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  1. /*
  2.  * Linux audio play 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.  
  24. #include <stdint.h>
  25.  
  26. #if HAVE_SOUNDCARD_H
  27. #include <soundcard.h>
  28. #else
  29. #include <sys/soundcard.h>
  30. #endif
  31.  
  32. #if HAVE_UNISTD_H
  33. #include <unistd.h>
  34. #endif
  35. #include <fcntl.h>
  36. #include <sys/ioctl.h>
  37.  
  38. #include "libavutil/internal.h"
  39. #include "libavutil/opt.h"
  40. #include "libavutil/time.h"
  41.  
  42. #include "libavcodec/avcodec.h"
  43.  
  44. #include "avdevice.h"
  45. #include "libavformat/internal.h"
  46.  
  47. #include "oss.h"
  48.  
  49. static int audio_read_header(AVFormatContext *s1)
  50. {
  51.     OSSAudioData *s = s1->priv_data;
  52.     AVStream *st;
  53.     int ret;
  54.  
  55.     st = avformat_new_stream(s1, NULL);
  56.     if (!st) {
  57.         return AVERROR(ENOMEM);
  58.     }
  59.  
  60.     ret = ff_oss_audio_open(s1, 0, s1->filename);
  61.     if (ret < 0) {
  62.         return AVERROR(EIO);
  63.     }
  64.  
  65.     /* take real parameters */
  66.     st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
  67.     st->codec->codec_id = s->codec_id;
  68.     st->codec->sample_rate = s->sample_rate;
  69.     st->codec->channels = s->channels;
  70.  
  71.     avpriv_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
  72.     return 0;
  73. }
  74.  
  75. static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
  76. {
  77.     OSSAudioData *s = s1->priv_data;
  78.     int ret, bdelay;
  79.     int64_t cur_time;
  80.     struct audio_buf_info abufi;
  81.  
  82.     if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
  83.         return ret;
  84.  
  85.     ret = read(s->fd, pkt->data, pkt->size);
  86.     if (ret <= 0){
  87.         av_free_packet(pkt);
  88.         pkt->size = 0;
  89.         if (ret<0)  return AVERROR(errno);
  90.         else        return AVERROR_EOF;
  91.     }
  92.     pkt->size = ret;
  93.  
  94.     /* compute pts of the start of the packet */
  95.     cur_time = av_gettime();
  96.     bdelay = ret;
  97.     if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
  98.         bdelay += abufi.bytes;
  99.     }
  100.     /* subtract time represented by the number of bytes in the audio fifo */
  101.     cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
  102.  
  103.     /* convert to wanted units */
  104.     pkt->pts = cur_time;
  105.  
  106.     if (s->flip_left && s->channels == 2) {
  107.         int i;
  108.         short *p = (short *) pkt->data;
  109.  
  110.         for (i = 0; i < ret; i += 4) {
  111.             *p = ~*p;
  112.             p += 2;
  113.         }
  114.     }
  115.     return 0;
  116. }
  117.  
  118. static int audio_read_close(AVFormatContext *s1)
  119. {
  120.     OSSAudioData *s = s1->priv_data;
  121.  
  122.     ff_oss_audio_close(s);
  123.     return 0;
  124. }
  125.  
  126. static const AVOption options[] = {
  127.     { "sample_rate", "", offsetof(OSSAudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
  128.     { "channels",    "", offsetof(OSSAudioData, channels),    AV_OPT_TYPE_INT, {.i64 = 2},     1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
  129.     { NULL },
  130. };
  131.  
  132. static const AVClass oss_demuxer_class = {
  133.     .class_name     = "OSS demuxer",
  134.     .item_name      = av_default_item_name,
  135.     .option         = options,
  136.     .version        = LIBAVUTIL_VERSION_INT,
  137.     .category       = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
  138. };
  139.  
  140. AVInputFormat ff_oss_demuxer = {
  141.     .name           = "oss",
  142.     .long_name      = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"),
  143.     .priv_data_size = sizeof(OSSAudioData),
  144.     .read_header    = audio_read_header,
  145.     .read_packet    = audio_read_packet,
  146.     .read_close     = audio_read_close,
  147.     .flags          = AVFMT_NOFILE,
  148.     .priv_class     = &oss_demuxer_class,
  149. };
  150.