📄 rtp.c
字号:
/* * RTP input/output format * Copyright (c) 2002 Fabrice Bellard. * * This library 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 of the License, or (at your option) any later version. * * This library 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 this library; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */#include "avformat.h"#include <unistd.h>#include <sys/types.h>#include <sys/socket.h>#include <netinet/in.h>#ifndef __BEOS__# include <arpa/inet.h>#else# include "barpainet.h"#endif#include <netdb.h>//#define DEBUG/* TODO: - add RTCP statistics reporting (should be optional). - add support for h263/mpeg4 packetized output : IDEA: send a buffer to 'rtp_write_packet' contains all the packets for ONE frame. Each packet should have a four byte header containing the length in big endian format (same trick as 'url_open_dyn_packet_buf') */#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 1000typedef 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;enum RTPPayloadType { RTP_PT_ULAW = 0, RTP_PT_GSM = 3, RTP_PT_G723 = 4, RTP_PT_ALAW = 8, RTP_PT_S16BE_STEREO = 10, RTP_PT_S16BE_MONO = 11, RTP_PT_MPEGAUDIO = 14, RTP_PT_JPEG = 26, RTP_PT_H261 = 31, RTP_PT_MPEGVIDEO = 32, RTP_PT_MPEG2TS = 33, RTP_PT_H263 = 34, /* old H263 encapsulation */ RTP_PT_PRIVATE = 96,};typedef struct RTPContext { int payload_type; UINT32 ssrc; UINT16 seq; UINT32 timestamp; UINT32 base_timestamp; UINT32 cur_timestamp; int max_payload_size; /* rtcp sender statistics receive */ INT64 last_rtcp_ntp_time; UINT32 last_rtcp_timestamp; /* rtcp sender statistics */ unsigned int packet_count; unsigned int octet_count; unsigned int last_octet_count; int first_packet; /* buffer for output */ UINT8 buf[RTP_MAX_PACKET_LENGTH]; UINT8 *buf_ptr;} RTPContext;int rtp_get_codec_info(AVCodecContext *codec, int payload_type){ switch(payload_type) { case RTP_PT_ULAW: codec->codec_id = CODEC_ID_PCM_MULAW; codec->channels = 1; codec->sample_rate = 8000; break; case RTP_PT_ALAW: codec->codec_id = CODEC_ID_PCM_ALAW; codec->channels = 1; codec->sample_rate = 8000; break; case RTP_PT_S16BE_STEREO: codec->codec_id = CODEC_ID_PCM_S16BE; codec->channels = 2; codec->sample_rate = 44100; break; case RTP_PT_S16BE_MONO: codec->codec_id = CODEC_ID_PCM_S16BE; codec->channels = 1; codec->sample_rate = 44100; break; case RTP_PT_MPEGAUDIO: codec->codec_id = CODEC_ID_MP2; break; case RTP_PT_JPEG: codec->codec_id = CODEC_ID_MJPEG; break; case RTP_PT_MPEGVIDEO: codec->codec_id = CODEC_ID_MPEG1VIDEO; break; default: return -1; } return 0;}/* return < 0 if unknown payload type */int rtp_get_payload_type(AVCodecContext *codec){ int payload_type; /* compute the payload type */ payload_type = -1; switch(codec->codec_id) { case CODEC_ID_PCM_MULAW: payload_type = RTP_PT_ULAW; break; case CODEC_ID_PCM_ALAW: payload_type = RTP_PT_ALAW; break; case CODEC_ID_PCM_S16BE: if (codec->channels == 1) { payload_type = RTP_PT_S16BE_MONO; } else if (codec->channels == 2) { payload_type = RTP_PT_S16BE_STEREO; } break; case CODEC_ID_MP2: case CODEC_ID_MP3LAME: payload_type = RTP_PT_MPEGAUDIO; break; case CODEC_ID_MJPEG: payload_type = RTP_PT_JPEG; break; case CODEC_ID_MPEG1VIDEO: payload_type = RTP_PT_MPEGVIDEO; break; default: break; } return payload_type;}static inline UINT32 decode_be32(const UINT8 *p){ return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];}static inline UINT32 decode_be64(const UINT8 *p){ return ((UINT64)decode_be32(p) << 32) | decode_be32(p + 4);}static int rtcp_parse_packet(AVFormatContext *s1, const unsigned char *buf, int len){ RTPContext *s = s1->priv_data; if (buf[1] != 200) return -1; s->last_rtcp_ntp_time = decode_be64(buf + 8); s->last_rtcp_timestamp = decode_be32(buf + 16); return 0;}/** * Parse an RTP packet directly sent as raw data. Can only be used if * 'raw' is given as input file * @param s1 media file context * @param pkt returned packet * @param buf input buffer * @param len buffer len * @return zero if no error. */int rtp_parse_packet(AVFormatContext *s1, AVPacket *pkt, const unsigned char *buf, int len){ RTPContext *s = s1->priv_data; unsigned int ssrc, h; int payload_type, seq, delta_timestamp; AVStream *st; UINT32 timestamp; if (len < 12) return -1; if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) return -1; if (buf[1] >= 200 && buf[1] <= 204) { rtcp_parse_packet(s1, buf, len); return -1; } payload_type = buf[1] & 0x7f; seq = (buf[2] << 8) | buf[3]; timestamp = decode_be32(buf + 4); ssrc = decode_be32(buf + 8); if (s->payload_type < 0) { s->payload_type = payload_type; if (payload_type == RTP_PT_MPEG2TS) { /* XXX: special case : not a single codec but a whole stream */ return -1; } else { st = av_new_stream(s1, 0); if (!st) return -1; rtp_get_codec_info(&st->codec, payload_type); } } /* NOTE: we can handle only one payload type */ if (s->payload_type != payload_type) return -1;#if defined(DEBUG) || 1 if (seq != ((s->seq + 1) & 0xffff)) { printf("RTP: PT=%02x: bad cseq %04x expected=%04x\n", payload_type, seq, ((s->seq + 1) & 0xffff)); } s->seq = seq;#endif len -= 12; buf += 12; st = s1->streams[0]; switch(st->codec.codec_id) { case CODEC_ID_MP2: /* better than nothing: skip mpeg audio RTP header */ if (len <= 4) return -1; h = decode_be32(buf); len -= 4; buf += 4; av_new_packet(pkt, len); memcpy(pkt->data, buf, len); break; case CODEC_ID_MPEG1VIDEO: /* better than nothing: skip mpeg audio RTP header */ if (len <= 4) return -1; h = decode_be32(buf); buf += 4; len -= 4; if (h & (1 << 26)) { /* mpeg2 */ if (len <= 4) return -1; buf += 4; len -= 4; } av_new_packet(pkt, len); memcpy(pkt->data, buf, len); break; default: av_new_packet(pkt, len); memcpy(pkt->data, buf, len); break; } if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { /* compute pts from timestamp with received ntp_time */ delta_timestamp = timestamp - s->last_rtcp_timestamp; /* XXX: do conversion, but not needed for mpeg at 90 KhZ */ pkt->pts = s->last_rtcp_ntp_time + delta_timestamp; } return 0;}static int rtp_read_header(AVFormatContext *s1, AVFormatParameters *ap){ RTPContext *s = s1->priv_data; s->payload_type = -1; s->last_rtcp_ntp_time = AV_NOPTS_VALUE; return 0;}static int rtp_read_packet(AVFormatContext *s1, AVPacket *pkt){ char buf[RTP_MAX_PACKET_LENGTH]; int ret; /* XXX: needs a better API for packet handling ? */ for(;;) { ret = url_read(url_fileno(&s1->pb), buf, sizeof(buf)); if (ret < 0) return AVERROR_IO; if (rtp_parse_packet(s1, pkt, buf, ret) == 0) break; } return 0;}static int rtp_read_close(AVFormatContext *s1){ // RTPContext *s = s1->priv_data; return 0;}static int rtp_probe(AVProbeData *p){ if (strstart(p->filename, "rtp://", NULL)) return AVPROBE_SCORE_MAX;
⌨️ 快捷键说明
复制代码
Ctrl + C
搜索代码
Ctrl + F
全屏模式
F11
切换主题
Ctrl + Shift + D
显示快捷键
?
增大字号
Ctrl + =
减小字号
Ctrl + -