05055a17818f1a3391c5f7fda95f7cc312a9c22b
[libav.git] / libavformat / audio.c
1 /*
2 * Linux audio play and grab interface
3 * Copyright (c) 2000, 2001 Fabrice Bellard.
4 *
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.
9 *
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Lesser General Public License for more details.
14 *
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
18 */
19 #include "avformat.h"
20
21 #include <stdlib.h>
22 #include <stdio.h>
23 #include <string.h>
24 #include <sys/soundcard.h>
25 #include <unistd.h>
26 #include <fcntl.h>
27 #include <sys/ioctl.h>
28 #include <sys/mman.h>
29 #include <sys/time.h>
30
31 #define AUDIO_BLOCK_SIZE 4096
32
33 typedef struct {
34 int fd;
35 int sample_rate;
36 int channels;
37 int frame_size; /* in bytes ! */
38 int codec_id;
39 int flip_left : 1;
40 UINT8 buffer[AUDIO_BLOCK_SIZE];
41 int buffer_ptr;
42 } AudioData;
43
44 static int audio_open(AudioData *s, int is_output, const char *audio_device)
45 {
46 int audio_fd;
47 int tmp, err;
48 char *flip = getenv("AUDIO_FLIP_LEFT");
49
50 /* open linux audio device */
51 if (!audio_device)
52 audio_device = "/dev/dsp";
53
54 if (is_output)
55 audio_fd = open(audio_device, O_WRONLY);
56 else
57 audio_fd = open(audio_device, O_RDONLY);
58 if (audio_fd < 0) {
59 perror(audio_device);
60 return -EIO;
61 }
62
63 if (flip && *flip == '1') {
64 s->flip_left = 1;
65 }
66
67 /* non blocking mode */
68 if (!is_output)
69 fcntl(audio_fd, F_SETFL, O_NONBLOCK);
70
71 s->frame_size = AUDIO_BLOCK_SIZE;
72 #if 0
73 tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
74 err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
75 if (err < 0) {
76 perror("SNDCTL_DSP_SETFRAGMENT");
77 }
78 #endif
79
80 /* select format : favour native format */
81 err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
82
83 #ifdef WORDS_BIGENDIAN
84 if (tmp & AFMT_S16_BE) {
85 tmp = AFMT_S16_BE;
86 } else if (tmp & AFMT_S16_LE) {
87 tmp = AFMT_S16_LE;
88 } else {
89 tmp = 0;
90 }
91 #else
92 if (tmp & AFMT_S16_LE) {
93 tmp = AFMT_S16_LE;
94 } else if (tmp & AFMT_S16_BE) {
95 tmp = AFMT_S16_BE;
96 } else {
97 tmp = 0;
98 }
99 #endif
100
101 switch(tmp) {
102 case AFMT_S16_LE:
103 s->codec_id = CODEC_ID_PCM_S16LE;
104 break;
105 case AFMT_S16_BE:
106 s->codec_id = CODEC_ID_PCM_S16BE;
107 break;
108 default:
109 fprintf(stderr, "Soundcard does not support 16 bit sample format\n");
110 close(audio_fd);
111 return -EIO;
112 }
113 err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
114 if (err < 0) {
115 perror("SNDCTL_DSP_SETFMT");
116 goto fail;
117 }
118
119 tmp = (s->channels == 2);
120 err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
121 if (err < 0) {
122 perror("SNDCTL_DSP_STEREO");
123 goto fail;
124 }
125 if (tmp)
126 s->channels = 2;
127
128 tmp = s->sample_rate;
129 err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
130 if (err < 0) {
131 perror("SNDCTL_DSP_SPEED");
132 goto fail;
133 }
134 s->sample_rate = tmp; /* store real sample rate */
135 s->fd = audio_fd;
136
137 return 0;
138 fail:
139 close(audio_fd);
140 return -EIO;
141 }
142
143 static int audio_close(AudioData *s)
144 {
145 close(s->fd);
146 return 0;
147 }
148
149 /* sound output support */
150 static int audio_write_header(AVFormatContext *s1)
151 {
152 AudioData *s = s1->priv_data;
153 AVStream *st;
154 int ret;
155
156 st = s1->streams[0];
157 s->sample_rate = st->codec.sample_rate;
158 s->channels = st->codec.channels;
159 ret = audio_open(s, 1, NULL);
160 if (ret < 0) {
161 return -EIO;
162 } else {
163 return 0;
164 }
165 }
166
167 static int audio_write_packet(AVFormatContext *s1, int stream_index,
168 UINT8 *buf, int size, int force_pts)
169 {
170 AudioData *s = s1->priv_data;
171 int len, ret;
172
173 while (size > 0) {
174 len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
175 if (len > size)
176 len = size;
177 memcpy(s->buffer + s->buffer_ptr, buf, len);
178 s->buffer_ptr += len;
179 if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
180 for(;;) {
181 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
182 if (ret > 0)
183 break;
184 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
185 return -EIO;
186 }
187 s->buffer_ptr = 0;
188 }
189 buf += len;
190 size -= len;
191 }
192 return 0;
193 }
194
195 static int audio_write_trailer(AVFormatContext *s1)
196 {
197 AudioData *s = s1->priv_data;
198
199 audio_close(s);
200 return 0;
201 }
202
203 /* grab support */
204
205 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
206 {
207 AudioData *s = s1->priv_data;
208 AVStream *st;
209 int ret;
210
211 if (!ap || ap->sample_rate <= 0 || ap->channels <= 0)
212 return -1;
213
214 st = av_new_stream(s1, 0);
215 if (!st) {
216 return -ENOMEM;
217 }
218 s->sample_rate = ap->sample_rate;
219 s->channels = ap->channels;
220
221 ret = audio_open(s, 0, ap->device);
222 if (ret < 0) {
223 av_free(st);
224 return -EIO;
225 }
226
227 /* take real parameters */
228 st->codec.codec_type = CODEC_TYPE_AUDIO;
229 st->codec.codec_id = s->codec_id;
230 st->codec.sample_rate = s->sample_rate;
231 st->codec.channels = s->channels;
232
233 av_set_pts_info(s1, 48, 1, 1000000); /* 48 bits pts in us */
234 return 0;
235 }
236
237 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
238 {
239 AudioData *s = s1->priv_data;
240 int ret, bdelay;
241 int64_t cur_time;
242 struct audio_buf_info abufi;
243
244 if (av_new_packet(pkt, s->frame_size) < 0)
245 return -EIO;
246 for(;;) {
247 ret = read(s->fd, pkt->data, pkt->size);
248 if (ret > 0)
249 break;
250 if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
251 av_free_packet(pkt);
252 pkt->size = 0;
253 return 0;
254 }
255 if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
256 av_free_packet(pkt);
257 return -EIO;
258 }
259 }
260 pkt->size = ret;
261
262 /* compute pts of the start of the packet */
263 cur_time = av_gettime();
264 bdelay = ret;
265 if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
266 bdelay += abufi.bytes;
267 }
268 /* substract time represented by the number of bytes in the audio fifo */
269 cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
270
271 /* convert to wanted units */
272 pkt->pts = cur_time & ((1LL << 48) - 1);
273
274 if (s->flip_left && s->channels == 2) {
275 int i;
276 short *p = (short *) pkt->data;
277
278 for (i = 0; i < ret; i += 4) {
279 *p = ~*p;
280 p += 2;
281 }
282 }
283 return 0;
284 }
285
286 static int audio_read_close(AVFormatContext *s1)
287 {
288 AudioData *s = s1->priv_data;
289
290 audio_close(s);
291 return 0;
292 }
293
294 static AVInputFormat audio_in_format = {
295 "audio_device",
296 "audio grab and output",
297 sizeof(AudioData),
298 NULL,
299 audio_read_header,
300 audio_read_packet,
301 audio_read_close,
302 .flags = AVFMT_NOFILE,
303 };
304
305 static AVOutputFormat audio_out_format = {
306 "audio_device",
307 "audio grab and output",
308 "",
309 "",
310 sizeof(AudioData),
311 /* XXX: we make the assumption that the soundcard accepts this format */
312 /* XXX: find better solution with "preinit" method, needed also in
313 other formats */
314 #ifdef WORDS_BIGENDIAN
315 CODEC_ID_PCM_S16BE,
316 #else
317 CODEC_ID_PCM_S16LE,
318 #endif
319 CODEC_ID_NONE,
320 audio_write_header,
321 audio_write_packet,
322 audio_write_trailer,
323 .flags = AVFMT_NOFILE,
324 };
325
326 int audio_init(void)
327 {
328 av_register_input_format(&audio_in_format);
329 av_register_output_format(&audio_out_format);
330 return 0;
331 }