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https://github.com/xenia-project/FFmpeg.git
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367cac7827
This fixes heap-buffer-overflows in libopenmpt caused by interpreting the negative size value as unsigned size_t. Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com> Reviewed-by: Jörn Heusipp <osmanx@problemloesungsmaschine.de> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
212 lines
7.5 KiB
C
212 lines
7.5 KiB
C
/*
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* Tracked MOD demuxer (libopenmpt)
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* Copyright (c) 2016 Josh de Kock
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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 <libopenmpt/libopenmpt.h>
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#include <libopenmpt/libopenmpt_stream_callbacks_file.h>
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#include "libavutil/avstring.h"
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#include "libavutil/opt.h"
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#include "avformat.h"
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#include "internal.h"
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typedef struct OpenMPTContext {
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const AVClass *class;
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openmpt_module *module;
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int channels;
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double duration;
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/* options */
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int sample_rate;
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int64_t layout;
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int subsong;
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} OpenMPTContext;
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#define OFFSET(x) offsetof(OpenMPTContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM
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#define D AV_OPT_FLAG_DECODING_PARAM
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static const AVOption options[] = {
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{ "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, { .i64 = 48000 }, 1000, INT_MAX, A | D },
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{ "layout", "set channel layout", OFFSET(layout), AV_OPT_TYPE_CHANNEL_LAYOUT, { .i64 = AV_CH_LAYOUT_STEREO }, 0, INT64_MAX, A | D },
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{ "subsong", "set subsong", OFFSET(subsong), AV_OPT_TYPE_INT, { .i64 = -2 }, -2, INT_MAX, A | D, "subsong"},
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{ "all", "all", 0, AV_OPT_TYPE_CONST, { .i64 = -1}, 0, 0, A | D, "subsong" },
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{ "auto", "auto", 0, AV_OPT_TYPE_CONST, { .i64 = -2}, 0, 0, A | D, "subsong" },
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{ NULL }
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};
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static void openmpt_logfunc(const char *message, void *userdata)
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{
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int level = AV_LOG_INFO;
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if (strstr(message, "ERROR") != NULL) {
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level = AV_LOG_ERROR;
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}
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av_log(userdata, level, "%s\n", message);
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}
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#define add_meta(s, name, meta) \
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do { \
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const char *value = meta; \
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if (value && value[0]) \
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av_dict_set(&s->metadata, name, value, 0); \
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openmpt_free_string(value); \
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} while(0)
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static int read_header_openmpt(AVFormatContext *s)
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{
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AVStream *st;
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OpenMPTContext *openmpt = s->priv_data;
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int64_t size = avio_size(s->pb);
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if (!size)
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return AVERROR_INVALIDDATA;
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char *buf = av_malloc(size);
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int ret;
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if (!buf)
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return AVERROR(ENOMEM);
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size = avio_read(s->pb, buf, size);
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if (size < 0) {
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av_log(s, AV_LOG_ERROR, "Reading input buffer failed.\n");
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av_freep(&buf);
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return size;
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}
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openmpt->module = openmpt_module_create_from_memory(buf, size, openmpt_logfunc, s, NULL);
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av_freep(&buf);
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if (!openmpt->module)
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return AVERROR_INVALIDDATA;
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openmpt->channels = av_get_channel_layout_nb_channels(openmpt->layout);
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openmpt->duration = openmpt_module_get_duration_seconds(openmpt->module);
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add_meta(s, "artist", openmpt_module_get_metadata(openmpt->module, "artist"));
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add_meta(s, "title", openmpt_module_get_metadata(openmpt->module, "title"));
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add_meta(s, "encoder", openmpt_module_get_metadata(openmpt->module, "tracker"));
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add_meta(s, "comment", openmpt_module_get_metadata(openmpt->module, "message"));
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add_meta(s, "date", openmpt_module_get_metadata(openmpt->module, "date"));
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if (openmpt->subsong >= openmpt_module_get_num_subsongs(openmpt->module)) {
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openmpt_module_destroy(openmpt->module);
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av_log(s, AV_LOG_ERROR, "Invalid subsong index: %d\n", openmpt->subsong);
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return AVERROR(EINVAL);
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}
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if (openmpt->subsong != -2) {
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if (openmpt->subsong >= 0) {
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av_dict_set_int(&s->metadata, "track", openmpt->subsong + 1, 0);
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}
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ret = openmpt_module_select_subsong(openmpt->module, openmpt->subsong);
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if (!ret){
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openmpt_module_destroy(openmpt->module);
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av_log(s, AV_LOG_ERROR, "Could not select requested subsong: %d", openmpt->subsong);
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return AVERROR(EINVAL);
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}
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}
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st = avformat_new_stream(s, NULL);
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if (!st) {
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openmpt_module_destroy(openmpt->module);
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openmpt->module = NULL;
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return AVERROR(ENOMEM);
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}
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avpriv_set_pts_info(st, 64, 1, AV_TIME_BASE);
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st->duration = llrint(openmpt->duration*AV_TIME_BASE);
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st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codecpar->codec_id = AV_NE(AV_CODEC_ID_PCM_F32BE, AV_CODEC_ID_PCM_F32LE);
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st->codecpar->channels = openmpt->channels;
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st->codecpar->sample_rate = openmpt->sample_rate;
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return 0;
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}
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#define AUDIO_PKT_SIZE 2048
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static int read_packet_openmpt(AVFormatContext *s, AVPacket *pkt)
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{
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OpenMPTContext *openmpt = s->priv_data;
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int n_samples = AUDIO_PKT_SIZE / (openmpt->channels ? openmpt->channels*4 : 4);
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int ret;
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if ((ret = av_new_packet(pkt, AUDIO_PKT_SIZE)) < 0)
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return ret;
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switch (openmpt->channels) {
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case 1:
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ret = openmpt_module_read_float_mono(openmpt->module, openmpt->sample_rate,
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n_samples, (float *)pkt->data);
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break;
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case 2:
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ret = openmpt_module_read_interleaved_float_stereo(openmpt->module, openmpt->sample_rate,
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n_samples, (float *)pkt->data);
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break;
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case 4:
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ret = openmpt_module_read_interleaved_float_quad(openmpt->module, openmpt->sample_rate,
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n_samples, (float *)pkt->data);
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break;
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default:
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av_log(s, AV_LOG_ERROR, "Unsupported number of channels: %d", openmpt->channels);
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return AVERROR(EINVAL);
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}
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if (ret < 1) {
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pkt->size = 0;
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return AVERROR_EOF;
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}
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pkt->size = ret * (openmpt->channels * 4);
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return 0;
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}
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static int read_close_openmpt(AVFormatContext *s)
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{
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OpenMPTContext *openmpt = s->priv_data;
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openmpt_module_destroy(openmpt->module);
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openmpt->module = NULL;
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return 0;
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}
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static int read_seek_openmpt(AVFormatContext *s, int stream_idx, int64_t ts, int flags)
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{
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OpenMPTContext *openmpt = s->priv_data;
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openmpt_module_set_position_seconds(openmpt->module, (double)ts/AV_TIME_BASE);
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return 0;
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}
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static const AVClass class_openmpt = {
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.class_name = "libopenmpt",
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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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AVInputFormat ff_libopenmpt_demuxer = {
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.name = "libopenmpt",
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.long_name = NULL_IF_CONFIG_SMALL("Tracker formats (libopenmpt)"),
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.priv_data_size = sizeof(OpenMPTContext),
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.read_header = read_header_openmpt,
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.read_packet = read_packet_openmpt,
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.read_close = read_close_openmpt,
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.read_seek = read_seek_openmpt,
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.priv_class = &class_openmpt,
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.extensions = "669,amf,ams,dbm,digi,dmf,dsm,far,gdm,imf,it,j2b,m15,mdl,med,mmcmp,mms,mo3,mod,mptm,mt2,mtm,nst,okt,plm,ppm,psm,pt36,ptm,s3m,sfx,sfx2,stk,stm,ult,umx,wow,xm,xpk",
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};
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