libavdevice/oss_dec.c
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 /*
  * Linux audio play interface
  * Copyright (c) 2000, 2001 Fabrice Bellard
  *
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  * This file is part of FFmpeg.
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  *
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  * FFmpeg is free software; you can redistribute it and/or
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  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
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  * FFmpeg is distributed in the hope that it will be useful,
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  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
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  * License along with FFmpeg; if not, write to the Free Software
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  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #include "config.h"
 
 #include <stdint.h>
 
 #if HAVE_SOUNDCARD_H
 #include <soundcard.h>
 #else
 #include <sys/soundcard.h>
 #endif
 
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 #if HAVE_UNISTD_H
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 #include <unistd.h>
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 #endif
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 #include <fcntl.h>
 #include <sys/ioctl.h>
 
 #include "libavutil/internal.h"
 #include "libavutil/opt.h"
 #include "libavutil/time.h"
 
 #include "libavcodec/avcodec.h"
 
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 #include "avdevice.h"
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 #include "libavformat/internal.h"
 
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 #include "oss.h"
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 static int audio_read_header(AVFormatContext *s1)
 {
     OSSAudioData *s = s1->priv_data;
     AVStream *st;
     int ret;
 
     st = avformat_new_stream(s1, NULL);
     if (!st) {
         return AVERROR(ENOMEM);
     }
 
     ret = ff_oss_audio_open(s1, 0, s1->filename);
     if (ret < 0) {
         return AVERROR(EIO);
     }
 
     /* take real parameters */
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     st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
     st->codecpar->codec_id = s->codec_id;
     st->codecpar->sample_rate = s->sample_rate;
     st->codecpar->channels = s->channels;
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     avpriv_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
     return 0;
 }
 
 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
 {
     OSSAudioData *s = s1->priv_data;
     int ret, bdelay;
     int64_t cur_time;
     struct audio_buf_info abufi;
 
     if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
         return ret;
 
     ret = read(s->fd, pkt->data, pkt->size);
     if (ret <= 0){
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         av_packet_unref(pkt);
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         pkt->size = 0;
         if (ret<0)  return AVERROR(errno);
         else        return AVERROR_EOF;
     }
     pkt->size = ret;
 
     /* compute pts of the start of the packet */
     cur_time = av_gettime();
     bdelay = ret;
     if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
         bdelay += abufi.bytes;
     }
     /* subtract time represented by the number of bytes in the audio fifo */
     cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
 
     /* convert to wanted units */
     pkt->pts = cur_time;
 
     if (s->flip_left && s->channels == 2) {
         int i;
         short *p = (short *) pkt->data;
 
         for (i = 0; i < ret; i += 4) {
             *p = ~*p;
             p += 2;
         }
     }
     return 0;
 }
 
 static int audio_read_close(AVFormatContext *s1)
 {
     OSSAudioData *s = s1->priv_data;
 
     ff_oss_audio_close(s);
     return 0;
 }
 
 static const AVOption options[] = {
     { "sample_rate", "", offsetof(OSSAudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
     { "channels",    "", offsetof(OSSAudioData, channels),    AV_OPT_TYPE_INT, {.i64 = 2},     1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
     { NULL },
 };
 
 static const AVClass oss_demuxer_class = {
     .class_name     = "OSS demuxer",
     .item_name      = av_default_item_name,
     .option         = options,
     .version        = LIBAVUTIL_VERSION_INT,
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     .category       = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
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 };
 
 AVInputFormat ff_oss_demuxer = {
     .name           = "oss",
     .long_name      = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
     .priv_data_size = sizeof(OSSAudioData),
     .read_header    = audio_read_header,
     .read_packet    = audio_read_packet,
     .read_close     = audio_read_close,
     .flags          = AVFMT_NOFILE,
     .priv_class     = &oss_demuxer_class,
 };