forked from KolibriOS/kolibrios
83d5b68cdf
- Fixed speed. - Fixed lags. - Returned --res parameter git-svn-id: svn://kolibrios.org@9097 a494cfbc-eb01-0410-851d-a64ba20cac60
1497 lines
35 KiB
C
1497 lines
35 KiB
C
/*
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SDL_mixer: An audio mixer library based on the SDL library
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Copyright (C) 1997-2012 Sam Lantinga <slouken@libsdl.org>
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This software is provided 'as-is', without any express or implied
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warranty. In no event will the authors be held liable for any damages
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arising from the use of this software.
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Permission is granted to anyone to use this software for any purpose,
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including commercial applications, and to alter it and redistribute it
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freely, subject to the following restrictions:
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1. The origin of this software must not be misrepresented; you must not
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claim that you wrote the original software. If you use this software
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in a product, an acknowledgment in the product documentation would be
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appreciated but is not required.
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2. Altered source versions must be plainly marked as such, and must not be
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misrepresented as being the original software.
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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>
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#include <stdlib.h>
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#include <string.h>
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#include "SDL_mutex.h"
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#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"
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#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"
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#include "dynamic_mod.h"
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#include "dynamic_mp3.h"
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#include "dynamic_ogg.h"
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#define __MIX_INTERNAL_EFFECT__
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#include "effects_internal.h"
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#ifdef _KOLIBRI
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void uSDL_Delay(unsigned int time);
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unsigned uSDL_GetTicks();
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#else
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#define uSDL_Delay SDL_Delay
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#define uSDL_GetTicks SDL_GetTicks
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#endif
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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" */
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#define FORM 0x4d524f46 /* "FORM" */
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#define OGGS 0x5367674f /* "OggS" */
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#define CREA 0x61657243 /* "Crea" */
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#define FLAC 0x43614C66 /* "fLaC" */
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static int audio_opened = 0;
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static SDL_AudioSpec mixer;
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typedef struct _Mix_effectinfo
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{
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Mix_EffectFunc_t callback;
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Mix_EffectDone_t done_callback;
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void *udata;
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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;
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int playing;
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int paused;
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Uint8 *samples;
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int volume;
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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;
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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 (*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 (*channel_done_callback)(int channel) = NULL;
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/* Music function declarations */
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extern int open_music(SDL_AudioSpec *mixer);
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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);
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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. */
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static const char **chunk_decoders = NULL;
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static int num_decoders = 0;
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/* Semicolon-separated SoundFont paths */
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#ifdef MID_MUSIC
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extern char* soundfont_paths;
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#endif
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int Mix_GetNumChunkDecoders(void)
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{
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return(num_decoders);
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}
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const char *Mix_GetChunkDecoder(int index)
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{
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if ((index < 0) || (index >= num_decoders)) {
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return NULL;
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}
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return(chunk_decoders[index]);
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}
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static void add_chunk_decoder(const char *decoder)
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{
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void *ptr = SDL_realloc(chunk_decoders, (num_decoders + 1) * sizeof (const char **));
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if (ptr == NULL) {
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return; /* oh well, go on without it. */
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}
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chunk_decoders = (const char **) ptr;
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chunk_decoders[num_decoders++] = decoder;
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}
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/* rcg06192001 get linked library's version. */
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const SDL_version *Mix_Linked_Version(void)
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{
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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;
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int Mix_Init(int flags)
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{
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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) {
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result |= MIX_INIT_FLUIDSYNTH;
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}
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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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}
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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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}
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#else
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Mix_SetError("Mixer not built with FLAC support");
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#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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}
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#else
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Mix_SetError("Mixer not built with MOD support");
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#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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}
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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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}
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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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}
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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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}
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initialized |= result;
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return (result);
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}
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void Mix_Quit()
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{
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#ifdef USE_FLUIDSYNTH_MIDI
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if (initialized & MIX_INIT_FLUIDSYNTH) {
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Mix_QuitFluidSynth();
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}
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#endif
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#ifdef FLAC_MUSIC
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if (initialized & MIX_INIT_FLAC) {
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Mix_QuitFLAC();
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}
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#endif
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#ifdef MOD_MUSIC
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if (initialized & MIX_INIT_MOD) {
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Mix_QuitMOD();
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}
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#endif
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#ifdef MP3_MUSIC
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if (initialized & MIX_INIT_MP3) {
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Mix_QuitMP3();
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}
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#endif
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#ifdef OGG_MUSIC
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if (initialized & MIX_INIT_OGG) {
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Mix_QuitOgg();
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}
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#endif
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#ifdef MID_MUSIC
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if (soundfont_paths) {
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SDL_free(soundfont_paths);
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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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/*
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* rcg06122001 Cleanup effect callbacks.
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* MAKE SURE SDL_LockAudio() is called before this (or you're in the
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* audio callback).
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*/
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static void _Mix_channel_done_playing(int channel)
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{
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if (channel_done_callback) {
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channel_done_callback(channel);
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}
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/*
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* Call internal function directly, to avoid locking audio from
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* inside audio callback.
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*/
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_Mix_remove_all_effects(channel, &mix_channel[channel].effects);
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}
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static void *Mix_DoEffects(int chan, void *snd, int len)
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{
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int posteffect = (chan == MIX_CHANNEL_POST);
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effect_info *e = ((posteffect) ? posteffects : mix_channel[chan].effects);
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void *buf = snd;
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if (e != NULL) { /* are there any registered effects? */
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/* if this is the postmix, we can just overwrite the original. */
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if (!posteffect) {
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buf = SDL_malloc(len);
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if (buf == NULL) {
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return(snd);
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}
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memcpy(buf, snd, len);
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}
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for (; e != NULL; e = e->next) {
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if (e->callback != NULL) {
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e->callback(chan, buf, len, e->udata);
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}
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}
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}
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/* be sure to SDL_free() the return value if != snd ... */
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return(buf);
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}
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/* Mixing function */
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static void 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)
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/* 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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}
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/* Mix any playing channels... */
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sdl_ticks = uSDL_GetTicks();
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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 ) {
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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;
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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 ) {
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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 */
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if( mix_channel[i].fading == MIX_FADING_OUT ) {
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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].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 ) {
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Mix_Volume(i, (mix_channel[i].fade_volume * (mix_channel[i].fade_length-ticks))
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/ 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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}
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if ( mix_channel[i].playing > 0 ) {
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int index = 0;
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int remaining = len;
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while (mix_channel[i].playing > 0 && index < len) {
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remaining = len - index;
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volume = (mix_channel[i].volume*mix_channel[i].chunk->volume) / MIX_MAX_VOLUME;
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mixable = mix_channel[i].playing;
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if ( mixable > remaining ) {
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mixable = remaining;
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}
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mix_input = Mix_DoEffects(i, mix_channel[i].samples, mixable);
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SDL_MixAudio(stream+index,mix_input,mixable,volume);
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if (mix_input != mix_channel[i].samples)
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SDL_free(mix_input);
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mix_channel[i].samples += mixable;
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mix_channel[i].playing -= mixable;
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index += mixable;
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/* rcg06072001 Alert app if channel is done playing. */
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if (!mix_channel[i].playing && !mix_channel[i].looping) {
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_Mix_channel_done_playing(i);
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}
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}
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/* If looping the sample and we are at its end, make sure
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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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--mix_channel[i].looping;
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mix_channel[i].samples = mix_channel[i].chunk->abuf + remaining;
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mix_channel[i].playing = mix_channel[i].chunk->alen - remaining;
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index += remaining;
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}
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if ( ! mix_channel[i].playing && mix_channel[i].looping ) {
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--mix_channel[i].looping;
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mix_channel[i].samples = mix_channel[i].chunk->abuf;
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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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}
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/* rcg06122001 run posteffects... */
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Mix_DoEffects(MIX_CHANNEL_POST, stream, len);
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if ( mix_postmix ) {
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mix_postmix(mix_postmix_data, stream, len);
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}
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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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{
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printf("%s: %d bit %s audio (%s) at %u Hz\n", title, (fmt->format&0xFF),
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(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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}
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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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{
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int i;
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SDL_AudioSpec desired;
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/* If the mixer is already opened, increment open count */
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if ( audio_opened ) {
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if ( format == mixer.format && nchannels == mixer.channels ) {
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++audio_opened;
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return(0);
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}
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while ( audio_opened ) {
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Mix_CloseAudio();
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}
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}
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/* Set the desired format and frequency */
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desired.freq = frequency;
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desired.format = format;
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desired.channels = nchannels;
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desired.samples = chunksize;
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desired.callback = mix_channels;
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desired.userdata = NULL;
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/* Accept nearly any audio format */
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if ( SDL_OpenAudio(&desired, &mixer) < 0 ) {
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return(-1);
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}
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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 */
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if ( open_music(&mixer) < 0 ) {
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SDL_CloseAudio();
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return(-1);
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}
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num_channels = MIX_CHANNELS;
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mix_channel = (struct _Mix_Channel *) SDL_malloc(num_channels * sizeof(struct _Mix_Channel));
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/* Clear out the audio channels */
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for ( i=0; i<num_channels; ++i ) {
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mix_channel[i].chunk = NULL;
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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].volume = SDL_MIX_MAXVOLUME;
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mix_channel[i].fade_volume = SDL_MIX_MAXVOLUME;
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mix_channel[i].fade_volume_reset = SDL_MIX_MAXVOLUME;
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mix_channel[i].fading = MIX_NO_FADING;
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mix_channel[i].tag = -1;
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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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}
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Mix_VolumeMusic(SDL_MIX_MAXVOLUME);
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_Mix_InitEffects();
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/* This list is (currently) decided at build time. */
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add_chunk_decoder("WAVE");
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add_chunk_decoder("AIFF");
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add_chunk_decoder("VOC");
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#ifdef OGG_MUSIC
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add_chunk_decoder("OGG");
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#endif
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#ifdef FLAC_MUSIC
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add_chunk_decoder("FLAC");
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#endif
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audio_opened = 1;
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SDL_PauseAudio(0);
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return(0);
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}
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|
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/* Dynamically change the number of channels managed by the mixer.
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If decreasing the number of channels, the upper channels are
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stopped.
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*/
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int Mix_AllocateChannels(int numchans)
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{
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if ( numchans<0 || numchans==num_channels )
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return(num_channels);
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if ( numchans < num_channels ) {
|
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/* Stop the affected channels */
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int i;
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for(i=numchans; i < num_channels; i++) {
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Mix_UnregisterAllEffects(i);
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Mix_HaltChannel(i);
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}
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}
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SDL_LockAudio();
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mix_channel = (struct _Mix_Channel *) SDL_realloc(mix_channel, numchans * sizeof(struct _Mix_Channel));
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if ( numchans > num_channels ) {
|
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/* Initialize the new channels */
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int i;
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for(i=num_channels; i < numchans; i++) {
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mix_channel[i].chunk = NULL;
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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;
|
|
SDL_UnlockAudio();
|
|
return(num_channels);
|
|
}
|
|
|
|
/* 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;
|
|
}
|
|
if ( channels ) {
|
|
*channels = mixer.channels;
|
|
}
|
|
}
|
|
return(audio_opened);
|
|
}
|
|
|
|
|
|
/*
|
|
* !!! FIXME: Ideally, we want a Mix_LoadSample_RW(), which will handle the
|
|
* generic setup, then call the correct file format loader.
|
|
*/
|
|
|
|
/* Load a wave file */
|
|
Mix_Chunk *Mix_LoadWAV_RW(SDL_RWops *src, int freesrc)
|
|
{
|
|
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");
|
|
if ( freesrc && src ) {
|
|
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), 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;
|
|
#ifdef OGG_MUSIC
|
|
case OGGS:
|
|
loaded = Mix_LoadOGG_RW(src, freesrc, &wavespec,
|
|
(Uint8 **)&chunk->abuf, &chunk->alen);
|
|
break;
|
|
#endif
|
|
#ifdef FLAC_MUSIC
|
|
case FLAC:
|
|
loaded = Mix_LoadFLAC_RW(src, freesrc, &wavespec,
|
|
(Uint8 **)&chunk->abuf, &chunk->alen);
|
|
break;
|
|
#endif
|
|
case CREA:
|
|
loaded = Mix_LoadVOC_RW(src, freesrc, &wavespec,
|
|
(Uint8 **)&chunk->abuf, &chunk->alen);
|
|
break;
|
|
default:
|
|
SDL_SetError("Unrecognized sound file type");
|
|
return(0);
|
|
}
|
|
if ( !loaded ) {
|
|
SDL_free(chunk);
|
|
if ( freesrc ) {
|
|
SDL_RWclose(src);
|
|
}
|
|
return(NULL);
|
|
}
|
|
|
|
#if 0
|
|
PrintFormat("Audio device", &mixer);
|
|
PrintFormat("-- Wave file", &wavespec);
|
|
#endif
|
|
|
|
/* 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);
|
|
}
|
|
memcpy(wavecvt.buf, chunk->abuf, chunk->alen);
|
|
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);
|
|
}
|
|
|
|
/* 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 *)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 {
|
|
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);
|
|
}
|
|
|
|
/* 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 *)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);
|
|
}
|
|
|
|
/* 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 */
|
|
SDL_LockAudio();
|
|
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;
|
|
}
|
|
}
|
|
}
|
|
SDL_UnlockAudio();
|
|
/* Actually free the chunk */
|
|
if ( chunk->allocated ) {
|
|
SDL_free(chunk->abuf);
|
|
}
|
|
SDL_free(chunk);
|
|
}
|
|
}
|
|
|
|
/* 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();
|
|
}
|
|
|
|
/* 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);
|
|
}
|
|
|
|
void Mix_ChannelFinished(void (*channel_finished)(int channel))
|
|
{
|
|
SDL_LockAudio();
|
|
channel_done_callback = channel_finished;
|
|
SDL_UnlockAudio();
|
|
}
|
|
|
|
|
|
/* 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)
|
|
{
|
|
if (num > num_channels)
|
|
num = num_channels;
|
|
reserved_channels = num;
|
|
return num;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
/* Play an audio chunk on a specific channel.
|
|
If the specified channel is -1, play on the first free channel.
|
|
'ticks' is the number of milliseconds at most to play the sample, or -1
|
|
if there is no limit.
|
|
Returns which channel was used to play the sound.
|
|
*/
|
|
int Mix_PlayChannelTimed(int which, Mix_Chunk *chunk, int loops, int ticks)
|
|
{
|
|
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 */
|
|
SDL_LockAudio();
|
|
{
|
|
/* 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 = uSDL_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;
|
|
}
|
|
}
|
|
SDL_UnlockAudio();
|
|
|
|
/* Return the channel on which the sound is being played */
|
|
return(which);
|
|
}
|
|
|
|
/* 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 ) {
|
|
SDL_LockAudio();
|
|
mix_channel[which].expire = (ticks>0) ? ( uSDL_GetTicks() + ticks) : 0;
|
|
SDL_UnlockAudio();
|
|
++ status;
|
|
}
|
|
return(status);
|
|
}
|
|
|
|
/* Fade in a sound on a channel, over ms milliseconds */
|
|
int Mix_FadeInChannelTimed(int which, Mix_Chunk *chunk, int loops, int ms, int ticks)
|
|
{
|
|
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 */
|
|
SDL_LockAudio();
|
|
{
|
|
/* 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 = uSDL_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;
|
|
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;
|
|
}
|
|
}
|
|
SDL_UnlockAudio();
|
|
|
|
/* 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 = 0;
|
|
|
|
if ( which == -1 ) {
|
|
for ( i=0; i<num_channels; ++i ) {
|
|
prev_volume += Mix_Volume(i, volume);
|
|
}
|
|
prev_volume /= num_channels;
|
|
} else if ( which < num_channels ) {
|
|
prev_volume = mix_channel[which].volume;
|
|
if ( volume >= 0 ) {
|
|
if ( volume > SDL_MIX_MAXVOLUME ) {
|
|
volume = SDL_MIX_MAXVOLUME;
|
|
}
|
|
mix_channel[which].volume = volume;
|
|
}
|
|
}
|
|
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 ) {
|
|
for ( i=0; i<num_channels; ++i ) {
|
|
Mix_HaltChannel(i);
|
|
}
|
|
} else if ( which < num_channels ) {
|
|
SDL_LockAudio();
|
|
if (mix_channel[which].playing) {
|
|
_Mix_channel_done_playing(which);
|
|
mix_channel[which].playing = 0;
|
|
mix_channel[which].looping = 0;
|
|
}
|
|
mix_channel[which].expire = 0;
|
|
if(mix_channel[which].fading != MIX_NO_FADING) /* Restore volume */
|
|
mix_channel[which].volume = mix_channel[which].fade_volume_reset;
|
|
mix_channel[which].fading = MIX_NO_FADING;
|
|
SDL_UnlockAudio();
|
|
}
|
|
return(0);
|
|
}
|
|
|
|
/* Halt playing of a particular group of channels */
|
|
int Mix_HaltGroup(int tag)
|
|
{
|
|
int i;
|
|
|
|
for ( i=0; i<num_channels; ++i ) {
|
|
if( mix_channel[i].tag == tag ) {
|
|
Mix_HaltChannel(i);
|
|
}
|
|
}
|
|
return(0);
|
|
}
|
|
|
|
/* Fade out a channel and then stop it automatically */
|
|
int Mix_FadeOutChannel(int which, int ms)
|
|
{
|
|
int status;
|
|
|
|
status = 0;
|
|
if ( audio_opened ) {
|
|
if ( which == -1 ) {
|
|
int i;
|
|
|
|
for ( i=0; i<num_channels; ++i ) {
|
|
status += Mix_FadeOutChannel(i, ms);
|
|
}
|
|
} else if ( which < num_channels ) {
|
|
SDL_LockAudio();
|
|
if ( mix_channel[which].playing &&
|
|
(mix_channel[which].volume > 0) &&
|
|
(mix_channel[which].fading != MIX_FADING_OUT) ) {
|
|
mix_channel[which].fade_volume = mix_channel[which].volume;
|
|
mix_channel[which].fading = MIX_FADING_OUT;
|
|
mix_channel[which].fade_length = ms;
|
|
mix_channel[which].ticks_fade = uSDL_GetTicks();
|
|
|
|
/* only change fade_volume_reset if we're not fading. */
|
|
if (mix_channel[which].fading == MIX_NO_FADING) {
|
|
mix_channel[which].fade_volume_reset = mix_channel[which].volume;
|
|
}
|
|
++status;
|
|
}
|
|
SDL_UnlockAudio();
|
|
}
|
|
}
|
|
return(status);
|
|
}
|
|
|
|
/* 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 ) {
|
|
if( mix_channel[i].tag == tag ) {
|
|
status += Mix_FadeOutChannel(i,ms);
|
|
}
|
|
}
|
|
return(status);
|
|
}
|
|
|
|
Mix_Fading Mix_FadingChannel(int which)
|
|
{
|
|
if ( which < 0 || which >= num_channels ) {
|
|
return MIX_NO_FADING;
|
|
}
|
|
return mix_channel[which].fading;
|
|
}
|
|
|
|
/* Check the status of a specific channel.
|
|
If the specified mix_channel is -1, check all mix channels.
|
|
*/
|
|
int Mix_Playing(int which)
|
|
{
|
|
int status;
|
|
|
|
status = 0;
|
|
if ( which == -1 ) {
|
|
int i;
|
|
|
|
for ( i=0; i<num_channels; ++i ) {
|
|
if ((mix_channel[i].playing > 0) ||
|
|
(mix_channel[i].looping > 0))
|
|
{
|
|
++status;
|
|
}
|
|
}
|
|
} else if ( which < num_channels ) {
|
|
if ( (mix_channel[which].playing > 0) ||
|
|
(mix_channel[which].looping > 0) )
|
|
{
|
|
++status;
|
|
}
|
|
}
|
|
return(status);
|
|
}
|
|
|
|
/* 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);
|
|
}
|
|
|
|
/* Close the mixer, halting all playing audio */
|
|
void Mix_CloseAudio(void)
|
|
{
|
|
int i;
|
|
|
|
if ( audio_opened ) {
|
|
if ( audio_opened == 1 ) {
|
|
for (i = 0; i < num_channels; i++) {
|
|
Mix_UnregisterAllEffects(i);
|
|
}
|
|
Mix_UnregisterAllEffects(MIX_CHANNEL_POST);
|
|
close_music();
|
|
Mix_HaltChannel(-1);
|
|
_Mix_DeinitEffects();
|
|
SDL_CloseAudio();
|
|
SDL_free(mix_channel);
|
|
mix_channel = NULL;
|
|
|
|
/* rcg06042009 report available decoders at runtime. */
|
|
SDL_free(chunk_decoders);
|
|
chunk_decoders = NULL;
|
|
num_decoders = 0;
|
|
}
|
|
--audio_opened;
|
|
}
|
|
}
|
|
|
|
/* Pause a particular channel (or all) */
|
|
void Mix_Pause(int which)
|
|
{
|
|
Uint32 sdl_ticks = uSDL_GetTicks();
|
|
if ( which == -1 ) {
|
|
int i;
|
|
|
|
for ( i=0; i<num_channels; ++i ) {
|
|
if ( mix_channel[i].playing > 0 ) {
|
|
mix_channel[i].paused = sdl_ticks;
|
|
}
|
|
}
|
|
} else if ( which < num_channels ) {
|
|
if ( mix_channel[which].playing > 0 ) {
|
|
mix_channel[which].paused = sdl_ticks;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Resume a paused channel */
|
|
void Mix_Resume(int which)
|
|
{
|
|
Uint32 sdl_ticks = uSDL_GetTicks();
|
|
|
|
SDL_LockAudio();
|
|
if ( which == -1 ) {
|
|
int i;
|
|
|
|
for ( i=0; i<num_channels; ++i ) {
|
|
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;
|
|
}
|
|
}
|
|
} else if ( which < num_channels ) {
|
|
if ( mix_channel[which].playing > 0 ) {
|
|
if(mix_channel[which].expire > 0)
|
|
mix_channel[which].expire += sdl_ticks - mix_channel[which].paused;
|
|
mix_channel[which].paused = 0;
|
|
}
|
|
}
|
|
SDL_UnlockAudio();
|
|
}
|
|
|
|
int Mix_Paused(int which)
|
|
{
|
|
if ( which < 0 ) {
|
|
int status = 0;
|
|
int i;
|
|
for( i=0; i < num_channels; ++i ) {
|
|
if ( mix_channel[i].paused ) {
|
|
++ status;
|
|
}
|
|
}
|
|
return(status);
|
|
} else if ( which < num_channels ) {
|
|
return(mix_channel[which].paused != 0);
|
|
} else {
|
|
return(0);
|
|
}
|
|
}
|
|
|
|
/* Change the group of a channel */
|
|
int Mix_GroupChannel(int which, int tag)
|
|
{
|
|
if ( which < 0 || which > num_channels )
|
|
return(0);
|
|
|
|
SDL_LockAudio();
|
|
mix_channel[which].tag = tag;
|
|
SDL_UnlockAudio();
|
|
return(1);
|
|
}
|
|
|
|
/* Assign several consecutive channels to a group */
|
|
int Mix_GroupChannels(int from, int to, int tag)
|
|
{
|
|
int status = 0;
|
|
for( ; from <= to; ++ from ) {
|
|
status += Mix_GroupChannel(from, tag);
|
|
}
|
|
return(status);
|
|
}
|
|
|
|
/* Finds the first available channel in a group of channels */
|
|
int Mix_GroupAvailable(int tag)
|
|
{
|
|
int i;
|
|
for( i=0; i < num_channels; i ++ ) {
|
|
if ( ((tag == -1) || (tag == mix_channel[i].tag)) &&
|
|
(mix_channel[i].playing <= 0) )
|
|
return i;
|
|
}
|
|
return(-1);
|
|
}
|
|
|
|
int Mix_GroupCount(int tag)
|
|
{
|
|
int count = 0;
|
|
int i;
|
|
for( i=0; i < num_channels; i ++ ) {
|
|
if ( mix_channel[i].tag==tag || tag==-1 )
|
|
++ count;
|
|
}
|
|
return(count);
|
|
}
|
|
|
|
/* Finds the "oldest" sample playing in a group of channels */
|
|
int Mix_GroupOldest(int tag)
|
|
{
|
|
int chan = -1;
|
|
Uint32 mintime = uSDL_GetTicks();
|
|
int i;
|
|
for( i=0; i < num_channels; i ++ ) {
|
|
if ( (mix_channel[i].tag==tag || tag==-1) && mix_channel[i].playing > 0
|
|
&& mix_channel[i].start_time <= mintime ) {
|
|
mintime = mix_channel[i].start_time;
|
|
chan = i;
|
|
}
|
|
}
|
|
return(chan);
|
|
}
|
|
|
|
/* Finds the "most recent" (i.e. last) sample playing in a group of channels */
|
|
int Mix_GroupNewer(int tag)
|
|
{
|
|
int chan = -1;
|
|
Uint32 maxtime = 0;
|
|
int i;
|
|
for( i=0; i < num_channels; i ++ ) {
|
|
if ( (mix_channel[i].tag==tag || tag==-1) && mix_channel[i].playing > 0
|
|
&& mix_channel[i].start_time >= maxtime ) {
|
|
maxtime = mix_channel[i].start_time;
|
|
chan = i;
|
|
}
|
|
}
|
|
return(chan);
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
* rcg06122001 The special effects exportable API.
|
|
* Please see effect_*.c for internally-implemented effects, such
|
|
* as Mix_SetPanning().
|
|
*/
|
|
|
|
/* MAKE SURE you hold the audio lock (SDL_LockAudio()) before calling this! */
|
|
static int _Mix_register_effect(effect_info **e, Mix_EffectFunc_t f,
|
|
Mix_EffectDone_t d, void *arg)
|
|
{
|
|
effect_info *new_e;
|
|
|
|
if (!e) {
|
|
Mix_SetError("Internal error");
|
|
return(0);
|
|
}
|
|
|
|
if (f == NULL) {
|
|
Mix_SetError("NULL effect callback");
|
|
return(0);
|
|
}
|
|
|
|
new_e = SDL_malloc(sizeof (effect_info));
|
|
if (new_e == NULL) {
|
|
Mix_SetError("Out of memory");
|
|
return(0);
|
|
}
|
|
|
|
new_e->callback = f;
|
|
new_e->done_callback = d;
|
|
new_e->udata = arg;
|
|
new_e->next = NULL;
|
|
|
|
/* add new effect to end of linked list... */
|
|
if (*e == NULL) {
|
|
*e = new_e;
|
|
} else {
|
|
effect_info *cur = *e;
|
|
while (1) {
|
|
if (cur->next == NULL) {
|
|
cur->next = new_e;
|
|
break;
|
|
}
|
|
cur = cur->next;
|
|
}
|
|
}
|
|
|
|
return(1);
|
|
}
|
|
|
|
|
|
/* MAKE SURE you hold the audio lock (SDL_LockAudio()) before calling this! */
|
|
static int _Mix_remove_effect(int channel, effect_info **e, Mix_EffectFunc_t f)
|
|
{
|
|
effect_info *cur;
|
|
effect_info *prev = NULL;
|
|
effect_info *next = NULL;
|
|
|
|
if (!e) {
|
|
Mix_SetError("Internal error");
|
|
return(0);
|
|
}
|
|
|
|
for (cur = *e; cur != NULL; cur = cur->next) {
|
|
if (cur->callback == f) {
|
|
next = cur->next;
|
|
if (cur->done_callback != NULL) {
|
|
cur->done_callback(channel, cur->udata);
|
|
}
|
|
SDL_free(cur);
|
|
|
|
if (prev == NULL) { /* removing first item of list? */
|
|
*e = next;
|
|
} else {
|
|
prev->next = next;
|
|
}
|
|
return(1);
|
|
}
|
|
prev = cur;
|
|
}
|
|
|
|
Mix_SetError("No such effect registered");
|
|
return(0);
|
|
}
|
|
|
|
|
|
/* MAKE SURE you hold the audio lock (SDL_LockAudio()) before calling this! */
|
|
static int _Mix_remove_all_effects(int channel, effect_info **e)
|
|
{
|
|
effect_info *cur;
|
|
effect_info *next;
|
|
|
|
if (!e) {
|
|
Mix_SetError("Internal error");
|
|
return(0);
|
|
}
|
|
|
|
for (cur = *e; cur != NULL; cur = next) {
|
|
next = cur->next;
|
|
if (cur->done_callback != NULL) {
|
|
cur->done_callback(channel, cur->udata);
|
|
}
|
|
SDL_free(cur);
|
|
}
|
|
*e = NULL;
|
|
|
|
return(1);
|
|
}
|
|
|
|
|
|
/* MAKE SURE you hold the audio lock (SDL_LockAudio()) before calling this! */
|
|
int _Mix_RegisterEffect_locked(int channel, Mix_EffectFunc_t f,
|
|
Mix_EffectDone_t d, void *arg)
|
|
{
|
|
effect_info **e = NULL;
|
|
|
|
if (channel == MIX_CHANNEL_POST) {
|
|
e = &posteffects;
|
|
} else {
|
|
if ((channel < 0) || (channel >= num_channels)) {
|
|
Mix_SetError("Invalid channel number");
|
|
return(0);
|
|
}
|
|
e = &mix_channel[channel].effects;
|
|
}
|
|
|
|
return _Mix_register_effect(e, f, d, arg);
|
|
}
|
|
|
|
int Mix_RegisterEffect(int channel, Mix_EffectFunc_t f,
|
|
Mix_EffectDone_t d, void *arg)
|
|
{
|
|
int retval;
|
|
SDL_LockAudio();
|
|
retval = _Mix_RegisterEffect_locked(channel, f, d, arg);
|
|
SDL_UnlockAudio();
|
|
return retval;
|
|
}
|
|
|
|
|
|
/* MAKE SURE you hold the audio lock (SDL_LockAudio()) before calling this! */
|
|
int _Mix_UnregisterEffect_locked(int channel, Mix_EffectFunc_t f)
|
|
{
|
|
effect_info **e = NULL;
|
|
|
|
if (channel == MIX_CHANNEL_POST) {
|
|
e = &posteffects;
|
|
} else {
|
|
if ((channel < 0) || (channel >= num_channels)) {
|
|
Mix_SetError("Invalid channel number");
|
|
return(0);
|
|
}
|
|
e = &mix_channel[channel].effects;
|
|
}
|
|
|
|
return _Mix_remove_effect(channel, e, f);
|
|
}
|
|
|
|
int Mix_UnregisterEffect(int channel, Mix_EffectFunc_t f)
|
|
{
|
|
int retval;
|
|
SDL_LockAudio();
|
|
retval = _Mix_UnregisterEffect_locked(channel, f);
|
|
SDL_UnlockAudio();
|
|
return(retval);
|
|
}
|
|
|
|
/* MAKE SURE you hold the audio lock (SDL_LockAudio()) before calling this! */
|
|
int _Mix_UnregisterAllEffects_locked(int channel)
|
|
{
|
|
effect_info **e = NULL;
|
|
|
|
if (channel == MIX_CHANNEL_POST) {
|
|
e = &posteffects;
|
|
} else {
|
|
if ((channel < 0) || (channel >= num_channels)) {
|
|
Mix_SetError("Invalid channel number");
|
|
return(0);
|
|
}
|
|
e = &mix_channel[channel].effects;
|
|
}
|
|
|
|
return _Mix_remove_all_effects(channel, e);
|
|
}
|
|
|
|
int Mix_UnregisterAllEffects(int channel)
|
|
{
|
|
int retval;
|
|
SDL_LockAudio();
|
|
retval = _Mix_UnregisterAllEffects_locked(channel);
|
|
SDL_UnlockAudio();
|
|
return(retval);
|
|
}
|
|
|
|
/* end of mixer.c ... */
|
|
|