/* * RTSP/SDP client * 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 */ #include "avformat.h" #include #include /* for select() prototype */ #include "network.h" #include "rtp_internal.h" //#define DEBUG //#define DEBUG_RTP_TCP enum RTSPClientState { RTSP_STATE_IDLE, RTSP_STATE_PLAYING, RTSP_STATE_PAUSED, }; typedef struct RTSPState { URLContext *rtsp_hd; /* RTSP TCP connexion handle */ int nb_rtsp_streams; struct RTSPStream **rtsp_streams; enum RTSPClientState state; int64_t seek_timestamp; /* XXX: currently we use unbuffered input */ // ByteIOContext rtsp_gb; int seq; /* RTSP command sequence number */ char session_id[512]; enum RTSPProtocol protocol; char last_reply[2048]; /* XXX: allocate ? */ RTPDemuxContext *cur_rtp; } RTSPState; typedef struct RTSPStream { URLContext *rtp_handle; /* RTP stream handle */ RTPDemuxContext *rtp_ctx; /* RTP parse context */ int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */ int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */ char control_url[1024]; /* url for this stream (from SDP) */ int sdp_port; /* port (from SDP content - not used in RTSP) */ struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */ int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */ int sdp_payload_type; /* payload type - only used in SDP */ rtp_payload_data_t rtp_payload_data; /* rtp payload parsing infos from SDP */ RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure) void *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol) } RTSPStream; static int rtsp_read_play(AVFormatContext *s); /* XXX: currently, the only way to change the protocols consists in changing this variable */ int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_UDP); FFRTSPCallback *ff_rtsp_callback = NULL; static int rtsp_probe(AVProbeData *p) { if (strstart(p->filename, "rtsp:", NULL)) return AVPROBE_SCORE_MAX; return 0; } static int redir_isspace(int c) { return (c == ' ' || c == '\t' || c == '\n' || c == '\r'); } static void skip_spaces(const char **pp) { const char *p; p = *pp; while (redir_isspace(*p)) p++; *pp = p; } static void get_word_sep(char *buf, int buf_size, const char *sep, const char **pp) { const char *p; char *q; p = *pp; if (*p == '/') p++; skip_spaces(&p); q = buf; while (!strchr(sep, *p) && *p != '\0') { if ((q - buf) < buf_size - 1) *q++ = *p; p++; } if (buf_size > 0) *q = '\0'; *pp = p; } static void get_word(char *buf, int buf_size, const char **pp) { const char *p; char *q; p = *pp; skip_spaces(&p); q = buf; while (!redir_isspace(*p) && *p != '\0') { if ((q - buf) < buf_size - 1) *q++ = *p; p++; } if (buf_size > 0) *q = '\0'; *pp = p; } /* parse the rtpmap description: /[/] */ static int sdp_parse_rtpmap(AVCodecContext *codec, RTSPStream *rtsp_st, int payload_type, const char *p) { char buf[256]; int i; AVCodec *c; const char *c_name; /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and see if we can handle this kind of payload */ get_word_sep(buf, sizeof(buf), "/", &p); if (payload_type >= RTP_PT_PRIVATE) { RTPDynamicProtocolHandler *handler= RTPFirstDynamicPayloadHandler; while(handler) { if (!strcmp(buf, handler->enc_name) && (codec->codec_type == handler->codec_type)) { codec->codec_id = handler->codec_id; rtsp_st->dynamic_handler= handler; if(handler->open) { rtsp_st->dynamic_protocol_context= handler->open(); } break; } handler= handler->next; } } else { /* We are in a standard case ( from http://www.iana.org/assignments/rtp-parameters) */ /* search into AVRtpPayloadTypes[] */ for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i) if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpPayloadTypes[i].codec_type)){ codec->codec_id = AVRtpPayloadTypes[i].codec_id; break; } } c = avcodec_find_decoder(codec->codec_id); if (c && c->name) c_name = c->name; else c_name = (char *)NULL; if (c_name) { get_word_sep(buf, sizeof(buf), "/", &p); i = atoi(buf); switch (codec->codec_type) { case CODEC_TYPE_AUDIO: av_log(codec, AV_LOG_DEBUG, " audio codec set to : %s\n", c_name); codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE; codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS; if (i > 0) { codec->sample_rate = i; get_word_sep(buf, sizeof(buf), "/", &p); i = atoi(buf); if (i > 0) codec->channels = i; // TODO: there is a bug here; if it is a mono stream, and less than 22000Hz, faad upconverts to stereo and twice the // frequency. No problem, but the sample rate is being set here by the sdp line. Upcoming patch forthcoming. (rdm) } av_log(codec, AV_LOG_DEBUG, " audio samplerate set to : %i\n", codec->sample_rate); av_log(codec, AV_LOG_DEBUG, " audio channels set to : %i\n", codec->channels); break; case CODEC_TYPE_VIDEO: av_log(codec, AV_LOG_DEBUG, " video codec set to : %s\n", c_name); break; default: break; } return 0; } return -1; } /* return the length and optionnaly the data */ static int hex_to_data(uint8_t *data, const char *p) { int c, len, v; len = 0; v = 1; for(;;) { skip_spaces(&p); if (p == '\0') break; c = toupper((unsigned char)*p++); if (c >= '0' && c <= '9') c = c - '0'; else if (c >= 'A' && c <= 'F') c = c - 'A' + 10; else break; v = (v << 4) | c; if (v & 0x100) { if (data) data[len] = v; len++; v = 1; } } return len; } static void sdp_parse_fmtp_config(AVCodecContext *codec, char *attr, char *value) { switch (codec->codec_id) { case CODEC_ID_MPEG4: case CODEC_ID_AAC: if (!strcmp(attr, "config")) { /* decode the hexa encoded parameter */ int len = hex_to_data(NULL, value); codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE); if (!codec->extradata) return; codec->extradata_size = len; hex_to_data(codec->extradata, value); } break; default: break; } return; } typedef struct attrname_map { const char *str; uint16_t type; uint32_t offset; } attrname_map_t; /* All known fmtp parmeters and the corresping RTPAttrTypeEnum */ #define ATTR_NAME_TYPE_INT 0 #define ATTR_NAME_TYPE_STR 1 static attrname_map_t attr_names[]= { {"SizeLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, sizelength)}, {"IndexLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexlength)}, {"IndexDeltaLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexdeltalength)}, {"profile-level-id", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, profile_level_id)}, {"StreamType", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, streamtype)}, {"mode", ATTR_NAME_TYPE_STR, offsetof(rtp_payload_data_t, mode)}, {NULL, -1, -1}, }; /** parse the attribute line from the fmtp a line of an sdp resonse. This is broken out as a function * because it is used in rtp_h264.c, which is forthcoming. */ int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size) { skip_spaces(p); if(**p) { get_word_sep(attr, attr_size, "=", p); if (**p == '=') (*p)++; get_word_sep(value, value_size, ";", p); if (**p == ';') (*p)++; return 1; } return 0; } /* parse a SDP line and save stream attributes */ static void sdp_parse_fmtp(AVStream *st, const char *p) { char attr[256]; char value[4096]; int i; RTSPStream *rtsp_st = st->priv_data; AVCodecContext *codec = st->codec; rtp_payload_data_t *rtp_payload_data = &rtsp_st->rtp_payload_data; /* loop on each attribute */ while(rtsp_next_attr_and_value(&p, attr, sizeof(attr), value, sizeof(value))) { /* grab the codec extra_data from the config parameter of the fmtp line */ sdp_parse_fmtp_config(codec, attr, value); /* Looking for a known attribute */ for (i = 0; attr_names[i].str; ++i) { if (!strcasecmp(attr, attr_names[i].str)) { if (attr_names[i].type == ATTR_NAME_TYPE_INT) *(int *)((char *)rtp_payload_data + attr_names[i].offset) = atoi(value); else if (attr_names[i].type == ATTR_NAME_TYPE_STR) *(char **)((char *)rtp_payload_data + attr_names[i].offset) = av_strdup(value); } } } } /** Parse a string \p in the form of Range:npt=xx-xx, and determine the start * and end time. * Used for seeking in the rtp stream. */ static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end) { char buf[256]; skip_spaces(&p); if (!stristart(p, "npt=", &p)) return; *start = AV_NOPTS_VALUE; *end = AV_NOPTS_VALUE; get_word_sep(buf, sizeof(buf), "-", &p); *start = parse_date(buf, 1); if (*p == '-') { p++; get_word_sep(buf, sizeof(buf), "-", &p); *end = parse_date(buf, 1); } // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start); // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end); } typedef struct SDPParseState { /* SDP only */ struct in_addr default_ip; int default_ttl; } SDPParseState; static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1, int letter, const char *buf) { RTSPState *rt = s->priv_data; char buf1[64], st_type[64]; const char *p; int codec_type, payload_type, i; AVStream *st; RTSPStream *rtsp_st; struct in_addr sdp_ip; int ttl; #ifdef DEBUG printf("sdp: %c='%s'\n", letter, buf); #endif p = buf; switch(letter) { case 'c': get_word(buf1, sizeof(buf1), &p); if (strcmp(buf1, "IN") != 0) return; get_word(buf1, sizeof(buf1), &p); if (strcmp(buf1, "IP4") != 0) return; get_word_sep(buf1, sizeof(buf1), "/", &p); if (inet_aton(buf1, &sdp_ip) == 0) return; ttl = 16; if (*p == '/') { p++; get_word_sep(buf1, sizeof(buf1), "/", &p); ttl = atoi(buf1); } if (s->nb_streams == 0) { s1->default_ip = sdp_ip; s1->default_ttl = ttl; } else { st = s->streams[s->nb_streams - 1]; rtsp_st = st->priv_data; rtsp_st->sdp_ip = sdp_ip; rtsp_st->sdp_ttl = ttl; } break; case 's': pstrcpy(s->title, sizeof(s->title), p); break; case 'i': if (s->nb_streams == 0) { pstrcpy(s->comment, sizeof(s->comment), p); break; } break; case 'm': /* new stream */ get_word(st_type, sizeof(st_type), &p); if (!strcmp(st_type, "audio")) { codec_type = CODEC_TYPE_AUDIO; } else if (!strcmp(st_type, "video")) { codec_type = CODEC_TYPE_VIDEO; } else { return; } rtsp_st = av_mallocz(sizeof(RTSPStream)); if (!rtsp_st) return; rtsp_st->stream_index = -1; dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st); rtsp_st->sdp_ip = s1->default_ip; rtsp_st->sdp_ttl = s1->default_ttl; get_word(buf1, sizeof(buf1), &p); /* port */ rtsp_st->sdp_port = atoi(buf1); get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */ /* XXX: handle list of formats */ get_word(buf1, sizeof(buf1), &p); /* format list */ rtsp_st->sdp_payload_type = atoi(buf1); if (!strcmp(AVRtpPayloadTypes[rtsp_st->sdp_payload_type].enc_name, "MP2T")) { /* no corresponding stream */ } else { st = av_new_stream(s, 0); if (!st) return; st->priv_data = rtsp_st; rtsp_st->stream_index = st->index; st->codec->codec_type = codec_type; if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) { /* if standard payload type, we can find the codec right now */ rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type); } } /* put a default control url */ pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), s->filename); break; case 'a': if (strstart(p, "control:", &p) && s->nb_streams > 0) { char proto[32]; /* get the control url */ st = s->streams[s->nb_streams - 1]; rtsp_st = st->priv_data; /* XXX: may need to add full url resolution */ url_split(proto, sizeof(proto), NULL, 0, NULL, 0, NULL, NULL, 0, p); if (proto[0] == '\0') { /* relative control URL */ pstrcat(rtsp_st->control_url, sizeof(rtsp_st->control_url), "/"); pstrcat(rtsp_st->control_url, sizeof(rtsp_st->control_url), p); } else { pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), p); } } else if (strstart(p, "rtpmap:", &p)) { /* NOTE: rtpmap is only supported AFTER the 'm=' tag */ get_word(buf1, sizeof(buf1), &p); payload_type = atoi(buf1); for(i = 0; i < s->nb_streams;i++) { st = s->streams[i]; rtsp_st = st->priv_data; if (rtsp_st->sdp_payload_type == payload_type) { sdp_parse_rtpmap(st->codec, rtsp_st, payload_type, p); } } } else if (strstart(p, "fmtp:", &p)) { /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */ get_word(buf1, sizeof(buf1), &p); payload_type = atoi(buf1); for(i = 0; i < s->nb_streams;i++) { st = s->streams[i]; rtsp_st = st->priv_data; if (rtsp_st->sdp_payload_type == payload_type) { if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) { if(!rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf)) { sdp_parse_fmtp(st, p); } } else { sdp_parse_fmtp(st, p); } } } } else if(strstart(p, "framesize:", &p)) { // let dynamic protocol handlers have a stab at the line. get_word(buf1, sizeof(buf1), &p); payload_type = atoi(buf1); for(i = 0; i < s->nb_streams;i++) { st = s->streams[i]; rtsp_st = st->priv_data; if (rtsp_st->sdp_payload_type == payload_type) { if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) { rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf); } } } } else if(strstart(p, "range:", &p)) { int64_t start, end; // this is so that seeking on a streamed file can work. rtsp_parse_range_npt(p, &start, &end); s->start_time= start; s->duration= (end==AV_NOPTS_VALUE)?AV_NOPTS_VALUE:end-start; // AV_NOPTS_VALUE means live broadcast (and can't seek) } break; } } static int sdp_parse(AVFormatContext *s, const char *content) { const char *p; int letter; char buf[1024], *q; SDPParseState sdp_parse_state, *s1 = &sdp_parse_state; memset(s1, 0, sizeof(SDPParseState)); p = content; for(;;) { skip_spaces(&p); letter = *p; if (letter == '\0') break; p++; if (*p != '=') goto next_line; p++; /* get the content */ q = buf; while (*p != '\n' && *p != '\r' && *p != '\0') { if ((q - buf) < sizeof(buf) - 1) *q++ = *p; p++; } *q = '\0'; sdp_parse_line(s, s1, letter, buf); next_line: while (*p != '\n' && *p != '\0') p++; if (*p == '\n') p++; } return 0; } static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp) { const char *p; int v; p = *pp; skip_spaces(&p); v = strtol(p, (char **)&p, 10); if (*p == '-') { p++; *min_ptr = v; v = strtol(p, (char **)&p, 10); *max_ptr = v; } else { *min_ptr = v; *max_ptr = v; } *pp = p; } /* XXX: only one transport specification is parsed */ static void rtsp_parse_transport(RTSPHeader *reply, const char *p) { char transport_protocol[16]; char profile[16]; char lower_transport[16]; char parameter[16]; RTSPTransportField *th; char buf[256]; reply->nb_transports = 0; for(;;) { skip_spaces(&p); if (*p == '\0') break; th = &reply->transports[reply->nb_transports]; get_word_sep(transport_protocol, sizeof(transport_protocol), "/", &p); if (*p == '/') p++; get_word_sep(profile, sizeof(profile), "/;,", &p); lower_transport[0] = '\0'; if (*p == '/') { p++; get_word_sep(lower_transport, sizeof(lower_transport), ";,", &p); } if (!strcasecmp(lower_transport, "TCP")) th->protocol = RTSP_PROTOCOL_RTP_TCP; else th->protocol = RTSP_PROTOCOL_RTP_UDP; if (*p == ';') p++; /* get each parameter */ while (*p != '\0' && *p != ',') { get_word_sep(parameter, sizeof(parameter), "=;,", &p); if (!strcmp(parameter, "port")) { if (*p == '=') { p++; rtsp_parse_range(&th->port_min, &th->port_max, &p); } } else if (!strcmp(parameter, "client_port")) { if (*p == '=') { p++; rtsp_parse_range(&th->client_port_min, &th->client_port_max, &p); } } else if (!strcmp(parameter, "server_port")) { if (*p == '=') { p++; rtsp_parse_range(&th->server_port_min, &th->server_port_max, &p); } } else if (!strcmp(parameter, "interleaved")) { if (*p == '=') { p++; rtsp_parse_range(&th->interleaved_min, &th->interleaved_max, &p); } } else if (!strcmp(parameter, "multicast")) { if (th->protocol == RTSP_PROTOCOL_RTP_UDP) th->protocol = RTSP_PROTOCOL_RTP_UDP_MULTICAST; } else if (!strcmp(parameter, "ttl")) { if (*p == '=') { p++; th->ttl = strtol(p, (char **)&p, 10); } } else if (!strcmp(parameter, "destination")) { struct in_addr ipaddr; if (*p == '=') { p++; get_word_sep(buf, sizeof(buf), ";,", &p); if (inet_aton(buf, &ipaddr)) th->destination = ntohl(ipaddr.s_addr); } } while (*p != ';' && *p != '\0' && *p != ',') p++; if (*p == ';') p++; } if (*p == ',') p++; reply->nb_transports++; } } void rtsp_parse_line(RTSPHeader *reply, const char *buf) { const char *p; /* NOTE: we do case independent match for broken servers */ p = buf; if (stristart(p, "Session:", &p)) { get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p); } else if (stristart(p, "Content-Length:", &p)) { reply->content_length = strtol(p, NULL, 10); } else if (stristart(p, "Transport:", &p)) { rtsp_parse_transport(reply, p); } else if (stristart(p, "CSeq:", &p)) { reply->seq = strtol(p, NULL, 10); } else if (stristart(p, "Range:", &p)) { rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end); } } static int url_readbuf(URLContext *h, unsigned char *buf, int size) { int ret, len; len = 0; while (len < size) { ret = url_read(h, buf+len, size-len); if (ret < 1) return ret; len += ret; } return len; } /* skip a RTP/TCP interleaved packet */ static void rtsp_skip_packet(AVFormatContext *s) { RTSPState *rt = s->priv_data; int ret, len, len1; uint8_t buf[1024]; ret = url_readbuf(rt->rtsp_hd, buf, 3); if (ret != 3) return; len = (buf[1] << 8) | buf[2]; #ifdef DEBUG printf("skipping RTP packet len=%d\n", len); #endif /* skip payload */ while (len > 0) { len1 = len; if (len1 > sizeof(buf)) len1 = sizeof(buf); ret = url_readbuf(rt->rtsp_hd, buf, len1); if (ret != len1) return; len -= len1; } } static void rtsp_send_cmd(AVFormatContext *s, const char *cmd, RTSPHeader *reply, unsigned char **content_ptr) { RTSPState *rt = s->priv_data; char buf[4096], buf1[1024], *q; unsigned char ch; const char *p; int content_length, line_count; unsigned char *content = NULL; memset(reply, 0, sizeof(RTSPHeader)); rt->seq++; pstrcpy(buf, sizeof(buf), cmd); snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq); pstrcat(buf, sizeof(buf), buf1); if (rt->session_id[0] != '\0' && !strstr(cmd, "\nIf-Match:")) { snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id); pstrcat(buf, sizeof(buf), buf1); } pstrcat(buf, sizeof(buf), "\r\n"); #ifdef DEBUG printf("Sending:\n%s--\n", buf); #endif url_write(rt->rtsp_hd, buf, strlen(buf)); /* parse reply (XXX: use buffers) */ line_count = 0; rt->last_reply[0] = '\0'; for(;;) { q = buf; for(;;) { if (url_readbuf(rt->rtsp_hd, &ch, 1) != 1) break; if (ch == '\n') break; if (ch == '$') { /* XXX: only parse it if first char on line ? */ rtsp_skip_packet(s); } else if (ch != '\r') { if ((q - buf) < sizeof(buf) - 1) *q++ = ch; } } *q = '\0'; #ifdef DEBUG printf("line='%s'\n", buf); #endif /* test if last line */ if (buf[0] == '\0') break; p = buf; if (line_count == 0) { /* get reply code */ get_word(buf1, sizeof(buf1), &p); get_word(buf1, sizeof(buf1), &p); reply->status_code = atoi(buf1); } else { rtsp_parse_line(reply, p); pstrcat(rt->last_reply, sizeof(rt->last_reply), p); pstrcat(rt->last_reply, sizeof(rt->last_reply), "\n"); } line_count++; } if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0') pstrcpy(rt->session_id, sizeof(rt->session_id), reply->session_id); content_length = reply->content_length; if (content_length > 0) { /* leave some room for a trailing '\0' (useful for simple parsing) */ content = av_malloc(content_length + 1); (void)url_readbuf(rt->rtsp_hd, content, content_length); content[content_length] = '\0'; } if (content_ptr) *content_ptr = content; } /* useful for modules: set RTSP callback function */ void rtsp_set_callback(FFRTSPCallback *rtsp_cb) { ff_rtsp_callback = rtsp_cb; } /* close and free RTSP streams */ static void rtsp_close_streams(RTSPState *rt) { int i; RTSPStream *rtsp_st; for(i=0;inb_rtsp_streams;i++) { rtsp_st = rt->rtsp_streams[i]; if (rtsp_st) { if (rtsp_st->rtp_ctx) rtp_parse_close(rtsp_st->rtp_ctx); if (rtsp_st->rtp_handle) url_close(rtsp_st->rtp_handle); if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context) rtsp_st->dynamic_handler->close(rtsp_st->dynamic_protocol_context); } av_free(rtsp_st); } av_free(rt->rtsp_streams); } static int rtsp_read_header(AVFormatContext *s, AVFormatParameters *ap) { RTSPState *rt = s->priv_data; char host[1024], path[1024], tcpname[1024], cmd[2048]; URLContext *rtsp_hd; int port, i, j, ret, err; RTSPHeader reply1, *reply = &reply1; unsigned char *content = NULL; RTSPStream *rtsp_st; int protocol_mask; AVStream *st; /* extract hostname and port */ url_split(NULL, 0, NULL, 0, host, sizeof(host), &port, path, sizeof(path), s->filename); if (port < 0) port = RTSP_DEFAULT_PORT; /* open the tcp connexion */ snprintf(tcpname, sizeof(tcpname), "tcp://%s:%d", host, port); if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0) return AVERROR_IO; rt->rtsp_hd = rtsp_hd; rt->seq = 0; /* describe the stream */ snprintf(cmd, sizeof(cmd), "DESCRIBE %s RTSP/1.0\r\n" "Accept: application/sdp\r\n", s->filename); rtsp_send_cmd(s, cmd, reply, &content); if (!content) { err = AVERROR_INVALIDDATA; goto fail; } if (reply->status_code != RTSP_STATUS_OK) { err = AVERROR_INVALIDDATA; goto fail; } /* now we got the SDP description, we parse it */ ret = sdp_parse(s, (const char *)content); av_freep(&content); if (ret < 0) { err = AVERROR_INVALIDDATA; goto fail; } protocol_mask = rtsp_default_protocols; /* for each stream, make the setup request */ /* XXX: we assume the same server is used for the control of each RTSP stream */ for(j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) { char transport[2048]; rtsp_st = rt->rtsp_streams[i]; /* compute available transports */ transport[0] = '\0'; /* RTP/UDP */ if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP)) { char buf[256]; /* first try in specified port range */ if (RTSP_RTP_PORT_MIN != 0) { while(j <= RTSP_RTP_PORT_MAX) { snprintf(buf, sizeof(buf), "rtp://?localport=%d", j); if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0) { j += 2; /* we will use two port by rtp stream (rtp and rtcp) */ goto rtp_opened; } } } /* then try on any port ** if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) { ** err = AVERROR_INVALIDDATA; ** goto fail; ** } */ rtp_opened: port = rtp_get_local_port(rtsp_st->rtp_handle); if (transport[0] != '\0') pstrcat(transport, sizeof(transport), ","); snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1, "RTP/AVP/UDP;unicast;client_port=%d-%d", port, port + 1); } /* RTP/TCP */ else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_TCP)) { if (transport[0] != '\0') pstrcat(transport, sizeof(transport), ","); snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1, "RTP/AVP/TCP"); } else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST)) { if (transport[0] != '\0') pstrcat(transport, sizeof(transport), ","); snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1, "RTP/AVP/UDP;multicast"); } snprintf(cmd, sizeof(cmd), "SETUP %s RTSP/1.0\r\n" "Transport: %s\r\n", rtsp_st->control_url, transport); rtsp_send_cmd(s, cmd, reply, NULL); if (reply->status_code != RTSP_STATUS_OK || reply->nb_transports != 1) { err = AVERROR_INVALIDDATA; goto fail; } /* XXX: same protocol for all streams is required */ if (i > 0) { if (reply->transports[0].protocol != rt->protocol) { err = AVERROR_INVALIDDATA; goto fail; } } else { rt->protocol = reply->transports[0].protocol; } /* close RTP connection if not choosen */ if (reply->transports[0].protocol != RTSP_PROTOCOL_RTP_UDP && (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP))) { url_close(rtsp_st->rtp_handle); rtsp_st->rtp_handle = NULL; } switch(reply->transports[0].protocol) { case RTSP_PROTOCOL_RTP_TCP: rtsp_st->interleaved_min = reply->transports[0].interleaved_min; rtsp_st->interleaved_max = reply->transports[0].interleaved_max; break; case RTSP_PROTOCOL_RTP_UDP: { char url[1024]; /* XXX: also use address if specified */ snprintf(url, sizeof(url), "rtp://%s:%d", host, reply->transports[0].server_port_min); if (rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) { err = AVERROR_INVALIDDATA; goto fail; } } break; case RTSP_PROTOCOL_RTP_UDP_MULTICAST: { char url[1024]; int ttl; ttl = reply->transports[0].ttl; if (!ttl) ttl = 16; snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d", host, reply->transports[0].server_port_min, ttl); if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) { err = AVERROR_INVALIDDATA; goto fail; } } break; } /* open the RTP context */ st = NULL; if (rtsp_st->stream_index >= 0) st = s->streams[rtsp_st->stream_index]; if (!st) s->ctx_flags |= AVFMTCTX_NOHEADER; rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data); if (!rtsp_st->rtp_ctx) { err = AVERROR_NOMEM; goto fail; } else { if(rtsp_st->dynamic_handler) { rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context; rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet; } } } /* use callback if available to extend setup */ if (ff_rtsp_callback) { if (ff_rtsp_callback(RTSP_ACTION_CLIENT_SETUP, rt->session_id, NULL, 0, rt->last_reply) < 0) { err = AVERROR_INVALIDDATA; goto fail; } } rt->state = RTSP_STATE_IDLE; rt->seek_timestamp = 0; /* default is to start stream at position zero */ if (ap->initial_pause) { /* do not start immediately */ } else { if (rtsp_read_play(s) < 0) { err = AVERROR_INVALIDDATA; goto fail; } } return 0; fail: rtsp_close_streams(rt); av_freep(&content); url_close(rt->rtsp_hd); return err; } static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, uint8_t *buf, int buf_size) { RTSPState *rt = s->priv_data; int id, len, i, ret; RTSPStream *rtsp_st; #ifdef DEBUG_RTP_TCP printf("tcp_read_packet:\n"); #endif redo: for(;;) { ret = url_readbuf(rt->rtsp_hd, buf, 1); #ifdef DEBUG_RTP_TCP printf("ret=%d c=%02x [%c]\n", ret, buf[0], buf[0]); #endif if (ret != 1) return -1; if (buf[0] == '$') break; } ret = url_readbuf(rt->rtsp_hd, buf, 3); if (ret != 3) return -1; id = buf[0]; len = (buf[1] << 8) | buf[2]; #ifdef DEBUG_RTP_TCP printf("id=%d len=%d\n", id, len); #endif if (len > buf_size || len < 12) goto redo; /* get the data */ ret = url_readbuf(rt->rtsp_hd, buf, len); if (ret != len) return -1; /* find the matching stream */ for(i = 0; i < rt->nb_rtsp_streams; i++) { rtsp_st = rt->rtsp_streams[i]; if (id >= rtsp_st->interleaved_min && id <= rtsp_st->interleaved_max) goto found; } goto redo; found: *prtsp_st = rtsp_st; return len; } static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, uint8_t *buf, int buf_size) { RTSPState *rt = s->priv_data; RTSPStream *rtsp_st; fd_set rfds; int fd1, fd2, fd_max, n, i, ret; struct timeval tv; for(;;) { if (url_interrupt_cb()) return -1; FD_ZERO(&rfds); fd_max = -1; for(i = 0; i < rt->nb_rtsp_streams; i++) { rtsp_st = rt->rtsp_streams[i]; /* currently, we cannot probe RTCP handle because of blocking restrictions */ rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2); if (fd1 > fd_max) fd_max = fd1; FD_SET(fd1, &rfds); } tv.tv_sec = 0; tv.tv_usec = 100 * 1000; n = select(fd_max + 1, &rfds, NULL, NULL, &tv); if (n > 0) { for(i = 0; i < rt->nb_rtsp_streams; i++) { rtsp_st = rt->rtsp_streams[i]; rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2); if (FD_ISSET(fd1, &rfds)) { ret = url_read(rtsp_st->rtp_handle, buf, buf_size); if (ret > 0) { *prtsp_st = rtsp_st; return ret; } } } } } } static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt) { RTSPState *rt = s->priv_data; RTSPStream *rtsp_st; int ret, len; uint8_t buf[RTP_MAX_PACKET_LENGTH]; /* get next frames from the same RTP packet */ if (rt->cur_rtp) { ret = rtp_parse_packet(rt->cur_rtp, pkt, NULL, 0); if (ret == 0) { rt->cur_rtp = NULL; return 0; } else if (ret == 1) { return 0; } else { rt->cur_rtp = NULL; } } /* read next RTP packet */ redo: switch(rt->protocol) { default: case RTSP_PROTOCOL_RTP_TCP: len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf)); break; case RTSP_PROTOCOL_RTP_UDP: case RTSP_PROTOCOL_RTP_UDP_MULTICAST: len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf)); if (rtsp_st->rtp_ctx) rtp_check_and_send_back_rr(rtsp_st->rtp_ctx, len); break; } if (len < 0) return AVERROR_IO; ret = rtp_parse_packet(rtsp_st->rtp_ctx, pkt, buf, len); if (ret < 0) goto redo; if (ret == 1) { /* more packets may follow, so we save the RTP context */ rt->cur_rtp = rtsp_st->rtp_ctx; } return 0; } static int rtsp_read_play(AVFormatContext *s) { RTSPState *rt = s->priv_data; RTSPHeader reply1, *reply = &reply1; char cmd[1024]; av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state); if (rt->state == RTSP_STATE_PAUSED) { snprintf(cmd, sizeof(cmd), "PLAY %s RTSP/1.0\r\n", s->filename); } else { snprintf(cmd, sizeof(cmd), "PLAY %s RTSP/1.0\r\n" "Range: npt=%0.3f-\r\n", s->filename, (double)rt->seek_timestamp / AV_TIME_BASE); } rtsp_send_cmd(s, cmd, reply, NULL); if (reply->status_code != RTSP_STATUS_OK) { return -1; } else { rt->state = RTSP_STATE_PLAYING; return 0; } } /* pause the stream */ static int rtsp_read_pause(AVFormatContext *s) { RTSPState *rt = s->priv_data; RTSPHeader reply1, *reply = &reply1; char cmd[1024]; rt = s->priv_data; if (rt->state != RTSP_STATE_PLAYING) return 0; snprintf(cmd, sizeof(cmd), "PAUSE %s RTSP/1.0\r\n", s->filename); rtsp_send_cmd(s, cmd, reply, NULL); if (reply->status_code != RTSP_STATUS_OK) { return -1; } else { rt->state = RTSP_STATE_PAUSED; return 0; } } static int rtsp_read_seek(AVFormatContext *s, int stream_index, int64_t timestamp, int flags) { RTSPState *rt = s->priv_data; rt->seek_timestamp = timestamp; switch(rt->state) { default: case RTSP_STATE_IDLE: break; case RTSP_STATE_PLAYING: if (rtsp_read_play(s) != 0) return -1; break; case RTSP_STATE_PAUSED: rt->state = RTSP_STATE_IDLE; break; } return 0; } static int rtsp_read_close(AVFormatContext *s) { RTSPState *rt = s->priv_data; RTSPHeader reply1, *reply = &reply1; char cmd[1024]; #if 0 /* NOTE: it is valid to flush the buffer here */ if (rt->protocol == RTSP_PROTOCOL_RTP_TCP) { url_fclose(&rt->rtsp_gb); } #endif snprintf(cmd, sizeof(cmd), "TEARDOWN %s RTSP/1.0\r\n", s->filename); rtsp_send_cmd(s, cmd, reply, NULL); if (ff_rtsp_callback) { ff_rtsp_callback(RTSP_ACTION_CLIENT_TEARDOWN, rt->session_id, NULL, 0, NULL); } rtsp_close_streams(rt); url_close(rt->rtsp_hd); return 0; } AVInputFormat rtsp_demuxer = { "rtsp", "RTSP input format", sizeof(RTSPState), rtsp_probe, rtsp_read_header, rtsp_read_packet, rtsp_read_close, rtsp_read_seek, .flags = AVFMT_NOFILE, .read_play = rtsp_read_play, .read_pause = rtsp_read_pause, }; static int sdp_probe(AVProbeData *p1) { const char *p = p1->buf, *p_end = p1->buf + p1->buf_size; /* we look for a line beginning "c=IN IP4" */ while (p < p_end && *p != '\0') { if (p + sizeof("c=IN IP4") - 1 < p_end && strstart(p, "c=IN IP4", NULL)) return AVPROBE_SCORE_MAX / 2; while(p < p_end - 1 && *p != '\n') p++; if (++p >= p_end) break; if (*p == '\r') p++; } return 0; } #define SDP_MAX_SIZE 8192 static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap) { RTSPState *rt = s->priv_data; RTSPStream *rtsp_st; int size, i, err; char *content; char url[1024]; AVStream *st; /* read the whole sdp file */ /* XXX: better loading */ content = av_malloc(SDP_MAX_SIZE); size = get_buffer(&s->pb, content, SDP_MAX_SIZE - 1); if (size <= 0) { av_free(content); return AVERROR_INVALIDDATA; } content[size] ='\0'; sdp_parse(s, content); av_free(content); /* open each RTP stream */ for(i=0;inb_rtsp_streams;i++) { rtsp_st = rt->rtsp_streams[i]; snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d", inet_ntoa(rtsp_st->sdp_ip), rtsp_st->sdp_port, rtsp_st->sdp_ttl); if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) { err = AVERROR_INVALIDDATA; goto fail; } /* open the RTP context */ st = NULL; if (rtsp_st->stream_index >= 0) st = s->streams[rtsp_st->stream_index]; if (!st) s->ctx_flags |= AVFMTCTX_NOHEADER; rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data); if (!rtsp_st->rtp_ctx) { err = AVERROR_NOMEM; goto fail; } else { if(rtsp_st->dynamic_handler) { rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context; rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet; } } } return 0; fail: rtsp_close_streams(rt); return err; } static int sdp_read_packet(AVFormatContext *s, AVPacket *pkt) { return rtsp_read_packet(s, pkt); } static int sdp_read_close(AVFormatContext *s) { RTSPState *rt = s->priv_data; rtsp_close_streams(rt); return 0; } #ifdef CONFIG_SDP_DEMUXER AVInputFormat sdp_demuxer = { "sdp", "SDP", sizeof(RTSPState), sdp_probe, sdp_read_header, sdp_read_packet, sdp_read_close, }; #endif /* dummy redirector format (used directly in av_open_input_file now) */ static int redir_probe(AVProbeData *pd) { const char *p; p = pd->buf; while (redir_isspace(*p)) p++; if (strstart(p, "http://", NULL) || strstart(p, "rtsp://", NULL)) return AVPROBE_SCORE_MAX; return 0; } /* called from utils.c */ int redir_open(AVFormatContext **ic_ptr, ByteIOContext *f) { char buf[4096], *q; int c; AVFormatContext *ic = NULL; /* parse each URL and try to open it */ c = url_fgetc(f); while (c != URL_EOF) { /* skip spaces */ for(;;) { if (!redir_isspace(c)) break; c = url_fgetc(f); } if (c == URL_EOF) break; /* record url */ q = buf; for(;;) { if (c == URL_EOF || redir_isspace(c)) break; if ((q - buf) < sizeof(buf) - 1) *q++ = c; c = url_fgetc(f); } *q = '\0'; //printf("URL='%s'\n", buf); /* try to open the media file */ if (av_open_input_file(&ic, buf, NULL, 0, NULL) == 0) break; } *ic_ptr = ic; if (!ic) return AVERROR_IO; else return 0; } AVInputFormat redir_demuxer = { "redir", "Redirector format", 0, redir_probe, NULL, NULL, NULL, };