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* Copyright (c) 2010 Tomas Härdin
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* This file is part of Libav.
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* Libav 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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* Libav 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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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; 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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#include "libavutil/intreadwrite.h"
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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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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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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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static int lxf_probe(AVProbeData *p)
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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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* Verify the checksum of an LXF packet header
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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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static int check_checksum(const uint8_t *header)
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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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* Read input until we find the next ident. If found, copy it to the header buffer
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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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static int sync(AVFormatContext *s, uint8_t *header)
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uint8_t buf[LXF_IDENT_LENGTH];
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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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while (memcmp(buf, LXF_IDENT, LXF_IDENT_LENGTH)) {
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if (s->pb->eof_reached)
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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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memcpy(header, LXF_IDENT, LXF_IDENT_LENGTH);
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* Read and checksum the next packet header
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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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static int get_packet_header(AVFormatContext *s, uint8_t *header, uint32_t *format)
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AVIOContext *pb = s->pb;
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int track_size, samples, ret;
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//find and read the ident
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if ((ret = sync(s, header)) < 0)
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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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if (check_checksum(header))
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av_log(s, AV_LOG_ERROR, "checksum error\n");
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*format = AV_RL32(&header[32]);
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ret = AV_RL32(&header[36]);
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switch (AV_RL32(&header[16])) {
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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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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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//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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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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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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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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track_size = AV_RL32(&header[48]);
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samples = track_size * 8 / st->codec->bits_per_coded_sample;
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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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av_set_pts_info(s->streams[0], 64, 1001, 30000);
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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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av_set_pts_info(s->streams[0], 64, 1, 25);
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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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static int lxf_read_header(AVFormatContext *s, AVFormatParameters *ap)
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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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uint32_t format, video_params, disk_params;
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uint16_t record_date, expiration_date;
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if ((ret = get_packet_header(s, header, &format)) < 0)
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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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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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if (!(st = av_new_stream(s, 0)))
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return AVERROR(ENOMEM);
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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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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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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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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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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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if ((lxf->channels = (disk_params >> 2) & 0xF)) {
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if (!(st = av_new_stream(s, 1)))
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return AVERROR(ENOMEM);
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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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av_set_pts_info(st, 64, 1, st->codec->sample_rate);
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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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* 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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* @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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static void deplanarize(LXFDemuxContext *lxf, AVStream *ast, uint8_t *out, int bytes)
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int x, y, z, i, bytes_per_sample = ast->codec->bits_per_coded_sample >> 3;
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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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static int lxf_read_packet(AVFormatContext *s, AVPacket *pkt)
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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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if ((ret = get_packet_header(s, header, &format)) < 0)
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stream = AV_RL32(&header[16]);
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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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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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//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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if ((ret2 = av_new_packet(pkt, ret)) < 0)
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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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if ((ret2 = avio_read(pb, buf, ret)) != ret) {
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return ret2 < 0 ? ret2 : AVERROR_EOF;
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pkt->stream_index = stream;
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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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//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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pkt->dts = lxf->frame_number++;
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AVInputFormat ff_lxf_demuxer = {
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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},