PeerStreamer

/*

 * RTSP/SDP client

 * Copyright (c) 2002 Fabrice Bellard

 *

 * This file is part of FFmpeg.

 *

 * FFmpeg is free software; you can redistribute it and/or

 * modify it under the terms of the GNU Lesser General Public

 * License as published by the Free Software Foundation; either

 * version 2.1 of the License, or (at your option) any later version.

 *

 * FFmpeg is distributed in the hope that it will be useful,

 * but WITHOUT ANY WARRANTY; without even the implied warranty of

 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

 * Lesser General Public License for more details.

 *

 * You should have received a copy of the GNU Lesser General Public

 * License along with FFmpeg; if not, write to the Free Software

 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

 */

#include "libavutil/base64.h"

#include "libavutil/avstring.h"

#include "libavutil/intreadwrite.h"

#include "avformat.h"

#include <sys/time.h>

#if HAVE_SYS_SELECT_H

#include <sys/select.h>

#endif

#include <strings.h>

#include "internal.h"

#include "network.h"

#include "os_support.h"

#include "rtsp.h"

#include "rtpdec.h"

#include "rdt.h"

#include "rtpdec_asf.h"

#include "rtpdec_vorbis.h"

//#define DEBUG

//#define DEBUG_RTP_TCP

#if LIBAVFORMAT_VERSION_INT < (53 << 16)

int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP);

#endif

#define SPACE_CHARS " \t\r\n"

/* we use memchr() instead of strchr() here because strchr() will return

 * the terminating '\0' of SPACE_CHARS instead of NULL if c is '\0'. */

#define redir_isspace(c) memchr(SPACE_CHARS, c, 4)

static void skip_spaces(const char **pp)

{

    const char *p;

    p = *pp;

    while (redir_isspace(*p))

        p++;

    *pp = p;

}

static void get_word_until_chars(char *buf, int buf_size,

                                 const char *sep, const char **pp)

{

    const char *p;

    char *q;

    p = *pp;

    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_sep(char *buf, int buf_size, const char *sep,

                         const char **pp)

{

    if (**pp == '/') (*pp)++;

    get_word_until_chars(buf, buf_size, sep, pp);

}

static void get_word(char *buf, int buf_size, const char **pp)

{

    get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);

}

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

static int sdp_parse_rtpmap(AVFormatContext *s,

                            AVCodecContext *codec, RTSPStream *rtsp_st,

                            int payload_type, const char *p)

{

    char buf[256];

    int i;

    AVCodec *c;

    const char *c_name;

    /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and

     * see if we can handle this kind of payload.

     * The space should normally not be there but some Real streams or

     * particular servers ("RealServer Version 6.1.3.970", see issue 1658)

     * have a trailing space. */

    get_word_sep(buf, sizeof(buf), "/ ", &p);

    if (payload_type >= RTP_PT_PRIVATE) {

        RTPDynamicProtocolHandler *handler;

        for (handler = RTPFirstDynamicPayloadHandler;

             handler; handler = handler->next) {

            if (!strcasecmp(buf, handler->enc_name) &&

                codec->codec_type == handler->codec_type) {

                codec->codec_id          = handler->codec_id;

                rtsp_st->dynamic_handler = handler;

                if (handler->open)

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

                break;

            }

        }

    } else {

        /* We are in a standard case

         * (from http://www.iana.org/assignments/rtp-parameters). */

        /* search into AVRtpPayloadTypes[] */

        codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);

    }

    c = avcodec_find_decoder(codec->codec_id);

    if (c && c->name)

        c_name = c->name;

    else

        c_name = "(null)";

    get_word_sep(buf, sizeof(buf), "/", &p);

    i = atoi(buf);

    switch (codec->codec_type) {

    case CODEC_TYPE_AUDIO:

        av_log(s, AV_LOG_DEBUG, "audio codec set to: %s\n", c_name);

        codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;

        codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;

        if (i > 0) {

            codec->sample_rate = i;

            get_word_sep(buf, sizeof(buf), "/", &p);

            i = atoi(buf);

            if (i > 0)

                codec->channels = i;

            // TODO: there is a bug here; if it is a mono stream, and

            // less than 22000Hz, faad upconverts to stereo and twice

            // the frequency.  No problem, but the sample rate is being

            // set here by the sdp line. Patch on its way. (rdm)

        }

        av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",

               codec->sample_rate);

        av_log(s, AV_LOG_DEBUG, "audio channels set to: %i\n",

               codec->channels);

        break;

    case CODEC_TYPE_VIDEO:

        av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name);

        break;

    default:

        break;

    }

    return 0;

}

/* return the length and optionally 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, void *ctx,

                                  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);

            if (codec->extradata)

                av_free(codec->extradata);

            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;

    case CODEC_ID_VORBIS:

        ff_vorbis_parse_fmtp_config(codec, ctx, attr, value);

        break;

    default:

        break;

    }

    return;

}

typedef struct {

    const char *str;

    uint16_t    type;

    uint32_t    offset;

} AttrNameMap;

/* All known fmtp parmeters and the corresping RTPAttrTypeEnum */

#define ATTR_NAME_TYPE_INT 0

#define ATTR_NAME_TYPE_STR 1

static const AttrNameMap attr_names[]=

{

    { "SizeLength",       ATTR_NAME_TYPE_INT,

      offsetof(RTPPayloadData, sizelength) },

    { "IndexLength",      ATTR_NAME_TYPE_INT,

      offsetof(RTPPayloadData, indexlength) },

    { "IndexDeltaLength", ATTR_NAME_TYPE_INT,

      offsetof(RTPPayloadData, indexdeltalength) },

    { "profile-level-id", ATTR_NAME_TYPE_INT,

      offsetof(RTPPayloadData, profile_level_id) },

    { "StreamType",       ATTR_NAME_TYPE_INT,

      offsetof(RTPPayloadData, streamtype) },

    { "mode",             ATTR_NAME_TYPE_STR,

      offsetof(RTPPayloadData, 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 ff_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];

    /* Vorbis setup headers can be up to 12KB and are sent base64

     * encoded, giving a 12KB * (4/3) = 16KB FMTP line. */

    char value[16384];

    int i;

    RTSPStream *rtsp_st = st->priv_data;

    AVCodecContext *codec = st->codec;

    RTPPayloadData *rtp_payload_data = &rtsp_st->rtp_payload_data;

    /* loop on each attribute */

    while (ff_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, rtsp_st->dynamic_protocol_context,

                              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 (!av_stristart(p, "npt=", &p))

        return;

    *start = AV_NOPTS_VALUE;

    *end = AV_NOPTS_VALUE;

    get_word_sep(buf, sizeof(buf), "-", &p);

    *start = parse_date(buf, 1);

    if (*p == '-') {

        p++;

        get_word_sep(buf, sizeof(buf), "-", &p);

        *end = parse_date(buf, 1);

    }

//    av_log(NULL, AV_LOG_DEBUG, "Range Start: %lld\n", *start);

//    av_log(NULL, AV_LOG_DEBUG, "Range End: %lld\n", *end);

}

typedef struct SDPParseState {

    /* SDP only */

    struct in_addr default_ip;

    int            default_ttl;

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

} SDPParseState;

static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,

                           int letter, const char *buf)

{

    RTSPState *rt = s->priv_data;

    char buf1[64], st_type[64];

    const char *p;

    enum CodecType codec_type;

    int payload_type, i;

    AVStream *st;

    RTSPStream *rtsp_st;

    struct in_addr sdp_ip;

    int ttl;

    dprintf(s, "sdp: %c='%s'\n", letter, buf);

    p = buf;

    if (s1->skip_media && letter != 'm')

        return;

    switch (letter) {

    case 'c':

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

        if (strcmp(buf1, "IN") != 0)

            return;

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

        if (strcmp(buf1, "IP4") != 0)

            return;

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

        if (ff_inet_aton(buf1, &sdp_ip) == 0)

            return;

        ttl = 16;

        if (*p == '/') {

            p++;

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

            ttl = atoi(buf1);

        }

        if (s->nb_streams == 0) {

            s1->default_ip = sdp_ip;

            s1->default_ttl = ttl;

        } else {

            st = s->streams[s->nb_streams - 1];

            rtsp_st = st->priv_data;

            rtsp_st->sdp_ip = sdp_ip;

            rtsp_st->sdp_ttl = ttl;

        }

        break;

    case 's':

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

        break;

    case 'i':

        if (s->nb_streams == 0) {

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

            break;

        }

        break;

    case 'm':

        /* new stream */

        s1->skip_media = 0;

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

        if (!strcmp(st_type, "audio")) {

            codec_type = CODEC_TYPE_AUDIO;

        } else if (!strcmp(st_type, "video")) {

            codec_type = CODEC_TYPE_VIDEO;

        } else if (!strcmp(st_type, "application")) {

            codec_type = CODEC_TYPE_DATA;

        } else {

            s1->skip_media = 1;

            return;

        }

        rtsp_st = av_mallocz(sizeof(RTSPStream));

        if (!rtsp_st)

            return;

        rtsp_st->stream_index = -1;

        dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);

        rtsp_st->sdp_ip = s1->default_ip;

        rtsp_st->sdp_ttl = s1->default_ttl;

        get_word(buf1, sizeof(buf1), &p); /* port */

        rtsp_st->sdp_port = atoi(buf1);

        get_word(buf1, sizeof(buf1), &p); /* protocol (ignored) */

        /* XXX: handle list of formats */

        get_word(buf1, sizeof(buf1), &p); /* format list */

        rtsp_st->sdp_payload_type = atoi(buf1);

        if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {

            /* no corresponding stream */

        } else {

            st = av_new_stream(s, 0);

            if (!st)

                return;

            st->priv_data = rtsp_st;

            rtsp_st->stream_index = st->index;

            st->codec->codec_type = codec_type;

            if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {

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

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

            }

        }

        /* put a default control url */

        av_strlcpy(rtsp_st->control_url, rt->control_uri,

                   sizeof(rtsp_st->control_url));

        break;

    case 'a':

        if (av_strstart(p, "control:", &p)) {

            if (s->nb_streams == 0) {

                if (!strncmp(p, "rtsp://", 7))

                    av_strlcpy(rt->control_uri, p,

                               sizeof(rt->control_uri));

            } else {

            char proto[32];

            /* get the control url */

            st = s->streams[s->nb_streams - 1];

            rtsp_st = st->priv_data;

            /* XXX: may need to add full url resolution */

            ff_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,

                         NULL, NULL, 0, p);

            if (proto[0] == '\0') {

                /* relative control URL */

                if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')

                av_strlcat(rtsp_st->control_url, "/",

                           sizeof(rtsp_st->control_url));

                av_strlcat(rtsp_st->control_url, p,

                           sizeof(rtsp_st->control_url));

            } else

                av_strlcpy(rtsp_st->control_url, p,

                           sizeof(rtsp_st->control_url));

            }

        } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {

            /* NOTE: rtpmap is only supported AFTER the 'm=' tag */

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

            payload_type = atoi(buf1);

            st = s->streams[s->nb_streams - 1];

            rtsp_st = st->priv_data;

            sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p);

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

            /* 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 &&

                          rtsp_st->dynamic_handler->parse_sdp_a_line(s,

                              i, rtsp_st->dynamic_protocol_context, buf)))

                        sdp_parse_fmtp(st, p);

                }

            }

        } else if (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);

            for (i = 0; i < s->nb_streams; i++) {

                st      = s->streams[i];

                rtsp_st = st->priv_data;

                if (rtsp_st->sdp_payload_type == payload_type &&

                    rtsp_st->dynamic_handler &&

                    rtsp_st->dynamic_handler->parse_sdp_a_line)

                    rtsp_st->dynamic_handler->parse_sdp_a_line(s, i,

                        rtsp_st->dynamic_protocol_context, buf);

            }

        } else if (av_strstart(p, "range:", &p)) {

            int64_t start, end;

            // this is so that seeking on a streamed file can work.

            rtsp_parse_range_npt(p, &start, &end);

            s->start_time = start;

            /* AV_NOPTS_VALUE means live broadcast (and can't seek) */

            s->duration   = (end == AV_NOPTS_VALUE) ?

                            AV_NOPTS_VALUE : end - start;

        } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {

            if (atoi(p) == 1)

                rt->transport = RTSP_TRANSPORT_RDT;

        } else {

            if (rt->server_type == RTSP_SERVER_WMS)

                ff_wms_parse_sdp_a_line(s, p);

            if (s->nb_streams > 0) {

                if (rt->server_type == RTSP_SERVER_REAL)

                    ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);

                rtsp_st = s->streams[s->nb_streams - 1]->priv_data;

                if (rtsp_st->dynamic_handler &&

                    rtsp_st->dynamic_handler->parse_sdp_a_line)

                    rtsp_st->dynamic_handler->parse_sdp_a_line(s,

                        s->nb_streams - 1,

                        rtsp_st->dynamic_protocol_context, buf);

            }

        }

        break;

    }

}

static int sdp_parse(AVFormatContext *s, const char *content)

{

    const char *p;

    int letter;

    /* Some SDP lines, particularly for Realmedia or ASF RTSP streams,

     * contain long SDP lines containing complete ASF Headers (several

     * kB) or arrays of MDPR (RM stream descriptor) headers plus

     * "rulebooks" describing their properties. Therefore, the SDP line

     * buffer is large.

     *

     * The Vorbis FMTP line can be up to 16KB - see sdp_parse_fmtp. */

    char buf[16384], *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;

}

/* close and free RTSP streams */

void ff_rtsp_close_streams(AVFormatContext *s)

{

    RTSPState *rt = s->priv_data;

    int i;

    RTSPStream *rtsp_st;

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

        rtsp_st = rt->rtsp_streams[i];

        if (rtsp_st) {

            if (rtsp_st->transport_priv) {

                if (s->oformat) {

                    AVFormatContext *rtpctx = rtsp_st->transport_priv;

                    av_write_trailer(rtpctx);

                    if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {

                        uint8_t *ptr;

                        url_close_dyn_buf(rtpctx->pb, &ptr);

                        av_free(ptr);

                    } else {

                    url_fclose(rtpctx->pb);

                    }

                    av_metadata_free(&rtpctx->streams[0]->metadata);

                    av_metadata_free(&rtpctx->metadata);

                    av_free(rtpctx->streams[0]);

                    av_free(rtpctx);

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

                    ff_rdt_parse_close(rtsp_st->transport_priv);

                else

                    rtp_parse_close(rtsp_st->transport_priv);

            }

            if (rtsp_st->rtp_handle)

                url_close(rtsp_st->rtp_handle);

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

                rtsp_st->dynamic_handler->close(

                    rtsp_st->dynamic_protocol_context);

        }

    }

    av_free(rt->rtsp_streams);

    if (rt->asf_ctx) {

        av_close_input_stream (rt->asf_ctx);

        rt->asf_ctx = NULL;

    }

    av_freep(&rt->auth_b64);

}

static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st,

                               URLContext *handle)

{

    RTSPState *rt = s->priv_data;

    AVFormatContext *rtpctx;

    int ret;

    AVOutputFormat *rtp_format = av_guess_format("rtp", NULL, NULL);

    if (!rtp_format)

        return NULL;

    /* Allocate an AVFormatContext for each output stream */

    rtpctx = avformat_alloc_context();

    if (!rtpctx)

        return NULL;

    rtpctx->oformat = rtp_format;

    if (!av_new_stream(rtpctx, 0)) {

        av_free(rtpctx);

        return NULL;

    }

    /* Copy the max delay setting; the rtp muxer reads this. */

    rtpctx->max_delay = s->max_delay;

    /* Copy other stream parameters. */

    rtpctx->streams[0]->sample_aspect_ratio = st->sample_aspect_ratio;

    /* Set the synchronized start time. */

    rtpctx->start_time_realtime = rt->start_time;

    /* Remove the local codec, link to the original codec

     * context instead, to give the rtp muxer access to

     * codec parameters. */

    av_free(rtpctx->streams[0]->codec);

    rtpctx->streams[0]->codec = st->codec;

    if (handle) {

    url_fdopen(&rtpctx->pb, handle);

    } else

        url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);

    ret = av_write_header(rtpctx);

    if (ret) {

        if (handle) {

        url_fclose(rtpctx->pb);

        } else {

            uint8_t *ptr;

            url_close_dyn_buf(rtpctx->pb, &ptr);

            av_free(ptr);

        }

        av_free(rtpctx->streams[0]);

        av_free(rtpctx);

        return NULL;

    }

    /* Copy the RTP AVStream timebase back to the original AVStream */

    st->time_base = rtpctx->streams[0]->time_base;

    return rtpctx;

}

static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)

{

    RTSPState *rt = s->priv_data;

    AVStream *st = NULL;

    /* open the RTP context */

    if (rtsp_st->stream_index >= 0)

        st = s->streams[rtsp_st->stream_index];

    if (!st)

        s->ctx_flags |= AVFMTCTX_NOHEADER;

    if (s->oformat) {

        rtsp_st->transport_priv = rtsp_rtp_mux_open(s, st, rtsp_st->rtp_handle);

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

        rtsp_st->rtp_handle = NULL;

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

        rtsp_st->transport_priv = ff_rdt_parse_open(s, st->index,

                                            rtsp_st->dynamic_protocol_context,

                                            rtsp_st->dynamic_handler);

    else

        rtsp_st->transport_priv = rtp_parse_open(s, st, rtsp_st->rtp_handle,

                                         rtsp_st->sdp_payload_type,

                                         &rtsp_st->rtp_payload_data);

    if (!rtsp_st->transport_priv) {

         return AVERROR(ENOMEM);

    } else if (rt->transport != RTSP_TRANSPORT_RDT) {

        if (rtsp_st->dynamic_handler) {

            rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,

                                           rtsp_st->dynamic_protocol_context,

                                           rtsp_st->dynamic_handler);

        }

    }

    return 0;

}

#if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER

static int rtsp_probe(AVProbeData *p)

{

    if (av_strstart(p->filename, "rtsp:", NULL))

        return AVPROBE_SCORE_MAX;

    return 0;

}

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

{

    const char *p;

    int v;

    p = *pp;

    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(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 (;;) {

        skip_spaces(&p);

        if (*p == '\0')

            break;

        th = &reply->transports[reply->nb_transports];

        get_word_sep(transport_protocol, sizeof(transport_protocol),

                     "/", &p);

        if (!strcasecmp (transport_protocol, "rtp")) {

            get_word_sep(profile, sizeof(profile), "/;,", &p);

            lower_transport[0] = '\0';

            /* rtp/avp/<protocol> */

            if (*p == '/') {

                get_word_sep(lower_transport, sizeof(lower_transport),

                             ";,", &p);

            }

            th->transport = RTSP_TRANSPORT_RTP;

        } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||

                   !strcasecmp (transport_protocol, "x-real-rdt")) {

            /* x-pn-tng/<protocol> */

            get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);

            profile[0] = '\0';

            th->transport = RTSP_TRANSPORT_RDT;

        }

        if (!strcasecmp(lower_transport, "TCP"))

            th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;

        else

            th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;

        if (*p == ';')

            p++;

        /* get each parameter */

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

            get_word_sep(parameter, sizeof(parameter), "=;,", &p);

            if (!strcmp(parameter, "port")) {

                if (*p == '=') {

                    p++;

                    rtsp_parse_range(&th->port_min, &th->port_max, &p);

                }

            } else if (!strcmp(parameter, "client_port")) {

                if (*p == '=') {

                    p++;

                    rtsp_parse_range(&th->client_port_min,

                                     &th->client_port_max, &p);

                }

            } else if (!strcmp(parameter, "server_port")) {

                if (*p == '=') {

                    p++;

                    rtsp_parse_range(&th->server_port_min,

                                     &th->server_port_max, &p);

                }

            } else if (!strcmp(parameter, "interleaved")) {

                if (*p == '=') {

                    p++;

                    rtsp_parse_range(&th->interleaved_min,

                                     &th->interleaved_max, &p);

                }

            } else if (!strcmp(parameter, "multicast")) {

                if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)

                    th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;

            } else if (!strcmp(parameter, "ttl")) {

                if (*p == '=') {

                    p++;

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

                }

            } else if (!strcmp(parameter, "destination")) {

                struct in_addr ipaddr;

                if (*p == '=') {

                    p++;

                    get_word_sep(buf, sizeof(buf), ";,", &p);

                    if (ff_inet_aton(buf, &ipaddr))

                        th->destination = ntohl(ipaddr.s_addr);

                }

            }

            while (*p != ';' && *p != '\0' && *p != ',')

                p++;

            if (*p == ';')

                p++;

        }

        if (*p == ',')

            p++;

        reply->nb_transports++;

    }

}

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

{

    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)) {

        skip_spaces(&p);

        av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));

    } else if (av_stristart(p, "Server:", &p)) {

        skip_spaces(&p);

        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)) {

        skip_spaces(&p);

        av_strlcpy(reply->location, p , sizeof(reply->location));

    }

}

/* skip a RTP/TCP interleaved packet */

void ff_rtsp_skip_packet(AVFormatContext *s)

{

    RTSPState *rt = s->priv_data;

    int ret, len, len1;

    uint8_t buf[1024];

    ret = url_read_complete(rt->rtsp_hd, buf, 3);

    if (ret != 3)

        return;

    len = AV_RB16(buf + 1);

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

    /* skip payload */

    while (len > 0) {

        len1 = len;

        if (len1 > sizeof(buf))

            len1 = sizeof(buf);

        ret = url_read_complete(rt->rtsp_hd, buf, len1);

        if (ret != len1)

            return;

        len -= len1;

    }

}

int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,

                       unsigned char **content_ptr,

                       int return_on_interleaved_data)

{

    RTSPState *rt = s->priv_data;

    char buf[4096], buf1[1024], *q;

    unsigned char ch;

    const char *p;

    int ret, content_length, line_count = 0;

    unsigned char *content = NULL;

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

    /* parse reply (XXX: use buffers) */

    rt->last_reply[0] = '\0';

    for (;;) {

        q = buf;

        for (;;) {

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

#ifdef DEBUG_RTP_TCP

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

#endif

            if (ret != 1)

                return -1;

            if (ch == '\n')

                break;

            if (ch == '$') {

                /* XXX: only parse it if first char on line ? */

                if (return_on_interleaved_data) {

                    return 1;

                } else

                    ff_rtsp_skip_packet(s);

            } else if (ch != '\r') {

                if ((q - buf) < sizeof(buf) - 1)

                    *q++ = ch;

            }

        }

        *q = '\0';

        dprintf(s, "line='%s'\n", buf);

        /* test if last line */

        if (buf[0] == '\0')

            break;

        p = buf;

        if (line_count == 0) {

            /* get reply code */

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

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

            reply->status_code = atoi(buf1);

        } else {

            ff_rtsp_parse_line(reply, p);

            av_strlcat(rt->last_reply, p,    sizeof(rt->last_reply));

            av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));

        }

        line_count++;

    }

    if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')

        av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));

    content_length = reply->content_length;

    if (content_length > 0) {

        /* leave some room for a trailing '\0' (useful for simple parsing) */

        content = av_malloc(content_length + 1);

        (void)url_read_complete(rt->rtsp_hd, content, content_length);

        content[content_length] = '\0';

    }

    if (content_ptr)

        *content_ptr = content;

    else

        av_free(content);

    /* 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;

}

void ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,

                                         const char *cmd,

                                         const unsigned char *send_content,

                                         int send_content_length)

{

    RTSPState *rt = s->priv_data;

    char buf[4096], buf1[1024];

    rt->seq++;

    av_strlcpy(buf, cmd, sizeof(buf));

    snprintf(buf1, sizeof(buf1), "CSeq: %d\r\n", rt->seq);

    av_strlcat(buf, buf1, sizeof(buf));

    if (rt->session_id[0] != '\0' && !strstr(cmd, "\nIf-Match:")) {

        snprintf(buf1, sizeof(buf1), "Session: %s\r\n", rt->session_id);

        av_strlcat(buf, buf1, sizeof(buf));

    }

    if (rt->auth_b64)

        av_strlcatf(buf, sizeof(buf),

                    "Authorization: Basic %s\r\n",

                    rt->auth_b64);

    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));

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

    url_write(rt->rtsp_hd, buf, strlen(buf));

    if (send_content_length > 0 && send_content)

        url_write(rt->rtsp_hd, send_content, send_content_length);

    rt->last_cmd_time = av_gettime();

}

void ff_rtsp_send_cmd_async(AVFormatContext *s, const char *cmd)

{

    ff_rtsp_send_cmd_with_content_async(s, cmd, NULL, 0);

}

void ff_rtsp_send_cmd(AVFormatContext *s,

                      const char *cmd, RTSPMessageHeader *reply,

                      unsigned char **content_ptr)

{

    ff_rtsp_send_cmd_async(s, cmd);

    ff_rtsp_read_reply(s, reply, content_ptr, 0);

}

void ff_rtsp_send_cmd_with_content(AVFormatContext *s,

                                const char *cmd,

                                RTSPMessageHeader *reply,

                                unsigned char **content_ptr,

                                const unsigned char *send_content,

                                int send_content_length)

{

    ff_rtsp_send_cmd_with_content_async(s, cmd, send_content, send_content_length);

    ff_rtsp_read_reply(s, reply, content_ptr, 0);

}

/**

 * @returns 0 on success, <0 on error, 1 if protocol is unavailable.

 */

static int make_setup_request(AVFormatContext *s, const char *host, int port,

                              int lower_transport, const char *real_challenge)

{

    RTSPState *rt = s->priv_data;

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

    RTSPStream *rtsp_st;

    RTSPMessageHeader reply1, *reply = &reply1;

    char cmd[2048];

    const char *trans_pref;

    if (rt->transport == RTSP_TRANSPORT_RDT)

        trans_pref = "x-pn-tng";

    else

        trans_pref = "RTP/AVP";

    /* default timeout: 1 minute */

    rt->timeout = 60;

    /* for each stream, make the setup request */

    /* XXX: we assume the same server is used for the control of each

     * RTSP stream */

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

        char transport[2048];

        /**

         * WMS serves all UDP data over a single connection, the RTX, which

         * isn't necessarily the first in the SDP but has to be the first

         * to be set up, else the second/third SETUP will fail with a 461.

         */

        if (lower_transport == RTSP_LOWER_TRANSPORT_UDP &&

             rt->server_type == RTSP_SERVER_WMS) {

            if (i == 0) {

                /* rtx first */

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

                    int len = strlen(rt->rtsp_streams[rtx]->control_url);

                    if (len >= 4 &&

                        !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,

                                "/rtx"))

                        break;

                }

                if (rtx == rt->nb_rtsp_streams)

                    return -1; /* no RTX found */

                rtsp_st = rt->rtsp_streams[rtx];

            } else

                rtsp_st = rt->rtsp_streams[i > rtx ? i : i - 1];

        } else

            rtsp_st = rt->rtsp_streams[i];

        /* RTP/UDP */

        if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {

            char buf[256];

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

                port = reply->transports[0].client_port_min;

                goto have_port;

            }

            /* first try in specified port range */

            if (RTSP_RTP_PORT_MIN != 0) {

                while (j <= RTSP_RTP_PORT_MAX) {

                    ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,

                                "?localport=%d", j);

                    /* we will use two ports per rtp stream (rtp and rtcp) */

                    j += 2;

                    if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)

                        goto rtp_opened;

                }

            }

#if 0

            /* then try on any port */

            if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {

                err = AVERROR_INVALIDDATA;

                goto fail;

            }

#endif

        rtp_opened:

            port = rtp_get_local_port(rtsp_st->rtp_handle);

        have_port:

            snprintf(transport, sizeof(transport) - 1,

                     "%s/UDP;", trans_pref);

            if (rt->server_type != RTSP_SERVER_REAL)

                av_strlcat(transport, "unicast;", sizeof(transport));

            av_strlcatf(transport, sizeof(transport),

                     "client_port=%d", port);

            if (rt->transport == RTSP_TRANSPORT_RTP &&

                !(rt->server_type == RTSP_SERVER_WMS && i > 0))

                av_strlcatf(transport, sizeof(transport), "-%d", port + 1);

        }

        /* RTP/TCP */

        else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {

            /** For WMS streams, the application streams are only used for

             * UDP. When trying to set it up for TCP streams, the server

             * will return an error. Therefore, we skip those streams. */

            if (rt->server_type == RTSP_SERVER_WMS &&

                s->streams[rtsp_st->stream_index]->codec->codec_type ==

                    CODEC_TYPE_DATA)

                continue;

            snprintf(transport, sizeof(transport) - 1,

                     "%s/TCP;", trans_pref);

            if (rt->server_type == RTSP_SERVER_WMS)

                av_strlcat(transport, "unicast;", sizeof(transport));

            av_strlcatf(transport, sizeof(transport),

                        "interleaved=%d-%d",

                        interleave, interleave + 1);

            interleave += 2;

        }

        else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {

            snprintf(transport, sizeof(transport) - 1,

                     "%s/UDP;multicast", trans_pref);

        }

        if (s->oformat) {

            av_strlcat(transport, ";mode=receive", sizeof(transport));

        } else if (rt->server_type == RTSP_SERVER_REAL ||

                   rt->server_type == RTSP_SERVER_WMS)

            av_strlcat(transport, ";mode=play", sizeof(transport));

        snprintf(cmd, sizeof(cmd),

                 "SETUP %s RTSP/1.0\r\n"

                 "Transport: %s\r\n",

                 rtsp_st->control_url, transport);

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

            char real_res[41], real_csum[9];

            ff_rdt_calc_response_and_checksum(real_res, real_csum,

                                              real_challenge);

            av_strlcatf(cmd, sizeof(cmd),

                        "If-Match: %s\r\n"

                        "RealChallenge2: %s, sd=%s\r\n",

                        rt->session_id, real_res, real_csum);

        }

        ff_rtsp_send_cmd(s, cmd, reply, NULL);

        if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {

            err = 1;

            goto fail;

        } else if (reply->status_code != RTSP_STATUS_OK ||

                   reply->nb_transports != 1) {

            err = AVERROR_INVALIDDATA;

            goto fail;

        }

        /* XXX: same protocol for all streams is required */

        if (i > 0) {

            if (reply->transports[0].lower_transport != rt->lower_transport ||

                reply->transports[0].transport != rt->transport) {

                err = AVERROR_INVALIDDATA;

                goto fail;

            }

        } else {

            rt->lower_transport = reply->transports[0].lower_transport;

            rt->transport = reply->transports[0].transport;

        }

        /* close RTP connection if not choosen */

        if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&

            (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {

            url_close(rtsp_st->rtp_handle);

            rtsp_st->rtp_handle = NULL;

        }

        switch(reply->transports[0].lower_transport) {

        case RTSP_LOWER_TRANSPORT_TCP:

            rtsp_st->interleaved_min = reply->transports[0].interleaved_min;

            rtsp_st->interleaved_max = reply->transports[0].interleaved_max;

            break;

        case RTSP_LOWER_TRANSPORT_UDP: {

            char url[1024];

            /* XXX: also use address if specified */

            ff_url_join(url, sizeof(url), "rtp", NULL, host,

                        reply->transports[0].server_port_min, NULL);

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

                rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {

                err = AVERROR_INVALIDDATA;

                goto fail;

            }

            /* Try to initialize the connection state in a

             * potential NAT router by sending dummy packets.

             * RTP/RTCP dummy packets are used for RDT, too.

             */

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

                rtp_send_punch_packets(rtsp_st->rtp_handle);

            break;

        }

        case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {

            char url[1024];

            struct in_addr in;

            int port, ttl;

            if (reply->transports[0].destination) {

                in.s_addr = htonl(reply->transports[0].destination);

                port      = reply->transports[0].port_min;

                ttl       = reply->transports[0].ttl;

            } else {

                in        = rtsp_st->sdp_ip;

                port      = rtsp_st->sdp_port;

                ttl       = rtsp_st->sdp_ttl;

            }

            ff_url_join(url, sizeof(url), "rtp", NULL, inet_ntoa(in),

                        port, "?ttl=%d", ttl);

            if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {

                err = AVERROR_INVALIDDATA;

                goto fail;

            }

            break;

        }

        }

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

            goto fail;

    }

    if (reply->timeout > 0)

        rt->timeout = reply->timeout;

    if (rt->server_type == RTSP_SERVER_REAL)

        rt->need_subscription = 1;

    return 0;

fail:

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

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

            url_close(rt->rtsp_streams[i]->rtp_handle);

            rt->rtsp_streams[i]->rtp_handle = NULL;

        }