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
 133     tmp = s->sample_rate;
 134     err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
 135     if (err < 0) {
 136         av_log(NULL, 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 */
 155 static int audio_write_header(AVFormatContext *s1)
 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(s, 1, s1->filename);
 165     if (ret < 0) {
 166         return AVERROR(EIO);
 167     } else {
 168         return 0;
 169     }
 170 }
 171
 172 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
 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 = AUDIO_BLOCK_SIZE - s->buffer_ptr;
 181         if (len > size)
 182             len = size;
 183         memcpy(s->buffer + s->buffer_ptr, buf, len);
 184         s->buffer_ptr += len;
 185         if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
 186             for(;;) {
 187                 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
 188                 if (ret > 0)
 189                     break;
 190                 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
 191                     return AVERROR(EIO);
 192             }
 193             s->buffer_ptr = 0;
 194         }
 195         buf += len;
 196         size -= len;
 197     }
 198     return 0;
 199 }
 200
 201 static int audio_write_trailer(AVFormatContext *s1)
 202 {
 203     AudioData *s = s1->priv_data;
 204
 205     audio_close(s);
 206     return 0;
 207 }
 208
 209 /* grab support */
 210
 211 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
 212 {
 213     AudioData *s = s1->priv_data;
 214     AVStream *st;
 215     int ret;
 216
 217     if (ap->sample_rate <= 0 || ap->channels <= 0)
 218         return -1;
 219
 220     st = av_new_stream(s1, 0);
 221     if (!st) {
 222         return AVERROR(ENOMEM);
 223     }
 224     s->sample_rate = ap->sample_rate;
 225     s->channels = ap->channels;
 226
 227     ret = audio_open(s, 0, s1->filename);
 228     if (ret < 0) {
 229         av_free(st);
 230         return AVERROR(EIO);
 231     }
 232
 233     /* take real parameters */
 234     st->codec->codec_type = CODEC_TYPE_AUDIO;
 235     st->codec->codec_id = s->codec_id;
 236     st->codec->sample_rate = s->sample_rate;
 237     st->codec->channels = s->channels;
 238
 239     av_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
 240     return 0;
 241 }
 242
 243 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
 244 {
 245     AudioData *s = s1->priv_data;
 246     int ret, bdelay;
 247     int64_t cur_time;
 248     struct audio_buf_info abufi;
 249
 250     if (av_new_packet(pkt, s->frame_size) < 0)
 251         return AVERROR(EIO);
 252     for(;;) {
 253         struct timeval tv;
 254         fd_set fds;
 255
 256         tv.tv_sec = 0;
 257         tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */
 258
 259         FD_ZERO(&fds);
 260         FD_SET(s->fd, &fds);
 261
 262         /* This will block until data is available or we get a timeout */
 263         (void) select(s->fd + 1, &fds, 0, 0, &tv);
 264
 265         ret = read(s->fd, pkt->data, pkt->size);
 266         if (ret > 0)
 267             break;
 268         if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
 269             av_free_packet(pkt);
 270             pkt->size = 0;
 271             pkt->pts = av_gettime();
 272             return 0;
 273         }
 274         if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
 275             av_free_packet(pkt);
 276             return AVERROR(EIO);
 277         }
 278     }
 279     pkt->size = ret;
 280
 281     /* compute pts of the start of the packet */
 282     cur_time = av_gettime();
 283     bdelay = ret;
 284     if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
 285         bdelay += abufi.bytes;
 286     }
 287     /* subtract time represented by the number of bytes in the audio fifo */
 288     cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
 289
 290     /* convert to wanted units */
 291     pkt->pts = cur_time;
 292
 293     if (s->flip_left && s->channels == 2) {
 294         int i;
 295         short *p = (short *) pkt->data;
 296
 297         for (i = 0; i < ret; i += 4) {
 298             *p = ~*p;
 299             p += 2;
 300         }
 301     }
 302     return 0;
 303 }
 304
 305 static int audio_read_close(AVFormatContext *s1)
 306 {
 307     AudioData *s = s1->priv_data;
 308
 309     audio_close(s);
 310     return 0;
 311 }
 312
 313 #ifdef CONFIG_OSS_DEMUXER
 314 AVInputFormat oss_demuxer = {
 315     "oss",
 316     NULL_IF_CONFIG_SMALL("Open Sound System capture"),
 317     sizeof(AudioData),
 318     NULL,
 319     audio_read_header,
 320     audio_read_packet,
 321     audio_read_close,
 322     .flags = AVFMT_NOFILE,
 323 };
 324 #endif
 325
 326 #ifdef CONFIG_OSS_MUXER
 327 AVOutputFormat oss_muxer = {
 328     "oss",
 329     NULL_IF_CONFIG_SMALL("Open Sound System playback"),
 330     "",
 331     "",
 332     sizeof(AudioData),
 333     /* XXX: we make the assumption that the soundcard accepts this format */
 334     /* XXX: find better solution with "preinit" method, needed also in
 335        other formats */
 336 #ifdef WORDS_BIGENDIAN
 337     CODEC_ID_PCM_S16BE,
 338 #else
 339     CODEC_ID_PCM_S16LE,
 340 #endif
 341     CODEC_ID_NONE,
 342     audio_write_header,
 343     audio_write_packet,
 344     audio_write_trailer,
 345     .flags = AVFMT_NOFILE,
 346 };
 347 #endif