~ppsspp/ppsspp/ffmpeg-upstream

/*

 * RTSP/SDP client

 * Copyright (c) 2002 Fabrice Bellard

 *

 * This file is part of FFmpeg.

 *

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

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

 * License as published by the Free Software Foundation; either

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

 *

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

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

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

 * Lesser General Public License for more details.

 *

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

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

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

 */

#include "libavutil/avassert.h"

#include "libavutil/base64.h"

#include "libavutil/avstring.h"

#include "libavutil/intreadwrite.h"

#include "libavutil/mathematics.h"

#include "libavutil/parseutils.h"

#include "libavutil/random_seed.h"

#include "libavutil/dict.h"

#include "libavutil/opt.h"

#include "libavutil/time.h"

#include "avformat.h"

#include "avio_internal.h"

#if HAVE_POLL_H

#include <poll.h>

#endif

#include "internal.h"

#include "network.h"

#include "os_support.h"

#include "http.h"

#include "rtsp.h"

#include "rtpdec.h"

#include "rtpproto.h"

#include "rdt.h"

#include "rtpdec_formats.h"

#include "rtpenc_chain.h"

#include "url.h"

#include "rtpenc.h"

#include "mpegts.h"

/* Timeout values for socket poll, in ms,

 * and read_packet(), in seconds  */

#define POLL_TIMEOUT_MS 100

#define READ_PACKET_TIMEOUT_S 10

#define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS

#define SDP_MAX_SIZE 16384

#define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH

#define DEFAULT_REORDERING_DELAY 100000

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

#define DEC AV_OPT_FLAG_DECODING_PARAM

#define ENC AV_OPT_FLAG_ENCODING_PARAM

#define RTSP_FLAG_OPTS(name, longname) \

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

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

#define RTSP_MEDIATYPE_OPTS(name, longname) \

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

    { "video", "Video", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \

    { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \

    { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }, \

    { "subtitle", "Subtitle", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_SUBTITLE}, 0, 0, DEC, "allowed_media_types" }

#define COMMON_OPTS() \

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

    { "buffer_size",        "Underlying protocol send/receive buffer size",                  OFFSET(buffer_size),           AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC|ENC } \

const AVOption ff_rtsp_options[] = {

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

    FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),

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

    { "udp", "UDP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP}, 0, 0, DEC|ENC, "rtsp_transport" }, \

    { "tcp", "TCP", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_TCP}, 0, 0, DEC|ENC, "rtsp_transport" }, \

    { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },

    { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {.i64 = (1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },

    RTSP_FLAG_OPTS("rtsp_flags", "set RTSP flags"),

    { "listen", "wait for incoming connections", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_LISTEN}, 0, 0, DEC, "rtsp_flags" },

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

    RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),

    { "min_port", "set minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },

    { "max_port", "set maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },

    { "timeout", "set maximum timeout (in seconds) to wait for incoming connections (-1 is infinite, imply flag listen)", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },

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

    COMMON_OPTS(),

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

    { NULL },

};

static const AVOption sdp_options[] = {

    RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),

    { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },

    { "rtcp_to_source", "send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },

    RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),

    COMMON_OPTS(),

    { NULL },

};

static const AVOption rtp_options[] = {

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

    COMMON_OPTS(),

    { NULL },

};

static AVDictionary *map_to_opts(RTSPState *rt)

{

    AVDictionary *opts = NULL;

    char buf[256];

    snprintf(buf, sizeof(buf), "%d", rt->buffer_size);

    av_dict_set(&opts, "buffer_size", buf, 0);

    return opts;

}

static void get_word_until_chars(char *buf, int buf_size,

                                 const char *sep, const char **pp)

{

    const char *p;

    char *q;

    p = *pp;

    p += strspn(p, SPACE_CHARS);

    q = buf;

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

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

            *q++ = *p;

        p++;

    }

    if (buf_size > 0)

        *q = '\0';

    *pp = p;

}

static void get_word_sep(char *buf, int buf_size, const char *sep,

                         const char **pp)

{

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

    get_word_until_chars(buf, buf_size, sep, pp);

}

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

{

    get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);

}

/** Parse a string p in the form of Range:npt=xx-xx, and determine the start

 *  and end time.

 *  Used for seeking in the rtp stream.

 */

static void rtsp_parse_range_npt(const char *p, int64_t *start, int64_t *end)

{

    char buf[256];

    p += strspn(p, SPACE_CHARS);

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

        return;

    *start = AV_NOPTS_VALUE;

    *end = AV_NOPTS_VALUE;

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

    if (av_parse_time(start, buf, 1) < 0)

        return;

    if (*p == '-') {

        p++;

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

        if (av_parse_time(end, buf, 1) < 0)

            av_log(NULL, AV_LOG_DEBUG, "Failed to parse interval end specification '%s'\n", buf);

    }

}

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

{

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

    hints.ai_flags = AI_NUMERICHOST;

    if (getaddrinfo(buf, NULL, &hints, &ai))

        return -1;

    memcpy(sock, ai->ai_addr, FFMIN(sizeof(*sock), ai->ai_addrlen));

    freeaddrinfo(ai);

    return 0;

}

#if CONFIG_RTPDEC

static void init_rtp_handler(RTPDynamicProtocolHandler *handler,

                             RTSPStream *rtsp_st, AVStream *st)

{

    AVCodecContext *codec = st ? st->codec : NULL;

    if (!handler)

        return;

    if (codec)

        codec->codec_id          = handler->codec_id;

    rtsp_st->dynamic_handler = handler;

    if (st)

        st->need_parsing = handler->need_parsing;

    if (handler->priv_data_size) {

        rtsp_st->dynamic_protocol_context = av_mallocz(handler->priv_data_size);

        if (!rtsp_st->dynamic_protocol_context)

            rtsp_st->dynamic_handler = NULL;

    }

}

static void finalize_rtp_handler_init(AVFormatContext *s, RTSPStream *rtsp_st,

                                      AVStream *st)

{

    if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) {

        int ret = rtsp_st->dynamic_handler->init(s, st ? st->index : -1,

                                                 rtsp_st->dynamic_protocol_context);

        if (ret < 0) {

            if (rtsp_st->dynamic_protocol_context) {

                if (rtsp_st->dynamic_handler->close)

                    rtsp_st->dynamic_handler->close(

                        rtsp_st->dynamic_protocol_context);

                av_free(rtsp_st->dynamic_protocol_context);

            }

            rtsp_st->dynamic_protocol_context = NULL;

            rtsp_st->dynamic_handler = NULL;

        }

    }

}

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

static int sdp_parse_rtpmap(AVFormatContext *s,

                            AVStream *st, RTSPStream *rtsp_st,

                            int payload_type, const char *p)

{

    AVCodecContext *codec = st->codec;

    char buf[256];

    int i;

    AVCodec *c;

    const char *c_name;

    /* See if we can handle this kind of payload.

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

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

     * have a trailing space. */

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

    if (payload_type < RTP_PT_PRIVATE) {

        /* We are in a standard case

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

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

    }

    if (codec->codec_id == AV_CODEC_ID_NONE) {

        RTPDynamicProtocolHandler *handler =

            ff_rtp_handler_find_by_name(buf, codec->codec_type);

        init_rtp_handler(handler, rtsp_st, st);

        /* If no dynamic handler was found, check with the list of standard

         * allocated types, if such a stream for some reason happens to

         * use a private payload type. This isn't handled in rtpdec.c, since

         * the format name from the rtpmap line never is passed into rtpdec. */

        if (!rtsp_st->dynamic_handler)

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

    }

    c = avcodec_find_decoder(codec->codec_id);

    if (c && c->name)

        c_name = c->name;

    else

        c_name = "(null)";

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

    i = atoi(buf);

    switch (codec->codec_type) {

    case AVMEDIA_TYPE_AUDIO:

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

        codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;

        codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;

        if (i > 0) {

            codec->sample_rate = i;

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

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

            i = atoi(buf);

            if (i > 0)

                codec->channels = i;

        }

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

               codec->sample_rate);

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

               codec->channels);

        break;

    case AVMEDIA_TYPE_VIDEO:

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

        if (i > 0)

            avpriv_set_pts_info(st, 32, 1, i);

        break;

    default:

        break;

    }

    finalize_rtp_handler_init(s, rtsp_st, st);

    return 0;

}

/* parse the attribute line from the fmtp a line of an sdp response. This

 * is broken out as a function because it is used in rtp_h264.c, which is

 * forthcoming. */

int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,

                                char *value, int value_size)

{

    *p += strspn(*p, SPACE_CHARS);

    if (**p) {

        get_word_sep(attr, attr_size, "=", p);

        if (**p == '=')

            (*p)++;

        get_word_sep(value, value_size, ";", p);

        if (**p == ';')

            (*p)++;

        return 1;

    }

    return 0;

}

typedef struct SDPParseState {

    /* SDP only */

    struct sockaddr_storage default_ip;

    int            default_ttl;

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

    int nb_default_include_source_addrs; /**< Number of source-specific multicast include source IP address (from SDP content) */

    struct RTSPSource **default_include_source_addrs; /**< Source-specific multicast include source IP address (from SDP content) */

    int nb_default_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP address (from SDP content) */

    struct RTSPSource **default_exclude_source_addrs; /**< Source-specific multicast exclude source IP address (from SDP content) */

    int seen_rtpmap;

    int seen_fmtp;

    char delayed_fmtp[2048];

} SDPParseState;

static void copy_default_source_addrs(struct RTSPSource **addrs, int count,

                                      struct RTSPSource ***dest, int *dest_count)

{

    RTSPSource *rtsp_src, *rtsp_src2;

    int i;

    for (i = 0; i < count; i++) {

        rtsp_src = addrs[i];

        rtsp_src2 = av_malloc(sizeof(*rtsp_src2));

        if (!rtsp_src2)

            continue;

        memcpy(rtsp_src2, rtsp_src, sizeof(*rtsp_src));

        dynarray_add(dest, dest_count, rtsp_src2);

    }

}

static void parse_fmtp(AVFormatContext *s, RTSPState *rt,

                       int payload_type, const char *line)

{

    int i;

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

        RTSPStream *rtsp_st = rt->rtsp_streams[i];

        if (rtsp_st->sdp_payload_type == payload_type &&

            rtsp_st->dynamic_handler &&

            rtsp_st->dynamic_handler->parse_sdp_a_line) {

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

                rtsp_st->dynamic_protocol_context, line);

        }

    }

}

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

                           int letter, const char *buf)

{

    RTSPState *rt = s->priv_data;

    char buf1[64], st_type[64];

    const char *p;

    enum AVMediaType codec_type;

    int payload_type;

    AVStream *st;

    RTSPStream *rtsp_st;

    RTSPSource *rtsp_src;

    struct sockaddr_storage sdp_ip;

    int ttl;

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

    p = buf;

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

        return;

    switch (letter) {

    case 'c':

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

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

            return;

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

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

            return;

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

        if (get_sockaddr(buf1, &sdp_ip))

            return;

        ttl = 16;

        if (*p == '/') {

            p++;

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

            ttl = atoi(buf1);

        }

        if (s->nb_streams == 0) {

            s1->default_ip = sdp_ip;

            s1->default_ttl = ttl;

        } else {

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

            rtsp_st->sdp_ip = sdp_ip;

            rtsp_st->sdp_ttl = ttl;

        }

        break;

    case 's':

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

        break;

    case 'i':

        if (s->nb_streams == 0) {

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

            break;

        }

        break;

    case 'm':

        /* new stream */

        s1->skip_media  = 0;

        s1->seen_fmtp   = 0;

        s1->seen_rtpmap = 0;

        codec_type = AVMEDIA_TYPE_UNKNOWN;

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

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

            codec_type = AVMEDIA_TYPE_AUDIO;

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

            codec_type = AVMEDIA_TYPE_VIDEO;

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

            codec_type = AVMEDIA_TYPE_DATA;

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

            codec_type = AVMEDIA_TYPE_SUBTITLE;

        }

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

            s1->skip_media = 1;

            return;

        }

        rtsp_st = av_mallocz(sizeof(RTSPStream));

        if (!rtsp_st)

            return;

        rtsp_st->stream_index = -1;

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

        rtsp_st->sdp_ip = s1->default_ip;

        rtsp_st->sdp_ttl = s1->default_ttl;

        copy_default_source_addrs(s1->default_include_source_addrs,

                                  s1->nb_default_include_source_addrs,

                                  &rtsp_st->include_source_addrs,

                                  &rtsp_st->nb_include_source_addrs);

        copy_default_source_addrs(s1->default_exclude_source_addrs,

                                  s1->nb_default_exclude_source_addrs,

                                  &rtsp_st->exclude_source_addrs,

                                  &rtsp_st->nb_exclude_source_addrs);

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

        rtsp_st->sdp_port = atoi(buf1);

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

        if (!strcmp(buf1, "udp"))

            rt->transport = RTSP_TRANSPORT_RAW;

        else if (strstr(buf1, "/AVPF") || strstr(buf1, "/SAVPF"))

            rtsp_st->feedback = 1;

        /* XXX: handle list of formats */

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

        rtsp_st->sdp_payload_type = atoi(buf1);

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

            /* no corresponding stream */

            if (rt->transport == RTSP_TRANSPORT_RAW) {

                if (CONFIG_RTPDEC && !rt->ts)

                    rt->ts = avpriv_mpegts_parse_open(s);

            } else {

                RTPDynamicProtocolHandler *handler;

                handler = ff_rtp_handler_find_by_id(

                              rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);

                init_rtp_handler(handler, rtsp_st, NULL);

                finalize_rtp_handler_init(s, rtsp_st, NULL);

            }

        } else if (rt->server_type == RTSP_SERVER_WMS &&

                   codec_type == AVMEDIA_TYPE_DATA) {

            /* RTX stream, a stream that carries all the other actual

             * audio/video streams. Don't expose this to the callers. */

        } else {

            st = avformat_new_stream(s, NULL);

            if (!st)

                return;

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

            rtsp_st->stream_index = st->index;

            st->codec->codec_type = codec_type;

            if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {

                RTPDynamicProtocolHandler *handler;

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

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

                if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&

                    st->codec->sample_rate > 0)

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

                /* Even static payload types may need a custom depacketizer */

                handler = ff_rtp_handler_find_by_id(

                              rtsp_st->sdp_payload_type, st->codec->codec_type);

                init_rtp_handler(handler, rtsp_st, st);

                finalize_rtp_handler_init(s, rtsp_st, st);

            }

            if (rt->default_lang[0])

                av_dict_set(&st->metadata, "language", rt->default_lang, 0);

        }

        /* put a default control url */

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

                   sizeof(rtsp_st->control_url));

        break;

    case 'a':

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

            if (s->nb_streams == 0) {

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

                    av_strlcpy(rt->control_uri, p,

                               sizeof(rt->control_uri));

            } else {

                char proto[32];

                /* get the control url */

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

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

                av_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,

                             NULL, NULL, 0, p);

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

                    /* relative control URL */

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

                    av_strlcat(rtsp_st->control_url, "/",

                               sizeof(rtsp_st->control_url));

                    av_strlcat(rtsp_st->control_url, p,

                               sizeof(rtsp_st->control_url));

                } else

                    av_strlcpy(rtsp_st->control_url, p,

                               sizeof(rtsp_st->control_url));

            }

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

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

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

            payload_type = atoi(buf1);

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

            if (rtsp_st->stream_index >= 0) {

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

                sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);

            }

            s1->seen_rtpmap = 1;

            if (s1->seen_fmtp) {

                parse_fmtp(s, rt, payload_type, s1->delayed_fmtp);

            }

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

                   av_strstart(p, "framesize:", &p)) {

            // let dynamic protocol handlers have a stab at the line.

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

            payload_type = atoi(buf1);

            if (s1->seen_rtpmap) {

                parse_fmtp(s, rt, payload_type, buf);

            } else {

                s1->seen_fmtp = 1;

                av_strlcpy(s1->delayed_fmtp, buf, sizeof(s1->delayed_fmtp));

            }

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

            int64_t start, end;

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

            rtsp_parse_range_npt(p, &start, &end);

            s->start_time = start;

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

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

                            AV_NOPTS_VALUE : end - start;

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

            if (s->nb_streams > 0) {

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

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

                if (rtsp_st->stream_index >= 0) {

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

                    av_dict_set(&st->metadata, "language", buf1, 0);

                }

            } else

                get_word(rt->default_lang, sizeof(rt->default_lang), &p);

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

            if (atoi(p) == 1)

                rt->transport = RTSP_TRANSPORT_RDT;

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

                   s->nb_streams > 0) {

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

            st->codec->sample_rate = atoi(p);

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

            // RFC 4568

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

            get_word(buf1, sizeof(buf1), &p); // ignore tag

            get_word(rtsp_st->crypto_suite, sizeof(rtsp_st->crypto_suite), &p);

            p += strspn(p, SPACE_CHARS);

            if (av_strstart(p, "inline:", &p))

                get_word(rtsp_st->crypto_params, sizeof(rtsp_st->crypto_params), &p);

        } else if (av_strstart(p, "source-filter:", &p)) {

            int exclude = 0;

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

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

                return;

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

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

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

                return;

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

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

                return;

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

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

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

                rtsp_src = av_mallocz(sizeof(*rtsp_src));

                if (!rtsp_src)

                    return;

                get_word(rtsp_src->addr, sizeof(rtsp_src->addr), &p);

                if (exclude) {

                    if (s->nb_streams == 0) {

                        dynarray_add(&s1->default_exclude_source_addrs, &s1->nb_default_exclude_source_addrs, rtsp_src);

                    } else {

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

                        dynarray_add(&rtsp_st->exclude_source_addrs, &rtsp_st->nb_exclude_source_addrs, rtsp_src);

                    }

                } else {

                    if (s->nb_streams == 0) {

                        dynarray_add(&s1->default_include_source_addrs, &s1->nb_default_include_source_addrs, rtsp_src);

                    } else {

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

                        dynarray_add(&rtsp_st->include_source_addrs, &rtsp_st->nb_include_source_addrs, rtsp_src);

                    }

                }

            }

        } else {

            if (rt->server_type == RTSP_SERVER_WMS)

                ff_wms_parse_sdp_a_line(s, p);

            if (s->nb_streams > 0) {

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

                if (rt->server_type == RTSP_SERVER_REAL)

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

                if (rtsp_st->dynamic_handler &&

                    rtsp_st->dynamic_handler->parse_sdp_a_line)

                    rtsp_st->dynamic_handler->parse_sdp_a_line(s,

                        rtsp_st->stream_index,

                        rtsp_st->dynamic_protocol_context, buf);

            }

        }

        break;

    }

}

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

{

    RTSPState *rt = s->priv_data;

    const char *p;

    int letter, i;

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

     * contain long SDP lines containing complete ASF Headers (several

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

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

     * buffer is large.

     *

     * The Vorbis FMTP line can be up to 16KB - see xiph_parse_sdp_line

     * in rtpdec_xiph.c. */

    char buf[16384], *q;

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

    p = content;

    for (;;) {

        p += strspn(p, SPACE_CHARS);