1
mirror of https://git.videolan.org/git/ffmpeg.git synced 2024-08-09 02:45:45 +02:00
ffmpeg/libavformat/rtpdec.h
Martin Storsjö 3a1cdcc798 rtpdec: Emit timestamps for packets before the first RTCP packet, too
Emitted timestamps in each stream start from 0, for the first received
RTP packet. Once an RTCP packet is received, that one is used for
sync, emitting timestamps that fit seamlessly into the earlier ones.

Originally committed as revision 26187 to svn://svn.ffmpeg.org/ffmpeg/trunk
2011-01-01 22:27:16 +00:00

209 lines
8.6 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_REORDER_QUEUE_DEFAULT_SIZE 10
#define RTP_NOTS_VALUE ((uint32_t)-1)
typedef struct RTPDemuxContext RTPDemuxContext;
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size);
void rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
RTPDynamicProtocolHandler *handler);
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
uint8_t **buf, int len);
void rtp_parse_close(RTPDemuxContext *s);
int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s);
void ff_rtp_reset_packet_queue(RTPDemuxContext *s);
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);
/**
* 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;
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. */
// 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;
};
typedef struct RTPPacket {
uint16_t seq;
uint8_t *buf;
int len;
int64_t recvtime;
struct RTPPacket *next;
} RTPPacket;
// 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)
/** 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; // TODO: move into statistics
int64_t first_rtcp_ntp_time; // TODO: move into statistics
uint32_t last_rtcp_timestamp; // TODO: move into statistics
int64_t rtcp_ts_offset;
/* 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);
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);
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 */