libavdevice/oss_audio.c

   1 /*
   2  * Linux audio play and grab interface
   3  * Copyright (c) 2000, 2001 Fabrice Bellard
   4  *
   5  * This file is part of Libav.
   6  *
   7  * Libav 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  * Libav 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 Libav; 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 #if 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
  37 #include "libavutil/log.h"
  38 #include "libavutil/opt.h"
  39 #include "libavutil/time.h"
  40 #include "libavcodec/avcodec.h"
  41 #include "libavformat/avformat.h"
  42 #include "libavformat/internal.h"
  43
  44 #define AUDIO_BLOCK_SIZE 4096
  45
  46 typedef struct {
  47     AVClass *class;
  48     int fd;
  49     int sample_rate;
  50     int channels;
  51     int frame_size; /* in bytes ! */
  52     enum AVCodecID codec_id;
  53     unsigned int flip_left : 1;
  54     uint8_t buffer[AUDIO_BLOCK_SIZE];
  55     int buffer_ptr;
  56 } AudioData;
  57
  58 static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device)
  59 {
  60     AudioData *s = s1->priv_data;
  61     int audio_fd;
  62     int tmp, err;
  63     char *flip = getenv("AUDIO_FLIP_LEFT");
  64
  65     if (is_output)
  66         audio_fd = open(audio_device, O_WRONLY);
  67     else
  68         audio_fd = open(audio_device, O_RDONLY);
  69     if (audio_fd < 0) {
  70         av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
  71         return AVERROR(EIO);
  72     }
  73
  74     if (flip && *flip == '1') {
  75         s->flip_left = 1;
  76     }
  77
  78     /* non blocking mode */
  79     if (!is_output)
  80         fcntl(audio_fd, F_SETFL, O_NONBLOCK);
  81
  82     s->frame_size = AUDIO_BLOCK_SIZE;
  83
  84     /* select format : favour native format */
  85     err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
  86
  87 #if HAVE_BIGENDIAN
  88     if (tmp & AFMT_S16_BE) {
  89         tmp = AFMT_S16_BE;
  90     } else if (tmp & AFMT_S16_LE) {
  91         tmp = AFMT_S16_LE;
  92     } else {
  93         tmp = 0;
  94     }
  95 #else
  96     if (tmp & AFMT_S16_LE) {
  97         tmp = AFMT_S16_LE;
  98     } else if (tmp & AFMT_S16_BE) {
  99         tmp = AFMT_S16_BE;
 100     } else {
 101         tmp = 0;
 102     }
 103 #endif
 104
 105     switch(tmp) {
 106     case AFMT_S16_LE:
 107         s->codec_id = AV_CODEC_ID_PCM_S16LE;
 108         break;
 109     case AFMT_S16_BE:
 110         s->codec_id = AV_CODEC_ID_PCM_S16BE;
 111         break;
 112     default:
 113         av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
 114         close(audio_fd);
 115         return AVERROR(EIO);
 116     }
 117     err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
 118     if (err < 0) {
 119         av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
 120         goto fail;
 121     }
 122
 123     tmp = (s->channels == 2);
 124     err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
 125     if (err < 0) {
 126         av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
 127         goto fail;
 128     }
 129
 130     tmp = s->sample_rate;
 131     err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
 132     if (err < 0) {
 133         av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
 134         goto fail;
 135     }
 136     s->sample_rate = tmp; /* store real sample rate */
 137     s->fd = audio_fd;
 138
 139     return 0;
 140  fail:
 141     close(audio_fd);
 142     return AVERROR(EIO);
 143 }
 144
 145 static int audio_close(AudioData *s)
 146 {
 147     close(s->fd);
 148     return 0;
 149 }
 150
 151 /* sound output support */
 152 static int audio_write_header(AVFormatContext *s1)
 153 {
 154     AudioData *s = s1->priv_data;
 155     AVStream *st;
 156     int ret;
 157
 158     st = s1->streams[0];
 159     s->sample_rate = st->codec->sample_rate;
 160     s->channels = st->codec->channels;
 161     ret = audio_open(s1, 1, s1->filename);
 162     if (ret < 0) {
 163         return AVERROR(EIO);
 164     } else {
 165         return 0;
 166     }
 167 }
 168
 169 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
 170 {
 171     AudioData *s = s1->priv_data;
 172     int len, ret;
 173     int size= pkt->size;
 174     uint8_t *buf= pkt->data;
 175
 176     while (size > 0) {
 177         len = FFMIN(AUDIO_BLOCK_SIZE - s->buffer_ptr, 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(EIO);
 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)
 207 {
 208     AudioData *s = s1->priv_data;
 209     AVStream *st;
 210     int ret;
 211
 212     st = avformat_new_stream(s1, NULL);
 213     if (!st) {
 214         return AVERROR(ENOMEM);
 215     }
 216
 217     ret = audio_open(s1, 0, s1->filename);
 218     if (ret < 0) {
 219         return AVERROR(EIO);
 220     }
 221
 222     /* take real parameters */
 223     st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
 224     st->codec->codec_id = s->codec_id;
 225     st->codec->sample_rate = s->sample_rate;
 226     st->codec->channels = s->channels;
 227
 228     avpriv_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
 229     return 0;
 230 }
 231
 232 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
 233 {
 234     AudioData *s = s1->priv_data;
 235     int ret, bdelay;
 236     int64_t cur_time;
 237     struct audio_buf_info abufi;
 238
 239     if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
 240         return ret;
 241
 242     ret = read(s->fd, pkt->data, pkt->size);
 243     if (ret <= 0){
 244         av_free_packet(pkt);
 245         pkt->size = 0;
 246         if (ret<0)  return AVERROR(errno);
 247         else        return AVERROR_EOF;
 248     }
 249     pkt->size = ret;
 250
 251     /* compute pts of the start of the packet */
 252     cur_time = av_gettime();
 253     bdelay = ret;
 254     if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
 255         bdelay += abufi.bytes;
 256     }
 257     /* subtract time represented by the number of bytes in the audio fifo */
 258     cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
 259
 260     /* convert to wanted units */
 261     pkt->pts = cur_time;
 262
 263     if (s->flip_left && s->channels == 2) {
 264         int i;
 265         short *p = (short *) pkt->data;
 266
 267         for (i = 0; i < ret; i += 4) {
 268             *p = ~*p;
 269             p += 2;
 270         }
 271     }
 272     return 0;
 273 }
 274
 275 static int audio_read_close(AVFormatContext *s1)
 276 {
 277     AudioData *s = s1->priv_data;
 278
 279     audio_close(s);
 280     return 0;
 281 }
 282
 283 #if CONFIG_OSS_INDEV
 284 static const AVOption options[] = {
 285     { "sample_rate", "", offsetof(AudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
 286     { "channels",    "", offsetof(AudioData, channels),    AV_OPT_TYPE_INT, {.i64 = 2},     1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
 287     { NULL },
 288 };
 289
 290 static const AVClass oss_demuxer_class = {
 291     .class_name     = "OSS demuxer",
 292     .item_name      = av_default_item_name,
 293     .option         = options,
 294     .version        = LIBAVUTIL_VERSION_INT,
 295 };
 296
 297 AVInputFormat ff_oss_demuxer = {
 298     .name           = "oss",
 299     .long_name      = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
 300     .priv_data_size = sizeof(AudioData),
 301     .read_header    = audio_read_header,
 302     .read_packet    = audio_read_packet,
 303     .read_close     = audio_read_close,
 304     .flags          = AVFMT_NOFILE,
 305     .priv_class     = &oss_demuxer_class,
 306 };
 307 #endif
 308
 309 #if CONFIG_OSS_OUTDEV
 310 AVOutputFormat ff_oss_muxer = {
 311     .name           = "oss",
 312     .long_name      = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) playback"),
 313     .priv_data_size = sizeof(AudioData),
 314     /* XXX: we make the assumption that the soundcard accepts this format */
 315     /* XXX: find better solution with "preinit" method, needed also in
 316        other formats */
 317     .audio_codec    = AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE),
 318     .video_codec    = AV_CODEC_ID_NONE,
 319     .write_header   = audio_write_header,
 320     .write_packet   = audio_write_packet,
 321     .write_trailer  = audio_write_trailer,
 322     .flags          = AVFMT_NOFILE,
 323 };
 324 #endif