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FFmpeg/libavformat/pcmdec.c
Anton Khirnov 08bebeb1be Revert "all: Don't set AVClass.item_name to its default value" 
Some callers assume that item_name is always set, so this may be
considered an API break.

This reverts commit 0c6203c97a.
2024-01-20 10:34:48 +01:00

212 lines
8.9 KiB
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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 "config_components.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#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;
#if FF_API_OLD_CHANNEL_LAYOUT
int channels;
#endif
AVChannelLayout ch_layout;
} PCMAudioDemuxerContext;
static int pcm_read_header(AVFormatContext *s)
{
PCMAudioDemuxerContext *s1 = s->priv_data;
AVCodecParameters *par;
AVStream *st;
uint8_t *mime_type = NULL;
int ret;
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
par = st->codecpar;
par->codec_type = AVMEDIA_TYPE_AUDIO;
par->codec_id = s->iformat->raw_codec_id;
par->sample_rate = s1->sample_rate;
#if FF_API_OLD_CHANNEL_LAYOUT
if (s1->ch_layout.nb_channels) {
#endif
ret = av_channel_layout_copy(&par->ch_layout, &s1->ch_layout);
if (ret < 0)
return ret;
#if FF_API_OLD_CHANNEL_LAYOUT
} else
par->ch_layout.nb_channels = s1->channels;
#endif
av_opt_get(s->pb, "mime_type", AV_OPT_SEARCH_CHILDREN, &mime_type);
if (mime_type && s->iformat->mime_type) {
int rate = 0, channels = 0, little_endian = 0;
const char *options;
if (av_stristart(mime_type, s->iformat->mime_type, &options)) { /* audio/L16 */
while (options = strchr(options, ';')) {
options++;
if (!rate)
sscanf(options, " rate=%d", &rate);
if (!channels)
sscanf(options, " channels=%d", &channels);
if (!little_endian) {
char val[sizeof("little-endian")];
if (sscanf(options, " endianness=%13s", val) == 1) {
little_endian = strcmp(val, "little-endian") == 0;
}
}
}
if (rate <= 0) {
av_log(s, AV_LOG_ERROR,
"Invalid sample_rate found in mime_type \"%s\"\n",
mime_type);
av_freep(&mime_type);
return AVERROR_INVALIDDATA;
}
par->sample_rate = rate;
if (channels > 0) {
av_channel_layout_uninit(&par->ch_layout);
par->ch_layout.nb_channels = channels;
}
if (little_endian)
par->codec_id = AV_CODEC_ID_PCM_S16LE;
}
}
av_freep(&mime_type);
par->bits_per_coded_sample = av_get_bits_per_sample(par->codec_id);
av_assert0(par->bits_per_coded_sample > 0);
par->block_align = par->bits_per_coded_sample * par->ch_layout.nb_channels / 8;
avpriv_set_pts_info(st, 64, 1, par->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 },
#if FF_API_OLD_CHANNEL_LAYOUT
{ "channels", "", offsetof(PCMAudioDemuxerContext, channels), AV_OPT_TYPE_INT, {.i64 = 1}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_DEPRECATED },
{ "ch_layout", "", offsetof(PCMAudioDemuxerContext, ch_layout), AV_OPT_TYPE_CHLAYOUT, {.str = NULL}, 0, 0, AV_OPT_FLAG_DECODING_PARAM },
#else
{ "ch_layout", "", offsetof(PCMAudioDemuxerContext, ch_layout), AV_OPT_TYPE_CHLAYOUT, {.str = "mono"}, 0, 0, AV_OPT_FLAG_DECODING_PARAM },
#endif
{ NULL },
};
static const AVClass pcm_demuxer_class = {
.class_name = "pcm demuxer",
.item_name = av_default_item_name,
.option = pcm_options,
.version = LIBAVUTIL_VERSION_INT,
};
#define PCMDEF_0(name_, long_name_, ext, codec, ...)
#define PCMDEF_1(name_, long_name_, ext, codec, ...) \
const 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 = &pcm_demuxer_class, \
__VA_ARGS__ \
};
#define PCMDEF_2(name, long_name, ext, codec, enabled, ...) \
PCMDEF_ ## enabled(name, long_name, ext, codec, __VA_ARGS__)
#define PCMDEF_3(name, long_name, ext, codec, config, ...) \
PCMDEF_2(name, long_name, ext, codec, config, __VA_ARGS__)
#define PCMDEF_EXT(name, long_name, ext, uppercase, ...) \
PCMDEF_3(name, long_name, ext, AV_CODEC_ID_PCM_ ## uppercase, \
CONFIG_PCM_ ## uppercase ## _DEMUXER, __VA_ARGS__)
#define PCMDEF(name, long_name, ext, uppercase) \
PCMDEF_EXT(name, long_name, ext, uppercase, )
PCMDEF(f64be, "PCM 64-bit floating-point big-endian", NULL, F64BE)
PCMDEF(f64le, "PCM 64-bit floating-point little-endian", NULL, F64LE)
PCMDEF(f32be, "PCM 32-bit floating-point big-endian", NULL, F32BE)
PCMDEF(f32le, "PCM 32-bit floating-point little-endian", NULL, F32LE)
PCMDEF(s32be, "PCM signed 32-bit big-endian", NULL, S32BE)
PCMDEF(s32le, "PCM signed 32-bit little-endian", NULL, S32LE)
PCMDEF(s24be, "PCM signed 24-bit big-endian", NULL, S24BE)
PCMDEF(s24le, "PCM signed 24-bit little-endian", NULL, S24LE)
PCMDEF_EXT(s16be, "PCM signed 16-bit big-endian",
AV_NE("sw", NULL), S16BE, .mime_type = "audio/L16")
PCMDEF(s16le, "PCM signed 16-bit little-endian", AV_NE(NULL, "sw"), S16LE)
PCMDEF(s8, "PCM signed 8-bit", "sb", S8)
PCMDEF(u32be, "PCM unsigned 32-bit big-endian", NULL, U32BE)
PCMDEF(u32le, "PCM unsigned 32-bit little-endian", NULL, U32LE)
PCMDEF(u24be, "PCM unsigned 24-bit big-endian", NULL, U24BE)
PCMDEF(u24le, "PCM unsigned 24-bit little-endian", NULL, U24LE)
PCMDEF(u16be, "PCM unsigned 16-bit big-endian", AV_NE("uw", NULL), U16BE)
PCMDEF(u16le, "PCM unsigned 16-bit little-endian", AV_NE(NULL, "uw"), U16LE)
PCMDEF(u8, "PCM unsigned 8-bit", "ub", U8)
PCMDEF(alaw, "PCM A-law", "al", ALAW)
PCMDEF(mulaw, "PCM mu-law", "ul", MULAW)
PCMDEF(vidc, "PCM Archimedes VIDC", NULL, VIDC)
#if CONFIG_SLN_DEMUXER
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 },
#if FF_API_OLD_CHANNEL_LAYOUT
{ "channels", "", offsetof(PCMAudioDemuxerContext, channels), AV_OPT_TYPE_INT, {.i64 = 1}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_DEPRECATED },
{ "ch_layout", "", offsetof(PCMAudioDemuxerContext, ch_layout), AV_OPT_TYPE_CHLAYOUT, {.str = NULL}, 0, 0, AV_OPT_FLAG_DECODING_PARAM },
#else
{ "ch_layout", "", offsetof(PCMAudioDemuxerContext, ch_layout), AV_OPT_TYPE_CHLAYOUT, {.str = "mono"}, 0, 0, AV_OPT_FLAG_DECODING_PARAM },
#endif
{ NULL },
};
static const AVClass sln_demuxer_class = {
.class_name = "sln demuxer",
.item_name = av_default_item_name,
.option = sln_options,
.version = LIBAVUTIL_VERSION_INT,
};
const 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,
};
#endif
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