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/*
* RAW PCM demuxers
* Copyright (c) 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* 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.
*
* FFmpeg is distributed in the hope that it will be useful,
* 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
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avformat.h"
#include "internal.h"
#include "pcm.h"
#include "libavutil/log.h"
#include "libavutil/opt.h"
#include "libavutil/avassert.h"
typedef struct PCMAudioDemuxerContext {
AVClass *class;
int sample_rate;
int channels;
} PCMAudioDemuxerContext;
static int pcm_read_header(AVFormatContext *s)
{
PCMAudioDemuxerContext *s1 = s->priv_data;
AVStream *st;
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = s->iformat->raw_codec_id;
st->codecpar->sample_rate = s1->sample_rate;
st->codecpar->channels = s1->channels;
st->codecpar->bits_per_coded_sample =
av_get_bits_per_sample(st->codecpar->codec_id);
av_assert0(st->codecpar->bits_per_coded_sample > 0);
st->codecpar->block_align =
st->codecpar->bits_per_coded_sample * st->codecpar->channels / 8;
avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
return 0;
}
static const AVOption pcm_options[] = {
{ "sample_rate", "", offsetof(PCMAudioDemuxerContext, sample_rate), AV_OPT_TYPE_INT, {.i64 = 44100}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ "channels", "", offsetof(PCMAudioDemuxerContext, channels), AV_OPT_TYPE_INT, {.i64 = 1}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ NULL },
};
#define PCMDEF(name_, long_name_, ext, codec) \
static const AVClass name_ ## _demuxer_class = { \
.class_name = #name_ " demuxer", \
.item_name = av_default_item_name, \
.option = pcm_options, \
.version = LIBAVUTIL_VERSION_INT, \
}; \
AVInputFormat ff_pcm_ ## name_ ## _demuxer = { \
.name = #name_, \
.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
.priv_data_size = sizeof(PCMAudioDemuxerContext), \
.read_header = pcm_read_header, \
.read_packet = ff_pcm_read_packet, \
.read_seek = ff_pcm_read_seek, \
.flags = AVFMT_GENERIC_INDEX, \
.extensions = ext, \
.raw_codec_id = codec, \
.priv_class = &name_ ## _demuxer_class, \
};
PCMDEF(f64be, "PCM 64-bit floating-point big-endian",
NULL, AV_CODEC_ID_PCM_F64BE)
PCMDEF(f64le, "PCM 64-bit floating-point little-endian",
NULL, AV_CODEC_ID_PCM_F64LE)
PCMDEF(f32be, "PCM 32-bit floating-point big-endian",
NULL, AV_CODEC_ID_PCM_F32BE)
PCMDEF(f32le, "PCM 32-bit floating-point little-endian",
NULL, AV_CODEC_ID_PCM_F32LE)
PCMDEF(s32be, "PCM signed 32-bit big-endian",
NULL, AV_CODEC_ID_PCM_S32BE)
PCMDEF(s32le, "PCM signed 32-bit little-endian",
NULL, AV_CODEC_ID_PCM_S32LE)
PCMDEF(s24be, "PCM signed 24-bit big-endian",
NULL, AV_CODEC_ID_PCM_S24BE)
PCMDEF(s24le, "PCM signed 24-bit little-endian",
NULL, AV_CODEC_ID_PCM_S24LE)
PCMDEF(s16be, "PCM signed 16-bit big-endian",
AV_NE("sw", NULL), AV_CODEC_ID_PCM_S16BE)
PCMDEF(s16le, "PCM signed 16-bit little-endian",
AV_NE(NULL, "sw"), AV_CODEC_ID_PCM_S16LE)
PCMDEF(s8, "PCM signed 8-bit",
"sb", AV_CODEC_ID_PCM_S8)
PCMDEF(u32be, "PCM unsigned 32-bit big-endian",
NULL, AV_CODEC_ID_PCM_U32BE)
PCMDEF(u32le, "PCM unsigned 32-bit little-endian",
NULL, AV_CODEC_ID_PCM_U32LE)
PCMDEF(u24be, "PCM unsigned 24-bit big-endian",
NULL, AV_CODEC_ID_PCM_U24BE)
PCMDEF(u24le, "PCM unsigned 24-bit little-endian",
NULL, AV_CODEC_ID_PCM_U24LE)
PCMDEF(u16be, "PCM unsigned 16-bit big-endian",
AV_NE("uw", NULL), AV_CODEC_ID_PCM_U16BE)
PCMDEF(u16le, "PCM unsigned 16-bit little-endian",
AV_NE(NULL, "uw"), AV_CODEC_ID_PCM_U16LE)
PCMDEF(u8, "PCM unsigned 8-bit",
"ub", AV_CODEC_ID_PCM_U8)
PCMDEF(alaw, "PCM A-law",
"al", AV_CODEC_ID_PCM_ALAW)
PCMDEF(mulaw, "PCM mu-law",
"ul", AV_CODEC_ID_PCM_MULAW)
static const AVOption sln_options[] = {
{ "sample_rate", "", offsetof(PCMAudioDemuxerContext, sample_rate), AV_OPT_TYPE_INT, {.i64 = 8000}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ "channels", "", offsetof(PCMAudioDemuxerContext, channels), AV_OPT_TYPE_INT, {.i64 = 1}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
{ NULL },
};
static const AVClass sln_demuxer_class = {
.class_name = "sln demuxer",
.item_name = av_default_item_name,
.option = sln_options,
.version = LIBAVUTIL_VERSION_INT,
};
AVInputFormat ff_sln_demuxer = {
.name = "sln",
.long_name = NULL_IF_CONFIG_SMALL("Asterisk raw pcm"),
.priv_data_size = sizeof(PCMAudioDemuxerContext),
.read_header = pcm_read_header,
.read_packet = ff_pcm_read_packet,
.read_seek = ff_pcm_read_seek,
.flags = AVFMT_GENERIC_INDEX,
.extensions = "sln",
.raw_codec_id = AV_CODEC_ID_PCM_S16LE,
.priv_class = &sln_demuxer_class,
};
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