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 
37 #include "libavutil/log.h"
38 #include "libavutil/opt.h"
39 #include "libavutil/time.h"
40 #include "libavcodec/avcodec.h"
41 #include "libavformat/avformat.h"
42 #include "libavformat/internal.h"
43 
44 #define AUDIO_BLOCK_SIZE 4096
45 
46 typedef struct {
47  AVClass *class;
48  int fd;
50  int channels;
51  int frame_size; /* in bytes ! */
53  unsigned int flip_left : 1;
56 } AudioData;
57 
58 static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device)
59 {
60  AudioData *s = s1->priv_data;
61  int audio_fd;
62  int tmp, err;
63  char *flip = getenv("AUDIO_FLIP_LEFT");
64 
65  if (is_output)
66  audio_fd = open(audio_device, O_WRONLY);
67  else
68  audio_fd = open(audio_device, O_RDONLY);
69  if (audio_fd < 0) {
70  av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
71  return AVERROR(EIO);
72  }
73 
74  if (flip && *flip == '1') {
75  s->flip_left = 1;
76  }
77 
78  /* non blocking mode */
79  if (!is_output)
80  fcntl(audio_fd, F_SETFL, O_NONBLOCK);
81 
83 
84  /* select format : favour native format */
85  err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
86 
87 #if HAVE_BIGENDIAN
88  if (tmp & AFMT_S16_BE) {
89  tmp = AFMT_S16_BE;
90  } else if (tmp & AFMT_S16_LE) {
91  tmp = AFMT_S16_LE;
92  } else {
93  tmp = 0;
94  }
95 #else
96  if (tmp & AFMT_S16_LE) {
97  tmp = AFMT_S16_LE;
98  } else if (tmp & AFMT_S16_BE) {
99  tmp = AFMT_S16_BE;
100  } else {
101  tmp = 0;
102  }
103 #endif
104 
105  switch(tmp) {
106  case AFMT_S16_LE:
108  break;
109  case AFMT_S16_BE:
111  break;
112  default:
113  av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
114  close(audio_fd);
115  return AVERROR(EIO);
116  }
117  err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
118  if (err < 0) {
119  av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
120  goto fail;
121  }
122 
123  tmp = (s->channels == 2);
124  err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
125  if (err < 0) {
126  av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
127  goto fail;
128  }
129 
130  tmp = s->sample_rate;
131  err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
132  if (err < 0) {
133  av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
134  goto fail;
135  }
136  s->sample_rate = tmp; /* store real sample rate */
137  s->fd = audio_fd;
138 
139  return 0;
140  fail:
141  close(audio_fd);
142  return AVERROR(EIO);
143 }
144 
145 static int audio_close(AudioData *s)
146 {
147  close(s->fd);
148  return 0;
149 }
150 
151 /* sound output support */
153 {
154  AudioData *s = s1->priv_data;
155  AVStream *st;
156  int ret;
157 
158  st = s1->streams[0];
159  s->sample_rate = st->codec->sample_rate;
160  s->channels = st->codec->channels;
161  ret = audio_open(s1, 1, s1->filename);
162  if (ret < 0) {
163  return AVERROR(EIO);
164  } else {
165  return 0;
166  }
167 }
168 
170 {
171  AudioData *s = s1->priv_data;
172  int len, ret;
173  int size= pkt->size;
174  uint8_t *buf= pkt->data;
175 
176  while (size > 0) {
177  len = FFMIN(AUDIO_BLOCK_SIZE - s->buffer_ptr, size);
178  memcpy(s->buffer + s->buffer_ptr, buf, len);
179  s->buffer_ptr += len;
180  if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
181  for(;;) {
182  ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
183  if (ret > 0)
184  break;
185  if (ret < 0 && (errno != EAGAIN && errno != EINTR))
186  return AVERROR(EIO);
187  }
188  s->buffer_ptr = 0;
189  }
190  buf += len;
191  size -= len;
192  }
193  return 0;
194 }
195 
197 {
198  AudioData *s = s1->priv_data;
199 
200  audio_close(s);
201  return 0;
202 }
203 
204 /* grab support */
205 
207 {
208  AudioData *s = s1->priv_data;
209  AVStream *st;
210  int ret;
211 
212  st = avformat_new_stream(s1, NULL);
213  if (!st) {
214  return AVERROR(ENOMEM);
215  }
216 
217  ret = audio_open(s1, 0, s1->filename);
218  if (ret < 0) {
219  return AVERROR(EIO);
220  }
221 
222  /* take real parameters */
224  st->codec->codec_id = s->codec_id;
225  st->codec->sample_rate = s->sample_rate;
226  st->codec->channels = s->channels;
227 
228  avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
229  return 0;
230 }
231 
233 {
234  AudioData *s = s1->priv_data;
235  int ret, bdelay;
236  int64_t cur_time;
237  struct audio_buf_info abufi;
238 
239  if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
240  return ret;
241 
242  ret = read(s->fd, pkt->data, pkt->size);
243  if (ret <= 0){
244  av_free_packet(pkt);
245  pkt->size = 0;
246  if (ret<0) return AVERROR(errno);
247  else return AVERROR_EOF;
248  }
249  pkt->size = ret;
250 
251  /* compute pts of the start of the packet */
252  cur_time = av_gettime();
253  bdelay = ret;
254  if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
255  bdelay += abufi.bytes;
256  }
257  /* subtract time represented by the number of bytes in the audio fifo */
258  cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
259 
260  /* convert to wanted units */
261  pkt->pts = cur_time;
262 
263  if (s->flip_left && s->channels == 2) {
264  int i;
265  short *p = (short *) pkt->data;
266 
267  for (i = 0; i < ret; i += 4) {
268  *p = ~*p;
269  p += 2;
270  }
271  }
272  return 0;
273 }
274 
276 {
277  AudioData *s = s1->priv_data;
278 
279  audio_close(s);
280  return 0;
281 }
282 
283 #if CONFIG_OSS_INDEV
284 static const AVOption options[] = {
285  { "sample_rate", "", offsetof(AudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
286  { "channels", "", offsetof(AudioData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
287  { NULL },
288 };
289 
290 static const AVClass oss_demuxer_class = {
291  .class_name = "OSS demuxer",
292  .item_name = av_default_item_name,
293  .option = options,
294  .version = LIBAVUTIL_VERSION_INT,
295 };
296 
297 AVInputFormat ff_oss_demuxer = {
298  .name = "oss",
299  .long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
300  .priv_data_size = sizeof(AudioData),
304  .flags = AVFMT_NOFILE,
305  .priv_class = &oss_demuxer_class,
306 };
307 #endif
308 
309 #if CONFIG_OSS_OUTDEV
310 AVOutputFormat ff_oss_muxer = {
311  .name = "oss",
312  .long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) playback"),
313  .priv_data_size = sizeof(AudioData),
314  /* XXX: we make the assumption that the soundcard accepts this format */
315  /* XXX: find better solution with "preinit" method, needed also in
316  other formats */
318  .video_codec = AV_CODEC_ID_NONE,
319  .write_header = audio_write_header,
320  .write_packet = audio_write_packet,
321  .write_trailer = audio_write_trailer,
322  .flags = AVFMT_NOFILE,
323 };
324 #endif