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_t 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 AVERROR_IO;
  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         av_log(NULL, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
 110         close(audio_fd);
 111         return AVERROR_IO;
 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 AVERROR_IO;
 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 AVERROR_IO;
 162     } else {
 163         return 0;
 164     }
 165 }
 166
 167 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
 168 {
 169     AudioData *s = s1->priv_data;
 170     int len, ret;
 171     int size= pkt->size;
 172     uint8_t *buf= pkt->data;
 173
 174     while (size > 0) {
 175         len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
 176         if (len > size)
 177             len = 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_IO;
 187             }
 188             s->buffer_ptr = 0;
 189         }
 190         buf += len;
 191         size -= len;
 192     }
 193     return 0;
 194 }
 195
 196 static int audio_write_trailer(AVFormatContext *s1)
 197 {
 198     AudioData *s = s1->priv_data;
 199
 200     audio_close(s);
 201     return 0;
 202 }
 203
 204 /* grab support */
 205
 206 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
 207 {
 208     AudioData *s = s1->priv_data;
 209     AVStream *st;
 210     int ret;
 211
 212     if (!ap || ap->sample_rate <= 0 || ap->channels <= 0)
 213         return -1;
 214
 215     st = av_new_stream(s1, 0);
 216     if (!st) {
 217         return -ENOMEM;
 218     }
 219     s->sample_rate = ap->sample_rate;
 220     s->channels = ap->channels;
 221
 222     ret = audio_open(s, 0, ap->device);
 223     if (ret < 0) {
 224         av_free(st);
 225         return AVERROR_IO;
 226     }
 227
 228     /* take real parameters */
 229     st->codec.codec_type = CODEC_TYPE_AUDIO;
 230     st->codec.codec_id = s->codec_id;
 231     st->codec.sample_rate = s->sample_rate;
 232     st->codec.channels = s->channels;
 233
 234     av_set_pts_info(st, 48, 1, 1000000);  /* 48 bits pts in us */
 235     return 0;
 236 }
 237
 238 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
 239 {
 240     AudioData *s = s1->priv_data;
 241     int ret, bdelay;
 242     int64_t cur_time;
 243     struct audio_buf_info abufi;
 244
 245     if (av_new_packet(pkt, s->frame_size) < 0)
 246         return AVERROR_IO;
 247     for(;;) {
 248         struct timeval tv;
 249         fd_set fds;
 250
 251         tv.tv_sec = 0;
 252         tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */
 253
 254         FD_ZERO(&fds);
 255         FD_SET(s->fd, &fds);
 256
 257         /* This will block until data is available or we get a timeout */
 258         (void) select(s->fd + 1, &fds, 0, 0, &tv);
 259
 260         ret = read(s->fd, pkt->data, pkt->size);
 261         if (ret > 0)
 262             break;
 263         if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
 264             av_free_packet(pkt);
 265             pkt->size = 0;
 266             pkt->pts = av_gettime() & ((1LL << 48) - 1);
 267             return 0;
 268         }
 269         if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
 270             av_free_packet(pkt);
 271             return AVERROR_IO;
 272         }
 273     }
 274     pkt->size = ret;
 275
 276     /* compute pts of the start of the packet */
 277     cur_time = av_gettime();
 278     bdelay = ret;
 279     if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
 280         bdelay += abufi.bytes;
 281     }
 282     /* substract time represented by the number of bytes in the audio fifo */
 283     cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
 284
 285     /* convert to wanted units */
 286     pkt->pts = cur_time & ((1LL << 48) - 1);
 287
 288     if (s->flip_left && s->channels == 2) {
 289         int i;
 290         short *p = (short *) pkt->data;
 291
 292         for (i = 0; i < ret; i += 4) {
 293             *p = ~*p;
 294             p += 2;
 295         }
 296     }
 297     return 0;
 298 }
 299
 300 static int audio_read_close(AVFormatContext *s1)
 301 {
 302     AudioData *s = s1->priv_data;
 303
 304     audio_close(s);
 305     return 0;
 306 }
 307
 308 static AVInputFormat audio_in_format = {
 309     "audio_device",
 310     "audio grab and output",
 311     sizeof(AudioData),
 312     NULL,
 313     audio_read_header,
 314     audio_read_packet,
 315     audio_read_close,
 316     .flags = AVFMT_NOFILE,
 317 };
 318
 319 static AVOutputFormat audio_out_format = {
 320     "audio_device",
 321     "audio grab and output",
 322     "",
 323     "",
 324     sizeof(AudioData),
 325     /* XXX: we make the assumption that the soundcard accepts this format */
 326     /* XXX: find better solution with "preinit" method, needed also in
 327        other formats */
 328 #ifdef WORDS_BIGENDIAN
 329     CODEC_ID_PCM_S16BE,
 330 #else
 331     CODEC_ID_PCM_S16LE,
 332 #endif
 333     CODEC_ID_NONE,
 334     audio_write_header,
 335     audio_write_packet,
 336     audio_write_trailer,
 337     .flags = AVFMT_NOFILE,
 338 };
 339
 340 int audio_init(void)
 341 {
 342     av_register_input_format(&audio_in_format);
 343     av_register_output_format(&audio_out_format);
 344     return 0;
 345 }