1 /*
2 * audio_out_oss.c
3 * Copyright (C) 2000-2002 Michel Lespinasse <walken@zoy.org>
4 * Copyright (C) 1999-2000 Aaron Holtzman <aholtzma@ess.engr.uvic.ca>
5 *
6 * This file is part of a52dec, a free ATSC A-52 stream decoder.
7 * See http://liba52.sourceforge.net/ for updates.
8 *
9 * a52dec is free software; you can redistribute it and/or modify
10 * it under the terms of the GNU General Public License as published by
11 * the Free Software Foundation; either version 2 of the License, or
12 * (at your option) any later version.
13 *
14 * a52dec is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 * GNU General Public License for more details.
18 *
19 * You should have received a copy of the GNU General Public License
20 * along with this program; if not, write to the Free Software
21 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
22 */
23
24 #include "config.h"
25
26 #ifdef LIBAO_OSS
27
28 #include <stdio.h>
29 #include <stdlib.h>
30 #include <sys/ioctl.h>
31 #include <unistd.h>
32 #include <fcntl.h>
33 #include <inttypes.h>
34
35 #if defined(__OpenBSD__)
36 #include <soundcard.h>
37 #elif defined(__FreeBSD__)
38 #include <machine/soundcard.h>
39 #ifndef AFMT_S16_NE
40 #include <machine/endian.h>
41 #if BYTE_ORDER == LITTLE_ENDIAN
42 #define AFMT_S16_NE AFMT_S16_LE
43 #else
44 #define AFMT_S16_NE AFMT_S16_BE
45 #endif
46 #endif
47 #else
48 #include <sys/soundcard.h>
49 #endif
50
51 #include "a52.h"
52 #include "audio_out.h"
53 #include "audio_out_internal.h"
54
55 typedef struct oss_instance_s {
56 ao_instance_t ao;
57 int fd;
58 int sample_rate;
59 int set_params;
60 int flags;
61 } oss_instance_t;
62
oss_setup(ao_instance_t * _instance,int sample_rate,int * flags,sample_t * level,sample_t * bias)63 static int oss_setup (ao_instance_t * _instance, int sample_rate, int * flags,
64 sample_t * level, sample_t * bias)
65 {
66 oss_instance_t * instance = (oss_instance_t *) _instance;
67
68 if ((instance->set_params == 0) && (instance->sample_rate != sample_rate))
69 return 1;
70 instance->sample_rate = sample_rate;
71
72 *flags = instance->flags;
73 *level = 1;
74 *bias = 384;
75
76 return 0;
77 }
78
oss_play(ao_instance_t * _instance,int flags,sample_t * _samples)79 static int oss_play (ao_instance_t * _instance, int flags, sample_t * _samples)
80 {
81 oss_instance_t * instance = (oss_instance_t *) _instance;
82 int16_t int16_samples[256*6];
83 int chans = -1;
84
85 #ifdef LIBA52_DOUBLE
86 float samples[256 * 6];
87 int i;
88
89 for (i = 0; i < 256 * 6; i++)
90 samples[i] = _samples[i];
91 #else
92 float * samples = _samples;
93 #endif
94
95 chans = channels_multi (flags);
96 flags &= A52_CHANNEL_MASK | A52_LFE;
97
98 if (instance->set_params) {
99 int tmp;
100
101 tmp = chans;
102 if ((ioctl (instance->fd, SNDCTL_DSP_CHANNELS, &tmp) < 0) ||
103 (tmp != chans)) {
104 fprintf (stderr, "Can not set number of channels\n");
105 return 1;
106 }
107
108 tmp = instance->sample_rate;
109 if ((ioctl (instance->fd, SNDCTL_DSP_SPEED, &tmp) < 0) ||
110 (tmp != instance->sample_rate)) {
111 fprintf (stderr, "Can not set sample rate\n");
112 return 1;
113 }
114
115 instance->flags = flags;
116 instance->set_params = 0;
117 } else if ((flags == A52_DOLBY) && (instance->flags == A52_STEREO)) {
118 fprintf (stderr, "Switching from stereo to dolby surround\n");
119 instance->flags = A52_DOLBY;
120 } else if ((flags == A52_STEREO) && (instance->flags == A52_DOLBY)) {
121 fprintf (stderr, "Switching from dolby surround to stereo\n");
122 instance->flags = A52_STEREO;
123 } else if (flags != instance->flags)
124 return 1;
125
126 float2s16_multi (samples, int16_samples, flags);
127 write (instance->fd, int16_samples, 256 * sizeof (int16_t) * chans);
128
129 return 0;
130 }
131
oss_close(ao_instance_t * _instance)132 static void oss_close (ao_instance_t * _instance)
133 {
134 oss_instance_t * instance = (oss_instance_t *) _instance;
135
136 close (instance->fd);
137 }
138
oss_open(int flags)139 static ao_instance_t * oss_open (int flags)
140 {
141 oss_instance_t * instance;
142 int format;
143
144 instance = malloc (sizeof (oss_instance_t));
145 if (instance == NULL)
146 return NULL;
147
148 instance->ao.setup = oss_setup;
149 instance->ao.play = oss_play;
150 instance->ao.close = oss_close;
151
152 instance->sample_rate = 0;
153 instance->set_params = 1;
154 instance->flags = flags;
155
156 instance->fd = open ("/dev/dsp", O_WRONLY);
157 if (instance->fd < 0) {
158 fprintf (stderr, "Can not open /dev/dsp\n");
159 free (instance);
160 return NULL;
161 }
162
163 format = AFMT_S16_NE;
164 if ((ioctl (instance->fd, SNDCTL_DSP_SETFMT, &format) < 0) ||
165 (format != AFMT_S16_NE)) {
166 fprintf (stderr, "Can not set sample format\n");
167 free (instance);
168 return NULL;
169 }
170
171 return (ao_instance_t *) instance;
172 }
173
ao_oss_open(void)174 ao_instance_t * ao_oss_open (void)
175 {
176 return oss_open (A52_STEREO);
177 }
178
ao_ossdolby_open(void)179 ao_instance_t * ao_ossdolby_open (void)
180 {
181 return oss_open (A52_DOLBY);
182 }
183
ao_oss4_open(void)184 ao_instance_t * ao_oss4_open (void)
185 {
186 return oss_open (A52_2F2R);
187 }
188
ao_oss6_open(void)189 ao_instance_t * ao_oss6_open (void)
190 {
191 return oss_open (A52_3F2R | A52_LFE);
192 }
193
194 #endif
195