view rtp.h @ 2660:022174d849d5 libavformat

fix issue 225, instead of stoping when wrong atom size is found, limit atom size to what is left, assuming container atom has correct size.. cricket4.3g2 has incorrect moov atom size which indicates that file size should be 2 bytes bigger than it is and quicktime reads it correctly though.

/*
 * 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 */