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	9d76cf0b18
	
	
	
		
			
			* qatar/master: rtpdec: Templatize the code for different g726 bitrate variants rv40: move loop filter to rv34dsp context lavf: make av_set_pts_info private. rtpdec: Add support for G726 audio rtpdec: Add an init function that can do custom codec context initialization avconv: make copy_tb on by default. matroskadec: don't set codec timebase. rmdec: don't set codec timebase. avconv: compute next_pts from input packet duration when possible. lavf: estimate frame duration from r_frame_rate. avconv: update InputStream.pts in the streamcopy case. Conflicts: avconv.c libavdevice/alsa-audio-dec.c libavdevice/bktr.c libavdevice/fbdev.c libavdevice/libdc1394.c libavdevice/oss_audio.c libavdevice/v4l.c libavdevice/v4l2.c libavdevice/vfwcap.c libavdevice/x11grab.c libavformat/au.c libavformat/eacdata.c libavformat/flvdec.c libavformat/mpegts.c libavformat/mxfenc.c libavformat/rtpdec_g726.c libavformat/wtv.c libavformat/xmv.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
		
			
				
	
	
		
			350 lines
		
	
	
		
			11 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			350 lines
		
	
	
		
			11 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
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|  * LXF demuxer
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|  * Copyright (c) 2010 Tomas Härdin
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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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| 
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| #include "libavutil/intreadwrite.h"
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| #include "avformat.h"
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| #include "internal.h"
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| #include "riff.h"
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| 
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| #define LXF_PACKET_HEADER_SIZE  60
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| #define LXF_HEADER_DATA_SIZE    120
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| #define LXF_IDENT               "LEITCH\0"
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| #define LXF_IDENT_LENGTH        8
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| #define LXF_SAMPLERATE          48000
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| #define LXF_MAX_AUDIO_PACKET    (8008*15*4) ///< 15-channel 32-bit NTSC audio frame
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| 
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| static const AVCodecTag lxf_tags[] = {
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|     { CODEC_ID_MJPEG,       0 },
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|     { CODEC_ID_MPEG1VIDEO,  1 },
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|     { CODEC_ID_MPEG2VIDEO,  2 },    //MpMl, 4:2:0
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|     { CODEC_ID_MPEG2VIDEO,  3 },    //MpPl, 4:2:2
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|     { CODEC_ID_DVVIDEO,     4 },    //DV25
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|     { CODEC_ID_DVVIDEO,     5 },    //DVCPRO
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|     { CODEC_ID_DVVIDEO,     6 },    //DVCPRO50
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|     { CODEC_ID_RAWVIDEO,    7 },    //PIX_FMT_ARGB, where alpha is used for chroma keying
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|     { CODEC_ID_RAWVIDEO,    8 },    //16-bit chroma key
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|     { CODEC_ID_MPEG2VIDEO,  9 },    //4:2:2 CBP ("Constrained Bytes per Gop")
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|     { CODEC_ID_NONE,        0 },
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| };
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| 
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| typedef struct {
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|     int channels;                       ///< number of audio channels. zero means no audio
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|     uint8_t temp[LXF_MAX_AUDIO_PACKET]; ///< temp buffer for de-planarizing the audio data
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|     int frame_number;                   ///< current video frame
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| } LXFDemuxContext;
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| 
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| static int lxf_probe(AVProbeData *p)
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| {
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|     if (!memcmp(p->buf, LXF_IDENT, LXF_IDENT_LENGTH))
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|         return AVPROBE_SCORE_MAX;
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| 
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|     return 0;
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| }
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| 
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| /**
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|  * Verify the checksum of an LXF packet header
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|  *
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|  * @param[in] header the packet header to check
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|  * @return zero if the checksum is OK, non-zero otherwise
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|  */
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| static int check_checksum(const uint8_t *header)
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| {
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|     int x;
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|     uint32_t sum = 0;
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| 
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|     for (x = 0; x < LXF_PACKET_HEADER_SIZE; x += 4)
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|         sum += AV_RL32(&header[x]);
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| 
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|     return sum;
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| }
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| 
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| /**
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|  * Read input until we find the next ident. If found, copy it to the header buffer
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|  *
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|  * @param[out] header where to copy the ident to
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|  * @return 0 if an ident was found, < 0 on I/O error
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|  */
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| static int sync(AVFormatContext *s, uint8_t *header)
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| {
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|     uint8_t buf[LXF_IDENT_LENGTH];
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|     int ret;
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| 
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|     if ((ret = avio_read(s->pb, buf, LXF_IDENT_LENGTH)) != LXF_IDENT_LENGTH)
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|         return ret < 0 ? ret : AVERROR_EOF;
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| 
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|     while (memcmp(buf, LXF_IDENT, LXF_IDENT_LENGTH)) {
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|         if (url_feof(s->pb))
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|             return AVERROR_EOF;
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| 
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|         memmove(buf, &buf[1], LXF_IDENT_LENGTH-1);
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|         buf[LXF_IDENT_LENGTH-1] = avio_r8(s->pb);
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|     }
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| 
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|     memcpy(header, LXF_IDENT, LXF_IDENT_LENGTH);
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| 
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|     return 0;
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| }
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| 
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| /**
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|  * Read and checksum the next packet header
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|  *
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|  * @param[out] header the read packet header
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|  * @param[out] format context dependent format information
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|  * @return the size of the payload following the header or < 0 on failure
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|  */
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| static int get_packet_header(AVFormatContext *s, uint8_t *header, uint32_t *format)
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| {
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|     AVIOContext   *pb  = s->pb;
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|     int track_size, samples, ret;
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|     AVStream *st;
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| 
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|     //find and read the ident
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|     if ((ret = sync(s, header)) < 0)
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|         return ret;
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| 
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|     //read the rest of the packet header
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|     if ((ret = avio_read(pb, header + LXF_IDENT_LENGTH,
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|                           LXF_PACKET_HEADER_SIZE - LXF_IDENT_LENGTH)) !=
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|                           LXF_PACKET_HEADER_SIZE - LXF_IDENT_LENGTH) {
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|         return ret < 0 ? ret : AVERROR_EOF;
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|     }
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| 
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|     if (check_checksum(header))
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|         av_log(s, AV_LOG_ERROR, "checksum error\n");
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| 
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|     *format = AV_RL32(&header[32]);
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|     ret     = AV_RL32(&header[36]);
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| 
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|     //type
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|     switch (AV_RL32(&header[16])) {
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|     case 0:
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|         //video
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|         //skip VBI data and metadata
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|         avio_skip(pb, (int64_t)(uint32_t)AV_RL32(&header[44]) +
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|                       (int64_t)(uint32_t)AV_RL32(&header[52]));
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|         break;
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|     case 1:
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|         //audio
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|         if (!(st = s->streams[1])) {
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|             av_log(s, AV_LOG_INFO, "got audio packet, but no audio stream present\n");
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|             break;
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|         }
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| 
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|         //set codec based on specified audio bitdepth
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|         //we only support tightly packed 16-, 20-, 24- and 32-bit PCM at the moment
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|         *format                          = AV_RL32(&header[40]);
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|         st->codec->bits_per_coded_sample = (*format >> 6) & 0x3F;
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| 
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|         if (st->codec->bits_per_coded_sample != (*format & 0x3F)) {
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|             av_log(s, AV_LOG_WARNING, "only tightly packed PCM currently supported\n");
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|             return AVERROR_PATCHWELCOME;
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|         }
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| 
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|         switch (st->codec->bits_per_coded_sample) {
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|         case 16: st->codec->codec_id = CODEC_ID_PCM_S16LE; break;
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|         case 20: st->codec->codec_id = CODEC_ID_PCM_LXF;   break;
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|         case 24: st->codec->codec_id = CODEC_ID_PCM_S24LE; break;
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|         case 32: st->codec->codec_id = CODEC_ID_PCM_S32LE; break;
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|         default:
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|             av_log(s, AV_LOG_WARNING,
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|                    "only 16-, 20-, 24- and 32-bit PCM currently supported\n");
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|             return AVERROR_PATCHWELCOME;
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|         }
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| 
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|         track_size = AV_RL32(&header[48]);
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|         samples = track_size * 8 / st->codec->bits_per_coded_sample;
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| 
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|         //use audio packet size to determine video standard
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|         //for NTSC we have one 8008-sample audio frame per five video frames
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|         if (samples == LXF_SAMPLERATE * 5005 / 30000) {
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|             avpriv_set_pts_info(s->streams[0], 64, 1001, 30000);
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|         } else {
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|             //assume PAL, but warn if we don't have 1920 samples
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|             if (samples != LXF_SAMPLERATE / 25)
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|                 av_log(s, AV_LOG_WARNING,
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|                        "video doesn't seem to be PAL or NTSC. guessing PAL\n");
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| 
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|             avpriv_set_pts_info(s->streams[0], 64, 1, 25);
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|         }
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| 
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|         //TODO: warning if track mask != (1 << channels) - 1?
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|         ret = av_popcount(AV_RL32(&header[44])) * track_size;
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| 
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|         break;
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|     default:
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|         break;
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|     }
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| 
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|     return ret;
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| }
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| 
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| static int lxf_read_header(AVFormatContext *s, AVFormatParameters *ap)
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| {
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|     LXFDemuxContext *lxf = s->priv_data;
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|     AVIOContext   *pb  = s->pb;
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|     uint8_t header[LXF_PACKET_HEADER_SIZE], header_data[LXF_HEADER_DATA_SIZE];
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|     int ret;
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|     AVStream *st;
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|     uint32_t format, video_params, disk_params;
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|     uint16_t record_date, expiration_date;
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| 
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|     if ((ret = get_packet_header(s, header, &format)) < 0)
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|         return ret;
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| 
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|     if (ret != LXF_HEADER_DATA_SIZE) {
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|         av_log(s, AV_LOG_ERROR, "expected %d B size header, got %d\n",
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|                LXF_HEADER_DATA_SIZE, ret);
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|         return AVERROR_INVALIDDATA;
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|     }
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| 
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|     if ((ret = avio_read(pb, header_data, LXF_HEADER_DATA_SIZE)) != LXF_HEADER_DATA_SIZE)
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|         return ret < 0 ? ret : AVERROR_EOF;
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| 
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|     if (!(st = avformat_new_stream(s, NULL)))
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|         return AVERROR(ENOMEM);
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| 
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|     st->duration          = AV_RL32(&header_data[32]);
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|     video_params          = AV_RL32(&header_data[40]);
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|     record_date           = AV_RL16(&header_data[56]);
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|     expiration_date       = AV_RL16(&header_data[58]);
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|     disk_params           = AV_RL32(&header_data[116]);
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| 
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|     st->codec->codec_type = AVMEDIA_TYPE_VIDEO;
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|     st->codec->bit_rate   = 1000000 * ((video_params >> 14) & 0xFF);
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|     st->codec->codec_tag  = video_params & 0xF;
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|     st->codec->codec_id   = ff_codec_get_id(lxf_tags, st->codec->codec_tag);
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| 
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|     av_log(s, AV_LOG_DEBUG, "record: %x = %i-%02i-%02i\n",
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|            record_date, 1900 + (record_date & 0x7F), (record_date >> 7) & 0xF,
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|            (record_date >> 11) & 0x1F);
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| 
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|     av_log(s, AV_LOG_DEBUG, "expire: %x = %i-%02i-%02i\n",
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|            expiration_date, 1900 + (expiration_date & 0x7F), (expiration_date >> 7) & 0xF,
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|            (expiration_date >> 11) & 0x1F);
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| 
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|     if ((video_params >> 22) & 1)
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|         av_log(s, AV_LOG_WARNING, "VBI data not yet supported\n");
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| 
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|     if ((lxf->channels = (disk_params >> 2) & 0xF)) {
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|         if (!(st = avformat_new_stream(s, NULL)))
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|             return AVERROR(ENOMEM);
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| 
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|         st->codec->codec_type  = AVMEDIA_TYPE_AUDIO;
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|         st->codec->sample_rate = LXF_SAMPLERATE;
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|         st->codec->channels    = lxf->channels;
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| 
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|         avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
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|     }
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| 
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|     if (format == 1) {
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|         //skip extended field data
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|         avio_skip(s->pb, (uint32_t)AV_RL32(&header[40]));
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|     }
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| 
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|     return 0;
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| }
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| 
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| /**
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|  * De-planerize the PCM data in lxf->temp
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|  * FIXME: remove this once support for planar audio is added to libavcodec
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|  *
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|  * @param[out] out where to write the de-planerized data to
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|  * @param[in] bytes the total size of the PCM data
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|  */
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| static void deplanarize(LXFDemuxContext *lxf, AVStream *ast, uint8_t *out, int bytes)
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| {
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|     int x, y, z, i, bytes_per_sample = ast->codec->bits_per_coded_sample >> 3;
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| 
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|     for (z = i = 0; z < lxf->channels; z++)
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|         for (y = 0; y < bytes / bytes_per_sample / lxf->channels; y++)
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|             for (x = 0; x < bytes_per_sample; x++, i++)
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|                 out[x + bytes_per_sample*(z + y*lxf->channels)] = lxf->temp[i];
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| }
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| 
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| static int lxf_read_packet(AVFormatContext *s, AVPacket *pkt)
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| {
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|     LXFDemuxContext *lxf = s->priv_data;
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|     AVIOContext   *pb  = s->pb;
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|     uint8_t header[LXF_PACKET_HEADER_SIZE], *buf;
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|     AVStream *ast = NULL;
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|     uint32_t stream, format;
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|     int ret, ret2;
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| 
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|     if ((ret = get_packet_header(s, header, &format)) < 0)
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|         return ret;
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| 
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|     stream = AV_RL32(&header[16]);
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| 
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|     if (stream > 1) {
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|         av_log(s, AV_LOG_WARNING, "got packet with illegal stream index %u\n", stream);
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|         return AVERROR(EAGAIN);
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|     }
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| 
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|     if (stream == 1 && !(ast = s->streams[1])) {
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|         av_log(s, AV_LOG_ERROR, "got audio packet without having an audio stream\n");
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|         return AVERROR_INVALIDDATA;
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|     }
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| 
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|     //make sure the data fits in the de-planerization buffer
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|     if (ast && ret > LXF_MAX_AUDIO_PACKET) {
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|         av_log(s, AV_LOG_ERROR, "audio packet too large (%i > %i)\n",
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|             ret, LXF_MAX_AUDIO_PACKET);
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|         return AVERROR_INVALIDDATA;
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|     }
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| 
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|     if ((ret2 = av_new_packet(pkt, ret)) < 0)
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|         return ret2;
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| 
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|     //read non-20-bit audio data into lxf->temp so we can deplanarize it
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|     buf = ast && ast->codec->codec_id != CODEC_ID_PCM_LXF ? lxf->temp : pkt->data;
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| 
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|     if ((ret2 = avio_read(pb, buf, ret)) != ret) {
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|         av_free_packet(pkt);
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|         return ret2 < 0 ? ret2 : AVERROR_EOF;
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|     }
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| 
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|     pkt->stream_index = stream;
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| 
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|     if (ast) {
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|         if(ast->codec->codec_id != CODEC_ID_PCM_LXF)
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|             deplanarize(lxf, ast, pkt->data, ret);
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|     } else {
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|         //picture type (0 = closed I, 1 = open I, 2 = P, 3 = B)
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|         if (((format >> 22) & 0x3) < 2)
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|             pkt->flags |= AV_PKT_FLAG_KEY;
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| 
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|         pkt->dts = lxf->frame_number++;
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|     }
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| 
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|     return ret;
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| }
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| 
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| AVInputFormat ff_lxf_demuxer = {
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|     .name           = "lxf",
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|     .long_name      = NULL_IF_CONFIG_SMALL("VR native stream format (LXF)"),
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|     .priv_data_size = sizeof(LXFDemuxContext),
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|     .read_probe     = lxf_probe,
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|     .read_header    = lxf_read_header,
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|     .read_packet    = lxf_read_packet,
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|     .codec_tag      = (const AVCodecTag* const []){lxf_tags, 0},
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| };
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| 
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