/*

  * 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/avassert.h"

 #include "libavutil/base64.h"

 #include "libavutil/avstring.h"

 #include "libavutil/intreadwrite.h"

 #include "libavutil/mathematics.h"

 #include "libavutil/parseutils.h"

 #include "libavutil/random_seed.h"

 #include "libavutil/dict.h"

 #include "libavutil/opt.h"

 #include "libavutil/time.h"

 #include "avformat.h"

 #include "avio_internal.h"

 #if HAVE_POLL_H
 #include <poll.h>

 #endif

 #include "internal.h"

 #include "network.h"

 #include "os_support.h"

 #include "http.h"

 #include "rtsp.h"

 #include "rtpdec.h"

 #include "rtpproto.h"

 #include "rdt.h"

 #include "rtpdec_formats.h"

 #include "rtpenc_chain.h"

 #include "url.h"

 #include "rtpenc.h"

 #include "mpegts.h"

 /* Timeout values for socket poll, in ms,

  * and read_packet(), in seconds  */

 #define POLL_TIMEOUT_MS 100

 #define READ_PACKET_TIMEOUT_S 10

 #define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS

 #define SDP_MAX_SIZE 16384

 #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH

 #define DEFAULT_REORDERING_DELAY 100000

 #define OFFSET(x) offsetof(RTSPState, x)
 #define DEC AV_OPT_FLAG_DECODING_PARAM

 #define ENC AV_OPT_FLAG_ENCODING_PARAM

 
 #define RTSP_FLAG_OPTS(name, longname) \

     { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \

     { "filter_src", "only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }

 #define RTSP_MEDIATYPE_OPTS(name, longname) \

     { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { .i64 = (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \

     { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
     { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
     { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }

 #define RTSP_REORDERING_OPTS() \

     { "reorder_queue_size", "set number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }

 const AVOption ff_rtsp_options[] = {

     { "initial_pause",  "do not start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },

     FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),

     { "rtsp_transport", "set RTSP transport protocols", OFFSET(lower_transport_mask), AV_OPT_TYPE_FLAGS, {.i64 = 0}, INT_MIN, INT_MAX, DEC|ENC, "rtsp_transport" }, \

     { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
     { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \
     { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
     { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },

     RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),
     { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },

     { "prefer_tcp", "try RTP via TCP first, if available", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_PREFER_TCP}, 0, 0, DEC|ENC, "rtsp_flags" },

     RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
     { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
     { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
     { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },

     { "stimeout", "set timeout (in microseconds) of socket TCP I/O operations", OFFSET(stimeout), AV_OPT_TYPE_INT, {.i64 = 0}, INT_MIN, INT_MAX, DEC },

     RTSP_REORDERING_OPTS(),

     { "user-agent", "override User-Agent header", OFFSET(user_agent), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, DEC },

     { NULL },
 };
 

 static const AVOption sdp_options[] = {
     RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),

     { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
     { "rtcp_to_source", "send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },
     RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),

     RTSP_REORDERING_OPTS(),

     { NULL },
 };
 
 static const AVOption rtp_options[] = {

     RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),

     RTSP_REORDERING_OPTS(),

     { NULL },
 };
 

 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 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);

     av_parse_time(start, buf, 1);

     if (*p == '-') {
         p++;
         get_word_sep(buf, sizeof(buf), "-", &p);

         av_parse_time(end, buf, 1);

     }
 }
 
 static int get_sockaddr(const char *buf, struct sockaddr_storage *sock)
 {

     struct addrinfo hints = { 0 }, *ai = NULL;

     hints.ai_flags = AI_NUMERICHOST;
     if (getaddrinfo(buf, NULL, &hints, &ai))
         return -1;
     memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));
     freeaddrinfo(ai);
     return 0;
 }
 

 #if CONFIG_RTPDEC

 static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
                              RTSPStream *rtsp_st, AVCodecContext *codec)
 {
     if (!handler)
         return;

     if (codec)
         codec->codec_id          = handler->codec_id;

     rtsp_st->dynamic_handler = handler;

     if (handler->alloc) {

         rtsp_st->dynamic_protocol_context = handler->alloc();

         if (!rtsp_st->dynamic_protocol_context)
             rtsp_st->dynamic_handler = NULL;
     }

 }
 

 /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */

 static int sdp_parse_rtpmap(AVFormatContext *s,

                             AVStream *st, RTSPStream *rtsp_st,

                             int payload_type, const char *p)

 {

     AVCodecContext *codec = st->codec;

     char buf[256];

     int i;
     AVCodec *c;

     const char *c_name;

     /* 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) {
         /* We are in a standard case
          * (from http://www.iana.org/assignments/rtp-parameters). */
         codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
     }
 

     if (codec->codec_id == AV_CODEC_ID_NONE) {

         RTPDynamicProtocolHandler *handler =
             ff_rtp_handler_find_by_name(buf, codec->codec_type);
         init_rtp_handler(handler, rtsp_st, codec);

         /* If no dynamic handler was found, check with the list of standard
          * allocated types, if such a stream for some reason happens to
          * use a private payload type. This isn't handled in rtpdec.c, since
          * the format name from the rtpmap line never is passed into rtpdec. */
         if (!rtsp_st->dynamic_handler)
             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;

             avpriv_set_pts_info(st, 32, 1, codec->sample_rate);

             get_word_sep(buf, sizeof(buf), "/", &p);
             i = atoi(buf);
             if (i > 0)
                 codec->channels = i;

         }

         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);

         if (i > 0)

             avpriv_set_pts_info(st, 32, 1, i);

         break;
     default:
         break;
     }

     if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init)
         rtsp_st->dynamic_handler->init(s, st->index,
                                        rtsp_st->dynamic_protocol_context);

     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;
 }
 

 typedef struct SDPParseState {
     /* SDP only */

     struct sockaddr_storage default_ip;

     int            default_ttl;
     int            skip_media;  ///< set if an unknown m= line occurs

     int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */
     struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */
     int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */
     struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */

     int seen_rtpmap;
     int seen_fmtp;
     char delayed_fmtp[2048];

 } SDPParseState;
 

 static void copy_default_source_addrs(struct RTSPSource **addrs, int count,
                                       struct RTSPSource ***dest, int *dest_count)
 {
     RTSPSource *rtsp_src, *rtsp_src2;
     int i;
     for (i = 0; i < count; i++) {
         rtsp_src = addrs[i];
         rtsp_src2 = av_malloc(sizeof(*rtsp_src2));
         if (!rtsp_src2)
             continue;
         memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));
         dynarray_add(dest, dest_count, rtsp_src2);
     }
 }
 

 static void parse_fmtp(AVFormatContext *s, RTSPState *rt,
                        int payload_type, const char *line)
 {
     int i;
 
     for (i = 0; i < rt->nb_rtsp_streams; i++) {
         RTSPStream *rtsp_st = rt->rtsp_streams[i];
         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, line);
         }
     }
 }
 

 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;

     AVStream *st;
     RTSPStream *rtsp_st;

     RTSPSource *rtsp_src;

     struct sockaddr_storage sdp_ip;

     int ttl;
 

     av_dlog(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") && strcmp(buf1, "IP6"))

             return;
         get_word_sep(buf1, sizeof(buf1), "/", &p);

         if (get_sockaddr(buf1, &sdp_ip))

             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 {

             rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];

             rtsp_st->sdp_ip = sdp_ip;
             rtsp_st->sdp_ttl = ttl;
         }
         break;

     case 's':

         av_dict_set(&s->metadata, "title", p, 0);

         break;
     case 'i':
         if (s->nb_streams == 0) {

             av_dict_set(&s->metadata, "comment", p, 0);

             break;
         }
         break;
     case 'm':
         /* new stream */

         s1->skip_media  = 0;
         s1->seen_fmtp   = 0;
         s1->seen_rtpmap = 0;

         codec_type = AVMEDIA_TYPE_UNKNOWN;

         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;

         }
         if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {

             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;
 

         copy_default_source_addrs(s1->default_include_source_addrs,
                                   s1->nb_default_include_source_addrs,
                                   &rtsp_st->include_source_addrs,
                                   &rtsp_st->nb_include_source_addrs);
         copy_default_source_addrs(s1->default_exclude_source_addrs,
                                   s1->nb_default_exclude_source_addrs,
                                   &rtsp_st->exclude_source_addrs,
                                   &rtsp_st->nb_exclude_source_addrs);
 

         get_word(buf1, sizeof(buf1), &p); /* port */
         rtsp_st->sdp_port = atoi(buf1);
 

         get_word(buf1, sizeof(buf1), &p); /* protocol */
         if (!strcmp(buf1, "udp"))
             rt->transport = RTSP_TRANSPORT_RAW;

         else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))
             rtsp_st->feedback = 1;

         /* 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 */

             if (rt->transport == RTSP_TRANSPORT_RAW) {
                 if (!rt->ts && CONFIG_RTPDEC)

                     rt->ts = avpriv_mpegts_parse_open(s);

             } else {
                 RTPDynamicProtocolHandler *handler;
                 handler = ff_rtp_handler_find_by_id(
                               rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
                 init_rtp_handler(handler, rtsp_st, NULL);
                 if (handler && handler->init)
                     handler->init(s, -1, rtsp_st->dynamic_protocol_context);
             }

         } else if (rt->server_type == RTSP_SERVER_WMS &&
                    codec_type == AVMEDIA_TYPE_DATA) {
             /* RTX stream, a stream that carries all the other actual
              * audio/video streams. Don't expose this to the callers. */

         } else {

             st = avformat_new_stream(s, NULL);

             if (!st)
                 return;

             st->id = rt->nb_rtsp_streams - 1;

             rtsp_st->stream_index = st->index;

             st->codec->codec_type = codec_type;

             if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {

                 RTPDynamicProtocolHandler *handler;

                 /* if standard payload type, we can find the codec right now */

                 ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);

                 if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
                     st->codec->sample_rate > 0)

                     avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);

                 /* Even static payload types may need a custom depacketizer */
                 handler = ff_rtp_handler_find_by_id(
                               rtsp_st->sdp_payload_type, st->codec->codec_type);
                 init_rtp_handler(handler, rtsp_st, st->codec);

                 if (handler && handler->init)
                     handler->init(s, st->index,
                                   rtsp_st->dynamic_protocol_context);

             }
         }

         /* 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 */

                 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];

                 /* 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);

             rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];

             if (rtsp_st->stream_index >= 0) {
                 st = s->streams[rtsp_st->stream_index];
                 sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
             }

             s1->seen_rtpmap = 1;
             if (s1->seen_fmtp) {
                 parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);
             }

         } else if (av_strstart(p, "fmtp:", &p) ||
                    av_strstart(p, "framesize:", &p)) {

             // let dynamic protocol handlers have a stab at the line.
             get_word(buf1, sizeof(buf1), &p);
             payload_type = atoi(buf1);

             if (s1->seen_rtpmap) {
                 parse_fmtp(s, rt, payload_type, buf);
             } else {
                 s1->seen_fmtp = 1;
                 av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));
             }

         } 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 (av_strstart(p, "SampleRate:integer;", &p) &&
                    s->nb_streams > 0) {
             st = s->streams[s->nb_streams - 1];
             st->codec->sample_rate = atoi(p);

         } else if (av_strstart(p, "crypto:", &p) && s->nb_streams > 0) {
             // RFC 4568
             rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
             get_word(buf1, sizeof(buf1), &p); // ignore tag
             get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);
             p += strspn(p, SPACE_CHARS);
             if (av_strstart(p, "inline:", &p))
                 get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);

         } else if (av_strstart(p, "source-filter:", &p)) {
             int exclude = 0;

             get_word(buf1, sizeof(buf1), &p);

             if (strcmp(buf1, "incl") && strcmp(buf1, "excl"))

                 return;

             exclude = !strcmp(buf1, "excl");

 
             get_word(buf1, sizeof(buf1), &p);
             if (strcmp(buf1, "IN") != 0)

                 return;

             get_word(buf1, sizeof(buf1), &p);

             if (strcmp(buf1, "IP4") && strcmp(buf1, "IP6") && strcmp(buf1, "*"))

                 return;

             // not checking that the destination address actually matches or is wildcard

             get_word(buf1, sizeof(buf1), &p);
 

             while (*p != '\0') {
                 rtsp_src = av_mallocz(sizeof(*rtsp_src));
                 if (!rtsp_src)
                     return;
                 get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);
                 if (exclude) {
                     if (s->nb_streams == 0) {
                         dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);
                     } else {
                         rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
                         dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);
                     }
                 } else {
                     if (s->nb_streams == 0) {
                         dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);
                     } else {
                         rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
                         dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);
                     }
                 }
             }

         } else {
             if (rt->server_type == RTSP_SERVER_WMS)
                 ff_wms_parse_sdp_a_line(s, p);
             if (s->nb_streams > 0) {

                 rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
 

                 if (rt->server_type == RTSP_SERVER_REAL)

                     ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);

 
                 if (rtsp_st->dynamic_handler &&
                     rtsp_st->dynamic_handler->parse_sdp_a_line)

                     rtsp_st->dynamic_handler->parse_sdp_a_line(s,

                         rtsp_st->stream_index,

                         rtsp_st->dynamic_protocol_context, buf);

             }

         }
         break;
     }
 }
 

 int ff_sdp_parse(AVFormatContext *s, const char *content)

 {

     RTSPState *rt = s->priv_data;

     const char *p;

     int letter, i;

     /* 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 = { { 0 } }, *s1 = &sdp_parse_state;

     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++;
     }

 
     for (i = 0; i < s1->nb_default_include_source_addrs; i++)
         av_free(s1->default_include_source_addrs[i]);
     av_freep(&s1->default_include_source_addrs);
     for (i = 0; i < s1->nb_default_exclude_source_addrs; i++)
         av_free(s1->default_exclude_source_addrs[i]);
     av_freep(&s1->default_exclude_source_addrs);
 

     rt->p = av_malloc_array(rt->nb_rtsp_streams + 1, sizeof(struct pollfd) * 2);

     if (!rt->p) return AVERROR(ENOMEM);

     return 0;
 }

 #endif /* CONFIG_RTPDEC */

 void ff_rtsp_undo_setup(AVFormatContext *s, int send_packets)

 {
     RTSPState *rt = s->priv_data;
     int i;
 
     for (i = 0; i < rt->nb_rtsp_streams; i++) {
         RTSPStream *rtsp_st = rt->rtsp_streams[i];
         if (!rtsp_st)
             continue;
         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;

                     if (CONFIG_RTSP_MUXER && rtpctx->pb && send_packets)
                         ff_rtsp_tcp_write_packet(s, rtsp_st);

                     avio_close_dyn_buf(rtpctx->pb, &ptr);

                     av_free(ptr);
                 } else {

                     avio_close(rtpctx->pb);

                 }

                 avformat_free_context(rtpctx);

             } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)
                 ff_rdt_parse_close(rtsp_st->transport_priv);

             else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC)

                 ff_rtp_parse_close(rtsp_st->transport_priv);

         }
         rtsp_st->transport_priv = NULL;
         if (rtsp_st->rtp_handle)

             ffurl_close(rtsp_st->rtp_handle);

         rtsp_st->rtp_handle = NULL;
     }
 }
 

 /* close and free RTSP streams */

 void ff_rtsp_close_streams(AVFormatContext *s)

 {

     RTSPState *rt = s->priv_data;

     int i, j;

     RTSPStream *rtsp_st;
 

     ff_rtsp_undo_setup(s, 0);

     for (i = 0; i < rt->nb_rtsp_streams; i++) {

         rtsp_st = rt->rtsp_streams[i];
         if (rtsp_st) {
             if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)

                 rtsp_st->dynamic_handler->free(

                     rtsp_st->dynamic_protocol_context);

             for (j = 0; j < rtsp_st->nb_include_source_addrs; j++)
                 av_free(rtsp_st->include_source_addrs[j]);
             av_freep(&rtsp_st->include_source_addrs);
             for (j = 0; j < rtsp_st->nb_exclude_source_addrs; j++)
                 av_free(rtsp_st->exclude_source_addrs[j]);
             av_freep(&rtsp_st->exclude_source_addrs);
 

             av_free(rtsp_st);

         }
     }
     av_free(rt->rtsp_streams);
     if (rt->asf_ctx) {

         avformat_close_input(&rt->asf_ctx);

     }

     if (rt->ts && CONFIG_RTPDEC)

         avpriv_mpegts_parse_close(rt->ts);

     av_free(rt->p);

     av_free(rt->recvbuf);

 }
 

 int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)

 {
     RTSPState *rt = s->priv_data;
     AVStream *st = NULL;

     int reordering_queue_size = rt->reordering_queue_size;
     if (reordering_queue_size < 0) {
         if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
             reordering_queue_size = 0;
         else
             reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
     }

 
     /* 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 && CONFIG_RTSP_MUXER) {

         int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv,
                                         s, st, rtsp_st->rtp_handle,

                                         RTSP_TCP_MAX_PACKET_SIZE,
                                         rtsp_st->stream_index);

         /* Ownership of rtp_handle is passed to the rtp mux context */

         rtsp_st->rtp_handle = NULL;

         if (ret < 0)
             return ret;

         st->time_base = ((AVFormatContext*)rtsp_st->transport_priv)->streams[0]->time_base;

     } else if (rt->transport == RTSP_TRANSPORT_RAW) {
         return 0; // Don't need to open any parser here

     } else if (rt->transport == RTSP_TRANSPORT_RDT && CONFIG_RTPDEC)

         rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,
                                             rtsp_st->dynamic_protocol_context,
                                             rtsp_st->dynamic_handler);

     else if (CONFIG_RTPDEC)

         rtsp_st->transport_priv = ff_rtp_parse_open(s, st,

                                          rtsp_st->sdp_payload_type,

                                          reordering_queue_size);

 
     if (!rtsp_st->transport_priv) {
          return AVERROR(ENOMEM);

     } else if (rt->transport == RTSP_TRANSPORT_RTP && CONFIG_RTPDEC) {

         if (rtsp_st->dynamic_handler) {

             ff_rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
                                               rtsp_st->dynamic_protocol_context,
                                               rtsp_st->dynamic_handler);

         }

         if (rtsp_st->crypto_suite[0])
             ff_rtp_parse_set_crypto(rtsp_st->transport_priv,
                                     rtsp_st->crypto_suite,
                                     rtsp_st->crypto_params);

     }
 
     return 0;
 }
 

 #if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER

 static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
 {

     const char *q;
     char *p;

     int v;
 

     q = *pp;
     q += strspn(q, SPACE_CHARS);
     v = strtol(q, &p, 10);

     if (*p == '-') {
         p++;
         *min_ptr = v;

         v = strtol(p, &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 (!av_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 (!av_strcasecmp (transport_protocol, "x-pn-tng") ||
                    !av_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;

         } else if (!av_strcasecmp(transport_protocol, "raw")) {
             get_word_sep(profile, sizeof(profile), "/;,", &p);
             lower_transport[0] = '\0';
             /* raw/raw/<protocol> */
             if (*p == '/') {
                 get_word_sep(lower_transport, sizeof(lower_transport),
                              ";,", &p);
             }
             th->transport = RTSP_TRANSPORT_RAW;

         }

         if (!av_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 == '=') {

                     char *end;

                     p++;

                     th->ttl = strtol(p, &end, 10);
                     p = end;

                 }
             } else if (!strcmp(parameter, "destination")) {
                 if (*p == '=') {
                     p++;
                     get_word_sep(buf, sizeof(buf), ";,", &p);

                     get_sockaddr(buf, &th->destination);

                 }

             } else if (!strcmp(parameter, "source")) {
                 if (*p == '=') {
                     p++;
                     get_word_sep(buf, sizeof(buf), ";,", &p);
                     av_strlcpy(th->source, buf, sizeof(th->source));
                 }

             } else if (!strcmp(parameter, "mode")) {
                 if (*p == '=') {
                     p++;
                     get_word_sep(buf, sizeof(buf), ";, ", &p);
                     if (!strcmp(buf, "record") ||
                         !strcmp(buf, "receive"))
                         th->mode_record = 1;
                 }

             }

             while (*p != ';' && *p != '\0' && *p != ',')
                 p++;
             if (*p == ';')
                 p++;
         }
         if (*p == ',')
             p++;
 
         reply->nb_transports++;
     }
 }
 

 static void handle_rtp_info(RTSPState *rt, const char *url,
                             uint32_t seq, uint32_t rtptime)
 {
     int i;
     if (!rtptime || !url[0])
         return;
     if (rt->transport != RTSP_TRANSPORT_RTP)
         return;
     for (i = 0; i < rt->nb_rtsp_streams; i++) {
         RTSPStream *rtsp_st = rt->rtsp_streams[i];
         RTPDemuxContext *rtpctx = rtsp_st->transport_priv;
         if (!rtpctx)
             continue;
         if (!strcmp(rtsp_st->control_url, url)) {
             rtpctx->base_timestamp = rtptime;
             break;
         }
     }
 }
 
 static void rtsp_parse_rtp_info(RTSPState *rt, const char *p)
 {
     int read = 0;
     char key[20], value[1024], url[1024] = "";
     uint32_t seq = 0, rtptime = 0;
 
     for (;;) {
         p += strspn(p, SPACE_CHARS);
         if (!*p)
             break;
         get_word_sep(key, sizeof(key), "=", &p);
         if (*p != '=')
             break;
         p++;
         get_word_sep(value, sizeof(value), ";, ", &p);
         read++;
         if (!strcmp(key, "url"))
             av_strlcpy(url, value, sizeof(url));
         else if (!strcmp(key, "seq"))

             seq = strtoul(value, NULL, 10);

         else if (!strcmp(key, "rtptime"))

             rtptime = strtoul(value, NULL, 10);

         if (*p == ',') {
             handle_rtp_info(rt, url, seq, rtptime);
             url[0] = '\0';
             seq = rtptime = 0;
             read = 0;
         }
         if (*p)
             p++;
     }
     if (read > 0)
         handle_rtp_info(rt, url, seq, rtptime);
 }
 

 void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,

                         RTSPState *rt, const char *method)

 {
     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) && rt) {

         p += strspn(p, SPACE_CHARS);

         ff_http_auth_handle_header(&rt->auth_state, "WWW-Authenticate", p);
     } else if (av_stristart(p, "Authentication-Info:", &p) && rt) {

         p += strspn(p, SPACE_CHARS);

         ff_http_auth_handle_header(&rt->auth_state, "Authentication-Info", p);

     } else if (av_stristart(p, "Content-Base:", &p) && rt) {

         p += strspn(p, SPACE_CHARS);

         if (method && !strcmp(method, "DESCRIBE"))
             av_strlcpy(rt->control_uri, p , sizeof(rt->control_uri));

     } else if (av_stristart(p, "RTP-Info:", &p) && rt) {
         p += strspn(p, SPACE_CHARS);
         if (method && !strcmp(method, "PLAY"))
             rtsp_parse_rtp_info(rt, p);

     } else if (av_stristart(p, "Public:", &p) && rt) {
         if (strstr(p, "GET_PARAMETER") &&
             method && !strcmp(method, "OPTIONS"))
             rt->get_parameter_supported = 1;

     } else if (av_stristart(p, "x-Accept-Dynamic-Rate:", &p) && rt) {
         p += strspn(p, SPACE_CHARS);
         rt->accept_dynamic_rate = atoi(p);

     } else if (av_stristart(p, "Content-Type:", &p)) {
         p += strspn(p, SPACE_CHARS);
         av_strlcpy(reply->content_type, p, sizeof(reply->content_type));

     }
 }
 

 /* 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 = ffurl_read_complete(rt->rtsp_hd, buf, 3);

     if (ret != 3)
         return;

     len = AV_RB16(buf + 1);

     av_dlog(s, "skipping RTP packet len=%d\n", len);

     /* skip payload */
     while (len > 0) {
         len1 = len;
         if (len1 > sizeof(buf))
             len1 = sizeof(buf);

         ret = ffurl_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, const char *method)

 {
     RTSPState *rt = s->priv_data;
     char buf[4096], buf1[1024], *q;
     unsigned char ch;
     const char *p;

     int ret, content_length, line_count = 0, request = 0;

     unsigned char *content = NULL;
 

 start:
     line_count = 0;
     request = 0;
     content = NULL;

     memset(reply, 0, sizeof(*reply));

 
     /* parse reply (XXX: use buffers) */
     rt->last_reply[0] = '\0';

     for (;;) {

         q = buf;

         for (;;) {

             ret = ffurl_read_complete(rt->rtsp_hd, &ch, 1);

             av_dlog(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);

             if (ret != 1)

                 return AVERROR_EOF;

             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';

         av_dlog(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);

             if (!strncmp(buf1, "RTSP/", 5)) {
                 get_word(buf1, sizeof(buf1), &p);
                 reply->status_code = atoi(buf1);
                 av_strlcpy(reply->reason, p, sizeof(reply->reason));
             } else {
                 av_strlcpy(reply->reason, buf1, sizeof(reply->reason)); // method
                 get_word(buf1, sizeof(buf1), &p); // object
                 request = 1;
             }

         } else {

             ff_rtsp_parse_line(reply, p, rt, method);

             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' && !request)

         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);

         ffurl_read_complete(rt->rtsp_hd, content, content_length);

         content[content_length] = '\0';
     }
     if (content_ptr)
         *content_ptr = content;

     else
         av_free(content);

     if (request) {
         char buf[1024];
         char base64buf[AV_BASE64_SIZE(sizeof(buf))];
         const char* ptr = buf;
 
         if (!strcmp(reply->reason, "OPTIONS")) {
             snprintf(buf, sizeof(buf), "RTSP/1.0 200 OK\r\n");
             if (reply->seq)
                 av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", reply->seq);
             if (reply->session_id[0])
                 av_strlcatf(buf, sizeof(buf), "Session: %s\r\n",
                                               reply->session_id);
         } else {
             snprintf(buf, sizeof(buf), "RTSP/1.0 501 Not Implemented\r\n");
         }
         av_strlcat(buf, "\r\n", sizeof(buf));
 
         if (rt->control_transport == RTSP_MODE_TUNNEL) {
             av_base64_encode(base64buf, sizeof(base64buf), buf, strlen(buf));
             ptr = base64buf;
         }
         ffurl_write(rt->rtsp_hd_out, ptr, strlen(ptr));
 
         rt->last_cmd_time = av_gettime();
         /* Even if the request from the server had data, it is not the data
          * that the caller wants or expects. The memory could also be leaked
          * if the actual following reply has content data. */
         if (content_ptr)
             av_freep(content_ptr);
         /* If method is set, this is called from ff_rtsp_send_cmd,
          * where a reply to exactly this request is awaited. For

          * callers from within packet receiving, we just want to

          * return to the caller and go back to receiving packets. */
         if (method)
             goto start;
         return 0;
     }
 

     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;

 }
 

 /**
  * Send a command to the RTSP server without waiting for the reply.
  *
  * @param s RTSP (de)muxer context
  * @param method the method for the request
  * @param url the target url for the request
  * @param headers extra header lines to include in the request
  * @param send_content if non-null, the data to send as request body content
  * @param send_content_length the length of the send_content data, or 0 if
  *                            send_content is null
  *
  * @return zero if success, nonzero otherwise
  */

 static int 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);

     av_strlcatf(buf, sizeof(buf), "User-Agent: %s\r\n",  rt->user_agent);

     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;
     }
 

     av_dlog(s, "Sending:\n%s--\n", buf);

     ffurl_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;
         }

         ffurl_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 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, attempts = 0;

 
 retry:
     cur_auth_type = rt->auth_state.auth_type;

     if ((ret = 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, method) ) < 0)

         return ret;

     attempts++;

     if (reply->status_code == 401 &&
         (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
         rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)

         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;

 }
 

 int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,

                               int lower_transport, const char *real_challenge)

 {
     RTSPState *rt = s->priv_data;

     int rtx = 0, j, i, err, interleave = 0, port_off;

     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 if (rt->transport == RTSP_TRANSPORT_RAW)
         trans_pref = "RAW/RAW";

     else
         trans_pref = "RTP/AVP";

     /* default timeout: 1 minute */
     rt->timeout = 60;
 

     /* Choose a random starting offset within the first half of the
      * port range, to allow for a number of ports to try even if the offset
      * happens to be at the end of the random range. */
     port_off = av_get_random_seed() % ((rt->rtp_port_max - rt->rtp_port_min)/2);

     /* even random offset */
     port_off -= port_off & 0x01;

     for (j = rt->rtp_port_min + port_off, 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 */

             while (j <= rt->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 (!ffurl_open(&rtsp_st->rtp_handle, buf, AVIO_FLAG_READ_WRITE,
                                &s->interrupt_callback, NULL))
                     goto rtp_opened;

             }

             av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
             err = AVERROR(EIO);
             goto fail;

 
         rtp_opened:

             port = ff_rtp_get_local_rtp_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 &&

                 (rtsp_st->stream_index < 0 ||
                  s->streams[rtsp_st->stream_index]->codec->codec_type ==
                     AVMEDIA_TYPE_DATA))

                 continue;

             snprintf(transport, sizeof(transport) - 1,

                      "%s/TCP;", trans_pref);

             if (rt->transport != RTSP_TRANSPORT_RDT)

                 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=record", 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 (rt->accept_dynamic_rate)
             av_strlcat(cmd, "x-Dynamic-Rate: 0\r\n", sizeof(cmd));

         if (i == 0 && rt->server_type == RTSP_SERVER_REAL && CONFIG_RTPDEC) {

             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;

         }
 

         /* Fail if the server responded with another lower transport mode
          * than what we requested. */
         if (reply->transports[0].lower_transport != lower_transport) {
             av_log(s, AV_LOG_ERROR, "Nonmatching transport in server reply\n");
             err = AVERROR_INVALIDDATA;
             goto fail;

         }
 

         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], options[30] = "";

             const char *peer = host;

             if (rt->rtsp_flags & RTSP_FLAG_FILTER_SRC)

                 av_strlcpy(options, "?connect=1", sizeof(options));

             /* Use source address if specified */

             if (reply->transports[0].source[0])
                 peer = reply->transports[0].source;
             ff_url_join(url, sizeof(url), "rtp", NULL, peer,
                         reply->transports[0].server_port_min, "%s", options);

             if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&

                 ff_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 &&
                 CONFIG_RTPDEC)

                 ff_rtp_send_punch_packets(rtsp_st->rtp_handle);

             break;

         }
         case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {

             char url[1024], namebuf[50], optbuf[20] = "";

             struct sockaddr_storage addr;

             int port, ttl;
 

             if (reply->transports[0].destination.ss_family) {
                 addr      = reply->transports[0].destination;

                 port      = reply->transports[0].port_min;
                 ttl       = reply->transports[0].ttl;
             } else {

                 addr      = rtsp_st->sdp_ip;

                 port      = rtsp_st->sdp_port;
                 ttl       = rtsp_st->sdp_ttl;
             }

             if (ttl > 0)
                 snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);

             getnameinfo((struct sockaddr*) &addr, sizeof(addr),
                         namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
             ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,

                         port, "%s", optbuf);

             if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,

                            &s->interrupt_callback, NULL) < 0) {

                 err = AVERROR_INVALIDDATA;
                 goto fail;

             }
             break;
         }

         }

         if ((err = ff_rtsp_open_transport_ctx(s, rtsp_st)))

             goto fail;

     }
 

     if (rt->nb_rtsp_streams && reply->timeout > 0)

         rt->timeout = reply->timeout;
 

     if (rt->server_type == RTSP_SERVER_REAL)

         rt->need_subscription = 1;
 

     return 0;
 
 fail:

     ff_rtsp_undo_setup(s, 0);

     return err;
 }
 

 void ff_rtsp_close_connections(AVFormatContext *s)
 {
     RTSPState *rt = s->priv_data;

     if (rt->rtsp_hd_out != rt->rtsp_hd) ffurl_close(rt->rtsp_hd_out);
     ffurl_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];

     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 (rt->rtp_port_max < rt->rtp_port_min) {
         av_log(s, AV_LOG_ERROR, "Invalid UDP port range, max port %d less "
                                 "than min port %d\n", rt->rtp_port_max,
                                                       rt->rtp_port_min);
         return AVERROR(EINVAL);
     }
 

     if (!ff_network_init())
         return AVERROR(EIO);

     if (s->max_delay < 0) /* Not set by the caller */

         s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;

     rt->control_transport = RTSP_MODE_PLAIN;

     if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
         rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
         rt->control_transport = RTSP_MODE_TUNNEL;
     }
     /* Only pass through valid flags from here */
     rt->lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
 
 redirect:
     lower_transport_mask = rt->lower_transport_mask;

     /* 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;
 

     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 (ffurl_alloc(&rt->rtsp_hd, httpname, AVIO_FLAG_READ,
                         &s->interrupt_callback) < 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);

         av_opt_set(rt->rtsp_hd->priv_data, "headers", headers, 0);

 
         /* complete the connection */

         if (ffurl_connect(rt->rtsp_hd, NULL)) {

             err = AVERROR(EIO);
             goto fail;
         }
 
         /* POST requests */

         if (ffurl_alloc(&rt->rtsp_hd_out, httpname, AVIO_FLAG_WRITE,
                         &s->interrupt_callback) < 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);

         av_opt_set(rt->rtsp_hd_out->priv_data, "headers", headers, 0);

         av_opt_set(rt->rtsp_hd_out->priv_data, "chunked_post", "0", 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 (ffurl_connect(rt->rtsp_hd_out, NULL)) {

             err = AVERROR(EIO);
             goto fail;
         }

     } else {

         /* open the tcp connection */

         ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port,
                     "?timeout=%d", rt->stimeout);

         if (ffurl_open(&rt->rtsp_hd, tcpname, AVIO_FLAG_READ_WRITE,

                        &s->interrupt_callback, NULL) < 0) {

             err = AVERROR(EIO);
             goto fail;
         }

         rt->rtsp_hd_out = rt->rtsp_hd;

     }

     rt->seq = 0;
 

     tcp_fd = ffurl_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.

                         *   ClientChallenge is a 16-byte ID in hex
                         *   CompanyID is a 16-byte ID in base64

                         */
                        "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 (!av_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 && CONFIG_RTSP_DEMUXER)

         err = ff_rtsp_setup_input_streams(s, reply);

     else if (CONFIG_RTSP_MUXER)

         err = ff_rtsp_setup_output_streams(s, host);

     if (err)

         goto fail;
 

     do {

         int lower_transport = ff_log2_tab[lower_transport_mask &
                                   ~(lower_transport_mask - 1)];

         if ((lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_TCP))
                 && (rt->rtsp_flags & RTSP_FLAG_PREFER_TCP))
             lower_transport = RTSP_LOWER_TRANSPORT_TCP;
 

         err = ff_rtsp_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 = AVERROR(EPROTONOSUPPORT);

             goto fail;
         }
     } while (err);

     rt->lower_transport_mask = lower_transport_mask;
     av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));

     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 /* CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER */

 #if CONFIG_RTPDEC

 static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,

                            uint8_t *buf, int buf_size, int64_t wait_end)

 {
     RTSPState *rt = s->priv_data;
     RTSPStream *rtsp_st;

     int n, i, ret, tcp_fd, timeout_cnt = 0;
     int max_p = 0;
     struct pollfd *p = rt->p;

     int *fds = NULL, fdsnum, fdsidx;

     for (;;) {

         if (ff_check_interrupt(&s->interrupt_callback))

             return AVERROR_EXIT;

         if (wait_end && wait_end - av_gettime() < 0)
             return AVERROR(EAGAIN);

         max_p = 0;

         if (rt->rtsp_hd) {

             tcp_fd = ffurl_get_file_handle(rt->rtsp_hd);

             p[max_p].fd = tcp_fd;
             p[max_p++].events = POLLIN;

         } else {
             tcp_fd = -1;
         }

         for (i = 0; i < rt->nb_rtsp_streams; i++) {

             rtsp_st = rt->rtsp_streams[i];
             if (rtsp_st->rtp_handle) {

                 if (ret = ffurl_get_multi_file_handle(rtsp_st->rtp_handle,
                                                       &fds, &fdsnum)) {
                     av_log(s, AV_LOG_ERROR, "Unable to recover rtp ports\n");
                     return ret;
                 }
                 if (fdsnum != 2) {
                     av_log(s, AV_LOG_ERROR,
                            "Number of fds %d not supported\n", fdsnum);
                     return AVERROR_INVALIDDATA;
                 }
                 for (fdsidx = 0; fdsidx < fdsnum; fdsidx++) {
                     p[max_p].fd       = fds[fdsidx];
                     p[max_p++].events = POLLIN;
                 }
                 av_free(fds);

             }
         }

         n = poll(p, max_p, POLL_TIMEOUT_MS);

         if (n > 0) {

             int j = 1 - (tcp_fd == -1);

             timeout_cnt = 0;

             for (i = 0; i < rt->nb_rtsp_streams; i++) {

                 rtsp_st = rt->rtsp_streams[i];
                 if (rtsp_st->rtp_handle) {

                     if (p[j].revents & POLLIN || p[j+1].revents & POLLIN) {

                         ret = ffurl_read(rtsp_st->rtp_handle, buf, buf_size);

                         if (ret > 0) {
                             *prtsp_st = rtsp_st;
                             return ret;
                         }
                     }

                     j+=2;

                 }
             }

 #if CONFIG_RTSP_DEMUXER

             if (tcp_fd != -1 && p[0].revents & POLLIN) {

                 if (rt->rtsp_flags & RTSP_FLAG_LISTEN) {
                     if (rt->state == RTSP_STATE_STREAMING) {
                         if (!ff_rtsp_parse_streaming_commands(s))
                             return AVERROR_EOF;
                         else
                             av_log(s, AV_LOG_WARNING,
                                    "Unable to answer to TEARDOWN\n");
                     } else
                         return 0;
                 } else {
                     RTSPMessageHeader reply;
                     ret = ff_rtsp_read_reply(s, &reply, NULL, 0, NULL);
                     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 AVERROR(ETIMEDOUT);

         } else if (n < 0 && errno != EINTR)
             return AVERROR(errno);

     }
 }
 

 static int pick_stream(AVFormatContext *s, RTSPStream **rtsp_st,
                        const uint8_t *buf, int len)
 {
     RTSPState *rt = s->priv_data;
     int i;
     if (len < 0)
         return len;
     if (rt->nb_rtsp_streams == 1) {
         *rtsp_st = rt->rtsp_streams[0];
         return len;
     }
     if (len >= 8 && rt->transport == RTSP_TRANSPORT_RTP) {
         if (RTP_PT_IS_RTCP(rt->recvbuf[1])) {
             int no_ssrc = 0;
             for (i = 0; i < rt->nb_rtsp_streams; i++) {
                 RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;
                 if (!rtpctx)
                     continue;
                 if (rtpctx->ssrc == AV_RB32(&buf[4])) {
                     *rtsp_st = rt->rtsp_streams[i];
                     return len;
                 }
                 if (!rtpctx->ssrc)
                     no_ssrc = 1;
             }
             if (no_ssrc) {
                 av_log(s, AV_LOG_WARNING,
                        "Unable to pick stream for packet - SSRC not known for "
                        "all streams\n");
                 return AVERROR(EAGAIN);
             }
         } else {
             for (i = 0; i < rt->nb_rtsp_streams; i++) {
                 if ((buf[1] & 0x7f) == rt->rtsp_streams[i]->sdp_payload_type) {
                     *rtsp_st = rt->rtsp_streams[i];
                     return len;
                 }
             }
         }
     }
     av_log(s, AV_LOG_WARNING, "Unable to pick stream for packet\n");
     return AVERROR(EAGAIN);
 }
 

 int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)

 {
     RTSPState *rt = s->priv_data;
     int ret, len;

     RTSPStream *rtsp_st, *first_queue_st = NULL;
     int64_t wait_end = 0;

     if (rt->nb_byes == rt->nb_rtsp_streams)
         return AVERROR_EOF;
 

     /* 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 if (rt->transport == RTSP_TRANSPORT_RTP) {

             ret = ff_rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);

         } else if (rt->ts && CONFIG_RTPDEC) {

             ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf + rt->recvbuf_pos, rt->recvbuf_len - rt->recvbuf_pos);

             if (ret >= 0) {
                 rt->recvbuf_pos += ret;
                 ret = rt->recvbuf_pos < rt->recvbuf_len;
             }

         } else

             ret = -1;

         if (ret == 0) {
             rt->cur_transport_priv = NULL;
             return 0;
         } else if (ret == 1) {
             return 0;

         } else

             rt->cur_transport_priv = NULL;
     }
 

 redo:

     if (rt->transport == RTSP_TRANSPORT_RTP) {
         int i;
         int64_t first_queue_time = 0;
         for (i = 0; i < rt->nb_rtsp_streams; i++) {
             RTPDemuxContext *rtpctx = rt->rtsp_streams[i]->transport_priv;

             int64_t queue_time;
             if (!rtpctx)
                 continue;
             queue_time = ff_rtp_queued_packet_time(rtpctx);

             if (queue_time && (queue_time - first_queue_time < 0 ||
                                !first_queue_time)) {
                 first_queue_time = queue_time;
                 first_queue_st   = rt->rtsp_streams[i];
             }
         }

         if (first_queue_time) {

             wait_end = first_queue_time + s->max_delay;

         } else {
             wait_end = 0;
             first_queue_st = NULL;
         }

     }
 

     /* read next RTP packet */

     if (!rt->recvbuf) {
         rt->recvbuf = av_malloc(RECVBUF_SIZE);
         if (!rt->recvbuf)
             return AVERROR(ENOMEM);
     }
 

     switch(rt->lower_transport) {
     default:

 #if CONFIG_RTSP_DEMUXER

     case RTSP_LOWER_TRANSPORT_TCP:

         len = ff_rtsp_tcp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE);

         break;

 #endif

     case RTSP_LOWER_TRANSPORT_UDP:
     case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:

         len = udp_read_packet(s, &rtsp_st, rt->recvbuf, RECVBUF_SIZE, wait_end);

         if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)

             ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, rtsp_st->rtp_handle, NULL, len);
         break;
     case RTSP_LOWER_TRANSPORT_CUSTOM:

         if (first_queue_st && rt->transport == RTSP_TRANSPORT_RTP &&
             wait_end && wait_end < av_gettime())
             len = AVERROR(EAGAIN);
         else
             len = ffio_read_partial(s->pb, rt->recvbuf, RECVBUF_SIZE);

         len = pick_stream(s, &rtsp_st, rt->recvbuf, len);
         if (len > 0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
             ff_rtp_check_and_send_back_rr(rtsp_st->transport_priv, NULL, s->pb, len);

         break;
     }

     if (len == AVERROR(EAGAIN) && first_queue_st &&
         rt->transport == RTSP_TRANSPORT_RTP) {
         rtsp_st = first_queue_st;

         ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, NULL, 0);

         goto end;
     }

     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, &rt->recvbuf, len);

     } else if (rt->transport == RTSP_TRANSPORT_RTP) {

         ret = ff_rtp_parse_packet(rtsp_st->transport_priv, pkt, &rt->recvbuf, len);

         if (rtsp_st->feedback) {
             AVIOContext *pb = NULL;
             if (rt->lower_transport == RTSP_LOWER_TRANSPORT_CUSTOM)
                 pb = s->pb;
             ff_rtp_send_rtcp_feedback(rtsp_st->transport_priv, rtsp_st->rtp_handle, pb);
         }

         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;

                 AVStream *st = NULL;
                 if (rtsp_st->stream_index >= 0)
                     st = s->streams[rtsp_st->stream_index];

                 for (i = 0; i < rt->nb_rtsp_streams; i++) {

                     RTPDemuxContext *rtpctx2 = rt->rtsp_streams[i]->transport_priv;

                     AVStream *st2 = NULL;
                     if (rt->rtsp_streams[i]->stream_index >= 0)
                         st2 = s->streams[rt->rtsp_streams[i]->stream_index];
                     if (rtpctx2 && st && st2 &&
                         rtpctx2->first_rtcp_ntp_time == AV_NOPTS_VALUE) {

                         rtpctx2->first_rtcp_ntp_time = rtpctx->first_rtcp_ntp_time;

                         rtpctx2->rtcp_ts_offset = av_rescale_q(
                             rtpctx->rtcp_ts_offset, st->time_base,
                             st2->time_base);
                     }

                 }

                 // Make real NTP start time available in AVFormatContext
                 if (s->start_time_realtime == AV_NOPTS_VALUE) {
                     s->start_time_realtime = av_rescale (rtpctx->first_rtcp_ntp_time - (NTP_OFFSET << 32), 1000000, 1LL << 32);
                     if (rtpctx->st) {
                         s->start_time_realtime -=
                             av_rescale (rtpctx->rtcp_ts_offset,
                                         (uint64_t) rtpctx->st->time_base.num * 1000000,
                                                    rtpctx->st->time_base.den);
                     }
                 }

             }

             if (ret == -RTCP_BYE) {
                 rt->nb_byes++;
 
                 av_log(s, AV_LOG_DEBUG, "Received BYE for stream %d (%d/%d)\n",
                        rtsp_st->stream_index, rt->nb_byes, rt->nb_rtsp_streams);
 
                 if (rt->nb_byes == rt->nb_rtsp_streams)
                     return AVERROR_EOF;
             }

         }

     } else if (rt->ts && CONFIG_RTPDEC) {

         ret = avpriv_mpegts_parse_packet(rt->ts, pkt, rt->recvbuf, len);

         if (ret >= 0) {
             if (ret < len) {
                 rt->recvbuf_len = len;
                 rt->recvbuf_pos = ret;
                 rt->cur_transport_priv = rt->ts;
                 return 1;
             } else {
                 ret = 0;
             }
         }

     } else {
         return AVERROR_INVALIDDATA;

     }

 end:

     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;
 }

 #endif /* CONFIG_RTPDEC */

 #if CONFIG_SDP_DEMUXER

 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 IP" */

     while (p < p_end && *p != '\0') {

         if (p + sizeof("c=IN IP") - 1 < p_end &&
             av_strstart(p, "c=IN IP", NULL))

             return AVPROBE_SCORE_EXTENSION;

         while (p < p_end - 1 && *p != '\n') p++;

         if (++p >= p_end)

             break;
         if (*p == '\r')
             p++;
     }

     return 0;
 }
 

 static void append_source_addrs(char *buf, int size, const char *name,
                                 int count, struct RTSPSource **addrs)
 {
     int i;
     if (!count)
         return;
     av_strlcatf(buf, size, "&%s=%s", name, addrs[0]->addr);
     for (i = 1; i < count; i++)
         av_strlcatf(buf, size, ",%s", addrs[i]->addr);
 }
 

 static int sdp_read_header(AVFormatContext *s)

 {

     RTSPState *rt = s->priv_data;

     RTSPStream *rtsp_st;
     int size, i, err;
     char *content;
     char url[1024];
 

     if (!ff_network_init())
         return AVERROR(EIO);
 

     if (s->max_delay < 0) /* Not set by the caller */

         s->max_delay = DEFAULT_REORDERING_DELAY;

     if (rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)
         rt->lower_transport = RTSP_LOWER_TRANSPORT_CUSTOM;

     /* read the whole sdp file */
     /* XXX: better loading */
     content = av_malloc(SDP_MAX_SIZE);

     size = avio_read(s->pb, content, SDP_MAX_SIZE - 1);

     if (size <= 0) {
         av_free(content);
         return AVERROR_INVALIDDATA;
     }
     content[size] ='\0';
 

     err = ff_sdp_parse(s, content);

     av_free(content);

     if (err) goto fail;

 
     /* open each RTP stream */

     for (i = 0; i < rt->nb_rtsp_streams; i++) {

         char namebuf[50];

         rtsp_st = rt->rtsp_streams[i];

         if (!(rt->rtsp_flags & RTSP_FLAG_CUSTOM_IO)) {
             getnameinfo((struct sockaddr*) &rtsp_st->sdp_ip, sizeof(rtsp_st->sdp_ip),
                         namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
             ff_url_join(url, sizeof(url), "rtp", NULL,
                         namebuf, rtsp_st->sdp_port,

                         "?localport=%d&ttl=%d&connect=%d&write_to_source=%d",
                         rtsp_st->sdp_port, rtsp_st->sdp_ttl,
                         rt->rtsp_flags & RTSP_FLAG_FILTER_SRC ? 1 : 0,
                         rt->rtsp_flags & RTSP_FLAG_RTCP_TO_SOURCE ? 1 : 0);

 
             append_source_addrs(url, sizeof(url), "sources",
                                 rtsp_st->nb_include_source_addrs,
                                 rtsp_st->include_source_addrs);
             append_source_addrs(url, sizeof(url), "block",
                                 rtsp_st->nb_exclude_source_addrs,
                                 rtsp_st->exclude_source_addrs);

             if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
                            &s->interrupt_callback, NULL) < 0) {
                 err = AVERROR_INVALIDDATA;
                 goto fail;
             }

         }

         if ((err = ff_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;
 }
 

 static const AVClass sdp_demuxer_class = {
     .class_name     = "SDP demuxer",
     .item_name      = av_default_item_name,
     .option         = sdp_options,
     .version        = LIBAVUTIL_VERSION_INT,
 };
 

 AVInputFormat ff_sdp_demuxer = {

     .name           = "sdp",
     .long_name      = NULL_IF_CONFIG_SMALL("SDP"),
     .priv_data_size = sizeof(RTSPState),
     .read_probe     = sdp_probe,
     .read_header    = sdp_read_header,
     .read_packet    = ff_rtsp_fetch_packet,
     .read_close     = sdp_read_close,

     .priv_class     = &sdp_demuxer_class,

 };

 #endif /* CONFIG_SDP_DEMUXER */

 #if CONFIG_RTP_DEMUXER

 static int rtp_probe(AVProbeData *p)
 {
     if (av_strstart(p->filename, "rtp:", NULL))
         return AVPROBE_SCORE_MAX;
     return 0;
 }
 

 static int rtp_read_header(AVFormatContext *s)

 {

     uint8_t recvbuf[RTP_MAX_PACKET_LENGTH];

     char host[500], sdp[500];
     int ret, port;
     URLContext* in = NULL;
     int payload_type;

     AVCodecContext codec = { 0 };

     struct sockaddr_storage addr;

     AVIOContext pb;

     socklen_t addrlen = sizeof(addr);

     RTSPState *rt = s->priv_data;

 
     if (!ff_network_init())
         return AVERROR(EIO);
 

     ret = ffurl_open(&in, s->filename, AVIO_FLAG_READ,

                      &s->interrupt_callback, NULL);

     if (ret)
         goto fail;
 
     while (1) {

         ret = ffurl_read(in, recvbuf, sizeof(recvbuf));

         if (ret == AVERROR(EAGAIN))
             continue;
         if (ret < 0)
             goto fail;
         if (ret < 12) {
             av_log(s, AV_LOG_WARNING, "Received too short packet\n");
             continue;
         }
 
         if ((recvbuf[0] & 0xc0) != 0x80) {
             av_log(s, AV_LOG_WARNING, "Unsupported RTP version packet "
                                       "received\n");
             continue;
         }
 

         if (RTP_PT_IS_RTCP(recvbuf[1]))

             continue;
 

         payload_type = recvbuf[1] & 0x7f;
         break;
     }

     getsockname(ffurl_get_file_handle(in), (struct sockaddr*) &addr, &addrlen);

     ffurl_close(in);

     in = NULL;
 
     if (ff_rtp_get_codec_info(&codec, payload_type)) {
         av_log(s, AV_LOG_ERROR, "Unable to receive RTP payload type %d "
                                 "without an SDP file describing it\n",
                                  payload_type);
         goto fail;
     }
     if (codec.codec_type != AVMEDIA_TYPE_DATA) {
         av_log(s, AV_LOG_WARNING, "Guessing on RTP content - if not received "
                                   "properly you need an SDP file "
                                   "describing it\n");
     }
 
     av_url_split(NULL, 0, NULL, 0, host, sizeof(host), &port,
                  NULL, 0, s->filename);
 
     snprintf(sdp, sizeof(sdp),
              "v=0\r\nc=IN IP%d %s\r\nm=%s %d RTP/AVP %d\r\n",
              addr.ss_family == AF_INET ? 4 : 6, host,
              codec.codec_type == AVMEDIA_TYPE_DATA  ? "application" :
              codec.codec_type == AVMEDIA_TYPE_VIDEO ? "video" : "audio",
              port, payload_type);
     av_log(s, AV_LOG_VERBOSE, "SDP:\n%s\n", sdp);
 

     ffio_init_context(&pb, sdp, strlen(sdp), 0, NULL, NULL, NULL, NULL);

     s->pb = &pb;
 
     /* sdp_read_header initializes this again */
     ff_network_close();
 

     rt->media_type_mask = (1 << (AVMEDIA_TYPE_DATA+1)) - 1;
 

     ret = sdp_read_header(s);

     s->pb = NULL;
     return ret;
 
 fail:
     if (in)

         ffurl_close(in);

     ff_network_close();
     return ret;
 }
 

 static const AVClass rtp_demuxer_class = {
     .class_name     = "RTP demuxer",
     .item_name      = av_default_item_name,
     .option         = rtp_options,
     .version        = LIBAVUTIL_VERSION_INT,
 };
 

 AVInputFormat ff_rtp_demuxer = {

     .name           = "rtp",

     .long_name      = NULL_IF_CONFIG_SMALL("RTP input"),

     .priv_data_size = sizeof(RTSPState),
     .read_probe     = rtp_probe,
     .read_header    = rtp_read_header,
     .read_packet    = ff_rtsp_fetch_packet,
     .read_close     = sdp_read_close,

     .flags          = AVFMT_NOFILE,
     .priv_class     = &rtp_demuxer_class,

 };

 #endif /* CONFIG_RTP_DEMUXER */