3 * Copyright (c) 2002 Fabrice Bellard.
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24 #include <unistd.h> /* for select() prototype */
27 #include "rtp_internal.h"
30 //#define DEBUG_RTP_TCP
32 enum RTSPClientState {
38 typedef struct RTSPState {
39 URLContext *rtsp_hd; /* RTSP TCP connexion handle */
41 struct RTSPStream **rtsp_streams;
43 enum RTSPClientState state;
44 int64_t seek_timestamp;
46 /* XXX: currently we use unbuffered input */
47 // ByteIOContext rtsp_gb;
48 int seq; /* RTSP command sequence number */
50 enum RTSPProtocol protocol;
51 char last_reply[2048]; /* XXX: allocate ? */
52 RTPDemuxContext *cur_rtp;
55 typedef struct RTSPStream {
56 URLContext *rtp_handle; /* RTP stream handle */
57 RTPDemuxContext *rtp_ctx; /* RTP parse context */
59 int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */
60 int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */
61 char control_url[1024]; /* url for this stream (from SDP) */
63 int sdp_port; /* port (from SDP content - not used in RTSP) */
64 struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */
65 int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */
66 int sdp_payload_type; /* payload type - only used in SDP */
67 rtp_payload_data_t rtp_payload_data; /* rtp payload parsing infos from SDP */
69 RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure)
70 void *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol)
73 static int rtsp_read_play(AVFormatContext *s);
75 /* XXX: currently, the only way to change the protocols consists in
76 changing this variable */
78 int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_UDP);
80 FFRTSPCallback *ff_rtsp_callback = NULL;
82 static int rtsp_probe(AVProbeData *p)
84 if (strstart(p->filename, "rtsp:", NULL))
85 return AVPROBE_SCORE_MAX;
89 static int redir_isspace(int c)
91 return (c == ' ' || c == '\t' || c == '\n' || c == '\r');
94 static void skip_spaces(const char **pp)
98 while (redir_isspace(*p))
103 static void get_word_sep(char *buf, int buf_size, const char *sep,
114 while (!strchr(sep, *p) && *p != '\0') {
115 if ((q - buf) < buf_size - 1)
124 static void get_word(char *buf, int buf_size, const char **pp)
132 while (!redir_isspace(*p) && *p != '\0') {
133 if ((q - buf) < buf_size - 1)
142 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other
144 static int sdp_parse_rtpmap(AVCodecContext *codec, RTSPStream *rtsp_st, int payload_type, const char *p)
151 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
152 see if we can handle this kind of payload */
153 get_word_sep(buf, sizeof(buf), "/", &p);
154 if (payload_type >= RTP_PT_PRIVATE) {
155 RTPDynamicProtocolHandler *handler= RTPFirstDynamicPayloadHandler;
157 if (!strcmp(buf, handler->enc_name) && (codec->codec_type == handler->codec_type)) {
158 codec->codec_id = handler->codec_id;
159 rtsp_st->dynamic_handler= handler;
161 rtsp_st->dynamic_protocol_context= handler->open();
165 handler= handler->next;
168 /* We are in a standard case ( from http://www.iana.org/assignments/rtp-parameters) */
169 /* search into AVRtpPayloadTypes[] */
170 for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
171 if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpPayloadTypes[i].codec_type)){
172 codec->codec_id = AVRtpPayloadTypes[i].codec_id;
177 c = avcodec_find_decoder(codec->codec_id);
181 c_name = (char *)NULL;
184 get_word_sep(buf, sizeof(buf), "/", &p);
186 switch (codec->codec_type) {
187 case CODEC_TYPE_AUDIO:
188 av_log(codec, AV_LOG_DEBUG, " audio codec set to : %s\n", c_name);
189 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
190 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
192 codec->sample_rate = i;
193 get_word_sep(buf, sizeof(buf), "/", &p);
197 // TODO: there is a bug here; if it is a mono stream, and less than 22000Hz, faad upconverts to stereo and twice the
198 // frequency. No problem, but the sample rate is being set here by the sdp line. Upcoming patch forthcoming. (rdm)
200 av_log(codec, AV_LOG_DEBUG, " audio samplerate set to : %i\n", codec->sample_rate);
201 av_log(codec, AV_LOG_DEBUG, " audio channels set to : %i\n", codec->channels);
203 case CODEC_TYPE_VIDEO:
204 av_log(codec, AV_LOG_DEBUG, " video codec set to : %s\n", c_name);
215 /* return the length and optionnaly the data */
216 static int hex_to_data(uint8_t *data, const char *p)
226 c = toupper((unsigned char)*p++);
227 if (c >= '0' && c <= '9')
229 else if (c >= 'A' && c <= 'F')
244 static void sdp_parse_fmtp_config(AVCodecContext *codec, char *attr, char *value)
246 switch (codec->codec_id) {
249 if (!strcmp(attr, "config")) {
250 /* decode the hexa encoded parameter */
251 int len = hex_to_data(NULL, value);
252 codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
253 if (!codec->extradata)
255 codec->extradata_size = len;
256 hex_to_data(codec->extradata, value);
265 typedef struct attrname_map
272 /* All known fmtp parmeters and the corresping RTPAttrTypeEnum */
273 #define ATTR_NAME_TYPE_INT 0
274 #define ATTR_NAME_TYPE_STR 1
275 static attrname_map_t attr_names[]=
277 {"SizeLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, sizelength)},
278 {"IndexLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexlength)},
279 {"IndexDeltaLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexdeltalength)},
280 {"profile-level-id", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, profile_level_id)},
281 {"StreamType", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, streamtype)},
282 {"mode", ATTR_NAME_TYPE_STR, offsetof(rtp_payload_data_t, mode)},
286 /** parse the attribute line from the fmtp a line of an sdp resonse. This is broken out as a function
287 * because it is used in rtp_h264.c, which is forthcoming.
289 int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
294 get_word_sep(attr, attr_size, "=", p);
297 get_word_sep(value, value_size, ";", p);
305 /* parse a SDP line and save stream attributes */
306 static void sdp_parse_fmtp(AVStream *st, const char *p)
312 RTSPStream *rtsp_st = st->priv_data;
313 AVCodecContext *codec = st->codec;
314 rtp_payload_data_t *rtp_payload_data = &rtsp_st->rtp_payload_data;
316 /* loop on each attribute */
317 while(rtsp_next_attr_and_value(&p, attr, sizeof(attr), value, sizeof(value)))
319 /* grab the codec extra_data from the config parameter of the fmtp line */
320 sdp_parse_fmtp_config(codec, attr, value);
321 /* Looking for a known attribute */
322 for (i = 0; attr_names[i].str; ++i) {
323 if (!strcasecmp(attr, attr_names[i].str)) {
324 if (attr_names[i].type == ATTR_NAME_TYPE_INT)
325 *(int *)((char *)rtp_payload_data + attr_names[i].offset) = atoi(value);
326 else if (attr_names[i].type == ATTR_NAME_TYPE_STR)
327 *(char **)((char *)rtp_payload_data + attr_names[i].offset) = av_strdup(value);
333 /** Parse a string \p in the form of Range:npt=xx-xx, and determine the start
335 * Used for seeking in the rtp stream.
337 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
342 if (!stristart(p, "npt=", &p))
345 *start = AV_NOPTS_VALUE;
346 *end = AV_NOPTS_VALUE;
348 get_word_sep(buf, sizeof(buf), "-", &p);
349 *start = parse_date(buf, 1);
352 get_word_sep(buf, sizeof(buf), "-", &p);
353 *end = parse_date(buf, 1);
355 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
356 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
359 typedef struct SDPParseState {
361 struct in_addr default_ip;
365 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
366 int letter, const char *buf)
368 RTSPState *rt = s->priv_data;
369 char buf1[64], st_type[64];
371 int codec_type, payload_type, i;
374 struct in_addr sdp_ip;
378 printf("sdp: %c='%s'\n", letter, buf);
384 get_word(buf1, sizeof(buf1), &p);
385 if (strcmp(buf1, "IN") != 0)
387 get_word(buf1, sizeof(buf1), &p);
388 if (strcmp(buf1, "IP4") != 0)
390 get_word_sep(buf1, sizeof(buf1), "/", &p);
391 if (inet_aton(buf1, &sdp_ip) == 0)
396 get_word_sep(buf1, sizeof(buf1), "/", &p);
399 if (s->nb_streams == 0) {
400 s1->default_ip = sdp_ip;
401 s1->default_ttl = ttl;
403 st = s->streams[s->nb_streams - 1];
404 rtsp_st = st->priv_data;
405 rtsp_st->sdp_ip = sdp_ip;
406 rtsp_st->sdp_ttl = ttl;
410 pstrcpy(s->title, sizeof(s->title), p);
413 if (s->nb_streams == 0) {
414 pstrcpy(s->comment, sizeof(s->comment), p);
420 get_word(st_type, sizeof(st_type), &p);
421 if (!strcmp(st_type, "audio")) {
422 codec_type = CODEC_TYPE_AUDIO;
423 } else if (!strcmp(st_type, "video")) {
424 codec_type = CODEC_TYPE_VIDEO;
428 rtsp_st = av_mallocz(sizeof(RTSPStream));
431 rtsp_st->stream_index = -1;
432 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
434 rtsp_st->sdp_ip = s1->default_ip;
435 rtsp_st->sdp_ttl = s1->default_ttl;
437 get_word(buf1, sizeof(buf1), &p); /* port */
438 rtsp_st->sdp_port = atoi(buf1);
440 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
442 /* XXX: handle list of formats */
443 get_word(buf1, sizeof(buf1), &p); /* format list */
444 rtsp_st->sdp_payload_type = atoi(buf1);
446 if (!strcmp(AVRtpPayloadTypes[rtsp_st->sdp_payload_type].enc_name, "MP2T")) {
447 /* no corresponding stream */
449 st = av_new_stream(s, 0);
452 st->priv_data = rtsp_st;
453 rtsp_st->stream_index = st->index;
454 st->codec->codec_type = codec_type;
455 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
456 /* if standard payload type, we can find the codec right now */
457 rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
460 /* put a default control url */
461 pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), s->filename);
464 if (strstart(p, "control:", &p) && s->nb_streams > 0) {
466 /* get the control url */
467 st = s->streams[s->nb_streams - 1];
468 rtsp_st = st->priv_data;
470 /* XXX: may need to add full url resolution */
471 url_split(proto, sizeof(proto), NULL, 0, NULL, 0, NULL, NULL, 0, p);
472 if (proto[0] == '\0') {
473 /* relative control URL */
474 pstrcat(rtsp_st->control_url, sizeof(rtsp_st->control_url), "/");
475 pstrcat(rtsp_st->control_url, sizeof(rtsp_st->control_url), p);
477 pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), p);
479 } else if (strstart(p, "rtpmap:", &p)) {
480 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
481 get_word(buf1, sizeof(buf1), &p);
482 payload_type = atoi(buf1);
483 for(i = 0; i < s->nb_streams;i++) {
485 rtsp_st = st->priv_data;
486 if (rtsp_st->sdp_payload_type == payload_type) {
487 sdp_parse_rtpmap(st->codec, rtsp_st, payload_type, p);
490 } else if (strstart(p, "fmtp:", &p)) {
491 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
492 get_word(buf1, sizeof(buf1), &p);
493 payload_type = atoi(buf1);
494 for(i = 0; i < s->nb_streams;i++) {
496 rtsp_st = st->priv_data;
497 if (rtsp_st->sdp_payload_type == payload_type) {
498 if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) {
499 if(!rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf)) {
500 sdp_parse_fmtp(st, p);
503 sdp_parse_fmtp(st, p);
507 } else if(strstart(p, "framesize:", &p)) {
508 // let dynamic protocol handlers have a stab at the line.
509 get_word(buf1, sizeof(buf1), &p);
510 payload_type = atoi(buf1);
511 for(i = 0; i < s->nb_streams;i++) {
513 rtsp_st = st->priv_data;
514 if (rtsp_st->sdp_payload_type == payload_type) {
515 if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) {
516 rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf);
520 } else if(strstart(p, "range:", &p)) {
523 // this is so that seeking on a streamed file can work.
524 rtsp_parse_range_npt(p, &start, &end);
525 s->start_time= start;
526 s->duration= (end==AV_NOPTS_VALUE)?AV_NOPTS_VALUE:end-start; // AV_NOPTS_VALUE means live broadcast (and can't seek)
532 static int sdp_parse(AVFormatContext *s, const char *content)
537 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
539 memset(s1, 0, sizeof(SDPParseState));
550 /* get the content */
552 while (*p != '\n' && *p != '\r' && *p != '\0') {
553 if ((q - buf) < sizeof(buf) - 1)
558 sdp_parse_line(s, s1, letter, buf);
560 while (*p != '\n' && *p != '\0')
568 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
575 v = strtol(p, (char **)&p, 10);
579 v = strtol(p, (char **)&p, 10);
588 /* XXX: only one transport specification is parsed */
589 static void rtsp_parse_transport(RTSPHeader *reply, const char *p)
591 char transport_protocol[16];
593 char lower_transport[16];
595 RTSPTransportField *th;
598 reply->nb_transports = 0;
605 th = &reply->transports[reply->nb_transports];
607 get_word_sep(transport_protocol, sizeof(transport_protocol),
611 get_word_sep(profile, sizeof(profile), "/;,", &p);
612 lower_transport[0] = '\0';
615 get_word_sep(lower_transport, sizeof(lower_transport),
618 if (!strcasecmp(lower_transport, "TCP"))
619 th->protocol = RTSP_PROTOCOL_RTP_TCP;
621 th->protocol = RTSP_PROTOCOL_RTP_UDP;
625 /* get each parameter */
626 while (*p != '\0' && *p != ',') {
627 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
628 if (!strcmp(parameter, "port")) {
631 rtsp_parse_range(&th->port_min, &th->port_max, &p);
633 } else if (!strcmp(parameter, "client_port")) {
636 rtsp_parse_range(&th->client_port_min,
637 &th->client_port_max, &p);
639 } else if (!strcmp(parameter, "server_port")) {
642 rtsp_parse_range(&th->server_port_min,
643 &th->server_port_max, &p);
645 } else if (!strcmp(parameter, "interleaved")) {
648 rtsp_parse_range(&th->interleaved_min,
649 &th->interleaved_max, &p);
651 } else if (!strcmp(parameter, "multicast")) {
652 if (th->protocol == RTSP_PROTOCOL_RTP_UDP)
653 th->protocol = RTSP_PROTOCOL_RTP_UDP_MULTICAST;
654 } else if (!strcmp(parameter, "ttl")) {
657 th->ttl = strtol(p, (char **)&p, 10);
659 } else if (!strcmp(parameter, "destination")) {
660 struct in_addr ipaddr;
664 get_word_sep(buf, sizeof(buf), ";,", &p);
665 if (inet_aton(buf, &ipaddr))
666 th->destination = ntohl(ipaddr.s_addr);
669 while (*p != ';' && *p != '\0' && *p != ',')
677 reply->nb_transports++;
681 void rtsp_parse_line(RTSPHeader *reply, const char *buf)
685 /* NOTE: we do case independent match for broken servers */
687 if (stristart(p, "Session:", &p)) {
688 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
689 } else if (stristart(p, "Content-Length:", &p)) {
690 reply->content_length = strtol(p, NULL, 10);
691 } else if (stristart(p, "Transport:", &p)) {
692 rtsp_parse_transport(reply, p);
693 } else if (stristart(p, "CSeq:", &p)) {
694 reply->seq = strtol(p, NULL, 10);
695 } else if (stristart(p, "Range:", &p)) {
696 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
700 static int url_readbuf(URLContext *h, unsigned char *buf, int size)
706 ret = url_read(h, buf+len, size-len);
714 /* skip a RTP/TCP interleaved packet */
715 static void rtsp_skip_packet(AVFormatContext *s)
717 RTSPState *rt = s->priv_data;
721 ret = url_readbuf(rt->rtsp_hd, buf, 3);
724 len = (buf[1] << 8) | buf[2];
726 printf("skipping RTP packet len=%d\n", len);
731 if (len1 > sizeof(buf))
733 ret = url_readbuf(rt->rtsp_hd, buf, len1);
740 static void rtsp_send_cmd(AVFormatContext *s,
741 const char *cmd, RTSPHeader *reply,
742 unsigned char **content_ptr)
744 RTSPState *rt = s->priv_data;
745 char buf[4096], buf1[1024], *q;
748 int content_length, line_count;
749 unsigned char *content = NULL;
751 memset(reply, 0, sizeof(RTSPHeader));
754 pstrcpy(buf, sizeof(buf), cmd);
755 snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq);
756 pstrcat(buf, sizeof(buf), buf1);
757 if (rt->session_id[0] != '\0' && !strstr(cmd, "\nIf-Match:")) {
758 snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id);
759 pstrcat(buf, sizeof(buf), buf1);
761 pstrcat(buf, sizeof(buf), "\r\n");
763 printf("Sending:\n%s--\n", buf);
765 url_write(rt->rtsp_hd, buf, strlen(buf));
767 /* parse reply (XXX: use buffers) */
769 rt->last_reply[0] = '\0';
773 if (url_readbuf(rt->rtsp_hd, &ch, 1) != 1)
778 /* XXX: only parse it if first char on line ? */
780 } else if (ch != '\r') {
781 if ((q - buf) < sizeof(buf) - 1)
787 printf("line='%s'\n", buf);
789 /* test if last line */
793 if (line_count == 0) {
795 get_word(buf1, sizeof(buf1), &p);
796 get_word(buf1, sizeof(buf1), &p);
797 reply->status_code = atoi(buf1);
799 rtsp_parse_line(reply, p);
800 pstrcat(rt->last_reply, sizeof(rt->last_reply), p);
801 pstrcat(rt->last_reply, sizeof(rt->last_reply), "\n");
806 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
807 pstrcpy(rt->session_id, sizeof(rt->session_id), reply->session_id);
809 content_length = reply->content_length;
810 if (content_length > 0) {
811 /* leave some room for a trailing '\0' (useful for simple parsing) */
812 content = av_malloc(content_length + 1);
813 (void)url_readbuf(rt->rtsp_hd, content, content_length);
814 content[content_length] = '\0';
817 *content_ptr = content;
821 void rtsp_set_callback(FFRTSPCallback *rtsp_cb)
823 ff_rtsp_callback = rtsp_cb;
827 /* close and free RTSP streams */
828 static void rtsp_close_streams(RTSPState *rt)
833 for(i=0;i<rt->nb_rtsp_streams;i++) {
834 rtsp_st = rt->rtsp_streams[i];
836 if (rtsp_st->rtp_ctx)
837 rtp_parse_close(rtsp_st->rtp_ctx);
838 if (rtsp_st->rtp_handle)
839 url_close(rtsp_st->rtp_handle);
840 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
841 rtsp_st->dynamic_handler->close(rtsp_st->dynamic_protocol_context);
845 av_free(rt->rtsp_streams);
848 static int rtsp_read_header(AVFormatContext *s,
849 AVFormatParameters *ap)
851 RTSPState *rt = s->priv_data;
852 char host[1024], path[1024], tcpname[1024], cmd[2048];
854 int port, i, j, ret, err;
855 RTSPHeader reply1, *reply = &reply1;
856 unsigned char *content = NULL;
861 /* extract hostname and port */
862 url_split(NULL, 0, NULL, 0,
863 host, sizeof(host), &port, path, sizeof(path), s->filename);
865 port = RTSP_DEFAULT_PORT;
867 /* open the tcp connexion */
868 snprintf(tcpname, sizeof(tcpname), "tcp://%s:%d", host, port);
869 if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0)
871 rt->rtsp_hd = rtsp_hd;
874 /* describe the stream */
875 snprintf(cmd, sizeof(cmd),
876 "DESCRIBE %s RTSP/1.0\r\n"
877 "Accept: application/sdp\r\n",
879 rtsp_send_cmd(s, cmd, reply, &content);
881 err = AVERROR_INVALIDDATA;
884 if (reply->status_code != RTSP_STATUS_OK) {
885 err = AVERROR_INVALIDDATA;
889 /* now we got the SDP description, we parse it */
890 ret = sdp_parse(s, (const char *)content);
893 err = AVERROR_INVALIDDATA;
897 protocol_mask = rtsp_default_protocols;
899 /* for each stream, make the setup request */
900 /* XXX: we assume the same server is used for the control of each
903 for(j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
904 char transport[2048];
906 rtsp_st = rt->rtsp_streams[i];
908 /* compute available transports */
912 if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP)) {
915 /* first try in specified port range */
916 if (RTSP_RTP_PORT_MIN != 0) {
917 while(j <= RTSP_RTP_PORT_MAX) {
918 snprintf(buf, sizeof(buf), "rtp://?localport=%d", j);
919 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0) {
920 j += 2; /* we will use two port by rtp stream (rtp and rtcp) */
926 /* then try on any port
927 ** if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
928 ** err = AVERROR_INVALIDDATA;
934 port = rtp_get_local_port(rtsp_st->rtp_handle);
935 if (transport[0] != '\0')
936 pstrcat(transport, sizeof(transport), ",");
937 snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1,
938 "RTP/AVP/UDP;unicast;client_port=%d-%d",
943 else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_TCP)) {
944 if (transport[0] != '\0')
945 pstrcat(transport, sizeof(transport), ",");
946 snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1,
950 else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST)) {
951 if (transport[0] != '\0')
952 pstrcat(transport, sizeof(transport), ",");
953 snprintf(transport + strlen(transport),
954 sizeof(transport) - strlen(transport) - 1,
955 "RTP/AVP/UDP;multicast");
957 snprintf(cmd, sizeof(cmd),
958 "SETUP %s RTSP/1.0\r\n"
960 rtsp_st->control_url, transport);
961 rtsp_send_cmd(s, cmd, reply, NULL);
962 if (reply->status_code != RTSP_STATUS_OK ||
963 reply->nb_transports != 1) {
964 err = AVERROR_INVALIDDATA;
968 /* XXX: same protocol for all streams is required */
970 if (reply->transports[0].protocol != rt->protocol) {
971 err = AVERROR_INVALIDDATA;
975 rt->protocol = reply->transports[0].protocol;
978 /* close RTP connection if not choosen */
979 if (reply->transports[0].protocol != RTSP_PROTOCOL_RTP_UDP &&
980 (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP))) {
981 url_close(rtsp_st->rtp_handle);
982 rtsp_st->rtp_handle = NULL;
985 switch(reply->transports[0].protocol) {
986 case RTSP_PROTOCOL_RTP_TCP:
987 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
988 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
991 case RTSP_PROTOCOL_RTP_UDP:
995 /* XXX: also use address if specified */
996 snprintf(url, sizeof(url), "rtp://%s:%d",
997 host, reply->transports[0].server_port_min);
998 if (rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
999 err = AVERROR_INVALIDDATA;
1004 case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
1009 ttl = reply->transports[0].ttl;
1012 snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d",
1014 reply->transports[0].server_port_min,
1016 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1017 err = AVERROR_INVALIDDATA;
1023 /* open the RTP context */
1025 if (rtsp_st->stream_index >= 0)
1026 st = s->streams[rtsp_st->stream_index];
1028 s->ctx_flags |= AVFMTCTX_NOHEADER;
1029 rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
1031 if (!rtsp_st->rtp_ctx) {
1032 err = AVERROR_NOMEM;
1035 if(rtsp_st->dynamic_handler) {
1036 rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context;
1037 rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet;
1042 /* use callback if available to extend setup */
1043 if (ff_rtsp_callback) {
1044 if (ff_rtsp_callback(RTSP_ACTION_CLIENT_SETUP, rt->session_id,
1045 NULL, 0, rt->last_reply) < 0) {
1046 err = AVERROR_INVALIDDATA;
1052 rt->state = RTSP_STATE_IDLE;
1053 rt->seek_timestamp = 0; /* default is to start stream at position
1055 if (ap->initial_pause) {
1056 /* do not start immediately */
1058 if (rtsp_read_play(s) < 0) {
1059 err = AVERROR_INVALIDDATA;
1065 rtsp_close_streams(rt);
1067 url_close(rt->rtsp_hd);
1071 static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1072 uint8_t *buf, int buf_size)
1074 RTSPState *rt = s->priv_data;
1075 int id, len, i, ret;
1076 RTSPStream *rtsp_st;
1078 #ifdef DEBUG_RTP_TCP
1079 printf("tcp_read_packet:\n");
1083 ret = url_readbuf(rt->rtsp_hd, buf, 1);
1084 #ifdef DEBUG_RTP_TCP
1085 printf("ret=%d c=%02x [%c]\n", ret, buf[0], buf[0]);
1092 ret = url_readbuf(rt->rtsp_hd, buf, 3);
1096 len = (buf[1] << 8) | buf[2];
1097 #ifdef DEBUG_RTP_TCP
1098 printf("id=%d len=%d\n", id, len);
1100 if (len > buf_size || len < 12)
1103 ret = url_readbuf(rt->rtsp_hd, buf, len);
1107 /* find the matching stream */
1108 for(i = 0; i < rt->nb_rtsp_streams; i++) {
1109 rtsp_st = rt->rtsp_streams[i];
1110 if (id >= rtsp_st->interleaved_min &&
1111 id <= rtsp_st->interleaved_max)
1116 *prtsp_st = rtsp_st;
1120 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1121 uint8_t *buf, int buf_size)
1123 RTSPState *rt = s->priv_data;
1124 RTSPStream *rtsp_st;
1126 int fd1, fd2, fd_max, n, i, ret;
1130 if (url_interrupt_cb())
1134 for(i = 0; i < rt->nb_rtsp_streams; i++) {
1135 rtsp_st = rt->rtsp_streams[i];
1136 /* currently, we cannot probe RTCP handle because of blocking restrictions */
1137 rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2);
1143 tv.tv_usec = 100 * 1000;
1144 n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
1146 for(i = 0; i < rt->nb_rtsp_streams; i++) {
1147 rtsp_st = rt->rtsp_streams[i];
1148 rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2);
1149 if (FD_ISSET(fd1, &rfds)) {
1150 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1152 *prtsp_st = rtsp_st;
1161 static int rtsp_read_packet(AVFormatContext *s,
1164 RTSPState *rt = s->priv_data;
1165 RTSPStream *rtsp_st;
1167 uint8_t buf[RTP_MAX_PACKET_LENGTH];
1169 /* get next frames from the same RTP packet */
1171 ret = rtp_parse_packet(rt->cur_rtp, pkt, NULL, 0);
1175 } else if (ret == 1) {
1182 /* read next RTP packet */
1184 switch(rt->protocol) {
1186 case RTSP_PROTOCOL_RTP_TCP:
1187 len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1189 case RTSP_PROTOCOL_RTP_UDP:
1190 case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
1191 len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1192 if (rtsp_st->rtp_ctx)
1193 rtp_check_and_send_back_rr(rtsp_st->rtp_ctx, len);
1198 ret = rtp_parse_packet(rtsp_st->rtp_ctx, pkt, buf, len);
1202 /* more packets may follow, so we save the RTP context */
1203 rt->cur_rtp = rtsp_st->rtp_ctx;
1208 static int rtsp_read_play(AVFormatContext *s)
1210 RTSPState *rt = s->priv_data;
1211 RTSPHeader reply1, *reply = &reply1;
1214 av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
1216 if (rt->state == RTSP_STATE_PAUSED) {
1217 snprintf(cmd, sizeof(cmd),
1218 "PLAY %s RTSP/1.0\r\n",
1221 snprintf(cmd, sizeof(cmd),
1222 "PLAY %s RTSP/1.0\r\n"
1223 "Range: npt=%0.3f-\r\n",
1225 (double)rt->seek_timestamp / AV_TIME_BASE);
1227 rtsp_send_cmd(s, cmd, reply, NULL);
1228 if (reply->status_code != RTSP_STATUS_OK) {
1231 rt->state = RTSP_STATE_PLAYING;
1236 /* pause the stream */
1237 static int rtsp_read_pause(AVFormatContext *s)
1239 RTSPState *rt = s->priv_data;
1240 RTSPHeader reply1, *reply = &reply1;
1245 if (rt->state != RTSP_STATE_PLAYING)
1248 snprintf(cmd, sizeof(cmd),
1249 "PAUSE %s RTSP/1.0\r\n",
1251 rtsp_send_cmd(s, cmd, reply, NULL);
1252 if (reply->status_code != RTSP_STATUS_OK) {
1255 rt->state = RTSP_STATE_PAUSED;
1260 static int rtsp_read_seek(AVFormatContext *s, int stream_index,
1261 int64_t timestamp, int flags)
1263 RTSPState *rt = s->priv_data;
1265 rt->seek_timestamp = timestamp;
1268 case RTSP_STATE_IDLE:
1270 case RTSP_STATE_PLAYING:
1271 if (rtsp_read_play(s) != 0)
1274 case RTSP_STATE_PAUSED:
1275 rt->state = RTSP_STATE_IDLE;
1281 static int rtsp_read_close(AVFormatContext *s)
1283 RTSPState *rt = s->priv_data;
1284 RTSPHeader reply1, *reply = &reply1;
1288 /* NOTE: it is valid to flush the buffer here */
1289 if (rt->protocol == RTSP_PROTOCOL_RTP_TCP) {
1290 url_fclose(&rt->rtsp_gb);
1293 snprintf(cmd, sizeof(cmd),
1294 "TEARDOWN %s RTSP/1.0\r\n",
1296 rtsp_send_cmd(s, cmd, reply, NULL);
1298 if (ff_rtsp_callback) {
1299 ff_rtsp_callback(RTSP_ACTION_CLIENT_TEARDOWN, rt->session_id,
1303 rtsp_close_streams(rt);
1304 url_close(rt->rtsp_hd);
1308 AVInputFormat rtsp_demuxer = {
1310 "RTSP input format",
1317 .flags = AVFMT_NOFILE,
1318 .read_play = rtsp_read_play,
1319 .read_pause = rtsp_read_pause,
1322 static int sdp_probe(AVProbeData *p1)
1324 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1326 /* we look for a line beginning "c=IN IP4" */
1327 while (p < p_end && *p != '\0') {
1328 if (p + sizeof("c=IN IP4") - 1 < p_end && strstart(p, "c=IN IP4", NULL))
1329 return AVPROBE_SCORE_MAX / 2;
1331 while(p < p_end - 1 && *p != '\n') p++;
1340 #define SDP_MAX_SIZE 8192
1342 static int sdp_read_header(AVFormatContext *s,
1343 AVFormatParameters *ap)
1345 RTSPState *rt = s->priv_data;
1346 RTSPStream *rtsp_st;
1352 /* read the whole sdp file */
1353 /* XXX: better loading */
1354 content = av_malloc(SDP_MAX_SIZE);
1355 size = get_buffer(&s->pb, content, SDP_MAX_SIZE - 1);
1358 return AVERROR_INVALIDDATA;
1360 content[size] ='\0';
1362 sdp_parse(s, content);
1365 /* open each RTP stream */
1366 for(i=0;i<rt->nb_rtsp_streams;i++) {
1367 rtsp_st = rt->rtsp_streams[i];
1369 snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d",
1370 inet_ntoa(rtsp_st->sdp_ip),
1373 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1374 err = AVERROR_INVALIDDATA;
1377 /* open the RTP context */
1379 if (rtsp_st->stream_index >= 0)
1380 st = s->streams[rtsp_st->stream_index];
1382 s->ctx_flags |= AVFMTCTX_NOHEADER;
1383 rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
1384 if (!rtsp_st->rtp_ctx) {
1385 err = AVERROR_NOMEM;
1388 if(rtsp_st->dynamic_handler) {
1389 rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context;
1390 rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet;
1396 rtsp_close_streams(rt);
1400 static int sdp_read_packet(AVFormatContext *s,
1403 return rtsp_read_packet(s, pkt);
1406 static int sdp_read_close(AVFormatContext *s)
1408 RTSPState *rt = s->priv_data;
1409 rtsp_close_streams(rt);
1413 #ifdef CONFIG_SDP_DEMUXER
1414 AVInputFormat sdp_demuxer = {
1425 /* dummy redirector format (used directly in av_open_input_file now) */
1426 static int redir_probe(AVProbeData *pd)
1430 while (redir_isspace(*p))
1432 if (strstart(p, "http://", NULL) ||
1433 strstart(p, "rtsp://", NULL))
1434 return AVPROBE_SCORE_MAX;
1438 /* called from utils.c */
1439 int redir_open(AVFormatContext **ic_ptr, ByteIOContext *f)
1443 AVFormatContext *ic = NULL;
1445 /* parse each URL and try to open it */
1447 while (c != URL_EOF) {
1450 if (!redir_isspace(c))
1459 if (c == URL_EOF || redir_isspace(c))
1461 if ((q - buf) < sizeof(buf) - 1)
1466 //printf("URL='%s'\n", buf);
1467 /* try to open the media file */
1468 if (av_open_input_file(&ic, buf, NULL, 0, NULL) == 0)
1478 AVInputFormat redir_demuxer = {
1480 "Redirector format",