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
23 #include <unistd.h> /* for select() prototype */
25 #include <netinet/in.h>
26 #include <sys/socket.h>
27 #include <arpa/inet.h>
29 #include "rtp_internal.h"
32 //#define DEBUG_RTP_TCP
34 enum RTSPClientState {
40 typedef struct RTSPState {
41 URLContext *rtsp_hd; /* RTSP TCP connexion handle */
43 struct RTSPStream **rtsp_streams;
45 enum RTSPClientState state;
46 int64_t seek_timestamp;
48 /* XXX: currently we use unbuffered input */
49 // ByteIOContext rtsp_gb;
50 int seq; /* RTSP command sequence number */
52 enum RTSPProtocol protocol;
53 char last_reply[2048]; /* XXX: allocate ? */
54 RTPDemuxContext *cur_rtp;
57 typedef struct RTSPStream {
58 URLContext *rtp_handle; /* RTP stream handle */
59 RTPDemuxContext *rtp_ctx; /* RTP parse context */
61 int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */
62 int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */
63 char control_url[1024]; /* url for this stream (from SDP) */
65 int sdp_port; /* port (from SDP content - not used in RTSP) */
66 struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */
67 int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */
68 int sdp_payload_type; /* payload type - only used in SDP */
69 rtp_payload_data_t rtp_payload_data; /* rtp payload parsing infos from SDP */
71 RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure)
72 void *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol)
75 static int rtsp_read_play(AVFormatContext *s);
77 /* XXX: currently, the only way to change the protocols consists in
78 changing this variable */
80 int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_UDP);
82 FFRTSPCallback *ff_rtsp_callback = NULL;
84 static int rtsp_probe(AVProbeData *p)
86 if (strstart(p->filename, "rtsp:", NULL))
87 return AVPROBE_SCORE_MAX;
91 static int redir_isspace(int c)
93 return (c == ' ' || c == '\t' || c == '\n' || c == '\r');
96 static void skip_spaces(const char **pp)
100 while (redir_isspace(*p))
105 static void get_word_sep(char *buf, int buf_size, const char *sep,
116 while (!strchr(sep, *p) && *p != '\0') {
117 if ((q - buf) < buf_size - 1)
126 static void get_word(char *buf, int buf_size, const char **pp)
134 while (!redir_isspace(*p) && *p != '\0') {
135 if ((q - buf) < buf_size - 1)
144 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other
146 static int sdp_parse_rtpmap(AVCodecContext *codec, RTSPStream *rtsp_st, int payload_type, const char *p)
153 /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
154 see if we can handle this kind of payload */
155 get_word_sep(buf, sizeof(buf), "/", &p);
156 if (payload_type >= RTP_PT_PRIVATE) {
157 RTPDynamicProtocolHandler *handler= RTPFirstDynamicPayloadHandler;
159 if (!strcmp(buf, handler->enc_name) && (codec->codec_type == handler->codec_type)) {
160 codec->codec_id = handler->codec_id;
161 rtsp_st->dynamic_handler= handler;
163 rtsp_st->dynamic_protocol_context= handler->open();
167 handler= handler->next;
170 /* We are in a standard case ( from http://www.iana.org/assignments/rtp-parameters) */
171 /* search into AVRtpPayloadTypes[] */
172 for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
173 if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpPayloadTypes[i].codec_type)){
174 codec->codec_id = AVRtpPayloadTypes[i].codec_id;
179 c = avcodec_find_decoder(codec->codec_id);
183 c_name = (char *)NULL;
186 get_word_sep(buf, sizeof(buf), "/", &p);
188 switch (codec->codec_type) {
189 case CODEC_TYPE_AUDIO:
190 av_log(codec, AV_LOG_DEBUG, " audio codec set to : %s\n", c_name);
191 codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
192 codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
194 codec->sample_rate = i;
195 get_word_sep(buf, sizeof(buf), "/", &p);
199 // TODO: there is a bug here; if it is a mono stream, and less than 22000Hz, faad upconverts to stereo and twice the
200 // frequency. No problem, but the sample rate is being set here by the sdp line. Upcoming patch forthcoming. (rdm)
202 av_log(codec, AV_LOG_DEBUG, " audio samplerate set to : %i\n", codec->sample_rate);
203 av_log(codec, AV_LOG_DEBUG, " audio channels set to : %i\n", codec->channels);
205 case CODEC_TYPE_VIDEO:
206 av_log(codec, AV_LOG_DEBUG, " video codec set to : %s\n", c_name);
217 /* return the length and optionnaly the data */
218 static int hex_to_data(uint8_t *data, const char *p)
228 c = toupper((unsigned char)*p++);
229 if (c >= '0' && c <= '9')
231 else if (c >= 'A' && c <= 'F')
246 static void sdp_parse_fmtp_config(AVCodecContext *codec, char *attr, char *value)
248 switch (codec->codec_id) {
251 if (!strcmp(attr, "config")) {
252 /* decode the hexa encoded parameter */
253 int len = hex_to_data(NULL, value);
254 codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
255 if (!codec->extradata)
257 codec->extradata_size = len;
258 hex_to_data(codec->extradata, value);
267 typedef struct attrname_map
274 /* All known fmtp parmeters and the corresping RTPAttrTypeEnum */
275 #define ATTR_NAME_TYPE_INT 0
276 #define ATTR_NAME_TYPE_STR 1
277 static attrname_map_t attr_names[]=
279 {"SizeLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, sizelength)},
280 {"IndexLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexlength)},
281 {"IndexDeltaLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexdeltalength)},
282 {"profile-level-id", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, profile_level_id)},
283 {"StreamType", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, streamtype)},
284 {"mode", ATTR_NAME_TYPE_STR, offsetof(rtp_payload_data_t, mode)},
288 /** parse the attribute line from the fmtp a line of an sdp resonse. This is broken out as a function
289 * because it is used in rtp_h264.c, which is forthcoming.
291 int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
296 get_word_sep(attr, attr_size, "=", p);
299 get_word_sep(value, value_size, ";", p);
307 /* parse a SDP line and save stream attributes */
308 static void sdp_parse_fmtp(AVStream *st, const char *p)
314 RTSPStream *rtsp_st = st->priv_data;
315 AVCodecContext *codec = st->codec;
316 rtp_payload_data_t *rtp_payload_data = &rtsp_st->rtp_payload_data;
318 /* loop on each attribute */
319 while(rtsp_next_attr_and_value(&p, attr, sizeof(attr), value, sizeof(value)))
321 /* grab the codec extra_data from the config parameter of the fmtp line */
322 sdp_parse_fmtp_config(codec, attr, value);
323 /* Looking for a known attribute */
324 for (i = 0; attr_names[i].str; ++i) {
325 if (!strcasecmp(attr, attr_names[i].str)) {
326 if (attr_names[i].type == ATTR_NAME_TYPE_INT)
327 *(int *)((char *)rtp_payload_data + attr_names[i].offset) = atoi(value);
328 else if (attr_names[i].type == ATTR_NAME_TYPE_STR)
329 *(char **)((char *)rtp_payload_data + attr_names[i].offset) = av_strdup(value);
335 /** Parse a string \p in the form of Range:npt=xx-xx, and determine the start
337 * Used for seeking in the rtp stream.
339 static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
344 if (!stristart(p, "npt=", &p))
347 *start = AV_NOPTS_VALUE;
348 *end = AV_NOPTS_VALUE;
350 get_word_sep(buf, sizeof(buf), "-", &p);
351 *start = parse_date(buf, 1);
354 get_word_sep(buf, sizeof(buf), "-", &p);
355 *end = parse_date(buf, 1);
357 // av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
358 // av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
361 typedef struct SDPParseState {
363 struct in_addr default_ip;
367 static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
368 int letter, const char *buf)
370 RTSPState *rt = s->priv_data;
371 char buf1[64], st_type[64];
373 int codec_type, payload_type, i;
376 struct in_addr sdp_ip;
380 printf("sdp: %c='%s'\n", letter, buf);
386 get_word(buf1, sizeof(buf1), &p);
387 if (strcmp(buf1, "IN") != 0)
389 get_word(buf1, sizeof(buf1), &p);
390 if (strcmp(buf1, "IP4") != 0)
392 get_word_sep(buf1, sizeof(buf1), "/", &p);
393 if (inet_aton(buf1, &sdp_ip) == 0)
398 get_word_sep(buf1, sizeof(buf1), "/", &p);
401 if (s->nb_streams == 0) {
402 s1->default_ip = sdp_ip;
403 s1->default_ttl = ttl;
405 st = s->streams[s->nb_streams - 1];
406 rtsp_st = st->priv_data;
407 rtsp_st->sdp_ip = sdp_ip;
408 rtsp_st->sdp_ttl = ttl;
412 pstrcpy(s->title, sizeof(s->title), p);
415 if (s->nb_streams == 0) {
416 pstrcpy(s->comment, sizeof(s->comment), p);
422 get_word(st_type, sizeof(st_type), &p);
423 if (!strcmp(st_type, "audio")) {
424 codec_type = CODEC_TYPE_AUDIO;
425 } else if (!strcmp(st_type, "video")) {
426 codec_type = CODEC_TYPE_VIDEO;
430 rtsp_st = av_mallocz(sizeof(RTSPStream));
433 rtsp_st->stream_index = -1;
434 dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
436 rtsp_st->sdp_ip = s1->default_ip;
437 rtsp_st->sdp_ttl = s1->default_ttl;
439 get_word(buf1, sizeof(buf1), &p); /* port */
440 rtsp_st->sdp_port = atoi(buf1);
442 get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
444 /* XXX: handle list of formats */
445 get_word(buf1, sizeof(buf1), &p); /* format list */
446 rtsp_st->sdp_payload_type = atoi(buf1);
448 if (!strcmp(AVRtpPayloadTypes[rtsp_st->sdp_payload_type].enc_name, "MP2T")) {
449 /* no corresponding stream */
451 st = av_new_stream(s, 0);
454 st->priv_data = rtsp_st;
455 rtsp_st->stream_index = st->index;
456 st->codec->codec_type = codec_type;
457 if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
458 /* if standard payload type, we can find the codec right now */
459 rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
462 /* put a default control url */
463 pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), s->filename);
466 if (strstart(p, "control:", &p) && s->nb_streams > 0) {
468 /* get the control url */
469 st = s->streams[s->nb_streams - 1];
470 rtsp_st = st->priv_data;
472 /* XXX: may need to add full url resolution */
473 url_split(proto, sizeof(proto), NULL, 0, NULL, 0, NULL, NULL, 0, p);
474 if (proto[0] == '\0') {
475 /* relative control URL */
476 pstrcat(rtsp_st->control_url, sizeof(rtsp_st->control_url), "/");
477 pstrcat(rtsp_st->control_url, sizeof(rtsp_st->control_url), p);
479 pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), p);
481 } else if (strstart(p, "rtpmap:", &p)) {
482 /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
483 get_word(buf1, sizeof(buf1), &p);
484 payload_type = atoi(buf1);
485 for(i = 0; i < s->nb_streams;i++) {
487 rtsp_st = st->priv_data;
488 if (rtsp_st->sdp_payload_type == payload_type) {
489 sdp_parse_rtpmap(st->codec, rtsp_st, payload_type, p);
492 } else if (strstart(p, "fmtp:", &p)) {
493 /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
494 get_word(buf1, sizeof(buf1), &p);
495 payload_type = atoi(buf1);
496 for(i = 0; i < s->nb_streams;i++) {
498 rtsp_st = st->priv_data;
499 if (rtsp_st->sdp_payload_type == payload_type) {
500 if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) {
501 if(!rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf)) {
502 sdp_parse_fmtp(st, p);
505 sdp_parse_fmtp(st, p);
509 } else if(strstart(p, "framesize:", &p)) {
510 // let dynamic protocol handlers have a stab at the line.
511 get_word(buf1, sizeof(buf1), &p);
512 payload_type = atoi(buf1);
513 for(i = 0; i < s->nb_streams;i++) {
515 rtsp_st = st->priv_data;
516 if (rtsp_st->sdp_payload_type == payload_type) {
517 if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) {
518 rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf);
522 } else if(strstart(p, "range:", &p)) {
525 // this is so that seeking on a streamed file can work.
526 rtsp_parse_range_npt(p, &start, &end);
527 s->start_time= start;
528 s->duration= (end==AV_NOPTS_VALUE)?AV_NOPTS_VALUE:end-start; // AV_NOPTS_VALUE means live broadcast (and can't seek)
534 static int sdp_parse(AVFormatContext *s, const char *content)
539 SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
541 memset(s1, 0, sizeof(SDPParseState));
552 /* get the content */
554 while (*p != '\n' && *p != '\r' && *p != '\0') {
555 if ((q - buf) < sizeof(buf) - 1)
560 sdp_parse_line(s, s1, letter, buf);
562 while (*p != '\n' && *p != '\0')
570 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
577 v = strtol(p, (char **)&p, 10);
581 v = strtol(p, (char **)&p, 10);
590 /* XXX: only one transport specification is parsed */
591 static void rtsp_parse_transport(RTSPHeader *reply, const char *p)
593 char transport_protocol[16];
595 char lower_transport[16];
597 RTSPTransportField *th;
600 reply->nb_transports = 0;
607 th = &reply->transports[reply->nb_transports];
609 get_word_sep(transport_protocol, sizeof(transport_protocol),
613 get_word_sep(profile, sizeof(profile), "/;,", &p);
614 lower_transport[0] = '\0';
617 get_word_sep(lower_transport, sizeof(lower_transport),
620 if (!strcasecmp(lower_transport, "TCP"))
621 th->protocol = RTSP_PROTOCOL_RTP_TCP;
623 th->protocol = RTSP_PROTOCOL_RTP_UDP;
627 /* get each parameter */
628 while (*p != '\0' && *p != ',') {
629 get_word_sep(parameter, sizeof(parameter), "=;,", &p);
630 if (!strcmp(parameter, "port")) {
633 rtsp_parse_range(&th->port_min, &th->port_max, &p);
635 } else if (!strcmp(parameter, "client_port")) {
638 rtsp_parse_range(&th->client_port_min,
639 &th->client_port_max, &p);
641 } else if (!strcmp(parameter, "server_port")) {
644 rtsp_parse_range(&th->server_port_min,
645 &th->server_port_max, &p);
647 } else if (!strcmp(parameter, "interleaved")) {
650 rtsp_parse_range(&th->interleaved_min,
651 &th->interleaved_max, &p);
653 } else if (!strcmp(parameter, "multicast")) {
654 if (th->protocol == RTSP_PROTOCOL_RTP_UDP)
655 th->protocol = RTSP_PROTOCOL_RTP_UDP_MULTICAST;
656 } else if (!strcmp(parameter, "ttl")) {
659 th->ttl = strtol(p, (char **)&p, 10);
661 } else if (!strcmp(parameter, "destination")) {
662 struct in_addr ipaddr;
666 get_word_sep(buf, sizeof(buf), ";,", &p);
667 if (inet_aton(buf, &ipaddr))
668 th->destination = ntohl(ipaddr.s_addr);
671 while (*p != ';' && *p != '\0' && *p != ',')
679 reply->nb_transports++;
683 void rtsp_parse_line(RTSPHeader *reply, const char *buf)
687 /* NOTE: we do case independent match for broken servers */
689 if (stristart(p, "Session:", &p)) {
690 get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
691 } else if (stristart(p, "Content-Length:", &p)) {
692 reply->content_length = strtol(p, NULL, 10);
693 } else if (stristart(p, "Transport:", &p)) {
694 rtsp_parse_transport(reply, p);
695 } else if (stristart(p, "CSeq:", &p)) {
696 reply->seq = strtol(p, NULL, 10);
697 } else if (stristart(p, "Range:", &p)) {
698 rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
702 static int url_readbuf(URLContext *h, unsigned char *buf, int size)
708 ret = url_read(h, buf+len, size-len);
716 /* skip a RTP/TCP interleaved packet */
717 static void rtsp_skip_packet(AVFormatContext *s)
719 RTSPState *rt = s->priv_data;
723 ret = url_readbuf(rt->rtsp_hd, buf, 3);
726 len = (buf[1] << 8) | buf[2];
728 printf("skipping RTP packet len=%d\n", len);
733 if (len1 > sizeof(buf))
735 ret = url_readbuf(rt->rtsp_hd, buf, len1);
742 static void rtsp_send_cmd(AVFormatContext *s,
743 const char *cmd, RTSPHeader *reply,
744 unsigned char **content_ptr)
746 RTSPState *rt = s->priv_data;
747 char buf[4096], buf1[1024], *q;
750 int content_length, line_count;
751 unsigned char *content = NULL;
753 memset(reply, 0, sizeof(RTSPHeader));
756 pstrcpy(buf, sizeof(buf), cmd);
757 snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq);
758 pstrcat(buf, sizeof(buf), buf1);
759 if (rt->session_id[0] != '\0' && !strstr(cmd, "\nIf-Match:")) {
760 snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id);
761 pstrcat(buf, sizeof(buf), buf1);
763 pstrcat(buf, sizeof(buf), "\r\n");
765 printf("Sending:\n%s--\n", buf);
767 url_write(rt->rtsp_hd, buf, strlen(buf));
769 /* parse reply (XXX: use buffers) */
771 rt->last_reply[0] = '\0';
775 if (url_readbuf(rt->rtsp_hd, &ch, 1) != 1)
780 /* XXX: only parse it if first char on line ? */
782 } else if (ch != '\r') {
783 if ((q - buf) < sizeof(buf) - 1)
789 printf("line='%s'\n", buf);
791 /* test if last line */
795 if (line_count == 0) {
797 get_word(buf1, sizeof(buf1), &p);
798 get_word(buf1, sizeof(buf1), &p);
799 reply->status_code = atoi(buf1);
801 rtsp_parse_line(reply, p);
802 pstrcat(rt->last_reply, sizeof(rt->last_reply), p);
803 pstrcat(rt->last_reply, sizeof(rt->last_reply), "\n");
808 if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
809 pstrcpy(rt->session_id, sizeof(rt->session_id), reply->session_id);
811 content_length = reply->content_length;
812 if (content_length > 0) {
813 /* leave some room for a trailing '\0' (useful for simple parsing) */
814 content = av_malloc(content_length + 1);
815 (void)url_readbuf(rt->rtsp_hd, content, content_length);
816 content[content_length] = '\0';
819 *content_ptr = content;
822 /* useful for modules: set RTSP callback function */
824 void rtsp_set_callback(FFRTSPCallback *rtsp_cb)
826 ff_rtsp_callback = rtsp_cb;
830 /* close and free RTSP streams */
831 static void rtsp_close_streams(RTSPState *rt)
836 for(i=0;i<rt->nb_rtsp_streams;i++) {
837 rtsp_st = rt->rtsp_streams[i];
839 if (rtsp_st->rtp_ctx)
840 rtp_parse_close(rtsp_st->rtp_ctx);
841 if (rtsp_st->rtp_handle)
842 url_close(rtsp_st->rtp_handle);
843 if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
844 rtsp_st->dynamic_handler->close(rtsp_st->dynamic_protocol_context);
848 av_free(rt->rtsp_streams);
851 static int rtsp_read_header(AVFormatContext *s,
852 AVFormatParameters *ap)
854 RTSPState *rt = s->priv_data;
855 char host[1024], path[1024], tcpname[1024], cmd[2048];
857 int port, i, j, ret, err;
858 RTSPHeader reply1, *reply = &reply1;
859 unsigned char *content = NULL;
864 /* extract hostname and port */
865 url_split(NULL, 0, NULL, 0,
866 host, sizeof(host), &port, path, sizeof(path), s->filename);
868 port = RTSP_DEFAULT_PORT;
870 /* open the tcp connexion */
871 snprintf(tcpname, sizeof(tcpname), "tcp://%s:%d", host, port);
872 if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0)
874 rt->rtsp_hd = rtsp_hd;
877 /* describe the stream */
878 snprintf(cmd, sizeof(cmd),
879 "DESCRIBE %s RTSP/1.0\r\n"
880 "Accept: application/sdp\r\n",
882 rtsp_send_cmd(s, cmd, reply, &content);
884 err = AVERROR_INVALIDDATA;
887 if (reply->status_code != RTSP_STATUS_OK) {
888 err = AVERROR_INVALIDDATA;
892 /* now we got the SDP description, we parse it */
893 ret = sdp_parse(s, (const char *)content);
896 err = AVERROR_INVALIDDATA;
900 protocol_mask = rtsp_default_protocols;
902 /* for each stream, make the setup request */
903 /* XXX: we assume the same server is used for the control of each
906 for(j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
907 char transport[2048];
909 rtsp_st = rt->rtsp_streams[i];
911 /* compute available transports */
915 if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP)) {
918 /* first try in specified port range */
919 if (RTSP_RTP_PORT_MIN != 0) {
920 while(j <= RTSP_RTP_PORT_MAX) {
921 snprintf(buf, sizeof(buf), "rtp://?localport=%d", j);
922 if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0) {
923 j += 2; /* we will use two port by rtp stream (rtp and rtcp) */
929 /* then try on any port
930 ** if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
931 ** err = AVERROR_INVALIDDATA;
937 port = rtp_get_local_port(rtsp_st->rtp_handle);
938 if (transport[0] != '\0')
939 pstrcat(transport, sizeof(transport), ",");
940 snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1,
941 "RTP/AVP/UDP;unicast;client_port=%d-%d",
946 else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_TCP)) {
947 if (transport[0] != '\0')
948 pstrcat(transport, sizeof(transport), ",");
949 snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1,
953 else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST)) {
954 if (transport[0] != '\0')
955 pstrcat(transport, sizeof(transport), ",");
956 snprintf(transport + strlen(transport),
957 sizeof(transport) - strlen(transport) - 1,
958 "RTP/AVP/UDP;multicast");
960 snprintf(cmd, sizeof(cmd),
961 "SETUP %s RTSP/1.0\r\n"
963 rtsp_st->control_url, transport);
964 rtsp_send_cmd(s, cmd, reply, NULL);
965 if (reply->status_code != RTSP_STATUS_OK ||
966 reply->nb_transports != 1) {
967 err = AVERROR_INVALIDDATA;
971 /* XXX: same protocol for all streams is required */
973 if (reply->transports[0].protocol != rt->protocol) {
974 err = AVERROR_INVALIDDATA;
978 rt->protocol = reply->transports[0].protocol;
981 /* close RTP connection if not choosen */
982 if (reply->transports[0].protocol != RTSP_PROTOCOL_RTP_UDP &&
983 (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP))) {
984 url_close(rtsp_st->rtp_handle);
985 rtsp_st->rtp_handle = NULL;
988 switch(reply->transports[0].protocol) {
989 case RTSP_PROTOCOL_RTP_TCP:
990 rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
991 rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
994 case RTSP_PROTOCOL_RTP_UDP:
998 /* XXX: also use address if specified */
999 snprintf(url, sizeof(url), "rtp://%s:%d",
1000 host, reply->transports[0].server_port_min);
1001 if (rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
1002 err = AVERROR_INVALIDDATA;
1007 case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
1012 ttl = reply->transports[0].ttl;
1015 snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d",
1017 reply->transports[0].server_port_min,
1019 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1020 err = AVERROR_INVALIDDATA;
1026 /* open the RTP context */
1028 if (rtsp_st->stream_index >= 0)
1029 st = s->streams[rtsp_st->stream_index];
1031 s->ctx_flags |= AVFMTCTX_NOHEADER;
1032 rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
1034 if (!rtsp_st->rtp_ctx) {
1035 err = AVERROR_NOMEM;
1038 if(rtsp_st->dynamic_handler) {
1039 rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context;
1040 rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet;
1045 /* use callback if available to extend setup */
1046 if (ff_rtsp_callback) {
1047 if (ff_rtsp_callback(RTSP_ACTION_CLIENT_SETUP, rt->session_id,
1048 NULL, 0, rt->last_reply) < 0) {
1049 err = AVERROR_INVALIDDATA;
1055 rt->state = RTSP_STATE_IDLE;
1056 rt->seek_timestamp = 0; /* default is to start stream at position
1058 if (ap->initial_pause) {
1059 /* do not start immediately */
1061 if (rtsp_read_play(s) < 0) {
1062 err = AVERROR_INVALIDDATA;
1068 rtsp_close_streams(rt);
1070 url_close(rt->rtsp_hd);
1074 static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1075 uint8_t *buf, int buf_size)
1077 RTSPState *rt = s->priv_data;
1078 int id, len, i, ret;
1079 RTSPStream *rtsp_st;
1081 #ifdef DEBUG_RTP_TCP
1082 printf("tcp_read_packet:\n");
1086 ret = url_readbuf(rt->rtsp_hd, buf, 1);
1087 #ifdef DEBUG_RTP_TCP
1088 printf("ret=%d c=%02x [%c]\n", ret, buf[0], buf[0]);
1095 ret = url_readbuf(rt->rtsp_hd, buf, 3);
1099 len = (buf[1] << 8) | buf[2];
1100 #ifdef DEBUG_RTP_TCP
1101 printf("id=%d len=%d\n", id, len);
1103 if (len > buf_size || len < 12)
1106 ret = url_readbuf(rt->rtsp_hd, buf, len);
1110 /* find the matching stream */
1111 for(i = 0; i < rt->nb_rtsp_streams; i++) {
1112 rtsp_st = rt->rtsp_streams[i];
1113 if (id >= rtsp_st->interleaved_min &&
1114 id <= rtsp_st->interleaved_max)
1119 *prtsp_st = rtsp_st;
1123 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
1124 uint8_t *buf, int buf_size)
1126 RTSPState *rt = s->priv_data;
1127 RTSPStream *rtsp_st;
1129 int fd1, fd2, fd_max, n, i, ret;
1133 if (url_interrupt_cb())
1137 for(i = 0; i < rt->nb_rtsp_streams; i++) {
1138 rtsp_st = rt->rtsp_streams[i];
1139 /* currently, we cannot probe RTCP handle because of blocking restrictions */
1140 rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2);
1146 tv.tv_usec = 100 * 1000;
1147 n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
1149 for(i = 0; i < rt->nb_rtsp_streams; i++) {
1150 rtsp_st = rt->rtsp_streams[i];
1151 rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2);
1152 if (FD_ISSET(fd1, &rfds)) {
1153 ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
1155 *prtsp_st = rtsp_st;
1164 static int rtsp_read_packet(AVFormatContext *s,
1167 RTSPState *rt = s->priv_data;
1168 RTSPStream *rtsp_st;
1170 uint8_t buf[RTP_MAX_PACKET_LENGTH];
1172 /* get next frames from the same RTP packet */
1174 ret = rtp_parse_packet(rt->cur_rtp, pkt, NULL, 0);
1178 } else if (ret == 1) {
1185 /* read next RTP packet */
1187 switch(rt->protocol) {
1189 case RTSP_PROTOCOL_RTP_TCP:
1190 len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1192 case RTSP_PROTOCOL_RTP_UDP:
1193 case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
1194 len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
1195 if (rtsp_st->rtp_ctx)
1196 rtp_check_and_send_back_rr(rtsp_st->rtp_ctx, len);
1201 ret = rtp_parse_packet(rtsp_st->rtp_ctx, pkt, buf, len);
1205 /* more packets may follow, so we save the RTP context */
1206 rt->cur_rtp = rtsp_st->rtp_ctx;
1211 static int rtsp_read_play(AVFormatContext *s)
1213 RTSPState *rt = s->priv_data;
1214 RTSPHeader reply1, *reply = &reply1;
1217 av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
1219 if (rt->state == RTSP_STATE_PAUSED) {
1220 snprintf(cmd, sizeof(cmd),
1221 "PLAY %s RTSP/1.0\r\n",
1224 snprintf(cmd, sizeof(cmd),
1225 "PLAY %s RTSP/1.0\r\n"
1226 "Range: npt=%0.3f-\r\n",
1228 (double)rt->seek_timestamp / AV_TIME_BASE);
1230 rtsp_send_cmd(s, cmd, reply, NULL);
1231 if (reply->status_code != RTSP_STATUS_OK) {
1234 rt->state = RTSP_STATE_PLAYING;
1239 /* pause the stream */
1240 static int rtsp_read_pause(AVFormatContext *s)
1242 RTSPState *rt = s->priv_data;
1243 RTSPHeader reply1, *reply = &reply1;
1248 if (rt->state != RTSP_STATE_PLAYING)
1251 snprintf(cmd, sizeof(cmd),
1252 "PAUSE %s RTSP/1.0\r\n",
1254 rtsp_send_cmd(s, cmd, reply, NULL);
1255 if (reply->status_code != RTSP_STATUS_OK) {
1258 rt->state = RTSP_STATE_PAUSED;
1263 static int rtsp_read_seek(AVFormatContext *s, int stream_index,
1264 int64_t timestamp, int flags)
1266 RTSPState *rt = s->priv_data;
1268 rt->seek_timestamp = timestamp;
1271 case RTSP_STATE_IDLE:
1273 case RTSP_STATE_PLAYING:
1274 if (rtsp_read_play(s) != 0)
1277 case RTSP_STATE_PAUSED:
1278 rt->state = RTSP_STATE_IDLE;
1284 static int rtsp_read_close(AVFormatContext *s)
1286 RTSPState *rt = s->priv_data;
1287 RTSPHeader reply1, *reply = &reply1;
1291 /* NOTE: it is valid to flush the buffer here */
1292 if (rt->protocol == RTSP_PROTOCOL_RTP_TCP) {
1293 url_fclose(&rt->rtsp_gb);
1296 snprintf(cmd, sizeof(cmd),
1297 "TEARDOWN %s RTSP/1.0\r\n",
1299 rtsp_send_cmd(s, cmd, reply, NULL);
1301 if (ff_rtsp_callback) {
1302 ff_rtsp_callback(RTSP_ACTION_CLIENT_TEARDOWN, rt->session_id,
1306 rtsp_close_streams(rt);
1307 url_close(rt->rtsp_hd);
1311 AVInputFormat rtsp_demuxer = {
1313 "RTSP input format",
1320 .flags = AVFMT_NOFILE,
1321 .read_play = rtsp_read_play,
1322 .read_pause = rtsp_read_pause,
1325 static int sdp_probe(AVProbeData *p1)
1327 const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1329 /* we look for a line beginning "c=IN IP4" */
1330 while (p < p_end && *p != '\0') {
1331 if (p + sizeof("c=IN IP4") - 1 < p_end && strstart(p, "c=IN IP4", NULL))
1332 return AVPROBE_SCORE_MAX / 2;
1334 while(p < p_end - 1 && *p != '\n') p++;
1343 #define SDP_MAX_SIZE 8192
1345 static int sdp_read_header(AVFormatContext *s,
1346 AVFormatParameters *ap)
1348 RTSPState *rt = s->priv_data;
1349 RTSPStream *rtsp_st;
1355 /* read the whole sdp file */
1356 /* XXX: better loading */
1357 content = av_malloc(SDP_MAX_SIZE);
1358 size = get_buffer(&s->pb, content, SDP_MAX_SIZE - 1);
1361 return AVERROR_INVALIDDATA;
1363 content[size] ='\0';
1365 sdp_parse(s, content);
1368 /* open each RTP stream */
1369 for(i=0;i<rt->nb_rtsp_streams;i++) {
1370 rtsp_st = rt->rtsp_streams[i];
1372 snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d",
1373 inet_ntoa(rtsp_st->sdp_ip),
1376 if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
1377 err = AVERROR_INVALIDDATA;
1380 /* open the RTP context */
1382 if (rtsp_st->stream_index >= 0)
1383 st = s->streams[rtsp_st->stream_index];
1385 s->ctx_flags |= AVFMTCTX_NOHEADER;
1386 rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
1387 if (!rtsp_st->rtp_ctx) {
1388 err = AVERROR_NOMEM;
1391 if(rtsp_st->dynamic_handler) {
1392 rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context;
1393 rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet;
1399 rtsp_close_streams(rt);
1403 static int sdp_read_packet(AVFormatContext *s,
1406 return rtsp_read_packet(s, pkt);
1409 static int sdp_read_close(AVFormatContext *s)
1411 RTSPState *rt = s->priv_data;
1412 rtsp_close_streams(rt);
1416 #ifdef CONFIG_SDP_DEMUXER
1417 AVInputFormat sdp_demuxer = {
1428 /* dummy redirector format (used directly in av_open_input_file now) */
1429 static int redir_probe(AVProbeData *pd)
1433 while (redir_isspace(*p))
1435 if (strstart(p, "http://", NULL) ||
1436 strstart(p, "rtsp://", NULL))
1437 return AVPROBE_SCORE_MAX;
1441 /* called from utils.c */
1442 int redir_open(AVFormatContext **ic_ptr, ByteIOContext *f)
1446 AVFormatContext *ic = NULL;
1448 /* parse each URL and try to open it */
1450 while (c != URL_EOF) {
1453 if (!redir_isspace(c))
1462 if (c == URL_EOF || redir_isspace(c))
1464 if ((q - buf) < sizeof(buf) - 1)
1469 //printf("URL='%s'\n", buf);
1470 /* try to open the media file */
1471 if (av_open_input_file(&ic, buf, NULL, 0, NULL) == 0)
1481 AVInputFormat redir_demuxer = {
1483 "Redirector format",