Import SDL 1.2.14 and GLEW 1.5.4.

I am not replacing the following header files used by the Windows build,
as I don't have a Windows build setup to test. Please point the Windows
build at the header files inside these clean distribution packages and
garbage collect the duplicates: GLew/*.h and SDL/Include_1.2.


git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@5711 8ced0084-cf51-0410-be5f-012b33b47a6e

Externals/SDL/src/SDL.c

@@ -0,0 +1,350 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Initialization code for SDL */
+
+#include "SDL.h"
+#include "SDL_fatal.h"
+#if !SDL_VIDEO_DISABLED
+#include "video/SDL_leaks.h"
+#endif
+
+#if SDL_THREAD_PTH
+#include <pth.h>
+#endif
+
+/* Initialization/Cleanup routines */
+#if !SDL_JOYSTICK_DISABLED
+extern int  SDL_JoystickInit(void);
+extern void SDL_JoystickQuit(void);
+#endif
+#if !SDL_CDROM_DISABLED
+extern int  SDL_CDROMInit(void);
+extern void SDL_CDROMQuit(void);
+#endif
+#if !SDL_TIMERS_DISABLED
+extern void SDL_StartTicks(void);
+extern int  SDL_TimerInit(void);
+extern void SDL_TimerQuit(void);
+#endif
+
+/* The current SDL version */
+static SDL_version version = 
+	{ SDL_MAJOR_VERSION, SDL_MINOR_VERSION, SDL_PATCHLEVEL };
+
+/* The initialized subsystems */
+static Uint32 SDL_initialized = 0;
+#if !SDL_TIMERS_DISABLED
+static Uint32 ticks_started = 0;
+#endif
+
+#ifdef CHECK_LEAKS
+int surfaces_allocated = 0;
+#endif
+
+int SDL_InitSubSystem(Uint32 flags)
+{
+#if !SDL_VIDEO_DISABLED
+	/* Initialize the video/event subsystem */
+	if ( (flags & SDL_INIT_VIDEO) && !(SDL_initialized & SDL_INIT_VIDEO) ) {
+		if ( SDL_VideoInit(SDL_getenv("SDL_VIDEODRIVER"),
+		                   (flags&SDL_INIT_EVENTTHREAD)) < 0 ) {
+			return(-1);
+		}
+		SDL_initialized |= SDL_INIT_VIDEO;
+	}
+#else
+	if ( flags & SDL_INIT_VIDEO ) {
+		SDL_SetError("SDL not built with video support");
+		return(-1);
+	}
+#endif
+
+#if !SDL_AUDIO_DISABLED
+	/* Initialize the audio subsystem */
+	if ( (flags & SDL_INIT_AUDIO) && !(SDL_initialized & SDL_INIT_AUDIO) ) {
+		if ( SDL_AudioInit(SDL_getenv("SDL_AUDIODRIVER")) < 0 ) {
+			return(-1);
+		}
+		SDL_initialized |= SDL_INIT_AUDIO;
+	}
+#else
+	if ( flags & SDL_INIT_AUDIO ) {
+		SDL_SetError("SDL not built with audio support");
+		return(-1);
+	}
+#endif
+
+#if !SDL_TIMERS_DISABLED
+	/* Initialize the timer subsystem */
+	if ( ! ticks_started ) {
+		SDL_StartTicks();
+		ticks_started = 1;
+	}
+	if ( (flags & SDL_INIT_TIMER) && !(SDL_initialized & SDL_INIT_TIMER) ) {
+		if ( SDL_TimerInit() < 0 ) {
+			return(-1);
+		}
+		SDL_initialized |= SDL_INIT_TIMER;
+	}
+#else
+	if ( flags & SDL_INIT_TIMER ) {
+		SDL_SetError("SDL not built with timer support");
+		return(-1);
+	}
+#endif
+
+#if !SDL_JOYSTICK_DISABLED
+	/* Initialize the joystick subsystem */
+	if ( (flags & SDL_INIT_JOYSTICK) &&
+	     !(SDL_initialized & SDL_INIT_JOYSTICK) ) {
+		if ( SDL_JoystickInit() < 0 ) {
+			return(-1);
+		}
+		SDL_initialized |= SDL_INIT_JOYSTICK;
+	}
+#else
+	if ( flags & SDL_INIT_JOYSTICK ) {
+		SDL_SetError("SDL not built with joystick support");
+		return(-1);
+	}
+#endif
+
+#if !SDL_CDROM_DISABLED
+	/* Initialize the CD-ROM subsystem */
+	if ( (flags & SDL_INIT_CDROM) && !(SDL_initialized & SDL_INIT_CDROM) ) {
+		if ( SDL_CDROMInit() < 0 ) {
+			return(-1);
+		}
+		SDL_initialized |= SDL_INIT_CDROM;
+	}
+#else
+	if ( flags & SDL_INIT_CDROM ) {
+		SDL_SetError("SDL not built with cdrom support");
+		return(-1);
+	}
+#endif
+	return(0);
+}
+
+int SDL_Init(Uint32 flags)
+{
+#if !SDL_THREADS_DISABLED && SDL_THREAD_PTH
+	if (!pth_init()) {
+		return -1;
+	}
+#endif
+
+	/* Clear the error message */
+	SDL_ClearError();
+
+	/* Initialize the desired subsystems */
+	if ( SDL_InitSubSystem(flags) < 0 ) {
+		return(-1);
+	}
+
+	/* Everything is initialized */
+	if ( !(flags & SDL_INIT_NOPARACHUTE) ) {
+		SDL_InstallParachute();
+	}
+	return(0);
+}
+
+void SDL_QuitSubSystem(Uint32 flags)
+{
+	/* Shut down requested initialized subsystems */
+#if !SDL_CDROM_DISABLED
+	if ( (flags & SDL_initialized & SDL_INIT_CDROM) ) {
+		SDL_CDROMQuit();
+		SDL_initialized &= ~SDL_INIT_CDROM;
+	}
+#endif
+#if !SDL_JOYSTICK_DISABLED
+	if ( (flags & SDL_initialized & SDL_INIT_JOYSTICK) ) {
+		SDL_JoystickQuit();
+		SDL_initialized &= ~SDL_INIT_JOYSTICK;
+	}
+#endif
+#if !SDL_TIMERS_DISABLED
+	if ( (flags & SDL_initialized & SDL_INIT_TIMER) ) {
+		SDL_TimerQuit();
+		SDL_initialized &= ~SDL_INIT_TIMER;
+	}
+#endif
+#if !SDL_AUDIO_DISABLED
+	if ( (flags & SDL_initialized & SDL_INIT_AUDIO) ) {
+		SDL_AudioQuit();
+		SDL_initialized &= ~SDL_INIT_AUDIO;
+	}
+#endif
+#if !SDL_VIDEO_DISABLED
+	if ( (flags & SDL_initialized & SDL_INIT_VIDEO) ) {
+		SDL_VideoQuit();
+		SDL_initialized &= ~SDL_INIT_VIDEO;
+	}
+#endif
+}
+
+Uint32 SDL_WasInit(Uint32 flags)
+{
+	if ( ! flags ) {
+		flags = SDL_INIT_EVERYTHING;
+	}
+	return (SDL_initialized&flags);
+}
+
+void SDL_Quit(void)
+{
+	/* Quit all subsystems */
+#ifdef DEBUG_BUILD
+  printf("[SDL_Quit] : Enter! Calling QuitSubSystem()\n"); fflush(stdout);
+#endif
+	SDL_QuitSubSystem(SDL_INIT_EVERYTHING);
+
+#ifdef CHECK_LEAKS
+#ifdef DEBUG_BUILD
+  printf("[SDL_Quit] : CHECK_LEAKS\n"); fflush(stdout);
+#endif
+
+	/* Print the number of surfaces not freed */
+	if ( surfaces_allocated != 0 ) {
+		fprintf(stderr, "SDL Warning: %d SDL surfaces extant\n", 
+							surfaces_allocated);
+	}
+#endif
+#ifdef DEBUG_BUILD
+  printf("[SDL_Quit] : SDL_UninstallParachute()\n"); fflush(stdout);
+#endif
+
+	/* Uninstall any parachute signal handlers */
+	SDL_UninstallParachute();
+
+#if !SDL_THREADS_DISABLED && SDL_THREAD_PTH
+	pth_kill();
+#endif
+#ifdef DEBUG_BUILD
+  printf("[SDL_Quit] : Returning!\n"); fflush(stdout);
+#endif
+
+}
+
+/* Return the library version number */
+const SDL_version * SDL_Linked_Version(void)
+{
+	return(&version);
+}
+
+#if defined(__OS2__)
+/* Building for OS/2 */
+#ifdef __WATCOMC__
+
+#define INCL_DOSERRORS
+#define INCL_DOSEXCEPTIONS
+#include <os2.h>
+
+/* Exception handler to prevent the Audio thread hanging, making a zombie process! */
+ULONG _System SDL_Main_ExceptionHandler(PEXCEPTIONREPORTRECORD pERepRec,
+                                        PEXCEPTIONREGISTRATIONRECORD pERegRec,
+                                        PCONTEXTRECORD pCtxRec,
+                                        PVOID p)
+{
+  if (pERepRec->fHandlerFlags & EH_EXIT_UNWIND)
+    return XCPT_CONTINUE_SEARCH;
+  if (pERepRec->fHandlerFlags & EH_UNWINDING)
+    return XCPT_CONTINUE_SEARCH;
+  if (pERepRec->fHandlerFlags & EH_NESTED_CALL)
+    return XCPT_CONTINUE_SEARCH;
+
+  /* Do cleanup at every fatal exception! */
+  if (((pERepRec->ExceptionNum & XCPT_SEVERITY_CODE) == XCPT_FATAL_EXCEPTION) &&
+      (pERepRec->ExceptionNum != XCPT_BREAKPOINT) &&
+      (pERepRec->ExceptionNum != XCPT_SINGLE_STEP)
+     )
+  {
+    if (SDL_initialized & SDL_INIT_AUDIO)
+    {
+      /* This removes the zombie audio thread in case of emergency. */
+#ifdef DEBUG_BUILD
+      printf("[SDL_Main_ExceptionHandler] : Calling SDL_CloseAudio()!\n");
+#endif
+      SDL_CloseAudio();
+    }
+  }
+  return (XCPT_CONTINUE_SEARCH);
+}
+
+
+EXCEPTIONREGISTRATIONRECORD SDL_Main_xcpthand = {0, SDL_Main_ExceptionHandler};
+
+/* The main DLL entry for DLL Initialization and Uninitialization: */
+unsigned _System LibMain(unsigned hmod, unsigned termination)
+{
+  if (termination)
+  {
+#ifdef DEBUG_BUILD
+/*    printf("[SDL DLL Unintialization] : Removing exception handler\n"); */
+#endif
+    DosUnsetExceptionHandler(&SDL_Main_xcpthand);
+    return 1;
+  } else
+  {
+#ifdef DEBUG_BUILD
+    /* Make stdout and stderr unbuffered! */
+    setbuf(stdout, NULL);
+    setbuf(stderr, NULL);
+#endif
+    /* Fire up exception handler */
+#ifdef DEBUG_BUILD
+/*    printf("[SDL DLL Initialization] : Setting exception handler\n"); */
+#endif
+    /* Set exception handler */
+    DosSetExceptionHandler(&SDL_Main_xcpthand);
+
+    return 1;
+  }
+}
+#endif /* __WATCOMC__ */
+
+#elif defined(__WIN32__)  && !defined(__SYMBIAN32__)
+
+#if !defined(HAVE_LIBC) || (defined(__WATCOMC__) && defined(BUILD_DLL))
+/* Need to include DllMain() on Watcom C for some reason.. */
+#define WIN32_LEAN_AND_MEAN
+#include <windows.h>
+
+BOOL APIENTRY _DllMainCRTStartup( HANDLE hModule, 
+                       DWORD  ul_reason_for_call, 
+                       LPVOID lpReserved )
+{
+	switch (ul_reason_for_call) {
+		case DLL_PROCESS_ATTACH:
+		case DLL_THREAD_ATTACH:
+		case DLL_THREAD_DETACH:
+		case DLL_PROCESS_DETACH:
+			break;
+	}
+	return TRUE;
+}
+#endif /* building DLL with Watcom C */
+
+#endif /* OS/2 elif __WIN32__ */

Externals/SDL/src/SDL_error.c

@@ -0,0 +1,238 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Simple error handling in SDL */
+
+#include "SDL_error.h"
+#include "SDL_error_c.h"
+
+/* Routine to get the thread-specific error variable */
+#if SDL_THREADS_DISABLED
+/* The  SDL_arraysize(The ),default (non-thread-safe) global error variable */
+static SDL_error SDL_global_error;
+#define SDL_GetErrBuf()	(&SDL_global_error)
+#else
+extern SDL_error *SDL_GetErrBuf(void);
+#endif /* SDL_THREADS_DISABLED */
+
+#define SDL_ERRBUFIZE	1024
+
+/* Private functions */
+
+static const char *SDL_LookupString(const char *key)
+{
+	/* FIXME: Add code to lookup key in language string hash-table */
+	return key;
+}
+
+/* Public functions */
+
+void SDL_SetError (const char *fmt, ...)
+{
+	va_list ap;
+	SDL_error *error;
+
+	/* Copy in the key, mark error as valid */
+	error = SDL_GetErrBuf();
+	error->error = 1;
+	SDL_strlcpy((char *)error->key, fmt, sizeof(error->key));
+
+	va_start(ap, fmt);
+	error->argc = 0;
+	while ( *fmt ) {
+		if ( *fmt++ == '%' ) {
+			while ( *fmt == '.' || (*fmt >= '0' && *fmt <= '9') ) {
+				++fmt;
+			}
+			switch (*fmt++) {
+			    case 0:  /* Malformed format string.. */
+				--fmt;
+				break;
+			    case 'c':
+			    case 'i':
+			    case 'd':
+			    case 'u':
+			    case 'o':
+			    case 'x':
+			    case 'X':
+				error->args[error->argc++].value_i =
+							va_arg(ap, int);
+				break;
+			    case 'f':
+				error->args[error->argc++].value_f =
+							va_arg(ap, double);
+				break;
+			    case 'p':
+				error->args[error->argc++].value_ptr =
+							va_arg(ap, void *);
+				break;
+			    case 's':
+				{
+				  int i = error->argc;
+				  const char *str = va_arg(ap, const char *);
+				  if (str == NULL)
+				      str = "(null)";
+				  SDL_strlcpy((char *)error->args[i].buf, str, ERR_MAX_STRLEN);
+				  error->argc++;
+				}
+				break;
+			    default:
+				break;
+			}
+			if ( error->argc >= ERR_MAX_ARGS ) {
+				break;
+			}
+		}
+	}
+	va_end(ap);
+
+	/* If we are in debug mode, print out an error message */
+#ifdef DEBUG_ERROR
+	fprintf(stderr, "SDL_SetError: %s\n", SDL_GetError());
+#endif
+}
+
+/* This function has a bit more overhead than most error functions
+   so that it supports internationalization and thread-safe errors.
+*/
+char *SDL_GetErrorMsg(char *errstr, unsigned int maxlen)
+{
+	SDL_error *error;
+
+	/* Clear the error string */
+	*errstr = '\0'; --maxlen;
+
+	/* Get the thread-safe error, and print it out */
+	error = SDL_GetErrBuf();
+	if ( error->error ) {
+		const char *fmt;
+		char *msg = errstr;
+		int len;
+		int argi;
+
+		fmt = SDL_LookupString(error->key);
+		argi = 0;
+		while ( *fmt && (maxlen > 0) ) {
+			if ( *fmt == '%' ) {
+				char tmp[32], *spot = tmp;
+				*spot++ = *fmt++;
+				while ( (*fmt == '.' || (*fmt >= '0' && *fmt <= '9')) && spot < (tmp+SDL_arraysize(tmp)-2) ) {
+					*spot++ = *fmt++;
+				}
+				*spot++ = *fmt++;
+				*spot++ = '\0';
+				switch (spot[-2]) {
+				    case '%':
+					*msg++ = '%';
+					maxlen -= 1;
+					break;
+				    case 'c':
+				    case 'i':
+			            case 'd':
+			            case 'u':
+			            case 'o':
+				    case 'x':
+				    case 'X':
+					len = SDL_snprintf(msg, maxlen, tmp, error->args[argi++].value_i);
+					msg += len;
+					maxlen -= len;
+					break;
+				    case 'f':
+					len = SDL_snprintf(msg, maxlen, tmp, error->args[argi++].value_f);
+					msg += len;
+					maxlen -= len;
+					break;
+				    case 'p':
+					len = SDL_snprintf(msg, maxlen, tmp, error->args[argi++].value_ptr);
+					msg += len;
+					maxlen -= len;
+					break;
+				    case 's':
+					len = SDL_snprintf(msg, maxlen, tmp, SDL_LookupString(error->args[argi++].buf));
+					msg += len;
+					maxlen -= len;
+					break;
+				}
+			} else {
+				*msg++ = *fmt++;
+				maxlen -= 1;
+			}
+		}
+		*msg = 0;	/* NULL terminate the string */
+	}
+	return(errstr);
+}
+
+/* Available for backwards compatibility */
+char *SDL_GetError (void)
+{
+	static char errmsg[SDL_ERRBUFIZE];
+
+	return((char *)SDL_GetErrorMsg(errmsg, SDL_ERRBUFIZE));
+}
+
+void SDL_ClearError(void)
+{
+	SDL_error *error;
+
+	error = SDL_GetErrBuf();
+	error->error = 0;
+}
+
+/* Very common errors go here */
+void SDL_Error(SDL_errorcode code)
+{
+	switch (code) {
+		case SDL_ENOMEM:
+			SDL_SetError("Out of memory");
+			break;
+		case SDL_EFREAD:
+			SDL_SetError("Error reading from datastream");
+			break;
+		case SDL_EFWRITE:
+			SDL_SetError("Error writing to datastream");
+			break;
+		case SDL_EFSEEK:
+			SDL_SetError("Error seeking in datastream");
+			break;
+		default:
+			SDL_SetError("Unknown SDL error");
+			break;
+	}
+}
+
+#ifdef TEST_ERROR
+int main(int argc, char *argv[])
+{
+	char buffer[BUFSIZ+1];
+
+	SDL_SetError("Hi there!");
+	printf("Error 1: %s\n", SDL_GetError());
+	SDL_ClearError();
+	SDL_memset(buffer, '1', BUFSIZ);
+	buffer[BUFSIZ] = 0;
+	SDL_SetError("This is the error: %s (%f)", buffer, 1.0);
+	printf("Error 2: %s\n", SDL_GetError());
+	exit(0);
+}
+#endif

Externals/SDL/src/SDL_error_c.h

@@ -0,0 +1,58 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* This file defines a structure that carries language-independent
+   error messages
+*/
+
+#ifndef _SDL_error_c_h
+#define _SDL_error_c_h
+
+#define ERR_MAX_STRLEN	128
+#define ERR_MAX_ARGS	5
+
+typedef struct SDL_error {
+	/* This is a numeric value corresponding to the current error */
+	int error;
+
+	/* This is a key used to index into a language hashtable containing
+	   internationalized versions of the SDL error messages.  If the key
+	   is not in the hashtable, or no hashtable is available, the key is
+	   used directly as an error message format string.
+	*/
+	char key[ERR_MAX_STRLEN];
+
+	/* These are the arguments for the error functions */
+	int argc;
+	union {
+		void *value_ptr;
+#if 0	/* What is a character anyway?  (UNICODE issues) */
+		unsigned char value_c;
+#endif
+		int value_i;
+		double value_f;
+		char buf[ERR_MAX_STRLEN];
+	} args[ERR_MAX_ARGS];
+} SDL_error;
+
+#endif /* _SDL_error_c_h */

Externals/SDL/src/SDL_fatal.c

@@ -0,0 +1,134 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* General fatal signal handling code for SDL */
+
+#ifdef HAVE_SIGNAL_H
+
+#include <signal.h>
+
+#include "SDL.h"
+#include "SDL_fatal.h"
+
+/* This installs some signal handlers for the more common fatal signals,
+   so that if the programmer is lazy, the app doesn't die so horribly if
+   the program crashes.
+*/
+
+static void SDL_Parachute(int sig)
+{
+	signal(sig, SIG_DFL);
+	SDL_Quit();
+	raise(sig);
+}
+
+static int SDL_fatal_signals[] = {
+	SIGSEGV,
+#ifdef SIGBUS
+	SIGBUS,
+#endif
+#ifdef SIGFPE
+	SIGFPE,
+#endif
+#ifdef SIGQUIT
+	SIGQUIT,
+#endif
+	0
+};
+
+void SDL_InstallParachute(void)
+{
+	/* Set a handler for any fatal signal not already handled */
+	int i;
+#ifdef HAVE_SIGACTION
+	struct sigaction action;
+
+	for ( i=0; SDL_fatal_signals[i]; ++i ) {
+		sigaction(SDL_fatal_signals[i], NULL, &action);
+		if ( action.sa_handler == SIG_DFL ) {
+			action.sa_handler = SDL_Parachute;
+			sigaction(SDL_fatal_signals[i], &action, NULL);
+		}
+	}
+#ifdef SIGALRM
+	/* Set SIGALRM to be ignored -- necessary on Solaris */
+	sigaction(SIGALRM, NULL, &action);
+	if ( action.sa_handler == SIG_DFL ) {
+		action.sa_handler = SIG_IGN;
+		sigaction(SIGALRM, &action, NULL);
+	}
+#endif
+#else
+	void (*ohandler)(int);
+
+	for ( i=0; SDL_fatal_signals[i]; ++i ) {
+		ohandler = signal(SDL_fatal_signals[i], SDL_Parachute);
+		if ( ohandler != SIG_DFL ) {
+			signal(SDL_fatal_signals[i], ohandler);
+		}
+	}
+#endif /* HAVE_SIGACTION */
+	return;
+}
+
+void SDL_UninstallParachute(void)
+{
+	/* Remove a handler for any fatal signal handled */
+	int i;
+#ifdef HAVE_SIGACTION
+	struct sigaction action;
+
+	for ( i=0; SDL_fatal_signals[i]; ++i ) {
+		sigaction(SDL_fatal_signals[i], NULL, &action);
+		if ( action.sa_handler == SDL_Parachute ) {
+			action.sa_handler = SIG_DFL;
+			sigaction(SDL_fatal_signals[i], &action, NULL);
+		}
+	}
+#else
+	void (*ohandler)(int);
+
+	for ( i=0; SDL_fatal_signals[i]; ++i ) {
+		ohandler = signal(SDL_fatal_signals[i], SIG_DFL);
+		if ( ohandler != SDL_Parachute ) {
+			signal(SDL_fatal_signals[i], ohandler);
+		}
+	}
+#endif /* HAVE_SIGACTION */
+}
+
+#else
+
+/* No signals on this platform, nothing to do.. */
+
+void SDL_InstallParachute(void)
+{
+	return;
+}
+
+void SDL_UninstallParachute(void)
+{
+	return;
+}
+
+#endif /* HAVE_SIGNAL_H */

Externals/SDL/src/SDL_fatal.h

@@ -0,0 +1,28 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* General fatal signal handling code for SDL */
+
+extern void SDL_InstallParachute(void);
+extern void SDL_UninstallParachute(void);
+

Externals/SDL/src/audio/SDL_audio.c

@@ -0,0 +1,695 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Allow access to a raw mixing buffer */
+
+#include "SDL.h"
+#include "SDL_audio_c.h"
+#include "SDL_audiomem.h"
+#include "SDL_sysaudio.h"
+
+#ifdef __OS2__
+/* We'll need the DosSetPriority() API! */
+#define INCL_DOSPROCESS
+#include <os2.h>
+#endif
+
+/* Available audio drivers */
+static AudioBootStrap *bootstrap[] = {
+#if SDL_AUDIO_DRIVER_BSD
+	&BSD_AUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_PULSE
+	&PULSE_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_ALSA
+	&ALSA_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_OSS
+	&DSP_bootstrap,
+	&DMA_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_QNXNTO
+	&QNXNTOAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_SUNAUDIO
+	&SUNAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_DMEDIA
+	&DMEDIA_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_ARTS
+	&ARTS_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_ESD
+	&ESD_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_NAS
+	&NAS_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_DSOUND
+	&DSOUND_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_WAVEOUT
+	&WAVEOUT_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_PAUD
+	&Paud_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_BAUDIO
+	&BAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_COREAUDIO
+	&COREAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_SNDMGR
+	&SNDMGR_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_MINT
+	&MINTAUDIO_GSXB_bootstrap,
+	&MINTAUDIO_MCSN_bootstrap,
+	&MINTAUDIO_STFA_bootstrap,
+	&MINTAUDIO_XBIOS_bootstrap,
+	&MINTAUDIO_DMA8_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_DISK
+	&DISKAUD_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_DUMMY
+	&DUMMYAUD_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_DC
+	&DCAUD_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_NDS
+	&NDSAUD_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_MMEAUDIO
+	&MMEAUDIO_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_DART
+	&DART_bootstrap,
+#endif
+#if SDL_AUDIO_DRIVER_EPOCAUDIO
+	&EPOCAudio_bootstrap,
+#endif
+	NULL
+};
+SDL_AudioDevice *current_audio = NULL;
+
+/* Various local functions */
+int SDL_AudioInit(const char *driver_name);
+void SDL_AudioQuit(void);
+
+/* The general mixing thread function */
+int SDLCALL SDL_RunAudio(void *audiop)
+{
+	SDL_AudioDevice *audio = (SDL_AudioDevice *)audiop;
+	Uint8 *stream;
+	int    stream_len;
+	void  *udata;
+	void (SDLCALL *fill)(void *userdata,Uint8 *stream, int len);
+	int    silence;
+
+	/* Perform any thread setup */
+	if ( audio->ThreadInit ) {
+		audio->ThreadInit(audio);
+	}
+	audio->threadid = SDL_ThreadID();
+
+	/* Set up the mixing function */
+	fill  = audio->spec.callback;
+	udata = audio->spec.userdata;
+
+	if ( audio->convert.needed ) {
+		if ( audio->convert.src_format == AUDIO_U8 ) {
+			silence = 0x80;
+		} else {
+			silence = 0;
+		}
+		stream_len = audio->convert.len;
+	} else {
+		silence = audio->spec.silence;
+		stream_len = audio->spec.size;
+	}
+
+#ifdef __OS2__
+        /* Increase the priority of this thread to make sure that
+           the audio will be continuous all the time! */
+#ifdef USE_DOSSETPRIORITY
+        if (SDL_getenv("SDL_USE_TIMECRITICAL_AUDIO"))
+        {
+#ifdef DEBUG_BUILD
+          printf("[SDL_RunAudio] : Setting priority to TimeCritical+0! (TID%d)\n", SDL_ThreadID());
+#endif
+          DosSetPriority(PRTYS_THREAD, PRTYC_TIMECRITICAL, 0, 0);
+        }
+        else
+        {
+#ifdef DEBUG_BUILD
+          printf("[SDL_RunAudio] : Setting priority to ForegroundServer+0! (TID%d)\n", SDL_ThreadID());
+#endif
+          DosSetPriority(PRTYS_THREAD, PRTYC_FOREGROUNDSERVER, 0, 0);
+        }
+#endif
+#endif
+
+	/* Loop, filling the audio buffers */
+	while ( audio->enabled ) {
+
+		/* Fill the current buffer with sound */
+		if ( audio->convert.needed ) {
+			if ( audio->convert.buf ) {
+				stream = audio->convert.buf;
+			} else {
+				continue;
+			}
+		} else {
+			stream = audio->GetAudioBuf(audio);
+			if ( stream == NULL ) {
+				stream = audio->fake_stream;
+			}
+		}
+
+		SDL_memset(stream, silence, stream_len);
+
+		if ( ! audio->paused ) {
+			SDL_mutexP(audio->mixer_lock);
+			(*fill)(udata, stream, stream_len);
+			SDL_mutexV(audio->mixer_lock);
+		}
+
+		/* Convert the audio if necessary */
+		if ( audio->convert.needed ) {
+			SDL_ConvertAudio(&audio->convert);
+			stream = audio->GetAudioBuf(audio);
+			if ( stream == NULL ) {
+				stream = audio->fake_stream;
+			}
+			SDL_memcpy(stream, audio->convert.buf,
+			               audio->convert.len_cvt);
+		}
+
+		/* Ready current buffer for play and change current buffer */
+		if ( stream != audio->fake_stream ) {
+			audio->PlayAudio(audio);
+		}
+
+		/* Wait for an audio buffer to become available */
+		if ( stream == audio->fake_stream ) {
+			SDL_Delay((audio->spec.samples*1000)/audio->spec.freq);
+		} else {
+			audio->WaitAudio(audio);
+		}
+	}
+
+	/* Wait for the audio to drain.. */
+	if ( audio->WaitDone ) {
+		audio->WaitDone(audio);
+	}
+
+#ifdef __OS2__
+#ifdef DEBUG_BUILD
+        printf("[SDL_RunAudio] : Task exiting. (TID%d)\n", SDL_ThreadID());
+#endif
+#endif
+	return(0);
+}
+
+static void SDL_LockAudio_Default(SDL_AudioDevice *audio)
+{
+	if ( audio->thread && (SDL_ThreadID() == audio->threadid) ) {
+		return;
+	}
+	SDL_mutexP(audio->mixer_lock);
+}
+
+static void SDL_UnlockAudio_Default(SDL_AudioDevice *audio)
+{
+	if ( audio->thread && (SDL_ThreadID() == audio->threadid) ) {
+		return;
+	}
+	SDL_mutexV(audio->mixer_lock);
+}
+
+static Uint16 SDL_ParseAudioFormat(const char *string)
+{
+	Uint16 format = 0;
+
+	switch (*string) {
+	    case 'U':
+		++string;
+		format |= 0x0000;
+		break;
+	    case 'S':
+		++string;
+		format |= 0x8000;
+		break;
+	    default:
+		return 0;
+	}
+	switch (SDL_atoi(string)) {
+	    case 8:
+		string += 1;
+		format |= 8;
+		break;
+	    case 16:
+		string += 2;
+		format |= 16;
+		if ( SDL_strcmp(string, "LSB") == 0
+#if SDL_BYTEORDER == SDL_LIL_ENDIAN
+		     || SDL_strcmp(string, "SYS") == 0
+#endif
+		    ) {
+			format |= 0x0000;
+		}
+		if ( SDL_strcmp(string, "MSB") == 0
+#if SDL_BYTEORDER == SDL_BIG_ENDIAN
+		     || SDL_strcmp(string, "SYS") == 0
+#endif
+		    ) {
+			format |= 0x1000;
+		}
+		break;
+	    default:
+		return 0;
+	}
+	return format;
+}
+
+int SDL_AudioInit(const char *driver_name)
+{
+	SDL_AudioDevice *audio;
+	int i = 0, idx;
+
+	/* Check to make sure we don't overwrite 'current_audio' */
+	if ( current_audio != NULL ) {
+		SDL_AudioQuit();
+	}
+
+	/* Select the proper audio driver */
+	audio = NULL;
+	idx = 0;
+#if SDL_AUDIO_DRIVER_ESD
+	if ( (driver_name == NULL) && (SDL_getenv("ESPEAKER") != NULL) ) {
+		/* Ahem, we know that if ESPEAKER is set, user probably wants
+		   to use ESD, but don't start it if it's not already running.
+		   This probably isn't the place to do this, but... Shh! :)
+		 */
+		for ( i=0; bootstrap[i]; ++i ) {
+			if ( SDL_strcasecmp(bootstrap[i]->name, "esd") == 0 ) {
+#ifdef HAVE_PUTENV
+				const char *esd_no_spawn;
+
+				/* Don't start ESD if it's not running */
+				esd_no_spawn = getenv("ESD_NO_SPAWN");
+				if ( esd_no_spawn == NULL ) {
+					putenv("ESD_NO_SPAWN=1");
+				}
+#endif
+				if ( bootstrap[i]->available() ) {
+					audio = bootstrap[i]->create(0);
+					break;
+				}
+#ifdef HAVE_UNSETENV
+				if ( esd_no_spawn == NULL ) {
+					unsetenv("ESD_NO_SPAWN");
+				}
+#endif
+			}
+		}
+	}
+#endif /* SDL_AUDIO_DRIVER_ESD */
+	if ( audio == NULL ) {
+		if ( driver_name != NULL ) {
+#if 0	/* This will be replaced with a better driver selection API */
+			if ( SDL_strrchr(driver_name, ':') != NULL ) {
+				idx = atoi(SDL_strrchr(driver_name, ':')+1);
+			}
+#endif
+			for ( i=0; bootstrap[i]; ++i ) {
+				if (SDL_strcasecmp(bootstrap[i]->name, driver_name) == 0) {
+					if ( bootstrap[i]->available() ) {
+						audio=bootstrap[i]->create(idx);
+						break;
+					}
+				}
+			}
+		} else {
+			for ( i=0; bootstrap[i]; ++i ) {
+				if ( bootstrap[i]->available() ) {
+					audio = bootstrap[i]->create(idx);
+					if ( audio != NULL ) {
+						break;
+					}
+				}
+			}
+		}
+		if ( audio == NULL ) {
+			SDL_SetError("No available audio device");
+#if 0 /* Don't fail SDL_Init() if audio isn't available.
+         SDL_OpenAudio() will handle it at that point.  *sigh*
+       */
+			return(-1);
+#endif
+		}
+	}
+	current_audio = audio;
+	if ( current_audio ) {
+		current_audio->name = bootstrap[i]->name;
+		if ( !current_audio->LockAudio && !current_audio->UnlockAudio ) {
+			current_audio->LockAudio = SDL_LockAudio_Default;
+			current_audio->UnlockAudio = SDL_UnlockAudio_Default;
+		}
+	}
+	return(0);
+}
+
+char *SDL_AudioDriverName(char *namebuf, int maxlen)
+{
+	if ( current_audio != NULL ) {
+		SDL_strlcpy(namebuf, current_audio->name, maxlen);
+		return(namebuf);
+	}
+	return(NULL);
+}
+
+int SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained)
+{
+	SDL_AudioDevice *audio;
+	const char *env;
+
+	/* Start up the audio driver, if necessary */
+	if ( ! current_audio ) {
+		if ( (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) ||
+		     (current_audio == NULL) ) {
+			return(-1);
+		}
+	}
+	audio = current_audio;
+
+	if (audio->opened) {
+		SDL_SetError("Audio device is already opened");
+		return(-1);
+	}
+
+	/* Verify some parameters */
+	if ( desired->freq == 0 ) {
+		env = SDL_getenv("SDL_AUDIO_FREQUENCY");
+		if ( env ) {
+			desired->freq = SDL_atoi(env);
+		}
+	}
+	if ( desired->freq == 0 ) {
+		/* Pick some default audio frequency */
+		desired->freq = 22050;
+	}
+	if ( desired->format == 0 ) {
+		env = SDL_getenv("SDL_AUDIO_FORMAT");
+		if ( env ) {
+			desired->format = SDL_ParseAudioFormat(env);
+		}
+	}
+	if ( desired->format == 0 ) {
+		/* Pick some default audio format */
+		desired->format = AUDIO_S16;
+	}
+	if ( desired->channels == 0 ) {
+		env = SDL_getenv("SDL_AUDIO_CHANNELS");
+		if ( env ) {
+			desired->channels = (Uint8)SDL_atoi(env);
+		}
+	}
+	if ( desired->channels == 0 ) {
+		/* Pick a default number of channels */
+		desired->channels = 2;
+	}
+	switch ( desired->channels ) {
+	    case 1:	/* Mono */
+	    case 2:	/* Stereo */
+	    case 4:	/* surround */
+	    case 6:	/* surround with center and lfe */
+		break;
+	    default:
+		SDL_SetError("1 (mono) and 2 (stereo) channels supported");
+		return(-1);
+	}
+	if ( desired->samples == 0 ) {
+		env = SDL_getenv("SDL_AUDIO_SAMPLES");
+		if ( env ) {
+			desired->samples = (Uint16)SDL_atoi(env);
+		}
+	}
+	if ( desired->samples == 0 ) {
+		/* Pick a default of ~46 ms at desired frequency */
+		int samples = (desired->freq / 1000) * 46;
+		int power2 = 1;
+		while ( power2 < samples ) {
+			power2 *= 2;
+		}
+		desired->samples = power2;
+	}
+	if ( desired->callback == NULL ) {
+		SDL_SetError("SDL_OpenAudio() passed a NULL callback");
+		return(-1);
+	}
+
+#if SDL_THREADS_DISABLED
+	/* Uses interrupt driven audio, without thread */
+#else
+	/* Create a semaphore for locking the sound buffers */
+	audio->mixer_lock = SDL_CreateMutex();
+	if ( audio->mixer_lock == NULL ) {
+		SDL_SetError("Couldn't create mixer lock");
+		SDL_CloseAudio();
+		return(-1);
+	}
+#endif /* SDL_THREADS_DISABLED */
+
+	/* Calculate the silence and size of the audio specification */
+	SDL_CalculateAudioSpec(desired);
+
+	/* Open the audio subsystem */
+	SDL_memcpy(&audio->spec, desired, sizeof(audio->spec));
+	audio->convert.needed = 0;
+	audio->enabled = 1;
+	audio->paused  = 1;
+
+	audio->opened = audio->OpenAudio(audio, &audio->spec)+1;
+
+	if ( ! audio->opened ) {
+		SDL_CloseAudio();
+		return(-1);
+	}
+
+	/* If the audio driver changes the buffer size, accept it */
+	if ( audio->spec.samples != desired->samples ) {
+		desired->samples = audio->spec.samples;
+		SDL_CalculateAudioSpec(desired);
+	}
+
+	/* Allocate a fake audio memory buffer */
+	audio->fake_stream = SDL_AllocAudioMem(audio->spec.size);
+	if ( audio->fake_stream == NULL ) {
+		SDL_CloseAudio();
+		SDL_OutOfMemory();
+		return(-1);
+	}
+
+	/* See if we need to do any conversion */
+	if ( obtained != NULL ) {
+		SDL_memcpy(obtained, &audio->spec, sizeof(audio->spec));
+	} else if ( desired->freq != audio->spec.freq ||
+                    desired->format != audio->spec.format ||
+	            desired->channels != audio->spec.channels ) {
+		/* Build an audio conversion block */
+		if ( SDL_BuildAudioCVT(&audio->convert,
+			desired->format, desired->channels,
+					desired->freq,
+			audio->spec.format, audio->spec.channels,
+					audio->spec.freq) < 0 ) {
+			SDL_CloseAudio();
+			return(-1);
+		}
+		if ( audio->convert.needed ) {
+			audio->convert.len = (int) ( ((double) audio->spec.size) /
+                                          audio->convert.len_ratio );
+			audio->convert.buf =(Uint8 *)SDL_AllocAudioMem(
+			   audio->convert.len*audio->convert.len_mult);
+			if ( audio->convert.buf == NULL ) {
+				SDL_CloseAudio();
+				SDL_OutOfMemory();
+				return(-1);
+			}
+		}
+	}
+
+	/* Start the audio thread if necessary */
+	switch (audio->opened) {
+		case  1:
+			/* Start the audio thread */
+#if (defined(__WIN32__) && !defined(_WIN32_WCE)) && !defined(HAVE_LIBC) && !defined(__SYMBIAN32__)
+#undef SDL_CreateThread
+			audio->thread = SDL_CreateThread(SDL_RunAudio, audio, NULL, NULL);
+#else
+			audio->thread = SDL_CreateThread(SDL_RunAudio, audio);
+#endif
+			if ( audio->thread == NULL ) {
+				SDL_CloseAudio();
+				SDL_SetError("Couldn't create audio thread");
+				return(-1);
+			}
+			break;
+
+		default:
+			/* The audio is now playing */
+			break;
+	}
+
+	return(0);
+}
+
+SDL_audiostatus SDL_GetAudioStatus(void)
+{
+	SDL_AudioDevice *audio = current_audio;
+	SDL_audiostatus status;
+
+	status = SDL_AUDIO_STOPPED;
+	if ( audio && audio->enabled ) {
+		if ( audio->paused ) {
+			status = SDL_AUDIO_PAUSED;
+		} else {
+			status = SDL_AUDIO_PLAYING;
+		}
+	}
+	return(status);
+}
+
+void SDL_PauseAudio (int pause_on)
+{
+	SDL_AudioDevice *audio = current_audio;
+
+	if ( audio ) {
+		audio->paused = pause_on;
+	}
+}
+
+void SDL_LockAudio (void)
+{
+	SDL_AudioDevice *audio = current_audio;
+
+	/* Obtain a lock on the mixing buffers */
+	if ( audio && audio->LockAudio ) {
+		audio->LockAudio(audio);
+	}
+}
+
+void SDL_UnlockAudio (void)
+{
+	SDL_AudioDevice *audio = current_audio;
+
+	/* Release lock on the mixing buffers */
+	if ( audio && audio->UnlockAudio ) {
+		audio->UnlockAudio(audio);
+	}
+}
+
+void SDL_CloseAudio (void)
+{
+	SDL_QuitSubSystem(SDL_INIT_AUDIO);
+}
+
+void SDL_AudioQuit(void)
+{
+	SDL_AudioDevice *audio = current_audio;
+
+	if ( audio ) {
+		audio->enabled = 0;
+		if ( audio->thread != NULL ) {
+			SDL_WaitThread(audio->thread, NULL);
+		}
+		if ( audio->mixer_lock != NULL ) {
+			SDL_DestroyMutex(audio->mixer_lock);
+		}
+		if ( audio->fake_stream != NULL ) {
+			SDL_FreeAudioMem(audio->fake_stream);
+		}
+		if ( audio->convert.needed ) {
+			SDL_FreeAudioMem(audio->convert.buf);
+
+		}
+		if ( audio->opened ) {
+			audio->CloseAudio(audio);
+			audio->opened = 0;
+		}
+		/* Free the driver data */
+		audio->free(audio);
+		current_audio = NULL;
+	}
+}
+
+#define NUM_FORMATS	6
+static int format_idx;
+static int format_idx_sub;
+static Uint16 format_list[NUM_FORMATS][NUM_FORMATS] = {
+ { AUDIO_U8, AUDIO_S8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB },
+ { AUDIO_S8, AUDIO_U8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB },
+ { AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_U8, AUDIO_S8 },
+ { AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_U8, AUDIO_S8 },
+ { AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U8, AUDIO_S8 },
+ { AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_U8, AUDIO_S8 },
+};
+
+Uint16 SDL_FirstAudioFormat(Uint16 format)
+{
+	for ( format_idx=0; format_idx < NUM_FORMATS; ++format_idx ) {
+		if ( format_list[format_idx][0] == format ) {
+			break;
+		}
+	}
+	format_idx_sub = 0;
+	return(SDL_NextAudioFormat());
+}
+
+Uint16 SDL_NextAudioFormat(void)
+{
+	if ( (format_idx == NUM_FORMATS) || (format_idx_sub == NUM_FORMATS) ) {
+		return(0);
+	}
+	return(format_list[format_idx][format_idx_sub++]);
+}
+
+void SDL_CalculateAudioSpec(SDL_AudioSpec *spec)
+{
+	switch (spec->format) {
+		case AUDIO_U8:
+			spec->silence = 0x80;
+			break;
+		default:
+			spec->silence = 0x00;
+			break;
+	}
+	spec->size = (spec->format&0xFF)/8;
+	spec->size *= spec->channels;
+	spec->size *= spec->samples;
+}

Externals/SDL/src/audio/SDL_audio_c.h

@@ -0,0 +1,34 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Functions and variables exported from SDL_audio.c for SDL_sysaudio.c */
+
+/* Functions to get a list of "close" audio formats */
+extern Uint16 SDL_FirstAudioFormat(Uint16 format);
+extern Uint16 SDL_NextAudioFormat(void);
+
+/* Function to calculate the size and silence for a SDL_AudioSpec */
+extern void SDL_CalculateAudioSpec(SDL_AudioSpec *spec);
+
+/* The actual mixing thread function */
+extern int SDLCALL SDL_RunAudio(void *audiop);

Externals/SDL/src/audio/SDL_audiodev.c

@@ -0,0 +1,179 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Get the name of the audio device we use for output */
+
+#if SDL_AUDIO_DRIVER_BSD || SDL_AUDIO_DRIVER_OSS || SDL_AUDIO_DRIVER_SUNAUDIO
+
+#include <fcntl.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+
+#include "SDL_stdinc.h"
+#include "SDL_audiodev_c.h"
+
+#ifndef _PATH_DEV_DSP
+#if defined(__NETBSD__) || defined(__OPENBSD__)
+#define _PATH_DEV_DSP  "/dev/audio"
+#else
+#define _PATH_DEV_DSP  "/dev/dsp"
+#endif
+#endif
+#ifndef _PATH_DEV_DSP24
+#define _PATH_DEV_DSP24	"/dev/sound/dsp"
+#endif
+#ifndef _PATH_DEV_AUDIO
+#define _PATH_DEV_AUDIO	"/dev/audio"
+#endif
+
+
+int SDL_OpenAudioPath(char *path, int maxlen, int flags, int classic)
+{
+	const char *audiodev;
+	int audio_fd;
+	char audiopath[1024];
+
+	/* Figure out what our audio device is */
+	if ( ((audiodev=SDL_getenv("SDL_PATH_DSP")) == NULL) &&
+	     ((audiodev=SDL_getenv("AUDIODEV")) == NULL) ) {
+		if ( classic ) {
+			audiodev = _PATH_DEV_AUDIO;
+		} else {
+			struct stat sb;
+
+			/* Added support for /dev/sound/\* in Linux 2.4 */
+			if ( ((stat("/dev/sound", &sb) == 0) && S_ISDIR(sb.st_mode)) &&
+				 ((stat(_PATH_DEV_DSP24, &sb) == 0) && S_ISCHR(sb.st_mode)) ) {
+				audiodev = _PATH_DEV_DSP24;
+			} else {
+				audiodev = _PATH_DEV_DSP;
+			}
+		}
+	}
+	audio_fd = open(audiodev, flags, 0);
+
+	/* If the first open fails, look for other devices */
+	if ( (audio_fd < 0) && (SDL_strlen(audiodev) < (sizeof(audiopath)-3)) ) {
+		int exists, instance;
+		struct stat sb;
+
+		instance = 1;
+		do { /* Don't use errno ENOENT - it may not be thread-safe */
+			SDL_snprintf(audiopath, SDL_arraysize(audiopath),
+			             "%s%d", audiodev, instance++);
+			exists = 0;
+			if ( stat(audiopath, &sb) == 0 ) {
+				exists = 1;
+				audio_fd = open(audiopath, flags, 0); 
+			}
+		} while ( exists && (audio_fd < 0) );
+		audiodev = audiopath;
+	}
+	if ( path != NULL ) {
+		SDL_strlcpy(path, audiodev, maxlen);
+		path[maxlen-1] = '\0';
+	}
+	return(audio_fd);
+}
+
+#elif SDL_AUDIO_DRIVER_PAUD
+
+/* Get the name of the audio device we use for output */
+
+#include <sys/types.h>
+#include <sys/stat.h>
+
+#include "SDL_stdinc.h"
+#include "SDL_audiodev_c.h"
+
+#ifndef _PATH_DEV_DSP
+#define _PATH_DEV_DSP	"/dev/%caud%c/%c"
+#endif
+
+char devsettings[][3] =
+{
+    { 'p', '0', '1' }, { 'p', '0', '2' }, { 'p', '0', '3' }, { 'p', '0', '4' },
+    { 'p', '1', '1' }, { 'p', '1', '2' }, { 'p', '1', '3' }, { 'p', '1', '4' },
+    { 'p', '2', '1' }, { 'p', '2', '2' }, { 'p', '2', '3' }, { 'p', '2', '4' },
+    { 'p', '3', '1' }, { 'p', '3', '2' }, { 'p', '3', '3' }, { 'p', '3', '4' },
+    { 'b', '0', '1' }, { 'b', '0', '2' }, { 'b', '0', '3' }, { 'b', '0', '4' },
+    { 'b', '1', '1' }, { 'b', '1', '2' }, { 'b', '1', '3' }, { 'b', '1', '4' },
+    { 'b', '2', '1' }, { 'b', '2', '2' }, { 'b', '2', '3' }, { 'b', '2', '4' },
+    { 'b', '3', '1' }, { 'b', '3', '2' }, { 'b', '3', '3' }, { 'b', '3', '4' },
+    { '\0', '\0', '\0' }
+};
+
+static int OpenUserDefinedDevice(char *path, int maxlen, int flags)
+{
+	const char *audiodev;
+	int  audio_fd;
+
+	/* Figure out what our audio device is */
+	if ((audiodev=SDL_getenv("SDL_PATH_DSP")) == NULL) {
+	    audiodev=SDL_getenv("AUDIODEV");
+	}
+	if ( audiodev == NULL ) {
+	    return -1;
+	}
+	audio_fd = open(audiodev, flags, 0);
+	if ( path != NULL ) {
+		SDL_strlcpy(path, audiodev, maxlen);
+		path[maxlen-1] = '\0';
+	}
+	return audio_fd;
+}
+
+int SDL_OpenAudioPath(char *path, int maxlen, int flags, int classic)
+{
+    struct stat sb;
+    int         audio_fd;
+    char        audiopath[1024];
+    int         cycle;
+
+    audio_fd = OpenUserDefinedDevice(path,maxlen,flags);
+    if ( audio_fd != -1 ) {
+        return audio_fd;
+    }
+
+    cycle    = 0;
+    while( devsettings[cycle][0] != '\0' ) {
+        SDL_snprintf( audiopath, SDL_arraysize(audiopath),
+                 _PATH_DEV_DSP,
+                 devsettings[cycle][0],
+                 devsettings[cycle][1],
+                 devsettings[cycle][2]);
+
+	if ( stat(audiopath, &sb) == 0 ) {
+	    audio_fd = open(audiopath, flags, 0);
+	    if ( audio_fd > 0 ) {
+		if ( path != NULL ) {
+		    SDL_strlcpy( path, audiopath, maxlen );
+		}
+	        return audio_fd;
+	    }
+	}
+    }
+    return -1;
+}
+
+#endif /* Audio driver selection */

Externals/SDL/src/audio/SDL_audiodev_c.h

@@ -0,0 +1,26 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Open the audio device, storing the pathname in 'path'  */
+extern int SDL_OpenAudioPath(char *path, int maxlen, int flags, int classic);
+

Externals/SDL/src/audio/SDL_audiomem.h

@@ -0,0 +1,25 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#define SDL_AllocAudioMem	SDL_malloc
+#define SDL_FreeAudioMem	SDL_free

Externals/SDL/src/audio/SDL_mixer.c

@@ -0,0 +1,264 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* This provides the default mixing callback for the SDL audio routines */
+
+#include "SDL_cpuinfo.h"
+#include "SDL_timer.h"
+#include "SDL_audio.h"
+#include "SDL_sysaudio.h"
+#include "SDL_mixer_MMX.h"
+#include "SDL_mixer_MMX_VC.h"
+#include "SDL_mixer_m68k.h"
+
+/* This table is used to add two sound values together and pin
+ * the value to avoid overflow.  (used with permission from ARDI)
+ * Changed to use 0xFE instead of 0xFF for better sound quality.
+ */
+static const Uint8 mix8[] =
+{
+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x02, 0x03,
+  0x04, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0A, 0x0B, 0x0C, 0x0D, 0x0E,
+  0x0F, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0x17, 0x18, 0x19,
+  0x1A, 0x1B, 0x1C, 0x1D, 0x1E, 0x1F, 0x20, 0x21, 0x22, 0x23, 0x24,
+  0x25, 0x26, 0x27, 0x28, 0x29, 0x2A, 0x2B, 0x2C, 0x2D, 0x2E, 0x2F,
+  0x30, 0x31, 0x32, 0x33, 0x34, 0x35, 0x36, 0x37, 0x38, 0x39, 0x3A,
+  0x3B, 0x3C, 0x3D, 0x3E, 0x3F, 0x40, 0x41, 0x42, 0x43, 0x44, 0x45,
+  0x46, 0x47, 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, 0x50,
+  0x51, 0x52, 0x53, 0x54, 0x55, 0x56, 0x57, 0x58, 0x59, 0x5A, 0x5B,
+  0x5C, 0x5D, 0x5E, 0x5F, 0x60, 0x61, 0x62, 0x63, 0x64, 0x65, 0x66,
+  0x67, 0x68, 0x69, 0x6A, 0x6B, 0x6C, 0x6D, 0x6E, 0x6F, 0x70, 0x71,
+  0x72, 0x73, 0x74, 0x75, 0x76, 0x77, 0x78, 0x79, 0x7A, 0x7B, 0x7C,
+  0x7D, 0x7E, 0x7F, 0x80, 0x81, 0x82, 0x83, 0x84, 0x85, 0x86, 0x87,
+  0x88, 0x89, 0x8A, 0x8B, 0x8C, 0x8D, 0x8E, 0x8F, 0x90, 0x91, 0x92,
+  0x93, 0x94, 0x95, 0x96, 0x97, 0x98, 0x99, 0x9A, 0x9B, 0x9C, 0x9D,
+  0x9E, 0x9F, 0xA0, 0xA1, 0xA2, 0xA3, 0xA4, 0xA5, 0xA6, 0xA7, 0xA8,
+  0xA9, 0xAA, 0xAB, 0xAC, 0xAD, 0xAE, 0xAF, 0xB0, 0xB1, 0xB2, 0xB3,
+  0xB4, 0xB5, 0xB6, 0xB7, 0xB8, 0xB9, 0xBA, 0xBB, 0xBC, 0xBD, 0xBE,
+  0xBF, 0xC0, 0xC1, 0xC2, 0xC3, 0xC4, 0xC5, 0xC6, 0xC7, 0xC8, 0xC9,
+  0xCA, 0xCB, 0xCC, 0xCD, 0xCE, 0xCF, 0xD0, 0xD1, 0xD2, 0xD3, 0xD4,
+  0xD5, 0xD6, 0xD7, 0xD8, 0xD9, 0xDA, 0xDB, 0xDC, 0xDD, 0xDE, 0xDF,
+  0xE0, 0xE1, 0xE2, 0xE3, 0xE4, 0xE5, 0xE6, 0xE7, 0xE8, 0xE9, 0xEA,
+  0xEB, 0xEC, 0xED, 0xEE, 0xEF, 0xF0, 0xF1, 0xF2, 0xF3, 0xF4, 0xF5,
+  0xF6, 0xF7, 0xF8, 0xF9, 0xFA, 0xFB, 0xFC, 0xFD, 0xFE, 0xFE, 0xFE,
+  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
+  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
+  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
+  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
+  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
+  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
+  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
+  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
+  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
+  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
+  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE,
+  0xFE, 0xFE, 0xFE, 0xFE, 0xFE, 0xFE
+};
+
+/* The volume ranges from 0 - 128 */
+#define ADJUST_VOLUME(s, v)	(s = (s*v)/SDL_MIX_MAXVOLUME)
+#define ADJUST_VOLUME_U8(s, v)	(s = (((s-128)*v)/SDL_MIX_MAXVOLUME)+128)
+
+void SDL_MixAudio (Uint8 *dst, const Uint8 *src, Uint32 len, int volume)
+{
+	Uint16 format;
+
+	if ( volume == 0 ) {
+		return;
+	}
+	/* Mix the user-level audio format */
+	if ( current_audio ) {
+		if ( current_audio->convert.needed ) {
+			format = current_audio->convert.src_format;
+		} else {
+			format = current_audio->spec.format;
+		}
+	} else {
+  		/* HACK HACK HACK */
+		format = AUDIO_S16;
+	}
+	switch (format) {
+
+		case AUDIO_U8: {
+#if defined(__GNUC__) && defined(__M68000__) && defined(SDL_ASSEMBLY_ROUTINES)
+			SDL_MixAudio_m68k_U8((char*)dst,(char*)src,(unsigned long)len,(long)volume,(char *)mix8);
+#else
+			Uint8 src_sample;
+
+			while ( len-- ) {
+				src_sample = *src;
+				ADJUST_VOLUME_U8(src_sample, volume);
+				*dst = mix8[*dst+src_sample];
+				++dst;
+				++src;
+			}
+#endif
+		}
+		break;
+
+		case AUDIO_S8: {
+#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */
+#if defined(__GNUC__) && defined(__i386__) && defined(SDL_ASSEMBLY_ROUTINES)
+			if (SDL_HasMMX())
+			{
+				SDL_MixAudio_MMX_S8((char*)dst,(char*)src,(unsigned int)len,(int)volume);
+			}
+			else
+#elif ((defined(_MSC_VER) && defined(_M_IX86)) || defined(__WATCOMC__)) && defined(SDL_ASSEMBLY_ROUTINES)
+			if (SDL_HasMMX())
+			{
+				SDL_MixAudio_MMX_S8_VC((char*)dst,(char*)src,(unsigned int)len,(int)volume);
+			}
+			else
+#endif
+#endif
+
+#if defined(__GNUC__) && defined(__M68000__) && defined(SDL_ASSEMBLY_ROUTINES)
+			SDL_MixAudio_m68k_S8((char*)dst,(char*)src,(unsigned long)len,(long)volume);
+#else
+			{
+			Sint8 *dst8, *src8;
+			Sint8 src_sample;
+			int dst_sample;
+			const int max_audioval = ((1<<(8-1))-1);
+			const int min_audioval = -(1<<(8-1));
+
+			src8 = (Sint8 *)src;
+			dst8 = (Sint8 *)dst;
+			while ( len-- ) {
+				src_sample = *src8;
+				ADJUST_VOLUME(src_sample, volume);
+				dst_sample = *dst8 + src_sample;
+				if ( dst_sample > max_audioval ) {
+					*dst8 = max_audioval;
+				} else
+				if ( dst_sample < min_audioval ) {
+					*dst8 = min_audioval;
+				} else {
+					*dst8 = dst_sample;
+				}
+				++dst8;
+				++src8;
+			}
+			}
+#endif
+		}
+		break;
+
+		case AUDIO_S16LSB: {
+#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */
+#if defined(__GNUC__) && defined(__i386__) && defined(SDL_ASSEMBLY_ROUTINES)
+			if (SDL_HasMMX())
+			{
+				SDL_MixAudio_MMX_S16((char*)dst,(char*)src,(unsigned int)len,(int)volume);
+			}
+                        else
+#elif ((defined(_MSC_VER) && defined(_M_IX86)) || defined(__WATCOMC__)) && defined(SDL_ASSEMBLY_ROUTINES)
+			if (SDL_HasMMX())
+			{
+				SDL_MixAudio_MMX_S16_VC((char*)dst,(char*)src,(unsigned int)len,(int)volume);
+			}
+			else
+#endif
+#endif
+
+#if defined(__GNUC__) && defined(__M68000__) && defined(SDL_ASSEMBLY_ROUTINES)
+			SDL_MixAudio_m68k_S16LSB((short*)dst,(short*)src,(unsigned long)len,(long)volume);
+#else
+			{
+			Sint16 src1, src2;
+			int dst_sample;
+			const int max_audioval = ((1<<(16-1))-1);
+			const int min_audioval = -(1<<(16-1));
+
+			len /= 2;
+			while ( len-- ) {
+				src1 = ((src[1])<<8|src[0]);
+				ADJUST_VOLUME(src1, volume);
+				src2 = ((dst[1])<<8|dst[0]);
+				src += 2;
+				dst_sample = src1+src2;
+				if ( dst_sample > max_audioval ) {
+					dst_sample = max_audioval;
+				} else
+				if ( dst_sample < min_audioval ) {
+					dst_sample = min_audioval;
+				}
+				dst[0] = dst_sample&0xFF;
+				dst_sample >>= 8;
+				dst[1] = dst_sample&0xFF;
+				dst += 2;
+			}
+			}
+#endif
+		}
+		break;
+
+		case AUDIO_S16MSB: {
+#if defined(__GNUC__) && defined(__M68000__) && defined(SDL_ASSEMBLY_ROUTINES)
+			SDL_MixAudio_m68k_S16MSB((short*)dst,(short*)src,(unsigned long)len,(long)volume);
+#else
+			Sint16 src1, src2;
+			int dst_sample;
+			const int max_audioval = ((1<<(16-1))-1);
+			const int min_audioval = -(1<<(16-1));
+
+			len /= 2;
+			while ( len-- ) {
+				src1 = ((src[0])<<8|src[1]);
+				ADJUST_VOLUME(src1, volume);
+				src2 = ((dst[0])<<8|dst[1]);
+				src += 2;
+				dst_sample = src1+src2;
+				if ( dst_sample > max_audioval ) {
+					dst_sample = max_audioval;
+				} else
+				if ( dst_sample < min_audioval ) {
+					dst_sample = min_audioval;
+				}
+				dst[1] = dst_sample&0xFF;
+				dst_sample >>= 8;
+				dst[0] = dst_sample&0xFF;
+				dst += 2;
+			}
+#endif
+		}
+		break;
+
+		default: /* If this happens... FIXME! */
+			SDL_SetError("SDL_MixAudio(): unknown audio format");
+			return;
+	}
+}
+

Externals/SDL/src/audio/SDL_mixer_MMX.c

@@ -0,0 +1,207 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/*
+    MMX assembler version of SDL_MixAudio for signed little endian 16 bit samples and signed 8 bit samples
+    Copyright 2002 Stephane Marchesin (stephane.marchesin@wanadoo.fr)
+    This code is licensed under the LGPL (see COPYING for details)
+ 
+    Assumes buffer size in bytes is a multiple of 16
+    Assumes SDL_MIX_MAXVOLUME = 128
+*/
+
+
+/***********************************************
+*   Mixing for 16 bit signed buffers
+***********************************************/
+
+#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */
+#if defined(__GNUC__) && defined(__i386__) && defined(SDL_ASSEMBLY_ROUTINES)
+void SDL_MixAudio_MMX_S16(char* dst,char* src,unsigned int size,int volume)
+{
+    __asm__ __volatile__ (
+
+"	movl %3,%%eax\n"	/* eax = volume */
+
+"	movl %2,%%edx\n"	/* edx = size */
+
+"	shrl $4,%%edx\n"	/* process 16 bytes per iteration = 8 samples */
+
+"	jz .endS16\n"
+
+"	pxor %%mm0,%%mm0\n"
+
+"	movd %%eax,%%mm0\n"
+"	movq %%mm0,%%mm1\n"
+"	psllq $16,%%mm0\n"
+"	por %%mm1,%%mm0\n"
+"	psllq $16,%%mm0\n"
+"	por %%mm1,%%mm0\n"
+"	psllq $16,%%mm0\n"
+"	por %%mm1,%%mm0\n"		/* mm0 = vol|vol|vol|vol */
+
+".align 8\n"
+"	.mixloopS16:\n"
+
+"	movq (%1),%%mm1\n" /* mm1 = a|b|c|d */
+
+"	movq %%mm1,%%mm2\n" /* mm2 = a|b|c|d */
+
+"	movq 8(%1),%%mm4\n" /* mm4 = e|f|g|h */
+
+	/* pr<70> charger le buffer dst dans mm7 */
+"	movq (%0),%%mm7\n" /* mm7 = dst[0] */
+
+	/* multiplier par le volume */
+"	pmullw %%mm0,%%mm1\n" /* mm1 = l(a*v)|l(b*v)|l(c*v)|l(d*v) */
+
+"	pmulhw %%mm0,%%mm2\n" /* mm2 = h(a*v)|h(b*v)|h(c*v)|h(d*v) */
+"	movq %%mm4,%%mm5\n" /* mm5 = e|f|g|h */
+
+"	pmullw %%mm0,%%mm4\n" /* mm4 = l(e*v)|l(f*v)|l(g*v)|l(h*v) */
+
+"	pmulhw %%mm0,%%mm5\n" /* mm5 = h(e*v)|h(f*v)|h(g*v)|h(h*v) */
+"	movq %%mm1,%%mm3\n" /* mm3 = l(a*v)|l(b*v)|l(c*v)|l(d*v) */
+
+"	punpckhwd %%mm2,%%mm1\n" /* mm1 = a*v|b*v */
+
+"	movq %%mm4,%%mm6\n" /* mm6 = l(e*v)|l(f*v)|l(g*v)|l(h*v) */
+"	punpcklwd %%mm2,%%mm3\n" /* mm3 = c*v|d*v */
+
+"	punpckhwd %%mm5,%%mm4\n" /* mm4 = e*f|f*v */
+
+"	punpcklwd %%mm5,%%mm6\n" /* mm6 = g*v|h*v */
+
+	/* pr<70> charger le buffer dst dans mm5 */
+"	movq 8(%0),%%mm5\n" /* mm5 = dst[1] */
+
+	/* diviser par 128 */
+"	psrad $7,%%mm1\n" /* mm1 = a*v/128|b*v/128 , 128 = SDL_MIX_MAXVOLUME */
+"	add $16,%1\n"
+
+"	psrad $7,%%mm3\n" /* mm3 = c*v/128|d*v/128 */
+
+"	psrad $7,%%mm4\n" /* mm4 = e*v/128|f*v/128 */
+
+	/* mm1 = le sample avec le volume modifi<66> */
+"	packssdw %%mm1,%%mm3\n" /* mm3 = s(a*v|b*v|c*v|d*v) */
+
+"	psrad $7,%%mm6\n" /* mm6= g*v/128|h*v/128 */
+"	paddsw %%mm7,%%mm3\n" /* mm3 = adjust_volume(src)+dst */
+
+	/* mm4 = le sample avec le volume modifi<66> */
+"	packssdw %%mm4,%%mm6\n" /* mm6 = s(e*v|f*v|g*v|h*v) */
+"	movq %%mm3,(%0)\n"
+
+"	paddsw %%mm5,%%mm6\n" /* mm6 = adjust_volume(src)+dst */
+
+"	movq %%mm6,8(%0)\n"
+
+"	add $16,%0\n"
+
+"	dec %%edx\n"
+
+"	jnz .mixloopS16\n"
+
+"	emms\n"
+
+".endS16:\n"
+	 :
+	 : "r" (dst), "r"(src),"m"(size),
+	 "m"(volume)
+	 : "eax","edx","memory"
+	 );
+}
+
+
+
+/*////////////////////////////////////////////// */
+/* Mixing for 8 bit signed buffers */
+/*////////////////////////////////////////////// */
+
+void SDL_MixAudio_MMX_S8(char* dst,char* src,unsigned int size,int volume)
+{
+    __asm__ __volatile__ (
+
+"	movl %3,%%eax\n"	/* eax = volume */
+
+"	movd %%eax,%%mm0\n"
+"	movq %%mm0,%%mm1\n"
+"	psllq $16,%%mm0\n"
+"	por %%mm1,%%mm0\n"
+"	psllq $16,%%mm0\n"
+"	por %%mm1,%%mm0\n"
+"	psllq $16,%%mm0\n"
+"	por %%mm1,%%mm0\n"
+
+"	movl %2,%%edx\n"	/* edx = size */
+"	shr $3,%%edx\n"	/* process 8 bytes per iteration = 8 samples */
+
+"	cmp $0,%%edx\n"
+"	je .endS8\n"
+
+".align 8\n"
+"	.mixloopS8:\n"
+
+"	pxor %%mm2,%%mm2\n"		/* mm2 = 0 */
+"	movq (%1),%%mm1\n"	/* mm1 = a|b|c|d|e|f|g|h */
+
+"	movq %%mm1,%%mm3\n" 	/* mm3 = a|b|c|d|e|f|g|h */
+
+	/* on va faire le "sign extension" en faisant un cmp avec 0 qui retourne 1 si <0, 0 si >0 */
+"	pcmpgtb %%mm1,%%mm2\n"	/* mm2 = 11111111|00000000|00000000.... */
+
+"	punpckhbw %%mm2,%%mm1\n"	/* mm1 = 0|a|0|b|0|c|0|d */
+
+"	punpcklbw %%mm2,%%mm3\n"	/* mm3 = 0|e|0|f|0|g|0|h */
+"	movq (%0),%%mm2\n"	/* mm2 = destination */
+
+"	pmullw %%mm0,%%mm1\n"	/* mm1 = v*a|v*b|v*c|v*d */
+"	add $8,%1\n"
+
+"	pmullw %%mm0,%%mm3\n"	/* mm3 = v*e|v*f|v*g|v*h */
+"	psraw $7,%%mm1\n"		/* mm1 = v*a/128|v*b/128|v*c/128|v*d/128  */
+
+"	psraw $7,%%mm3\n"		/* mm3 = v*e/128|v*f/128|v*g/128|v*h/128 */
+
+"	packsswb %%mm1,%%mm3\n"	/* mm1 = v*a/128|v*b/128|v*c/128|v*d/128|v*e/128|v*f/128|v*g/128|v*h/128 */
+
+"	paddsb %%mm2,%%mm3\n"	/* add to destination buffer */
+
+"	movq %%mm3,(%0)\n"	/* store back to ram */
+"	add $8,%0\n"
+
+"	dec %%edx\n"
+
+"	jnz .mixloopS8\n"
+
+".endS8:\n"
+"	emms\n"
+	 :
+	 : "r" (dst), "r"(src),"m"(size),
+	 "m"(volume)
+	 : "eax","edx","memory"
+	 );
+}
+#endif
+#endif

Externals/SDL/src/audio/SDL_mixer_MMX.h

@@ -0,0 +1,17 @@
+/*
+    headers for MMX assembler version of SDL_MixAudio
+    Copyright 2002 Stephane Marchesin (stephane.marchesin@wanadoo.fr)
+    This code is licensed under the LGPL (see COPYING for details)
+
+    Assumes buffer size in bytes is a multiple of 16
+    Assumes SDL_MIX_MAXVOLUME = 128
+*/
+#include "SDL_config.h"
+
+#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */
+#if defined(__GNUC__) && defined(__i386__) && defined(SDL_ASSEMBLY_ROUTINES)
+void SDL_MixAudio_MMX_S16(char* ,char* ,unsigned int ,int );
+void SDL_MixAudio_MMX_S8(char* ,char* ,unsigned int ,int );
+#endif
+#endif
+

Externals/SDL/src/audio/SDL_mixer_MMX_VC.c

@@ -0,0 +1,183 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#include "SDL_mixer_MMX_VC.h"
+
+#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */
+#if ((defined(_MSC_VER) && defined(_M_IX86)) || defined(__WATCOMC__)) && defined(SDL_ASSEMBLY_ROUTINES)
+// MMX assembler version of SDL_MixAudio for signed little endian 16 bit samples and signed 8 bit samples
+// Copyright 2002 Stephane Marchesin (stephane.marchesin@wanadoo.fr)
+// Converted to Intel ASM notation by Cth
+// This code is licensed under the LGPL (see COPYING for details)
+// 
+// Assumes buffer size in bytes is a multiple of 16
+// Assumes SDL_MIX_MAXVOLUME = 128
+
+
+////////////////////////////////////////////////
+// Mixing for 16 bit signed buffers
+////////////////////////////////////////////////
+
+void SDL_MixAudio_MMX_S16_VC(char* dst,char* src,unsigned int nSize,int volume)
+{
+	__asm
+	{
+
+		push	edi
+		push	esi
+		push	ebx
+		
+		mov		edi, dst		// edi = dst
+		mov		esi, src		// esi = src
+		mov		eax, volume		// eax = volume
+		mov		ebx, nSize		// ebx = size
+		shr		ebx, 4			// process 16 bytes per iteration = 8 samples
+		jz		endS16
+		
+		pxor	mm0, mm0
+		movd	mm0, eax		//%%eax,%%mm0
+		movq	mm1, mm0		//%%mm0,%%mm1
+		psllq	mm0, 16			//$16,%%mm0
+		por		mm0, mm1		//%%mm1,%%mm0
+		psllq	mm0, 16			//$16,%%mm0
+		por		mm0, mm1		//%%mm1,%%mm0
+		psllq	mm0, 16			//$16,%%mm0
+		por		mm0, mm1		//%%mm1,%%mm0			// mm0 = vol|vol|vol|vol
+
+		#ifndef __WATCOMC__
+		align	16
+		#endif
+mixloopS16:
+		movq	mm1, [esi]		//(%%esi),%%mm1\n" // mm1 = a|b|c|d
+		movq	mm2, mm1		//%%mm1,%%mm2\n" // mm2 = a|b|c|d
+		movq	mm4, [esi + 8]	//8(%%esi),%%mm4\n" // mm4 = e|f|g|h
+		// pre charger le buffer dst dans mm7
+		movq	mm7, [edi]		//(%%edi),%%mm7\n" // mm7 = dst[0]"
+		// multiplier par le volume
+		pmullw	mm1, mm0		//%%mm0,%%mm1\n" // mm1 = l(a*v)|l(b*v)|l(c*v)|l(d*v)
+		pmulhw	mm2, mm0		//%%mm0,%%mm2\n" // mm2 = h(a*v)|h(b*v)|h(c*v)|h(d*v)
+		movq	mm5, mm4		//%%mm4,%%mm5\n" // mm5 = e|f|g|h
+		pmullw	mm4, mm0		//%%mm0,%%mm4\n" // mm4 = l(e*v)|l(f*v)|l(g*v)|l(h*v)
+		pmulhw	mm5, mm0		//%%mm0,%%mm5\n" // mm5 = h(e*v)|h(f*v)|h(g*v)|h(h*v)
+		movq	mm3, mm1		//%%mm1,%%mm3\n" // mm3 = l(a*v)|l(b*v)|l(c*v)|l(d*v)
+		punpckhwd	mm1, mm2	//%%mm2,%%mm1\n" // mm1 = a*v|b*v
+		movq		mm6, mm4	//%%mm4,%%mm6\n" // mm6 = l(e*v)|l(f*v)|l(g*v)|l(h*v)
+		punpcklwd	mm3, mm2	//%%mm2,%%mm3\n" // mm3 = c*v|d*v
+		punpckhwd	mm4, mm5	//%%mm5,%%mm4\n" // mm4 = e*f|f*v
+		punpcklwd	mm6, mm5	//%%mm5,%%mm6\n" // mm6 = g*v|h*v
+		// pre charger le buffer dst dans mm5
+		movq	mm5, [edi + 8]	//8(%%edi),%%mm5\n" // mm5 = dst[1]
+		// diviser par 128
+		psrad	mm1, 7			//$7,%%mm1\n" // mm1 = a*v/128|b*v/128 , 128 = SDL_MIX_MAXVOLUME
+		add		esi, 16			//$16,%%esi\n"
+		psrad	mm3, 7			//$7,%%mm3\n" // mm3 = c*v/128|d*v/128
+		psrad	mm4, 7			//$7,%%mm4\n" // mm4 = e*v/128|f*v/128
+		// mm1 = le sample avec le volume modifie
+		packssdw	mm3, mm1	//%%mm1,%%mm3\n" // mm3 = s(a*v|b*v|c*v|d*v)
+		psrad	mm6, 7			//$7,%%mm6\n" // mm6= g*v/128|h*v/128
+		paddsw	mm3, mm7		//%%mm7,%%mm3\n" // mm3 = adjust_volume(src)+dst
+		// mm4 = le sample avec le volume modifie
+		packssdw	mm6, mm4	//%%mm4,%%mm6\n" // mm6 = s(e*v|f*v|g*v|h*v)
+		movq	[edi], mm3		//%%mm3,(%%edi)\n"
+		paddsw	mm6, mm5		//%%mm5,%%mm6\n" // mm6 = adjust_volume(src)+dst
+		movq	[edi + 8], mm6	//%%mm6,8(%%edi)\n"
+		add		edi, 16			//$16,%%edi\n"
+		dec		ebx				//%%ebx\n"
+		jnz mixloopS16
+
+endS16:
+		emms
+		
+		pop		ebx
+		pop		esi
+		pop		edi
+	}
+
+}
+
+////////////////////////////////////////////////
+// Mixing for 8 bit signed buffers
+////////////////////////////////////////////////
+
+void SDL_MixAudio_MMX_S8_VC(char* dst,char* src,unsigned int nSize,int volume)
+{
+	_asm
+	{
+
+		push	edi
+		push	esi
+		push	ebx
+		
+		mov		edi, dst	//movl	%0,%%edi	// edi = dst
+		mov		esi, src	//%1,%%esi	// esi = src
+		mov		eax, volume	//%3,%%eax	// eax = volume
+
+		movd	mm0, eax	//%%eax,%%mm0
+		movq	mm1, mm0	//%%mm0,%%mm1
+		psllq	mm0, 16		//$16,%%mm0
+		por		mm0, mm1	//%%mm1,%%mm0
+		psllq	mm0, 16		//$16,%%mm0
+		por		mm0, mm1	//%%mm1,%%mm0
+		psllq	mm0, 16		//$16,%%mm0
+		por		mm0, mm1	//%%mm1,%%mm0
+		
+		mov		ebx, nSize	//%2,%%ebx	// ebx = size
+		shr		ebx, 3		//$3,%%ebx	// process 8 bytes per iteration = 8 samples
+		cmp		ebx, 0		//$0,%%ebx
+		je		endS8
+
+		#ifndef __WATCOMC__
+		align 16
+		#endif
+mixloopS8:
+		pxor	mm2, mm2	//%%mm2,%%mm2		// mm2 = 0
+		movq	mm1, [esi]	//(%%esi),%%mm1	// mm1 = a|b|c|d|e|f|g|h
+		movq	mm3, mm1	//%%mm1,%%mm3 	// mm3 = a|b|c|d|e|f|g|h
+		// on va faire le "sign extension" en faisant un cmp avec 0 qui retourne 1 si <0, 0 si >0
+		pcmpgtb		mm2, mm1	//%%mm1,%%mm2	// mm2 = 11111111|00000000|00000000....
+		punpckhbw	mm1, mm2	//%%mm2,%%mm1	// mm1 = 0|a|0|b|0|c|0|d
+		punpcklbw	mm3, mm2	//%%mm2,%%mm3	// mm3 = 0|e|0|f|0|g|0|h
+		movq	mm2, [edi]	//(%%edi),%%mm2	// mm2 = destination
+		pmullw	mm1, mm0	//%%mm0,%%mm1	// mm1 = v*a|v*b|v*c|v*d
+		add		esi, 8		//$8,%%esi
+		pmullw	mm3, mm0	//%%mm0,%%mm3	// mm3 = v*e|v*f|v*g|v*h
+		psraw	mm1, 7		//$7,%%mm1		// mm1 = v*a/128|v*b/128|v*c/128|v*d/128 
+		psraw	mm3, 7		//$7,%%mm3		// mm3 = v*e/128|v*f/128|v*g/128|v*h/128
+		packsswb mm3, mm1	//%%mm1,%%mm3	// mm1 = v*a/128|v*b/128|v*c/128|v*d/128|v*e/128|v*f/128|v*g/128|v*h/128
+		paddsb	mm3, mm2	//%%mm2,%%mm3	// add to destination buffer
+		movq	[edi], mm3	//%%mm3,(%%edi)	// store back to ram
+		add		edi, 8		//$8,%%edi
+		dec		ebx			//%%ebx
+		jnz		mixloopS8
+		
+endS8:
+		emms
+		
+		pop		ebx
+		pop		esi
+		pop		edi
+	}
+}
+
+#endif /* SDL_ASSEMBLY_ROUTINES */
+#endif /* SDL_BUGGY_MMX_MIXERS */

Externals/SDL/src/audio/SDL_mixer_MMX_VC.h

@@ -0,0 +1,38 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+
+#if defined(SDL_BUGGY_MMX_MIXERS) /* buggy, so we're disabling them. --ryan. */
+#if ((defined(_MSC_VER) && defined(_M_IX86)) || defined(__WATCOMC__)) && defined(SDL_ASSEMBLY_ROUTINES)
+/* headers for MMX assembler version of SDL_MixAudio
+   Copyright 2002 Stephane Marchesin (stephane.marchesin@wanadoo.fr)
+   Converted to Intel ASM notation by Cth
+   This code is licensed under the LGPL (see COPYING for details)
+   
+   Assumes buffer size in bytes is a multiple of 16
+   Assumes SDL_MIX_MAXVOLUME = 128
+*/
+void SDL_MixAudio_MMX_S16_VC(char* ,char* ,unsigned int ,int );
+void SDL_MixAudio_MMX_S8_VC(char* ,char* ,unsigned int ,int );
+#endif
+#endif

Externals/SDL/src/audio/SDL_mixer_m68k.c

@@ -0,0 +1,211 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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"
+
+/*
+	m68k assembly mix routines
+
+	Patrice Mandin
+*/
+
+#if defined(__M68000__) && defined(__GNUC__)
+void SDL_MixAudio_m68k_U8(char* dst, char* src, long len, long volume, char* mix8)
+{
+    __asm__ __volatile__ (
+
+	"tstl	%2\n"
+"	beqs	stoploop_u8\n"
+"mixloop_u8:\n"
+
+	/* Mix a sample */
+
+"	moveq	#0,%%d0\n"
+"	moveq	#0,%%d1\n"
+
+"	moveb	%1@+,%%d0\n"	/* d0 = *src++ */
+"	sub	#128,%%d0\n"	/* d0 -= 128 */
+"	muls	%3,%%d0\n"	/* d0 *= volume (0<=volume<=128) */
+"	moveb	%0@,%%d1\n"	/* d1 = *dst */
+"	asr	#7,%%d0\n"	/* d0 /= 128 (SDL_MIX_MAXVOLUME) */
+"	add	#128,%%d0\n"	/* d0 += 128 */
+
+"	add	%%d1,%%d0\n"
+
+"	moveb	%4@(%%d0:w),%0@+\n"
+
+	/* Loop till done */
+
+"	subql	#1,%2\n"
+"	bhis	mixloop_u8\n"
+"stoploop_u8:\n"
+
+	 : /* no return value */
+	 : /* input */
+	 	"a"(dst), "a"(src), "d"(len), "d"(volume), "a"(mix8)	
+	 : /* clobbered registers */
+	 	"d0", "d1", "cc", "memory" 
+	 );
+}
+
+void SDL_MixAudio_m68k_S8(char* dst, char* src, long len, long volume)
+{
+    __asm__ __volatile__ (
+
+	"tstl	%2\n"
+"	beqs	stoploop_s8\n"
+"	moveq	#-128,%%d2\n"
+"	moveq	#127,%%d3\n"
+"mixloop_s8:\n"
+
+	/* Mix a sample */
+
+"	moveq	#0,%%d0\n"
+"	moveq	#0,%%d1\n"
+
+"	moveb	%1@+,%%d0\n"	/* d0 = *src++ */
+"	muls	%3,%%d0\n"	/* d0 *= volume (0<=volume<=128) */
+"	moveb	%0@,%%d1\n"	/* d1 = *dst */
+"	asr	#7,%%d0\n"	/* d0 /= 128 (SDL_MIX_MAXVOLUME) */
+
+"	add	%%d1,%%d0\n"
+
+"	cmp	%%d2,%%d0\n"
+"	bges	lower_limit_s8\n"
+"	move	%%d2,%%d0\n"
+"lower_limit_s8:\n"
+
+"	cmp	%%d3,%%d0\n"
+"	bles	upper_limit_s8\n"
+"	move	%%d3,%%d0\n"
+"upper_limit_s8:\n"
+"	moveb	%%d0,%0@+\n"
+
+	/* Loop till done */
+
+"	subql	#1,%2\n"
+"	bhis	mixloop_s8\n"
+"stoploop_s8:\n"
+
+	 : /* no return value */
+	 : /* input */
+	 	"a"(dst), "a"(src), "d"(len), "d"(volume)	
+	 : /* clobbered registers */
+	 	"d0", "d1", "d2", "d3", "cc", "memory" 
+	 );
+}
+
+void SDL_MixAudio_m68k_S16MSB(short* dst, short* src, long len, long volume)
+{
+    __asm__ __volatile__ (
+
+	"tstl	%2\n"
+"	beqs	stoploop_s16msb\n"
+"	movel	#-32768,%%d2\n"
+"	movel	#32767,%%d3\n"
+"	lsrl	#1,%2\n"
+"mixloop_s16msb:\n"
+
+	/* Mix a sample */
+
+"	move	%1@+,%%d0\n"	/* d0 = *src++ */
+"	muls	%3,%%d0\n"	/* d0 *= volume (0<=volume<=128) */
+"	move	%0@,%%d1\n"	/* d1 = *dst */
+"	extl	%%d1\n"		/* extend d1 to 32 bits */
+"	asrl	#7,%%d0\n"	/* d0 /= 128 (SDL_MIX_MAXVOLUME) */
+
+"	addl	%%d1,%%d0\n"
+
+"	cmpl	%%d2,%%d0\n"
+"	bges	lower_limit_s16msb\n"
+"	move	%%d2,%%d0\n"
+"lower_limit_s16msb:\n"
+
+"	cmpl	%%d3,%%d0\n"
+"	bles	upper_limit_s16msb\n"
+"	move	%%d3,%%d0\n"
+"upper_limit_s16msb:\n"
+"	move	%%d0,%0@+\n"
+
+	/* Loop till done */
+
+"	subql	#1,%2\n"
+"	bhis	mixloop_s16msb\n"
+"stoploop_s16msb:\n"
+
+	 : /* no return value */
+	 : /* input */
+	 	"a"(dst), "a"(src), "d"(len), "d"(volume)	
+	 : /* clobbered registers */
+	 	"d0", "d1", "d2", "d3", "cc", "memory" 
+	 );
+}
+
+void SDL_MixAudio_m68k_S16LSB(short* dst, short* src, long len, long volume)
+{
+    __asm__ __volatile__ (
+
+	"tstl	%2\n"
+"	beqs	stoploop_s16lsb\n"
+"	movel	#-32768,%%d2\n"
+"	movel	#32767,%%d3\n"
+"	lsrl	#1,%2\n"
+"mixloop_s16lsb:\n"
+
+	/* Mix a sample */
+
+"	move	%1@+,%%d0\n"	/* d0 = *src++ */
+"	rorw	#8,%%d0\n"
+"	muls	%3,%%d0\n"	/* d0 *= volume (0<=volume<=128) */
+"	move	%0@,%%d1\n"	/* d1 = *dst */
+"	rorw	#8,%%d1\n"
+"	extl	%%d1\n"		/* extend d1 to 32 bits */
+"	asrl	#7,%%d0\n"	/* d0 /= 128 (SDL_MIX_MAXVOLUME) */
+
+"	addl	%%d1,%%d0\n"
+
+"	cmpl	%%d2,%%d0\n"
+"	bges	lower_limit_s16lsb\n"
+"	move	%%d2,%%d0\n"
+"lower_limit_s16lsb:\n"
+
+"	cmpl	%%d3,%%d0\n"
+"	bles	upper_limit_s16lsb\n"
+"	move	%%d3,%%d0\n"
+"upper_limit_s16lsb:\n"
+"	rorw	#8,%%d0\n"
+"	move	%%d0,%0@+\n"
+
+	/* Loop till done */
+
+"	subql	#1,%2\n"
+"	bhis	mixloop_s16lsb\n"
+"stoploop_s16lsb:\n"
+
+	 : /* no return value */
+	 : /* input */
+	 	"a"(dst), "a"(src), "d"(len), "d"(volume)	
+	 : /* clobbered registers */
+	 	"d0", "d1", "d2", "d3", "cc", "memory" 
+	 );
+}
+#endif
+

Externals/SDL/src/audio/SDL_mixer_m68k.h

@@ -0,0 +1,36 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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"
+
+/*
+	m68k assembly mix routines
+
+	Patrice Mandin
+*/
+
+#if defined(__M68000__) && defined(__GNUC__)
+void SDL_MixAudio_m68k_U8(char* dst,char* src, long len, long volume, char* mix8);
+void SDL_MixAudio_m68k_S8(char* dst,char* src, long len, long volume);
+
+void SDL_MixAudio_m68k_S16MSB(short* dst,short* src, long len, long volume);
+void SDL_MixAudio_m68k_S16LSB(short* dst,short* src, long len, long volume);
+#endif

Externals/SDL/src/audio/SDL_sysaudio.h

@@ -0,0 +1,184 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is SDL_free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_sysaudio_h
+#define _SDL_sysaudio_h
+
+#include "SDL_mutex.h"
+#include "SDL_thread.h"
+
+/* The SDL audio driver */
+typedef struct SDL_AudioDevice SDL_AudioDevice;
+
+/* Define the SDL audio driver structure */
+#define _THIS	SDL_AudioDevice *_this
+#ifndef _STATUS
+#define _STATUS	SDL_status *status
+#endif
+struct SDL_AudioDevice {
+	/* * * */
+	/* The name of this audio driver */
+	const char *name;
+
+	/* * * */
+	/* The description of this audio driver */
+	const char *desc;
+
+	/* * * */
+	/* Public driver functions */
+	int  (*OpenAudio)(_THIS, SDL_AudioSpec *spec);
+	void (*ThreadInit)(_THIS);	/* Called by audio thread at start */
+	void (*WaitAudio)(_THIS);
+	void (*PlayAudio)(_THIS);
+	Uint8 *(*GetAudioBuf)(_THIS);
+	void (*WaitDone)(_THIS);
+	void (*CloseAudio)(_THIS);
+
+	/* * * */
+	/* Lock / Unlock functions added for the Mac port */
+	void (*LockAudio)(_THIS);
+	void (*UnlockAudio)(_THIS);
+
+	/* * * */
+	/* Data common to all devices */
+
+	/* The current audio specification (shared with audio thread) */
+	SDL_AudioSpec spec;
+
+	/* An audio conversion block for audio format emulation */
+	SDL_AudioCVT convert;
+
+	/* Current state flags */
+	int enabled;
+	int paused;
+	int opened;
+
+	/* Fake audio buffer for when the audio hardware is busy */
+	Uint8 *fake_stream;
+
+	/* A semaphore for locking the mixing buffers */
+	SDL_mutex *mixer_lock;
+
+	/* A thread to feed the audio device */
+	SDL_Thread *thread;
+	Uint32 threadid;
+
+	/* * * */
+	/* Data private to this driver */
+	struct SDL_PrivateAudioData *hidden;
+
+	/* * * */
+	/* The function used to dispose of this structure */
+	void (*free)(_THIS);
+};
+#undef _THIS
+
+typedef struct AudioBootStrap {
+	const char *name;
+	const char *desc;
+	int (*available)(void);
+	SDL_AudioDevice *(*create)(int devindex);
+} AudioBootStrap;
+
+#if SDL_AUDIO_DRIVER_BSD
+extern AudioBootStrap BSD_AUDIO_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_PULSE
+extern AudioBootStrap PULSE_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_ALSA
+extern AudioBootStrap ALSA_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_OSS
+extern AudioBootStrap DSP_bootstrap;
+extern AudioBootStrap DMA_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_QNXNTO
+extern AudioBootStrap QNXNTOAUDIO_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_SUNAUDIO
+extern AudioBootStrap SUNAUDIO_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_DMEDIA
+extern AudioBootStrap DMEDIA_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_ARTS
+extern AudioBootStrap ARTS_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_ESD
+extern AudioBootStrap ESD_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_NAS
+extern AudioBootStrap NAS_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_DSOUND
+extern AudioBootStrap DSOUND_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_WAVEOUT
+extern AudioBootStrap WAVEOUT_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_PAUD
+extern AudioBootStrap Paud_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_BAUDIO
+extern AudioBootStrap BAUDIO_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_COREAUDIO
+extern AudioBootStrap COREAUDIO_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_SNDMGR
+extern AudioBootStrap SNDMGR_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_MINT
+extern AudioBootStrap MINTAUDIO_GSXB_bootstrap;
+extern AudioBootStrap MINTAUDIO_MCSN_bootstrap;
+extern AudioBootStrap MINTAUDIO_STFA_bootstrap;
+extern AudioBootStrap MINTAUDIO_XBIOS_bootstrap;
+extern AudioBootStrap MINTAUDIO_DMA8_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_DISK
+extern AudioBootStrap DISKAUD_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_DUMMY
+extern AudioBootStrap DUMMYAUD_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_DC
+extern AudioBootStrap DCAUD_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_NDS
+extern AudioBootStrap NDSAUD_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_MMEAUDIO
+extern AudioBootStrap MMEAUDIO_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_DART
+extern AudioBootStrap DART_bootstrap;
+#endif
+#if SDL_AUDIO_DRIVER_EPOCAUDIO
+extern AudioBootStrap EPOCAudio_bootstrap; 
+#endif
+
+/* This is the current audio device */
+extern SDL_AudioDevice *current_audio;
+
+#endif /* _SDL_sysaudio_h */

Externals/SDL/src/audio/SDL_wave.c

@@ -0,0 +1,600 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Microsoft WAVE file loading routines */
+
+#include "SDL_audio.h"
+#include "SDL_wave.h"
+
+
+static int ReadChunk(SDL_RWops *src, Chunk *chunk);
+
+struct MS_ADPCM_decodestate {
+	Uint8 hPredictor;
+	Uint16 iDelta;
+	Sint16 iSamp1;
+	Sint16 iSamp2;
+};
+static struct MS_ADPCM_decoder {
+	WaveFMT wavefmt;
+	Uint16 wSamplesPerBlock;
+	Uint16 wNumCoef;
+	Sint16 aCoeff[7][2];
+	/* * * */
+	struct MS_ADPCM_decodestate state[2];
+} MS_ADPCM_state;
+
+static int InitMS_ADPCM(WaveFMT *format)
+{
+	Uint8 *rogue_feel;
+	Uint16 extra_info;
+	int i;
+
+	/* Set the rogue pointer to the MS_ADPCM specific data */
+	MS_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding);
+	MS_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels);
+	MS_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency);
+	MS_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate);
+	MS_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign);
+	MS_ADPCM_state.wavefmt.bitspersample =
+					 SDL_SwapLE16(format->bitspersample);
+	rogue_feel = (Uint8 *)format+sizeof(*format);
+	if ( sizeof(*format) == 16 ) {
+		extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]);
+		rogue_feel += sizeof(Uint16);
+	}
+	MS_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]);
+	rogue_feel += sizeof(Uint16);
+	MS_ADPCM_state.wNumCoef = ((rogue_feel[1]<<8)|rogue_feel[0]);
+	rogue_feel += sizeof(Uint16);
+	if ( MS_ADPCM_state.wNumCoef != 7 ) {
+		SDL_SetError("Unknown set of MS_ADPCM coefficients");
+		return(-1);
+	}
+	for ( i=0; i<MS_ADPCM_state.wNumCoef; ++i ) {
+		MS_ADPCM_state.aCoeff[i][0] = ((rogue_feel[1]<<8)|rogue_feel[0]);
+		rogue_feel += sizeof(Uint16);
+		MS_ADPCM_state.aCoeff[i][1] = ((rogue_feel[1]<<8)|rogue_feel[0]);
+		rogue_feel += sizeof(Uint16);
+	}
+	return(0);
+}
+
+static Sint32 MS_ADPCM_nibble(struct MS_ADPCM_decodestate *state,
+					Uint8 nybble, Sint16 *coeff)
+{
+	const Sint32 max_audioval = ((1<<(16-1))-1);
+	const Sint32 min_audioval = -(1<<(16-1));
+	const Sint32 adaptive[] = {
+		230, 230, 230, 230, 307, 409, 512, 614,
+		768, 614, 512, 409, 307, 230, 230, 230
+	};
+	Sint32 new_sample, delta;
+
+	new_sample = ((state->iSamp1 * coeff[0]) +
+		      (state->iSamp2 * coeff[1]))/256;
+	if ( nybble & 0x08 ) {
+		new_sample += state->iDelta * (nybble-0x10);
+	} else {
+		new_sample += state->iDelta * nybble;
+	}
+	if ( new_sample < min_audioval ) {
+		new_sample = min_audioval;
+	} else
+	if ( new_sample > max_audioval ) {
+		new_sample = max_audioval;
+	}
+	delta = ((Sint32)state->iDelta * adaptive[nybble])/256;
+	if ( delta < 16 ) {
+		delta = 16;
+	}
+	state->iDelta = (Uint16)delta;
+	state->iSamp2 = state->iSamp1;
+	state->iSamp1 = (Sint16)new_sample;
+	return(new_sample);
+}
+
+static int MS_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len)
+{
+	struct MS_ADPCM_decodestate *state[2];
+	Uint8 *freeable, *encoded, *decoded;
+	Sint32 encoded_len, samplesleft;
+	Sint8 nybble, stereo;
+	Sint16 *coeff[2];
+	Sint32 new_sample;
+
+	/* Allocate the proper sized output buffer */
+	encoded_len = *audio_len;
+	encoded = *audio_buf;
+	freeable = *audio_buf;
+	*audio_len = (encoded_len/MS_ADPCM_state.wavefmt.blockalign) * 
+				MS_ADPCM_state.wSamplesPerBlock*
+				MS_ADPCM_state.wavefmt.channels*sizeof(Sint16);
+	*audio_buf = (Uint8 *)SDL_malloc(*audio_len);
+	if ( *audio_buf == NULL ) {
+		SDL_Error(SDL_ENOMEM);
+		return(-1);
+	}
+	decoded = *audio_buf;
+
+	/* Get ready... Go! */
+	stereo = (MS_ADPCM_state.wavefmt.channels == 2);
+	state[0] = &MS_ADPCM_state.state[0];
+	state[1] = &MS_ADPCM_state.state[stereo];
+	while ( encoded_len >= MS_ADPCM_state.wavefmt.blockalign ) {
+		/* Grab the initial information for this block */
+		state[0]->hPredictor = *encoded++;
+		if ( stereo ) {
+			state[1]->hPredictor = *encoded++;
+		}
+		state[0]->iDelta = ((encoded[1]<<8)|encoded[0]);
+		encoded += sizeof(Sint16);
+		if ( stereo ) {
+			state[1]->iDelta = ((encoded[1]<<8)|encoded[0]);
+			encoded += sizeof(Sint16);
+		}
+		state[0]->iSamp1 = ((encoded[1]<<8)|encoded[0]);
+		encoded += sizeof(Sint16);
+		if ( stereo ) {
+			state[1]->iSamp1 = ((encoded[1]<<8)|encoded[0]);
+			encoded += sizeof(Sint16);
+		}
+		state[0]->iSamp2 = ((encoded[1]<<8)|encoded[0]);
+		encoded += sizeof(Sint16);
+		if ( stereo ) {
+			state[1]->iSamp2 = ((encoded[1]<<8)|encoded[0]);
+			encoded += sizeof(Sint16);
+		}
+		coeff[0] = MS_ADPCM_state.aCoeff[state[0]->hPredictor];
+		coeff[1] = MS_ADPCM_state.aCoeff[state[1]->hPredictor];
+
+		/* Store the two initial samples we start with */
+		decoded[0] = state[0]->iSamp2&0xFF;
+		decoded[1] = state[0]->iSamp2>>8;
+		decoded += 2;
+		if ( stereo ) {
+			decoded[0] = state[1]->iSamp2&0xFF;
+			decoded[1] = state[1]->iSamp2>>8;
+			decoded += 2;
+		}
+		decoded[0] = state[0]->iSamp1&0xFF;
+		decoded[1] = state[0]->iSamp1>>8;
+		decoded += 2;
+		if ( stereo ) {
+			decoded[0] = state[1]->iSamp1&0xFF;
+			decoded[1] = state[1]->iSamp1>>8;
+			decoded += 2;
+		}
+
+		/* Decode and store the other samples in this block */
+		samplesleft = (MS_ADPCM_state.wSamplesPerBlock-2)*
+					MS_ADPCM_state.wavefmt.channels;
+		while ( samplesleft > 0 ) {
+			nybble = (*encoded)>>4;
+			new_sample = MS_ADPCM_nibble(state[0],nybble,coeff[0]);
+			decoded[0] = new_sample&0xFF;
+			new_sample >>= 8;
+			decoded[1] = new_sample&0xFF;
+			decoded += 2;
+
+			nybble = (*encoded)&0x0F;
+			new_sample = MS_ADPCM_nibble(state[1],nybble,coeff[1]);
+			decoded[0] = new_sample&0xFF;
+			new_sample >>= 8;
+			decoded[1] = new_sample&0xFF;
+			decoded += 2;
+
+			++encoded;
+			samplesleft -= 2;
+		}
+		encoded_len -= MS_ADPCM_state.wavefmt.blockalign;
+	}
+	SDL_free(freeable);
+	return(0);
+}
+
+struct IMA_ADPCM_decodestate {
+	Sint32 sample;
+	Sint8 index;
+};
+static struct IMA_ADPCM_decoder {
+	WaveFMT wavefmt;
+	Uint16 wSamplesPerBlock;
+	/* * * */
+	struct IMA_ADPCM_decodestate state[2];
+} IMA_ADPCM_state;
+
+static int InitIMA_ADPCM(WaveFMT *format)
+{
+	Uint8 *rogue_feel;
+	Uint16 extra_info;
+
+	/* Set the rogue pointer to the IMA_ADPCM specific data */
+	IMA_ADPCM_state.wavefmt.encoding = SDL_SwapLE16(format->encoding);
+	IMA_ADPCM_state.wavefmt.channels = SDL_SwapLE16(format->channels);
+	IMA_ADPCM_state.wavefmt.frequency = SDL_SwapLE32(format->frequency);
+	IMA_ADPCM_state.wavefmt.byterate = SDL_SwapLE32(format->byterate);
+	IMA_ADPCM_state.wavefmt.blockalign = SDL_SwapLE16(format->blockalign);
+	IMA_ADPCM_state.wavefmt.bitspersample =
+					 SDL_SwapLE16(format->bitspersample);
+	rogue_feel = (Uint8 *)format+sizeof(*format);
+	if ( sizeof(*format) == 16 ) {
+		extra_info = ((rogue_feel[1]<<8)|rogue_feel[0]);
+		rogue_feel += sizeof(Uint16);
+	}
+	IMA_ADPCM_state.wSamplesPerBlock = ((rogue_feel[1]<<8)|rogue_feel[0]);
+	return(0);
+}
+
+static Sint32 IMA_ADPCM_nibble(struct IMA_ADPCM_decodestate *state,Uint8 nybble)
+{
+	const Sint32 max_audioval = ((1<<(16-1))-1);
+	const Sint32 min_audioval = -(1<<(16-1));
+	const int index_table[16] = {
+		-1, -1, -1, -1,
+		 2,  4,  6,  8,
+		-1, -1, -1, -1,
+		 2,  4,  6,  8
+	};
+	const Sint32 step_table[89] = {
+		7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31,
+		34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130,
+		143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408,
+		449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282,
+		1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327,
+		3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630,
+		9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350,
+		22385, 24623, 27086, 29794, 32767
+	};
+	Sint32 delta, step;
+
+	/* Compute difference and new sample value */
+	step = step_table[state->index];
+	delta = step >> 3;
+	if ( nybble & 0x04 ) delta += step;
+	if ( nybble & 0x02 ) delta += (step >> 1);
+	if ( nybble & 0x01 ) delta += (step >> 2);
+	if ( nybble & 0x08 ) delta = -delta;
+	state->sample += delta;
+
+	/* Update index value */
+	state->index += index_table[nybble];
+	if ( state->index > 88 ) {
+		state->index = 88;
+	} else
+	if ( state->index < 0 ) {
+		state->index = 0;
+	}
+
+	/* Clamp output sample */
+	if ( state->sample > max_audioval ) {
+		state->sample = max_audioval;
+	} else
+	if ( state->sample < min_audioval ) {
+		state->sample = min_audioval;
+	}
+	return(state->sample);
+}
+
+/* Fill the decode buffer with a channel block of data (8 samples) */
+static void Fill_IMA_ADPCM_block(Uint8 *decoded, Uint8 *encoded,
+	int channel, int numchannels, struct IMA_ADPCM_decodestate *state)
+{
+	int i;
+	Sint8 nybble;
+	Sint32 new_sample;
+
+	decoded += (channel * 2);
+	for ( i=0; i<4; ++i ) {
+		nybble = (*encoded)&0x0F;
+		new_sample = IMA_ADPCM_nibble(state, nybble);
+		decoded[0] = new_sample&0xFF;
+		new_sample >>= 8;
+		decoded[1] = new_sample&0xFF;
+		decoded += 2 * numchannels;
+
+		nybble = (*encoded)>>4;
+		new_sample = IMA_ADPCM_nibble(state, nybble);
+		decoded[0] = new_sample&0xFF;
+		new_sample >>= 8;
+		decoded[1] = new_sample&0xFF;
+		decoded += 2 * numchannels;
+
+		++encoded;
+	}
+}
+
+static int IMA_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len)
+{
+	struct IMA_ADPCM_decodestate *state;
+	Uint8 *freeable, *encoded, *decoded;
+	Sint32 encoded_len, samplesleft;
+	unsigned int c, channels;
+
+	/* Check to make sure we have enough variables in the state array */
+	channels = IMA_ADPCM_state.wavefmt.channels;
+	if ( channels > SDL_arraysize(IMA_ADPCM_state.state) ) {
+		SDL_SetError("IMA ADPCM decoder can only handle %d channels",
+					SDL_arraysize(IMA_ADPCM_state.state));
+		return(-1);
+	}
+	state = IMA_ADPCM_state.state;
+
+	/* Allocate the proper sized output buffer */
+	encoded_len = *audio_len;
+	encoded = *audio_buf;
+	freeable = *audio_buf;
+	*audio_len = (encoded_len/IMA_ADPCM_state.wavefmt.blockalign) * 
+				IMA_ADPCM_state.wSamplesPerBlock*
+				IMA_ADPCM_state.wavefmt.channels*sizeof(Sint16);
+	*audio_buf = (Uint8 *)SDL_malloc(*audio_len);
+	if ( *audio_buf == NULL ) {
+		SDL_Error(SDL_ENOMEM);
+		return(-1);
+	}
+	decoded = *audio_buf;
+
+	/* Get ready... Go! */
+	while ( encoded_len >= IMA_ADPCM_state.wavefmt.blockalign ) {
+		/* Grab the initial information for this block */
+		for ( c=0; c<channels; ++c ) {
+			/* Fill the state information for this block */
+			state[c].sample = ((encoded[1]<<8)|encoded[0]);
+			encoded += 2;
+			if ( state[c].sample & 0x8000 ) {
+				state[c].sample -= 0x10000;
+			}
+			state[c].index = *encoded++;
+			/* Reserved byte in buffer header, should be 0 */
+			if ( *encoded++ != 0 ) {
+				/* Uh oh, corrupt data?  Buggy code? */;
+			}
+
+			/* Store the initial sample we start with */
+			decoded[0] = (Uint8)(state[c].sample&0xFF);
+			decoded[1] = (Uint8)(state[c].sample>>8);
+			decoded += 2;
+		}
+
+		/* Decode and store the other samples in this block */
+		samplesleft = (IMA_ADPCM_state.wSamplesPerBlock-1)*channels;
+		while ( samplesleft > 0 ) {
+			for ( c=0; c<channels; ++c ) {
+				Fill_IMA_ADPCM_block(decoded, encoded,
+						c, channels, &state[c]);
+				encoded += 4;
+				samplesleft -= 8;
+			}
+			decoded += (channels * 8 * 2);
+		}
+		encoded_len -= IMA_ADPCM_state.wavefmt.blockalign;
+	}
+	SDL_free(freeable);
+	return(0);
+}
+
+SDL_AudioSpec * SDL_LoadWAV_RW (SDL_RWops *src, int freesrc,
+		SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len)
+{
+	int was_error;
+	Chunk chunk;
+	int lenread;
+	int MS_ADPCM_encoded, IMA_ADPCM_encoded;
+	int samplesize;
+
+	/* WAV magic header */
+	Uint32 RIFFchunk;
+	Uint32 wavelen = 0;
+	Uint32 WAVEmagic;
+	Uint32 headerDiff = 0;
+
+	/* FMT chunk */
+	WaveFMT *format = NULL;
+
+	/* Make sure we are passed a valid data source */
+	was_error = 0;
+	if ( src == NULL ) {
+		was_error = 1;
+		goto done;
+	}
+		
+	/* Check the magic header */
+	RIFFchunk	= SDL_ReadLE32(src);
+	wavelen		= SDL_ReadLE32(src);
+	if ( wavelen == WAVE ) { /* The RIFFchunk has already been read */
+		WAVEmagic = wavelen;
+		wavelen   = RIFFchunk;
+		RIFFchunk = RIFF;
+	} else {
+		WAVEmagic = SDL_ReadLE32(src);
+	}
+	if ( (RIFFchunk != RIFF) || (WAVEmagic != WAVE) ) {
+		SDL_SetError("Unrecognized file type (not WAVE)");
+		was_error = 1;
+		goto done;
+	}
+	headerDiff += sizeof(Uint32); /* for WAVE */
+
+	/* Read the audio data format chunk */
+	chunk.data = NULL;
+	do {
+		if ( chunk.data != NULL ) {
+			SDL_free(chunk.data);
+			chunk.data = NULL;
+		}
+		lenread = ReadChunk(src, &chunk);
+		if ( lenread < 0 ) {
+			was_error = 1;
+			goto done;
+		}
+		/* 2 Uint32's for chunk header+len, plus the lenread */
+		headerDiff += lenread + 2 * sizeof(Uint32);
+	} while ( (chunk.magic == FACT) || (chunk.magic == LIST) );
+
+	/* Decode the audio data format */
+	format = (WaveFMT *)chunk.data;
+	if ( chunk.magic != FMT ) {
+		SDL_SetError("Complex WAVE files not supported");
+		was_error = 1;
+		goto done;
+	}
+	MS_ADPCM_encoded = IMA_ADPCM_encoded = 0;
+	switch (SDL_SwapLE16(format->encoding)) {
+		case PCM_CODE:
+			/* We can understand this */
+			break;
+		case MS_ADPCM_CODE:
+			/* Try to understand this */
+			if ( InitMS_ADPCM(format) < 0 ) {
+				was_error = 1;
+				goto done;
+			}
+			MS_ADPCM_encoded = 1;
+			break;
+		case IMA_ADPCM_CODE:
+			/* Try to understand this */
+			if ( InitIMA_ADPCM(format) < 0 ) {
+				was_error = 1;
+				goto done;
+			}
+			IMA_ADPCM_encoded = 1;
+			break;
+		case MP3_CODE:
+			SDL_SetError("MPEG Layer 3 data not supported",
+					SDL_SwapLE16(format->encoding));
+			was_error = 1;
+			goto done;
+		default:
+			SDL_SetError("Unknown WAVE data format: 0x%.4x",
+					SDL_SwapLE16(format->encoding));
+			was_error = 1;
+			goto done;
+	}
+	SDL_memset(spec, 0, (sizeof *spec));
+	spec->freq = SDL_SwapLE32(format->frequency);
+	switch (SDL_SwapLE16(format->bitspersample)) {
+		case 4:
+			if ( MS_ADPCM_encoded || IMA_ADPCM_encoded ) {
+				spec->format = AUDIO_S16;
+			} else {
+				was_error = 1;
+			}
+			break;
+		case 8:
+			spec->format = AUDIO_U8;
+			break;
+		case 16:
+			spec->format = AUDIO_S16;
+			break;
+		default:
+			was_error = 1;
+			break;
+	}
+	if ( was_error ) {
+		SDL_SetError("Unknown %d-bit PCM data format",
+			SDL_SwapLE16(format->bitspersample));
+		goto done;
+	}
+	spec->channels = (Uint8)SDL_SwapLE16(format->channels);
+	spec->samples = 4096;		/* Good default buffer size */
+
+	/* Read the audio data chunk */
+	*audio_buf = NULL;
+	do {
+		if ( *audio_buf != NULL ) {
+			SDL_free(*audio_buf);
+			*audio_buf = NULL;
+		}
+		lenread = ReadChunk(src, &chunk);
+		if ( lenread < 0 ) {
+			was_error = 1;
+			goto done;
+		}
+		*audio_len = lenread;
+		*audio_buf = chunk.data;
+		if(chunk.magic != DATA) headerDiff += lenread + 2 * sizeof(Uint32);
+	} while ( chunk.magic != DATA );
+	headerDiff += 2 * sizeof(Uint32); /* for the data chunk and len */
+
+	if ( MS_ADPCM_encoded ) {
+		if ( MS_ADPCM_decode(audio_buf, audio_len) < 0 ) {
+			was_error = 1;
+			goto done;
+		}
+	}
+	if ( IMA_ADPCM_encoded ) {
+		if ( IMA_ADPCM_decode(audio_buf, audio_len) < 0 ) {
+			was_error = 1;
+			goto done;
+		}
+	}
+
+	/* Don't return a buffer that isn't a multiple of samplesize */
+	samplesize = ((spec->format & 0xFF)/8)*spec->channels;
+	*audio_len &= ~(samplesize-1);
+
+done:
+	if ( format != NULL ) {
+		SDL_free(format);
+	}
+	if ( src ) {
+		if ( freesrc ) {
+			SDL_RWclose(src);
+		} else {
+			/* seek to the end of the file (given by the RIFF chunk) */
+			SDL_RWseek(src, wavelen - chunk.length - headerDiff, RW_SEEK_CUR);
+		}
+	}
+	if ( was_error ) {
+		spec = NULL;
+	}
+	return(spec);
+}
+
+/* Since the WAV memory is allocated in the shared library, it must also
+   be freed here.  (Necessary under Win32, VC++)
+ */
+void SDL_FreeWAV(Uint8 *audio_buf)
+{
+	if ( audio_buf != NULL ) {
+		SDL_free(audio_buf);
+	}
+}
+
+static int ReadChunk(SDL_RWops *src, Chunk *chunk)
+{
+	chunk->magic	= SDL_ReadLE32(src);
+	chunk->length	= SDL_ReadLE32(src);
+	chunk->data = (Uint8 *)SDL_malloc(chunk->length);
+	if ( chunk->data == NULL ) {
+		SDL_Error(SDL_ENOMEM);
+		return(-1);
+	}
+	if ( SDL_RWread(src, chunk->data, chunk->length, 1) != 1 ) {
+		SDL_Error(SDL_EFREAD);
+		SDL_free(chunk->data);
+		chunk->data = NULL;
+		return(-1);
+	}
+	return(chunk->length);
+}

Externals/SDL/src/audio/SDL_wave.h

@@ -0,0 +1,62 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is SDL_free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* WAVE files are little-endian */
+
+/*******************************************/
+/* Define values for Microsoft WAVE format */
+/*******************************************/
+#define RIFF		0x46464952		/* "RIFF" */
+#define WAVE		0x45564157		/* "WAVE" */
+#define FACT		0x74636166		/* "fact" */
+#define LIST		0x5453494c		/* "LIST" */
+#define FMT		0x20746D66		/* "fmt " */
+#define DATA		0x61746164		/* "data" */
+#define PCM_CODE	0x0001
+#define MS_ADPCM_CODE	0x0002
+#define IMA_ADPCM_CODE	0x0011
+#define MP3_CODE	0x0055
+#define WAVE_MONO	1
+#define WAVE_STEREO	2
+
+/* Normally, these three chunks come consecutively in a WAVE file */
+typedef struct WaveFMT {
+/* Not saved in the chunk we read:
+	Uint32	FMTchunk;
+	Uint32	fmtlen;
+*/
+	Uint16	encoding;	
+	Uint16	channels;		/* 1 = mono, 2 = stereo */
+	Uint32	frequency;		/* One of 11025, 22050, or 44100 Hz */
+	Uint32	byterate;		/* Average bytes per second */
+	Uint16	blockalign;		/* Bytes per sample block */
+	Uint16	bitspersample;		/* One of 8, 12, 16, or 4 for ADPCM */
+} WaveFMT;
+
+/* The general chunk found in the WAVE file */
+typedef struct Chunk {
+	Uint32 magic;
+	Uint32 length;
+	Uint8 *data;
+} Chunk;
+

Externals/SDL/src/audio/alsa/SDL_alsa_audio.c

@@ -0,0 +1,612 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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_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_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; \
+    const Uint32 count = (this->spec.samples / 6); \
+    Uint32 i; \
+    for (i = 0; i < count; 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_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);
+}

Externals/SDL/src/audio/alsa/SDL_alsa_audio.h

@@ -0,0 +1,48 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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 */

Externals/SDL/src/audio/arts/SDL_artsaudio.c

@@ -0,0 +1,348 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Allow access to 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_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_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 Audio_Available(void)
+{
+	int available = 0;
+
+	if ( LoadARTSLibrary() < 0 ) {
+		return available;
+	}
+	if ( SDL_NAME(arts_init)() == 0 ) {
+		if ( SDL_NAME(arts_suspended)() ) {
+			/* 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 ( ! SDL_NAME(arts_suspended)() ) {
+		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);
+}

Externals/SDL/src/audio/arts/SDL_artsaudio.h

@@ -0,0 +1,60 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */
+

Externals/SDL/src/audio/baudio/SDL_beaudio.cc

@@ -0,0 +1,225 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Allow access to the audio stream on BeOS */
+
+#include <SoundPlayer.h>
+
+#include "../../main/beos/SDL_BeApp.h"
+
+extern "C" {
+
+#include "SDL_audio.h"
+#include "../SDL_audio_c.h"
+#include "../SDL_sysaudio.h"
+#include "../../thread/beos/SDL_systhread_c.h"
+#include "SDL_beaudio.h"
+
+
+/* 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 */

Externals/SDL/src/audio/baudio/SDL_beaudio.h

@@ -0,0 +1,39 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */

Externals/SDL/src/audio/bsd/SDL_bsdaudio.c

@@ -0,0 +1,404 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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);
+}

Externals/SDL/src/audio/bsd/SDL_bsdaudio.h

@@ -0,0 +1,58 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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 */

Externals/SDL/src/audio/dart/SDL_dart.c

@@ -0,0 +1,441 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Allow access to 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
+};
+

Externals/SDL/src/audio/dart/SDL_dart.h

@@ -0,0 +1,63 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */

Externals/SDL/src/audio/dc/SDL_dcaudio.c

@@ -0,0 +1,246 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+
+*/
+#include "SDL_config.h"
+
+/* 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);
+}

Externals/SDL/src/audio/dc/SDL_dcaudio.h

@@ -0,0 +1,41 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */

Externals/SDL/src/audio/dc/aica.c

@@ -0,0 +1,271 @@
+/* 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
+}

Externals/SDL/src/audio/dc/aica.h

@@ -0,0 +1,40 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _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

Externals/SDL/src/audio/disk/SDL_diskaudio.c

@@ -0,0 +1,186 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+
+    This file written by Ryan C. Gordon (icculus@icculus.org)
+*/
+#include "SDL_config.h"
+
+/* 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);
+}
+

Externals/SDL/src/audio/disk/SDL_diskaudio.h

@@ -0,0 +1,41 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */

Externals/SDL/src/audio/dma/SDL_dmaaudio.c

@@ -0,0 +1,455 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Allow access to 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);
+}

Externals/SDL/src/audio/dma/SDL_dmaaudio.h

@@ -0,0 +1,59 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */

Externals/SDL/src/audio/dmedia/SDL_irixaudio.c

@@ -0,0 +1,242 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Allow access to 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);
+}
+

Externals/SDL/src/audio/dmedia/SDL_irixaudio.h

@@ -0,0 +1,45 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */

Externals/SDL/src/audio/dsp/SDL_dspaudio.c

@@ -0,0 +1,340 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+
+    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);
+}

Externals/SDL/src/audio/dsp/SDL_dspaudio.h

@@ -0,0 +1,53 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */

Externals/SDL/src/audio/dummy/SDL_dummyaudio.c

@@ -0,0 +1,156 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+
+    This file written by Ryan C. Gordon (icculus@icculus.org)
+*/
+#include "SDL_config.h"
+
+/* Output audio to nowhere... */
+
+#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_dummyaudio.h"
+
+/* The tag name used by DUMMY audio */
+#define DUMMYAUD_DRIVER_NAME         "dummy"
+
+/* Audio driver functions */
+static int DUMMYAUD_OpenAudio(_THIS, SDL_AudioSpec *spec);
+static void DUMMYAUD_WaitAudio(_THIS);
+static void DUMMYAUD_PlayAudio(_THIS);
+static Uint8 *DUMMYAUD_GetAudioBuf(_THIS);
+static void DUMMYAUD_CloseAudio(_THIS);
+
+/* Audio driver bootstrap functions */
+static int DUMMYAUD_Available(void)
+{
+	const char *envr = SDL_getenv("SDL_AUDIODRIVER");
+	if (envr && (SDL_strcmp(envr, DUMMYAUD_DRIVER_NAME) == 0)) {
+		return(1);
+	}
+	return(0);
+}
+
+static void DUMMYAUD_DeleteDevice(SDL_AudioDevice *device)
+{
+	SDL_free(device->hidden);
+	SDL_free(device);
+}
+
+static SDL_AudioDevice *DUMMYAUD_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 = DUMMYAUD_OpenAudio;
+	this->WaitAudio = DUMMYAUD_WaitAudio;
+	this->PlayAudio = DUMMYAUD_PlayAudio;
+	this->GetAudioBuf = DUMMYAUD_GetAudioBuf;
+	this->CloseAudio = DUMMYAUD_CloseAudio;
+
+	this->free = DUMMYAUD_DeleteDevice;
+
+	return this;
+}
+
+AudioBootStrap DUMMYAUD_bootstrap = {
+	DUMMYAUD_DRIVER_NAME, "SDL dummy audio driver",
+	DUMMYAUD_Available, DUMMYAUD_CreateDevice
+};
+
+/* This function waits until it is possible to write a full sound buffer */
+static void DUMMYAUD_WaitAudio(_THIS)
+{
+	/* Don't block on first calls to simulate initial fragment filling. */
+	if (this->hidden->initial_calls)
+		this->hidden->initial_calls--;
+	else
+		SDL_Delay(this->hidden->write_delay);
+}
+
+static void DUMMYAUD_PlayAudio(_THIS)
+{
+	/* no-op...this is a null driver. */
+}
+
+static Uint8 *DUMMYAUD_GetAudioBuf(_THIS)
+{
+	return(this->hidden->mixbuf);
+}
+
+static void DUMMYAUD_CloseAudio(_THIS)
+{
+	if ( this->hidden->mixbuf != NULL ) {
+		SDL_FreeAudioMem(this->hidden->mixbuf);
+		this->hidden->mixbuf = NULL;
+	}
+}
+
+static int DUMMYAUD_OpenAudio(_THIS, SDL_AudioSpec *spec)
+{
+	float bytes_per_sec = 0.0f;
+
+	/* 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);
+
+	bytes_per_sec = (float) (((spec->format & 0xFF) / 8) *
+	                   spec->channels * spec->freq);
+
+	/*
+	 * We try to make this request more audio at the correct rate for
+	 *  a given audio spec, so timing stays fairly faithful.
+	 * Also, we have it not block at all for the first two calls, so
+	 *  it seems like we're filling two audio fragments right out of the
+	 *  gate, like other SDL drivers tend to do.
+	 */
+	this->hidden->initial_calls = 2;
+	this->hidden->write_delay =
+	               (Uint32) ((((float) spec->size) / bytes_per_sec) * 1000.0f);
+
+	/* We're ready to rock and roll. :-) */
+	return(0);
+}
+

Externals/SDL/src/audio/dummy/SDL_dummyaudio.h

@@ -0,0 +1,40 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_dummyaudio_h
+#define _SDL_dummyaudio_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;
+	Uint32 write_delay;
+	Uint32 initial_calls;
+};
+
+#endif /* _SDL_dummyaudio_h */

Externals/SDL/src/audio/esd/SDL_esdaudio.c

@@ -0,0 +1,323 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Allow access to 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);
+}

Externals/SDL/src/audio/esd/SDL_esdaudio.h

@@ -0,0 +1,57 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */

Externals/SDL/src/audio/macosx/SDL_coreaudio.c

@@ -0,0 +1,291 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#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);
+}

Externals/SDL/src/audio/macosx/SDL_coreaudio.h

@@ -0,0 +1,45 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */

Externals/SDL/src/audio/macrom/SDL_romaudio.c

@@ -0,0 +1,496 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#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 */
+

Externals/SDL/src/audio/macrom/SDL_romaudio.h

@@ -0,0 +1,50 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */

Externals/SDL/src/audio/mint/SDL_mintaudio.c

@@ -0,0 +1,215 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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)
+{
+	unsigned 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();
+	}
+}

Externals/SDL/src/audio/mint/SDL_mintaudio.h

@@ -0,0 +1,157 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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
+
+/* MFP 68901 interrupt sources */
+#ifndef MFP_PARALLEL
+enum {
+	MFP_PARALLEL=0,
+	MFP_DCD,
+	MFP_CTS,
+	MFP_BITBLT,
+	MFP_TIMERD,
+	MFP_BAUDRATE=MFP_TIMERD,
+	MFP_TIMERC,
+	MFP_200HZ=MFP_TIMERC,
+	MFP_ACIA,
+	MFP_DISK,
+	MFP_TIMERB,
+	MFP_HBLANK=MFP_TIMERB,
+	MFP_TERR,
+	MFP_TBE,
+	MFP_RERR,
+	MFP_RBF,
+	MFP_TIMERA,
+	MFP_DMASOUND=MFP_TIMERA,
+	MFP_RING,
+	MFP_MONODETECT
+};
+#endif
+
+/* Xbtimer() timers */
+#ifndef XB_TIMERA
+enum {
+	XB_TIMERA=0,
+	XB_TIMERB,
+	XB_TIMERC,
+	XB_TIMERD
+};
+#endif
+
+/* 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 */

Externals/SDL/src/audio/mint/SDL_mintaudio_dma8.c

@@ -0,0 +1,361 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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 unsigned 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);
+static void Mint_InitAudio(_THIS, SDL_AudioSpec *spec);
+
+/*--- 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)
+{
+	void *oldpile;
+
+	/* Stop replay */
+	oldpile=(void *)Super(0);
+	DMAAUDIO_IO.control=0;
+	Super(oldpile);
+}
+
+static void Mint_UnlockAudio(_THIS)
+{
+	void *oldpile;
+
+	/* Restart replay */
+	oldpile=(void *)Super(0);
+	DMAAUDIO_IO.control=3;
+	Super(oldpile);
+}
+
+static void Mint_CloseAudio(_THIS)
+{
+	void *oldpile;
+
+	/* Stop replay */
+	oldpile=(void *)Super(0);
+	DMAAUDIO_IO.control=0;
+	Super(oldpile);
+
+	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 void Mint_InitAudio(_THIS, SDL_AudioSpec *spec)
+{
+	void *oldpile;
+	unsigned long buffer;
+	unsigned char mode;
+	
+	/* Set replay tracks */
+	if (cookie_snd & SND_16BIT) {
+		Settracks(0,0);
+		Setmontracks(0);
+	}
+
+	oldpile=(void *)Super(0);
+
+	/* Stop currently playing sound */
+	DMAAUDIO_IO.control=0;
+
+	/* 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 (spec->channels==1) {
+		mode |= 1<<7;
+	}
+	DMAAUDIO_IO.sound_ctrl = mode;	
+
+	/* 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"));
+		}
+	}
+
+	/* Go */
+	DMAAUDIO_IO.control = 3;	/* playback + repeat */
+
+	Super(oldpile);
+}
+
+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 */
+}

Externals/SDL/src/audio/mint/SDL_mintaudio_dma8.h

@@ -0,0 +1,85 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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 M<>quignon
+*/
+
+#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 */

Externals/SDL/src/audio/mint/SDL_mintaudio_gsxb.c

@@ -0,0 +1,436 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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 unsigned 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;
+	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)
+{
+}

Externals/SDL/src/audio/mint/SDL_mintaudio_gsxb.h

@@ -0,0 +1,108 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/*
+ * GSXB audio definitions
+ * 
+ * Patrice Mandin
+ */
+
+#ifndef _SDL_mintaudio_gsxb_h
+#define _SDL_mintaudio_gsxb_h
+
+#include <mint/falcon.h>	/* for trap_14_xxx macros */
+
+/* GSXB Cookie */
+
+#define C_GSXB		0x47535842L
+
+/* 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 */

Externals/SDL/src/audio/mint/SDL_mintaudio_it.S

@@ -0,0 +1,281 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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 M<EFBFBD>quignon
+ */
+
+	.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
+
+/*--- Xbios interrupt vector to measure Falcon external clock ---*/
+
+_SDL_MintAudio_XbiosInterruptMeasureClock:          /* 1 mS */
+
+	btst	#0,0xFFFF8901:w	/* state DMA sound */
+	beqs	SDL_MintAudio_EndIntMeasure
+	addql	#1,_SDL_MintAudio_clocktics
+SDL_MintAudio_EndIntMeasure:
+	bclr	#5,0xFFFFFA0F:w	/* Clear service bit */
+	rte
+
+/*--- Xbios interrupt vector ---*/
+
+_SDL_MintAudio_XbiosInterrupt:
+
+	/* Reenable interrupts, so other interrupts can work */
+	movew	#0x2300,sr
+
+	/* Clear service bit, so other MFP interrupts can work */
+	bclr	#5,0xfffffa0f:w
+
+	/* Check if we are not already running */
+	tstw	_SDL_MintAudio_mutex
+	bne	SDL_MintAudio_XbiosEnd
+	notw	_SDL_MintAudio_mutex
+	
+	/* Swap buffers */
+	eorw	#1,_SDL_MintAudio_numbuf
+
+	moveml	d0-d7/a0-a6,sp@-
+
+	/* Save FPU if needed */
+	tstw	_SDL_MintAudio_hasfpu
+	beqs	SDL_MintAudio_Xbios_nofpu1
+	.chip	68060
+	fsave	sp@-
+	fmoveml fpcr/fpsr/fpiar,sp@-
+	fmovemx	fp0-fp7,sp@-
+	.chip	68000
+SDL_MintAudio_Xbios_nofpu1:
+
+	/* Callback */
+	jsr	_SDL_MintAudio_Callback
+
+	/* Restore FPU if needed */
+	tstw	_SDL_MintAudio_hasfpu
+	beqs	SDL_MintAudio_Xbios_nofpu2
+	.chip	68060
+	fmovemx	sp@+,fp0-fp7
+	fmoveml	sp@+,fpcr/fpsr/fpiar
+	frestore	sp@+
+	.chip	68000
+SDL_MintAudio_Xbios_nofpu2:
+
+	/* Reserve space for registers */
+	subl	#savamt,savptr
+
+	/* 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 */
+	addl	#savamt,savptr
+
+	moveml	sp@+,d0-d7/a0-a6
+
+	clrw	_SDL_MintAudio_mutex
+SDL_MintAudio_XbiosEnd:
+	rte
+
+/*--- DMA 8 bits interrupt vector ---*/
+
+_SDL_MintAudio_Dma8Interrupt:
+
+	/* Reenable interrupts, so other interrupts can work */
+	movew	#0x2300,sr
+
+	/* Clear service bit, so other MFP interrupts can work */
+	bclr	#5,0xfffffa0f:w
+
+	/* Check if we are not already running */
+	tstw	_SDL_MintAudio_mutex
+	bne	SDL_MintAudio_Dma8End
+	notw	_SDL_MintAudio_mutex
+	
+	/* Swap buffers */
+	eorw	#1,_SDL_MintAudio_numbuf
+
+	moveml	d0-d1/a0-a1,sp@-
+
+	/* Save FPU if needed */
+	tstw	_SDL_MintAudio_hasfpu
+	beqs	SDL_MintAudio_Dma8_nofpu1
+	.chip	68060
+	fsave	sp@-
+	fmoveml fpcr/fpsr/fpiar,sp@-
+	fmovemx	fp0-fp7,sp@-
+	.chip	68000
+SDL_MintAudio_Dma8_nofpu1:
+
+	/* Callback */
+	jsr	_SDL_MintAudio_Callback
+
+	/* Restore FPU if needed */
+	tstw	_SDL_MintAudio_hasfpu
+	beqs	SDL_MintAudio_Dma8_nofpu2
+	.chip	68060
+	fmovemx	sp@+,fp0-fp7
+	fmoveml	sp@+,fpcr/fpsr/fpiar
+	frestore	sp@+
+	.chip	68000
+SDL_MintAudio_Dma8_nofpu2:
+
+	/* Set new buffer */
+
+	moveq	#0,d0
+
+	movew	_SDL_MintAudio_numbuf,d0
+	lslw	#2,d0
+	lea	_SDL_MintAudio_audiobuf,a0
+	movel	a0@(d0:w),d1
+
+	/* Modify DMA addresses */
+	lea	0xffff8900:w,a0
+
+	moveb	d1,a0@(0x07)	/* Start address */
+	rorl	#8,d1
+	moveb	d1,a0@(0x05)
+	rorl	#8,d1
+	moveb	d1,a0@(0x03)
+	swap	d1
+
+	addl	_SDL_MintAudio_audiosize,d1
+
+	moveb	d1,a0@(0x13)	/* End address */
+	rorl	#8,d1
+	moveb	d1,a0@(0x11)
+	rorl	#8,d1
+	moveb	d1,a0@(0x0f)
+
+	moveml	sp@+,d0-d1/a0-a1
+
+	clrw	_SDL_MintAudio_mutex
+SDL_MintAudio_Dma8End:
+	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
+	bnes	SDL_MintAudio_StfaEnd
+	notw	_SDL_MintAudio_mutex
+	
+	/* Swap buffers */
+	eorw	#1,_SDL_MintAudio_numbuf
+
+	moveml	d0-d7/a0-a6,sp@-
+
+	/* Save FPU if needed */
+	tstw	_SDL_MintAudio_hasfpu
+	beqs	SDL_MintAudio_Stfa_nofpu1
+	.chip	68060
+	fsave	sp@-
+	fmoveml fpcr/fpsr/fpiar,sp@-
+	fmovemx	fp0-fp7,sp@-
+	.chip	68000
+SDL_MintAudio_Stfa_nofpu1:
+
+	/* Callback */
+	jsr	_SDL_MintAudio_Callback
+
+	/* Restore FPU if needed */
+	tstw	_SDL_MintAudio_hasfpu
+	beqs	SDL_MintAudio_Stfa_nofpu2
+	.chip	68060
+	fmovemx	sp@+,fp0-fp7
+	fmoveml	sp@+,fpcr/fpsr/fpiar
+	frestore	sp@+
+	.chip	68000
+SDL_MintAudio_Stfa_nofpu2:
+
+	/* Set new buffer */
+
+	moveq	#0,d0
+	movel	_SDL_MintAudio_stfa,a1
+
+	movew	_SDL_MintAudio_numbuf,d0
+	lslw	#2,d0
+	lea	_SDL_MintAudio_audiobuf,a0
+	movel	a0@(d0:w),d1
+
+	/* Modify STFA replay buffers */
+	movel	d1,a1@(STFA_SOUND_START)
+	addl	_SDL_MintAudio_audiosize,d1
+	movel	d1,a1@(STFA_SOUND_END)
+
+	moveml	sp@+,d0-d7/a0-a6
+
+	clrw	_SDL_MintAudio_mutex
+SDL_MintAudio_StfaEnd:
+	rte

Externals/SDL/src/audio/mint/SDL_mintaudio_mcsn.c

@@ -0,0 +1,404 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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 unsigned 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)
+{
+	unsigned 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, (long *) &cookie_mcsn) != C_FOUND) {
+		DEBUG_PRINT((DEBUG_NAME "no MCSN audio\n"));
+		return(0);
+	}
+
+	/* 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 */
+}

Externals/SDL/src/audio/mint/SDL_mintaudio_mcsn.h

@@ -0,0 +1,59 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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 __attribute__((packed));
+
+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 */

Externals/SDL/src/audio/mint/SDL_mintaudio_stfa.c

@@ -0,0 +1,323 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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 "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 unsigned 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)
+{
+	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, (long *) &cookie_stfa) != C_FOUND) {
+		DEBUG_PRINT((DEBUG_NAME "no STFA audio\n"));
+		return(0);
+	}
+
+	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;
+	Super(oldpile);
+}
+
+static void Mint_UnlockAudio(_THIS)
+{
+	void *oldpile;
+
+	/* Restart replay */
+	oldpile=(void *)Super(0);
+	cookie_stfa->sound_enable=STFA_PLAY_ENABLE|STFA_PLAY_REPEAT;
+	Super(oldpile);
+}
+
+static void Mint_CloseAudio(_THIS)
+{
+	void *oldpile;
+
+	/* Stop replay */
+	oldpile=(void *)Super(0);
+	cookie_stfa->sound_enable=STFA_PLAY_DISABLE;
+	Super(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;
+
+	Super(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 */
+}

Externals/SDL/src/audio/mint/SDL_mintaudio_stfa.h

@@ -0,0 +1,100 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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 C_STFA	0x53544641L	/* Sound treiber f<>r atari (seb/The removers) */
+
+#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 __attribute__((packed));
+
+#endif /* _SDL_mintaudio_stfa_h */

Externals/SDL/src/audio/mint/SDL_mintaudio_xbios.c

@@ -0,0 +1,495 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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 M<>quignon
+*/
+
+#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 "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 unsigned 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;
+
+	Super(oldstack);
+}
+
+static void Mint_CheckExternalClock(_THIS)
+{
+#define SIZE_BUF_CLOCK_MEASURE (44100/10)
+
+	unsigned long cookie_snd;
+	char *buffer;
+	int i, j;
+
+	/* DSP present with its GPIO port ? */
+	if (Getcookie(C__SND, &cookie_snd) == C_NOTFOUND) {
+		return;
+	}
+	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;
+	Uint32 extclock;
+
+	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, dmaclock, 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"));
+	}
+
+	dmaclock = MINTAUDIO_frequencies[MINTAUDIO_numfreq].masterclock;
+	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 */
+}

Externals/SDL/src/audio/mme/SDL_mmeaudio.c

@@ -0,0 +1,264 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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;
+    }
+}
+

Externals/SDL/src/audio/mme/SDL_mmeaudio.h

@@ -0,0 +1,51 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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 */

Externals/SDL/src/audio/nas/SDL_nasaudio.c

@@ -0,0 +1,423 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+
+    This driver was written by:
+    Erik Inge Bols<6C>
+    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);
+}

Externals/SDL/src/audio/nas/SDL_nasaudio.h

@@ -0,0 +1,62 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+
+    This driver was written by:
+    Erik Inge Bols<6C>
+    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 */
+

Externals/SDL/src/audio/nds/SDL_ndsaudio.c

@@ -0,0 +1,335 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Allow access to 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);
+}

Externals/SDL/src/audio/nds/SDL_ndsaudio.h

@@ -0,0 +1,40 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */

Externals/SDL/src/audio/nds/sound9.c

@@ -0,0 +1,61 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+#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;
+}

Externals/SDL/src/audio/nds/soundcommon.h

@@ -0,0 +1,80 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef __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

Externals/SDL/src/audio/nto/SDL_nto_audio.c

@@ -0,0 +1,507 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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);
+}

Externals/SDL/src/audio/nto/SDL_nto_audio.h

@@ -0,0 +1,68 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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__ */

Externals/SDL/src/audio/paudio/SDL_paudio.c

@@ -0,0 +1,511 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    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;
+}
+

Externals/SDL/src/audio/paudio/SDL_paudio.h

@@ -0,0 +1,57 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */

Externals/SDL/src/audio/pulse/SDL_pulseaudio.c

@@ -0,0 +1,534 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    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 "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);
+
+#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
+);
+
+pa_mainloop * (*SDL_NAME(pa_mainloop_new))(void);
+pa_mainloop_api * (*SDL_NAME(pa_mainloop_get_api))(pa_mainloop *m);
+int (*SDL_NAME(pa_mainloop_iterate))(pa_mainloop *m, int block, int *retval);
+void (*SDL_NAME(pa_mainloop_free))(pa_mainloop *m);
+
+pa_operation_state_t (*SDL_NAME(pa_operation_get_state))(pa_operation *o);
+void (*SDL_NAME(pa_operation_cancel))(pa_operation *o);
+void (*SDL_NAME(pa_operation_unref))(pa_operation *o);
+
+pa_context * (*SDL_NAME(pa_context_new))(
+	pa_mainloop_api *m, const char *name);
+int (*SDL_NAME(pa_context_connect))(
+	pa_context *c, const char *server,
+	pa_context_flags_t flags, const pa_spawn_api *api);
+pa_context_state_t (*SDL_NAME(pa_context_get_state))(pa_context *c);
+void (*SDL_NAME(pa_context_disconnect))(pa_context *c);
+void (*SDL_NAME(pa_context_unref))(pa_context *c);
+
+pa_stream * (*SDL_NAME(pa_stream_new))(pa_context *c,
+	const char *name, const pa_sample_spec *ss, const pa_channel_map *map);
+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);
+pa_stream_state_t (*SDL_NAME(pa_stream_get_state))(pa_stream *s);
+size_t (*SDL_NAME(pa_stream_writable_size))(pa_stream *s);
+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);
+pa_operation * (*SDL_NAME(pa_stream_drain))(pa_stream *s,
+	pa_stream_success_cb_t cb, void *userdata);
+int (*SDL_NAME(pa_stream_disconnect))(pa_stream *s);
+void (*SDL_NAME(pa_stream_unref))(pa_stream *s);
+
+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)		},
+};
+
+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);
+	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->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 stream_drain_complete(pa_stream *s, int success, void *userdata) {
+}
+
+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;
+	}
+	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);
+	}
+
+	mainloop_api = SDL_NAME(pa_mainloop_get_api)(mainloop);
+	if (!(context = SDL_NAME(pa_context_new)(mainloop_api, get_progname()))) {
+		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);
+}

Externals/SDL/src/audio/pulse/SDL_pulseaudio.h

@@ -0,0 +1,71 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    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;
+
+	/* 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 */
+

Externals/SDL/src/audio/sun/SDL_sunaudio.c

@@ -0,0 +1,432 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Allow access to 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 <20>-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);
+}

Externals/SDL/src/audio/sun/SDL_sunaudio.h

@@ -0,0 +1,55 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */

Externals/SDL/src/audio/symbian/SDL_epocaudio.cpp

@@ -0,0 +1,614 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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();
+	}
+
+
+

Externals/SDL/src/audio/symbian/SDL_epocaudio.h

@@ -0,0 +1,37 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU 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 */

Externals/SDL/src/audio/symbian/streamplayer.cpp

@@ -0,0 +1,279 @@
+#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);
+	*/
+	
+
+
+
+

Externals/SDL/src/audio/symbian/streamplayer.h

@@ -0,0 +1,89 @@
+#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
+

Externals/SDL/src/audio/ums/SDL_umsaudio.c

@@ -0,0 +1,547 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    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 );
+}
+

Externals/SDL/src/audio/ums/SDL_umsaudio.h

@@ -0,0 +1,50 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    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 */
+

Externals/SDL/src/audio/windib/SDL_dibaudio.c

@@ -0,0 +1,322 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Allow access to 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);
+}

Externals/SDL/src/audio/windib/SDL_dibaudio.h

@@ -0,0 +1,49 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */

Externals/SDL/src/audio/windx5/SDL_dx5audio.c

@@ -0,0 +1,705 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* Allow access to 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);
+}
+

Externals/SDL/src/audio/windx5/SDL_dx5audio.h

@@ -0,0 +1,55 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifndef _SDL_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 */

Externals/SDL/src/audio/windx5/directx.h

@@ -0,0 +1,84 @@
+
+#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
+
+#ifdef __GNUC__
+#define NONAMELESSUNION
+#endif
+#include <ddraw.h>
+#include <dsound.h>
+#include <dinput.h>
+
+#endif /* _directx_h */

Externals/SDL/src/cdrom/SDL_cdrom.c

@@ -0,0 +1,341 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* This is the CD-audio control API for Simple DirectMedia Layer */
+
+#include "SDL_cdrom.h"
+#include "SDL_syscdrom.h"
+
+#if !defined(__MACOS__)
+#define CLIP_FRAMES	10	/* Some CD-ROMs won't go all the way */
+#endif
+
+static int SDL_cdinitted = 0;
+static SDL_CD *default_cdrom;
+
+/* The system level CD-ROM control functions */
+struct CDcaps SDL_CDcaps = {
+	NULL,					/* Name */
+	NULL,					/* Open */
+	NULL,					/* GetTOC */
+	NULL,					/* Status */
+	NULL,					/* Play */
+	NULL,					/* Pause */
+	NULL,					/* Resume */
+	NULL,					/* Stop */
+	NULL,					/* Eject */
+	NULL,					/* Close */
+};
+int SDL_numcds;
+
+int SDL_CDROMInit(void)
+{
+	int retval;
+
+	SDL_numcds = 0;
+	retval = SDL_SYS_CDInit();
+	if ( retval == 0 ) {
+		SDL_cdinitted = 1;
+	}
+	default_cdrom = NULL;
+	return(retval);
+}
+
+/* Check to see if the CD-ROM subsystem has been initialized */
+static int CheckInit(int check_cdrom, SDL_CD **cdrom)
+{
+	int okay;
+
+	okay = SDL_cdinitted;
+	if ( check_cdrom && (*cdrom == NULL) ) {
+		*cdrom = default_cdrom;
+		if ( *cdrom == NULL ) {
+			SDL_SetError("CD-ROM not opened");
+			okay = 0;
+		}
+	}
+	if ( ! SDL_cdinitted ) {
+		SDL_SetError("CD-ROM subsystem not initialized");
+	}
+	return(okay);
+}
+
+int SDL_CDNumDrives(void)
+{
+	if ( ! CheckInit(0, NULL) ) {
+		return(-1);
+	}
+	return(SDL_numcds);
+}
+
+const char *SDL_CDName(int drive)
+{
+	if ( ! CheckInit(0, NULL) ) {
+		return(NULL);
+	}
+	if ( drive >= SDL_numcds ) {
+		SDL_SetError("Invalid CD-ROM drive index");
+		return(NULL);
+	}
+	if ( SDL_CDcaps.Name ) {
+		return(SDL_CDcaps.Name(drive));
+	} else {
+		return("");
+	}
+}
+
+SDL_CD *SDL_CDOpen(int drive)
+{
+	struct SDL_CD *cdrom;
+
+	if ( ! CheckInit(0, NULL) ) {
+		return(NULL);
+	}
+	if ( drive >= SDL_numcds ) {
+		SDL_SetError("Invalid CD-ROM drive index");
+		return(NULL);
+	}
+	cdrom = (SDL_CD *)SDL_malloc(sizeof(*cdrom));
+	if ( cdrom == NULL ) {
+		SDL_OutOfMemory();
+		return(NULL);
+	}
+	SDL_memset(cdrom, 0, sizeof(*cdrom));
+	cdrom->id = SDL_CDcaps.Open(drive);
+	if ( cdrom->id < 0 ) {
+		SDL_free(cdrom);
+		return(NULL);
+	}
+	default_cdrom = cdrom;
+	return(cdrom);
+}
+
+CDstatus SDL_CDStatus(SDL_CD *cdrom)
+{
+	CDstatus status;
+	int i;
+	Uint32 position;
+
+	/* Check if the CD-ROM subsystem has been initialized */
+	if ( ! CheckInit(1, &cdrom) ) {
+		return(CD_ERROR);
+	}
+
+	/* Get the current status of the drive */
+	cdrom->numtracks = 0;
+	cdrom->cur_track = 0;
+	cdrom->cur_frame = 0;
+	status = SDL_CDcaps.Status(cdrom, &i);
+	position = (Uint32)i;
+	cdrom->status = status;
+
+	/* Get the table of contents, if there's a CD available */
+	if ( CD_INDRIVE(status) ) {
+		if ( SDL_CDcaps.GetTOC(cdrom) < 0 ) {
+			status = CD_ERROR;
+		}
+		/* If the drive is playing, get current play position */
+		if ( (status == CD_PLAYING) || (status == CD_PAUSED) ) {
+			for ( i=1; cdrom->track[i].offset <= position; ++i ) {
+				/* Keep looking */;
+			}
+#ifdef DEBUG_CDROM
+  fprintf(stderr, "Current position: %d, track = %d (offset is %d)\n",
+				position, i-1, cdrom->track[i-1].offset);
+#endif
+			cdrom->cur_track = i-1;
+			position -= cdrom->track[cdrom->cur_track].offset;
+			cdrom->cur_frame = position;
+		}
+	}
+	return(status);
+}
+
+int SDL_CDPlayTracks(SDL_CD *cdrom,
+			int strack, int sframe, int ntracks, int nframes)
+{
+	int etrack, eframe;
+	int start, length;
+
+	/* Check if the CD-ROM subsystem has been initialized */
+	if ( ! CheckInit(1, &cdrom) ) {
+		return(CD_ERROR);
+	}
+
+	/* Determine the starting and ending tracks */
+	if ( (strack < 0) || (strack >= cdrom->numtracks) ) {
+		SDL_SetError("Invalid starting track");
+		return(CD_ERROR);
+	}
+	if ( ! ntracks && ! nframes ) {
+		etrack = cdrom->numtracks;
+		eframe = 0;
+	} else {
+		etrack = strack+ntracks;
+		if ( etrack == strack ) {
+			eframe = sframe + nframes;
+		} else {
+			eframe = nframes;
+		}
+	}
+	if ( etrack > cdrom->numtracks ) {
+		SDL_SetError("Invalid play length");
+		return(CD_ERROR);
+	}
+
+	/* Skip data tracks and verify frame offsets */
+	while ( (strack <= etrack) &&
+			(cdrom->track[strack].type == SDL_DATA_TRACK) ) {
+		++strack;
+	}
+	if ( sframe >= (int)cdrom->track[strack].length ) {
+		SDL_SetError("Invalid starting frame for track %d", strack);
+		return(CD_ERROR);
+	}
+	while ( (etrack > strack) &&
+			(cdrom->track[etrack-1].type == SDL_DATA_TRACK) ) {
+		--etrack;
+	}
+	if ( eframe > (int)cdrom->track[etrack].length ) {
+		SDL_SetError("Invalid ending frame for track %d", etrack);
+		return(CD_ERROR);
+	}
+
+	/* Determine start frame and play length */
+	start = (cdrom->track[strack].offset+sframe);
+	length = (cdrom->track[etrack].offset+eframe)-start;
+#ifdef CLIP_FRAMES
+	/* I've never seen this necessary, but xmcd does it.. */
+	length -= CLIP_FRAMES;	/* CLIP_FRAMES == 10 */
+#endif
+	if ( length < 0 ) {
+		return(0);
+	}
+
+	/* Play! */
+#ifdef DEBUG_CDROM
+  fprintf(stderr, "Playing %d frames at offset %d\n", length, start);
+#endif
+	return(SDL_CDcaps.Play(cdrom, start, length));
+}
+
+int SDL_CDPlay(SDL_CD *cdrom, int sframe, int length)
+{
+	/* Check if the CD-ROM subsystem has been initialized */
+	if ( ! CheckInit(1, &cdrom) ) {
+		return(CD_ERROR);
+	}
+
+	return(SDL_CDcaps.Play(cdrom, sframe, length));
+}
+
+int SDL_CDPause(SDL_CD *cdrom)
+{
+	CDstatus status;
+	int retval;
+
+	/* Check if the CD-ROM subsystem has been initialized */
+	if ( ! CheckInit(1, &cdrom) ) {
+		return(CD_ERROR);
+	}
+
+	status = SDL_CDcaps.Status(cdrom, NULL);
+	switch (status) {
+		case CD_PLAYING:
+			retval = SDL_CDcaps.Pause(cdrom);
+			break;
+		default:
+			retval = 0;
+			break;
+	}
+	return(retval);
+}
+
+int SDL_CDResume(SDL_CD *cdrom)
+{
+	CDstatus status;
+	int retval;
+
+	/* Check if the CD-ROM subsystem has been initialized */
+	if ( ! CheckInit(1, &cdrom) ) {
+		return(CD_ERROR);
+	}
+
+	status = SDL_CDcaps.Status(cdrom, NULL);
+	switch (status) {
+		case CD_PAUSED:
+			retval = SDL_CDcaps.Resume(cdrom);
+		default:
+			retval = 0;
+			break;
+	}
+	return(retval);
+}
+
+int SDL_CDStop(SDL_CD *cdrom)
+{
+	CDstatus status;
+	int retval;
+
+	/* Check if the CD-ROM subsystem has been initialized */
+	if ( ! CheckInit(1, &cdrom) ) {
+		return(CD_ERROR);
+	}
+
+	status = SDL_CDcaps.Status(cdrom, NULL);
+	switch (status) {
+		case CD_PLAYING:
+		case CD_PAUSED:
+			retval = SDL_CDcaps.Stop(cdrom);
+		default:
+			retval = 0;
+			break;
+	}
+	return(retval);
+}
+
+int SDL_CDEject(SDL_CD *cdrom)
+{
+	/* Check if the CD-ROM subsystem has been initialized */
+	if ( ! CheckInit(1, &cdrom) ) {
+		return(CD_ERROR);
+	}
+	return(SDL_CDcaps.Eject(cdrom));
+}
+
+void SDL_CDClose(SDL_CD *cdrom)
+{
+	/* Check if the CD-ROM subsystem has been initialized */
+	if ( ! CheckInit(1, &cdrom) ) {
+		return;
+	}
+	SDL_CDcaps.Close(cdrom);
+	SDL_free(cdrom);
+	default_cdrom = NULL;
+}
+
+void SDL_CDROMQuit(void)
+{
+	SDL_SYS_CDQuit();
+	SDL_cdinitted = 0;
+}

Externals/SDL/src/cdrom/SDL_syscdrom.h

@@ -0,0 +1,76 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is SDL_free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+/* This is the system specific header for the SDL CD-ROM API */
+
+/* Structure of CD audio control functions */
+extern struct CDcaps {
+	/* Get the name of the specified drive */
+	const char *(*Name)(int drive);
+
+	/* Open the specified drive, returning a drive id, or -1 on error */
+	int (*Open)(int drive);
+
+	/* Get table-of-contents (number of tracks + track info) for disk.
+	   The TOC information should be stored in the cdrom structure.
+	   This function should return 0 on success, or -1 on error.
+	 */
+	int (*GetTOC)(SDL_CD *cdrom);
+
+	/* Return the current status and play position, in frames, of the
+	   drive.  'position' may be NULL, and if so, should be ignored.
+	 */
+	CDstatus (*Status)(SDL_CD *cdrom, int *position);
+
+	/* Play from frame 'start' to 'start+len' */
+	int (*Play)(SDL_CD *cdrom, int start, int len); 
+
+	/* Pause play */
+	int (*Pause)(SDL_CD *cdrom);
+
+	/* Resume play */
+	int (*Resume)(SDL_CD *cdrom);
+
+	/* Stop play */
+	int (*Stop)(SDL_CD *cdrom);
+
+	/* Eject the current disk */
+	int (*Eject)(SDL_CD *cdrom);
+
+	/* Close the specified drive */
+	void (*Close)(SDL_CD *cdrom);
+} SDL_CDcaps;
+
+/* The number of available CD-ROM drives on the system */
+extern int SDL_numcds;
+
+/* Function to scan the system for CD-ROM drives and fill SDL_CDcaps.
+ * This function should set SDL_numcds to the number of available CD
+ * drives.  Drive 0 should be the system default CD-ROM.
+ * It should return 0, or -1 on an unrecoverable fatal error.
+*/
+extern int  SDL_SYS_CDInit(void);
+
+/* Function to perform any system-specific CD-ROM related cleanup */
+extern void SDL_SYS_CDQuit(void);
+

Externals/SDL/src/cdrom/aix/SDL_syscdrom.c

@@ -0,0 +1,660 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    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;
+}
+
+static const char *SDL_SYS_CDName(int drive)
+{
+	return(SDL_cdlist[drive]);
+}
+
+static int SDL_SYS_CDOpen(int drive)
+{
+    int   fd;
+    char* lastsl;
+    char* cdromname;
+    size_t len;
+
+    /*
+     * We found /dev/cd? drives and that is in our list. But we can
+     * open only the /dev/rcd? versions of those devices for Audio CD.
+     */
+    len = SDL_strlen(SDL_cdlist[drive])+2;
+    cdromname = (char*)SDL_malloc(len);
+    SDL_strlcpy(cdromname,SDL_cdlist[drive],len);
+    lastsl = SDL_strrchr(cdromname,'/');
+    if (lastsl) {
+	*lastsl = 0;
+	SDL_strlcat(cdromname,"/r",len);
+	lastsl = SDL_strrchr(SDL_cdlist[drive],'/');
+	if (lastsl) {
+	    lastsl++;
+	    SDL_strlcat(cdromname,lastsl,len);
+	}
+    }
+
+#ifdef DEBUG_CDROM
+    fprintf(stderr, "Should open drive %s, opening %s\n", SDL_cdlist[drive], cdromname);
+#endif
+
+    /*
+     * Use exclusive access. Don't use SC_DIAGNOSTICS as xmcd does because they
+     * require root priviledges, and we don't want that. SC_SINGLE provides
+     * exclusive access with less trouble.
+     */
+    fd = openx(cdromname, O_RDONLY, NULL, SC_SINGLE);
+    if ( fd < 0 )
+    {
+#ifdef DEBUG_CDROM
+            fprintf(stderr, "Could not open drive %s (%s)\n", cdromname, strerror(errno));
+#endif
+    }
+    else
+    {
+	struct mode_form_op cdMode;
+	int                 ret;
+#ifdef DEBUG_CDROM
+	cdMode.action = CD_GET_MODE;
+	ret = SDL_SYS_CDioctl(fd, DK_CD_MODE, &cdMode);
+	if ( ret < 0 ) {
+            fprintf(stderr,
+	            "Could not get drive mode for %s (%s)\n",
+		    cdromname, strerror(errno));
+	} else {
+	    switch(cdMode.cd_mode_form) {
+		case CD_MODE1 :
+                    fprintf(stderr,
+	                "Drive mode for %s is %s\n",
+		        cdromname, "CD-ROM Data Mode 1");
+		    break;
+		case CD_MODE2_FORM1 :
+                    fprintf(stderr,
+	                "Drive mode for %s is %s\n",
+		        cdromname, "CD-ROM XA Data Mode 2 Form 1");
+		    break;
+		case CD_MODE2_FORM2 :
+                    fprintf(stderr,
+	                "Drive mode for %s is %s\n",
+		        cdromname, "CD-ROM XA Data Mode 2 Form 2");
+		    break;
+		case CD_DA :
+                    fprintf(stderr,
+	                "Drive mode for %s is %s\n",
+		        cdromname, "CD-DA");
+		    break;
+		default :
+                    fprintf(stderr,
+	                "Drive mode for %s is %s\n",
+		        cdromname, "unknown");
+		    break;
+	    }
+	}
+#endif
+
+	cdMode.action       = CD_CHG_MODE;
+	cdMode.cd_mode_form = CD_DA;
+	ret = SDL_SYS_CDioctl(fd, DK_CD_MODE, &cdMode);
+	if ( ret < 0 ) {
+#ifdef DEBUG_CDROM
+            fprintf(stderr,
+	            "Could not set drive mode for %s (%s)\n",
+		    cdromname, strerror(errno));
+#endif
+            SDL_SetError("ioctl() error: Could not set CD drive mode, %s",
+	                 strerror(errno));
+	} else {
+#ifdef DEBUG_CDROM
+            fprintf(stderr,
+	            "Drive mode for %s set to CD_DA\n",
+		    cdromname);
+#endif
+	}
+    }
+    SDL_free(cdromname);
+    return fd;
+}
+
+static int SDL_SYS_CDGetTOC(SDL_CD *cdrom)
+{
+    struct cd_audio_cmd cmd;
+    struct cd_audio_cmd entry;
+    int                 i;
+    int                 okay;
+
+    cmd.audio_cmds = CD_TRK_INFO_AUDIO;
+    cmd.msf_flag   = FALSE;
+    if ( SDL_SYS_CDioctl(cdrom->id, DKAUDIO, &cmd) < 0 ) {
+	return -1;
+    }
+
+    okay = 0;
+    cdrom->numtracks = cmd.indexing.track_index.last_track
+		     - cmd.indexing.track_index.first_track+1;
+    if ( cdrom->numtracks > SDL_MAX_TRACKS ) {
+        cdrom->numtracks = SDL_MAX_TRACKS;
+    }
+
+    /* Read all the track TOC entries */
+    for ( i=0; i<=cdrom->numtracks; ++i ) {
+        if ( i == cdrom->numtracks ) {
+            cdrom->track[i].id = 0xAA;;
+        } else {
+            cdrom->track[i].id = cmd.indexing.track_index.first_track+i;
+        }
+        entry.audio_cmds         = CD_GET_TRK_MSF;
+	entry.indexing.track_msf.track = cdrom->track[i].id;
+	if ( SDL_SYS_CDioctl(cdrom->id, DKAUDIO, &entry) < 0 ) {
+            break;
+        } else {
+            cdrom->track[i].type = 0;    /* don't know how to detect 0x04 data track */
+            cdrom->track[i].offset = MSF_TO_FRAMES(
+                entry.indexing.track_msf.mins,
+                entry.indexing.track_msf.secs,
+                entry.indexing.track_msf.frames);
+            cdrom->track[i].length = 0;
+            if ( i > 0 ) {
+                cdrom->track[i-1].length = cdrom->track[i].offset
+		                         - cdrom->track[i-1].offset;
+            }
+        }
+    }
+    if ( i == (cdrom->numtracks+1) ) {
+        okay = 1;
+    }
+    return(okay ? 0 : -1);
+}
+
+/* Get CD-ROM status */
+static CDstatus SDL_SYS_CDStatus(SDL_CD *cdrom, int *position)
+{
+    CDstatus            status;
+    struct cd_audio_cmd cmd;
+    cmd.audio_cmds = CD_INFO_AUDIO;
+
+    if ( SDL_SYS_CDioctl(cdrom->id, DKAUDIO, &cmd) < 0 ) {
+#ifdef DEBUG_CDROM
+    fprintf(stderr, "ioctl failed in SDL_SYS_CDStatus (%s)\n", SDL_GetError());
+#endif
+        status = CD_ERROR;
+    } else {
+        switch (cmd.status) {
+            case CD_NO_AUDIO:
+            case CD_COMPLETED:
+                status = CD_STOPPED;
+                break;
+            case CD_PLAY_AUDIO:
+                status = CD_PLAYING;
+                break;
+            case CD_PAUSE_AUDIO:
+                status = CD_PAUSED;
+                break;
+            case CD_NOT_VALID:
+#ifdef DEBUG_CDROM
+    fprintf(stderr, "cdStatus failed with CD_NOT_VALID\n");
+#endif
+                status = CD_ERROR;
+                break;
+            case CD_STATUS_ERROR:
+#ifdef DEBUG_CDROM
+    fprintf(stderr, "cdStatus failed with CD_STATUS_ERROR\n");
+#endif
+                status = CD_ERROR;
+                break;
+            default:
+#ifdef DEBUG_CDROM
+    fprintf(stderr, "cdStatus failed with unknown error\n");
+#endif
+                status = CD_ERROR;
+                break;
+        }
+    }
+    if ( position ) {
+        if ( status == CD_PLAYING || (status == CD_PAUSED) ) {
+            *position = MSF_TO_FRAMES( cmd.indexing.info_audio.current_mins,
+                                       cmd.indexing.info_audio.current_secs,
+                                       cmd.indexing.info_audio.current_frames);
+        } else {
+            *position = 0;
+        }
+    }
+    return status;
+}
+
+/* Start play */
+static int SDL_SYS_CDPlay(SDL_CD *cdrom, int start, int length)
+{
+    struct cd_audio_cmd cmd;
+
+    /*
+     * My CD Rom is muted by default. I think I read that this is new with
+     * AIX 4.3. SDL does not change the volume, so I need a kludge. Maybe
+     * its better to do this elsewhere?
+     */
+    cmd.audio_cmds = CD_PLAY_AUDIO | CD_SET_VOLUME;
+    cmd.msf_flag   = TRUE;
+    FRAMES_TO_MSF(start,
+                  &cmd.indexing.msf.first_mins,
+                  &cmd.indexing.msf.first_secs,
+                  &cmd.indexing.msf.first_frames);
+    FRAMES_TO_MSF(start+length,
+                  &cmd.indexing.msf.last_mins,
+                  &cmd.indexing.msf.last_secs,
+                  &cmd.indexing.msf.last_frames);
+    cmd.volume_type     = CD_VOLUME_ALL;
+    cmd.all_channel_vol = 255;   /* This is a uchar. What is a good value? No docu! */
+    cmd.out_port_0_sel  = CD_AUDIO_CHNL_0;
+    cmd.out_port_1_sel  = CD_AUDIO_CHNL_1;
+    cmd.out_port_2_sel  = CD_AUDIO_CHNL_2;
+    cmd.out_port_3_sel  = CD_AUDIO_CHNL_3;
+
+#ifdef DEBUG_CDROM
+  fprintf(stderr, "Trying to play from %d:%d:%d to %d:%d:%d\n",
+	cmd.indexing.msf.first_mins,
+	cmd.indexing.msf.first_secs,
+	cmd.indexing.msf.first_frames,
+	cmd.indexing.msf.last_mins,
+	cmd.indexing.msf.last_secs,
+	cmd.indexing.msf.last_frames);
+#endif
+	return(SDL_SYS_CDioctl(cdrom->id, DKAUDIO, &cmd));
+}
+
+/* Pause play */
+static int SDL_SYS_CDPause(SDL_CD *cdrom)
+{
+    struct cd_audio_cmd cmd;
+    cmd.audio_cmds = CD_PAUSE_AUDIO;
+    return(SDL_SYS_CDioctl(cdrom->id, DKAUDIO, &cmd));
+}
+
+/* Resume play */
+static int SDL_SYS_CDResume(SDL_CD *cdrom)
+{
+    struct cd_audio_cmd cmd;
+    cmd.audio_cmds = CD_RESUME_AUDIO;
+    return(SDL_SYS_CDioctl(cdrom->id, DKAUDIO, &cmd));
+}
+
+/* Stop play */
+static int SDL_SYS_CDStop(SDL_CD *cdrom)
+{
+    struct cd_audio_cmd cmd;
+    cmd.audio_cmds = CD_STOP_AUDIO;
+    return(SDL_SYS_CDioctl(cdrom->id, DKAUDIO, &cmd));
+}
+
+/* Eject the CD-ROM */
+static int SDL_SYS_CDEject(SDL_CD *cdrom)
+{
+    return(SDL_SYS_CDioctl(cdrom->id, DKEJECT, 0));
+}
+
+/* Close the CD-ROM handle */
+static void SDL_SYS_CDClose(SDL_CD *cdrom)
+{
+    close(cdrom->id);
+}
+
+void SDL_SYS_CDQuit(void)
+{
+	int i;
+
+	if ( SDL_numcds > 0 ) {
+		for ( i=0; i<SDL_numcds; ++i ) {
+			SDL_free(SDL_cdlist[i]);
+		}
+		SDL_numcds = 0;
+	}
+}
+
+#endif /* SDL_CDROM_AIX */

Externals/SDL/src/cdrom/beos/SDL_syscdrom.cc

@@ -0,0 +1,412 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifdef SDL_CDROM_BEOS
+
+/* Functions for system-level CD-ROM audio control on BeOS
+   (not completely implemented yet)
+ */
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <unistd.h>
+
+#include <scsi.h>
+#include <Directory.h>
+#include <Entry.h>
+#include <Path.h>
+
+#include "SDL_cdrom.h"
+extern "C" {
+#include "../SDL_syscdrom.h"
+}
+
+/* Constants to help us get at the SCSI table-of-contents info */
+#define CD_NUMTRACKS(toc)	toc.toc_data[3]
+#define CD_TRACK(toc, track)	(&toc.toc_data[6+(track)*8])
+#define CD_TRACK_N(toc, track)	CD_TRACK(toc, track)[0]
+#define CD_TRACK_M(toc, track)	CD_TRACK(toc, track)[3]
+#define CD_TRACK_S(toc, track)	CD_TRACK(toc, track)[4]
+#define CD_TRACK_F(toc, track)	CD_TRACK(toc, track)[5]
+
+/* Constants to help us get at the SCSI position info */
+#define POS_TRACK(pos)	pos.position[6]
+#define POS_ABS_M(pos)	pos.position[9]
+#define POS_ABS_S(pos)	pos.position[10]
+#define POS_ABS_F(pos)	pos.position[11]
+#define POS_REL_M(pos)	pos.position[13]
+#define POS_REL_S(pos)	pos.position[14]
+#define POS_REL_F(pos)	pos.position[15]
+
+/* 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];
+
+/* 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);
+int try_dir(const char *directory);
+
+
+/* Check a drive to see if it is a CD-ROM */
+static int CheckDrive(char *drive)
+{
+	struct stat stbuf;
+	int is_cd, cdfd;
+	device_geometry 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;
+	cdfd = open(drive, 0);
+	if ( cdfd >= 0 ) {
+		if ( ioctl(cdfd, B_GET_GEOMETRY, &info) == B_NO_ERROR ) {
+			if ( info.device_type == B_CD ) {
+				is_cd = 1;
+			}
+		}
+		close(cdfd);
+	} else {
+		/* This can happen when the drive is open .. (?) */;
+		is_cd = 1;
+	}
+	return(is_cd);
+}
+
+/* Add a CD-ROM drive to our list of valid drives */
+static void AddDrive(char *drive)
+{
+	int i;
+	size_t len;
+
+	if ( SDL_numcds < MAX_DRIVES ) {
+		/* Add this drive to our list */
+		i = SDL_numcds;
+		len = SDL_strlen(drive)+1;
+		SDL_cdlist[i] = (char *)SDL_malloc(len);
+		if ( SDL_cdlist[i] == NULL ) {
+			SDL_OutOfMemory();
+			return;
+		}
+		SDL_strlcpy(SDL_cdlist[i], drive, len);
+		++SDL_numcds;
+#ifdef CDROM_DEBUG
+  fprintf(stderr, "Added CD-ROM drive: %s\n", drive);
+#endif
+	}
+}
+
+/* IDE bus scanning magic */
+enum {
+	IDE_GET_DEVICES_INFO = B_DEVICE_OP_CODES_END + 50,
+};
+struct ide_ctrl_info {
+	bool	ide_0_present;
+	bool	ide_0_master_present;
+	bool	ide_0_slave_present;
+	int	ide_0_master_type;
+	int	ide_0_slave_type;
+	bool	ide_1_present;
+	bool	ide_1_master_present;
+	bool	ide_1_slave_present;
+	int	ide_1_master_type;
+	int	ide_1_slave_type;
+};
+
+int  SDL_SYS_CDInit(void)
+{
+	char *SDLcdrom;
+	int raw_fd;
+	struct ide_ctrl_info info;
+
+	/* 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';
+				}
+				if ( CheckDrive(SDLcdrom) > 0 ) {
+					AddDrive(SDLcdrom);
+				}
+				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);
+		}
+	}
+	
+	/* Scan the system for CD-ROM drives */
+	try_dir("/dev/disk");
+	return 0;
+}
+
+
+int try_dir(const char *directory)
+{ 
+	BDirectory dir; 
+	dir.SetTo(directory); 
+	if(dir.InitCheck() != B_NO_ERROR) { 
+		return false; 
+	} 
+	dir.Rewind(); 
+	BEntry entry; 
+	while(dir.GetNextEntry(&entry) >= 0) { 
+		BPath path; 
+		const char *name; 
+		entry_ref e; 
+		
+		if(entry.GetPath(&path) != B_NO_ERROR) 
+			continue; 
+		name = path.Path(); 
+		
+		if(entry.GetRef(&e) != B_NO_ERROR) 
+			continue; 
+
+		if(entry.IsDirectory()) { 
+			if(SDL_strcmp(e.name, "floppy") == 0) 
+				continue; /* ignore floppy (it is not silent)  */
+			int devfd = try_dir(name);
+			if(devfd >= 0)
+				return devfd;
+		} 
+		else { 
+			int devfd; 
+			device_geometry g; 
+
+			if(SDL_strcmp(e.name, "raw") != 0) 
+				continue; /* ignore partitions */
+
+			devfd = open(name, O_RDONLY); 
+			if(devfd < 0) 
+				continue; 
+
+			if(ioctl(devfd, B_GET_GEOMETRY, &g, sizeof(g)) >= 0) {
+				if(g.device_type == B_CD)
+				{
+				AddDrive(strdup(name));
+				}
+			}
+			close(devfd);
+		} 
+	}
+	return B_ERROR;
+}
+
+
+/* General ioctl() CD-ROM command function */
+static int SDL_SYS_CDioctl(int index, int command, void *arg)
+{
+	int okay;
+	int fd;
+
+	okay = 0;
+	fd = open(SDL_cdlist[index], 0);
+	if ( fd >= 0 ) {
+		if ( ioctl(fd, command, arg) == B_NO_ERROR ) {
+			okay = 1;
+		}
+		close(fd);
+	}
+	return(okay ? 0 : -1);
+}
+
+static const char *SDL_SYS_CDName(int drive)
+{
+	return(SDL_cdlist[drive]);
+} 
+
+static int SDL_SYS_CDOpen(int drive)
+{
+	return(drive);
+}
+
+static int SDL_SYS_CDGetTOC(SDL_CD *cdrom)
+{
+	int i;
+	scsi_toc toc;
+
+	if ( SDL_SYS_CDioctl(cdrom->id, B_SCSI_GET_TOC, &toc) == 0 ) {
+		cdrom->numtracks = CD_NUMTRACKS(toc);
+		if ( cdrom->numtracks > SDL_MAX_TRACKS ) {
+			cdrom->numtracks = SDL_MAX_TRACKS;
+		}
+		for ( i=0; i<=cdrom->numtracks; ++i ) {
+			cdrom->track[i].id = CD_TRACK_N(toc, i);
+			/* FIXME:  How do we tell on BeOS? */
+			cdrom->track[i].type = SDL_AUDIO_TRACK;
+			cdrom->track[i].offset = MSF_TO_FRAMES(
+							CD_TRACK_M(toc, i),
+							CD_TRACK_S(toc, i),
+							CD_TRACK_F(toc, i));
+			cdrom->track[i].length = 0;
+			if ( i > 0 ) {
+				cdrom->track[i-1].length =
+						cdrom->track[i].offset-
+						cdrom->track[i-1].offset;
+			}
+		}
+		return(0);
+	} else {
+		return(-1);
+	}
+}
+
+/* Get CD-ROM status */
+static CDstatus SDL_SYS_CDStatus(SDL_CD *cdrom, int *position)
+{
+	CDstatus status;
+	int fd;
+	int cur_frame;
+	scsi_position pos;
+
+	fd = open(SDL_cdlist[cdrom->id], 0);
+	cur_frame = 0;
+	if ( fd >= 0 ) {
+		if ( ioctl(fd, B_SCSI_GET_POSITION, &pos) == B_NO_ERROR ) {
+			cur_frame = MSF_TO_FRAMES(
+				POS_ABS_M(pos), POS_ABS_S(pos), POS_ABS_F(pos));
+		}
+		if ( ! pos.position[1] || (pos.position[1] >= 0x13) ||
+			((pos.position[1] == 0x12) && (!pos.position[6])) ) {
+			status = CD_STOPPED;
+		} else
+		if ( pos.position[1] == 0x11 ) {
+			status = CD_PLAYING;
+		} else {
+			status = CD_PAUSED;
+		}
+		close(fd);
+	} else {
+		status = CD_TRAYEMPTY;
+	}
+	if ( position ) {
+		*position = cur_frame;
+	}
+	return(status);
+}
+
+/* Start play */
+static int SDL_SYS_CDPlay(SDL_CD *cdrom, int start, int length)
+{
+	int okay;
+	int fd;
+	scsi_play_position pos;
+
+	okay = 0;
+	fd = open(SDL_cdlist[cdrom->id], 0);
+	if ( fd >= 0 ) {
+		FRAMES_TO_MSF(start, &pos.start_m, &pos.start_s, &pos.start_f);
+		FRAMES_TO_MSF(start+length, &pos.end_m, &pos.end_s, &pos.end_f);
+		if ( ioctl(fd, B_SCSI_PLAY_POSITION, &pos) == B_NO_ERROR ) {
+			okay = 1;
+		}
+		close(fd);
+	}
+	return(okay ? 0 : -1);
+}
+
+/* Pause play */
+static int SDL_SYS_CDPause(SDL_CD *cdrom)
+{
+	return(SDL_SYS_CDioctl(cdrom->id, B_SCSI_PAUSE_AUDIO, 0));
+}
+
+/* Resume play */
+static int SDL_SYS_CDResume(SDL_CD *cdrom)
+{
+	return(SDL_SYS_CDioctl(cdrom->id, B_SCSI_RESUME_AUDIO, 0));
+}
+
+/* Stop play */
+static int SDL_SYS_CDStop(SDL_CD *cdrom)
+{
+	return(SDL_SYS_CDioctl(cdrom->id, B_SCSI_STOP_AUDIO, 0));
+}
+
+/* Eject the CD-ROM */
+static int SDL_SYS_CDEject(SDL_CD *cdrom)
+{
+	return(SDL_SYS_CDioctl(cdrom->id, B_SCSI_EJECT, 0));
+}
+
+/* Close the CD-ROM handle */
+static void SDL_SYS_CDClose(SDL_CD *cdrom)
+{
+	close(cdrom->id);
+}
+
+void SDL_SYS_CDQuit(void)
+{
+	int i;
+
+	if ( SDL_numcds > 0 ) {
+		for ( i=0; i<SDL_numcds; ++i ) {
+			SDL_free(SDL_cdlist[i]);
+		}
+		SDL_numcds = 0;
+	}
+}
+
+#endif /* SDL_CDROM_BEOS */

Externals/SDL/src/cdrom/bsdi/SDL_syscdrom.c

@@ -0,0 +1,542 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifdef SDL_CDROM_BSDI
+
+/*
+ * Functions for system-level CD-ROM audio control for BSD/OS 4.x
+ * This started life out as a copy of the freebsd/SDL_cdrom.c file but was
+ * heavily modified.   Works for standard (MMC) SCSI and ATAPI CDrom drives.
+ *
+ * Steven Schultz - sms@to.gd-es.com
+*/
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <err.h>
+#include <unistd.h>
+#include <sys/ioctl.h>
+#include </sys/dev/scsi/scsi.h>
+#include </sys/dev/scsi/scsi_ioctl.h>
+
+#include "SDL_cdrom.h"
+#include "../SDL_syscdrom.h"
+
+/*
+ * The msf_to_frame and frame_to_msf were yanked from libcdrom and inlined
+ * here so that -lcdrom doesn't have to be dragged in for something so simple.
+*/
+
+#define	FRAMES_PER_SECOND	75
+#define	FRAMES_PER_MINUTE	(FRAMES_PER_SECOND * 60)
+
+int
+msf_to_frame(int minute, int second, int frame)
+	{
+	return(minute * FRAMES_PER_MINUTE + second * FRAMES_PER_SECOND + frame);
+	}
+
+void
+frame_to_msf(int frame, int *minp, int *secp, int *framep)
+	{
+	*minp = frame / FRAMES_PER_MINUTE;
+	*secp = (frame % FRAMES_PER_MINUTE) / FRAMES_PER_SECOND;
+	*framep = frame % FRAMES_PER_SECOND;
+	}
+
+/* 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);
+
+typedef	struct	scsi_cdb cdb_t;
+
+static int scsi_cmd(int fd,
+		struct scsi_cdb *cdb,
+		int cdblen, 
+		int rw,
+		caddr_t data,
+		int datalen,
+		struct scsi_user_cdb *sus)
+	{
+	int	scsistatus;
+	unsigned char	*cp;
+	struct	scsi_user_cdb suc;
+
+    /* safety checks */
+	if	(!cdb) return(-1);
+	if	(rw != SUC_READ && rw != SUC_WRITE) return(-1);
+
+	suc.suc_flags = rw;
+	suc.suc_cdblen = cdblen;
+	bcopy(cdb, suc.suc_cdb, cdblen);
+	suc.suc_datalen = datalen;
+	suc.suc_data = data;
+	suc.suc_timeout = 10;		/* 10 secs max for TUR or SENSE */
+	if	(ioctl(fd, SCSIRAWCDB, &suc) == -1)
+		return(-11);
+	scsistatus = suc.suc_sus.sus_status;
+	cp = suc.suc_sus.sus_sense;
+
+/*
+ * If a place to copy the sense data back to has been provided then the
+ * caller is responsible for checking the errors and printing any information
+ * out if the status was not successful.
+*/
+	if	(scsistatus != 0 && !sus)
+		{
+		fprintf(stderr,"scsistatus = %x cmd = %x\n",
+			scsistatus, cdb[0]);
+		fprintf(stderr, "sense %x %x %x %x %x %x %x %x %x %x %x %x %x %x %x %x\n", 
+			cp[0], cp[1], cp[2], cp[3], cp[4], cp[5],
+			cp[6], cp[7], cp[8], cp[9], cp[10], cp[11],
+			cp[12], cp[13], cp[14], cp[15]);
+		return(1);
+		}
+	if	(sus)
+		bcopy(&suc, sus, sizeof (struct scsi_user_cdb));
+	if	(scsistatus)
+		return(1);	/* Return non-zero for unsuccessful status */
+	return(0);
+	}
+
+/* request vendor brand and model */
+unsigned char *Inquiry(int fd)
+	{
+	static struct scsi_cdb6 cdb = 
+		{
+		0x12, 
+		0, 0, 0,
+		56,
+		0
+		};
+	static unsigned char Inqbuffer[56];
+
+	if	(scsi_cmd(fd, (cdb_t *)&cdb, 6, SUC_READ, Inqbuffer, 
+			sizeof(Inqbuffer), 0))
+		return("\377");
+	return(Inqbuffer);
+	}
+
+#define ADD_SENSECODE 12
+#define ADD_SC_QUALIFIER 13
+
+int TestForMedium(int fd)
+	{
+	int	sts, asc, ascq;
+	struct	scsi_user_cdb sus;
+	static struct scsi_cdb6 cdb =
+		{
+		CMD_TEST_UNIT_READY, /* command */
+		0,	/* reserved */
+		0,	/* reserved */
+		0,	/* reserved */
+		0,	/* reserved */
+		0	/* reserved */
+		};
+
+again:	sts = scsi_cmd(fd, (cdb_t *)&cdb, 6, SUC_READ, 0, 0, &sus);
+	asc = sus.suc_sus.sus_sense[ADD_SENSECODE];
+	ascq = sus.suc_sus.sus_sense[ADD_SC_QUALIFIER];
+	if	(asc == 0x3a && ascq == 0x0)	/* no medium */
+		return(0);
+	if	(asc == 0x28 && ascq == 0x0)	/* medium changed */
+		goto again;
+	if	(asc == 0x4 && ascq == 0x1 )	/* coming ready */
+		{
+		sleep(2);
+		goto again;
+		}
+	return(1);
+	}
+
+/* Check a drive to see if it is a CD-ROM */
+static int CheckDrive(char *drive, struct stat *stbuf)
+{
+	int is_cd = 0, cdfd;
+	char *p;
+
+	/* 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 */
+	cdfd = open(drive, (O_RDONLY|O_EXCL|O_NONBLOCK), 0);
+	if ( cdfd >= 0 ) {
+		p = Inquiry(cdfd);
+		if (*p == TYPE_ROM)
+			is_cd = 1;
+		close(cdfd);
+	}
+	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
+	}
+}
+
+int  SDL_SYS_CDInit(void)
+{
+	/* checklist: /dev/rsr?c */
+	static char *checklist[] = {
+	"?0 rsr?", NULL
+	};
+	char *SDLcdrom;
+	int i, j, exists;
+	char drive[32];
+	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';
+				}
+				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);
+		}
+	}
+
+	/* Scan the system for CD-ROM drives */
+	for ( i=0; checklist[i]; ++i ) {
+		if ( checklist[i][0] == '?' ) {
+			char *insert;
+			exists = 1;
+			for ( j=checklist[i][1]; exists; ++j ) {
+				SDL_snprintf(drive, SDL_arraysize(drive), "/dev/%sc", &checklist[i][3]);
+				insert = SDL_strchr(drive, '?');
+				if ( insert != NULL ) {
+					*insert = j;
+				}
+				switch (CheckDrive(drive, &stbuf)) {
+					/* Drive exists and is a CD-ROM */
+					case 1:
+						AddDrive(drive, &stbuf);
+						break;
+					/* Drive exists, but isn't a CD-ROM */
+					case 0:
+						break;
+					/* Drive doesn't exist */
+					case -1:
+						exists = 0;
+						break;
+				}
+			}
+		} else {
+			SDL_snprintf(drive, SDL_arraysize(drive), "/dev/%s", checklist[i]);
+			if ( CheckDrive(drive, &stbuf) > 0 ) {
+				AddDrive(drive, &stbuf);
+			}
+		}
+	}
+	return(0);
+}
+
+static const char *SDL_SYS_CDName(int drive)
+{
+	return(SDL_cdlist[drive]);
+}
+
+static int SDL_SYS_CDOpen(int drive)
+{
+	return(open(SDL_cdlist[drive], O_RDONLY | O_NONBLOCK | O_EXCL, 0));
+}
+
+static int SDL_SYS_CDGetTOC(SDL_CD *cdrom)
+	{
+	u_char cdb[10], buf[4], *p, *toc;
+	struct scsi_user_cdb sus;
+	int i, sts, first_track, last_track, ntracks, toc_size;
+
+	bzero(cdb, sizeof (cdb));
+	cdb[0] = 0x43;		/* Read TOC */
+	cdb[1] = 0x2;		/* MSF */
+	cdb[8] = 4;		/* size TOC header */
+	sts = scsi_cmd(cdrom->id, (cdb_t *)cdb, 10, SUC_READ, buf, 4, &sus);
+	if	(sts < 0)
+		return(-1);
+	first_track = buf[2];
+	last_track = buf[3];
+	ntracks = last_track - first_track + 1;
+	cdrom->numtracks = ntracks;
+	toc_size = 4 + (ntracks + 1) * 8;
+	toc = (u_char *)SDL_malloc(toc_size);
+	if	(toc == NULL)
+		return(-1);
+	bzero(cdb, sizeof (cdb));
+	cdb[0] = 0x43;
+	cdb[1] = 0x2;
+	cdb[6] = first_track;
+	cdb[7] = toc_size >> 8;
+	cdb[8] = toc_size & 0xff;
+	sts = scsi_cmd(cdrom->id, (cdb_t *)cdb, 10, SUC_READ, toc, toc_size, 
+			&sus);
+	if	(sts < 0)
+		{
+		SDL_free(toc);
+		return(-1);
+		}
+
+	for	(i = 0, p = toc+4; i <= ntracks; i++, p+= 8)
+		{
+		if	(i == ntracks)
+			cdrom->track[i].id = 0xAA;	/* Leadout */
+		else
+			cdrom->track[i].id = first_track + i;
+		if	(p[1] & 0x20)
+			cdrom->track[i].type = SDL_DATA_TRACK;
+		else
+			cdrom->track[i].type = SDL_AUDIO_TRACK;
+		cdrom->track[i].offset = msf_to_frame(p[5], p[6], p[7]);
+		cdrom->track[i].length = 0;
+		if	(i > 0)
+			cdrom->track[i-1].length = cdrom->track[i].offset -
+						   cdrom->track[i-1].offset;
+		}
+	SDL_free(toc);
+	return(0);
+	}
+
+/* Get CD-ROM status */
+static CDstatus SDL_SYS_CDStatus(SDL_CD *cdrom, int *position)
+	{
+	CDstatus status;
+	u_char	cdb[10], buf[16];
+	int	sts;
+	struct	scsi_user_cdb sus;
+
+	bzero(cdb, sizeof (cdb));
+	cdb[0] = 0x42;		/* read subq */
+	cdb[1] = 0x2;		/* MSF */
+	cdb[2] = 0x40;		/* q channel */
+	cdb[3] = 1;		/* current pos */
+	cdb[7] = sizeof (buf) >> 8;
+	cdb[8] = sizeof (buf) & 0xff;
+	sts = scsi_cmd(cdrom->id, (cdb_t *)cdb, 10, SUC_READ, buf, sizeof (buf),
+			&sus);
+	if	(sts < 0)
+		return(-1);
+	if	(sts)
+		{
+		if	(TestForMedium(cdrom->id) == 0)
+			status = CD_TRAYEMPTY;
+		else
+			status = CD_ERROR;
+		}
+	else
+		{
+		switch	(buf[1])
+			{
+			case	0x11:
+				status = CD_PLAYING;
+				break;
+			case	0x12:
+				status = CD_PAUSED;
+				break;
+			case	0x13:
+			case	0x14:
+			case	0x15:
+				status = CD_STOPPED;
+				break;
+			default:
+				status = CD_ERROR;
+				break;
+			}
+		}
+	if	(position)
+		{
+		if	( status == CD_PLAYING || (status == CD_PAUSED) )
+			*position = msf_to_frame(buf[9], buf[10], buf[11]);
+		else
+			*position = 0;
+		}
+	return(status);
+	}
+
+/* Start play */
+static int SDL_SYS_CDPlay(SDL_CD *cdrom, int start, int length)
+	{
+	u_char	cdb[10];
+	int	sts, minute, second, frame, eminute, esecond, eframe;
+	struct	scsi_user_cdb sus;
+
+	bzero(cdb, sizeof(cdb));
+	cdb[0] = 0x47;		/* Play */
+	frame_to_msf(start, &minute, &second, &frame);
+	frame_to_msf(start + length, &eminute, &esecond, &eframe);
+	cdb[3] = minute;
+	cdb[4] = second;
+	cdb[5] = frame;
+	cdb[6] = eminute;
+	cdb[7] = esecond;
+	cdb[8] = eframe;
+	sts = scsi_cmd(cdrom->id, (cdb_t *)cdb, 10, SUC_READ, 0, 0, &sus);
+	return(sts);
+	}
+
+static	int
+pauseresume(SDL_CD *cdrom, int flag)
+	{
+	u_char	cdb[10];
+	struct	scsi_user_cdb sus;
+
+	bzero(cdb, sizeof (cdb));
+	cdb[0] = 0x4b;
+	cdb[8] = flag & 0x1;
+	return(scsi_cmd(cdrom->id, (cdb_t *)cdb, 10, SUC_READ, 0, 0, &sus));
+	}
+
+/* Pause play */
+static int SDL_SYS_CDPause(SDL_CD *cdrom)
+{
+	return(pauseresume(cdrom, 0));
+}
+
+/* Resume play */
+static int SDL_SYS_CDResume(SDL_CD *cdrom)
+{
+	return(pauseresume(cdrom, 1));
+}
+
+/* Stop play */
+static int SDL_SYS_CDStop(SDL_CD *cdrom)
+{
+	u_char cdb[6];
+	struct	scsi_user_cdb sus;
+
+	bzero(cdb, sizeof (cdb));
+	cdb[0] = 0x1b;		/* stop */
+	cdb[1] = 1;		/* immediate */
+	return(scsi_cmd(cdrom->id, (cdb_t *)cdb, 6, SUC_READ, 0, 0, &sus));
+}
+
+/* Eject the CD-ROM */
+static int SDL_SYS_CDEject(SDL_CD *cdrom)
+{
+	u_char cdb[6];
+	struct	scsi_user_cdb sus;
+
+	bzero(cdb, sizeof (cdb));
+	cdb[0] = 0x1b;		/* stop */
+	cdb[1] = 1;		/* immediate */
+	cdb[4] = 2;		/* eject */
+	return(scsi_cmd(cdrom->id, (cdb_t *)cdb, 6, SUC_READ, 0, 0, &sus));
+}
+
+/* Close the CD-ROM handle */
+static void SDL_SYS_CDClose(SDL_CD *cdrom)
+	{
+	close(cdrom->id);
+	}
+
+void SDL_SYS_CDQuit(void)
+{
+	int i;
+
+	if ( SDL_numcds > 0 ) {
+		for ( i=0; i<SDL_numcds; ++i ) {
+			SDL_free(SDL_cdlist[i]);
+			}
+		}
+	SDL_numcds = 0;
+}
+
+#endif /* SDL_CDROM_BSDI */

Externals/SDL/src/cdrom/dc/SDL_syscdrom.c

@@ -0,0 +1,167 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifdef SDL_CDROM_DC
+
+/* Functions for system-level CD-ROM audio control */
+
+#include <dc/cdrom.h>
+#include <dc/spu.h>
+
+#include "SDL_cdrom.h"
+#include "../SDL_syscdrom.h"
+
+/* 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);
+
+
+int  SDL_SYS_CDInit(void)
+{
+	/* 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;
+
+	return(0);
+}
+
+static const char *SDL_SYS_CDName(int drive)
+{
+	return "/cd";
+}
+
+static int SDL_SYS_CDOpen(int drive)
+{
+	return(drive);
+}
+
+#define	TRACK_CDDA	0
+static int SDL_SYS_CDGetTOC(SDL_CD *cdrom)
+{
+	CDROM_TOC toc;
+	int ret,i;
+
+	ret = cdrom_read_toc(&toc,0);
+	if (ret!=ERR_OK) {
+		return -1;
+	}
+
+	cdrom->numtracks = TOC_TRACK(toc.last)-TOC_TRACK(toc.first)+1;
+	for(i=0;i<cdrom->numtracks;i++) {
+		unsigned long entry = toc.entry[i];
+		cdrom->track[i].id = i+1;
+		cdrom->track[i].type = (TOC_CTRL(toc.entry[i])==TRACK_CDDA)?SDL_AUDIO_TRACK:SDL_DATA_TRACK;
+		cdrom->track[i].offset = TOC_LBA(entry)-150;
+		cdrom->track[i].length = TOC_LBA((i+1<toc.last)?toc.entry[i+1]:toc.leadout_sector)-TOC_LBA(entry);
+	}
+
+	return 0;
+}
+
+/* Get CD-ROM status */
+static CDstatus SDL_SYS_CDStatus(SDL_CD *cdrom, int *position)
+{
+	int ret,dc_status,disc_type;
+
+	ret = cdrom_get_status(&dc_status,&disc_type);
+	if (ret!=ERR_OK) return CD_ERROR;
+
+	switch(dc_status) {
+//	case CD_STATUS_BUSY:
+	case CD_STATUS_PAUSED:
+		return CD_PAUSED;
+	case CD_STATUS_STANDBY:
+		return CD_STOPPED;
+	case CD_STATUS_PLAYING:
+		return CD_PLAYING;
+//	case CD_STATUS_SEEKING:
+//	case CD_STATUS_SCANING:
+	case CD_STATUS_OPEN:
+	case CD_STATUS_NO_DISC:
+		return CD_TRAYEMPTY;
+	default:
+		return	CD_ERROR;
+	}
+}
+
+/* Start play */
+static int SDL_SYS_CDPlay(SDL_CD *cdrom, int start, int length)
+{
+	int ret = cdrom_cdda_play(start-150,start-150+length,1,CDDA_SECTORS);
+	return ret==ERR_OK?0:-1;
+}
+
+/* Pause play */
+static int SDL_SYS_CDPause(SDL_CD *cdrom)
+{
+	int ret=cdrom_cdda_pause();
+	return ret==ERR_OK?0:-1;
+}
+
+/* Resume play */
+static int SDL_SYS_CDResume(SDL_CD *cdrom)
+{
+	int ret=cdrom_cdda_resume();
+	return ret==ERR_OK?0:-1;
+}
+
+/* Stop play */
+static int SDL_SYS_CDStop(SDL_CD *cdrom)
+{
+	int ret=cdrom_spin_down();
+	return ret==ERR_OK?0:-1;
+}
+
+/* Eject the CD-ROM */
+static int SDL_SYS_CDEject(SDL_CD *cdrom)
+{
+	return -1;
+}
+
+/* Close the CD-ROM handle */
+static void SDL_SYS_CDClose(SDL_CD *cdrom)
+{
+}
+
+void SDL_SYS_CDQuit(void)
+{
+
+}
+
+#endif /* SDL_CDROM_DC */

Externals/SDL/src/cdrom/dummy/SDL_syscdrom.c

@@ -0,0 +1,41 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#if defined(SDL_CDROM_DUMMY) || defined(SDL_CDROM_DISABLED)
+
+/* Stub functions for system-level CD-ROM audio control */
+
+#include "SDL_cdrom.h"
+#include "../SDL_syscdrom.h"
+
+int  SDL_SYS_CDInit(void)
+{
+	return(0);
+}
+
+void SDL_SYS_CDQuit(void)
+{
+	return;
+}
+
+#endif /* SDL_CDROM_DUMMY || SDL_CDROM_DISABLED */

Externals/SDL/src/cdrom/freebsd/SDL_syscdrom.c

@@ -0,0 +1,406 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifdef SDL_CDROM_FREEBSD
+
+/* Functions for system-level CD-ROM audio control */
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <unistd.h>
+#include <sys/cdio.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);
+
+/* Some ioctl() errno values which occur when the tray is empty */
+#define ERRNO_TRAYEMPTY(errno)	\
+	((errno == EIO) || (errno == ENOENT) || (errno == EINVAL))
+
+/* Check a drive to see if it is a CD-ROM */
+static int CheckDrive(char *drive, struct stat *stbuf)
+{
+	int is_cd, cdfd;
+	struct ioc_read_subchannel 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 ) {
+			info.address_format = CD_MSF_FORMAT;
+			info.data_format = CD_CURRENT_POSITION;
+			info.data_len = 0;
+			info.data = NULL;
+			/* Under Linux, EIO occurs when a disk is not present.
+			   This isn't 100% reliable, so we use the USE_MNTENT
+			   code above instead.
+			 */
+			if ( (ioctl(cdfd, CDIOCREADSUBCHANNEL, &info) == 0) ||
+						ERRNO_TRAYEMPTY(errno) ) {
+				is_cd = 1;
+			}
+			close(cdfd);
+		}
+	}
+	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
+	}
+}
+
+int  SDL_SYS_CDInit(void)
+{
+	/* checklist: /dev/cdrom,/dev/cd?c /dev/acd?c
+			/dev/matcd?c /dev/mcd?c /dev/scd?c */
+	static char *checklist[] = {
+	"cdrom", "?0 cd?", "?0 acd?", "?0 matcd?", "?0 mcd?", "?0 scd?",NULL
+	};
+	char *SDLcdrom;
+	int i, j, exists;
+	char drive[32];
+	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';
+				}
+				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);
+		}
+	}
+
+	/* Scan the system for CD-ROM drives */
+	for ( i=0; checklist[i]; ++i ) {
+		if ( checklist[i][0] == '?' ) {
+			char *insert;
+			exists = 1;
+			for ( j=checklist[i][1]; exists; ++j ) {
+				SDL_snprintf(drive, SDL_arraysize(drive), "/dev/%sc", &checklist[i][3]);
+				insert = SDL_strchr(drive, '?');
+				if ( insert != NULL ) {
+					*insert = j;
+				}
+				switch (CheckDrive(drive, &stbuf)) {
+					/* Drive exists and is a CD-ROM */
+					case 1:
+						AddDrive(drive, &stbuf);
+						break;
+					/* Drive exists, but isn't a CD-ROM */
+					case 0:
+						break;
+					/* Drive doesn't exist */
+					case -1:
+						exists = 0;
+						break;
+				}
+			}
+		} else {
+			SDL_snprintf(drive, SDL_arraysize(drive), "/dev/%s", checklist[i]);
+			if ( CheckDrive(drive, &stbuf) > 0 ) {
+				AddDrive(drive, &stbuf);
+			}
+		}
+	}
+	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);
+}
+
+static const char *SDL_SYS_CDName(int drive)
+{
+	return(SDL_cdlist[drive]);
+}
+
+static int SDL_SYS_CDOpen(int drive)
+{
+	return(open(SDL_cdlist[drive], (O_RDONLY|O_EXCL|O_NONBLOCK), 0));
+}
+
+static int SDL_SYS_CDGetTOC(SDL_CD *cdrom)
+{
+	struct ioc_toc_header toc;
+	int i, okay;
+	struct ioc_read_toc_entry entry;
+	struct cd_toc_entry data;
+
+	okay = 0;
+	if ( SDL_SYS_CDioctl(cdrom->id, CDIOREADTOCHEADER, &toc) == 0 ) {
+		cdrom->numtracks = toc.ending_track-toc.starting_track+1;
+		if ( cdrom->numtracks > SDL_MAX_TRACKS ) {
+			cdrom->numtracks = SDL_MAX_TRACKS;
+		}
+		/* Read all the track TOC entries */
+		for ( i=0; i<=cdrom->numtracks; ++i ) {
+			if ( i == cdrom->numtracks ) {
+				cdrom->track[i].id = 0xAA; /* CDROM_LEADOUT */
+			} else {
+				cdrom->track[i].id = toc.starting_track+i;
+			}
+			entry.starting_track = cdrom->track[i].id;
+			entry.address_format = CD_MSF_FORMAT;
+			entry.data_len = sizeof(data);
+			entry.data = &data;
+			if ( SDL_SYS_CDioctl(cdrom->id, CDIOREADTOCENTRYS,
+								&entry) < 0 ) {
+				break;
+			} else {
+				cdrom->track[i].type = data.control;
+				cdrom->track[i].offset = MSF_TO_FRAMES(
+						data.addr.msf.minute,
+						data.addr.msf.second,
+						data.addr.msf.frame);
+				cdrom->track[i].length = 0;
+				if ( i > 0 ) {
+					cdrom->track[i-1].length =
+						cdrom->track[i].offset-
+						cdrom->track[i-1].offset;
+				}
+			}
+		}
+		if ( i == (cdrom->numtracks+1) ) {
+			okay = 1;
+		}
+	}
+	return(okay ? 0 : -1);
+}
+
+/* Get CD-ROM status */
+static CDstatus SDL_SYS_CDStatus(SDL_CD *cdrom, int *position)
+{
+	CDstatus status;
+	struct ioc_toc_header toc;
+	struct ioc_read_subchannel info;
+	struct cd_sub_channel_info data;
+
+	info.address_format = CD_MSF_FORMAT;
+	info.data_format = CD_CURRENT_POSITION;
+	info.track = 0;
+	info.data_len = sizeof(data);
+	info.data = &data;
+	if ( ioctl(cdrom->id, CDIOCREADSUBCHANNEL, &info) < 0 ) {
+		if ( ERRNO_TRAYEMPTY(errno) ) {
+			status = CD_TRAYEMPTY;
+		} else {
+			status = CD_ERROR;
+		}
+	} else {
+		switch (data.header.audio_status) {
+			case CD_AS_AUDIO_INVALID:
+			case CD_AS_NO_STATUS:
+				/* Try to determine if there's a CD available */
+				if (ioctl(cdrom->id,CDIOREADTOCHEADER,&toc)==0)
+					status = CD_STOPPED;
+				else
+					status = CD_TRAYEMPTY;
+				break;
+			case CD_AS_PLAY_COMPLETED:
+				status = CD_STOPPED;
+				break;
+			case CD_AS_PLAY_IN_PROGRESS:
+				status = CD_PLAYING;
+				break;
+			case CD_AS_PLAY_PAUSED:
+				status = CD_PAUSED;
+				break;
+			default:
+				status = CD_ERROR;
+				break;
+		}
+	}
+	if ( position ) {
+		if ( status == CD_PLAYING || (status == CD_PAUSED) ) {
+			*position = MSF_TO_FRAMES(
+					data.what.position.absaddr.msf.minute,
+					data.what.position.absaddr.msf.second,
+					data.what.position.absaddr.msf.frame);
+		} else {
+			*position = 0;
+		}
+	}
+	return(status);
+}
+
+/* Start play */
+static int SDL_SYS_CDPlay(SDL_CD *cdrom, int start, int length)
+{
+	struct ioc_play_msf playtime;
+
+	FRAMES_TO_MSF(start,
+		&playtime.start_m, &playtime.start_s, &playtime.start_f);
+	FRAMES_TO_MSF(start+length,
+		&playtime.end_m, &playtime.end_s, &playtime.end_f);
+#ifdef DEBUG_CDROM
+  fprintf(stderr, "Trying to play from %d:%d:%d to %d:%d:%d\n",
+	playtime.cdmsf_min0, playtime.cdmsf_sec0, playtime.cdmsf_frame0,
+	playtime.cdmsf_min1, playtime.cdmsf_sec1, playtime.cdmsf_frame1);
+#endif
+	ioctl(cdrom->id, CDIOCSTART, 0);
+	return(SDL_SYS_CDioctl(cdrom->id, CDIOCPLAYMSF, &playtime));
+}
+
+/* Pause play */
+static int SDL_SYS_CDPause(SDL_CD *cdrom)
+{
+	return(SDL_SYS_CDioctl(cdrom->id, CDIOCPAUSE, 0));
+}
+
+/* Resume play */
+static int SDL_SYS_CDResume(SDL_CD *cdrom)
+{
+	return(SDL_SYS_CDioctl(cdrom->id, CDIOCRESUME, 0));
+}
+
+/* Stop play */
+static int SDL_SYS_CDStop(SDL_CD *cdrom)
+{
+	return(SDL_SYS_CDioctl(cdrom->id, CDIOCSTOP, 0));
+}
+
+/* Eject the CD-ROM */
+static int SDL_SYS_CDEject(SDL_CD *cdrom)
+{
+	return(SDL_SYS_CDioctl(cdrom->id, CDIOCEJECT, 0));
+}
+
+/* Close the CD-ROM handle */
+static void SDL_SYS_CDClose(SDL_CD *cdrom)
+{
+	close(cdrom->id);
+}
+
+void SDL_SYS_CDQuit(void)
+{
+	int i;
+
+	if ( SDL_numcds > 0 ) {
+		for ( i=0; i<SDL_numcds; ++i ) {
+			SDL_free(SDL_cdlist[i]);
+		}
+		SDL_numcds = 0;
+	}
+}
+
+#endif /* SDL_CDROM_FREEBSD */

Externals/SDL/src/cdrom/linux/SDL_syscdrom.c

@@ -0,0 +1,564 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifdef SDL_CDROM_LINUX
+
+/* Functions for system-level CD-ROM audio control */
+
+#include <string.h>	/* For strerror() */
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <unistd.h>
+#ifdef __LINUX__
+#ifdef HAVE_LINUX_VERSION_H
+/* linux 2.6.9 workaround */
+#include <linux/version.h>
+#if LINUX_VERSION_CODE == KERNEL_VERSION(2,6,9)
+#include <asm/types.h>
+#define __le64 __u64
+#define __le32 __u32
+#define __le16 __u16
+#define __be64 __u64
+#define __be32 __u32
+#define __be16 __u16
+#endif /* linux 2.6.9 workaround */
+#endif /* HAVE_LINUX_VERSION_H */
+#include <linux/cdrom.h>
+#endif
+#ifdef __SVR4
+#include <sys/cdio.h>
+#endif
+
+/* Define this to use the alternative getmntent() code */
+#ifndef __SVR4
+#define USE_MNTENT
+#endif
+
+#ifdef USE_MNTENT
+#if defined(__USLC__)
+#include <sys/mntent.h>
+#else
+#include <mntent.h>
+#endif
+
+#ifndef _PATH_MNTTAB
+#ifdef MNTTAB
+#define _PATH_MNTTAB	MNTTAB
+#else
+#define _PATH_MNTTAB	"/etc/fstab"
+#endif
+#endif /* !_PATH_MNTTAB */
+
+#ifndef _PATH_MOUNTED
+#define _PATH_MOUNTED	"/etc/mtab"
+#endif /* !_PATH_MOUNTED */
+
+#ifndef MNTTYPE_CDROM
+#define MNTTYPE_CDROM	"iso9660"
+#endif
+#ifndef MNTTYPE_SUPER
+#define MNTTYPE_SUPER	"supermount"
+#endif
+#endif /* USE_MNTENT */
+
+#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);
+
+/* Some ioctl() errno values which occur when the tray is empty */
+#ifndef ENOMEDIUM
+#define ENOMEDIUM ENOENT
+#endif
+#define ERRNO_TRAYEMPTY(errno)	\
+	((errno == EIO)    || (errno == ENOENT) || \
+	 (errno == EINVAL) || (errno == ENOMEDIUM))
+
+/* Check a drive to see if it is a CD-ROM */
+static int CheckDrive(char *drive, char *mnttype, struct stat *stbuf)
+{
+	int is_cd, cdfd;
+	struct cdrom_subchnl 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_NONBLOCK), 0);
+		if ( cdfd >= 0 ) {
+			info.cdsc_format = CDROM_MSF;
+			/* Under Linux, EIO occurs when a disk is not present.
+			 */
+			if ( (ioctl(cdfd, CDROMSUBCHNL, &info) == 0) ||
+						ERRNO_TRAYEMPTY(errno) ) {
+				is_cd = 1;
+			}
+			close(cdfd);
+		}
+#ifdef USE_MNTENT
+		/* Even if we can't read it, it might be mounted */
+		else if ( mnttype && (SDL_strcmp(mnttype, MNTTYPE_CDROM) == 0) ) {
+			is_cd = 1;
+		}
+#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
+	}
+}
+
+#ifdef USE_MNTENT
+static void CheckMounts(const char *mtab)
+{
+	FILE *mntfp;
+	struct mntent *mntent;
+	struct stat stbuf;
+
+	mntfp = setmntent(mtab, "r");
+	if ( mntfp != NULL ) {
+		char *tmp;
+		char *mnt_type;
+		size_t mnt_type_len;
+		char *mnt_dev;
+		size_t mnt_dev_len;
+
+		while ( (mntent=getmntent(mntfp)) != NULL ) {
+			mnt_type_len = SDL_strlen(mntent->mnt_type) + 1;
+			mnt_type = SDL_stack_alloc(char, mnt_type_len);
+			if (mnt_type == NULL)
+				continue;  /* maybe you'll get lucky next time. */
+
+			mnt_dev_len = SDL_strlen(mntent->mnt_fsname) + 1;
+			mnt_dev = SDL_stack_alloc(char, mnt_dev_len);
+			if (mnt_dev == NULL) {
+				SDL_stack_free(mnt_type);
+				continue;
+			}
+
+			SDL_strlcpy(mnt_type, mntent->mnt_type, mnt_type_len);
+			SDL_strlcpy(mnt_dev, mntent->mnt_fsname, mnt_dev_len);
+
+			/* Handle "supermount" filesystem mounts */
+			if ( SDL_strcmp(mnt_type, MNTTYPE_SUPER) == 0 ) {
+				tmp = SDL_strstr(mntent->mnt_opts, "fs=");
+				if ( tmp ) {
+					SDL_stack_free(mnt_type);
+					mnt_type = SDL_strdup(tmp + SDL_strlen("fs="));
+					if ( mnt_type ) {
+						tmp = SDL_strchr(mnt_type, ',');
+						if ( tmp ) {
+							*tmp = '\0';
+						}
+					}
+				}
+				tmp = SDL_strstr(mntent->mnt_opts, "dev=");
+				if ( tmp ) {
+					SDL_stack_free(mnt_dev);
+					mnt_dev = SDL_strdup(tmp + SDL_strlen("dev="));
+					if ( mnt_dev ) {
+						tmp = SDL_strchr(mnt_dev, ',');
+						if ( tmp ) {
+							*tmp = '\0';
+						}
+					}
+				}
+			}
+			if ( SDL_strcmp(mnt_type, MNTTYPE_CDROM) == 0 ) {
+#ifdef DEBUG_CDROM
+  fprintf(stderr, "Checking mount path from %s: %s mounted on %s of %s\n",
+	mtab, mnt_dev, mntent->mnt_dir, mnt_type);
+#endif
+				if (CheckDrive(mnt_dev, mnt_type, &stbuf) > 0) {
+					AddDrive(mnt_dev, &stbuf);
+				}
+			}
+			SDL_stack_free(mnt_dev);
+			SDL_stack_free(mnt_type);
+		}
+		endmntent(mntfp);
+	}
+}
+#endif /* USE_MNTENT */
+
+int  SDL_SYS_CDInit(void)
+{
+	/* checklist: /dev/cdrom, /dev/hd?, /dev/scd? /dev/sr? */
+	static char *checklist[] = {
+		"cdrom", "?a hd?", "?0 scd?", "?0 sr?", NULL
+	};
+	char *SDLcdrom;
+	int i, j, exists;
+	char drive[32];
+	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, NULL, &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);
+		}
+	}
+
+#ifdef USE_MNTENT
+	/* Check /dev/cdrom first :-) */
+	if (CheckDrive("/dev/cdrom", NULL, &stbuf) > 0) {
+		AddDrive("/dev/cdrom", &stbuf);
+	}
+
+	/* Now check the currently mounted CD drives */
+	CheckMounts(_PATH_MOUNTED);
+
+	/* Finally check possible mountable drives in /etc/fstab */
+	CheckMounts(_PATH_MNTTAB);
+
+	/* If we found our drives, there's nothing left to do */
+	if ( SDL_numcds > 0 ) {
+		return(0);
+	}
+#endif /* USE_MNTENT */
+
+	/* Scan the system for CD-ROM drives.
+	   Not always 100% reliable, so use the USE_MNTENT code above first.
+	 */
+	for ( i=0; checklist[i]; ++i ) {
+		if ( checklist[i][0] == '?' ) {
+			char *insert;
+			exists = 1;
+			for ( j=checklist[i][1]; exists; ++j ) {
+				SDL_snprintf(drive, SDL_arraysize(drive), "/dev/%s", &checklist[i][3]);
+				insert = SDL_strchr(drive, '?');
+				if ( insert != NULL ) {
+					*insert = j;
+				}
+#ifdef DEBUG_CDROM
+  fprintf(stderr, "Checking possible CD-ROM drive: %s\n", drive);
+#endif
+				switch (CheckDrive(drive, NULL, &stbuf)) {
+					/* Drive exists and is a CD-ROM */
+					case 1:
+						AddDrive(drive, &stbuf);
+						break;
+					/* Drive exists, but isn't a CD-ROM */
+					case 0:
+						break;
+					/* Drive doesn't exist */
+					case -1:
+						exists = 0;
+						break;
+				}
+			}
+		} else {
+			SDL_snprintf(drive, SDL_arraysize(drive), "/dev/%s", checklist[i]);
+#ifdef DEBUG_CDROM
+  fprintf(stderr, "Checking possible CD-ROM drive: %s\n", drive);
+#endif
+			if ( CheckDrive(drive, NULL, &stbuf) > 0 ) {
+				AddDrive(drive, &stbuf);
+			}
+		}
+	}
+	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);
+}
+
+static const char *SDL_SYS_CDName(int drive)
+{
+	return(SDL_cdlist[drive]);
+}
+
+static int SDL_SYS_CDOpen(int drive)
+{
+	return(open(SDL_cdlist[drive], (O_RDONLY|O_NONBLOCK), 0));
+}
+
+static int SDL_SYS_CDGetTOC(SDL_CD *cdrom)
+{
+	struct cdrom_tochdr toc;
+	int i, okay;
+	struct cdrom_tocentry entry;
+
+	okay = 0;
+	if ( SDL_SYS_CDioctl(cdrom->id, CDROMREADTOCHDR, &toc) == 0 ) {
+		cdrom->numtracks = toc.cdth_trk1-toc.cdth_trk0+1;
+		if ( cdrom->numtracks > SDL_MAX_TRACKS ) {
+			cdrom->numtracks = SDL_MAX_TRACKS;
+		}
+		/* Read all the track TOC entries */
+		for ( i=0; i<=cdrom->numtracks; ++i ) {
+			if ( i == cdrom->numtracks ) {
+				cdrom->track[i].id = CDROM_LEADOUT;
+			} else {
+				cdrom->track[i].id = toc.cdth_trk0+i;
+			}
+			entry.cdte_track = cdrom->track[i].id;
+			entry.cdte_format = CDROM_MSF;
+			if ( SDL_SYS_CDioctl(cdrom->id, CDROMREADTOCENTRY,
+								&entry) < 0 ) {
+				break;
+			} else {
+				if ( entry.cdte_ctrl & CDROM_DATA_TRACK ) {
+					cdrom->track[i].type = SDL_DATA_TRACK;
+				} else {
+					cdrom->track[i].type = SDL_AUDIO_TRACK;
+				}
+				cdrom->track[i].offset = MSF_TO_FRAMES(
+						entry.cdte_addr.msf.minute,
+						entry.cdte_addr.msf.second,
+						entry.cdte_addr.msf.frame);
+				cdrom->track[i].length = 0;
+				if ( i > 0 ) {
+					cdrom->track[i-1].length =
+						cdrom->track[i].offset-
+						cdrom->track[i-1].offset;
+				}
+			}
+		}
+		if ( i == (cdrom->numtracks+1) ) {
+			okay = 1;
+		}
+	}
+	return(okay ? 0 : -1);
+}
+
+/* Get CD-ROM status */
+static CDstatus SDL_SYS_CDStatus(SDL_CD *cdrom, int *position)
+{
+	CDstatus status;
+	struct cdrom_tochdr toc;
+	struct cdrom_subchnl info;
+
+	info.cdsc_format = CDROM_MSF;
+	if ( ioctl(cdrom->id, CDROMSUBCHNL, &info) < 0 ) {
+		if ( ERRNO_TRAYEMPTY(errno) ) {
+			status = CD_TRAYEMPTY;
+		} else {
+			status = CD_ERROR;
+		}
+	} else {
+		switch (info.cdsc_audiostatus) {
+			case CDROM_AUDIO_INVALID:
+			case CDROM_AUDIO_NO_STATUS:
+				/* Try to determine if there's a CD available */
+				if (ioctl(cdrom->id, CDROMREADTOCHDR, &toc)==0)
+					status = CD_STOPPED;
+				else
+					status = CD_TRAYEMPTY;
+				break;
+			case CDROM_AUDIO_COMPLETED:
+				status = CD_STOPPED;
+				break;
+			case CDROM_AUDIO_PLAY:
+				status = CD_PLAYING;
+				break;
+			case CDROM_AUDIO_PAUSED:
+				/* Workaround buggy CD-ROM drive */
+				if ( info.cdsc_trk == CDROM_LEADOUT ) {
+					status = CD_STOPPED;
+				} else {
+					status = CD_PAUSED;
+				}
+				break;
+			default:
+				status = CD_ERROR;
+				break;
+		}
+	}
+	if ( position ) {
+		if ( status == CD_PLAYING || (status == CD_PAUSED) ) {
+			*position = MSF_TO_FRAMES(
+					info.cdsc_absaddr.msf.minute,
+					info.cdsc_absaddr.msf.second,
+					info.cdsc_absaddr.msf.frame);
+		} else {
+			*position = 0;
+		}
+	}
+	return(status);
+}
+
+/* Start play */
+static int SDL_SYS_CDPlay(SDL_CD *cdrom, int start, int length)
+{
+	struct cdrom_msf playtime;
+
+	FRAMES_TO_MSF(start,
+	   &playtime.cdmsf_min0, &playtime.cdmsf_sec0, &playtime.cdmsf_frame0);
+	FRAMES_TO_MSF(start+length,
+	   &playtime.cdmsf_min1, &playtime.cdmsf_sec1, &playtime.cdmsf_frame1);
+#ifdef DEBUG_CDROM
+  fprintf(stderr, "Trying to play from %d:%d:%d to %d:%d:%d\n",
+	playtime.cdmsf_min0, playtime.cdmsf_sec0, playtime.cdmsf_frame0,
+	playtime.cdmsf_min1, playtime.cdmsf_sec1, playtime.cdmsf_frame1);
+#endif
+	return(SDL_SYS_CDioctl(cdrom->id, CDROMPLAYMSF, &playtime));
+}
+
+/* Pause play */
+static int SDL_SYS_CDPause(SDL_CD *cdrom)
+{
+	return(SDL_SYS_CDioctl(cdrom->id, CDROMPAUSE, 0));
+}
+
+/* Resume play */
+static int SDL_SYS_CDResume(SDL_CD *cdrom)
+{
+	return(SDL_SYS_CDioctl(cdrom->id, CDROMRESUME, 0));
+}
+
+/* Stop play */
+static int SDL_SYS_CDStop(SDL_CD *cdrom)
+{
+	return(SDL_SYS_CDioctl(cdrom->id, CDROMSTOP, 0));
+}
+
+/* Eject the CD-ROM */
+static int SDL_SYS_CDEject(SDL_CD *cdrom)
+{
+	return(SDL_SYS_CDioctl(cdrom->id, CDROMEJECT, 0));
+}
+
+/* Close the CD-ROM handle */
+static void SDL_SYS_CDClose(SDL_CD *cdrom)
+{
+	close(cdrom->id);
+}
+
+void SDL_SYS_CDQuit(void)
+{
+	int i;
+
+	if ( SDL_numcds > 0 ) {
+		for ( i=0; i<SDL_numcds; ++i ) {
+			SDL_free(SDL_cdlist[i]);
+		}
+		SDL_numcds = 0;
+	}
+}
+
+#endif /* SDL_CDROM_LINUX */

Externals/SDL/src/cdrom/macos/SDL_syscdrom.c

@@ -0,0 +1,525 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997-2009 Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Lesser General Public
+    License as published by the Free Software Foundation; either
+    version 2.1 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Lesser General Public License for more details.
+
+    You should have received a copy of the GNU Lesser General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+#include "SDL_config.h"
+
+#ifdef SDL_CDROM_MACOS
+
+/* MacOS functions for system-level CD-ROM audio control */
+
+#include <Devices.h>
+#include <Files.h>
+#include <LowMem.h> /* Use entry table macros, not functions in InterfaceLib  */
+
+#include "SDL_cdrom.h"
+#include "../SDL_syscdrom.h"
+#include "SDL_syscdrom_c.h"
+
+/* Added by Matt Slot */
+#if !defined(LMGetUnitTableEntryCount)
+  #define LMGetUnitTableEntryCount()   *(short *)0x01D2
+#endif
+
+/* The maximum number of CD-ROM drives we'll detect */
+#define MAX_DRIVES	26	
+
+/* A list of available CD-ROM drives */
+static long SDL_cdversion = 0;
+static struct {
+	short		dRefNum;
+	short		driveNum;
+	long		frames;
+	char		name[256];
+	Boolean		hasAudio;
+	} SDL_cdlist[MAX_DRIVES];
+static StringPtr gDriverName = "\p.AppleCD";
+
+/* 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 short SDL_SYS_ShortToBCD(short value)
+{
+	return((value % 10) + (value / 10) * 0x10); /* Convert value to BCD */
+}
+
+static short SDL_SYS_BCDToShort(short value)
+{
+	return((value % 0x10) + (value / 0x10) * 10); /* Convert value from BCD */
+}
+
+int  SDL_SYS_CDInit(void)
+{
+	SInt16			dRefNum = 0;
+	SInt16			first, last;
+
+	SDL_numcds = 0;
+
+	/* Check that the software is available */
+	if (Gestalt(kGestaltAudioCDSelector, &SDL_cdversion) || 
+			!SDL_cdversion) return(0);
+
+	/* 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;
+
+	/* Walk the list, count each AudioCD driver, and save the refnums */
+	first = -1;
+	last = 0 - LMGetUnitTableEntryCount();
+	for(dRefNum = first; dRefNum >= last; dRefNum--) {
+		Str255		driverName;
+		StringPtr	namePtr;
+		DCtlHandle	deviceEntry;
+
+		deviceEntry = GetDCtlEntry(dRefNum);
+		if (! deviceEntry) continue;
+		
+		/* Is this an .AppleCD ? */
+		namePtr = (*deviceEntry)->dCtlFlags & (1L << dRAMBased) ?
+				((StringPtr) ((DCtlPtr) deviceEntry)->dCtlDriver + 18) :
+				((StringPtr) (*deviceEntry)->dCtlDriver + 18);
+		BlockMoveData(namePtr, driverName, namePtr[0]+1);
+		if (driverName[0] > gDriverName[0]) driverName[0] = gDriverName[0];
+		if (! EqualString(driverName, gDriverName, false, false)) continue;
+
+		/* Record the basic info for each drive */
+		SDL_cdlist[SDL_numcds].dRefNum = dRefNum;
+		BlockMoveData(namePtr + 1, SDL_cdlist[SDL_numcds].name, namePtr[0]);
+		SDL_cdlist[SDL_numcds].name[namePtr[0]] = 0;
+		SDL_cdlist[SDL_numcds].hasAudio = false;
+		SDL_numcds++;
+	}
+	return(0);
+}
+
+static const char *SDL_SYS_CDName(int drive)
+{
+	return(SDL_cdlist[drive].name);
+}
+
+static int get_drivenum(int drive)
+{
+	QHdr *driveQ = GetDrvQHdr();
+	DrvQEl *driveElem;
+
+	/* Update the drive number */
+	SDL_cdlist[drive].driveNum = 0;
+	if ( driveQ->qTail ) {
+		driveQ->qTail->qLink = 0;
+	}
+	for ( driveElem=(DrvQEl *)driveQ->qHead; driveElem;
+	      driveElem = (DrvQEl *)driveElem->qLink ) {
+		if ( driveElem->dQRefNum == SDL_cdlist[drive].dRefNum ) {
+			SDL_cdlist[drive].driveNum = driveElem->dQDrive;
+			break;
+		}
+	}
+	return(SDL_cdlist[drive].driveNum);
+}
+
+static int SDL_SYS_CDOpen(int drive)
+{
+	return(drive);
+}
+
+static int SDL_SYS_CDGetTOC(SDL_CD *cdrom)
+{
+	CDCntrlParam		cdpb;
+	CDTrackData			tracks[SDL_MAX_TRACKS];
+	long				i, leadout;
+
+	/* Get the number of tracks on the CD by examining the TOC */
+	SDL_memset(&cdpb, 0, sizeof(cdpb));
+	cdpb.ioVRefNum = SDL_cdlist[cdrom->id].driveNum;
+	cdpb.ioCRefNum = SDL_cdlist[cdrom->id].dRefNum;
+	cdpb.csCode = kReadTOC;
+	cdpb.csParam.words[0] = kGetTrackRange;
+	if ( PBControlSync((ParmBlkPtr)&cdpb) != noErr ) {
+		SDL_SetError("PBControlSync() failed");
+		return(-1);
+	}
+
+	cdrom->numtracks = 
+			SDL_SYS_BCDToShort(cdpb.csParam.bytes[1]) - 
+			SDL_SYS_BCDToShort(cdpb.csParam.bytes[0]) + 1;
+	if ( cdrom->numtracks > SDL_MAX_TRACKS )
+		cdrom->numtracks = SDL_MAX_TRACKS;
+	cdrom->status = CD_STOPPED;
+	cdrom->cur_track = 0; /* Apparently these are set elsewhere */
+	cdrom->cur_frame = 0; /* Apparently these are set elsewhere */
+
+
+	/* Get the lead out area of the CD by examining the TOC */
+	SDL_memset(&cdpb, 0, sizeof(cdpb));
+	cdpb.ioVRefNum = SDL_cdlist[cdrom->id].driveNum;
+	cdpb.ioCRefNum = SDL_cdlist[cdrom->id].dRefNum;
+	cdpb.csCode = kReadTOC;
+	cdpb.csParam.words[0] = kGetLeadOutArea;
+	if ( PBControlSync((ParmBlkPtr)&cdpb) != noErr ) {
+		SDL_SetError("PBControlSync() failed");
+		return(-1);
+	}
+
+	leadout = MSF_TO_FRAMES(
+			SDL_SYS_BCDToShort(cdpb.csParam.bytes[0]),
+			SDL_SYS_BCDToShort(cdpb.csParam.bytes[1]),
+			SDL_SYS_BCDToShort(cdpb.csParam.bytes[2]));
+
+	/* Get an array of track locations by examining the TOC */
+	SDL_memset(tracks, 0, sizeof(tracks));
+	SDL_memset(&cdpb, 0, sizeof(cdpb));
+	cdpb.ioVRefNum = SDL_cdlist[cdrom->id].driveNum;
+	cdpb.ioCRefNum = SDL_cdlist[cdrom->id].dRefNum;
+	cdpb.csCode = kReadTOC;
+	cdpb.csParam.words[0] = kGetTrackEntries;	/* Type of Query */
+	* ((long *) (cdpb.csParam.words+1)) = (long) tracks;				
+	cdpb.csParam.words[3] = cdrom->numtracks * sizeof(tracks[0]);		
+	* ((char *) (cdpb.csParam.words+4)) = 1;	/* First track */
+	if ( PBControlSync((ParmBlkPtr)&cdpb) != noErr ) {
+		SDL_SetError("PBControlSync() failed");
+		return(-1);
+	}
+
+	/* Read all the track TOC entries */
+	SDL_cdlist[cdrom->id].hasAudio = false;
+	for ( i=0; i<cdrom->numtracks; ++i ) 
+		{
+		cdrom->track[i].id = i+1;
+		if (tracks[i].entry.control & kDataTrackMask)
+			cdrom->track[i].type = SDL_DATA_TRACK;
+		else
+			{
+			cdrom->track[i].type = SDL_AUDIO_TRACK;
+			SDL_cdlist[SDL_numcds].hasAudio = true;
+			}
+		
+		cdrom->track[i].offset = MSF_TO_FRAMES(
+				SDL_SYS_BCDToShort(tracks[i].entry.min),
+				SDL_SYS_BCDToShort(tracks[i].entry.min),
+				SDL_SYS_BCDToShort(tracks[i].entry.frame));
+		cdrom->track[i].length = MSF_TO_FRAMES(
+				SDL_SYS_BCDToShort(tracks[i+1].entry.min),
+				SDL_SYS_BCDToShort(tracks[i+1].entry.min),
+				SDL_SYS_BCDToShort(tracks[i+1].entry.frame)) -
+				cdrom->track[i].offset;
+		}
+	
+	/* Apparently SDL wants a fake last entry */
+	cdrom->track[i].offset = leadout;
+	cdrom->track[i].length = 0;
+
+	return(0);
+}
+
+/* Get CD-ROM status */
+static CDstatus SDL_SYS_CDStatus(SDL_CD *cdrom, int *position)
+{
+	CDCntrlParam cdpb;
+	CDstatus status = CD_ERROR;
+	Boolean spinning = false;
+
+	if (position) *position = 0;
+
+	/* Get the number of tracks on the CD by examining the TOC */
+	if ( ! get_drivenum(cdrom->id) ) {
+		return(CD_TRAYEMPTY);
+	}
+	SDL_memset(&cdpb, 0, sizeof(cdpb));
+	cdpb.ioVRefNum = SDL_cdlist[cdrom->id].driveNum;
+	cdpb.ioCRefNum = SDL_cdlist[cdrom->id].dRefNum;
+	cdpb.csCode = kReadTOC;
+	cdpb.csParam.words[0] = kGetTrackRange;
+	if ( PBControlSync((ParmBlkPtr)&cdpb) != noErr ) {
+		SDL_SetError("PBControlSync() failed");
+		return(CD_ERROR);
+	}
+
+	cdrom->numtracks = 
+			SDL_SYS_BCDToShort(cdpb.csParam.bytes[1]) - 
+			SDL_SYS_BCDToShort(cdpb.csParam.bytes[0]) + 1;
+	if ( cdrom->numtracks > SDL_MAX_TRACKS )
+		cdrom->numtracks = SDL_MAX_TRACKS;
+	cdrom->cur_track = 0; /* Apparently these are set elsewhere */
+	cdrom->cur_frame = 0; /* Apparently these are set elsewhere */
+
+
+	if (1 || SDL_cdlist[cdrom->id].hasAudio) {
+		/* Get the current playback status */
+		SDL_memset(&cdpb, 0, sizeof(cdpb));
+		cdpb.ioVRefNum = SDL_cdlist[cdrom->id].driveNum;
+		cdpb.ioCRefNum = SDL_cdlist[cdrom->id].dRefNum;
+		cdpb.csCode = kAudioStatus;
+		if ( PBControlSync((ParmBlkPtr)&cdpb) != noErr ) {
+			SDL_SetError("PBControlSync() failed");
+			return(-1);
+		}
+	
+		switch(cdpb.csParam.cd.status) {
+			case kStatusPlaying:
+				status = CD_PLAYING;
+				spinning = true;
+				break;
+			case kStatusPaused:
+				status = CD_PAUSED;
+				spinning = true;
+				break;
+			case kStatusMuted:
+				status = CD_PLAYING; /* What should I do here? */
+				spinning = true;
+				break;
+			case kStatusDone:
+				status = CD_STOPPED;
+				spinning = true;
+				break;
+			case kStatusStopped:
+				status = CD_STOPPED;
+				spinning = false;
+				break;
+			case kStatusError:
+			default:
+				status = CD_ERROR;
+				spinning = false;
+				break;
+			}
+
+		if (spinning && position) *position = MSF_TO_FRAMES(
+				SDL_SYS_BCDToShort(cdpb.csParam.cd.minute),
+				SDL_SYS_BCDToShort(cdpb.csParam.cd.second),
+				SDL_SYS_BCDToShort(cdpb.csParam.cd.frame));
+		}
+	else
+		status = CD_ERROR; /* What should I do here? */
+
+	return(status);
+}
+
+/* Start play */
+static int SDL_SYS_CDPlay(SDL_CD *cdrom, int start, int length)
+{
+	CDCntrlParam cdpb;
+
+	/* Pause the current audio playback to avoid audible artifacts */
+	if ( SDL_SYS_CDPause(cdrom) < 0 ) {
+		return(-1);
+	}
+
+	/* Specify the AudioCD playback mode */
+	SDL_memset(&cdpb, 0, sizeof(cdpb));
+	cdpb.ioVRefNum = SDL_cdlist[cdrom->id].driveNum;
+	cdpb.ioCRefNum = SDL_cdlist[cdrom->id].dRefNum;
+	cdpb.csCode = kSetPlayMode;
+	cdpb.csParam.bytes[0] = false;			/* Repeat? */
+	cdpb.csParam.bytes[1] = kPlayModeSequential;	/* Play mode */
+	/* <20><><EFBFBD><EFBFBD>Treat as soft error, NEC Drive doesnt support this call <20><><EFBFBD> */
+	PBControlSync((ParmBlkPtr) &cdpb);
+
+#if 1
+	/* Specify the end of audio playback */
+	SDL_memset(&cdpb, 0, sizeof(cdpb));
+	cdpb.ioVRefNum = SDL_cdlist[cdrom->id].driveNum;
+	cdpb.ioCRefNum = SDL_cdlist[cdrom->id].dRefNum;
+	cdpb.csCode = kAudioStop;
+	cdpb.csParam.words[0] = kBlockPosition;		/* Position Mode */
+	*(long *) (cdpb.csParam.words + 1) = start+length-1; /* Search Address */
+	if ( PBControlSync((ParmBlkPtr)&cdpb) != noErr ) {
+		SDL_SetError("PBControlSync() failed");
+		return(-1);
+	}
+
+	/* Specify the start of audio playback, and start it */
+	SDL_memset(&cdpb, 0, sizeof(cdpb));
+	cdpb.ioVRefNum = SDL_cdlist[cdrom->id].driveNum;
+	cdpb.ioCRefNum = SDL_cdlist[cdrom->id].dRefNum;
+	cdpb.csCode = kAudioPlay;
+	cdpb.csParam.words[0] = kBlockPosition;			/* Position Mode */
+	*(long *) (cdpb.csParam.words + 1) = start+1;	/* Search Address */
+	cdpb.csParam.words[3] = false;					/* Stop address? */
+	cdpb.csParam.words[4] = kStereoPlayMode;		/* Audio Play Mode */
+	if ( PBControlSync((ParmBlkPtr)&cdpb) != noErr ) {
+		SDL_SetError("PBControlSync() failed");
+		return(-1);
+	}
+#else
+	/* Specify the end of audio playback */
+	FRAMES_TO_MSF(start+length, &m, &s, &f);
+	SDL_memset(&cdpb, 0, sizeof(cdpb));
+	cdpb.ioVRefNum = SDL_cdlist[cdrom->id].driveNum;
+	cdpb.ioCRefNum = SDL_cdlist[cdrom->id].dRefNum;
+	cdpb.csCode = kAudioStop;
+	cdpb.csParam.words[0] = kTrackPosition;			/* Position Mode */
+	cdpb.csParam.words[1] = 0;						/* Search Address (hiword)*/
+	cdpb.csParam.words[2] = 						/* Search Address (loword)*/
+			SDL_SYS_ShortToBCD(cdrom->numtracks);	
+	if ( PBControlSync((ParmBlkPtr)&cdpb) != noErr ) {
+		SDL_SetError("PBControlSync() failed");
+		return(-1);
+	}
+
+	/* Specify the start of audio playback, and start it */
+	FRAMES_TO_MSF(start, &m, &s, &f);
+	SDL_memset(&cdpb, 0, sizeof(cdpb));
+	cdpb.ioVRefNum = SDL_cdlist[cdrom->id].driveNum;
+	cdpb.ioCRefNum = SDL_cdlist[cdrom->id].dRefNum;
+	cdpb.csCode = kAudioPlay;
+	cdpb.csParam.words[0] = kTrackPosition;			/* Position Mode */
+	cdpb.csParam.words[1] = 0;						/* Search Address (hiword)*/
+	cdpb.csParam.words[2] = SDL_SYS_ShortToBCD(1);	/* Search Address (loword)*/
+	cdpb.csParam.words[3] = false;					/* Stop address? */
+	cdpb.csParam.words[4] = kStereoPlayMode;		/* Audio Play Mode */
+	if ( PBControlSync((ParmBlkPtr)&cdpb) != noErr ) {
+		SDL_SetError("PBControlSync() failed");
+		return(-1);
+	}
+#endif
+
+	return(0);
+}
+
+/* Pause play */
+static int SDL_SYS_CDPause(SDL_CD *cdrom)
+{
+	CDCntrlParam cdpb;
+
+	SDL_memset(&cdpb, 0, sizeof(cdpb));
+	cdpb.ioVRefNum = SDL_cdlist[cdrom->id].driveNum;
+	cdpb.ioCRefNum = SDL_cdlist[cdrom->id].dRefNum;
+	cdpb.csCode = kAudioPause;
+	cdpb.csParam.words[0] = 0;	/* Pause/Continue Flag (hiword) */
+	cdpb.csParam.words[1] = 1;	/* Pause/Continue Flag (loword) */
+	if ( PBControlSync((ParmBlkPtr)&cdpb) != noErr ) {
+		SDL_SetError("PBControlSync() failed");
+		return(-1);
+	}
+	return(0);
+}
+
+/* Resume play */
+static int SDL_SYS_CDResume(SDL_CD *cdrom)
+{
+	CDCntrlParam cdpb;
+
+	SDL_memset(&cdpb, 0, sizeof(cdpb));
+	cdpb.ioVRefNum = SDL_cdlist[cdrom->id].driveNum;
+	cdpb.ioCRefNum = SDL_cdlist[cdrom->id].dRefNum;
+	cdpb.csCode = kAudioPause;
+	cdpb.csParam.words[0] = 0;	/* Pause/Continue Flag (hiword) */
+	cdpb.csParam.words[1] = 0;	/* Pause/Continue Flag (loword) */
+	if ( PBControlSync((ParmBlkPtr)&cdpb) != noErr ) {
+		SDL_SetError("PBControlSync() failed");
+		return(-1);
+	}
+	return(0);
+}
+
+/* Stop play */
+static int SDL_SYS_CDStop(SDL_CD *cdrom)
+{
+	CDCntrlParam cdpb;
+
+	SDL_memset(&cdpb, 0, sizeof(cdpb));
+	cdpb.ioVRefNum = SDL_cdlist[cdrom->id].driveNum;
+	cdpb.ioCRefNum = SDL_cdlist[cdrom->id].dRefNum;
+	cdpb.csCode = kAudioStop;
+	cdpb.csParam.words[0] = 0;		/* Position Mode */
+	cdpb.csParam.words[1] = 0;		/* Search Address (hiword) */
+	cdpb.csParam.words[2] = 0;		/* Search Address (loword) */
+	if ( PBControlSync((ParmBlkPtr)&cdpb) != noErr ) {
+		SDL_SetError("PBControlSync() failed");
+		return(-1);
+	}
+	return(0);
+}
+
+/* Eject the CD-ROM */
+static int SDL_SYS_CDEject(SDL_CD *cdrom)
+{
+	Boolean	disk = false;
+	QHdr *driveQ = GetDrvQHdr();
+	DrvQEl *driveElem;
+	HParamBlockRec hpb;
+	ParamBlockRec cpb;
+
+	for ( driveElem = (DrvQEl *) driveQ->qHead; driveElem; driveElem = 
+			  (driveElem) ? ((DrvQEl *) driveElem->qLink) : 
+			  ((DrvQEl *) driveQ->qHead) ) {
+		if ( driveQ->qTail ) {
+			driveQ->qTail->qLink = 0;
+		}
+		if ( driveElem->dQRefNum != SDL_cdlist[cdrom->id].dRefNum ) {
+			continue;
+		}
+	
+		/* Does drive contain mounted volume? If not, skip */
+		SDL_memset(&hpb, 0, sizeof(hpb));
+		hpb.volumeParam.ioVRefNum = driveElem->dQDrive;
+		if ( PBHGetVInfoSync(&hpb) != noErr ) {
+			continue;
+		}
+		if ( (UnmountVol(0, driveElem->dQDrive) == noErr) && 
+		     (Eject(0, driveElem->dQDrive) == noErr) ) {
+			driveElem = 0; /* Clear pointer to reset our loop */
+			disk = true;
+		}
+	}
+
+	/* If no disk is present, just eject the tray */
+	if (! disk) {
+		SDL_memset(&cpb, 0, sizeof(cpb));
+		cpb.cntrlParam.ioVRefNum = 0; /* No Drive */
+		cpb.cntrlParam.ioCRefNum = SDL_cdlist[cdrom->id].dRefNum;
+		cpb.cntrlParam.csCode = kEjectTheDisc;
+		if ( PBControlSync((ParmBlkPtr)&cpb) != noErr ) {
+			SDL_SetError("PBControlSync() failed");
+			return(-1);
+		}
+	}
+	return(0);
+}
+
+/* Close the CD-ROM handle */
+static void SDL_SYS_CDClose(SDL_CD *cdrom)
+{
+	return;
+}
+
+void SDL_SYS_CDQuit(void)
+{
+	while(SDL_numcds--) 
+		SDL_memset(SDL_cdlist + SDL_numcds, 0, sizeof(SDL_cdlist[0]));
+}
+
+#endif /* SDL_CDROM_MACOS */