release/src/router/ffmpeg/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 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 #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 #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(AVFormatContext *s1, int is_output, const char *audio_device)
  57 {
  58     AudioData *s = s1->priv_data;
  59     int audio_fd;
  60     int tmp, err;
  61     char *flip = getenv("AUDIO_FLIP_LEFT");
  62
  63     if (is_output)
  64         audio_fd = open(audio_device, O_WRONLY);
  65     else
  66         audio_fd = open(audio_device, O_RDONLY);
  67     if (audio_fd < 0) {
  68         av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
  69         return AVERROR(EIO);
  70     }
  71
  72     if (flip && *flip == '1') {
  73         s->flip_left = 1;
  74     }
  75
  76     /* non blocking mode */
  77     if (!is_output)
  78         fcntl(audio_fd, F_SETFL, O_NONBLOCK);
  79
  80     s->frame_size = AUDIO_BLOCK_SIZE;
  81 #if 0
  82     tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
  83     err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
  84     if (err < 0) {
  85         perror("SNDCTL_DSP_SETFRAGMENT");
  86     }
  87 #endif
  88
  89     /* select format : favour native format */
  90     err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
  91
  92 #if HAVE_BIGENDIAN
  93     if (tmp & AFMT_S16_BE) {
  94         tmp = AFMT_S16_BE;
  95     } else if (tmp & AFMT_S16_LE) {
  96         tmp = AFMT_S16_LE;
  97     } else {
  98         tmp = 0;
  99     }
 100 #else
 101     if (tmp & AFMT_S16_LE) {
 102         tmp = AFMT_S16_LE;
 103     } else if (tmp & AFMT_S16_BE) {
 104         tmp = AFMT_S16_BE;
 105     } else {
 106         tmp = 0;
 107     }
 108 #endif
 109
 110     switch(tmp) {
 111     case AFMT_S16_LE:
 112         s->codec_id = CODEC_ID_PCM_S16LE;
 113         break;
 114     case AFMT_S16_BE:
 115         s->codec_id = CODEC_ID_PCM_S16BE;
 116         break;
 117     default:
 118         av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
 119         close(audio_fd);
 120         return AVERROR(EIO);
 121     }
 122     err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
 123     if (err < 0) {
 124         av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
 125         goto fail;
 126     }
 127
 128     tmp = (s->channels == 2);
 129     err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
 130     if (err < 0) {
 131         av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
 132         goto fail;
 133     }
 134
 135     tmp = s->sample_rate;
 136     err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
 137     if (err < 0) {
 138         av_log(s1, 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(s1, 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(s1, 0, s1->filename);
 230     if (ret < 0) {
 231         return AVERROR(EIO);
 232     }
 233
 234     /* take real parameters */
 235     st->codec->codec_type = AVMEDIA_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 ((ret=av_new_packet(pkt, s->frame_size)) < 0)
 252         return ret;
 253
 254     ret = read(s->fd, pkt->data, pkt->size);
 255     if (ret <= 0){
 256         av_free_packet(pkt);
 257         pkt->size = 0;
 258         if (ret<0)  return AVERROR(errno);
 259         else        return AVERROR_EOF;
 260     }
 261     pkt->size = ret;
 262
 263     /* compute pts of the start of the packet */
 264     cur_time = av_gettime();
 265     bdelay = ret;
 266     if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
 267         bdelay += abufi.bytes;
 268     }
 269     /* subtract time represented by the number of bytes in the audio fifo */
 270     cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
 271
 272     /* convert to wanted units */
 273     pkt->pts = cur_time;
 274
 275     if (s->flip_left && s->channels == 2) {
 276         int i;
 277         short *p = (short *) pkt->data;
 278
 279         for (i = 0; i < ret; i += 4) {
 280             *p = ~*p;
 281             p += 2;
 282         }
 283     }
 284     return 0;
 285 }
 286
 287 static int audio_read_close(AVFormatContext *s1)
 288 {
 289     AudioData *s = s1->priv_data;
 290
 291     audio_close(s);
 292     return 0;
 293 }
 294
 295 #if CONFIG_OSS_INDEV
 296 AVInputFormat oss_demuxer = {
 297     "oss",
 298     NULL_IF_CONFIG_SMALL("Open Sound System capture"),
 299     sizeof(AudioData),
 300     NULL,
 301     audio_read_header,
 302     audio_read_packet,
 303     audio_read_close,
 304     .flags = AVFMT_NOFILE,
 305 };
 306 #endif
 307
 308 #if CONFIG_OSS_OUTDEV
 309 AVOutputFormat oss_muxer = {
 310     "oss",
 311     NULL_IF_CONFIG_SMALL("Open Sound System playback"),
 312     "",
 313     "",
 314     sizeof(AudioData),
 315     /* XXX: we make the assumption that the soundcard accepts this format */
 316     /* XXX: find better solution with "preinit" method, needed also in
 317        other formats */
 318 #if HAVE_BIGENDIAN
 319     CODEC_ID_PCM_S16BE,
 320 #else
 321     CODEC_ID_PCM_S16LE,
 322 #endif
 323     CODEC_ID_NONE,
 324     audio_write_header,
 325     audio_write_packet,
 326     audio_write_trailer,
 327     .flags = AVFMT_NOFILE,
 328 };
 329 #endif