/* * audio_out_solaris.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_SOLARIS #include #include #include #include #include #include #include #include "a52.h" #include "audio_out.h" #include "audio_out_internal.h" typedef struct solaris_instance_s { ao_instance_t ao; int fd; int sample_rate; int set_params; int flags; } solaris_instance_t; static int solaris_setup (ao_instance_t * _instance, int sample_rate, int * flags, sample_t * level, sample_t * bias) { solaris_instance_t * instance = (solaris_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 solaris_play (ao_instance_t * _instance, int flags, sample_t * _samples) { solaris_instance_t * instance = (solaris_instance_t *) _instance; int16_t int16_samples[256*2]; #ifdef LIBA52_DOUBLE float samples[256 * 2]; int i; for (i = 0; i < 256 * 2; i++) samples[i] = _samples[i]; #else float * samples = _samples; #endif if (instance->set_params) { audio_info_t info; /* Setup our parameters */ AUDIO_INITINFO (&info); info.play.sample_rate = instance->sample_rate; info.play.precision = 16; info.play.channels = 2; info.play.encoding = AUDIO_ENCODING_LINEAR; /* Write our configuration */ /* An implicit GETINFO is also performed. */ if (ioctl (instance->fd, AUDIO_SETINFO, &info) < 0) { perror ("Writing audio config block"); return 1; } if ((info.play.sample_rate != instance->sample_rate) || (info.play.precision != 16) || (info.play.channels != 2)) { fprintf (stderr, "Wanted %dHz %d bits %d channels\n", instance->sample_rate, 16, 2); fprintf (stderr, "Got %dHz %d bits %d channels\n", info.play.sample_rate, info.play.precision, info.play.channels); 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_2 (samples, int16_samples); write (instance->fd, int16_samples, 256 * sizeof (int16_t) * 2); return 0; } static void solaris_close (ao_instance_t * _instance) { solaris_instance_t * instance = (solaris_instance_t *) _instance; close (instance->fd); } static ao_instance_t * solaris_open (int flags) { solaris_instance_t * instance; instance = malloc (sizeof (solaris_instance_t)); if (instance == NULL) return NULL; instance->ao.setup = solaris_setup; instance->ao.play = solaris_play; instance->ao.close = solaris_close; instance->sample_rate = 0; instance->set_params = 1; instance->flags = flags; instance->fd = open ("/dev/audio", O_WRONLY); if (instance->fd < 0) { fprintf (stderr, "Can not open /dev/audio\n"); free (instance); return NULL; } return (ao_instance_t *) instance; } ao_instance_t * ao_solaris_open (void) { return solaris_open (A52_STEREO); } ao_instance_t * ao_solarisdolby_open (void) { return solaris_open (A52_DOLBY); } #endif