--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/src/audio/SDL_audio.c	Thu Apr 26 16:45:43 2001 +0000
@@ -0,0 +1,516 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997, 1998, 1999, 2000, 2001  Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Library General Public
+    License as published by the Free Software Foundation; either
+    version 2 of the License, or (at your option) any later version.
+
+    This library is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+    Library General Public License for more details.
+
+    You should have received a copy of the GNU Library General Public
+    License along with this library; if not, write to the Free
+    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+
+    Sam Lantinga
+    slouken@devolution.com
+*/
+
+#ifdef SAVE_RCSID
+static char rcsid =
+ "@(#) $Id$";
+#endif
+
+/* Allow access to a raw mixing buffer */
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "SDL.h"
+#include "SDL_audio.h"
+#include "SDL_timer.h"
+#include "SDL_error.h"
+#include "SDL_audio_c.h"
+#include "SDL_audiomem.h"
+#include "SDL_sysaudio.h"
+
+/* Available audio drivers */
+static AudioBootStrap *bootstrap[] = {
+#if defined(unix) && \
+   !defined(linux) && !defined(__FreeBSD__) && !defined(__CYGWIN32__) \
+   && !defined(__bsdi__)
+	&AUDIO_bootstrap,
+#endif
+#ifdef OSS_SUPPORT
+	&DSP_bootstrap,
+	&DMA_bootstrap,
+#endif
+#ifdef ALSA_SUPPORT
+	&ALSA_bootstrap,
+#endif
+#ifdef ARTSC_SUPPORT
+	&ARTSC_bootstrap,
+#endif
+#ifdef ESD_SUPPORT
+	&ESD_bootstrap,
+#endif
+#ifdef NAS_SUPPORT
+	&NAS_bootstrap,
+#endif
+#ifdef ENABLE_DIRECTX
+	&DSOUND_bootstrap,
+#endif
+#ifdef ENABLE_WINDIB
+	&WAVEOUT_bootstrap,
+#endif
+#ifdef __BEOS__
+	&BAUDIO_bootstrap,
+#endif
+#if defined(macintosh) || TARGET_API_MAC_CARBON
+	&SNDMGR_bootstrap,
+#endif
+#ifdef _AIX
+	&Paud_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 SDL_RunAudio(void *audiop)
+{
+	SDL_AudioDevice *audio = (SDL_AudioDevice *)audiop;
+	Uint8 *stream;
+	int    stream_len;
+	void  *udata;
+	void (*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;
+	}
+	stream = audio->fake_stream;
+
+	/* Loop, filling the audio buffers */
+	while ( audio->enabled ) {
+
+		/* Wait for new current buffer to finish playing */
+		if ( stream == audio->fake_stream ) {
+			SDL_Delay((audio->spec.samples*1000)/audio->spec.freq);
+		} else {
+			audio->WaitAudio(audio);
+		}
+
+		/* Fill the current buffer with sound */
+		if ( audio->convert.needed ) {
+			/* The buffer may not be allocated yet */
+			if ( audio->convert.buf ) {
+				stream = audio->convert.buf;
+			} else {
+				continue;
+			}
+		} else {
+			stream = audio->GetAudioBuf(audio);
+			if ( stream == NULL ) {
+				stream = audio->fake_stream;
+			}
+		}
+		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;
+			}
+			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 the audio to drain.. */
+	if ( audio->WaitDone ) {
+		audio->WaitDone(audio);
+	}
+	return(0);
+}
+
+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;
+#ifdef unix
+	if ( (driver_name == NULL) && (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 ( strcmp(bootstrap[i]->name, "esd") == 0 ) {
+				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");
+				}
+				if ( bootstrap[i]->available() ) {
+					audio = bootstrap[i]->create(0);
+					break;
+				}
+#ifdef linux	/* No unsetenv() on most platforms */
+				if ( esd_no_spawn == NULL ) {
+					unsetenv("ESD_NO_SPAWN");
+				}
+#endif
+			}
+		}
+	}
+#endif /* unix */
+	if ( audio == NULL ) {
+		if ( driver_name != NULL ) {
+#if 0	/* This will be replaced with a better driver selection API */
+			if ( strrchr(driver_name, ':') != NULL ) {
+				idx = atoi(strrchr(driver_name, ':')+1);
+			}
+#endif
+			for ( i=0; bootstrap[i]; ++i ) {
+				if (strncmp(bootstrap[i]->name, driver_name,
+				            strlen(bootstrap[i]->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;
+	}
+	return(0);
+}
+
+char *SDL_AudioDriverName(char *namebuf, int maxlen)
+{
+	if ( current_audio != NULL ) {
+		strncpy(namebuf, current_audio->name, maxlen-1);
+		namebuf[maxlen-1] = '\0';
+		return(namebuf);
+	}
+	return(NULL);
+}
+
+int SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained)
+{
+	SDL_AudioDevice *audio;
+
+	/* 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;
+
+	/* Verify some parameters */
+	if ( desired->callback == NULL ) {
+		SDL_SetError("SDL_OpenAudio() passed a NULL callback");
+		return(-1);
+	}
+	switch ( desired->channels ) {
+	    case 1:	/* Mono */
+	    case 2:	/* Stereo */
+		break;
+	    default:
+		SDL_SetError("1 (mono) and 2 (stereo) channels supported");
+		return(-1);
+	}
+
+#ifdef macintosh
+	/* FIXME: Need to implement PPC interrupt asm for SDL_LockAudio() */
+#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
+
+	/* Calculate the silence and size of the audio specification */
+	SDL_CalculateAudioSpec(desired);
+
+	/* Open the audio subsystem */
+	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 ( memcmp(desired, &audio->spec, sizeof(audio->spec)) == 0 ) {
+		/* Just copy over the desired audio specification */
+		if ( obtained != NULL ) {
+			memcpy(obtained, &audio->spec, sizeof(audio->spec));
+		}
+	} else {
+		/* Copy over the audio specification if possible */
+		if ( obtained != NULL ) {
+			memcpy(obtained, &audio->spec, sizeof(audio->spec));
+		} else {
+			/* 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 = desired->size;
+				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 */
+			audio->thread = SDL_CreateThread(SDL_RunAudio, audio);
+			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 ) {
+		if ( audio->thread && (SDL_ThreadID() == audio->threadid) ) {
+			return;
+		}
+		SDL_mutexP(audio->mixer_lock);
+	}
+}
+
+void SDL_UnlockAudio (void)
+{
+	SDL_AudioDevice *audio = current_audio;
+
+	/* Release lock on the mixing buffers */
+	if ( audio ) {
+		if ( audio->thread && (SDL_ThreadID() == audio->threadid) ) {
+			return;
+		}
+		SDL_mutexV(audio->mixer_lock);
+	}
+}
+
+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;
+}