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b800327f4c
This commit does for AVInputFormat what commit59c9dc82f4
did for AVOutputFormat: It adds a new type FFInputFormat, moves all the internals of AVInputFormat to it and adds a now reduced AVInputFormat as first member. This does not affect/improve extensibility of both public or private fields for demuxers (it is still a mess due to lavd). This is possible since50f34172e0
(which removed the last usage of an internal field of AVInputFormat in fftools). (Hint: tools/probetest.c accesses the internals of FFInputFormat as well, but given that it is a testing tool this is not considered a problem.) Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
193 lines
8.0 KiB
C
193 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 "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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