Mercurial > libavformat.hg
view rtsp.c @ 6408:3a89d4044e01 libavformat
Simplify resolve_destination in sdp.c further, now that we don't enforce IPv4 any longer
| author | mstorsjo |
|---|---|
| date | Wed, 25 Aug 2010 13:30:06 +0000 |
| parents | f8c3cb8c503e |
| children | 31aef55204bd |
line wrap: on
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/* * 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 "libavutil/base64.h" #include "libavutil/avstring.h" #include "libavutil/intreadwrite.h" #include "libavutil/random_seed.h" #include "avformat.h" #include <sys/time.h> #if HAVE_SYS_SELECT_H #include <sys/select.h> #endif #include <strings.h> #include "internal.h" #include "network.h" #include "os_support.h" #include "http.h" #include "rtsp.h" #include "rtpdec.h" #include "rdt.h" #include "rtpdec_formats.h" //#define DEBUG //#define DEBUG_RTP_TCP #if LIBAVFORMAT_VERSION_INT < (53 << 16) int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP); #endif /* Timeout values for socket select, in ms, * and read_packet(), in seconds */ #define SELECT_TIMEOUT_MS 100 #define READ_PACKET_TIMEOUT_S 10 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / SELECT_TIMEOUT_MS #define SDP_MAX_SIZE 16384 static void get_word_until_chars(char *buf, int buf_size, const char *sep, const char **pp) { const char *p; char *q; p = *pp; p += strspn(p, SPACE_CHARS); 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_sep(char *buf, int buf_size, const char *sep, const char **pp) { if (**pp == '/') (*pp)++; get_word_until_chars(buf, buf_size, sep, pp); } static void get_word(char *buf, int buf_size, const char **pp) { get_word_until_chars(buf, buf_size, SPACE_CHARS, pp); } /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */ static int sdp_parse_rtpmap(AVFormatContext *s, 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. * The space should normally not be there but some Real streams or * particular servers ("RealServer Version 6.1.3.970", see issue 1658) * have a trailing space. */ get_word_sep(buf, sizeof(buf), "/ ", &p); if (payload_type >= RTP_PT_PRIVATE) { RTPDynamicProtocolHandler *handler; for (handler = RTPFirstDynamicPayloadHandler; handler; handler = handler->next) { if (!strcasecmp(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; } } } else { /* We are in a standard case * (from http://www.iana.org/assignments/rtp-parameters). */ /* search into AVRtpPayloadTypes[] */ codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type); } c = avcodec_find_decoder(codec->codec_id); if (c && c->name) c_name = c->name; else c_name = "(null)"; get_word_sep(buf, sizeof(buf), "/", &p); i = atoi(buf); switch (codec->codec_type) { case AVMEDIA_TYPE_AUDIO: av_log(s, 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. Patch on its way. (rdm) } av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n", codec->sample_rate); av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n", codec->channels); break; case AVMEDIA_TYPE_VIDEO: av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name); break; default: break; } return 0; } /* parse the attribute line from the fmtp a line of an sdp response. This * is broken out as a function because it is used in rtp_h264.c, which is * forthcoming. */ int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size) { *p += strspn(*p, SPACE_CHARS); 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 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]; p += strspn(p, SPACE_CHARS); if (!av_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; int skip_media; ///< set if an unknown m= line occurs } 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; enum AVMediaType codec_type; int payload_type, i; AVStream *st; RTSPStream *rtsp_st; struct in_addr sdp_ip; int ttl; dprintf(s, "sdp: %c='%s'\n", letter, buf); p = buf; if (s1->skip_media && letter != 'm') return; 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 (ff_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': av_metadata_set2(&s->metadata, "title", p, 0); break; case 'i': if (s->nb_streams == 0) { av_metadata_set2(&s->metadata, "comment", p, 0); break; } break; case 'm': /* new stream */ s1->skip_media = 0; get_word(st_type, sizeof(st_type), &p); if (!strcmp(st_type, "audio")) { codec_type = AVMEDIA_TYPE_AUDIO; } else if (!strcmp(st_type, "video")) { codec_type = AVMEDIA_TYPE_VIDEO; } else if (!strcmp(st_type, "application")) { codec_type = AVMEDIA_TYPE_DATA; } else { s1->skip_media = 1; 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(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "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 */ ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type); } } /* put a default control url */ av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url)); break; case 'a': if (av_strstart(p, "control:", &p)) { if (s->nb_streams == 0) { if (!strncmp(p, "rtsp://", 7)) av_strlcpy(rt->control_uri, p, sizeof(rt->control_uri)); } else { 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 */ av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0, NULL, NULL, 0, p); if (proto[0] == '\0') { /* relative control URL */ if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/') av_strlcat(rtsp_st->control_url, "/", sizeof(rtsp_st->control_url)); av_strlcat(rtsp_st->control_url, p, sizeof(rtsp_st->control_url)); } else av_strlcpy(rtsp_st->control_url, p, sizeof(rtsp_st->control_url)); } } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) { /* NOTE: rtpmap is only supported AFTER the 'm=' tag */ get_word(buf1, sizeof(buf1), &p); payload_type = atoi(buf1); st = s->streams[s->nb_streams - 1]; rtsp_st = st->priv_data; sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p); } else if (av_strstart(p, "fmtp:", &p) || av_strstart(p, "framesize:", &p)) { /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */ // 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 && rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) rtsp_st->dynamic_handler->parse_sdp_a_line(s, i, rtsp_st->dynamic_protocol_context, buf); } } else if (av_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; /* AV_NOPTS_VALUE means live broadcast (and can't seek) */ s->duration = (end == AV_NOPTS_VALUE) ? AV_NOPTS_VALUE : end - start; } else if (av_strstart(p, "IsRealDataType:integer;",&p)) { if (atoi(p) == 1) rt->transport = RTSP_TRANSPORT_RDT; } else { if (rt->server_type == RTSP_SERVER_WMS) ff_wms_parse_sdp_a_line(s, p); if (s->nb_streams > 0) { if (rt->server_type == RTSP_SERVER_REAL) ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p); rtsp_st = s->streams[s->nb_streams - 1]->priv_data; if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) rtsp_st->dynamic_handler->parse_sdp_a_line(s, s->nb_streams - 1, rtsp_st->dynamic_protocol_context, buf); } } break; } } static int sdp_parse(AVFormatContext *s, const char *content) { const char *p; int letter; /* Some SDP lines, particularly for Realmedia or ASF RTSP streams, * contain long SDP lines containing complete ASF Headers (several * kB) or arrays of MDPR (RM stream descriptor) headers plus * "rulebooks" describing their properties. Therefore, the SDP line * buffer is large. * * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line * in rtpdec_xiph.c. */ char buf[16384], *q; SDPParseState sdp_parse_state, *s1 = &sdp_parse_state; memset(s1, 0, sizeof(SDPParseState)); p = content; for (;;) { p += strspn(p, SPACE_CHARS); 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; } /* close and free RTSP streams */ void ff_rtsp_close_streams(AVFormatContext *s) { RTSPState *rt = s->priv_data; int i; RTSPStream *rtsp_st; for (i = 0; i < rt->nb_rtsp_streams; i++) { rtsp_st = rt->rtsp_streams[i]; if (rtsp_st) { if (rtsp_st->transport_priv) { if (s->oformat) { AVFormatContext *rtpctx = rtsp_st->transport_priv; av_write_trailer(rtpctx); if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) { uint8_t *ptr; url_close_dyn_buf(rtpctx->pb, &ptr); av_free(ptr); } else { url_fclose(rtpctx->pb); } av_metadata_free(&rtpctx->streams[0]->metadata); av_metadata_free(&rtpctx->metadata); av_free(rtpctx->streams[0]); av_free(rtpctx); } else if (rt->transport == RTSP_TRANSPORT_RDT) ff_rdt_parse_close(rtsp_st->transport_priv); else rtp_parse_close(rtsp_st->transport_priv); } 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(rt->rtsp_streams); if (rt->asf_ctx) { av_close_input_stream (rt->asf_ctx); rt->asf_ctx = NULL; } } static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st, URLContext *handle) { RTSPState *rt = s->priv_data; AVFormatContext *rtpctx; int ret; AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL); if (!rtp_format) return NULL; /* Allocate an AVFormatContext for each output stream */ rtpctx = avformat_alloc_context(); if (!rtpctx) return NULL; rtpctx->oformat = rtp_format; if (!av_new_stream(rtpctx, 0)) { av_free(rtpctx); return NULL; } /* Copy the max delay setting; the rtp muxer reads this. */ rtpctx->max_delay = s->max_delay; /* Copy other stream parameters. */ rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio; /* Set the synchronized start time. */ rtpctx->start_time_realtime = rt->start_time; /* Remove the local codec, link to the original codec * context instead, to give the rtp muxer access to * codec parameters. */ av_free(rtpctx->streams[0]->codec); rtpctx->streams[0]->codec = st->codec; if (handle) { url_fdopen(&rtpctx->pb, handle); } else url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE); ret = av_write_header(rtpctx); if (ret) { if (handle) { url_fclose(rtpctx->pb); } else { uint8_t *ptr; url_close_dyn_buf(rtpctx->pb, &ptr); av_free(ptr); } av_free(rtpctx->streams[0]); av_free(rtpctx); return NULL; } /* Copy the RTP AVStream timebase back to the original AVStream */ st->time_base = rtpctx->streams[0]->time_base; return rtpctx; } static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st) { RTSPState *rt = s->priv_data; AVStream *st = NULL; /* open the RTP context */ if (rtsp_st->stream_index >= 0) st = s->streams[rtsp_st->stream_index]; if (!st) s->ctx_flags |= AVFMTCTX_NOHEADER; if (s->oformat) { rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle); /* Ownership of rtp_handle is passed to the rtp mux context */ rtsp_st->rtp_handle = NULL; } else if (rt->transport == RTSP_TRANSPORT_RDT) rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index, rtsp_st->dynamic_protocol_context, rtsp_st->dynamic_handler); else rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type); if (!rtsp_st->transport_priv) { return AVERROR(ENOMEM); } else if (rt->transport != RTSP_TRANSPORT_RDT) { if (rtsp_st->dynamic_handler) { rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv, rtsp_st->dynamic_protocol_context, rtsp_st->dynamic_handler); } } return 0; } #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER static int rtsp_probe(AVProbeData *p) { if (av_strstart(p->filename, "rtsp:", NULL)) return AVPROBE_SCORE_MAX; return 0; } static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp) { const char *p; int v; p = *pp; p += strspn(p, SPACE_CHARS); 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(RTSPMessageHeader *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 (;;) { p += strspn(p, SPACE_CHARS); if (*p == '\0') break; th = &reply->transports[reply->nb_transports]; get_word_sep(transport_protocol, sizeof(transport_protocol), "/", &p); if (!strcasecmp (transport_protocol, "rtp")) { get_word_sep(profile, sizeof(profile), "/;,", &p); lower_transport[0] = '\0'; /* rtp/avp/<protocol> */ if (*p == '/') { get_word_sep(lower_transport, sizeof(lower_transport), ";,", &p); } th->transport = RTSP_TRANSPORT_RTP; } else if (!strcasecmp (transport_protocol, "x-pn-tng") || !strcasecmp (transport_protocol, "x-real-rdt")) { /* x-pn-tng/<protocol> */ get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p); profile[0] = '\0'; th->transport = RTSP_TRANSPORT_RDT; } if (!strcasecmp(lower_transport, "TCP")) th->lower_transport = RTSP_LOWER_TRANSPORT_TCP; else th->lower_transport = RTSP_LOWER_TRANSPORT_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->lower_transport == RTSP_LOWER_TRANSPORT_UDP) th->lower_transport = RTSP_LOWER_TRANSPORT_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 (ff_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 ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf, HTTPAuthState *auth_state) { const char *p; /* NOTE: we do case independent match for broken servers */ p = buf; if (av_stristart(p, "Session:", &p)) { int t; get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p); if (av_stristart(p, ";timeout=", &p) && (t = strtol(p, NULL, 10)) > 0) { reply->timeout = t; } } else if (av_stristart(p, "Content-Length:", &p)) { reply->content_length = strtol(p, NULL, 10); } else if (av_stristart(p, "Transport:", &p)) { rtsp_parse_transport(reply, p); } else if (av_stristart(p, "CSeq:", &p)) { reply->seq = strtol(p, NULL, 10); } else if (av_stristart(p, "Range:", &p)) { rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end); } else if (av_stristart(p, "RealChallenge1:", &p)) { p += strspn(p, SPACE_CHARS); av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge)); } else if (av_stristart(p, "Server:", &p)) { p += strspn(p, SPACE_CHARS); av_strlcpy(reply->server, p, sizeof(reply->server)); } else if (av_stristart(p, "Notice:", &p) || av_stristart(p, "X-Notice:", &p)) { reply->notice = strtol(p, NULL, 10); } else if (av_stristart(p, "Location:", &p)) { p += strspn(p, SPACE_CHARS); av_strlcpy(reply->location, p , sizeof(reply->location)); } else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) { p += strspn(p, SPACE_CHARS); ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p); } else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) { p += strspn(p, SPACE_CHARS); ff_http_auth_handle_header(auth_state, "Authentication-Info", p); } } /* skip a RTP/TCP interleaved packet */ void ff_rtsp_skip_packet(AVFormatContext *s) { RTSPState *rt = s->priv_data; int ret, len, len1; uint8_t buf[1024]; ret = url_read_complete(rt->rtsp_hd, buf, 3); if (ret != 3) return; len = AV_RB16(buf + 1); dprintf(s, "skipping RTP packet len=%d\n", len); /* skip payload */ while (len > 0) { len1 = len; if (len1 > sizeof(buf)) len1 = sizeof(buf); ret = url_read_complete(rt->rtsp_hd, buf, len1); if (ret != len1) return; len -= len1; } } int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply, unsigned char **content_ptr, int return_on_interleaved_data) { RTSPState *rt = s->priv_data; char buf[4096], buf1[1024], *q; unsigned char ch; const char *p; int ret, content_length, line_count = 0; unsigned char *content = NULL; memset(reply, 0, sizeof(*reply)); /* parse reply (XXX: use buffers) */ rt->last_reply[0] = '\0'; for (;;) { q = buf; for (;;) { ret = url_read_complete(rt->rtsp_hd, &ch, 1); #ifdef DEBUG_RTP_TCP dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch); #endif if (ret != 1) return -1; if (ch == '\n') break; if (ch == '$') { /* XXX: only parse it if first char on line ? */ if (return_on_interleaved_data) { return 1; } else ff_rtsp_skip_packet(s); } else if (ch != '\r') { if ((q - buf) < sizeof(buf) - 1) *q++ = ch; } } *q = '\0'; dprintf(s, "line='%s'\n", buf); /* 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); av_strlcpy(reply->reason, p, sizeof(reply->reason)); } else { ff_rtsp_parse_line(reply, p, &rt->auth_state); av_strlcat(rt->last_reply, p, sizeof(rt->last_reply)); av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply)); } line_count++; } if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0') av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->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_read_complete(rt->rtsp_hd, content, content_length); content[content_length] = '\0'; } if (content_ptr) *content_ptr = content; else av_free(content); if (rt->seq != reply->seq) { av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n", rt->seq, reply->seq); } /* EOS */ if (reply->notice == 2101 /* End-of-Stream Reached */ || reply->notice == 2104 /* Start-of-Stream Reached */ || reply->notice == 2306 /* Continuous Feed Terminated */) { rt->state = RTSP_STATE_IDLE; } else if (reply->notice >= 4400 && reply->notice < 5500) { return AVERROR(EIO); /* data or server error */ } else if (reply->notice == 2401 /* Ticket Expired */ || (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ ) return AVERROR(EPERM); return 0; } int ff_rtsp_send_cmd_with_content_async(AVFormatContext *s, const char *method, const char *url, const char *headers, const unsigned char *send_content, int send_content_length) { RTSPState *rt = s->priv_data; char buf[4096], *out_buf; char base64buf[AV_BASE64_SIZE(sizeof(buf))]; /* Add in RTSP headers */ out_buf = buf; rt->seq++; snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url); if (headers) av_strlcat(buf, headers, sizeof(buf)); av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq); if (rt->session_id[0] != '\0' && (!headers || !strstr(headers, "\nIf-Match:"))) { av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id); } if (rt->auth[0]) { char *str = ff_http_auth_create_response(&rt->auth_state, rt->auth, url, method); if (str) av_strlcat(buf, str, sizeof(buf)); av_free(str); } if (send_content_length > 0 && send_content) av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length); av_strlcat(buf, "\r\n", sizeof(buf)); /* base64 encode rtsp if tunneling */ if (rt->control_transport == RTSP_MODE_TUNNEL) { av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf)); out_buf = base64buf; } dprintf(s, "Sending:\n%s--\n", buf); url_write(rt->rtsp_hd_out, out_buf, strlen(out_buf)); if (send_content_length > 0 && send_content) { if (rt->control_transport == RTSP_MODE_TUNNEL) { av_log(s, AV_LOG_ERROR, "tunneling of RTSP requests " "with content data not supported\n"); return AVERROR_PATCHWELCOME; } url_write(rt->rtsp_hd_out, send_content, send_content_length); } rt->last_cmd_time = av_gettime(); return 0; } int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method, const char *url, const char *headers) { return ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0); } int ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url, const char *headers, RTSPMessageHeader *reply, unsigned char **content_ptr) { return ff_rtsp_send_cmd_with_content(s, method, url, headers, reply, content_ptr, NULL, 0); } int ff_rtsp_send_cmd_with_content(AVFormatContext *s, const char *method, const char *url, const char *header, RTSPMessageHeader *reply, unsigned char **content_ptr, const unsigned char *send_content, int send_content_length) { RTSPState *rt = s->priv_data; HTTPAuthType cur_auth_type; int ret; retry: cur_auth_type = rt->auth_state.auth_type; if ((ret = ff_rtsp_send_cmd_with_content_async(s, method, url, header, send_content, send_content_length))) return ret; if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0) ) < 0) return ret; if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE && rt->auth_state.auth_type != HTTP_AUTH_NONE) goto retry; if (reply->status_code > 400){ av_log(s, AV_LOG_ERROR, "method %s failed: %d%s\n", method, reply->status_code, reply->reason); av_log(s, AV_LOG_DEBUG, "%s\n", rt->last_reply); } return 0; } /** * @return 0 on success, <0 on error, 1 if protocol is unavailable. */ static int make_setup_request(AVFormatContext *s, const char *host, int port, int lower_transport, const char *real_challenge) { RTSPState *rt = s->priv_data; int rtx, j, i, err, interleave = 0; RTSPStream *rtsp_st; RTSPMessageHeader reply1, *reply = &reply1; char cmd[2048]; const char *trans_pref; if (rt->transport == RTSP_TRANSPORT_RDT) trans_pref = "x-pn-tng"; else trans_pref = "RTP/AVP"; /* default timeout: 1 minute */ rt->timeout = 60; /* 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]; /** * WMS serves all UDP data over a single connection, the RTX, which * isn't necessarily the first in the SDP but has to be the first * to be set up, else the second/third SETUP will fail with a 461. */ if (lower_transport == RTSP_LOWER_TRANSPORT_UDP && rt->server_type == RTSP_SERVER_WMS) { if (i == 0) { /* rtx first */ for (rtx = 0; rtx < rt->nb_rtsp_streams; rtx++) { int len = strlen(rt->rtsp_streams[rtx]->control_url); if (len >= 4 && !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4, "/rtx")) break; } if (rtx == rt->nb_rtsp_streams) return -1; /* no RTX found */ rtsp_st = rt->rtsp_streams[rtx]; } else rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1]; } else rtsp_st = rt->rtsp_streams[i]; /* RTP/UDP */ if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) { char buf[256]; if (rt->server_type == RTSP_SERVER_WMS && i > 1) { port = reply->transports[0].client_port_min; goto have_port; } /* first try in specified port range */ if (RTSP_RTP_PORT_MIN != 0) { while (j <= RTSP_RTP_PORT_MAX) { ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1, "?localport=%d", j); /* we will use two ports per rtp stream (rtp and rtcp) */ j += 2; if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0) goto rtp_opened; } } #if 0 /* then try on any port */ if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) { err = AVERROR_INVALIDDATA; goto fail; } #endif rtp_opened: port = rtp_get_local_port(rtsp_st->rtp_handle); have_port: snprintf(transport, sizeof(transport) - 1, "%s/UDP;", trans_pref); if (rt->server_type != RTSP_SERVER_REAL) av_strlcat(transport, "unicast;", sizeof(transport)); av_strlcatf(transport, sizeof(transport), "client_port=%d", port); if (rt->transport == RTSP_TRANSPORT_RTP && !(rt->server_type == RTSP_SERVER_WMS && i > 0)) av_strlcatf(transport, sizeof(transport), "-%d", port + 1); } /* RTP/TCP */ else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) { /** For WMS streams, the application streams are only used for * UDP. When trying to set it up for TCP streams, the server * will return an error. Therefore, we skip those streams. */ if (rt->server_type == RTSP_SERVER_WMS && s->streams[rtsp_st->stream_index]->codec->codec_type == AVMEDIA_TYPE_DATA) continue; snprintf(transport, sizeof(transport) - 1, "%s/TCP;", trans_pref); if (rt->server_type == RTSP_SERVER_WMS) av_strlcat(transport, "unicast;", sizeof(transport)); av_strlcatf(transport, sizeof(transport), "interleaved=%d-%d", interleave, interleave + 1); interleave += 2; } else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) { snprintf(transport, sizeof(transport) - 1, "%s/UDP;multicast", trans_pref); } if (s->oformat) { av_strlcat(transport, ";mode=receive", sizeof(transport)); } else if (rt->server_type == RTSP_SERVER_REAL || rt->server_type == RTSP_SERVER_WMS) av_strlcat(transport, ";mode=play", sizeof(transport)); snprintf(cmd, sizeof(cmd), "Transport: %s\r\n", transport); if (i == 0 && rt->server_type == RTSP_SERVER_REAL) { char real_res[41], real_csum[9]; ff_rdt_calc_response_and_checksum(real_res, real_csum, real_challenge); av_strlcatf(cmd, sizeof(cmd), "If-Match: %s\r\n" "RealChallenge2: %s, sd=%s\r\n", rt->session_id, real_res, real_csum); } ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL); if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) { err = 1; goto fail; } else 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].lower_transport != rt->lower_transport || reply->transports[0].transport != rt->transport) { err = AVERROR_INVALIDDATA; goto fail; } } else { rt->lower_transport = reply->transports[0].lower_transport; rt->transport = reply->transports[0].transport; } /* close RTP connection if not chosen */ if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP && (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) { url_close(rtsp_st->rtp_handle); rtsp_st->rtp_handle = NULL; } switch(reply->transports[0].lower_transport) { case RTSP_LOWER_TRANSPORT_TCP: rtsp_st->interleaved_min = reply->transports[0].interleaved_min; rtsp_st->interleaved_max = reply->transports[0].interleaved_max; break; case RTSP_LOWER_TRANSPORT_UDP: { char url[1024]; /* XXX: also use address if specified */ ff_url_join(url, sizeof(url), "rtp", NULL, host, reply->transports[0].server_port_min, NULL); if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) { err = AVERROR_INVALIDDATA; goto fail; } /* Try to initialize the connection state in a * potential NAT router by sending dummy packets. * RTP/RTCP dummy packets are used for RDT, too. */ if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat) rtp_send_punch_packets(rtsp_st->rtp_handle); break; } case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: { char url[1024]; struct in_addr in; int port, ttl; if (reply->transports[0].destination) { in.s_addr = htonl(reply->transports[0].destination); port = reply->transports[0].port_min; ttl = reply->transports[0].ttl; } else { in = rtsp_st->sdp_ip; port = rtsp_st->sdp_port; ttl = rtsp_st->sdp_ttl; } ff_url_join(url, sizeof(url), "rtp", NULL, inet_ntoa(in), port, "?ttl=%d", ttl); if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) { err = AVERROR_INVALIDDATA; goto fail; } break; } } if ((err = rtsp_open_transport_ctx(s, rtsp_st))) goto fail; } if (reply->timeout > 0) rt->timeout = reply->timeout; if (rt->server_type == RTSP_SERVER_REAL) rt->need_subscription = 1; return 0; fail: for (i = 0; i < rt->nb_rtsp_streams; i++) { if (rt->rtsp_streams[i]->rtp_handle) { url_close(rt->rtsp_streams[i]->rtp_handle); rt->rtsp_streams[i]->rtp_handle = NULL; } } return err; } static int rtsp_read_play(AVFormatContext *s) { RTSPState *rt = s->priv_data; RTSPMessageHeader reply1, *reply = &reply1; int i; char cmd[1024]; av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state); if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) { if (rt->state == RTSP_STATE_PAUSED) { cmd[0] = 0; } else { snprintf(cmd, sizeof(cmd), "Range: npt=%0.3f-\r\n", (double)rt->seek_timestamp / AV_TIME_BASE); } ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL); if (reply->status_code != RTSP_STATUS_OK) { return -1; } if (reply->range_start != AV_NOPTS_VALUE && rt->transport == RTSP_TRANSPORT_RTP) { for (i = 0; i < rt->nb_rtsp_streams; i++) { RTSPStream *rtsp_st = rt->rtsp_streams[i]; RTPDemuxContext *rtpctx = rtsp_st->transport_priv; AVStream *st = NULL; if (!rtpctx) continue; if (rtsp_st->stream_index >= 0) st = s->streams[rtsp_st->stream_index]; rtpctx->last_rtcp_ntp_time = AV_NOPTS_VALUE; rtpctx->first_rtcp_ntp_time = AV_NOPTS_VALUE; if (st) rtpctx->range_start_offset = av_rescale_q(reply->range_start, AV_TIME_BASE_Q, st->time_base); } } } rt->state = RTSP_STATE_STREAMING; return 0; } static int rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply) { RTSPState *rt = s->priv_data; char cmd[1024]; unsigned char *content = NULL; int ret; /* describe the stream */ snprintf(cmd, sizeof(cmd), "Accept: application/sdp\r\n"); if (rt->server_type == RTSP_SERVER_REAL) { /** * The Require: attribute is needed for proper streaming from * Realmedia servers. */ av_strlcat(cmd, "Require: com.real.retain-entity-for-setup\r\n", sizeof(cmd)); } ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content); if (!content) return AVERROR_INVALIDDATA; if (reply->status_code != RTSP_STATUS_OK) { av_freep(&content); return AVERROR_INVALIDDATA; } /* now we got the SDP description, we parse it */ ret = sdp_parse(s, (const char *)content); av_freep(&content); if (ret < 0) return AVERROR_INVALIDDATA; return 0; } static int rtsp_setup_output_streams(AVFormatContext *s, const char *addr) { RTSPState *rt = s->priv_data; RTSPMessageHeader reply1, *reply = &reply1; int i; char *sdp; AVFormatContext sdp_ctx, *ctx_array[1]; rt->start_time = av_gettime(); /* Announce the stream */ sdp = av_mallocz(SDP_MAX_SIZE); if (sdp == NULL) return AVERROR(ENOMEM); /* We create the SDP based on the RTSP AVFormatContext where we * aren't allowed to change the filename field. (We create the SDP * based on the RTSP context since the contexts for the RTP streams * don't exist yet.) In order to specify a custom URL with the actual * peer IP instead of the originally specified hostname, we create * a temporary copy of the AVFormatContext, where the custom URL is set. * * FIXME: Create the SDP without copying the AVFormatContext. * This either requires setting up the RTP stream AVFormatContexts * already here (complicating things immensely) or getting a more * flexible SDP creation interface. */ sdp_ctx = *s; ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename), "rtsp", NULL, addr, -1, NULL); ctx_array[0] = &sdp_ctx; if (avf_sdp_create(ctx_array, 1, sdp, SDP_MAX_SIZE)) { av_free(sdp); return AVERROR_INVALIDDATA; } av_log(s, AV_LOG_INFO, "SDP:\n%s\n", sdp); ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri, "Content-Type: application/sdp\r\n", reply, NULL, sdp, strlen(sdp)); av_free(sdp); if (reply->status_code != RTSP_STATUS_OK) return AVERROR_INVALIDDATA; /* Set up the RTSPStreams for each AVStream */ for (i = 0; i < s->nb_streams; i++) { RTSPStream *rtsp_st; AVStream *st = s->streams[i]; rtsp_st = av_mallocz(sizeof(RTSPStream)); if (!rtsp_st) return AVERROR(ENOMEM); dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st); st->priv_data = rtsp_st; rtsp_st->stream_index = i; av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url)); /* Note, this must match the relative uri set in the sdp content */ av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url), "/streamid=%d", i); } return 0; } void ff_rtsp_close_connections(AVFormatContext *s) { RTSPState *rt = s->priv_data; if (rt->rtsp_hd_out != rt->rtsp_hd) url_close(rt->rtsp_hd_out); url_close(rt->rtsp_hd); rt->rtsp_hd = rt->rtsp_hd_out = NULL; } int ff_rtsp_connect(AVFormatContext *s) { RTSPState *rt = s->priv_data; char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128]; char *option_list, *option, *filename; int port, err, tcp_fd; RTSPMessageHeader reply1 = {0}, *reply = &reply1; int lower_transport_mask = 0; char real_challenge[64]; struct sockaddr_storage peer; socklen_t peer_len = sizeof(peer); if (!ff_network_init()) return AVERROR(EIO); redirect: rt->control_transport = RTSP_MODE_PLAIN; /* extract hostname and port */ av_url_split(NULL, 0, auth, sizeof(auth), host, sizeof(host), &port, path, sizeof(path), s->filename); if (*auth) { av_strlcpy(rt->auth, auth, sizeof(rt->auth)); } if (port < 0) port = RTSP_DEFAULT_PORT; /* search for options */ option_list = strrchr(path, '?'); if (option_list) { /* Strip out the RTSP specific options, write out the rest of * the options back into the same string. */ filename = option_list; while (option_list) { /* move the option pointer */ option = ++option_list; option_list = strchr(option_list, '&'); if (option_list) *option_list = 0; /* handle the options */ if (!strcmp(option, "udp")) { lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP); } else if (!strcmp(option, "multicast")) { lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST); } else if (!strcmp(option, "tcp")) { lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP); } else if(!strcmp(option, "http")) { lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP); rt->control_transport = RTSP_MODE_TUNNEL; } else { /* Write options back into the buffer, using memmove instead * of strcpy since the strings may overlap. */ int len = strlen(option); memmove(++filename, option, len); filename += len; if (option_list) *filename = '&'; } } *filename = 0; } if (!lower_transport_mask) lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1; if (s->oformat) { /* Only UDP or TCP - UDP multicast isn't supported. */ lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) | (1 << RTSP_LOWER_TRANSPORT_TCP); if (!lower_transport_mask || rt->control_transport == RTSP_MODE_TUNNEL) { av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, " "only UDP and TCP are supported for output.\n"); err = AVERROR(EINVAL); goto fail; } } /* Construct the URI used in request; this is similar to s->filename, * but with authentication credentials removed and RTSP specific options * stripped out. */ ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL, host, port, "%s", path); if (rt->control_transport == RTSP_MODE_TUNNEL) { /* set up initial handshake for tunneling */ char httpname[1024]; char sessioncookie[17]; char headers[1024]; ff_url_join(httpname, sizeof(httpname), "http", auth, host, port, "%s", path); snprintf(sessioncookie, sizeof(sessioncookie), "%08x%08x", av_get_random_seed(), av_get_random_seed()); /* GET requests */ if (url_alloc(&rt->rtsp_hd, httpname, URL_RDONLY) < 0) { err = AVERROR(EIO); goto fail; } /* generate GET headers */ snprintf(headers, sizeof(headers), "x-sessioncookie: %s\r\n" "Accept: application/x-rtsp-tunnelled\r\n" "Pragma: no-cache\r\n" "Cache-Control: no-cache\r\n", sessioncookie); ff_http_set_headers(rt->rtsp_hd, headers); /* complete the connection */ if (url_connect(rt->rtsp_hd)) { err = AVERROR(EIO); goto fail; } /* POST requests */ if (url_alloc(&rt->rtsp_hd_out, httpname, URL_WRONLY) < 0 ) { err = AVERROR(EIO); goto fail; } /* generate POST headers */ snprintf(headers, sizeof(headers), "x-sessioncookie: %s\r\n" "Content-Type: application/x-rtsp-tunnelled\r\n" "Pragma: no-cache\r\n" "Cache-Control: no-cache\r\n" "Content-Length: 32767\r\n" "Expires: Sun, 9 Jan 1972 00:00:00 GMT\r\n", sessioncookie); ff_http_set_headers(rt->rtsp_hd_out, headers); ff_http_set_chunked_transfer_encoding(rt->rtsp_hd_out, 0); /* Initialize the authentication state for the POST session. The HTTP * protocol implementation doesn't properly handle multi-pass * authentication for POST requests, since it would require one of * the following: * - implementing Expect: 100-continue, which many HTTP servers * don't support anyway, even less the RTSP servers that do HTTP * tunneling * - sending the whole POST data until getting a 401 reply specifying * what authentication method to use, then resending all that data * - waiting for potential 401 replies directly after sending the * POST header (waiting for some unspecified time) * Therefore, we copy the full auth state, which works for both basic * and digest. (For digest, we would have to synchronize the nonce * count variable between the two sessions, if we'd do more requests * with the original session, though.) */ ff_http_init_auth_state(rt->rtsp_hd_out, rt->rtsp_hd); /* complete the connection */ if (url_connect(rt->rtsp_hd_out)) { err = AVERROR(EIO); goto fail; } } else { /* open the tcp connection */ ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL); if (url_open(&rt->rtsp_hd, tcpname, URL_RDWR) < 0) { err = AVERROR(EIO); goto fail; } rt->rtsp_hd_out = rt->rtsp_hd; } rt->seq = 0; tcp_fd = url_get_file_handle(rt->rtsp_hd); if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) { getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host), NULL, 0, NI_NUMERICHOST); } /* request options supported by the server; this also detects server * type */ for (rt->server_type = RTSP_SERVER_RTP;;) { cmd[0] = 0; if (rt->server_type == RTSP_SERVER_REAL) av_strlcat(cmd, /** * The following entries are required for proper * streaming from a Realmedia server. They are * interdependent in some way although we currently * don't quite understand how. Values were copied * from mplayer SVN r23589. * @param CompanyID is a 16-byte ID in base64 * @param ClientChallenge is a 16-byte ID in hex */ "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n" "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n" "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n" "GUID: 00000000-0000-0000-0000-000000000000\r\n", sizeof(cmd)); ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL); if (reply->status_code != RTSP_STATUS_OK) { err = AVERROR_INVALIDDATA; goto fail; } /* detect server type if not standard-compliant RTP */ if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) { rt->server_type = RTSP_SERVER_REAL; continue; } else if (!strncasecmp(reply->server, "WMServer/", 9)) { rt->server_type = RTSP_SERVER_WMS; } else if (rt->server_type == RTSP_SERVER_REAL) strcpy(real_challenge, reply->real_challenge); break; } if (s->iformat) err = rtsp_setup_input_streams(s, reply); else err = rtsp_setup_output_streams(s, host); if (err) goto fail; do { int lower_transport = ff_log2_tab[lower_transport_mask & ~(lower_transport_mask - 1)]; err = make_setup_request(s, host, port, lower_transport, rt->server_type == RTSP_SERVER_REAL ? real_challenge : NULL); if (err < 0) goto fail; lower_transport_mask &= ~(1 << lower_transport); if (lower_transport_mask == 0 && err == 1) { err = FF_NETERROR(EPROTONOSUPPORT); goto fail; } } while (err); rt->state = RTSP_STATE_IDLE; rt->seek_timestamp = 0; /* default is to start stream at position zero */ return 0; fail: ff_rtsp_close_streams(s); ff_rtsp_close_connections(s); if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) { av_strlcpy(s->filename, reply->location, sizeof(s->filename)); av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n", reply->status_code, s->filename); goto redirect; } ff_network_close(); return err; } #endif #if CONFIG_RTSP_DEMUXER static int rtsp_read_header(AVFormatContext *s, AVFormatParameters *ap) { RTSPState *rt = s->priv_data; int ret; ret = ff_rtsp_connect(s); if (ret) return ret; rt->real_setup_cache = av_mallocz(2 * s->nb_streams * sizeof(*rt->real_setup_cache)); if (!rt->real_setup_cache) return AVERROR(ENOMEM); rt->real_setup = rt->real_setup_cache + s->nb_streams * sizeof(*rt->real_setup); if (ap->initial_pause) { /* do not start immediately */ } else { if (rtsp_read_play(s) < 0) { ff_rtsp_close_streams(s); ff_rtsp_close_connections(s); return AVERROR_INVALIDDATA; } } return 0; } 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 fd, fd_max, n, i, ret, tcp_fd, timeout_cnt = 0; struct timeval tv; for (;;) { if (url_interrupt_cb()) return AVERROR(EINTR); FD_ZERO(&rfds); if (rt->rtsp_hd) { tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd); FD_SET(tcp_fd, &rfds); } else { fd_max = 0; tcp_fd = -1; } for (i = 0; i < rt->nb_rtsp_streams; i++) { rtsp_st = rt->rtsp_streams[i]; if (rtsp_st->rtp_handle) { /* currently, we cannot probe RTCP handle because of * blocking restrictions */ fd = url_get_file_handle(rtsp_st->rtp_handle); if (fd > fd_max) fd_max = fd; FD_SET(fd, &rfds); } } tv.tv_sec = 0; tv.tv_usec = SELECT_TIMEOUT_MS * 1000; n = select(fd_max + 1, &rfds, NULL, NULL, &tv); if (n > 0) { timeout_cnt = 0; for (i = 0; i < rt->nb_rtsp_streams; i++) { rtsp_st = rt->rtsp_streams[i]; if (rtsp_st->rtp_handle) { fd = url_get_file_handle(rtsp_st->rtp_handle); if (FD_ISSET(fd, &rfds)) { ret = url_read(rtsp_st->rtp_handle, buf, buf_size); if (ret > 0) { *prtsp_st = rtsp_st; return ret; } } } } #if CONFIG_RTSP_DEMUXER if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) { RTSPMessageHeader reply; ret = ff_rtsp_read_reply(s, &reply, NULL, 0); if (ret < 0) return ret; /* XXX: parse message */ if (rt->state != RTSP_STATE_STREAMING) return 0; } #endif } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) { return FF_NETERROR(ETIMEDOUT); } else if (n < 0 && errno != EINTR) return AVERROR(errno); } } 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 dprintf(s, "tcp_read_packet:\n"); #endif redo: for (;;) { RTSPMessageHeader reply; ret = ff_rtsp_read_reply(s, &reply, NULL, 1); if (ret == -1) return -1; if (ret == 1) /* received '$' */ break; /* XXX: parse message */ if (rt->state != RTSP_STATE_STREAMING) return 0; } ret = url_read_complete(rt->rtsp_hd, buf, 3); if (ret != 3) return -1; id = buf[0]; len = AV_RB16(buf + 1); #ifdef DEBUG_RTP_TCP dprintf(s, "id=%d len=%d\n", id, len); #endif if (len > buf_size || len < 12) goto redo; /* get the data */ ret = url_read_complete(rt->rtsp_hd, buf, len); if (ret != len) return -1; if (rt->transport == RTSP_TRANSPORT_RDT && ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL) < 0) 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 rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt) { RTSPState *rt = s->priv_data; int ret, len; uint8_t buf[10 * RTP_MAX_PACKET_LENGTH]; RTSPStream *rtsp_st; /* get next frames from the same RTP packet */ if (rt->cur_transport_priv) { if (rt->transport == RTSP_TRANSPORT_RDT) { ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0); } else ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0); if (ret == 0) { rt->cur_transport_priv = NULL; return 0; } else if (ret == 1) { return 0; } else rt->cur_transport_priv = NULL; } /* read next RTP packet */ redo: switch(rt->lower_transport) { default: #if CONFIG_RTSP_DEMUXER case RTSP_LOWER_TRANSPORT_TCP: len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf)); break; #endif case RTSP_LOWER_TRANSPORT_UDP: case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf)); if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP) rtp_check_and_send_back_rr(rtsp_st->transport_priv, len); break; } if (len < 0) return len; if (len == 0) return AVERROR_EOF; if (rt->transport == RTSP_TRANSPORT_RDT) { ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, buf, len); } else { ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, buf, len); if (ret < 0) { /* Either bad packet, or a RTCP packet. Check if the * first_rtcp_ntp_time field was initialized. */ RTPDemuxContext *rtpctx = rtsp_st->transport_priv; if (rtpctx->first_rtcp_ntp_time != AV_NOPTS_VALUE) { /* first_rtcp_ntp_time has been initialized for this stream, * copy the same value to all other uninitialized streams, * in order to map their timestamp origin to the same ntp time * as this one. */ int i; for (i = 0; i < rt->nb_rtsp_streams; i++) { RTPDemuxContext *rtpctx2 = rtsp_st->transport_priv; if (rtpctx2 && rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time; } } } } if (ret < 0) goto redo; if (ret == 1) /* more packets may follow, so we save the RTP context */ rt->cur_transport_priv = rtsp_st->transport_priv; return ret; } static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt) { RTSPState *rt = s->priv_data; int ret; RTSPMessageHeader reply1, *reply = &reply1; char cmd[1024]; if (rt->server_type == RTSP_SERVER_REAL) { int i; for (i = 0; i < s->nb_streams; i++) rt->real_setup[i] = s->streams[i]->discard; if (!rt->need_subscription) { if (memcmp (rt->real_setup, rt->real_setup_cache, sizeof(enum AVDiscard) * s->nb_streams)) { snprintf(cmd, sizeof(cmd), "Unsubscribe: %s\r\n", rt->last_subscription); ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri, cmd, reply, NULL); if (reply->status_code != RTSP_STATUS_OK) return AVERROR_INVALIDDATA; rt->need_subscription = 1; } } if (rt->need_subscription) { int r, rule_nr, first = 1; memcpy(rt->real_setup_cache, rt->real_setup, sizeof(enum AVDiscard) * s->nb_streams); rt->last_subscription[0] = 0; snprintf(cmd, sizeof(cmd), "Subscribe: "); for (i = 0; i < rt->nb_rtsp_streams; i++) { rule_nr = 0; for (r = 0; r < s->nb_streams; r++) { if (s->streams[r]->priv_data == rt->rtsp_streams[i]) { if (s->streams[r]->discard != AVDISCARD_ALL) { if (!first) av_strlcat(rt->last_subscription, ",", sizeof(rt->last_subscription)); ff_rdt_subscribe_rule( rt->last_subscription, sizeof(rt->last_subscription), i, rule_nr); first = 0; } rule_nr++; } } } av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription); ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri, cmd, reply, NULL); if (reply->status_code != RTSP_STATUS_OK) return AVERROR_INVALIDDATA; rt->need_subscription = 0; if (rt->state == RTSP_STATE_STREAMING) rtsp_read_play (s); } } ret = rtsp_fetch_packet(s, pkt); if (ret < 0) return ret; /* send dummy request to keep TCP connection alive */ if ((av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) { if (rt->server_type == RTSP_SERVER_WMS) { ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL); } else { ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL); } } return 0; } /* pause the stream */ static int rtsp_read_pause(AVFormatContext *s) { RTSPState *rt = s->priv_data; RTSPMessageHeader reply1, *reply = &reply1; if (rt->state != RTSP_STATE_STREAMING) return 0; else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) { ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL); if (reply->status_code != RTSP_STATUS_OK) { return -1; } } 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 = av_rescale_q(timestamp, s->streams[stream_index]->time_base, AV_TIME_BASE_Q); switch(rt->state) { default: case RTSP_STATE_IDLE: break; case RTSP_STATE_STREAMING: if (rtsp_read_pause(s) != 0) return -1; rt->state = RTSP_STATE_SEEKING; 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; #if 0 /* NOTE: it is valid to flush the buffer here */ if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) { url_fclose(&rt->rtsp_gb); } #endif ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL); ff_rtsp_close_streams(s); ff_rtsp_close_connections(s); ff_network_close(); rt->real_setup = NULL; av_freep(&rt->real_setup_cache); return 0; } AVInputFormat rtsp_demuxer = { "rtsp", NULL_IF_CONFIG_SMALL("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, }; #endif 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 && av_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; } 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]; if (!ff_network_init()) return AVERROR(EIO); /* 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; i < rt->nb_rtsp_streams; i++) { rtsp_st = rt->rtsp_streams[i]; ff_url_join(url, sizeof(url), "rtp", NULL, inet_ntoa(rtsp_st->sdp_ip), rtsp_st->sdp_port, "?localport=%d&ttl=%d", rtsp_st->sdp_port, rtsp_st->sdp_ttl); if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) { err = AVERROR_INVALIDDATA; goto fail; } if ((err = rtsp_open_transport_ctx(s, rtsp_st))) goto fail; } return 0; fail: ff_rtsp_close_streams(s); ff_network_close(); return err; } static int sdp_read_close(AVFormatContext *s) { ff_rtsp_close_streams(s); ff_network_close(); return 0; } AVInputFormat sdp_demuxer = { "sdp", NULL_IF_CONFIG_SMALL("SDP"), sizeof(RTSPState), sdp_probe, sdp_read_header, rtsp_fetch_packet, sdp_read_close, };