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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) 1997-2016 Sam Lantinga <slouken@libsdl.org>
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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 "mixer.h"
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#include "load_aiff.h"
#include "load_voc.h"
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#include "load_mp3.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"
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#include "dynamic_fluidsynth.h"
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#include "dynamic_modplug.h"
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#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" */
#define WAVE 0x45564157 /* "WAVE" */
#define FORM 0x4d524f46 /* "FORM" */
#define OGGS 0x5367674f /* "OggS" */
#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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static SDL_AudioDeviceID audio_device;
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typedef struct _Mix_effectinfo
{
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Mix_EffectFunc_t callback;
Mix_EffectDone_t done_callback;
void *udata;
struct _Mix_effectinfo *next;
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} effect_info;
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static struct _Mix_Channel {
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Mix_Chunk *chunk;
int playing;
int paused;
Uint8 *samples;
int volume;
int looping;
int tag;
Uint32 expire;
Uint32 start_time;
Mix_Fading fading;
int fade_volume;
int fade_volume_reset;
Uint32 fade_length;
Uint32 ticks_fade;
effect_info *effects;
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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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{
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return(num_decoders);
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}
const char *Mix_GetChunkDecoder(int index)
{
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if ((index < 0) || (index >= num_decoders)) {
return NULL;
}
return(chunk_decoders[index]);
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}
static void add_chunk_decoder(const char *decoder)
{
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void *ptr = SDL_realloc((void *)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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}
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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;
SDL_MIXER_VERSION(&linked_version);
return(&linked_version);
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}
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static int initialized = 0;
int Mix_Init(int flags)
{
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int result = 0;
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if (flags & MIX_INIT_FLUIDSYNTH) {
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#ifdef USE_FLUIDSYNTH_MIDI
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if ((initialized & MIX_INIT_FLUIDSYNTH) || Mix_InitFluidSynth() == 0) {
result |= MIX_INIT_FLUIDSYNTH;
}
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#else
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Mix_SetError("Mixer not built with FluidSynth support");
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#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) {
result |= MIX_INIT_FLAC;
}
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#else
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Mix_SetError("Mixer not built with FLAC support");
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#endif
}
if (flags & MIX_INIT_MODPLUG) {
#ifdef MODPLUG_MUSIC
if ((initialized & MIX_INIT_MODPLUG) || Mix_InitModPlug() == 0) {
result |= MIX_INIT_MODPLUG;
}
#else
Mix_SetError("Mixer not built with MOD modplug support");
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#endif
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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) {
result |= MIX_INIT_MOD;
}
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#else
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Mix_SetError("Mixer not built with MOD mikmod support");
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#endif
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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) {
result |= MIX_INIT_MP3;
}
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#elif defined(MP3_MAD_MUSIC)
result |= MIX_INIT_MP3;
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#else
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Mix_SetError("Mixer not built with MP3 support");
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#endif
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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) {
result |= MIX_INIT_OGG;
}
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#else
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Mix_SetError("Mixer not built with Ogg Vorbis support");
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#endif
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}
initialized |= result;
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return (result);
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}
void Mix_Quit()
{
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#ifdef USE_FLUIDSYNTH_MIDI
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if (initialized & MIX_INIT_FLUIDSYNTH) {
Mix_QuitFluidSynth();
}
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#endif
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#ifdef FLAC_MUSIC
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if (initialized & MIX_INIT_FLAC) {
Mix_QuitFLAC();
}
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#endif
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#ifdef MODPLUG_MUSIC
if (initialized & MIX_INIT_MODPLUG) {
Mix_QuitModPlug();
}
#endif
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#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
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if (soundfont_paths) {
SDL_free(soundfont_paths);
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soundfont_paths = NULL;
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}
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#endif
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initialized = 0;
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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.
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* MAKE SURE Mix_LockAudio() is called before this (or you're in the
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* audio callback).
*/
static void _Mix_channel_done_playing(int channel)
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{
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if (channel_done_callback) {
channel_done_callback(channel);
}
/*
* Call internal function directly, to avoid locking audio from
* inside audio callback.
*/
_Mix_remove_all_effects(channel, &mix_channel[channel].effects);
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}
static void *Mix_DoEffects(int chan, void *snd, int len)
{
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int posteffect = (chan == MIX_CHANNEL_POST);
effect_info *e = ((posteffect) ? posteffects : mix_channel[chan].effects);
void *buf = snd;
if (e != NULL) { /* are there any registered effects? */
/* if this is the postmix, we can just overwrite the original. */
if (!posteffect) {
buf = SDL_malloc(len);
if (buf == NULL) {
return(snd);
}
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SDL_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);
}
}
}
/* be sure to SDL_free() the return value if != snd ... */
return(buf);
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}
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/* Mixing function */
static void mix_channels(void *udata, Uint8 *stream, int len)
{
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Uint8 *mix_input;
int i, mixable, volume = SDL_MIX_MAXVOLUME;
Uint32 sdl_ticks;
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#if SDL_VERSION_ATLEAST(1, 3, 0)
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/* Need to initialize the stream in SDL 1.3+ */
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SDL_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) */
if ( music_active || (mix_music != music_mixer) ) {
mix_music(music_data, stream, len);
}
/* Mix any playing channels... */
sdl_ticks = SDL_GetTicks();
for ( i=0; i<num_channels; ++i ) {
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if ( !mix_channel[i].paused ) {
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if ( mix_channel[i].expire > 0 && mix_channel[i].expire < sdl_ticks ) {
/* Expiration delay for that channel is reached */
mix_channel[i].playing = 0;
mix_channel[i].looping = 0;
mix_channel[i].fading = MIX_NO_FADING;
mix_channel[i].expire = 0;
_Mix_channel_done_playing(i);
} else if ( mix_channel[i].fading != MIX_NO_FADING ) {
Uint32 ticks = sdl_ticks - mix_channel[i].ticks_fade;
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if ( ticks >= mix_channel[i].fade_length ) {
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Mix_Volume(i, mix_channel[i].fade_volume_reset); /* Restore the volume */
if( mix_channel[i].fading == MIX_FADING_OUT ) {
mix_channel[i].playing = 0;
mix_channel[i].looping = 0;
mix_channel[i].expire = 0;
_Mix_channel_done_playing(i);
}
mix_channel[i].fading = MIX_NO_FADING;
} else {
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if ( mix_channel[i].fading == MIX_FADING_OUT ) {
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Mix_Volume(i, (mix_channel[i].fade_volume * (mix_channel[i].fade_length-ticks))
/ mix_channel[i].fade_length );
} else {
Mix_Volume(i, (mix_channel[i].fade_volume * ticks) / mix_channel[i].fade_length );
}
}
}
if ( mix_channel[i].playing > 0 ) {
int index = 0;
int remaining = len;
while (mix_channel[i].playing > 0 && index < len) {
remaining = len - index;
volume = (mix_channel[i].volume*mix_channel[i].chunk->volume) / MIX_MAX_VOLUME;
mixable = mix_channel[i].playing;
if ( mixable > remaining ) {
mixable = remaining;
}
mix_input = Mix_DoEffects(i, mix_channel[i].samples, mixable);
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SDL_MixAudioFormat(stream+index,mix_input,mixer.format,mixable,volume);
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if (mix_input != mix_channel[i].samples)
SDL_free(mix_input);
mix_channel[i].samples += mixable;
mix_channel[i].playing -= mixable;
index += mixable;
/* rcg06072001 Alert app if channel is done playing. */
if (!mix_channel[i].playing && !mix_channel[i].looping) {
_Mix_channel_done_playing(i);
}
}
/* If looping the sample and we are at its end, make sure
we will still return a full buffer */
while ( mix_channel[i].looping && index < len ) {
int alen = mix_channel[i].chunk->alen;
remaining = len - index;
if (remaining > alen) {
remaining = alen;
}
mix_input = Mix_DoEffects(i, mix_channel[i].chunk->abuf, remaining);
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SDL_MixAudioFormat(stream+index, mix_input, mixer.format, remaining, volume);
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if (mix_input != mix_channel[i].chunk->abuf)
SDL_free(mix_input);
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if (mix_channel[i].looping > 0) {
--mix_channel[i].looping;
}
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mix_channel[i].samples = mix_channel[i].chunk->abuf + remaining;
mix_channel[i].playing = mix_channel[i].chunk->alen - remaining;
index += remaining;
}
if ( ! mix_channel[i].playing && mix_channel[i].looping ) {
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if (mix_channel[i].looping > 0) {
--mix_channel[i].looping;
}
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mix_channel[i].samples = mix_channel[i].chunk->abuf;
mix_channel[i].playing = mix_channel[i].chunk->alen;
}
}
}
}
/* rcg06122001 run posteffects... */
Mix_DoEffects(MIX_CHANNEL_POST, stream, len);
if ( mix_postmix ) {
mix_postmix(mix_postmix_data, stream, len);
}
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}
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#if 0
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static void PrintFormat(char *title, SDL_AudioSpec *fmt)
{
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printf("%s: %d bit %s audio (%s) at %u Hz\n", title, (fmt->format&0xFF),
(fmt->format&0x8000) ? "signed" : "unsigned",
(fmt->channels > 2) ? "surround" :
(fmt->channels > 1) ? "stereo" : "mono", fmt->freq);
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}
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#endif
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/* Open the mixer with a certain desired audio format */
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int Mix_OpenAudioDevice(int frequency, Uint16 format, int nchannels, int chunksize,
const char* device, int allowed_changes)
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{
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int i;
SDL_AudioSpec desired;
/* If the mixer is already opened, increment open count */
if ( audio_opened ) {
if ( format == mixer.format && nchannels == mixer.channels ) {
++audio_opened;
return(0);
}
while ( audio_opened ) {
Mix_CloseAudio();
}
}
/* Set the desired format and frequency */
desired.freq = frequency;
desired.format = format;
desired.channels = nchannels;
desired.samples = chunksize;
desired.callback = mix_channels;
desired.userdata = NULL;
/* Accept nearly any audio format */
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if ( (audio_device = SDL_OpenAudioDevice(device, 0, &desired, &mixer, allowed_changes)) == 0 ) {
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return(-1);
}
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#if 0
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PrintFormat("Audio device", &mixer);
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#endif
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/* Initialize the music players */
if ( open_music(&mixer) < 0 ) {
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SDL_CloseAudioDevice(audio_device);
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return(-1);
}
num_channels = MIX_CHANNELS;
mix_channel = (struct _Mix_Channel *) SDL_malloc(num_channels * sizeof(struct _Mix_Channel));
/* Clear out the audio channels */
for ( i=0; i<num_channels; ++i ) {
mix_channel[i].chunk = NULL;
mix_channel[i].playing = 0;
mix_channel[i].looping = 0;
mix_channel[i].volume = SDL_MIX_MAXVOLUME;
mix_channel[i].fade_volume = SDL_MIX_MAXVOLUME;
mix_channel[i].fade_volume_reset = SDL_MIX_MAXVOLUME;
mix_channel[i].fading = MIX_NO_FADING;
mix_channel[i].tag = -1;
mix_channel[i].expire = 0;
mix_channel[i].effects = NULL;
mix_channel[i].paused = 0;
}
Mix_VolumeMusic(SDL_MIX_MAXVOLUME);
_Mix_InitEffects();
/* This list is (currently) decided at build time. */
add_chunk_decoder("WAVE");
add_chunk_decoder("AIFF");
add_chunk_decoder("VOC");
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#ifdef OGG_MUSIC
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add_chunk_decoder("OGG");
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#endif
#ifdef FLAC_MUSIC
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add_chunk_decoder("FLAC");
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#endif
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#if defined(MP3_MUSIC) || defined(MP3_MAD_MUSIC)
add_chunk_decoder("MP3");
#endif
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audio_opened = 1;
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SDL_PauseAudioDevice(audio_device, 0);
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return(0);
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}
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/* Open the mixer with a certain desired audio format */
int Mix_OpenAudio(int frequency, Uint16 format, int nchannels, int chunksize)
{
return Mix_OpenAudioDevice(frequency, format, nchannels, chunksize,
NULL, SDL_AUDIO_ALLOW_ANY_CHANGE);
}
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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)
{
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if ( numchans<0 || numchans==num_channels )
return(num_channels);
if ( numchans < num_channels ) {
/* Stop the affected channels */
int i;
for(i=numchans; i < num_channels; i++) {
Mix_UnregisterAllEffects(i);
Mix_HaltChannel(i);
}
}
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Mix_LockAudio();
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mix_channel = (struct _Mix_Channel *) SDL_realloc(mix_channel, numchans * sizeof(struct _Mix_Channel));
if ( numchans > num_channels ) {
/* Initialize the new channels */
int i;
for(i=num_channels; i < numchans; i++) {
mix_channel[i].chunk = NULL;
mix_channel[i].playing = 0;
mix_channel[i].looping = 0;
mix_channel[i].volume = SDL_MIX_MAXVOLUME;
mix_channel[i].fade_volume = SDL_MIX_MAXVOLUME;
mix_channel[i].fade_volume_reset = SDL_MIX_MAXVOLUME;
mix_channel[i].fading = MIX_NO_FADING;
mix_channel[i].tag = -1;
mix_channel[i].expire = 0;
mix_channel[i].effects = NULL;
mix_channel[i].paused = 0;
}
}
num_channels = numchans;
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Mix_UnlockAudio();
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return(num_channels);
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}
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/* Return the actual mixer parameters */
int Mix_QuerySpec(int *frequency, Uint16 *format, int *channels)
{
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if ( audio_opened ) {
if ( frequency ) {
*frequency = mixer.freq;
}
if ( format ) {
*format = mixer.format;
}
if ( channels ) {
*channels = mixer.channels;
}
}
return(audio_opened);
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}
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static int detect_mp3(Uint8 *magic)
{
if ( strncmp((char *)magic, "ID3", 3) == 0 ) {
return 1;
}
/* Detection code lifted from SMPEG */
if(((magic[0] & 0xff) != 0xff) || // No sync bits
((magic[1] & 0xf0) != 0xf0) || //
((magic[2] & 0xf0) == 0x00) || // Bitrate is 0
((magic[2] & 0xf0) == 0xf0) || // Bitrate is 15
((magic[2] & 0x0c) == 0x0c) || // Frequency is 3
((magic[1] & 0x06) == 0x00)) { // Layer is 4
return(0);
}
return 1;
}
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/* Load a wave file */
Mix_Chunk *Mix_LoadWAV_RW(SDL_RWops *src, int freesrc)
{
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Uint32 magic;
Mix_Chunk *chunk;
SDL_AudioSpec wavespec, *loaded;
SDL_AudioCVT wavecvt;
int samplesize;
/* rcg06012001 Make sure src is valid */
if ( ! src ) {
SDL_SetError("Mix_LoadWAV_RW with NULL src");
return(NULL);
}
/* Make sure audio has been opened */
if ( ! audio_opened ) {
SDL_SetError("Audio device hasn't been opened");
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if ( freesrc ) {
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SDL_RWclose(src);
}
return(NULL);
}
/* Allocate the chunk memory */
chunk = (Mix_Chunk *)SDL_malloc(sizeof(Mix_Chunk));
if ( chunk == NULL ) {
SDL_SetError("Out of memory");
if ( freesrc ) {
SDL_RWclose(src);
}
return(NULL);
}
/* Find out what kind of audio file this is */
magic = SDL_ReadLE32(src);
/* Seek backwards for compatibility with older loaders */
SDL_RWseek(src, -(int)sizeof(Uint32), RW_SEEK_CUR);
switch (magic) {
case WAVE:
case RIFF:
loaded = SDL_LoadWAV_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
break;
case FORM:
loaded = Mix_LoadAIFF_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
break;
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#ifdef OGG_MUSIC
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case OGGS:
loaded = Mix_LoadOGG_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
break;
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#endif
#ifdef FLAC_MUSIC
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case FLAC:
loaded = Mix_LoadFLAC_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
break;
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#endif
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case CREA:
loaded = Mix_LoadVOC_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
break;
default:
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#if defined(MP3_MUSIC) || defined(MP3_MAD_MUSIC)
if (detect_mp3((Uint8*)&magic))
{
/* note: send a copy of the mixer spec */
wavespec = mixer;
loaded = Mix_LoadMP3_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
break;
}
#endif
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SDL_SetError("Unrecognized sound file type");
if ( freesrc ) {
SDL_RWclose(src);
}
loaded = NULL;
break;
}
if ( !loaded ) {
/* The individual loaders have closed src if needed */
SDL_free(chunk);
return(NULL);
}
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#if 0
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PrintFormat("Audio device", &mixer);
PrintFormat("-- Wave file", &wavespec);
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#endif
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/* Build the audio converter and create conversion buffers */
if ( wavespec.format != mixer.format ||
wavespec.channels != mixer.channels ||
wavespec.freq != mixer.freq ) {
if ( SDL_BuildAudioCVT(&wavecvt,
wavespec.format, wavespec.channels, wavespec.freq,
mixer.format, mixer.channels, mixer.freq) < 0 ) {
SDL_free(chunk->abuf);
SDL_free(chunk);
return(NULL);
}
samplesize = ((wavespec.format & 0xFF)/8)*wavespec.channels;
wavecvt.len = chunk->alen & ~(samplesize-1);
wavecvt.buf = (Uint8 *)SDL_calloc(1, wavecvt.len*wavecvt.len_mult);
if ( wavecvt.buf == NULL ) {
SDL_SetError("Out of memory");
SDL_free(chunk->abuf);
SDL_free(chunk);
return(NULL);
}
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SDL_memcpy(wavecvt.buf, chunk->abuf, chunk->alen);
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SDL_free(chunk->abuf);
/* Run the audio converter */
if ( SDL_ConvertAudio(&wavecvt) < 0 ) {
SDL_free(wavecvt.buf);
SDL_free(chunk);
return(NULL);
}
chunk->abuf = wavecvt.buf;
chunk->alen = wavecvt.len_cvt;
}
chunk->allocated = 1;
chunk->volume = MIX_MAX_VOLUME;
return(chunk);
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}
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/* Load a wave file of the mixer format from a memory buffer */
Mix_Chunk *Mix_QuickLoad_WAV(Uint8 *mem)
{
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Mix_Chunk *chunk;
Uint8 magic[4];
/* Make sure audio has been opened */
if ( ! audio_opened ) {
SDL_SetError("Audio device hasn't been opened");
return(NULL);
}
/* Allocate the chunk memory */
chunk = (Mix_Chunk *)SDL_calloc(1,sizeof(Mix_Chunk));
if ( chunk == NULL ) {
SDL_SetError("Out of memory");
return(NULL);
}
/* Essentially just skip to the audio data (no error checking - fast) */
chunk->allocated = 0;
mem += 12; /* WAV header */
do {
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SDL_memcpy(magic, mem, 4);
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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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}
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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)
{
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Mix_Chunk *chunk;
/* Make sure audio has been opened */
if ( ! audio_opened ) {
SDL_SetError("Audio device hasn't been opened");
return(NULL);
}
/* Allocate the chunk memory */
chunk = (Mix_Chunk *)SDL_malloc(sizeof(Mix_Chunk));
if ( chunk == NULL ) {
SDL_SetError("Out of memory");
return(NULL);
}
/* Essentially just point at the audio data (no error checking - fast) */
chunk->allocated = 0;
chunk->alen = len;
chunk->abuf = mem;
chunk->volume = MIX_MAX_VOLUME;
return(chunk);
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}
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/* Free an audio chunk previously loaded */
void Mix_FreeChunk(Mix_Chunk *chunk)
{
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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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Mix_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;
mix_channel[i].looping = 0;
}
}
}
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Mix_UnlockAudio();
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/* Actually free the chunk */
if ( chunk->allocated ) {
SDL_free(chunk->abuf);
}
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)
{
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Mix_LockAudio();
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mix_postmix_data = arg;
mix_postmix = mix_func;
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Mix_UnlockAudio();
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}
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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)
{
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Mix_LockAudio();
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if ( mix_func != NULL ) {
music_data = arg;
mix_music = mix_func;
} else {
music_data = NULL;
mix_music = music_mixer;
}
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Mix_UnlockAudio();
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}
void *Mix_GetMusicHookData(void)
{
848
return(music_data);
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}
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void Mix_ChannelFinished(void (*channel_finished)(int channel))
{
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Mix_LockAudio();
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channel_done_callback = channel_finished;
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Mix_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;
reserved_channels = num;
return num;
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}
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static int checkchunkintegral(Mix_Chunk *chunk)
{
873
int frame_width = 1;
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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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}
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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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{
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int i;
/* Don't play null pointers :-) */
if ( chunk == NULL ) {
Mix_SetError("Tried to play a NULL chunk");
return(-1);
}
if ( !checkchunkintegral(chunk)) {
Mix_SetError("Tried to play a chunk with a bad frame");
return(-1);
}
/* Lock the mixer while modifying the playing channels */
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Mix_LockAudio();
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{
/* If which is -1, play on the first free channel */
if ( which == -1 ) {
for ( i=reserved_channels; i<num_channels; ++i ) {
if ( mix_channel[i].playing <= 0 )
break;
}
if ( i == num_channels ) {
Mix_SetError("No free channels available");
which = -1;
} else {
which = i;
}
}
/* Queue up the audio data for this channel */
if ( which >= 0 && which < num_channels ) {
Uint32 sdl_ticks = SDL_GetTicks();
if (Mix_Playing(which))
_Mix_channel_done_playing(which);
mix_channel[which].samples = chunk->abuf;
mix_channel[which].playing = chunk->alen;
mix_channel[which].looping = loops;
mix_channel[which].chunk = chunk;
mix_channel[which].paused = 0;
mix_channel[which].fading = MIX_NO_FADING;
mix_channel[which].start_time = sdl_ticks;
mix_channel[which].expire = (ticks>0) ? (sdl_ticks + ticks) : 0;
}
}
933
Mix_UnlockAudio();
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/* Return the channel on which the sound is being played */
return(which);
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}
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/* Change the expiration delay for a channel */
int Mix_ExpireChannel(int which, int ticks)
{
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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 ) {
950
Mix_LockAudio();
951
mix_channel[which].expire = (ticks>0) ? (SDL_GetTicks() + ticks) : 0;
952
Mix_UnlockAudio();
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++ status;
}
return(status);
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957
}
958
/* Fade in a sound on a channel, over ms milliseconds */
959
int Mix_FadeInChannelTimed(int which, Mix_Chunk *chunk, int loops, int ms, int ticks)
960
{
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971
972
int i;
/* Don't play null pointers :-) */
if ( chunk == NULL ) {
return(-1);
}
if ( !checkchunkintegral(chunk)) {
Mix_SetError("Tried to play a chunk with a bad frame");
return(-1);
}
/* Lock the mixer while modifying the playing channels */
973
Mix_LockAudio();
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{
/* If which is -1, play on the first free channel */
if ( which == -1 ) {
for ( i=reserved_channels; i<num_channels; ++i ) {
if ( mix_channel[i].playing <= 0 )
break;
}
if ( i == num_channels ) {
which = -1;
} else {
which = i;
}
}
/* Queue up the audio data for this channel */
if ( which >= 0 && which < num_channels ) {
Uint32 sdl_ticks = SDL_GetTicks();
if (Mix_Playing(which))
_Mix_channel_done_playing(which);
mix_channel[which].samples = chunk->abuf;
mix_channel[which].playing = chunk->alen;
mix_channel[which].looping = loops;
mix_channel[which].chunk = chunk;
mix_channel[which].paused = 0;
mix_channel[which].fading = MIX_FADING_IN;
mix_channel[which].fade_volume = mix_channel[which].volume;
mix_channel[which].fade_volume_reset = mix_channel[which].volume;