mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-21 06:50:44 +00:00
3008a93b4d
Possible since 6197453761
.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
227 lines
8.9 KiB
C
227 lines
8.9 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/codec_id.h"
|
|
#include "libavcodec/packet.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 500
|
|
|
|
#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,
|
|
const 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);
|
|
|
|
/**
|
|
* 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;
|
|
enum AVMediaType codec_type;
|
|
enum AVCodecID codec_id;
|
|
enum AVStreamParseType need_parsing;
|
|
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. */
|
|
int priv_data_size;
|
|
|
|
/** 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);
|
|
/** Free any data needed by the rtp parsing for this dynamic data.
|
|
* Don't free the protocol_data pointer itself, that is freed by the
|
|
* caller. This is called even if the init method failed. */
|
|
void (*close)(PayloadContext *protocol_data);
|
|
/** Parse handler for this dynamic packet */
|
|
DynamicPayloadPacketHandlerProc parse_packet;
|
|
int (*need_keyframe)(PayloadContext *context);
|
|
};
|
|
|
|
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;
|
|
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 */
|
|
uint64_t last_rtcp_ntp_time;
|
|
int64_t last_rtcp_reception_time;
|
|
uint64_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;
|
|
};
|
|
|
|
/**
|
|
* Find a registered rtp dynamic protocol handler with the specified name.
|
|
*
|
|
* @param name name of the requested rtp dynamic protocol handler
|
|
* @return A rtp dynamic protocol handler if one was found, NULL otherwise.
|
|
*/
|
|
const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
|
|
enum AVMediaType codec_type);
|
|
/**
|
|
* Find a registered rtp dynamic protocol handler with a matching codec ID.
|
|
*
|
|
* @param id AVCodecID of the requested rtp dynamic protocol handler.
|
|
* @return A rtp dynamic protocol handler if one was found, NULL otherwise.
|
|
*/
|
|
const 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(AVFormatContext *s,
|
|
AVStream *stream, PayloadContext *data, const char *p,
|
|
int (*parse_fmtp)(AVFormatContext *s,
|
|
AVStream *stream,
|
|
PayloadContext *data,
|
|
const char *attr, const char *value));
|
|
|
|
/**
|
|
* 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 */
|