mirror of https://git.ffmpeg.org/ffmpeg.git
131 lines
3.6 KiB
C
131 lines
3.6 KiB
C
/*
|
|
* RTP definitions
|
|
* Copyright (c) 2002 Fabrice Bellard.
|
|
*
|
|
* 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 RTP_H
|
|
#define RTP_H
|
|
|
|
#include "avcodec.h"
|
|
#include "avformat.h"
|
|
|
|
#define RTP_MIN_PACKET_LENGTH 12
|
|
#define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
|
|
|
|
int rtp_init(void);
|
|
int rtp_get_codec_info(AVCodecContext *codec, int payload_type);
|
|
|
|
/** return < 0 if unknown payload type */
|
|
int rtp_get_payload_type(AVCodecContext *codec);
|
|
|
|
typedef struct RTPDemuxContext RTPDemuxContext;
|
|
typedef struct rtp_payload_data_s rtp_payload_data_s;
|
|
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_s *rtp_payload_data);
|
|
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
|
|
const uint8_t *buf, int len);
|
|
void rtp_parse_close(RTPDemuxContext *s);
|
|
|
|
extern AVOutputFormat rtp_muxer;
|
|
extern AVInputFormat rtp_demuxer;
|
|
|
|
int rtp_get_local_port(URLContext *h);
|
|
int rtp_set_remote_url(URLContext *h, const char *uri);
|
|
void rtp_get_file_handles(URLContext *h, int *prtp_fd, int *prtcp_fd);
|
|
|
|
/**
|
|
* 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);
|
|
|
|
extern URLProtocol rtp_protocol;
|
|
|
|
#define RTP_PT_PRIVATE 96
|
|
#define RTP_VERSION 2
|
|
#define RTP_MAX_SDES 256 /**< maximum text length for SDES */
|
|
|
|
/* RTCP paquets use 0.5 % of the bandwidth */
|
|
#define RTCP_TX_RATIO_NUM 5
|
|
#define RTCP_TX_RATIO_DEN 1000
|
|
|
|
/** Structure listing useful vars to parse RTP packet payload*/
|
|
typedef struct rtp_payload_data_s
|
|
{
|
|
int sizelength;
|
|
int indexlength;
|
|
int indexdeltalength;
|
|
int profile_level_id;
|
|
int streamtype;
|
|
int objecttype;
|
|
char *mode;
|
|
|
|
/** mpeg 4 AU headers */
|
|
struct AUHeaders {
|
|
int size;
|
|
int index;
|
|
int cts_flag;
|
|
int cts;
|
|
int dts_flag;
|
|
int dts;
|
|
int rap_flag;
|
|
int streamstate;
|
|
} *au_headers;
|
|
int nb_au_headers;
|
|
int au_headers_length_bytes;
|
|
int cur_au_index;
|
|
} rtp_payload_data_t;
|
|
|
|
typedef struct AVRtpPayloadType_s
|
|
{
|
|
int pt;
|
|
const char enc_name[50]; /* XXX: why 50 ? */
|
|
enum CodecType codec_type;
|
|
enum CodecID codec_id;
|
|
int clock_rate;
|
|
int audio_channels;
|
|
} AVRtpPayloadType_t;
|
|
|
|
#if 0
|
|
typedef enum {
|
|
RTCP_SR = 200,
|
|
RTCP_RR = 201,
|
|
RTCP_SDES = 202,
|
|
RTCP_BYE = 203,
|
|
RTCP_APP = 204
|
|
} rtcp_type_t;
|
|
|
|
typedef enum {
|
|
RTCP_SDES_END = 0,
|
|
RTCP_SDES_CNAME = 1,
|
|
RTCP_SDES_NAME = 2,
|
|
RTCP_SDES_EMAIL = 3,
|
|
RTCP_SDES_PHONE = 4,
|
|
RTCP_SDES_LOC = 5,
|
|
RTCP_SDES_TOOL = 6,
|
|
RTCP_SDES_NOTE = 7,
|
|
RTCP_SDES_PRIV = 8,
|
|
RTCP_SDES_IMG = 9,
|
|
RTCP_SDES_DOOR = 10,
|
|
RTCP_SDES_SOURCE = 11
|
|
} rtcp_sdes_type_t;
|
|
#endif
|
|
|
|
extern AVRtpPayloadType_t AVRtpPayloadTypes[];
|
|
#endif /* RTP_H */
|