Start working on a branch where FFmpeg is not copied, patched and carved to be built with automake but instead imported inline and built using its own build system. This is an import of a slightly modified FFmpeg current tree. xine-lib builds, install and run fine with it, but there are of course plenty of things that needs to be fixed before it can even be considered for a 1.2.x series. Work will continue in the next days of course.

CVS patchset: 8397
CVS date: 2006/12/02 01:19:48

1 files changed, 1493 insertions, 0 deletions

diff --git a/contrib/ffmpeg/libavformat/rtsp.c b/contrib/ffmpeg/libavformat/rtsp.c
new file mode 100644
index 000000000..787cdd685
--- /dev/null
+++ b/contrib/ffmpeg/libavformat/rtsp.c
@@ -0,0 +1,1493 @@
+/*
+ * RTSP/SDP client
+ * Copyright (c) 2002 Fabrice Bellard.
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+#include "avformat.h"
+
+#include <unistd.h> /* for select() prototype */
+#include <sys/time.h>
+#include <netinet/in.h>
+#include <sys/socket.h>
+#ifndef __BEOS__
+# include <arpa/inet.h>
+#else
+# include "barpainet.h"
+#endif
+
+#include "rtp_internal.h"
+
+//#define DEBUG
+//#define DEBUG_RTP_TCP
+
+enum RTSPClientState {
+    RTSP_STATE_IDLE,
+    RTSP_STATE_PLAYING,
+    RTSP_STATE_PAUSED,
+};
+
+typedef struct RTSPState {
+    URLContext *rtsp_hd; /* RTSP TCP connexion handle */
+    int nb_rtsp_streams;
+    struct RTSPStream **rtsp_streams;
+
+    enum RTSPClientState state;
+    int64_t seek_timestamp;
+
+    /* XXX: currently we use unbuffered input */
+    //    ByteIOContext rtsp_gb;
+    int seq;        /* RTSP command sequence number */
+    char session_id[512];
+    enum RTSPProtocol protocol;
+    char last_reply[2048]; /* XXX: allocate ? */
+    RTPDemuxContext *cur_rtp;
+} RTSPState;
+
+typedef struct RTSPStream {
+    URLContext *rtp_handle; /* RTP stream handle */
+    RTPDemuxContext *rtp_ctx; /* RTP parse context */
+
+    int stream_index; /* corresponding stream index, if any. -1 if none (MPEG2TS case) */
+    int interleaved_min, interleaved_max;  /* interleave ids, if TCP transport */
+    char control_url[1024]; /* url for this stream (from SDP) */
+
+    int sdp_port; /* port (from SDP content - not used in RTSP) */
+    struct in_addr sdp_ip; /* IP address  (from SDP content - not used in RTSP) */
+    int sdp_ttl;  /* IP TTL (from SDP content - not used in RTSP) */
+    int sdp_payload_type; /* payload type - only used in SDP */
+    rtp_payload_data_t rtp_payload_data; /* rtp payload parsing infos from SDP */
+
+    RTPDynamicProtocolHandler *dynamic_handler; ///< Only valid if it's a dynamic protocol. (This is the handler structure)
+    void *dynamic_protocol_context; ///< Only valid if it's a dynamic protocol. (This is any private data associated with the dynamic protocol)
+} RTSPStream;
+
+static int rtsp_read_play(AVFormatContext *s);
+
+/* XXX: currently, the only way to change the protocols consists in
+   changing this variable */
+
+int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_UDP);
+
+FFRTSPCallback *ff_rtsp_callback = NULL;
+
+static int rtsp_probe(AVProbeData *p)
+{
+    if (strstart(p->filename, "rtsp:", NULL))
+        return AVPROBE_SCORE_MAX;
+    return 0;
+}
+
+static int redir_isspace(int c)
+{
+    return (c == ' ' || c == '\t' || c == '\n' || c == '\r');
+}
+
+static void skip_spaces(const char **pp)
+{
+    const char *p;
+    p = *pp;
+    while (redir_isspace(*p))
+        p++;
+    *pp = p;
+}
+
+static void get_word_sep(char *buf, int buf_size, const char *sep,
+                         const char **pp)
+{
+    const char *p;
+    char *q;
+
+    p = *pp;
+    if (*p == '/')
+        p++;
+    skip_spaces(&p);
+    q = buf;
+    while (!strchr(sep, *p) && *p != '\0') {
+        if ((q - buf) < buf_size - 1)
+            *q++ = *p;
+        p++;
+    }
+    if (buf_size > 0)
+        *q = '\0';
+    *pp = p;
+}
+
+static void get_word(char *buf, int buf_size, const char **pp)
+{
+    const char *p;
+    char *q;
+
+    p = *pp;
+    skip_spaces(&p);
+    q = buf;
+    while (!redir_isspace(*p) && *p != '\0') {
+        if ((q - buf) < buf_size - 1)
+            *q++ = *p;
+        p++;
+    }
+    if (buf_size > 0)
+        *q = '\0';
+    *pp = p;
+}
+
+/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other
+   params>] */
+static int sdp_parse_rtpmap(AVCodecContext *codec, RTSPStream *rtsp_st, int payload_type, const char *p)
+{
+    char buf[256];
+    int i;
+    AVCodec *c;
+    const char *c_name;
+
+    /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
+       see if we can handle this kind of payload */
+    get_word_sep(buf, sizeof(buf), "/", &p);
+    if (payload_type >= RTP_PT_PRIVATE) {
+        RTPDynamicProtocolHandler *handler= RTPFirstDynamicPayloadHandler;
+        while(handler) {
+            if (!strcmp(buf, handler->enc_name) && (codec->codec_type == handler->codec_type)) {
+                codec->codec_id = handler->codec_id;
+                rtsp_st->dynamic_handler= handler;
+                if(handler->open) {
+                    rtsp_st->dynamic_protocol_context= handler->open();
+                }
+                break;
+            }
+            handler= handler->next;
+        }
+    } else {
+        /* We are in a standard case ( from http://www.iana.org/assignments/rtp-parameters) */
+        /* search into AVRtpPayloadTypes[] */
+        for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
+            if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpPayloadTypes[i].codec_type)){
+                codec->codec_id = AVRtpPayloadTypes[i].codec_id;
+                break;
+            }
+    }
+
+    c = avcodec_find_decoder(codec->codec_id);
+    if (c && c->name)
+        c_name = c->name;
+    else
+        c_name = (char *)NULL;
+
+    if (c_name) {
+        get_word_sep(buf, sizeof(buf), "/", &p);
+        i = atoi(buf);
+        switch (codec->codec_type) {
+            case CODEC_TYPE_AUDIO:
+                av_log(codec, AV_LOG_DEBUG, " audio codec set to : %s\n", c_name);
+                codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
+                codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
+                if (i > 0) {
+                    codec->sample_rate = i;
+                    get_word_sep(buf, sizeof(buf), "/", &p);
+                    i = atoi(buf);
+                    if (i > 0)
+                        codec->channels = i;
+                    // TODO: there is a bug here; if it is a mono stream, and less than 22000Hz, faad upconverts to stereo and twice the
+                    //  frequency.  No problem, but the sample rate is being set here by the sdp line.  Upcoming patch forthcoming. (rdm)
+                }
+                av_log(codec, AV_LOG_DEBUG, " audio samplerate set to : %i\n", codec->sample_rate);
+                av_log(codec, AV_LOG_DEBUG, " audio channels set to : %i\n", codec->channels);
+                break;
+            case CODEC_TYPE_VIDEO:
+                av_log(codec, AV_LOG_DEBUG, " video codec set to : %s\n", c_name);
+                break;
+            default:
+                break;
+        }
+        return 0;
+    }
+
+    return -1;
+}
+
+/* return the length and optionnaly the data */
+static int hex_to_data(uint8_t *data, const char *p)
+{
+    int c, len, v;
+
+    len = 0;
+    v = 1;
+    for(;;) {
+        skip_spaces(&p);
+        if (p == '\0')
+            break;
+        c = toupper((unsigned char)*p++);
+        if (c >= '0' && c <= '9')
+            c = c - '0';
+        else if (c >= 'A' && c <= 'F')
+            c = c - 'A' + 10;
+        else
+            break;
+        v = (v << 4) | c;
+        if (v & 0x100) {
+            if (data)
+                data[len] = v;
+            len++;
+            v = 1;
+        }
+    }
+    return len;
+}
+
+static void sdp_parse_fmtp_config(AVCodecContext *codec, char *attr, char *value)
+{
+    switch (codec->codec_id) {
+        case CODEC_ID_MPEG4:
+        case CODEC_ID_AAC:
+            if (!strcmp(attr, "config")) {
+                /* decode the hexa encoded parameter */
+                int len = hex_to_data(NULL, value);
+                codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
+                if (!codec->extradata)
+                    return;
+                codec->extradata_size = len;
+                hex_to_data(codec->extradata, value);
+            }
+            break;
+        default:
+            break;
+    }
+    return;
+}
+
+typedef struct attrname_map
+{
+    const char *str;
+    uint16_t type;
+    uint32_t offset;
+} attrname_map_t;
+
+/* All known fmtp parmeters and the corresping RTPAttrTypeEnum */
+#define ATTR_NAME_TYPE_INT 0
+#define ATTR_NAME_TYPE_STR 1
+static attrname_map_t attr_names[]=
+{
+    {"SizeLength",       ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, sizelength)},
+    {"IndexLength",      ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexlength)},
+    {"IndexDeltaLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexdeltalength)},
+    {"profile-level-id", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, profile_level_id)},
+    {"StreamType",       ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, streamtype)},
+    {"mode",             ATTR_NAME_TYPE_STR, offsetof(rtp_payload_data_t, mode)},
+    {NULL, -1, -1},
+};
+
+/** parse the attribute line from the fmtp a line of an sdp resonse.  This is broken out as a function
+* because it is used in rtp_h264.c, which is forthcoming.
+*/
+int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size)
+{
+    skip_spaces(p);
+    if(**p)
+    {
+        get_word_sep(attr, attr_size, "=", p);
+        if (**p == '=')
+            (*p)++;
+        get_word_sep(value, value_size, ";", p);
+        if (**p == ';')
+            (*p)++;
+        return 1;
+    }
+    return 0;
+}
+
+/* parse a SDP line and save stream attributes */
+static void sdp_parse_fmtp(AVStream *st, const char *p)
+{
+    char attr[256];
+    char value[4096];
+    int i;
+
+    RTSPStream *rtsp_st = st->priv_data;
+    AVCodecContext *codec = st->codec;
+    rtp_payload_data_t *rtp_payload_data = &rtsp_st->rtp_payload_data;
+
+    /* loop on each attribute */
+    while(rtsp_next_attr_and_value(&p, attr, sizeof(attr), value, sizeof(value)))
+    {
+        /* grab the codec extra_data from the config parameter of the fmtp line */
+        sdp_parse_fmtp_config(codec, attr, value);
+        /* Looking for a known attribute */
+        for (i = 0; attr_names[i].str; ++i) {
+            if (!strcasecmp(attr, attr_names[i].str)) {
+                if (attr_names[i].type == ATTR_NAME_TYPE_INT)
+                    *(int *)((char *)rtp_payload_data + attr_names[i].offset) = atoi(value);
+                else if (attr_names[i].type == ATTR_NAME_TYPE_STR)
+                    *(char **)((char *)rtp_payload_data + attr_names[i].offset) = av_strdup(value);
+            }
+        }
+    }
+}
+
+/** Parse a string \p in the form of Range:npt=xx-xx, and determine the start
+ *  and end time.
+ *  Used for seeking in the rtp stream.
+ */
+static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)
+{
+    char buf[256];
+
+    skip_spaces(&p);
+    if (!stristart(p, "npt=", &p))
+        return;
+
+    *start = AV_NOPTS_VALUE;
+    *end = AV_NOPTS_VALUE;
+
+    get_word_sep(buf, sizeof(buf), "-", &p);
+    *start = parse_date(buf, 1);
+    if (*p == '-') {
+        p++;
+        get_word_sep(buf, sizeof(buf), "-", &p);
+        *end = parse_date(buf, 1);
+    }
+//    av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);
+//    av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);
+}
+
+typedef struct SDPParseState {
+    /* SDP only */
+    struct in_addr default_ip;
+    int default_ttl;
+} SDPParseState;
+
+static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
+                           int letter, const char *buf)
+{
+    RTSPState *rt = s->priv_data;
+    char buf1[64], st_type[64];
+    const char *p;
+    int codec_type, payload_type, i;
+    AVStream *st;
+    RTSPStream *rtsp_st;
+    struct in_addr sdp_ip;
+    int ttl;
+
+#ifdef DEBUG
+    printf("sdp: %c='%s'\n", letter, buf);
+#endif
+
+    p = buf;
+    switch(letter) {
+    case 'c':
+        get_word(buf1, sizeof(buf1), &p);
+        if (strcmp(buf1, "IN") != 0)
+            return;
+        get_word(buf1, sizeof(buf1), &p);
+        if (strcmp(buf1, "IP4") != 0)
+            return;
+        get_word_sep(buf1, sizeof(buf1), "/", &p);
+        if (inet_aton(buf1, &sdp_ip) == 0)
+            return;
+        ttl = 16;
+        if (*p == '/') {
+            p++;
+            get_word_sep(buf1, sizeof(buf1), "/", &p);
+            ttl = atoi(buf1);
+        }
+        if (s->nb_streams == 0) {
+            s1->default_ip = sdp_ip;
+            s1->default_ttl = ttl;
+        } else {
+            st = s->streams[s->nb_streams - 1];
+            rtsp_st = st->priv_data;
+            rtsp_st->sdp_ip = sdp_ip;
+            rtsp_st->sdp_ttl = ttl;
+        }
+        break;
+    case 's':
+        pstrcpy(s->title, sizeof(s->title), p);
+        break;
+    case 'i':
+        if (s->nb_streams == 0) {
+            pstrcpy(s->comment, sizeof(s->comment), p);
+            break;
+        }
+        break;
+    case 'm':
+        /* new stream */
+        get_word(st_type, sizeof(st_type), &p);
+        if (!strcmp(st_type, "audio")) {
+            codec_type = CODEC_TYPE_AUDIO;
+        } else if (!strcmp(st_type, "video")) {
+            codec_type = CODEC_TYPE_VIDEO;
+        } else {
+            return;
+        }
+        rtsp_st = av_mallocz(sizeof(RTSPStream));
+        if (!rtsp_st)
+            return;
+        rtsp_st->stream_index = -1;
+        dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
+
+        rtsp_st->sdp_ip = s1->default_ip;
+        rtsp_st->sdp_ttl = s1->default_ttl;
+
+        get_word(buf1, sizeof(buf1), &p); /* port */
+        rtsp_st->sdp_port = atoi(buf1);
+
+        get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */
+
+        /* XXX: handle list of formats */
+        get_word(buf1, sizeof(buf1), &p); /* format list */
+        rtsp_st->sdp_payload_type = atoi(buf1);
+
+        if (!strcmp(AVRtpPayloadTypes[rtsp_st->sdp_payload_type].enc_name, "MP2T")) {
+            /* no corresponding stream */
+        } else {
+            st = av_new_stream(s, 0);
+            if (!st)
+                return;
+            st->priv_data = rtsp_st;
+            rtsp_st->stream_index = st->index;
+            st->codec->codec_type = codec_type;
+            if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
+                /* if standard payload type, we can find the codec right now */
+                rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
+            }
+        }
+        /* put a default control url */
+        pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), s->filename);
+        break;
+    case 'a':
+        if (strstart(p, "control:", &p) && s->nb_streams > 0) {
+            char proto[32];
+            /* get the control url */
+            st = s->streams[s->nb_streams - 1];
+            rtsp_st = st->priv_data;
+
+            /* XXX: may need to add full url resolution */
+            url_split(proto, sizeof(proto), NULL, 0, NULL, 0, NULL, NULL, 0, p);
+            if (proto[0] == '\0') {
+                /* relative control URL */
+                pstrcat(rtsp_st->control_url, sizeof(rtsp_st->control_url), "/");
+                pstrcat(rtsp_st->control_url, sizeof(rtsp_st->control_url), p);
+            } else {
+                pstrcpy(rtsp_st->control_url, sizeof(rtsp_st->control_url), p);
+            }
+        } else if (strstart(p, "rtpmap:", &p)) {
+            /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
+            get_word(buf1, sizeof(buf1), &p);
+            payload_type = atoi(buf1);
+            for(i = 0; i < s->nb_streams;i++) {
+                st = s->streams[i];
+                rtsp_st = st->priv_data;
+                if (rtsp_st->sdp_payload_type == payload_type) {
+                    sdp_parse_rtpmap(st->codec, rtsp_st, payload_type, p);
+                }
+            }
+        } else if (strstart(p, "fmtp:", &p)) {
+            /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
+            get_word(buf1, sizeof(buf1), &p);
+            payload_type = atoi(buf1);
+            for(i = 0; i < s->nb_streams;i++) {
+                st = s->streams[i];
+                rtsp_st = st->priv_data;
+                if (rtsp_st->sdp_payload_type == payload_type) {
+                    if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) {
+                        if(!rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf)) {
+                            sdp_parse_fmtp(st, p);
+                        }
+                    } else {
+                        sdp_parse_fmtp(st, p);
+                    }
+                }
+            }
+        } else if(strstart(p, "framesize:", &p)) {
+            // let dynamic protocol handlers have a stab at the line.
+            get_word(buf1, sizeof(buf1), &p);
+            payload_type = atoi(buf1);
+            for(i = 0; i < s->nb_streams;i++) {
+                st = s->streams[i];
+                rtsp_st = st->priv_data;
+                if (rtsp_st->sdp_payload_type == payload_type) {
+                    if(rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->parse_sdp_a_line) {
+                        rtsp_st->dynamic_handler->parse_sdp_a_line(st, rtsp_st->dynamic_protocol_context, buf);
+                    }
+                }
+            }
+        } else if(strstart(p, "range:", &p)) {
+            int64_t start, end;
+
+            // this is so that seeking on a streamed file can work.
+            rtsp_parse_range_npt(p, &start, &end);
+            s->start_time= start;
+            s->duration= (end==AV_NOPTS_VALUE)?AV_NOPTS_VALUE:end-start; // AV_NOPTS_VALUE means live broadcast (and can't seek)
+        }
+        break;
+    }
+}
+
+static int sdp_parse(AVFormatContext *s, const char *content)
+{
+    const char *p;
+    int letter;
+    char buf[1024], *q;
+    SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
+
+    memset(s1, 0, sizeof(SDPParseState));
+    p = content;
+    for(;;) {
+        skip_spaces(&p);
+        letter = *p;
+        if (letter == '\0')
+            break;
+        p++;
+        if (*p != '=')
+            goto next_line;
+        p++;
+        /* get the content */
+        q = buf;
+        while (*p != '\n' && *p != '\r' && *p != '\0') {
+            if ((q - buf) < sizeof(buf) - 1)
+                *q++ = *p;
+            p++;
+        }
+        *q = '\0';
+        sdp_parse_line(s, s1, letter, buf);
+    next_line:
+        while (*p != '\n' && *p != '\0')
+            p++;
+        if (*p == '\n')
+            p++;
+    }
+    return 0;
+}
+
+static void rtsp_parse_range(int *min_ptr, int *max_ptr, const char **pp)
+{
+    const char *p;
+    int v;
+
+    p = *pp;
+    skip_spaces(&p);
+    v = strtol(p, (char **)&p, 10);
+    if (*p == '-') {
+        p++;
+        *min_ptr = v;
+        v = strtol(p, (char **)&p, 10);
+        *max_ptr = v;
+    } else {
+        *min_ptr = v;
+        *max_ptr = v;
+    }
+    *pp = p;
+}
+
+/* XXX: only one transport specification is parsed */
+static void rtsp_parse_transport(RTSPHeader *reply, const char *p)
+{
+    char transport_protocol[16];
+    char profile[16];
+    char lower_transport[16];
+    char parameter[16];
+    RTSPTransportField *th;
+    char buf[256];
+
+    reply->nb_transports = 0;
+
+    for(;;) {
+        skip_spaces(&p);
+        if (*p == '\0')
+            break;
+
+        th = &reply->transports[reply->nb_transports];
+
+        get_word_sep(transport_protocol, sizeof(transport_protocol),
+                     "/", &p);
+        if (*p == '/')
+            p++;
+        get_word_sep(profile, sizeof(profile), "/;,", &p);
+        lower_transport[0] = '\0';
+        if (*p == '/') {
+            p++;
+            get_word_sep(lower_transport, sizeof(lower_transport),
+                         ";,", &p);
+        }
+        if (!strcasecmp(lower_transport, "TCP"))
+            th->protocol = RTSP_PROTOCOL_RTP_TCP;
+        else
+            th->protocol = RTSP_PROTOCOL_RTP_UDP;
+
+        if (*p == ';')
+            p++;
+        /* get each parameter */
+        while (*p != '\0' && *p != ',') {
+            get_word_sep(parameter, sizeof(parameter), "=;,", &p);
+            if (!strcmp(parameter, "port")) {
+                if (*p == '=') {
+                    p++;
+                    rtsp_parse_range(&th->port_min, &th->port_max, &p);
+                }
+            } else if (!strcmp(parameter, "client_port")) {
+                if (*p == '=') {
+                    p++;
+                    rtsp_parse_range(&th->client_port_min,
+                                     &th->client_port_max, &p);
+                }
+            } else if (!strcmp(parameter, "server_port")) {
+                if (*p == '=') {
+                    p++;
+                    rtsp_parse_range(&th->server_port_min,
+                                     &th->server_port_max, &p);
+                }
+            } else if (!strcmp(parameter, "interleaved")) {
+                if (*p == '=') {
+                    p++;
+                    rtsp_parse_range(&th->interleaved_min,
+                                     &th->interleaved_max, &p);
+                }
+            } else if (!strcmp(parameter, "multicast")) {
+                if (th->protocol == RTSP_PROTOCOL_RTP_UDP)
+                    th->protocol = RTSP_PROTOCOL_RTP_UDP_MULTICAST;
+            } else if (!strcmp(parameter, "ttl")) {
+                if (*p == '=') {
+                    p++;
+                    th->ttl = strtol(p, (char **)&p, 10);
+                }
+            } else if (!strcmp(parameter, "destination")) {
+                struct in_addr ipaddr;
+
+                if (*p == '=') {
+                    p++;
+                    get_word_sep(buf, sizeof(buf), ";,", &p);
+                    if (inet_aton(buf, &ipaddr))
+                        th->destination = ntohl(ipaddr.s_addr);
+                }
+            }
+            while (*p != ';' && *p != '\0' && *p != ',')
+                p++;
+            if (*p == ';')
+                p++;
+        }
+        if (*p == ',')
+            p++;
+
+        reply->nb_transports++;
+    }
+}
+
+void rtsp_parse_line(RTSPHeader *reply, const char *buf)
+{
+    const char *p;
+
+    /* NOTE: we do case independent match for broken servers */
+    p = buf;
+    if (stristart(p, "Session:", &p)) {
+        get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
+    } else if (stristart(p, "Content-Length:", &p)) {
+        reply->content_length = strtol(p, NULL, 10);
+    } else if (stristart(p, "Transport:", &p)) {
+        rtsp_parse_transport(reply, p);
+    } else if (stristart(p, "CSeq:", &p)) {
+        reply->seq = strtol(p, NULL, 10);
+    } else if (stristart(p, "Range:", &p)) {
+        rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
+    }
+}
+
+static int url_readbuf(URLContext *h, unsigned char *buf, int size)
+{
+    int ret, len;
+
+    len = 0;
+    while (len < size) {
+        ret = url_read(h, buf+len, size-len);
+        if (ret < 1)
+            return ret;
+        len += ret;
+    }
+    return len;
+}
+
+/* skip a RTP/TCP interleaved packet */
+static void rtsp_skip_packet(AVFormatContext *s)
+{
+    RTSPState *rt = s->priv_data;
+    int ret, len, len1;
+    uint8_t buf[1024];
+
+    ret = url_readbuf(rt->rtsp_hd, buf, 3);
+    if (ret != 3)
+        return;
+    len = (buf[1] << 8) | buf[2];
+#ifdef DEBUG
+    printf("skipping RTP packet len=%d\n", len);
+#endif
+    /* skip payload */
+    while (len > 0) {
+        len1 = len;
+        if (len1 > sizeof(buf))
+            len1 = sizeof(buf);
+        ret = url_readbuf(rt->rtsp_hd, buf, len1);
+        if (ret != len1)
+            return;
+        len -= len1;
+    }
+}
+
+static void rtsp_send_cmd(AVFormatContext *s,
+                          const char *cmd, RTSPHeader *reply,
+                          unsigned char **content_ptr)
+{
+    RTSPState *rt = s->priv_data;
+    char buf[4096], buf1[1024], *q;
+    unsigned char ch;
+    const char *p;
+    int content_length, line_count;
+    unsigned char *content = NULL;
+
+    memset(reply, 0, sizeof(RTSPHeader));
+
+    rt->seq++;
+    pstrcpy(buf, sizeof(buf), cmd);
+    snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq);
+    pstrcat(buf, sizeof(buf), buf1);
+    if (rt->session_id[0] != '\0' && !strstr(cmd, "\nIf-Match:")) {
+        snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id);
+        pstrcat(buf, sizeof(buf), buf1);
+    }
+    pstrcat(buf, sizeof(buf), "\r\n");
+#ifdef DEBUG
+    printf("Sending:\n%s--\n", buf);
+#endif
+    url_write(rt->rtsp_hd, buf, strlen(buf));
+
+    /* parse reply (XXX: use buffers) */
+    line_count = 0;
+    rt->last_reply[0] = '\0';
+    for(;;) {
+        q = buf;
+        for(;;) {
+            if (url_readbuf(rt->rtsp_hd, &ch, 1) != 1)
+                break;
+            if (ch == '\n')
+                break;
+            if (ch == '$') {
+                /* XXX: only parse it if first char on line ? */
+                rtsp_skip_packet(s);
+            } else if (ch != '\r') {
+                if ((q - buf) < sizeof(buf) - 1)
+                    *q++ = ch;
+            }
+        }
+        *q = '\0';
+#ifdef DEBUG
+        printf("line='%s'\n", buf);
+#endif
+        /* test if last line */
+        if (buf[0] == '\0')
+            break;
+        p = buf;
+        if (line_count == 0) {
+            /* get reply code */
+            get_word(buf1, sizeof(buf1), &p);
+            get_word(buf1, sizeof(buf1), &p);
+            reply->status_code = atoi(buf1);
+        } else {
+            rtsp_parse_line(reply, p);
+            pstrcat(rt->last_reply, sizeof(rt->last_reply), p);
+            pstrcat(rt->last_reply, sizeof(rt->last_reply), "\n");
+        }
+        line_count++;
+    }
+
+    if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
+        pstrcpy(rt->session_id, sizeof(rt->session_id), reply->session_id);
+
+    content_length = reply->content_length;
+    if (content_length > 0) {
+        /* leave some room for a trailing '\0' (useful for simple parsing) */
+        content = av_malloc(content_length + 1);
+        (void)url_readbuf(rt->rtsp_hd, content, content_length);
+        content[content_length] = '\0';
+    }
+    if (content_ptr)
+        *content_ptr = content;
+}
+
+/* useful for modules: set RTSP callback function */
+
+void rtsp_set_callback(FFRTSPCallback *rtsp_cb)
+{
+    ff_rtsp_callback = rtsp_cb;
+}
+
+
+/* close and free RTSP streams */
+static void rtsp_close_streams(RTSPState *rt)
+{
+    int i;
+    RTSPStream *rtsp_st;
+
+    for(i=0;i<rt->nb_rtsp_streams;i++) {
+        rtsp_st = rt->rtsp_streams[i];
+        if (rtsp_st) {
+            if (rtsp_st->rtp_ctx)
+                rtp_parse_close(rtsp_st->rtp_ctx);
+            if (rtsp_st->rtp_handle)
+                url_close(rtsp_st->rtp_handle);
+            if (rtsp_st->dynamic_handler && rtsp_st->dynamic_protocol_context)
+                rtsp_st->dynamic_handler->close(rtsp_st->dynamic_protocol_context);
+        }
+        av_free(rtsp_st);
+    }
+    av_free(rt->rtsp_streams);
+}
+
+static int rtsp_read_header(AVFormatContext *s,
+                            AVFormatParameters *ap)
+{
+    RTSPState *rt = s->priv_data;
+    char host[1024], path[1024], tcpname[1024], cmd[2048];
+    URLContext *rtsp_hd;
+    int port, i, j, ret, err;
+    RTSPHeader reply1, *reply = &reply1;
+    unsigned char *content = NULL;
+    RTSPStream *rtsp_st;
+    int protocol_mask;
+    AVStream *st;
+
+    /* extract hostname and port */
+    url_split(NULL, 0, NULL, 0,
+              host, sizeof(host), &port, path, sizeof(path), s->filename);
+    if (port < 0)
+        port = RTSP_DEFAULT_PORT;
+
+    /* open the tcp connexion */
+    snprintf(tcpname, sizeof(tcpname), "tcp://%s:%d", host, port);
+    if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0)
+        return AVERROR_IO;
+    rt->rtsp_hd = rtsp_hd;
+    rt->seq = 0;
+
+    /* describe the stream */
+    snprintf(cmd, sizeof(cmd),
+             "DESCRIBE %s RTSP/1.0\r\n"
+             "Accept: application/sdp\r\n",
+             s->filename);
+    rtsp_send_cmd(s, cmd, reply, &content);
+    if (!content) {
+        err = AVERROR_INVALIDDATA;
+        goto fail;
+    }
+    if (reply->status_code != RTSP_STATUS_OK) {
+        err = AVERROR_INVALIDDATA;
+        goto fail;
+    }
+
+    /* now we got the SDP description, we parse it */
+    ret = sdp_parse(s, (const char *)content);
+    av_freep(&content);
+    if (ret < 0) {
+        err = AVERROR_INVALIDDATA;
+        goto fail;
+    }
+
+    protocol_mask = rtsp_default_protocols;
+
+    /* for each stream, make the setup request */
+    /* XXX: we assume the same server is used for the control of each
+       RTSP stream */
+
+    for(j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
+        char transport[2048];
+
+        rtsp_st = rt->rtsp_streams[i];
+
+        /* compute available transports */
+        transport[0] = '\0';
+
+        /* RTP/UDP */
+        if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP)) {
+            char buf[256];
+
+            /* first try in specified port range */
+            if (RTSP_RTP_PORT_MIN != 0) {
+                while(j <= RTSP_RTP_PORT_MAX) {
+                    snprintf(buf, sizeof(buf), "rtp://?localport=%d", j);
+                    if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0) {
+                        j += 2; /* we will use two port by rtp stream (rtp and rtcp) */
+                        goto rtp_opened;
+                    }
+                }
+            }
+
+/*            then try on any port
+**            if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
+**                err = AVERROR_INVALIDDATA;
+**                goto fail;
+**            }
+*/
+
+        rtp_opened:
+            port = rtp_get_local_port(rtsp_st->rtp_handle);
+            if (transport[0] != '\0')
+                pstrcat(transport, sizeof(transport), ",");
+            snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1,
+                     "RTP/AVP/UDP;unicast;client_port=%d-%d",
+                     port, port + 1);
+        }
+
+        /* RTP/TCP */
+        else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_TCP)) {
+            if (transport[0] != '\0')
+                pstrcat(transport, sizeof(transport), ",");
+            snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1,
+                     "RTP/AVP/TCP");
+        }
+
+        else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST)) {
+            if (transport[0] != '\0')
+                pstrcat(transport, sizeof(transport), ",");
+            snprintf(transport + strlen(transport),
+                     sizeof(transport) - strlen(transport) - 1,
+                     "RTP/AVP/UDP;multicast");
+        }
+        snprintf(cmd, sizeof(cmd),
+                 "SETUP %s RTSP/1.0\r\n"
+                 "Transport: %s\r\n",
+                 rtsp_st->control_url, transport);
+        rtsp_send_cmd(s, cmd, reply, NULL);
+        if (reply->status_code != RTSP_STATUS_OK ||
+            reply->nb_transports != 1) {
+            err = AVERROR_INVALIDDATA;
+            goto fail;
+        }
+
+        /* XXX: same protocol for all streams is required */
+        if (i > 0) {
+            if (reply->transports[0].protocol != rt->protocol) {
+                err = AVERROR_INVALIDDATA;
+                goto fail;
+            }
+        } else {
+            rt->protocol = reply->transports[0].protocol;
+        }
+
+        /* close RTP connection if not choosen */
+        if (reply->transports[0].protocol != RTSP_PROTOCOL_RTP_UDP &&
+            (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP))) {
+            url_close(rtsp_st->rtp_handle);
+            rtsp_st->rtp_handle = NULL;
+        }
+
+        switch(reply->transports[0].protocol) {
+        case RTSP_PROTOCOL_RTP_TCP:
+            rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
+            rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
+            break;
+
+        case RTSP_PROTOCOL_RTP_UDP:
+            {
+                char url[1024];
+
+                /* XXX: also use address if specified */
+                snprintf(url, sizeof(url), "rtp://%s:%d",
+                         host, reply->transports[0].server_port_min);
+                if (rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
+                    err = AVERROR_INVALIDDATA;
+                    goto fail;
+                }
+            }
+            break;
+        case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
+            {
+                char url[1024];
+                int ttl;
+
+                ttl = reply->transports[0].ttl;
+                if (!ttl)
+                    ttl = 16;
+                snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d",
+                         host,
+                         reply->transports[0].server_port_min,
+                         ttl);
+                if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
+                    err = AVERROR_INVALIDDATA;
+                    goto fail;
+                }
+            }
+            break;
+        }
+        /* open the RTP context */
+        st = NULL;
+        if (rtsp_st->stream_index >= 0)
+            st = s->streams[rtsp_st->stream_index];
+        if (!st)
+            s->ctx_flags |= AVFMTCTX_NOHEADER;
+        rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
+
+        if (!rtsp_st->rtp_ctx) {
+            err = AVERROR_NOMEM;
+            goto fail;
+        } else {
+            if(rtsp_st->dynamic_handler) {
+                rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context;
+                rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet;
+            }
+        }
+    }
+
+    /* use callback if available to extend setup */
+    if (ff_rtsp_callback) {
+        if (ff_rtsp_callback(RTSP_ACTION_CLIENT_SETUP, rt->session_id,
+                             NULL, 0, rt->last_reply) < 0) {
+            err = AVERROR_INVALIDDATA;
+            goto fail;
+        }
+    }
+
+
+    rt->state = RTSP_STATE_IDLE;
+    rt->seek_timestamp = 0; /* default is to start stream at position
+                               zero */
+    if (ap->initial_pause) {
+        /* do not start immediately */
+    } else {
+        if (rtsp_read_play(s) < 0) {
+            err = AVERROR_INVALIDDATA;
+            goto fail;
+        }
+    }
+    return 0;
+ fail:
+    rtsp_close_streams(rt);
+    av_freep(&content);
+    url_close(rt->rtsp_hd);
+    return err;
+}
+
+static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
+                           uint8_t *buf, int buf_size)
+{
+    RTSPState *rt = s->priv_data;
+    int id, len, i, ret;
+    RTSPStream *rtsp_st;
+
+#ifdef DEBUG_RTP_TCP
+    printf("tcp_read_packet:\n");
+#endif
+ redo:
+    for(;;) {
+        ret = url_readbuf(rt->rtsp_hd, buf, 1);
+#ifdef DEBUG_RTP_TCP
+        printf("ret=%d c=%02x [%c]\n", ret, buf[0], buf[0]);
+#endif
+        if (ret != 1)
+            return -1;
+        if (buf[0] == '$')
+            break;
+    }
+    ret = url_readbuf(rt->rtsp_hd, buf, 3);
+    if (ret != 3)
+        return -1;
+    id = buf[0];
+    len = (buf[1] << 8) | buf[2];
+#ifdef DEBUG_RTP_TCP
+    printf("id=%d len=%d\n", id, len);
+#endif
+    if (len > buf_size || len < 12)
+        goto redo;
+    /* get the data */
+    ret = url_readbuf(rt->rtsp_hd, buf, len);
+    if (ret != len)
+        return -1;
+
+    /* find the matching stream */
+    for(i = 0; i < rt->nb_rtsp_streams; i++) {
+        rtsp_st = rt->rtsp_streams[i];
+        if (id >= rtsp_st->interleaved_min &&
+            id <= rtsp_st->interleaved_max)
+            goto found;
+    }
+    goto redo;
+ found:
+    *prtsp_st = rtsp_st;
+    return len;
+}
+
+static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
+                           uint8_t *buf, int buf_size)
+{
+    RTSPState *rt = s->priv_data;
+    RTSPStream *rtsp_st;
+    fd_set rfds;
+    int fd1, fd2, fd_max, n, i, ret;
+    struct timeval tv;
+
+    for(;;) {
+        if (url_interrupt_cb())
+            return -1;
+        FD_ZERO(&rfds);
+        fd_max = -1;
+        for(i = 0; i < rt->nb_rtsp_streams; i++) {
+            rtsp_st = rt->rtsp_streams[i];
+            /* currently, we cannot probe RTCP handle because of blocking restrictions */
+            rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2);
+            if (fd1 > fd_max)
+                fd_max = fd1;
+            FD_SET(fd1, &rfds);
+        }
+        tv.tv_sec = 0;
+        tv.tv_usec = 100 * 1000;
+        n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
+        if (n > 0) {
+            for(i = 0; i < rt->nb_rtsp_streams; i++) {
+                rtsp_st = rt->rtsp_streams[i];
+                rtp_get_file_handles(rtsp_st->rtp_handle, &fd1, &fd2);
+                if (FD_ISSET(fd1, &rfds)) {
+                    ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
+                    if (ret > 0) {
+                        *prtsp_st = rtsp_st;
+                        return ret;
+                    }
+                }
+            }
+        }
+    }
+}
+
+static int rtsp_read_packet(AVFormatContext *s,
+                            AVPacket *pkt)
+{
+    RTSPState *rt = s->priv_data;
+    RTSPStream *rtsp_st;
+    int ret, len;
+    uint8_t buf[RTP_MAX_PACKET_LENGTH];
+
+    /* get next frames from the same RTP packet */
+    if (rt->cur_rtp) {
+        ret = rtp_parse_packet(rt->cur_rtp, pkt, NULL, 0);
+        if (ret == 0) {
+            rt->cur_rtp = NULL;
+            return 0;
+        } else if (ret == 1) {
+            return 0;
+        } else {
+            rt->cur_rtp = NULL;
+        }
+    }
+
+    /* read next RTP packet */
+ redo:
+    switch(rt->protocol) {
+    default:
+    case RTSP_PROTOCOL_RTP_TCP:
+        len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
+        break;
+    case RTSP_PROTOCOL_RTP_UDP:
+    case RTSP_PROTOCOL_RTP_UDP_MULTICAST:
+        len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
+        if (rtsp_st->rtp_ctx)
+            rtp_check_and_send_back_rr(rtsp_st->rtp_ctx, len);
+        break;
+    }
+    if (len < 0)
+        return AVERROR_IO;
+    ret = rtp_parse_packet(rtsp_st->rtp_ctx, pkt, buf, len);
+    if (ret < 0)
+        goto redo;
+    if (ret == 1) {
+        /* more packets may follow, so we save the RTP context */
+        rt->cur_rtp = rtsp_st->rtp_ctx;
+    }
+    return 0;
+}
+
+static int rtsp_read_play(AVFormatContext *s)
+{
+    RTSPState *rt = s->priv_data;
+    RTSPHeader reply1, *reply = &reply1;
+    char cmd[1024];
+
+    av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
+
+    if (rt->state == RTSP_STATE_PAUSED) {
+        snprintf(cmd, sizeof(cmd),
+                 "PLAY %s RTSP/1.0\r\n",
+                 s->filename);
+    } else {
+        snprintf(cmd, sizeof(cmd),
+                 "PLAY %s RTSP/1.0\r\n"
+                 "Range: npt=%0.3f-\r\n",
+                 s->filename,
+                 (double)rt->seek_timestamp / AV_TIME_BASE);
+    }
+    rtsp_send_cmd(s, cmd, reply, NULL);
+    if (reply->status_code != RTSP_STATUS_OK) {
+        return -1;
+    } else {
+        rt->state = RTSP_STATE_PLAYING;
+        return 0;
+    }
+}
+
+/* pause the stream */
+static int rtsp_read_pause(AVFormatContext *s)
+{
+    RTSPState *rt = s->priv_data;
+    RTSPHeader reply1, *reply = &reply1;
+    char cmd[1024];
+
+    rt = s->priv_data;
+
+    if (rt->state != RTSP_STATE_PLAYING)
+        return 0;
+
+    snprintf(cmd, sizeof(cmd),
+             "PAUSE %s RTSP/1.0\r\n",
+             s->filename);
+    rtsp_send_cmd(s, cmd, reply, NULL);
+    if (reply->status_code != RTSP_STATUS_OK) {
+        return -1;
+    } else {
+        rt->state = RTSP_STATE_PAUSED;
+        return 0;
+    }
+}
+
+static int rtsp_read_seek(AVFormatContext *s, int stream_index,
+                          int64_t timestamp, int flags)
+{
+    RTSPState *rt = s->priv_data;
+
+    rt->seek_timestamp = timestamp;
+    switch(rt->state) {
+    default:
+    case RTSP_STATE_IDLE:
+        break;
+    case RTSP_STATE_PLAYING:
+        if (rtsp_read_play(s) != 0)
+            return -1;
+        break;
+    case RTSP_STATE_PAUSED:
+        rt->state = RTSP_STATE_IDLE;
+        break;
+    }
+    return 0;
+}
+
+static int rtsp_read_close(AVFormatContext *s)
+{
+    RTSPState *rt = s->priv_data;
+    RTSPHeader reply1, *reply = &reply1;
+    char cmd[1024];
+
+#if 0
+    /* NOTE: it is valid to flush the buffer here */
+    if (rt->protocol == RTSP_PROTOCOL_RTP_TCP) {
+        url_fclose(&rt->rtsp_gb);
+    }
+#endif
+    snprintf(cmd, sizeof(cmd),
+             "TEARDOWN %s RTSP/1.0\r\n",
+             s->filename);
+    rtsp_send_cmd(s, cmd, reply, NULL);
+
+    if (ff_rtsp_callback) {
+        ff_rtsp_callback(RTSP_ACTION_CLIENT_TEARDOWN, rt->session_id,
+                         NULL, 0, NULL);
+    }
+
+    rtsp_close_streams(rt);
+    url_close(rt->rtsp_hd);
+    return 0;
+}
+
+AVInputFormat rtsp_demuxer = {
+    "rtsp",
+    "RTSP input format",
+    sizeof(RTSPState),
+    rtsp_probe,
+    rtsp_read_header,
+    rtsp_read_packet,
+    rtsp_read_close,
+    rtsp_read_seek,
+    .flags = AVFMT_NOFILE,
+    .read_play = rtsp_read_play,
+    .read_pause = rtsp_read_pause,
+};
+
+static int sdp_probe(AVProbeData *p1)
+{
+    const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
+
+    /* we look for a line beginning "c=IN IP4" */
+    while (p < p_end && *p != '\0') {
+        if (p + sizeof("c=IN IP4") - 1 < p_end && strstart(p, "c=IN IP4", NULL))
+            return AVPROBE_SCORE_MAX / 2;
+
+        while(p < p_end - 1 && *p != '\n') p++;
+        if (++p >= p_end)
+            break;
+        if (*p == '\r')
+            p++;
+    }
+    return 0;
+}
+
+#define SDP_MAX_SIZE 8192
+
+static int sdp_read_header(AVFormatContext *s,
+                           AVFormatParameters *ap)
+{
+    RTSPState *rt = s->priv_data;
+    RTSPStream *rtsp_st;
+    int size, i, err;
+    char *content;
+    char url[1024];
+    AVStream *st;
+
+    /* read the whole sdp file */
+    /* XXX: better loading */
+    content = av_malloc(SDP_MAX_SIZE);
+    size = get_buffer(&s->pb, content, SDP_MAX_SIZE - 1);
+    if (size <= 0) {
+        av_free(content);
+        return AVERROR_INVALIDDATA;
+    }
+    content[size] ='\0';
+
+    sdp_parse(s, content);
+    av_free(content);
+
+    /* open each RTP stream */
+    for(i=0;i<rt->nb_rtsp_streams;i++) {
+        rtsp_st = rt->rtsp_streams[i];
+
+        snprintf(url, sizeof(url), "rtp://%s:%d?multicast=1&ttl=%d",
+                 inet_ntoa(rtsp_st->sdp_ip),
+                 rtsp_st->sdp_port,
+                 rtsp_st->sdp_ttl);
+        if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
+            err = AVERROR_INVALIDDATA;
+            goto fail;
+        }
+        /* open the RTP context */
+        st = NULL;
+        if (rtsp_st->stream_index >= 0)
+            st = s->streams[rtsp_st->stream_index];
+        if (!st)
+            s->ctx_flags |= AVFMTCTX_NOHEADER;
+        rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->rtp_handle, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
+        if (!rtsp_st->rtp_ctx) {
+            err = AVERROR_NOMEM;
+            goto fail;
+        } else {
+            if(rtsp_st->dynamic_handler) {
+                rtsp_st->rtp_ctx->dynamic_protocol_context= rtsp_st->dynamic_protocol_context;
+                rtsp_st->rtp_ctx->parse_packet= rtsp_st->dynamic_handler->parse_packet;
+            }
+        }
+    }
+    return 0;
+ fail:
+    rtsp_close_streams(rt);
+    return err;
+}
+
+static int sdp_read_packet(AVFormatContext *s,
+                            AVPacket *pkt)
+{
+    return rtsp_read_packet(s, pkt);
+}
+
+static int sdp_read_close(AVFormatContext *s)
+{
+    RTSPState *rt = s->priv_data;
+    rtsp_close_streams(rt);
+    return 0;
+}
+
+#ifdef CONFIG_SDP_DEMUXER
+AVInputFormat sdp_demuxer = {
+    "sdp",
+    "SDP",
+    sizeof(RTSPState),
+    sdp_probe,
+    sdp_read_header,
+    sdp_read_packet,
+    sdp_read_close,
+};
+#endif
+
+/* dummy redirector format (used directly in av_open_input_file now) */
+static int redir_probe(AVProbeData *pd)
+{
+    const char *p;
+    p = pd->buf;
+    while (redir_isspace(*p))
+        p++;
+    if (strstart(p, "http://", NULL) ||
+        strstart(p, "rtsp://", NULL))
+        return AVPROBE_SCORE_MAX;
+    return 0;
+}
+
+/* called from utils.c */
+int redir_open(AVFormatContext **ic_ptr, ByteIOContext *f)
+{
+    char buf[4096], *q;
+    int c;
+    AVFormatContext *ic = NULL;
+
+    /* parse each URL and try to open it */
+    c = url_fgetc(f);
+    while (c != URL_EOF) {
+        /* skip spaces */
+        for(;;) {
+            if (!redir_isspace(c))
+                break;
+            c = url_fgetc(f);
+        }
+        if (c == URL_EOF)
+            break;
+        /* record url */
+        q = buf;
+        for(;;) {
+            if (c == URL_EOF || redir_isspace(c))
+                break;
+            if ((q - buf) < sizeof(buf) - 1)
+                *q++ = c;
+            c = url_fgetc(f);
+        }
+        *q = '\0';
+        //printf("URL='%s'\n", buf);
+        /* try to open the media file */
+        if (av_open_input_file(&ic, buf, NULL, 0, NULL) == 0)
+            break;
+    }
+    *ic_ptr = ic;
+    if (!ic)
+        return AVERROR_IO;
+    else
+        return 0;
+}
+
+AVInputFormat redir_demuxer = {
+    "redir",
+    "Redirector format",
+    0,
+    redir_probe,
+    NULL,
+    NULL,
+    NULL,
+};