sound.c

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/*
 * See Licensing and Copyright notice in naev.h
 */
#include <sys/stat.h>
#include "physfs.h"
#include "SDL.h"
#include "SDL_mutex.h"
#include "SDL_thread.h"
 
#include "naev.h"
#include "physfsrwops.h"
 
#include "sound.h"
 
#include "array.h"
#include "camera.h"
#include "conf.h"
#include "env.h"
#include "log.h"
#include "music.h"
#include "ndata.h"
#include "nstring.h"
#include "physics.h"
#include "player.h"
#include "nopenal.h"
#include "nlua_spfx.h"
 
#define SOUND_FADEOUT         100
#define SOUND_VOICES           64   
#define SOUND_SUFFIX_WAV   ".wav" 
#define SOUND_SUFFIX_OGG   ".ogg" 
#define voiceLock()        SDL_LockMutex(voice_mutex)
#define voiceUnlock()      SDL_UnlockMutex(voice_mutex)
 
typedef struct alSound_ {
   char *filename; 
   char *name; 
   double length; 
   int channels; 
   ALuint buf; 
} alSound;
 
typedef enum voice_state_ {
   VOICE_STOPPED, 
   VOICE_PLAYING, 
   VOICE_FADEOUT, 
   VOICE_DESTROY  
} voice_state_t;
 
typedef struct alVoice_ {
   struct alVoice_ *prev; 
   struct alVoice_ *next; 
   int id; 
   voice_state_t state; 
   unsigned int flags; 
   ALfloat pos[3]; 
   ALfloat vel[3]; 
   ALuint source; 
   ALuint buffer; 
} alVoice;
 
typedef struct alGroup_s {
   int id; 
   /* Sources. */
   ALuint *sources; 
   int nsources; 
   voice_state_t state; 
   int fade_timer; 
   int speed; 
   double volume; 
   double pitch; 
} alGroup_t;
 
/*
 * Global sound properties.
 */
int sound_disabled            = 0; 
static int sound_initialized  = 0; 
/*
 * Sound list.
 */
static alSound *sound_list    = NULL; 
/*
 * Voices.
 */
static int voice_genid        = 0; 
static alVoice *voice_active  = NULL; 
static alVoice *voice_pool    = NULL; 
static SDL_mutex *voice_mutex = NULL; 
/*
 * Internally used sounds.
 */
static int snd_compression    = -1; 
static int snd_compressionG   = -1; 
static double snd_compression_gain = 0.; 
/*
 * prototypes
 */
/* General. */
static int sound_makeList (void);
static void sound_free( alSound *snd );
/* Voices. */
 
/*
 * Global sound lock.
 */
SDL_mutex *sound_lock = NULL; 
/*
 * Global device and context.
 */
static ALCcontext *al_context = NULL; 
static ALCdevice *al_device   = NULL; 
static ALfloat svolume        = 1.; 
static ALfloat svolume_lin    = 1.; 
static ALfloat svolume_speed  = 1.; 
alInfo_t al_info; 
/*
 * struct to hold all the sources and currently attached voice
 */
static ALuint *source_stack   = NULL; 
static ALuint *source_total   = NULL; 
static ALuint *source_all     = NULL; 
static int source_nstack      = 0; 
static int source_ntotal      = 0; 
static int source_nall        = 0; 
static int source_mstack      = 0; 
/*
 * EFX stuff.
 */
ALuint sound_efx_directSlot   = 0; 
static ALuint efx_reverb      = 0; 
static ALuint efx_echo        = 0; 
/*
 * Sound speed.
 */
static double sound_speed     = 1.; 
/*
 * Group management.
 */
static alGroup_t *al_groups = NULL; 
static int al_ngroups       = 0; 
static int al_groupidgen    = 0; 
/*
 * Prototypes.
 */
/*
 * Loading.
 */
static int sound_al_init (void);
static const char* vorbis_getErr( int err );
static int al_enableEFX (void);
/*
 * General.
 */
static int al_playVoice( alVoice *v, alSound *s,
      ALfloat px, ALfloat py, ALfloat vx, ALfloat vy, ALint relative );
static int al_load( alSound *snd, SDL_RWops *rw, const char *name );
static int al_loadWav( ALuint *buf, SDL_RWops *rw );
static int al_loadOgg( ALuint *buf, OggVorbis_File *vf );
/*
 * Pausing.
 */
static void al_pausev( ALint n, ALuint *s );
static void al_resumev( ALint n, ALuint *s );
/*
 * Groups.
 */
static alGroup_t *al_getGroup( int group );
/*
 * Voice management.
 */
static alVoice* voice_new (void);
static int voice_add( alVoice* v );
static alVoice* voice_get( int id );
/*
 * Sound playing.
 */
static void al_updateVoice( alVoice *v );
static void al_volumeUpdate (void);
/*
 * Vorbis stuff.
 */
static size_t ovpack_read( void *ptr, size_t size, size_t nmemb, void *datasource )
{
   SDL_RWops *rw = datasource;
   return (size_t) SDL_RWread( rw, ptr, size, nmemb );
}
static int ovpack_seek( void *datasource, ogg_int64_t offset, int whence )
{
   SDL_RWops *rw = datasource;
   return SDL_RWseek( rw, offset, whence );
}
static int ovpack_close( void *datasource )
{
   SDL_RWops *rw = datasource;
   return SDL_RWclose( rw );
}
static int ovpack_closeFake( void *datasource )
{
   (void) datasource;
   return 0;
}
static long ovpack_tell( void *datasource )
{
   SDL_RWops *rw = datasource;
   return SDL_RWseek( rw, 0, SEEK_CUR );
}
ov_callbacks sound_al_ovcall = {
   .read_func  = ovpack_read,
   .seek_func  = ovpack_seek,
   .close_func = ovpack_close,
   .tell_func  = ovpack_tell
}; 
ov_callbacks sound_al_ovcall_noclose = {
   .read_func  = ovpack_read,
   .seek_func  = ovpack_seek,
   .close_func = ovpack_closeFake,
   .tell_func  = ovpack_tell
}; 
static int sound_al_init (void)
{
   int ret, nattribs = 0;
   ALuint s;
   ALint attribs[10] = { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 };
 
   /* Default values. */
   ret = 0;
 
   /* We'll need a mutex */
   sound_lock = SDL_CreateMutex();
   soundLock();
 
   /* opening the default device */
   al_device = alcOpenDevice(NULL);
   if (al_device == NULL) {
      WARN(_("Unable to open default sound device"));
      ret = -1;
      goto snderr_dev;
   }
 
   /* Set good default for sources. */
   attribs[0] = ALC_MONO_SOURCES;
   attribs[1] = 512;
   attribs[2] = ALC_STEREO_SOURCES;
   attribs[3] = 32;
   nattribs = 4;
 
   /* Query EFX extension. */
   if (conf.al_efx) {
      al_info.efx = alcIsExtensionPresent( al_device, "ALC_EXT_EFX" );
      if (al_info.efx == AL_TRUE) {
         attribs[nattribs+0] = ALC_MAX_AUXILIARY_SENDS;
         attribs[nattribs+1] = 4;
         nattribs += 2;
      }
   }
   else
      al_info.efx = AL_FALSE;
 
   /* Check more extensions. */
   al_info.output_limiter = alcIsExtensionPresent( al_device, "ALC_SOFT_output_limiter" );
   if (al_info.output_limiter) {
      attribs[nattribs+0] = ALC_OUTPUT_LIMITER_SOFT;
      attribs[nattribs+1] = ALC_TRUE;
      nattribs += 2;
   }
 
   /* Create the OpenAL context */
   al_context = alcCreateContext( al_device, attribs );
   if (al_context == NULL) {
      WARN(_("Unable to create OpenAL context"));
      ret = -2;
      goto snderr_ctx;
   }
 
   /* Set active context */
   if (alcMakeContextCurrent( al_context )==AL_FALSE) {
      WARN(_("Failure to set default context"));
      ret = -4;
      goto snderr_act;
   }
 
   /* Clear the errors */
   alGetError();
 
   /* Query some extensions. */
   if (al_info.output_limiter) {
      ALint limiter;
      alcGetIntegerv( al_device, ALC_OUTPUT_LIMITER_SOFT, 1, &limiter );
      if (limiter != ALC_TRUE)
         WARN(_("Failed to set ALC_OUTPUT_LIMITER_SOFT"));
   }
 
   /* Get context information. */
   alcGetIntegerv( al_device, ALC_FREQUENCY, 1, &al_info.freq );
   alcGetIntegerv( al_device, ALC_MONO_SOURCES, 1, &al_info.nmono );
   alcGetIntegerv( al_device, ALC_STEREO_SOURCES, 1, &al_info.nstereo );
 
   /* Try to enable EFX. */
   if (al_info.efx == AL_TRUE) {
      al_enableEFX();
   }
   else {
      al_info.efx_reverb = AL_FALSE;
      al_info.efx_echo   = AL_FALSE;
   }
 
   /* Check for errors. */
   al_checkErr();
 
   /* Start allocating the sources - music has already taken theirs */
   source_nstack  = 0;
   source_mstack  = SOUND_VOICES;
   source_stack   = malloc( sizeof( ALuint ) * source_mstack );
   while (source_nstack < SOUND_VOICES) {
      alGenSources( 1, &s );
      source_stack[source_nstack] = s;
 
      /* How OpenAL distance model works:
       *
       * Clamped:
       *  gain = distance_function( CLAMP( AL_REFERENCE_DISTANCE, AL_MAX_DISTANCE, distance ) );
       *
       * Distance functions:
       *                                       AL_REFERENCE_DISTANCE
       *  * Inverse = ------------------------------------------------------------------------------
       *              AL_REFERENCE_DISTANCE + AL_ROLLOFF_FACTOR ( distance - AL_REFERENCE_DISTANCE )
       *
       *             1 - AL_ROLLOFF_FACTOR ( distance - AL_REFERENCE_DISTANCE )
       *  * Linear = ----------------------------------------------------------
       *                      AL_MAX_DISTANCE - AL_REFERENCE_DISTANCE
       *
       *                  /       distance        \ -AL_ROLLOFF_FACTOR
       *  * Exponential = | --------------------- |
       *                  \ AL_REFERENCE_DISTANCE /
       *
       *
       * Some values:
       *
       *  model    falloff  reference   100     1000    5000   10000
       *  linear     1        500      1.000   0.947   0.526   0.000
       *  inverse    1        500      1.000   0.500   0.100   0.050
       *  exponent   1        500      1.000   0.500   0.100   0.050
       *  inverse   0.5       500      1.000   0.667   0.182   0.095
       *  exponent  0.5       500      1.000   0.707   0.316   0.223
       *  inverse    2        500      1.000   0.333   0.052   0.026
       *  exponent   2        500      1.000   0.250   0.010   0.003
       */
      alSourcef( s, AL_REFERENCE_DISTANCE, SOUND_REFERENCE_DISTANCE ); /* Close distance to clamp at (doesn't get louder). */
      alSourcef( s, AL_MAX_DISTANCE,       SOUND_MAX_DISTANCE ); /* Max distance to clamp at (doesn't get quieter). */
      alSourcef( s, AL_ROLLOFF_FACTOR,     1. ); /* Determines how it drops off. */
 
      /* Set the filter. */
      if (al_info.efx == AL_TRUE)
         alSource3i( s, AL_AUXILIARY_SEND_FILTER, sound_efx_directSlot, 0, AL_FILTER_NULL );
 
      /* Check for error. */
      if (alGetError() == AL_NO_ERROR)
         source_nstack++;
      else
         break;
   }
 
   if ( source_nstack == 0 ) {
      WARN( _( "OpenAL failed to initialize sources" ) );
      source_mstack = 0;
      free( source_stack );
      source_stack = NULL;
   }
   else {
      /* Reduce ram usage. */
      source_mstack = source_nstack;
      source_stack  = realloc( source_stack, sizeof( ALuint ) * source_mstack );
      /* Copy allocated sources to total stack. */
      source_ntotal = source_mstack;
      source_total  = malloc( sizeof( ALuint ) * source_mstack );
      memcpy( source_total, source_stack, sizeof( ALuint ) * source_mstack );
 
      /* Copy allocated sources to all stack. */
      source_nall = source_mstack;
      source_all  = malloc( sizeof( ALuint ) * source_mstack );
      memcpy( source_all, source_stack, sizeof( ALuint ) * source_mstack );
   }
 
   /* Set up how sound works. */
   alDistanceModel( AL_INVERSE_DISTANCE_CLAMPED ); /* Clamping is fundamental so it doesn't sound like crap. */
   sound_env( SOUND_ENV_NORMAL, 0. );
 
   /* Check for errors. */
   al_checkErr();
 
   /* we can unlock now */
   soundUnlock();
 
   /* debug magic */
   DEBUG(_("OpenAL started: %d Hz"), al_info.freq);
   DEBUG(_("Renderer: %s"), alGetString(AL_RENDERER));
   if (al_info.efx)
      DEBUG(_("Version: %s with EFX %d.%d"), alGetString(AL_VERSION),
            al_info.efx_major, al_info.efx_minor);
   else
      DEBUG(_("Version: %s without EFX"), alGetString(AL_VERSION));
   DEBUG_BLANK();
 
   return ret;
 
   /*
    * error handling
    */
snderr_act:
   alcDestroyContext( al_context );
snderr_ctx:
   al_context = NULL;
   alcCloseDevice( al_device );
snderr_dev:
   al_device = NULL;
   soundUnlock();
   SDL_DestroyMutex( sound_lock );
   sound_lock = NULL;
   return ret;
}
 
static int al_enableEFX (void)
{
   /* Issues with ALSOFT 1.19.1 crashes so we work around it.
    * TODO: Disable someday. */
   if (strcmp(alGetString(AL_VERSION), "1.1 ALSOFT 1.19.1")==0) {
      DEBUG(_("Crashing ALSOFT version detected, disabling EFX"));
      al_info.efx = AL_FALSE;
      return -1;
   }
 
   /* Get general information. */
   alcGetIntegerv( al_device, ALC_MAX_AUXILIARY_SENDS, 1, &al_info.efx_auxSends );
   alcGetIntegerv( al_device, ALC_EFX_MAJOR_VERSION, 1, &al_info.efx_major );
   alcGetIntegerv( al_device, ALC_EFX_MINOR_VERSION, 1, &al_info.efx_minor );
 
   /* Get function pointers. */
   nalGenAuxiliaryEffectSlots  = alGetProcAddress( "alGenAuxiliaryEffectSlots" );
   nalDeleteAuxiliaryEffectSlots = alGetProcAddress( "alDeleteAuxiliaryEffectSlots" );
   nalIsAuxiliaryEffectSlot    = alGetProcAddress( "alIsAuxiliaryEffectSlot" );
   nalAuxiliaryEffectSloti     = alGetProcAddress( "alAuxiliaryEffectSloti" );
   nalAuxiliaryEffectSlotiv    = alGetProcAddress( "alAuxiliaryEffectSlotiv" );
   nalAuxiliaryEffectSlotf     = alGetProcAddress( "alAuxiliaryEffectSlotf" );
   nalAuxiliaryEffectSlotfv    = alGetProcAddress( "alAuxiliaryEffectSlotfv" );
   nalGetAuxiliaryEffectSloti  = alGetProcAddress( "alGetAuxiliaryEffectSloti" );
   nalGetAuxiliaryEffectSlotiv = alGetProcAddress( "alGetAuxiliaryEffectSlotiv" );
   nalGetAuxiliaryEffectSlotf  = alGetProcAddress( "alGetAuxiliaryEffectSlotf" );
   nalGetAuxiliaryEffectSlotfv = alGetProcAddress( "alGetAuxiliaryEffectSlotfv" );
   nalGenFilters               = alGetProcAddress( "alGenFilters" );
   nalDeleteFilters            = alGetProcAddress( "alDeleteFilters" );
   nalFilteri                  = alGetProcAddress( "alFilteri" );
   nalFilteriv                 = alGetProcAddress( "alFilteriv" );
   nalFilterf                  = alGetProcAddress( "alFilterf" );
   nalFilterfv                 = alGetProcAddress( "alFilterfv" );
   nalGenEffects               = alGetProcAddress( "alGenEffects" );
   nalDeleteEffects            = alGetProcAddress( "alDeleteEffects" );
   nalEffecti                  = alGetProcAddress( "alEffecti" );
   nalEffectiv                 = alGetProcAddress( "alEffectiv" );
   nalEffectf                  = alGetProcAddress( "alEffectf" );
   nalEffectfv                 = alGetProcAddress( "alEffectfv" );
   if (!nalGenAuxiliaryEffectSlots || !nalDeleteAuxiliaryEffectSlots ||
         !nalIsAuxiliaryEffectSlot ||
         !nalAuxiliaryEffectSloti || !nalAuxiliaryEffectSlotiv ||
         !nalAuxiliaryEffectSlotf || !nalAuxiliaryEffectSlotfv ||
         !nalGetAuxiliaryEffectSloti || !nalGetAuxiliaryEffectSlotiv ||
         !nalGetAuxiliaryEffectSlotf || !nalGetAuxiliaryEffectSlotfv ||
         !nalGenFilters || !nalDeleteFilters ||
         !nalFilteri || !nalFilteriv || !nalFilterf || !nalFilterfv ||
         !nalGenEffects || !nalDeleteEffects ||
         !nalEffecti || !nalEffectiv || !nalEffectf || !nalEffectfv) {
      DEBUG(_("OpenAL EFX functions not found, disabling EFX."));
      al_info.efx = AL_FALSE;
      return -1;
   }
 
   /* Create auxiliary slot. */
   nalGenAuxiliaryEffectSlots( 1, &sound_efx_directSlot );
 
   /* Create reverb effect. */
   nalGenEffects( 1, &efx_reverb );
   nalEffecti( efx_reverb, AL_EFFECT_TYPE, AL_EFFECT_REVERB );
   if (alGetError() != AL_NO_ERROR) {
      DEBUG(_("OpenAL Reverb not found, disabling."));
      al_info.efx_reverb = AL_FALSE;
      nalDeleteEffects( 1, &efx_reverb );
   }
   else {
      al_info.efx_reverb = AL_TRUE;
 
      /* Set Reverb parameters. */
      /*nalEffectf( efx_reverb, AL_REVERB_DECAY_TIME, 15. );*/
   }
 
   /* Create echo effect. */
   nalGenEffects( 1, &efx_echo );
   nalEffecti( efx_echo, AL_EFFECT_TYPE, AL_EFFECT_ECHO );
   if (alGetError() != AL_NO_ERROR) {
      DEBUG(_("OpenAL Echo not found, disabling."));
      al_info.efx_echo = AL_FALSE;
      nalDeleteEffects( 1, &efx_echo );
   }
   else {
      al_info.efx_echo = AL_TRUE;
 
      /* Set Echo parameters. */
      nalEffectf( efx_echo, AL_ECHO_DELAY, 0.207 );
   }
 
   /* Set up the listener. */
   alListenerf( AL_METERS_PER_UNIT, 5. );
 
   /* Check for errors. */
   al_checkErr();
 
   return 0;
}
 
int sound_init (void)
{
   int ret;
 
   /* See if sound is disabled. */
   if (conf.nosound) {
      sound_disabled = 1;
      music_disabled = 1;
   }
 
   /* Parse conf. */
   if (sound_disabled && music_disabled)
      return 0;
 
   /* Initialize sound backend. */
   ret = sound_al_init();
   if (ret != 0) {
      sound_disabled = 1;
      music_disabled = 1;
      WARN(_("Sound disabled."));
      return ret;
   }
 
   /* Create voice lock. */
   voice_mutex = SDL_CreateMutex();
   if (voice_mutex == NULL)
      WARN(_("Unable to create voice mutex."));
 
   /* Load available sounds. */
   ret = sound_makeList();
   if (ret != 0)
      return ret;
 
   /* Set volume. */
   if ((conf.sound > 1.) || (conf.sound < 0.)) {
      WARN(_("Sound has invalid value, clamping to [0:1]."));
      conf.sound = CLAMP( 0., 1., conf.sound );
   }
   sound_volume(conf.sound);
 
   /* Initialized. */
   sound_initialized = 1;
 
   /* Initialize music. */
   ret = music_init();
   if (ret != 0) {
      music_disabled = 1;
      WARN(_("Music disabled."));
   }
 
   /* Load compression noise. */
   snd_compression = sound_get( "compression" );
   if (snd_compression >= 0) {
      snd_compressionG = sound_createGroup( 1 );
      sound_speedGroup( snd_compressionG, 0 );
   }
 
   return 0;
}
 
void sound_exit (void)
{
   /* Nothing to disable. */
   if (sound_disabled || !sound_initialized)
      return;
 
   if (voice_mutex != NULL) {
      voiceLock();
      /* free the voices. */
      while (voice_active != NULL) {
         alVoice *v = voice_active;
         voice_active = v->next;
         free(v);
      }
      while (voice_pool != NULL) {
         alVoice *v = voice_pool;
         voice_pool = v->next;
         free(v);
      }
      voiceUnlock();
 
      /* Destroy voice lock. */
      SDL_DestroyMutex(voice_mutex);
      voice_mutex = NULL;
   }
 
   soundLock();
 
   /* Free groups. */
   for (int i=0; i<al_ngroups; i++) {
      alGroup_t *g = &al_groups[i];
      free(g->sources);
      g->sources  = NULL;
      g->nsources = 0;
   }
   free(al_groups);
   al_groups  = NULL;
   al_ngroups = 0;
 
   /* Free stacks. */
   if (source_all != NULL) {
      alSourceStopv(   source_nall, source_all );
      alDeleteSources( source_nall, source_all );
      free(source_all);
   }
   source_all        = NULL;
   source_nall       = 0;
   free(source_total);
   source_total      = NULL;
   source_ntotal     = 0;
   free(source_stack);
   source_stack      = NULL;
   source_nstack     = 0;
   source_mstack     = 0;
 
   /* free the sounds */
   for (int i=0; i<array_size(sound_list); i++)
      sound_free( &sound_list[i] );
   array_free( sound_list );
 
   /* Clean up EFX stuff. */
   if (al_info.efx == AL_TRUE) {
      nalDeleteAuxiliaryEffectSlots( 1, &sound_efx_directSlot );
      if (al_info.efx_reverb == AL_TRUE)
         nalDeleteEffects( 1, &efx_reverb );
      if (al_info.efx_echo == AL_TRUE)
         nalDeleteEffects( 1, &efx_echo );
   }
 
   /* Clean up global stuff. */
   if (al_context) {
      alcMakeContextCurrent(NULL);
      alcDestroyContext( al_context );
   }
   if (al_device)
      alcCloseDevice( al_device );
 
   soundUnlock();
 
   SDL_DestroyMutex( sound_lock );
 
   /* Sound is done. */
   sound_initialized = 0;
}
 
int sound_get( const char* name )
{
   if (sound_disabled)
      return 0;
 
   for (int i=0; i<array_size(sound_list); i++)
      if (strcmp(name, sound_list[i].name)==0)
         return i;
 
   WARN(_("Sound '%s' not found in sound list"), name);
   return -1;
}
 
double sound_getLength( int sound )
{
   if (sound_disabled)
      return 0.;
 
   return sound_list[sound].length;
}
 
int sound_play( int sound )
{
   alVoice *v;
   alSound *s;
 
   if (sound_disabled)
      return 0;
 
   if ((sound < 0) || (sound >= array_size(sound_list)))
      return -1;
 
   /* Gets a new voice. */
   v = voice_new();
 
   /* Get the sound. */
   s = &sound_list[sound];
 
   /* Try to play the sound. */
   if (al_playVoice( v, s, 0., 0., 0., 0., AL_TRUE ))
      return -1;
 
   /* Set state and add to list. */
   v->state = VOICE_PLAYING;
   v->id = ++voice_genid;
   voice_add(v);
   return v->id;
}
 
int sound_playPos( int sound, double px, double py, double vx, double vy )
{
   alVoice *v;
   alSound *s;
   Pilot *p;
   double cx, cy, dist;
   int target;
 
   if (sound_disabled)
      return 0;
 
   if ((sound < 0) || (sound >= array_size(sound_list)))
      return -1;
 
   target = cam_getTarget();
 
   /* Following a pilot. */
   p = pilot_get(target);
   if (target && (p != NULL)) {
      if (!pilot_inRange( p, px, py ))
         return 0;
   }
   /* Set to a position. */
   else {
      cam_getPos(&cx, &cy);
      dist = pow2(px - cx) + pow2(py - cy);
      if (dist > pilot_sensorRange())
         return 0;
   }
 
   /* Gets a new voice. */
   v = voice_new();
 
   /* Get the sound. */
   s = &sound_list[sound];
 
   /* Try to play the sound. */
   if (al_playVoice( v, s, px, py, vx, vy, AL_FALSE ))
      return -1;
 
   /* Actually add the voice to the list. */
   v->state = VOICE_PLAYING;
   v->id = ++voice_genid;
   voice_add(v);
 
   return v->id;
}
 
int sound_updatePos( int voice, double px, double py, double vx, double vy )
{
   alVoice *v;
 
   if (sound_disabled)
      return 0;
 
   v = voice_get(voice);
   if (v == NULL)
      return 0;
 
   /* Update the voice. */
   v->pos[0] = px;
   v->pos[1] = py;
   v->vel[0] = vx;
   v->vel[1] = vy;
   return 0;
}
 
int sound_update( double dt )
{
   unsigned int t = SDL_GetTicks();
 
   /* Update music if needed. */
   music_update(dt);
 
   if (sound_disabled)
      return 0;
 
   /* System update. */
   for (int i=0; i<al_ngroups; i++) {
      unsigned int f;
      alGroup_t *g = &al_groups[i];
      /* Handle fadeout. */
      if (g->state != VOICE_FADEOUT)
         continue;
 
      /* Calculate fadeout. */
      f = t - g->fade_timer;
      if (f < SOUND_FADEOUT) {
         ALfloat d, v;
         d = 1. - (ALfloat) f / (ALfloat) SOUND_FADEOUT;
         v = d * svolume * g->volume;
         if (g->speed)
            v *= svolume_speed;
         soundLock();
         for (int j=0; j<g->nsources; j++)
            alSourcef( g->sources[j], AL_GAIN, v );
         /* Check for errors. */
         al_checkErr();
         soundUnlock();
      }
      /* Fadeout done. */
      else {
         ALfloat v;
         soundLock();
         v = svolume * g->volume;
         if (g->speed)
            v *= svolume_speed;
         for (int j=0; j<g->nsources; j++) {
            alSourceStop( g->sources[j] );
            alSourcei( g->sources[j], AL_BUFFER, AL_NONE );
            alSourcef( g->sources[j], AL_GAIN, v );
         }
         /* Check for errors. */
         al_checkErr();
         soundUnlock();
 
         /* Mark as done. */
         g->state = VOICE_PLAYING;
      }
   }
 
   if (voice_active == NULL)
      return 0;
 
   voiceLock();
 
   /* The actual control loop. */
   for (alVoice *v=voice_active; v!=NULL; v=v->next) {
 
      /* Run first to clear in same iteration. */
      al_updateVoice( v );
 
      /* Destroy and toss into pool. */
      if ((v->state == VOICE_STOPPED) || (v->state == VOICE_DESTROY)) {
         /* Remove from active list. */
         alVoice *tv = v->prev;
         if (tv == NULL) {
            voice_active = v->next;
            if (voice_active != NULL)
               voice_active->prev = NULL;
         }
         else {
            tv->next = v->next;
            if (tv->next != NULL)
               tv->next->prev = tv;
         }
 
         /* Add to free pool. */
         v->next = voice_pool;
         v->prev = NULL;
         voice_pool = v;
         if (v->next != NULL)
            v->next->prev = v;
 
         /* Avoid loop blockage. */
         v = (tv != NULL) ? tv->next : voice_active;
         if (v == NULL)
            break;
      }
   }
 
   voiceUnlock();
 
   return 0;
}
 
void sound_pause (void)
{
   if (sound_disabled)
      return;
 
   soundLock();
   al_pausev( source_ntotal, source_total );
   al_checkErr();
   soundUnlock();
 
   if (snd_compression >= 0)
      sound_pauseGroup( snd_compressionG );
}
 
void sound_resume (void)
{
   if (sound_disabled)
      return;
 
   soundLock();
   al_resumev( source_ntotal, source_total );
   al_checkErr();
   soundUnlock();
 
   if (snd_compression >= 0)
      sound_resumeGroup( snd_compressionG );
}
 
void sound_stopAll (void)
{
   if (sound_disabled)
      return;
 
   /* Make sure there are voices. */
   if (voice_active==NULL)
      return;
 
   voiceLock();
   for (alVoice *v=voice_active; v!=NULL; v=v->next) {
      if ((v->state == VOICE_STOPPED) || (v->state == VOICE_DESTROY))
         continue;
      if (v->source != 0) {
         /* TODO not sure if we want to move the locks outside of the loop. Worried it might deadlock somewhere. */
         soundLock();
         alSourceStop( v->source );
         al_checkErr();
         soundUnlock();
      }
      v->state = VOICE_STOPPED;
   }
   voiceUnlock();
}
 
void sound_stop( int voice )
{
   alVoice *v;
 
   if (sound_disabled)
      return;
 
   v = voice_get(voice);
   if (v == NULL)
      return;
 
   if ((v->state == VOICE_STOPPED) || (v->state == VOICE_DESTROY))
      return;
 
   if (v->source != 0) {
      soundLock();
      alSourceStop( v->source );
      al_checkErr();
      soundUnlock();
   }
   v->state = VOICE_STOPPED;
}
 
int sound_updateListener( double dir, double px, double py,
      double vx, double vy )
{
   ALfloat ori[6], pos[3], vel[3];
   double c, s;
 
   if (sound_disabled)
      return 0;
 
   c = cos(dir);
   s = sin(dir);
 
   soundLock();
 
   ori[0] = c;
   ori[1] = s;
   ori[2] = 0.;
   ori[3] = 0.;
   ori[4] = 0.;
   ori[5] = 1.;
   alListenerfv( AL_ORIENTATION, ori );
   pos[0] = px;
   pos[1] = py;
   pos[2] = 100.;
   alListenerfv( AL_POSITION, pos );
   vel[0] = vx;
   vel[1] = vy;
   vel[2] = 0.;
   alListenerfv( AL_VELOCITY, vel );
 
   /* Check for errors. */
   al_checkErr();
 
   soundUnlock();
 
   return 0;
}
 
void sound_setSpeed( double s )
{
   double v;
   int playing;
 
   if (sound_disabled)
      return;
 
   /* We implement the brown noise here. */
   playing = (snd_compression_gain > 0.);
   v = CLAMP( 0, 1., (s-2)/10. );
 
   if (v > 0.) {
      if (snd_compression >= 0) {
         if (!playing)
            sound_playGroup( snd_compressionG, snd_compression, 0 ); /* Start playing only if it's not playing. */
         sound_volumeGroup( snd_compressionG, v );
      }
      svolume_speed = 1.-v;
      al_volumeUpdate();
   }
   else if (playing) {
      if (snd_compression >= 0)
         sound_stopGroup( snd_compressionG ); /* Stop compression sound. */
      svolume_speed = 1.;
      al_volumeUpdate();
   }
   snd_compression_gain = v;
 
   soundLock();
   sound_speed = s; /* Set the speed. */
   /* Do all the groupless. */
   for (int i=0; i<source_ntotal; i++)
      alSourcef( source_total[i], AL_PITCH, s );
   /* Do specific groups. */
   for (int i=0; i<al_ngroups; i++) {
      alGroup_t *g = &al_groups[i];
      if (!g->speed)
         continue;
      for (int j=0; j<g->nsources; j++)
         alSourcef( g->sources[j], AL_PITCH, s*g->pitch );
   }
   /* Check for errors. */
   al_checkErr();
   soundUnlock();
}
 
static int sound_makeList (void)
{
   char** files;
   int suflen;
 
   if (sound_disabled)
      return 0;
 
   /* get the file list */
   files = PHYSFS_enumerateFiles( SOUND_PATH );
 
   /* Create the list. */
   sound_list = array_create( alSound );
 
   /* load the profiles */
   suflen = strlen(SOUND_SUFFIX_WAV);
   for (size_t i=0; files[i]!=NULL; i++) {
      int len;
      char path[PATH_MAX];
      SDL_RWops *rw;
      int flen = strlen(files[i]);
 
      /* Must be longer than suffix. */
      if (flen < suflen)
         continue;
 
      /* Make sure is wav or ogg. */
      if ((strncmp( &files[i][flen - suflen], SOUND_SUFFIX_WAV, suflen)!=0) &&
            (strncmp( &files[i][flen - suflen], SOUND_SUFFIX_OGG, suflen)!=0))
         continue;
 
      /* Load the sound. */
      snprintf( path, sizeof(path), SOUND_PATH"%s", files[i] );
      rw = PHYSFSRWOPS_openRead( path );
 
      /* remove the suffix */
      len = flen - suflen;
      files[i][len] = '\0';
 
      source_newRW( rw, files[i], 0 );
      SDL_RWclose( rw );
   }
 
   DEBUG( n_("Loaded %d Sound", "Loaded %d Sounds", array_size(sound_list)), array_size(sound_list) );
 
   /* Clean up. */
   PHYSFS_freeList( files );
 
   return 0;
}
 
int sound_volume( const double vol )
{
   if (sound_disabled)
      return 0;
 
   /* Calculate volume level. */
   svolume_lin = vol;
   if (vol > 0.) /* Floor of -48 dB (0.00390625 amplitude) */
      svolume = (ALfloat) 1. / pow(2., (1. - vol) * 8.);
   else
      svolume  = 0.;
 
   /* Update volume. */
   al_volumeUpdate();
 
   return 0;
}
 
double sound_getVolume (void)
{
   if (sound_disabled)
      return 0.;
 
   return svolume_lin;
}
 
double sound_getVolumeLog (void)
{
   if (sound_disabled)
      return 0.;
 
   return svolume;
}
 
static void sound_free( alSound *snd )
{
   /* Free general stuff. */
   free(snd->name);
   free(snd->filename);
 
   /* Free internals. */
   soundLock();
 
   alDeleteBuffers( 1, &snd->buf );
   al_checkErr();
 
   soundUnlock();
}
 
int sound_createGroup( int size )
{
   alGroup_t *g;
   int id;
 
   if (sound_disabled)
      return 0;
 
   /* Get new ID. */
   id  = ++al_groupidgen;
 
   /* Grow group list. */
   al_ngroups++;
   al_groups = realloc( al_groups, sizeof(alGroup_t) * al_ngroups );
   g        = &al_groups[ al_ngroups-1 ];
   memset( g, 0, sizeof(alGroup_t) );
   g->id    = id;
   g->state = VOICE_PLAYING;
   g->speed = 1;
   g->volume = 1.;
   g->pitch = 1.;
 
   /* Allocate sources. */
   g->sources  = calloc( size, sizeof(ALuint) );
   g->nsources = size;
 
   /* Add some sources. */
   for (int i=0; i<size; i++) {
      /* Make sure there's enough. */
      if (source_nstack <= 0)
         goto group_err;
 
      /* Pull one off the stack. */
      source_nstack--;
      g->sources[i] = source_stack[source_nstack];
 
      /* Disable EFX, they don't affect groups. */
      if (al_info.efx_reverb == AL_TRUE) {
         alSourcef(  g->sources[i], AL_AIR_ABSORPTION_FACTOR, 0. );
         alSource3i( g->sources[i], AL_AUXILIARY_SEND_FILTER,
               AL_EFFECTSLOT_NULL, 0, AL_FILTER_NULL );
      }
 
      /* Remove from total too. */
      for (int j=0; j<source_ntotal; j++) {
         if (g->sources[i] == source_total[j]) {
            source_ntotal--;
            memmove( &source_total[j], &source_total[j+1],
                  sizeof(ALuint) * (source_ntotal - j) );
            break;
         }
      }
   }
 
   al_checkErr();
   return id;
 
group_err:
   free(g->sources);
   al_ngroups--;
   al_checkErr();
   return 0;
}
 
int sound_playGroup( int group, int sound, int once )
{
   alSound *s;
 
   if (sound_disabled)
      return 0;
 
   if ((sound < 0) || (sound >= array_size(sound_list)))
      return -1;
 
   s = &sound_list[sound];
   for (int i=0; i<al_ngroups; i++) {
      alGroup_t *g;
 
      /* Find group. */
      if (al_groups[i].id != group)
         continue;
 
      g = &al_groups[i];
      g->state = VOICE_PLAYING;
      soundLock();
      for (int j=0; j<g->nsources; j++) {
         double v;
         ALint state;
         alGetSourcei( g->sources[j], AL_SOURCE_STATE, &state );
 
         /* No free ones, just smash the last one. */
         if (j == g->nsources-1) {
            if (state != AL_STOPPED)
               alSourceStop( g->sources[j] );
         }
         /* Ignore playing/paused. */
         else if ((state == AL_PLAYING) || (state == AL_PAUSED))
            continue;
 
         /* Attach buffer. */
         alSourcei( g->sources[j], AL_BUFFER, s->buf );
 
         /* Do not do positional sound. */
         alSourcei( g->sources[j], AL_SOURCE_RELATIVE, AL_TRUE );
 
         /* See if should loop. */
         alSourcei( g->sources[j], AL_LOOPING, (once) ? AL_FALSE : AL_TRUE );
 
         /* Set volume. */
         v = svolume * g->volume;
         if (g->speed)
            v *= svolume_speed;
         alSourcef( g->sources[j], AL_GAIN, v );
 
         /* Start playing. */
         alSourcePlay( g->sources[j] );
 
         /* Check for errors. */
         al_checkErr();
 
         soundUnlock();
         return 0;
      }
      al_checkErr();
      soundUnlock();
 
      /* Group matched but no free source found.. */
      WARN(_("Group '%d' has no free sounds."), group );
      return -1;
   }
 
   WARN(_("Group '%d' not found."), group);
   return -1;
}
 
void sound_stopGroup( int group )
{
   alGroup_t *g;
 
   if (sound_disabled)
      return;
 
   g = al_getGroup( group );
   if (g == NULL)
      return;
 
   g->state      = VOICE_FADEOUT;
   g->fade_timer = SDL_GetTicks();
}
 
void sound_pauseGroup( int group )
{
   alGroup_t *g;
 
   if (sound_disabled)
      return;
 
   g = al_getGroup( group );
   if (g == NULL)
      return;
 
   soundLock();
   al_pausev( g->nsources, g->sources );
   al_checkErr();
   soundUnlock();
}
 
void sound_resumeGroup( int group )
{
   alGroup_t *g;
 
   if (sound_disabled)
      return;
 
   g = al_getGroup( group );
   if (g == NULL)
      return;
 
   soundLock();
   al_resumev( g->nsources, g->sources );
   al_checkErr();
   soundUnlock();
}
 
static void groupSpeedReset( alGroup_t *g )
{
   for (int i=0; i<g->nsources; i++) {
      if (g->speed)
         alSourcef( g->sources[i], AL_PITCH, sound_speed*g->pitch );
      else
         alSourcef( g->sources[i], AL_PITCH, 1. );
   }
}
void sound_speedGroup( int group, int enable )
{
   alGroup_t *g;
 
   if (sound_disabled)
      return;
 
   g = al_getGroup( group );
   if (g == NULL)
      return;
 
   soundLock();
   g->speed = enable;
   groupSpeedReset(g);
   al_checkErr();
   soundUnlock();
}
 
void sound_volumeGroup( int group, double volume )
{
   double v;
   alGroup_t *g;
 
   if (sound_disabled)
      return;
 
   g = al_getGroup( group );
   if (g == NULL)
      return;
 
   g->volume = volume;
 
   soundLock();
   v = svolume * g->volume;
   if (g->speed)
      v *= svolume_speed;
   for (int j=0; j<g->nsources; j++)
      alSourcef( g->sources[j], AL_GAIN, v );
   al_checkErr();
   soundUnlock();
}
 
void sound_pitchGroup( int group, double pitch )
{
   alGroup_t *g;
 
   if (sound_disabled)
      return;
 
   g = al_getGroup( group );
   if (g == NULL)
      return;
 
   soundLock();
   g->pitch = pitch;
   groupSpeedReset(g);
   al_checkErr();
   soundUnlock();
}
 
void sound_setAbsorption( double value )
{
   if (sound_disabled)
      return;
 
   soundLock();
   for (int i=0; i<source_ntotal; i++) {
      ALuint s = source_total[i];
      /* Value is from 0. (normal) to 10..
      * It represents the attenuation per meter. In this case it decreases by
      * 0.05*AB_FACTOR dB/meter where AB_FACTOR is the air absorption factor.
      * In our case each pixel represents 5 meters.
      */
      alSourcef( s, AL_AIR_ABSORPTION_FACTOR, value );
   }
   al_checkErr();
   soundUnlock();
}
 
int sound_env( SoundEnv_t env_type, double param )
{
   ALfloat f;
 
   if (sound_disabled)
      return 0;
 
   soundLock();
   switch (env_type) {
      case SOUND_ENV_NORMAL:
         /* Set global parameters. */
         alSpeedOfSound( 3433. );
         alDopplerFactor( 0.3 );
 
         if (al_info.efx == AL_TRUE) {
            /* Disconnect the effect. */
            nalAuxiliaryEffectSloti( sound_efx_directSlot,
                  AL_EFFECTSLOT_EFFECT, AL_EFFECT_NULL );
 
            /* Set per-source parameters. */
            sound_setAbsorption( 0. );
         }
         break;
 
      case SOUND_ENV_NEBULA:
         f = param / 1000.;
 
         /* Set global parameters. */
         alSpeedOfSound( 3433./(1. + f*2.) );
         alDopplerFactor( 1.0 );
 
         if (al_info.efx == AL_TRUE) {
            if (al_info.efx_reverb == AL_TRUE) {
               /* Tweak the reverb. */
               nalEffectf( efx_reverb, AL_REVERB_DECAY_TIME,    10. );
               nalEffectf( efx_reverb, AL_REVERB_DECAY_HFRATIO, 0.5 );
 
               /* Connect the effect. */
               nalAuxiliaryEffectSloti( sound_efx_directSlot,
                     AL_EFFECTSLOT_EFFECT, efx_reverb );
            }
 
            /* Set per-source parameters. */
            sound_setAbsorption( 3.*f );
         }
         break;
   }
 
   /* Check for errors. */
   al_checkErr();
 
   soundUnlock();
 
   return 0;
}
 
alVoice* voice_new (void)
{
   alVoice *v;
 
   /* No free voices, allocate a new one. */
   if (voice_pool == NULL) {
      v = calloc( 1, sizeof(alVoice) );
      voice_pool = v;
      return v;
   }
 
   /* First free voice. */
   v = voice_pool; /* We do not touch the next nor prev, it's still in the pool. */
   return v;
}
 
int voice_add( alVoice* v )
{
   alVoice *tv;
 
   /* Remove from pool. */
   if (v->prev != NULL) {
      tv = v->prev;
      tv->next = v->next;
      if (tv->next != NULL)
         tv->next->prev = tv;
   }
   else { /* Set pool to be the next. */
      voice_pool = v->next;
      if (voice_pool != NULL)
         voice_pool->prev = NULL;
   }
 
   /* Insert to the front of active voices. */
   voiceLock();
   tv = voice_active;
   v->next = tv;
   v->prev = NULL;
   voice_active = v;
   if (tv != NULL)
      tv->prev = v;
   voiceUnlock();
   return 0;
}
 
alVoice* voice_get( int id )
{
   alVoice *v;
   /* Make sure there are voices. */
   if (voice_active==NULL)
      return NULL;
 
   voiceLock();
   for (v=voice_active; v!=NULL; v=v->next)
      if (v->id == id)
         break;
   voiceUnlock();
 
   return v;
}
 
int source_newRW( SDL_RWops *rw, const char *name, unsigned int flags )
{
   int ret;
   alSound snd, *sndl;
   (void) flags;
 
   if (sound_disabled)
      return -1;
 
   memset( &snd, 0, sizeof(alSound) );
   ret = al_load( &snd, rw, name );
   if (ret)
      return -1;
 
   sndl = &array_grow( &sound_list );
   memcpy( sndl, &snd, sizeof(alSound) );
   sndl->name = strdup( name );
 
   return sndl-sound_list;
}
 
int source_new( const char* filename, unsigned int flags )
{
   SDL_RWops *rw = PHYSFSRWOPS_openRead( filename );
   int id = source_newRW( rw, filename, flags );
   SDL_RWclose( rw );
   return id;
}
 
static void al_pausev( ALint n, ALuint *s )
{
   for (int i=0; i<n; i++) {
      ALint state;
      alGetSourcei( s[i], AL_SOURCE_STATE, &state );
      if (state == AL_PLAYING)
         alSourcePause( s[i] );
   }
}
 
static void al_resumev( ALint n, ALuint *s )
{
   for (int i=0; i<n; i++) {
      ALint state;
      alGetSourcei( s[i], AL_SOURCE_STATE, &state );
      if (state == AL_PAUSED)
         alSourcePlay( s[i] );
   }
}
 
static alGroup_t *al_getGroup( int group )
{
   for (int i=0; i<al_ngroups; i++) {
      if (al_groups[i].id != group)
         continue;
      return &al_groups[i];
   }
   WARN(_("Group '%d' not found."), group);
   return NULL;
}
 
static int al_loadWav( ALuint *buf, SDL_RWops *rw )
{
   SDL_AudioSpec wav_spec;
   Uint32 wav_length;
   Uint8 *wav_buffer;
   ALenum format;
 
   SDL_RWseek( rw, 0, SEEK_SET );
 
   /* Load WAV. */
   if (SDL_LoadWAV_RW( rw, 0, &wav_spec, &wav_buffer, &wav_length) == NULL) {
      WARN(_("SDL_LoadWav_RW failed: %s"), SDL_GetError());
      return -1;
   }
 
   /* Handle format. */
   switch (wav_spec.format) {
      case AUDIO_U8:
      case AUDIO_S8:
         format = (wav_spec.channels==1) ? AL_FORMAT_MONO8 : AL_FORMAT_STEREO8;
         break;
      case AUDIO_U16LSB:
      case AUDIO_S16LSB:
         format = (wav_spec.channels==1) ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
         break;
      case AUDIO_U16MSB:
      case AUDIO_S16MSB:
         WARN( _("Big endian WAVs unsupported!") );
         return -1;
      default:
         WARN( _("Invalid WAV format!") );
         return -1;
   }
 
   /* Load into openal. */
   soundLock();
   /* Create new buffer. */
   alGenBuffers( 1, buf );
   /* Put into the buffer. */
   alBufferData( *buf, format, wav_buffer, wav_length, wav_spec.freq );
   al_checkErr();
   soundUnlock();
 
   /* Clean up. */
   SDL_FreeWAV( wav_buffer );
   return 0;
}
 
static const char* vorbis_getErr( int err )
{
   switch (err) {
      case OV_EREAD:       return _("A read from media returned an error.");
      case OV_EFAULT:      return _("Internal logic fault; indicates a bug or heap/stack corruption.");
      case OV_EIMPL:       return _("Feature not implemented.");
      case OV_EINVAL:      return _("Either an invalid argument, or incompletely initialized argument passed to libvorbisfile call");
      case OV_ENOTVORBIS:  return _("Bitstream is not Vorbis data.");
      case OV_EBADHEADER:  return _("Invalid Vorbis bitstream header.");
      case OV_EVERSION:    return _("Vorbis version mismatch.");
      case OV_EBADLINK:    return _("The given link exists in the Vorbis data stream, but is not decipherable due to garbage or corruption.");
      case OV_ENOSEEK:     return _("The given stream is not seekable.");
 
      default: return _("Unknown vorbisfile error.");
   }
}
 
static int al_loadOgg( ALuint *buf, OggVorbis_File *vf )
{
   int ret;
   long i;
   int section;
   vorbis_info *info;
   ALenum format;
   ogg_int64_t len;
   char *data;
   long bytes_read;
 
   /* Finish opening the file. */
   ret = ov_test_open(vf);
   if (ret) {
      WARN(_("Failed to finish loading Ogg file: %s"), vorbis_getErr(ret) );
      return -1;
   }
 
   /* Get file information. */
   info   = ov_info( vf, -1 );
   format = (info->channels == 1) ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
   len    = ov_pcm_total( vf, -1 ) * info->channels * sizeof(short);
 
   /* Allocate memory. */
   data = malloc( len );
 
   /* Fill buffer. */
   i = 0;
   bytes_read = 1;
   while (bytes_read > 0) {
      /* Fill buffer with data ibytes_read the 16 bit signed samples format. */
      bytes_read = ov_read( vf, &data[i], 4096, (SDL_BYTEORDER == SDL_BIG_ENDIAN), 2, 1, &section );
      if (bytes_read==OV_HOLE || bytes_read==OV_EBADLINK || bytes_read==OV_EINVAL) {
         WARN(_("Error reading from OGG file!"));
         continue;
      }
      i += bytes_read;
   }
 
   soundLock();
   /* Create new buffer. */
   alGenBuffers( 1, buf );
   /* Put into buffer. */
   alBufferData( *buf, format, data, len, info->rate );
   al_checkErr();
   soundUnlock();
 
   /* Clean up. */
   free(data);
   ov_clear(vf);
 
   return 0;
}
 
int sound_al_buffer( ALuint *buf, SDL_RWops *rw, const char *name )
{
   int ret;
   OggVorbis_File vf;
 
   /* Check to see if it's an Ogg. */
   if (ov_test_callbacks( rw, &vf, NULL, 0, sound_al_ovcall_noclose )==0)
      ret = al_loadOgg( buf, &vf );
 
   /* Otherwise try WAV. */
   else {
      /* Destroy the partially loaded vorbisfile. */
      ov_clear(&vf);
 
      /* Try to load Wav. */
      ret = al_loadWav( buf, rw );
   }
 
   /* Failed to load. */
   if (ret != 0) {
      WARN(_("Failed to load sound file '%s'."), name);
      return ret;
   }
 
   soundLock();
 
   /* Check for errors. */
   al_checkErr();
 
   soundUnlock();
 
   return 0;
}
 
int al_load( alSound *snd, SDL_RWops *rw, const char *name )
{
   ALint freq, bits, channels, size;
   int ret = sound_al_buffer( &snd->buf, rw, name );
   if (ret != 0) {
      WARN(_("Failed to load sound file '%s'."), name);
      return ret;
   }
 
   soundLock();
 
   /* Get the length of the sound. */
   alGetBufferi( snd->buf, AL_FREQUENCY, &freq );
   alGetBufferi( snd->buf, AL_BITS, &bits );
   alGetBufferi( snd->buf, AL_CHANNELS, &channels );
   alGetBufferi( snd->buf, AL_SIZE, &size );
   if ((freq==0) || (bits==0) || (channels==0)) {
      WARN(_("Something went wrong when loading sound file '%s'."), name);
      snd->length = 0;
   }
   else
      snd->length = (double)size / (double)(freq * (bits/8) * channels);
   snd->channels = channels;
 
   /* Check for errors. */
   al_checkErr();
 
   soundUnlock();
 
   return 0;
}
 
static void al_volumeUpdate (void)
{
   soundLock();
   /* Do generic ones. */
   for (int i=0; i<source_ntotal; i++)
      alSourcef( source_total[i], AL_GAIN, svolume*svolume_speed );
   /* Do specific groups. */
   for (int i=0; i<al_ngroups; i++) {
      alGroup_t *g = &al_groups[i];
      double v = svolume * g->volume;
      if (g->speed)
         v *= svolume_speed;
      for (int j=0; j<g->nsources; j++)
         alSourcef( g->sources[j], AL_GAIN, v );
   }
   al_checkErr();
   soundUnlock();
 
   /* Do special effects. */
   spfxL_setSpeedVolume( svolume * svolume_speed );
}
 
static int al_playVoice( alVoice *v, alSound *s,
      ALfloat px, ALfloat py, ALfloat vx, ALfloat vy, ALint relative )
{
   ALuint source;
 
   /* Make sure there's enough. */
   if (source_nstack <= 0)
      return 0;
 
   /* Pull one off the stack. */
   source_nstack--;
   source = source_stack[source_nstack];
 
   /* Set up the source and buffer. */
   v->source = source;
 
   if (v->source == 0)
      return -1;
   v->buffer = s->buf;
 
   soundLock();
 
   /* Attach buffer. */
   alSourcei( v->source, AL_BUFFER, v->buffer );
 
   /* Enable positional sound. */
   alSourcei( v->source, AL_SOURCE_RELATIVE, relative );
 
#if DEBUGGING
   if ((relative==AL_FALSE) && (s->channels>1))
      WARN(_("Sound '%s' has %d channels but is being played as positional. It should be mono!"), s->name, s->channels );
#endif /* DEBUGGING */
 
   /* Update position. */
   v->pos[0] = px;
   v->pos[1] = py;
   v->pos[2] = 0.;
   v->vel[0] = vx;
   v->vel[1] = vy;
   v->vel[2] = 0.;
 
   /* Set up properties. */
   alSourcef(  v->source, AL_GAIN, svolume*svolume_speed );
   alSourcefv( v->source, AL_POSITION, v->pos );
   alSourcefv( v->source, AL_VELOCITY, v->vel );
 
   /* Defaults just in case. */
   alSourcei( v->source, AL_LOOPING, AL_FALSE );
 
   /* Start playing. */
   alSourcePlay( v->source );
 
   /* Check for errors. */
   al_checkErr();
 
   soundUnlock();
 
   return 0;
}
 
int sound_al_updatePos( alVoice *v,
            double px, double py, double vx, double vy )
{
   v->pos[0] = px;
   v->pos[1] = py;
   v->vel[0] = vx;
   v->vel[1] = vy;
   return 0;
}
 
void al_updateVoice( alVoice *v )
{
   ALint state;
 
   /* Invalid source, mark to delete. */
   if (v->source == 0) {
      v->state = VOICE_DESTROY;
      return;
   }
 
   soundLock();
 
   /* Get status. */
   alGetSourcei( v->source, AL_SOURCE_STATE, &state );
   if (state == AL_STOPPED) {
 
      /* Remove buffer so it doesn't start up again if resume is called. */
      alSourcei( v->source, AL_BUFFER, AL_NONE );
 
      /* Check for errors. */
      al_checkErr();
 
      soundUnlock();
 
      /* Put source back on the list. */
      source_stack[source_nstack] = v->source;
      source_nstack++;
      v->source = 0;
 
      /* Mark as stopped - erased next iteration. */
      v->state = VOICE_STOPPED;
      return;
   }
 
   /* Set up properties. */
   alSourcef(  v->source, AL_GAIN, svolume*svolume_speed );
   alSourcefv( v->source, AL_POSITION, v->pos );
   alSourcefv( v->source, AL_VELOCITY, v->vel );
 
   /* Check for errors. */
   al_checkErr();
 
   soundUnlock();
}