/
mixer.c
1488 lines (1304 loc) · 34.9 KB
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/*
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SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
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.
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*/
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/* $Id$ */
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#include <stdio.h>
#include <stdlib.h>
#include <string.h>
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#include "SDL_mutex.h"
#include "SDL_endian.h"
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#include "SDL_timer.h"
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#include "SDL_mixer.h"
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#include "load_aiff.h"
#include "load_voc.h"
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#include "load_ogg.h"
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#include "load_flac.h"
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#include "dynamic_flac.h"
#include "dynamic_mod.h"
#include "dynamic_mp3.h"
#include "dynamic_ogg.h"
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#define __MIX_INTERNAL_EFFECT__
#include "effects_internal.h"
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/* Magic numbers for various audio file formats */
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#define RIFF 0x46464952 /* "RIFF" */
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#define WAVE 0x45564157 /* "WAVE" */
#define FORM 0x4d524f46 /* "FORM" */
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#define OGGS 0x5367674f /* "OggS" */
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#define CREA 0x61657243 /* "Crea" */
#define FLAC 0x43614C66 /* "fLaC" */
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static int audio_opened = 0;
static SDL_AudioSpec mixer;
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typedef struct _Mix_effectinfo
{
Mix_EffectFunc_t callback;
Mix_EffectDone_t done_callback;
void *udata;
struct _Mix_effectinfo *next;
} effect_info;
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static struct _Mix_Channel {
Mix_Chunk *chunk;
int playing;
int paused;
Uint8 *samples;
int volume;
int looping;
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int tag;
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Uint32 expire;
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Uint32 start_time;
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Mix_Fading fading;
int fade_volume;
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int fade_volume_reset;
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Uint32 fade_length;
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Uint32 ticks_fade;
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effect_info *effects;
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} *mix_channel = NULL;
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static effect_info *posteffects = NULL;
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static int num_channels;
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static int reserved_channels = 0;
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/* Support for hooking into the mixer callback system */
static void (*mix_postmix)(void *udata, Uint8 *stream, int len) = NULL;
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static void *mix_postmix_data = NULL;
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/* rcg07062001 callback to alert when channels are done playing. */
static void (*channel_done_callback)(int channel) = NULL;
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/* Music function declarations */
extern int open_music(SDL_AudioSpec *mixer);
extern void close_music(void);
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/* Support for user defined music functions, plus the default one */
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extern int volatile music_active;
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extern void music_mixer(void *udata, Uint8 *stream, int len);
static void (*mix_music)(void *udata, Uint8 *stream, int len) = music_mixer;
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static void *music_data = NULL;
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/* rcg06042009 report available decoders at runtime. */
static const char **chunk_decoders = NULL;
static int num_decoders = 0;
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/* Semicolon-separated SoundFont paths */
#ifdef MID_MUSIC
extern char* soundfont_paths;
#endif
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int Mix_GetNumChunkDecoders(void)
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{
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)
{
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void *ptr = SDL_realloc(chunk_decoders, (num_decoders + 1) * sizeof (const char **));
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if (ptr == NULL) {
return; /* oh well, go on without it. */
}
chunk_decoders = (const char **) ptr;
chunk_decoders[num_decoders++] = decoder;
}
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/* rcg06192001 get linked library's version. */
const SDL_version *Mix_Linked_Version(void)
{
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static SDL_version linked_version;
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SDL_MIXER_VERSION(&linked_version);
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return(&linked_version);
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}
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static int initialized = 0;
int Mix_Init(int flags)
{
int result = 0;
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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
}
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if (flags & MIX_INIT_FLAC) {
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#ifdef FLAC_MUSIC
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if ((initialized & MIX_INIT_FLAC) || Mix_InitFLAC() == 0) {
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result |= MIX_INIT_FLAC;
}
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#else
Mix_SetError("Mixer not built with FLAC support");
#endif
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}
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if (flags & MIX_INIT_MOD) {
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#ifdef MOD_MUSIC
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if ((initialized & MIX_INIT_MOD) || Mix_InitMOD() == 0) {
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result |= MIX_INIT_MOD;
}
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#else
Mix_SetError("Mixer not built with MOD support");
#endif
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}
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if (flags & MIX_INIT_MP3) {
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#ifdef MP3_MUSIC
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if ((initialized & MIX_INIT_MP3) || Mix_InitMP3() == 0) {
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result |= MIX_INIT_MP3;
}
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#else
Mix_SetError("Mixer not built with MP3 support");
#endif
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}
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if (flags & MIX_INIT_OGG) {
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#ifdef OGG_MUSIC
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if ((initialized & MIX_INIT_OGG) || Mix_InitOgg() == 0) {
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result |= MIX_INIT_OGG;
}
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#else
Mix_SetError("Mixer not built with Ogg Vorbis support");
#endif
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}
initialized |= result;
return (result);
}
void Mix_Quit()
{
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#ifdef USE_FLUIDSYNTH_MIDI
if (initialized & MIX_INIT_FLUIDSYNTH) {
Mix_QuitFluidSynth();
}
#endif
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#ifdef FLAC_MUSIC
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if (initialized & MIX_INIT_FLAC) {
Mix_QuitFLAC();
}
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#endif
#ifdef MOD_MUSIC
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if (initialized & MIX_INIT_MOD) {
Mix_QuitMOD();
}
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#endif
#ifdef MP3_MUSIC
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if (initialized & MIX_INIT_MP3) {
Mix_QuitMP3();
}
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#endif
#ifdef OGG_MUSIC
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if (initialized & MIX_INIT_OGG) {
Mix_QuitOgg();
}
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#endif
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#ifdef MID_MUSIC
if (soundfont_paths) {
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SDL_free(soundfont_paths);
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}
#endif
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initialized = 0;
}
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static int _Mix_remove_all_effects(int channel, effect_info **e);
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/*
* 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)
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{
if (channel_done_callback) {
channel_done_callback(channel);
}
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/*
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* Call internal function directly, to avoid locking audio from
* inside audio callback.
*/
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_Mix_remove_all_effects(channel, &mix_channel[channel].effects);
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}
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) {
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buf = SDL_malloc(len);
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if (buf == NULL) {
return(snd);
}
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memcpy(buf, snd, len);
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}
for (; e != NULL; e = e->next) {
if (e->callback != NULL) {
e->callback(chan, buf, len, e->udata);
}
}
}
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/* be sure to SDL_free() the return value if != snd ... */
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return(buf);
}
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/* Mixing function */
static void mix_channels(void *udata, Uint8 *stream, int len)
{
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Uint8 *mix_input;
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int i, mixable, volume = SDL_MIX_MAXVOLUME;
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Uint32 sdl_ticks;
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#if SDL_VERSION_ATLEAST(1, 3, 0)
/* Need to initialize the stream in SDL 1.3+ */
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memset(stream, mixer.silence, len);
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#endif
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/* Mix the music (must be done before the channels are added) */
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if ( music_active || (mix_music != music_mixer) ) {
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mix_music(music_data, stream, len);
}
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/* Mix any playing channels... */
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sdl_ticks = SDL_GetTicks();
for ( i=0; i<num_channels; ++i ) {
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if( ! mix_channel[i].paused ) {
if ( mix_channel[i].expire > 0 && mix_channel[i].expire < sdl_ticks ) {
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/* Expiration delay for that channel is reached */
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mix_channel[i].playing = 0;
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mix_channel[i].looping = 0;
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mix_channel[i].fading = MIX_NO_FADING;
mix_channel[i].expire = 0;
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_Mix_channel_done_playing(i);
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} 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 ) {
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Mix_Volume(i, mix_channel[i].fade_volume_reset); /* Restore the volume */
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if( mix_channel[i].fading == MIX_FADING_OUT ) {
mix_channel[i].playing = 0;
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mix_channel[i].looping = 0;
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mix_channel[i].expire = 0;
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_Mix_channel_done_playing(i);
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}
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mix_channel[i].fading = MIX_NO_FADING;
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} else {
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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 );
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} else {
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Mix_Volume(i, (mix_channel[i].fade_volume * ticks) / mix_channel[i].fade_length );
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}
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}
}
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if ( mix_channel[i].playing > 0 ) {
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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)
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SDL_free(mix_input);
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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);
}
}
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/* If looping the sample and we are at its end, make sure
we will still return a full buffer */
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while ( mix_channel[i].looping && index < len ) {
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int alen = mix_channel[i].chunk->alen;
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remaining = len - index;
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if (remaining > alen) {
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remaining = alen;
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mix_input = Mix_DoEffects(i, mix_channel[i].chunk->abuf, remaining);
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SDL_MixAudio(stream+index, mix_input, remaining, volume);
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if (mix_input != mix_channel[i].chunk->abuf)
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SDL_free(mix_input);
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--mix_channel[i].looping;
mix_channel[i].samples = mix_channel[i].chunk->abuf + remaining;
mix_channel[i].playing = mix_channel[i].chunk->alen - remaining;
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index += remaining;
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}
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if ( ! mix_channel[i].playing && mix_channel[i].looping ) {
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--mix_channel[i].looping;
mix_channel[i].samples = mix_channel[i].chunk->abuf;
mix_channel[i].playing = mix_channel[i].chunk->alen;
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}
}
}
}
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/* rcg06122001 run posteffects... */
Mix_DoEffects(MIX_CHANNEL_POST, stream, len);
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if ( mix_postmix ) {
mix_postmix(mix_postmix_data, stream, len);
}
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}
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#if 0
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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",
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(fmt->channels > 2) ? "surround" :
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(fmt->channels > 1) ? "stereo" : "mono", fmt->freq);
}
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#endif
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/* Open the mixer with a certain desired audio format */
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int Mix_OpenAudio(int frequency, Uint16 format, int nchannels, int chunksize)
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{
int i;
SDL_AudioSpec desired;
/* If the mixer is already opened, increment open count */
if ( audio_opened ) {
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if ( format == mixer.format && nchannels == mixer.channels ) {
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++audio_opened;
return(0);
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}
while ( audio_opened ) {
Mix_CloseAudio();
}
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}
/* Set the desired format and frequency */
desired.freq = frequency;
desired.format = format;
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desired.channels = nchannels;
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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);
}
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num_channels = MIX_CHANNELS;
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mix_channel = (struct _Mix_Channel *) SDL_malloc(num_channels * sizeof(struct _Mix_Channel));
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/* Clear out the audio channels */
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for ( i=0; i<num_channels; ++i ) {
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mix_channel[i].chunk = NULL;
mix_channel[i].playing = 0;
mix_channel[i].looping = 0;
mix_channel[i].volume = SDL_MIX_MAXVOLUME;
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mix_channel[i].fade_volume = SDL_MIX_MAXVOLUME;
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mix_channel[i].fade_volume_reset = SDL_MIX_MAXVOLUME;
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mix_channel[i].fading = MIX_NO_FADING;
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mix_channel[i].tag = -1;
mix_channel[i].expire = 0;
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mix_channel[i].effects = NULL;
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mix_channel[i].paused = 0;
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}
Mix_VolumeMusic(SDL_MIX_MAXVOLUME);
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_Mix_InitEffects();
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/* 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
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audio_opened = 1;
SDL_PauseAudio(0);
return(0);
}
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/* 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++) {
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Mix_UnregisterAllEffects(i);
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Mix_HaltChannel(i);
}
}
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SDL_LockAudio();
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mix_channel = (struct _Mix_Channel *) SDL_realloc(mix_channel, numchans * sizeof(struct _Mix_Channel));
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if ( numchans > num_channels ) {
/* Initialize the new channels */
int i;
for(i=num_channels; i < numchans; i++) {
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mix_channel[i].chunk = NULL;
mix_channel[i].playing = 0;
mix_channel[i].looping = 0;
mix_channel[i].volume = SDL_MIX_MAXVOLUME;
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mix_channel[i].fade_volume = SDL_MIX_MAXVOLUME;
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mix_channel[i].fade_volume_reset = SDL_MIX_MAXVOLUME;
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mix_channel[i].fading = MIX_NO_FADING;
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mix_channel[i].tag = -1;
mix_channel[i].expire = 0;
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mix_channel[i].effects = NULL;
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mix_channel[i].paused = 0;
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}
}
num_channels = numchans;
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SDL_UnlockAudio();
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return(num_channels);
}
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/* 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;
}
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if ( channels ) {
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*channels = mixer.channels;
}
}
return(audio_opened);
}
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/*
* !!! FIXME: Ideally, we want a Mix_LoadSample_RW(), which will handle the
* generic setup, then call the correct file format loader.
*/
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/* Load a wave file */
Mix_Chunk *Mix_LoadWAV_RW(SDL_RWops *src, int freesrc)
{
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Uint32 magic;
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Mix_Chunk *chunk;
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SDL_AudioSpec wavespec, *loaded;
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SDL_AudioCVT wavecvt;
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int samplesize;
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/* rcg06012001 Make sure src is valid */
if ( ! src ) {
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SDL_SetError("Mix_LoadWAV_RW with NULL src");
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return(NULL);
}
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/* Make sure audio has been opened */
if ( ! audio_opened ) {
SDL_SetError("Audio device hasn't been opened");
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if ( freesrc && src ) {
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SDL_RWclose(src);
}
return(NULL);
}
/* Allocate the chunk memory */
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chunk = (Mix_Chunk *)SDL_malloc(sizeof(Mix_Chunk));
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if ( chunk == NULL ) {
SDL_SetError("Out of memory");
if ( freesrc ) {
SDL_RWclose(src);
}
return(NULL);
}
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/* Find out what kind of audio file this is */
magic = SDL_ReadLE32(src);
/* Seek backwards for compatibility with older loaders */
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SDL_RWseek(src, -(int)sizeof(Uint32), RW_SEEK_CUR);
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switch (magic) {
case WAVE:
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case RIFF:
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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;
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#ifdef OGG_MUSIC
case OGGS:
loaded = Mix_LoadOGG_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
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break;
#endif
#ifdef FLAC_MUSIC
case FLAC:
loaded = Mix_LoadFLAC_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
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break;
#endif
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case CREA:
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loaded = Mix_LoadVOC_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
break;
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default:
SDL_SetError("Unrecognized sound file type");
return(0);
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}
if ( !loaded ) {
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SDL_free(chunk);
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if ( freesrc ) {
SDL_RWclose(src);
}
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return(NULL);
}
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#if 0
PrintFormat("Audio device", &mixer);
PrintFormat("-- Wave file", &wavespec);
#endif
/* Build the audio converter and create conversion buffers */
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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 ) {
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SDL_free(chunk->abuf);
SDL_free(chunk);
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return(NULL);
}
samplesize = ((wavespec.format & 0xFF)/8)*wavespec.channels;
wavecvt.len = chunk->alen & ~(samplesize-1);
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wavecvt.buf = (Uint8 *)SDL_calloc(1, wavecvt.len*wavecvt.len_mult);
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if ( wavecvt.buf == NULL ) {
SDL_SetError("Out of memory");
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SDL_free(chunk->abuf);
SDL_free(chunk);
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return(NULL);
}
memcpy(wavecvt.buf, chunk->abuf, chunk->alen);
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SDL_free(chunk->abuf);
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/* Run the audio converter */
if ( SDL_ConvertAudio(&wavecvt) < 0 ) {
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SDL_free(wavecvt.buf);
SDL_free(chunk);
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return(NULL);
}
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chunk->abuf = wavecvt.buf;
chunk->alen = wavecvt.len_cvt;
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}
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chunk->allocated = 1;
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chunk->volume = MIX_MAX_VOLUME;
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return(chunk);
}
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/* 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 */
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chunk = (Mix_Chunk *)SDL_calloc(1,sizeof(Mix_Chunk));
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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);
}
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/* 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 */
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chunk = (Mix_Chunk *)SDL_malloc(sizeof(Mix_Chunk));
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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);
}
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/* 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 */
738
SDL_LockAudio();
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if ( mix_channel ) {
for ( i=0; i<num_channels; ++i ) {
if ( chunk == mix_channel[i].chunk ) {
mix_channel[i].playing = 0;
743
mix_channel[i].looping = 0;
744
}
745
746
}
}
747
SDL_UnlockAudio();
748
/* Actually free the chunk */
749
if ( chunk->allocated ) {
750
SDL_free(chunk->abuf);
751
}
752
SDL_free(chunk);
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}
}
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/* 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();
}
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/* 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);
}
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void Mix_ChannelFinished(void (*channel_finished)(int channel))
{
793
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795
SDL_LockAudio();
channel_done_callback = channel_finished;
SDL_UnlockAudio();
796
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798
}
799
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803
804
/* 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)
{
805
806
if (num > num_channels)
num = num_channels;
807
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809
810
reserved_channels = num;
return num;
}
811
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816
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819
820
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;
}
821
822
/* Play an audio chunk on a specific channel.
If the specified channel is -1, play on the first free channel.
823
824
'ticks' is the number of milliseconds at most to play the sample, or -1
if there is no limit.
825
826
Returns which channel was used to play the sound.
*/
827
int Mix_PlayChannelTimed(int which, Mix_Chunk *chunk, int loops, int ticks)
828
829
830
831
832
{
int i;
/* Don't play null pointers :-) */
if ( chunk == NULL ) {
833
Mix_SetError("Tried to play a NULL chunk");
834
835
return(-1);
}
836
837
838
839
if ( !checkchunkintegral(chunk)) {
Mix_SetError("Tried to play a chunk with a bad frame");
return(-1);
}
840
841
/* Lock the mixer while modifying the playing channels */
842
SDL_LockAudio();
843
844
845
{
/* If which is -1, play on the first free channel */
if ( which == -1 ) {
846
for ( i=reserved_channels; i<num_channels; ++i ) {
847
if ( mix_channel[i].playing <= 0 )
848
849
break;
}
850
if ( i == num_channels ) {
851
Mix_SetError("No free channels available");
852
853
854
855
856
857
858
which = -1;
} else {
which = i;
}
}
/* Queue up the audio data for this channel */
859
if ( which >= 0 && which < num_channels ) {
860
Uint32 sdl_ticks = SDL_GetTicks();
861
if (Mix_Playing(which))
862
_Mix_channel_done_playing(which);
863
864
865
866
867
868
869
870
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;
871
872
}
}
873
SDL_UnlockAudio();
874
875
876
877
878
/* Return the channel on which the sound is being played */
return(which);
}
879
880
881
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883
884
885
886
887
888
889
/* 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 ) {
890
SDL_LockAudio();
891
mix_channel[which].expire = (ticks>0) ? (SDL_GetTicks() + ticks) : 0;
892
SDL_UnlockAudio();
893
894
895
896
897
++ status;
}
return(status);
}
898
/* Fade in a sound on a channel, over ms milliseconds */
899
int Mix_FadeInChannelTimed(int which, Mix_Chunk *chunk, int loops, int ms, int ticks)
900
901
902
903
904
905
906
{
int i;
/* Don't play null pointers :-) */
if ( chunk == NULL ) {
return(-1);
}
907
908
909
910
if ( !checkchunkintegral(chunk)) {
Mix_SetError("Tried to play a chunk with a bad frame");
return(-1);
}
911
912
/* Lock the mixer while modifying the playing channels */
913
SDL_LockAudio();
914
915
916
{
/* If which is -1, play on the first free channel */
if ( which == -1 ) {
917
for ( i=reserved_channels; i<num_channels; ++i ) {
918
if ( mix_channel[i].playing <= 0 )
919
920
break;
}
921
if ( i == num_channels ) {
922
923
924
925
926
927
928
which = -1;
} else {
which = i;
}
}
/* Queue up the audio data for this channel */
929
if ( which >= 0 && which < num_channels ) {
930
Uint32 sdl_ticks = SDL_GetTicks();
931
if (Mix_Playing(which))
932
_Mix_channel_done_playing(which);
933
934
935
936
937
938
939
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;
940
mix_channel[which].fade_volume_reset = mix_channel[which].volume;
941
942
943
944
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;
945
946
}
}
947
SDL_UnlockAudio();
948
949
950
951
952
953
954
955
956
/* 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;
957
int prev_volume = 0;
958
959
if ( which == -1 ) {
960
for ( i=0; i<num_channels; ++i ) {
961
962
prev_volume += Mix_Volume(i, volume);
}
963
prev_volume /= num_channels;
964
} else if ( which < num_channels ) {
965
prev_volume = mix_channel[which].volume;
966
967
968
969
970
if ( volume >= 0 ) {
if ( volume > SDL_MIX_MAXVOLUME ) {
volume = SDL_MIX_MAXVOLUME;
}
mix_channel[which].volume = volume;
971
}
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
}
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 ) {
996
for ( i=0; i<num_channels; ++i ) {
997
998
Mix_HaltChannel(i);
}
999
} else if ( which < num_channels ) {
1000
SDL_LockAudio();