audio specification */
-	SDL_CalculateAudioSpec(spec);
-
-	/* Determine the power of two of the fragment size */
-	for ( frag_spec = 0; (0x01<<frag_spec) < spec->size; ++frag_spec );
-	if ( (0x01<<frag_spec) != spec->size ) {
-		SDL_SetError("Fragment size must be a power of two");
-		return(-1);
-	}
-
-	/* Set the audio buffering parameters */
-	if ( ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_spec) < 0 ) {
-		SDL_SetError("Couldn't set audio fragment spec");
-		return(-1);
-	}
-
-	/* Set the audio format */
-	value = format;
-	if ( (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) ||
-						(value != format) ) {
-		SDL_SetError("Couldn't set audio format");
-		return(-1);
-	}
-
-	/* Set mono or stereo audio */
-	value = (spec->channels > 1);
-	if ( (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) < 0) ||
-						(value != stereo) ) {
-		SDL_SetError("Couldn't set audio channels");
-		return(-1);
-	}
-
-	/* Set the DSP frequency */
-	value = spec->freq;
-	if ( ioctl(audio_fd, SNDCTL_DSP_SPEED, &value) < 0 ) {
-		SDL_SetError("Couldn't set audio frequency");
-		return(-1);
-	}
-	spec->freq = value;
-
-	/* We successfully re-opened the audio */
-	return(0);
-}
-
-static int DMA_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	char audiodev[1024];
-	int format;
-	int stereo;
-	int value;
-	Uint16 test_format;
-	struct audio_buf_info info;
-
-	/* Reset the timer synchronization flag */
-	frame_ticks = 0.0;
-
-	/* Open the audio device */
-	audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
-	if ( audio_fd < 0 ) {
-		SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
-		return(-1);
-	}
-	dma_buf = NULL;
-	ioctl(audio_fd, SNDCTL_DSP_RESET, 0);
-
-	/* Get a list of supported hardware formats */
-	if ( ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &value) < 0 ) {
-		SDL_SetError("Couldn't get audio format list");
-		return(-1);
-	}
-
-	/* Try for a closest match on audio format */
-	format = 0;
-	for ( test_format = SDL_FirstAudioFormat(spec->format);
-						! format && test_format; ) {
-#ifdef DEBUG_AUDIO
-		fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
-#endif
-		switch ( test_format ) {
-			case AUDIO_U8:
-				if ( value & AFMT_U8 ) {
-					format = AFMT_U8;
-				}
-				break;
-			case AUDIO_S8:
-				if ( value & AFMT_S8 ) {
-					format = AFMT_S8;
-				}
-				break;
-			case AUDIO_S16LSB:
-				if ( value & AFMT_S16_LE ) {
-					format = AFMT_S16_LE;
-				}
-				break;
-			case AUDIO_S16MSB:
-				if ( value & AFMT_S16_BE ) {
-					format = AFMT_S16_BE;
-				}
-				break;
-			case AUDIO_U16LSB:
-				if ( value & AFMT_U16_LE ) {
-					format = AFMT_U16_LE;
-				}
-				break;
-			case AUDIO_U16MSB:
-				if ( value & AFMT_U16_BE ) {
-					format = AFMT_U16_BE;
-				}
-				break;
-			default:
-				format = 0;
-				break;
-		}
-		if ( ! format ) {
-			test_format = SDL_NextAudioFormat();
-		}
-	}
-	if ( format == 0 ) {
-		SDL_SetError("Couldn't find any hardware audio formats");
-		return(-1);
-	}
-	spec->format = test_format;
-
-	/* Set the audio format */
-	value = format;
-	if ( (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) ||
-						(value != format) ) {
-		SDL_SetError("Couldn't set audio format");
-		return(-1);
-	}
-
-	/* Set mono or stereo audio (currently only two channels supported) */
-	stereo = (spec->channels > 1);
-	ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
-	if ( stereo ) {
-		spec->channels = 2;
-	} else {
-		spec->channels = 1;
-	}
-
-	/* Because some drivers don't allow setting the buffer size
-	   after setting the format, we must re-open the audio device
-	   once we know what format and channels are supported
-	 */
-	if ( DMA_ReopenAudio(this, audiodev, format, stereo, spec) < 0 ) {
-		/* Error is set by DMA_ReopenAudio() */
-		return(-1);
-	}
-
-	/* Memory map the audio buffer */
-	if ( ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &info) < 0 ) {
-		SDL_SetError("Couldn't get OSPACE parameters");
-		return(-1);
-	}
-	spec->size = info.fragsize;
-	spec->samples = spec->size / ((spec->format & 0xFF) / 8);
-	spec->samples /= spec->channels;
-	num_buffers = info.fragstotal;
-	dma_len = num_buffers*spec->size;
-	dma_buf = (Uint8 *)mmap(NULL, dma_len, PROT_WRITE, MAP_SHARED,
-							audio_fd, 0);
-	if ( dma_buf == MAP_FAILED ) {
-		SDL_SetError("DMA memory map failed");
-		dma_buf = NULL;
-		return(-1);
-	}
-	SDL_memset(dma_buf, spec->silence, dma_len);
-
-	/* Check to see if we need to use select() workaround */
-	{ char *workaround;
-		workaround = SDL_getenv("SDL_DSP_NOSELECT");
-		if ( workaround ) {
-			frame_ticks = (float)(spec->samples*1000)/spec->freq;
-			next_frame = SDL_GetTicks()+frame_ticks;
-		}
-	}
-
-	/* Trigger audio playback */
-	value = 0;
-	ioctl(audio_fd, SNDCTL_DSP_SETTRIGGER, &value);
-	value = PCM_ENABLE_OUTPUT;
-	if ( ioctl(audio_fd, SNDCTL_DSP_SETTRIGGER, &value) < 0 ) {
-		SDL_SetError("Couldn't trigger audio output");
-		return(-1);
-	}
-
-	/* Get the parent process id (we're the parent of the audio thread) */
-	parent = getpid();
-
-	/* We're ready to rock and roll. :-) */
-	return(0);
-}
diff --git a/apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.h b/apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.h
deleted file mode 100644
index 9a45f732a1..0000000000
--- a/apps/plugins/sdl/src/audio/dma/SDL_dmaaudio.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_dspaudio_h
-#define _SDL_dspaudio_h
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData {
-	/* The file descriptor for the audio device */
-	int audio_fd;
-
-	/* The parent process id, to detect when application quits */
-	pid_t parent;
-
-	/* Raw mixing buffer */
-	Uint8 *dma_buf;
-	int    dma_len;
-	int num_buffers;
-
-	/* Support for audio timing using a timer, in addition to select() */
-	float frame_ticks;
-	float next_frame;
-};
-#define FUDGE_TICKS	10	/* The scheduler overhead ticks per frame */
-
-/* Old variable names */
-#define audio_fd		(this->hidden->audio_fd)
-#define parent			(this->hidden->parent)
-#define dma_buf			(this->hidden->dma_buf)
-#define dma_len			(this->hidden->dma_len)
-#define num_buffers		(this->hidden->num_buffers)
-#define frame_ticks		(this->hidden->frame_ticks)
-#define next_frame		(this->hidden->next_frame)
-
-#endif /* _SDL_dspaudio_h */
diff --git a/apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.c b/apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.c
deleted file mode 100644
index 1dcd2421ec..0000000000
--- a/apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.c
+++ /dev/null
@@ -1,242 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/* Allow access to a raw mixing buffer (For IRIX 6.5 and higher) */
-/* patch for IRIX 5 by Georg Schwarz 18/07/2004 */
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "SDL_irixaudio.h"
-
-
-#ifndef AL_RESOURCE /* as a test whether we use the old IRIX audio libraries */
-#define OLD_IRIX_AUDIO
-#define alClosePort(x) ALcloseport(x)
-#define alFreeConfig(x) ALfreeconfig(x)
-#define alGetFillable(x) ALgetfillable(x)
-#define alNewConfig() ALnewconfig()
-#define alOpenPort(x,y,z) ALopenport(x,y,z)
-#define alSetChannels(x,y) ALsetchannels(x,y)
-#define alSetQueueSize(x,y) ALsetqueuesize(x,y)
-#define alSetSampFmt(x,y) ALsetsampfmt(x,y)
-#define alSetWidth(x,y) ALsetwidth(x,y)
-#endif
-
-/* Audio driver functions */
-static int AL_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void AL_WaitAudio(_THIS);
-static void AL_PlayAudio(_THIS);
-static Uint8 *AL_GetAudioBuf(_THIS);
-static void AL_CloseAudio(_THIS);
-
-/* Audio driver bootstrap functions */
-
-static int Audio_Available(void)
-{
-	return 1;
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-	SDL_free(device->hidden);
-	SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-	if ( this ) {
-		SDL_memset(this, 0, (sizeof *this));
-		this->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *this->hidden));
-	}
-	if ( (this == NULL) || (this->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( this ) {
-			SDL_free(this);
-		}
-		return(0);
-	}
-	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-	/* Set the function pointers */
-	this->OpenAudio = AL_OpenAudio;
-	this->WaitAudio = AL_WaitAudio;
-	this->PlayAudio = AL_PlayAudio;
-	this->GetAudioBuf = AL_GetAudioBuf;
-	this->CloseAudio = AL_CloseAudio;
-
-	this->free = Audio_DeleteDevice;
-
-	return this;
-}
-
-AudioBootStrap DMEDIA_bootstrap = {
-	"AL", "IRIX DMedia audio",
-	Audio_Available, Audio_CreateDevice
-};
-
-
-void static AL_WaitAudio(_THIS)
-{
-	Sint32 timeleft;
-
-	timeleft = this->spec.samples - alGetFillable(audio_port);
-	if ( timeleft > 0 ) {
-		timeleft /= (this->spec.freq/1000);
-		SDL_Delay((Uint32)timeleft);
-	}
-}
-
-static void AL_PlayAudio(_THIS)
-{
-	/* Write the audio data out */
-	if ( alWriteFrames(audio_port, mixbuf, this->spec.samples) < 0 ) {
-		/* Assume fatal error, for now */
-		this->enabled = 0;
-	}
-}
-
-static Uint8 *AL_GetAudioBuf(_THIS)
-{
-	return(mixbuf);
-}
-
-static void AL_CloseAudio(_THIS)
-{
-	if ( mixbuf != NULL ) {
-		SDL_FreeAudioMem(mixbuf);
-		mixbuf = NULL;
-	}
-	if ( audio_port != NULL ) {
-		alClosePort(audio_port);
-		audio_port = NULL;
-	}
-}
-
-static int AL_OpenAudio(_THIS, SDL_AudioSpec * spec)
-{
-	Uint16 test_format = SDL_FirstAudioFormat(spec->format);
-	long width = 0;
-	long fmt = 0;
-	int valid = 0;
-
-#ifdef OLD_IRIX_AUDIO
-	{
-		long audio_param[2];
-		audio_param[0] = AL_OUTPUT_RATE;
-		audio_param[1] = spec->freq;
-		valid = (ALsetparams(AL_DEFAULT_DEVICE, audio_param, 2) < 0);
-	}
-#else
-	{
-		ALpv audio_param;
-		audio_param.param = AL_RATE;
-		audio_param.value.i = spec->freq;
-		valid = (alSetParams(AL_DEFAULT_OUTPUT, &audio_param, 1) < 0);
-	}
-#endif
-
-	while ((!valid) && (test_format)) {
-		valid = 1;
-		spec->format = test_format;
-
-		switch (test_format) {
-			case AUDIO_S8:
-				width = AL_SAMPLE_8;
-				fmt = AL_SAMPFMT_TWOSCOMP;
-				break;
-
-			case AUDIO_S16SYS:
-				width = AL_SAMPLE_16;
-				fmt = AL_SAMPFMT_TWOSCOMP;
-				break;
-
-			default:
-				valid = 0;
-				test_format = SDL_NextAudioFormat();
-				break;
-		}
-
-		if (valid) {
-			ALconfig audio_config = alNewConfig();
-			valid = 0;
-			if (audio_config) {
-				if (alSetChannels(audio_config, spec->channels) < 0) {
-					if (spec->channels > 2) {  /* can't handle > stereo? */
-						spec->channels = 2;  /* try again below. */
-					}
-				}
-
-				if ((alSetSampFmt(audio_config, fmt) >= 0) &&
-				    ((!width) || (alSetWidth(audio_config, width) >= 0)) &&
-				    (alSetQueueSize(audio_config, spec->samples * 2) >= 0) &&
-				    (alSetChannels(audio_config, spec->channels) >= 0)) {
-
-					audio_port = alOpenPort("SDL audio", "w", audio_config);
-					if (audio_port == NULL) {
-						/* docs say AL_BAD_CHANNELS happens here, too. */
-						int err = oserror();
-						if (err == AL_BAD_CHANNELS) {
-							spec->channels = 2;
-							alSetChannels(audio_config, spec->channels);
-							audio_port = alOpenPort("SDL audio", "w",
-							                        audio_config);
-						}
-					}
-
-					if (audio_port != NULL) {
-						valid = 1;
-					}
-				}
-
-				alFreeConfig(audio_config);
-			}
-		}
-	}
-
-	if (!valid) {
-		SDL_SetError("Unsupported audio format");
-		return (-1);
-	}
-
-	/* Update the fragment size as size in bytes */
-	SDL_CalculateAudioSpec(spec);
-
-	/* Allocate mixing buffer */
-	mixbuf = (Uint8 *) SDL_AllocAudioMem(spec->size);
-	if (mixbuf == NULL) {
-		SDL_OutOfMemory();
-		return (-1);
-	}
-	SDL_memset(mixbuf, spec->silence, spec->size);
-
-	/* We're ready to rock and roll. :-) */
-	return (0);
-}
-
diff --git a/apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.h b/apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.h
deleted file mode 100644
index c04f497cea..0000000000
--- a/apps/plugins/sdl/src/audio/dmedia/SDL_irixaudio.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_lowaudio_h
-#define _SDL_lowaudio_h
-
-#include <dmedia/audio.h>
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the audio functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData {
-	/* The handle for the audio device */
-	ALport audio_port;
-
-	Uint8 *mixbuf;           /* The app mixing buffer */
-};
-
-/* Old variable names */
-#define audio_port		(this->hidden->audio_port)
-#define mixbuf			(this->hidden->mixbuf)
-
-#endif /* _SDL_lowaudio_h */
diff --git a/apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.c b/apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.c
deleted file mode 100644
index 256c547f9b..0000000000
--- a/apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.c
+++ /dev/null
@@ -1,340 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-
-    Modified in Oct 2004 by Hannu Savolainen 
-    hannu@opensound.com
-*/
-#include "SDL_config.h"
-
-/* Allow access to a raw mixing buffer */
-
-#include <stdio.h>	/* For perror() */
-#include <string.h>	/* For strerror() */
-#include <errno.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include <signal.h>
-#include <sys/time.h>
-#include <sys/ioctl.h>
-#include <sys/stat.h>
-
-#if SDL_AUDIO_DRIVER_OSS_SOUNDCARD_H
-/* This is installed on some systems */
-#include <soundcard.h>
-#else
-/* This is recommended by OSS */
-#include <sys/soundcard.h>
-#endif
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_audiodev_c.h"
-#include "SDL_dspaudio.h"
-
-/* The tag name used by DSP audio */
-#define DSP_DRIVER_NAME         "dsp"
-
-/* Open the audio device for playback, and don't block if busy */
-#define OPEN_FLAGS	(O_WRONLY|O_NONBLOCK)
-
-/* Audio driver functions */
-static int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void DSP_WaitAudio(_THIS);
-static void DSP_PlayAudio(_THIS);
-static Uint8 *DSP_GetAudioBuf(_THIS);
-static void DSP_CloseAudio(_THIS);
-
-/* Audio driver bootstrap functions */
-
-static int Audio_Available(void)
-{
-	int fd;
-	int available;
-
-	available = 0;
-	fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0);
-	if ( fd >= 0 ) {
-		available = 1;
-		close(fd);
-	}
-	return(available);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-	SDL_free(device->hidden);
-	SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-	if ( this ) {
-		SDL_memset(this, 0, (sizeof *this));
-		this->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *this->hidden));
-	}
-	if ( (this == NULL) || (this->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( this ) {
-			SDL_free(this);
-		}
-		return(0);
-	}
-	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-	audio_fd = -1;
-
-	/* Set the function pointers */
-	this->OpenAudio = DSP_OpenAudio;
-	this->WaitAudio = DSP_WaitAudio;
-	this->PlayAudio = DSP_PlayAudio;
-	this->GetAudioBuf = DSP_GetAudioBuf;
-	this->CloseAudio = DSP_CloseAudio;
-
-	this->free = Audio_DeleteDevice;
-
-	return this;
-}
-
-AudioBootStrap DSP_bootstrap = {
-	DSP_DRIVER_NAME, "OSS /dev/dsp standard audio",
-	Audio_Available, Audio_CreateDevice
-};
-
-/* This function waits until it is possible to write a full sound buffer */
-static void DSP_WaitAudio(_THIS)
-{
-	/* Not needed at all since OSS handles waiting automagically */
-}
-
-static void DSP_PlayAudio(_THIS)
-{
-	if (write(audio_fd, mixbuf, mixlen)==-1)
-	{
-		perror("Audio write");
-		this->enabled = 0;
-	}
-
-#ifdef DEBUG_AUDIO
-	fprintf(stderr, "Wrote %d bytes of audio data\n", mixlen);
-#endif
-}
-
-static Uint8 *DSP_GetAudioBuf(_THIS)
-{
-	return(mixbuf);
-}
-
-static void DSP_CloseAudio(_THIS)
-{
-	if ( mixbuf != NULL ) {
-		SDL_FreeAudioMem(mixbuf);
-		mixbuf = NULL;
-	}
-	if ( audio_fd >= 0 ) {
-		close(audio_fd);
-		audio_fd = -1;
-	}
-}
-
-static int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	char audiodev[1024];
-	int format;
-	int value;
-	int frag_spec;
-	Uint16 test_format;
-
-    /* Make sure fragment size stays a power of 2, or OSS fails. */
-    /* I don't know which of these are actually legal values, though... */
-    if (spec->channels > 8)
-        spec->channels = 8;
-    else if (spec->channels > 4)
-        spec->channels = 4;
-    else if (spec->channels > 2)
-        spec->channels = 2;
-
-	/* Open the audio device */
-	audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
-	if ( audio_fd < 0 ) {
-		SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
-		return(-1);
-	}
-	mixbuf = NULL;
-
-	/* Make the file descriptor use blocking writes with fcntl() */
-	{ long flags;
-		flags = fcntl(audio_fd, F_GETFL);
-		flags &= ~O_NONBLOCK;
-		if ( fcntl(audio_fd, F_SETFL, flags) < 0 ) {
-			SDL_SetError("Couldn't set audio blocking mode");
-			DSP_CloseAudio(this);
-			return(-1);
-		}
-	}
-
-	/* Get a list of supported hardware formats */
-	if ( ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &value) < 0 ) {
-		perror("SNDCTL_DSP_GETFMTS");
-		SDL_SetError("Couldn't get audio format list");
-		DSP_CloseAudio(this);
-		return(-1);
-	}
-
-	/* Try for a closest match on audio format */
-	format = 0;
-	for ( test_format = SDL_FirstAudioFormat(spec->format);
-						! format && test_format; ) {
-#ifdef DEBUG_AUDIO
-		fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
-#endif
-		switch ( test_format ) {
-			case AUDIO_U8:
-				if ( value & AFMT_U8 ) {
-					format = AFMT_U8;
-				}
-				break;
-			case AUDIO_S16LSB:
-				if ( value & AFMT_S16_LE ) {
-					format = AFMT_S16_LE;
-				}
-				break;
-			case AUDIO_S16MSB:
-				if ( value & AFMT_S16_BE ) {
-					format = AFMT_S16_BE;
-				}
-				break;
-#if 0
-/*
- * These formats are not used by any real life systems so they are not 
- * needed here.
- */
-			case AUDIO_S8:
-				if ( value & AFMT_S8 ) {
-					format = AFMT_S8;
-				}
-				break;
-			case AUDIO_U16LSB:
-				if ( value & AFMT_U16_LE ) {
-					format = AFMT_U16_LE;
-				}
-				break;
-			case AUDIO_U16MSB:
-				if ( value & AFMT_U16_BE ) {
-					format = AFMT_U16_BE;
-				}
-				break;
-#endif
-			default:
-				format = 0;
-				break;
-		}
-		if ( ! format ) {
-			test_format = SDL_NextAudioFormat();
-		}
-	}
-	if ( format == 0 ) {
-		SDL_SetError("Couldn't find any hardware audio formats");
-		DSP_CloseAudio(this);
-		return(-1);
-	}
-	spec->format = test_format;
-
-	/* Set the audio format */
-	value = format;
-	if ( (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) ||
-						(value != format) ) {
-		perror("SNDCTL_DSP_SETFMT");
-		SDL_SetError("Couldn't set audio format");
-		DSP_CloseAudio(this);
-		return(-1);
-	}
-
-	/* Set the number of channels of output */
-	value = spec->channels;
-	if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &value) < 0 ) {
-		perror("SNDCTL_DSP_CHANNELS");
-		SDL_SetError("Cannot set the number of channels");
-		DSP_CloseAudio(this);
-		return(-1);
-	}
-	spec->channels = value;
-
-	/* Set the DSP frequency */
-	value = spec->freq;
-	if ( ioctl(audio_fd, SNDCTL_DSP_SPEED, &value) < 0 ) {
-		perror("SNDCTL_DSP_SPEED");
-		SDL_SetError("Couldn't set audio frequency");
-		DSP_CloseAudio(this);
-		return(-1);
-	}
-	spec->freq = value;
-
-	/* Calculate the final parameters for this audio specification */
-	SDL_CalculateAudioSpec(spec);
-
-	/* Determine the power of two of the fragment size */
-	for ( frag_spec = 0; (0x01U<<frag_spec) < spec->size; ++frag_spec );
-	if ( (0x01U<<frag_spec) != spec->size ) {
-		SDL_SetError("Fragment size must be a power of two");
-		DSP_CloseAudio(this);
-		return(-1);
-	}
-	frag_spec |= 0x00020000;	/* two fragments, for low latency */
-
-	/* Set the audio buffering parameters */
-#ifdef DEBUG_AUDIO
-	fprintf(stderr, "Requesting %d fragments of size %d\n",
-		(frag_spec >> 16), 1<<(frag_spec&0xFFFF));
-#endif
-	if ( ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_spec) < 0 ) {
-		perror("SNDCTL_DSP_SETFRAGMENT");
-	}
-#ifdef DEBUG_AUDIO
-	{ audio_buf_info info;
-	  ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &info);
-	  fprintf(stderr, "fragments = %d\n", info.fragments);
-	  fprintf(stderr, "fragstotal = %d\n", info.fragstotal);
-	  fprintf(stderr, "fragsize = %d\n", info.fragsize);
-	  fprintf(stderr, "bytes = %d\n", info.bytes);
-	}
-#endif
-
-	/* Allocate mixing buffer */
-	mixlen = spec->size;
-	mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
-	if ( mixbuf == NULL ) {
-		DSP_CloseAudio(this);
-		return(-1);
-	}
-	SDL_memset(mixbuf, spec->silence, spec->size);
-
-	/* Get the parent process id (we're the parent of the audio thread) */
-	parent = getpid();
-
-	/* We're ready to rock and roll. :-) */
-	return(0);
-}
diff --git a/apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.h b/apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.h
deleted file mode 100644
index 382544f967..0000000000
--- a/apps/plugins/sdl/src/audio/dsp/SDL_dspaudio.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_dspaudio_h
-#define _SDL_dspaudio_h
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData {
-	/* The file descriptor for the audio device */
-	int audio_fd;
-
-	/* The parent process id, to detect when application quits */
-	pid_t parent;
-
-	/* Raw mixing buffer */
-	Uint8 *mixbuf;
-	int    mixlen;
-};
-#define FUDGE_TICKS	10	/* The scheduler overhead ticks per frame */
-
-/* Old variable names */
-#define audio_fd		(this->hidden->audio_fd)
-#define parent			(this->hidden->parent)
-#define mixbuf			(this->hidden->mixbuf)
-#define mixlen			(this->hidden->mixlen)
-#define frame_ticks		(this->hidden->frame_ticks)
-#define next_frame		(this->hidden->next_frame)
-
-#endif /* _SDL_dspaudio_h */
diff --git a/apps/plugins/sdl/src/audio/esd/SDL_esdaudio.c b/apps/plugins/sdl/src/audio/esd/SDL_esdaudio.c
deleted file mode 100644
index f54b0ea9c5..0000000000
--- a/apps/plugins/sdl/src/audio/esd/SDL_esdaudio.c
+++ /dev/null
@@ -1,323 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/* Allow access to an ESD network stream mixing buffer */
-
-#include <sys/types.h>
-#include <unistd.h>
-#include <signal.h>
-#include <errno.h>
-#include <esd.h>
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_audiodev_c.h"
-#include "SDL_esdaudio.h"
-
-#ifdef SDL_AUDIO_DRIVER_ESD_DYNAMIC
-#include "SDL_name.h"
-#include "SDL_loadso.h"
-#else
-#define SDL_NAME(X)	X
-#endif
-
-/* The tag name used by ESD audio */
-#define ESD_DRIVER_NAME		"esd"
-
-/* Audio driver functions */
-static int ESD_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void ESD_WaitAudio(_THIS);
-static void ESD_PlayAudio(_THIS);
-static Uint8 *ESD_GetAudioBuf(_THIS);
-static void ESD_CloseAudio(_THIS);
-
-#ifdef SDL_AUDIO_DRIVER_ESD_DYNAMIC
-
-static const char *esd_library = SDL_AUDIO_DRIVER_ESD_DYNAMIC;
-static void *esd_handle = NULL;
-static int esd_loaded = 0;
-
-static int (*SDL_NAME(esd_open_sound))( const char *host );
-static int (*SDL_NAME(esd_close))( int esd );
-static int (*SDL_NAME(esd_play_stream))( esd_format_t format, int rate,
-                                         const char *host, const char *name );
-static struct {
-	const char *name;
-	void **func;
-} esd_functions[] = {
-	{ "esd_open_sound",	(void **)&SDL_NAME(esd_open_sound)	},
-	{ "esd_close",		(void **)&SDL_NAME(esd_close)		},
-	{ "esd_play_stream",	(void **)&SDL_NAME(esd_play_stream)	},
-};
-
-static void UnloadESDLibrary()
-{
-	if ( esd_loaded ) {
-		SDL_UnloadObject(esd_handle);
-		esd_handle = NULL;
-		esd_loaded = 0;
-	}
-}
-
-static int LoadESDLibrary(void)
-{
-	int i, retval = -1;
-
-	esd_handle = SDL_LoadObject(esd_library);
-	if ( esd_handle ) {
-		esd_loaded = 1;
-		retval = 0;
-		for ( i=0; i<SDL_arraysize(esd_functions); ++i ) {
-			*esd_functions[i].func = SDL_LoadFunction(esd_handle, esd_functions[i].name);
-			if ( !*esd_functions[i].func ) {
-				retval = -1;
-				UnloadESDLibrary();
-				break;
-			}
-		}
-	}
-	return retval;
-}
-
-#else
-
-static void UnloadESDLibrary()
-{
-	return;
-}
-
-static int LoadESDLibrary(void)
-{
-	return 0;
-}
-
-#endif /* SDL_AUDIO_DRIVER_ESD_DYNAMIC */
-
-/* Audio driver bootstrap functions */
-
-static int Audio_Available(void)
-{
-	int connection;
-	int available;
-
-	available = 0;
-	if ( LoadESDLibrary() < 0 ) {
-		return available;
-	}
-	connection = SDL_NAME(esd_open_sound)(NULL);
-	if ( connection >= 0 ) {
-		available = 1;
-		SDL_NAME(esd_close)(connection);
-	}
-	UnloadESDLibrary();
-	return(available);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-	SDL_free(device->hidden);
-	SDL_free(device);
-	UnloadESDLibrary();
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	LoadESDLibrary();
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-	if ( this ) {
-		SDL_memset(this, 0, (sizeof *this));
-		this->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *this->hidden));
-	}
-	if ( (this == NULL) || (this->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( this ) {
-			SDL_free(this);
-		}
-		return(0);
-	}
-	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-	audio_fd = -1;
-
-	/* Set the function pointers */
-	this->OpenAudio = ESD_OpenAudio;
-	this->WaitAudio = ESD_WaitAudio;
-	this->PlayAudio = ESD_PlayAudio;
-	this->GetAudioBuf = ESD_GetAudioBuf;
-	this->CloseAudio = ESD_CloseAudio;
-
-	this->free = Audio_DeleteDevice;
-
-	return this;
-}
-
-AudioBootStrap ESD_bootstrap = {
-	ESD_DRIVER_NAME, "Enlightened Sound Daemon",
-	Audio_Available, Audio_CreateDevice
-};
-
-/* This function waits until it is possible to write a full sound buffer */
-static void ESD_WaitAudio(_THIS)
-{
-	Sint32 ticks;
-
-	/* Check to see if the thread-parent process is still alive */
-	{ static int cnt = 0;
-		/* Note that this only works with thread implementations 
-		   that use a different process id for each thread.
-		*/
-		if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */
-			if ( kill(parent, 0) < 0 ) {
-				this->enabled = 0;
-			}
-		}
-	}
-
-	/* Use timer for general audio synchronization */
-	ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS;
-	if ( ticks > 0 ) {
-		SDL_Delay(ticks);
-	}
-}
-
-static void ESD_PlayAudio(_THIS)
-{
-	int written;
-
-	/* Write the audio data, checking for EAGAIN on broken audio drivers */
-	do {
-		written = write(audio_fd, mixbuf, mixlen);
-		if ( (written < 0) && ((errno == 0) || (errno == EAGAIN)) ) {
-			SDL_Delay(1);	/* Let a little CPU time go by */
-		}
-	} while ( (written < 0) && 
-	          ((errno == 0) || (errno == EAGAIN) || (errno == EINTR)) );
-
-	/* Set the next write frame */
-	next_frame += frame_ticks;
-
-	/* If we couldn't write, assume fatal error for now */
-	if ( written < 0 ) {
-		this->enabled = 0;
-	}
-}
-
-static Uint8 *ESD_GetAudioBuf(_THIS)
-{
-	return(mixbuf);
-}
-
-static void ESD_CloseAudio(_THIS)
-{
-	if ( mixbuf != NULL ) {
-		SDL_FreeAudioMem(mixbuf);
-		mixbuf = NULL;
-	}
-	if ( audio_fd >= 0 ) {
-		SDL_NAME(esd_close)(audio_fd);
-		audio_fd = -1;
-	}
-}
-
-/* Try to get the name of the program */
-static char *get_progname(void)
-{
-	char *progname = NULL;
-#ifdef __LINUX__
-	FILE *fp;
-	static char temp[BUFSIZ];
-
-	SDL_snprintf(temp, SDL_arraysize(temp), "/proc/%d/cmdline", getpid());
-	fp = fopen(temp, "r");
-	if ( fp != NULL ) {
-		if ( fgets(temp, sizeof(temp)-1, fp) ) {
-			progname = SDL_strrchr(temp, '/');
-			if ( progname == NULL ) {
-				progname = temp;
-			} else {
-				progname = progname+1;
-			}
-		}
-		fclose(fp);
-	}
-#endif
-	return(progname);
-}
-
-static int ESD_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	esd_format_t format;
-
-	/* Convert audio spec to the ESD audio format */
-	format = (ESD_STREAM | ESD_PLAY);
-	switch ( spec->format & 0xFF ) {
-		case 8:
-			format |= ESD_BITS8;
-			break;
-		case 16:
-			format |= ESD_BITS16;
-			break;
-		default:
-			SDL_SetError("Unsupported ESD audio format");
-			return(-1);
-	}
-	if ( spec->channels == 1 ) {
-		format |= ESD_MONO;
-	} else {
-		format |= ESD_STEREO;
-	}
-#if 0
-	spec->samples = ESD_BUF_SIZE;	/* Darn, no way to change this yet */
-#endif
-
-	/* Open a connection to the ESD audio server */
-	audio_fd = SDL_NAME(esd_play_stream)(format, spec->freq, NULL, get_progname());
-	if ( audio_fd < 0 ) {
-		SDL_SetError("Couldn't open ESD connection");
-		return(-1);
-	}
-
-	/* Calculate the final parameters for this audio specification */
-	SDL_CalculateAudioSpec(spec);
-	frame_ticks = (float)(spec->samples*1000)/spec->freq;
-	next_frame = SDL_GetTicks()+frame_ticks;
-
-	/* Allocate mixing buffer */
-	mixlen = spec->size;
-	mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
-	if ( mixbuf == NULL ) {
-		return(-1);
-	}
-	SDL_memset(mixbuf, spec->silence, spec->size);
-
-	/* Get the parent process id (we're the parent of the audio thread) */
-	parent = getpid();
-
-	/* We're ready to rock and roll. :-) */
-	return(0);
-}
diff --git a/apps/plugins/sdl/src/audio/esd/SDL_esdaudio.h b/apps/plugins/sdl/src/audio/esd/SDL_esdaudio.h
deleted file mode 100644
index da4ae6a04b..0000000000
--- a/apps/plugins/sdl/src/audio/esd/SDL_esdaudio.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_esdaudio_h
-#define _SDL_esdaudio_h
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData {
-	/* The file descriptor for the audio device */
-	int audio_fd;
-
-	/* The parent process id, to detect when application quits */
-	pid_t parent;
-
-	/* Raw mixing buffer */
-	Uint8 *mixbuf;
-	int    mixlen;
-
-	/* Support for audio timing using a timer */
-	float frame_ticks;
-	float next_frame;
-};
-#define FUDGE_TICKS	10	/* The scheduler overhead ticks per frame */
-
-/* Old variable names */
-#define audio_fd		(this->hidden->audio_fd)
-#define parent			(this->hidden->parent)
-#define mixbuf			(this->hidden->mixbuf)
-#define mixlen			(this->hidden->mixlen)
-#define frame_ticks		(this->hidden->frame_ticks)
-#define next_frame		(this->hidden->next_frame)
-
-#endif /* _SDL_esdaudio_h */
diff --git a/apps/plugins/sdl/src/audio/macosx/SDL_coreaudio.c b/apps/plugins/sdl/src/audio/macosx/SDL_coreaudio.c
deleted file mode 100644
index 31316d1fd9..0000000000
--- a/apps/plugins/sdl/src/audio/macosx/SDL_coreaudio.c
+++ /dev/null
@@ -1,291 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#include <CoreAudio/CoreAudio.h>
-#include <CoreServices/CoreServices.h>
-#include <AudioUnit/AudioUnit.h>
-#if MAC_OS_X_VERSION_MAX_ALLOWED <= 1050
-#include <AudioUnit/AUNTComponent.h>
-#endif
-
-#include "SDL_audio.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_sysaudio.h"
-#include "SDL_coreaudio.h"
-
-
-/* Audio driver functions */
-
-static int Core_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void Core_WaitAudio(_THIS);
-static void Core_PlayAudio(_THIS);
-static Uint8 *Core_GetAudioBuf(_THIS);
-static void Core_CloseAudio(_THIS);
-
-/* Audio driver bootstrap functions */
-
-static int Audio_Available(void)
-{
-    return(1);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-    SDL_free(device->hidden);
-    SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-    SDL_AudioDevice *this;
-
-    /* Initialize all variables that we clean on shutdown */
-    this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-    if ( this ) {
-        SDL_memset(this, 0, (sizeof *this));
-        this->hidden = (struct SDL_PrivateAudioData *)
-                SDL_malloc((sizeof *this->hidden));
-    }
-    if ( (this == NULL) || (this->hidden == NULL) ) {
-        SDL_OutOfMemory();
-        if ( this ) {
-            SDL_free(this);
-        }
-        return(0);
-    }
-    SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-    /* Set the function pointers */
-    this->OpenAudio = Core_OpenAudio;
-    this->WaitAudio = Core_WaitAudio;
-    this->PlayAudio = Core_PlayAudio;
-    this->GetAudioBuf = Core_GetAudioBuf;
-    this->CloseAudio = Core_CloseAudio;
-
-    this->free = Audio_DeleteDevice;
-
-    return this;
-}
-
-AudioBootStrap COREAUDIO_bootstrap = {
-    "coreaudio", "Mac OS X CoreAudio",
-    Audio_Available, Audio_CreateDevice
-};
-
-/* The CoreAudio callback */
-static OSStatus     audioCallback (void                            *inRefCon,
-                                   AudioUnitRenderActionFlags      *ioActionFlags,
-                                   const AudioTimeStamp            *inTimeStamp,
-                                   UInt32                          inBusNumber,
-                                   UInt32                          inNumberFrames,
-                                   AudioBufferList                 *ioData)
-{
-    SDL_AudioDevice *this = (SDL_AudioDevice *)inRefCon;
-    UInt32 remaining, len;
-    AudioBuffer *abuf;
-    void *ptr;
-    UInt32 i;
-
-    /* Only do anything if audio is enabled and not paused */
-    if ( ! this->enabled || this->paused ) {
-        for (i = 0; i < ioData->mNumberBuffers; i++) {
-            abuf = &ioData->mBuffers[i];
-            SDL_memset(abuf->mData, this->spec.silence, abuf->mDataByteSize);
-        }
-        return 0;
-    }
-    
-    /* No SDL conversion should be needed here, ever, since we accept
-       any input format in OpenAudio, and leave the conversion to CoreAudio.
-     */
-    /*
-    assert(!this->convert.needed);
-    assert(this->spec.channels == ioData->mNumberChannels);
-     */
-
-    for (i = 0; i < ioData->mNumberBuffers; i++) {
-        abuf = &ioData->mBuffers[i];
-        remaining = abuf->mDataByteSize;
-        ptr = abuf->mData;
-        while (remaining > 0) {
-            if (bufferOffset >= bufferSize) {
-                /* Generate the data */
-                SDL_memset(buffer, this->spec.silence, bufferSize);
-                SDL_mutexP(this->mixer_lock);
-                (*this->spec.callback)(this->spec.userdata,
-                            buffer, bufferSize);
-                SDL_mutexV(this->mixer_lock);
-                bufferOffset = 0;
-            }
-        
-            len = bufferSize - bufferOffset;
-            if (len > remaining)
-                len = remaining;
-            SDL_memcpy(ptr, (char *)buffer + bufferOffset, len);
-            ptr = (char *)ptr + len;
-            remaining -= len;
-            bufferOffset += len;
-        }
-    }
-
-    return 0;
-}
-
-/* Dummy functions -- we don't use thread-based audio */
-void Core_WaitAudio(_THIS)
-{
-    return;
-}
-
-void Core_PlayAudio(_THIS)
-{
-    return;
-}
-
-Uint8 *Core_GetAudioBuf(_THIS)
-{
-    return(NULL);
-}
-
-void Core_CloseAudio(_THIS)
-{
-    OSStatus result;
-    struct AURenderCallbackStruct callback;
-
-    /* stop processing the audio unit */
-    result = AudioOutputUnitStop (outputAudioUnit);
-    if (result != noErr) {
-        SDL_SetError("Core_CloseAudio: AudioOutputUnitStop");
-        return;
-    }
-
-    /* Remove the input callback */
-    callback.inputProc = 0;
-    callback.inputProcRefCon = 0;
-    result = AudioUnitSetProperty (outputAudioUnit, 
-                        kAudioUnitProperty_SetRenderCallback,
-                        kAudioUnitScope_Input, 
-                        0,
-                        &callback, 
-                        sizeof(callback));
-    if (result != noErr) {
-        SDL_SetError("Core_CloseAudio: AudioUnitSetProperty (kAudioUnitProperty_SetInputCallback)");
-        return;
-    }
-
-    result = CloseComponent(outputAudioUnit);
-    if (result != noErr) {
-        SDL_SetError("Core_CloseAudio: CloseComponent");
-        return;
-    }
-    
-    SDL_free(buffer);
-}
-
-#define CHECK_RESULT(msg) \
-    if (result != noErr) { \
-        SDL_SetError("Failed to start CoreAudio: " msg); \
-        return -1; \
-    }
-
-
-int Core_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-    OSStatus result = noErr;
-    Component comp;
-    ComponentDescription desc;
-    struct AURenderCallbackStruct callback;
-    AudioStreamBasicDescription requestedDesc;
-
-    /* Setup a AudioStreamBasicDescription with the requested format */
-    requestedDesc.mFormatID = kAudioFormatLinearPCM;
-    requestedDesc.mFormatFlags = kLinearPCMFormatFlagIsPacked;
-    requestedDesc.mChannelsPerFrame = spec->channels;
-    requestedDesc.mSampleRate = spec->freq;
-    
-    requestedDesc.mBitsPerChannel = spec->format & 0xFF;
-    if (spec->format & 0x8000)
-        requestedDesc.mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
-    if (spec->format & 0x1000)
-        requestedDesc.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
-
-    requestedDesc.mFramesPerPacket = 1;
-    requestedDesc.mBytesPerFrame = requestedDesc.mBitsPerChannel * requestedDesc.mChannelsPerFrame / 8;
-    requestedDesc.mBytesPerPacket = requestedDesc.mBytesPerFrame * requestedDesc.mFramesPerPacket;
-
-
-    /* Locate the default output audio unit */
-    desc.componentType = kAudioUnitType_Output;
-    desc.componentSubType = kAudioUnitSubType_DefaultOutput;
-    desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-    desc.componentFlags = 0;
-    desc.componentFlagsMask = 0;
-    
-    comp = FindNextComponent (NULL, &desc);
-    if (comp == NULL) {
-        SDL_SetError ("Failed to start CoreAudio: FindNextComponent returned NULL");
-        return -1;
-    }
-    
-    /* Open & initialize the default output audio unit */
-    result = OpenAComponent (comp, &outputAudioUnit);
-    CHECK_RESULT("OpenAComponent")
-
-    result = AudioUnitInitialize (outputAudioUnit);
-    CHECK_RESULT("AudioUnitInitialize")
-                
-    /* Set the input format of the audio unit. */
-    result = AudioUnitSetProperty (outputAudioUnit,
-                               kAudioUnitProperty_StreamFormat,
-                               kAudioUnitScope_Input,
-                               0,
-                               &requestedDesc,
-                               sizeof (requestedDesc));
-    CHECK_RESULT("AudioUnitSetProperty (kAudioUnitProperty_StreamFormat)")
-
-    /* Set the audio callback */
-    callback.inputProc = audioCallback;
-    callback.inputProcRefCon = this;
-    result = AudioUnitSetProperty (outputAudioUnit, 
-                        kAudioUnitProperty_SetRenderCallback,
-                        kAudioUnitScope_Input, 
-                        0,
-                        &callback, 
-                        sizeof(callback));
-    CHECK_RESULT("AudioUnitSetProperty (kAudioUnitProperty_SetInputCallback)")
-
-    /* Calculate the final parameters for this audio specification */
-    SDL_CalculateAudioSpec(spec);
-    
-    /* Allocate a sample buffer */
-    bufferOffset = bufferSize = this->spec.size;
-    buffer = SDL_malloc(bufferSize);
-
-    /* Finally, start processing of the audio unit */
-    result = AudioOutputUnitStart (outputAudioUnit);
-    CHECK_RESULT("AudioOutputUnitStart")    
-    
-
-    /* We're running! */
-    return(1);
-}
diff --git a/apps/plugins/sdl/src/audio/macosx/SDL_coreaudio.h b/apps/plugins/sdl/src/audio/macosx/SDL_coreaudio.h
deleted file mode 100644
index c11bc03a2b..0000000000
--- a/apps/plugins/sdl/src/audio/macosx/SDL_coreaudio.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_coreaudio_h
-#define _SDL_coreaudio_h
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData {
-	AudioUnit outputAudioUnit;
-	void *buffer;
-	UInt32 bufferOffset;
-	UInt32 bufferSize;
-};
-
-/* Old variable names */
-#define outputAudioUnit		(this->hidden->outputAudioUnit)
-#define buffer		(this->hidden->buffer)
-#define bufferOffset		(this->hidden->bufferOffset)
-#define bufferSize		(this->hidden->bufferSize)
-
-#endif /* _SDL_coreaudio_h */
diff --git a/apps/plugins/sdl/src/audio/macrom/SDL_romaudio.c b/apps/plugins/sdl/src/audio/macrom/SDL_romaudio.c
deleted file mode 100644
index 1b3d49e198..0000000000
--- a/apps/plugins/sdl/src/audio/macrom/SDL_romaudio.c
+++ /dev/null
@@ -1,496 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#if defined(__APPLE__) && defined(__MACH__)
-#  include <Carbon/Carbon.h>
-#elif TARGET_API_MAC_CARBON && (UNIVERSAL_INTERFACES_VERSION > 0x0335)
-#  include <Carbon.h>
-#else
-#  include <Sound.h> /* SoundManager interface */
-#  include <Gestalt.h>
-#  include <DriverServices.h>
-#endif
-
-#if !defined(NewSndCallBackUPP) && (UNIVERSAL_INTERFACES_VERSION < 0x0335)
-#if !defined(NewSndCallBackProc) /* avoid circular redefinition... */
-#define NewSndCallBackUPP NewSndCallBackProc
-#endif
-#if !defined(NewSndCallBackUPP)
-#define NewSndCallBackUPP NewSndCallBackProc
-#endif
-#endif
-
-#include "SDL_audio.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_sysaudio.h"
-#include "SDL_romaudio.h"
-
-/* Audio driver functions */
-
-static void Mac_CloseAudio(_THIS);
-static int Mac_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void Mac_LockAudio(_THIS);
-static void Mac_UnlockAudio(_THIS);
-
-/* Audio driver bootstrap functions */
-
-
-static int Audio_Available(void)
-{
-    return(1);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-    SDL_free(device->hidden);
-    SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-    SDL_AudioDevice *this;
-
-    /* Initialize all variables that we clean on shutdown */
-    this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-    if ( this ) {
-        SDL_memset(this, 0, (sizeof *this));
-        this->hidden = (struct SDL_PrivateAudioData *)
-                SDL_malloc((sizeof *this->hidden));
-    }
-    if ( (this == NULL) || (this->hidden == NULL) ) {
-        SDL_OutOfMemory();
-        if ( this ) {
-            SDL_free(this);
-        }
-        return(0);
-    }
-    SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-    /* Set the function pointers */
-    this->OpenAudio   = Mac_OpenAudio;
-    this->CloseAudio  = Mac_CloseAudio;
-    this->LockAudio   = Mac_LockAudio;
-    this->UnlockAudio = Mac_UnlockAudio;
-    this->free        = Audio_DeleteDevice;
-
-#ifdef __MACOSX__	/* Mac OS X uses threaded audio, so normal thread code is okay */
-    this->LockAudio   = NULL;
-    this->UnlockAudio = NULL;
-#endif
-    return this;
-}
-
-AudioBootStrap SNDMGR_bootstrap = {
-	"sndmgr", "MacOS SoundManager 3.0",
-	Audio_Available, Audio_CreateDevice
-};
-
-#if defined(TARGET_API_MAC_CARBON) || defined(USE_RYANS_SOUNDCODE)
-/* This works correctly on Mac OS X */
-
-#pragma options align=power
-
-static volatile SInt32 audio_is_locked = 0;
-static volatile SInt32 need_to_mix = 0;
-
-static UInt8  *buffer[2];
-static volatile UInt32 running = 0;
-static CmpSoundHeader header;
-static volatile Uint32 fill_me = 0;
-
-static void mix_buffer(SDL_AudioDevice *audio, UInt8 *buffer)
-{
-   if ( ! audio->paused ) {
-#ifdef __MACOSX__
-        SDL_mutexP(audio->mixer_lock);
-#endif
-        if ( audio->convert.needed ) {
-            audio->spec.callback(audio->spec.userdata,
-                    (Uint8 *)audio->convert.buf,audio->convert.len);
-            SDL_ConvertAudio(&audio->convert);
-            if ( audio->convert.len_cvt != audio->spec.size ) {
-                /* Uh oh... probably crashes here */;
-            }
-            SDL_memcpy(buffer, audio->convert.buf, audio->convert.len_cvt);
-        } else {
-            audio->spec.callback(audio->spec.userdata, buffer, audio->spec.size);
-        }
-#ifdef __MACOSX__
-        SDL_mutexV(audio->mixer_lock);
-#endif
-    }
-
-    DecrementAtomic((SInt32 *) &need_to_mix);
-}
-
-static void Mac_LockAudio(_THIS)
-{
-    IncrementAtomic((SInt32 *) &audio_is_locked);
-}
-
-static void Mac_UnlockAudio(_THIS)
-{
-    SInt32 oldval;
-         
-    oldval = DecrementAtomic((SInt32 *) &audio_is_locked);
-    if ( oldval != 1 )  /* != 1 means audio is still locked. */
-        return;
-
-    /* Did we miss the chance to mix in an interrupt? Do it now. */
-    if ( BitAndAtomic (0xFFFFFFFF, (UInt32 *) &need_to_mix) ) {
-        /*
-         * Note that this could be a problem if you missed an interrupt
-         *  while the audio was locked, and get preempted by a second
-         *  interrupt here, but that means you locked for way too long anyhow.
-         */
-        mix_buffer (this, buffer[fill_me]);
-    }
-}
-
-static void callBackProc (SndChannel *chan, SndCommand *cmd_passed ) {
-   UInt32 play_me;
-   SndCommand cmd; 
-   SDL_AudioDevice *audio = (SDL_AudioDevice *)chan->userInfo;
-
-   IncrementAtomic((SInt32 *) &need_to_mix);
-
-   fill_me = cmd_passed->param2;  /* buffer that has just finished playing, so fill it */      
-   play_me = ! fill_me;           /* filled buffer to play _now_ */
-
-   if ( ! audio->enabled ) {
-      return;
-   }
-   
-   /* queue previously mixed buffer for playback. */
-   header.samplePtr = (Ptr)buffer[play_me];
-   cmd.cmd = bufferCmd;
-   cmd.param1 = 0; 
-   cmd.param2 = (long)&header;
-   SndDoCommand (chan, &cmd, 0);
-
-   memset (buffer[fill_me], 0, audio->spec.size);
-
-   /*
-    * if audio device isn't locked, mix the next buffer to be queued in
-    *  the memory block that just finished playing.
-    */
-   if ( ! BitAndAtomic(0xFFFFFFFF, (UInt32 *) &audio_is_locked) ) {
-      mix_buffer (audio, buffer[fill_me]);
-   } 
-
-   /* set this callback to run again when current buffer drains. */
-   if ( running ) {
-      cmd.cmd = callBackCmd;
-      cmd.param1 = 0;
-      cmd.param2 = play_me;
-   
-      SndDoCommand (chan, &cmd, 0);
-   }
-}
-
-static int Mac_OpenAudio(_THIS, SDL_AudioSpec *spec) {
-
-   SndCallBackUPP callback;
-   int sample_bits;
-   int i;
-   long initOptions;
-      
-   /* Very few conversions are required, but... */
-    switch (spec->format) {
-        case AUDIO_S8:
-        spec->format = AUDIO_U8;
-        break;
-        case AUDIO_U16LSB:
-        spec->format = AUDIO_S16LSB;
-        break;
-        case AUDIO_U16MSB:
-        spec->format = AUDIO_S16MSB;
-        break;
-    }
-    SDL_CalculateAudioSpec(spec);
-    
-    /* initialize bufferCmd header */
-    memset (&header, 0, sizeof(header));
-    callback = (SndCallBackUPP) NewSndCallBackUPP (callBackProc);
-    sample_bits = spec->size / spec->samples / spec->channels * 8;
-
-#ifdef DEBUG_AUDIO
-    fprintf(stderr,
-	"Audio format 0x%x, channels = %d, sample_bits = %d, frequency = %d\n",
-	spec->format, spec->channels, sample_bits, spec->freq);
-#endif /* DEBUG_AUDIO */
-    
-    header.numChannels = spec->channels;
-    header.sampleSize  = sample_bits;
-    header.sampleRate  = spec->freq << 16;
-    header.numFrames   = spec->samples;
-    header.encode      = cmpSH;
-    
-    /* Note that we install the 16bitLittleEndian Converter if needed. */
-    if ( spec->format == 0x8010 ) {
-        header.compressionID = fixedCompression;
-        header.format = k16BitLittleEndianFormat;
-    }
-    
-    /* allocate 2 buffers */
-    for (i=0; i<2; i++) {
-       buffer[i] = (UInt8*)malloc (sizeof(UInt8) * spec->size);
-      if (buffer[i] == NULL) {
-         SDL_OutOfMemory();
-         return (-1);
-      }
-     memset (buffer[i], 0, spec->size);
-   }
-   
-   /* Create the sound manager channel */
-    channel = (SndChannelPtr)SDL_malloc(sizeof(*channel));
-    if ( channel == NULL ) {
-        SDL_OutOfMemory();
-        return(-1);
-    }
-    if ( spec->channels >= 2 ) {
-        initOptions = initStereo;
-    } else {
-        initOptions = initMono;
-    }
-    channel->userInfo = (long)this;
-    channel->qLength = 128;
-    if ( SndNewChannel(&channel, sampledSynth, initOptions, callback) != noErr ) {
-        SDL_SetError("Unable to create audio channel");
-        SDL_free(channel);
-        channel = NULL;
-        return(-1);
-    }
-   
-   /* start playback */
-   {
-      SndCommand cmd;
-      cmd.cmd = callBackCmd;
-      cmd.param2 = 0;
-      running = 1;
-      SndDoCommand (channel, &cmd, 0);
-   }
-   
-   return 1;
-}
-
-static void Mac_CloseAudio(_THIS) {
-   
-   int i;
-   
-   running = 0;
-   
-   if (channel) {
-      SndDisposeChannel (channel, true);
-      channel = NULL;
-   }
-   
-    for ( i=0; i<2; ++i ) {
-        if ( buffer[i] ) {
-            SDL_free(buffer[i]);
-            buffer[i] = NULL;
-        }
-    }
-}
-
-#else /* !TARGET_API_MAC_CARBON && !USE_RYANS_SOUNDCODE */
-
-static void Mac_LockAudio(_THIS)
-{
-    /* no-op. */
-}
-
-static void Mac_UnlockAudio(_THIS)
-{
-    /* no-op. */
-}
-
-
-/* This function is called by Sound Manager when it has exhausted one of
-   the buffers, so we'll zero it to silence and fill it with audio if
-   we're not paused.
-*/
-static pascal
-void sndDoubleBackProc (SndChannelPtr chan, SndDoubleBufferPtr newbuf)
-{
-    SDL_AudioDevice *audio = (SDL_AudioDevice *)newbuf->dbUserInfo[0];
-
-    /* If audio is quitting, don't do anything */
-    if ( ! audio->enabled ) {
-        return;
-    }
-    memset (newbuf->dbSoundData, 0, audio->spec.size);
-    newbuf->dbNumFrames = audio->spec.samples;
-    if ( ! audio->paused ) {
-        if ( audio->convert.needed ) {
-            audio->spec.callback(audio->spec.userdata,
-                (Uint8 *)audio->convert.buf,audio->convert.len);
-            SDL_ConvertAudio(&audio->convert);
-#if 0
-            if ( audio->convert.len_cvt != audio->spec.size ) {
-                /* Uh oh... probably crashes here */;
-            }
-#endif
-            SDL_memcpy(newbuf->dbSoundData, audio->convert.buf,
-                            audio->convert.len_cvt);
-        } else {
-            audio->spec.callback(audio->spec.userdata,
-                (Uint8 *)newbuf->dbSoundData, audio->spec.size);
-        }
-    }
-    newbuf->dbFlags    |= dbBufferReady;
-}
-
-static int DoubleBufferAudio_Available(void)
-{
-    int available;
-    NumVersion sndversion;
-    long response;
-
-    available = 0;
-    sndversion = SndSoundManagerVersion();
-    if ( sndversion.majorRev >= 3 ) {
-        if ( Gestalt(gestaltSoundAttr, &response) == noErr ) {
-            if ( (response & (1 << gestaltSndPlayDoubleBuffer)) ) {
-                available = 1;
-            }
-        }
-    } else {
-        if ( Gestalt(gestaltSoundAttr, &response) == noErr ) {
-            if ( (response & (1 << gestaltHasASC)) ) {
-                available = 1;
-            }
-        }
-    }
-    return(available);
-}
-
-static void Mac_CloseAudio(_THIS)
-{
-    int i;
-
-    if ( channel != NULL ) {
-        /* Clean up the audio channel */
-        SndDisposeChannel(channel, true);
-        channel = NULL;
-    }
-    for ( i=0; i<2; ++i ) {
-        if ( audio_buf[i] ) {
-            SDL_free(audio_buf[i]);
-            audio_buf[i] = NULL;
-        }
-    }
-}
-
-static int Mac_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-    SndDoubleBufferHeader2 audio_dbh;
-    int i;
-    long initOptions;
-    int sample_bits;
-    SndDoubleBackUPP doubleBackProc;
-
-    /* Check to make sure double-buffered audio is available */
-    if ( ! DoubleBufferAudio_Available() ) {
-        SDL_SetError("Sound manager doesn't support double-buffering");
-        return(-1);
-    }
-
-    /* Very few conversions are required, but... */
-    switch (spec->format) {
-        case AUDIO_S8:
-        spec->format = AUDIO_U8;
-        break;
-        case AUDIO_U16LSB:
-        spec->format = AUDIO_S16LSB;
-        break;
-        case AUDIO_U16MSB:
-        spec->format = AUDIO_S16MSB;
-        break;
-    }
-    SDL_CalculateAudioSpec(spec);
-
-    /* initialize the double-back header */
-    SDL_memset(&audio_dbh, 0, sizeof(audio_dbh));
-    doubleBackProc = NewSndDoubleBackProc (sndDoubleBackProc);
-    sample_bits = spec->size / spec->samples / spec->channels * 8;
-    
-    audio_dbh.dbhNumChannels = spec->channels;
-    audio_dbh.dbhSampleSize    = sample_bits;
-    audio_dbh.dbhCompressionID = 0;
-    audio_dbh.dbhPacketSize    = 0;
-    audio_dbh.dbhSampleRate    = spec->freq << 16;
-    audio_dbh.dbhDoubleBack    = doubleBackProc;
-    audio_dbh.dbhFormat    = 0;
-
-    /* Note that we install the 16bitLittleEndian Converter if needed. */
-    if ( spec->format == 0x8010 ) {
-        audio_dbh.dbhCompressionID = fixedCompression;
-        audio_dbh.dbhFormat = k16BitLittleEndianFormat;
-    }
-
-    /* allocate the 2 double-back buffers */
-    for ( i=0; i<2; ++i ) {
-        audio_buf[i] = SDL_calloc(1, sizeof(SndDoubleBuffer)+spec->size);
-        if ( audio_buf[i] == NULL ) {
-            SDL_OutOfMemory();
-            return(-1);
-        }
-        audio_buf[i]->dbNumFrames = spec->samples;
-        audio_buf[i]->dbFlags = dbBufferReady;
-        audio_buf[i]->dbUserInfo[0] = (long)this;
-        audio_dbh.dbhBufferPtr[i] = audio_buf[i];
-    }
-
-    /* Create the sound manager channel */
-    channel = (SndChannelPtr)SDL_malloc(sizeof(*channel));
-    if ( channel == NULL ) {
-        SDL_OutOfMemory();
-        return(-1);
-    }
-    if ( spec->channels >= 2 ) {
-        initOptions = initStereo;
-    } else {
-        initOptions = initMono;
-    }
-    channel->userInfo = 0;
-    channel->qLength = 128;
-    if ( SndNewChannel(&channel, sampledSynth, initOptions, 0L) != noErr ) {
-        SDL_SetError("Unable to create audio channel");
-        SDL_free(channel);
-        channel = NULL;
-        return(-1);
-    }
- 
-    /* Start playback */
-    if ( SndPlayDoubleBuffer(channel, (SndDoubleBufferHeaderPtr)&audio_dbh)
-                                != noErr ) {
-        SDL_SetError("Unable to play double buffered audio");
-        return(-1);
-    }
-    
-    return 1;
-}
-
-#endif /* TARGET_API_MAC_CARBON || USE_RYANS_SOUNDCODE */
-
diff --git a/apps/plugins/sdl/src/audio/macrom/SDL_romaudio.h b/apps/plugins/sdl/src/audio/macrom/SDL_romaudio.h
deleted file mode 100644
index 90e19c0695..0000000000
--- a/apps/plugins/sdl/src/audio/macrom/SDL_romaudio.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_romaudio_h
-#define _SDL_romaudio_h
-
-#include "../SDL_sysaudio.h"
-
-/* This is Ryan's improved MacOS sound code, with locking support */
-#define USE_RYANS_SOUNDCODE
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData {
-	/* Sound manager audio channel */
-	SndChannelPtr channel;
-#if defined(TARGET_API_MAC_CARBON) || defined(USE_RYANS_SOUNDCODE)
-	/* FIXME: Add Ryan's static data here */
-#else
-	/* Double buffering variables */
-	SndDoubleBufferPtr audio_buf[2];
-#endif
-};
-
-/* Old variable names */
-#define channel		(this->hidden->channel)
-#define audio_buf	(this->hidden->audio_buf)
-
-#endif /* _SDL_romaudio_h */
diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio.c b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio.c
deleted file mode 100644
index 46ba690c3e..0000000000
--- a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio.c
+++ /dev/null
@@ -1,215 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/*
-	Audio interrupt variables and callback function
-
-	Patrice Mandin
-*/
-
-#include <unistd.h>
-
-#include <mint/osbind.h>
-#include <mint/falcon.h>
-#include <mint/mintbind.h>
-#include <mint/cookie.h>
-
-#include "SDL_audio.h"
-#include "SDL_mintaudio.h"
-#include "SDL_mintaudio_stfa.h"
-
-/* The audio device */
-
-SDL_AudioDevice *SDL_MintAudio_device;
-Uint8 *SDL_MintAudio_audiobuf[2];	/* Pointers to buffers */
-unsigned long SDL_MintAudio_audiosize;		/* Length of audio buffer=spec->size */
-volatile unsigned short SDL_MintAudio_numbuf;		/* Buffer to play */
-volatile unsigned short SDL_MintAudio_mutex;
-volatile unsigned long SDL_MintAudio_clocktics;
-cookie_stfa_t	*SDL_MintAudio_stfa;
-unsigned short SDL_MintAudio_hasfpu;
-
-/* MiNT thread variables */
-SDL_bool SDL_MintAudio_mint_present;
-SDL_bool SDL_MintAudio_quit_thread;
-SDL_bool SDL_MintAudio_thread_finished;
-long SDL_MintAudio_thread_pid;
-
-/* The callback function, called by each driver whenever needed */
-
-void SDL_MintAudio_Callback(void)
-{
-	Uint8 *buffer;
-	SDL_AudioDevice *audio = SDL_MintAudio_device;
-
- 	buffer = SDL_MintAudio_audiobuf[SDL_MintAudio_numbuf];
-	SDL_memset(buffer, audio->spec.silence, audio->spec.size);
-
-	if (audio->paused)
-		return;
-
-	if (audio->convert.needed) {
-		int silence;
-
-		if ( audio->convert.src_format == AUDIO_U8 ) {
-			silence = 0x80;
-		} else {
-			silence = 0;
-		}
-		SDL_memset(audio->convert.buf, silence, audio->convert.len);
-		audio->spec.callback(audio->spec.userdata,
-			(Uint8 *)audio->convert.buf,audio->convert.len);
-		SDL_ConvertAudio(&audio->convert);
-		SDL_memcpy(buffer, audio->convert.buf, audio->convert.len_cvt);
-	} else {
-		audio->spec.callback(audio->spec.userdata, buffer, audio->spec.size);
-	}
-}
-
-/* Add a new frequency/clock/predivisor to the current list */
-void SDL_MintAudio_AddFrequency(_THIS, Uint32 frequency, Uint32 clock,
-	Uint32 prediv, int gpio_bits)
-{
-	int i, p;
-
-	if (MINTAUDIO_freqcount==MINTAUDIO_maxfreqs) {
-		return;
-	}
-
-	/* Search where to insert the frequency (highest first) */
-	for (p=0; p<MINTAUDIO_freqcount; p++) {
-		if (frequency > MINTAUDIO_frequencies[p].frequency) {
-			break;
-		}
-	}
-
-	/* Put all following ones farer */
-	if (MINTAUDIO_freqcount>0) {
-		for (i=MINTAUDIO_freqcount; i>p; i--) {
-			SDL_memcpy(&MINTAUDIO_frequencies[i], &MINTAUDIO_frequencies[i-1], sizeof(mint_frequency_t));
-		}
-	}
-
-	/* And insert new one */
-	MINTAUDIO_frequencies[p].frequency = frequency;
-	MINTAUDIO_frequencies[p].masterclock = clock;
-	MINTAUDIO_frequencies[p].predivisor = prediv;
-	MINTAUDIO_frequencies[p].gpio_bits = gpio_bits;
-
-	MINTAUDIO_freqcount++;
-}
-
-/* Search for the nearest frequency */
-int SDL_MintAudio_SearchFrequency(_THIS, int desired_freq)
-{
-	int i;
-
-	/* Only 1 freq ? */
-	if (MINTAUDIO_freqcount==1) {
-		return 0;
-	}
-
-	/* Check the array */
-	for (i=0; i<MINTAUDIO_freqcount; i++) {
-		if (desired_freq >= ((MINTAUDIO_frequencies[i].frequency+
-			MINTAUDIO_frequencies[i+1].frequency)>>1)) {
-			return i;
-		}
-	}
-
-	/* Not in the array, give the latest */
-	return MINTAUDIO_freqcount-1;
-}
-
-/* Check if FPU is present */
-void SDL_MintAudio_CheckFpu(void)
-{
-	long cookie_fpu;
-
-	SDL_MintAudio_hasfpu = 0;
-	if (Getcookie(C__FPU, &cookie_fpu) != C_FOUND) {
-		return;
-	}
-	switch ((cookie_fpu>>16)&0xfffe) {
-		case 2:
-		case 4:
-		case 6:
-		case 8:
-		case 16:
-			SDL_MintAudio_hasfpu = 1;
-			break;
-	}
-}
-
-/* The thread function, used under MiNT with xbios */
-int SDL_MintAudio_Thread(long param)
-{
-	SndBufPtr	pointers;
-	SDL_bool	buffers_filled[2] = {SDL_FALSE, SDL_FALSE};
-
-	SDL_MintAudio_thread_finished = SDL_FALSE;
-	while (!SDL_MintAudio_quit_thread) {
-		if (Buffptr(&pointers)!=0)
-			continue;
-
-		if (( (unsigned long)pointers.play>=(unsigned long)SDL_MintAudio_audiobuf[0])
-			&& ( (unsigned long)pointers.play<=(unsigned long)SDL_MintAudio_audiobuf[1])) 
-		{
-			/* DMA is reading buffer #0, setup buffer #1 if not already done */
-			if (!buffers_filled[1]) {
-				SDL_MintAudio_numbuf = 1;
-				SDL_MintAudio_Callback();
-				Setbuffer(0, SDL_MintAudio_audiobuf[1], SDL_MintAudio_audiobuf[1] + SDL_MintAudio_audiosize);
-				buffers_filled[1]=SDL_TRUE;
-				buffers_filled[0]=SDL_FALSE;
-			}
-		} else {
-			/* DMA is reading buffer #1, setup buffer #0 if not already done */
-			if (!buffers_filled[0]) {
-				SDL_MintAudio_numbuf = 0;
-				SDL_MintAudio_Callback();
-				Setbuffer(0, SDL_MintAudio_audiobuf[0], SDL_MintAudio_audiobuf[0] + SDL_MintAudio_audiosize);
-				buffers_filled[0]=SDL_TRUE;
-				buffers_filled[1]=SDL_FALSE;
-			}
-		}
-
-		usleep(100);
-	}
-	SDL_MintAudio_thread_finished = SDL_TRUE;
-	return 0;
-}
-
-void SDL_MintAudio_WaitThread(void)
-{
-	if (!SDL_MintAudio_mint_present)
-		return;
-
-	if (SDL_MintAudio_thread_finished)
-		return;
-
-	SDL_MintAudio_quit_thread = SDL_TRUE;
-	while (!SDL_MintAudio_thread_finished) {
-		Syield();
-	}
-}
diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio.h b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio.h
deleted file mode 100644
index ba6056ee3a..0000000000
--- a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio.h
+++ /dev/null
@@ -1,121 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/*
-	MiNT audio driver
-
-	Patrice Mandin
-*/
-
-#ifndef _SDL_mintaudio_h
-#define _SDL_mintaudio_h
-
-#include "../SDL_sysaudio.h"
-#include "SDL_mintaudio_stfa.h"
-
-/* Hidden "this" pointer for the audio functions */
-#define _THIS	SDL_AudioDevice *this
-
-/* 16 predivisors with 3 clocks max. */
-#define MINTAUDIO_maxfreqs		(16*3)		
-
-typedef struct {
-	Uint32	frequency;
-	Uint32	masterclock;
-	Uint32	predivisor;
-	int	gpio_bits;	/* in case of external clock */
-} mint_frequency_t;
-
-struct SDL_PrivateAudioData {
-	mint_frequency_t	frequencies[MINTAUDIO_maxfreqs];
-	int 	freq_count;		/* Number of frequencies in the array */
-	int		numfreq;		/* Number of selected frequency */
-};
-
-/* Old variable names */
-
-#define MINTAUDIO_frequencies	(this->hidden->frequencies)
-#define MINTAUDIO_freqcount		(this->hidden->freq_count)
-#define MINTAUDIO_numfreq		(this->hidden->numfreq)
-
-/* _MCH cookie (values>>16) */
-enum {
-	MCH_ST=0,
-	MCH_STE,
-	MCH_TT,
-	MCH_F30,
-	MCH_CLONE,
-	MCH_ARANYM
-};
-
-/* Master clocks for replay frequencies */
-#define MASTERCLOCK_STE		8010666		/* Not sure of this one */
-#define MASTERCLOCK_TT		16107953	/* Not sure of this one */
-#define MASTERCLOCK_FALCON1	25175000
-#define MASTERCLOCK_FALCON2	32000000	/* Only usable for DSP56K */
-#define MASTERCLOCK_FALCONEXT	-1		/* Clock on DSP56K port, unknown */
-#define MASTERCLOCK_44K		22579200	/* Standard clock for 44.1 Khz */
-#define MASTERCLOCK_48K		24576000	/* Standard clock for 48 Khz */
-
-/* Master clock predivisors */
-#define MASTERPREDIV_STE	160
-#define MASTERPREDIV_TT		320
-#define MASTERPREDIV_FALCON	256
-#define MASTERPREDIV_MILAN	256
-
-/* Variables */
-extern SDL_AudioDevice *SDL_MintAudio_device;
-extern Uint8 *SDL_MintAudio_audiobuf[2];	/* Pointers to buffers */
-extern unsigned long SDL_MintAudio_audiosize;		/* Length of audio buffer=spec->size */
-extern volatile unsigned short SDL_MintAudio_numbuf;		/* Buffer to play */
-extern volatile unsigned short SDL_MintAudio_mutex;
-extern cookie_stfa_t *SDL_MintAudio_stfa;
-extern volatile unsigned long SDL_MintAudio_clocktics;
-extern unsigned short SDL_MintAudio_hasfpu;	/* To preserve fpu registers if needed */
-
-/* MiNT thread variables */
-extern SDL_bool	SDL_MintAudio_mint_present;
-extern SDL_bool SDL_MintAudio_quit_thread;
-extern SDL_bool SDL_MintAudio_thread_finished;
-extern long SDL_MintAudio_thread_pid;
-
-/* Functions */
-void SDL_MintAudio_Callback(void);
-void SDL_MintAudio_AddFrequency(_THIS, Uint32 frequency, Uint32 clock,
-	Uint32 prediv, int gpio_bits);
-int SDL_MintAudio_SearchFrequency(_THIS, int desired_freq);
-void SDL_MintAudio_CheckFpu(void);
-
-/* MiNT thread functions */
-int SDL_MintAudio_Thread(long param);
-void SDL_MintAudio_WaitThread(void);
-
-/* ASM interrupt functions */
-void SDL_MintAudio_GsxbInterrupt(void);
-void SDL_MintAudio_EmptyGsxbInterrupt(void);
-void SDL_MintAudio_XbiosInterruptMeasureClock(void);
-void SDL_MintAudio_XbiosInterrupt(void);
-void SDL_MintAudio_Dma8Interrupt(void);
-void SDL_MintAudio_StfaInterrupt(void);
-
-#endif /* _SDL_mintaudio_h */
diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_dma8.c b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_dma8.c
deleted file mode 100644
index 61feba3d64..0000000000
--- a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_dma8.c
+++ /dev/null
@@ -1,357 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/*
-	MiNT audio driver
-	using DMA 8bits (hardware access)
-
-	Patrice Mandin
-*/
-
-/* Mint includes */
-#include <mint/osbind.h>
-#include <mint/falcon.h>
-#include <mint/cookie.h>
-
-#include "SDL_audio.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_sysaudio.h"
-
-#include "../../video/ataricommon/SDL_atarimxalloc_c.h"
-
-#include "SDL_mintaudio.h"
-#include "SDL_mintaudio_dma8.h"
-
-/*--- Defines ---*/
-
-#define MINT_AUDIO_DRIVER_NAME "mint_dma8"
-
-/* Debug print info */
-#define DEBUG_NAME "audio:dma8: "
-#if 0
-#define DEBUG_PRINT(what) \
-	{ \
-		printf what; \
-	}
-#else
-#define DEBUG_PRINT(what)
-#endif
-
-/*--- Static variables ---*/
-
-static long cookie_snd, cookie_mch;
-
-/*--- Audio driver functions ---*/
-
-static void Mint_CloseAudio(_THIS);
-static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void Mint_LockAudio(_THIS);
-static void Mint_UnlockAudio(_THIS);
-
-/* To check/init hardware audio */
-static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec);
-
-/* Functions called in supervisor mode */
-static void Mint_InitDma(void);
-static void Mint_StopReplay(void);
-static void Mint_StartReplay(void);
-
-/*--- Audio driver bootstrap functions ---*/
-
-static int Audio_Available(void)
-{
-	const char *envr = SDL_getenv("SDL_AUDIODRIVER");
-
-	/* Check if user asked a different audio driver */
-	if ((envr) && (SDL_strcmp(envr, MINT_AUDIO_DRIVER_NAME)!=0)) {
-		DEBUG_PRINT((DEBUG_NAME "user asked a different audio driver\n"));
-		return 0;
-	}
-
-	/* Cookie _MCH present ? if not, assume ST machine */
-	if (Getcookie(C__MCH, &cookie_mch) == C_NOTFOUND) {
-		cookie_mch = MCH_ST;
-	}
-
-	/* Cookie _SND present ? if not, assume ST machine */
-	if (Getcookie(C__SND, &cookie_snd) == C_NOTFOUND) {
-		cookie_snd = SND_PSG;
-	}
-
-	/* Check if we have 8 bits audio */
-	if ((cookie_snd & SND_8BIT)==0) {
-		DEBUG_PRINT((DEBUG_NAME "no 8 bits sound\n"));
-	    return(0);
-	}
-
-	/* Check if audio is lockable */
-	if (cookie_snd & SND_16BIT) {
-		if (Locksnd()!=1) {
-			DEBUG_PRINT((DEBUG_NAME "audio locked by other application\n"));
-			return(0);
-		}
-
-		Unlocksnd();
-	}
-
-	DEBUG_PRINT((DEBUG_NAME "8 bits audio available!\n"));
-	return(1);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-    SDL_free(device->hidden);
-    SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-    if ( this ) {
-        SDL_memset(this, 0, (sizeof *this));
-        this->hidden = (struct SDL_PrivateAudioData *)
-                SDL_malloc((sizeof *this->hidden));
-    }
-    if ( (this == NULL) || (this->hidden == NULL) ) {
-        SDL_OutOfMemory();
-        if ( this ) {
-            SDL_free(this);
-        }
-        return(0);
-    }
-    SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-    /* Set the function pointers */
-    this->OpenAudio   = Mint_OpenAudio;
-    this->CloseAudio  = Mint_CloseAudio;
-    this->LockAudio   = Mint_LockAudio;
-    this->UnlockAudio = Mint_UnlockAudio;
-    this->free        = Audio_DeleteDevice;
-
-    return this;
-}
-
-AudioBootStrap MINTAUDIO_DMA8_bootstrap = {
-	MINT_AUDIO_DRIVER_NAME, "MiNT DMA 8 bits audio driver",
-	Audio_Available, Audio_CreateDevice
-};
-
-static void Mint_LockAudio(_THIS)
-{
-	Supexec(Mint_StopReplay);
-}
-
-static void Mint_UnlockAudio(_THIS)
-{
-	Supexec(Mint_StartReplay);
-}
-
-static void Mint_CloseAudio(_THIS)
-{
-	Supexec(Mint_StopReplay);
-
-	DEBUG_PRINT((DEBUG_NAME "closeaudio: replay stopped\n"));
-
-	/* Disable interrupt */
-	Jdisint(MFP_DMASOUND);
-
-	DEBUG_PRINT((DEBUG_NAME "closeaudio: interrupt disabled\n"));
-
-	/* Wait if currently playing sound */
-	while (SDL_MintAudio_mutex != 0) {
-	}
-
-	DEBUG_PRINT((DEBUG_NAME "closeaudio: no more interrupt running\n"));
-
-	/* Clear buffers */
-	if (SDL_MintAudio_audiobuf[0]) {
-		Mfree(SDL_MintAudio_audiobuf[0]);
-		SDL_MintAudio_audiobuf[0] = SDL_MintAudio_audiobuf[1] = NULL;
-	}
-
-	DEBUG_PRINT((DEBUG_NAME "closeaudio: buffers freed\n"));
-}
-
-static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec)
-{
-	int i, masterprediv, sfreq;
-	unsigned long masterclock;
-
-	DEBUG_PRINT((DEBUG_NAME "asked: %d bits, ",spec->format & 0x00ff));
-	DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0)));
-	DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0)));
-	DEBUG_PRINT(("channels=%d, ", spec->channels));
-	DEBUG_PRINT(("freq=%d\n", spec->freq));
-
-	if (spec->channels > 2)
-		spec->channels = 2;
-
-	/* Check formats available */
-	spec->format = AUDIO_S8;
-	
-	/* Calculate and select the closest frequency */
-	sfreq=0;
-	masterclock=MASTERCLOCK_STE;
-	masterprediv=MASTERPREDIV_STE;
-	switch(cookie_mch>>16) {
-/*
-		case MCH_STE:
-			masterclock=MASTERCLOCK_STE;
-			masterprediv=MASTERPREDIV_STE;
-			break;
-*/
-		case MCH_TT:
-			masterclock=MASTERCLOCK_TT;
-			masterprediv=MASTERPREDIV_TT;
-			break;
-		case MCH_F30:
-		case MCH_ARANYM:
-			masterclock=MASTERCLOCK_FALCON1;
-			masterprediv=MASTERPREDIV_FALCON;
-			sfreq=1;
-			break;
-	}
-	
-	MINTAUDIO_freqcount=0;
-	for (i=sfreq;i<4;i++) {
-		SDL_MintAudio_AddFrequency(this, masterclock/(masterprediv*(1<<i)),
-			masterclock, i-sfreq, -1);
-	}
-
-#if 1
-	for (i=0; i<MINTAUDIO_freqcount; i++) {
-		DEBUG_PRINT((DEBUG_NAME "freq %d: %lu Hz, clock %lu, prediv %d\n",
-			i, MINTAUDIO_frequencies[i].frequency, MINTAUDIO_frequencies[i].masterclock,
-			MINTAUDIO_frequencies[i].predivisor
-		));
-	}
-#endif
-
-	MINTAUDIO_numfreq=SDL_MintAudio_SearchFrequency(this, spec->freq);
-	spec->freq=MINTAUDIO_frequencies[MINTAUDIO_numfreq].frequency;
-
-	DEBUG_PRINT((DEBUG_NAME "obtained: %d bits, ",spec->format & 0x00ff));
-	DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0)));
-	DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0)));
-	DEBUG_PRINT(("channels=%d, ", spec->channels));
-	DEBUG_PRINT(("freq=%d\n", spec->freq));
-
-	return 0;
-}
-
-static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	SDL_MintAudio_device = this;
-
-	/* Check audio capabilities */
-	if (Mint_CheckAudio(this, spec)==-1) {
-		return -1;
-	}
-
-	SDL_CalculateAudioSpec(spec);
-
-	/* Allocate memory for audio buffers in DMA-able RAM */
-	DEBUG_PRINT((DEBUG_NAME "buffer size=%d\n", spec->size));
-
-	SDL_MintAudio_audiobuf[0] = Atari_SysMalloc(spec->size *2, MX_STRAM);
-	if (SDL_MintAudio_audiobuf[0]==NULL) {
-		SDL_SetError("MINT_OpenAudio: Not enough memory for audio buffer");
-		return (-1);
-	}
-	SDL_MintAudio_audiobuf[1] = SDL_MintAudio_audiobuf[0] + spec->size ;
-	SDL_MintAudio_numbuf=0;
-	SDL_memset(SDL_MintAudio_audiobuf[0], spec->silence, spec->size *2);
-	SDL_MintAudio_audiosize = spec->size;
-	SDL_MintAudio_mutex = 0;
-
-	DEBUG_PRINT((DEBUG_NAME "buffer 0 at 0x%08x\n", SDL_MintAudio_audiobuf[0]));
-	DEBUG_PRINT((DEBUG_NAME "buffer 1 at 0x%08x\n", SDL_MintAudio_audiobuf[1]));
-
-	SDL_MintAudio_CheckFpu();
-
-	/* Set replay tracks */
-	if (cookie_snd & SND_16BIT) {
-		Settracks(0,0);
-		Setmontracks(0);
-	}
-
-	Supexec(Mint_InitDma);
-
-	/* Set interrupt */
-	Jdisint(MFP_DMASOUND);
-	Xbtimer(XB_TIMERA, 8, 1, SDL_MintAudio_Dma8Interrupt);
-	Jenabint(MFP_DMASOUND);
-
-	if (cookie_snd & SND_16BIT) {
-		if (Setinterrupt(SI_TIMERA, SI_PLAY)<0) {
-			DEBUG_PRINT((DEBUG_NAME "Setinterrupt() failed\n"));
-		}
-	}
-
-	Supexec(Mint_StartReplay);
-
-    return(1);	/* We don't use threaded audio */
-}
-
-/* Functions called in supervisor mode */
-
-static void Mint_InitDma(void)
-{
-	unsigned long buffer;
-	unsigned char mode;
-	SDL_AudioDevice *this = SDL_MintAudio_device;
-
-	Mint_StopReplay();
-
-	/* Set buffer */
-	buffer = (unsigned long) SDL_MintAudio_audiobuf[SDL_MintAudio_numbuf];
-	DMAAUDIO_IO.start_high = (buffer>>16) & 255;
-	DMAAUDIO_IO.start_mid = (buffer>>8) & 255;
-	DMAAUDIO_IO.start_low = buffer & 255;
-
-	buffer += SDL_MintAudio_audiosize;
-	DMAAUDIO_IO.end_high = (buffer>>16) & 255;
-	DMAAUDIO_IO.end_mid = (buffer>>8) & 255;
-	DMAAUDIO_IO.end_low = buffer & 255;
-
-	mode = 3-MINTAUDIO_frequencies[MINTAUDIO_numfreq].predivisor;
-	if (this->spec.channels==1) {
-		mode |= 1<<7;
-	}
-	DMAAUDIO_IO.sound_ctrl = mode;	
-}
-
-static void Mint_StopReplay(void)
-{
-	/* Stop replay */
-	DMAAUDIO_IO.control=0;
-}
-
-static void Mint_StartReplay(void)
-{
-	/* Start replay */
-	DMAAUDIO_IO.control=3;
-}
diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_dma8.h b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_dma8.h
deleted file mode 100644
index a52e5db7a5..0000000000
--- a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_dma8.h
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/*
-	DMA 8bits and Falcon Codec audio definitions
-
-	Patrice Mandin, Didier Méquignon
-*/
-
-#ifndef _SDL_mintaudio_dma8_h
-#define _SDL_mintaudio_dma8_h
-
-#define DMAAUDIO_IO_BASE (0xffff8900)
-struct DMAAUDIO_IO_S {
-	unsigned char int_ctrl;
-	unsigned char control;
-
-	unsigned char dummy1;
-	unsigned char start_high;
-	unsigned char dummy2;
-	unsigned char start_mid;
-	unsigned char dummy3;
-	unsigned char start_low;
-
-	unsigned char dummy4;
-	unsigned char cur_high;
-	unsigned char dummy5;
-	unsigned char cur_mid;
-	unsigned char dummy6;
-	unsigned char cur_low;
-
-	unsigned char dummy7;
-	unsigned char end_high;
-	unsigned char dummy8;
-	unsigned char end_mid;
-	unsigned char dummy9;
-	unsigned char end_low;
-
-	unsigned char dummy10[12];
-
-	unsigned char track_ctrl; /* CODEC only */
-	unsigned char sound_ctrl;
-	unsigned short sound_data;
-	unsigned short sound_mask;
-
-	unsigned char dummy11[10];
-	
-	unsigned short dev_ctrl;
-	unsigned short dest_ctrl;
-	unsigned short sync_div;
-	unsigned char track_rec;
-	unsigned char adderin_input;
-	unsigned char channel_input;
-	unsigned char channel_amplification;
-	unsigned char channel_reduction;
-	
-	unsigned char dummy12[6];
-
-	unsigned char data_direction;
-	unsigned char dummy13;
-	unsigned char dev_data;
-};
-#define DMAAUDIO_IO ((*(volatile struct DMAAUDIO_IO_S *)DMAAUDIO_IO_BASE))
-
-#endif /* _SDL_mintaudio_dma8_h */
diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_gsxb.c b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_gsxb.c
deleted file mode 100644
index 8d7716a137..0000000000
--- a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_gsxb.c
+++ /dev/null
@@ -1,436 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/*
-	MiNT audio driver
-	using XBIOS functions (GSXB compatible driver)
-
-	Patrice Mandin
-*/
-
-/* Mint includes */
-#include <mint/osbind.h>
-#include <mint/falcon.h>
-#include <mint/cookie.h>
-
-#include "SDL_audio.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_sysaudio.h"
-
-#include "../../video/ataricommon/SDL_atarimxalloc_c.h"
-
-#include "SDL_mintaudio.h"
-#include "SDL_mintaudio_gsxb.h"
-
-/*--- Defines ---*/
-
-#define MINT_AUDIO_DRIVER_NAME "mint_gsxb"
-
-/* Debug print info */
-#define DEBUG_NAME "audio:gsxb: "
-#if 0
-#define DEBUG_PRINT(what) \
-	{ \
-		printf what; \
-	}
-#else
-#define DEBUG_PRINT(what)
-#endif
-
-/*--- Static variables ---*/
-
-static long cookie_snd, cookie_gsxb;
-
-/*--- Audio driver functions ---*/
-
-static void Mint_CloseAudio(_THIS);
-static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void Mint_LockAudio(_THIS);
-static void Mint_UnlockAudio(_THIS);
-
-/* To check/init hardware audio */
-static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec);
-static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec);
-
-/* GSXB callbacks */
-static void Mint_GsxbInterrupt(void);
-static void Mint_GsxbNullInterrupt(void);
-
-/*--- Audio driver bootstrap functions ---*/
-
-static int Audio_Available(void)
-{
-	const char *envr = SDL_getenv("SDL_AUDIODRIVER");
-
-	/* Check if user asked a different audio driver */
-	if ((envr) && (SDL_strcmp(envr, MINT_AUDIO_DRIVER_NAME)!=0)) {
-		DEBUG_PRINT((DEBUG_NAME "user asked a different audio driver\n"));
-		return(0);
-	}
-
-	/* Cookie _SND present ? if not, assume ST machine */
-	if (Getcookie(C__SND, &cookie_snd) == C_NOTFOUND) {
-		cookie_snd = SND_PSG;
-	}
-
-	/* Check if we have 16 bits audio */
-	if ((cookie_snd & SND_16BIT)==0) {
-		DEBUG_PRINT((DEBUG_NAME "no 16 bits sound\n"));
-	    return(0);
-	}
-
-	/* Cookie GSXB present ? */
-	cookie_gsxb = (Getcookie(C_GSXB, &cookie_gsxb) == C_FOUND);
-
-	/* Is it GSXB ? */
-	if (((cookie_snd & SND_GSXB)==0) || (cookie_gsxb==0)) {
-		DEBUG_PRINT((DEBUG_NAME "no GSXB audio\n"));
-		return(0);
-	}
-
-	/* Check if audio is lockable */
-	if (Locksnd()!=1) {
-		DEBUG_PRINT((DEBUG_NAME "audio locked by other application\n"));
-		return(0);
-	}
-
-	Unlocksnd();
-
-	DEBUG_PRINT((DEBUG_NAME "GSXB audio available!\n"));
-	return(1);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-    SDL_free(device->hidden);
-    SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-    if ( this ) {
-        SDL_memset(this, 0, (sizeof *this));
-        this->hidden = (struct SDL_PrivateAudioData *)
-                SDL_malloc((sizeof *this->hidden));
-    }
-    if ( (this == NULL) || (this->hidden == NULL) ) {
-        SDL_OutOfMemory();
-        if ( this ) {
-            SDL_free(this);
-        }
-        return(0);
-    }
-    SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-    /* Set the function pointers */
-    this->OpenAudio   = Mint_OpenAudio;
-    this->CloseAudio  = Mint_CloseAudio;
-    this->LockAudio   = Mint_LockAudio;
-    this->UnlockAudio = Mint_UnlockAudio;
-    this->free        = Audio_DeleteDevice;
-
-    return this;
-}
-
-AudioBootStrap MINTAUDIO_GSXB_bootstrap = {
-	MINT_AUDIO_DRIVER_NAME, "MiNT GSXB audio driver",
-	Audio_Available, Audio_CreateDevice
-};
-
-static void Mint_LockAudio(_THIS)
-{
-	/* Stop replay */
-	Buffoper(0);
-}
-
-static void Mint_UnlockAudio(_THIS)
-{
-	/* Restart replay */
-	Buffoper(SB_PLA_ENA|SB_PLA_RPT);
-}
-
-static void Mint_CloseAudio(_THIS)
-{
-	/* Stop replay */
-	Buffoper(0);
-
-	/* Uninstall interrupt */
-	if (NSetinterrupt(2, SI_NONE, Mint_GsxbNullInterrupt)<0) {
-		DEBUG_PRINT((DEBUG_NAME "NSetinterrupt() failed in close\n"));
-	}
-
-	/* Wait if currently playing sound */
-	while (SDL_MintAudio_mutex != 0) {
-	}
-
-	/* Clear buffers */
-	if (SDL_MintAudio_audiobuf[0]) {
-		Mfree(SDL_MintAudio_audiobuf[0]);
-		SDL_MintAudio_audiobuf[0] = SDL_MintAudio_audiobuf[1] = NULL;
-	}
-
-	/* Unlock sound system */
-	Unlocksnd();
-}
-
-static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec)
-{
-	long snd_format = 0;
-	int i, resolution, format_signed, format_bigendian;
-    Uint16 test_format = SDL_FirstAudioFormat(spec->format);
-    int valid_datatype = 0;
-
-	resolution = spec->format & 0x00ff;
-	format_signed = ((spec->format & 0x8000)!=0);
-	format_bigendian = ((spec->format & 0x1000)!=0);
-
-	DEBUG_PRINT((DEBUG_NAME "asked: %d bits, ",spec->format & 0x00ff));
-	DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0)));
-	DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0)));
-	DEBUG_PRINT(("channels=%d, ", spec->channels));
-	DEBUG_PRINT(("freq=%d\n", spec->freq));
-
-    if (spec->channels > 2) {
-        spec->channels = 2;  /* no more than stereo! */
-    }
-
-    while ((!valid_datatype) && (test_format)) {
-        /* Check formats available */
-        snd_format = Sndstatus(SND_QUERYFORMATS);
-        spec->format = test_format;
-        resolution = spec->format & 0xff;
-        format_signed = (spec->format & (1<<15));
-        format_bigendian = (spec->format & (1<<12));
-        switch (test_format) {
-            case AUDIO_U8:
-            case AUDIO_S8:
-                if (snd_format & SND_FORMAT8) {
-                    valid_datatype = 1;
-                    snd_format = Sndstatus(SND_QUERY8BIT);
-                }
-                break;
-
-            case AUDIO_U16LSB:
-            case AUDIO_S16LSB:
-            case AUDIO_U16MSB:
-            case AUDIO_S16MSB:
-                if (snd_format & SND_FORMAT16) {
-                    valid_datatype = 1;
-                    snd_format = Sndstatus(SND_QUERY16BIT);
-                }
-                break;
-
-            default:
-                test_format = SDL_NextAudioFormat();
-                break;
-        }
-    }
-
-    if (!valid_datatype) {
-        SDL_SetError("Unsupported audio format");
-        return (-1);
-    }
-
-	/* Check signed/unsigned format */
-	if (format_signed) {
-		if (snd_format & SND_FORMATSIGNED) {
-			/* Ok */
-		} else if (snd_format & SND_FORMATUNSIGNED) {
-			/* Give unsigned format */
-			spec->format = spec->format & (~0x8000);
-		}
-	} else {
-		if (snd_format & SND_FORMATUNSIGNED) {
-			/* Ok */
-		} else if (snd_format & SND_FORMATSIGNED) {
-			/* Give signed format */
-			spec->format |= 0x8000;
-		}
-	}
-
-	if (format_bigendian) {
-		if (snd_format & SND_FORMATBIGENDIAN) {
-			/* Ok */
-		} else if (snd_format & SND_FORMATLITTLEENDIAN) {
-			/* Give little endian format */
-			spec->format = spec->format & (~0x1000);
-		}
-	} else {
-		if (snd_format & SND_FORMATLITTLEENDIAN) {
-			/* Ok */
-		} else if (snd_format & SND_FORMATBIGENDIAN) {
-			/* Give big endian format */
-			spec->format |= 0x1000;
-		}
-	}
-	
-	/* Calculate and select the closest frequency */
-	MINTAUDIO_freqcount=0;
-	for (i=1;i<4;i++) {
-		SDL_MintAudio_AddFrequency(this,
-			MASTERCLOCK_44K/(MASTERPREDIV_MILAN*(1<<i)), MASTERCLOCK_44K,
-			(1<<i)-1, -1);
-	}
-
-#if 1
-	for (i=0; i<MINTAUDIO_freqcount; i++) {
-		DEBUG_PRINT((DEBUG_NAME "freq %d: %lu Hz, clock %lu, prediv %d\n",
-			i, MINTAUDIO_frequencies[i].frequency, MINTAUDIO_frequencies[i].masterclock,
-			MINTAUDIO_frequencies[i].predivisor
-		));
-	}
-#endif
-
-	MINTAUDIO_numfreq=SDL_MintAudio_SearchFrequency(this, spec->freq);
-	spec->freq=MINTAUDIO_frequencies[MINTAUDIO_numfreq].frequency;
-
-	DEBUG_PRINT((DEBUG_NAME "obtained: %d bits, ",spec->format & 0x00ff));
-	DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0)));
-	DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0)));
-	DEBUG_PRINT(("channels=%d, ", spec->channels));
-	DEBUG_PRINT(("freq=%d\n", spec->freq));
-
-	return 0;
-}
-
-static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec)
-{
-	int channels_mode, prediv;
-	void *buffer;
-
-	/* Stop currently playing sound */
-	Buffoper(0);
-
-	/* Set replay tracks */
-	Settracks(0,0);
-	Setmontracks(0);
-
-	/* Select replay format */
-	switch (spec->format & 0xff) {
-		case 8:
-			if (spec->channels==2) {
-				channels_mode=STEREO8;
-			} else {
-				channels_mode=MONO8;
-			}
-			break;
-		case 16:
-			if (spec->channels==2) {
-				channels_mode=STEREO16;
-			} else {
-				channels_mode=MONO16;
-			}
-			break;
-		default:
-			channels_mode=STEREO16;
-			break;
-	}
-	if (Setmode(channels_mode)<0) {
-		DEBUG_PRINT((DEBUG_NAME "Setmode() failed\n"));
-	}
-
-	prediv = MINTAUDIO_frequencies[MINTAUDIO_numfreq].predivisor;
-	Devconnect(DMAPLAY, DAC, CLKEXT, prediv, 1);
-
-	/* Set buffer */
-	buffer = SDL_MintAudio_audiobuf[SDL_MintAudio_numbuf];
-	if (Setbuffer(0, buffer, buffer + spec->size)<0) {
-		DEBUG_PRINT((DEBUG_NAME "Setbuffer() failed\n"));
-	}
-	
-	/* Install interrupt */
-	if (NSetinterrupt(2, SI_PLAY, Mint_GsxbInterrupt)<0) {
-		DEBUG_PRINT((DEBUG_NAME "NSetinterrupt() failed\n"));
-	}
-
-	/* Go */
-	Buffoper(SB_PLA_ENA|SB_PLA_RPT);
-	DEBUG_PRINT((DEBUG_NAME "hardware initialized\n"));
-}
-
-static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	/* Lock sound system */
-	if (Locksnd()!=1) {
-   	    SDL_SetError("Mint_OpenAudio: Audio system already in use");
-        return(-1);
-	}
-
-	SDL_MintAudio_device = this;
-
-	/* Check audio capabilities */
-	if (Mint_CheckAudio(this, spec)==-1) {
-		return -1;
-	}
-
-	SDL_CalculateAudioSpec(spec);
-
-	/* Allocate memory for audio buffers in DMA-able RAM */
-	DEBUG_PRINT((DEBUG_NAME "buffer size=%d\n", spec->size));
-
-	SDL_MintAudio_audiobuf[0] = Atari_SysMalloc(spec->size *2, MX_STRAM);
-	if (SDL_MintAudio_audiobuf[0]==NULL) {
-		SDL_SetError("MINT_OpenAudio: Not enough memory for audio buffer");
-		return (-1);
-	}
-	SDL_MintAudio_audiobuf[1] = SDL_MintAudio_audiobuf[0] + spec->size ;
-	SDL_MintAudio_numbuf=0;
-	SDL_memset(SDL_MintAudio_audiobuf[0], spec->silence, spec->size *2);
-	SDL_MintAudio_audiosize = spec->size;
-	SDL_MintAudio_mutex = 0;
-
-	DEBUG_PRINT((DEBUG_NAME "buffer 0 at 0x%08x\n", SDL_MintAudio_audiobuf[0]));
-	DEBUG_PRINT((DEBUG_NAME "buffer 1 at 0x%08x\n", SDL_MintAudio_audiobuf[1]));
-
-	SDL_MintAudio_CheckFpu();
-
-	/* Setup audio hardware */
-	Mint_InitAudio(this, spec);
-
-    return(1);	/* We don't use threaded audio */
-}
-
-static void Mint_GsxbInterrupt(void)
-{
-	Uint8 *newbuf;
-
-	if (SDL_MintAudio_mutex)
-		return;
-
-	SDL_MintAudio_mutex=1;
-
-	SDL_MintAudio_numbuf ^= 1;
-	SDL_MintAudio_Callback();
-	newbuf = SDL_MintAudio_audiobuf[SDL_MintAudio_numbuf];
-	Setbuffer(0, newbuf, newbuf + SDL_MintAudio_audiosize);
-
-	SDL_MintAudio_mutex=0;
-}
-
-static void Mint_GsxbNullInterrupt(void)
-{
-}
diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_gsxb.h b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_gsxb.h
deleted file mode 100644
index aee26b7ee3..0000000000
--- a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_gsxb.h
+++ /dev/null
@@ -1,104 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/*
- * GSXB audio definitions
- * 
- * Patrice Mandin
- */
-
-#ifndef _SDL_mintaudio_gsxb_h
-#define _SDL_mintaudio_gsxb_h
-
-#include <mint/falcon.h>	/* for trap_14_xxx macros */
-
-/* Bit 5 in cookie _SND */
-
-#define SND_GSXB	(1<<5)
-
-/* NSoundcmd modes */
-
-#define SETRATE			7	/* Set sample rate */
-#define SET8BITFORMAT	8	/* 8 bits format */
-#define SET16BITFORMAT	9	/* 16 bits format */
-#define SET24BITFORMAT	10	/* 24 bits format */
-#define SET32BITFORMAT	11	/* 32 bits format */
-#define LTATTEN_MASTER	12	/* Attenuation */
-#define RTATTEN_MASTER	13
-#define LTATTEN_MICIN	14
-#define RTATTEN_MICIN	15
-#define LTATTEN_FMGEN	16
-#define RTATTEN_FMGEN	17
-#define LTATTEN_LINEIN	18
-#define RTATTEN_LINEIN	19
-#define LTATTEN_CDIN	20
-#define RTATTEN_CDIN	21
-#define LTATTEN_VIDIN	22
-#define RTATTEN_VIDIN	23
-#define LTATTEN_AUXIN	24
-#define RTATTEN_AUXIN	25
-
-/* Setmode modes */
-
-#define MONO16		3
-#define STEREO24	4
-#define STEREO32	5
-#define MONO24		6
-#define MONO32		7
-
-/* Sndstatus modes */
-
-#define SND_QUERYFORMATS	2
-#define SND_QUERYMIXERS		3
-#define SND_QUERYSOURCES	4
-#define SND_QUERYDUPLEX		5
-#define SND_QUERY8BIT		8
-#define SND_QUERY16BIT		9
-#define SND_QUERY24BIT		10
-#define SND_QUERY32BIT		11
-
-#define SND_FORMAT8		(1<<0)
-#define SND_FORMAT16	(1<<1)
-#define SND_FORMAT24	(1<<2)
-#define SND_FORMAT32	(1<<3)
-
-#define SND_FORMATSIGNED		(1<<0)
-#define SND_FORMATUNSIGNED		(1<<1)
-#define SND_FORMATBIGENDIAN		(1<<2)
-#define SND_FORMATLITTLEENDIAN	(1<<3)
-
-/* Devconnect prescalers */
-
-#define CLK_44K		1
-#define CLK_22K		3
-#define CLK_11K		7
-
-/* Extra xbios functions */
-
-#define NSoundcmd(mode,data,data2)	\
-	(long)trap_14_wwl((short)130,(short)(mode),(short)(data),(long)(data2))
-#define NSetinterrupt(src_inter,cause,inth_addr)	\
-	(long)trap_14_wwwl((short)135,(short)(src_inter),(short)(cause),	\
-		(long)(inth_addr))
-
-#endif /* _SDL_mintaudio_gsxb_h */
diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_it.S b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_it.S
deleted file mode 100644
index a2ecac4c65..0000000000
--- a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_it.S
+++ /dev/null
@@ -1,386 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-
-/*
-	Audio interrupts
-
-	Patrice Mandin, Didier Méquignon
- */
-
-	.text
-
-	.globl	_SDL_MintAudio_Callback
-
-	.globl	_SDL_MintAudio_XbiosInterrupt
-	.globl	_SDL_MintAudio_XbiosInterruptMeasureClock
-	.globl	_SDL_MintAudio_Dma8Interrupt
-	.globl	_SDL_MintAudio_StfaInterrupt
-
-	.globl	_SDL_MintAudio_mutex
-	.globl	_SDL_MintAudio_audiobuf
-	.globl	_SDL_MintAudio_numbuf
-	.globl	_SDL_MintAudio_audiosize
-	.globl	_SDL_MintAudio_clocktics
-	.globl	_SDL_MintAudio_hasfpu
-
-	.globl	_SDL_MintAudio_stfa
-
-/*
-	How it works:
-	- Audio is playing buffer #0 (resp. #1)
-	- We must calculate a sample in buffer #1 (resp. #0)
-	  so we first call the callback to do it
-	- Then we swap the buffers
-*/
-
-#define	savptr	0x4a2
-#define	savamt	0x46
-
-/*--- Save/restore FPU context ---*/
-
-#if defined(__mcoldfire__)
-
-#define SAVE_FPU_CONTEXT \
-	lea		sp@(-216),sp;	\
-	fsave		sp@;	\
-	fmovel		fpiar,sp@-;	\
-	lea		sp@(-64),sp;	\
-	fmovemd	fp0-fp7,sp@
-
-#define RESTORE_FPU_CONTEXT	\
-	fmovemd		sp@,fp0-fp7;	\
-	lea		sp@(64),sp;	\
-	fmovel		sp@+,fpiar;	\
-	frestore	sp@;	\
-	lea		sp@(216),sp
-
-#else
-
-#define SAVE_FPU_CONTEXT	\
-	.chip	68k/68881;	\
-	fsave	sp@-;	\
-	fmoveml fpcr/fpsr/fpiar,sp@-;	\
-	fmovemx	fp0-fp7,sp@-;	\
-	.chip	68k
-
-#define RESTORE_FPU_CONTEXT	\
-	.chip	68k/68881;	\
-	fmovemx	sp@+,fp0-fp7;	\
-	fmoveml	sp@+,fpcr/fpsr/fpiar;	\
-	frestore	sp@+;	\
-	.chip	68k
-
-#endif
-
-/*--- Xbios interrupt vector to measure Falcon external clock ---*/
-
-_SDL_MintAudio_XbiosInterruptMeasureClock:          /* 1 mS */
-#if defined(__mcoldfire__)
-	movel	d0,sp@-
-	
-	moveql	#0,d0
-	btst	d0,0xFFFF8901:w	/* state DMA sound */
-#else
-	btst	#0,0xFFFF8901:w	/* state DMA sound */
-#endif
-	beqs	SDL_MintAudio_EndIntMeasure
-	addql	#1,_SDL_MintAudio_clocktics
-SDL_MintAudio_EndIntMeasure:
-#if defined(__mcoldfire__)
-	moveql	#5,d0
-	bclr	d0,0xFFFFFA0F:w	/* Clear service bit */
-
-	movel	sp@+,d0
-#else
-	bclr	#5,0xFFFFFA0F:w	/* Clear service bit */
-#endif
-	rte
-
-/*--- Xbios interrupt vector ---*/
-
-_SDL_MintAudio_XbiosInterrupt:
-#if defined(__mcoldfire__)
-	lea	sp@(-60),sp
-	moveml	d0-d7/a0-a6,sp@
-#else
-	moveml	d0-d7/a0-a6,sp@-
-#endif
-
-	/* Reenable interrupts, so other interrupts can work */
-	movew	#0x2300,sr
-
-	/* Clear service bit, so other MFP interrupts can work */
-#if defined(__mcoldfire__)
-	moveql	#5,d0
-	bclr	d0,0xfffffa0f:w
-#else
-	bclr	#5,0xfffffa0f:w
-#endif
-
-	/* Check if we are not already running */
-	tstw	_SDL_MintAudio_mutex
-	bne	SDL_MintAudio_XbiosEnd
-
-#if defined(__mcoldfire__)
-	movew	_SDL_MintAudio_mutex,d0
-	notl	d0
-	movew	d0,_SDL_MintAudio_mutex
-
-	movew	_SDL_MintAudio_numbuf,d1
-	eorl	#1,d1
-	movew	d1,_SDL_MintAudio_numbuf
-#else
-	notw	_SDL_MintAudio_mutex
-
-	/* Swap buffers */
-	eorw	#1,_SDL_MintAudio_numbuf
-#endif
-	
-	/* Save FPU if needed */
-	tstw	_SDL_MintAudio_hasfpu
-	beqs	SDL_MintAudio_Xbios_nofpu1
-	SAVE_FPU_CONTEXT
-SDL_MintAudio_Xbios_nofpu1:
-
-	/* Callback */
-	jsr	_SDL_MintAudio_Callback
-
-	/* Restore FPU if needed */
-	tstw	_SDL_MintAudio_hasfpu
-	beqs	SDL_MintAudio_Xbios_nofpu2
-	RESTORE_FPU_CONTEXT
-SDL_MintAudio_Xbios_nofpu2:
-
-	/* Reserve space for registers */
-#if defined(__mcoldfire__)
-	movel	#savamt,d0
-	subl	d0,savptr
-#else
-	subl	#savamt,savptr
-#endif
-
-	/* Set new buffer */
-
-	moveq	#0,d0
-	movel	_SDL_MintAudio_audiosize,d1
-
-	movew	_SDL_MintAudio_numbuf,d0
-	lsll	#2,d0
-	lea	_SDL_MintAudio_audiobuf,a0
-	movel	a0@(d0:l),a1
-
-	lea	a1@(d1:l),a2
-
-	movel	a2,sp@-
-	movel	a1,sp@-
-	clrw	sp@-
-	movew	#131,sp@-
-	trap	#14
-	lea	sp@(12),sp
-
-	/* Restore registers space */
-#if defined(__mcoldfire__)
-	movel	#savamt,d0
-	addl	d0,savptr
-#else
-	addl	#savamt,savptr
-#endif
-
-	clrw	_SDL_MintAudio_mutex
-SDL_MintAudio_XbiosEnd:
-#if defined(__mcoldfire__)
-	moveml	sp@,d0-d7/a0-a6
-	lea	sp@(60),sp
-#else
-	moveml	sp@+,d0-d7/a0-a6
-#endif
-	rte
-
-/*--- DMA 8 bits interrupt vector ---*/
-
-_SDL_MintAudio_Dma8Interrupt:
-#if defined(__mcoldfire__)
-	lea	sp@(-16),sp
-	moveml	d0-d1/a0-a1,sp@
-#else
-	moveml	d0-d1/a0-a1,sp@-
-#endif
-
-	/* Reenable interrupts, so other interrupts can work */
-	movew	#0x2300,sr
-
-	/* Clear service bit, so other MFP interrupts can work */
-#if defined(__mcoldfire__)
-	moveql	#5,d0
-	bclr	d0,0xfffffa0f:w
-#else
-	bclr	#5,0xfffffa0f:w
-#endif
-	/* Check if we are not already running */
-	tstw	_SDL_MintAudio_mutex
-	bne	SDL_MintAudio_Dma8End
-
-#if defined(__mcoldfire__)
-	movew	_SDL_MintAudio_mutex,d0
-	notl	d0
-	movew	d0,_SDL_MintAudio_mutex
-
-	movew	_SDL_MintAudio_numbuf,d1
-	eorl	#1,d1
-	movew	d1,_SDL_MintAudio_numbuf
-#else
-	notw	_SDL_MintAudio_mutex
-
-	/* Swap buffers */
-	eorw	#1,_SDL_MintAudio_numbuf
-#endif
-
-	/* Save FPU if needed */
-	tstw	_SDL_MintAudio_hasfpu
-	beqs	SDL_MintAudio_Dma8_nofpu1
-	SAVE_FPU_CONTEXT
-SDL_MintAudio_Dma8_nofpu1:
-
-	/* Callback */
-	jsr	_SDL_MintAudio_Callback
-
-	/* Restore FPU if needed */
-	tstw	_SDL_MintAudio_hasfpu
-	beqs	SDL_MintAudio_Dma8_nofpu2
-	RESTORE_FPU_CONTEXT
-SDL_MintAudio_Dma8_nofpu2:
-
-	/* Set new buffer */
-
-	moveq	#0,d0
-
-	movew	_SDL_MintAudio_numbuf,d0
-	lsll	#2,d0
-	lea	_SDL_MintAudio_audiobuf,a0
-	movel	a0@(d0:l),d1
-
-	/* Modify DMA addresses */
-	lea	0xffff8900:w,a0
-
-	movel	d1,d0
-
-	moveb	d0,a0@(0x07)	/* Start address */
-	lsrl	#8,d0
-	moveb	d0,a0@(0x05)
-	lsrl	#8,d0
-	moveb	d0,a0@(0x03)
-
-	addl	_SDL_MintAudio_audiosize,d1
-
-	movel	d1,d0
-
-	moveb	d0,a0@(0x13)	/* End address */
-	lsrl	#8,d0
-	moveb	d0,a0@(0x11)
-	lsrl	#8,d0
-	moveb	d0,a0@(0x0f)
-
-	clrw	_SDL_MintAudio_mutex
-SDL_MintAudio_Dma8End:
-#if defined(__mcoldfire__)
-	moveml	sp@,d0-d1/a0-a1
-	lea	sp@(16),sp
-#else
-	moveml	sp@+,d0-d1/a0-a1
-#endif
-	rte
-
-/*--- STFA interrupt vector ---*/
-
-STFA_SOUND_START	=	6
-STFA_SOUND_END		=	STFA_SOUND_START+8
-
-_SDL_MintAudio_StfaInterrupt:
-	/* Reenable interrupts, so other interrupts can work */
-	movew	#0x2300,sr
-
-	/* Check if we are not already running */
-	tstw	_SDL_MintAudio_mutex
-
-#if defined(__mcoldfire__)
-	bne	SDL_MintAudio_StfaEnd
-
-	lea	sp@(-60),sp
-	moveml	d0-d7/a0-a6,sp@
-
-	movew	_SDL_MintAudio_mutex,d0
-	notl	d0
-	movew	d0,_SDL_MintAudio_mutex
-
-	movew	_SDL_MintAudio_numbuf,d1
-	eorl	#1,d1
-	movew	d1,_SDL_MintAudio_numbuf
-#else
-	bnes	SDL_MintAudio_StfaEnd
-
-	moveml	d0-d7/a0-a6,sp@-
-
-	notw	_SDL_MintAudio_mutex
-
-	/* Swap buffers */
-	eorw	#1,_SDL_MintAudio_numbuf
-#endif
-	
-	/* Save FPU if needed */
-	tstw	_SDL_MintAudio_hasfpu
-	beqs	SDL_MintAudio_Stfa_nofpu1
-	SAVE_FPU_CONTEXT
-SDL_MintAudio_Stfa_nofpu1:
-
-	/* Callback */
-	jsr	_SDL_MintAudio_Callback
-
-	/* Restore FPU if needed */
-	tstw	_SDL_MintAudio_hasfpu
-	beqs	SDL_MintAudio_Stfa_nofpu2
-	RESTORE_FPU_CONTEXT
-SDL_MintAudio_Stfa_nofpu2:
-
-	/* Set new buffer */
-
-	moveq	#0,d0
-	movel	_SDL_MintAudio_stfa,a1
-
-	movew	_SDL_MintAudio_numbuf,d0
-	lsll	#2,d0
-	lea	_SDL_MintAudio_audiobuf,a0
-	movel	a0@(d0:l),d1
-
-	/* Modify STFA replay buffers */
-	movel	d1,a1@(STFA_SOUND_START)
-	addl	_SDL_MintAudio_audiosize,d1
-	movel	d1,a1@(STFA_SOUND_END)
-
-#if defined(__mcoldfire__)
-	moveml	sp@,d0-d7/a0-a6
-	lea	sp@(60),sp
-#else
-	moveml	sp@+,d0-d7/a0-a6
-#endif
-	clrw	_SDL_MintAudio_mutex
-SDL_MintAudio_StfaEnd:
-	rte
diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_mcsn.c b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_mcsn.c
deleted file mode 100644
index 387609b168..0000000000
--- a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_mcsn.c
+++ /dev/null
@@ -1,405 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/*
-	MiNT audio driver
-	using XBIOS functions (MacSound compatible driver)
-
-	Patrice Mandin
-*/
-
-#include <support.h>
-
-/* Mint includes */
-#include <mint/osbind.h>
-#include <mint/falcon.h>
-#include <mint/cookie.h>
-
-#include "SDL_audio.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_sysaudio.h"
-
-#include "../../video/ataricommon/SDL_atarimxalloc_c.h"
-
-#include "SDL_mintaudio.h"
-#include "SDL_mintaudio_mcsn.h"
-
-/*--- Defines ---*/
-
-#define MINT_AUDIO_DRIVER_NAME "mint_mcsn"
-
-/* Debug print info */
-#define DEBUG_NAME "audio:mcsn: "
-#if 0
-#define DEBUG_PRINT(what) \
-	{ \
-		printf what; \
-	}
-#else
-#define DEBUG_PRINT(what)
-#endif
-
-/*--- Static variables ---*/
-
-static long cookie_snd, cookie_mch;
-static cookie_mcsn_t *cookie_mcsn;
-
-/*--- Audio driver functions ---*/
-
-static void Mint_CloseAudio(_THIS);
-static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void Mint_LockAudio(_THIS);
-static void Mint_UnlockAudio(_THIS);
-
-/* To check/init hardware audio */
-static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec);
-static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec);
-
-/*--- Audio driver bootstrap functions ---*/
-
-static int Audio_Available(void)
-{
-	long dummy;
-	const char *envr = SDL_getenv("SDL_AUDIODRIVER");
-
-	SDL_MintAudio_mint_present = (Getcookie(C_MiNT, &dummy) == C_FOUND);
-
-	/* We can't use XBIOS in interrupt with Magic, don't know about thread */
-	if (Getcookie(C_MagX, &dummy) == C_FOUND) {
-		return(0);
-	}
-
-	/* Check if user asked a different audio driver */
-	if ((envr) && (SDL_strcmp(envr, MINT_AUDIO_DRIVER_NAME)!=0)) {
-		DEBUG_PRINT((DEBUG_NAME "user asked a different audio driver\n"));
-		return(0);
-	}
-
-	/* Cookie _MCH present ? if not, assume ST machine */
-	if (Getcookie(C__MCH, &cookie_mch) == C_NOTFOUND) {
-		cookie_mch = MCH_ST;
-	}
-
-	/* Cookie _SND present ? if not, assume ST machine */
-	if (Getcookie(C__SND, &cookie_snd) == C_NOTFOUND) {
-		cookie_snd = SND_PSG;
-	}
-
-	/* Check if we have 16 bits audio */
-	if ((cookie_snd & SND_16BIT)==0) {
-		DEBUG_PRINT((DEBUG_NAME "no 16 bits sound\n"));
-	    return(0);
-	}
-
-	/* Cookie MCSN present ? */
-	if (Getcookie(C_McSn, &dummy) != C_FOUND) {
-		DEBUG_PRINT((DEBUG_NAME "no MCSN audio\n"));
-		return(0);
-	}
-	cookie_mcsn = (cookie_mcsn_t *) dummy;
-
-	/* Check if interrupt at end of replay */
-	if (cookie_mcsn->pint == 0) {
-		DEBUG_PRINT((DEBUG_NAME "no interrupt at end of replay\n"));
-		return(0);
-	}
-
-	/* Check if audio is lockable */
-	if (Locksnd()!=1) {
-		DEBUG_PRINT((DEBUG_NAME "audio locked by other application\n"));
-		return(0);
-	}
-
-	Unlocksnd();
-
-	DEBUG_PRINT((DEBUG_NAME "MCSN audio available!\n"));
-	return(1);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-    SDL_free(device->hidden);
-    SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-    if ( this ) {
-        SDL_memset(this, 0, (sizeof *this));
-        this->hidden = (struct SDL_PrivateAudioData *)
-                SDL_malloc((sizeof *this->hidden));
-    }
-    if ( (this == NULL) || (this->hidden == NULL) ) {
-        SDL_OutOfMemory();
-        if ( this ) {
-            SDL_free(this);
-        }
-        return(0);
-    }
-    SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-    /* Set the function pointers */
-    this->OpenAudio   = Mint_OpenAudio;
-    this->CloseAudio  = Mint_CloseAudio;
-    this->LockAudio   = Mint_LockAudio;
-    this->UnlockAudio = Mint_UnlockAudio;
-    this->free        = Audio_DeleteDevice;
-
-    return this;
-}
-
-AudioBootStrap MINTAUDIO_MCSN_bootstrap = {
-	MINT_AUDIO_DRIVER_NAME, "MiNT MCSN audio driver",
-	Audio_Available, Audio_CreateDevice
-};
-
-static void Mint_LockAudio(_THIS)
-{
-	/* Stop replay */
-	Buffoper(0);
-}
-
-static void Mint_UnlockAudio(_THIS)
-{
-	/* Restart replay */
-	Buffoper(SB_PLA_ENA|SB_PLA_RPT);
-}
-
-static void Mint_CloseAudio(_THIS)
-{
-	/* Stop replay */
-	SDL_MintAudio_WaitThread();
-	Buffoper(0);
-
-	if (!SDL_MintAudio_mint_present) {
-		/* Uninstall interrupt */
-		Jdisint(MFP_DMASOUND);
-	}
-
-	/* Wait if currently playing sound */
-	while (SDL_MintAudio_mutex != 0) {
-	}
-
-	/* Clear buffers */
-	if (SDL_MintAudio_audiobuf[0]) {
-		Mfree(SDL_MintAudio_audiobuf[0]);
-		SDL_MintAudio_audiobuf[0] = SDL_MintAudio_audiobuf[1] = NULL;
-	}
-
-	/* Unlock sound system */
-	Unlocksnd();
-}
-
-static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec)
-{
-	int i;
-	unsigned long masterclock, masterprediv;
-
-	DEBUG_PRINT((DEBUG_NAME "asked: %d bits, ",spec->format & 0x00ff));
-	DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0)));
-	DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0)));
-	DEBUG_PRINT(("channels=%d, ", spec->channels));
-	DEBUG_PRINT(("freq=%d\n", spec->freq));
-
-    if (spec->channels > 2) {
-        spec->channels = 2;  /* no more than stereo! */
-    }
-
-	/* Check formats available */
-	MINTAUDIO_freqcount=0;
-	switch(cookie_mcsn->play) {
-		case MCSN_ST:
-			spec->channels=1;
-			spec->format=8; /* FIXME: is it signed or unsigned ? */
-			SDL_MintAudio_AddFrequency(this, 12500, 0, 0, -1);
-			break;
-		case MCSN_TT:	/* Also STE, Mega STE */
-			spec->format=AUDIO_S8;
-			masterclock=MASTERCLOCK_STE;
-			masterprediv=MASTERPREDIV_STE;
-			if ((cookie_mch>>16)==MCH_TT) {
-				masterclock=MASTERCLOCK_TT;
-				masterprediv=MASTERPREDIV_TT;
-			}
-			for (i=0; i<4; i++) {
-				SDL_MintAudio_AddFrequency(this, masterclock/(masterprediv*(1<<i)),
-					masterclock, 3-i, -1);
-			}
-			break;
-		case MCSN_FALCON:	/* Also Mac */
-			for (i=1; i<12; i++) {
-				/* Remove unusable Falcon codec predivisors */
-				if ((i==6) || (i==8) || (i==10)) {
-					continue;
-				}
-				SDL_MintAudio_AddFrequency(this, MASTERCLOCK_FALCON1/(MASTERPREDIV_FALCON*(i+1)),
-					CLK25M, i+1, -1);
-			}
-			if (cookie_mcsn->res1 != 0) {
-				for (i=1; i<4; i++) {
-					SDL_MintAudio_AddFrequency(this, (cookie_mcsn->res1)/(MASTERPREDIV_FALCON*(1<<i)),
-						CLKEXT, (1<<i)-1, -1);
-				}
-			}
-			spec->format |= 0x8000;	/* Audio is always signed */
-			if ((spec->format & 0x00ff)==16) {
-				spec->format |= 0x1000;	/* Audio is always big endian */
-				spec->channels=2;	/* 16 bits always stereo */
-			}
-			break;
-	}
-
-#if 0
-	for (i=0; i<MINTAUDIO_freqcount; i++) {
-		DEBUG_PRINT((DEBUG_NAME "freq %d: %lu Hz, clock %lu, prediv %d\n",
-			i, MINTAUDIO_frequencies[i].frequency, MINTAUDIO_frequencies[i].masterclock,
-			MINTAUDIO_frequencies[i].predivisor
-		));
-	}
-#endif
-
-	MINTAUDIO_numfreq=SDL_MintAudio_SearchFrequency(this, spec->freq);
-	spec->freq=MINTAUDIO_frequencies[MINTAUDIO_numfreq].frequency;
-
-	DEBUG_PRINT((DEBUG_NAME "obtained: %d bits, ",spec->format & 0x00ff));
-	DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0)));
-	DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0)));
-	DEBUG_PRINT(("channels=%d, ", spec->channels));
-	DEBUG_PRINT(("freq=%d\n", spec->freq));
-
-	return 0;
-}
-
-static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec)
-{
-	int channels_mode, prediv, dmaclock;
-	void *buffer;
-
-	/* Stop currently playing sound */
-	SDL_MintAudio_quit_thread = SDL_FALSE;
-	SDL_MintAudio_thread_finished = SDL_TRUE;
-	SDL_MintAudio_WaitThread();
-	Buffoper(0);
-
-	/* Set replay tracks */
-	Settracks(0,0);
-	Setmontracks(0);
-
-	/* Select replay format */
-	channels_mode=STEREO16;
-	switch (spec->format & 0xff) {
-		case 8:
-			if (spec->channels==2) {
-				channels_mode=STEREO8;
-			} else {
-				channels_mode=MONO8;
-			}
-			break;
-	}
-	if (Setmode(channels_mode)<0) {
-		DEBUG_PRINT((DEBUG_NAME "Setmode() failed\n"));
-	}
-
-	dmaclock = MINTAUDIO_frequencies[MINTAUDIO_numfreq].masterclock;
-	prediv = MINTAUDIO_frequencies[MINTAUDIO_numfreq].predivisor;
-	switch(cookie_mcsn->play) {
-		case MCSN_TT:
-			Devconnect(DMAPLAY, DAC, CLK25M, CLKOLD, 1);
-			Soundcmd(SETPRESCALE, prediv);
-			DEBUG_PRINT((DEBUG_NAME "STE/TT prescaler selected\n"));
-			break;
-		case MCSN_FALCON:
-			Devconnect(DMAPLAY, DAC, dmaclock, prediv, 1);
-			DEBUG_PRINT((DEBUG_NAME "Falcon prescaler selected\n"));
-			break;
-	}
-
-	/* Set buffer */
-	buffer = SDL_MintAudio_audiobuf[SDL_MintAudio_numbuf];
-	if (Setbuffer(0, buffer, buffer + spec->size)<0) {
-		DEBUG_PRINT((DEBUG_NAME "Setbuffer() failed\n"));
-	}
-	
-	if (SDL_MintAudio_mint_present) {
-		SDL_MintAudio_thread_pid = tfork(SDL_MintAudio_Thread, 0);
-	} else {
-		/* Install interrupt */
-		Jdisint(MFP_DMASOUND);
-		Xbtimer(XB_TIMERA, 8, 1, SDL_MintAudio_XbiosInterrupt);
-		Jenabint(MFP_DMASOUND);
-
-		if (Setinterrupt(SI_TIMERA, SI_PLAY)<0) {
-			DEBUG_PRINT((DEBUG_NAME "Setinterrupt() failed\n"));
-		}
-	}
-
-	/* Go */
-	Buffoper(SB_PLA_ENA|SB_PLA_RPT);
-	DEBUG_PRINT((DEBUG_NAME "hardware initialized\n"));
-}
-
-static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	/* Lock sound system */
-	if (Locksnd()!=1) {
-   	    SDL_SetError("Mint_OpenAudio: Audio system already in use");
-        return(-1);
-	}
-
-	SDL_MintAudio_device = this;
-
-	/* Check audio capabilities */
-	if (Mint_CheckAudio(this, spec)==-1) {
-		return -1;
-	}
-
-	SDL_CalculateAudioSpec(spec);
-
-	/* Allocate memory for audio buffers in DMA-able RAM */
-	DEBUG_PRINT((DEBUG_NAME "buffer size=%d\n", spec->size));
-
-	SDL_MintAudio_audiobuf[0] = Atari_SysMalloc(spec->size *2, MX_STRAM);
-	if (SDL_MintAudio_audiobuf[0]==NULL) {
-		SDL_SetError("MINT_OpenAudio: Not enough memory for audio buffer");
-		return (-1);
-	}
-	SDL_MintAudio_audiobuf[1] = SDL_MintAudio_audiobuf[0] + spec->size ;
-	SDL_MintAudio_numbuf=0;
-	SDL_memset(SDL_MintAudio_audiobuf[0], spec->silence, spec->size *2);
-	SDL_MintAudio_audiosize = spec->size;
-	SDL_MintAudio_mutex = 0;
-
-	DEBUG_PRINT((DEBUG_NAME "buffer 0 at 0x%08x\n", SDL_MintAudio_audiobuf[0]));
-	DEBUG_PRINT((DEBUG_NAME "buffer 1 at 0x%08x\n", SDL_MintAudio_audiobuf[1]));
-
-	SDL_MintAudio_CheckFpu();
-
-	/* Setup audio hardware */
-	Mint_InitAudio(this, spec);
-
-    return(1);	/* We don't use SDL threaded audio */
-}
diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_mcsn.h b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_mcsn.h
deleted file mode 100644
index b772fdab03..0000000000
--- a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_mcsn.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/*
-	MCSN control structure
-
-	Patrice Mandin
-*/
-
-#ifndef _SDL_mintaudio_mcsh_h
-#define _SDL_mintaudio_mcsh_h
-
-typedef struct {
-	unsigned short version;	/* Version */
-	unsigned short size;	/* Size of structure */
-
-	unsigned short play;	/* Replay capability */
-	unsigned short record;	/* Record capability */
-	unsigned short dsp;		/* DSP56K present */
-	unsigned short pint;	/* Interrupt at end of replay */
-	unsigned short rint;	/* Interrupt at end of record */
-
-	unsigned long res1;		/* Frequency of external clock */
-	unsigned long res2;
-	unsigned long res3;
-	unsigned long res4;
-} cookie_mcsn_t;
-
-enum {
-	MCSN_ST=0,
-	MCSN_TT,
-	MCSN_STE=MCSN_TT,
-	MCSN_FALCON,
-	MCSN_MAC=MCSN_FALCON
-};
-
-#define SETSMPFREQ	7	/* Set sample frequency */
-
-#endif /* _SDL_mintaudio_mcsh_h */
diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_stfa.c b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_stfa.c
deleted file mode 100644
index 4a581e0351..0000000000
--- a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_stfa.c
+++ /dev/null
@@ -1,326 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/*
-	MiNT audio driver
-	using XBIOS functions (STFA driver)
-
-	Patrice Mandin
-*/
-
-/* Mint includes */
-#include <mint/osbind.h>
-#include <mint/falcon.h>
-#include <mint/cookie.h>
-
-#include "SDL_audio.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_sysaudio.h"
-
-#include "../../video/ataricommon/SDL_atarimxalloc_c.h"
-#include "../../video/ataricommon/SDL_atarisuper.h"
-
-#include "SDL_mintaudio.h"
-#include "SDL_mintaudio_stfa.h"
-
-/*--- Defines ---*/
-
-#define MINT_AUDIO_DRIVER_NAME "mint_stfa"
-
-/* Debug print info */
-#define DEBUG_NAME "audio:stfa: "
-#if 0
-#define DEBUG_PRINT(what) \
-	{ \
-		printf what; \
-	}
-#else
-#define DEBUG_PRINT(what)
-#endif
-
-/*--- Static variables ---*/
-
-static long cookie_snd, cookie_mch;
-static cookie_stfa_t *cookie_stfa;
-
-static const int freqs[16]={
-	4995,	6269,	7493,	8192,
-	9830,	10971,	12538,	14985,
-	16384,	19819,	21943,	24576,
-	30720,	32336,	43885,	49152
-};
-
-/*--- Audio driver functions ---*/
-
-static void Mint_CloseAudio(_THIS);
-static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void Mint_LockAudio(_THIS);
-static void Mint_UnlockAudio(_THIS);
-
-/* To check/init hardware audio */
-static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec);
-static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec);
-
-/*--- Audio driver bootstrap functions ---*/
-
-static int Audio_Available(void)
-{
-	long dummy;
-	const char *envr = SDL_getenv("SDL_AUDIODRIVER");
-
-	/* Check if user asked a different audio driver */
-	if ((envr) && (SDL_strcmp(envr, MINT_AUDIO_DRIVER_NAME)!=0)) {
-		DEBUG_PRINT((DEBUG_NAME "user asked a different audio driver\n"));
-		return(0);
-	}
-
-	/* Cookie _MCH present ? if not, assume ST machine */
-	if (Getcookie(C__MCH, &cookie_mch) == C_NOTFOUND) {
-		cookie_mch = MCH_ST;
-	}
-
-	/* Cookie _SND present ? if not, assume ST machine */
-	if (Getcookie(C__SND, &cookie_snd) == C_NOTFOUND) {
-		cookie_snd = SND_PSG;
-	}
-
-	/* Cookie STFA present ? */
-	if (Getcookie(C_STFA, &dummy) != C_FOUND) {
-		DEBUG_PRINT((DEBUG_NAME "no STFA audio\n"));
-		return(0);
-	}
-	cookie_stfa = (cookie_stfa_t *) dummy;
-
-	SDL_MintAudio_stfa = cookie_stfa;
-
-	DEBUG_PRINT((DEBUG_NAME "STFA audio available!\n"));
-	return(1);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-    SDL_free(device->hidden);
-    SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-    if ( this ) {
-        SDL_memset(this, 0, (sizeof *this));
-        this->hidden = (struct SDL_PrivateAudioData *)
-                SDL_malloc((sizeof *this->hidden));
-    }
-    if ( (this == NULL) || (this->hidden == NULL) ) {
-        SDL_OutOfMemory();
-        if ( this ) {
-            SDL_free(this);
-        }
-        return(0);
-    }
-    SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-    /* Set the function pointers */
-    this->OpenAudio   = Mint_OpenAudio;
-    this->CloseAudio  = Mint_CloseAudio;
-    this->LockAudio   = Mint_LockAudio;
-    this->UnlockAudio = Mint_UnlockAudio;
-    this->free        = Audio_DeleteDevice;
-
-    return this;
-}
-
-AudioBootStrap MINTAUDIO_STFA_bootstrap = {
-	MINT_AUDIO_DRIVER_NAME, "MiNT STFA audio driver",
-	Audio_Available, Audio_CreateDevice
-};
-
-static void Mint_LockAudio(_THIS)
-{
-	void *oldpile;
-
-	/* Stop replay */
-	oldpile=(void *)Super(0);
-	cookie_stfa->sound_enable=STFA_PLAY_DISABLE;
-	SuperToUser(oldpile);
-}
-
-static void Mint_UnlockAudio(_THIS)
-{
-	void *oldpile;
-
-	/* Restart replay */
-	oldpile=(void *)Super(0);
-	cookie_stfa->sound_enable=STFA_PLAY_ENABLE|STFA_PLAY_REPEAT;
-	SuperToUser(oldpile);
-}
-
-static void Mint_CloseAudio(_THIS)
-{
-	void *oldpile;
-
-	/* Stop replay */
-	oldpile=(void *)Super(0);
-	cookie_stfa->sound_enable=STFA_PLAY_DISABLE;
-	SuperToUser(oldpile);
-
-	/* Wait if currently playing sound */
-	while (SDL_MintAudio_mutex != 0) {
-	}
-
-	/* Clear buffers */
-	if (SDL_MintAudio_audiobuf[0]) {
-		Mfree(SDL_MintAudio_audiobuf[0]);
-		SDL_MintAudio_audiobuf[0] = SDL_MintAudio_audiobuf[1] = NULL;
-	}
-}
-
-static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec)
-{
-	int i;
-
-	DEBUG_PRINT((DEBUG_NAME "asked: %d bits, ",spec->format & 0x00ff));
-	DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0)));
-	DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0)));
-	DEBUG_PRINT(("channels=%d, ", spec->channels));
-	DEBUG_PRINT(("freq=%d\n", spec->freq));
-
-    if (spec->channels > 2) {
-        spec->channels = 2;  /* no more than stereo! */
-    }
-
-	/* Check formats available */
-	MINTAUDIO_freqcount=0;
-	for (i=0;i<16;i++) {
-		SDL_MintAudio_AddFrequency(this, freqs[i], 0, i, -1);
-	}
-
-#if 1
-	for (i=0; i<MINTAUDIO_freqcount; i++) {
-		DEBUG_PRINT((DEBUG_NAME "freq %d: %lu Hz, clock %lu, prediv %d\n",
-			i, MINTAUDIO_frequencies[i].frequency, MINTAUDIO_frequencies[i].masterclock,
-			MINTAUDIO_frequencies[i].predivisor
-		));
-	}
-#endif
-
-	MINTAUDIO_numfreq=SDL_MintAudio_SearchFrequency(this, spec->freq);
-	spec->freq=MINTAUDIO_frequencies[MINTAUDIO_numfreq].frequency;
-
-	DEBUG_PRINT((DEBUG_NAME "obtained: %d bits, ",spec->format & 0x00ff));
-	DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0)));
-	DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0)));
-	DEBUG_PRINT(("channels=%d, ", spec->channels));
-	DEBUG_PRINT(("freq=%d\n", spec->freq));
-
-	return 0;
-}
-
-static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec)
-{
-	void *buffer;
-	void *oldpile;
-
-	buffer = SDL_MintAudio_audiobuf[SDL_MintAudio_numbuf];
-
-	oldpile=(void *)Super(0);
-
-	/* Stop replay */
-	cookie_stfa->sound_enable=STFA_PLAY_DISABLE;
-
-	/* Select replay format */
-	cookie_stfa->sound_control = MINTAUDIO_frequencies[MINTAUDIO_numfreq].predivisor;
-	if ((spec->format & 0xff)==8) {
-		cookie_stfa->sound_control |= STFA_FORMAT_8BIT;
-	} else {
-		cookie_stfa->sound_control |= STFA_FORMAT_16BIT;
-	}
-	if (spec->channels==2) {
-		cookie_stfa->sound_control |= STFA_FORMAT_STEREO;
-	} else {
-		cookie_stfa->sound_control |= STFA_FORMAT_MONO;
-	}
-	if ((spec->format & 0x8000)!=0) {
-		cookie_stfa->sound_control |= STFA_FORMAT_SIGNED;
-	} else {
-		cookie_stfa->sound_control |= STFA_FORMAT_UNSIGNED;
-	}
-	if ((spec->format & 0x1000)!=0) {
-		cookie_stfa->sound_control |= STFA_FORMAT_BIGENDIAN;
-	} else {
-		cookie_stfa->sound_control |= STFA_FORMAT_LITENDIAN;
-	}
-
-	/* Set buffer */
-	cookie_stfa->sound_start = (unsigned long) buffer;
-	cookie_stfa->sound_end = (unsigned long) (buffer + spec->size);
-
-	/* Set interrupt */
-	cookie_stfa->stfa_it = SDL_MintAudio_StfaInterrupt;
-
-	/* Restart replay */
-	cookie_stfa->sound_enable=STFA_PLAY_ENABLE|STFA_PLAY_REPEAT;
-
-	SuperToUser(oldpile);
-
-	DEBUG_PRINT((DEBUG_NAME "hardware initialized\n"));
-}
-
-static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	SDL_MintAudio_device = this;
-
-	/* Check audio capabilities */
-	if (Mint_CheckAudio(this, spec)==-1) {
-		return -1;
-	}
-
-	SDL_CalculateAudioSpec(spec);
-
-	/* Allocate memory for audio buffers in DMA-able RAM */
-	DEBUG_PRINT((DEBUG_NAME "buffer size=%d\n", spec->size));
-
-	SDL_MintAudio_audiobuf[0] = Atari_SysMalloc(spec->size *2, MX_STRAM);
-	if (SDL_MintAudio_audiobuf[0]==NULL) {
-		SDL_SetError("MINT_OpenAudio: Not enough memory for audio buffer");
-		return (-1);
-	}
-	SDL_MintAudio_audiobuf[1] = SDL_MintAudio_audiobuf[0] + spec->size ;
-	SDL_MintAudio_numbuf=0;
-	SDL_memset(SDL_MintAudio_audiobuf[0], spec->silence, spec->size *2);
-	SDL_MintAudio_audiosize = spec->size;
-	SDL_MintAudio_mutex = 0;
-
-	DEBUG_PRINT((DEBUG_NAME "buffer 0 at 0x%08x\n", SDL_MintAudio_audiobuf[0]));
-	DEBUG_PRINT((DEBUG_NAME "buffer 1 at 0x%08x\n", SDL_MintAudio_audiobuf[1]));
-
-	SDL_MintAudio_CheckFpu();
-
-	/* Setup audio hardware */
-	Mint_InitAudio(this, spec);
-
-    return(1);	/* We don't use threaded audio */
-}
diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_stfa.h b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_stfa.h
deleted file mode 100644
index 1789b4bb41..0000000000
--- a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_stfa.h
+++ /dev/null
@@ -1,97 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/*
-	STFA control structure
-
-	Patrice Mandin
-*/
-
-#ifndef _SDL_mintaudio_stfa_h
-#define _SDL_mintaudio_stfa_h
-
-/*--- Defines ---*/
-
-#define STFA_PLAY_ENABLE	(1<<0)
-#define STFA_PLAY_DISABLE	(0<<0)
-#define STFA_PLAY_REPEAT	(1<<1)
-#define STFA_PLAY_SINGLE	(0<<1)
-
-#define STFA_FORMAT_SIGNED		(1<<15)
-#define STFA_FORMAT_UNSIGNED	(0<<15)
-#define STFA_FORMAT_STEREO		(1<<14)
-#define STFA_FORMAT_MONO		(0<<14)
-#define STFA_FORMAT_16BIT		(1<<13)
-#define STFA_FORMAT_8BIT		(0<<13)
-#define STFA_FORMAT_LITENDIAN	(1<<9)
-#define STFA_FORMAT_BIGENDIAN	(0<<9)
-#define STFA_FORMAT_FREQ_MASK	0x0f
-enum {
-	STFA_FORMAT_F4995=0,
-	STFA_FORMAT_F6269,
-	STFA_FORMAT_F7493,
-	STFA_FORMAT_F8192,
-
-	STFA_FORMAT_F9830,
-	STFA_FORMAT_F10971,
-	STFA_FORMAT_F12538,
-	STFA_FORMAT_F14985,
-
-	STFA_FORMAT_F16384,
-	STFA_FORMAT_F19819,
-	STFA_FORMAT_F21943,
-	STFA_FORMAT_F24576,
-
-	STFA_FORMAT_F30720,
-	STFA_FORMAT_F32336,
-	STFA_FORMAT_F43885,
-	STFA_FORMAT_F49152
-};
-
-/*--- Types ---*/
-
-typedef struct {
-	unsigned short sound_enable;
-	unsigned short sound_control;
-	unsigned short sound_output;
-	unsigned long sound_start;
-	unsigned long sound_current;
-	unsigned long sound_end;
-	unsigned short version;
-	void *old_vbl;
-	void *old_timera;
-	unsigned long old_mfp_status;
-	void *new_vbl;
-	void *drivers_list;
-	void *play_stop;
-	unsigned short frequency;
-	void *set_frequency;
-	unsigned short frequency_threshold;
-	unsigned short *custom_freq_table;
-	unsigned short stfa_on_off;
-	void *new_drivers_list;
-	unsigned long old_bit_2_of_cookie_snd;
-	void (*stfa_it)(void);
-} cookie_stfa_t;
-
-#endif /* _SDL_mintaudio_stfa_h */
diff --git a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_xbios.c b/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_xbios.c
deleted file mode 100644
index 42a0d4a2f9..0000000000
--- a/apps/plugins/sdl/src/audio/mint/SDL_mintaudio_xbios.c
+++ /dev/null
@@ -1,490 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/*
-	MiNT audio driver
-	using XBIOS functions (Falcon)
-
-	Patrice Mandin, Didier Méquignon
-*/
-
-#include <unistd.h>
-#include <support.h>
-
-/* Mint includes */
-#include <mint/osbind.h>
-#include <mint/falcon.h>
-#include <mint/cookie.h>
-
-#include "SDL_audio.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_sysaudio.h"
-
-#include "../../video/ataricommon/SDL_atarimxalloc_c.h"
-#include "../../video/ataricommon/SDL_atarisuper.h"
-
-#include "SDL_mintaudio.h"
-#include "SDL_mintaudio_dma8.h"
-
-/*--- Defines ---*/
-
-#define MINT_AUDIO_DRIVER_NAME "mint_xbios"
-
-/* Debug print info */
-#define DEBUG_NAME "audio:xbios: "
-#if 0
-#define DEBUG_PRINT(what) \
-	{ \
-		printf what; \
-	}
-#else
-#define DEBUG_PRINT(what)
-#endif
-
-/*--- Static variables ---*/
-
-static long cookie_snd;
-
-/*--- Audio driver functions ---*/
-
-static void Mint_CloseAudio(_THIS);
-static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void Mint_LockAudio(_THIS);
-static void Mint_UnlockAudio(_THIS);
-
-/* To check/init hardware audio */
-static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec);
-static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec);
-
-/*--- Audio driver bootstrap functions ---*/
-
-static int Audio_Available(void)
-{
-/*	unsigned long dummy;*/
-	const char *envr = SDL_getenv("SDL_AUDIODRIVER");
-
-	/*SDL_MintAudio_mint_present = (Getcookie(C_MiNT, &dummy) == C_FOUND);*/
-	SDL_MintAudio_mint_present = SDL_FALSE;
-
-	/* We can't use XBIOS in interrupt with Magic, don't know about thread */
-	/*if (Getcookie(C_MagX, &dummy) == C_FOUND) {
-		return(0);
-	}*/
-
-	/* Check if user asked a different audio driver */
-	if ((envr) && (SDL_strcmp(envr, MINT_AUDIO_DRIVER_NAME)!=0)) {
-		DEBUG_PRINT((DEBUG_NAME "user asked a different audio driver\n"));
-		return(0);
-	}
-
-	/* Cookie _SND present ? if not, assume ST machine */
-	if (Getcookie(C__SND, &cookie_snd) == C_NOTFOUND) {
-		cookie_snd = SND_PSG;
-	}
-
-	/* Check if we have 16 bits audio */
-	if ((cookie_snd & SND_16BIT)==0) {
-		DEBUG_PRINT((DEBUG_NAME "no 16 bits sound\n"));
-		return(0);
-	}
-
-	/* Check if audio is lockable */
-	if (Locksnd()!=1) {
-		DEBUG_PRINT((DEBUG_NAME "audio locked by other application\n"));
-		return(0);
-	}
-
-	Unlocksnd();
-
-	DEBUG_PRINT((DEBUG_NAME "XBIOS audio available!\n"));
-	return(1);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-    SDL_free(device->hidden);
-    SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-    if ( this ) {
-        SDL_memset(this, 0, (sizeof *this));
-        this->hidden = (struct SDL_PrivateAudioData *)
-                SDL_malloc((sizeof *this->hidden));
-    }
-    if ( (this == NULL) || (this->hidden == NULL) ) {
-        SDL_OutOfMemory();
-        if ( this ) {
-            SDL_free(this);
-        }
-        return(0);
-    }
-    SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-    /* Set the function pointers */
-    this->OpenAudio   = Mint_OpenAudio;
-    this->CloseAudio  = Mint_CloseAudio;
-    this->LockAudio   = Mint_LockAudio;
-    this->UnlockAudio = Mint_UnlockAudio;
-    this->free        = Audio_DeleteDevice;
-
-    return this;
-}
-
-AudioBootStrap MINTAUDIO_XBIOS_bootstrap = {
-	MINT_AUDIO_DRIVER_NAME, "MiNT XBIOS audio driver",
-	Audio_Available, Audio_CreateDevice
-};
-
-static void Mint_LockAudio(_THIS)
-{
-	/* Stop replay */
-	Buffoper(0);
-}
-
-static void Mint_UnlockAudio(_THIS)
-{
-	/* Restart replay */
-	Buffoper(SB_PLA_ENA|SB_PLA_RPT);
-}
-
-static void Mint_CloseAudio(_THIS)
-{
-	/* Stop replay */
-	SDL_MintAudio_WaitThread();
-	Buffoper(0);
-
-	if (!SDL_MintAudio_mint_present) {
-		/* Uninstall interrupt */
-		Jdisint(MFP_DMASOUND);
-	}
-
-	/* Wait if currently playing sound */
-	while (SDL_MintAudio_mutex != 0) {
-	}
-
-	/* Clear buffers */
-	if (SDL_MintAudio_audiobuf[0]) {
-		Mfree(SDL_MintAudio_audiobuf[0]);
-		SDL_MintAudio_audiobuf[0] = SDL_MintAudio_audiobuf[1] = NULL;
-	}
-
-	/* Unlock sound system */
-	Unlocksnd();
-}
-
-/* Falcon XBIOS implementation of Devconnect() is buggy with external clock */
-static void Devconnect2(int src, int dst, int sclk, int pre)
-{		
-	static const unsigned short MASK1[3] = { 0, 0x6000, 0 };
-	static const unsigned short MASK2[4] = { 0xFFF0, 0xFF8F, 0xF0FF, 0x0FFF };
-	static const unsigned short INDEX1[4] = {  1, 3, 5, 7 };
-	static const unsigned short INDEX2[4] = {  0, 2, 4, 6 };
-	unsigned short sync_div,dev_ctrl,dest_ctrl;
-	void *oldstack;
-
-	if (dst==0) {
-		return;
-	}
-
-	oldstack=(void *)Super(0);
-
-	dev_ctrl = DMAAUDIO_IO.dev_ctrl;
-	dest_ctrl = DMAAUDIO_IO.dest_ctrl;
-	dev_ctrl &= MASK2[src];
-
-	if (src==ADC) {
-		dev_ctrl |= MASK1[sclk];
-	} else {
-		dev_ctrl |= (INDEX1[sclk] << (src<<4));
-	}
-
-	if (dst & DMAREC) {		
-		dest_ctrl &= 0xFFF0;
-		dest_ctrl |= INDEX1[src];
-	}
-
-	if (dst & DSPRECV) {		
-		dest_ctrl &= 0xFF8F;
-		dest_ctrl |= (INDEX1[src]<<4); 
-	}
-
-	if (dst & EXTOUT) {		
-		dest_ctrl &= 0xF0FF;
-		dest_ctrl |= (INDEX1[src]<<8); 
-	}
-
-	if (dst & DAC) {		
-		dev_ctrl &= 0x0FFF;
-		dev_ctrl |= MASK1[sclk]; 
-		dest_ctrl &=  0x0FFF;
-		dest_ctrl |= (INDEX2[src]<<12); 
-	}
-
-	sync_div = DMAAUDIO_IO.sync_div;
-	if (sclk==CLKEXT) {
-		pre<<=8;
-		sync_div &= 0xF0FF;
-	} else {
-		sync_div &= 0xFFF0;
-	}
-	sync_div |= pre;
-
-	DMAAUDIO_IO.dev_ctrl = dev_ctrl;
-	DMAAUDIO_IO.dest_ctrl = dest_ctrl;
-	DMAAUDIO_IO.sync_div = sync_div;
-
-	SuperToUser(oldstack);
-}
-
-static void Mint_CheckExternalClock(_THIS)
-{
-#define SIZE_BUF_CLOCK_MEASURE (44100/10)
-
-	char *buffer;
-	int i, j;
-
-	/* DSP present with its GPIO port ? */
-	if ((cookie_snd & SND_DSP)==0) {
-		return;
-	}
-
-	buffer = Atari_SysMalloc(SIZE_BUF_CLOCK_MEASURE, MX_STRAM);
-	if (buffer==NULL) {
-		DEBUG_PRINT((DEBUG_NAME "Not enough memory for the measure\n"));
-		return;
-	}
-	SDL_memset(buffer, 0, SIZE_BUF_CLOCK_MEASURE);
-
-	Buffoper(0);
-	Settracks(0,0);
-	Setmontracks(0);
-	Setmode(MONO8);
-	Jdisint(MFP_TIMERA);
-
-	for (i=0; i<2; i++) {
-		Gpio(GPIO_SET,7);      /* DSP port gpio outputs */
-		Gpio(GPIO_WRITE,2+i);  /* 22.5792/24.576 MHz for 44.1/48KHz */
-		Devconnect2(DMAPLAY, DAC, CLKEXT, CLK50K);  /* Matrix and clock source */
-		Setbuffer(0, buffer, buffer + SIZE_BUF_CLOCK_MEASURE);		           /* Set buffer */
-		Xbtimer(XB_TIMERA, 5, 38, SDL_MintAudio_XbiosInterruptMeasureClock); /* delay mode timer A, prediv /64, 1KHz */
-		Jenabint(MFP_TIMERA);
-		SDL_MintAudio_clocktics = 0;
-		Buffoper(SB_PLA_ENA);
-		usleep(110000);
-
-		if((Buffoper(-1) & 1)==0) {
-			if (SDL_MintAudio_clocktics) {
-				unsigned long khz;
-
-				khz = ((SIZE_BUF_CLOCK_MEASURE/SDL_MintAudio_clocktics) +1) & 0xFFFFFFFE;
-				DEBUG_PRINT((DEBUG_NAME "measure %d: freq=%lu KHz\n", i+1, khz));
-
-				if(khz==44) {
-					for (j=1; j<4; j++) {
-						SDL_MintAudio_AddFrequency(this, MASTERCLOCK_44K/(MASTERPREDIV_FALCON*(1<<j)), MASTERCLOCK_44K, (1<<j)-1, 2+i);
-					}
-				} else if (khz==48) {
-					for (j=1; j<4; j++) {
-						SDL_MintAudio_AddFrequency(this, MASTERCLOCK_48K/(MASTERPREDIV_FALCON*(1<<j)), MASTERCLOCK_48K, (1<<j)-1, 2+i);
-					}
-				}
-			} else {
-				DEBUG_PRINT((DEBUG_NAME "No measure\n"));
-			}
-		} else {
-			DEBUG_PRINT((DEBUG_NAME "No SDMA clock\n"));
-		}
-
-		Buffoper(0);             /* stop */
-		Jdisint(MFP_TIMERA);     /* Uninstall interrupt */
-	}
-
-	Mfree(buffer);
-}
-
-static int Mint_CheckAudio(_THIS, SDL_AudioSpec *spec)
-{
-	int i;
-
-	DEBUG_PRINT((DEBUG_NAME "asked: %d bits, ",spec->format & 0x00ff));
-	DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0)));
-	DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0)));
-	DEBUG_PRINT(("channels=%d, ", spec->channels));
-	DEBUG_PRINT(("freq=%d\n", spec->freq));
-
-    if (spec->channels > 2) {
-        spec->channels = 2;  /* no more than stereo! */
-    }
-
-	spec->format |= 0x8000;	/* Audio is always signed */
-	if ((spec->format & 0x00ff)==16) {
-		spec->format |= 0x1000;	/* Audio is always big endian */
-		spec->channels=2;	/* 16 bits always stereo */
-	}
-
-	MINTAUDIO_freqcount=0;
-
-	/* Add external clocks if present */
-	Mint_CheckExternalClock(this);
-
-	/* Standard clocks */
-	for (i=1;i<12;i++) {
-		/* Remove unusable Falcon codec predivisors */
-		if ((i==6) || (i==8) || (i==10)) {
-			continue;
-		}
-		SDL_MintAudio_AddFrequency(this, MASTERCLOCK_FALCON1/(MASTERPREDIV_FALCON*(i+1)), MASTERCLOCK_FALCON1, i, -1);
-	}
-
-#if 1
-	for (i=0; i<MINTAUDIO_freqcount; i++) {
-		DEBUG_PRINT((DEBUG_NAME "freq %d: %lu Hz, clock %lu, prediv %d\n",
-			i, MINTAUDIO_frequencies[i].frequency, MINTAUDIO_frequencies[i].masterclock,
-			MINTAUDIO_frequencies[i].predivisor
-		));
-	}
-#endif
-
-	MINTAUDIO_numfreq=SDL_MintAudio_SearchFrequency(this, spec->freq);
-	spec->freq=MINTAUDIO_frequencies[MINTAUDIO_numfreq].frequency;
-
-	DEBUG_PRINT((DEBUG_NAME "obtained: %d bits, ",spec->format & 0x00ff));
-	DEBUG_PRINT(("signed=%d, ", ((spec->format & 0x8000)!=0)));
-	DEBUG_PRINT(("big endian=%d, ", ((spec->format & 0x1000)!=0)));
-	DEBUG_PRINT(("channels=%d, ", spec->channels));
-	DEBUG_PRINT(("freq=%d\n", spec->freq));
-
-	return 0;
-}
-
-static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec)
-{
-	int channels_mode, prediv;
-	void *buffer;
-
-	/* Stop currently playing sound */
-	SDL_MintAudio_quit_thread = SDL_FALSE;
-	SDL_MintAudio_thread_finished = SDL_TRUE;
-	SDL_MintAudio_WaitThread();
-	Buffoper(0);
-
-	/* Set replay tracks */
-	Settracks(0,0);
-	Setmontracks(0);
-
-	/* Select replay format */
-	channels_mode=STEREO16;
-	switch (spec->format & 0xff) {
-		case 8:
-			if (spec->channels==2) {
-				channels_mode=STEREO8;
-			} else {
-				channels_mode=MONO8;
-			}
-			break;
-	}
-	if (Setmode(channels_mode)<0) {
-		DEBUG_PRINT((DEBUG_NAME "Setmode() failed\n"));
-	}
-
-	prediv = MINTAUDIO_frequencies[MINTAUDIO_numfreq].predivisor;
-	if (MINTAUDIO_frequencies[MINTAUDIO_numfreq].gpio_bits != -1) {
-		Gpio(GPIO_SET,7);		/* DSP port gpio outputs */
-		Gpio(GPIO_WRITE, MINTAUDIO_frequencies[MINTAUDIO_numfreq].gpio_bits);
-		Devconnect2(DMAPLAY, DAC|EXTOUT, CLKEXT, prediv);
-	} else {
-		Devconnect2(DMAPLAY, DAC, CLK25M, prediv);
-	}
-
-	/* Set buffer */
-	buffer = SDL_MintAudio_audiobuf[SDL_MintAudio_numbuf];
-	if (Setbuffer(0, buffer, buffer + spec->size)<0) {
-		DEBUG_PRINT((DEBUG_NAME "Setbuffer() failed\n"));
-	}
-	
-	if (SDL_MintAudio_mint_present) {
-		SDL_MintAudio_thread_pid = tfork(SDL_MintAudio_Thread, 0);
-	} else {
-		/* Install interrupt */
-		Jdisint(MFP_DMASOUND);
-		/*Xbtimer(XB_TIMERA, 8, 1, SDL_MintAudio_XbiosInterrupt);*/
-		Xbtimer(XB_TIMERA, 8, 1, SDL_MintAudio_Dma8Interrupt);
-		Jenabint(MFP_DMASOUND);
-
-		if (Setinterrupt(SI_TIMERA, SI_PLAY)<0) {
-			DEBUG_PRINT((DEBUG_NAME "Setinterrupt() failed\n"));
-		}
-	}
-
-	/* Go */
-	Buffoper(SB_PLA_ENA|SB_PLA_RPT);
-	DEBUG_PRINT((DEBUG_NAME "hardware initialized\n"));
-}
-
-static int Mint_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	/* Lock sound system */
-	if (Locksnd()!=1) {
-   	    SDL_SetError("Mint_OpenAudio: Audio system already in use");
-        return(-1);
-	}
-
-	SDL_MintAudio_device = this;
-
-	/* Check audio capabilities */
-	if (Mint_CheckAudio(this, spec)==-1) {
-		return -1;
-	}
-
-	SDL_CalculateAudioSpec(spec);
-
-	/* Allocate memory for audio buffers in DMA-able RAM */
-	DEBUG_PRINT((DEBUG_NAME "buffer size=%d\n", spec->size));
-
-	SDL_MintAudio_audiobuf[0] = Atari_SysMalloc(spec->size *2, MX_STRAM);
-	if (SDL_MintAudio_audiobuf[0]==NULL) {
-		SDL_SetError("MINT_OpenAudio: Not enough memory for audio buffer");
-		return (-1);
-	}
-	SDL_MintAudio_audiobuf[1] = SDL_MintAudio_audiobuf[0] + spec->size ;
-	SDL_MintAudio_numbuf=0;
-	SDL_memset(SDL_MintAudio_audiobuf[0], spec->silence, spec->size *2);
-	SDL_MintAudio_audiosize = spec->size;
-	SDL_MintAudio_mutex = 0;
-
-	DEBUG_PRINT((DEBUG_NAME "buffer 0 at 0x%08x\n", SDL_MintAudio_audiobuf[0]));
-	DEBUG_PRINT((DEBUG_NAME "buffer 1 at 0x%08x\n", SDL_MintAudio_audiobuf[1]));
-
-	SDL_MintAudio_CheckFpu();
-
-	/* Setup audio hardware */
-	Mint_InitAudio(this, spec);
-
-    return(1);	/* We don't use SDL threaded audio */
-}
diff --git a/apps/plugins/sdl/src/audio/mme/SDL_mmeaudio.c b/apps/plugins/sdl/src/audio/mme/SDL_mmeaudio.c
deleted file mode 100644
index 64a0ecc36d..0000000000
--- a/apps/plugins/sdl/src/audio/mme/SDL_mmeaudio.c
+++ /dev/null
@@ -1,264 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/* Tru64 UNIX MME support */
-#include <mme_api.h>
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audio_c.h"
-#include "SDL_mmeaudio.h"
-
-static BOOL inUse[NUM_BUFFERS];
-
-/* Audio driver functions */
-static int MME_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void MME_WaitAudio(_THIS);
-static Uint8 *MME_GetAudioBuf(_THIS);
-static void MME_PlayAudio(_THIS);
-static void MME_WaitDone(_THIS);
-static void MME_CloseAudio(_THIS);
-
-/* Audio driver bootstrap functions */
-static int Audio_Available(void)
-{
-    return(1);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-    if ( device ) {
-	if ( device->hidden ) {
-	    SDL_free(device->hidden);
-	    device->hidden = NULL;
-	}
-	SDL_free(device);
-	device = NULL;
-    }
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-    SDL_AudioDevice *this;
-
-/* Initialize all variables that we clean on shutdown */
-    this = SDL_malloc(sizeof(SDL_AudioDevice));
-    if ( this ) {
-	SDL_memset(this, 0, (sizeof *this));
-	this->hidden = SDL_malloc((sizeof *this->hidden));
-    }
-    if ( (this == NULL) || (this->hidden == NULL) ) {
-	SDL_OutOfMemory();
-	if ( this ) {
-	    SDL_free(this);
-	}
-	return(0);
-    }
-    SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-    /* Set the function pointers */
-    this->OpenAudio       =       MME_OpenAudio;
-    this->WaitAudio       =       MME_WaitAudio;
-    this->PlayAudio       =       MME_PlayAudio;
-    this->GetAudioBuf     =     MME_GetAudioBuf;
-    this->WaitDone        =        MME_WaitDone;
-    this->CloseAudio      =      MME_CloseAudio;
-    this->free            =  Audio_DeleteDevice;
-
-    return this;
-}
-
-AudioBootStrap MMEAUDIO_bootstrap = {
-    "waveout", "Tru64 MME WaveOut",
-    Audio_Available, Audio_CreateDevice
-};
-
-static void SetMMerror(char *function, MMRESULT code)
-{
-    int len;
-    char errbuf[MAXERRORLENGTH];
-
-    SDL_snprintf(errbuf, SDL_arraysize(errbuf), "%s: ", function);
-    len = SDL_strlen(errbuf);
-    waveOutGetErrorText(code, errbuf+len, MAXERRORLENGTH-len);
-    SDL_SetError("%s",errbuf);
-}
-
-static void CALLBACK MME_CALLBACK(HWAVEOUT hwo,
-				  UINT uMsg,
-				  DWORD dwInstance,
-				  LPARAM dwParam1,
-				  LPARAM dwParam2)
-{
-    WAVEHDR *wp = (WAVEHDR *) dwParam1;
-
-    if ( uMsg == WOM_DONE )
-	inUse[wp->dwUser] = FALSE;
-}
-
-static int MME_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-    MMRESULT result;
-    int i;
-
-    mixbuf = NULL;
-
-    /* Set basic WAVE format parameters */
-    shm = mmeAllocMem(sizeof(*shm));
-    if ( shm == NULL ) {
-	SDL_SetError("Out of memory: shm");
-	return(-1);
-    }
-    shm->sound = 0;
-    shm->wFmt.wf.wFormatTag = WAVE_FORMAT_PCM;
-
-    /* Determine the audio parameters from the AudioSpec */
-    switch ( spec->format & 0xFF ) {
-	case 8:
-	    /* Unsigned 8 bit audio data */
-	    spec->format = AUDIO_U8;
-	    shm->wFmt.wBitsPerSample = 8;
-	    break;
-	case 16:
-	    /* Signed 16 bit audio data */
-	    spec->format = AUDIO_S16;
-	    shm->wFmt.wBitsPerSample = 16;
-	    break;
-	    default:
-	    SDL_SetError("Unsupported audio format");
-	    return(-1);
-    }
-
-    shm->wFmt.wf.nChannels = spec->channels;
-    shm->wFmt.wf.nSamplesPerSec = spec->freq;
-    shm->wFmt.wf.nBlockAlign =
-	shm->wFmt.wf.nChannels * shm->wFmt.wBitsPerSample / 8;
-    shm->wFmt.wf.nAvgBytesPerSec =
-	shm->wFmt.wf.nSamplesPerSec * shm->wFmt.wf.nBlockAlign;
-
-    /* Check the buffer size -- minimum of 1/4 second (word aligned) */
-    if ( spec->samples < (spec->freq/4) )
-	spec->samples = ((spec->freq/4)+3)&~3;
-
-    /* Update the fragment size as size in bytes */
-    SDL_CalculateAudioSpec(spec);
-
-    /* Open the audio device */
-    result = waveOutOpen(&(shm->sound),
-			 WAVE_MAPPER,
-			 &(shm->wFmt.wf),
-			 MME_CALLBACK,
-			 NULL,
-			 (CALLBACK_FUNCTION|WAVE_OPEN_SHAREABLE));
-    if ( result != MMSYSERR_NOERROR ) {
-	    SetMMerror("waveOutOpen()", result);
-	    return(-1);
-    }
-
-    /* Create the sound buffers */
-    mixbuf = (Uint8 *)mmeAllocBuffer(NUM_BUFFERS * (spec->size));
-    if ( mixbuf == NULL ) {
-	SDL_SetError("Out of memory: mixbuf");
-	return(-1);
-    }
-
-    for (i = 0; i < NUM_BUFFERS; i++) {
-	shm->wHdr[i].lpData         = &mixbuf[i * (spec->size)];
-	shm->wHdr[i].dwBufferLength = spec->size;
-	shm->wHdr[i].dwFlags        = 0;
-	shm->wHdr[i].dwUser         = i;
-	shm->wHdr[i].dwLoops        = 0;       /* loop control counter */
-	shm->wHdr[i].lpNext         = NULL;    /* reserved for driver */
-	shm->wHdr[i].reserved       = 0;
-	inUse[i] = FALSE;
-    }
-    next_buffer = 0;
-    return 0;
-}
-
-static void MME_WaitAudio(_THIS)
-{
-    while ( inUse[next_buffer] ) {
-	mmeWaitForCallbacks();
-	mmeProcessCallbacks();
-    }
-}
-
-static Uint8 *MME_GetAudioBuf(_THIS)
-{
-    Uint8 *retval;
-
-    inUse[next_buffer] = TRUE;
-    retval = (Uint8 *)(shm->wHdr[next_buffer].lpData);
-    return retval;
-}
-
-static void MME_PlayAudio(_THIS)
-{
-    /* Queue it up */
-    waveOutWrite(shm->sound, &(shm->wHdr[next_buffer]), sizeof(WAVEHDR));
-    next_buffer = (next_buffer+1)%NUM_BUFFERS;
-}
-
-static void MME_WaitDone(_THIS)
-{
-    MMRESULT result;
-    int i;
-
-    if ( shm->sound ) {
-	for (i = 0; i < NUM_BUFFERS; i++)
-	    while ( inUse[i] ) {
-		mmeWaitForCallbacks();
-		mmeProcessCallbacks();
-	    }
-	result = waveOutReset(shm->sound);
-	if ( result != MMSYSERR_NOERROR )
-	    SetMMerror("waveOutReset()", result);
-	mmeProcessCallbacks();
-    }
-}
-
-static void MME_CloseAudio(_THIS)
-{
-    MMRESULT result;
-
-    if ( mixbuf ) {
-	result = mmeFreeBuffer(mixbuf);
-	if (result != MMSYSERR_NOERROR )
-	    SetMMerror("mmeFreeBuffer", result);
-	mixbuf = NULL;
-    }
-
-    if ( shm ) {
-	if ( shm->sound ) {
-	    result = waveOutClose(shm->sound);
-	    if (result != MMSYSERR_NOERROR )
-		SetMMerror("waveOutClose()", result);
-	    mmeProcessCallbacks();
-	}
-	result = mmeFreeMem(shm);
-	if (result != MMSYSERR_NOERROR )
-	    SetMMerror("mmeFreeMem()", result);
-	shm = NULL;
-    }
-}
-
diff --git a/apps/plugins/sdl/src/audio/mme/SDL_mmeaudio.h b/apps/plugins/sdl/src/audio/mme/SDL_mmeaudio.h
deleted file mode 100644
index 6bfaed32e0..0000000000
--- a/apps/plugins/sdl/src/audio/mme/SDL_mmeaudio.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/* Allow access to a raw mixing buffer */
-
-#ifndef _SDL_lowaudio_h
-#define _SDL_lowaudio_h
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-#define NUM_BUFFERS 2
-
-struct SharedMem {
-    HWAVEOUT sound;
-    WAVEHDR wHdr[NUM_BUFFERS];
-    PCMWAVEFORMAT wFmt;
-};
-
-struct SDL_PrivateAudioData {
-    Uint8 *mixbuf;          /* The raw allocated mixing buffer */
-    struct SharedMem *shm;
-    int next_buffer;
-};
-
-#define shm			(this->hidden->shm)
-#define mixbuf			(this->hidden->mixbuf)
-#define next_buffer		(this->hidden->next_buffer)
-/* Old variable names */
-#endif /* _SDL_lowaudio_h */
diff --git a/apps/plugins/sdl/src/audio/nas/SDL_nasaudio.c b/apps/plugins/sdl/src/audio/nas/SDL_nasaudio.c
deleted file mode 100644
index a561e62984..0000000000
--- a/apps/plugins/sdl/src/audio/nas/SDL_nasaudio.c
+++ /dev/null
@@ -1,423 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-
-    This driver was written by:
-    Erik Inge Bolsø
-    knan@mo.himolde.no
-*/
-#include "SDL_config.h"
-
-/* Allow access to a raw mixing buffer */
-
-#include <signal.h>
-#include <unistd.h>
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_audiodev_c.h"
-#include "SDL_nasaudio.h"
-
-#ifdef SDL_AUDIO_DRIVER_NAS_DYNAMIC
-#include "SDL_loadso.h"
-#endif
-
-/* The tag name used by artsc audio */
-#define NAS_DRIVER_NAME         "nas"
-
-static struct SDL_PrivateAudioData *this2 = NULL;
-
-static void (*NAS_AuCloseServer) (AuServer *);
-static void (*NAS_AuNextEvent) (AuServer *, AuBool, AuEvent *);
-static AuBool(*NAS_AuDispatchEvent) (AuServer *, AuEvent *);
-static AuFlowID(*NAS_AuCreateFlow) (AuServer *, AuStatus *);
-static void (*NAS_AuStartFlow) (AuServer *, AuFlowID, AuStatus *);
-static void (*NAS_AuSetElements)
-  (AuServer *, AuFlowID, AuBool, int, AuElement *, AuStatus *);
-static void (*NAS_AuWriteElement)
-  (AuServer *, AuFlowID, int, AuUint32, AuPointer, AuBool, AuStatus *);
-static AuServer *(*NAS_AuOpenServer)
-  (_AuConst char *, int, _AuConst char *, int, _AuConst char *, char **);
-static AuEventHandlerRec *(*NAS_AuRegisterEventHandler)
-  (AuServer *, AuMask, int, AuID, AuEventHandlerCallback, AuPointer);
-
-
-#ifdef SDL_AUDIO_DRIVER_NAS_DYNAMIC
-
-static const char *nas_library = SDL_AUDIO_DRIVER_NAS_DYNAMIC;
-static void *nas_handle = NULL;
-
-static int
-load_nas_sym(const char *fn, void **addr)
-{
-    *addr = SDL_LoadFunction(nas_handle, fn);
-    if (*addr == NULL) {
-        return 0;
-    }
-    return 1;
-}
-
-/* cast funcs to char* first, to please GCC's strict aliasing rules. */
-#define SDL_NAS_SYM(x) \
-    if (!load_nas_sym(#x, (void **) (char *) &NAS_##x)) return -1
-#else
-#define SDL_NAS_SYM(x) NAS_##x = x
-#endif
-
-static int
-load_nas_syms(void)
-{
-    SDL_NAS_SYM(AuCloseServer);
-    SDL_NAS_SYM(AuNextEvent);
-    SDL_NAS_SYM(AuDispatchEvent);
-    SDL_NAS_SYM(AuCreateFlow);
-    SDL_NAS_SYM(AuStartFlow);
-    SDL_NAS_SYM(AuSetElements);
-    SDL_NAS_SYM(AuWriteElement);
-    SDL_NAS_SYM(AuOpenServer);
-    SDL_NAS_SYM(AuRegisterEventHandler);
-    return 0;
-}
-
-#undef SDL_NAS_SYM
-
-#ifdef SDL_AUDIO_DRIVER_NAS_DYNAMIC
-
-static void
-UnloadNASLibrary(void)
-{
-    if (nas_handle != NULL) {
-        SDL_UnloadObject(nas_handle);
-        nas_handle = NULL;
-    }
-}
-
-static int
-LoadNASLibrary(void)
-{
-    int retval = 0;
-    if (nas_handle == NULL) {
-        nas_handle = SDL_LoadObject(nas_library);
-        if (nas_handle == NULL) {
-            /* Copy error string so we can use it in a new SDL_SetError(). */
-            char *origerr = SDL_GetError();
-            size_t len = SDL_strlen(origerr) + 1;
-            char *err = (char *) alloca(len);
-            SDL_strlcpy(err, origerr, len);
-            retval = -1;
-            SDL_SetError("NAS: SDL_LoadObject('%s') failed: %s\n",
-                         nas_library, err);
-        } else {
-            retval = load_nas_syms();
-            if (retval < 0) {
-                UnloadNASLibrary();
-            }
-        }
-    }
-    return retval;
-}
-
-#else
-
-static void
-UnloadNASLibrary(void)
-{
-}
-
-static int
-LoadNASLibrary(void)
-{
-    load_nas_syms();
-    return 0;
-}
-
-#endif /* SDL_AUDIO_DRIVER_NAS_DYNAMIC */
-
-
-/* Audio driver functions */
-static int NAS_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void NAS_WaitAudio(_THIS);
-static void NAS_PlayAudio(_THIS);
-static Uint8 *NAS_GetAudioBuf(_THIS);
-static void NAS_CloseAudio(_THIS);
-
-/* Audio driver bootstrap functions */
-
-static int Audio_Available(void)
-{
-	if (LoadNASLibrary() == 0) {
-		AuServer *aud = NAS_AuOpenServer("", 0, NULL, 0, NULL, NULL);
-		if (!aud) {
-			UnloadNASLibrary();
-			return 0;
-		}
-		NAS_AuCloseServer(aud);
-		UnloadNASLibrary();
-		return 1;
-	}
-	return 0;
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-	UnloadNASLibrary();
-	SDL_free(device->hidden);
-	SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	if (LoadNASLibrary() < 0) {
-		return NULL;
-	}
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-	if ( this ) {
-		SDL_memset(this, 0, (sizeof *this));
-		this->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *this->hidden));
-	}
-	if ( (this == NULL) || (this->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( this ) {
-			SDL_free(this);
-		}
-		return NULL;
-	}
-	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-	/* Set the function pointers */
-	this->OpenAudio = NAS_OpenAudio;
-	this->WaitAudio = NAS_WaitAudio;
-	this->PlayAudio = NAS_PlayAudio;
-	this->GetAudioBuf = NAS_GetAudioBuf;
-	this->CloseAudio = NAS_CloseAudio;
-
-	this->free = Audio_DeleteDevice;
-
-	return this;
-}
-
-AudioBootStrap NAS_bootstrap = {
-	NAS_DRIVER_NAME, "Network Audio System",
-	Audio_Available, Audio_CreateDevice
-};
-
-/* This function waits until it is possible to write a full sound buffer */
-static void NAS_WaitAudio(_THIS)
-{
-	while ( this->hidden->buf_free < this->hidden->mixlen ) {
-		AuEvent ev;
-		NAS_AuNextEvent(this->hidden->aud, AuTrue, &ev);
-		NAS_AuDispatchEvent(this->hidden->aud, &ev);
-	}
-}
-
-static void NAS_PlayAudio(_THIS)
-{
-	while (this->hidden->mixlen > this->hidden->buf_free) { /* We think the buffer is full? Yikes! Ask the server for events,
-				    in the hope that some of them is LowWater events telling us more
-				    of the buffer is free now than what we think. */
-		AuEvent ev;
-		NAS_AuNextEvent(this->hidden->aud, AuTrue, &ev);
-		NAS_AuDispatchEvent(this->hidden->aud, &ev);
-	}
-	this->hidden->buf_free -= this->hidden->mixlen;
-
-	/* Write the audio data */
-	NAS_AuWriteElement(this->hidden->aud, this->hidden->flow, 0, this->hidden->mixlen, this->hidden->mixbuf, AuFalse, NULL);
-
-	this->hidden->written += this->hidden->mixlen;
-	
-#ifdef DEBUG_AUDIO
-	fprintf(stderr, "Wrote %d bytes of audio data\n", this->hidden->mixlen);
-#endif
-}
-
-static Uint8 *NAS_GetAudioBuf(_THIS)
-{
-	return(this->hidden->mixbuf);
-}
-
-static void NAS_CloseAudio(_THIS)
-{
-	if ( this->hidden->mixbuf != NULL ) {
-		SDL_FreeAudioMem(this->hidden->mixbuf);
-		this->hidden->mixbuf = NULL;
-	}
-	if ( this->hidden->aud ) {
-		NAS_AuCloseServer(this->hidden->aud);
-		this->hidden->aud = 0;
-	}
-}
-
-static unsigned char sdlformat_to_auformat(unsigned int fmt)
-{
-  switch (fmt)
-    {
-    case AUDIO_U8:
-      return AuFormatLinearUnsigned8;
-    case AUDIO_S8:
-      return AuFormatLinearSigned8;
-    case AUDIO_U16LSB:
-      return AuFormatLinearUnsigned16LSB;
-    case AUDIO_U16MSB:
-      return AuFormatLinearUnsigned16MSB;
-    case AUDIO_S16LSB:
-      return AuFormatLinearSigned16LSB;
-    case AUDIO_S16MSB:
-      return AuFormatLinearSigned16MSB;
-    }
-  return AuNone;
-}
-
-static AuBool
-event_handler(AuServer* aud, AuEvent* ev, AuEventHandlerRec* hnd)
-{
-	switch (ev->type) {
-	case AuEventTypeElementNotify: {
-		AuElementNotifyEvent* event = (AuElementNotifyEvent *)ev;
-
-		switch (event->kind) {
-		case AuElementNotifyKindLowWater:
-			if (this2->buf_free >= 0) {
-				this2->really += event->num_bytes;
-				gettimeofday(&this2->last_tv, 0);
-				this2->buf_free += event->num_bytes;
-			} else {
-				this2->buf_free = event->num_bytes;
-			}
-			break;
-		case AuElementNotifyKindState:
-			switch (event->cur_state) {
-			case AuStatePause:
-				if (event->reason != AuReasonUser) {
-					if (this2->buf_free >= 0) {
-						this2->really += event->num_bytes;
-						gettimeofday(&this2->last_tv, 0);
-						this2->buf_free += event->num_bytes;
-					} else {
-						this2->buf_free = event->num_bytes;
-					}
-				}
-				break;
-			}
-		}
-	}
-	}
-	return AuTrue;
-}
-
-static AuDeviceID
-find_device(_THIS, int nch)
-{
-    /* These "Au" things are all macros, not functions... */
-	int i;
-	for (i = 0; i < AuServerNumDevices(this->hidden->aud); i++) {
-		if ((AuDeviceKind(AuServerDevice(this->hidden->aud, i)) ==
-				AuComponentKindPhysicalOutput) &&
-			AuDeviceNumTracks(AuServerDevice(this->hidden->aud, i)) == nch) {
-			return AuDeviceIdentifier(AuServerDevice(this->hidden->aud, i));
-		}
-	}
-	return AuNone;
-}
-
-static int NAS_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	AuElement elms[3];
-	int buffer_size;
-	Uint16 test_format, format;
-
-	this->hidden->mixbuf = NULL;
-
-	/* Try for a closest match on audio format */
-	format = 0;
-	for ( test_format = SDL_FirstAudioFormat(spec->format);
-						! format && test_format; ) {
-		format = sdlformat_to_auformat(test_format);
-
-		if (format == AuNone) {
-			test_format = SDL_NextAudioFormat();
-		}
-	}
-	if ( format == 0 ) {
-		SDL_SetError("Couldn't find any hardware audio formats");
-		return(-1);
-	}
-	spec->format = test_format;
-
-	this->hidden->aud = NAS_AuOpenServer("", 0, NULL, 0, NULL, NULL);
-	if (this->hidden->aud == 0)
-	{
-		SDL_SetError("Couldn't open connection to NAS server");
-		return (-1);
-	}
-	
-	this->hidden->dev = find_device(this, spec->channels);
-	if ((this->hidden->dev == AuNone) || (!(this->hidden->flow = NAS_AuCreateFlow(this->hidden->aud, NULL)))) {
-		NAS_AuCloseServer(this->hidden->aud);
-		this->hidden->aud = 0;
-		SDL_SetError("Couldn't find a fitting playback device on NAS server");
-		return (-1);
-	}
-	
-	buffer_size = spec->freq;
-	if (buffer_size < 4096)
-		buffer_size = 4096; 
-
-	if (buffer_size > 32768)
-		buffer_size = 32768; /* So that the buffer won't get unmanageably big. */
-
-	/* Calculate the final parameters for this audio specification */
-	SDL_CalculateAudioSpec(spec);
-
-	this2 = this->hidden;
-
-    /* These "Au" things without a NAS_ prefix are macros, not functions... */
-	AuMakeElementImportClient(elms, spec->freq, format, spec->channels, AuTrue,
-				buffer_size, buffer_size / 4, 0, NULL);
-	AuMakeElementExportDevice(elms+1, 0, this->hidden->dev, spec->freq,
-				AuUnlimitedSamples, 0, NULL);
-	NAS_AuSetElements(this->hidden->aud, this->hidden->flow, AuTrue, 2, elms, NULL);
-	NAS_AuRegisterEventHandler(this->hidden->aud, AuEventHandlerIDMask, 0, this->hidden->flow,
-				event_handler, (AuPointer) NULL);
-
-	NAS_AuStartFlow(this->hidden->aud, this->hidden->flow, NULL);
-
-	/* Allocate mixing buffer */
-	this->hidden->mixlen = spec->size;
-	this->hidden->mixbuf = (Uint8 *)SDL_AllocAudioMem(this->hidden->mixlen);
-	if ( this->hidden->mixbuf == NULL ) {
-		return(-1);
-	}
-	SDL_memset(this->hidden->mixbuf, spec->silence, spec->size);
-
-	/* Get the parent process id (we're the parent of the audio thread) */
-	this->hidden->parent = getpid();
-
-	/* We're ready to rock and roll. :-) */
-	return(0);
-}
diff --git a/apps/plugins/sdl/src/audio/nas/SDL_nasaudio.h b/apps/plugins/sdl/src/audio/nas/SDL_nasaudio.h
deleted file mode 100644
index 1c09630880..0000000000
--- a/apps/plugins/sdl/src/audio/nas/SDL_nasaudio.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-
-    This driver was written by:
-    Erik Inge Bolsø
-    knan@mo.himolde.no
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_nasaudio_h
-#define _SDL_nasaudio_h
-
-#ifdef __sgi
-#include <nas/audiolib.h>
-#else
-#include <audio/audiolib.h>
-#endif
-#include <sys/time.h>
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData {
-	AuServer*  aud;
-	AuFlowID   flow;
-	AuDeviceID dev;
-	
-	/* The parent process id, to detect when application quits */
-	pid_t parent;
-
-	/* Raw mixing buffer */
-	Uint8 *mixbuf;
-	int    mixlen;
-
-	int written;
-	int really;
-	int bps;
-	struct timeval last_tv;
-	int buf_free;
-};
-#endif /* _SDL_nasaudio_h */
-
diff --git a/apps/plugins/sdl/src/audio/nds/SDL_ndsaudio.c b/apps/plugins/sdl/src/audio/nds/SDL_ndsaudio.c
deleted file mode 100644
index afe141a567..0000000000
--- a/apps/plugins/sdl/src/audio/nds/SDL_ndsaudio.c
+++ /dev/null
@@ -1,335 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/* Allow access to a raw mixing buffer */
-#include <nds.h>
-#include "SDL.h"
-#include "SDL_endian.h"
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "SDL_ndsaudio.h"
-#include "soundcommon.h"
-
-
-/* Audio driver functions */
-static int NDS_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void NDS_WaitAudio(_THIS);
-static void NDS_PlayAudio(_THIS);
-static Uint8 *NDS_GetAudioBuf(_THIS);
-static void NDS_CloseAudio(_THIS);
-
-/* Audio driver bootstrap functions */
-
-u32 framecounter = 0,soundoffset = 0;
-static SDL_AudioDevice *sdl_nds_audiodevice; 
-
-//void SoundMixCallback(void *stream,u32 size)
-//{
-//	//printf("SoundMixCallback\n");
-//	
-//	Uint8 *buffer;
-//	 
-// 	buffer = sdl_nds_audiodevice->hidden->mixbuf;
-//	memset(buffer, sdl_nds_audiodevice->spec.silence, size);
-//	
-//	if (!sdl_nds_audiodevice->paused){ 
-//		 
-//
-//	//if (sdl_nds_audiodevice->convert.needed) {
-//	//	int silence;
-//
-//	//	if (sdl_nds_audiodevice->convert.src_format == AUDIO_U8 ) { 
-//	//		silence = 0x80;
-//	//	} else {
-//	//		silence =  0; 
-//	//	}
-//	//	memset(sdl_nds_audiodevice->convert.buf, silence, sdl_nds_audiodevice->convert.len);
-//	//	sdl_nds_audiodevice->spec.callback(sdl_nds_audiodevice->spec.userdata,
-//	//		(Uint8 *)sdl_nds_audiodevice->convert.buf,sdl_nds_audiodevice->convert.len);
-//	//	SDL_ConvertAudio(&sdl_nds_audiodevice->convert);
-//	//	memcpy(buffer, sdl_nds_audiodevice->convert.buf, sdl_nds_audiodevice->convert.len_cvt);
-//	//} else 
-//	{
-//		sdl_nds_audiodevice->spec.callback(sdl_nds_audiodevice->spec.userdata, buffer, size);
-//		//memcpy((Sint16 *)stream,buffer, size);
-//	}
-//
-//	}
-//
-//	if(soundsystem->format == 8)
-//	{
-//		int i;
-//		s32 *buffer32 = (s32 *)buffer; 
-//		s32 *stream32 = (s32 *)stream;
-//		for(i=0;i<size/4;i++){ *stream32++ = buffer32[i] ^ 0x80808080;}
-//		//for(i = 0; i < size; i++)
-//		//	((s8*)stream)[i]=(buffer[i]^0x80);
-//	}
-//	else
-//	{
-//		int i;
-//		for(i = 0; i < size; i++){
-//			//((short*)stream)[i] =(short)buffer[i] << 8;				// sound 8bit ---> buffer 16bit
-//			//if (buffer[i] &0x80)
-//				//((Sint16*)stream)[i] = 0xff00 | buffer[i];
-//			((Sint16*)stream)[i] = (buffer[i] - 128) << 8;
-//
-//			//else
-//			//	((Sint16*)stream)[i] = buffer[i];
-//		}
-//		//register signed char *pSrc =buffer;
-//		//register short *pDest =stream;
-//		//int x;
-//		//			for (x=size; x>0; x--)
-//		//			{
-//		//				register short temp = (((short)*pSrc)-128)<<8;
-//		//				pSrc++;
-//		//				*pDest++ = temp;
-//		//			}
-//
-//		//memcpy((Sint16 *)stream,buffer, size);
-//	}
-//}
-
-void SoundMixCallback(void *stream,u32 len)
-{
-	SDL_AudioDevice *audio = (SDL_AudioDevice *)sdl_nds_audiodevice;
-
-	/* Silence the buffer, since it's ours */
-	SDL_memset(stream, audio->spec.silence, len);
-
-	/* Only do soemthing if audio is enabled */
-	if ( ! audio->enabled )
-		return;
-
-	if ( ! audio->paused ) {
-		if ( audio->convert.needed ) {
-			//fprintf(stderr,"converting audio\n");
-			SDL_mutexP(audio->mixer_lock);
-			(*audio->spec.callback)(audio->spec.userdata,
-				(Uint8 *)audio->convert.buf,audio->convert.len);
-			SDL_mutexV(audio->mixer_lock);
-			SDL_ConvertAudio(&audio->convert);
-			SDL_memcpy(stream,audio->convert.buf,audio->convert.len_cvt);
-		} else {
-			SDL_mutexP(audio->mixer_lock);
-			(*audio->spec.callback)(audio->spec.userdata,
-						(Uint8 *)stream, len);
-			SDL_mutexV(audio->mixer_lock);
-		}
-	}
-	return;
-}
-void MixSound(void)
-{
-	int remain;
-
-	if(soundsystem->format == 8)
-	{
-		if((soundsystem->soundcursor + soundsystem->numsamples) > soundsystem->buffersize)
-		{
-			SoundMixCallback(&soundsystem->mixbuffer[soundsystem->soundcursor],soundsystem->buffersize - soundsystem->soundcursor);
-			remain = soundsystem->numsamples - (soundsystem->buffersize - soundsystem->soundcursor);
-			SoundMixCallback(soundsystem->mixbuffer,remain);
-		}
-		else
-		{
-			SoundMixCallback(&soundsystem->mixbuffer[soundsystem->soundcursor],soundsystem->numsamples);
-		}
-	}
-	else
-	{
-		if((soundsystem->soundcursor + soundsystem->numsamples) > (soundsystem->buffersize >> 1))
-		{
-			SoundMixCallback(&soundsystem->mixbuffer[soundsystem->soundcursor << 1],(soundsystem->buffersize >> 1) - soundsystem->soundcursor);
-			remain = soundsystem->numsamples - ((soundsystem->buffersize >> 1) - soundsystem->soundcursor);
-			SoundMixCallback(soundsystem->mixbuffer,remain);
-		}
-		else
-		{
-			SoundMixCallback(&soundsystem->mixbuffer[soundsystem->soundcursor << 1],soundsystem->numsamples);
-		}
-	}
-}
-
-void InterruptHandler(void)
-{
-	framecounter++;
-}
-void FiFoHandler(void)
-{
-	u32 command;
-	while ( !(REG_IPC_FIFO_CR & (IPC_FIFO_RECV_EMPTY)) ) 
-	{
-		command = REG_IPC_FIFO_RX;
-
-		switch(command)
-		{
-		case FIFO_NONE:
-			break;
-		case UPDATEON_ARM9:
-			REG_IME = 0;
-			MixSound();
-			REG_IME = 1;
-			SendCommandToArm7(MIXCOMPLETE_ONARM9);
-			break;
-		}
-	}
-}
-
-
-
-
-
-static int Audio_Available(void)
-{
-	return(1);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice));
-	if ( this ) {
-		SDL_memset(this, 0, (sizeof *this));
-		this->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *this->hidden));
-	}
-	if ( (this == NULL) || (this->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( this ) {
-			SDL_free(this);
-		}
-		return(0);
-	}
-	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-	/* Set the function pointers */
-	this->OpenAudio = NDS_OpenAudio;
-	this->WaitAudio = NDS_WaitAudio;
-	this->PlayAudio = NDS_PlayAudio;
-	this->GetAudioBuf = NDS_GetAudioBuf;
-	this->CloseAudio = NDS_CloseAudio;
-
-	this->free = Audio_DeleteDevice;
-//fprintf(stderr,"Audio_CreateDevice\n");
-	return this;
-}
-
-AudioBootStrap NDSAUD_bootstrap = {
-	"nds", "NDS audio",
-	Audio_Available, Audio_CreateDevice
-};
-
-
-void static NDS_WaitAudio(_THIS)
-{
-	//printf("NDS_WaitAudio\n");
-}
-
-static void NDS_PlayAudio(_THIS)
-{
-	//printf("playing audio\n");
-	if (this->paused)
-		return;
-	
-}
-
-static Uint8 *NDS_GetAudioBuf(_THIS)
-{
-	return NULL;//(this->hidden->mixbuf); 
-}
-
-static void NDS_CloseAudio(_THIS)
-{
-/*	if ( this->hidden->mixbuf != NULL ) {
-		SDL_FreeAudioMem(this->hidden->mixbuf);
-		this->hidden->mixbuf = NULL;
-	}*/ 
-}
-
-static int NDS_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	//printf("NDS_OpenAudio\n");
-	int format = 0;
-	//switch(spec->format&0xff) {
-	//case  8: spec->format = AUDIO_S8;format=8; break;
-	//case 16: spec->format = AUDIO_S16LSB;format=16; break;
-	//default:
-	//	SDL_SetError("Unsupported audio format");
-	//	return(-1);
-	//}
-	switch (spec->format&~0x1000) {
-		case AUDIO_S8:
-			/* Signed 8-bit audio supported */
-			format=8;
-			break;
-		case AUDIO_U8:
-			spec->format ^= 0x80;format=8;
-			break;
-		case AUDIO_U16:
-			/* Unsigned 16-bit audio unsupported, convert to S16 */
-			spec->format ^=0x8000;format=16;
-		case AUDIO_S16:
-			/* Signed 16-bit audio supported */
-			format=16;
-			break;
-	}
-	/* Update the fragment size as size in bytes */
-	SDL_CalculateAudioSpec(spec);
-
-	/* Allocate mixing buffer */
-	//this->hidden->mixlen = spec->size;
-	//this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
-	//if ( this->hidden->mixbuf == NULL ) {
-	//	SDL_SetError("Out of Memory");
-	//	return(-1);
-	//} 
-
-	SDL_NDSAudio_mutex = 0; 
-	sdl_nds_audiodevice=this;
-	
-	irqInit();
-	irqSet(IRQ_VBLANK,&InterruptHandler);
-	irqSet(IRQ_FIFO_NOT_EMPTY,&FiFoHandler);
-	irqEnable(IRQ_FIFO_NOT_EMPTY);
-	
-	REG_IPC_FIFO_CR = IPC_FIFO_ENABLE | IPC_FIFO_SEND_CLEAR | IPC_FIFO_RECV_IRQ;
-
-
-
-	SoundSystemInit(spec->freq,spec->size,0,format);
-	SoundStartMixer();
-
-	
-	return(1);
-}
diff --git a/apps/plugins/sdl/src/audio/nds/SDL_ndsaudio.h b/apps/plugins/sdl/src/audio/nds/SDL_ndsaudio.h
deleted file mode 100644
index cb6d1ea858..0000000000
--- a/apps/plugins/sdl/src/audio/nds/SDL_ndsaudio.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_lowaudio_h
-#define _SDL_lowaudio_h
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the audio functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData {
-	/* The file descriptor for the audio device */
-	//Uint8 *mixbuf;
-	//Uint32 mixlen;
-}; 
-unsigned short SDL_NDSAudio_mutex=0; 
- 
-
-#endif /* _SDL_lowaudio_h */
diff --git a/apps/plugins/sdl/src/audio/nds/sound9.c b/apps/plugins/sdl/src/audio/nds/sound9.c
deleted file mode 100644
index 59c1c219ae..0000000000
--- a/apps/plugins/sdl/src/audio/nds/sound9.c
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-#include "SDL_stdinc.h"
-
-#include "soundcommon.h"
-
-void SoundSystemInit(u32 rate,u32 buffersize,u8 channel,u8 format)
-{
-	soundsystem->rate = rate;
-	
-	if(format == 8) 
-		soundsystem->buffersize = buffersize;
-	else if(format == 16)
-		soundsystem->buffersize = buffersize * sizeof(short);
-
-	soundsystem->mixbuffer = (s8*)SDL_malloc(soundsystem->buffersize);
-	//soundsystem->soundbuffer = soundsystem->mixbuffer;
-	soundsystem->format = format;
-	soundsystem->channel = channel;
-	soundsystem->prevtimer = 0;
-	soundsystem->soundcursor = 0;
-	soundsystem->numsamples = 0;
-	soundsystem->period = 0x1000000 / rate;
-	soundsystem->cmd = INIT;
-}
-
-void SoundStartMixer(void)
-{
-	soundsystem->cmd |= MIX;
-}
-
-void SendCommandToArm7(u32 command)
-{
-    while (REG_IPC_FIFO_CR & IPC_FIFO_SEND_FULL);
-    if (REG_IPC_FIFO_CR & IPC_FIFO_ERROR)
-    {
-        REG_IPC_FIFO_CR |= IPC_FIFO_SEND_CLEAR;
-    } 
-    
-    REG_IPC_FIFO_TX = command;
-}
diff --git a/apps/plugins/sdl/src/audio/nds/soundcommon.h b/apps/plugins/sdl/src/audio/nds/soundcommon.h
deleted file mode 100644
index d38e37cf68..0000000000
--- a/apps/plugins/sdl/src/audio/nds/soundcommon.h
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef __SOUNDCOMMON_H
-#define __SOUNDCOMMON_H
-
-#include <nds.h>
-
-#define CLOCK (1 << 25)
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-typedef enum
-{
-	NONE = 0,
-	INIT = 1,
-	MIX = 2,
-	MIXING = 4,
-	STOP = 8
-}CommandType;
-
-typedef enum
-{
-	FIFO_NONE = 0,
-	UPDATEON_ARM9 = 1,
-	MIXCOMPLETE_ONARM9 = 2,
-}FifoType; 
-
-typedef struct
-{
-	s8 *mixbuffer;//,*soundbuffer;
-	u32 rate;
-	u32 buffersize;
-	u32 cmd;
-	u8 channel,format;
-	u32 soundcursor,numsamples;
-	s32 prevtimer;
-	s16 period;
-}S_SoundSystem;
-
-#define soundsystem ((S_SoundSystem*)((u32)(IPC)+sizeof(TransferRegion)))
-
-#ifdef ARM9
-extern void SoundSystemInit(u32 rate,u32 buffersize,u8 channel,u8 format);
-extern void SoundStartMixer(void);
-extern void SendCommandToArm7(u32 command);
-#else
-extern void SoundVBlankIrq(void);
-extern void SoundSwapAndMix(void);
-extern void SoundSetTimer(int period);
-extern void SoundFifoHandler(void);
-extern void SendCommandToArm9(u32 command);
-#endif
-
-#ifdef __cplusplus
-}
-#endif
-#endif
diff --git a/apps/plugins/sdl/src/audio/nto/SDL_nto_audio.c b/apps/plugins/sdl/src/audio/nto/SDL_nto_audio.c
deleted file mode 100644
index f951825460..0000000000
--- a/apps/plugins/sdl/src/audio/nto/SDL_nto_audio.c
+++ /dev/null
@@ -1,507 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#include <errno.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include <signal.h>
-#include <sys/types.h>
-#include <sys/time.h>
-#include <sched.h>
-#include <sys/select.h>
-#include <sys/neutrino.h>
-#include <sys/asoundlib.h>
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "SDL_nto_audio.h"
-
-/* The tag name used by NTO audio */
-#define DRIVER_NAME "qsa-nto"
-
-/* default channel communication parameters */
-#define DEFAULT_CPARAMS_RATE 22050
-#define DEFAULT_CPARAMS_VOICES 1
-/* FIXME: need to add in the near future flexible logic with frag_size and frags count */
-#define DEFAULT_CPARAMS_FRAG_SIZE 4096
-#define DEFAULT_CPARAMS_FRAGS_MIN 1
-#define DEFAULT_CPARAMS_FRAGS_MAX 1
-
-/* Open the audio device for playback, and don't block if busy */
-#define OPEN_FLAGS SND_PCM_OPEN_PLAYBACK
-
-#define QSA_NO_WORKAROUNDS  0x00000000
-#define QSA_MMAP_WORKAROUND 0x00000001
-
-struct BuggyCards
-{
-   char* cardname;
-   unsigned long bugtype;
-};
-
-#define QSA_WA_CARDS 3
-
-struct BuggyCards buggycards[QSA_WA_CARDS]=
-{
-   {"Sound Blaster Live!", QSA_MMAP_WORKAROUND},
-   {"Vortex 8820",         QSA_MMAP_WORKAROUND},
-   {"Vortex 8830",         QSA_MMAP_WORKAROUND},
-};
-
-/* Audio driver functions */
-static void NTO_ThreadInit(_THIS);
-static int NTO_OpenAudio(_THIS, SDL_AudioSpec* spec);
-static void NTO_WaitAudio(_THIS);
-static void NTO_PlayAudio(_THIS);
-static Uint8* NTO_GetAudioBuf(_THIS);
-static void NTO_CloseAudio(_THIS);
-
-/* card names check to apply the workarounds */
-static int NTO_CheckBuggyCards(_THIS, unsigned long checkfor)
-{
-    char scardname[33];
-    int it;
-    
-    if (snd_card_get_name(cardno, scardname, 32)<0)
-    {
-        return 0;
-    }
-
-    for (it=0; it<QSA_WA_CARDS; it++)
-    {
-       if (SDL_strcmp(buggycards[it].cardname, scardname)==0)
-       {
-          if (buggycards[it].bugtype==checkfor)
-          {
-              return 1;
-          }
-       }
-    }
-
-    return 0;
-}
-
-static void NTO_ThreadInit(_THIS)
-{
-   int status;
-   struct sched_param param;
-
-   /* increasing default 10 priority to 25 to avoid jerky sound */
-   status=SchedGet(0, 0, &param);
-   param.sched_priority=param.sched_curpriority+15;
-   status=SchedSet(0, 0, SCHED_NOCHANGE, &param);
-}
-
-/* PCM transfer channel parameters initialize function */
-static void NTO_InitAudioParams(snd_pcm_channel_params_t* cpars)
-{
-    SDL_memset(cpars, 0, sizeof(snd_pcm_channel_params_t));
-
-    cpars->channel = SND_PCM_CHANNEL_PLAYBACK;
-    cpars->mode = SND_PCM_MODE_BLOCK;
-    cpars->start_mode = SND_PCM_START_DATA;
-    cpars->stop_mode  = SND_PCM_STOP_STOP;
-    cpars->format.format = SND_PCM_SFMT_S16_LE;
-    cpars->format.interleave = 1;
-    cpars->format.rate = DEFAULT_CPARAMS_RATE;
-    cpars->format.voices = DEFAULT_CPARAMS_VOICES;
-    cpars->buf.block.frag_size = DEFAULT_CPARAMS_FRAG_SIZE;
-    cpars->buf.block.frags_min = DEFAULT_CPARAMS_FRAGS_MIN;
-    cpars->buf.block.frags_max = DEFAULT_CPARAMS_FRAGS_MAX;
-}
-
-static int NTO_AudioAvailable(void)
-{
-    /*  See if we can open a nonblocking channel.
-        Return value '1' means we can.
-        Return value '0' means we cannot. */
-
-    int available;
-    int rval;
-    snd_pcm_t* handle;
-
-    available = 0;
-    handle = NULL;
-
-    rval = snd_pcm_open_preferred(&handle, NULL, NULL, OPEN_FLAGS);
-
-    if (rval >= 0)
-    {
-        available = 1;
-
-        if ((rval = snd_pcm_close(handle)) < 0)
-        {
-            SDL_SetError("NTO_AudioAvailable(): snd_pcm_close failed: %s\n", snd_strerror(rval));
-            available = 0;
-        }
-    }
-    else
-    {
-        SDL_SetError("NTO_AudioAvailable(): there are no available audio devices.\n");
-    }
-
-    return (available);
-}
-
-static void NTO_DeleteAudioDevice(SDL_AudioDevice *device)
-{
-    if ((device)&&(device->hidden))
-    {
-        SDL_free(device->hidden);
-    }
-    if (device)
-    {
-        SDL_free(device);
-    }
-}
-
-static SDL_AudioDevice* NTO_CreateAudioDevice(int devindex)
-{
-    SDL_AudioDevice *this;
-
-    /* Initialize all variables that we clean on shutdown */
-    this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-    if (this)
-    {
-        SDL_memset(this, 0, sizeof(SDL_AudioDevice));
-        this->hidden = (struct SDL_PrivateAudioData *)SDL_malloc(sizeof(struct SDL_PrivateAudioData));
-    }
-    if ((this == NULL) || (this->hidden == NULL))
-    {
-        SDL_OutOfMemory();
-        if (this)
-        {
-            SDL_free(this);
-	}
-        return (0);
-    }
-    SDL_memset(this->hidden, 0, sizeof(struct SDL_PrivateAudioData));
-    audio_handle = NULL;
-
-    /* Set the function pointers */
-    this->ThreadInit = NTO_ThreadInit;
-    this->OpenAudio = NTO_OpenAudio;
-    this->WaitAudio = NTO_WaitAudio;
-    this->PlayAudio = NTO_PlayAudio;
-    this->GetAudioBuf = NTO_GetAudioBuf;
-    this->CloseAudio = NTO_CloseAudio;
-
-    this->free = NTO_DeleteAudioDevice;
-
-    return this;
-}
-
-AudioBootStrap QNXNTOAUDIO_bootstrap =
-{
-    DRIVER_NAME, "QNX6 QSA-NTO Audio",
-    NTO_AudioAvailable,
-    NTO_CreateAudioDevice
-};
-
-/* This function waits until it is possible to write a full sound buffer */
-static void NTO_WaitAudio(_THIS)
-{
-    fd_set wfds;
-    int selectret;
-
-    FD_ZERO(&wfds);
-    FD_SET(audio_fd, &wfds);
-
-    do {
-        selectret=select(audio_fd + 1, NULL, &wfds, NULL, NULL);
-        switch (selectret)
-        {
-            case -1:
-            case  0: SDL_SetError("NTO_WaitAudio(): select() failed: %s\n", strerror(errno));
-                     return;
-            default: if (FD_ISSET(audio_fd, &wfds))
-                     {
-                         return;
-                     }
-                     break;
-        }
-    } while(1);
-}
-
-static void NTO_PlayAudio(_THIS)
-{
-    int written, rval;
-    int towrite;
-    void* pcmbuffer;
-
-    if (!this->enabled)
-    {
-        return;
-    }
-    
-    towrite = this->spec.size;
-    pcmbuffer = pcm_buf;
-
-    /* Write the audio data, checking for EAGAIN (buffer full) and underrun */
-    do {
-        written = snd_pcm_plugin_write(audio_handle, pcm_buf, towrite);
-        if (written != towrite)
-        {
-            if ((errno == EAGAIN) || (errno == EWOULDBLOCK))
-            {
-                /* Let a little CPU time go by and try to write again */
-                SDL_Delay(1);
-                /* if we wrote some data */
-                towrite -= written;
-                pcmbuffer += written * this->spec.channels;
-                continue;
-            }		
-            else
-            {
-                if ((errno == EINVAL) || (errno == EIO))
-                {
-                    SDL_memset(&cstatus, 0, sizeof(cstatus));
-                    cstatus.channel = SND_PCM_CHANNEL_PLAYBACK;
-                    if ((rval = snd_pcm_plugin_status(audio_handle, &cstatus)) < 0)
-                    {
-                        SDL_SetError("NTO_PlayAudio(): snd_pcm_plugin_status failed: %s\n", snd_strerror(rval));
-                        return;
-                    }	
-                    if ((cstatus.status == SND_PCM_STATUS_UNDERRUN) || (cstatus.status == SND_PCM_STATUS_READY))
-                    {
-                        if ((rval = snd_pcm_plugin_prepare(audio_handle, SND_PCM_CHANNEL_PLAYBACK)) < 0)
-                        {
-                            SDL_SetError("NTO_PlayAudio(): snd_pcm_plugin_prepare failed: %s\n", snd_strerror(rval));
-                            return;
-                        }
-                    }		        		
-                    continue;
-                }
-                else
-                {
-                    return;
-                }
-            }
-        }
-        else
-        {
-            /* we wrote all remaining data */
-            towrite -= written;
-            pcmbuffer += written * this->spec.channels;
-        }
-    } while ((towrite > 0)  && (this->enabled));
-
-    /* If we couldn't write, assume fatal error for now */
-    if (towrite != 0)
-    {
-        this->enabled = 0;
-    }
-
-    return;
-}
-
-static Uint8* NTO_GetAudioBuf(_THIS)
-{
-    return pcm_buf;
-}
-
-static void NTO_CloseAudio(_THIS)
-{
-    int rval;
-
-    this->enabled = 0;
-
-    if (audio_handle != NULL)
-    {
-        if ((rval = snd_pcm_plugin_flush(audio_handle, SND_PCM_CHANNEL_PLAYBACK)) < 0)
-        {
-            SDL_SetError("NTO_CloseAudio(): snd_pcm_plugin_flush failed: %s\n", snd_strerror(rval));
-            return;
-        }
-        if ((rval = snd_pcm_close(audio_handle)) < 0)
-        {
-            SDL_SetError("NTO_CloseAudio(): snd_pcm_close failed: %s\n",snd_strerror(rval));
-            return;
-        }
-        audio_handle = NULL;
-    }
-}
-
-static int NTO_OpenAudio(_THIS, SDL_AudioSpec* spec)
-{
-    int rval;
-    int format;
-    Uint16 test_format;
-    int found;
-
-    audio_handle = NULL;
-    this->enabled = 0;
-
-    if (pcm_buf != NULL)
-    {
-        SDL_FreeAudioMem(pcm_buf); 
-        pcm_buf = NULL;
-    }
-
-    /* initialize channel transfer parameters to default */
-    NTO_InitAudioParams(&cparams);
-
-    /* Open the audio device */
-    rval = snd_pcm_open_preferred(&audio_handle, &cardno, &deviceno, OPEN_FLAGS);
-    if (rval < 0)
-    {
-        SDL_SetError("NTO_OpenAudio(): snd_pcm_open failed: %s\n", snd_strerror(rval));
-        return (-1);
-    }
-
-    if (!NTO_CheckBuggyCards(this, QSA_MMAP_WORKAROUND))
-    {
-        /* enable count status parameter */
-        if ((rval = snd_pcm_plugin_set_disable(audio_handle, PLUGIN_DISABLE_MMAP)) < 0)
-        {
-            SDL_SetError("snd_pcm_plugin_set_disable failed: %s\n", snd_strerror(rval));
-            return (-1);
-        }
-    }
-
-    /* Try for a closest match on audio format */
-    format = 0;
-    /* can't use format as SND_PCM_SFMT_U8 = 0 in nto */
-    found = 0;
-
-    for (test_format=SDL_FirstAudioFormat(spec->format); !found ;)
-    {
-        /* if match found set format to equivalent ALSA format */
-        switch (test_format)
-        {
-            case AUDIO_U8:
-                           format = SND_PCM_SFMT_U8;
-                           found = 1;
-                           break;
-            case AUDIO_S8:
-                           format = SND_PCM_SFMT_S8;
-                           found = 1;
-                           break;
-            case AUDIO_S16LSB:
-                           format = SND_PCM_SFMT_S16_LE;
-                           found = 1;
-                           break;
-            case AUDIO_S16MSB:
-                           format = SND_PCM_SFMT_S16_BE;
-                           found = 1;
-                           break;
-            case AUDIO_U16LSB:
-                           format = SND_PCM_SFMT_U16_LE;
-                           found = 1;
-                           break;
-            case AUDIO_U16MSB:
-                           format = SND_PCM_SFMT_U16_BE;
-                           found = 1;
-                           break;
-            default:
-                           break;
-        }
-
-        if (!found)
-        {
-            test_format = SDL_NextAudioFormat();
-        }
-    }
-
-    /* assumes test_format not 0 on success */
-    if (test_format == 0)
-    {
-        SDL_SetError("NTO_OpenAudio(): Couldn't find any hardware audio formats");
-        return (-1);
-    }
-
-    spec->format = test_format;
-
-    /* Set the audio format */
-    cparams.format.format = format;
-
-    /* Set mono or stereo audio (currently only two channels supported) */
-    cparams.format.voices = spec->channels;
-	
-    /* Set rate */
-    cparams.format.rate = spec->freq;
-
-    /* Setup the transfer parameters according to cparams */
-    rval = snd_pcm_plugin_params(audio_handle, &cparams);
-    if (rval < 0)
-    {
-        SDL_SetError("NTO_OpenAudio(): snd_pcm_channel_params failed: %s\n", snd_strerror(rval));
-        return (-1);
-    }
-
-    /* Make sure channel is setup right one last time */
-    SDL_memset(&csetup, 0x00, sizeof(csetup));
-    csetup.channel = SND_PCM_CHANNEL_PLAYBACK;
-    if (snd_pcm_plugin_setup(audio_handle, &csetup) < 0)
-    {
-        SDL_SetError("NTO_OpenAudio(): Unable to setup playback channel\n");
-        return -1;
-    }
-
-
-    /* Calculate the final parameters for this audio specification */
-    SDL_CalculateAudioSpec(spec);
-
-    pcm_len = spec->size;
-
-    if (pcm_len==0)
-    {
-        pcm_len = csetup.buf.block.frag_size * spec->channels * (snd_pcm_format_width(format)/8);
-    }
-
-    /* Allocate memory to the audio buffer and initialize with silence (Note that
-       buffer size must be a multiple of fragment size, so find closest multiple)
-    */
-    pcm_buf = (Uint8*)SDL_AllocAudioMem(pcm_len);
-    if (pcm_buf == NULL)
-    {
-        SDL_SetError("NTO_OpenAudio(): pcm buffer allocation failed\n");
-        return (-1);
-    }
-    SDL_memset(pcm_buf, spec->silence, pcm_len);
-
-    /* get the file descriptor */
-    if ((audio_fd = snd_pcm_file_descriptor(audio_handle, SND_PCM_CHANNEL_PLAYBACK)) < 0)
-    {
-        SDL_SetError("NTO_OpenAudio(): snd_pcm_file_descriptor failed with error code: %s\n", snd_strerror(rval));
-        return (-1);
-    }
-
-    /* Trigger audio playback */
-    rval = snd_pcm_plugin_prepare(audio_handle, SND_PCM_CHANNEL_PLAYBACK);
-    if (rval < 0)
-    {
-        SDL_SetError("snd_pcm_plugin_prepare failed: %s\n", snd_strerror(rval));
-        return (-1);
-    }
-
-    this->enabled = 1;
-
-    /* Get the parent process id (we're the parent of the audio thread) */
-    parent = getpid();
-
-    /* We're really ready to rock and roll. :-) */
-    return (0);
-}
diff --git a/apps/plugins/sdl/src/audio/nto/SDL_nto_audio.h b/apps/plugins/sdl/src/audio/nto/SDL_nto_audio.h
deleted file mode 100644
index cfd592c96b..0000000000
--- a/apps/plugins/sdl/src/audio/nto/SDL_nto_audio.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef __SDL_NTO_AUDIO_H__
-#define __SDL_NTO_AUDIO_H__
-
-#include <sys/asoundlib.h>
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the audio functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData
-{
-    /* The audio device handle */
-    int cardno;
-    int deviceno;
-    snd_pcm_t* audio_handle;
-
-    /* The audio file descriptor */
-    int audio_fd;
-
-    /* The parent process id, to detect when application quits */
-    pid_t parent;
-
-    /* Raw mixing buffer */
-    Uint8* pcm_buf;
-    Uint32 pcm_len;
-
-    /* QSA parameters */
-    snd_pcm_channel_status_t cstatus;
-    snd_pcm_channel_params_t cparams;
-    snd_pcm_channel_setup_t  csetup;
-};
-
-#define cardno          (this->hidden->cardno)
-#define deviceno        (this->hidden->deviceno)
-#define audio_handle    (this->hidden->audio_handle)
-#define audio_fd        (this->hidden->audio_fd)
-#define parent          (this->hidden->parent)
-#define pcm_buf         (this->hidden->pcm_buf)
-#define pcm_len         (this->hidden->pcm_len)
-#define cstatus         (this->hidden->cstatus)
-#define cparams         (this->hidden->cparams)
-#define csetup          (this->hidden->csetup)
-
-#endif /* __SDL_NTO_AUDIO_H__ */
diff --git a/apps/plugins/sdl/src/audio/paudio/SDL_paudio.c b/apps/plugins/sdl/src/audio/paudio/SDL_paudio.c
deleted file mode 100644
index 6270d8c0a8..0000000000
--- a/apps/plugins/sdl/src/audio/paudio/SDL_paudio.c
+++ /dev/null
@@ -1,511 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Carsten Griwodz
-    griff@kom.tu-darmstadt.de
-
-    based on linux/SDL_dspaudio.c by Sam Lantinga
-*/
-#include "SDL_config.h"
-
-/* Allow access to a raw mixing buffer */
-
-#include <errno.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include <sys/time.h>
-#include <sys/ioctl.h>
-#include <sys/stat.h>
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_audiodev_c.h"
-#include "SDL_paudio.h"
-
-#define DEBUG_AUDIO 1
-
-/* A conflict within AIX 4.3.3 <sys/> headers and probably others as well.
- * I guess nobody ever uses audio... Shame over AIX header files.  */
-#include <sys/machine.h>
-#undef BIG_ENDIAN
-#include <sys/audio.h>
-
-/* The tag name used by paud audio */
-#define Paud_DRIVER_NAME         "paud"
-
-/* Open the audio device for playback, and don't block if busy */
-/* #define OPEN_FLAGS	(O_WRONLY|O_NONBLOCK) */
-#define OPEN_FLAGS	O_WRONLY
-
-/* Audio driver functions */
-static int Paud_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void Paud_WaitAudio(_THIS);
-static void Paud_PlayAudio(_THIS);
-static Uint8 *Paud_GetAudioBuf(_THIS);
-static void Paud_CloseAudio(_THIS);
-
-/* Audio driver bootstrap functions */
-
-static int Audio_Available(void)
-{
-	int fd;
-	int available;
-
-	available = 0;
-	fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0);
-	if ( fd >= 0 ) {
-		available = 1;
-		close(fd);
-	}
-	return(available);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-	SDL_free(device->hidden);
-	SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-	if ( this ) {
-		SDL_memset(this, 0, (sizeof *this));
-		this->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *this->hidden));
-	}
-	if ( (this == NULL) || (this->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( this ) {
-			SDL_free(this);
-		}
-		return(0);
-	}
-	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-	audio_fd = -1;
-
-	/* Set the function pointers */
-	this->OpenAudio = Paud_OpenAudio;
-	this->WaitAudio = Paud_WaitAudio;
-	this->PlayAudio = Paud_PlayAudio;
-	this->GetAudioBuf = Paud_GetAudioBuf;
-	this->CloseAudio = Paud_CloseAudio;
-
-	this->free = Audio_DeleteDevice;
-
-	return this;
-}
-
-AudioBootStrap Paud_bootstrap = {
-	Paud_DRIVER_NAME, "AIX Paudio",
-	Audio_Available, Audio_CreateDevice
-};
-
-/* This function waits until it is possible to write a full sound buffer */
-static void Paud_WaitAudio(_THIS)
-{
-    fd_set fdset;
-
-    /* See if we need to use timed audio synchronization */
-    if ( frame_ticks ) {
-        /* Use timer for general audio synchronization */
-        Sint32 ticks;
-
-        ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS;
-        if ( ticks > 0 ) {
-	    SDL_Delay(ticks);
-        }
-    } else {
-        audio_buffer  paud_bufinfo;
-
-        /* Use select() for audio synchronization */
-        struct timeval timeout;
-        FD_ZERO(&fdset);
-        FD_SET(audio_fd, &fdset);
-
-        if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 ) {
-#ifdef DEBUG_AUDIO
-            fprintf(stderr, "Couldn't get audio buffer information\n");
-#endif
-            timeout.tv_sec  = 10;
-            timeout.tv_usec = 0;
-        } else {
-	    long ms_in_buf = paud_bufinfo.write_buf_time;
-            timeout.tv_sec  = ms_in_buf/1000;
-	    ms_in_buf       = ms_in_buf - timeout.tv_sec*1000;
-            timeout.tv_usec = ms_in_buf*1000;
-#ifdef DEBUG_AUDIO
-            fprintf( stderr,
-		     "Waiting for write_buf_time=%ld,%ld\n",
-		     timeout.tv_sec,
-		     timeout.tv_usec );
-#endif
-	}
-
-#ifdef DEBUG_AUDIO
-        fprintf(stderr, "Waiting for audio to get ready\n");
-#endif
-        if ( select(audio_fd+1, NULL, &fdset, NULL, &timeout) <= 0 ) {
-            const char *message = "Audio timeout - buggy audio driver? (disabled)";
-            /*
-	     * In general we should never print to the screen,
-             * but in this case we have no other way of letting
-             * the user know what happened.
-             */
-            fprintf(stderr, "SDL: %s - %s\n", strerror(errno), message);
-            this->enabled = 0;
-            /* Don't try to close - may hang */
-            audio_fd = -1;
-#ifdef DEBUG_AUDIO
-            fprintf(stderr, "Done disabling audio\n");
-#endif
-        }
-#ifdef DEBUG_AUDIO
-        fprintf(stderr, "Ready!\n");
-#endif
-    }
-}
-
-static void Paud_PlayAudio(_THIS)
-{
-	int written;
-
-	/* Write the audio data, checking for EAGAIN on broken audio drivers */
-	do {
-		written = write(audio_fd, mixbuf, mixlen);
-		if ( (written < 0) && ((errno == 0) || (errno == EAGAIN)) ) {
-			SDL_Delay(1);	/* Let a little CPU time go by */
-		}
-	} while ( (written < 0) && 
-	          ((errno == 0) || (errno == EAGAIN) || (errno == EINTR)) );
-
-	/* If timer synchronization is enabled, set the next write frame */
-	if ( frame_ticks ) {
-		next_frame += frame_ticks;
-	}
-
-	/* If we couldn't write, assume fatal error for now */
-	if ( written < 0 ) {
-		this->enabled = 0;
-	}
-#ifdef DEBUG_AUDIO
-	fprintf(stderr, "Wrote %d bytes of audio data\n", written);
-#endif
-}
-
-static Uint8 *Paud_GetAudioBuf(_THIS)
-{
-	return mixbuf;
-}
-
-static void Paud_CloseAudio(_THIS)
-{
-	if ( mixbuf != NULL ) {
-		SDL_FreeAudioMem(mixbuf);
-		mixbuf = NULL;
-	}
-	if ( audio_fd >= 0 ) {
-		close(audio_fd);
-		audio_fd = -1;
-	}
-}
-
-static int Paud_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	char          audiodev[1024];
-	int           format;
-	int           bytes_per_sample;
-	Uint16        test_format;
-	audio_init    paud_init;
-	audio_buffer  paud_bufinfo;
-	audio_status  paud_status;
-	audio_control paud_control;
-	audio_change  paud_change;
-
-	/* Reset the timer synchronization flag */
-	frame_ticks = 0.0;
-
-	/* Open the audio device */
-	audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);
-	if ( audio_fd < 0 ) {
-		SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
-		return -1;
-	}
-
-	/*
-	 * We can't set the buffer size - just ask the device for the maximum
-	 * that we can have.
-	 */
-	if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 ) {
-		SDL_SetError("Couldn't get audio buffer information");
-		return -1;
-	}
-
-	mixbuf = NULL;
-
-	if ( spec->channels > 1 )
-	    spec->channels = 2;
-	else
-	    spec->channels = 1;
-
-	/*
-	 * Fields in the audio_init structure:
-	 *
-	 * Ignored by us:
-	 *
-	 * paud.loadpath[LOAD_PATH]; * DSP code to load, MWave chip only?
-	 * paud.slot_number;         * slot number of the adapter
-	 * paud.device_id;           * adapter identification number
-	 *
-	 * Input:
-	 *
-	 * paud.srate;           * the sampling rate in Hz
-	 * paud.bits_per_sample; * 8, 16, 32, ...
-	 * paud.bsize;           * block size for this rate
-	 * paud.mode;            * ADPCM, PCM, MU_LAW, A_LAW, SOURCE_MIX
-	 * paud.channels;        * 1=mono, 2=stereo
-	 * paud.flags;           * FIXED - fixed length data
-	 *                       * LEFT_ALIGNED, RIGHT_ALIGNED (var len only)
-	 *                       * TWOS_COMPLEMENT - 2's complement data
-	 *                       * SIGNED - signed? comment seems wrong in sys/audio.h
-	 *                       * BIG_ENDIAN
-	 * paud.operation;       * PLAY, RECORD
-	 *
-	 * Output:
-	 *
-	 * paud.flags;           * PITCH            - pitch is supported
-	 *                       * INPUT            - input is supported
-	 *                       * OUTPUT           - output is supported
-	 *                       * MONITOR          - monitor is supported
-	 *                       * VOLUME           - volume is supported
-	 *                       * VOLUME_DELAY     - volume delay is supported
-	 *                       * BALANCE          - balance is supported
-	 *                       * BALANCE_DELAY    - balance delay is supported
-	 *                       * TREBLE           - treble control is supported
-	 *                       * BASS             - bass control is supported
-	 *                       * BESTFIT_PROVIDED - best fit returned
-	 *                       * LOAD_CODE        - DSP load needed
-	 * paud.rc;              * NO_PLAY         - DSP code can't do play requests
-	 *                       * NO_RECORD       - DSP code can't do record requests
-	 *                       * INVALID_REQUEST - request was invalid
-	 *                       * CONFLICT        - conflict with open's flags
-	 *                       * OVERLOADED      - out of DSP MIPS or memory
-	 * paud.position_resolution; * smallest increment for position
-	 */
-
-        paud_init.srate = spec->freq;
-	paud_init.mode = PCM;
-	paud_init.operation = PLAY;
-	paud_init.channels = spec->channels;
-
-	/* Try for a closest match on audio format */
-	format = 0;
-	for ( test_format = SDL_FirstAudioFormat(spec->format);
-						! format && test_format; ) {
-#ifdef DEBUG_AUDIO
-		fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
-#endif
-		switch ( test_format ) {
-			case AUDIO_U8:
-			    bytes_per_sample = 1;
-			    paud_init.bits_per_sample = 8;
-			    paud_init.flags = TWOS_COMPLEMENT | FIXED;
-			    format = 1;
-			    break;
-			case AUDIO_S8:
-			    bytes_per_sample = 1;
-			    paud_init.bits_per_sample = 8;
-			    paud_init.flags = SIGNED |
-					      TWOS_COMPLEMENT | FIXED;
-			    format = 1;
-			    break;
-			case AUDIO_S16LSB:
-			    bytes_per_sample = 2;
-			    paud_init.bits_per_sample = 16;
-			    paud_init.flags = SIGNED |
-					      TWOS_COMPLEMENT | FIXED;
-			    format = 1;
-			    break;
-			case AUDIO_S16MSB:
-			    bytes_per_sample = 2;
-			    paud_init.bits_per_sample = 16;
-			    paud_init.flags = BIG_ENDIAN |
-					      SIGNED |
-					      TWOS_COMPLEMENT | FIXED;
-			    format = 1;
-			    break;
-			case AUDIO_U16LSB:
-			    bytes_per_sample = 2;
-			    paud_init.bits_per_sample = 16;
-			    paud_init.flags = TWOS_COMPLEMENT | FIXED;
-			    format = 1;
-			    break;
-			case AUDIO_U16MSB:
-			    bytes_per_sample = 2;
-			    paud_init.bits_per_sample = 16;
-			    paud_init.flags = BIG_ENDIAN |
-					      TWOS_COMPLEMENT | FIXED;
-			    format = 1;
-			    break;
-			default:
-				break;
-		}
-		if ( ! format ) {
-			test_format = SDL_NextAudioFormat();
-		}
-	}
-	if ( format == 0 ) {
-#ifdef DEBUG_AUDIO
-            fprintf(stderr, "Couldn't find any hardware audio formats\n");
-#endif
-	    SDL_SetError("Couldn't find any hardware audio formats");
-	    return -1;
-	}
-	spec->format = test_format;
-
-	/*
-	 * We know the buffer size and the max number of subsequent writes
-	 * that can be pending. If more than one can pend, allow the application
-	 * to do something like double buffering between our write buffer and
-	 * the device's own buffer that we are filling with write() anyway.
-	 *
-	 * We calculate spec->samples like this because SDL_CalculateAudioSpec()
-	 * will give put paud_bufinfo.write_buf_cap (or paud_bufinfo.write_buf_cap/2)
-	 * into spec->size in return.
-	 */
-	if ( paud_bufinfo.request_buf_cap == 1 )
-	{
-	    spec->samples = paud_bufinfo.write_buf_cap
-			  / bytes_per_sample
-			  / spec->channels;
-	}
-	else
-	{
-	    spec->samples = paud_bufinfo.write_buf_cap
-			  / bytes_per_sample
-			  / spec->channels
-			  / 2;
-	}
-        paud_init.bsize = bytes_per_sample * spec->channels;
-
-	SDL_CalculateAudioSpec(spec);
-
-	/*
-	 * The AIX paud device init can't modify the values of the audio_init
-	 * structure that we pass to it. So we don't need any recalculation
-	 * of this stuff and no reinit call as in linux dsp and dma code.
-	 *
-	 * /dev/paud supports all of the encoding formats, so we don't need
-	 * to do anything like reopening the device, either.
-	 */
-	if ( ioctl(audio_fd, AUDIO_INIT, &paud_init) < 0 ) {
-	    switch ( paud_init.rc )
-	    {
-	    case 1 :
-		SDL_SetError("Couldn't set audio format: DSP can't do play requests");
-		return -1;
-		break;
-	    case 2 :
-		SDL_SetError("Couldn't set audio format: DSP can't do record requests");
-		return -1;
-		break;
-	    case 4 :
-		SDL_SetError("Couldn't set audio format: request was invalid");
-		return -1;
-		break;
-	    case 5 :
-		SDL_SetError("Couldn't set audio format: conflict with open's flags");
-		return -1;
-		break;
-	    case 6 :
-		SDL_SetError("Couldn't set audio format: out of DSP MIPS or memory");
-		return -1;
-		break;
-	    default :
-		SDL_SetError("Couldn't set audio format: not documented in sys/audio.h");
-		return -1;
-		break;
-	    }
-	}
-
-	/* Allocate mixing buffer */
-	mixlen = spec->size;
-	mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
-	if ( mixbuf == NULL ) {
-		return -1;
-	}
-	SDL_memset(mixbuf, spec->silence, spec->size);
-
-	/*
-	 * Set some paramters: full volume, first speaker that we can find.
-	 * Ignore the other settings for now.
-	 */
-	paud_change.input = AUDIO_IGNORE;         /* the new input source */
-        paud_change.output = OUTPUT_1;            /* EXTERNAL_SPEAKER,INTERNAL_SPEAKER,OUTPUT_1 */
-        paud_change.monitor = AUDIO_IGNORE;       /* the new monitor state */
-        paud_change.volume = 0x7fffffff;          /* volume level [0-0x7fffffff] */
-        paud_change.volume_delay = AUDIO_IGNORE;  /* the new volume delay */
-        paud_change.balance = 0x3fffffff;         /* the new balance */
-        paud_change.balance_delay = AUDIO_IGNORE; /* the new balance delay */
-        paud_change.treble = AUDIO_IGNORE;        /* the new treble state */
-        paud_change.bass = AUDIO_IGNORE;          /* the new bass state */
-        paud_change.pitch = AUDIO_IGNORE;         /* the new pitch state */
-
-	paud_control.ioctl_request = AUDIO_CHANGE;
-	paud_control.request_info = (char*)&paud_change;
-	if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 ) {
-#ifdef DEBUG_AUDIO
-            fprintf(stderr, "Can't change audio display settings\n" );
-#endif
-	}
-
-	/*
-	 * Tell the device to expect data. Actual start will wait for
-	 * the first write() call.
-	 */
-	paud_control.ioctl_request = AUDIO_START;
-	paud_control.position = 0;
-	if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 ) {
-#ifdef DEBUG_AUDIO
-            fprintf(stderr, "Can't start audio play\n" );
-#endif
-	    SDL_SetError("Can't start audio play");
-	    return -1;
-	}
-
-        /* Check to see if we need to use select() workaround */
-        { char *workaround;
-                workaround = SDL_getenv("SDL_DSP_NOSELECT");
-                if ( workaround ) {
-                        frame_ticks = (float)(spec->samples*1000)/spec->freq;
-                        next_frame = SDL_GetTicks()+frame_ticks;
-                }
-        }
-
-	/* Get the parent process id (we're the parent of the audio thread) */
-	parent = getpid();
-
-	/* We're ready to rock and roll. :-) */
-	return 0;
-}
-
diff --git a/apps/plugins/sdl/src/audio/paudio/SDL_paudio.h b/apps/plugins/sdl/src/audio/paudio/SDL_paudio.h
deleted file mode 100644
index 72eff1ddef..0000000000
--- a/apps/plugins/sdl/src/audio/paudio/SDL_paudio.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_paudaudio_h
-#define _SDL_paudaudio_h
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData {
-	/* The file descriptor for the audio device */
-	int audio_fd;
-
-	/* The parent process id, to detect when application quits */
-	pid_t parent;
-
-	/* Raw mixing buffer */
-	Uint8 *mixbuf;
-	int    mixlen;
-
-	/* Support for audio timing using a timer, in addition to select() */
-	float frame_ticks;
-	float next_frame;
-};
-#define FUDGE_TICKS	10	/* The scheduler overhead ticks per frame */
-
-/* Old variable names */
-#define audio_fd		(this->hidden->audio_fd)
-#define parent			(this->hidden->parent)
-#define mixbuf			(this->hidden->mixbuf)
-#define mixlen			(this->hidden->mixlen)
-#define frame_ticks		(this->hidden->frame_ticks)
-#define next_frame		(this->hidden->next_frame)
-
-#endif /* _SDL_paudaudio_h */
diff --git a/apps/plugins/sdl/src/audio/pulse/SDL_pulseaudio.c b/apps/plugins/sdl/src/audio/pulse/SDL_pulseaudio.c
deleted file mode 100644
index 29373f37fd..0000000000
--- a/apps/plugins/sdl/src/audio/pulse/SDL_pulseaudio.c
+++ /dev/null
@@ -1,570 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Stéphan Kochen
-    stephan@kochen.nl
-
-    Based on parts of the ALSA and ESounD output drivers.
-*/
-#include "SDL_config.h"
-
-/* Allow access to an PulseAudio network stream mixing buffer */
-
-#include <sys/types.h>
-#include <unistd.h>
-#include <signal.h>
-#include <errno.h>
-#include <pulse/pulseaudio.h>
-#include <pulse/simple.h>
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_audiodev_c.h"
-#include "../../../include/SDL_video.h"  /* for SDL_WM_GetCaption(). */
-#include "SDL_pulseaudio.h"
-
-#ifdef SDL_AUDIO_DRIVER_PULSE_DYNAMIC
-#include "SDL_name.h"
-#include "SDL_loadso.h"
-#else
-#define SDL_NAME(X)	X
-#endif
-
-/* The tag name used by the driver */
-#define PULSE_DRIVER_NAME	"pulse"
-
-/* Audio driver functions */
-static int PULSE_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void PULSE_WaitAudio(_THIS);
-static void PULSE_PlayAudio(_THIS);
-static Uint8 *PULSE_GetAudioBuf(_THIS);
-static void PULSE_CloseAudio(_THIS);
-static void PULSE_WaitDone(_THIS);
-static void PULSE_SetCaption(_THIS, const char *str);
-
-#ifdef SDL_AUDIO_DRIVER_PULSE_DYNAMIC
-
-static const char *pulse_library = SDL_AUDIO_DRIVER_PULSE_DYNAMIC;
-static void *pulse_handle = NULL;
-static int pulse_loaded = 0;
-
-static pa_simple* (*SDL_NAME(pa_simple_new))(
-	const char *server,
-	const char *name,
-	pa_stream_direction_t dir,
-	const char *dev,
-	const char *stream_name,
-	const pa_sample_spec *ss,
-	const pa_channel_map *map,
-	const pa_buffer_attr *attr,
-	int *error
-);
-static void (*SDL_NAME(pa_simple_free))(pa_simple *s);
-
-static pa_channel_map* (*SDL_NAME(pa_channel_map_init_auto))(
-	pa_channel_map *m,
-	unsigned channels,
-	pa_channel_map_def_t def
-);
-
-static pa_mainloop * (*SDL_NAME(pa_mainloop_new))(void);
-static pa_mainloop_api * (*SDL_NAME(pa_mainloop_get_api))(pa_mainloop *m);
-static int (*SDL_NAME(pa_mainloop_iterate))(pa_mainloop *m, int block, int *retval);
-static void (*SDL_NAME(pa_mainloop_free))(pa_mainloop *m);
-
-static pa_operation_state_t (*SDL_NAME(pa_operation_get_state))(pa_operation *o);
-static void (*SDL_NAME(pa_operation_cancel))(pa_operation *o);
-static void (*SDL_NAME(pa_operation_unref))(pa_operation *o);
-
-static pa_context * (*SDL_NAME(pa_context_new))(
-	pa_mainloop_api *m, const char *name);
-static int (*SDL_NAME(pa_context_connect))(
-	pa_context *c, const char *server,
-	pa_context_flags_t flags, const pa_spawn_api *api);
-static pa_context_state_t (*SDL_NAME(pa_context_get_state))(pa_context *c);
-static void (*SDL_NAME(pa_context_disconnect))(pa_context *c);
-static void (*SDL_NAME(pa_context_unref))(pa_context *c);
-
-static pa_stream * (*SDL_NAME(pa_stream_new))(pa_context *c,
-	const char *name, const pa_sample_spec *ss, const pa_channel_map *map);
-static int (*SDL_NAME(pa_stream_connect_playback))(pa_stream *s, const char *dev,
-	const pa_buffer_attr *attr, pa_stream_flags_t flags,
-	pa_cvolume *volume, pa_stream *sync_stream);
-static pa_stream_state_t (*SDL_NAME(pa_stream_get_state))(pa_stream *s);
-static size_t (*SDL_NAME(pa_stream_writable_size))(pa_stream *s);
-static int (*SDL_NAME(pa_stream_write))(pa_stream *s, const void *data, size_t nbytes,
-	pa_free_cb_t free_cb, int64_t offset, pa_seek_mode_t seek);
-static pa_operation * (*SDL_NAME(pa_stream_drain))(pa_stream *s,
-	pa_stream_success_cb_t cb, void *userdata);
-static int (*SDL_NAME(pa_stream_disconnect))(pa_stream *s);
-static void (*SDL_NAME(pa_stream_unref))(pa_stream *s);
-static pa_operation* (*SDL_NAME(pa_context_set_name))(pa_context *c,
-	const char *name, pa_context_success_cb_t cb, void *userdata);
-
-static struct {
-	const char *name;
-	void **func;
-} pulse_functions[] = {
-	{ "pa_simple_new",
-		(void **)&SDL_NAME(pa_simple_new)		},
-	{ "pa_simple_free",
-		(void **)&SDL_NAME(pa_simple_free)		},
-	{ "pa_channel_map_init_auto",
-		(void **)&SDL_NAME(pa_channel_map_init_auto)	},
-	{ "pa_mainloop_new",
-		(void **)&SDL_NAME(pa_mainloop_new)		},
-	{ "pa_mainloop_get_api",
-		(void **)&SDL_NAME(pa_mainloop_get_api)		},
-	{ "pa_mainloop_iterate",
-		(void **)&SDL_NAME(pa_mainloop_iterate)		},
-	{ "pa_mainloop_free",
-		(void **)&SDL_NAME(pa_mainloop_free)		},
-	{ "pa_operation_get_state",
-		(void **)&SDL_NAME(pa_operation_get_state)	},
-	{ "pa_operation_cancel",
-		(void **)&SDL_NAME(pa_operation_cancel)		},
-	{ "pa_operation_unref",
-		(void **)&SDL_NAME(pa_operation_unref)		},
-	{ "pa_context_new",
-		(void **)&SDL_NAME(pa_context_new)		},
-	{ "pa_context_connect",
-		(void **)&SDL_NAME(pa_context_connect)		},
-	{ "pa_context_get_state",
-		(void **)&SDL_NAME(pa_context_get_state)	},
-	{ "pa_context_disconnect",
-		(void **)&SDL_NAME(pa_context_disconnect)	},
-	{ "pa_context_unref",
-		(void **)&SDL_NAME(pa_context_unref)		},
-	{ "pa_stream_new",
-		(void **)&SDL_NAME(pa_stream_new)		},
-	{ "pa_stream_connect_playback",
-		(void **)&SDL_NAME(pa_stream_connect_playback)	},
-	{ "pa_stream_get_state",
-		(void **)&SDL_NAME(pa_stream_get_state)		},
-	{ "pa_stream_writable_size",
-		(void **)&SDL_NAME(pa_stream_writable_size)	},
-	{ "pa_stream_write",
-		(void **)&SDL_NAME(pa_stream_write)		},
-	{ "pa_stream_drain",
-		(void **)&SDL_NAME(pa_stream_drain)		},
-	{ "pa_stream_disconnect",
-		(void **)&SDL_NAME(pa_stream_disconnect)	},
-	{ "pa_stream_unref",
-		(void **)&SDL_NAME(pa_stream_unref)		},
-	{ "pa_context_set_name",
-		(void **)&SDL_NAME(pa_context_set_name)		},
-};
-
-static void UnloadPulseLibrary()
-{
-	if ( pulse_loaded ) {
-		SDL_UnloadObject(pulse_handle);
-		pulse_handle = NULL;
-		pulse_loaded = 0;
-	}
-}
-
-static int LoadPulseLibrary(void)
-{
-	int i, retval = -1;
-
-	pulse_handle = SDL_LoadObject(pulse_library);
-	if ( pulse_handle ) {
-		pulse_loaded = 1;
-		retval = 0;
-		for ( i=0; i<SDL_arraysize(pulse_functions); ++i ) {
-			*pulse_functions[i].func = SDL_LoadFunction(pulse_handle, pulse_functions[i].name);
-			if ( !*pulse_functions[i].func ) {
-				retval = -1;
-				UnloadPulseLibrary();
-				break;
-			}
-		}
-	}
-	return retval;
-}
-
-#else
-
-static void UnloadPulseLibrary()
-{
-	return;
-}
-
-static int LoadPulseLibrary(void)
-{
-	return 0;
-}
-
-#endif /* SDL_AUDIO_DRIVER_PULSE_DYNAMIC */
-
-/* Audio driver bootstrap functions */
-
-static int Audio_Available(void)
-{
-	pa_sample_spec paspec;
-	pa_simple *connection;
-	int available;
-
-	available = 0;
-	if ( LoadPulseLibrary() < 0 ) {
-		return available;
-	}
-
-	/* Connect with a dummy format. */
-	paspec.format = PA_SAMPLE_U8;
-	paspec.rate = 11025;
-	paspec.channels = 1;
-	connection = SDL_NAME(pa_simple_new)(
-		NULL,                        /* server */
-		"Test stream",               /* application name */
-		PA_STREAM_PLAYBACK,          /* playback mode */
-		NULL,                        /* device on the server */
-		"Simple DirectMedia Layer",  /* stream description */
-		&paspec,                     /* sample format spec */
-		NULL,                        /* channel map */
-		NULL,                        /* buffering attributes */
-		NULL                         /* error code */
-	);
-	if ( connection != NULL ) {
-		available = 1;
-		SDL_NAME(pa_simple_free)(connection);
-	}
-
-	UnloadPulseLibrary();
-	return(available);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-	SDL_free(device->hidden->caption);
-	SDL_free(device->hidden);
-	SDL_free(device);
-	UnloadPulseLibrary();
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	LoadPulseLibrary();
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-	if ( this ) {
-		SDL_memset(this, 0, (sizeof *this));
-		this->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *this->hidden));
-	}
-	if ( (this == NULL) || (this->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( this ) {
-			SDL_free(this);
-		}
-		return(0);
-	}
-	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-	/* Set the function pointers */
-	this->OpenAudio = PULSE_OpenAudio;
-	this->WaitAudio = PULSE_WaitAudio;
-	this->PlayAudio = PULSE_PlayAudio;
-	this->GetAudioBuf = PULSE_GetAudioBuf;
-	this->CloseAudio = PULSE_CloseAudio;
-	this->WaitDone = PULSE_WaitDone;
-	this->SetCaption = PULSE_SetCaption;
-
-	this->free = Audio_DeleteDevice;
-
-	return this;
-}
-
-AudioBootStrap PULSE_bootstrap = {
-	PULSE_DRIVER_NAME, "PulseAudio",
-	Audio_Available, Audio_CreateDevice
-};
-
-/* This function waits until it is possible to write a full sound buffer */
-static void PULSE_WaitAudio(_THIS)
-{
-	int size;
-	while(1) {
-		if (SDL_NAME(pa_context_get_state)(context) != PA_CONTEXT_READY ||
-		    SDL_NAME(pa_stream_get_state)(stream) != PA_STREAM_READY ||
-		    SDL_NAME(pa_mainloop_iterate)(mainloop, 1, NULL) < 0) {
-			this->enabled = 0;
-			return;
-		}
-		size = SDL_NAME(pa_stream_writable_size)(stream);
-		if (size >= mixlen)
-			return;
-	}
-}
-
-static void PULSE_PlayAudio(_THIS)
-{
-	/* Write the audio data */
-	if (SDL_NAME(pa_stream_write)(stream, mixbuf, mixlen, NULL, 0LL, PA_SEEK_RELATIVE) < 0)
-		this->enabled = 0;
-}
-
-static Uint8 *PULSE_GetAudioBuf(_THIS)
-{
-	return(mixbuf);
-}
-
-static void PULSE_CloseAudio(_THIS)
-{
-	if ( mixbuf != NULL ) {
-		SDL_FreeAudioMem(mixbuf);
-		mixbuf = NULL;
-	}
-	if ( stream != NULL ) {
-		SDL_NAME(pa_stream_disconnect)(stream);
-		SDL_NAME(pa_stream_unref)(stream);
-		stream = NULL;
-	}
-	if (context != NULL) {
-		SDL_NAME(pa_context_disconnect)(context);
-		SDL_NAME(pa_context_unref)(context);
-		context = NULL;
-	}
-	if (mainloop != NULL) {
-		SDL_NAME(pa_mainloop_free)(mainloop);
-		mainloop = NULL;
-	}
-}
-
-/* Try to get the name of the program */
-static char *get_progname(void)
-{
-#ifdef __LINUX__
-	char *progname = NULL;
-	FILE *fp;
-	static char temp[BUFSIZ];
-
-	SDL_snprintf(temp, SDL_arraysize(temp), "/proc/%d/cmdline", getpid());
-	fp = fopen(temp, "r");
-	if ( fp != NULL ) {
-		if ( fgets(temp, sizeof(temp)-1, fp) ) {
-			progname = SDL_strrchr(temp, '/');
-			if ( progname == NULL ) {
-				progname = temp;
-			} else {
-				progname = progname+1;
-			}
-		}
-		fclose(fp);
-	}
-	return(progname);
-#elif defined(__NetBSD__)
-	return getprogname();
-#else
-	return("unknown");
-#endif
-}
-
-static void caption_set_complete(pa_context *c, int success, void *userdata)
-{
-	/* no-op. */
-}
-
-static void PULSE_SetCaption(_THIS, const char *str)
-{
-	SDL_free(this->hidden->caption);
-	if ((str == NULL) || (*str == '\0')) {
-		str = get_progname();  /* set a default so SOMETHING shows up. */
-	}
-	this->hidden->caption = SDL_strdup(str);
-	if (context != NULL) {
-		SDL_NAME(pa_context_set_name)(context, this->hidden->caption,
-		                              caption_set_complete, 0);
-	}
-}
-
-static void stream_drain_complete(pa_stream *s, int success, void *userdata)
-{
-	/* no-op. */
-}
-
-static void PULSE_WaitDone(_THIS)
-{
-	pa_operation *o;
-
-	o = SDL_NAME(pa_stream_drain)(stream, stream_drain_complete, NULL);
-	if (!o)
-		return;
-
-	while (SDL_NAME(pa_operation_get_state)(o) != PA_OPERATION_DONE) {
-		if (SDL_NAME(pa_context_get_state)(context) != PA_CONTEXT_READY ||
-		    SDL_NAME(pa_stream_get_state)(stream) != PA_STREAM_READY ||
-		    SDL_NAME(pa_mainloop_iterate)(mainloop, 1, NULL) < 0) {
-			SDL_NAME(pa_operation_cancel)(o);
-			break;
-		}
-	}
-	SDL_NAME(pa_operation_unref)(o);
-}
-
-static int PULSE_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	int             state;
-	Uint16          test_format;
-	pa_sample_spec  paspec;
-	pa_buffer_attr  paattr;
-	pa_channel_map  pacmap;
-	pa_stream_flags_t flags = 0;
-
-	paspec.format = PA_SAMPLE_INVALID;
-	for ( test_format = SDL_FirstAudioFormat(spec->format); test_format; ) {
-		switch ( test_format ) {
-			case AUDIO_U8:
-				paspec.format = PA_SAMPLE_U8;
-				break;
-			case AUDIO_S16LSB:
-				paspec.format = PA_SAMPLE_S16LE;
-				break;
-			case AUDIO_S16MSB:
-				paspec.format = PA_SAMPLE_S16BE;
-				break;
-		}
-		if ( paspec.format != PA_SAMPLE_INVALID )
-			break;
-		test_format = SDL_NextAudioFormat();
-	}
-	if (paspec.format == PA_SAMPLE_INVALID ) {
-		SDL_SetError("Couldn't find any suitable audio formats");
-		return(-1);
-	}
-	spec->format = test_format;
-
-	paspec.channels = spec->channels;
-	paspec.rate = spec->freq;
-
-	/* Calculate the final parameters for this audio specification */
-#ifdef PA_STREAM_ADJUST_LATENCY
-	spec->samples /= 2; /* Mix in smaller chunck to avoid underruns */
-#endif
-	SDL_CalculateAudioSpec(spec);
-
-	/* Allocate mixing buffer */
-	mixlen = spec->size;
-	mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
-	if ( mixbuf == NULL ) {
-		return(-1);
-	}
-	SDL_memset(mixbuf, spec->silence, spec->size);
-
-	/* Reduced prebuffering compared to the defaults. */
-#ifdef PA_STREAM_ADJUST_LATENCY
-	paattr.tlength = mixlen * 4; /* 2x original requested bufsize */
-	paattr.prebuf = -1;
-	paattr.maxlength = -1;
-	paattr.minreq = mixlen; /* -1 can lead to pa_stream_writable_size()
-				   >= mixlen never becoming true */
-	flags = PA_STREAM_ADJUST_LATENCY;
-#else
-	paattr.tlength = mixlen*2;
-	paattr.prebuf = mixlen*2;
-	paattr.maxlength = mixlen*2;
-	paattr.minreq = mixlen;
-#endif
-
-	/* The SDL ALSA output hints us that we use Windows' channel mapping */
-	/* http://bugzilla.libsdl.org/show_bug.cgi?id=110 */
-	SDL_NAME(pa_channel_map_init_auto)(
-		&pacmap, spec->channels, PA_CHANNEL_MAP_WAVEEX);
-
-	/* Set up a new main loop */
-	if (!(mainloop = SDL_NAME(pa_mainloop_new)())) {
-		PULSE_CloseAudio(this);
-		SDL_SetError("pa_mainloop_new() failed");
-		return(-1);
-	}
-
-	if (this->hidden->caption == NULL) {
-		char *title = NULL;
-		SDL_WM_GetCaption(&title, NULL);
-		PULSE_SetCaption(this, title);
-	}
-
-	mainloop_api = SDL_NAME(pa_mainloop_get_api)(mainloop);
-	if (!(context = SDL_NAME(pa_context_new)(mainloop_api,
-	                                         this->hidden->caption))) {
-		PULSE_CloseAudio(this);
-		SDL_SetError("pa_context_new() failed");
-		return(-1);
-	}
-
-	/* Connect to the PulseAudio server */
-	if (SDL_NAME(pa_context_connect)(context, NULL, 0, NULL) < 0) {
-		PULSE_CloseAudio(this);
-		SDL_SetError("Could not setup connection to PulseAudio");
-		return(-1);
-	}
-
-	do {
-		if (SDL_NAME(pa_mainloop_iterate)(mainloop, 1, NULL) < 0) {
-			PULSE_CloseAudio(this);
-			SDL_SetError("pa_mainloop_iterate() failed");
-			return(-1);
-		}
-		state = SDL_NAME(pa_context_get_state)(context);
-		if (!PA_CONTEXT_IS_GOOD(state)) {
-			PULSE_CloseAudio(this);
-			SDL_SetError("Could not connect to PulseAudio");
-			return(-1);
-		}
-	} while (state != PA_CONTEXT_READY);
-
-	stream = SDL_NAME(pa_stream_new)(
-		context,
-		"Simple DirectMedia Layer",  /* stream description */
-		&paspec,                     /* sample format spec */
-		&pacmap                      /* channel map */
-	);
-	if ( stream == NULL ) {
-		PULSE_CloseAudio(this);
-		SDL_SetError("Could not setup PulseAudio stream");
-		return(-1);
-	}
-
-	if (SDL_NAME(pa_stream_connect_playback)(stream, NULL, &paattr, flags,
-			NULL, NULL) < 0) {
-		PULSE_CloseAudio(this);
-		SDL_SetError("Could not connect PulseAudio stream");
-		return(-1);
-	}
-
-	do {
-		if (SDL_NAME(pa_mainloop_iterate)(mainloop, 1, NULL) < 0) {
-			PULSE_CloseAudio(this);
-			SDL_SetError("pa_mainloop_iterate() failed");
-			return(-1);
-		}
-		state = SDL_NAME(pa_stream_get_state)(stream);
-		if (!PA_STREAM_IS_GOOD(state)) {
-			PULSE_CloseAudio(this);
-			SDL_SetError("Could not create to PulseAudio stream");
-			return(-1);
-		}
-	} while (state != PA_STREAM_READY);
-
-	return(0);
-}
diff --git a/apps/plugins/sdl/src/audio/pulse/SDL_pulseaudio.h b/apps/plugins/sdl/src/audio/pulse/SDL_pulseaudio.h
deleted file mode 100644
index 63ee751ef6..0000000000
--- a/apps/plugins/sdl/src/audio/pulse/SDL_pulseaudio.h
+++ /dev/null
@@ -1,73 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Stéphan Kochen
-    stephan@kochen.nl
-    
-    Based on parts of the ALSA and ESounD output drivers.
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_pulseaudio_h
-#define _SDL_pulseaudio_h
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData {
-	pa_mainloop *mainloop;
-	pa_mainloop_api *mainloop_api;
-	pa_context *context;
-	pa_stream *stream;
-
-	char *caption;
-
-	/* Raw mixing buffer */
-	Uint8 *mixbuf;
-	int    mixlen;
-};
-
-#if (PA_API_VERSION < 12)
-/** Return non-zero if the passed state is one of the connected states */
-static inline int PA_CONTEXT_IS_GOOD(pa_context_state_t x) {
-    return
-        x == PA_CONTEXT_CONNECTING ||
-        x == PA_CONTEXT_AUTHORIZING ||
-        x == PA_CONTEXT_SETTING_NAME ||
-        x == PA_CONTEXT_READY;
-}
-/** Return non-zero if the passed state is one of the connected states */
-static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x) {
-    return
-        x == PA_STREAM_CREATING ||
-        x == PA_STREAM_READY;
-}
-#endif	/* pulseaudio <= 0.9.10 */
-
-/* Old variable names */
-#define mainloop		(this->hidden->mainloop)
-#define mainloop_api		(this->hidden->mainloop_api)
-#define context			(this->hidden->context)
-#define stream			(this->hidden->stream)
-#define mixbuf			(this->hidden->mixbuf)
-#define mixlen			(this->hidden->mixlen)
-
-#endif /* _SDL_pulseaudio_h */
-
diff --git a/apps/plugins/sdl/src/audio/sun/SDL_sunaudio.c b/apps/plugins/sdl/src/audio/sun/SDL_sunaudio.c
deleted file mode 100644
index 7a39e71d1b..0000000000
--- a/apps/plugins/sdl/src/audio/sun/SDL_sunaudio.c
+++ /dev/null
@@ -1,432 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/* Allow access to a raw mixing buffer */
-
-#include <fcntl.h>
-#include <errno.h>
-#ifdef __NETBSD__
-#include <sys/ioctl.h>
-#include <sys/audioio.h>
-#endif
-#ifdef __SVR4
-#include <sys/audioio.h>
-#else
-#include <sys/time.h>
-#include <sys/types.h>
-#endif
-#include <unistd.h>
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audiomem.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_audiodev_c.h"
-#include "SDL_sunaudio.h"
-
-/* Open the audio device for playback, and don't block if busy */
-#define OPEN_FLAGS	(O_WRONLY|O_NONBLOCK)
-
-/* Audio driver functions */
-static int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void DSP_WaitAudio(_THIS);
-static void DSP_PlayAudio(_THIS);
-static Uint8 *DSP_GetAudioBuf(_THIS);
-static void DSP_CloseAudio(_THIS);
-
-static Uint8 snd2au(int sample);
-
-/* Audio driver bootstrap functions */
-
-static int Audio_Available(void)
-{
-	int fd;
-	int available;
-
-	available = 0;
-	fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 1);
-	if ( fd >= 0 ) {
-		available = 1;
-		close(fd);
-	}
-	return(available);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-	SDL_free(device->hidden);
-	SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-	if ( this ) {
-		SDL_memset(this, 0, (sizeof *this));
-		this->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *this->hidden));
-	}
-	if ( (this == NULL) || (this->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( this ) {
-			SDL_free(this);
-		}
-		return(0);
-	}
-	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-	audio_fd = -1;
-
-	/* Set the function pointers */
-	this->OpenAudio = DSP_OpenAudio;
-	this->WaitAudio = DSP_WaitAudio;
-	this->PlayAudio = DSP_PlayAudio;
-	this->GetAudioBuf = DSP_GetAudioBuf;
-	this->CloseAudio = DSP_CloseAudio;
-
-	this->free = Audio_DeleteDevice;
-
-	return this;
-}
-
-AudioBootStrap SUNAUDIO_bootstrap = {
-	"audio", "UNIX /dev/audio interface",
-	Audio_Available, Audio_CreateDevice
-};
-
-#ifdef DEBUG_AUDIO
-void CheckUnderflow(_THIS)
-{
-#ifdef AUDIO_GETINFO
-	audio_info_t info;
-	int left;
-
-	ioctl(audio_fd, AUDIO_GETINFO, &info);
-	left = (written - info.play.samples);
-	if ( written && (left == 0) ) {
-		fprintf(stderr, "audio underflow!\n");
-	}
-#endif
-}
-#endif
-
-void DSP_WaitAudio(_THIS)
-{
-#ifdef AUDIO_GETINFO
-#define SLEEP_FUDGE	10		/* 10 ms scheduling fudge factor */
-	audio_info_t info;
-	Sint32 left;
-
-	ioctl(audio_fd, AUDIO_GETINFO, &info);
-	left = (written - info.play.samples);
-	if ( left > fragsize ) {
-		Sint32 sleepy;
-
-		sleepy = ((left - fragsize)/frequency);
-		sleepy -= SLEEP_FUDGE;
-		if ( sleepy > 0 ) {
-			SDL_Delay(sleepy);
-		}
-	}
-#else
-	fd_set fdset;
-
-	FD_ZERO(&fdset);
-	FD_SET(audio_fd, &fdset);
-	select(audio_fd+1, NULL, &fdset, NULL, NULL);
-#endif
-}
-
-void DSP_PlayAudio(_THIS)
-{
-	/* Write the audio data */
-	if ( ulaw_only ) {
-		/* Assuming that this->spec.freq >= 8000 Hz */
-		int accum, incr, pos;
-		Uint8 *aubuf;
-
-		accum = 0;
-		incr  = this->spec.freq/8;
-		aubuf = ulaw_buf;
-		switch (audio_fmt & 0xFF) {
-			case 8: {
-				Uint8 *sndbuf;
-
-				sndbuf = mixbuf;
-				for ( pos=0; pos < fragsize; ++pos ) {
-					*aubuf = snd2au((0x80-*sndbuf)*64);
-					accum += incr;
-					while ( accum > 0 ) {
-						accum -= 1000;
-						sndbuf += 1;
-					}
-					aubuf += 1;
-				}
-			}
-			break;
-			case 16: {
-				Sint16 *sndbuf;
-
-				sndbuf = (Sint16 *)mixbuf;
-				for ( pos=0; pos < fragsize; ++pos ) {
-					*aubuf = snd2au(*sndbuf/4);
-					accum += incr;
-					while ( accum > 0 ) {
-						accum -= 1000;
-						sndbuf += 1;
-					}
-					aubuf += 1;
-				}
-			}
-			break;
-		}
-#ifdef DEBUG_AUDIO
-		CheckUnderflow(this);
-#endif
-		if ( write(audio_fd, ulaw_buf, fragsize) < 0 ) {
-			/* Assume fatal error, for now */
-			this->enabled = 0;
-		}
-		written += fragsize;
-	} else {
-#ifdef DEBUG_AUDIO
-		CheckUnderflow(this);
-#endif
-		if ( write(audio_fd, mixbuf, this->spec.size) < 0 ) {
-			/* Assume fatal error, for now */
-			this->enabled = 0;
-		}
-		written += fragsize;
-	}
-}
-
-Uint8 *DSP_GetAudioBuf(_THIS)
-{
-	return(mixbuf);
-}
-
-void DSP_CloseAudio(_THIS)
-{
-	if ( mixbuf != NULL ) {
-		SDL_FreeAudioMem(mixbuf);
-		mixbuf = NULL;
-	}
-	if ( ulaw_buf != NULL ) {
-		SDL_free(ulaw_buf);
-		ulaw_buf = NULL;
-	}
-	close(audio_fd);
-}
-
-int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	char audiodev[1024];
-#ifdef AUDIO_SETINFO
-	int enc;
-#endif
-	int desired_freq = spec->freq;
-
-	/* Initialize our freeable variables, in case we fail*/
-	audio_fd = -1;
-	mixbuf = NULL;
-	ulaw_buf = NULL;
-
-	/* Determine the audio parameters from the AudioSpec */
-	switch ( spec->format & 0xFF ) {
-
-		case 8: { /* Unsigned 8 bit audio data */
-			spec->format = AUDIO_U8;
-#ifdef AUDIO_SETINFO
-			enc = AUDIO_ENCODING_LINEAR8;
-#endif
-		}
-		break;
-
-		case 16: { /* Signed 16 bit audio data */
-		        spec->format = AUDIO_S16SYS;
-#ifdef AUDIO_SETINFO
-			enc = AUDIO_ENCODING_LINEAR;
-#endif
-		}
-		break;
-
-		default: {
-			SDL_SetError("Unsupported audio format");
-			return(-1);
-		}
-	}
-	audio_fmt = spec->format;
-
-	/* Open the audio device */
-	audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 1);
-	if ( audio_fd < 0 ) {
-		SDL_SetError("Couldn't open %s: %s", audiodev,
-			     strerror(errno));
-		return(-1);
-	}
-
-	ulaw_only = 0;		/* modern Suns do support linear audio */
-#ifdef AUDIO_SETINFO
-	for(;;) {
-	    audio_info_t info;
-	    AUDIO_INITINFO(&info); /* init all fields to "no change" */
-
-	    /* Try to set the requested settings */
-	    info.play.sample_rate = spec->freq;
-	    info.play.channels = spec->channels;
-	    info.play.precision = (enc == AUDIO_ENCODING_ULAW)
-		                  ? 8 : spec->format & 0xff;
-	    info.play.encoding = enc;
-	    if( ioctl(audio_fd, AUDIO_SETINFO, &info) == 0 ) {
-
-		/* Check to be sure we got what we wanted */
-		if(ioctl(audio_fd, AUDIO_GETINFO, &info) < 0) {
-		    SDL_SetError("Error getting audio parameters: %s",
-				 strerror(errno));
-		    return -1;
-		}
-		if(info.play.encoding == enc
-		   && info.play.precision == (spec->format & 0xff)
-		   && info.play.channels == spec->channels) {
-		    /* Yow! All seems to be well! */
-		    spec->freq = info.play.sample_rate;
-		    break;
-		}
-	    }
-
-	    switch(enc) {
-	    case AUDIO_ENCODING_LINEAR8:
-		/* unsigned 8bit apparently not supported here */
-		enc = AUDIO_ENCODING_LINEAR;
-		spec->format = AUDIO_S16SYS;
-		break;	/* try again */
-
-	    case AUDIO_ENCODING_LINEAR:
-		/* linear 16bit didn't work either, resort to µ-law */
-		enc = AUDIO_ENCODING_ULAW;
-		spec->channels = 1;
-		spec->freq = 8000;
-		spec->format = AUDIO_U8;
-		ulaw_only = 1;
-		break;
-
-	    default:
-		/* oh well... */
-		SDL_SetError("Error setting audio parameters: %s",
-			     strerror(errno));
-		return -1;
-	    }
-	}
-#endif /* AUDIO_SETINFO */
-	written = 0;
-
-	/* We can actually convert on-the-fly to U-Law */
-	if ( ulaw_only ) {
-	        spec->freq = desired_freq;
-		fragsize = (spec->samples*1000)/(spec->freq/8);
-		frequency = 8;
-		ulaw_buf = (Uint8 *)SDL_malloc(fragsize);
-		if ( ulaw_buf == NULL ) {
-			SDL_OutOfMemory();
-			return(-1);
-		}
-		spec->channels = 1;
-	} else {
-		fragsize = spec->samples;
-		frequency = spec->freq/1000;
-	}
-#ifdef DEBUG_AUDIO
-	fprintf(stderr, "Audio device %s U-Law only\n", 
-				ulaw_only ? "is" : "is not");
-	fprintf(stderr, "format=0x%x chan=%d freq=%d\n",
-		spec->format, spec->channels, spec->freq);
-#endif
-
-	/* Update the fragment size as size in bytes */
-	SDL_CalculateAudioSpec(spec);
-
-	/* Allocate mixing buffer */
-	mixbuf = (Uint8 *)SDL_AllocAudioMem(spec->size);
-	if ( mixbuf == NULL ) {
-		SDL_OutOfMemory();
-		return(-1);
-	}
-	SDL_memset(mixbuf, spec->silence, spec->size);
-
-	/* We're ready to rock and roll. :-) */
-	return(0);
-}
-
-/************************************************************************/
-/* This function (snd2au()) copyrighted:                                */
-/************************************************************************/
-/*      Copyright 1989 by Rich Gopstein and Harris Corporation          */
-/*                                                                      */
-/*      Permission to use, copy, modify, and distribute this software   */
-/*      and its documentation for any purpose and without fee is        */
-/*      hereby granted, provided that the above copyright notice        */
-/*      appears in all copies and that both that copyright notice and   */
-/*      this permission notice appear in supporting documentation, and  */
-/*      that the name of Rich Gopstein and Harris Corporation not be    */
-/*      used in advertising or publicity pertaining to distribution     */
-/*      of the software without specific, written prior permission.     */
-/*      Rich Gopstein and Harris Corporation make no representations    */
-/*      about the suitability of this software for any purpose.  It     */
-/*      provided "as is" without express or implied warranty.           */
-/************************************************************************/
-
-static Uint8 snd2au(int sample)
-{
-
-	int mask;
-
-	if (sample < 0) {
-		sample = -sample;
-		mask = 0x7f;
-	} else {
-		mask = 0xff;
-	}
-
-	if (sample < 32) {
-		sample = 0xF0 | (15 - sample / 2);
-	} else if (sample < 96) {
-		sample = 0xE0 | (15 - (sample - 32) / 4);
-	} else if (sample < 224) {
-		sample = 0xD0 | (15 - (sample - 96) / 8);
-	} else if (sample < 480) {
-		sample = 0xC0 | (15 - (sample - 224) / 16);
-	} else if (sample < 992) {
-		sample = 0xB0 | (15 - (sample - 480) / 32);
-	} else if (sample < 2016) {
-		sample = 0xA0 | (15 - (sample - 992) / 64);
-	} else if (sample < 4064) {
-		sample = 0x90 | (15 - (sample - 2016) / 128);
-	} else if (sample < 8160) {
-		sample = 0x80 | (15 - (sample - 4064) /  256);
-	} else {
-		sample = 0x80;
-	}
-	return (mask & sample);
-}
diff --git a/apps/plugins/sdl/src/audio/sun/SDL_sunaudio.h b/apps/plugins/sdl/src/audio/sun/SDL_sunaudio.h
deleted file mode 100644
index e6be419376..0000000000
--- a/apps/plugins/sdl/src/audio/sun/SDL_sunaudio.h
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_lowaudio_h
-#define _SDL_lowaudio_h
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData {
-	/* The file descriptor for the audio device */
-	int audio_fd;
-
-	Uint16 audio_fmt;        /* The app audio format */
-	Uint8 *mixbuf;           /* The app mixing buffer */
-	int ulaw_only;           /* Flag -- does hardware only output U-law? */
-	Uint8 *ulaw_buf;         /* The U-law mixing buffer */
-	Sint32 written;          /* The number of samples written */
-	int fragsize;            /* The audio fragment size in samples */
-	int frequency;           /* The audio frequency in KHz */
-};
-
-/* Old variable names */
-#define audio_fd		(this->hidden->audio_fd)
-#define audio_fmt		(this->hidden->audio_fmt)
-#define mixbuf			(this->hidden->mixbuf)
-#define ulaw_only		(this->hidden->ulaw_only)
-#define ulaw_buf		(this->hidden->ulaw_buf)
-#define written			(this->hidden->written)
-#define fragsize		(this->hidden->fragsize)
-#define frequency		(this->hidden->frequency)
-
-#endif /* _SDL_lowaudio_h */
diff --git a/apps/plugins/sdl/src/audio/symbian/SDL_epocaudio.cpp b/apps/plugins/sdl/src/audio/symbian/SDL_epocaudio.cpp
deleted file mode 100644
index 72a4eaf4e1..0000000000
--- a/apps/plugins/sdl/src/audio/symbian/SDL_epocaudio.cpp
+++ /dev/null
@@ -1,614 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@devolution.com
-*/
-
-/*
-    SDL_epocaudio.cpp
-    Epoc based SDL audio driver implementation
-    
-    Markus Mertama
-*/
-
-#ifdef SAVE_RCSID
-static char rcsid =
- "@(#) $Id: SDL_epocaudio.c,v 0.0.0.0 2001/06/19 17:19:56 hercules Exp $";
-#endif
-
-
-#include <stdlib.h>
-#include <stdio.h>
-#include <string.h>
-#include <errno.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include <signal.h>
-#include <sys/time.h>
-#include <sys/ioctl.h>
-#include <sys/stat.h>
-
-#include "epoc_sdl.h"
-
-#include <e32hal.h>
-
-
-extern "C" {
-#include "SDL_audio.h"
-#include "SDL_error.h"
-#include "SDL_audiomem.h"
-#include "SDL_audio_c.h"
-#include "SDL_timer.h"
-#include "SDL_audiodev_c.h"
-}
-
-#include "SDL_epocaudio.h"
-
-#include "streamplayer.h"
-
-
-//#define DEBUG_AUDIO
-
-
-/* Audio driver functions */
-
-static int EPOC_OpenAudio(SDL_AudioDevice *thisdevice, SDL_AudioSpec *spec);
-static void EPOC_WaitAudio(SDL_AudioDevice *thisdevice);
-static void EPOC_PlayAudio(SDL_AudioDevice *thisdevice);
-static Uint8 *EPOC_GetAudioBuf(SDL_AudioDevice *thisdevice);
-static void EPOC_CloseAudio(SDL_AudioDevice *thisdevice);
-static void EPOC_ThreadInit(SDL_AudioDevice *thisdevice);
-
-static int Audio_Available(void);
-static SDL_AudioDevice *Audio_CreateDevice(int devindex);
-static void Audio_DeleteDevice(SDL_AudioDevice *device);
-
-
-//void sos_adump(SDL_AudioDevice* thisdevice, void* data, int len);
-
-#ifdef __WINS__
-#define DODUMP
-#endif
-
-#ifdef DODUMP
-NONSHARABLE_CLASS(TDump)
-	{
-	public:
-	TInt Open();
-	void Close();
-	void Dump(const TDesC8& aDes);
-	private:
-		RFile iFile;
-    	RFs iFs; 
-	};
-	
-TInt TDump::Open()
-	{
-	TInt err = iFs.Connect();
-	if(err == KErrNone)
-		{
-#ifdef __WINS__
-_LIT(target, "C:\\sdlau.raw");
-#else
-_LIT(target, "E:\\sdlau.raw");
-#endif 
-		err = iFile.Replace(iFs, target, EFileWrite);
-		}
-	return err;
-	}
-void TDump::Close()
-	{
-	iFile.Close();
-	iFs.Close();
-	}
-void TDump::Dump(const TDesC8& aDes)
-	{
-	iFile.Write(aDes);
-	}
-#endif
-
-
-NONSHARABLE_CLASS(CSimpleWait) : public CTimer
-	{
-	public:
-		void Wait(TTimeIntervalMicroSeconds32 aWait);
-		static CSimpleWait* NewL();
-	private:
-		CSimpleWait();
-		void RunL();
-	};
-
-
-CSimpleWait* CSimpleWait::NewL()
-	{
-	CSimpleWait* wait = new (ELeave) CSimpleWait();
-	CleanupStack::PushL(wait);
-	wait->ConstructL();
-	CleanupStack::Pop();
-	return wait;
-	}
-
-void CSimpleWait::Wait(TTimeIntervalMicroSeconds32 aWait)
-	{
-	After(aWait);
-	CActiveScheduler::Start();
-	}
-	
-CSimpleWait::CSimpleWait() : CTimer(CActive::EPriorityStandard)	
-	{
-	CActiveScheduler::Add(this);
-	}
-
-void CSimpleWait::RunL()
-	{
-	CActiveScheduler::Stop();
-	}
-
-const TInt KAudioBuffers(2);
-	
-
-NONSHARABLE_CLASS(CEpocAudio) : public CBase, public MStreamObs, public MStreamProvider
-    {
-    public:
-    	static void* NewL(TInt BufferSize, TInt aFill);
-    	inline static CEpocAudio& Current(SDL_AudioDevice* thisdevice);
-    	
-    	static void Free(SDL_AudioDevice* thisdevice);
- 		
-    	void Wait();
-    	void Play();
-    //	void SetBuffer(const TDesC8& aBuffer);
-    	void ThreadInitL(TAny* aDevice);
-    	void Open(TInt iRate, TInt iChannels, TUint32 aType, TInt aBytes);
-    	~CEpocAudio();
-    	TUint8* Buffer();
-    	TBool SetPause(TBool aPause);
-    #ifdef DODUMP
-    	void Dump(const TDesC8& aBuf) {iDump.Dump(aBuf);}
-    #endif
-    private:
-    	CEpocAudio(TInt aBufferSize);
-    	void Complete(TInt aState, TInt aError);
-    	TPtrC8 Data();
-    	void ConstructL(TInt aFill);
-    private:
-    	TInt iBufferSize;
-    	CStreamPlayer* iPlayer;
-    	TInt iBufferRate;
-    	TInt iRate;
-    	TInt iChannels;
-    	TUint32 iType;
-    	TInt iPosition;
-    	TThreadId iTid;
-    	TUint8* iAudioPtr;
-    	TUint8* iBuffer;
-    //	TTimeIntervalMicroSeconds iStart;
-    	TTime iStart;
-    	TInt iTune;
-    	CSimpleWait* iWait;
-    #ifdef DODUMP
-    	TDump iDump;
-    #endif
-    };
-
-inline CEpocAudio& CEpocAudio::Current(SDL_AudioDevice* thisdevice)
-	{
-	return *static_cast<CEpocAudio*>((void*)thisdevice->hidden);
-	}
-	
-/*
-
-TBool EndSc(TAny*)
-	{	
-	CActiveScheduler::Stop();
-	}
-	
-LOCAL_C void CleanScL()
-	{
-	CIdle* d = CIdle::NewLC(CActive:::EPriorityIdle);
-	d->Start(TCallBack(EndSc));
-	CActiveScheduler::Start();
-	
-	}
-*/
-	
-void CEpocAudio::Free(SDL_AudioDevice* thisdevice)
-	{
-    CEpocAudio* ea = static_cast<CEpocAudio*>((void*)thisdevice->hidden);
-    if(ea)
-    	{
-		ASSERT(ea->iTid == RThread().Id());
-    	delete ea;
-    	thisdevice->hidden = NULL;	
-   
-    	CActiveScheduler* as =  CActiveScheduler::Current();
-    	ASSERT(as->StackDepth() == 0);    	
-    	delete as;
-    	CActiveScheduler::Install(NULL);
-    	}
-    ASSERT(thisdevice->hidden == NULL);
-	}
-	
-CEpocAudio::CEpocAudio(TInt aBufferSize) : iBufferSize(aBufferSize), iPosition(-1) 
-	{
-	}
-
-void* CEpocAudio::NewL(TInt aBufferSize, TInt aFill)
-	{
-	CEpocAudio* eAudioLib = new (ELeave) CEpocAudio(aBufferSize);
-	CleanupStack::PushL(eAudioLib);
-	eAudioLib->ConstructL(aFill);
-	CleanupStack::Pop();
-	return eAudioLib;
-	}
-	
-void CEpocAudio::ConstructL(TInt aFill)
-	{
-	iBuffer = (TUint8*) User::AllocL(KAudioBuffers * iBufferSize);
-	memset(iBuffer, aFill, KAudioBuffers * iBufferSize);
-	iAudioPtr = iBuffer;
-	}
-
-
-TBool CEpocAudio::SetPause(TBool aPause)
-	{
-	if(aPause && iPosition >= 0)
-		{
-		iPosition = -1;
-		if(iPlayer != NULL)
-			iPlayer->Stop();
-		}
-	if(!aPause && iPosition < 0)
-		{
-		iPosition = 0;
-		if(iPlayer != NULL)
-			iPlayer->Start();
-		}
-	return iPosition < 0;
-	}
-	
-void CEpocAudio::ThreadInitL(TAny* aDevice)
-	{
-	iTid = RThread().Id(); 
-	CActiveScheduler* as =  new (ELeave) CActiveScheduler();
-    CActiveScheduler::Install(as);
-    
-    EpocSdlEnv::AppendCleanupItem(TSdlCleanupItem((TSdlCleanupOperation)EPOC_CloseAudio, aDevice));
-   
-    iWait = CSimpleWait::NewL();
-   
-    iPlayer = new (ELeave) CStreamPlayer(*this, *this);
-    iPlayer->ConstructL();	
-    iPlayer->OpenStream(iRate, iChannels, iType);
-    
-    #ifdef DODUMP
-    User::LeaveIfError(iDump.Open());
-    #endif
-	}
-	
-	
-	
-TUint8* CEpocAudio::Buffer()
-	{
-	iStart.UniversalTime();
-//	iStart = iPlayer->Position();		
-	return iAudioPtr;
-
-	}
-	
-CEpocAudio::~CEpocAudio()
-	{
-	if(iWait != NULL)
-		iWait->Cancel();
-	delete iWait; 
-	if(iPlayer != NULL)
-		iPlayer->Close();
-	delete iPlayer;
-	delete iBuffer;
-	}
-	
-void CEpocAudio::Complete(TInt aState, TInt aError)
-	{
-	if(aState == MStreamObs::EClose)
-		{
-		}
-	if(iPlayer->Closed())
-		return;
-	switch(aError)
-		{
-		case KErrUnderflow:
-		case KErrInUse:
-			iPlayer->Start();
-			break;
-		case KErrAbort:
-			iPlayer->Open();
-		}
-	}
-	
-
-void sos_adump(SDL_AudioDevice* thisdevice, void* data, int len)
-	{
-#ifdef DODUMP
-	const TPtrC8 buf((TUint8*)data, len);
-	CEpocAudio::Current(thisdevice).Dump(buf);
-#endif
-	}
-
-const TInt KClip(256);
-	
-TPtrC8 CEpocAudio::Data()
-	{
-	if(iPosition < 0)
-		return KNullDesC8();
-	
-	TPtrC8 data(iAudioPtr + iPosition, KClip);
-	
-#ifdef DODUMP
-	iDump.Dump(data);
-#endif
-	
-	iPosition += KClip;
-	if(iPosition >= iBufferSize) 
-		{
-		
-/*		if(iAudioPtr == iBuffer)
-			iAudioPtr = iBuffer + iBufferSize;
-		else
-			iAudioPtr = iBuffer;
-*/		
-		iAudioPtr += iBufferSize;
-		
-		if((iAudioPtr - iBuffer) >= KAudioBuffers * iBufferSize)
-			iAudioPtr = iBuffer;
-		
-		iPosition = -1;
-		if(iWait->IsActive())
-			{
-			iWait->Cancel();
-			CActiveScheduler::Stop();
-			}
-		}
-	return data;
-	}
-		
-
-
-	
-void CEpocAudio::Play()
-	{
-	iPosition = 0;
-	}
-
-void CEpocAudio::Wait()
-	{
-	if(iPosition >= 0 /*&& iPlayer->Playing()*/)
-		{
-		const TInt64 bufMs = TInt64(iBufferSize - KClip) * TInt64(1000000);
-		const TInt64 specTime =  bufMs / TInt64(iRate * iChannels * 2);
-		iWait->After(specTime);
-		
-		CActiveScheduler::Start();
-		TTime end;
-		end.UniversalTime();
-		const TTimeIntervalMicroSeconds delta = end.MicroSecondsFrom(iStart);
-	
-	
-//		const TTimeIntervalMicroSeconds end = iPlayer->Position();
-		
-		
-	
-		
-		const TInt diff = specTime - delta.Int64();
-		
-		if(diff > 0 && diff < 200000)
-			{
-			User::After(diff);
-			}
-		
-		}
-	else
-		{
-	User::After(10000); 
-//	iWait->Wait(10000); //just give some time...	
-		}	
-	}
-	
-void CEpocAudio::Open(TInt aRate, TInt aChannels, TUint32 aType, TInt aBytes)	
-	{
-	iRate = aRate;
-	iChannels = aChannels;
-	iType = aType;
-    iBufferRate = iRate * iChannels * aBytes; //1/x
-	}
-	
-
-/* Audio driver bootstrap functions */
-
-AudioBootStrap EPOCAudio_bootstrap = {
-	"epoc\0\0\0",
-	"EPOC streaming audio\0\0\0",
-	Audio_Available,
-	Audio_CreateDevice
-};
-
-
-static SDL_AudioDevice *Audio_CreateDevice(int /*devindex*/)
-{
-	SDL_AudioDevice *thisdevice;
-
-	/* Initialize all variables that we clean on shutdown */
-	thisdevice = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice));
-	if ( thisdevice ) {
-		memset(thisdevice, 0, (sizeof *thisdevice));
-		thisdevice->hidden = NULL; /*(struct SDL_PrivateAudioData *)
-			 malloc((sizeof thisdevice->hidden)); */
-	}
-	if ( (thisdevice == NULL) /*|| (thisdevice->hidden == NULL) */) {
-		SDL_OutOfMemory();
-		if ( thisdevice ) {
-			free(thisdevice);
-		}
-		return(0);
-	}
-//	memset(thisdevice->hidden, 0, (sizeof *thisdevice->hidden));
-
-	/* Set the function pointers */
-	thisdevice->OpenAudio = EPOC_OpenAudio;
-	thisdevice->WaitAudio = EPOC_WaitAudio;
-	thisdevice->PlayAudio = EPOC_PlayAudio;
-	thisdevice->GetAudioBuf = EPOC_GetAudioBuf;
-	thisdevice->CloseAudio = EPOC_CloseAudio;
-    thisdevice->ThreadInit = EPOC_ThreadInit;
-	thisdevice->free = Audio_DeleteDevice;
-
-	return thisdevice;
-}
-
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-    {
-	//free(device->hidden);
-	free(device);
-    }
-
-static int Audio_Available(void)
-{
-	return(1); // Audio stream modules should be always there!
-}
-
-
-static int EPOC_OpenAudio(SDL_AudioDevice *thisdevice, SDL_AudioSpec *spec)
-{
-	SDL_TRACE("SDL:EPOC_OpenAudio");
-
-	
-	TUint32 type = KMMFFourCCCodePCM16;
-	TInt bytes = 2;
-	
-	switch(spec->format)
-		{
-		case AUDIO_U16LSB: 
-			type = KMMFFourCCCodePCMU16; 
-			break;
-		case AUDIO_S16LSB: 
-			type = KMMFFourCCCodePCM16; 
-			break;
-		case AUDIO_U16MSB: 
-			type = KMMFFourCCCodePCMU16B; 
-			break;
-		case AUDIO_S16MSB: 
-			type = KMMFFourCCCodePCM16B; 
-			break; 
-			//8 bit not supported!
-		case AUDIO_U8: 
-		case AUDIO_S8:
-		default:
-			spec->format = AUDIO_S16LSB;
-		};
-	
-
-	
-	if(spec->channels > 2)
-		spec->channels = 2;
-	
-	spec->freq = CStreamPlayer::ClosestSupportedRate(spec->freq);
-	
-
-	/* Allocate mixing buffer */
-	const TInt buflen = spec->size;// * bytes * spec->channels;
-//	audiobuf = NULL;
-    
-    TRAPD(err, thisdevice->hidden = static_cast<SDL_PrivateAudioData*>(CEpocAudio::NewL(buflen, spec->silence)));
-    if(err != KErrNone)
-        return -1;
-
-	CEpocAudio::Current(thisdevice).Open(spec->freq, spec->channels, type, bytes);
-	
-	CEpocAudio::Current(thisdevice).SetPause(ETrue);
-	
-   // isSDLAudioPaused = 1;
-
-    thisdevice->enabled = 0; /* enable only after audio engine has been initialized!*/
-
-	/* We're ready to rock and roll. :-) */
-	return(0);
-}
-
-
-static void EPOC_CloseAudio(SDL_AudioDevice* thisdevice)
-    {
-#ifdef DEBUG_AUDIO
-    SDL_TRACE("Close audio\n");
-#endif
-
-	CEpocAudio::Free(thisdevice);
-	}
-
-
-static void EPOC_ThreadInit(SDL_AudioDevice *thisdevice)
-    {
-	SDL_TRACE("SDL:EPOC_ThreadInit");
-    CEpocAudio::Current(thisdevice).ThreadInitL(thisdevice);
-    RThread().SetPriority(EPriorityMore);
-    thisdevice->enabled = 1;
-    }
-
-/* This function waits until it is possible to write a full sound buffer */
-static void EPOC_WaitAudio(SDL_AudioDevice* thisdevice)
-{
-#ifdef DEBUG_AUDIO
-    SDL_TRACE1("wait %d audio\n", CEpocAudio::AudioLib().StreamPlayer(KSfxChannel).SyncTime());
-    TInt tics = User::TickCount();
-#endif
-
-	CEpocAudio::Current(thisdevice).Wait();
-
-#ifdef DEBUG_AUDIO
-    TInt ntics =  User::TickCount() - tics;
-    SDL_TRACE1("audio waited %d\n", ntics);
-    SDL_TRACE1("audio at %d\n", tics);
-#endif
-}
-
-
- 
-static void EPOC_PlayAudio(SDL_AudioDevice* thisdevice)
-	{
- 	if(CEpocAudio::Current(thisdevice).SetPause(SDL_GetAudioStatus() == SDL_AUDIO_PAUSED))
- 		SDL_Delay(500); //hold on the busy loop
- 	else
- 		CEpocAudio::Current(thisdevice).Play();
-
-#ifdef DEBUG_AUDIO
-    SDL_TRACE("buffer has audio data\n");
-#endif
-
-	
-#ifdef DEBUG_AUDIO
-	SDL_TRACE1("Wrote %d bytes of audio data\n", buflen);
-#endif
-}
-
-static Uint8 *EPOC_GetAudioBuf(SDL_AudioDevice* thisdevice)
-	{
-	return CEpocAudio::Current(thisdevice).Buffer();
-	}
-
-
-
diff --git a/apps/plugins/sdl/src/audio/symbian/SDL_epocaudio.h b/apps/plugins/sdl/src/audio/symbian/SDL_epocaudio.h
deleted file mode 100644
index 5c95c86158..0000000000
--- a/apps/plugins/sdl/src/audio/symbian/SDL_epocaudio.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Library General Public
-    License as published by the Free Software Foundation; either
-    version 2 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Library General Public License for more details.
-
-    You should have received a copy of the GNU Library General Public
-    License along with this library; if not, write to the Free
-    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
-
-    Sam Lantinga
-    slouken@devolution.com
-*/
-
-
-#ifdef SAVE_RCSID
-static char rcsid =
- "@(#) $Id: SDL_epocaudio.h,v 1.1.2.2 2001/02/10 07:20:03 hercules Exp $";
-#endif
-
-#ifndef _SDL_EPOCAUDIO_H
-#define _SDL_EPOCAUDIO_H
-
-extern "C" {
-#include "SDL_sysaudio.h"
-}
-
-
-#endif /* _SDL_EPOCAUDIO_H */
diff --git a/apps/plugins/sdl/src/audio/symbian/streamplayer.cpp b/apps/plugins/sdl/src/audio/symbian/streamplayer.cpp
deleted file mode 100644
index dd733a1d17..0000000000
--- a/apps/plugins/sdl/src/audio/symbian/streamplayer.cpp
+++ /dev/null
@@ -1,279 +0,0 @@
-#include "streamplayer.h"
-#include<mda/common/audio.h>
-
-
-
-const TInt KMaxVolume(256);
-
-LOCAL_C TInt GetSampleRate(TInt aRate)
-    {
-    switch(aRate)
-        {
-        case 8000: return TMdaAudioDataSettings::ESampleRate8000Hz;
-        case 11025: return TMdaAudioDataSettings::ESampleRate11025Hz;
-        case 12000: return TMdaAudioDataSettings::ESampleRate12000Hz;
-        case 16000: return TMdaAudioDataSettings::ESampleRate16000Hz;
-        case 22050: return TMdaAudioDataSettings::ESampleRate22050Hz;
-        case 24000: return TMdaAudioDataSettings::ESampleRate24000Hz;
-        case 32000: return TMdaAudioDataSettings::ESampleRate32000Hz;
-        case 44100: return TMdaAudioDataSettings::ESampleRate44100Hz;
-        case 48000: return TMdaAudioDataSettings::ESampleRate48000Hz;
-        case 96000: return TMdaAudioDataSettings::ESampleRate96000Hz;
-        case 64000: return TMdaAudioDataSettings::ESampleRate64000Hz;
-        }
-    return KErrNotFound;
-    }
-
-LOCAL_C TInt GetChannels(TInt aChannels)
-    {
-    switch(aChannels)
-        {
-        case 1: return TMdaAudioDataSettings::EChannelsMono;
-        case 2: return TMdaAudioDataSettings::EChannelsStereo;
-        }
-    return KErrNotFound;
-    }
-    
-TInt CStreamPlayer::ClosestSupportedRate(TInt aRate)
-	{
-	if(aRate > 96000)
-		return 96000;
-	TInt rate = aRate;
-	while(GetSampleRate(rate) == KErrNotFound)
-		{
-		++rate;
-		}
-	return rate;
-	}
-
-CStreamPlayer::CStreamPlayer(MStreamProvider& aProvider,  MStreamObs& aObs) :
- iProvider(aProvider), iObs(aObs), iVolume(KMaxVolume)
-	{	
-	}
-		
-CStreamPlayer::~CStreamPlayer()
-	{
-	iState |= EDied;
-	if(iState & EInited)
-		Close();
-	User::After(100000); //wait buffer to be flushed
-	ASSERT(iPtr.Length() == 0);
-	delete iStream;
-	}
-		
-		
-void CStreamPlayer::ConstructL()
-	{
-	iStream = CMdaAudioOutputStream::NewL(*this, EMdaPriorityMax);
-	iSilence.SetMax();
-	iSilence.FillZ();
-	}
-		
-
-TInt CStreamPlayer::OpenStream(TInt aRate, TInt aChannels, TUint32 aType)
-	{
-	Close();
-	
-	iType = aType;
-	
-	iRate = GetSampleRate(aRate);
-	if(iRate == KErrNotFound)
-		return KErrNotSupported;
-	
-	iChannels = GetChannels(aChannels);
-	if(iChannels == KErrNotFound)
-		return KErrNotSupported;
-	
-	Open();
-    
-    return KErrNone;
-	}
-
-
-TInt CStreamPlayer::MaxVolume() const
-	{
-	return KMaxVolume;
-	}
-
-void CStreamPlayer::SetVolume(TInt aNew)
-    {
-    
-    const TInt maxi = MaxVolume();
-    if(aNew > maxi)
-		return;
-    if(aNew < 0)
-    	return;
-    
-    iVolume = aNew;
-    
-    iState |= EVolumeChange;
-    }
-    
- TInt CStreamPlayer::Volume() const
-    {
-	return iVolume;
-    }
-
-void CStreamPlayer::Open()
-	{
-	TMdaAudioDataSettings audioSettings;
-	audioSettings.Query();
-    audioSettings.iCaps = TMdaAudioDataSettings::ERealTime |
-     	TMdaAudioDataSettings::ESampleRateFixed; 
-    audioSettings.iSampleRate = iRate; 
-    audioSettings.iChannels = iChannels;
-	audioSettings.iFlags = TMdaAudioDataSettings::ENoNetworkRouting;
-	audioSettings.iVolume = 0;
-	
-	iState &= ~EStopped;
-    iStream->Open(&audioSettings);    
-	}
-	
-void CStreamPlayer::Stop()
-	{
-	if(iState & (EStarted | EInited))
-		{
-		Close();
-		iState |= EStopped;
-		}
-	}
-	
-void CStreamPlayer::Start()
-	{
-	if(iPtr.Length() == 0)
-		{
-		iState |= EStarted;
-		if(iState & EInited)
-			{
-			Request();
-			}
-		else if(iState & EStopped)
-			{
-			Open();
-			}
-		}
-	}
-
-void CStreamPlayer::Close()
-	{
-	iState &= ~EInited;
-	iStream->Stop();
-	iState &= ~EStarted;
-	}
-	
-void CStreamPlayer::Request()
-	{
-	if(iState & EInited)
-		{
-		iPtr.Set(KNullDesC8);
-		
-		if(iState & EVolumeChange)
-			{
-       		const TReal newVol = iVolume;
-       		const TReal newMax = MaxVolume();
-       		const TInt maxVol = iStream->MaxVolume();
-       		const TReal max = static_cast<TReal>(maxVol);
-       		const TReal newvolume = (newVol * max) / newMax;
-       		const TInt vol = static_cast<TReal>(newvolume);
-       		iStream->SetVolume(vol);
-			iState &= ~EVolumeChange;
-			}
-			
-		if(iState & EStarted)
-			{
-	 		iPtr.Set(iProvider.Data());
-			}
-		if(iPtr.Length() == 0)
-			{
-			iPtr.Set(iSilence);
-			}
-		TRAPD(err, iStream->WriteL(iPtr));
-		if(err != KErrNone)
-			{
-			iObs.Complete(MStreamObs::EWrite, err);	
-			}
-	/*	else
-			{
-			iProvider.Written(iPtr.Length());
-			}*/
-		}
-	}
-	
-
-void CStreamPlayer::SetCapsL()
-	{
-	iStream->SetDataTypeL(iType);
-	iStream->SetAudioPropertiesL(iRate, iChannels);
-	}
-
-void CStreamPlayer::MaoscOpenComplete(TInt aError) 
-	{
-	if(aError == KErrNone)
-		{
-		TRAPD(err, SetCapsL());
-    	if(err == KErrNone)
-    		{
-    		iStream->SetPriority(EPriorityNormal, EMdaPriorityPreferenceTime);
-    		iState |= EInited;
-    		
-    		
-        	SetVolume(Volume());	
-    		
-    		if(iState & EStarted)
-    			{
-    			Request();
-    			}
-    			
-			}
-		aError = err;
-		}
-	if(!(iState & EDied))
-		iObs.Complete(MStreamObs::EInit, aError);
-	}
-	
-void CStreamPlayer::MaoscBufferCopied(TInt aError, const TDesC8& /*aBuffer*/)
-	{
-	iPtr.Set(KNullDesC8);
-	if(aError == KErrNone)
-		{
-		if(iState & EInited)
-			Request();
-		else
-			iStream->Stop();
-		}
-	else if(!(iState & EDied))
-		iObs.Complete(MStreamObs::EPlay, aError);
-	}
-	
-void CStreamPlayer::MaoscPlayComplete(TInt aError)
-	{	
-	iPtr.Set(KNullDesC8);
-	iState &= ~EStarted;
-	if(!(iState & EDied))
-		iObs.Complete(MStreamObs::EClose, aError);
-	}
-	
-TBool CStreamPlayer::Playing() const
-	{
-	return (iState & EInited) && (iState & EStarted);
-	}
-
-TBool CStreamPlayer::Closed() const
-	{
-	return !(iState & EInited) && !(iState & EDied);
-	}
-	
-	/*
-void CStreamPlayer::Request()
-	{
-	SetActive();
-	TRequestStatus* s = &iStatus;
-	User::RequestComplete(s, KErrNone);
-	}
-	//		iTimer.After(0);
-	*/
-	
-
-
-
-
diff --git a/apps/plugins/sdl/src/audio/symbian/streamplayer.h b/apps/plugins/sdl/src/audio/symbian/streamplayer.h
deleted file mode 100644
index 8c6e74f920..0000000000
--- a/apps/plugins/sdl/src/audio/symbian/streamplayer.h
+++ /dev/null
@@ -1,89 +0,0 @@
-#ifndef STREAMPLAYER_H
-#define STREAMPLAYER_H
-
-#include<MdaAudioOutputStream.h>
-
-const TInt KSilenceBuffer = 256;
-
-class MStreamObs
-    {
-    public:
-    	enum 
-    	{
-    	EInit, 
-    	EPlay,
-    	EWrite,
-    	EClose,	
-    	};
-        virtual void Complete(TInt aState, TInt aError) = 0;
-    };
-
-class MStreamProvider
-	{
-	public:
-		virtual TPtrC8 Data() = 0;	
-	};
-
-NONSHARABLE_CLASS(CStreamPlayer) : public CBase, public MMdaAudioOutputStreamCallback
-	{
-	public:
-		CStreamPlayer(MStreamProvider& aProvider, MStreamObs& aObs);
-		~CStreamPlayer();
-		void ConstructL();
-		
-		static TInt ClosestSupportedRate(TInt aRate);
-		
-		TInt OpenStream(TInt aRate, TInt aChannels, TUint32 aType = KMMFFourCCCodePCM16);
-		
-		void SetVolume(TInt aNew);
-		TInt Volume() const;
-		TInt MaxVolume() const;
-		
-		void Stop();
-		void Start();
-		void Open();
-		void Close();
-		
-		TBool Playing() const;
-		TBool Closed() const;
-		
-	private:
-
-		void MaoscOpenComplete(TInt aError) ;
-		void MaoscBufferCopied(TInt aError, const TDesC8& aBuffer);
-		void MaoscPlayComplete(TInt aError);
-	
-	private:
-		void Request();
-		void SetCapsL();
-
-	private:
-		MStreamProvider& iProvider;
-		MStreamObs& iObs;	
-		TInt iVolume;
-	
-		CMdaAudioOutputStream* iStream;
-	
-		TInt iRate;
-		TInt iChannels;
-		TUint32 iType;
-		
-		enum 
-			{
-				ENone = 0,
-				EInited = 0x1,
-				EStarted = 0x2,
-				EStopped = 0x4,
-				EVolumeChange = 0x8,
-				EDied		  = 0x10
-			};
-		
-		TInt iState;
-		TBuf8<KSilenceBuffer> iSilence;
-		TPtrC8 iPtr;
-	
-	};
-
-
-#endif
-
diff --git a/apps/plugins/sdl/src/audio/ums/SDL_umsaudio.c b/apps/plugins/sdl/src/audio/ums/SDL_umsaudio.c
deleted file mode 100644
index 9488911db6..0000000000
--- a/apps/plugins/sdl/src/audio/ums/SDL_umsaudio.c
+++ /dev/null
@@ -1,547 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Carsten Griwodz
-    griff@kom.tu-darmstadt.de
-
-    based on linux/SDL_dspaudio.c by Sam Lantinga
-*/
-#include "SDL_config.h"
-
-/* Allow access to a raw mixing buffer */
-
-#include <errno.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include <sys/types.h>
-#include <sys/time.h>
-#include <sys/ioctl.h>
-#include <sys/stat.h>
-#include <sys/mman.h>
-
-#include "SDL_audio.h"
-#include "../SDL_audio_c.h"
-#include "../SDL_audiodev_c.h"
-#include "SDL_umsaudio.h"
-
-/* The tag name used by UMS audio */
-#define UMS_DRIVER_NAME         "ums"
-
-#define DEBUG_AUDIO 1
-
-/* Audio driver functions */
-static int UMS_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void UMS_PlayAudio(_THIS);
-static Uint8 *UMS_GetAudioBuf(_THIS);
-static void UMS_CloseAudio(_THIS);
-
-static UMSAudioDevice_ReturnCode UADOpen(_THIS,  string device, string mode, long flags);
-static UMSAudioDevice_ReturnCode UADClose(_THIS);
-static UMSAudioDevice_ReturnCode UADGetBitsPerSample(_THIS, long* bits);
-static UMSAudioDevice_ReturnCode UADSetBitsPerSample(_THIS, long bits);
-static UMSAudioDevice_ReturnCode UADSetSampleRate(_THIS, long rate, long* set_rate);
-static UMSAudioDevice_ReturnCode UADSetByteOrder(_THIS, string byte_order);
-static UMSAudioDevice_ReturnCode UADSetAudioFormatType(_THIS, string fmt);
-static UMSAudioDevice_ReturnCode UADSetNumberFormat(_THIS, string fmt);
-static UMSAudioDevice_ReturnCode UADInitialize(_THIS);
-static UMSAudioDevice_ReturnCode UADStart(_THIS);
-static UMSAudioDevice_ReturnCode UADStop(_THIS);
-static UMSAudioDevice_ReturnCode UADSetTimeFormat(_THIS,  UMSAudioTypes_TimeFormat fmt );
-static UMSAudioDevice_ReturnCode UADWriteBuffSize(_THIS,  long* buff_size );
-static UMSAudioDevice_ReturnCode UADWriteBuffRemain(_THIS,  long* buff_size );
-static UMSAudioDevice_ReturnCode UADWriteBuffUsed(_THIS,  long* buff_size );
-static UMSAudioDevice_ReturnCode UADSetDMABufferSize(_THIS,  long bytes, long* bytes_ret );
-static UMSAudioDevice_ReturnCode UADSetVolume(_THIS,  long volume );
-static UMSAudioDevice_ReturnCode UADSetBalance(_THIS,  long balance );
-static UMSAudioDevice_ReturnCode UADSetChannels(_THIS,  long channels );
-static UMSAudioDevice_ReturnCode UADPlayRemainingData(_THIS,  boolean block );
-static UMSAudioDevice_ReturnCode UADEnableOutput(_THIS,  string output, long* left_gain, long* right_gain);
-static UMSAudioDevice_ReturnCode UADWrite(_THIS,  UMSAudioTypes_Buffer* buff, long samples, long* samples_written);
-
-/* Audio driver bootstrap functions */
-static int Audio_Available(void)
-{
-    return 1;
-}
-
-static void Audio_DeleteDevice(_THIS)
-{
-    if(this->hidden->playbuf._buffer) SDL_free(this->hidden->playbuf._buffer);
-    if(this->hidden->fillbuf._buffer) SDL_free(this->hidden->fillbuf._buffer);
-    _somFree( this->hidden->umsdev );
-    SDL_free(this->hidden);
-    SDL_free(this);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-    SDL_AudioDevice *this;
-
-    /*
-     * Allocate and initialize management storage and private management
-     * storage for this SDL-using library.
-     */
-    this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-    if ( this ) {
-        SDL_memset(this, 0, (sizeof *this));
-        this->hidden = (struct SDL_PrivateAudioData *)SDL_malloc((sizeof *this->hidden));
-    }
-    if ( (this == NULL) || (this->hidden == NULL) ) {
-        SDL_OutOfMemory();
-        if ( this ) {
-            SDL_free(this);
-        }
-        return(0);
-    }
-    SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-#ifdef DEBUG_AUDIO
-    fprintf(stderr, "Creating UMS Audio device\n");
-#endif
-
-    /*
-     * Calls for UMS env initialization and audio object construction.
-     */
-    this->hidden->ev     = somGetGlobalEnvironment();
-    this->hidden->umsdev = UMSAudioDeviceNew();
-
-    /*
-     * Set the function pointers.
-     */
-    this->OpenAudio   = UMS_OpenAudio;
-    this->WaitAudio   = NULL;           /* we do blocking output */
-    this->PlayAudio   = UMS_PlayAudio;
-    this->GetAudioBuf = UMS_GetAudioBuf;
-    this->CloseAudio  = UMS_CloseAudio;
-    this->free        = Audio_DeleteDevice;
-
-#ifdef DEBUG_AUDIO
-    fprintf(stderr, "done\n");
-#endif
-    return this;
-}
-
-AudioBootStrap UMS_bootstrap = {
-	UMS_DRIVER_NAME, "AIX UMS audio",
-	Audio_Available, Audio_CreateDevice
-};
-
-static Uint8 *UMS_GetAudioBuf(_THIS)
-{
-#ifdef DEBUG_AUDIO
-    fprintf(stderr, "enter UMS_GetAudioBuf\n");
-#endif
-    return this->hidden->fillbuf._buffer;
-/*
-    long                      bufSize;
-    UMSAudioDevice_ReturnCode rc;
-
-    rc = UADSetTimeFormat(this, UMSAudioTypes_Bytes );
-    rc = UADWriteBuffSize(this,  bufSize );
-*/
-}
-
-static void UMS_CloseAudio(_THIS)
-{
-    UMSAudioDevice_ReturnCode rc;
-
-#ifdef DEBUG_AUDIO
-    fprintf(stderr, "enter UMS_CloseAudio\n");
-#endif
-    rc = UADPlayRemainingData(this, TRUE);
-    rc = UADStop(this);
-    rc = UADClose(this);
-}
-
-static void UMS_PlayAudio(_THIS)
-{
-    UMSAudioDevice_ReturnCode rc;
-    long                      samplesToWrite;
-    long                      samplesWritten;
-    UMSAudioTypes_Buffer      swpbuf;
-
-#ifdef DEBUG_AUDIO
-    fprintf(stderr, "enter UMS_PlayAudio\n");
-#endif
-    samplesToWrite = this->hidden->playbuf._length/this->hidden->bytesPerSample;
-    do
-    {
-        rc = UADWrite(this,  &this->hidden->playbuf,
-		       samplesToWrite,
-	               &samplesWritten );
-	samplesToWrite -= samplesWritten;
-
-	/* rc values: UMSAudioDevice_Success
-	 *            UMSAudioDevice_Failure
-	 *            UMSAudioDevice_Preempted
-	 *            UMSAudioDevice_Interrupted
-	 *            UMSAudioDevice_DeviceError
-	 */
-	if ( rc == UMSAudioDevice_DeviceError ) {
-#ifdef DEBUG_AUDIO
-	    fprintf(stderr, "Returning from PlayAudio with devices error\n");
-#endif
-	    return;
-	}
-    }
-    while(samplesToWrite>0);
-
-    SDL_LockAudio();
-    SDL_memcpy( &swpbuf,                &this->hidden->playbuf, sizeof(UMSAudioTypes_Buffer) );
-    SDL_memcpy( &this->hidden->playbuf, &this->hidden->fillbuf, sizeof(UMSAudioTypes_Buffer) );
-    SDL_memcpy( &this->hidden->fillbuf, &swpbuf,                sizeof(UMSAudioTypes_Buffer) );
-    SDL_UnlockAudio();
-
-#ifdef DEBUG_AUDIO
-    fprintf(stderr, "Wrote audio data and swapped buffer\n");
-#endif
-}
-
-#if 0
-// 	/* Set the DSP frequency */
-// 	value = spec->freq;
-// 	if ( ioctl(this->hidden->audio_fd, SOUND_PCM_WRITE_RATE, &value) < 0 ) {
-// 		SDL_SetError("Couldn't set audio frequency");
-// 		return(-1);
-// 	}
-// 	spec->freq = value;
-#endif
-
-static int UMS_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-    char*  audiodev = "/dev/paud0";
-    long   lgain;
-    long   rgain;
-    long   outRate;
-    long   outBufSize;
-    long   bitsPerSample;
-    long   samplesPerSec;
-    long   success;
-    Uint16 test_format;
-    int    frag_spec;
-    UMSAudioDevice_ReturnCode rc;
-
-#ifdef DEBUG_AUDIO
-    fprintf(stderr, "enter UMS_OpenAudio\n");
-#endif
-    rc = UADOpen(this, audiodev,"PLAY", UMSAudioDevice_BlockingIO);
-    if ( rc != UMSAudioDevice_Success ) {
-	SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
-	return -1;
-    }
- 
-    rc = UADSetAudioFormatType(this, "PCM"); 
-
-    success = 0;
-    test_format = SDL_FirstAudioFormat(spec->format);
-    do
-    {
-#ifdef DEBUG_AUDIO
-        fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
-#endif
-        switch ( test_format )
-        {
-        case AUDIO_U8:
-/* from the mac code: better ? */
-/* sample_bits = spec->size / spec->samples / spec->channels * 8; */
-	    success       = 1;
-            bitsPerSample = 8;
-            rc = UADSetSampleRate(this,  spec->freq << 16, &outRate );
-            rc = UADSetByteOrder(this, "MSB");       /* irrelevant */
-            rc = UADSetNumberFormat(this, "UNSIGNED");
-            break;
-        case AUDIO_S8:
-	    success       = 1;
-            bitsPerSample = 8;
-            rc = UADSetSampleRate(this,  spec->freq << 16, &outRate );
-            rc = UADSetByteOrder(this, "MSB");       /* irrelevant */
-            rc = UADSetNumberFormat(this, "SIGNED");
-            break;
-        case AUDIO_S16LSB:
-	    success       = 1;
-            bitsPerSample = 16;
-            rc = UADSetSampleRate(this,  spec->freq << 16, &outRate );
-            rc = UADSetByteOrder(this, "LSB");
-            rc = UADSetNumberFormat(this, "SIGNED");
-            break;
-        case AUDIO_S16MSB:
-	    success       = 1;
-            bitsPerSample = 16;
-            rc = UADSetSampleRate(this,  spec->freq << 16, &outRate );
-            rc = UADSetByteOrder(this, "MSB");
-            rc = UADSetNumberFormat(this, "SIGNED");
-            break;
-        case AUDIO_U16LSB:
-	    success       = 1;
-            bitsPerSample = 16;
-            rc = UADSetSampleRate(this,  spec->freq << 16, &outRate );
-            rc = UADSetByteOrder(this, "LSB");
-            rc = UADSetNumberFormat(this, "UNSIGNED");
-            break;
-        case AUDIO_U16MSB:
-	    success       = 1;
-            bitsPerSample = 16;
-            rc = UADSetSampleRate(this,  spec->freq << 16, &outRate );
-            rc = UADSetByteOrder(this, "MSB");
-            rc = UADSetNumberFormat(this, "UNSIGNED");
-            break;
-        default:
-            break;
-        }
-        if ( ! success ) {
-            test_format = SDL_NextAudioFormat();
-        }
-    }
-    while ( ! success && test_format );
-
-    if ( success == 0 ) {
-        SDL_SetError("Couldn't find any hardware audio formats");
-        return -1;
-    }
-
-    spec->format = test_format;
-
-    for ( frag_spec = 0; (0x01<<frag_spec) < spec->size; ++frag_spec );
-    if ( (0x01<<frag_spec) != spec->size ) {
-        SDL_SetError("Fragment size must be a power of two");
-        return -1;
-    }
-    if ( frag_spec > 2048 ) frag_spec = 2048;
-
-    this->hidden->bytesPerSample   = (bitsPerSample / 8) * spec->channels;
-    samplesPerSec                  = this->hidden->bytesPerSample * outRate;
-
-    this->hidden->playbuf._length  = 0;
-    this->hidden->playbuf._maximum = spec->size;
-    this->hidden->playbuf._buffer  = (unsigned char*)SDL_malloc(spec->size);
-    this->hidden->fillbuf._length  = 0;
-    this->hidden->fillbuf._maximum = spec->size;
-    this->hidden->fillbuf._buffer  = (unsigned char*)SDL_malloc(spec->size);
-
-    rc = UADSetBitsPerSample(this,  bitsPerSample );
-    rc = UADSetDMABufferSize(this,  frag_spec, &outBufSize );
-    rc = UADSetChannels(this, spec->channels);      /* functions reduces to mono or stereo */
-
-    lgain = 100; /*maximum left input gain*/
-    rgain = 100; /*maimum right input gain*/
-    rc = UADEnableOutput(this, "LINE_OUT",&lgain,&rgain);
-    rc = UADInitialize(this);
-    rc = UADStart(this);
-    rc = UADSetVolume(this, 100);
-    rc = UADSetBalance(this, 0);
-
-    /* We're ready to rock and roll. :-) */
-    return 0;
-}
-
-
-static UMSAudioDevice_ReturnCode UADGetBitsPerSample(_THIS, long* bits)
-{
-    return UMSAudioDevice_get_bits_per_sample( this->hidden->umsdev,
-					       this->hidden->ev,
-					       bits );
-}
-
-static UMSAudioDevice_ReturnCode UADSetBitsPerSample(_THIS, long bits)
-{
-    return UMSAudioDevice_set_bits_per_sample( this->hidden->umsdev,
-					       this->hidden->ev,
-					       bits );
-}
-
-static UMSAudioDevice_ReturnCode UADSetSampleRate(_THIS, long rate, long* set_rate)
-{
-    /* from the mac code: sample rate = spec->freq << 16; */
-    return UMSAudioDevice_set_sample_rate( this->hidden->umsdev,
-					   this->hidden->ev,
-					   rate,
-					   set_rate );
-}
-
-static UMSAudioDevice_ReturnCode UADSetByteOrder(_THIS, string byte_order)
-{
-    return UMSAudioDevice_set_byte_order( this->hidden->umsdev,
-					  this->hidden->ev,
-					  byte_order );
-}
-
-static UMSAudioDevice_ReturnCode UADSetAudioFormatType(_THIS, string fmt)
-{
-    /* possible PCM, A_LAW or MU_LAW */
-    return UMSAudioDevice_set_audio_format_type( this->hidden->umsdev,
-						 this->hidden->ev,
-						 fmt );
-}
-
-static UMSAudioDevice_ReturnCode UADSetNumberFormat(_THIS, string fmt)
-{
-    /* possible SIGNED, UNSIGNED, or TWOS_COMPLEMENT */
-    return UMSAudioDevice_set_number_format( this->hidden->umsdev,
-					     this->hidden->ev,
-					     fmt );
-}
-
-static UMSAudioDevice_ReturnCode UADInitialize(_THIS)
-{
-    return UMSAudioDevice_initialize( this->hidden->umsdev,
-				      this->hidden->ev );
-}
-
-static UMSAudioDevice_ReturnCode UADStart(_THIS)
-{
-    return UMSAudioDevice_start( this->hidden->umsdev,
-				 this->hidden->ev );
-}
-
-static UMSAudioDevice_ReturnCode UADSetTimeFormat(_THIS,  UMSAudioTypes_TimeFormat fmt )
-{
-    /*
-     * Switches the time format to the new format, immediately.
-     * possible UMSAudioTypes_Msecs, UMSAudioTypes_Bytes or UMSAudioTypes_Samples
-     */
-    return UMSAudioDevice_set_time_format( this->hidden->umsdev,
-					   this->hidden->ev,
-					   fmt );
-}
-
-static UMSAudioDevice_ReturnCode UADWriteBuffSize(_THIS,  long* buff_size )
-{
-    /*
-     * returns write buffer size in the current time format
-     */
-    return UMSAudioDevice_write_buff_size( this->hidden->umsdev,
-                                           this->hidden->ev,
-					   buff_size );
-}
-
-static UMSAudioDevice_ReturnCode UADWriteBuffRemain(_THIS,  long* buff_size )
-{
-    /*
-     * returns amount of available space in the write buffer
-     * in the current time format
-     */
-    return UMSAudioDevice_write_buff_remain( this->hidden->umsdev,
-                                             this->hidden->ev,
-					     buff_size );
-}
-
-static UMSAudioDevice_ReturnCode UADWriteBuffUsed(_THIS,  long* buff_size )
-{
-    /*
-     * returns amount of filled space in the write buffer
-     * in the current time format
-     */
-    return UMSAudioDevice_write_buff_used( this->hidden->umsdev,
-                                           this->hidden->ev,
-					   buff_size );
-}
-
-static UMSAudioDevice_ReturnCode UADSetDMABufferSize(_THIS,  long bytes, long* bytes_ret )
-{
-    /*
-     * Request a new DMA buffer size, maximum requested size 2048.
-     * Takes effect with next initialize() call.
-     * Devices may or may not support DMA.
-     */
-    return UMSAudioDevice_set_DMA_buffer_size( this->hidden->umsdev,
-					       this->hidden->ev,
-					       bytes,
-					       bytes_ret );
-}
-
-static UMSAudioDevice_ReturnCode UADSetVolume(_THIS,  long volume )
-{
-    /*
-     * Set the volume.
-     * Takes effect immediately.
-     */
-    return UMSAudioDevice_set_volume( this->hidden->umsdev,
-				      this->hidden->ev,
-				      volume );
-}
-
-static UMSAudioDevice_ReturnCode UADSetBalance(_THIS,  long balance )
-{
-    /*
-     * Set the balance.
-     * Takes effect immediately.
-     */
-    return UMSAudioDevice_set_balance( this->hidden->umsdev,
-				       this->hidden->ev,
-				       balance );
-}
-
-static UMSAudioDevice_ReturnCode UADSetChannels(_THIS,  long channels )
-{
-    /*
-     * Set mono or stereo.
-     * Takes effect with next initialize() call.
-     */
-    if ( channels != 1 ) channels = 2;
-    return UMSAudioDevice_set_number_of_channels( this->hidden->umsdev,
-				                  this->hidden->ev,
-				                  channels );
-}
-
-static UMSAudioDevice_ReturnCode UADOpen(_THIS,  string device, string mode, long flags)
-{
-    return UMSAudioDevice_open( this->hidden->umsdev,
-				this->hidden->ev,
-				device,
-				mode,
-				flags );
-}
-
-static UMSAudioDevice_ReturnCode UADWrite(_THIS,  UMSAudioTypes_Buffer* buff,
-                                           long samples,
-					   long* samples_written)
-{
-    return UMSAudioDevice_write( this->hidden->umsdev,
-				 this->hidden->ev,
-				 buff,
-				 samples,
-				 samples_written );
-}
-
-static UMSAudioDevice_ReturnCode UADPlayRemainingData(_THIS,  boolean block )
-{
-    return UMSAudioDevice_play_remaining_data( this->hidden->umsdev,
-					       this->hidden->ev,
-					       block);
-}
-
-static UMSAudioDevice_ReturnCode UADStop(_THIS)
-{
-    return UMSAudioDevice_stop( this->hidden->umsdev,
-				this->hidden->ev );
-}
-
-static UMSAudioDevice_ReturnCode UADClose(_THIS)
-{
-    return UMSAudioDevice_close( this->hidden->umsdev,
-				 this->hidden->ev );
-}
-
-static UMSAudioDevice_ReturnCode UADEnableOutput(_THIS,  string output, long* left_gain, long* right_gain)
-{
-    return UMSAudioDevice_enable_output( this->hidden->umsdev,
-					 this->hidden->ev,
-					 output,
-					 left_gain,
-					 right_gain );
-}
-
diff --git a/apps/plugins/sdl/src/audio/ums/SDL_umsaudio.h b/apps/plugins/sdl/src/audio/ums/SDL_umsaudio.h
deleted file mode 100644
index 367fe853b6..0000000000
--- a/apps/plugins/sdl/src/audio/ums/SDL_umsaudio.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Carsten Griwodz
-    griff@kom.tu-darmstadt.de
-
-    based on linux/SDL_dspaudio.h by Sam Lantinga
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_UMSaudio_h
-#define _SDL_UMSaudio_h
-
-#include <UMS/UMSAudioDevice.h>
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-struct SDL_PrivateAudioData
-{
-    /* Pointer to the (open) UMS audio device */
-    Environment*   ev;
-    UMSAudioDevice umsdev;
-
-    /* Raw mixing buffer */
-    UMSAudioTypes_Buffer playbuf;
-    UMSAudioTypes_Buffer fillbuf;
-
-    long bytesPerSample;
-};
-
-#endif /* _SDL_UMSaudio_h */
-
diff --git a/apps/plugins/sdl/src/audio/windib/SDL_dibaudio.c b/apps/plugins/sdl/src/audio/windib/SDL_dibaudio.c
deleted file mode 100644
index 51a9a4d60a..0000000000
--- a/apps/plugins/sdl/src/audio/windib/SDL_dibaudio.c
+++ /dev/null
@@ -1,322 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/* Allow access to a raw mixing buffer */
-
-#define WIN32_LEAN_AND_MEAN
-#include <windows.h>
-#include <mmsystem.h>
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audio_c.h"
-#include "SDL_dibaudio.h"
-#if defined(_WIN32_WCE) && (_WIN32_WCE < 300)
-#include "win_ce_semaphore.h"
-#endif
-
-
-/* Audio driver functions */
-static int DIB_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void DIB_ThreadInit(_THIS);
-static void DIB_WaitAudio(_THIS);
-static Uint8 *DIB_GetAudioBuf(_THIS);
-static void DIB_PlayAudio(_THIS);
-static void DIB_WaitDone(_THIS);
-static void DIB_CloseAudio(_THIS);
-
-/* Audio driver bootstrap functions */
-
-static int Audio_Available(void)
-{
-	return(1);
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-	SDL_free(device->hidden);
-	SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-	if ( this ) {
-		SDL_memset(this, 0, (sizeof *this));
-		this->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *this->hidden));
-	}
-	if ( (this == NULL) || (this->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( this ) {
-			SDL_free(this);
-		}
-		return(0);
-	}
-	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-	/* Set the function pointers */
-	this->OpenAudio = DIB_OpenAudio;
-	this->ThreadInit = DIB_ThreadInit;
-	this->WaitAudio = DIB_WaitAudio;
-	this->PlayAudio = DIB_PlayAudio;
-	this->GetAudioBuf = DIB_GetAudioBuf;
-	this->WaitDone = DIB_WaitDone;
-	this->CloseAudio = DIB_CloseAudio;
-
-	this->free = Audio_DeleteDevice;
-
-	return this;
-}
-
-AudioBootStrap WAVEOUT_bootstrap = {
-	"waveout", "Win95/98/NT/2000 WaveOut",
-	Audio_Available, Audio_CreateDevice
-};
-
-
-/* The Win32 callback for filling the WAVE device */
-static void CALLBACK FillSound(HWAVEOUT hwo, UINT uMsg, DWORD_PTR dwInstance,
-						DWORD dwParam1, DWORD dwParam2)
-{
-	SDL_AudioDevice *this = (SDL_AudioDevice *)dwInstance;
-
-	/* Only service "buffer done playing" messages */
-	if ( uMsg != WOM_DONE )
-		return;
-
-	/* Signal that we are done playing a buffer */
-#if defined(_WIN32_WCE) && (_WIN32_WCE < 300)
-	ReleaseSemaphoreCE(audio_sem, 1, NULL);
-#else
-	ReleaseSemaphore(audio_sem, 1, NULL);
-#endif
-}
-
-static void SetMMerror(char *function, MMRESULT code)
-{
-	size_t len;
-	char errbuf[MAXERRORLENGTH];
-#ifdef _WIN32_WCE
-	wchar_t werrbuf[MAXERRORLENGTH];
-#endif
-
-	SDL_snprintf(errbuf, SDL_arraysize(errbuf), "%s: ", function);
-	len = SDL_strlen(errbuf);
-
-#ifdef _WIN32_WCE
-	/* UNICODE version */
-	waveOutGetErrorText(code, werrbuf, MAXERRORLENGTH-len);
-	WideCharToMultiByte(CP_ACP,0,werrbuf,-1,errbuf+len,MAXERRORLENGTH-len,NULL,NULL);
-#else
-	waveOutGetErrorText(code, errbuf+len, (UINT)(MAXERRORLENGTH-len));
-#endif
-
-	SDL_SetError("%s",errbuf);
-}
-
-/* Set high priority for the audio thread */
-static void DIB_ThreadInit(_THIS)
-{
-	SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_HIGHEST);
-}
-
-void DIB_WaitAudio(_THIS)
-{
-	/* Wait for an audio chunk to finish */
-#if defined(_WIN32_WCE) && (_WIN32_WCE < 300)
-	WaitForSemaphoreCE(audio_sem, INFINITE);
-#else
-	WaitForSingleObject(audio_sem, INFINITE);
-#endif
-}
-
-Uint8 *DIB_GetAudioBuf(_THIS)
-{
-        Uint8 *retval;
-
-	retval = (Uint8 *)(wavebuf[next_buffer].lpData);
-	return retval;
-}
-
-void DIB_PlayAudio(_THIS)
-{
-	/* Queue it up */
-	waveOutWrite(sound, &wavebuf[next_buffer], sizeof(wavebuf[0]));
-	next_buffer = (next_buffer+1)%NUM_BUFFERS;
-}
-
-void DIB_WaitDone(_THIS)
-{
-	int i, left;
-
-	do {
-		left = NUM_BUFFERS;
-		for ( i=0; i<NUM_BUFFERS; ++i ) {
-			if ( wavebuf[i].dwFlags & WHDR_DONE ) {
-				--left;
-			}
-		}
-		if ( left > 0 ) {
-			SDL_Delay(100);
-		}
-	} while ( left > 0 );
-}
-
-void DIB_CloseAudio(_THIS)
-{
-	int i;
-
-	/* Close up audio */
-	if ( audio_sem ) {
-#if defined(_WIN32_WCE) && (_WIN32_WCE < 300)
-		CloseSynchHandle(audio_sem);
-#else
-		CloseHandle(audio_sem);
-#endif
-	}
-	if ( sound ) {
-		waveOutClose(sound);
-	}
-
-	/* Clean up mixing buffers */
-	for ( i=0; i<NUM_BUFFERS; ++i ) {
-		if ( wavebuf[i].dwUser != 0xFFFF ) {
-			waveOutUnprepareHeader(sound, &wavebuf[i],
-						sizeof(wavebuf[i]));
-			wavebuf[i].dwUser = 0xFFFF;
-		}
-	}
-	/* Free raw mixing buffer */
-	if ( mixbuf != NULL ) {
-		SDL_free(mixbuf);
-		mixbuf = NULL;
-	}
-}
-
-int DIB_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	MMRESULT result;
-	int i;
-	WAVEFORMATEX waveformat;
-
-	/* Initialize the wavebuf structures for closing */
-	sound = NULL;
-	audio_sem = NULL;
-	for ( i = 0; i < NUM_BUFFERS; ++i )
-		wavebuf[i].dwUser = 0xFFFF;
-	mixbuf = NULL;
-
-	/* Set basic WAVE format parameters */
-	SDL_memset(&waveformat, 0, sizeof(waveformat));
-	waveformat.wFormatTag = WAVE_FORMAT_PCM;
-
-	/* Determine the audio parameters from the AudioSpec */
-	switch ( spec->format & 0xFF ) {
-		case 8:
-			/* Unsigned 8 bit audio data */
-			spec->format = AUDIO_U8;
-			waveformat.wBitsPerSample = 8;
-			break;
-		case 16:
-			/* Signed 16 bit audio data */
-			spec->format = AUDIO_S16;
-			waveformat.wBitsPerSample = 16;
-			break;
-		default:
-			SDL_SetError("Unsupported audio format");
-			return(-1);
-	}
-	waveformat.nChannels = spec->channels;
-	waveformat.nSamplesPerSec = spec->freq;
-	waveformat.nBlockAlign =
-		waveformat.nChannels * (waveformat.wBitsPerSample/8);
-	waveformat.nAvgBytesPerSec = 
-		waveformat.nSamplesPerSec * waveformat.nBlockAlign;
-
-	/* Check the buffer size -- minimum of 1/4 second (word aligned) */
-	if ( spec->samples < (spec->freq/4) )
-		spec->samples = ((spec->freq/4)+3)&~3;
-
-	/* Update the fragment size as size in bytes */
-	SDL_CalculateAudioSpec(spec);
-
-	/* Open the audio device */
-	result = waveOutOpen(&sound, WAVE_MAPPER, &waveformat,
-			(DWORD_PTR)FillSound, (DWORD_PTR)this, CALLBACK_FUNCTION);
-	if ( result != MMSYSERR_NOERROR ) {
-		SetMMerror("waveOutOpen()", result);
-		return(-1);
-	}
-
-#ifdef SOUND_DEBUG
-	/* Check the sound device we retrieved */
-	{
-		WAVEOUTCAPS caps;
-
-		result = waveOutGetDevCaps((UINT)sound, &caps, sizeof(caps));
-		if ( result != MMSYSERR_NOERROR ) {
-			SetMMerror("waveOutGetDevCaps()", result);
-			return(-1);
-		}
-		printf("Audio device: %s\n", caps.szPname);
-	}
-#endif
-
-	/* Create the audio buffer semaphore */
-#if defined(_WIN32_WCE) && (_WIN32_WCE < 300)
-	audio_sem = CreateSemaphoreCE(NULL, NUM_BUFFERS-1, NUM_BUFFERS, NULL);
-#else
-	audio_sem = CreateSemaphore(NULL, NUM_BUFFERS-1, NUM_BUFFERS, NULL);
-#endif
-	if ( audio_sem == NULL ) {
-		SDL_SetError("Couldn't create semaphore");
-		return(-1);
-	}
-
-	/* Create the sound buffers */
-	mixbuf = (Uint8 *)SDL_malloc(NUM_BUFFERS*spec->size);
-	if ( mixbuf == NULL ) {
-		SDL_SetError("Out of memory");
-		return(-1);
-	}
-	for ( i = 0; i < NUM_BUFFERS; ++i ) {
-		SDL_memset(&wavebuf[i], 0, sizeof(wavebuf[i]));
-		wavebuf[i].lpData = (LPSTR) &mixbuf[i*spec->size];
-		wavebuf[i].dwBufferLength = spec->size;
-		wavebuf[i].dwFlags = WHDR_DONE;
-		result = waveOutPrepareHeader(sound, &wavebuf[i],
-							sizeof(wavebuf[i]));
-		if ( result != MMSYSERR_NOERROR ) {
-			SetMMerror("waveOutPrepareHeader()", result);
-			return(-1);
-		}
-	}
-
-	/* Ready to go! */
-	next_buffer = 0;
-	return(0);
-}
diff --git a/apps/plugins/sdl/src/audio/windib/SDL_dibaudio.h b/apps/plugins/sdl/src/audio/windib/SDL_dibaudio.h
deleted file mode 100644
index d2c62280b5..0000000000
--- a/apps/plugins/sdl/src/audio/windib/SDL_dibaudio.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_lowaudio_h
-#define _SDL_lowaudio_h
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-#define NUM_BUFFERS 2			/* -- Don't lower this! */
-
-struct SDL_PrivateAudioData {
-	HWAVEOUT sound;
-	HANDLE audio_sem;
-	Uint8 *mixbuf;		/* The raw allocated mixing buffer */
-	WAVEHDR wavebuf[NUM_BUFFERS];	/* Wave audio fragments */
-	int next_buffer;
-};
-
-/* Old variable names */
-#define sound			(this->hidden->sound)
-#define audio_sem 		(this->hidden->audio_sem)
-#define mixbuf			(this->hidden->mixbuf)
-#define wavebuf			(this->hidden->wavebuf)
-#define next_buffer		(this->hidden->next_buffer)
-
-#endif /* _SDL_lowaudio_h */
diff --git a/apps/plugins/sdl/src/audio/windx5/SDL_dx5audio.c b/apps/plugins/sdl/src/audio/windx5/SDL_dx5audio.c
deleted file mode 100644
index c3d42aeda1..0000000000
--- a/apps/plugins/sdl/src/audio/windx5/SDL_dx5audio.c
+++ /dev/null
@@ -1,705 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-/* Allow access to a raw mixing buffer */
-
-#include "SDL_timer.h"
-#include "SDL_audio.h"
-#include "../SDL_audio_c.h"
-#include "SDL_dx5audio.h"
-
-/* Define this if you want to use DirectX 6 DirectSoundNotify interface */
-//#define USE_POSITION_NOTIFY
-
-/* DirectX function pointers for audio */
-HRESULT (WINAPI *DSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
-
-/* Audio driver functions */
-static int DX5_OpenAudio(_THIS, SDL_AudioSpec *spec);
-static void DX5_ThreadInit(_THIS);
-static void DX5_WaitAudio_BusyWait(_THIS);
-#ifdef USE_POSITION_NOTIFY
-static void DX6_WaitAudio_EventWait(_THIS);
-#endif
-static void DX5_PlayAudio(_THIS);
-static Uint8 *DX5_GetAudioBuf(_THIS);
-static void DX5_WaitDone(_THIS);
-static void DX5_CloseAudio(_THIS);
-
-/* Audio driver bootstrap functions */
-
-static int Audio_Available(void)
-{
-	HINSTANCE DSoundDLL;
-	int dsound_ok;
-
-	/* Version check DSOUND.DLL (Is DirectX okay?) */
-	dsound_ok = 0;
-	DSoundDLL = LoadLibrary(TEXT("DSOUND.DLL"));
-	if ( DSoundDLL != NULL ) {
-		/* We just use basic DirectSound, we're okay */
-		/* Yay! */
-		/* Unfortunately, the sound drivers on NT have
-		   higher latencies than the audio buffers used
-		   by many SDL applications, so there are gaps
-		   in the audio - it sounds terrible.  Punt for now.
-		 */
-		OSVERSIONINFO ver;
-		ver.dwOSVersionInfoSize = sizeof (OSVERSIONINFO);
-		GetVersionEx(&ver);
-		switch (ver.dwPlatformId) {
-			case VER_PLATFORM_WIN32_NT:
-				if ( ver.dwMajorVersion > 4 ) {
-					/* Win2K */
-					dsound_ok = 1;
-				} else {
-					/* WinNT */
-					dsound_ok = 0;
-				}
-				break;
-			default:
-				/* Win95 or Win98 */
-				dsound_ok = 1;
-				break;
-		}
-		/* Now check for DirectX 5 or better - otherwise
-		 * we will fail later in DX5_OpenAudio without a chance
-		 * to fall back to the DIB driver. */
-		if (dsound_ok) {
-			/* DirectSoundCaptureCreate was added in DX5 */
-			if (!GetProcAddress(DSoundDLL, TEXT("DirectSoundCaptureCreate")))
-				dsound_ok = 0;
-
-		}
-		/* Clean up.. */
-		FreeLibrary(DSoundDLL);
-	}
-	return(dsound_ok);
-}
-
-/* Functions for loading the DirectX functions dynamically */
-static HINSTANCE DSoundDLL = NULL;
-
-static void DX5_Unload(void)
-{
-	if ( DSoundDLL != NULL ) {
-		FreeLibrary(DSoundDLL);
-		DSoundCreate = NULL;
-		DSoundDLL = NULL;
-	}
-}
-static int DX5_Load(void)
-{
-	int status;
-
-	DX5_Unload();
-	DSoundDLL = LoadLibrary(TEXT("DSOUND.DLL"));
-	if ( DSoundDLL != NULL ) {
-		DSoundCreate = (void *)GetProcAddress(DSoundDLL,
-					TEXT("DirectSoundCreate"));
-	}
-	if ( DSoundDLL && DSoundCreate ) {
-		status = 0;
-	} else {
-		DX5_Unload();
-		status = -1;
-	}
-	return status;
-}
-
-static void Audio_DeleteDevice(SDL_AudioDevice *device)
-{
-	DX5_Unload();
-	SDL_free(device->hidden);
-	SDL_free(device);
-}
-
-static SDL_AudioDevice *Audio_CreateDevice(int devindex)
-{
-	SDL_AudioDevice *this;
-
-	/* Load DirectX */
-	if ( DX5_Load() < 0 ) {
-		return(NULL);
-	}
-
-	/* Initialize all variables that we clean on shutdown */
-	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));
-	if ( this ) {
-		SDL_memset(this, 0, (sizeof *this));
-		this->hidden = (struct SDL_PrivateAudioData *)
-				SDL_malloc((sizeof *this->hidden));
-	}
-	if ( (this == NULL) || (this->hidden == NULL) ) {
-		SDL_OutOfMemory();
-		if ( this ) {
-			SDL_free(this);
-		}
-		return(0);
-	}
-	SDL_memset(this->hidden, 0, (sizeof *this->hidden));
-
-	/* Set the function pointers */
-	this->OpenAudio = DX5_OpenAudio;
-	this->ThreadInit = DX5_ThreadInit;
-	this->WaitAudio = DX5_WaitAudio_BusyWait;
-	this->PlayAudio = DX5_PlayAudio;
-	this->GetAudioBuf = DX5_GetAudioBuf;
-	this->WaitDone = DX5_WaitDone;
-	this->CloseAudio = DX5_CloseAudio;
-
-	this->free = Audio_DeleteDevice;
-
-	return this;
-}
-
-AudioBootStrap DSOUND_bootstrap = {
-	"dsound", "Win95/98/2000 DirectSound",
-	Audio_Available, Audio_CreateDevice
-};
-
-static void SetDSerror(const char *function, int code)
-{
-	static const char *error;
-	static char  errbuf[1024];
-
-	errbuf[0] = 0;
-	switch (code) {
-		case E_NOINTERFACE:
-			error = 
-		"Unsupported interface\n-- Is DirectX 5.0 or later installed?";
-			break;
-		case DSERR_ALLOCATED:
-			error = "Audio device in use";
-			break;
-		case DSERR_BADFORMAT:
-			error = "Unsupported audio format";
-			break;
-		case DSERR_BUFFERLOST:
-			error = "Mixing buffer was lost";
-			break;
-		case DSERR_CONTROLUNAVAIL:
-			error = "Control requested is not available";
-			break;
-		case DSERR_INVALIDCALL:
-			error = "Invalid call for the current state";
-			break;
-		case DSERR_INVALIDPARAM:
-			error = "Invalid parameter";
-			break;
-		case DSERR_NODRIVER:
-			error = "No audio device found";
-			break;
-		case DSERR_OUTOFMEMORY:
-			error = "Out of memory";
-			break;
-		case DSERR_PRIOLEVELNEEDED:
-			error = "Caller doesn't have priority";
-			break;
-		case DSERR_UNSUPPORTED:
-			error = "Function not supported";
-			break;
-		default:
-			SDL_snprintf(errbuf, SDL_arraysize(errbuf),
-			         "%s: Unknown DirectSound error: 0x%x",
-								function, code);
-			break;
-	}
-	if ( ! errbuf[0] ) {
-		SDL_snprintf(errbuf, SDL_arraysize(errbuf), "%s: %s", function, error);
-	}
-	SDL_SetError("%s", errbuf);
-	return;
-}
-
-/* DirectSound needs to be associated with a window */
-static HWND mainwin = NULL;
-/* */
-void DX5_SoundFocus(HWND hwnd)
-{
-	mainwin = hwnd;
-}
-
-static void DX5_ThreadInit(_THIS)
-{
-	SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_HIGHEST);
-}
-
-static void DX5_WaitAudio_BusyWait(_THIS)
-{
-	DWORD status;
-	DWORD cursor, junk;
-	HRESULT result;
-
-	/* Semi-busy wait, since we have no way of getting play notification
-	   on a primary mixing buffer located in hardware (DirectX 5.0)
-	*/
-	result = IDirectSoundBuffer_GetCurrentPosition(mixbuf, &junk, &cursor);
-	if ( result != DS_OK ) {
-		if ( result == DSERR_BUFFERLOST ) {
-			IDirectSoundBuffer_Restore(mixbuf);
-		}
-#ifdef DEBUG_SOUND
-		SetDSerror("DirectSound GetCurrentPosition", result);
-#endif
-		return;
-	}
-
-	while ( (cursor/mixlen) == lastchunk ) {
-		/* FIXME: find out how much time is left and sleep that long */
-		SDL_Delay(1);
-
-		/* Try to restore a lost sound buffer */
-		IDirectSoundBuffer_GetStatus(mixbuf, &status);
-		if ( (status&DSBSTATUS_BUFFERLOST) ) {
-			IDirectSoundBuffer_Restore(mixbuf);
-			IDirectSoundBuffer_GetStatus(mixbuf, &status);
-			if ( (status&DSBSTATUS_BUFFERLOST) ) {
-				break;
-			}
-		}
-		if ( ! (status&DSBSTATUS_PLAYING) ) {
-			result = IDirectSoundBuffer_Play(mixbuf, 0, 0, DSBPLAY_LOOPING);
-			if ( result == DS_OK ) {
-				continue;
-			}
-#ifdef DEBUG_SOUND
-			SetDSerror("DirectSound Play", result);
-#endif
-			return;
-		}
-
-		/* Find out where we are playing */
-		result = IDirectSoundBuffer_GetCurrentPosition(mixbuf,
-								&junk, &cursor);
-		if ( result != DS_OK ) {
-			SetDSerror("DirectSound GetCurrentPosition", result);
-			return;
-		}
-	}
-}
-
-#ifdef USE_POSITION_NOTIFY
-static void DX6_WaitAudio_EventWait(_THIS)
-{
-	DWORD status;
-	HRESULT result;
-
-	/* Try to restore a lost sound buffer */
-	IDirectSoundBuffer_GetStatus(mixbuf, &status);
-	if ( (status&DSBSTATUS_BUFFERLOST) ) {
-		IDirectSoundBuffer_Restore(mixbuf);
-		IDirectSoundBuffer_GetStatus(mixbuf, &status);
-		if ( (status&DSBSTATUS_BUFFERLOST) ) {
-			return;
-		}
-	}
-	if ( ! (status&DSBSTATUS_PLAYING) ) {
-		result = IDirectSoundBuffer_Play(mixbuf, 0, 0, DSBPLAY_LOOPING);
-		if ( result != DS_OK ) {
-#ifdef DEBUG_SOUND
-			SetDSerror("DirectSound Play", result);
-#endif
-			return;
-		}
-	}
-	WaitForSingleObject(audio_event, INFINITE);
-}
-#endif /* USE_POSITION_NOTIFY */
-
-static void DX5_PlayAudio(_THIS)
-{
-	/* Unlock the buffer, allowing it to play */
-	if ( locked_buf ) {
-		IDirectSoundBuffer_Unlock(mixbuf, locked_buf, mixlen, NULL, 0);
-	}
-
-}
-
-static Uint8 *DX5_GetAudioBuf(_THIS)
-{
-	DWORD   cursor, junk;
-	HRESULT result;
-	DWORD   rawlen;
-
-	/* Figure out which blocks to fill next */
-	locked_buf = NULL;
-	result = IDirectSoundBuffer_GetCurrentPosition(mixbuf, &junk, &cursor);
-	if ( result == DSERR_BUFFERLOST ) {
-		IDirectSoundBuffer_Restore(mixbuf);
-		result = IDirectSoundBuffer_GetCurrentPosition(mixbuf,
-								&junk, &cursor);
-	}
-	if ( result != DS_OK ) {
-		SetDSerror("DirectSound GetCurrentPosition", result);
-		return(NULL);
-	}
-	cursor /= mixlen;
-#ifdef DEBUG_SOUND
-	/* Detect audio dropouts */
-	{ DWORD spot = cursor;
-	  if ( spot < lastchunk ) {
-	    spot += NUM_BUFFERS;
-	  }
-	  if ( spot > lastchunk+1 ) {
-	    fprintf(stderr, "Audio dropout, missed %d fragments\n",
-	            (spot - (lastchunk+1)));
-	  }
-	}
-#endif
-	lastchunk = cursor;
-	cursor = (cursor+1)%NUM_BUFFERS;
-	cursor *= mixlen;
-
-	/* Lock the audio buffer */
-	result = IDirectSoundBuffer_Lock(mixbuf, cursor, mixlen,
-				(LPVOID *)&locked_buf, &rawlen, NULL, &junk, 0);
-	if ( result == DSERR_BUFFERLOST ) {
-		IDirectSoundBuffer_Restore(mixbuf);
-		result = IDirectSoundBuffer_Lock(mixbuf, cursor, mixlen,
-				(LPVOID *)&locked_buf, &rawlen, NULL, &junk, 0);
-	}
-	if ( result != DS_OK ) {
-		SetDSerror("DirectSound Lock", result);
-		return(NULL);
-	}
-	return(locked_buf);
-}
-
-static void DX5_WaitDone(_THIS)
-{
-	Uint8 *stream;
-
-	/* Wait for the playing chunk to finish */
-	stream = this->GetAudioBuf(this);
-	if ( stream != NULL ) {
-		SDL_memset(stream, silence, mixlen);
-		this->PlayAudio(this);
-	}
-	this->WaitAudio(this);
-
-	/* Stop the looping sound buffer */
-	IDirectSoundBuffer_Stop(mixbuf);
-}
-
-static void DX5_CloseAudio(_THIS)
-{
-	if ( sound != NULL ) {
-		if ( mixbuf != NULL ) {
-			/* Clean up the audio buffer */
-			IDirectSoundBuffer_Release(mixbuf);
-			mixbuf = NULL;
-		}
-		if ( audio_event != NULL ) {
-			CloseHandle(audio_event);
-			audio_event = NULL;
-		}
-		IDirectSound_Release(sound);
-		sound = NULL;
-	}
-}
-
-#ifdef USE_PRIMARY_BUFFER
-/* This function tries to create a primary audio buffer, and returns the
-   number of audio chunks available in the created buffer.
-*/
-static int CreatePrimary(LPDIRECTSOUND sndObj, HWND focus, 
-	LPDIRECTSOUNDBUFFER *sndbuf, WAVEFORMATEX *wavefmt, Uint32 chunksize)
-{
-	HRESULT result;
-	DSBUFFERDESC format;
-	DSBCAPS caps;
-	int numchunks;
-
-	/* Try to set primary mixing privileges */
-	result = IDirectSound_SetCooperativeLevel(sndObj, focus,
-							DSSCL_WRITEPRIMARY);
-	if ( result != DS_OK ) {
-#ifdef DEBUG_SOUND
-		SetDSerror("DirectSound SetCooperativeLevel", result);
-#endif
-		return(-1);
-	}
-
-	/* Try to create the primary buffer */
-	SDL_memset(&format, 0, sizeof(format));
-	format.dwSize = sizeof(format);
-	format.dwFlags=(DSBCAPS_PRIMARYBUFFER|DSBCAPS_GETCURRENTPOSITION2);
-	format.dwFlags |= DSBCAPS_STICKYFOCUS;
-#ifdef USE_POSITION_NOTIFY
-	format.dwFlags |= DSBCAPS_CTRLPOSITIONNOTIFY;
-#endif
-	result = IDirectSound_CreateSoundBuffer(sndObj, &format, sndbuf, NULL);
-	if ( result != DS_OK ) {
-#ifdef DEBUG_SOUND
-		SetDSerror("DirectSound CreateSoundBuffer", result);
-#endif
-		return(-1);
-	}
-
-	/* Check the size of the fragment buffer */
-	SDL_memset(&caps, 0, sizeof(caps));
-	caps.dwSize = sizeof(caps);
-	result = IDirectSoundBuffer_GetCaps(*sndbuf, &caps);
-	if ( result != DS_OK ) {
-#ifdef DEBUG_SOUND
-		SetDSerror("DirectSound GetCaps", result);
-#endif
-		IDirectSoundBuffer_Release(*sndbuf);
-		return(-1);
-	}
-	if ( (chunksize > caps.dwBufferBytes) ||
-				((caps.dwBufferBytes%chunksize) != 0) ) {
-		/* The primary buffer size is not a multiple of 'chunksize'
-		   -- this hopefully doesn't happen when 'chunksize' is a 
-		      power of 2.
-		*/
-		IDirectSoundBuffer_Release(*sndbuf);
-		SDL_SetError(
-"Primary buffer size is: %d, cannot break it into chunks of %d bytes\n",
-					caps.dwBufferBytes, chunksize);
-		return(-1);
-	}
-	numchunks = (caps.dwBufferBytes/chunksize);
-
-	/* Set the primary audio format */
-	result = IDirectSoundBuffer_SetFormat(*sndbuf, wavefmt);
-	if ( result != DS_OK ) {
-#ifdef DEBUG_SOUND
-		SetDSerror("DirectSound SetFormat", result);
-#endif
-		IDirectSoundBuffer_Release(*sndbuf);
-		return(-1);
-	}
-	return(numchunks);
-}
-#endif /* USE_PRIMARY_BUFFER */
-
-/* This function tries to create a secondary audio buffer, and returns the
-   number of audio chunks available in the created buffer.
-*/
-static int CreateSecondary(LPDIRECTSOUND sndObj, HWND focus,
-	LPDIRECTSOUNDBUFFER *sndbuf, WAVEFORMATEX *wavefmt, Uint32 chunksize)
-{
-	const int numchunks = 8;
-	HRESULT result;
-	DSBUFFERDESC format;
-	LPVOID pvAudioPtr1, pvAudioPtr2;
-	DWORD  dwAudioBytes1, dwAudioBytes2;
-
-	/* Try to set primary mixing privileges */
-	if ( focus ) {
-		result = IDirectSound_SetCooperativeLevel(sndObj,
-					focus, DSSCL_PRIORITY);
-	} else {
-		result = IDirectSound_SetCooperativeLevel(sndObj,
-					GetDesktopWindow(), DSSCL_NORMAL);
-	}
-	if ( result != DS_OK ) {
-#ifdef DEBUG_SOUND
-		SetDSerror("DirectSound SetCooperativeLevel", result);
-#endif
-		return(-1);
-	}
-
-	/* Try to create the secondary buffer */
-	SDL_memset(&format, 0, sizeof(format));
-	format.dwSize = sizeof(format);
-	format.dwFlags = DSBCAPS_GETCURRENTPOSITION2;
-#ifdef USE_POSITION_NOTIFY
-	format.dwFlags |= DSBCAPS_CTRLPOSITIONNOTIFY;
-#endif
-	if ( ! focus ) {
-		format.dwFlags |= DSBCAPS_GLOBALFOCUS;
-	} else {
-		format.dwFlags |= DSBCAPS_STICKYFOCUS;
-	}
-	format.dwBufferBytes = numchunks*chunksize;
-	if ( (format.dwBufferBytes < DSBSIZE_MIN) ||
-	     (format.dwBufferBytes > DSBSIZE_MAX) ) {
-		SDL_SetError("Sound buffer size must be between %d and %d",
-				DSBSIZE_MIN/numchunks, DSBSIZE_MAX/numchunks);
-		return(-1);
-	}
-	format.dwReserved = 0;
-	format.lpwfxFormat = wavefmt;
-	result = IDirectSound_CreateSoundBuffer(sndObj, &format, sndbuf, NULL);
-	if ( result != DS_OK ) {
-		SetDSerror("DirectSound CreateSoundBuffer", result);
-		return(-1);
-	}
-	IDirectSoundBuffer_SetFormat(*sndbuf, wavefmt);
-
-	/* Silence the initial audio buffer */
-	result = IDirectSoundBuffer_Lock(*sndbuf, 0, format.dwBufferBytes,
-	                                 (LPVOID *)&pvAudioPtr1, &dwAudioBytes1,
-	                                 (LPVOID *)&pvAudioPtr2, &dwAudioBytes2,
-	                                 DSBLOCK_ENTIREBUFFER);
-	if ( result == DS_OK ) {
-		if ( wavefmt->wBitsPerSample == 8 ) {
-			SDL_memset(pvAudioPtr1, 0x80, dwAudioBytes1);
-		} else {
-			SDL_memset(pvAudioPtr1, 0x00, dwAudioBytes1);
-		}
-		IDirectSoundBuffer_Unlock(*sndbuf,
-		                          (LPVOID)pvAudioPtr1, dwAudioBytes1,
-		                          (LPVOID)pvAudioPtr2, dwAudioBytes2);
-	}
-
-	/* We're ready to go */
-	return(numchunks);
-}
-
-/* This function tries to set position notify events on the mixing buffer */
-#ifdef USE_POSITION_NOTIFY
-static int CreateAudioEvent(_THIS)
-{
-	LPDIRECTSOUNDNOTIFY notify;
-	DSBPOSITIONNOTIFY *notify_positions;
-	int i, retval;
-	HRESULT result;
-
-	/* Default to fail on exit */
-	retval = -1;
-	notify = NULL;
-
-	/* Query for the interface */
-	result = IDirectSoundBuffer_QueryInterface(mixbuf,
-			&IID_IDirectSoundNotify, (void *)&notify);
-	if ( result != DS_OK ) {
-		goto done;
-	}
-
-	/* Allocate the notify structures */
-	notify_positions = (DSBPOSITIONNOTIFY *)SDL_malloc(NUM_BUFFERS*
-					sizeof(*notify_positions));
-	if ( notify_positions == NULL ) {
-		goto done;
-	}
-
-	/* Create the notify event */
-	audio_event = CreateEvent(NULL, FALSE, FALSE, NULL);
-	if ( audio_event == NULL ) {
-		goto done;
-	}
-
-	/* Set up the notify structures */
-	for ( i=0; i<NUM_BUFFERS; ++i ) {
-		notify_positions[i].dwOffset = i*mixlen;
-		notify_positions[i].hEventNotify = audio_event;
-	}
-	result = IDirectSoundNotify_SetNotificationPositions(notify,
-					NUM_BUFFERS, notify_positions);
-	if ( result == DS_OK ) {
-		retval = 0;
-	}
-done:
-	if ( notify != NULL ) {
-		IDirectSoundNotify_Release(notify);
-	}
-	return(retval);
-}
-#endif /* USE_POSITION_NOTIFY */
-
-static int DX5_OpenAudio(_THIS, SDL_AudioSpec *spec)
-{
-	HRESULT      result;
-	WAVEFORMATEX waveformat;
-
-	/* Set basic WAVE format parameters */
-	SDL_memset(&waveformat, 0, sizeof(waveformat));
-	waveformat.wFormatTag = WAVE_FORMAT_PCM;
-
-	/* Determine the audio parameters from the AudioSpec */
-	switch ( spec->format & 0xFF ) {
-		case 8:
-			/* Unsigned 8 bit audio data */
-			spec->format = AUDIO_U8;
-			silence = 0x80;
-			waveformat.wBitsPerSample = 8;
-			break;
-		case 16:
-			/* Signed 16 bit audio data */
-			spec->format = AUDIO_S16;
-			silence = 0x00;
-			waveformat.wBitsPerSample = 16;
-			break;
-		default:
-			SDL_SetError("Unsupported audio format");
-			return(-1);
-	}
-	waveformat.nChannels = spec->channels;
-	waveformat.nSamplesPerSec = spec->freq;
-	waveformat.nBlockAlign =
-		waveformat.nChannels * (waveformat.wBitsPerSample/8);
-	waveformat.nAvgBytesPerSec = 
-		waveformat.nSamplesPerSec * waveformat.nBlockAlign;
-
-	/* Update the fragment size as size in bytes */
-	SDL_CalculateAudioSpec(spec);
-
-	/* Open the audio device */
-	result = DSoundCreate(NULL, &sound, NULL);
-	if ( result != DS_OK ) {
-		SetDSerror("DirectSoundCreate", result);
-		return(-1);
-	}
-
-	/* Create the audio buffer to which we write */
-	NUM_BUFFERS = -1;
-#ifdef USE_PRIMARY_BUFFER
-	if ( mainwin ) {
-		NUM_BUFFERS = CreatePrimary(sound, mainwin, &mixbuf,
-						&waveformat, spec->size);
-	}
-#endif /* USE_PRIMARY_BUFFER */
-	if ( NUM_BUFFERS < 0 ) {
-		NUM_BUFFERS = CreateSecondary(sound, mainwin, &mixbuf,
-						&waveformat, spec->size);
-		if ( NUM_BUFFERS < 0 ) {
-			return(-1);
-		}
-#ifdef DEBUG_SOUND
-		fprintf(stderr, "Using secondary audio buffer\n");
-#endif
-	}
-#ifdef DEBUG_SOUND
-	else
-		fprintf(stderr, "Using primary audio buffer\n");
-#endif
-
-	/* The buffer will auto-start playing in DX5_WaitAudio() */
-	lastchunk = 0;
-	mixlen = spec->size;
-
-#ifdef USE_POSITION_NOTIFY
-	/* See if we can use DirectX 6 event notification */
-	if ( CreateAudioEvent(this) == 0 ) {
-		this->WaitAudio = DX6_WaitAudio_EventWait;
-	} else {
-		this->WaitAudio = DX5_WaitAudio_BusyWait;
-	}
-#endif
-	return(0);
-}
-
diff --git a/apps/plugins/sdl/src/audio/windx5/SDL_dx5audio.h b/apps/plugins/sdl/src/audio/windx5/SDL_dx5audio.h
deleted file mode 100644
index bc4022fc9c..0000000000
--- a/apps/plugins/sdl/src/audio/windx5/SDL_dx5audio.h
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Sam Lantinga
-    slouken@libsdl.org
-*/
-#include "SDL_config.h"
-
-#ifndef _SDL_lowaudio_h
-#define _SDL_lowaudio_h
-
-#include "directx.h"
-
-#include "../SDL_sysaudio.h"
-
-/* Hidden "this" pointer for the video functions */
-#define _THIS	SDL_AudioDevice *this
-
-/* The DirectSound objects */
-struct SDL_PrivateAudioData {
-	LPDIRECTSOUND sound;
-	LPDIRECTSOUNDBUFFER mixbuf;
-	int NUM_BUFFERS;
-	int mixlen, silence;
-	DWORD lastchunk;
-	Uint8 *locked_buf;
-	HANDLE audio_event;
-};
-
-/* Old variable names */
-#define sound			(this->hidden->sound)
-#define mixbuf			(this->hidden->mixbuf)
-#define NUM_BUFFERS		(this->hidden->NUM_BUFFERS)
-#define mixlen			(this->hidden->mixlen)
-#define silence			(this->hidden->silence)
-#define lastchunk		(this->hidden->lastchunk)
-#define locked_buf		(this->hidden->locked_buf)
-#define audio_event		(this->hidden->audio_event)
-
-#endif /* _SDL_lowaudio_h */
diff --git a/apps/plugins/sdl/src/audio/windx5/directx.h b/apps/plugins/sdl/src/audio/windx5/directx.h
deleted file mode 100644
index 5f339f2de8..0000000000
--- a/apps/plugins/sdl/src/audio/windx5/directx.h
+++ /dev/null
@@ -1,81 +0,0 @@
-
-#ifndef _directx_h
-#define _directx_h
-
-/* Include all of the DirectX 5.0 headers and adds any necessary tweaks */
-
-#define WIN32_LEAN_AND_MEAN
-#include <windows.h>
-#include <mmsystem.h>
-#ifndef WIN32
-#define WIN32
-#endif
-#undef  WINNT
-
-/* Far pointers don't exist in 32-bit code */
-#ifndef FAR
-#define FAR
-#endif
-
-/* Error codes not yet included in Win32 API header files */
-#ifndef MAKE_HRESULT
-#define MAKE_HRESULT(sev,fac,code) \
-	((HRESULT)(((unsigned long)(sev)<<31) | ((unsigned long)(fac)<<16) | ((unsigned long)(code))))
-#endif
-
-#ifndef S_OK
-#define S_OK		(HRESULT)0x00000000L
-#endif
-
-#ifndef SUCCEEDED
-#define SUCCEEDED(x)	((HRESULT)(x) >= 0)
-#endif
-#ifndef FAILED
-#define FAILED(x)	((HRESULT)(x)<0)
-#endif
-
-#ifndef E_FAIL
-#define E_FAIL		(HRESULT)0x80000008L
-#endif
-#ifndef E_NOINTERFACE
-#define E_NOINTERFACE	(HRESULT)0x80004002L
-#endif
-#ifndef E_OUTOFMEMORY
-#define E_OUTOFMEMORY	(HRESULT)0x8007000EL
-#endif
-#ifndef E_INVALIDARG
-#define E_INVALIDARG	(HRESULT)0x80070057L
-#endif
-#ifndef E_NOTIMPL
-#define E_NOTIMPL	(HRESULT)0x80004001L
-#endif
-#ifndef REGDB_E_CLASSNOTREG
-#define REGDB_E_CLASSNOTREG	(HRESULT)0x80040154L
-#endif
-
-/* Severity codes */
-#ifndef SEVERITY_ERROR
-#define SEVERITY_ERROR	1
-#endif
-
-/* Error facility codes */
-#ifndef FACILITY_WIN32
-#define FACILITY_WIN32	7
-#endif
-
-#ifndef FIELD_OFFSET
-#define FIELD_OFFSET(type, field)    ((LONG)&(((type *)0)->field))
-#endif
-
-/* DirectX headers (if it isn't included, I haven't tested it yet)
- */
-/* We need these defines to mark what version of DirectX API we use */
-#define DIRECTDRAW_VERSION  0x0700
-#define DIRECTSOUND_VERSION 0x0500
-#define DIRECTINPUT_VERSION 0x0500
-
-#include <ddraw.h>
-#include <dsound.h>
-#include <dinput.h>
-
-#endif /* _directx_h */
diff --git a/apps/plugins/sdl/src/cdrom/aix/SDL_syscdrom.c b/apps/plugins/sdl/src/cdrom/aix/SDL_syscdrom.c
deleted file mode 100644
index e7e05585e3..0000000000
--- a/apps/plugins/sdl/src/cdrom/aix/SDL_syscdrom.c
+++ /dev/null
@@ -1,660 +0,0 @@
-/*
-    SDL - Simple DirectMedia Layer
-    Copyright (C) 1997-2012 Sam Lantinga
-
-    This library is free software; you can redistribute it and/or
-    modify it under the terms of the GNU Lesser General Public
-    License as published by the Free Software Foundation; either
-    version 2.1 of the License, or (at your option) any later version.
-
-    This library is distributed in the hope that it will be useful,
-    but WITHOUT ANY WARRANTY; without even the implied warranty of
-    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-    Lesser General Public License for more details.
-
-    You should have received a copy of the GNU Lesser General Public
-    License along with this library; if not, write to the Free Software
-    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
-
-    Carsten Griwodz
-    griff@kom.tu-darmstadt.de
-
-    based on linux/SDL_syscdrom.c by Sam Lantinga
-*/
-#include "SDL_config.h"
-
-#ifdef SDL_CDROM_AIX
-
-/* Functions for system-level CD-ROM audio control */
-
-/*#define DEBUG_CDROM 1*/
-
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-#include <errno.h>
-#include <unistd.h>
-
-#include <sys/ioctl.h>
-#include <sys/devinfo.h>
-#include <sys/mntctl.h>
-#include <sys/statfs.h>
-#include <sys/vmount.h>
-#include <fstab.h>
-#include <sys/scdisk.h>
-
-#include "SDL_cdrom.h"
-#include "../SDL_syscdrom.h"
-
-/* The maximum number of CD-ROM drives we'll detect */
-#define MAX_DRIVES	16	
-
-/* A list of available CD-ROM drives */
-static char *SDL_cdlist[MAX_DRIVES];
-static dev_t SDL_cdmode[MAX_DRIVES];
-
-/* The system-dependent CD control functions */
-static const char *SDL_SYS_CDName(int drive);
-static int         SDL_SYS_CDOpen(int drive);
-static int         SDL_SYS_CDGetTOC(SDL_CD *cdrom);
-static CDstatus    SDL_SYS_CDStatus(SDL_CD *cdrom, int *position);
-static int         SDL_SYS_CDPlay(SDL_CD *cdrom, int start, int length);
-static int         SDL_SYS_CDPause(SDL_CD *cdrom);
-static int         SDL_SYS_CDResume(SDL_CD *cdrom);
-static int         SDL_SYS_CDStop(SDL_CD *cdrom);
-static int         SDL_SYS_CDEject(SDL_CD *cdrom);
-static void        SDL_SYS_CDClose(SDL_CD *cdrom);
-static int         SDL_SYS_CDioctl(int id, int command, void *arg);
-
-/* Check a drive to see if it is a CD-ROM */
-static int CheckDrive(char *drive, struct stat *stbuf)
-{
-    int is_cd;
-    int cdfd;
-    int ret;
-    struct devinfo info;
-
-    /* If it doesn't exist, return -1 */
-    if ( stat(drive, stbuf) < 0 ) {
-        return -1;
-    }
-
-    /* If it does exist, verify that it's an available CD-ROM */
-    is_cd = 0;
-    if ( S_ISCHR(stbuf->st_mode) || S_ISBLK(stbuf->st_mode) ) {
-        cdfd = open(drive, (O_RDONLY|O_EXCL|O_NONBLOCK), 0);
-        if ( cdfd >= 0 ) {
-            ret = SDL_SYS_CDioctl( cdfd, IOCINFO, &info );
-	    if ( ret < 0 ) {
-		/* Some kind of error */
-		is_cd = 0;
-	    } else {
-		if ( info.devtype == DD_CDROM ) {
-		    is_cd = 1;
-		} else {
-		    is_cd = 0;
-		}
-	    }
-            close(cdfd);
-	}
-#ifdef DEBUG_CDROM
-	else
-	{
-            fprintf(stderr, "Could not open drive %s (%s)\n", drive, strerror(errno));
-	}
-#endif
-    }
-    return is_cd;
-}
-
-/* Add a CD-ROM drive to our list of valid drives */
-static void AddDrive(char *drive, struct stat *stbuf)
-{
-	int i;
-
-	if ( SDL_numcds < MAX_DRIVES ) {
-		/* Check to make sure it's not already in our list.
-	 	   This can happen when we see a drive via symbolic link.
-		 */
-		for ( i=0; i<SDL_numcds; ++i ) {
-			if ( stbuf->st_rdev == SDL_cdmode[i] ) {
-#ifdef DEBUG_CDROM
-  fprintf(stderr, "Duplicate drive detected: %s == %s\n", drive, SDL_cdlist[i]);
-#endif
-				return;
-			}
-		}
-
-		/* Add this drive to our list */
-		i = SDL_numcds;
-		SDL_cdlist[i] = SDL_strdup(drive);
-		if ( SDL_cdlist[i] == NULL ) {
-			SDL_OutOfMemory();
-			return;
-		}
-		SDL_cdmode[i] = stbuf->st_rdev;
-		++SDL_numcds;
-#ifdef DEBUG_CDROM
-  fprintf(stderr, "Added CD-ROM drive: %s\n", drive);
-#endif
-	}
-}
-
-static void CheckMounts()
-{
-    char*          buffer;
-    int            bufsz;
-    struct vmount* ptr;
-    int            ret;
-
-    buffer = (char*)SDL_malloc(10);
-    bufsz  = 10;
-    if ( buffer==NULL )
-    {
-        fprintf(stderr, "Could not allocate 10 bytes in aix/SDL_syscdrom.c:CheckMounts\n" );
-	exit ( -10 );
-    }
-
-    do
-    {
-	/* mntctrl() returns an array of all mounted filesystems */
-        ret = mntctl ( MCTL_QUERY, bufsz, buffer );
-        if ( ret == 0 )
-        {
-				   /* Buffer was too small, realloc.    */
-            bufsz = *(int*)buffer; /* Required size is in first word.   */
-				   /* (whatever a word is in AIX 4.3.3) */
-				   /* int seems to be OK in 32bit mode. */
-            SDL_free(buffer);
-            buffer = (char*)SDL_malloc(bufsz);
-            if ( buffer==NULL )
-            {
-                fprintf(stderr,
-			"Could not allocate %d bytes in aix/SDL_syscdrom.c:CheckMounts\n",
-			bufsz );
-	        exit ( -10 );
-            }
-        }
-	else if ( ret < 0 )
-	{
-#ifdef DEBUG_CDROM
-            fprintf(stderr, "Error reading vmount structures\n");
-#endif
-	    return;
-	}
-    }
-    while ( ret == 0 );
-
-#ifdef DEBUG_CDROM
-    fprintf ( stderr, "Read %d vmount structures\n",ret );
-#endif
-    ptr = (struct vmount*)buffer;
-    do
-    {
-            switch(ptr->vmt_gfstype)
-            {
-            case MNT_CDROM :
-                {
-		    struct stat stbuf;
-		    char*       text;
-
-		    text = (char*)ptr + ptr->vmt_data[VMT_OBJECT].vmt_off;
-#ifdef DEBUG_CDROM
-  fprintf(stderr, "Checking mount path: %s mounted on %s\n",
-	text, (char*)ptr + ptr->vmt_data[VMT_STUB].vmt_off );
-#endif
-		    if ( CheckDrive( text, &stbuf) > 0)
-		    {
-		        AddDrive( text, &stbuf);
-		    }
-                }
-                break;
-            default :
-                break;
-            }
-            ptr = (struct vmount*)((char*)ptr + ptr->vmt_length);
-            ret--;
-    }
-    while ( ret > 0 );
-
-    free ( buffer );
-}
-
-static int CheckNonmounts()
-{
-#ifdef _THREAD_SAFE
-    AFILE_t      fsFile = NULL;
-    int          passNo = 0;
-    int          ret;
-    struct fstab entry;
-    struct stat  stbuf;
-
-    ret = setfsent_r( &fsFile, &passNo );
-    if ( ret != 0 ) return -1;
-    do
-    {
-        ret = getfsent_r ( &entry, &fsFile, &passNo );
-        if ( ret == 0 ) {
-            char* l = SDL_strrchr(entry.fs_spec,'/');
-            if ( l != NULL ) {
-                if ( !SDL_strncmp("cd",++l,2) ) {
-#ifdef DEBUG_CDROM
-                    fprintf(stderr,
-			    "Found unmounted CD ROM drive with device name %s\n",
-			    entry.fs_spec);
-#endif
-		    if ( CheckDrive( entry.fs_spec, &stbuf) > 0)
-		    {
-		        AddDrive( entry.fs_spec, &stbuf);
-		    }
-                }
-            }
-        }
-    }
-    while ( ret == 0 );
-    ret = endfsent_r ( &fsFile );
-    if ( ret != 0 ) return -1;
-    return 0;
-#else
-    struct fstab* entry;
-    struct stat  stbuf;
-
-    setfsent();
-    do
-    {
-        entry = getfsent();
-        if ( entry != NULL ) {
-            char* l = SDL_strrchr(entry->fs_spec,'/');
-            if ( l != NULL ) {
-                if ( !SDL_strncmp("cd",++l,2) ) {
-#ifdef DEBUG_CDROM
-                    fprintf(stderr,"Found unmounted CD ROM drive with device name %s", entry->fs_spec);
-#endif
-		    if ( CheckDrive( entry->fs_spec, &stbuf) > 0)
-		    {
-		        AddDrive( entry->fs_spec, &stbuf);
-		    }
-                }
-            }
-        }
-    }
-    while ( entry != NULL );
-    endfsent();
-#endif
-}
-
-int  SDL_SYS_CDInit(void)
-{
-	char *SDLcdrom;
-	struct stat stbuf;
-
-	/* Fill in our driver capabilities */
-	SDL_CDcaps.Name = SDL_SYS_CDName;
-	SDL_CDcaps.Open = SDL_SYS_CDOpen;
-	SDL_CDcaps.GetTOC = SDL_SYS_CDGetTOC;
-	SDL_CDcaps.Status = SDL_SYS_CDStatus;
-	SDL_CDcaps.Play = SDL_SYS_CDPlay;
-	SDL_CDcaps.Pause = SDL_SYS_CDPause;
-	SDL_CDcaps.Resume = SDL_SYS_CDResume;
-	SDL_CDcaps.Stop = SDL_SYS_CDStop;
-	SDL_CDcaps.Eject = SDL_SYS_CDEject;
-	SDL_CDcaps.Close = SDL_SYS_CDClose;
-
-	/* Look in the environment for our CD-ROM drive list */
-	SDLcdrom = SDL_getenv("SDL_CDROM");	/* ':' separated list of devices */
-	if ( SDLcdrom != NULL ) {
-		char *cdpath, *delim;
-		size_t len = SDL_strlen(SDLcdrom)+1;
-		cdpath = SDL_stack_alloc(char, len);
-		if ( cdpath != NULL ) {
-			SDL_strlcpy(cdpath, SDLcdrom, len);
-			SDLcdrom = cdpath;
-			do {
-				delim = SDL_strchr(SDLcdrom, ':');
-				if ( delim ) {
-					*delim++ = '\0';
-				}
-#ifdef DEBUG_CDROM
-  fprintf(stderr, "Checking CD-ROM drive from SDL_CDROM: %s\n", SDLcdrom);
-#endif
-				if ( CheckDrive(SDLcdrom, &stbuf) > 0 ) {
-					AddDrive(SDLcdrom, &stbuf);
-				}
-				if ( delim ) {
-					SDLcdrom = delim;
-				} else {
-					SDLcdrom = NULL;
-				}
-			} while ( SDLcdrom );
-			SDL_stack_free(cdpath);
-		}
-
-		/* If we found our drives, there's nothing left to do */
-		if ( SDL_numcds > 0 ) {
-			return(0);
-		}
-	}
-
-	CheckMounts();
-	CheckNonmounts();
-
-	return 0;
-}
-
-/* General ioctl() CD-ROM command function */
-static int SDL_SYS_CDioctl(int id, int command, void *arg)
-{
-    int retval;
-
-    retval = ioctl(id, command, arg);
-    if ( retval < 0 ) {
-        SDL_SetError("ioctl() error: %s", strerror(errno));
-    }
-    return retval;
-}