1/* 2 * RAW PCM demuxers 3 * Copyright (c) 2002 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 "avformat.h" 23#include "internal.h" 24#include "pcm.h" 25#include "libavutil/log.h" 26#include "libavutil/opt.h" 27#include "libavutil/avassert.h" 28 29typedef struct PCMAudioDemuxerContext { 30 AVClass *class; 31 int sample_rate; 32 int channels; 33} PCMAudioDemuxerContext; 34 35static int pcm_read_header(AVFormatContext *s) 36{ 37 PCMAudioDemuxerContext *s1 = s->priv_data; 38 AVStream *st; 39 40 st = avformat_new_stream(s, NULL); 41 if (!st) 42 return AVERROR(ENOMEM); 43 44 45 st->codec->codec_type = AVMEDIA_TYPE_AUDIO; 46 st->codec->codec_id = s->iformat->raw_codec_id; 47 st->codec->sample_rate = s1->sample_rate; 48 st->codec->channels = s1->channels; 49 50 st->codec->bits_per_coded_sample = 51 av_get_bits_per_sample(st->codec->codec_id); 52 53 av_assert0(st->codec->bits_per_coded_sample > 0); 54 55 st->codec->block_align = 56 st->codec->bits_per_coded_sample * st->codec->channels / 8; 57 58 avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate); 59 return 0; 60} 61 62static const AVOption pcm_options[] = { 63 { "sample_rate", "", offsetof(PCMAudioDemuxerContext, sample_rate), AV_OPT_TYPE_INT, {.i64 = 44100}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, 64 { "channels", "", offsetof(PCMAudioDemuxerContext, channels), AV_OPT_TYPE_INT, {.i64 = 1}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, 65 { NULL }, 66}; 67 68#define PCMDEF(name_, long_name_, ext, codec) \ 69static const AVClass name_ ## _demuxer_class = { \ 70 .class_name = #name_ " demuxer", \ 71 .item_name = av_default_item_name, \ 72 .option = pcm_options, \ 73 .version = LIBAVUTIL_VERSION_INT, \ 74}; \ 75AVInputFormat ff_pcm_ ## name_ ## _demuxer = { \ 76 .name = #name_, \ 77 .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ 78 .priv_data_size = sizeof(PCMAudioDemuxerContext), \ 79 .read_header = pcm_read_header, \ 80 .read_packet = ff_pcm_read_packet, \ 81 .read_seek = ff_pcm_read_seek, \ 82 .flags = AVFMT_GENERIC_INDEX, \ 83 .extensions = ext, \ 84 .raw_codec_id = codec, \ 85 .priv_class = &name_ ## _demuxer_class, \ 86}; 87 88PCMDEF(f64be, "PCM 64-bit floating-point big-endian", 89 NULL, AV_CODEC_ID_PCM_F64BE) 90 91PCMDEF(f64le, "PCM 64-bit floating-point little-endian", 92 NULL, AV_CODEC_ID_PCM_F64LE) 93 94PCMDEF(f32be, "PCM 32-bit floating-point big-endian", 95 NULL, AV_CODEC_ID_PCM_F32BE) 96 97PCMDEF(f32le, "PCM 32-bit floating-point little-endian", 98 NULL, AV_CODEC_ID_PCM_F32LE) 99 100PCMDEF(s32be, "PCM signed 32-bit big-endian", 101 NULL, AV_CODEC_ID_PCM_S32BE) 102 103PCMDEF(s32le, "PCM signed 32-bit little-endian", 104 NULL, AV_CODEC_ID_PCM_S32LE) 105 106PCMDEF(s24be, "PCM signed 24-bit big-endian", 107 NULL, AV_CODEC_ID_PCM_S24BE) 108 109PCMDEF(s24le, "PCM signed 24-bit little-endian", 110 NULL, AV_CODEC_ID_PCM_S24LE) 111 112PCMDEF(s16be, "PCM signed 16-bit big-endian", 113 AV_NE("sw", NULL), AV_CODEC_ID_PCM_S16BE) 114 115PCMDEF(s16le, "PCM signed 16-bit little-endian", 116 AV_NE(NULL, "sw"), AV_CODEC_ID_PCM_S16LE) 117 118PCMDEF(s8, "PCM signed 8-bit", 119 "sb", AV_CODEC_ID_PCM_S8) 120 121PCMDEF(u32be, "PCM unsigned 32-bit big-endian", 122 NULL, AV_CODEC_ID_PCM_U32BE) 123 124PCMDEF(u32le, "PCM unsigned 32-bit little-endian", 125 NULL, AV_CODEC_ID_PCM_U32LE) 126 127PCMDEF(u24be, "PCM unsigned 24-bit big-endian", 128 NULL, AV_CODEC_ID_PCM_U24BE) 129 130PCMDEF(u24le, "PCM unsigned 24-bit little-endian", 131 NULL, AV_CODEC_ID_PCM_U24LE) 132 133PCMDEF(u16be, "PCM unsigned 16-bit big-endian", 134 AV_NE("uw", NULL), AV_CODEC_ID_PCM_U16BE) 135 136PCMDEF(u16le, "PCM unsigned 16-bit little-endian", 137 AV_NE(NULL, "uw"), AV_CODEC_ID_PCM_U16LE) 138 139PCMDEF(u8, "PCM unsigned 8-bit", 140 "ub", AV_CODEC_ID_PCM_U8) 141 142PCMDEF(alaw, "PCM A-law", 143 "al", AV_CODEC_ID_PCM_ALAW) 144 145PCMDEF(mulaw, "PCM mu-law", 146 "ul", AV_CODEC_ID_PCM_MULAW) 147 148static const AVOption sln_options[] = { 149 { "sample_rate", "", offsetof(PCMAudioDemuxerContext, sample_rate), AV_OPT_TYPE_INT, {.i64 = 8000}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, 150 { "channels", "", offsetof(PCMAudioDemuxerContext, channels), AV_OPT_TYPE_INT, {.i64 = 1}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, 151 { NULL }, 152}; 153 154static const AVClass sln_demuxer_class = { 155 .class_name = "sln demuxer", 156 .item_name = av_default_item_name, 157 .option = sln_options, 158 .version = LIBAVUTIL_VERSION_INT, 159}; 160 161AVInputFormat ff_sln_demuxer = { 162 .name = "sln", 163 .long_name = NULL_IF_CONFIG_SMALL("Asterisk raw pcm"), 164 .priv_data_size = sizeof(PCMAudioDemuxerContext), 165 .read_header = pcm_read_header, 166 .read_packet = ff_pcm_read_packet, 167 .read_seek = ff_pcm_read_seek, 168 .flags = AVFMT_GENERIC_INDEX, 169 .extensions = "sln", 170 .raw_codec_id = AV_CODEC_ID_PCM_S16LE, 171 .priv_class = &sln_demuxer_class, 172}; 173