| 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_audio.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 | }; |