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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  18  */
  19 #include "avformat.h"
  20
  21 #include <stdlib.h>
  22 #include <stdio.h>
  23 #include <string.h>
  24 #ifdef __OpenBSD__
  25 #include <soundcard.h>
  26 #else
  27 #include <sys/soundcard.h>
  28 #endif
  29 #include <unistd.h>
  30 #include <fcntl.h>
  31 #include <sys/ioctl.h>
  32 #include <sys/mman.h>
  33 #include <sys/time.h>
  34
  35 #define AUDIO_BLOCK_SIZE 4096
  36
  37 typedef struct {
  38     int fd;
  39     int sample_rate;
  40     int channels;
  41     int frame_size; /* in bytes ! */
  42     int codec_id;
  43     int flip_left : 1;
  44     uint8_t buffer[AUDIO_BLOCK_SIZE];
  45     int buffer_ptr;
  46 } AudioData;
  47
  48 static int audio_open(AudioData *s, int is_output, const char *audio_device)
  49 {
  50     int audio_fd;
  51     int tmp, err;
  52     char *flip = getenv("AUDIO_FLIP_LEFT");
  53
  54     /* open linux audio device */
  55     if (!audio_device)
  56 #ifdef __OpenBSD__
  57         audio_device = "/dev/sound";
  58 #else
  59         audio_device = "/dev/dsp";
  60 #endif
  61
  62     if (is_output)
  63         audio_fd = open(audio_device, O_WRONLY);
  64     else
  65         audio_fd = open(audio_device, O_RDONLY);
  66     if (audio_fd < 0) {
  67         perror(audio_device);
  68         return AVERROR_IO;
  69     }
  70
  71     if (flip && *flip == '1') {
  72         s->flip_left = 1;
  73     }
  74
  75     /* non blocking mode */
  76     if (!is_output)
  77         fcntl(audio_fd, F_SETFL, O_NONBLOCK);
  78
  79     s->frame_size = AUDIO_BLOCK_SIZE;
  80 #if 0
  81     tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
  82     err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
  83     if (err < 0) {
  84         perror("SNDCTL_DSP_SETFRAGMENT");
  85     }
  86 #endif
  87
  88     /* select format : favour native format */
  89     err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
  90
  91 #ifdef WORDS_BIGENDIAN
  92     if (tmp & AFMT_S16_BE) {
  93         tmp = AFMT_S16_BE;
  94     } else if (tmp & AFMT_S16_LE) {
  95         tmp = AFMT_S16_LE;
  96     } else {
  97         tmp = 0;
  98     }
  99 #else
 100     if (tmp & AFMT_S16_LE) {
 101         tmp = AFMT_S16_LE;
 102     } else if (tmp & AFMT_S16_BE) {
 103         tmp = AFMT_S16_BE;
 104     } else {
 105         tmp = 0;
 106     }
 107 #endif
 108
 109     switch(tmp) {
 110     case AFMT_S16_LE:
 111         s->codec_id = CODEC_ID_PCM_S16LE;
 112         break;
 113     case AFMT_S16_BE:
 114         s->codec_id = CODEC_ID_PCM_S16BE;
 115         break;
 116     default:
 117         av_log(NULL, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
 118         close(audio_fd);
 119         return AVERROR_IO;
 120     }
 121     err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
 122     if (err < 0) {
 123         perror("SNDCTL_DSP_SETFMT");
 124         goto fail;
 125     }
 126
 127     tmp = (s->channels == 2);
 128     err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
 129     if (err < 0) {
 130         perror("SNDCTL_DSP_STEREO");
 131         goto fail;
 132     }
 133     if (tmp)
 134         s->channels = 2;
 135
 136     tmp = s->sample_rate;
 137     err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
 138     if (err < 0) {
 139         perror("SNDCTL_DSP_SPEED");
 140         goto fail;
 141     }
 142     s->sample_rate = tmp; /* store real sample rate */
 143     s->fd = audio_fd;
 144
 145     return 0;
 146  fail:
 147     close(audio_fd);
 148     return AVERROR_IO;
 149 }
 150
 151 static int audio_close(AudioData *s)
 152 {
 153     close(s->fd);
 154     return 0;
 155 }
 156
 157 /* sound output support */
 158 static int audio_write_header(AVFormatContext *s1)
 159 {
 160     AudioData *s = s1->priv_data;
 161     AVStream *st;
 162     int ret;
 163
 164     st = s1->streams[0];
 165     s->sample_rate = st->codec->sample_rate;
 166     s->channels = st->codec->channels;
 167     ret = audio_open(s, 1, NULL);
 168     if (ret < 0) {
 169         return AVERROR_IO;
 170     } else {
 171         return 0;
 172     }
 173 }
 174
 175 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
 176 {
 177     AudioData *s = s1->priv_data;
 178     int len, ret;
 179     int size= pkt->size;
 180     uint8_t *buf= pkt->data;
 181
 182     while (size > 0) {
 183         len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
 184         if (len > size)
 185             len = size;
 186         memcpy(s->buffer + s->buffer_ptr, buf, len);
 187         s->buffer_ptr += len;
 188         if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
 189             for(;;) {
 190                 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
 191                 if (ret > 0)
 192                     break;
 193                 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
 194                     return AVERROR_IO;
 195             }
 196             s->buffer_ptr = 0;
 197         }
 198         buf += len;
 199         size -= len;
 200     }
 201     return 0;
 202 }
 203
 204 static int audio_write_trailer(AVFormatContext *s1)
 205 {
 206     AudioData *s = s1->priv_data;
 207
 208     audio_close(s);
 209     return 0;
 210 }
 211
 212 /* grab support */
 213
 214 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
 215 {
 216     AudioData *s = s1->priv_data;
 217     AVStream *st;
 218     int ret;
 219
 220     if (ap->sample_rate <= 0 || ap->channels <= 0)
 221         return -1;
 222
 223     st = av_new_stream(s1, 0);
 224     if (!st) {
 225         return -ENOMEM;
 226     }
 227     s->sample_rate = ap->sample_rate;
 228     s->channels = ap->channels;
 229
 230     ret = audio_open(s, 0, ap->device);
 231     if (ret < 0) {
 232         av_free(st);
 233         return AVERROR_IO;
 234     }
 235
 236     /* take real parameters */
 237     st->codec->codec_type = CODEC_TYPE_AUDIO;
 238     st->codec->codec_id = s->codec_id;
 239     st->codec->sample_rate = s->sample_rate;
 240     st->codec->channels = s->channels;
 241
 242     av_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
 243     return 0;
 244 }
 245
 246 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
 247 {
 248     AudioData *s = s1->priv_data;
 249     int ret, bdelay;
 250     int64_t cur_time;
 251     struct audio_buf_info abufi;
 252
 253     if (av_new_packet(pkt, s->frame_size) < 0)
 254         return AVERROR_IO;
 255     for(;;) {
 256         struct timeval tv;
 257         fd_set fds;
 258
 259         tv.tv_sec = 0;
 260         tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */
 261
 262         FD_ZERO(&fds);
 263         FD_SET(s->fd, &fds);
 264
 265         /* This will block until data is available or we get a timeout */
 266         (void) select(s->fd + 1, &fds, 0, 0, &tv);
 267
 268         ret = read(s->fd, pkt->data, pkt->size);
 269         if (ret > 0)
 270             break;
 271         if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
 272             av_free_packet(pkt);
 273             pkt->size = 0;
 274             pkt->pts = av_gettime();
 275             return 0;
 276         }
 277         if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
 278             av_free_packet(pkt);
 279             return AVERROR_IO;
 280         }
 281     }
 282     pkt->size = ret;
 283
 284     /* compute pts of the start of the packet */
 285     cur_time = av_gettime();
 286     bdelay = ret;
 287     if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
 288         bdelay += abufi.bytes;
 289     }
 290     /* substract time represented by the number of bytes in the audio fifo */
 291     cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
 292
 293     /* convert to wanted units */
 294     pkt->pts = cur_time;
 295
 296     if (s->flip_left && s->channels == 2) {
 297         int i;
 298         short *p = (short *) pkt->data;
 299
 300         for (i = 0; i < ret; i += 4) {
 301             *p = ~*p;
 302             p += 2;
 303         }
 304     }
 305     return 0;
 306 }
 307
 308 static int audio_read_close(AVFormatContext *s1)
 309 {
 310     AudioData *s = s1->priv_data;
 311
 312     audio_close(s);
 313     return 0;
 314 }
 315
 316 #ifdef CONFIG_AUDIO_DEMUXER
 317 AVInputFormat audio_demuxer = {
 318     "audio_device",
 319     "audio grab and output",
 320     sizeof(AudioData),
 321     NULL,
 322     audio_read_header,
 323     audio_read_packet,
 324     audio_read_close,
 325     .flags = AVFMT_NOFILE,
 326 };
 327 #endif
 328
 329 #ifdef CONFIG_AUDIO_MUXER
 330 AVOutputFormat audio_muxer = {
 331     "audio_device",
 332     "audio grab and output",
 333     "",
 334     "",
 335     sizeof(AudioData),
 336     /* XXX: we make the assumption that the soundcard accepts this format */
 337     /* XXX: find better solution with "preinit" method, needed also in
 338        other formats */
 339 #ifdef WORDS_BIGENDIAN
 340     CODEC_ID_PCM_S16BE,
 341 #else
 342     CODEC_ID_PCM_S16LE,
 343 #endif
 344     CODEC_ID_NONE,
 345     audio_write_header,
 346     audio_write_packet,
 347     audio_write_trailer,
 348     .flags = AVFMT_NOFILE,
 349 };
 350 #endif