rtp.c

/*****************************************************************************
 * rtp.c: rtp stream output module
 *****************************************************************************
 * Copyright (C) 2003-2004 the VideoLAN team
 * Copyright © 2007-2008 Rémi Denis-Courmont
 *
 * Authors: Laurent Aimar <fenrir@via.ecp.fr>
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * This program 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 General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
 *****************************************************************************/

/*****************************************************************************
 * Preamble
 *****************************************************************************/

#ifdef HAVE_CONFIG_H
# include "config.h"
#endif

#include <vlc_common.h>
#include <vlc_plugin.h>
#include <vlc_sout.h>
#include <vlc_block.h>

#include <vlc_httpd.h>
#include <vlc_url.h>
#include <vlc_network.h>
#include <vlc_charset.h>
#include <vlc_strings.h>
#include <srtp.h>

#include "rtp.h"

#ifdef HAVE_UNISTD_H
#   include <sys/types.h>
#   include <unistd.h>
#   include <fcntl.h>
#   include <sys/stat.h>
#endif
#ifdef HAVE_LINUX_DCCP_H
#   include <linux/dccp.h>
#endif
#ifndef IPPROTO_DCCP
# define IPPROTO_DCCP 33
#endif
#ifndef IPPROTO_UDPLITE
# define IPPROTO_UDPLITE 136
#endif

#include <errno.h>

#include <assert.h>

/*****************************************************************************
 * Module descriptor
 *****************************************************************************/

#define DEST_TEXT N_("Destination")
#define DEST_LONGTEXT N_( \
    "This is the output URL that will be used." )
#define SDP_TEXT N_("SDP")
#define SDP_LONGTEXT N_( \
    "This allows you to specify how the SDP (Session Descriptor) for this RTP "\
    "session will be made available. You must use an url: http://location to " \
    "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
    "for the SDP to be announced via SAP." )
#define SAP_TEXT N_("SAP announcing")
#define SAP_LONGTEXT N_("Announce this session with SAP.")
#define MUX_TEXT N_("Muxer")
#define MUX_LONGTEXT N_( \
    "This allows you to specify the muxer used for the streaming output. " \
    "Default is to use no muxer (standard RTP stream)." )

#define NAME_TEXT N_("Session name")
#define NAME_LONGTEXT N_( \
    "This is the name of the session that will be announced in the SDP " \
    "(Session Descriptor)." )
#define DESC_TEXT N_("Session description")
#define DESC_LONGTEXT N_( \
    "This allows you to give a short description with details about the stream, " \
    "that will be announced in the SDP (Session Descriptor)." )
#define URL_TEXT N_("Session URL")
#define URL_LONGTEXT N_( \
    "This allows you to give an URL with more details about the stream " \
    "(often the website of the streaming organization), that will " \
    "be announced in the SDP (Session Descriptor)." )
#define EMAIL_TEXT N_("Session email")
#define EMAIL_LONGTEXT N_( \
    "This allows you to give a contact mail address for the stream, that will " \
    "be announced in the SDP (Session Descriptor)." )
#define PHONE_TEXT N_("Session phone number")
#define PHONE_LONGTEXT N_( \
    "This allows you to give a contact telephone number for the stream, that will " \
    "be announced in the SDP (Session Descriptor)." )

#define PORT_TEXT N_("Port")
#define PORT_LONGTEXT N_( \
    "This allows you to specify the base port for the RTP streaming." )
#define PORT_AUDIO_TEXT N_("Audio port")
#define PORT_AUDIO_LONGTEXT N_( \
    "This allows you to specify the default audio port for the RTP streaming." )
#define PORT_VIDEO_TEXT N_("Video port")
#define PORT_VIDEO_LONGTEXT N_( \
    "This allows you to specify the default video port for the RTP streaming." )

#define TTL_TEXT N_("Hop limit (TTL)")
#define TTL_LONGTEXT N_( \
    "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
    "the multicast packets sent by the stream output (0 = use operating " \
    "system built-in default).")

#define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing")
#define RTCP_MUX_LONGTEXT N_( \
    "This sends and receives RTCP packet multiplexed over the same port " \
    "as RTP packets." )

#define PROTO_TEXT N_("Transport protocol")
#define PROTO_LONGTEXT N_( \
    "This selects which transport protocol to use for RTP." )

#define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)")
#define SRTP_KEY_LONGTEXT N_( \
    "RTP packets will be integrity-protected and ciphered "\
    "with this Secure RTP master shared secret key.")

#define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
#define SRTP_SALT_LONGTEXT N_( \
    "Secure RTP requires a (non-secret) master salt value.")

static const char *const ppsz_protos[] = {
    "dccp", "sctp", "tcp", "udp", "udplite",
};

static const char *const ppsz_protocols[] = {
    "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite",
};

#define RFC3016_TEXT N_("MP4A LATM")
#define RFC3016_LONGTEXT N_( \
    "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )

static int  Open ( vlc_object_t * );
static void Close( vlc_object_t * );

#define SOUT_CFG_PREFIX "sout-rtp-"
#define MAX_EMPTY_BLOCKS 200

vlc_module_begin();
    set_shortname( N_("RTP"));
    set_description( N_("RTP stream output") );
    set_capability( "sout stream", 0 );
    add_shortcut( "rtp" );
    set_category( CAT_SOUT );
    set_subcategory( SUBCAT_SOUT_STREAM );

    add_string( SOUT_CFG_PREFIX "dst", "", NULL, DEST_TEXT,
                DEST_LONGTEXT, true );
        change_unsafe();
    add_string( SOUT_CFG_PREFIX "sdp", "", NULL, SDP_TEXT,
                SDP_LONGTEXT, true );
    add_string( SOUT_CFG_PREFIX "mux", "", NULL, MUX_TEXT,
                MUX_LONGTEXT, true );
    add_bool( SOUT_CFG_PREFIX "sap", false, NULL, SAP_TEXT, SAP_LONGTEXT,
              true );

    add_string( SOUT_CFG_PREFIX "name", "", NULL, NAME_TEXT,
                NAME_LONGTEXT, true );
    add_string( SOUT_CFG_PREFIX "description", "", NULL, DESC_TEXT,
                DESC_LONGTEXT, true );
    add_string( SOUT_CFG_PREFIX "url", "", NULL, URL_TEXT,
                URL_LONGTEXT, true );
    add_string( SOUT_CFG_PREFIX "email", "", NULL, EMAIL_TEXT,
                EMAIL_LONGTEXT, true );
    add_string( SOUT_CFG_PREFIX "phone", "", NULL, PHONE_TEXT,
                PHONE_LONGTEXT, true );

    add_string( SOUT_CFG_PREFIX "proto", "udp", NULL, PROTO_TEXT,
                PROTO_LONGTEXT, false );
        change_string_list( ppsz_protos, ppsz_protocols, NULL );
    add_integer( SOUT_CFG_PREFIX "port", 50004, NULL, PORT_TEXT,
                 PORT_LONGTEXT, true );
    add_integer( SOUT_CFG_PREFIX "port-audio", 50000, NULL, PORT_AUDIO_TEXT,
                 PORT_AUDIO_LONGTEXT, true );
    add_integer( SOUT_CFG_PREFIX "port-video", 50002, NULL, PORT_VIDEO_TEXT,
                 PORT_VIDEO_LONGTEXT, true );

    add_integer( SOUT_CFG_PREFIX "ttl", 0, NULL, TTL_TEXT,
                 TTL_LONGTEXT, true );
    add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL,
              RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false );

    add_string( SOUT_CFG_PREFIX "key", "", NULL,
                SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false );
    add_string( SOUT_CFG_PREFIX "salt", "", NULL,
                SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false );

    add_bool( SOUT_CFG_PREFIX "mp4a-latm", 0, NULL, RFC3016_TEXT,
                 RFC3016_LONGTEXT, false );

    set_callbacks( Open, Close );
vlc_module_end();

/*****************************************************************************
 * Exported prototypes
 *****************************************************************************/
static const char *const ppsz_sout_options[] = {
    "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
    "sap", "description", "url", "email", "phone",
    "proto", "rtcp-mux", "key", "salt",
    "mp4a-latm", NULL
};

static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
static int               Del ( sout_stream_t *, sout_stream_id_t * );
static int               Send( sout_stream_t *, sout_stream_id_t *,
                               block_t* );
static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * );
static int               MuxDel ( sout_stream_t *, sout_stream_id_t * );
static int               MuxSend( sout_stream_t *, sout_stream_id_t *,
                                  block_t* );

static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
static void* ThreadSend( vlc_object_t *p_this );

static void SDPHandleUrl( sout_stream_t *, const char * );

static int SapSetup( sout_stream_t *p_stream );
static int FileSetup( sout_stream_t *p_stream );
static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );

struct sout_stream_sys_t
{
    /* SDP */
    char    *psz_sdp;
    vlc_mutex_t  lock_sdp;

    /* SDP to disk */
    bool b_export_sdp_file;
    char *psz_sdp_file;

    /* SDP via SAP */
    bool b_export_sap;
    session_descriptor_t *p_session;

    /* SDP via HTTP */
    httpd_host_t *p_httpd_host;
    httpd_file_t *p_httpd_file;

    /* RTSP */
    rtsp_stream_t *rtsp;

    /* */
    char     *psz_destination;
    uint8_t   proto;
    uint8_t   i_ttl;
    uint16_t  i_port;
    uint16_t  i_port_audio;
    uint16_t  i_port_video;
    bool b_latm;
    bool rtcp_mux;

    /* when need to use a private one or when using muxer */
    int i_payload_type;

    /* in case we do TS/PS over rtp */
    sout_mux_t        *p_mux;
    sout_access_out_t *p_grab;
    block_t           *packet;

    /* */
    vlc_mutex_t      lock_es;
    int              i_es;
    sout_stream_id_t **es;
};

typedef int (*pf_rtp_packetizer_t)( sout_stream_id_t *, block_t * );

typedef struct rtp_sink_t
{
    int rtp_fd;
    rtcp_sender_t *rtcp;
} rtp_sink_t;

struct sout_stream_id_t
{
    VLC_COMMON_MEMBERS

    sout_stream_t *p_stream;
    /* rtp field */
    uint16_t    i_sequence;
    uint8_t     i_payload_type;
    uint8_t     ssrc[4];

    /* for sdp */
    const char  *psz_enc;
    char        *psz_fmtp;
    int          i_clock_rate;
    int          i_port;
    int          i_cat;
    int          i_channels;
    int          i_bitrate;

    /* Packetizer specific fields */
    int                 i_mtu;
    srtp_session_t     *srtp;
    pf_rtp_packetizer_t pf_packetize;

    /* Packets sinks */
    vlc_mutex_t       lock_sink;
    int               sinkc;
    rtp_sink_t       *sinkv;
    rtsp_stream_id_t *rtsp_id;
    int              *listen_fd;

    block_fifo_t     *p_fifo;
    int64_t           i_caching;
};

/*****************************************************************************
 * Open:
 *****************************************************************************/
static int Open( vlc_object_t *p_this )
{
    sout_stream_t       *p_stream = (sout_stream_t*)p_this;
    sout_instance_t     *p_sout = p_stream->p_sout;
    sout_stream_sys_t   *p_sys = NULL;
    config_chain_t      *p_cfg = NULL;
    char                *psz;
    bool          b_rtsp = false;

    config_ChainParse( p_stream, SOUT_CFG_PREFIX,
                       ppsz_sout_options, p_stream->p_cfg );

    p_sys = malloc( sizeof( sout_stream_sys_t ) );
    if( p_sys == NULL )
        return VLC_ENOMEM;

    p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );

    p_sys->i_port       = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" );
    p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" );
    p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
    p_sys->rtcp_mux   = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );

    p_sys->psz_sdp_file = NULL;

    if( p_sys->i_port_audio == p_sys->i_port_video )
    {
        msg_Err( p_stream, "audio and video port cannot be the same" );
        p_sys->i_port_audio = 0;
        p_sys->i_port_video = 0;
    }

    for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
    {
        if( !strcmp( p_cfg->psz_name, "sdp" )
         && ( p_cfg->psz_value != NULL )
         && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
        {
            b_rtsp = true;
            break;
        }
    }
    if( !b_rtsp )
    {
        psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
        if( psz != NULL )
        {
            if( !strncasecmp( psz, "rtsp:", 5 ) )
                b_rtsp = true;
            free( psz );
        }
    }

    /* Transport protocol */
    p_sys->proto = IPPROTO_UDP;
    psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto");

    if ((psz == NULL) || !strcasecmp (psz, "udp"))
        (void)0; /* default */
    else
    if (!strcasecmp (psz, "dccp"))
    {
        p_sys->proto = IPPROTO_DCCP;
        p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
    }
#if 0
    else
    if (!strcasecmp (psz, "sctp"))
    {
        p_sys->proto = IPPROTO_TCP;
        p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
    }
#endif
#if 0
    else
    if (!strcasecmp (psz, "tcp"))
    {
        p_sys->proto = IPPROTO_TCP;
        p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */
    }
#endif
    else
    if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite"))
        p_sys->proto = IPPROTO_UDPLITE;
    else
        msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"",
                  psz);
    free (psz);
    var_Create (p_this, "dccp-service", VLC_VAR_STRING);

    if( ( p_sys->psz_destination == NULL ) && !b_rtsp )
    {
        msg_Err( p_stream, "missing destination and not in RTSP mode" );
        free( p_sys );
        return VLC_EGENERIC;
    }

    p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
    if( p_sys->i_ttl == 0 )
    {
        /* Normally, we should let the default hop limit up to the core,
         * but we have to know it to build our SDP properly, which is why
         * we ask the core. FIXME: broken when neither sout-rtp-ttl nor
         * ttl are set. */
        p_sys->i_ttl = config_GetInt( p_stream, "ttl" );
    }

    p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );

    p_sys->i_payload_type = 96;
    p_sys->i_es = 0;
    p_sys->es   = NULL;
    p_sys->rtsp = NULL;
    p_sys->psz_sdp = NULL;

    p_sys->b_export_sap = false;
    p_sys->b_export_sdp_file = false;
    p_sys->p_session = NULL;

    p_sys->p_httpd_host = NULL;
    p_sys->p_httpd_file = NULL;

    p_stream->p_sys     = p_sys;

    vlc_mutex_init( &p_sys->lock_sdp );
    vlc_mutex_init( &p_sys->lock_es );

    psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
    if( psz != NULL )
    {
        sout_stream_id_t *id;

        /* Check muxer type */
        if( strncasecmp( psz, "ps", 2 )
         && strncasecmp( psz, "mpeg1", 5 )
         && strncasecmp( psz, "ts", 2 ) )
        {
            msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
            free( psz );
            vlc_mutex_destroy( &p_sys->lock_sdp );
            vlc_mutex_destroy( &p_sys->lock_es );
            free( p_sys );
            return VLC_EGENERIC;
        }

        p_sys->p_grab = GrabberCreate( p_stream );
        p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab );
        free( psz );

        if( p_sys->p_mux == NULL )
        {
            msg_Err( p_stream, "cannot create muxer" );
            sout_AccessOutDelete( p_sys->p_grab );
            vlc_mutex_destroy( &p_sys->lock_sdp );
            vlc_mutex_destroy( &p_sys->lock_es );
            free( p_sys );
            return VLC_EGENERIC;
        }

        id = Add( p_stream, NULL );
        if( id == NULL )
        {
            sout_MuxDelete( p_sys->p_mux );
            sout_AccessOutDelete( p_sys->p_grab );
            vlc_mutex_destroy( &p_sys->lock_sdp );
            vlc_mutex_destroy( &p_sys->lock_es );
            free( p_sys );
            return VLC_EGENERIC;
        }

        p_sys->packet = NULL;

        p_stream->pf_add  = MuxAdd;
        p_stream->pf_del  = MuxDel;
        p_stream->pf_send = MuxSend;
    }
    else
    {
        p_sys->p_mux    = NULL;
        p_sys->p_grab   = NULL;

        p_stream->pf_add    = Add;
        p_stream->pf_del    = Del;
        p_stream->pf_send   = Send;
    }

    if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) )
        SDPHandleUrl( p_stream, "sap" );

    psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
    if( psz != NULL )
    {
        config_chain_t *p_cfg;

        SDPHandleUrl( p_stream, psz );

        for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
        {
            if( !strcmp( p_cfg->psz_name, "sdp" ) )
            {
                if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
                    continue;

                /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
                if( !strcmp( p_cfg->psz_value, psz ) )
                    continue;

                SDPHandleUrl( p_stream, p_cfg->psz_value );
            }
        }
        free( psz );
    }

    /* update p_sout->i_out_pace_nocontrol */
    p_stream->p_sout->i_out_pace_nocontrol++;

    return VLC_SUCCESS;
}

/*****************************************************************************
 * Close:
 *****************************************************************************/
static void Close( vlc_object_t * p_this )
{
    sout_stream_t     *p_stream = (sout_stream_t*)p_this;
    sout_stream_sys_t *p_sys = p_stream->p_sys;

    /* update p_sout->i_out_pace_nocontrol */
    p_stream->p_sout->i_out_pace_nocontrol--;

    if( p_sys->p_mux )
    {
        assert( p_sys->i_es == 1 );
        Del( p_stream, p_sys->es[0] );

        sout_MuxDelete( p_sys->p_mux );
        sout_AccessOutDelete( p_sys->p_grab );
        if( p_sys->packet )
        {
            block_Release( p_sys->packet );
        }
        if( p_sys->b_export_sap )
        {
            p_sys->p_mux = NULL;
            SapSetup( p_stream );
        }
    }

    if( p_sys->rtsp != NULL )
        RtspUnsetup( p_sys->rtsp );

    vlc_mutex_destroy( &p_sys->lock_sdp );
    vlc_mutex_destroy( &p_sys->lock_es );

    if( p_sys->p_httpd_file )
        httpd_FileDelete( p_sys->p_httpd_file );

    if( p_sys->p_httpd_host )
        httpd_HostDelete( p_sys->p_httpd_host );

    free( p_sys->psz_sdp );

    if( p_sys->b_export_sdp_file )
    {
#ifdef HAVE_UNISTD_H
        unlink( p_sys->psz_sdp_file );
#endif
        free( p_sys->psz_sdp_file );
    }
    free( p_sys->psz_destination );
    free( p_sys );
}

/*****************************************************************************
 * SDPHandleUrl:
 *****************************************************************************/
static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url )
{
    sout_stream_sys_t *p_sys = p_stream->p_sys;
    vlc_url_t url;

    vlc_UrlParse( &url, psz_url, 0 );
    if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) )
    {
        if( p_sys->p_httpd_file )
        {
            msg_Err( p_stream, "you can use sdp=http:// only once" );
            goto out;
        }

        if( HttpSetup( p_stream, &url ) )
        {
            msg_Err( p_stream, "cannot export SDP as HTTP" );
        }
    }
    else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) )
    {
        if( p_sys->rtsp != NULL )
        {
            msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
            goto out;
        }

        /* FIXME test if destination is multicast or no destination at all */
        p_sys->rtsp = RtspSetup( p_stream, &url );
        if( p_sys->rtsp == NULL )
        {
            msg_Err( p_stream, "cannot export SDP as RTSP" );
        }

        if( p_sys->p_mux != NULL )
        {
            sout_stream_id_t *id = p_sys->es[0];
            id->rtsp_id = RtspAddId( p_sys->rtsp, id, 0, GetDWBE( id->ssrc ),
                                     p_sys->psz_destination, p_sys->i_ttl,
                                     id->i_port, id->i_port + 1 );
        }
    }
    else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
             ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
    {
        p_sys->b_export_sap = true;
        SapSetup( p_stream );
    }
    else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
    {
        if( p_sys->b_export_sdp_file )
        {
            msg_Err( p_stream, "you can use sdp=file:// only once" );
            goto out;
        }
        p_sys->b_export_sdp_file = true;
        psz_url = &psz_url[5];
        if( psz_url[0] == '/' && psz_url[1] == '/' )
            psz_url += 2;
        p_sys->psz_sdp_file = strdup( psz_url );
    }
    else
    {
        msg_Warn( p_stream, "unknown protocol for SDP (%s)",
                  url.psz_protocol );
    }

out:
    vlc_UrlClean( &url );
}

/*****************************************************************************
 * SDPGenerate
 *****************************************************************************/
/*static*/
char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url )
{
    const sout_stream_sys_t *p_sys = p_stream->p_sys;
    char *psz_sdp;
    struct sockaddr_storage dst;
    socklen_t dstlen;
    int i;
    /*
     * When we have a fixed destination (typically when we do multicast),
     * we need to put the actual port numbers in the SDP.
     * When there is no fixed destination, we only support RTSP unicast
     * on-demand setup, so we should rather let the clients decide which ports
     * to use.
     * When there is both a fixed destination and RTSP unicast, we need to
     * put port numbers used by the fixed destination, otherwise the SDP would
     * become totally incorrect for multicast use. It should be noted that
     * port numbers from SDP with RTSP are only "recommendation" from the
     * server to the clients (per RFC2326), so only broken clients will fail
     * to handle this properly. There is no solution but to use two differents
     * output chain with two different RTSP URLs if you need to handle this
     * scenario.
     */
    int inclport;

    if( p_sys->psz_destination != NULL )
    {
        inclport = 1;

        /* Oh boy, this is really ugly! (+ race condition on lock_es) */
        dstlen = sizeof( dst );
        if( p_sys->es[0]->listen_fd != NULL )
            getsockname( p_sys->es[0]->listen_fd[0],
                         (struct sockaddr *)&dst, &dstlen );
        else
            getpeername( p_sys->es[0]->sinkv[0].rtp_fd,
                         (struct sockaddr *)&dst, &dstlen );
    }
    else
    {
        inclport = 0;

        /* Dummy destination address for RTSP */
        memset (&dst, 0, sizeof( struct sockaddr_in ) );
        dst.ss_family = AF_INET;
#ifdef HAVE_SA_LEN
        dst.ss_len =
#endif
        dstlen = sizeof( struct sockaddr_in );
    }

    psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
                             NULL, 0, (struct sockaddr *)&dst, dstlen );
    if( psz_sdp == NULL )
        return NULL;

    /* TODO: a=source-filter */
    if( p_sys->rtcp_mux )
        sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );

    if( rtsp_url != NULL )
        sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );

    /* FIXME: locking?! */
    for( i = 0; i < p_sys->i_es; i++ )
    {
        sout_stream_id_t *id = p_sys->es[i];
        const char *mime_major; /* major MIME type */
        const char *proto = "RTP/AVP"; /* protocol */

        switch( id->i_cat )
        {
            case VIDEO_ES:
                mime_major = "video";
                break;
            case AUDIO_ES:
                mime_major = "audio";
                break;
            case SPU_ES:
                mime_major = "text";
                break;
            default:
                continue;
        }

        if( rtsp_url == NULL )
        {
            switch( p_sys->proto )
            {
                case IPPROTO_UDP:
                    break;
                case IPPROTO_TCP:
                    proto = "TCP/RTP/AVP";
                    break;
                case IPPROTO_DCCP:
                    proto = "DCCP/RTP/AVP";
                    break;
                case IPPROTO_UDPLITE:
                    continue;
            }
        }

        sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
                      id->i_payload_type, false, id->i_bitrate,
                      id->psz_enc, id->i_clock_rate, id->i_channels,
                      id->psz_fmtp);

        if( rtsp_url != NULL )
        {
            assert( strlen( rtsp_url ) > 0 );
            bool addslash = ( rtsp_url[strlen( rtsp_url ) - 1] != '/' );
            sdp_AddAttribute ( &psz_sdp, "control",
                               addslash ? "%s/trackID=%u" : "%strackID=%u",
                               rtsp_url, i );
        }
        else
        {
            if( id->listen_fd != NULL )
                sdp_AddAttribute( &psz_sdp, "setup", "passive" );
            if( p_sys->proto == IPPROTO_DCCP )
                sdp_AddAttribute( &psz_sdp, "dccp-service-code", 
                                  "SC:RTP%c", toupper( mime_major[0] ) );
        }
    }

    return psz_sdp;
}

/*****************************************************************************
 * RTP mux
 *****************************************************************************/

static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
{
    static const char hex[16] = "0123456789abcdef";
    int i;

    for( i = 0; i < i_data; i++ )
    {
        s[2*i+0] = hex[(p_data[i]>>4)&0xf];
        s[2*i+1] = hex[(p_data[i]   )&0xf];
    }
    s[2*i_data] = '\0';
}

/**
 * Shrink the MTU down to a fixed packetization time (for audio).
 */
static void
rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
{
    /* Samples per second */
    size_t spl = (id->i_clock_rate - 1) * ptime_ms / 1000 + 1;
    bytes *= id->i_channels;
    spl *= bytes;

    if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
        id->i_mtu = 12 + spl;
    else /* MTU is too small for ptime, align to a sample boundary */
        id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
}

/** Add an ES as a new RTP stream */
static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
{
    /* NOTE: As a special case, if we use a non-RTP
     * mux (TS/PS), then p_fmt is NULL. */
    sout_stream_sys_t *p_sys = p_stream->p_sys;
    sout_stream_id_t  *id;
    int               i_port, cscov = -1;
    char              *psz_sdp;

    id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) );
    if( id == NULL )
        return NULL;
    vlc_object_attach( id, p_stream );

    /* Choose the port */
    i_port = 0;
    if( p_fmt == NULL )
        ;
    else
    if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
    {
        i_port = p_sys->i_port_audio;
        p_sys->i_port_audio = 0;
    }
    else
    if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
    {
        i_port = p_sys->i_port_video;
        p_sys->i_port_video = 0;
    }

    while( i_port == 0 )
    {
        if( p_sys->i_port != p_sys->i_port_audio
         && p_sys->i_port != p_sys->i_port_video )
        {
            i_port = p_sys->i_port;
            p_sys->i_port += 2;
            break;
        }
        p_sys->i_port += 2;
    }

    id->p_stream   = p_stream;

    id->i_sequence = rand()&0xffff;
    id->i_payload_type = p_sys->i_payload_type;
    id->ssrc[0] = rand()&0xff;
    id->ssrc[1] = rand()&0xff;
    id->ssrc[2] = rand()&0xff;
    id->ssrc[3] = rand()&0xff;

    id->psz_enc    = NULL;
    id->psz_fmtp   = NULL;
    id->i_clock_rate = 90000; /* most common case for video */
    id->i_channels = 0;
    id->i_port     = i_port;
    if( p_fmt != NULL )
    {
        id->i_cat  = p_fmt->i_cat;
        if( p_fmt->i_cat == AUDIO_ES )
        {
            id->i_clock_rate = p_fmt->audio.i_rate;
            id->i_channels = p_fmt->audio.i_channels;
        }
        id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
    }
    else
    {
        id->i_cat  = VIDEO_ES;
        id->i_bitrate = 0;
    }

    id->i_mtu = config_GetInt( p_stream, "mtu" );
    if( id->i_mtu <= 12 + 16 )
        id->i_mtu = 576 - 20 - 8; /* pessimistic */
    msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );

    id->srtp = NULL;
    id->pf_packetize = NULL;

    char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
    if (key)
    {
        id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
                                   SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
        if (id->srtp == NULL)
        {
            free (key);
            goto error;
        }

        char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
        errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
        free (salt);
        free (key);
        if (errno)
        {
            msg_Err (p_stream, "bad SRTP key/salt combination (%m)");
            goto error;
        }