/
mixer.c
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/*
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SDL_mixer: An audio mixer library based on the SDL library
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Copyright (C) 1997-2004 Sam Lantinga
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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
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slouken@libsdl.org
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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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/* 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" */
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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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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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/* 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 _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) {
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);
}
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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;
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Uint32 sdl_ticks;
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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;
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 ) {
if( mix_channel[i].fading == MIX_FADING_OUT ) {
mix_channel[i].playing = 0;
mix_channel[i].expire = 0;
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Mix_Volume(i, mix_channel[i].fade_volume); /* Restore the volume */
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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)
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) {
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)
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 ) {
if ( --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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}
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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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}
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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void _Mix_InitEffects(void);
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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 ) {
++audio_opened;
return(0);
}
/* 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 *) 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;
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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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++) {
Mix_HaltChannel(i);
}
}
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SDL_LockAudio();
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mix_channel = (struct _Mix_Channel *) 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;
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 */
chunk = (Mix_Chunk *)malloc(sizeof(Mix_Chunk));
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), 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);
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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free(chunk);
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 */
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);
}
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samplesize = ((wavespec.format & 0xFF)/8)*wavespec.channels;
wavecvt.len = chunk->alen & ~(samplesize-1);
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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);
}
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chunk->allocated = 1;
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chunk->abuf = wavecvt.buf;
chunk->alen = wavecvt.len_cvt;
chunk->volume = MIX_MAX_VOLUME;
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 */
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);
}
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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 */
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);
}
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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 */
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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;
}
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}
}
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SDL_UnlockAudio();
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/* Actually free the chunk */
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if ( chunk->allocated ) {
free(chunk->abuf);
}
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free(chunk);
}
}
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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))
{
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SDL_LockAudio();
channel_done_callback = channel_finished;
SDL_UnlockAudio();
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}
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/* 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)
{
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if (num > num_channels)
num = num_channels;
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reserved_channels = num;
return num;
}
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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;
}
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/* Play an audio chunk on a specific channel.
If the specified channel is -1, play on the first free channel.
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'ticks' is the number of milliseconds at most to play the sample, or -1
if there is no limit.
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Returns which channel was used to play the sound.
*/
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int Mix_PlayChannelTimed(int which, Mix_Chunk *chunk, int loops, int ticks)
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{
int i;
/* Don't play null pointers :-) */
if ( chunk == NULL ) {
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Mix_SetError("Tried to play a NULL chunk");
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return(-1);
}
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665
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667
if ( !checkchunkintegral(chunk)) {
Mix_SetError("Tried to play a chunk with a bad frame");
return(-1);
}
668
669
/* Lock the mixer while modifying the playing channels */
670
SDL_LockAudio();
671
672
673
{
/* If which is -1, play on the first free channel */
if ( which == -1 ) {
674
for ( i=reserved_channels; i<num_channels; ++i ) {
675
if ( mix_channel[i].playing <= 0 )
676
677
break;
}
678
if ( i == num_channels ) {
679
Mix_SetError("No free channels available");
680
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682
683
684
685
686
687
which = -1;
} else {
which = i;
}
}
/* Queue up the audio data for this channel */
if ( which >= 0 ) {
688
Uint32 sdl_ticks = SDL_GetTicks();
689
if (Mix_Playing(which))
690
_Mix_channel_done_playing(which);
691
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698
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;
699
700
}
}
701
SDL_UnlockAudio();
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706
/* Return the channel on which the sound is being played */
return(which);
}
707
708
709
710
711
712
713
714
715
716
717
/* 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 ) {
718
SDL_LockAudio();
719
mix_channel[which].expire = (ticks>0) ? (SDL_GetTicks() + ticks) : 0;
720
SDL_UnlockAudio();
721
722
723
724
725
++ status;
}
return(status);
}
726
/* Fade in a sound on a channel, over ms milliseconds */
727
int Mix_FadeInChannelTimed(int which, Mix_Chunk *chunk, int loops, int ms, int ticks)
728
729
730
731
732
733
734
{
int i;
/* Don't play null pointers :-) */
if ( chunk == NULL ) {
return(-1);
}
735
736
737
738
if ( !checkchunkintegral(chunk)) {
Mix_SetError("Tried to play a chunk with a bad frame");
return(-1);
}
739
740
/* Lock the mixer while modifying the playing channels */
741
SDL_LockAudio();
742
743
744
{
/* If which is -1, play on the first free channel */
if ( which == -1 ) {
745
for ( i=reserved_channels; i<num_channels; ++i ) {
746
if ( mix_channel[i].playing <= 0 )
747
748
break;
}
749
if ( i == num_channels ) {
750
751
752
753
754
755
756
757
which = -1;
} else {
which = i;
}
}
/* Queue up the audio data for this channel */
if ( which >= 0 ) {
758
Uint32 sdl_ticks = SDL_GetTicks();
759
if (Mix_Playing(which))
760
_Mix_channel_done_playing(which);
761
762
763
764
765
766
767
768
769
770
771
mix_channel[which].samples = chunk->abuf;
mix_channel[which].playing = chunk->alen;
mix_channel[which].looping = loops;
mix_channel[which].chunk = chunk;
mix_channel[which].paused = 0;
mix_channel[which].fading = MIX_FADING_IN;
mix_channel[which].fade_volume = mix_channel[which].volume;
mix_channel[which].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;
772
773
}
}
774
SDL_UnlockAudio();
775
776
777
778
779
780
781
782
783
784
785
786
787
/* Return the channel on which the sound is being played */
return(which);
}
/* Set volume of a particular channel */
int Mix_Volume(int which, int volume)
{
int i;
int prev_volume;
if ( which == -1 ) {
prev_volume = 0;
788
for ( i=0; i<num_channels; ++i ) {
789
790
prev_volume += Mix_Volume(i, volume);
}
791
prev_volume /= num_channels;
792
} else {
793
prev_volume = mix_channel[which].volume;
794
795
796
797
798
if ( volume >= 0 ) {
if ( volume > SDL_MIX_MAXVOLUME ) {
volume = SDL_MIX_MAXVOLUME;
}
mix_channel[which].volume = volume;
799
}
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
}
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 ) {
824
for ( i=0; i<num_channels; ++i ) {
825
826
827
Mix_HaltChannel(i);
}
} else {
828
SDL_LockAudio();
829
if (mix_channel[which].playing) {
830
_Mix_channel_done_playing(which);
831
mix_channel[which].playing = 0;
832
}
833
834
835
836
mix_channel[which].expire = 0;
if(mix_channel[which].fading != MIX_NO_FADING) /* Restore volume */
mix_channel[which].volume = mix_channel[which].fade_volume;
mix_channel[which].fading = MIX_NO_FADING;
837
SDL_UnlockAudio();
838
839
840
841
}
return(0);
}
842
843
844
845
846
847
/* Halt playing of a particular group of channels */
int Mix_HaltGroup(int tag)
{
int i;
for ( i=0; i<num_channels; ++i ) {
848
if( mix_channel[i].tag == tag ) {
849
850
851
852
853
854
Mix_HaltChannel(i);
}
}
return(0);
}
855
856
857
858
859
860
/* Fade out a channel and then stop it automatically */
int Mix_FadeOutChannel(int which, int ms)
{
int status;
status = 0;
861
862
863
if ( audio_opened ) {
if ( which == -1 ) {
int i;
864
865
866
867
868
for ( i=0; i<num_channels; ++i ) {
status += Mix_FadeOutChannel(i, ms);
}
} else {
869
SDL_LockAudio();
870
871
872
873
874
875
876
877
878
879
if ( mix_channel[which].playing &&
(mix_channel[which].volume > 0) &&
(mix_channel[which].fading != MIX_FADING_OUT) ) {
mix_channel[which].fading = MIX_FADING_OUT;
mix_channel[which].fade_volume = mix_channel[which].volume;
mix_channel[which].fade_length = ms;
mix_channel[which].ticks_fade = SDL_GetTicks();
++status;
}
880
SDL_UnlockAudio();
881
882
883
884
885
}
}
return(status);
}
886
887
888
889
890
891
/* Halt playing of a particular group of channels */
int Mix_FadeOutGroup(int tag, int ms)
{
int i;
int status = 0;
for ( i=0; i<num_channels; ++i ) {
892
if( mix_channel[i].tag == tag ) {
893
894
895
896
897
898
status += Mix_FadeOutChannel(i,ms);
}
}
return(status);
}
899
900
Mix_Fading Mix_FadingChannel(int which)
{
901
return mix_channel[which].fading;
902
903
}
904
/* Check the status of a specific channel.
905
If the specified mix_channel is -1, check all mix channels.
906
907
908
909
910
911
912
913
914
*/
int Mix_Playing(int which)
{
int status;
status = 0;
if ( which == -1 ) {
int i;
915
for ( i=0; i<num_channels; ++i ) {
916
917
918
if ((mix_channel[i].playing > 0) ||
(mix_channel[i].looping > 0))
{
919
920
921
922
++status;
}
}
} else {
923
924
925
if ((mix_channel[which].playing > 0) ||
(mix_channel[which].looping > 0))
{
926
927
928
929
930
931
++status;
}
}
return(status);
}
932
933
934
935
936
937
938
939
940
941
942
943
/* rcg06072001 Get the chunk associated with a channel. */
Mix_Chunk *Mix_GetChunk(int channel)
{
Mix_Chunk *retval = NULL;
if ((channel >= 0) && (channel < num_channels)) {
retval = mix_channel[channel].chunk;
}
return(retval);
}
944
945
946
/* Close the mixer, halting all playing audio */
void Mix_CloseAudio(void)
{
947
948
int i;
949
950
if ( audio_opened ) {
if ( audio_opened == 1 ) {
951
952
953
954
for (i = 0; i < num_channels; i++) {
Mix_UnregisterAllEffects(i);
}
Mix_UnregisterAllEffects(MIX_CHANNEL_POST);
955
956
957
close_music();
Mix_HaltChannel(-1);
SDL_CloseAudio();
958
959
free(mix_channel);
mix_channel = NULL;
960
961
962
963
964
965
966
967
}
--audio_opened;
}
}
/* Pause a particular channel (or all) */
void Mix_Pause(int which)
{
968
Uint32 sdl_ticks = SDL_GetTicks();
969
970
971
if ( which == -1 ) {
int i;
972
for ( i=0; i<num_channels; ++i ) {
973
974
if ( mix_channel[i].playing > 0 ) {
mix_channel[i].paused = sdl_ticks;
975
976
977
}
}
} else {
978
979
if ( mix_channel[which].playing > 0 ) {
mix_channel[which].paused = sdl_ticks;
980
981
982
983
984
985
986
}
}
}
/* Resume a paused channel */
void Mix_Resume(int which)
{
987
Uint32 sdl_ticks = SDL_GetTicks();
988
989
SDL_LockAudio();
990
991
992
if ( which == -1 ) {
int i;
993
for ( i=0; i<num_channels; ++i ) {
994
995
996
997
if ( mix_channel[i].playing > 0 ) {
if(mix_channel[i].expire > 0)
mix_channel[i].expire += sdl_ticks - mix_channel[i].paused;
mix_channel[i].paused = 0;
998
999
1000
}
}
} else {