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078d43e23a
This is different from how it is handled in codecs/demuxers/muxers though (where the close function isn't called if the open function failed), but since the number of depacketizers that have an .init function is quite limited, this is easy to change. The main point is that if the init function failed, we shouldn't try to use that depacketizer at all - this makes sure that the parse function doesn't need to check for the things that were initialized in the init function. Signed-off-by: Martin Storsjö <martin@martin.st>
220 lines
8.6 KiB
C
220 lines
8.6 KiB
C
/*
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* RTP demuxer definitions
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* Copyright (c) 2002 Fabrice Bellard
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* Copyright (c) 2006 Ryan Martell <rdm4@martellventures.com>
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*
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* This file is part of Libav.
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*
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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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*
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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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*
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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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*/
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#ifndef AVFORMAT_RTPDEC_H
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#define AVFORMAT_RTPDEC_H
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#include "libavcodec/avcodec.h"
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#include "avformat.h"
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#include "rtp.h"
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#include "url.h"
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#include "srtp.h"
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typedef struct PayloadContext PayloadContext;
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typedef struct RTPDynamicProtocolHandler RTPDynamicProtocolHandler;
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#define RTP_MIN_PACKET_LENGTH 12
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#define RTP_MAX_PACKET_LENGTH 8192
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#define RTP_REORDER_QUEUE_DEFAULT_SIZE 10
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#define RTP_NOTS_VALUE ((uint32_t)-1)
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typedef struct RTPDemuxContext RTPDemuxContext;
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RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
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int payload_type, int queue_size);
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void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
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RTPDynamicProtocolHandler *handler);
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void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
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const char *params);
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int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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uint8_t **buf, int len);
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void ff_rtp_parse_close(RTPDemuxContext *s);
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int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s);
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void ff_rtp_reset_packet_queue(RTPDemuxContext *s);
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/**
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* Send a dummy packet on both port pairs to set up the connection
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* state in potential NAT routers, so that we're able to receive
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* packets.
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*
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* Note, this only works if the NAT router doesn't remap ports. This
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* isn't a standardized procedure, but it works in many cases in practice.
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*
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* The same routine is used with RDT too, even if RDT doesn't use normal
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* RTP packets otherwise.
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*/
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void ff_rtp_send_punch_packets(URLContext* rtp_handle);
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/**
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* some rtp servers assume client is dead if they don't hear from them...
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* so we send a Receiver Report to the provided URLContext or AVIOContext
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* (we don't have access to the rtcp handle from here)
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*/
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int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
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AVIOContext *avio, int count);
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int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
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AVIOContext *avio);
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// these statistics are used for rtcp receiver reports...
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typedef struct RTPStatistics {
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uint16_t max_seq; ///< highest sequence number seen
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uint32_t cycles; ///< shifted count of sequence number cycles
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uint32_t base_seq; ///< base sequence number
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uint32_t bad_seq; ///< last bad sequence number + 1
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int probation; ///< sequence packets till source is valid
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uint32_t received; ///< packets received
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uint32_t expected_prior; ///< packets expected in last interval
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uint32_t received_prior; ///< packets received in last interval
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uint32_t transit; ///< relative transit time for previous packet
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uint32_t jitter; ///< estimated jitter.
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} RTPStatistics;
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#define RTP_FLAG_KEY 0x1 ///< RTP packet contains a keyframe
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#define RTP_FLAG_MARKER 0x2 ///< RTP marker bit was set for this packet
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/**
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* Packet parsing for "private" payloads in the RTP specs.
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*
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* @param ctx RTSP demuxer context
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* @param s stream context
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* @param st stream that this packet belongs to
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* @param pkt packet in which to write the parsed data
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* @param timestamp pointer to the RTP timestamp of the input data, can be
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* updated by the function if returning older, buffered data
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* @param buf pointer to raw RTP packet data
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* @param len length of buf
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* @param seq RTP sequence number of the packet
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* @param flags flags from the RTP packet header (RTP_FLAG_*)
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*/
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typedef int (*DynamicPayloadPacketHandlerProc)(AVFormatContext *ctx,
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PayloadContext *s,
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AVStream *st, AVPacket *pkt,
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uint32_t *timestamp,
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const uint8_t * buf,
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int len, uint16_t seq, int flags);
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struct RTPDynamicProtocolHandler {
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const char *enc_name;
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enum AVMediaType codec_type;
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enum AVCodecID codec_id;
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enum AVStreamParseType need_parsing;
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int static_payload_id; /* 0 means no payload id is set. 0 is a valid
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* payload ID (PCMU), too, but that format doesn't
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* require any custom depacketization code. */
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int priv_data_size;
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/** Initialize dynamic protocol handler, called after the full rtpmap line is parsed, may be null */
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int (*init)(AVFormatContext *s, int st_index, PayloadContext *priv_data);
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/** Parse the a= line from the sdp field */
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int (*parse_sdp_a_line)(AVFormatContext *s, int st_index,
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PayloadContext *priv_data, const char *line);
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/** Free any data needed by the rtp parsing for this dynamic data.
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* Don't free the protocol_data pointer itself, that is freed by the
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* caller. This is called even if the init method failed. */
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void (*free)(PayloadContext *protocol_data);
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/** Parse handler for this dynamic packet */
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DynamicPayloadPacketHandlerProc parse_packet;
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int (*need_keyframe)(PayloadContext *context);
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struct RTPDynamicProtocolHandler *next;
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};
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typedef struct RTPPacket {
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uint16_t seq;
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uint8_t *buf;
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int len;
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int64_t recvtime;
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struct RTPPacket *next;
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} RTPPacket;
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struct RTPDemuxContext {
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AVFormatContext *ic;
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AVStream *st;
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int payload_type;
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uint32_t ssrc;
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uint16_t seq;
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uint32_t timestamp;
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uint32_t base_timestamp;
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uint32_t cur_timestamp;
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int64_t unwrapped_timestamp;
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int64_t range_start_offset;
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int max_payload_size;
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/* used to send back RTCP RR */
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char hostname[256];
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int srtp_enabled;
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struct SRTPContext srtp;
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/** Statistics for this stream (used by RTCP receiver reports) */
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RTPStatistics statistics;
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/** Fields for packet reordering @{ */
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int prev_ret; ///< The return value of the actual parsing of the previous packet
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RTPPacket* queue; ///< A sorted queue of buffered packets not yet returned
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int queue_len; ///< The number of packets in queue
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int queue_size; ///< The size of queue, or 0 if reordering is disabled
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/*@}*/
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/* rtcp sender statistics receive */
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int64_t last_rtcp_ntp_time;
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int64_t last_rtcp_reception_time;
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int64_t first_rtcp_ntp_time;
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uint32_t last_rtcp_timestamp;
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int64_t rtcp_ts_offset;
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/* rtcp sender statistics */
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unsigned int packet_count;
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unsigned int octet_count;
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unsigned int last_octet_count;
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int64_t last_feedback_time;
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/* dynamic payload stuff */
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const RTPDynamicProtocolHandler *handler;
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PayloadContext *dynamic_protocol_context;
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};
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void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler);
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RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
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enum AVMediaType codec_type);
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RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
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enum AVMediaType codec_type);
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/* from rtsp.c, but used by rtp dynamic protocol handlers. */
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int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
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char *value, int value_size);
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int ff_parse_fmtp(AVFormatContext *s,
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AVStream *stream, PayloadContext *data, const char *p,
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int (*parse_fmtp)(AVFormatContext *s,
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AVStream *stream,
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PayloadContext *data,
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const char *attr, const char *value));
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void ff_register_rtp_dynamic_payload_handlers(void);
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/**
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* Close the dynamic buffer and make a packet from it.
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*/
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int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx);
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#endif /* AVFORMAT_RTPDEC_H */
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