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https://github.com/xenia-project/FFmpeg.git
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6f69f7a8bf
* commit '9200514ad8717c63f82101dc394f4378854325bf': lavf: replace AVStream.codec with AVStream.codecpar This has been a HUGE effort from: - Derek Buitenhuis <derek.buitenhuis@gmail.com> - Hendrik Leppkes <h.leppkes@gmail.com> - wm4 <nfxjfg@googlemail.com> - Clément Bœsch <clement@stupeflix.com> - James Almer <jamrial@gmail.com> - Michael Niedermayer <michael@niedermayer.cc> - Rostislav Pehlivanov <atomnuker@gmail.com> Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
215 lines
6.4 KiB
C
215 lines
6.4 KiB
C
/*
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* AST muxer
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* Copyright (c) 2012 James Almer
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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 "avformat.h"
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#include "avio_internal.h"
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#include "internal.h"
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#include "ast.h"
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#include "libavutil/mathematics.h"
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#include "libavutil/opt.h"
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typedef struct ASTMuxContext {
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AVClass *class;
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int64_t size;
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int64_t samples;
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int64_t loopstart;
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int64_t loopend;
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int fbs;
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} ASTMuxContext;
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#define CHECK_LOOP(type) \
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if (ast->loop ## type > 0) { \
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ast->loop ## type = av_rescale_rnd(ast->loop ## type, par->sample_rate, 1000, AV_ROUND_DOWN); \
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if (ast->loop ## type < 0 || ast->loop ## type > UINT_MAX) { \
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av_log(s, AV_LOG_ERROR, "Invalid loop" #type " value\n"); \
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return AVERROR(EINVAL); \
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} \
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}
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static int ast_write_header(AVFormatContext *s)
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{
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ASTMuxContext *ast = s->priv_data;
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AVIOContext *pb = s->pb;
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AVCodecParameters *par;
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unsigned int codec_tag;
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if (s->nb_streams == 1) {
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par = s->streams[0]->codecpar;
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} else {
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av_log(s, AV_LOG_ERROR, "only one stream is supported\n");
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return AVERROR(EINVAL);
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}
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if (par->codec_id == AV_CODEC_ID_ADPCM_AFC) {
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av_log(s, AV_LOG_ERROR, "muxing ADPCM AFC is not implemented\n");
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return AVERROR_PATCHWELCOME;
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}
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codec_tag = ff_codec_get_tag(ff_codec_ast_tags, par->codec_id);
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if (!codec_tag) {
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av_log(s, AV_LOG_ERROR, "unsupported codec\n");
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return AVERROR(EINVAL);
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}
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if (ast->loopend > 0 && ast->loopstart >= ast->loopend) {
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av_log(s, AV_LOG_ERROR, "loopend can't be less or equal to loopstart\n");
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return AVERROR(EINVAL);
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}
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/* Convert milliseconds to samples */
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CHECK_LOOP(start)
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CHECK_LOOP(end)
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ffio_wfourcc(pb, "STRM");
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ast->size = avio_tell(pb);
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avio_wb32(pb, 0); /* File size minus header */
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avio_wb16(pb, codec_tag);
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avio_wb16(pb, 16); /* Bit depth */
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avio_wb16(pb, par->channels);
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avio_wb16(pb, 0); /* Loop flag */
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avio_wb32(pb, par->sample_rate);
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ast->samples = avio_tell(pb);
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avio_wb32(pb, 0); /* Number of samples */
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avio_wb32(pb, 0); /* Loopstart */
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avio_wb32(pb, 0); /* Loopend */
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avio_wb32(pb, 0); /* Size of first block */
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/* Unknown */
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avio_wb32(pb, 0);
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avio_wl32(pb, 0x7F);
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avio_wb64(pb, 0);
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avio_wb64(pb, 0);
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avio_wb32(pb, 0);
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avio_flush(pb);
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return 0;
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}
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static int ast_write_packet(AVFormatContext *s, AVPacket *pkt)
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{
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AVIOContext *pb = s->pb;
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ASTMuxContext *ast = s->priv_data;
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AVCodecParameters *par = s->streams[0]->codecpar;
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int size = pkt->size / par->channels;
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if (s->streams[0]->nb_frames == 0)
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ast->fbs = size;
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ffio_wfourcc(pb, "BLCK");
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avio_wb32(pb, size); /* Block size */
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/* padding */
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avio_wb64(pb, 0);
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avio_wb64(pb, 0);
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avio_wb64(pb, 0);
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avio_write(pb, pkt->data, pkt->size);
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return 0;
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}
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static int ast_write_trailer(AVFormatContext *s)
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{
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AVIOContext *pb = s->pb;
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ASTMuxContext *ast = s->priv_data;
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AVCodecParameters *par = s->streams[0]->codecpar;
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int64_t file_size = avio_tell(pb);
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int64_t samples = (file_size - 64 - (32 * s->streams[0]->nb_frames)) / par->block_align; /* PCM_S16BE_PLANAR */
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av_log(s, AV_LOG_DEBUG, "total samples: %"PRId64"\n", samples);
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if (s->pb->seekable) {
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/* Number of samples */
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avio_seek(pb, ast->samples, SEEK_SET);
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avio_wb32(pb, samples);
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/* Loopstart if provided */
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if (ast->loopstart > 0) {
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if (ast->loopstart >= samples) {
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av_log(s, AV_LOG_WARNING, "Loopstart value is out of range and will be ignored\n");
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ast->loopstart = -1;
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avio_skip(pb, 4);
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} else
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avio_wb32(pb, ast->loopstart);
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} else
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avio_skip(pb, 4);
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/* Loopend if provided. Otherwise number of samples again */
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if (ast->loopend && ast->loopstart >= 0) {
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if (ast->loopend > samples) {
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av_log(s, AV_LOG_WARNING, "Loopend value is out of range and will be ignored\n");
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ast->loopend = samples;
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}
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avio_wb32(pb, ast->loopend);
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} else {
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avio_wb32(pb, samples);
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}
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/* Size of first block */
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avio_wb32(pb, ast->fbs);
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/* File size minus header */
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avio_seek(pb, ast->size, SEEK_SET);
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avio_wb32(pb, file_size - 64);
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/* Loop flag */
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if (ast->loopstart >= 0) {
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avio_skip(pb, 6);
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avio_wb16(pb, 0xFFFF);
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}
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avio_seek(pb, file_size, SEEK_SET);
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avio_flush(pb);
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}
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return 0;
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}
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#define OFFSET(obj) offsetof(ASTMuxContext, obj)
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static const AVOption options[] = {
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{ "loopstart", "Loopstart position in milliseconds.", OFFSET(loopstart), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
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{ "loopend", "Loopend position in milliseconds.", OFFSET(loopend), AV_OPT_TYPE_INT64, { .i64 = 0 }, 0, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
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{ NULL },
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};
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static const AVClass ast_muxer_class = {
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.class_name = "AST muxer",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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AVOutputFormat ff_ast_muxer = {
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.name = "ast",
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.long_name = NULL_IF_CONFIG_SMALL("AST (Audio Stream)"),
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.extensions = "ast",
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.priv_data_size = sizeof(ASTMuxContext),
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.audio_codec = AV_CODEC_ID_PCM_S16BE_PLANAR,
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.video_codec = AV_CODEC_ID_NONE,
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.write_header = ast_write_header,
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.write_packet = ast_write_packet,
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.write_trailer = ast_write_trailer,
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.priv_class = &ast_muxer_class,
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.codec_tag = (const AVCodecTag* const []){ff_codec_ast_tags, 0},
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};
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