mirror of https://git.ffmpeg.org/ffmpeg.git
211 lines
8.8 KiB
C
211 lines
8.8 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 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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#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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typedef struct PayloadContext PayloadContext;
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typedef struct RTPDynamicProtocolHandler_s RTPDynamicProtocolHandler;
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#define RTP_MIN_PACKET_LENGTH 12
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#define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
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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, URLContext *rtpc, 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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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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int ff_rtp_get_local_rtp_port(URLContext *h);
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int ff_rtp_get_local_rtcp_port(URLContext *h);
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int ff_rtp_set_remote_url(URLContext *h, const char *uri);
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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 ByteIO context
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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, int count);
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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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int received; ///< packets received
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int expected_prior; ///< packets expected in last interval
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int 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 in which to write the timestamp of this RTP packet
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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 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,
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AVPacket * pkt,
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uint32_t *timestamp,
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const uint8_t * buf,
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int len, int flags);
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struct RTPDynamicProtocolHandler_s {
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// fields from AVRtpDynamicPayloadType_s
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const char enc_name[50]; /* XXX: still why 50 ? ;-) */
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enum AVMediaType codec_type;
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enum AVCodecID codec_id;
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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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// may be null
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int (*init)(AVFormatContext *s, int st_index, PayloadContext *priv_data); ///< Initialize dynamic protocol handler, called after the full rtpmap line is parsed
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int (*parse_sdp_a_line) (AVFormatContext *s,
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int st_index,
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PayloadContext *priv_data,
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const char *line); ///< Parse the a= line from the sdp field
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PayloadContext *(*alloc) (void); ///< allocate any data needed by the rtp parsing for this dynamic data.
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void (*free)(PayloadContext *protocol_data); ///< free any data needed by the rtp parsing for this dynamic data.
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DynamicPayloadPacketHandlerProc parse_packet; ///< parse handler for this dynamic packet.
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struct RTPDynamicProtocolHandler_s *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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// moved out of rtp.c, because the h264 decoder needs to know about this structure..
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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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struct MpegTSContext *ts; /* only used for MP2T payloads */
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int read_buf_index;
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int read_buf_size;
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/* used to send back RTCP RR */
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URLContext *rtp_ctx;
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char hostname[256];
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RTPStatistics statistics; ///< Statistics for this stream (used by RTCP receiver reports)
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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; // TODO: move into statistics
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int64_t first_rtcp_ntp_time; // TODO: move into statistics
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uint32_t last_rtcp_timestamp; // TODO: move into statistics
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int64_t rtcp_ts_offset;
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/* rtcp sender statistics */
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unsigned int packet_count; // TODO: move into statistics (outgoing)
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unsigned int octet_count; // TODO: move into statistics (outgoing)
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unsigned int last_octet_count; // TODO: move into statistics (outgoing)
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int first_packet;
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/* buffer for output */
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uint8_t buf[RTP_MAX_PACKET_LENGTH];
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uint8_t *buf_ptr;
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/* dynamic payload stuff */
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DynamicPayloadPacketHandlerProc parse_packet; ///< This is also copied from the dynamic protocol handler structure
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PayloadContext *dynamic_protocol_context; ///< This is a copy from the values setup from the sdp parsing, in rtsp.c don't free me.
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int max_frames_per_packet;
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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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int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size); ///< from rtsp.c, but used by rtp dynamic protocol handlers.
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int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
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int (*parse_fmtp)(AVStream *stream,
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PayloadContext *data,
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char *attr, char *value));
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void av_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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