/
mixer.c
1491 lines (1307 loc) · 35 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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soundfont_paths=NULL;
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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;
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if ( ticks >= mix_channel[i].fade_length ) {
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");
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if ( freesrc ) {
SDL_RWclose(src);
}
loaded = NULL;
break;
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}
if ( !loaded ) {
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/* The individual loaders have closed src if needed */
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SDL_free(chunk);
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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 */
718
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 */
741
SDL_LockAudio();
742
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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;
746
mix_channel[i].looping = 0;
747
}
748
749
}
}
750
SDL_UnlockAudio();
751
/* Actually free the chunk */
752
if ( chunk->allocated ) {
753
SDL_free(chunk->abuf);
754
}
755
SDL_free(chunk);
756
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758
}
}
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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);
}
794
795
void Mix_ChannelFinished(void (*channel_finished)(int channel))
{
796
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798
SDL_LockAudio();
channel_done_callback = channel_finished;
SDL_UnlockAudio();
799
800
801
}
802
803
804
805
806
807
/* 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)
{
808
809
if (num > num_channels)
num = num_channels;
810
811
812
813
reserved_channels = num;
return num;
}
814
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816
817
818
819
820
821
822
823
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;
}
824
825
/* Play an audio chunk on a specific channel.
If the specified channel is -1, play on the first free channel.
826
827
'ticks' is the number of milliseconds at most to play the sample, or -1
if there is no limit.
828
829
Returns which channel was used to play the sound.
*/
830
int Mix_PlayChannelTimed(int which, Mix_Chunk *chunk, int loops, int ticks)
831
832
833
834
835
{
int i;
/* Don't play null pointers :-) */
if ( chunk == NULL ) {
836
Mix_SetError("Tried to play a NULL chunk");
837
838
return(-1);
}
839
840
841
842
if ( !checkchunkintegral(chunk)) {
Mix_SetError("Tried to play a chunk with a bad frame");
return(-1);
}
843
844
/* Lock the mixer while modifying the playing channels */
845
SDL_LockAudio();
846
847
848
{
/* If which is -1, play on the first free channel */
if ( which == -1 ) {
849
for ( i=reserved_channels; i<num_channels; ++i ) {
850
if ( mix_channel[i].playing <= 0 )
851
852
break;
}
853
if ( i == num_channels ) {
854
Mix_SetError("No free channels available");
855
856
857
858
859
860
861
which = -1;
} else {
which = i;
}
}
/* Queue up the audio data for this channel */
862
if ( which >= 0 && which < num_channels ) {
863
Uint32 sdl_ticks = SDL_GetTicks();
864
if (Mix_Playing(which))
865
_Mix_channel_done_playing(which);
866
867
868
869
870
871
872
873
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;
874
875
}
}
876
SDL_UnlockAudio();
877
878
879
880
881
/* Return the channel on which the sound is being played */
return(which);
}
882
883
884
885
886
887
888
889
890
891
892
/* 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 ) {
893
SDL_LockAudio();
894
mix_channel[which].expire = (ticks>0) ? (SDL_GetTicks() + ticks) : 0;
895
SDL_UnlockAudio();
896
897
898
899
900
++ status;
}
return(status);
}
901
/* Fade in a sound on a channel, over ms milliseconds */
902
int Mix_FadeInChannelTimed(int which, Mix_Chunk *chunk, int loops, int ms, int ticks)
903
904
905
906
907
908
909
{
int i;
/* Don't play null pointers :-) */
if ( chunk == NULL ) {
return(-1);
}
910
911
912
913
if ( !checkchunkintegral(chunk)) {
Mix_SetError("Tried to play a chunk with a bad frame");
return(-1);
}
914
915
/* Lock the mixer while modifying the playing channels */
916
SDL_LockAudio();
917
918
919
{
/* If which is -1, play on the first free channel */
if ( which == -1 ) {
920
for ( i=reserved_channels; i<num_channels; ++i ) {
921
if ( mix_channel[i].playing <= 0 )
922
923
break;
}
924
if ( i == num_channels ) {
925
926
927
928
929
930
931
which = -1;
} else {
which = i;
}
}
/* Queue up the audio data for this channel */
932
if ( which >= 0 && which < num_channels ) {
933
Uint32 sdl_ticks = SDL_GetTicks();
934
if (Mix_Playing(which))
935
_Mix_channel_done_playing(which);
936
937
938
939
940
941
942
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;
943
mix_channel[which].fade_volume_reset = mix_channel[which].volume;
944
945
946
947
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;
948
949
}
}
950
SDL_UnlockAudio();
951
952
953
954
955
956
957
958
959
/* 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;
960
int prev_volume = 0;
961
962
if ( which == -1 ) {
963
for ( i=0; i<num_channels; ++i ) {
964
965
prev_volume += Mix_Volume(i, volume);
}
966
prev_volume /= num_channels;
967
} else if ( which < num_channels ) {
968
prev_volume = mix_channel[which].volume;
969
970
971
972
973
if ( volume >= 0 ) {
if ( volume > SDL_MIX_MAXVOLUME ) {
volume = SDL_MIX_MAXVOLUME;
}
mix_channel[which].volume = volume;
974
}
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
}
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 ) {
999
for ( i=0; i<num_channels; ++i ) {
1000
Mix_HaltChannel(i);