rtsp.c 67.4 KB
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/*
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 * RTSP/SDP client
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 * Copyright (c) 2002 Fabrice Bellard
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 *
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 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * 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
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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#include "libavutil/base64.h"
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#include "libavutil/avstring.h"
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#include "libavutil/intreadwrite.h"
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#include "avformat.h"

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#include <sys/time.h>
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#if HAVE_SYS_SELECT_H
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#include <sys/select.h>
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#endif
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#include <strings.h>
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#include "internal.h"
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#include "network.h"
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#include "os_support.h"
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#include "rtsp.h"
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#include "rtpdec.h"
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#include "rdt.h"
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#include "rtpdec_asf.h"
#include "rtpdec_vorbis.h"
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//#define DEBUG
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//#define DEBUG_RTP_TCP
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#if LIBAVFORMAT_VERSION_INT < (53 << 16)
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int rtsp_default_protocols = (1 << RTSP_LOWER_TRANSPORT_UDP);
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#endif
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#define SPACE_CHARS " \t\r\n"
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/* 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)
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static void skip_spaces(const char **pp)
{
    const char *p;
    p = *pp;
    while (redir_isspace(*p))
        p++;
    *pp = p;
}

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static void get_word_until_chars(char *buf, int buf_size,
                                 const char *sep, const char **pp)
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{
    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;
}

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static void get_word_sep(char *buf, int buf_size, const char *sep,
                         const char **pp)
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{
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    if (**pp == '/') (*pp)++;
    get_word_until_chars(buf, buf_size, sep, pp);
}
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static void get_word(char *buf, int buf_size, const char **pp)
{
    get_word_until_chars(buf, buf_size, SPACE_CHARS, pp);
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}

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/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */
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static int sdp_parse_rtpmap(AVFormatContext *s,
                            AVCodecContext *codec, RTSPStream *rtsp_st,
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                            int payload_type, const char *p)
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{
    char buf[256];
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    int i;
    AVCodec *c;
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    const char *c_name;
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    /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
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     * 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);
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    if (payload_type >= RTP_PT_PRIVATE) {
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        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();
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                break;
            }
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        }
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    } else {
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        /* We are in a standard case
         * (from http://www.iana.org/assignments/rtp-parameters). */
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        /* search into AVRtpPayloadTypes[] */
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        codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
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    }

    c = avcodec_find_decoder(codec->codec_id);
    if (c && c->name)
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        c_name = c->name;
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    else
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        c_name = "(null)";
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    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)
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        }
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        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;
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}

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/* return the length and optionally the data */
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static int hex_to_data(uint8_t *data, const char *p)
{
    int c, len, v;

    len = 0;
    v = 1;
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    for (;;) {
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        skip_spaces(&p);
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        if (*p == '\0')
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            break;
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        c = toupper((unsigned char) *p++);
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        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;
}

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static void sdp_parse_fmtp_config(AVCodecContext * codec, void *ctx,
                                  char *attr, char *value)
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{
    switch (codec->codec_id) {
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    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;
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    }
    return;
}

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typedef struct {
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    const char *str;
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    uint16_t    type;
    uint32_t    offset;
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} AttrNameMap;
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/* All known fmtp parmeters and the corresping RTPAttrTypeEnum */
#define ATTR_NAME_TYPE_INT 0
#define ATTR_NAME_TYPE_STR 1
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static const AttrNameMap attr_names[]=
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{
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    { "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 },
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};

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/* 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. */
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int ff_rtsp_next_attr_and_value(const char **p, char *attr, int attr_size,
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                                char *value, int value_size)
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{
    skip_spaces(p);
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    if (**p) {
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        get_word_sep(attr, attr_size, "=", p);
        if (**p == '=')
            (*p)++;
        get_word_sep(value, value_size, ";", p);
        if (**p == ';')
            (*p)++;
        return 1;
    }
    return 0;
}

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/* parse a SDP line and save stream attributes */
static void sdp_parse_fmtp(AVStream *st, const char *p)
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{
    char attr[256];
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    /* Vorbis setup headers can be up to 12KB and are sent base64
     * encoded, giving a 12KB * (4/3) = 16KB FMTP line. */
    char value[16384];
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    int i;
    RTSPStream *rtsp_st = st->priv_data;
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    AVCodecContext *codec = st->codec;
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    RTPPayloadData *rtp_payload_data = &rtsp_st->rtp_payload_data;
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    /* loop on each attribute */
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    while (ff_rtsp_next_attr_and_value(&p, attr, sizeof(attr),
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                                       value, sizeof(value))) {
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        /* grab the codec extra_data from the config parameter of the fmtp
         * line */
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        sdp_parse_fmtp_config(codec, rtsp_st->dynamic_protocol_context,
                              attr, value);
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        /* Looking for a known attribute */
        for (i = 0; attr_names[i].str; ++i) {
            if (!strcasecmp(attr, attr_names[i].str)) {
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                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);
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            }
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        }
    }
}

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/** Parse a string p in the form of Range:npt=xx-xx, and determine the start
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 *  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);
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    if (!av_stristart(p, "npt=", &p))
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        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);
}

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typedef struct SDPParseState {
    /* SDP only */
    struct in_addr default_ip;
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    int            default_ttl;
    int            skip_media;  ///< set if an unknown m= line occurs
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} SDPParseState;

static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
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                           int letter, const char *buf)
{
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    RTSPState *rt = s->priv_data;
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    char buf1[64], st_type[64];
    const char *p;
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    enum CodecType codec_type;
    int payload_type, i;
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    AVStream *st;
    RTSPStream *rtsp_st;
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    struct in_addr sdp_ip;
    int ttl;

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    dprintf(s, "sdp: %c='%s'\n", letter, buf);
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    p = buf;
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    if (s1->skip_media && letter != 'm')
        return;
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    switch (letter) {
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    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);
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        if (ff_inet_aton(buf1, &sdp_ip) == 0)
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            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;
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    case 's':
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        av_metadata_set(&s->metadata, "title", p);
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        break;
    case 'i':
        if (s->nb_streams == 0) {
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            av_metadata_set(&s->metadata, "comment", p);
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            break;
        }
        break;
    case 'm':
        /* new stream */
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        s1->skip_media = 0;
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        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;
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        } else if (!strcmp(st_type, "application")) {
            codec_type = CODEC_TYPE_DATA;
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        } else {
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            s1->skip_media = 1;
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            return;
        }
        rtsp_st = av_mallocz(sizeof(RTSPStream));
        if (!rtsp_st)
            return;
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        rtsp_st->stream_index = -1;
        dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);
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        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) */
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        /* XXX: handle list of formats */
        get_word(buf1, sizeof(buf1), &p); /* format list */
        rtsp_st->sdp_payload_type = atoi(buf1);

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        if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
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            /* no corresponding stream */
        } else {
            st = av_new_stream(s, 0);
            if (!st)
                return;
            st->priv_data = rtsp_st;
            rtsp_st->stream_index = st->index;
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            st->codec->codec_type = codec_type;
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            if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
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                /* if standard payload type, we can find the codec right now */
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                ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type);
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            }
        }
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        /* put a default control url */
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        av_strlcpy(rtsp_st->control_url, rt->control_uri,
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                   sizeof(rtsp_st->control_url));
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        break;
    case 'a':
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        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 {
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            char proto[32];
            /* get the control url */
            st = s->streams[s->nb_streams - 1];
            rtsp_st = st->priv_data;
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            /* XXX: may need to add full url resolution */
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            ff_url_split(proto, sizeof(proto), NULL, 0, NULL, 0,
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                         NULL, NULL, 0, p);
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            if (proto[0] == '\0') {
                /* relative control URL */
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                if (rtsp_st->control_url[strlen(rtsp_st->control_url)-1]!='/')
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                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));
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            }
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        } else if (av_strstart(p, "rtpmap:", &p) && s->nb_streams > 0) {
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            /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
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            get_word(buf1, sizeof(buf1), &p);
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            payload_type = atoi(buf1);
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            st = s->streams[s->nb_streams - 1];
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            rtsp_st = st->priv_data;
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            sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p);
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        } else if (av_strstart(p, "fmtp:", &p)) {
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            /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
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            get_word(buf1, sizeof(buf1), &p);
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            payload_type = atoi(buf1);
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            for (i = 0; i < s->nb_streams; i++) {
                st      = s->streams[i];
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                rtsp_st = st->priv_data;
                if (rtsp_st->sdp_payload_type == payload_type) {
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                    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)))
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                        sdp_parse_fmtp(st, p);
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                }
            }
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        } else if (av_strstart(p, "framesize:", &p)) {
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            // let dynamic protocol handlers have a stab at the line.
            get_word(buf1, sizeof(buf1), &p);
            payload_type = atoi(buf1);
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            for (i = 0; i < s->nb_streams; i++) {
                st      = s->streams[i];
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                rtsp_st = st->priv_data;
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                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);
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            }
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        } else if (av_strstart(p, "range:", &p)) {
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            int64_t start, end;

            // this is so that seeking on a streamed file can work.
            rtsp_parse_range_npt(p, &start, &end);
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            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;
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        } else if (av_strstart(p, "IsRealDataType:integer;",&p)) {
            if (atoi(p) == 1)
                rt->transport = RTSP_TRANSPORT_RDT;
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        } else {
            if (rt->server_type == RTSP_SERVER_WMS)
                ff_wms_parse_sdp_a_line(s, p);
            if (s->nb_streams > 0) {
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                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)
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                    rtsp_st->dynamic_handler->parse_sdp_a_line(s,
                        s->nb_streams - 1,
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                        rtsp_st->dynamic_protocol_context, buf);
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            }
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        }
        break;
    }
}

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static int sdp_parse(AVFormatContext *s, const char *content)
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{
    const char *p;
    int letter;
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    /* 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
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     * buffer is large.
     *
     * The Vorbis FMTP line can be up to 16KB - see sdp_parse_fmtp. */
    char buf[16384], *q;
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    SDPParseState sdp_parse_state, *s1 = &sdp_parse_state;
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    memset(s1, 0, sizeof(SDPParseState));
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    p = content;
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    for (;;) {
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        skip_spaces(&p);
        letter = *p;
        if (letter == '\0')
            break;
        p++;
        if (*p != '=')
            goto next_line;
        p++;
        /* get the content */
        q = buf;
F
Fabrice Bellard 已提交
557
        while (*p != '\n' && *p != '\r' && *p != '\0') {
558 559 560 561 562
            if ((q - buf) < sizeof(buf) - 1)
                *q++ = *p;
            p++;
        }
        *q = '\0';
563
        sdp_parse_line(s, s1, letter, buf);
564 565 566 567 568 569 570 571 572
    next_line:
        while (*p != '\n' && *p != '\0')
            p++;
        if (*p == '\n')
            p++;
    }
    return 0;
}

573
/* close and free RTSP streams */
574
void ff_rtsp_close_streams(AVFormatContext *s)
575
{
576
    RTSPState *rt = s->priv_data;
577 578 579
    int i;
    RTSPStream *rtsp_st;

580
    for (i = 0; i < rt->nb_rtsp_streams; i++) {
581 582 583
        rtsp_st = rt->rtsp_streams[i];
        if (rtsp_st) {
            if (rtsp_st->transport_priv) {
584 585 586
                if (s->oformat) {
                    AVFormatContext *rtpctx = rtsp_st->transport_priv;
                    av_write_trailer(rtpctx);
587 588 589 590 591
                    if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
                        uint8_t *ptr;
                        url_close_dyn_buf(rtpctx->pb, &ptr);
                        av_free(ptr);
                    } else {
M
Martin Storsjö 已提交
592
                        url_fclose(rtpctx->pb);
593
                    }
594 595
                    av_metadata_free(&rtpctx->streams[0]->metadata);
                    av_metadata_free(&rtpctx->metadata);
596 597 598
                    av_free(rtpctx->streams[0]);
                    av_free(rtpctx);
                } else if (rt->transport == RTSP_TRANSPORT_RDT)
599 600 601 602 603 604 605
                    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)
606 607
                rtsp_st->dynamic_handler->close(
                    rtsp_st->dynamic_protocol_context);
608 609 610 611 612 613 614 615 616
        }
    }
    av_free(rt->rtsp_streams);
    if (rt->asf_ctx) {
        av_close_input_stream (rt->asf_ctx);
        rt->asf_ctx = NULL;
    }
}

617 618 619
static void *rtsp_rtp_mux_open(AVFormatContext *s, AVStream *st,
                               URLContext *handle)
{
620
    RTSPState *rt = s->priv_data;
621 622 623 624 625 626 627 628 629 630 631 632 633 634 635 636 637 638 639 640 641 642
    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;

643 644 645
    /* Set the synchronized start time. */
    rtpctx->start_time_realtime = rt->start_time;

646 647 648 649 650 651
    /* 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;

652
    if (handle) {
M
Martin Storsjö 已提交
653
        url_fdopen(&rtpctx->pb, handle);
654 655
    } else
        url_open_dyn_packet_buf(&rtpctx->pb, RTSP_TCP_MAX_PACKET_SIZE);
656 657 658
    ret = av_write_header(rtpctx);

    if (ret) {
659
        if (handle) {
M
Martin Storsjö 已提交
660
            url_fclose(rtpctx->pb);
661 662 663 664 665
        } else {
            uint8_t *ptr;
            url_close_dyn_buf(rtpctx->pb, &ptr);
            av_free(ptr);
        }
666 667 668 669 670 671 672 673 674 675
        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;
}

676
static int rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
677 678 679 680 681 682 683 684 685 686
{
    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;

687 688 689 690 691
    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)
692 693 694 695 696 697 698 699 700 701 702
        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) {
703
        if (rtsp_st->dynamic_handler) {
704 705 706 707 708 709 710 711 712
            rtp_parse_set_dynamic_protocol(rtsp_st->transport_priv,
                                           rtsp_st->dynamic_protocol_context,
                                           rtsp_st->dynamic_handler);
        }
    }

    return 0;
}

M
Martin Storsjö 已提交
713
#if CONFIG_RTSP_DEMUXER || CONFIG_RTSP_MUXER
714 715 716 717 718 719 720
static int rtsp_probe(AVProbeData *p)
{
    if (av_strstart(p->filename, "rtsp:", NULL))
        return AVPROBE_SCORE_MAX;
    return 0;
}

721 722 723 724 725 726 727 728 729 730 731 732 733 734 735 736 737 738 739 740 741
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 */
742
static void rtsp_parse_transport(RTSPMessageHeader *reply, const char *p)
743 744 745 746 747 748 749
{
    char transport_protocol[16];
    char profile[16];
    char lower_transport[16];
    char parameter[16];
    RTSPTransportField *th;
    char buf[256];
750

751
    reply->nb_transports = 0;
752

753
    for (;;) {
754 755 756 757 758 759
        skip_spaces(&p);
        if (*p == '\0')
            break;

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

760
        get_word_sep(transport_protocol, sizeof(transport_protocol),
761
                     "/", &p);
762
        if (!strcasecmp (transport_protocol, "rtp")) {
763 764 765 766 767 768
            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);
769 770 771 772
            }
            th->transport = RTSP_TRANSPORT_RTP;
        } else if (!strcasecmp (transport_protocol, "x-pn-tng") ||
                   !strcasecmp (transport_protocol, "x-real-rdt")) {
773
            /* x-pn-tng/<protocol> */
774 775
            get_word_sep(lower_transport, sizeof(lower_transport), "/;,", &p);
            profile[0] = '\0';
776
            th->transport = RTSP_TRANSPORT_RDT;
777
        }
F
Fabrice Bellard 已提交
778
        if (!strcasecmp(lower_transport, "TCP"))
779
            th->lower_transport = RTSP_LOWER_TRANSPORT_TCP;
780
        else
781
            th->lower_transport = RTSP_LOWER_TRANSPORT_UDP;
782

783 784 785 786 787 788 789 790 791 792 793 794 795
        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++;
796
                    rtsp_parse_range(&th->client_port_min,
797 798 799 800 801
                                     &th->client_port_max, &p);
                }
            } else if (!strcmp(parameter, "server_port")) {
                if (*p == '=') {
                    p++;
802
                    rtsp_parse_range(&th->server_port_min,
803 804 805 806 807
                                     &th->server_port_max, &p);
                }
            } else if (!strcmp(parameter, "interleaved")) {
                if (*p == '=') {
                    p++;
808
                    rtsp_parse_range(&th->interleaved_min,
809 810 811
                                     &th->interleaved_max, &p);
                }
            } else if (!strcmp(parameter, "multicast")) {
812 813
                if (th->lower_transport == RTSP_LOWER_TRANSPORT_UDP)
                    th->lower_transport = RTSP_LOWER_TRANSPORT_UDP_MULTICAST;
814 815 816 817 818 819 820 821 822 823 824
            } 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);
825
                    if (ff_inet_aton(buf, &ipaddr))
826 827 828 829 830 831 832 833 834 835 836 837 838 839 840
                        th->destination = ntohl(ipaddr.s_addr);
                }
            }
            while (*p != ';' && *p != '\0' && *p != ',')
                p++;
            if (*p == ';')
                p++;
        }
        if (*p == ',')
            p++;

        reply->nb_transports++;
    }
}

841 842
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
                        HTTPAuthState *auth_state)
843 844 845 846 847
{
    const char *p;

    /* NOTE: we do case independent match for broken servers */
    p = buf;
848
    if (av_stristart(p, "Session:", &p)) {
849
        int t;
850
        get_word_sep(reply->session_id, sizeof(reply->session_id), ";", &p);
851 852 853 854
        if (av_stristart(p, ";timeout=", &p) &&
            (t = strtol(p, NULL, 10)) > 0) {
            reply->timeout = t;
        }
855
    } else if (av_stristart(p, "Content-Length:", &p)) {
856
        reply->content_length = strtol(p, NULL, 10);
857
    } else if (av_stristart(p, "Transport:", &p)) {
858
        rtsp_parse_transport(reply, p);
859
    } else if (av_stristart(p, "CSeq:", &p)) {
860
        reply->seq = strtol(p, NULL, 10);
861
    } else if (av_stristart(p, "Range:", &p)) {
862
        rtsp_parse_range_npt(p, &reply->range_start, &reply->range_end);
863 864 865
    } else if (av_stristart(p, "RealChallenge1:", &p)) {
        skip_spaces(&p);
        av_strlcpy(reply->real_challenge, p, sizeof(reply->real_challenge));
866 867 868
    } else if (av_stristart(p, "Server:", &p)) {
        skip_spaces(&p);
        av_strlcpy(reply->server, p, sizeof(reply->server));
869 870 871
    } else if (av_stristart(p, "Notice:", &p) ||
               av_stristart(p, "X-Notice:", &p)) {
        reply->notice = strtol(p, NULL, 10);
L
Luca Barbato 已提交
872 873 874
    } else if (av_stristart(p, "Location:", &p)) {
        skip_spaces(&p);
        av_strlcpy(reply->location, p , sizeof(reply->location));
875 876 877 878 879 880
    } else if (av_stristart(p, "WWW-Authenticate:", &p) && auth_state) {
        skip_spaces(&p);
        ff_http_auth_handle_header(auth_state, "WWW-Authenticate", p);
    } else if (av_stristart(p, "Authentication-Info:", &p) && auth_state) {
        skip_spaces(&p);
        ff_http_auth_handle_header(auth_state, "Authentication-Info", p);
881 882 883
    }
}

884
/* skip a RTP/TCP interleaved packet */
885
void ff_rtsp_skip_packet(AVFormatContext *s)
886 887 888 889 890
{
    RTSPState *rt = s->priv_data;
    int ret, len, len1;
    uint8_t buf[1024];

891
    ret = url_read_complete(rt->rtsp_hd, buf, 3);
892 893
    if (ret != 3)
        return;
894
    len = AV_RB16(buf + 1);
895 896 897

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

898 899 900 901 902
    /* skip payload */
    while (len > 0) {
        len1 = len;
        if (len1 > sizeof(buf))
            len1 = sizeof(buf);
903
        ret = url_read_complete(rt->rtsp_hd, buf, len1);
904 905 906 907 908
        if (ret != len1)
            return;
        len -= len1;
    }
}
909

910
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
M
Martin Storsjö 已提交
911 912
                       unsigned char **content_ptr,
                       int return_on_interleaved_data)
913 914 915 916 917
{
    RTSPState *rt = s->priv_data;
    char buf[4096], buf1[1024], *q;
    unsigned char ch;
    const char *p;
918
    int ret, content_length, line_count = 0;
919 920
    unsigned char *content = NULL;

921
    memset(reply, 0, sizeof(*reply));
922 923 924

    /* parse reply (XXX: use buffers) */
    rt->last_reply[0] = '\0';
925
    for (;;) {
926
        q = buf;
927
        for (;;) {
928
            ret = url_read_complete(rt->rtsp_hd, &ch, 1);
929
#ifdef DEBUG_RTP_TCP
930
            dprintf(s, "ret=%d c=%02x [%c]\n", ret, ch, ch);
931 932 933
#endif
            if (ret != 1)
                return -1;
934 935
            if (ch == '\n')
                break;
936 937
            if (ch == '$') {
                /* XXX: only parse it if first char on line ? */
938 939 940
                if (return_on_interleaved_data) {
                    return 1;
                } else
941
                    ff_rtsp_skip_packet(s);
942
            } else if (ch != '\r') {
943 944 945 946 947
                if ((q - buf) < sizeof(buf) - 1)
                    *q++ = ch;
            }
        }
        *q = '\0';
948 949 950

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

951 952 953 954 955 956 957 958 959 960
        /* 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 {
961
            ff_rtsp_parse_line(reply, p, &rt->auth_state);
M
Måns Rullgård 已提交
962 963
            av_strlcat(rt->last_reply, p,    sizeof(rt->last_reply));
            av_strlcat(rt->last_reply, "\n", sizeof(rt->last_reply));
964 965 966
        }
        line_count++;
    }
967

968
    if (rt->session_id[0] == '\0' && reply->session_id[0] != '\0')
M
Måns Rullgård 已提交
969
        av_strlcpy(rt->session_id, reply->session_id, sizeof(rt->session_id));
970

971 972 973 974
    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);
975
        (void)url_read_complete(rt->rtsp_hd, content, content_length);
976 977 978 979
        content[content_length] = '\0';
    }
    if (content_ptr)
        *content_ptr = content;
980 981
    else
        av_free(content);
982

983 984 985 986 987
    if (rt->seq != reply->seq) {
        av_log(s, AV_LOG_WARNING, "CSeq %d expected, %d received.\n",
            rt->seq, reply->seq);
    }

988 989 990
    /* EOS */
    if (reply->notice == 2101 /* End-of-Stream Reached */      ||
        reply->notice == 2104 /* Start-of-Stream Reached */    ||
991
        reply->notice == 2306 /* Continuous Feed Terminated */) {
992
        rt->state = RTSP_STATE_IDLE;
993
    } else if (reply->notice >= 4400 && reply->notice < 5500) {
994
        return AVERROR(EIO); /* data or server error */
995
    } else if (reply->notice == 2401 /* Ticket Expired */ ||
996 997 998
             (reply->notice >= 5500 && reply->notice < 5600) /* end of term */ )
        return AVERROR(EPERM);

999
    return 0;
1000 1001
}

1002
void ff_rtsp_send_cmd_with_content_async(AVFormatContext *s,
1003 1004
                                         const char *method, const char *url,
                                         const char *headers,
M
Martin Storsjö 已提交
1005 1006
                                         const unsigned char *send_content,
                                         int send_content_length)
1007 1008
{
    RTSPState *rt = s->priv_data;
1009
    char buf[4096];
1010 1011

    rt->seq++;
1012 1013 1014
    snprintf(buf, sizeof(buf), "%s %s RTSP/1.0\r\n", method, url);
    if (headers)
        av_strlcat(buf, headers, sizeof(buf));
1015
    av_strlcatf(buf, sizeof(buf), "CSeq: %d\r\n", rt->seq);
1016 1017
    if (rt->session_id[0] != '\0' && (!headers ||
        !strstr(headers, "\nIf-Match:"))) {
1018
        av_strlcatf(buf, sizeof(buf), "Session: %s\r\n", rt->session_id);
1019
    }
1020 1021 1022 1023 1024 1025 1026
    if (rt->auth[0]) {
        char *str = ff_http_auth_create_response(&rt->auth_state,
                                                 rt->auth, url, method);
        if (str)
            av_strlcat(buf, str, sizeof(buf));
        av_free(str);
    }
1027 1028
    if (send_content_length > 0 && send_content)
        av_strlcatf(buf, sizeof(buf), "Content-Length: %d\r\n", send_content_length);
1029
    av_strlcat(buf, "\r\n", sizeof(buf));
1030 1031 1032

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

1033
    url_write(rt->rtsp_hd, buf, strlen(buf));
1034 1035
    if (send_content_length > 0 && send_content)
        url_write(rt->rtsp_hd, send_content, send_content_length);
1036 1037 1038
    rt->last_cmd_time = av_gettime();
}

1039 1040
void ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
                            const char *url, const char *headers)
1041
{
1042
    ff_rtsp_send_cmd_with_content_async(s, method, url, headers, NULL, 0);
1043 1044
}

1045 1046
void ff_rtsp_send_cmd(AVFormatContext *s, const char *method, const char *url,
                      const char *headers, RTSPMessageHeader *reply,
M
Martin Storsjö 已提交
1047
                      unsigned char **content_ptr)
1048
{
1049 1050
    ff_rtsp_send_cmd_with_content(s, method, url, headers, reply,
                                  content_ptr, NULL, 0);
1051 1052
}

1053
void ff_rtsp_send_cmd_with_content(AVFormatContext *s,
1054 1055
                                   const char *method, const char *url,
                                   const char *header,
M
Martin Storsjö 已提交
1056 1057 1058 1059
                                   RTSPMessageHeader *reply,
                                   unsigned char **content_ptr,
                                   const unsigned char *send_content,
                                   int send_content_length)
1060
{
1061 1062 1063 1064 1065
    RTSPState *rt = s->priv_data;
    HTTPAuthType cur_auth_type;

retry:
    cur_auth_type = rt->auth_state.auth_type;
1066 1067
    ff_rtsp_send_cmd_with_content_async(s, method, url, header,
                                        send_content, send_content_length);
1068

1069
    ff_rtsp_read_reply(s, reply, content_ptr, 0);
1070 1071 1072 1073

    if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
        rt->auth_state.auth_type != HTTP_AUTH_NONE)
        goto retry;
1074 1075
}

1076 1077 1078
/**
 * @returns 0 on success, <0 on error, 1 if protocol is unavailable.
 */
1079 1080
static int make_setup_request(AVFormatContext *s, const char *host, int port,
                              int lower_transport, const char *real_challenge)
1081 1082
{
    RTSPState *rt = s->priv_data;
1083
    int rtx, j, i, err, interleave = 0;
1084
    RTSPStream *rtsp_st;
1085
    RTSPMessageHeader reply1, *reply = &reply1;
1086
    char cmd[2048];
1087 1088
    const char *trans_pref;

1089
    if (rt->transport == RTSP_TRANSPORT_RDT)
1090 1091 1092
        trans_pref = "x-pn-tng";
    else
        trans_pref = "RTP/AVP";
1093

1094 1095 1096
    /* default timeout: 1 minute */
    rt->timeout = 60;

1097 1098
    /* for each stream, make the setup request */
    /* XXX: we assume the same server is used for the control of each
1099
     * RTSP stream */
R
Romain Degez 已提交
1100

1101
    for (j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
1102 1103
        char transport[2048];

1104 1105 1106 1107 1108 1109 1110 1111 1112 1113 1114 1115
        /**
         * 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 &&
1116 1117
                        !strcmp(rt->rtsp_streams[rtx]->control_url + len - 4,
                                "/rtx"))
1118 1119 1120 1121 1122 1123 1124 1125
                        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
R
Ronald S. Bultje 已提交
1126
            rtsp_st = rt->rtsp_streams[i];
1127 1128

        /* RTP/UDP */
1129
        if (lower_transport == RTSP_LOWER_TRANSPORT_UDP) {
F
Fabrice Bellard 已提交
1130 1131
            char buf[256];

1132 1133 1134 1135 1136
            if (rt->server_type == RTSP_SERVER_WMS && i > 1) {
                port = reply->transports[0].client_port_min;
                goto have_port;
            }

F
Fabrice Bellard 已提交
1137
            /* first try in specified port range */
R
Romain Degez 已提交
1138
            if (RTSP_RTP_PORT_MIN != 0) {
1139
                while (j <= RTSP_RTP_PORT_MAX) {
1140 1141
                    ff_url_join(buf, sizeof(buf), "rtp", NULL, host, -1,
                                "?localport=%d", j);
1142 1143 1144
                    /* we will use two ports per rtp stream (rtp and rtcp) */
                    j += 2;
                    if (url_open(&rtsp_st->rtp_handle, buf, URL_RDWR) == 0)
F
Fabrice Bellard 已提交
1145 1146
                        goto rtp_opened;
                }
1147
            }
F
Fabrice Bellard 已提交
1148

1149 1150 1151 1152 1153 1154 1155
#if 0
            /* then try on any port */
            if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
                err = AVERROR_INVALIDDATA;
                goto fail;
            }
#endif
F
Fabrice Bellard 已提交
1156 1157

        rtp_opened:
1158
            port = rtp_get_local_port(rtsp_st->rtp_handle);
1159
        have_port:
1160
            snprintf(transport, sizeof(transport) - 1,
1161 1162 1163 1164 1165
                     "%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);
1166 1167
            if (rt->transport == RTSP_TRANSPORT_RTP &&
                !(rt->server_type == RTSP_SERVER_WMS && i > 0))
1168
                av_strlcatf(transport, sizeof(transport), "-%d", port + 1);
1169 1170 1171
        }

        /* RTP/TCP */
1172
        else if (lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1173 1174 1175 1176
            /** 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 &&
1177 1178
                s->streams[rtsp_st->stream_index]->codec->codec_type ==
                    CODEC_TYPE_DATA)
1179
                continue;
1180
            snprintf(transport, sizeof(transport) - 1,
1181 1182 1183 1184 1185 1186 1187
                     "%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;
1188 1189
        }

1190
        else if (lower_transport == RTSP_LOWER_TRANSPORT_UDP_MULTICAST) {
1191
            snprintf(transport, sizeof(transport) - 1,
1192
                     "%s/UDP;multicast", trans_pref);
1193
        }
1194 1195 1196
        if (s->oformat) {
            av_strlcat(transport, ";mode=receive", sizeof(transport));
        } else if (rt->server_type == RTSP_SERVER_REAL ||
1197
                   rt->server_type == RTSP_SERVER_WMS)
1198
            av_strlcat(transport, ";mode=play", sizeof(transport));
1199
        snprintf(cmd, sizeof(cmd),
F
Fabrice Bellard 已提交
1200
                 "Transport: %s\r\n",
1201
                 transport);
1202
        if (i == 0 && rt->server_type == RTSP_SERVER_REAL) {
1203 1204 1205 1206 1207 1208 1209 1210
            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);
        }
1211
        ff_rtsp_send_cmd(s, "SETUP", rtsp_st->control_url, cmd, reply, NULL);
1212 1213 1214
        if (reply->status_code == 461 /* Unsupported protocol */ && i == 0) {
            err = 1;
            goto fail;
1215 1216
        } else if (reply->status_code != RTSP_STATUS_OK ||
                   reply->nb_transports != 1) {
1217 1218 1219 1220 1221 1222
            err = AVERROR_INVALIDDATA;
            goto fail;
        }

        /* XXX: same protocol for all streams is required */
        if (i > 0) {
1223 1224
            if (reply->transports[0].lower_transport != rt->lower_transport ||
                reply->transports[0].transport != rt->transport) {
1225 1226 1227 1228
                err = AVERROR_INVALIDDATA;
                goto fail;
            }
        } else {
1229
            rt->lower_transport = reply->transports[0].lower_transport;
1230
            rt->transport = reply->transports[0].transport;
1231 1232 1233
        }

        /* close RTP connection if not choosen */
1234 1235
        if (reply->transports[0].lower_transport != RTSP_LOWER_TRANSPORT_UDP &&
            (lower_transport == RTSP_LOWER_TRANSPORT_UDP)) {
1236 1237
            url_close(rtsp_st->rtp_handle);
            rtsp_st->rtp_handle = NULL;
1238 1239
        }

1240 1241
        switch(reply->transports[0].lower_transport) {
        case RTSP_LOWER_TRANSPORT_TCP:
1242 1243 1244
            rtsp_st->interleaved_min = reply->transports[0].interleaved_min;
            rtsp_st->interleaved_max = reply->transports[0].interleaved_max;
            break;
1245

1246 1247 1248 1249
        case RTSP_LOWER_TRANSPORT_UDP: {
            char url[1024];

            /* XXX: also use address if specified */
1250 1251
            ff_url_join(url, sizeof(url), "rtp", NULL, host,
                        reply->transports[0].server_port_min, NULL);
1252 1253 1254 1255
            if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) &&
                rtp_set_remote_url(rtsp_st->rtp_handle, url) < 0) {
                err = AVERROR_INVALIDDATA;
                goto fail;
1256
            }
1257 1258 1259 1260
            /* Try to initialize the connection state in a
             * potential NAT router by sending dummy packets.
             * RTP/RTCP dummy packets are used for RDT, too.
             */
1261
            if (!(rt->server_type == RTSP_SERVER_WMS && i > 1) && s->iformat)
1262
                rtp_send_punch_packets(rtsp_st->rtp_handle);
1263
            break;
1264 1265 1266 1267 1268 1269 1270 1271 1272 1273 1274 1275 1276 1277 1278
        }
        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;
            }
1279 1280
            ff_url_join(url, sizeof(url), "rtp", NULL, inet_ntoa(in),
                        port, "?ttl=%d", ttl);
1281 1282 1283
            if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
                err = AVERROR_INVALIDDATA;
                goto fail;
1284 1285 1286
            }
            break;
        }
1287
        }
R
Romain Degez 已提交
1288

1289
        if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
1290
            goto fail;
1291 1292
    }

1293 1294 1295
    if (reply->timeout > 0)
        rt->timeout = reply->timeout;

1296
    if (rt->server_type == RTSP_SERVER_REAL)
1297 1298
        rt->need_subscription = 1;

1299 1300 1301
    return 0;

fail:
1302
    for (i = 0; i < rt->nb_rtsp_streams; i++) {
1303 1304 1305 1306 1307
        if (rt->rtsp_streams[i]->rtp_handle) {
            url_close(rt->rtsp_streams[i]->rtp_handle);
            rt->rtsp_streams[i]->rtp_handle = NULL;
        }
    }
1308 1309 1310
    return err;
}

1311 1312 1313 1314 1315 1316 1317 1318 1319 1320
static int rtsp_read_play(AVFormatContext *s)
{
    RTSPState *rt = s->priv_data;
    RTSPMessageHeader reply1, *reply = &reply1;
    char cmd[1024];

    av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);

    if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
        if (rt->state == RTSP_STATE_PAUSED) {
1321
            cmd[0] = 0;
1322 1323 1324 1325 1326
        } else {
            snprintf(cmd, sizeof(cmd),
                     "Range: npt=%0.3f-\r\n",
                     (double)rt->seek_timestamp / AV_TIME_BASE);
        }
1327
        ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
1328 1329 1330 1331
        if (reply->status_code != RTSP_STATUS_OK) {
            return -1;
        }
    }
1332
    rt->state = RTSP_STATE_STREAMING;
1333 1334 1335
    return 0;
}

1336
static int rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply)
1337 1338 1339 1340 1341 1342 1343 1344
{
    RTSPState *rt = s->priv_data;
    char cmd[1024];
    unsigned char *content = NULL;
    int ret;

    /* describe the stream */
    snprintf(cmd, sizeof(cmd),
1345
             "Accept: application/sdp\r\n");
1346 1347 1348 1349 1350 1351 1352 1353 1354
    if (rt->server_type == RTSP_SERVER_REAL) {
        /**
         * The Require: attribute is needed for proper streaming from
         * Realmedia servers.
         */
        av_strlcat(cmd,
                   "Require: com.real.retain-entity-for-setup\r\n",
                   sizeof(cmd));
    }
1355
    ff_rtsp_send_cmd(s, "DESCRIBE", rt->control_uri, cmd, reply, &content);
1356 1357 1358 1359 1360 1361 1362 1363 1364 1365 1366 1367 1368 1369 1370 1371
    if (!content)
        return AVERROR_INVALIDDATA;
    if (reply->status_code != RTSP_STATUS_OK) {
        av_freep(&content);
        return AVERROR_INVALIDDATA;
    }

    /* now we got the SDP description, we parse it */
    ret = sdp_parse(s, (const char *)content);
    av_freep(&content);
    if (ret < 0)
        return AVERROR_INVALIDDATA;

    return 0;
}

1372
static int rtsp_setup_output_streams(AVFormatContext *s, const char *addr)
1373 1374 1375 1376 1377
{
    RTSPState *rt = s->priv_data;
    RTSPMessageHeader reply1, *reply = &reply1;
    int i;
    char *sdp;
1378
    AVFormatContext sdp_ctx, *ctx_array[1];
1379 1380

    rt->start_time = av_gettime();
1381 1382 1383 1384 1385

    /* Announce the stream */
    sdp = av_mallocz(8192);
    if (sdp == NULL)
        return AVERROR(ENOMEM);
1386 1387 1388 1389 1390 1391 1392 1393 1394 1395 1396 1397 1398 1399 1400 1401 1402
    /* We create the SDP based on the RTSP AVFormatContext where we
     * aren't allowed to change the filename field. (We create the SDP
     * based on the RTSP context since the contexts for the RTP streams
     * don't exist yet.) In order to specify a custom URL with the actual
     * peer IP instead of the originally specified hostname, we create
     * a temporary copy of the AVFormatContext, where the custom URL is set.
     *
     * FIXME: Create the SDP without copying the AVFormatContext.
     * This either requires setting up the RTP stream AVFormatContexts
     * already here (complicating things immensely) or getting a more
     * flexible SDP creation interface.
     */
    sdp_ctx = *s;
    ff_url_join(sdp_ctx.filename, sizeof(sdp_ctx.filename),
                "rtsp", NULL, addr, -1, NULL);
    ctx_array[0] = &sdp_ctx;
    if (avf_sdp_create(ctx_array, 1, sdp, 8192)) {
1403 1404 1405 1406
        av_free(sdp);
        return AVERROR_INVALIDDATA;
    }
    av_log(s, AV_LOG_INFO, "SDP:\n%s\n", sdp);
1407 1408 1409
    ff_rtsp_send_cmd_with_content(s, "ANNOUNCE", rt->control_uri,
                                  "Content-Type: application/sdp\r\n",
                                  reply, NULL, sdp, strlen(sdp));
1410 1411 1412 1413 1414 1415 1416 1417 1418 1419 1420 1421 1422 1423 1424 1425 1426
    av_free(sdp);
    if (reply->status_code != RTSP_STATUS_OK)
        return AVERROR_INVALIDDATA;

    /* Set up the RTSPStreams for each AVStream */
    for (i = 0; i < s->nb_streams; i++) {
        RTSPStream *rtsp_st;
        AVStream *st = s->streams[i];

        rtsp_st = av_mallocz(sizeof(RTSPStream));
        if (!rtsp_st)
            return AVERROR(ENOMEM);
        dynarray_add(&rt->rtsp_streams, &rt->nb_rtsp_streams, rtsp_st);

        st->priv_data = rtsp_st;
        rtsp_st->stream_index = i;

1427
        av_strlcpy(rtsp_st->control_url, rt->control_uri, sizeof(rtsp_st->control_url));
1428 1429 1430 1431 1432 1433 1434 1435
        /* Note, this must match the relative uri set in the sdp content */
        av_strlcatf(rtsp_st->control_url, sizeof(rtsp_st->control_url),
                    "/streamid=%d", i);
    }

    return 0;
}

1436
int ff_rtsp_connect(AVFormatContext *s)
1437 1438
{
    RTSPState *rt = s->priv_data;
1439 1440
    char host[1024], path[1024], tcpname[1024], cmd[2048], auth[128];
    char *option_list, *option, *filename;
1441
    URLContext *rtsp_hd;
1442
    int port, err, tcp_fd;
1443
    RTSPMessageHeader reply1, *reply = &reply1;
1444
    int lower_transport_mask = 0;
1445
    char real_challenge[64];
1446 1447
    struct sockaddr_storage peer;
    socklen_t peer_len = sizeof(peer);
1448 1449 1450

    if (!ff_network_init())
        return AVERROR(EIO);
1451
redirect:
1452
    /* extract hostname and port */
1453
    ff_url_split(NULL, 0, auth, sizeof(auth),
M
Martin Storsjö 已提交
1454
                 host, sizeof(host), &port, path, sizeof(path), s->filename);
1455
    if (*auth) {
1456
        av_strlcpy(rt->auth, auth, sizeof(rt->auth));
1457
    }
1458 1459 1460 1461
    if (port < 0)
        port = RTSP_DEFAULT_PORT;

    /* search for options */
1462
    option_list = strrchr(path, '?');
1463
    if (option_list) {
1464 1465 1466
        /* Strip out the RTSP specific options, write out the rest of
         * the options back into the same string. */
        filename = option_list;
1467
        while (option_list) {
1468
            /* move the option pointer */
1469
            option = ++option_list;
1470 1471
            option_list = strchr(option_list, '&');
            if (option_list)
1472 1473
                *option_list = 0;

1474
            /* handle the options */
1475
            if (!strcmp(option, "udp")) {
1476
                lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP);
1477
            } else if (!strcmp(option, "multicast")) {
1478
                lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_UDP_MULTICAST);
1479
            } else if (!strcmp(option, "tcp")) {
1480
                lower_transport_mask |= (1<< RTSP_LOWER_TRANSPORT_TCP);
1481
            } else {
1482 1483 1484 1485 1486
                /* Write options back into the buffer, using memmove instead
                 * of strcpy since the strings may overlap. */
                int len = strlen(option);
                memmove(++filename, option, len);
                filename += len;
1487 1488
                if (option_list) *filename = '&';
            }
1489
        }
1490
        *filename = 0;
1491 1492
    }

1493
    if (!lower_transport_mask)
1494
        lower_transport_mask = (1 << RTSP_LOWER_TRANSPORT_NB) - 1;
1495

1496
    if (s->oformat) {
1497 1498 1499
        /* Only UDP or TCP - UDP multicast isn't supported. */
        lower_transport_mask &= (1 << RTSP_LOWER_TRANSPORT_UDP) |
                                (1 << RTSP_LOWER_TRANSPORT_TCP);
1500 1501
        if (!lower_transport_mask) {
            av_log(s, AV_LOG_ERROR, "Unsupported lower transport method, "
1502
                                    "only UDP and TCP are supported for output.\n");
1503 1504 1505 1506 1507
            err = AVERROR(EINVAL);
            goto fail;
        }
    }

1508 1509 1510 1511 1512 1513
    /* Construct the URI used in request; this is similar to s->filename,
     * but with authentication credentials removed and RTSP specific options
     * stripped out. */
    ff_url_join(rt->control_uri, sizeof(rt->control_uri), "rtsp", NULL,
                host, port, "%s", path);

1514
    /* open the tcp connexion */
1515
    ff_url_join(tcpname, sizeof(tcpname), "tcp", NULL, host, port, NULL);
1516 1517 1518 1519
    if (url_open(&rtsp_hd, tcpname, URL_RDWR) < 0) {
        err = AVERROR(EIO);
        goto fail;
    }
1520 1521 1522
    rt->rtsp_hd = rtsp_hd;
    rt->seq = 0;

1523 1524 1525 1526 1527 1528
    tcp_fd = url_get_file_handle(rtsp_hd);
    if (!getpeername(tcp_fd, (struct sockaddr*) &peer, &peer_len)) {
        getnameinfo((struct sockaddr*) &peer, peer_len, host, sizeof(host),
                    NULL, 0, NI_NUMERICHOST);
    }

1529 1530
    /* request options supported by the server; this also detects server
     * type */
1531
    for (rt->server_type = RTSP_SERVER_RTP;;) {
1532
        cmd[0] = 0;
1533
        if (rt->server_type == RTSP_SERVER_REAL)
1534 1535 1536 1537 1538 1539 1540 1541 1542 1543 1544 1545 1546 1547 1548
            av_strlcat(cmd,
                       /**
                        * The following entries are required for proper
                        * streaming from a Realmedia server. They are
                        * interdependent in some way although we currently
                        * don't quite understand how. Values were copied
                        * from mplayer SVN r23589.
                        * @param CompanyID is a 16-byte ID in base64
                        * @param ClientChallenge is a 16-byte ID in hex
                        */
                       "ClientChallenge: 9e26d33f2984236010ef6253fb1887f7\r\n"
                       "PlayerStarttime: [28/03/2003:22:50:23 00:00]\r\n"
                       "CompanyID: KnKV4M4I/B2FjJ1TToLycw==\r\n"
                       "GUID: 00000000-0000-0000-0000-000000000000\r\n",
                       sizeof(cmd));
1549
        ff_rtsp_send_cmd(s, "OPTIONS", rt->control_uri, cmd, reply, NULL);
1550 1551 1552 1553 1554 1555
        if (reply->status_code != RTSP_STATUS_OK) {
            err = AVERROR_INVALIDDATA;
            goto fail;
        }

        /* detect server type if not standard-compliant RTP */
1556 1557
        if (rt->server_type != RTSP_SERVER_REAL && reply->real_challenge[0]) {
            rt->server_type = RTSP_SERVER_REAL;
1558
            continue;
1559 1560
        } else if (!strncasecmp(reply->server, "WMServer/", 9)) {
            rt->server_type = RTSP_SERVER_WMS;
1561
        } else if (rt->server_type == RTSP_SERVER_REAL)
1562 1563 1564 1565
            strcpy(real_challenge, reply->real_challenge);
        break;
    }

1566
    if (s->iformat)
1567
        err = rtsp_setup_input_streams(s, reply);
1568
    else
1569
        err = rtsp_setup_output_streams(s, host);
1570
    if (err)
1571 1572
        goto fail;

1573
    do {
1574 1575
        int lower_transport = ff_log2_tab[lower_transport_mask &
                                  ~(lower_transport_mask - 1)];
1576

1577
        err = make_setup_request(s, host, port, lower_transport,
1578
                                 rt->server_type == RTSP_SERVER_REAL ?
1579
                                     real_challenge : NULL);
1580
        if (err < 0)
1581
            goto fail;
1582 1583
        lower_transport_mask &= ~(1 << lower_transport);
        if (lower_transport_mask == 0 && err == 1) {
1584
            err = AVERROR(FF_NETERROR(EPROTONOSUPPORT));
1585 1586 1587
            goto fail;
        }
    } while (err);
1588

1589
    rt->state = RTSP_STATE_IDLE;
1590
    rt->seek_timestamp = 0; /* default is to start stream at position zero */
1591 1592
    return 0;
 fail:
1593
    ff_rtsp_close_streams(s);
1594
    url_close(rt->rtsp_hd);
1595
    if (reply->status_code >=300 && reply->status_code < 400 && s->iformat) {
L
Luca Barbato 已提交
1596 1597 1598 1599 1600 1601
        av_strlcpy(s->filename, reply->location, sizeof(s->filename));
        av_log(s, AV_LOG_INFO, "Status %d: Redirecting to %s\n",
               reply->status_code,
               s->filename);
        goto redirect;
    }
1602
    ff_network_close();
1603 1604
    return err;
}
M
Martin Storsjö 已提交
1605
#endif
1606

M
Martin Storsjö 已提交
1607
#if CONFIG_RTSP_DEMUXER
1608 1609 1610 1611 1612 1613
static int rtsp_read_header(AVFormatContext *s,
                            AVFormatParameters *ap)
{
    RTSPState *rt = s->priv_data;
    int ret;

1614
    ret = ff_rtsp_connect(s);
1615 1616 1617 1618 1619 1620 1621
    if (ret)
        return ret;

    if (ap->initial_pause) {
         /* do not start immediately */
    } else {
         if (rtsp_read_play(s) < 0) {
1622
            ff_rtsp_close_streams(s);
1623 1624 1625 1626 1627 1628 1629 1630
            url_close(rt->rtsp_hd);
            return AVERROR_INVALIDDATA;
        }
    }

    return 0;
}

R
Ronald S. Bultje 已提交
1631 1632 1633 1634 1635 1636 1637 1638 1639
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 fd, fd_max, n, i, ret, tcp_fd;
    struct timeval tv;

1640
    for (;;) {
R
Ronald S. Bultje 已提交
1641 1642 1643 1644 1645 1646 1647 1648 1649 1650
        if (url_interrupt_cb())
            return AVERROR(EINTR);
        FD_ZERO(&rfds);
        if (rt->rtsp_hd) {
            tcp_fd = fd_max = url_get_file_handle(rt->rtsp_hd);
            FD_SET(tcp_fd, &rfds);
        } else {
            fd_max = 0;
            tcp_fd = -1;
        }
1651
        for (i = 0; i < rt->nb_rtsp_streams; i++) {
R
Ronald S. Bultje 已提交
1652 1653 1654 1655 1656 1657 1658 1659 1660 1661 1662 1663 1664 1665
            rtsp_st = rt->rtsp_streams[i];
            if (rtsp_st->rtp_handle) {
                /* currently, we cannot probe RTCP handle because of
                 * blocking restrictions */
                fd = url_get_file_handle(rtsp_st->rtp_handle);
                if (fd > fd_max)
                    fd_max = fd;
                FD_SET(fd, &rfds);
            }
        }
        tv.tv_sec = 0;
        tv.tv_usec = 100 * 1000;
        n = select(fd_max + 1, &rfds, NULL, NULL, &tv);
        if (n > 0) {
1666
            for (i = 0; i < rt->nb_rtsp_streams; i++) {
R
Ronald S. Bultje 已提交
1667 1668 1669 1670 1671 1672 1673 1674 1675 1676 1677 1678 1679
                rtsp_st = rt->rtsp_streams[i];
                if (rtsp_st->rtp_handle) {
                    fd = url_get_file_handle(rtsp_st->rtp_handle);
                    if (FD_ISSET(fd, &rfds)) {
                        ret = url_read(rtsp_st->rtp_handle, buf, buf_size);
                        if (ret > 0) {
                            *prtsp_st = rtsp_st;
                            return ret;
                        }
                    }
                }
            }
#if CONFIG_RTSP_DEMUXER
1680
            if (tcp_fd != -1 && FD_ISSET(tcp_fd, &rfds)) {
R
Ronald S. Bultje 已提交
1681 1682
                RTSPMessageHeader reply;

1683 1684 1685
                ret = ff_rtsp_read_reply(s, &reply, NULL, 0);
                if (ret < 0)
                    return ret;
R
Ronald S. Bultje 已提交
1686
                /* XXX: parse message */
1687
                if (rt->state != RTSP_STATE_STREAMING)
R
Ronald S. Bultje 已提交
1688 1689 1690 1691 1692 1693 1694
                    return 0;
            }
#endif
        }
    }
}

1695 1696
static int tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
                           uint8_t *buf, int buf_size)
1697 1698
{
    RTSPState *rt = s->priv_data;
F
Fabrice Bellard 已提交
1699
    int id, len, i, ret;
1700 1701
    RTSPStream *rtsp_st;

F
Fabrice Bellard 已提交
1702
#ifdef DEBUG_RTP_TCP
1703
    dprintf(s, "tcp_read_packet:\n");
F
Fabrice Bellard 已提交
1704
#endif
1705 1706
redo:
    for (;;) {
1707 1708
        RTSPMessageHeader reply;

1709
        ret = ff_rtsp_read_reply(s, &reply, NULL, 1);
1710
        if (ret == -1)
1711
            return -1;
1712
        if (ret == 1) /* received '$' */
1713
            break;
1714
        /* XXX: parse message */
1715
        if (rt->state != RTSP_STATE_STREAMING)
1716
            return 0;
1717
    }
1718
    ret = url_read_complete(rt->rtsp_hd, buf, 3);
F
Fabrice Bellard 已提交
1719
    if (ret != 3)
1720
        return -1;
1721
    id  = buf[0];
1722
    len = AV_RB16(buf + 1);
F
Fabrice Bellard 已提交
1723
#ifdef DEBUG_RTP_TCP
1724
    dprintf(s, "id=%d len=%d\n", id, len);
F
Fabrice Bellard 已提交
1725
#endif
1726
    if (len > buf_size || len < 12)
1727 1728
        goto redo;
    /* get the data */
1729
    ret = url_read_complete(rt->rtsp_hd, buf, len);
F
Fabrice Bellard 已提交
1730
    if (ret != len)
1731
        return -1;
1732
    if (rt->transport == RTSP_TRANSPORT_RDT &&
1733
        ff_rdt_parse_header(buf, len, &id, NULL, NULL, NULL, NULL) < 0)
1734
        return -1;
1735

1736
    /* find the matching stream */
1737
    for (i = 0; i < rt->nb_rtsp_streams; i++) {
1738
        rtsp_st = rt->rtsp_streams[i];
1739 1740
        if (id >= rtsp_st->interleaved_min &&
            id <= rtsp_st->interleaved_max)
1741 1742 1743
            goto found;
    }
    goto redo;
1744
found:
1745 1746
    *prtsp_st = rtsp_st;
    return len;
1747 1748
}

R
Ronald S. Bultje 已提交
1749 1750 1751 1752 1753 1754 1755 1756 1757
static int rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt)
{
    RTSPState *rt = s->priv_data;
    int ret, len;
    uint8_t buf[10 * RTP_MAX_PACKET_LENGTH];
    RTSPStream *rtsp_st;

    /* get next frames from the same RTP packet */
    if (rt->cur_transport_priv) {
1758
        if (rt->transport == RTSP_TRANSPORT_RDT) {
R
Ronald S. Bultje 已提交
1759
            ret = ff_rdt_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
1760
        } else
R
Ronald S. Bultje 已提交
1761 1762 1763 1764 1765 1766
            ret = rtp_parse_packet(rt->cur_transport_priv, pkt, NULL, 0);
        if (ret == 0) {
            rt->cur_transport_priv = NULL;
            return 0;
        } else if (ret == 1) {
            return 0;
1767
        } else
R
Ronald S. Bultje 已提交
1768 1769 1770 1771 1772 1773 1774 1775 1776 1777 1778 1779 1780 1781 1782 1783 1784 1785 1786 1787 1788 1789 1790
            rt->cur_transport_priv = NULL;
    }

    /* read next RTP packet */
 redo:
    switch(rt->lower_transport) {
    default:
#if CONFIG_RTSP_DEMUXER
    case RTSP_LOWER_TRANSPORT_TCP:
        len = tcp_read_packet(s, &rtsp_st, buf, sizeof(buf));
        break;
#endif
    case RTSP_LOWER_TRANSPORT_UDP:
    case RTSP_LOWER_TRANSPORT_UDP_MULTICAST:
        len = udp_read_packet(s, &rtsp_st, buf, sizeof(buf));
        if (len >=0 && rtsp_st->transport_priv && rt->transport == RTSP_TRANSPORT_RTP)
            rtp_check_and_send_back_rr(rtsp_st->transport_priv, len);
        break;
    }
    if (len < 0)
        return len;
    if (len == 0)
        return AVERROR_EOF;
1791
    if (rt->transport == RTSP_TRANSPORT_RDT) {
R
Ronald S. Bultje 已提交
1792
        ret = ff_rdt_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
1793
    } else
R
Ronald S. Bultje 已提交
1794 1795 1796
        ret = rtp_parse_packet(rtsp_st->transport_priv, pkt, buf, len);
    if (ret < 0)
        goto redo;
1797
    if (ret == 1)
R
Ronald S. Bultje 已提交
1798 1799 1800 1801 1802 1803
        /* more packets may follow, so we save the RTP context */
        rt->cur_transport_priv = rtsp_st->transport_priv;

    return ret;
}

1804
static int rtsp_read_packet(AVFormatContext *s, AVPacket *pkt)
1805 1806
{
    RTSPState *rt = s->priv_data;
1807
    int ret;
1808 1809
    RTSPMessageHeader reply1, *reply = &reply1;
    char cmd[1024];
1810

1811
    if (rt->server_type == RTSP_SERVER_REAL) {
1812
        int i;
1813
        enum AVDiscard cache[MAX_STREAMS];
1814

1815 1816 1817 1818 1819 1820
        for (i = 0; i < s->nb_streams; i++)
            cache[i] = s->streams[i]->discard;

        if (!rt->need_subscription) {
            if (memcmp (cache, rt->real_setup_cache,
                        sizeof(enum AVDiscard) * s->nb_streams)) {
1821
                snprintf(cmd, sizeof(cmd),
R
Ronald S. Bultje 已提交
1822
                         "Unsubscribe: %s\r\n",
1823 1824 1825
                         rt->last_subscription);
                ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
                                 cmd, reply, NULL);
1826 1827 1828 1829
                if (reply->status_code != RTSP_STATUS_OK)
                    return AVERROR_INVALIDDATA;
                rt->need_subscription = 1;
            }
1830 1831
        }

1832 1833 1834 1835 1836 1837 1838 1839
        if (rt->need_subscription) {
            int r, rule_nr, first = 1;

            memcpy(rt->real_setup_cache, cache,
                   sizeof(enum AVDiscard) * s->nb_streams);
            rt->last_subscription[0] = 0;

            snprintf(cmd, sizeof(cmd),
1840
                     "Subscribe: ");
1841 1842 1843 1844 1845 1846 1847 1848 1849 1850 1851 1852 1853 1854 1855 1856 1857 1858
            for (i = 0; i < rt->nb_rtsp_streams; i++) {
                rule_nr = 0;
                for (r = 0; r < s->nb_streams; r++) {
                    if (s->streams[r]->priv_data == rt->rtsp_streams[i]) {
                        if (s->streams[r]->discard != AVDISCARD_ALL) {
                            if (!first)
                                av_strlcat(rt->last_subscription, ",",
                                           sizeof(rt->last_subscription));
                            ff_rdt_subscribe_rule(
                                rt->last_subscription,
                                sizeof(rt->last_subscription), i, rule_nr);
                            first = 0;
                        }
                        rule_nr++;
                    }
                }
            }
            av_strlcatf(cmd, sizeof(cmd), "%s\r\n", rt->last_subscription);
1859 1860
            ff_rtsp_send_cmd(s, "SET_PARAMETER", rt->control_uri,
                             cmd, reply, NULL);
1861 1862 1863 1864
            if (reply->status_code != RTSP_STATUS_OK)
                return AVERROR_INVALIDDATA;
            rt->need_subscription = 0;

1865
            if (rt->state == RTSP_STATE_STREAMING)
1866 1867
                rtsp_read_play (s);
        }
1868 1869
    }

L
Luca Barbato 已提交
1870
    ret = rtsp_fetch_packet(s, pkt);
1871
    if (ret < 0)
1872
        return ret;
1873 1874 1875 1876 1877 1878

    /* send dummy request to keep TCP connection alive */
    if ((rt->server_type == RTSP_SERVER_WMS ||
         rt->server_type == RTSP_SERVER_REAL) &&
        (av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
        if (rt->server_type == RTSP_SERVER_WMS) {
1879
            ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
1880
        } else {
1881
            ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL);
1882 1883 1884
        }
    }

1885
    return 0;
1886 1887
}

1888 1889
/* pause the stream */
static int rtsp_read_pause(AVFormatContext *s)
1890
{
1891
    RTSPState *rt = s->priv_data;
1892
    RTSPMessageHeader reply1, *reply = &reply1;
1893 1894

    rt = s->priv_data;
1895

1896
    if (rt->state != RTSP_STATE_STREAMING)
1897
        return 0;
1898
    else if (!(rt->server_type == RTSP_SERVER_REAL && rt->need_subscription)) {
1899
        ff_rtsp_send_cmd(s, "PAUSE", rt->control_uri, NULL, reply, NULL);
1900 1901 1902
        if (reply->status_code != RTSP_STATUS_OK) {
            return -1;
        }
1903
    }
1904 1905
    rt->state = RTSP_STATE_PAUSED;
    return 0;
1906 1907
}

1908
static int rtsp_read_seek(AVFormatContext *s, int stream_index,
1909
                          int64_t timestamp, int flags)
1910 1911
{
    RTSPState *rt = s->priv_data;
1912

1913 1914 1915
    rt->seek_timestamp = av_rescale_q(timestamp,
                                      s->streams[stream_index]->time_base,
                                      AV_TIME_BASE_Q);
1916 1917 1918 1919
    switch(rt->state) {
    default:
    case RTSP_STATE_IDLE:
        break;
1920
    case RTSP_STATE_STREAMING:
1921 1922 1923
        if (rtsp_read_pause(s) != 0)
            return -1;
        rt->state = RTSP_STATE_SEEKING;
1924 1925 1926 1927 1928 1929 1930 1931 1932 1933
        if (rtsp_read_play(s) != 0)
            return -1;
        break;
    case RTSP_STATE_PAUSED:
        rt->state = RTSP_STATE_IDLE;
        break;
    }
    return 0;
}

1934 1935 1936 1937
static int rtsp_read_close(AVFormatContext *s)
{
    RTSPState *rt = s->priv_data;

F
Fabrice Bellard 已提交
1938
#if 0
1939
    /* NOTE: it is valid to flush the buffer here */
1940
    if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
1941 1942
        url_fclose(&rt->rtsp_gb);
    }
F
Fabrice Bellard 已提交
1943
#endif
1944
    ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
1945

1946
    ff_rtsp_close_streams(s);
1947
    url_close(rt->rtsp_hd);
1948
    ff_network_close();
1949 1950 1951
    return 0;
}

1952
AVInputFormat rtsp_demuxer = {
1953
    "rtsp",
1954
    NULL_IF_CONFIG_SMALL("RTSP input format"),
1955 1956 1957 1958 1959
    sizeof(RTSPState),
    rtsp_probe,
    rtsp_read_header,
    rtsp_read_packet,
    rtsp_read_close,
1960
    rtsp_read_seek,
1961
    .flags = AVFMT_NOFILE,
1962 1963
    .read_play = rtsp_read_play,
    .read_pause = rtsp_read_pause,
1964
};
1965
#endif
1966

1967
static int sdp_probe(AVProbeData *p1)
1968
{
1969
    const char *p = p1->buf, *p_end = p1->buf + p1->buf_size;
1970 1971

    /* we look for a line beginning "c=IN IP4" */
1972
    while (p < p_end && *p != '\0') {
1973 1974
        if (p + sizeof("c=IN IP4") - 1 < p_end &&
            av_strstart(p, "c=IN IP4", NULL))
1975
            return AVPROBE_SCORE_MAX / 2;
1976

1977
        while (p < p_end - 1 && *p != '\n') p++;
1978
        if (++p >= p_end)
1979 1980 1981 1982
            break;
        if (*p == '\r')
            p++;
    }
1983 1984 1985 1986 1987
    return 0;
}

#define SDP_MAX_SIZE 8192

1988
static int sdp_read_header(AVFormatContext *s, AVFormatParameters *ap)
1989
{
1990
    RTSPState *rt = s->priv_data;
1991 1992 1993 1994 1995
    RTSPStream *rtsp_st;
    int size, i, err;
    char *content;
    char url[1024];

1996 1997 1998
    if (!ff_network_init())
        return AVERROR(EIO);

1999 2000 2001
    /* read the whole sdp file */
    /* XXX: better loading */
    content = av_malloc(SDP_MAX_SIZE);
2002
    size = get_buffer(s->pb, content, SDP_MAX_SIZE - 1);
2003 2004 2005 2006 2007 2008 2009 2010 2011 2012
    if (size <= 0) {
        av_free(content);
        return AVERROR_INVALIDDATA;
    }
    content[size] ='\0';

    sdp_parse(s, content);
    av_free(content);

    /* open each RTP stream */
2013
    for (i = 0; i < rt->nb_rtsp_streams; i++) {
2014
        rtsp_st = rt->rtsp_streams[i];
2015

2016 2017 2018 2019
        ff_url_join(url, sizeof(url), "rtp", NULL,
                    inet_ntoa(rtsp_st->sdp_ip), rtsp_st->sdp_port,
                    "?localport=%d&ttl=%d", rtsp_st->sdp_port,
                    rtsp_st->sdp_ttl);
2020
        if (url_open(&rtsp_st->rtp_handle, url, URL_RDWR) < 0) {
2021 2022 2023
            err = AVERROR_INVALIDDATA;
            goto fail;
        }
2024
        if ((err = rtsp_open_transport_ctx(s, rtsp_st)))
2025
            goto fail;
2026 2027
    }
    return 0;
2028
fail:
2029
    ff_rtsp_close_streams(s);
2030
    ff_network_close();
2031 2032 2033 2034 2035
    return err;
}

static int sdp_read_close(AVFormatContext *s)
{
2036
    ff_rtsp_close_streams(s);
2037
    ff_network_close();
2038 2039 2040
    return 0;
}

2041
AVInputFormat sdp_demuxer = {
2042
    "sdp",
2043
    NULL_IF_CONFIG_SMALL("SDP"),
2044 2045 2046
    sizeof(RTSPState),
    sdp_probe,
    sdp_read_header,
L
Luca Barbato 已提交
2047
    rtsp_fetch_packet,
2048 2049
    sdp_read_close,
};