oss_audio.c
Go to the documentation of this file.
1 /*
2  * Linux audio play and grab interface
3  * Copyright (c) 2000, 2001 Fabrice Bellard
4  *
5  * This file is part of Libav.
6  *
7  * Libav is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * Libav is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with Libav; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include "config.h"
23 #include <stdlib.h>
24 #include <stdio.h>
25 #include <stdint.h>
26 #include <string.h>
27 #include <errno.h>
28 #if HAVE_SOUNDCARD_H
29 #include <soundcard.h>
30 #else
31 #include <sys/soundcard.h>
32 #endif
33 #include <unistd.h>
34 #include <fcntl.h>
35 #include <sys/ioctl.h>
36 #include <sys/time.h>
37 #include <sys/select.h>
38 
39 #include "libavutil/log.h"
40 #include "libavutil/opt.h"
41 #include "libavcodec/avcodec.h"
42 #include "libavformat/avformat.h"
43 #include "libavformat/internal.h"
44 
45 #define AUDIO_BLOCK_SIZE 4096
46 
47 typedef struct {
48  AVClass *class;
49  int fd;
51  int channels;
52  int frame_size; /* in bytes ! */
54  unsigned int flip_left : 1;
57 } AudioData;
58 
59 static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device)
60 {
61  AudioData *s = s1->priv_data;
62  int audio_fd;
63  int tmp, err;
64  char *flip = getenv("AUDIO_FLIP_LEFT");
65 
66  if (is_output)
67  audio_fd = open(audio_device, O_WRONLY);
68  else
69  audio_fd = open(audio_device, O_RDONLY);
70  if (audio_fd < 0) {
71  av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
72  return AVERROR(EIO);
73  }
74 
75  if (flip && *flip == '1') {
76  s->flip_left = 1;
77  }
78 
79  /* non blocking mode */
80  if (!is_output)
81  fcntl(audio_fd, F_SETFL, O_NONBLOCK);
82 
84 
85  /* select format : favour native format */
86  err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
87 
88 #if HAVE_BIGENDIAN
89  if (tmp & AFMT_S16_BE) {
90  tmp = AFMT_S16_BE;
91  } else if (tmp & AFMT_S16_LE) {
92  tmp = AFMT_S16_LE;
93  } else {
94  tmp = 0;
95  }
96 #else
97  if (tmp & AFMT_S16_LE) {
98  tmp = AFMT_S16_LE;
99  } else if (tmp & AFMT_S16_BE) {
100  tmp = AFMT_S16_BE;
101  } else {
102  tmp = 0;
103  }
104 #endif
105 
106  switch(tmp) {
107  case AFMT_S16_LE:
109  break;
110  case AFMT_S16_BE:
112  break;
113  default:
114  av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
115  close(audio_fd);
116  return AVERROR(EIO);
117  }
118  err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
119  if (err < 0) {
120  av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
121  goto fail;
122  }
123 
124  tmp = (s->channels == 2);
125  err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
126  if (err < 0) {
127  av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
128  goto fail;
129  }
130 
131  tmp = s->sample_rate;
132  err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
133  if (err < 0) {
134  av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
135  goto fail;
136  }
137  s->sample_rate = tmp; /* store real sample rate */
138  s->fd = audio_fd;
139 
140  return 0;
141  fail:
142  close(audio_fd);
143  return AVERROR(EIO);
144 }
145 
146 static int audio_close(AudioData *s)
147 {
148  close(s->fd);
149  return 0;
150 }
151 
152 /* sound output support */
154 {
155  AudioData *s = s1->priv_data;
156  AVStream *st;
157  int ret;
158 
159  st = s1->streams[0];
160  s->sample_rate = st->codec->sample_rate;
161  s->channels = st->codec->channels;
162  ret = audio_open(s1, 1, s1->filename);
163  if (ret < 0) {
164  return AVERROR(EIO);
165  } else {
166  return 0;
167  }
168 }
169 
171 {
172  AudioData *s = s1->priv_data;
173  int len, ret;
174  int size= pkt->size;
175  uint8_t *buf= pkt->data;
176 
177  while (size > 0) {
178  len = FFMIN(AUDIO_BLOCK_SIZE - s->buffer_ptr, size);
179  memcpy(s->buffer + s->buffer_ptr, buf, len);
180  s->buffer_ptr += len;
181  if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
182  for(;;) {
183  ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
184  if (ret > 0)
185  break;
186  if (ret < 0 && (errno != EAGAIN && errno != EINTR))
187  return AVERROR(EIO);
188  }
189  s->buffer_ptr = 0;
190  }
191  buf += len;
192  size -= len;
193  }
194  return 0;
195 }
196 
198 {
199  AudioData *s = s1->priv_data;
200 
201  audio_close(s);
202  return 0;
203 }
204 
205 /* grab support */
206 
208 {
209  AudioData *s = s1->priv_data;
210  AVStream *st;
211  int ret;
212 
213  st = avformat_new_stream(s1, NULL);
214  if (!st) {
215  return AVERROR(ENOMEM);
216  }
217 
218  ret = audio_open(s1, 0, s1->filename);
219  if (ret < 0) {
220  return AVERROR(EIO);
221  }
222 
223  /* take real parameters */
225  st->codec->codec_id = s->codec_id;
226  st->codec->sample_rate = s->sample_rate;
227  st->codec->channels = s->channels;
228 
229  avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
230  return 0;
231 }
232 
234 {
235  AudioData *s = s1->priv_data;
236  int ret, bdelay;
237  int64_t cur_time;
238  struct audio_buf_info abufi;
239 
240  if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
241  return ret;
242 
243  ret = read(s->fd, pkt->data, pkt->size);
244  if (ret <= 0){
245  av_free_packet(pkt);
246  pkt->size = 0;
247  if (ret<0) return AVERROR(errno);
248  else return AVERROR_EOF;
249  }
250  pkt->size = ret;
251 
252  /* compute pts of the start of the packet */
253  cur_time = av_gettime();
254  bdelay = ret;
255  if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
256  bdelay += abufi.bytes;
257  }
258  /* subtract time represented by the number of bytes in the audio fifo */
259  cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
260 
261  /* convert to wanted units */
262  pkt->pts = cur_time;
263 
264  if (s->flip_left && s->channels == 2) {
265  int i;
266  short *p = (short *) pkt->data;
267 
268  for (i = 0; i < ret; i += 4) {
269  *p = ~*p;
270  p += 2;
271  }
272  }
273  return 0;
274 }
275 
277 {
278  AudioData *s = s1->priv_data;
279 
280  audio_close(s);
281  return 0;
282 }
283 
284 #if CONFIG_OSS_INDEV
285 static const AVOption options[] = {
286  { "sample_rate", "", offsetof(AudioData, sample_rate), AV_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
287  { "channels", "", offsetof(AudioData, channels), AV_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
288  { NULL },
289 };
290 
291 static const AVClass oss_demuxer_class = {
292  .class_name = "OSS demuxer",
293  .item_name = av_default_item_name,
294  .option = options,
295  .version = LIBAVUTIL_VERSION_INT,
296 };
297 
298 AVInputFormat ff_oss_demuxer = {
299  .name = "oss",
300  .long_name = NULL_IF_CONFIG_SMALL("Open Sound System capture"),
301  .priv_data_size = sizeof(AudioData),
305  .flags = AVFMT_NOFILE,
306  .priv_class = &oss_demuxer_class,
307 };
308 #endif
309 
310 #if CONFIG_OSS_OUTDEV
311 AVOutputFormat ff_oss_muxer = {
312  .name = "oss",
313  .long_name = NULL_IF_CONFIG_SMALL("Open Sound System playback"),
314  .priv_data_size = sizeof(AudioData),
315  /* XXX: we make the assumption that the soundcard accepts this format */
316  /* XXX: find better solution with "preinit" method, needed also in
317  other formats */
319  .video_codec = CODEC_ID_NONE,
320  .write_header = audio_write_header,
321  .write_packet = audio_write_packet,
322  .write_trailer = audio_write_trailer,
323  .flags = AVFMT_NOFILE,
324 };
325 #endif