/* * audio_out_al.c * Copyright (C) 2000-2002 Michel Lespinasse * Copyright (C) 1999-2000 Aaron Holtzman * * 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 #include #include #include #include "a52.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, sample_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 = 1; *bias = 384; 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 LIBA52_DOUBLE float samples[256 * 6]; int i; for (i = 0; i < 256 * 6; i++) samples[i] = _samples[i]; #else float * samples = _samples; #endif chans = channels_multi (flags); flags &= A52_CHANNEL_MASK | A52_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 == A52_DOLBY) && (instance->flags == A52_STEREO)) { fprintf (stderr, "Switching from stereo to dolby surround\n"); instance->flags = A52_DOLBY; } else if ((flags == A52_STEREO) && (instance->flags == A52_DOLBY)) { fprintf (stderr, "Switching from dolby surround to stereo\n"); instance->flags = A52_STEREO; } else if (flags != instance->flags) return 1; float2s16_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 (A52_STEREO); } ao_instance_t * ao_aldolby_open (void) { return al_open (A52_DOLBY); } ao_instance_t * ao_al4_open (void) { return al_open (A52_2F2R); } ao_instance_t * ao_al6_open (void) { return al_open (A52_3F2R | A52_LFE); } #endif