virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00
Code Issues Releases Activity
21bf0d6f80
FFmpeg/libavformat/rtpdec.h
Andrew Van Til 8df46c65cf rtpdec: Increase max rtp packet size to 8192 
Fixes Ticket2314
Fixes connecting to "Ceton InfiniTV4 PCIe & USB"

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-04-05 01:48:55 +02:00

220 lines
8.5 KiB
C
Raw Blame History

/*
* 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"
#include "url.h"
#include "srtp.h"
typedef struct PayloadContext PayloadContext;
typedef struct RTPDynamicProtocolHandler RTPDynamicProtocolHandler;
#define RTP_MIN_PACKET_LENGTH 12
#define RTP_MAX_PACKET_LENGTH 8192
#define RTP_REORDER_QUEUE_DEFAULT_SIZE 10
#define RTP_NOTS_VALUE ((uint32_t)-1)
typedef struct RTPDemuxContext RTPDemuxContext;
RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
int payload_type, int queue_size);
void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
RTPDynamicProtocolHandler *handler);
void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
const char *params);
int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
uint8_t **buf, int len);
void ff_rtp_parse_close(RTPDemuxContext *s);
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s);
void ff_rtp_reset_packet_queue(RTPDemuxContext *s);
int ff_rtp_get_local_rtp_port(URLContext *h);
int ff_rtp_get_local_rtcp_port(URLContext *h);
int ff_rtp_set_remote_url(URLContext *h, const char *uri);
/**
* 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 ff_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 URLContext or AVIOContext
* (we don't have access to the rtcp handle from here)
*/
int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
AVIOContext *avio, int count);
int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
AVIOContext *avio);
// these statistics are used for rtcp receiver reports...
typedef struct RTPStatistics {
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
uint32_t received; ///< packets received
uint32_t expected_prior; ///< packets expected in last interval
uint32_t 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 to the RTP timestamp of the input data, can be
* updated by the function if returning older, buffered data
* @param buf pointer to raw RTP packet data
* @param len length of buf
* @param seq RTP sequence number of the packet
* @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, uint16_t seq, int flags);
struct RTPDynamicProtocolHandler {
const char enc_name[50];
enum AVMediaType codec_type;
enum AVCodecID codec_id;
int static_payload_id; /* 0 means no payload id is set. 0 is a valid
* payload ID (PCMU), too, but that format doesn't
* require any custom depacketization code. */
/** Initialize dynamic protocol handler, called after the full rtpmap line is parsed, may be null */
int (*init)(AVFormatContext *s, int st_index, PayloadContext *priv_data);
/** Parse the a= line from the sdp field */
int (*parse_sdp_a_line)(AVFormatContext *s, int st_index,
PayloadContext *priv_data, const char *line);
/** Allocate any data needed by the rtp parsing for this dynamic data. */
PayloadContext *(*alloc)(void);
/** Free any data needed by the rtp parsing for this dynamic data. */
void (*free)(PayloadContext *protocol_data);
/** Parse handler for this dynamic packet */
DynamicPayloadPacketHandlerProc parse_packet;
int (*need_keyframe)(PayloadContext *context);
struct RTPDynamicProtocolHandler *next;
};
typedef struct RTPPacket {
uint16_t seq;
uint8_t *buf;
int len;
int64_t recvtime;
struct RTPPacket *next;
} RTPPacket;
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 unwrapped_timestamp;
int64_t range_start_offset;
int max_payload_size;
/* used to send back RTCP RR */
char hostname[256];
int srtp_enabled;
struct SRTPContext srtp;
/** Statistics for this stream (used by RTCP receiver reports) */
RTPStatistics statistics;
/** Fields for packet reordering @{ */
int prev_ret; ///< The return value of the actual parsing of the previous packet
RTPPacket* queue; ///< A sorted queue of buffered packets not yet returned
int queue_len; ///< The number of packets in queue
int queue_size; ///< The size of queue, or 0 if reordering is disabled
/*@}*/
/* rtcp sender statistics receive */
int64_t last_rtcp_ntp_time;
int64_t last_rtcp_reception_time;
int64_t first_rtcp_ntp_time;
uint32_t last_rtcp_timestamp;
int64_t rtcp_ts_offset;
/* rtcp sender statistics */
unsigned int packet_count;
unsigned int octet_count;
unsigned int last_octet_count;
int64_t last_feedback_time;
/* dynamic payload stuff */
const RTPDynamicProtocolHandler *handler;
PayloadContext *dynamic_protocol_context;
};
void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler);
RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
enum AVMediaType codec_type);
RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
enum AVMediaType codec_type);
/* from rtsp.c, but used by rtp dynamic protocol handlers. */
int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
char *value, int value_size);
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);
/**
* Close the dynamic buffer and make a packet from it.
*/
int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx);
#endif /* AVFORMAT_RTPDEC_H */
Reference in New Issue View Git Blame Copy Permalink