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 #include <sys/select.h>
  38
  39 #include "libavutil/log.h"
  40 #include "libavcodec/avcodec.h"
  41 #include "libavformat/avformat.h"
  42
  43 #define AUDIO_BLOCK_SIZE 4096
  44
  45 typedef struct {
  46     int fd;
  47     int sample_rate;
  48     int channels;
  49     int frame_size; /* in bytes ! */
  50     enum CodecID codec_id;
  51     unsigned int flip_left : 1;
  52     uint8_t buffer[AUDIO_BLOCK_SIZE];
  53     int buffer_ptr;
  54 } AudioData;
  55
  56 static int audio_open(AudioData *s, int is_output, const char *audio_device)
  57 {
  58     int audio_fd;
  59     int tmp, err;
  60     char *flip = getenv("AUDIO_FLIP_LEFT");
  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         av_log(NULL, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
  68         return AVERROR(EIO);
  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(EIO);
 120     }
 121     err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
 122     if (err < 0) {
 123         av_log(NULL, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
 124         goto fail;
 125     }
 126
 127     tmp = (s->channels == 2);
 128     err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
 129     if (err < 0) {
 130         av_log(NULL, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
 131         goto fail;
 132     }
 133
 134     tmp = s->sample_rate;
 135     err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
 136     if (err < 0) {
 137         av_log(NULL, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
 138         goto fail;
 139     }
 140     s->sample_rate = tmp; /* store real sample rate */
 141     s->fd = audio_fd;
 142
 143     return 0;
 144  fail:
 145     close(audio_fd);
 146     return AVERROR(EIO);
 147 }
 148
 149 static int audio_close(AudioData *s)
 150 {
 151     close(s->fd);
 152     return 0;
 153 }
 154
 155 /* sound output support */
 156 static int audio_write_header(AVFormatContext *s1)
 157 {
 158     AudioData *s = s1->priv_data;
 159     AVStream *st;
 160     int ret;
 161
 162     st = s1->streams[0];
 163     s->sample_rate = st->codec->sample_rate;
 164     s->channels = st->codec->channels;
 165     ret = audio_open(s, 1, s1->filename);
 166     if (ret < 0) {
 167         return AVERROR(EIO);
 168     } else {
 169         return 0;
 170     }
 171 }
 172
 173 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
 174 {
 175     AudioData *s = s1->priv_data;
 176     int len, ret;
 177     int size= pkt->size;
 178     uint8_t *buf= pkt->data;
 179
 180     while (size > 0) {
 181         len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
 182         if (len > size)
 183             len = size;
 184         memcpy(s->buffer + s->buffer_ptr, buf, len);
 185         s->buffer_ptr += len;
 186         if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
 187             for(;;) {
 188                 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
 189                 if (ret > 0)
 190                     break;
 191                 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
 192                     return AVERROR(EIO);
 193             }
 194             s->buffer_ptr = 0;
 195         }
 196         buf += len;
 197         size -= len;
 198     }
 199     return 0;
 200 }
 201
 202 static int audio_write_trailer(AVFormatContext *s1)
 203 {
 204     AudioData *s = s1->priv_data;
 205
 206     audio_close(s);
 207     return 0;
 208 }
 209
 210 /* grab support */
 211
 212 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
 213 {
 214     AudioData *s = s1->priv_data;
 215     AVStream *st;
 216     int ret;
 217
 218     if (ap->sample_rate <= 0 || ap->channels <= 0)
 219         return -1;
 220
 221     st = av_new_stream(s1, 0);
 222     if (!st) {
 223         return AVERROR(ENOMEM);
 224     }
 225     s->sample_rate = ap->sample_rate;
 226     s->channels = ap->channels;
 227
 228     ret = audio_open(s, 0, s1->filename);
 229     if (ret < 0) {
 230         av_free(st);
 231         return AVERROR(EIO);
 232     }
 233
 234     /* take real parameters */
 235     st->codec->codec_type = CODEC_TYPE_AUDIO;
 236     st->codec->codec_id = s->codec_id;
 237     st->codec->sample_rate = s->sample_rate;
 238     st->codec->channels = s->channels;
 239
 240     av_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
 241     return 0;
 242 }
 243
 244 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
 245 {
 246     AudioData *s = s1->priv_data;
 247     int ret, bdelay;
 248     int64_t cur_time;
 249     struct audio_buf_info abufi;
 250
 251     if (av_new_packet(pkt, s->frame_size) < 0)
 252         return AVERROR(EIO);
 253     for(;;) {
 254         struct timeval tv;
 255         fd_set fds;
 256
 257         tv.tv_sec = 0;
 258         tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */
 259
 260         FD_ZERO(&fds);
 261         FD_SET(s->fd, &fds);
 262
 263         /* This will block until data is available or we get a timeout */
 264         (void) select(s->fd + 1, &fds, 0, 0, &tv);
 265
 266         ret = read(s->fd, pkt->data, pkt->size);
 267         if (ret > 0)
 268             break;
 269         if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
 270             av_free_packet(pkt);
 271             pkt->size = 0;
 272             pkt->pts = av_gettime();
 273             return 0;
 274         }
 275         if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
 276             av_free_packet(pkt);
 277             return AVERROR(EIO);
 278         }
 279     }
 280     pkt->size = ret;
 281
 282     /* compute pts of the start of the packet */
 283     cur_time = av_gettime();
 284     bdelay = ret;
 285     if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
 286         bdelay += abufi.bytes;
 287     }
 288     /* subtract time represented by the number of bytes in the audio fifo */
 289     cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
 290
 291     /* convert to wanted units */
 292     pkt->pts = cur_time;
 293
 294     if (s->flip_left && s->channels == 2) {
 295         int i;
 296         short *p = (short *) pkt->data;
 297
 298         for (i = 0; i < ret; i += 4) {
 299             *p = ~*p;
 300             p += 2;
 301         }
 302     }
 303     return 0;
 304 }
 305
 306 static int audio_read_close(AVFormatContext *s1)
 307 {
 308     AudioData *s = s1->priv_data;
 309
 310     audio_close(s);
 311     return 0;
 312 }
 313
 314 #ifdef CONFIG_OSS_DEMUXER
 315 AVInputFormat oss_demuxer = {
 316     "oss",
 317     NULL_IF_CONFIG_SMALL("Open Sound System capture"),
 318     sizeof(AudioData),
 319     NULL,
 320     audio_read_header,
 321     audio_read_packet,
 322     audio_read_close,
 323     .flags = AVFMT_NOFILE,
 324 };
 325 #endif
 326
 327 #ifdef CONFIG_OSS_MUXER
 328 AVOutputFormat oss_muxer = {
 329     "oss",
 330     NULL_IF_CONFIG_SMALL("Open Sound System playback"),
 331     "",
 332     "",
 333     sizeof(AudioData),
 334     /* XXX: we make the assumption that the soundcard accepts this format */
 335     /* XXX: find better solution with "preinit" method, needed also in
 336        other formats */
 337 #ifdef WORDS_BIGENDIAN
 338     CODEC_ID_PCM_S16BE,
 339 #else
 340     CODEC_ID_PCM_S16LE,
 341 #endif
 342     CODEC_ID_NONE,
 343     audio_write_header,
 344     audio_write_packet,
 345     audio_write_trailer,
 346     .flags = AVFMT_NOFILE,
 347 };
 348 #endif