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mixer.c

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1501 lines (1315 loc) · 35.3 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 */
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static void (SDLCALL *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. */
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static void (SDLCALL *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 SDLCALL music_mixer(void *udata, Uint8 *stream, int len);
static void (SDLCALL *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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#elif defined(MP3_MAD_MUSIC)
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);
}
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void Mix_Quit(void)
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{
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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 */
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static void SDLCALL mix_channels(void *udata, Uint8 *stream, int len)
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{
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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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}
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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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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;
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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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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;
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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);
}
Oct 26, 1999
Oct 26, 1999
451
452
num_channels = MIX_CHANNELS;
Jan 13, 2012
Jan 13, 2012
453
mix_channel = (struct _Mix_Channel *) SDL_malloc(num_channels * sizeof(struct _Mix_Channel));
Oct 26, 1999
Oct 26, 1999
454
Oct 21, 1999
Oct 21, 1999
455
/* Clear out the audio channels */
Oct 26, 1999
Oct 26, 1999
456
for ( i=0; i<num_channels; ++i ) {
Jun 26, 2000
Jun 26, 2000
457
458
459
460
mix_channel[i].chunk = NULL;
mix_channel[i].playing = 0;
mix_channel[i].looping = 0;
mix_channel[i].volume = SDL_MIX_MAXVOLUME;
May 28, 2002
May 28, 2002
461
mix_channel[i].fade_volume = SDL_MIX_MAXVOLUME;
Oct 11, 2009
Oct 11, 2009
462
mix_channel[i].fade_volume_reset = SDL_MIX_MAXVOLUME;
May 28, 2002
May 28, 2002
463
mix_channel[i].fading = MIX_NO_FADING;
Jun 26, 2000
Jun 26, 2000
464
465
mix_channel[i].tag = -1;
mix_channel[i].expire = 0;
Sep 11, 2001
Sep 11, 2001
466
mix_channel[i].effects = NULL;
Dec 2, 2002
Dec 2, 2002
467
mix_channel[i].paused = 0;
Oct 21, 1999
Oct 21, 1999
468
469
}
Mix_VolumeMusic(SDL_MIX_MAXVOLUME);
Sep 11, 2001
Sep 11, 2001
470
471
472
_Mix_InitEffects();
Jun 5, 2009
Jun 5, 2009
473
474
475
476
477
478
479
480
481
482
483
/* 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
Oct 21, 1999
Oct 21, 1999
484
485
486
487
488
audio_opened = 1;
SDL_PauseAudio(0);
return(0);
}
Oct 26, 1999
Oct 26, 1999
489
490
491
492
493
494
495
496
497
498
499
500
501
/* 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++) {
Jul 15, 2007
Jul 15, 2007
502
Mix_UnregisterAllEffects(i);
Oct 26, 1999
Oct 26, 1999
503
504
505
Mix_HaltChannel(i);
}
}
May 3, 2002
May 3, 2002
506
SDL_LockAudio();
Jan 13, 2012
Jan 13, 2012
507
mix_channel = (struct _Mix_Channel *) SDL_realloc(mix_channel, numchans * sizeof(struct _Mix_Channel));
Oct 26, 1999
Oct 26, 1999
508
509
510
511
if ( numchans > num_channels ) {
/* Initialize the new channels */
int i;
for(i=num_channels; i < numchans; i++) {
Jun 26, 2000
Jun 26, 2000
512
513
514
515
mix_channel[i].chunk = NULL;
mix_channel[i].playing = 0;
mix_channel[i].looping = 0;
mix_channel[i].volume = SDL_MIX_MAXVOLUME;
May 28, 2002
May 28, 2002
516
mix_channel[i].fade_volume = SDL_MIX_MAXVOLUME;
Oct 11, 2009
Oct 11, 2009
517
mix_channel[i].fade_volume_reset = SDL_MIX_MAXVOLUME;
May 28, 2002
May 28, 2002
518
mix_channel[i].fading = MIX_NO_FADING;
Jun 26, 2000
Jun 26, 2000
519
520
mix_channel[i].tag = -1;
mix_channel[i].expire = 0;
Sep 11, 2001
Sep 11, 2001
521
mix_channel[i].effects = NULL;
Dec 2, 2002
Dec 2, 2002
522
mix_channel[i].paused = 0;
Oct 26, 1999
Oct 26, 1999
523
524
525
}
}
num_channels = numchans;
May 3, 2002
May 3, 2002
526
SDL_UnlockAudio();
Oct 26, 1999
Oct 26, 1999
527
528
529
return(num_channels);
}
Oct 21, 1999
Oct 21, 1999
530
531
532
533
534
535
536
537
538
539
/* 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;
}
Mar 4, 2001
Mar 4, 2001
540
if ( channels ) {
Oct 21, 1999
Oct 21, 1999
541
542
543
544
545
546
*channels = mixer.channels;
}
}
return(audio_opened);
}
Jun 10, 2001
Jun 10, 2001
547
548
549
550
551
552
/*
* !!! FIXME: Ideally, we want a Mix_LoadSample_RW(), which will handle the
* generic setup, then call the correct file format loader.
*/
Oct 21, 1999
Oct 21, 1999
553
554
555
/* Load a wave file */
Mix_Chunk *Mix_LoadWAV_RW(SDL_RWops *src, int freesrc)
{
Sep 11, 2001
Sep 11, 2001
556
Uint32 magic;
Oct 21, 1999
Oct 21, 1999
557
Mix_Chunk *chunk;
Sep 11, 2001
Sep 11, 2001
558
SDL_AudioSpec wavespec, *loaded;
Oct 21, 1999
Oct 21, 1999
559
SDL_AudioCVT wavecvt;
Jan 28, 2000
Jan 28, 2000
560
int samplesize;
Oct 21, 1999
Oct 21, 1999
561
Jun 10, 2001
Jun 10, 2001
562
563
/* rcg06012001 Make sure src is valid */
if ( ! src ) {
Sep 11, 2001
Sep 11, 2001
564
SDL_SetError("Mix_LoadWAV_RW with NULL src");
Jun 10, 2001
Jun 10, 2001
565
566
567
return(NULL);
}
Oct 21, 1999
Oct 21, 1999
568
569
570
/* Make sure audio has been opened */
if ( ! audio_opened ) {
SDL_SetError("Audio device hasn't been opened");
May 10, 2001
May 10, 2001
571
if ( freesrc && src ) {
Oct 21, 1999
Oct 21, 1999
572
573
574
575
576
577
SDL_RWclose(src);
}
return(NULL);
}
/* Allocate the chunk memory */
Jan 13, 2012
Jan 13, 2012
578
chunk = (Mix_Chunk *)SDL_malloc(sizeof(Mix_Chunk));
Oct 21, 1999
Oct 21, 1999
579
580
581
582
583
584
585
586
if ( chunk == NULL ) {
SDL_SetError("Out of memory");
if ( freesrc ) {
SDL_RWclose(src);
}
return(NULL);
}
Sep 11, 2001
Sep 11, 2001
587
588
589
/* Find out what kind of audio file this is */
magic = SDL_ReadLE32(src);
/* Seek backwards for compatibility with older loaders */
Nov 8, 2009
Nov 8, 2009
590
SDL_RWseek(src, -(int)sizeof(Uint32), RW_SEEK_CUR);
Sep 11, 2001
Sep 11, 2001
591
592
593
switch (magic) {
case WAVE:
Sep 23, 2001
Sep 23, 2001
594
case RIFF:
Sep 11, 2001
Sep 11, 2001
595
596
597
598
599
600
601
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;
Sep 9, 2002
Sep 9, 2002
602
603
604
605
#ifdef OGG_MUSIC
case OGGS:
loaded = Mix_LoadOGG_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
Feb 27, 2008
Feb 27, 2008
606
607
608
609
610
611
break;
#endif
#ifdef FLAC_MUSIC
case FLAC:
loaded = Mix_LoadFLAC_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
Sep 9, 2002
Sep 9, 2002
612
613
break;
#endif
Oct 16, 2002
Oct 16, 2002
614
case CREA:
Sep 11, 2001
Sep 11, 2001
615
616
617
loaded = Mix_LoadVOC_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
break;
Oct 16, 2002
Oct 16, 2002
618
619
default:
SDL_SetError("Unrecognized sound file type");
Feb 12, 2012
Feb 12, 2012
620
621
622
623
624
if ( freesrc ) {
SDL_RWclose(src);
}
loaded = NULL;
break;
Sep 11, 2001
Sep 11, 2001
625
626
}
if ( !loaded ) {
Feb 12, 2012
Feb 12, 2012
627
/* The individual loaders have closed src if needed */
Jan 13, 2012
Jan 13, 2012
628
SDL_free(chunk);
Oct 21, 1999
Oct 21, 1999
629
630
return(NULL);
}
Jun 10, 2001
Jun 10, 2001
631
Oct 21, 1999
Oct 21, 1999
632
633
634
635
636
637
#if 0
PrintFormat("Audio device", &mixer);
PrintFormat("-- Wave file", &wavespec);
#endif
/* Build the audio converter and create conversion buffers */
Jan 13, 2012
Jan 13, 2012
638
639
640
641
642
643
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 ) {
Jan 13, 2012
Jan 13, 2012
644
645
SDL_free(chunk->abuf);
SDL_free(chunk);
Jan 13, 2012
Jan 13, 2012
646
647
648
649
return(NULL);
}
samplesize = ((wavespec.format & 0xFF)/8)*wavespec.channels;
wavecvt.len = chunk->alen & ~(samplesize-1);
Jan 13, 2012
Jan 13, 2012
650
wavecvt.buf = (Uint8 *)SDL_calloc(1, wavecvt.len*wavecvt.len_mult);
Jan 13, 2012
Jan 13, 2012
651
652
if ( wavecvt.buf == NULL ) {
SDL_SetError("Out of memory");
Jan 13, 2012
Jan 13, 2012
653
654
SDL_free(chunk->abuf);
SDL_free(chunk);
Jan 13, 2012
Jan 13, 2012
655
656
return(NULL);
}
Sep 3, 2019
Sep 3, 2019
657
memcpy(wavecvt.buf, chunk->abuf, wavecvt.len);
Jan 13, 2012
Jan 13, 2012
658
SDL_free(chunk->abuf);
Oct 21, 1999
Oct 21, 1999
659
Jan 13, 2012
Jan 13, 2012
660
661
/* Run the audio converter */
if ( SDL_ConvertAudio(&wavecvt) < 0 ) {
Jan 13, 2012
Jan 13, 2012
662
663
SDL_free(wavecvt.buf);
SDL_free(chunk);
Jan 13, 2012
Jan 13, 2012
664
665
return(NULL);
}
Jan 13, 2012
Jan 13, 2012
666
667
668
chunk->abuf = wavecvt.buf;
chunk->alen = wavecvt.len_cvt;
Oct 21, 1999
Oct 21, 1999
669
}
Jan 13, 2012
Jan 13, 2012
670
Oct 21, 1999
Oct 21, 1999
671
chunk->allocated = 1;
Oct 21, 1999
Oct 21, 1999
672
chunk->volume = MIX_MAX_VOLUME;
Sep 14, 2011
Sep 14, 2011
673
Oct 21, 1999
Oct 21, 1999
674
675
676
return(chunk);
}
Oct 21, 1999
Oct 21, 1999
677
678
679
680
681
682
683
684
685
686
687
688
689
/* 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 */
Jan 13, 2012
Jan 13, 2012
690
chunk = (Mix_Chunk *)SDL_calloc(1,sizeof(Mix_Chunk));
Oct 21, 1999
Oct 21, 1999
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
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);
}
May 18, 2002
May 18, 2002
712
713
714
715
716
717
718
719
720
721
722
723
/* 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 */
Jan 13, 2012
Jan 13, 2012
724
chunk = (Mix_Chunk *)SDL_malloc(sizeof(Mix_Chunk));
May 18, 2002
May 18, 2002
725
726
727
728
729
730
731
732
733
734
735
736
737
738
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);
}
Oct 21, 1999
Oct 21, 1999
739
740
741
742
743
744
745
746
/* 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 */
May 3, 2002
May 3, 2002
747
SDL_LockAudio();
Jun 26, 2000
Jun 26, 2000
748
749
750
751
if ( mix_channel ) {
for ( i=0; i<num_channels; ++i ) {
if ( chunk == mix_channel[i].chunk ) {
mix_channel[i].playing = 0;
Jan 3, 2012
Jan 3, 2012
752
mix_channel[i].looping = 0;
Jun 26, 2000
Jun 26, 2000
753
}
Oct 21, 1999
Oct 21, 1999
754
755
}
}
May 3, 2002
May 3, 2002
756
SDL_UnlockAudio();
Oct 21, 1999
Oct 21, 1999
757
/* Actually free the chunk */
Oct 21, 1999
Oct 21, 1999
758
if ( chunk->allocated ) {
Jan 13, 2012
Jan 13, 2012
759
SDL_free(chunk->abuf);
Oct 21, 1999
Oct 21, 1999
760
}
Jan 13, 2012
Jan 13, 2012
761
SDL_free(chunk);
Oct 21, 1999
Oct 21, 1999
762
763
764
}
}
Dec 26, 1999
Dec 26, 1999
765
766
767
768
/* 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.
*/
Oct 7, 2018
Oct 7, 2018
769
void Mix_SetPostMix(void (SDLCALL *mix_func)
Dec 26, 1999
Dec 26, 1999
770
771
772
773
774
775
776
777
(void *udata, Uint8 *stream, int len), void *arg)
{
SDL_LockAudio();
mix_postmix_data = arg;
mix_postmix = mix_func;
SDL_UnlockAudio();
}
Oct 21, 1999
Oct 21, 1999
778
779
780
/* Add your own music player or mixer function.
If 'mix_func' is NULL, the default music player is re-enabled.
*/
Oct 7, 2018
Oct 7, 2018
781
void Mix_HookMusic(void (SDLCALL *mix_func)(void *udata, Uint8 *stream, int len),
Oct 21, 1999
Oct 21, 1999
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
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);
}
Oct 7, 2018
Oct 7, 2018
800
void Mix_ChannelFinished(void (SDLCALL *channel_finished)(int channel))
Jun 10, 2001
Jun 10, 2001
801
{
May 3, 2002
May 3, 2002
802
803
804
SDL_LockAudio();
channel_done_callback = channel_finished;
SDL_UnlockAudio();
Jun 10, 2001
Jun 10, 2001
805
806
807
}
Oct 21, 1999
Oct 21, 1999
808
809
810
811
812
813
/* 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)
{
Oct 26, 1999
Oct 26, 1999
814
815
if (num > num_channels)
num = num_channels;
Oct 21, 1999
Oct 21, 1999
816
817
818
819
reserved_channels = num;
return num;
}
Aug 21, 2004
Aug 21, 2004
820
821
822
823
824
825
826
827
828
829
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;
}
Oct 21, 1999
Oct 21, 1999
830
831
/* Play an audio chunk on a specific channel.
If the specified channel is -1, play on the first free channel.
Oct 26, 1999
Oct 26, 1999
832
833
'ticks' is the number of milliseconds at most to play the sample, or -1
if there is no limit.
Oct 21, 1999
Oct 21, 1999
834
835
Returns which channel was used to play the sound.
*/
Oct 26, 1999
Oct 26, 1999
836
int Mix_PlayChannelTimed(int which, Mix_Chunk *chunk, int loops, int ticks)
Oct 21, 1999
Oct 21, 1999
837
838
839
840
841
{
int i;
/* Don't play null pointers :-) */
if ( chunk == NULL ) {
May 19, 2002
May 19, 2002
842
Mix_SetError("Tried to play a NULL chunk");
Oct 21, 1999
Oct 21, 1999
843
844
return(-1);
}
Aug 21, 2004
Aug 21, 2004
845
846
847
848
if ( !checkchunkintegral(chunk)) {
Mix_SetError("Tried to play a chunk with a bad frame");
return(-1);
}
Oct 21, 1999
Oct 21, 1999
849
850
/* Lock the mixer while modifying the playing channels */
May 3, 2002
May 3, 2002
851
SDL_LockAudio();
Oct 21, 1999
Oct 21, 1999
852
853
854
{
/* If which is -1, play on the first free channel */
if ( which == -1 ) {
Oct 26, 1999
Oct 26, 1999
855
for ( i=reserved_channels; i<num_channels; ++i ) {
Jun 26, 2000
Jun 26, 2000
856
if ( mix_channel[i].playing <= 0 )
Oct 21, 1999
Oct 21, 1999
857
858
break;
}
Oct 26, 1999
Oct 26, 1999
859
if ( i == num_channels ) {
May 19, 2002
May 19, 2002
860
Mix_SetError("No free channels available");
Oct 21, 1999
Oct 21, 1999
861
862
863
864
865
866
867
which = -1;
} else {
which = i;
}
}
/* Queue up the audio data for this channel */
Jan 13, 2011
Jan 13, 2011
868
if ( which >= 0 && which < num_channels ) {
Jan 14, 2002
Jan 14, 2002
869
Uint32 sdl_ticks = SDL_GetTicks();
Jan 14, 2002
Jan 14, 2002
870
if (Mix_Playing(which))
May 3, 2002
May 3, 2002
871
_Mix_channel_done_playing(which);
Jun 26, 2000
Jun 26, 2000
872
873
874
875
876
877
878
879
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;
Oct 23, 1999
Oct 23, 1999
880
881
}
}
May 3, 2002
May 3, 2002
882
SDL_UnlockAudio();
Oct 23, 1999
Oct 23, 1999
883
884
885
886
887
/* Return the channel on which the sound is being played */
return(which);
}
Oct 26, 1999
Oct 26, 1999
888
889
890
891
892
893
894
895
896
897
898
/* 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 ) {
May 3, 2002
May 3, 2002
899
SDL_LockAudio();
Jun 26, 2000
Jun 26, 2000
900
mix_channel[which].expire = (ticks>0) ? (SDL_GetTicks() + ticks) : 0;
May 3, 2002
May 3, 2002
901
SDL_UnlockAudio();
Oct 26, 1999
Oct 26, 1999
902
903
904
905
906
++ status;
}
return(status);
}
Oct 23, 1999
Oct 23, 1999
907
/* Fade in a sound on a channel, over ms milliseconds */
Oct 26, 1999
Oct 26, 1999
908
int Mix_FadeInChannelTimed(int which, Mix_Chunk *chunk, int loops, int ms, int ticks)
Oct 23, 1999
Oct 23, 1999
909
910
911
912
913
914
915
{
int i;
/* Don't play null pointers :-) */
if ( chunk == NULL ) {
return(-1);
}
Aug 21, 2004
Aug 21, 2004
916
917
918
919
if ( !checkchunkintegral(chunk)) {
Mix_SetError("Tried to play a chunk with a bad frame");
return(-1);
}
Oct 23, 1999
Oct 23, 1999
920
921
/* Lock the mixer while modifying the playing channels */
May 3, 2002
May 3, 2002
922
SDL_LockAudio();
Oct 23, 1999
Oct 23, 1999
923
924
925
{
/* If which is -1, play on the first free channel */
if ( which == -1 ) {
Oct 26, 1999
Oct 26, 1999
926
for ( i=reserved_channels; i<num_channels; ++i ) {
Jun 26, 2000
Jun 26, 2000
927
if ( mix_channel[i].playing <= 0 )
Oct 23, 1999
Oct 23, 1999
928
929
break;
}
Oct 26, 1999
Oct 26, 1999
930
if ( i == num_channels ) {
Oct 23, 1999
Oct 23, 1999
931
932
933
934
935
936
937
which = -1;
} else {
which = i;
}
}
/* Queue up the audio data for this channel */
Jan 13, 2011
Jan 13, 2011
938
if ( which >= 0 && which < num_channels ) {
Jan 14, 2002
Jan 14, 2002
939
Uint32 sdl_ticks = SDL_GetTicks();
Jan 14, 2002
Jan 14, 2002
940
if (Mix_Playing(which))
May 3, 2002
May 3, 2002
941
_Mix_channel_done_playing(which);
Jun 26, 2000
Jun 26, 2000
942
943
944
945
946
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;
Aug 27, 2019
Aug 27, 2019
947
948
949
if (mix_channel[which].fading == MIX_NO_FADING) {
mix_channel[which].fade_volume_reset = mix_channel[which].volume;
}
Jun 26, 2000
Jun 26, 2000
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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;
Oct 21, 1999
Oct 21, 1999
956
957
}
}
May 3, 2002
May 3, 2002
958
SDL_UnlockAudio();
Oct 21, 1999
Oct 21, 1999
959
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/* 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;
Jan 13, 2011
Jan 13, 2011
968
int prev_volume = 0;
Oct 21, 1999
Oct 21, 1999
969
970
if ( which == -1 ) {
Oct 26, 1999
Oct 26, 1999
971
for ( i=0; i<num_channels; ++i ) {
Oct 21, 1999
Oct 21, 1999
972
973
prev_volume += Mix_Volume(i, volume);
}
Oct 26, 1999
Oct 26, 1999
974
prev_volume /= num_channels;
Jan 13, 2011
Jan 13, 2011
975
} else if ( which < num_channels ) {
Jun 26, 2000
Jun 26, 2000
976
prev_volume = mix_channel[which].volume;
Mar 24, 2002
Mar 24, 2002
977
978
979
980
981
if ( volume >= 0 ) {
if ( volume > SDL_MIX_MAXVOLUME ) {
volume = SDL_MIX_MAXVOLUME;
}
mix_channel[which].volume = volume;
Oct 23, 1999
Oct 23, 1999
982
}
Oct 21, 1999
Oct 21, 1999
983
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1000
}
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);
}