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219 lines
8.5 KiB
219 lines
8.5 KiB
/* |
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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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#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 1500 |
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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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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 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[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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/** 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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/** Allocate any data needed by the rtp parsing for this dynamic data. */ |
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PayloadContext *(*alloc)(void); |
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/** Free any data needed by the rtp parsing for this dynamic data. */ |
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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(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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