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https://github.com/FFmpeg/FFmpeg.git
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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>
270 lines
7.8 KiB
C
270 lines
7.8 KiB
C
/*
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* VQF demuxer
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* Copyright (c) 2009 Vitor Sessak
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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 "internal.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/dict.h"
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#include "libavutil/mathematics.h"
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typedef struct VqfContext {
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int frame_bit_len;
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uint8_t last_frame_bits;
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int remaining_bits;
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} VqfContext;
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static int vqf_probe(AVProbeData *probe_packet)
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{
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if (AV_RL32(probe_packet->buf) != MKTAG('T','W','I','N'))
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return 0;
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if (!memcmp(probe_packet->buf + 4, "97012000", 8))
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return AVPROBE_SCORE_MAX;
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if (!memcmp(probe_packet->buf + 4, "00052200", 8))
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return AVPROBE_SCORE_MAX;
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return AVPROBE_SCORE_MAX/2;
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}
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static void add_metadata(AVFormatContext *s, const char *tag,
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unsigned int tag_len, unsigned int remaining)
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{
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int len = FFMIN(tag_len, remaining);
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char *buf;
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if (len == UINT_MAX)
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return;
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buf = av_malloc(len+1);
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if (!buf)
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return;
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avio_read(s->pb, buf, len);
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buf[len] = 0;
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av_dict_set(&s->metadata, tag, buf, AV_DICT_DONT_STRDUP_VAL);
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}
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static int vqf_read_header(AVFormatContext *s, AVFormatParameters *ap)
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{
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VqfContext *c = s->priv_data;
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AVStream *st = avformat_new_stream(s, NULL);
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int chunk_tag;
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int rate_flag = -1;
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int header_size;
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int read_bitrate = 0;
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int size;
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uint8_t comm_chunk[12];
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if (!st)
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return AVERROR(ENOMEM);
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avio_skip(s->pb, 12);
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header_size = avio_rb32(s->pb);
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_id = CODEC_ID_TWINVQ;
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st->start_time = 0;
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do {
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int len;
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chunk_tag = avio_rl32(s->pb);
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if (chunk_tag == MKTAG('D','A','T','A'))
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break;
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len = avio_rb32(s->pb);
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if ((unsigned) len > INT_MAX/2) {
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av_log(s, AV_LOG_ERROR, "Malformed header\n");
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return -1;
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}
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header_size -= 8;
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switch(chunk_tag){
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case MKTAG('C','O','M','M'):
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avio_read(s->pb, comm_chunk, 12);
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st->codec->channels = AV_RB32(comm_chunk ) + 1;
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read_bitrate = AV_RB32(comm_chunk + 4);
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rate_flag = AV_RB32(comm_chunk + 8);
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avio_skip(s->pb, len-12);
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st->codec->bit_rate = read_bitrate*1000;
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break;
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case MKTAG('N','A','M','E'):
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add_metadata(s, "title" , len, header_size);
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break;
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case MKTAG('(','c',')',' '):
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add_metadata(s, "copyright", len, header_size);
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break;
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case MKTAG('A','U','T','H'):
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add_metadata(s, "author" , len, header_size);
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break;
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case MKTAG('A','L','B','M'):
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add_metadata(s, "album" , len, header_size);
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break;
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case MKTAG('T','R','C','K'):
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add_metadata(s, "track" , len, header_size);
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break;
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case MKTAG('C','O','M','T'):
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add_metadata(s, "comment" , len, header_size);
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break;
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case MKTAG('F','I','L','E'):
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add_metadata(s, "filename" , len, header_size);
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break;
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case MKTAG('D','S','I','Z'):
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add_metadata(s, "size" , len, header_size);
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break;
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case MKTAG('D','A','T','E'):
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add_metadata(s, "date" , len, header_size);
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break;
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case MKTAG('G','E','N','R'):
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add_metadata(s, "genre" , len, header_size);
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break;
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default:
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av_log(s, AV_LOG_ERROR, "Unknown chunk: %c%c%c%c\n",
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((char*)&chunk_tag)[0], ((char*)&chunk_tag)[1],
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((char*)&chunk_tag)[2], ((char*)&chunk_tag)[3]);
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avio_skip(s->pb, FFMIN(len, header_size));
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break;
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}
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header_size -= len;
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} while (header_size >= 0);
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switch (rate_flag) {
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case -1:
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av_log(s, AV_LOG_ERROR, "COMM tag not found!\n");
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return -1;
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case 44:
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st->codec->sample_rate = 44100;
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break;
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case 22:
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st->codec->sample_rate = 22050;
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break;
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case 11:
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st->codec->sample_rate = 11025;
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break;
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default:
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st->codec->sample_rate = rate_flag*1000;
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break;
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}
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switch (((st->codec->sample_rate/1000) << 8) +
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read_bitrate/st->codec->channels) {
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case (11<<8) + 8 :
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case (8 <<8) + 8 :
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case (11<<8) + 10:
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case (22<<8) + 32:
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size = 512;
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break;
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case (16<<8) + 16:
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case (22<<8) + 20:
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case (22<<8) + 24:
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size = 1024;
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break;
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case (44<<8) + 40:
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case (44<<8) + 48:
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size = 2048;
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break;
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default:
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av_log(s, AV_LOG_ERROR, "Mode not suported: %d Hz, %d kb/s.\n",
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st->codec->sample_rate, st->codec->bit_rate);
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return -1;
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}
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c->frame_bit_len = st->codec->bit_rate*size/st->codec->sample_rate;
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avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
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/* put first 12 bytes of COMM chunk in extradata */
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if (!(st->codec->extradata = av_malloc(12 + FF_INPUT_BUFFER_PADDING_SIZE)))
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return AVERROR(ENOMEM);
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st->codec->extradata_size = 12;
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memcpy(st->codec->extradata, comm_chunk, 12);
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return 0;
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}
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static int vqf_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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VqfContext *c = s->priv_data;
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int ret;
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int size = (c->frame_bit_len - c->remaining_bits + 7)>>3;
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pkt->pos = avio_tell(s->pb);
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pkt->stream_index = 0;
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if (av_new_packet(pkt, size+2) < 0)
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return AVERROR(EIO);
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pkt->data[0] = 8 - c->remaining_bits; // Number of bits to skip
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pkt->data[1] = c->last_frame_bits;
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ret = avio_read(s->pb, pkt->data+2, size);
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if (ret<=0) {
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av_free_packet(pkt);
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return AVERROR(EIO);
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}
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c->last_frame_bits = pkt->data[size+1];
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c->remaining_bits = (size << 3) - c->frame_bit_len + c->remaining_bits;
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return size+2;
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}
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static int vqf_read_seek(AVFormatContext *s,
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int stream_index, int64_t timestamp, int flags)
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{
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VqfContext *c = s->priv_data;
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AVStream *st;
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int ret;
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int64_t pos;
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st = s->streams[stream_index];
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pos = av_rescale_rnd(timestamp * st->codec->bit_rate,
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st->time_base.num,
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st->time_base.den * (int64_t)c->frame_bit_len,
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(flags & AVSEEK_FLAG_BACKWARD) ?
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AV_ROUND_DOWN : AV_ROUND_UP);
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pos *= c->frame_bit_len;
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st->cur_dts = av_rescale(pos, st->time_base.den,
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st->codec->bit_rate * (int64_t)st->time_base.num);
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if ((ret = avio_seek(s->pb, ((pos-7) >> 3) + s->data_offset, SEEK_SET)) < 0)
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return ret;
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c->remaining_bits = -7 - ((pos-7)&7);
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return 0;
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}
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AVInputFormat ff_vqf_demuxer = {
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.name = "vqf",
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.long_name = NULL_IF_CONFIG_SMALL("Nippon Telegraph and Telephone Corporation (NTT) TwinVQ"),
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.priv_data_size = sizeof(VqfContext),
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.read_probe = vqf_probe,
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.read_header = vqf_read_header,
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.read_packet = vqf_read_packet,
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.read_seek = vqf_read_seek,
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.extensions = "vqf",
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};
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