mirror of
https://git.ffmpeg.org/ffmpeg.git
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a4b8cb3c55
when calling, but url_close_buf() doesn't free it. The better solution is to not allocate it at all, init it with init_put_byte() and then not have to close it at all. In the case where we do need to hold it around for longer than within the function context, we allocate it with av_alloc_put_byte() and free it with av_free() instead. Discussed in ML thread "[PATCH] fix small memleak in rdt.c". Originally committed as revision 15850 to svn://svn.ffmpeg.org/ffmpeg/trunk
417 lines
13 KiB
C
417 lines
13 KiB
C
/*
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* Realmedia RTSP protocol (RDT) support.
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* Copyright (c) 2007 Ronald S. Bultje
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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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* @file rdt.c
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* @brief Realmedia RTSP protocol (RDT) support
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* @author Ronald S. Bultje <rbultje@ronald.bitfreak.net>
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*/
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#include "avformat.h"
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#include "libavutil/avstring.h"
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#include "rtp_internal.h"
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#include "rdt.h"
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#include "libavutil/base64.h"
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#include "libavutil/md5.h"
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#include "rm.h"
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#include "internal.h"
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struct RDTDemuxContext {
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AVFormatContext *ic;
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AVStream *st;
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void *dynamic_protocol_context;
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DynamicPayloadPacketHandlerProc parse_packet;
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uint32_t prev_set_id, prev_timestamp;
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};
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RDTDemuxContext *
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ff_rdt_parse_open(AVFormatContext *ic, AVStream *st,
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void *priv_data, RTPDynamicProtocolHandler *handler)
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{
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RDTDemuxContext *s = av_mallocz(sizeof(RDTDemuxContext));
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if (!s)
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return NULL;
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s->ic = ic;
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s->st = st;
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s->prev_set_id = -1;
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s->prev_timestamp = -1;
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s->parse_packet = handler->parse_packet;
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s->dynamic_protocol_context = priv_data;
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return s;
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}
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void
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ff_rdt_parse_close(RDTDemuxContext *s)
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{
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av_free(s);
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}
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struct PayloadContext {
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AVFormatContext *rmctx;
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uint8_t *mlti_data;
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unsigned int mlti_data_size;
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char buffer[RTP_MAX_PACKET_LENGTH + FF_INPUT_BUFFER_PADDING_SIZE];
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};
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void
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ff_rdt_calc_response_and_checksum(char response[41], char chksum[9],
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const char *challenge)
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{
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int ch_len = strlen (challenge), i;
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unsigned char zres[16],
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buf[64] = { 0xa1, 0xe9, 0x14, 0x9d, 0x0e, 0x6b, 0x3b, 0x59 };
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#define XOR_TABLE_SIZE 37
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const unsigned char xor_table[XOR_TABLE_SIZE] = {
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0x05, 0x18, 0x74, 0xd0, 0x0d, 0x09, 0x02, 0x53,
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0xc0, 0x01, 0x05, 0x05, 0x67, 0x03, 0x19, 0x70,
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0x08, 0x27, 0x66, 0x10, 0x10, 0x72, 0x08, 0x09,
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0x63, 0x11, 0x03, 0x71, 0x08, 0x08, 0x70, 0x02,
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0x10, 0x57, 0x05, 0x18, 0x54 };
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/* some (length) checks */
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if (ch_len == 40) /* what a hack... */
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ch_len = 32;
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else if (ch_len > 56)
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ch_len = 56;
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memcpy(buf + 8, challenge, ch_len);
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/* xor challenge bytewise with xor_table */
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for (i = 0; i < XOR_TABLE_SIZE; i++)
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buf[8 + i] ^= xor_table[i];
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av_md5_sum(zres, buf, 64);
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ff_data_to_hex(response, zres, 16);
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for (i=0;i<32;i++) response[i] = tolower(response[i]);
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/* add tail */
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strcpy (response + 32, "01d0a8e3");
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/* calculate checksum */
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for (i = 0; i < 8; i++)
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chksum[i] = response[i * 4];
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chksum[8] = 0;
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}
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static int
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rdt_load_mdpr (PayloadContext *rdt, AVStream *st, int rule_nr)
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{
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ByteIOContext pb;
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int size;
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uint32_t tag;
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/**
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* Layout of the MLTI chunk:
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* 4:MLTI
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* 2:<number of streams>
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* Then for each stream ([number_of_streams] times):
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* 2:<mdpr index>
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* 2:<number of mdpr chunks>
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* Then for each mdpr chunk ([number_of_mdpr_chunks] times):
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* 4:<size>
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* [size]:<data>
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* we skip MDPR chunks until we reach the one of the stream
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* we're interested in, and forward that ([size]+[data]) to
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* the RM demuxer to parse the stream-specific header data.
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*/
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if (!rdt->mlti_data)
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return -1;
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init_put_byte(&pb, rdt->mlti_data, rdt->mlti_data_size, 0,
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NULL, NULL, NULL, NULL);
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tag = get_le32(&pb);
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if (tag == MKTAG('M', 'L', 'T', 'I')) {
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int num, chunk_nr;
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/* read index of MDPR chunk numbers */
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num = get_be16(&pb);
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if (rule_nr < 0 || rule_nr >= num)
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return -1;
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url_fskip(&pb, rule_nr * 2);
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chunk_nr = get_be16(&pb);
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url_fskip(&pb, (num - 1 - rule_nr) * 2);
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/* read MDPR chunks */
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num = get_be16(&pb);
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if (chunk_nr >= num)
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return -1;
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while (chunk_nr--)
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url_fskip(&pb, get_be32(&pb));
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size = get_be32(&pb);
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} else {
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size = rdt->mlti_data_size;
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url_fseek(&pb, 0, SEEK_SET);
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}
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if (ff_rm_read_mdpr_codecdata(rdt->rmctx, &pb, st, size) < 0)
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return -1;
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return 0;
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}
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/**
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* Actual data handling.
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*/
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int
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ff_rdt_parse_header(const uint8_t *buf, int len,
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int *set_id, int *seq_no, int *stream_id,
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int *is_keyframe, uint32_t *timestamp)
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{
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int consumed = 10;
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/* skip status packets */
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while (len >= 5 && buf[1] == 0xFF /* status packet */) {
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int pkt_len;
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if (!(buf[0] & 0x80))
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return -1; /* not followed by a data packet */
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pkt_len = AV_RB16(buf+3);
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buf += pkt_len;
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len -= pkt_len;
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consumed += pkt_len;
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}
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if (len < 10)
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return -1;
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/**
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* Layout of the header (in bits):
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* 1: len_included
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* Flag indicating whether this header includes a length field;
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* this can be used to concatenate multiple RDT packets in a
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* single UDP/TCP data frame and is used to precede RDT data
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* by stream status packets
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* 1: need_reliable
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* Flag indicating whether this header includes a "reliable
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* sequence number"; these are apparently sequence numbers of
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* data packets alone. For data packets, this flag is always
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* set, according to the Real documentation [1]
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* 5: set_id
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* ID of a set of streams of identical content, possibly with
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* different codecs or bitrates
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* 1: is_reliable
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* Flag set for certain streams deemed less tolerable for packet
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* loss
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* 16: seq_no
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* Packet sequence number; if >=0xFF00, this is a non-data packet
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* containing stream status info, the second byte indicates the
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* type of status packet (see wireshark docs / source code [2])
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* if (len_included) {
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* 16: packet_len
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* } else {
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* packet_len = remainder of UDP/TCP frame
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* }
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* 1: is_back_to_back
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* Back-to-Back flag; used for timing, set for one in every 10
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* packets, according to the Real documentation [1]
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* 1: is_slow_data
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* Slow-data flag; currently unused, according to Real docs [1]
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* 5: stream_id
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* ID of the stream within this particular set of streams
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* 1: is_no_keyframe
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* Non-keyframe flag (unset if packet belongs to a keyframe)
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* 32: timestamp (PTS)
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* if (set_id == 0x1F) {
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* 16: set_id (extended set-of-streams ID; see set_id)
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* }
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* if (need_reliable) {
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* 16: reliable_seq_no
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* Reliable sequence number (see need_reliable)
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* }
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* if (stream_id == 0x3F) {
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* 16: stream_id (extended stream ID; see stream_id)
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* }
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* [1] https://protocol.helixcommunity.org/files/2005/devdocs/RDT_Feature_Level_20.txt
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* [2] http://www.wireshark.org/docs/dfref/r/rdt.html and
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* http://anonsvn.wireshark.org/viewvc/trunk/epan/dissectors/packet-rdt.c
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*/
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if (set_id) *set_id = (buf[0]>>1) & 0x1f;
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if (seq_no) *seq_no = AV_RB16(buf+1);
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if (timestamp) *timestamp = AV_RB32(buf+4);
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if (stream_id) *stream_id = (buf[3]>>1) & 0x1f;
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if (is_keyframe) *is_keyframe = !(buf[3] & 0x1);
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return consumed;
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}
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/**< return 0 on packet, no more left, 1 on packet, 1 on partial packet... */
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static int
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rdt_parse_packet (PayloadContext *rdt, AVStream *st,
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AVPacket *pkt, uint32_t *timestamp,
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const uint8_t *buf, int len, int flags)
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{
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int seq = 1, res;
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ByteIOContext pb;
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RMContext *rm = rdt->rmctx->priv_data;
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if (rm->audio_pkt_cnt == 0) {
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int pos;
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init_put_byte(&pb, buf, len, 0, NULL, NULL, NULL, NULL);
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flags = (flags & PKT_FLAG_KEY) ? 2 : 0;
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res = ff_rm_parse_packet (rdt->rmctx, &pb, st, len, pkt,
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&seq, &flags, timestamp);
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pos = url_ftell(&pb);
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if (res < 0)
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return res;
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if (rm->audio_pkt_cnt > 0 &&
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st->codec->codec_id == CODEC_ID_AAC) {
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memcpy (rdt->buffer, buf + pos, len - pos);
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rdt->rmctx->pb = av_alloc_put_byte (rdt->buffer, len - pos, 0,
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NULL, NULL, NULL, NULL);
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}
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} else {
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ff_rm_retrieve_cache (rdt->rmctx, rdt->rmctx->pb, st, pkt);
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if (rm->audio_pkt_cnt == 0 &&
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st->codec->codec_id == CODEC_ID_AAC)
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av_freep(&rdt->rmctx->pb);
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}
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pkt->stream_index = st->index;
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pkt->pts = *timestamp;
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return rm->audio_pkt_cnt > 0;
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}
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int
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ff_rdt_parse_packet(RDTDemuxContext *s, AVPacket *pkt,
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const uint8_t *buf, int len)
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{
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int seq_no, flags = 0, stream_id, set_id, is_keyframe;
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uint32_t timestamp;
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int rv= 0;
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if (!s->parse_packet)
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return -1;
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if (!buf) {
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/* return the next packets, if any */
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timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
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rv= s->parse_packet(s->dynamic_protocol_context,
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s->st, pkt, ×tamp, NULL, 0, flags);
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return rv;
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}
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if (len < 12)
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return -1;
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rv = ff_rdt_parse_header(buf, len, &set_id, &seq_no, &stream_id, &is_keyframe, ×tamp);
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if (rv < 0)
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return rv;
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if (is_keyframe && (set_id != s->prev_set_id || timestamp != s->prev_timestamp)) {
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flags |= PKT_FLAG_KEY;
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s->prev_set_id = set_id;
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s->prev_timestamp = timestamp;
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}
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buf += rv;
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len -= rv;
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rv = s->parse_packet(s->dynamic_protocol_context,
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s->st, pkt, ×tamp, buf, len, flags);
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return rv;
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}
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void
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ff_rdt_subscribe_rule (char *cmd, int size,
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int stream_nr, int rule_nr)
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{
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av_strlcatf(cmd, size, "stream=%d;rule=%d,stream=%d;rule=%d",
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stream_nr, rule_nr * 2, stream_nr, rule_nr * 2 + 1);
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}
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void
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ff_rdt_subscribe_rule2 (RDTDemuxContext *s, char *cmd, int size,
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int stream_nr, int rule_nr)
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{
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PayloadContext *rdt = s->dynamic_protocol_context;
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rdt_load_mdpr(rdt, s->st, rule_nr * 2);
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}
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static unsigned char *
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rdt_parse_b64buf (unsigned int *target_len, const char *p)
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{
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unsigned char *target;
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int len = strlen(p);
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if (*p == '\"') {
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p++;
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len -= 2; /* skip embracing " at start/end */
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}
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*target_len = len * 3 / 4;
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target = av_mallocz(*target_len + FF_INPUT_BUFFER_PADDING_SIZE);
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av_base64_decode(target, p, *target_len);
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return target;
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}
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static int
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rdt_parse_sdp_line (AVStream *stream, PayloadContext *rdt, const char *line)
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{
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const char *p = line;
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if (av_strstart(p, "OpaqueData:buffer;", &p)) {
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rdt->mlti_data = rdt_parse_b64buf(&rdt->mlti_data_size, p);
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} else if (av_strstart(p, "StartTime:integer;", &p))
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stream->first_dts = atoi(p);
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return 0;
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}
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static PayloadContext *
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rdt_new_extradata (void)
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{
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PayloadContext *rdt = av_mallocz(sizeof(PayloadContext));
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av_open_input_stream(&rdt->rmctx, NULL, "", &rdt_demuxer, NULL);
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return rdt;
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}
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static void
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rdt_free_extradata (PayloadContext *rdt)
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{
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if (rdt->rmctx)
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av_close_input_stream(rdt->rmctx);
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av_freep(&rdt->mlti_data);
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av_free(rdt);
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}
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#define RDT_HANDLER(n, s, t) \
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static RTPDynamicProtocolHandler ff_rdt_ ## n ## _handler = { \
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s, \
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t, \
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CODEC_ID_NONE, \
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rdt_parse_sdp_line, \
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rdt_new_extradata, \
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rdt_free_extradata, \
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rdt_parse_packet \
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};
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RDT_HANDLER(live_video, "x-pn-multirate-realvideo-live", CODEC_TYPE_VIDEO);
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RDT_HANDLER(live_audio, "x-pn-multirate-realaudio-live", CODEC_TYPE_AUDIO);
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RDT_HANDLER(video, "x-pn-realvideo", CODEC_TYPE_VIDEO);
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RDT_HANDLER(audio, "x-pn-realaudio", CODEC_TYPE_AUDIO);
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void av_register_rdt_dynamic_payload_handlers(void)
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{
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ff_register_dynamic_payload_handler(&ff_rdt_video_handler);
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ff_register_dynamic_payload_handler(&ff_rdt_audio_handler);
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ff_register_dynamic_payload_handler(&ff_rdt_live_video_handler);
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ff_register_dynamic_payload_handler(&ff_rdt_live_audio_handler);
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}
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