libavdevice/audio.c

   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 #ifdef 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
  38 #include "libavutil/log.h"
  39 #include "libavcodec/avcodec.h"
  40 #include "libavformat/avformat.h"
  41
  42 #define AUDIO_BLOCK_SIZE 4096
  43
  44 typedef struct {
  45     int fd;
  46     int sample_rate;
  47     int channels;
  48     int frame_size; /* in bytes ! */
  49     int codec_id;
  50     unsigned int flip_left : 1;
  51     uint8_t buffer[AUDIO_BLOCK_SIZE];
  52     int buffer_ptr;
  53 } AudioData;
  54
  55 static int audio_open(AudioData *s, int is_output, const char *audio_device)
  56 {
  57     int audio_fd;
  58     int tmp, err;
  59     char *flip = getenv("AUDIO_FLIP_LEFT");
  60
  61     if (is_output)
  62         audio_fd = open(audio_device, O_WRONLY);
  63     else
  64         audio_fd = open(audio_device, O_RDONLY);
  65     if (audio_fd < 0) {
  66         av_log(NULL, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
  67         return AVERROR(EIO);
  68     }
  69
  70     if (flip && *flip == '1') {
  71         s->flip_left = 1;
  72     }
  73
  74     /* non blocking mode */
  75     if (!is_output)
  76         fcntl(audio_fd, F_SETFL, O_NONBLOCK);
  77
  78     s->frame_size = AUDIO_BLOCK_SIZE;
  79 #if 0
  80     tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
  81     err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
  82     if (err < 0) {
  83         perror("SNDCTL_DSP_SETFRAGMENT");
  84     }
  85 #endif
  86
  87     /* select format : favour native format */
  88     err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
  89
  90 #ifdef WORDS_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:
 110         s->codec_id = CODEC_ID_PCM_S16LE;
 111         break;
 112     case AFMT_S16_BE:
 113         s->codec_id = CODEC_ID_PCM_S16BE;
 114         break;
 115     default:
 116         av_log(NULL, 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(NULL, 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(NULL, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
 130         goto fail;
 131     }
 132     if (tmp)
 133         s->channels = 2;
 134
 135     tmp = s->sample_rate;
 136     err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
 137     if (err < 0) {
 138         av_log(NULL, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
 139         goto fail;
 140     }
 141     s->sample_rate = tmp; /* store real sample rate */
 142     s->fd = audio_fd;
 143
 144     return 0;
 145  fail:
 146     close(audio_fd);
 147     return AVERROR(EIO);
 148 }
 149
 150 static int audio_close(AudioData *s)
 151 {
 152     close(s->fd);
 153     return 0;
 154 }
 155
 156 /* sound output support */
 157 static int audio_write_header(AVFormatContext *s1)
 158 {
 159     AudioData *s = s1->priv_data;
 160     AVStream *st;
 161     int ret;
 162
 163     st = s1->streams[0];
 164     s->sample_rate = st->codec->sample_rate;
 165     s->channels = st->codec->channels;
 166     ret = audio_open(s, 1, s1->filename);
 167     if (ret < 0) {
 168         return AVERROR(EIO);
 169     } else {
 170         return 0;
 171     }
 172 }
 173
 174 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
 175 {
 176     AudioData *s = s1->priv_data;
 177     int len, ret;
 178     int size= pkt->size;
 179     uint8_t *buf= pkt->data;
 180
 181     while (size > 0) {
 182         len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
 183         if (len > size)
 184             len = size;
 185         memcpy(s->buffer + s->buffer_ptr, buf, len);
 186         s->buffer_ptr += len;
 187         if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
 188             for(;;) {
 189                 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
 190                 if (ret > 0)
 191                     break;
 192                 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
 193                     return AVERROR(EIO);
 194             }
 195             s->buffer_ptr = 0;
 196         }
 197         buf += len;
 198         size -= len;
 199     }
 200     return 0;
 201 }
 202
 203 static int audio_write_trailer(AVFormatContext *s1)
 204 {
 205     AudioData *s = s1->priv_data;
 206
 207     audio_close(s);
 208     return 0;
 209 }
 210
 211 /* grab support */
 212
 213 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
 214 {
 215     AudioData *s = s1->priv_data;
 216     AVStream *st;
 217     int ret;
 218
 219     if (ap->sample_rate <= 0 || ap->channels <= 0)
 220         return -1;
 221
 222     st = av_new_stream(s1, 0);
 223     if (!st) {
 224         return AVERROR(ENOMEM);
 225     }
 226     s->sample_rate = ap->sample_rate;
 227     s->channels = ap->channels;
 228
 229     ret = audio_open(s, 0, s1->filename);
 230     if (ret < 0) {
 231         av_free(st);
 232         return AVERROR(EIO);
 233     }
 234
 235     /* take real parameters */
 236     st->codec->codec_type = CODEC_TYPE_AUDIO;
 237     st->codec->codec_id = s->codec_id;
 238     st->codec->sample_rate = s->sample_rate;
 239     st->codec->channels = s->channels;
 240
 241     av_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
 242     return 0;
 243 }
 244
 245 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
 246 {
 247     AudioData *s = s1->priv_data;
 248     int ret, bdelay;
 249     int64_t cur_time;
 250     struct audio_buf_info abufi;
 251
 252     if (av_new_packet(pkt, s->frame_size) < 0)
 253         return AVERROR(EIO);
 254     for(;;) {
 255         struct timeval tv;
 256         fd_set fds;
 257
 258         tv.tv_sec = 0;
 259         tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */
 260
 261         FD_ZERO(&fds);
 262         FD_SET(s->fd, &fds);
 263
 264         /* This will block until data is available or we get a timeout */
 265         (void) select(s->fd + 1, &fds, 0, 0, &tv);
 266
 267         ret = read(s->fd, pkt->data, pkt->size);
 268         if (ret > 0)
 269             break;
 270         if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
 271             av_free_packet(pkt);
 272             pkt->size = 0;
 273             pkt->pts = av_gettime();
 274             return 0;
 275         }
 276         if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
 277             av_free_packet(pkt);
 278             return AVERROR(EIO);
 279         }
 280     }
 281     pkt->size = ret;
 282
 283     /* compute pts of the start of the packet */
 284     cur_time = av_gettime();
 285     bdelay = ret;
 286     if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
 287         bdelay += abufi.bytes;
 288     }
 289     /* subtract time represented by the number of bytes in the audio fifo */
 290     cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
 291
 292     /* convert to wanted units */
 293     pkt->pts = cur_time;
 294
 295     if (s->flip_left && s->channels == 2) {
 296         int i;
 297         short *p = (short *) pkt->data;
 298
 299         for (i = 0; i < ret; i += 4) {
 300             *p = ~*p;
 301             p += 2;
 302         }
 303     }
 304     return 0;
 305 }
 306
 307 static int audio_read_close(AVFormatContext *s1)
 308 {
 309     AudioData *s = s1->priv_data;
 310
 311     audio_close(s);
 312     return 0;
 313 }
 314
 315 #ifdef CONFIG_OSS_DEMUXER
 316 AVInputFormat oss_demuxer = {
 317     "oss",
 318     NULL_IF_CONFIG_SMALL("audio grab and output"),
 319     sizeof(AudioData),
 320     NULL,
 321     audio_read_header,
 322     audio_read_packet,
 323     audio_read_close,
 324     .flags = AVFMT_NOFILE,
 325 };
 326 #endif
 327
 328 #ifdef CONFIG_OSS_MUXER
 329 AVOutputFormat oss_muxer = {
 330     "oss",
 331     NULL_IF_CONFIG_SMALL("audio grab and output"),
 332     "",
 333     "",
 334     sizeof(AudioData),
 335     /* XXX: we make the assumption that the soundcard accepts this format */
 336     /* XXX: find better solution with "preinit" method, needed also in
 337        other formats */
 338 #ifdef WORDS_BIGENDIAN
 339     CODEC_ID_PCM_S16BE,
 340 #else
 341     CODEC_ID_PCM_S16LE,
 342 #endif
 343     CODEC_ID_NONE,
 344     audio_write_header,
 345     audio_write_packet,
 346     audio_write_trailer,
 347     .flags = AVFMT_NOFILE,
 348 };
 349 #endif