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/*
* audio_out_al.c
* Copyright (C) 2000-2003 Michel Lespinasse <walken@zoy.org>
* Copyright (C) 1999-2000 Aaron Holtzman <aholtzma@ess.engr.uvic.ca>
*
* This file is part of a52dec, a free ATSC A-52 stream decoder.
* See http://liba52.sourceforge.net/ for updates.
*
* a52dec is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* a52dec is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "config.h"
#ifdef LIBAO_AL
#include <stdio.h>
#include <stdlib.h>
#include <dmedia/audio.h>
#include <inttypes.h>
#include "dts.h"
#include "audio_out.h"
#include "audio_out_internal.h"
typedef struct al_instance_s {
ao_instance_t ao;
ALport port;
int sample_rate;
int set_params;
int flags;
} al_instance_t;
static int al_setup (ao_instance_t * _instance, int sample_rate, int * flags,
level_t * level, sample_t * bias)
{
al_instance_t * instance = (al_instance_t *) _instance;
if ((instance->set_params == 0) && (instance->sample_rate != sample_rate))
return 1;
instance->sample_rate = sample_rate;
*flags = instance->flags;
*level = CONVERT_LEVEL;
*bias = CONVERT_BIAS;
return 0;
}
static int al_play (ao_instance_t * _instance, int flags, sample_t * _samples)
{
al_instance_t * instance = (al_instance_t *) _instance;
int16_t int16_samples[256*6];
int chans = -1;
#ifdef LIBDTS_DOUBLE
convert_t samples[256 * 6];
int i;
for (i = 0; i < 256 * 6; i++)
samples[i] = _samples[i];
#else
convert_t * samples = _samples;
#endif
chans = channels_multi (flags);
flags &= DTS_CHANNEL_MASK | DTS_LFE;
if (instance->set_params) {
ALconfig config;
ALpv params[2];
config = alNewConfig ();
if (!config) {
fprintf (stderr, "alNewConfig failed\n");
return 1;
}
if (alSetChannels (config, chans)) {
fprintf (stderr, "alSetChannels failed\n");
return 1;
}
if (alSetConfig (instance->port, config)) {
fprintf (stderr, "alSetConfig failed\n");
return 1;
}
alFreeConfig (config);
params[0].param = AL_MASTER_CLOCK;
params[0].value.i = AL_CRYSTAL_MCLK_TYPE;
params[1].param = AL_RATE;
params[1].value.ll = alIntToFixed (instance->sample_rate);
if (alSetParams (alGetResource (instance->port), params, 2) < 0) {
fprintf (stderr, "alSetParams failed\n");
return 1;
}
instance->flags = flags;
instance->set_params = 0;
} else if ((flags == DTS_DOLBY) && (instance->flags == DTS_STEREO)) {
fprintf (stderr, "Switching from stereo to dolby surround\n");
instance->flags = DTS_DOLBY;
} else if ((flags == DTS_STEREO) && (instance->flags == DTS_DOLBY)) {
fprintf (stderr, "Switching from dolby surround to stereo\n");
instance->flags = DTS_STEREO;
} else if (flags != instance->flags)
return 1;
convert2s16_multi (samples, int16_samples, flags);
alWriteFrames (instance->port, int16_samples, 256);
return 0;
}
static void al_close (ao_instance_t * _instance)
{
al_instance_t * instance = (al_instance_t *) _instance;
alClosePort (instance->port);
}
static ao_instance_t * al_open (int flags)
{
al_instance_t * instance;
int format;
instance = malloc (sizeof (al_instance_t));
if (instance == NULL)
return NULL;
instance->ao.setup = al_setup;
instance->ao.play = al_play;
instance->ao.close = al_close;
instance->sample_rate = 0;
instance->set_params = 1;
instance->flags = flags;
instance->port = alOpenPort ("a52dec", "w", 0);
if (instance->port < 0) {
fprintf (stderr, "alOpenPort failed\n");
free (instance);
return NULL;
}
return (ao_instance_t *) instance;
}
ao_instance_t * ao_al_open (void)
{
return al_open (DTS_STEREO);
}
ao_instance_t * ao_aldolby_open (void)
{
return al_open (DTS_DOLBY);
}
ao_instance_t * ao_al4_open (void)
{
return al_open (DTS_2F2R);
}
ao_instance_t * ao_al6_open (void)
{
return al_open (DTS_3F2R | DTS_LFE);
}
#endif
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