mixer.c

SDL_mixer 是一个基于 SDL 的混音器

/*
    SDL_mixer:  An audio mixer library based on the SDL library
    Copyright (C) 1997-2004 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
    [e-mail]
*/

/* $Id: mixer.c 2290 2006-05-01 03:34:46Z slouken $ */

#include <stdio.h>
#include <stdlib.h>
#include <string.h>

#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"

/* 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" */

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;
	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;


/* 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 _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 = 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 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;
	Uint32 sdl_ticks;

	/* 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 = SDL_GetTicks();
	for ( i=0; i<num_channels; ++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].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 ) {
					if( mix_channel[i].fading == MIX_FADING_OUT ) {
						mix_channel[i].playing = 0;
						mix_channel[i].expire = 0;
						Mix_Volume(i, mix_channel[i].fade_volume); /* Restore the volume */
						_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)
						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)
						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 ) {
					if ( --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);
	}
}

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);
}


void _Mix_InitEffects(void);

/* 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 *) malloc(num_channels * sizeof(struct _Mix_Channel));

	/* Clear out the audio channels */
	for ( i=0; i<num_channels; ++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].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();

	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_HaltChannel(i);
		}
	}
	SDL_LockAudio();
	mix_channel = (struct _Mix_Channel *) 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].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 *)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
		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 ) {
		free(chunk);
		return(NULL);
	}

#if 0
	PrintFormat("Audio device", &mixer);
	PrintFormat("-- Wave file", &wavespec);
#endif

	/* Build the audio converter and create conversion buffers */
	if ( SDL_BuildAudioCVT(&wavecvt,
			wavespec.format, wavespec.channels, wavespec.freq,
			mixer.format, mixer.channels, mixer.freq) < 0 ) {
		SDL_FreeWAV(chunk->abuf);
		free(chunk);
		return(NULL);
	}
	samplesize = ((wavespec.format & 0xFF)/8)*wavespec.channels;
	wavecvt.len = chunk->alen & ~(samplesize-1);
	wavecvt.buf = (Uint8 *)malloc(wavecvt.len*wavecvt.len_mult);
	if ( wavecvt.buf == NULL ) {
		SDL_SetError("Out of memory");
		SDL_FreeWAV(chunk->abuf);
		free(chunk);
		return(NULL);
	}
	memcpy(wavecvt.buf, chunk->abuf, chunk->alen);
	SDL_FreeWAV(chunk->abuf);

	/* Run the audio converter */
	if ( SDL_ConvertAudio(&wavecvt) < 0 ) {
		free(wavecvt.buf);
		free(chunk);
		return(NULL);
	}
	chunk->allocated = 1;
	chunk->abuf = wavecvt.buf;
	chunk->alen = wavecvt.len_cvt;
	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 *)malloc(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 *)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<num_channels; ++i ) {
				if ( chunk == mix_channel[i].chunk ) {
					mix_channel[i].playing = 0;
				}
			}
		}
		SDL_UnlockAudio();
		/* Actually free the chunk */
		if ( chunk->allocated ) {
			free(chunk->abuf);
		}
		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)
{