mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-24 16:22:37 +00:00
016bc031eb
This will be used for cleaning up code that is common among RTP depacketizers. Patch by Josh Allmann, joshua dot allmann at gmail Originally committed as revision 23847 to svn://svn.ffmpeg.org/ffmpeg/trunk
183 lines
7.5 KiB
C
183 lines
7.5 KiB
C
/*
|
|
* RTP demuxer definitions
|
|
* Copyright (c) 2002 Fabrice Bellard
|
|
* Copyright (c) 2006 Ryan Martell <rdm4@martellventures.com>
|
|
*
|
|
* This file is part of FFmpeg.
|
|
*
|
|
* FFmpeg is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Lesser General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2.1 of the License, or (at your option) any later version.
|
|
*
|
|
* FFmpeg is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Lesser General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Lesser General Public
|
|
* License along with FFmpeg; if not, write to the Free Software
|
|
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
|
*/
|
|
#ifndef AVFORMAT_RTPDEC_H
|
|
#define AVFORMAT_RTPDEC_H
|
|
|
|
#include "libavcodec/avcodec.h"
|
|
#include "avformat.h"
|
|
#include "rtp.h"
|
|
|
|
#define SPACE_CHARS " \t\r\n"
|
|
|
|
typedef struct PayloadContext PayloadContext;
|
|
typedef struct RTPDynamicProtocolHandler_s RTPDynamicProtocolHandler;
|
|
|
|
#define RTP_MIN_PACKET_LENGTH 12
|
|
#define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
|
|
|
|
typedef struct RTPDemuxContext RTPDemuxContext;
|
|
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type);
|
|
void rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
|
|
RTPDynamicProtocolHandler *handler);
|
|
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
|
|
const uint8_t *buf, int len);
|
|
void rtp_parse_close(RTPDemuxContext *s);
|
|
#if (LIBAVFORMAT_VERSION_MAJOR <= 53)
|
|
int rtp_get_local_port(URLContext *h);
|
|
#endif
|
|
int rtp_get_local_rtp_port(URLContext *h);
|
|
int rtp_get_local_rtcp_port(URLContext *h);
|
|
|
|
int rtp_set_remote_url(URLContext *h, const char *uri);
|
|
#if (LIBAVFORMAT_VERSION_MAJOR <= 52)
|
|
void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd);
|
|
#endif
|
|
|
|
/**
|
|
* Send a dummy packet on both port pairs to set up the connection
|
|
* state in potential NAT routers, so that we're able to receive
|
|
* packets.
|
|
*
|
|
* Note, this only works if the NAT router doesn't remap ports. This
|
|
* isn't a standardized procedure, but it works in many cases in practice.
|
|
*
|
|
* The same routine is used with RDT too, even if RDT doesn't use normal
|
|
* RTP packets otherwise.
|
|
*/
|
|
void rtp_send_punch_packets(URLContext* rtp_handle);
|
|
|
|
/**
|
|
* some rtp servers assume client is dead if they don't hear from them...
|
|
* so we send a Receiver Report to the provided ByteIO context
|
|
* (we don't have access to the rtcp handle from here)
|
|
*/
|
|
int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count);
|
|
|
|
// these statistics are used for rtcp receiver reports...
|
|
typedef struct {
|
|
uint16_t max_seq; ///< highest sequence number seen
|
|
uint32_t cycles; ///< shifted count of sequence number cycles
|
|
uint32_t base_seq; ///< base sequence number
|
|
uint32_t bad_seq; ///< last bad sequence number + 1
|
|
int probation; ///< sequence packets till source is valid
|
|
int received; ///< packets received
|
|
int expected_prior; ///< packets expected in last interval
|
|
int received_prior; ///< packets received in last interval
|
|
uint32_t transit; ///< relative transit time for previous packet
|
|
uint32_t jitter; ///< estimated jitter.
|
|
} RTPStatistics;
|
|
|
|
#define RTP_FLAG_KEY 0x1 ///< RTP packet contains a keyframe
|
|
#define RTP_FLAG_MARKER 0x2 ///< RTP marker bit was set for this packet
|
|
/**
|
|
* Packet parsing for "private" payloads in the RTP specs.
|
|
*
|
|
* @param ctx RTSP demuxer context
|
|
* @param s stream context
|
|
* @param st stream that this packet belongs to
|
|
* @param pkt packet in which to write the parsed data
|
|
* @param timestamp pointer in which to write the timestamp of this RTP packet
|
|
* @param buf pointer to raw RTP packet data
|
|
* @param len length of buf
|
|
* @param flags flags from the RTP packet header (RTP_FLAG_*)
|
|
*/
|
|
typedef int (*DynamicPayloadPacketHandlerProc) (AVFormatContext *ctx,
|
|
PayloadContext *s,
|
|
AVStream *st,
|
|
AVPacket * pkt,
|
|
uint32_t *timestamp,
|
|
const uint8_t * buf,
|
|
int len, int flags);
|
|
|
|
struct RTPDynamicProtocolHandler_s {
|
|
// fields from AVRtpDynamicPayloadType_s
|
|
const char enc_name[50]; /* XXX: still why 50 ? ;-) */
|
|
enum AVMediaType codec_type;
|
|
enum CodecID codec_id;
|
|
|
|
// may be null
|
|
int (*parse_sdp_a_line) (AVFormatContext *s,
|
|
int st_index,
|
|
PayloadContext *priv_data,
|
|
const char *line); ///< Parse the a= line from the sdp field
|
|
PayloadContext *(*open) (void); ///< allocate any data needed by the rtp parsing for this dynamic data.
|
|
void (*close)(PayloadContext *protocol_data); ///< free any data needed by the rtp parsing for this dynamic data.
|
|
DynamicPayloadPacketHandlerProc parse_packet; ///< parse handler for this dynamic packet.
|
|
|
|
struct RTPDynamicProtocolHandler_s *next;
|
|
};
|
|
|
|
// moved out of rtp.c, because the h264 decoder needs to know about this structure..
|
|
struct RTPDemuxContext {
|
|
AVFormatContext *ic;
|
|
AVStream *st;
|
|
int payload_type;
|
|
uint32_t ssrc;
|
|
uint16_t seq;
|
|
uint32_t timestamp;
|
|
uint32_t base_timestamp;
|
|
uint32_t cur_timestamp;
|
|
int64_t range_start_offset;
|
|
int max_payload_size;
|
|
struct MpegTSContext *ts; /* only used for MP2T payloads */
|
|
int read_buf_index;
|
|
int read_buf_size;
|
|
/* used to send back RTCP RR */
|
|
URLContext *rtp_ctx;
|
|
char hostname[256];
|
|
|
|
RTPStatistics statistics; ///< Statistics for this stream (used by RTCP receiver reports)
|
|
|
|
/* rtcp sender statistics receive */
|
|
int64_t last_rtcp_ntp_time; // TODO: move into statistics
|
|
int64_t first_rtcp_ntp_time; // TODO: move into statistics
|
|
uint32_t last_rtcp_timestamp; // TODO: move into statistics
|
|
|
|
/* rtcp sender statistics */
|
|
unsigned int packet_count; // TODO: move into statistics (outgoing)
|
|
unsigned int octet_count; // TODO: move into statistics (outgoing)
|
|
unsigned int last_octet_count; // TODO: move into statistics (outgoing)
|
|
int first_packet;
|
|
/* buffer for output */
|
|
uint8_t buf[RTP_MAX_PACKET_LENGTH];
|
|
uint8_t *buf_ptr;
|
|
|
|
/* dynamic payload stuff */
|
|
DynamicPayloadPacketHandlerProc parse_packet; ///< This is also copied from the dynamic protocol handler structure
|
|
PayloadContext *dynamic_protocol_context; ///< This is a copy from the values setup from the sdp parsing, in rtsp.c don't free me.
|
|
int max_frames_per_packet;
|
|
};
|
|
|
|
extern RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler;
|
|
void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler);
|
|
|
|
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.
|
|
|
|
int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
|
|
int (*parse_fmtp)(AVStream *stream,
|
|
PayloadContext *data,
|
|
char *attr, char *value));
|
|
|
|
void av_register_rtp_dynamic_payload_handlers(void);
|
|
|
|
#endif /* AVFORMAT_RTPDEC_H */
|