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oss_audio.c
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1 /*
2  * Linux audio play and grab interface
3  * Copyright (c) 2000, 2001 Fabrice Bellard
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg 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  * FFmpeg 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 FFmpeg; 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 "avdevice.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  if (fcntl(audio_fd, F_SETFL, O_NONBLOCK) < 0) {
81  av_log(s1, AV_LOG_WARNING, "%s: Could not enable non block mode (%s)\n", audio_device, strerror(errno));
82  }
83  }
84 
86 
87  /* select format : favour native format */
88  err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
89 
90 #if HAVE_BIGENDIAN
91  if (tmp & AFMT_S16_BE) {
92  tmp = AFMT_S16_BE;
93  } else if (tmp & AFMT_S16_LE) {
94  tmp = AFMT_S16_LE;
95  } else {
96  tmp = 0;
97  }
98 #else
99  if (tmp & AFMT_S16_LE) {
100  tmp = AFMT_S16_LE;
101  } else if (tmp & AFMT_S16_BE) {
102  tmp = AFMT_S16_BE;
103  } else {
104  tmp = 0;
105  }
106 #endif
107 
108  switch(tmp) {
109  case AFMT_S16_LE:
111  break;
112  case AFMT_S16_BE:
114  break;
115  default:
116  av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
117  close(audio_fd);
118  return AVERROR(EIO);
119  }
120  err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
121  if (err < 0) {
122  av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
123  goto fail;
124  }
125 
126  tmp = (s->channels == 2);
127  err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
128  if (err < 0) {
129  av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
130  goto fail;
131  }
132 
133  tmp = s->sample_rate;
134  err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
135  if (err < 0) {
136  av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
137  goto fail;
138  }
139  s->sample_rate = tmp; /* store real sample rate */
140  s->fd = audio_fd;
141 
142  return 0;
143  fail:
144  close(audio_fd);
145  return AVERROR(EIO);
146 }
147 
148 static int audio_close(AudioData *s)
149 {
150  close(s->fd);
151  return 0;
152 }
153 
154 /* sound output support */
156 {
157  AudioData *s = s1->priv_data;
158  AVStream *st;
159  int ret;
160 
161  st = s1->streams[0];
162  s->sample_rate = st->codec->sample_rate;
163  s->channels = st->codec->channels;
164  ret = audio_open(s1, 1, s1->filename);
165  if (ret < 0) {
166  return AVERROR(EIO);
167  } else {
168  return 0;
169  }
170 }
171 
173 {
174  AudioData *s = s1->priv_data;
175  int len, ret;
176  int size= pkt->size;
177  uint8_t *buf= pkt->data;
178 
179  while (size > 0) {
180  len = FFMIN(AUDIO_BLOCK_SIZE - s->buffer_ptr, size);
181  memcpy(s->buffer + s->buffer_ptr, buf, len);
182  s->buffer_ptr += len;
183  if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
184  for(;;) {
185  ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
186  if (ret > 0)
187  break;
188  if (ret < 0 && (errno != EAGAIN && errno != EINTR))
189  return AVERROR(EIO);
190  }
191  s->buffer_ptr = 0;
192  }
193  buf += len;
194  size -= len;
195  }
196  return 0;
197 }
198 
200 {
201  AudioData *s = s1->priv_data;
202 
203  audio_close(s);
204  return 0;
205 }
206 
207 /* grab support */
208 
210 {
211  AudioData *s = s1->priv_data;
212  AVStream *st;
213  int ret;
214 
215  st = avformat_new_stream(s1, NULL);
216  if (!st) {
217  return AVERROR(ENOMEM);
218  }
219 
220  ret = audio_open(s1, 0, s1->filename);
221  if (ret < 0) {
222  return AVERROR(EIO);
223  }
224 
225  /* take real parameters */
227  st->codec->codec_id = s->codec_id;
228  st->codec->sample_rate = s->sample_rate;
229  st->codec->channels = s->channels;
230 
231  avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
232  return 0;
233 }
234 
236 {
237  AudioData *s = s1->priv_data;
238  int ret, bdelay;
239  int64_t cur_time;
240  struct audio_buf_info abufi;
241 
242  if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
243  return ret;
244 
245  ret = read(s->fd, pkt->data, pkt->size);
246  if (ret <= 0){
247  av_free_packet(pkt);
248  pkt->size = 0;
249  if (ret<0) return AVERROR(errno);
250  else return AVERROR_EOF;
251  }
252  pkt->size = ret;
253 
254  /* compute pts of the start of the packet */
255  cur_time = av_gettime();
256  bdelay = ret;
257  if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
258  bdelay += abufi.bytes;
259  }
260  /* subtract time represented by the number of bytes in the audio fifo */
261  cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
262 
263  /* convert to wanted units */
264  pkt->pts = cur_time;
265 
266  if (s->flip_left && s->channels == 2) {
267  int i;
268  short *p = (short *) pkt->data;
269 
270  for (i = 0; i < ret; i += 4) {
271  *p = ~*p;
272  p += 2;
273  }
274  }
275  return 0;
276 }
277 
279 {
280  AudioData *s = s1->priv_data;
281 
282  audio_close(s);
283  return 0;
284 }
285 
286 #if CONFIG_OSS_INDEV
287 static const AVOption options[] = {
288  { "sample_rate", "", offsetof(AudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
289  { "channels", "", offsetof(AudioData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
290  { NULL },
291 };
292 
293 static const AVClass oss_demuxer_class = {
294  .class_name = "OSS demuxer",
295  .item_name = av_default_item_name,
296  .option = options,
297  .version = LIBAVUTIL_VERSION_INT,
298 };
299 
300 AVInputFormat ff_oss_demuxer = {
301  .name = "oss",
302  .long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
303  .priv_data_size = sizeof(AudioData),
307  .flags = AVFMT_NOFILE,
308  .priv_class = &oss_demuxer_class,
309 };
310 #endif
311 
312 #if CONFIG_OSS_OUTDEV
313 AVOutputFormat ff_oss_muxer = {
314  .name = "oss",
315  .long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) playback"),
316  .priv_data_size = sizeof(AudioData),
317  /* XXX: we make the assumption that the soundcard accepts this format */
318  /* XXX: find better solution with "preinit" method, needed also in
319  other formats */
321  .video_codec = AV_CODEC_ID_NONE,
322  .write_header = audio_write_header,
323  .write_packet = audio_write_packet,
324  .write_trailer = audio_write_trailer,
325  .flags = AVFMT_NOFILE,
326 };
327 #endif