PeerStreamer

/*

 * 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 FFMPEG_RTP_H

#define FFMPEG_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);

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);

#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 /* FFMPEG_RTP_H */