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/*
* 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 */
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