mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-22 15:23:11 +00:00
186f1ec5f4
Patch by Josh Allmann, joshua dot allmann at gmail Originally committed as revision 24929 to svn://svn.ffmpeg.org/ffmpeg/trunk
188 lines
7.6 KiB
C
188 lines
7.6 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"
|
|
|
|
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 */
|
|
|
|
#define RTP_NOTS_VALUE ((uint32_t)-1)
|
|
|
|
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);
|
|
|
|
/**
|
|
* Get the file handle for the RTCP socket.
|
|
*/
|
|
int rtp_get_rtcp_file_handle(URLContext *h);
|
|
|
|
// 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 */
|