Subversion Repositories Kolibri OS

/*

  SDL_mixer:  An audio mixer library based on the SDL library

  Copyright (C) 1997-2012 Sam Lantinga 

  This software is provided 'as-is', without any express or implied

  warranty.  In no event will the authors be held liable for any damages

  arising from the use of this software.

  Permission is granted to anyone to use this software for any purpose,

  including commercial applications, and to alter it and redistribute it

  freely, subject to the following restrictions:

  1. The origin of this software must not be misrepresented; you must not

     claim that you wrote the original software. If you use this software

     in a product, an acknowledgment in the product documentation would be

     appreciated but is not required.

  2. Altered source versions must be plainly marked as such, and must not be

     misrepresented as being the original software.

  3. This notice may not be removed or altered from any source distribution.

*/

/* $Id$ */

#include 

#include 

#include 

#include "SDL_mutex.h"

#include "SDL_endian.h"

#include "SDL_timer.h"

#include "SDL_mixer.h"

#include "load_aiff.h"

#include "load_voc.h"

#include "load_ogg.h"

#include "load_flac.h"

#include "dynamic_flac.h"

#include "dynamic_mod.h"

#include "dynamic_mp3.h"

#include "dynamic_ogg.h"

#define __MIX_INTERNAL_EFFECT__

#include "effects_internal.h"

#define uSDL_Delay SDL_Delay

#define uSDL_GetTicks SDL_GetTicks

/* Magic numbers for various audio file formats */

#define RIFF		0x46464952		/* "RIFF" */

#define WAVE		0x45564157		/* "WAVE" */

#define FORM		0x4d524f46		/* "FORM" */

#define OGGS		0x5367674f		/* "OggS" */

#define CREA		0x61657243		/* "Crea" */

#define FLAC		0x43614C66		/* "fLaC" */

static int audio_opened = 0;

static SDL_AudioSpec mixer;

typedef struct _Mix_effectinfo

{

	Mix_EffectFunc_t callback;

	Mix_EffectDone_t done_callback;

	void *udata;

	struct _Mix_effectinfo *next;

} effect_info;

static struct _Mix_Channel {

	Mix_Chunk *chunk;

	int playing;

	int paused;

	Uint8 *samples;

	int volume;

	int looping;

	int tag;

	Uint32 expire;

	Uint32 start_time;

	Mix_Fading fading;

	int fade_volume;

	int fade_volume_reset;

	Uint32 fade_length;

	Uint32 ticks_fade;

	effect_info *effects;

} *mix_channel = NULL;

static effect_info *posteffects = NULL;

static int num_channels;

static int reserved_channels = 0;

/* Support for hooking into the mixer callback system */

static void (*mix_postmix)(void *udata, Uint8 *stream, int len) = NULL;

static void *mix_postmix_data = NULL;

/* rcg07062001 callback to alert when channels are done playing. */

static void (*channel_done_callback)(int channel) = NULL;

/* Music function declarations */

extern int open_music(SDL_AudioSpec *mixer);

extern void close_music(void);

/* Support for user defined music functions, plus the default one */

extern int volatile music_active;

extern void music_mixer(void *udata, Uint8 *stream, int len);

static void (*mix_music)(void *udata, Uint8 *stream, int len) = music_mixer;

static void *music_data = NULL;

/* rcg06042009 report available decoders at runtime. */

static const char **chunk_decoders = NULL;

static int num_decoders = 0;

/* Semicolon-separated SoundFont paths */

#ifdef MID_MUSIC

extern char* soundfont_paths;

#endif

int Mix_GetNumChunkDecoders(void)

{

	return(num_decoders);

}

const char *Mix_GetChunkDecoder(int index)

{

	if ((index < 0) || (index >= num_decoders)) {

		return NULL;

	}

	return(chunk_decoders[index]);

}

static void add_chunk_decoder(const char *decoder)

{

	void *ptr = SDL_realloc(chunk_decoders, (num_decoders + 1) * sizeof (const char **));

	if (ptr == NULL) {

		return;  /* oh well, go on without it. */

	}

	chunk_decoders = (const char **) ptr;

	chunk_decoders[num_decoders++] = decoder;

}

/* rcg06192001 get linked library's version. */

const SDL_version *Mix_Linked_Version(void)

{

	static SDL_version linked_version;

	SDL_MIXER_VERSION(&linked_version);

	return(&linked_version);

}

static int initialized = 0;

int Mix_Init(int flags)

{

	int result = 0;

	if (flags & MIX_INIT_FLUIDSYNTH) {

#ifdef USE_FLUIDSYNTH_MIDI

		if ((initialized & MIX_INIT_FLUIDSYNTH) || Mix_InitFluidSynth() == 0) {

			result |= MIX_INIT_FLUIDSYNTH;

		}

#else

		Mix_SetError("Mixer not built with FluidSynth support");

#endif

	}

	if (flags & MIX_INIT_FLAC) {

#ifdef FLAC_MUSIC

		if ((initialized & MIX_INIT_FLAC) || Mix_InitFLAC() == 0) {

			result |= MIX_INIT_FLAC;

		}

#else

		Mix_SetError("Mixer not built with FLAC support");

#endif

	}

	if (flags & MIX_INIT_MOD) {

#ifdef MOD_MUSIC

		if ((initialized & MIX_INIT_MOD) || Mix_InitMOD() == 0) {

			result |= MIX_INIT_MOD;

		}

#else

		Mix_SetError("Mixer not built with MOD support");

#endif

	}

	if (flags & MIX_INIT_MP3) {

#ifdef MP3_MUSIC

		if ((initialized & MIX_INIT_MP3) || Mix_InitMP3() == 0) {

			result |= MIX_INIT_MP3;

		}

#else

		Mix_SetError("Mixer not built with MP3 support");

#endif

	}

	if (flags & MIX_INIT_OGG) {

#ifdef OGG_MUSIC

		if ((initialized & MIX_INIT_OGG) || Mix_InitOgg() == 0) {

			result |= MIX_INIT_OGG;

		}

#else

		Mix_SetError("Mixer not built with Ogg Vorbis support");

#endif

	}

	initialized |= result;

	return (result);

}

void Mix_Quit()

{

#ifdef USE_FLUIDSYNTH_MIDI

	if (initialized & MIX_INIT_FLUIDSYNTH) {

		Mix_QuitFluidSynth();

	}

#endif

#ifdef FLAC_MUSIC

	if (initialized & MIX_INIT_FLAC) {

		Mix_QuitFLAC();

	}

#endif

#ifdef MOD_MUSIC

	if (initialized & MIX_INIT_MOD) {

		Mix_QuitMOD();

	}

#endif

#ifdef MP3_MUSIC

	if (initialized & MIX_INIT_MP3) {

		Mix_QuitMP3();

	}

#endif

#ifdef OGG_MUSIC

	if (initialized & MIX_INIT_OGG) {

		Mix_QuitOgg();

	}

#endif

#ifdef MID_MUSIC

	if (soundfont_paths) {

		SDL_free(soundfont_paths);

	}

#endif

	initialized = 0;

}

static int _Mix_remove_all_effects(int channel, effect_info **e);

/*

 * rcg06122001 Cleanup effect callbacks.

 *  MAKE SURE SDL_LockAudio() is called before this (or you're in the

 *   audio callback).

 */

static void _Mix_channel_done_playing(int channel)

{

	if (channel_done_callback) {

	    channel_done_callback(channel);

	}

	/*

	 * Call internal function directly, to avoid locking audio from

	 *   inside audio callback.

	 */

	_Mix_remove_all_effects(channel, &mix_channel[channel].effects);

}

static void *Mix_DoEffects(int chan, void *snd, int len)

{

	int posteffect = (chan == MIX_CHANNEL_POST);

	effect_info *e = ((posteffect) ? posteffects : mix_channel[chan].effects);

	void *buf = snd;

	if (e != NULL) {    /* are there any registered effects? */

		/* if this is the postmix, we can just overwrite the original. */

		if (!posteffect) {

			buf = SDL_malloc(len);

			if (buf == NULL) {

				return(snd);

			}

			memcpy(buf, snd, len);

		}

		for (; e != NULL; e = e->next) {

			if (e->callback != NULL) {

				e->callback(chan, buf, len, e->udata);

			}

		}

	}

	/* be sure to SDL_free() the return value if != snd ... */

	return(buf);

}

/* Mixing function */

static void mix_channels(void *udata, Uint8 *stream, int len)

{

	Uint8 *mix_input;

	int i, mixable, volume = SDL_MIX_MAXVOLUME;

	Uint32 sdl_ticks;

#if SDL_VERSION_ATLEAST(1, 3, 0)

	/* Need to initialize the stream in SDL 1.3+ */

	memset(stream, mixer.silence, len);

#endif

	/* Mix the music (must be done before the channels are added) */

	if ( music_active || (mix_music != music_mixer) ) {

		mix_music(music_data, stream, len);

	}

	/* Mix any playing channels... */

	sdl_ticks =  uSDL_GetTicks();

	for ( i=0; i

		if( ! mix_channel[i].paused ) {

			if ( mix_channel[i].expire > 0 && mix_channel[i].expire < sdl_ticks ) {

				/* Expiration delay for that channel is reached */

				mix_channel[i].playing = 0;

				mix_channel[i].looping = 0;

				mix_channel[i].fading = MIX_NO_FADING;

				mix_channel[i].expire = 0;

				_Mix_channel_done_playing(i);

			} else if ( mix_channel[i].fading != MIX_NO_FADING ) {

				Uint32 ticks = sdl_ticks - mix_channel[i].ticks_fade;

				if( ticks > mix_channel[i].fade_length ) {

				    Mix_Volume(i, mix_channel[i].fade_volume_reset); /* Restore the volume */

					if( mix_channel[i].fading == MIX_FADING_OUT ) {

						mix_channel[i].playing = 0;

						mix_channel[i].looping = 0;

						mix_channel[i].expire = 0;

						_Mix_channel_done_playing(i);

					}

					mix_channel[i].fading = MIX_NO_FADING;

				} else {

					if( mix_channel[i].fading == MIX_FADING_OUT ) {

						Mix_Volume(i, (mix_channel[i].fade_volume * (mix_channel[i].fade_length-ticks))

								   / mix_channel[i].fade_length );

					} else {

						Mix_Volume(i, (mix_channel[i].fade_volume * ticks) / mix_channel[i].fade_length );

					}

				}

			}

			if ( mix_channel[i].playing > 0 ) {

				int index = 0;

				int remaining = len;

				while (mix_channel[i].playing > 0 && index < len) {

					remaining = len - index;

					volume = (mix_channel[i].volume*mix_channel[i].chunk->volume) / MIX_MAX_VOLUME;

					mixable = mix_channel[i].playing;

					if ( mixable > remaining ) {

						mixable = remaining;

					}

					mix_input = Mix_DoEffects(i, mix_channel[i].samples, mixable);

					SDL_MixAudio(stream+index,mix_input,mixable,volume);

					if (mix_input != mix_channel[i].samples)

						SDL_free(mix_input);

					mix_channel[i].samples += mixable;

					mix_channel[i].playing -= mixable;

					index += mixable;

					/* rcg06072001 Alert app if channel is done playing. */

					if (!mix_channel[i].playing && !mix_channel[i].looping) {

						_Mix_channel_done_playing(i);

					}

				}

				/* If looping the sample and we are at its end, make sure

				   we will still return a full buffer */

				while ( mix_channel[i].looping && index < len ) {

					int alen = mix_channel[i].chunk->alen;

					remaining = len - index;

					if (remaining > alen) {

						remaining = alen;

					}

					mix_input = Mix_DoEffects(i, mix_channel[i].chunk->abuf, remaining);

					SDL_MixAudio(stream+index, mix_input, remaining, volume);

					if (mix_input != mix_channel[i].chunk->abuf)

						SDL_free(mix_input);

					--mix_channel[i].looping;

					mix_channel[i].samples = mix_channel[i].chunk->abuf + remaining;

					mix_channel[i].playing = mix_channel[i].chunk->alen - remaining;

					index += remaining;

				}

				if ( ! mix_channel[i].playing && mix_channel[i].looping ) {

					--mix_channel[i].looping;

					mix_channel[i].samples = mix_channel[i].chunk->abuf;

					mix_channel[i].playing = mix_channel[i].chunk->alen;

				}

			}

		}

	}

	/* rcg06122001 run posteffects... */

	Mix_DoEffects(MIX_CHANNEL_POST, stream, len);

	if ( mix_postmix ) {

		mix_postmix(mix_postmix_data, stream, len);

	}

}

#if 0

static void PrintFormat(char *title, SDL_AudioSpec *fmt)

{

	printf("%s: %d bit %s audio (%s) at %u Hz\n", title, (fmt->format&0xFF),

			(fmt->format&0x8000) ? "signed" : "unsigned",

			(fmt->channels > 2) ? "surround" :

			(fmt->channels > 1) ? "stereo" : "mono", fmt->freq);

}

#endif

/* Open the mixer with a certain desired audio format */

int Mix_OpenAudio(int frequency, Uint16 format, int nchannels, int chunksize)

{

	int i;

	SDL_AudioSpec desired;

	/* If the mixer is already opened, increment open count */

	if ( audio_opened ) {

		if ( format == mixer.format && nchannels == mixer.channels ) {

			++audio_opened;

			return(0);

		}

		while ( audio_opened ) {

			Mix_CloseAudio();

		}

	}

	/* Set the desired format and frequency */

	desired.freq = frequency;

	desired.format = format;

	desired.channels = nchannels;

	desired.samples = chunksize;

	desired.callback = mix_channels;

	desired.userdata = NULL;

	/* Accept nearly any audio format */

	if ( SDL_OpenAudio(&desired, &mixer) < 0 ) {

		return(-1);

	}

#if 0

	PrintFormat("Audio device", &mixer);

#endif

	/* Initialize the music players */

	if ( open_music(&mixer) < 0 ) {

		SDL_CloseAudio();

		return(-1);

	}

	num_channels = MIX_CHANNELS;

	mix_channel = (struct _Mix_Channel *) SDL_malloc(num_channels * sizeof(struct _Mix_Channel));

	/* Clear out the audio channels */

	for ( i=0; i

		mix_channel[i].chunk = NULL;

		mix_channel[i].playing = 0;

		mix_channel[i].looping = 0;

		mix_channel[i].volume = SDL_MIX_MAXVOLUME;

		mix_channel[i].fade_volume = SDL_MIX_MAXVOLUME;

		mix_channel[i].fade_volume_reset = SDL_MIX_MAXVOLUME;

		mix_channel[i].fading = MIX_NO_FADING;

		mix_channel[i].tag = -1;

		mix_channel[i].expire = 0;

		mix_channel[i].effects = NULL;

		mix_channel[i].paused = 0;

	}

	Mix_VolumeMusic(SDL_MIX_MAXVOLUME);

	_Mix_InitEffects();

	/* This list is (currently) decided at build time. */

	add_chunk_decoder("WAVE");

	add_chunk_decoder("AIFF");

	add_chunk_decoder("VOC");

#ifdef OGG_MUSIC

	add_chunk_decoder("OGG");

#endif

#ifdef FLAC_MUSIC

	add_chunk_decoder("FLAC");

#endif

	audio_opened = 1;

	SDL_PauseAudio(0);

	return(0);

}

/* Dynamically change the number of channels managed by the mixer.

   If decreasing the number of channels, the upper channels are

   stopped.

 */

int Mix_AllocateChannels(int numchans)

{

	if ( numchans<0 || numchans==num_channels )

		return(num_channels);

	if ( numchans < num_channels ) {

		/* Stop the affected channels */

		int i;

		for(i=numchans; i < num_channels; i++) {

			Mix_UnregisterAllEffects(i);

			Mix_HaltChannel(i);

		}

	}

	SDL_LockAudio();

	mix_channel = (struct _Mix_Channel *) SDL_realloc(mix_channel, numchans * sizeof(struct _Mix_Channel));

	if ( numchans > num_channels ) {

		/* Initialize the new channels */

		int i;

		for(i=num_channels; i < numchans; i++) {

			mix_channel[i].chunk = NULL;

			mix_channel[i].playing = 0;

			mix_channel[i].looping = 0;

			mix_channel[i].volume = SDL_MIX_MAXVOLUME;

			mix_channel[i].fade_volume = SDL_MIX_MAXVOLUME;

			mix_channel[i].fade_volume_reset = SDL_MIX_MAXVOLUME;

			mix_channel[i].fading = MIX_NO_FADING;

			mix_channel[i].tag = -1;

			mix_channel[i].expire = 0;

			mix_channel[i].effects = NULL;

			mix_channel[i].paused = 0;

		}

	}

	num_channels = numchans;

	SDL_UnlockAudio();

	return(num_channels);

}

/* Return the actual mixer parameters */

int Mix_QuerySpec(int *frequency, Uint16 *format, int *channels)

{

	if ( audio_opened ) {

		if ( frequency ) {

			*frequency = mixer.freq;

		}

		if ( format ) {

			*format = mixer.format;

		}

		if ( channels ) {

			*channels = mixer.channels;

		}

	}

	return(audio_opened);

}

/*

 * !!! FIXME: Ideally, we want a Mix_LoadSample_RW(), which will handle the

 *             generic setup, then call the correct file format loader.

 */

/* Load a wave file */

Mix_Chunk *Mix_LoadWAV_RW(SDL_RWops *src, int freesrc)

{

	Uint32 magic;

	Mix_Chunk *chunk;

	SDL_AudioSpec wavespec, *loaded;

	SDL_AudioCVT wavecvt;

	int samplesize;

	/* rcg06012001 Make sure src is valid */

	if ( ! src ) {

		SDL_SetError("Mix_LoadWAV_RW with NULL src");

		return(NULL);

	}

	/* Make sure audio has been opened */

	if ( ! audio_opened ) {

		SDL_SetError("Audio device hasn't been opened");

		if ( freesrc && src ) {

			SDL_RWclose(src);

		}

		return(NULL);

	}

	/* Allocate the chunk memory */

	chunk = (Mix_Chunk *)SDL_malloc(sizeof(Mix_Chunk));

	if ( chunk == NULL ) {

		SDL_SetError("Out of memory");

		if ( freesrc ) {

			SDL_RWclose(src);

		}

		return(NULL);

	}

	/* Find out what kind of audio file this is */

	magic = SDL_ReadLE32(src);

	/* Seek backwards for compatibility with older loaders */

	SDL_RWseek(src, -(int)sizeof(Uint32), SEEK_CUR);

	switch (magic) {

		case WAVE:

		case RIFF:

			loaded = SDL_LoadWAV_RW(src, freesrc, &wavespec,

					(Uint8 **)&chunk->abuf, &chunk->alen);

			break;

		case FORM:

			loaded = Mix_LoadAIFF_RW(src, freesrc, &wavespec,

					(Uint8 **)&chunk->abuf, &chunk->alen);

			break;

#ifdef OGG_MUSIC

		case OGGS:

			loaded = Mix_LoadOGG_RW(src, freesrc, &wavespec,

					(Uint8 **)&chunk->abuf, &chunk->alen);

			break;

#endif

#ifdef FLAC_MUSIC

		case FLAC:

			loaded = Mix_LoadFLAC_RW(src, freesrc, &wavespec,

					(Uint8 **)&chunk->abuf, &chunk->alen);

			break;

#endif

		case CREA:

			loaded = Mix_LoadVOC_RW(src, freesrc, &wavespec,

					(Uint8 **)&chunk->abuf, &chunk->alen);

			break;

		default:

			SDL_SetError("Unrecognized sound file type");

			return(0);

	}

	if ( !loaded ) {

		SDL_free(chunk);

		if ( freesrc ) {

			SDL_RWclose(src);

		}

		return(NULL);

	}

#if 0

	PrintFormat("Audio device", &mixer);

	PrintFormat("-- Wave file", &wavespec);

#endif

	/* Build the audio converter and create conversion buffers */

	if ( wavespec.format != mixer.format ||

		 wavespec.channels != mixer.channels ||

		 wavespec.freq != mixer.freq ) {

		if ( SDL_BuildAudioCVT(&wavecvt,

				wavespec.format, wavespec.channels, wavespec.freq,

				mixer.format, mixer.channels, mixer.freq) < 0 ) {

			SDL_free(chunk->abuf);

			SDL_free(chunk);

			return(NULL);

		}

		samplesize = ((wavespec.format & 0xFF)/8)*wavespec.channels;

		wavecvt.len = chunk->alen & ~(samplesize-1);

		wavecvt.buf = (Uint8 *)SDL_calloc(1, wavecvt.len*wavecvt.len_mult);

		if ( wavecvt.buf == NULL ) {

			SDL_SetError("Out of memory");

			SDL_free(chunk->abuf);

			SDL_free(chunk);

			return(NULL);

		}

		memcpy(wavecvt.buf, chunk->abuf, chunk->alen);

		SDL_free(chunk->abuf);

		/* Run the audio converter */

		if ( SDL_ConvertAudio(&wavecvt) < 0 ) {

			SDL_free(wavecvt.buf);

			SDL_free(chunk);

			return(NULL);

		}

		chunk->abuf = wavecvt.buf;

		chunk->alen = wavecvt.len_cvt;

	}

	chunk->allocated = 1;

	chunk->volume = MIX_MAX_VOLUME;

	return(chunk);

}

/* Load a wave file of the mixer format from a memory buffer */

Mix_Chunk *Mix_QuickLoad_WAV(Uint8 *mem)

{

	Mix_Chunk *chunk;

	Uint8 magic[4];

	/* Make sure audio has been opened */

	if ( ! audio_opened ) {

		SDL_SetError("Audio device hasn't been opened");

		return(NULL);

	}

	/* Allocate the chunk memory */

	chunk = (Mix_Chunk *)SDL_calloc(1,sizeof(Mix_Chunk));

	if ( chunk == NULL ) {

		SDL_SetError("Out of memory");

		return(NULL);

	}

	/* Essentially just skip to the audio data (no error checking - fast) */

	chunk->allocated = 0;

	mem += 12; /* WAV header */

	do {

		memcpy(magic, mem, 4);

		mem += 4;

		chunk->alen = ((mem[3]<<24)|(mem[2]<<16)|(mem[1]<<8)|(mem[0]));

		mem += 4;

		chunk->abuf = mem;

		mem += chunk->alen;

	} while ( memcmp(magic, "data", 4) != 0 );

	chunk->volume = MIX_MAX_VOLUME;

	return(chunk);

}

/* Load raw audio data of the mixer format from a memory buffer */

Mix_Chunk *Mix_QuickLoad_RAW(Uint8 *mem, Uint32 len)

{

	Mix_Chunk *chunk;

	/* Make sure audio has been opened */

	if ( ! audio_opened ) {

		SDL_SetError("Audio device hasn't been opened");

		return(NULL);

	}

	/* Allocate the chunk memory */

	chunk = (Mix_Chunk *)SDL_malloc(sizeof(Mix_Chunk));

	if ( chunk == NULL ) {

		SDL_SetError("Out of memory");

		return(NULL);

	}

	/* Essentially just point at the audio data (no error checking - fast) */

	chunk->allocated = 0;

	chunk->alen = len;

	chunk->abuf = mem;

	chunk->volume = MIX_MAX_VOLUME;

	return(chunk);

}

/* Free an audio chunk previously loaded */

void Mix_FreeChunk(Mix_Chunk *chunk)

{

	int i;

	/* Caution -- if the chunk is playing, the mixer will crash */

	if ( chunk ) {

		/* Guarantee that this chunk isn't playing */

		SDL_LockAudio();

		if ( mix_channel ) {

			for ( i=0; i

				if ( chunk == mix_channel[i].chunk ) {

					mix_channel[i].playing = 0;

					mix_channel[i].looping = 0;

				}

			}

		}

		SDL_UnlockAudio();

		/* Actually free the chunk */

		if ( chunk->allocated ) {

			SDL_free(chunk->abuf);

		}

		SDL_free(chunk);

	}

}

/* Set a function that is called after all mixing is performed.

   This can be used to provide real-time visual display of the audio stream

   or add a custom mixer filter for the stream data.

*/

void Mix_SetPostMix(void (*mix_func)

                    (void *udata, Uint8 *stream, int len), void *arg)

{

	SDL_LockAudio();

	mix_postmix_data = arg;

	mix_postmix = mix_func;

	SDL_UnlockAudio();

}

/* Add your own music player or mixer function.

   If 'mix_func' is NULL, the default music player is re-enabled.

 */

void Mix_HookMusic(void (*mix_func)(void *udata, Uint8 *stream, int len),

                                                                void *arg)

{

	SDL_LockAudio();

	if ( mix_func != NULL ) {

		music_data = arg;

		mix_music = mix_func;

	} else {

		music_data = NULL;

		mix_music = music_mixer;

	}

	SDL_UnlockAudio();

}

void *Mix_GetMusicHookData(void)

{

	return(music_data);

}

void Mix_ChannelFinished(void (*channel_finished)(int channel))

{

	SDL_LockAudio();

	channel_done_callback = channel_finished;

	SDL_UnlockAudio();

}

/* Reserve the first channels (0 -> n-1) for the application, i.e. don't allocate

   them dynamically to the next sample if requested with a -1 value below.

   Returns the number of reserved channels.

 */

int Mix_ReserveChannels(int num)

{

	if (num > num_channels)

		num = num_channels;

	reserved_channels = num;

	return num;

}

static int checkchunkintegral(Mix_Chunk *chunk)

{

	int frame_width = 1;

	if ((mixer.format & 0xFF) == 16) frame_width = 2;

	frame_width *= mixer.channels;

	while (chunk->alen % frame_width) chunk->alen--;

	return chunk->alen;

}

/* Play an audio chunk on a specific channel.

   If the specified channel is -1, play on the first free channel.

   'ticks' is the number of milliseconds at most to play the sample, or -1

   if there is no limit.

   Returns which channel was used to play the sound.

*/

int Mix_PlayChannelTimed(int which, Mix_Chunk *chunk, int loops, int ticks)

{

	int i;

	/* Don't play null pointers :-) */

	if ( chunk == NULL ) {

		Mix_SetError("Tried to play a NULL chunk");

		return(-1);

	}

	if ( !checkchunkintegral(chunk)) {

		Mix_SetError("Tried to play a chunk with a bad frame");

		return(-1);

	}

	/* Lock the mixer while modifying the playing channels */

	SDL_LockAudio();

	{

		/* If which is -1, play on the first free channel */

		if ( which == -1 ) {

			for ( i=reserved_channels; i

				if ( mix_channel[i].playing <= 0 )

					break;

			}

			if ( i == num_channels ) {

				Mix_SetError("No free channels available");

				which = -1;

			} else {

				which = i;

			}

		}

		/* Queue up the audio data for this channel */

		if ( which >= 0 && which < num_channels ) {

			Uint32 sdl_ticks =  uSDL_GetTicks();

			if (Mix_Playing(which))

				_Mix_channel_done_playing(which);

			mix_channel[which].samples = chunk->abuf;

			mix_channel[which].playing = chunk->alen;

			mix_channel[which].looping = loops;

			mix_channel[which].chunk = chunk;

			mix_channel[which].paused = 0;

			mix_channel[which].fading = MIX_NO_FADING;

			mix_channel[which].start_time = sdl_ticks;

			mix_channel[which].expire = (ticks>0) ? (sdl_ticks + ticks) : 0;

		}

	}

	SDL_UnlockAudio();

	/* Return the channel on which the sound is being played */

	return(which);

}

/* Change the expiration delay for a channel */

int Mix_ExpireChannel(int which, int ticks)

{

	int status = 0;

	if ( which == -1 ) {

		int i;

		for ( i=0; i < num_channels; ++ i ) {

			status += Mix_ExpireChannel(i, ticks);

		}

	} else if ( which < num_channels ) {

		SDL_LockAudio();

		mix_channel[which].expire = (ticks>0) ? ( uSDL_GetTicks() + ticks) : 0;

		SDL_UnlockAudio();

		++ status;

	}

	return(status);

}

/* Fade in a sound on a channel, over ms milliseconds */

int Mix_FadeInChannelTimed(int which, Mix_Chunk *chunk, int loops, int ms, int ticks)

{

	int i;

	/* Don't play null pointers :-) */

	if ( chunk == NULL ) {

		return(-1);

	}

	if ( !checkchunkintegral(chunk)) {

		Mix_SetError("Tried to play a chunk with a bad frame");

		return(-1);

	}

	/* Lock the mixer while modifying the playing channels */

	SDL_LockAudio();

	{

		/* If which is -1, play on the first free channel */

		if ( which == -1 ) {

			for ( i=reserved_channels; i

				if ( mix_channel[i].playing <= 0 )

					break;

			}

			if ( i == num_channels ) {

				which = -1;

			} else {

				which = i;

			}

		}

		/* Queue up the audio data for this channel */

		if ( which >= 0 && which < num_channels ) {

			Uint32 sdl_ticks =  uSDL_GetTicks();

			if (Mix_Playing(which))

				_Mix_channel_done_playing(which);

			mix_channel[which].samples = chunk->abuf;

			mix_channel[which].playing = chunk->alen;

			mix_channel[which].looping = loops;

			mix_channel[which].chunk = chunk;

			mix_channel[which].paused = 0;

			mix_channel[which].fading = MIX_FADING_IN;

			mix_channel[which].fade_volume = mix_channel[which].volume;

			mix_channel[which].fade_volume_reset = mix_channel[which].volume;

			mix_channel[which].volume = 0;

			mix_channel[which].fade_length = (Uint32)ms;

			mix_channel[which].start_time = mix_channel[which].ticks_fade = sdl_ticks;

			mix_channel[which].expire = (ticks > 0) ? (sdl_ticks+ticks) : 0;

		}

	}

	SDL_UnlockAudio();

	/* Return the channel on which the sound is being played */

	return(which);

}

/* Set volume of a particular channel */

int Mix_Volume(int which, int volume)

{

	int i;

	int prev_volume = 0;

	if ( which == -1 ) {

		for ( i=0; i

			prev_volume += Mix_Volume(i, volume);

		}

		prev_volume /= num_channels;

	} else if ( which < num_channels ) {

		prev_volume = mix_channel[which].volume;

		if ( volume >= 0 ) {

			if ( volume > SDL_MIX_MAXVOLUME ) {

				volume = SDL_MIX_MAXVOLUME;

			}

			mix_channel[which].volume = volume;

		}

	}

	return(prev_volume);

}

/* Set volume of a particular chunk */

int Mix_VolumeChunk(Mix_Chunk *chunk, int volume)

{

	int prev_volume;

	prev_volume = chunk->volume;

	if ( volume >= 0 ) {

		if ( volume > MIX_MAX_VOLUME ) {

			volume = MIX_MAX_VOLUME;

		}

		chunk->volume = volume;

	}

	return(prev_volume);

}

/* Halt playing of a particular channel */

int Mix_HaltChannel(int which)

{

	int i;

	if ( which == -1 ) {

		for ( i=0; i

			Mix_HaltChannel(i);

		}

	} else if ( which < num_channels ) {

		SDL_LockAudio();

		if (mix_channel[which].playing) {

			_Mix_channel_done_playing(which);

			mix_channel[which].playing = 0;

			mix_channel[which].looping = 0;

		}

		mix_channel[which].expire = 0;

		if(mix_channel[which].fading != MIX_NO_FADING) /* Restore volume */

			mix_channel[which].volume = mix_channel[which].fade_volume_reset;

		mix_channel[which].fading = MIX_NO_FADING;

		SDL_UnlockAudio();

	}

	return(0);

}

/* Halt playing of a particular group of channels */

int Mix_HaltGroup(int tag)

{

	int i;

	for ( i=0; i

		if( mix_channel[i].tag == tag ) {

			Mix_HaltChannel(i);

		}

	}

	return(0);

}

/* Fade out a channel and then stop it automatically */

int Mix_FadeOutChannel(int which, int ms)

{

	int status;

	status = 0;

	if ( audio_opened ) {

		if ( which == -1 ) {

			int i;

			for ( i=0; i

				status += Mix_FadeOutChannel(i, ms);

			}

		} else if ( which < num_channels ) {

			SDL_LockAudio();

			if ( mix_channel[which].playing &&

			    (mix_channel[which].volume > 0) &&

			    (mix_channel[which].fading != MIX_FADING_OUT) ) {

				mix_channel[which].fade_volume = mix_channel[which].volume;

				mix_channel[which].fading = MIX_FADING_OUT;

				mix_channel[which].fade_length = ms;

				mix_channel[which].ticks_fade =  uSDL_GetTicks();

				/* only change fade_volume_reset if we're not fading. */

				if (mix_channel[which].fading == MIX_NO_FADING) {

				    mix_channel[which].fade_volume_reset = mix_channel[which].volume;

				}

				++status;

			}

			SDL_UnlockAudio();

		}

	}

	return(status);

}

/* Halt playing of a particular group of channels */

int Mix_FadeOutGroup(int tag, int ms)

{

	int i;

	int status = 0;

	for ( i=0; i

		if( mix_channel[i].tag == tag ) {

			status += Mix_FadeOutChannel(i,ms);

		}

	}

	return(status);

}

Mix_Fading Mix_FadingChannel(int which)

{

	if ( which < 0 || which >= num_channels ) {

		return MIX_NO_FADING;

	}

	return mix_channel[which].fading;

}

/* Check the status of a specific channel.

   If the specified mix_channel is -1, check all mix channels.

*/

int Mix_Playing(int which)

{

	int status;

	status = 0;

	if ( which == -1 ) {

		int i;

		for ( i=0; i

			if ((mix_channel[i].playing > 0) ||

				(mix_channel[i].looping > 0))

			{

				++status;

			}

		}

	} else if ( which < num_channels ) {

		if ( (mix_channel[which].playing > 0) ||

		     (mix_channel[which].looping > 0) )

		{

			++status;

		}

	}

	return(status);

}

/* rcg06072001 Get the chunk associated with a channel. */

Mix_Chunk *Mix_GetChunk(int channel)

{

	Mix_Chunk *retval = NULL;

	if ((channel >= 0) && (channel < num_channels)) {

		retval = mix_channel[channel].chunk;

	}

	return(retval);

}