factorized All In Wonder code (not tested) - Added AIW auto detect (not tested) ...
[libav.git] / libav / audio.c
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1/*
2 * Linux audio play and grab interface
19720f15 3 * Copyright (c) 2000, 2001 Fabrice Bellard.
de6d9b64 4 *
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5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Lesser General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
de6d9b64 9 *
19720f15 10 * This library is distributed in the hope that it will be useful,
de6d9b64 11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Lesser General Public License for more details.
de6d9b64 14 *
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15 * You should have received a copy of the GNU Lesser General Public
16 * License along with this library; if not, write to the Free Software
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
de6d9b64 18 */
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19#include "avformat.h"
20
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21#include <stdlib.h>
22#include <stdio.h>
23#include <string.h>
b13788c5 24#include <sys/soundcard.h>
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25#include <unistd.h>
26#include <fcntl.h>
27#include <sys/ioctl.h>
28#include <sys/mman.h>
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29#include <sys/time.h>
30
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31const char *audio_device = "/dev/dsp";
32
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33#define AUDIO_BLOCK_SIZE 4096
34
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35typedef struct {
36 int fd;
4972b26f 37 int sample_rate;
de6d9b64 38 int channels;
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39 int frame_size; /* in bytes ! */
40 int codec_id;
1de1cce2 41 int flip_left : 1;
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42 UINT8 buffer[AUDIO_BLOCK_SIZE];
43 int buffer_ptr;
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44} AudioData;
45
4972b26f 46static int audio_open(AudioData *s, int is_output)
de6d9b64 47{
4972b26f 48 int audio_fd;
de6d9b64 49 int tmp, err;
1de1cce2 50 char *flip = getenv("AUDIO_FLIP_LEFT");
de6d9b64 51
de6d9b64 52 /* open linux audio device */
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53 if (is_output)
54 audio_fd = open(audio_device, O_WRONLY);
de6d9b64 55 else
4972b26f 56 audio_fd = open(audio_device, O_RDONLY);
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57 if (audio_fd < 0) {
58 perror(audio_device);
59 return -EIO;
60 }
61
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62 if (flip && *flip == '1') {
63 s->flip_left = 1;
64 }
65
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66 /* non blocking mode */
67 fcntl(audio_fd, F_SETFL, O_NONBLOCK);
68
4972b26f 69 s->frame_size = AUDIO_BLOCK_SIZE;
de6d9b64 70#if 0
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71 tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
72 err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
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73 if (err < 0) {
74 perror("SNDCTL_DSP_SETFRAGMENT");
75 }
76#endif
77
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78 /* select format : favour native format */
79 err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
80
81#ifdef WORDS_BIGENDIAN
82 if (tmp & AFMT_S16_BE) {
83 tmp = AFMT_S16_BE;
84 } else if (tmp & AFMT_S16_LE) {
85 tmp = AFMT_S16_LE;
86 } else {
87 tmp = 0;
88 }
89#else
90 if (tmp & AFMT_S16_LE) {
91 tmp = AFMT_S16_LE;
92 } else if (tmp & AFMT_S16_BE) {
93 tmp = AFMT_S16_BE;
94 } else {
95 tmp = 0;
96 }
97#endif
98
99 switch(tmp) {
100 case AFMT_S16_LE:
101 s->codec_id = CODEC_ID_PCM_S16LE;
102 break;
103 case AFMT_S16_BE:
104 s->codec_id = CODEC_ID_PCM_S16BE;
105 break;
106 default:
107 fprintf(stderr, "Soundcard does not support 16 bit sample format\n");
108 close(audio_fd);
109 return -EIO;
110 }
111 err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
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112 if (err < 0) {
113 perror("SNDCTL_DSP_SETFMT");
114 goto fail;
115 }
116
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117 tmp = (s->channels == 2);
118 err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
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119 if (err < 0) {
120 perror("SNDCTL_DSP_STEREO");
121 goto fail;
122 }
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123 if (tmp)
124 s->channels = 2;
de6d9b64 125
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126 tmp = s->sample_rate;
127 err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
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128 if (err < 0) {
129 perror("SNDCTL_DSP_SPEED");
130 goto fail;
131 }
4972b26f 132 s->sample_rate = tmp; /* store real sample rate */
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133 s->fd = audio_fd;
134
135 return 0;
136 fail:
137 close(audio_fd);
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138 return -EIO;
139}
140
4972b26f 141static int audio_close(AudioData *s)
de6d9b64 142{
de6d9b64 143 close(s->fd);
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144 return 0;
145}
146
147/* sound output support */
148static int audio_write_header(AVFormatContext *s1)
149{
c9a65ca8 150 AudioData *s = s1->priv_data;
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151 AVStream *st;
152 int ret;
153
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154 st = s1->streams[0];
155 s->sample_rate = st->codec.sample_rate;
156 s->channels = st->codec.channels;
157 ret = audio_open(s, 1);
158 if (ret < 0) {
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159 return -EIO;
160 } else {
161 return 0;
162 }
163}
164
165static int audio_write_packet(AVFormatContext *s1, int stream_index,
10bb7023 166 UINT8 *buf, int size, int force_pts)
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167{
168 AudioData *s = s1->priv_data;
169 int len, ret;
170
171 while (size > 0) {
172 len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
173 if (len > size)
174 len = size;
175 memcpy(s->buffer + s->buffer_ptr, buf, len);
176 s->buffer_ptr += len;
177 if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
178 for(;;) {
179 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
180 if (ret != 0)
181 break;
182 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
183 return -EIO;
184 }
185 s->buffer_ptr = 0;
186 }
187 buf += len;
188 size -= len;
189 }
190 return 0;
191}
192
193static int audio_write_trailer(AVFormatContext *s1)
194{
195 AudioData *s = s1->priv_data;
196
197 audio_close(s);
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198 return 0;
199}
200
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201/* grab support */
202
203static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
204{
c9a65ca8 205 AudioData *s = s1->priv_data;
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206 AVStream *st;
207 int ret;
208
209 if (!ap || ap->sample_rate <= 0 || ap->channels <= 0)
210 return -1;
211
c9a65ca8 212 st = av_new_stream(s1, 0);
4972b26f 213 if (!st) {
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214 return -ENOMEM;
215 }
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216 s->sample_rate = ap->sample_rate;
217 s->channels = ap->channels;
218
219 ret = audio_open(s, 0);
220 if (ret < 0) {
1ea4f593 221 av_free(st);
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222 return -EIO;
223 } else {
224 /* take real parameters */
225 st->codec.codec_type = CODEC_TYPE_AUDIO;
226 st->codec.codec_id = s->codec_id;
227 st->codec.sample_rate = s->sample_rate;
228 st->codec.channels = s->channels;
229 return 0;
230 }
231}
232
233static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
234{
235 AudioData *s = s1->priv_data;
236 int ret;
237
238 if (av_new_packet(pkt, s->frame_size) < 0)
239 return -EIO;
240 for(;;) {
241 ret = read(s->fd, pkt->data, pkt->size);
242 if (ret > 0)
243 break;
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244 if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
245 av_free_packet(pkt);
246 pkt->size = 0;
247 return 0;
248 }
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249 if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
250 av_free_packet(pkt);
251 return -EIO;
252 }
253 }
254 pkt->size = ret;
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255 if (s->flip_left && s->channels == 2) {
256 int i;
257 short *p = (short *) pkt->data;
258
259 for (i = 0; i < ret; i += 4) {
260 *p = ~*p;
261 p += 2;
262 }
263 }
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264 return 0;
265}
266
267static int audio_read_close(AVFormatContext *s1)
268{
269 AudioData *s = s1->priv_data;
270
271 audio_close(s);
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272 return 0;
273}
274
c18a2692 275static AVInputFormat audio_in_format = {
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276 "audio_device",
277 "audio grab and output",
278 sizeof(AudioData),
279 NULL,
280 audio_read_header,
281 audio_read_packet,
282 audio_read_close,
bb76a117 283 .flags = AVFMT_NOFILE,
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284};
285
c18a2692 286static AVOutputFormat audio_out_format = {
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287 "audio_device",
288 "audio grab and output",
289 "",
290 "",
c9a65ca8 291 sizeof(AudioData),
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292 /* XXX: we make the assumption that the soundcard accepts this format */
293 /* XXX: find better solution with "preinit" method, needed also in
294 other formats */
295#ifdef WORDS_BIGENDIAN
296 CODEC_ID_PCM_S16BE,
297#else
298 CODEC_ID_PCM_S16LE,
299#endif
300 CODEC_ID_NONE,
301 audio_write_header,
302 audio_write_packet,
303 audio_write_trailer,
bb76a117 304 .flags = AVFMT_NOFILE,
de6d9b64 305};
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306
307int audio_init(void)
308{
309 av_register_input_format(&audio_in_format);
310 av_register_output_format(&audio_out_format);
311 return 0;
312}