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790f793844
There are lots of files that don't need it: The number of object files that actually need it went down from 2011 to 884 here. Keep it for external users in order to not cause breakages. Also improve the other headers a bit while just at it. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
194 lines
8.0 KiB
C
194 lines
8.0 KiB
C
/*
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* RAW PCM demuxers
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* Copyright (c) 2002 Fabrice Bellard
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "config_components.h"
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#include "libavutil/avstring.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/mem.h"
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#include "avformat.h"
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#include "demux.h"
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#include "internal.h"
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#include "pcm.h"
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#include "libavutil/log.h"
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#include "libavutil/opt.h"
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#include "libavutil/avassert.h"
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typedef struct PCMAudioDemuxerContext {
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AVClass *class;
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int sample_rate;
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AVChannelLayout ch_layout;
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} PCMAudioDemuxerContext;
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static int pcm_read_header(AVFormatContext *s)
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{
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PCMAudioDemuxerContext *s1 = s->priv_data;
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AVCodecParameters *par;
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AVStream *st;
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uint8_t *mime_type = NULL;
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int ret;
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st = avformat_new_stream(s, NULL);
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if (!st)
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return AVERROR(ENOMEM);
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par = st->codecpar;
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par->codec_type = AVMEDIA_TYPE_AUDIO;
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par->codec_id = ffifmt(s->iformat)->raw_codec_id;
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par->sample_rate = s1->sample_rate;
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ret = av_channel_layout_copy(&par->ch_layout, &s1->ch_layout);
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if (ret < 0)
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return ret;
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av_opt_get(s->pb, "mime_type", AV_OPT_SEARCH_CHILDREN, &mime_type);
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if (mime_type && s->iformat->mime_type) {
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int rate = 0, channels = 0, little_endian = 0;
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const char *options;
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if (av_stristart(mime_type, s->iformat->mime_type, &options)) { /* audio/L16 */
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while (options = strchr(options, ';')) {
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options++;
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if (!rate)
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sscanf(options, " rate=%d", &rate);
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if (!channels)
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sscanf(options, " channels=%d", &channels);
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if (!little_endian) {
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char val[sizeof("little-endian")];
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if (sscanf(options, " endianness=%13s", val) == 1) {
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little_endian = strcmp(val, "little-endian") == 0;
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}
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}
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}
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if (rate <= 0) {
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av_log(s, AV_LOG_ERROR,
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"Invalid sample_rate found in mime_type \"%s\"\n",
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mime_type);
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av_freep(&mime_type);
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return AVERROR_INVALIDDATA;
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}
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par->sample_rate = rate;
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if (channels > 0) {
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av_channel_layout_uninit(&par->ch_layout);
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par->ch_layout.nb_channels = channels;
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}
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if (little_endian)
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par->codec_id = AV_CODEC_ID_PCM_S16LE;
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}
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}
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av_freep(&mime_type);
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par->bits_per_coded_sample = av_get_bits_per_sample(par->codec_id);
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av_assert0(par->bits_per_coded_sample > 0);
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par->block_align = par->bits_per_coded_sample * par->ch_layout.nb_channels / 8;
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avpriv_set_pts_info(st, 64, 1, par->sample_rate);
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return 0;
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}
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static const AVOption pcm_options[] = {
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{ "sample_rate", "", offsetof(PCMAudioDemuxerContext, sample_rate), AV_OPT_TYPE_INT, {.i64 = 44100}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ "ch_layout", "", offsetof(PCMAudioDemuxerContext, ch_layout), AV_OPT_TYPE_CHLAYOUT, {.str = "mono"}, 0, 0, AV_OPT_FLAG_DECODING_PARAM },
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{ NULL },
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};
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static const AVClass pcm_demuxer_class = {
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.class_name = "pcm demuxer",
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.item_name = av_default_item_name,
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.option = pcm_options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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#define PCMDEF_0(name_, long_name_, ext, codec, ...)
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#define PCMDEF_1(name_, long_name_, ext, codec, ...) \
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const FFInputFormat ff_pcm_ ## name_ ## _demuxer = { \
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.p.name = #name_, \
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.p.long_name = NULL_IF_CONFIG_SMALL(long_name_), \
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.p.flags = AVFMT_GENERIC_INDEX, \
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.p.extensions = ext, \
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.p.priv_class = &pcm_demuxer_class, \
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.priv_data_size = sizeof(PCMAudioDemuxerContext), \
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.read_header = pcm_read_header, \
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.read_packet = ff_pcm_read_packet, \
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.read_seek = ff_pcm_read_seek, \
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.raw_codec_id = codec, \
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__VA_ARGS__ \
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};
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#define PCMDEF_2(name, long_name, ext, codec, enabled, ...) \
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PCMDEF_ ## enabled(name, long_name, ext, codec, __VA_ARGS__)
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#define PCMDEF_3(name, long_name, ext, codec, config, ...) \
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PCMDEF_2(name, long_name, ext, codec, config, __VA_ARGS__)
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#define PCMDEF_EXT(name, long_name, ext, uppercase, ...) \
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PCMDEF_3(name, long_name, ext, AV_CODEC_ID_PCM_ ## uppercase, \
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CONFIG_PCM_ ## uppercase ## _DEMUXER, __VA_ARGS__)
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#define PCMDEF(name, long_name, ext, uppercase) \
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PCMDEF_EXT(name, long_name, ext, uppercase, )
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PCMDEF(f64be, "PCM 64-bit floating-point big-endian", NULL, F64BE)
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PCMDEF(f64le, "PCM 64-bit floating-point little-endian", NULL, F64LE)
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PCMDEF(f32be, "PCM 32-bit floating-point big-endian", NULL, F32BE)
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PCMDEF(f32le, "PCM 32-bit floating-point little-endian", NULL, F32LE)
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PCMDEF(s32be, "PCM signed 32-bit big-endian", NULL, S32BE)
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PCMDEF(s32le, "PCM signed 32-bit little-endian", NULL, S32LE)
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PCMDEF(s24be, "PCM signed 24-bit big-endian", NULL, S24BE)
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PCMDEF(s24le, "PCM signed 24-bit little-endian", NULL, S24LE)
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PCMDEF_EXT(s16be, "PCM signed 16-bit big-endian",
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AV_NE("sw", NULL), S16BE, .p.mime_type = "audio/L16")
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PCMDEF(s16le, "PCM signed 16-bit little-endian", AV_NE(NULL, "sw"), S16LE)
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PCMDEF(s8, "PCM signed 8-bit", "sb", S8)
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PCMDEF(u32be, "PCM unsigned 32-bit big-endian", NULL, U32BE)
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PCMDEF(u32le, "PCM unsigned 32-bit little-endian", NULL, U32LE)
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PCMDEF(u24be, "PCM unsigned 24-bit big-endian", NULL, U24BE)
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PCMDEF(u24le, "PCM unsigned 24-bit little-endian", NULL, U24LE)
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PCMDEF(u16be, "PCM unsigned 16-bit big-endian", AV_NE("uw", NULL), U16BE)
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PCMDEF(u16le, "PCM unsigned 16-bit little-endian", AV_NE(NULL, "uw"), U16LE)
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PCMDEF(u8, "PCM unsigned 8-bit", "ub", U8)
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PCMDEF(alaw, "PCM A-law", "al", ALAW)
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PCMDEF(mulaw, "PCM mu-law", "ul", MULAW)
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PCMDEF(vidc, "PCM Archimedes VIDC", NULL, VIDC)
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#if CONFIG_SLN_DEMUXER
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static const AVOption sln_options[] = {
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{ "sample_rate", "", offsetof(PCMAudioDemuxerContext, sample_rate), AV_OPT_TYPE_INT, {.i64 = 8000}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
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{ "ch_layout", "", offsetof(PCMAudioDemuxerContext, ch_layout), AV_OPT_TYPE_CHLAYOUT, {.str = "mono"}, 0, 0, AV_OPT_FLAG_DECODING_PARAM },
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{ NULL },
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};
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static const AVClass sln_demuxer_class = {
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.class_name = "sln demuxer",
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.item_name = av_default_item_name,
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.option = sln_options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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const FFInputFormat ff_sln_demuxer = {
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.p.name = "sln",
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.p.long_name = NULL_IF_CONFIG_SMALL("Asterisk raw pcm"),
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.p.flags = AVFMT_GENERIC_INDEX,
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.p.extensions = "sln",
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.p.priv_class = &sln_demuxer_class,
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.priv_data_size = sizeof(PCMAudioDemuxerContext),
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.read_header = pcm_read_header,
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.read_packet = ff_pcm_read_packet,
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.read_seek = ff_pcm_read_seek,
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.raw_codec_id = AV_CODEC_ID_PCM_S16LE,
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};
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#endif
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