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rtp.c
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/*****************************************************************************
* rtp.c: rtp stream output module
*****************************************************************************
* Copyright (C) 2003-2004, 2010 the VideoLAN team
* Copyright © 2007-2008 Rémi Denis-Courmont
*
* Authors: Laurent Aimar <[email protected]>
* Pierre Ynard
*
* 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
#define VLC_MODULE_LICENSE VLC_LICENSE_GPL_2_PLUS
#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_fs.h>
#include <vlc_rand.h>
#ifdef HAVE_SRTP
# include <srtp.h>
# include <gcrypt.h>
# include <vlc_gcrypt.h>
#endif
#include "rtp.h"
#include <sys/types.h>
#include <unistd.h>
#ifdef HAVE_ARPA_INET_H
# include <arpa/inet.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 <ctype.h>
#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 a 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 CAT_TEXT N_("Session category")
#define CAT_LONGTEXT N_( \
"This allows you to specify a category for the session, " \
"that will be announced if you choose to use SAP." )
#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 a 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 (-1 = 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 CACHING_TEXT N_("Caching value (ms)")
#define CACHING_LONGTEXT N_( \
"Default caching value for outbound RTP streams. This " \
"value should be set in milliseconds." )
#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. "\
"This must be a 32-character-long hexadecimal string.")
#define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
#define SRTP_SALT_LONGTEXT N_( \
"Secure RTP requires a (non-secret) master salt value. " \
"This must be a 28-character-long hexadecimal string.")
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)." )
#define RTSP_TIMEOUT_TEXT N_( "RTSP session timeout (s)" )
#define RTSP_TIMEOUT_LONGTEXT N_( "RTSP sessions will be closed after " \
"not receiving any RTSP request for this long. Setting it to a " \
"negative value or zero disables timeouts. The default is 60 (one " \
"minute)." )
#define RTSP_USER_TEXT N_("Username")
#define RTSP_USER_LONGTEXT N_("Username that will be " \
"requested to access the stream." )
#define RTSP_PASS_TEXT N_("Password")
#define RTSP_PASS_LONGTEXT N_("Password that will be " \
"requested to access the stream." )
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", "vod" )
set_category( CAT_SOUT )
set_subcategory( SUBCAT_SOUT_STREAM )
add_string( SOUT_CFG_PREFIX "dst", "", DEST_TEXT,
DEST_LONGTEXT, true )
add_string( SOUT_CFG_PREFIX "sdp", "", SDP_TEXT,
SDP_LONGTEXT, true )
add_string( SOUT_CFG_PREFIX "mux", "", MUX_TEXT,
MUX_LONGTEXT, true )
add_bool( SOUT_CFG_PREFIX "sap", false, SAP_TEXT, SAP_LONGTEXT,
true )
add_string( SOUT_CFG_PREFIX "name", "", NAME_TEXT,
NAME_LONGTEXT, true )
add_string( SOUT_CFG_PREFIX "cat", "", CAT_TEXT, CAT_LONGTEXT, true )
add_string( SOUT_CFG_PREFIX "description", "", DESC_TEXT,
DESC_LONGTEXT, true )
add_string( SOUT_CFG_PREFIX "url", "", URL_TEXT,
URL_LONGTEXT, true )
add_string( SOUT_CFG_PREFIX "email", "", EMAIL_TEXT,
EMAIL_LONGTEXT, true )
add_string( SOUT_CFG_PREFIX "phone", "", PHONE_TEXT,
PHONE_LONGTEXT, true )
add_string( SOUT_CFG_PREFIX "proto", "udp", PROTO_TEXT,
PROTO_LONGTEXT, false )
change_string_list( ppsz_protos, ppsz_protocols )
add_integer( SOUT_CFG_PREFIX "port", 5004, PORT_TEXT,
PORT_LONGTEXT, true )
add_integer( SOUT_CFG_PREFIX "port-audio", 0, PORT_AUDIO_TEXT,
PORT_AUDIO_LONGTEXT, true )
add_integer( SOUT_CFG_PREFIX "port-video", 0, PORT_VIDEO_TEXT,
PORT_VIDEO_LONGTEXT, true )
add_integer( SOUT_CFG_PREFIX "ttl", -1, TTL_TEXT,
TTL_LONGTEXT, true )
add_bool( SOUT_CFG_PREFIX "rtcp-mux", false,
RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000,
CACHING_TEXT, CACHING_LONGTEXT, true )
#ifdef HAVE_SRTP
add_string( SOUT_CFG_PREFIX "key", "",
SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false )
add_string( SOUT_CFG_PREFIX "salt", "",
SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false )
#endif
add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, RFC3016_TEXT,
RFC3016_LONGTEXT, false )
set_callbacks( Open, Close )
add_submodule ()
set_shortname( N_("RTSP VoD" ) )
set_description( N_("RTSP VoD server") )
set_category( CAT_SOUT )
set_subcategory( SUBCAT_SOUT_VOD )
set_capability( "vod server", 10 )
set_callbacks( OpenVoD, CloseVoD )
add_shortcut( "rtsp" )
add_integer( "rtsp-timeout", 60, RTSP_TIMEOUT_TEXT,
RTSP_TIMEOUT_LONGTEXT, true )
add_string( "sout-rtsp-user", "",
RTSP_USER_TEXT, RTSP_USER_LONGTEXT, true )
add_password( "sout-rtsp-pwd", "",
RTSP_PASS_TEXT, RTSP_PASS_LONGTEXT, true )
vlc_module_end ()
/*****************************************************************************
* Exported prototypes
*****************************************************************************/
static const char *const ppsz_sout_options[] = {
"dst", "name", "cat", "port", "port-audio", "port-video", "*sdp", "ttl",
"mux", "sap", "description", "url", "email", "phone",
"proto", "rtcp-mux", "caching",
#ifdef HAVE_SRTP
"key", "salt",
#endif
"mp4a-latm", NULL
};
static sout_stream_id_sys_t *Add( sout_stream_t *, const es_format_t * );
static void Del ( sout_stream_t *, sout_stream_id_sys_t * );
static int Send( sout_stream_t *, sout_stream_id_sys_t *,
block_t* );
static sout_stream_id_sys_t *MuxAdd( sout_stream_t *, const es_format_t * );
static void MuxDel ( sout_stream_t *, sout_stream_id_sys_t * );
static int MuxSend( sout_stream_t *, sout_stream_id_sys_t *,
block_t* );
static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
static void* ThreadSend( void * );
static void *rtp_listen_thread( void * );
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 * );
static int64_t rtp_init_ts( const vod_media_t *p_media,
const char *psz_vod_session );
struct sout_stream_sys_t
{
/* SDP */
char *psz_sdp;
vlc_mutex_t lock_sdp;
/* SDP to disk */
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;
/* RTSP NPT and timestamp computations */
mtime_t i_npt_zero; /* when NPT=0 packet is sent */
int64_t i_pts_zero; /* predicts PTS of NPT=0 packet */
int64_t i_pts_offset; /* matches actual PTS to prediction */
vlc_mutex_t lock_ts;
/* */
char *psz_destination;
uint16_t i_port;
uint16_t i_port_audio;
uint16_t i_port_video;
uint8_t proto;
bool rtcp_mux;
bool b_latm;
/* VoD */
vod_media_t *p_vod_media;
char *psz_vod_session;
/* 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_sys_t **es;
};
typedef struct rtp_sink_t
{
int rtp_fd;
rtcp_sender_t *rtcp;
} rtp_sink_t;
struct sout_stream_id_sys_t
{
sout_stream_t *p_stream;
/* rtp field */
/* For RFC 4175, seqnum is extended to 32-bits */
uint32_t i_sequence;
bool b_first_packet;
bool b_ts_init;
uint32_t i_ts_offset;
uint8_t ssrc[4];
/* for rtsp */
uint16_t i_seq_sent_next;
/* for sdp */
rtp_format_t rtp_fmt;
int i_port;
/* Packetizer specific fields */
int i_mtu;
#ifdef HAVE_SRTP
srtp_session_t *srtp;
#endif
/* Packets sinks */
vlc_thread_t thread;
vlc_mutex_t lock_sink;
int sinkc;
rtp_sink_t *sinkv;
rtsp_stream_id_t *rtsp_id;
struct {
int *fd;
vlc_thread_t thread;
} listen;
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_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" );
if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
{
msg_Err( p_stream, "audio and video RTP port must be distinct" );
free( p_sys->psz_destination );
free( p_sys );
return VLC_EGENERIC;
}
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);
p_sys->p_vod_media = NULL;
p_sys->psz_vod_session = NULL;
if (! strcmp(p_stream->psz_name, "vod"))
{
/* The VLM stops all instances before deleting a media, so this
* reference will remain valid during the lifetime of the rtp
* stream output. */
p_sys->p_vod_media = var_InheritAddress(p_stream, "vod-media");
if (p_sys->p_vod_media != NULL)
{
p_sys->psz_vod_session = var_InheritString(p_stream, "vod-session");
if (p_sys->psz_vod_session == NULL)
{
msg_Err(p_stream, "missing VoD session");
free(p_sys);
return VLC_EGENERIC;
}
const char *mux = vod_get_mux(p_sys->p_vod_media);
var_SetString(p_stream, SOUT_CFG_PREFIX "mux", mux);
}
}
if( p_sys->psz_destination == NULL && !b_rtsp
&& p_sys->p_vod_media == NULL )
{
msg_Err( p_stream, "missing destination and not in RTSP mode" );
free( p_sys );
return VLC_EGENERIC;
}
int i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
if( i_ttl != -1 )
{
var_Create( p_stream, "ttl", VLC_VAR_INTEGER );
var_SetInteger( p_stream, "ttl", i_ttl );
}
p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
/* NPT=0 time will be determined when we packetize the first packet
* (of any ES). But we want to be able to report rtptime in RTSP
* without waiting (and already did in the VoD case). So until then,
* we use an arbitrary reference PTS for timestamp computations, and
* then actual PTS will catch up using offsets. */
p_sys->i_npt_zero = VLC_TS_INVALID;
p_sys->i_pts_zero = rtp_init_ts(p_sys->p_vod_media,
p_sys->psz_vod_session);
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->p_session = NULL;
p_sys->psz_sdp_file = 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_ts );
vlc_mutex_init( &p_sys->lock_es );
psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
if( psz != NULL )
{
/* 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_ts );
vlc_mutex_destroy( &p_sys->lock_es );
free( p_sys->psz_vod_session );
free( p_sys->psz_destination );
free( p_sys );
return VLC_EGENERIC;
}
p_sys->p_grab = GrabberCreate( p_stream );
p_sys->p_mux = sout_MuxNew( p_stream->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_ts );
vlc_mutex_destroy( &p_sys->lock_es );
free( p_sys->psz_vod_session );
free( p_sys->psz_destination );
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;
}
p_stream->pace_nocontrol = true;
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 );
}
if( p_sys->p_mux != NULL )
{
sout_stream_id_sys_t *id = Add( p_stream, NULL );
if( id == NULL )
{
Close( p_this );
return VLC_EGENERIC;
}
}
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;
if( p_sys->p_mux )
{
assert( p_sys->i_es <= 1 );
sout_MuxDelete( p_sys->p_mux );
if ( p_sys->i_es > 0 )
Del( p_stream, p_sys->es[0] );
sout_AccessOutDelete( p_sys->p_grab );
if( p_sys->packet )
{
block_Release( p_sys->packet );
}
}
if( p_sys->rtsp != NULL )
RtspUnsetup( p_sys->rtsp );
vlc_mutex_destroy( &p_sys->lock_sdp );
vlc_mutex_destroy( &p_sys->lock_ts );
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->psz_sdp_file != NULL )
{
unlink( p_sys->psz_sdp_file );
free( p_sys->psz_sdp_file );
}
free( p_sys->psz_vod_session );
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 );
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;
}
if( url.psz_host != NULL && *url.psz_host )
{
msg_Warn( p_stream, "\"%s\" RTSP host might be ignored in "
"multiple-host configurations, use at your own risks.",
url.psz_host );
msg_Info( p_stream, "Consider passing --rtsp-host=IP on the "
"command line instead." );
var_Create( p_stream, "rtsp-host", VLC_VAR_STRING );
var_SetString( p_stream, "rtsp-host", url.psz_host );
}
if( url.i_port != 0 )
{
/* msg_Info( p_stream, "Consider passing --rtsp-port=%u on "
"the command line instead.", url.i_port ); */
var_Create( p_stream, "rtsp-port", VLC_VAR_INTEGER );
var_SetInteger( p_stream, "rtsp-port", url.i_port );
}
p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), NULL, url.psz_path );
if( p_sys->rtsp == NULL )
msg_Err( p_stream, "cannot export SDP as RTSP" );
}
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->psz_sdp_file != NULL )
{
msg_Err( p_stream, "you can use sdp=file:// only once" );
goto out;
}
p_sys->psz_sdp_file = vlc_uri2path( psz_url );
if( p_sys->psz_sdp_file == NULL )
goto out;
FileSetup( p_stream );
}
else
{
msg_Warn( p_stream, "unknown protocol for SDP (%s)",
url.psz_protocol );
}
out:
vlc_UrlClean( &url );
}
/*****************************************************************************
* SDPGenerate
*****************************************************************************/
/*static*/
char *SDPGenerate( sout_stream_t *p_stream, const char *rtsp_url )
{
sout_stream_sys_t *p_sys = p_stream->p_sys;
char *psz_sdp = NULL;
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;
vlc_mutex_lock( &p_sys->lock_es );
if( unlikely(p_sys->i_es == 0 || (rtsp_url != NULL && !p_sys->es[0]->rtsp_id)) )
goto out; /* hmm... */
if( p_sys->psz_destination != NULL )
{
inclport = 1;
/* Oh boy, this is really ugly! */
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;
/* Check against URL format rtsp://[<ipv6>]:<port>/<path> */
bool ipv6 = rtsp_url != NULL && strlen( rtsp_url ) > 7
&& rtsp_url[7] == '[';
/* Dummy destination address for RTSP */
dstlen = ipv6 ? sizeof( struct sockaddr_in6 )
: sizeof( struct sockaddr_in );
memset (&dst, 0, dstlen);
dst.ss_family = ipv6 ? AF_INET6 : AF_INET;
#ifdef HAVE_SA_LEN
dst.ss_len = dstlen;
#endif
}
psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
NULL, 0, (struct sockaddr *)&dst, dstlen );
if( psz_sdp == NULL )
goto out;
/* 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 );
const char *proto = "RTP/AVP"; /* protocol */
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:
return psz_sdp;
}
}
for( i = 0; i < p_sys->i_es; i++ )
{
sout_stream_id_sys_t *id = p_sys->es[i];
rtp_format_t *rtp_fmt = &id->rtp_fmt;
const char *mime_major; /* major MIME type */
switch( rtp_fmt->cat )
{
case VIDEO_ES:
mime_major = "video";
break;
case AUDIO_ES:
mime_major = "audio";
break;
case SPU_ES:
mime_major = "text";
break;
default:
continue;
}
sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
rtp_fmt->payload_type, false, rtp_fmt->bitrate,
rtp_fmt->ptname, rtp_fmt->clock_rate, rtp_fmt->channels,
rtp_fmt->fmtp);
/* cf RFC4566 §5.14 */
if( inclport && !p_sys->rtcp_mux && (id->i_port & 1) )
sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
if( rtsp_url != NULL )
{
char *track_url = RtspAppendTrackPath( id->rtsp_id, rtsp_url );
if( track_url != NULL )
{
sdp_AddAttribute ( &psz_sdp, "control", "%s", track_url );
free( track_url );
}
}
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( (unsigned char)mime_major[0] ) );
}
}
out:
vlc_mutex_unlock( &p_sys->lock_es );
return psz_sdp;
}
/*****************************************************************************
* RTP mux
*****************************************************************************/
/**
* Shrink the MTU down to a fixed packetization time (for audio).
*/
static void
rtp_set_ptime (sout_stream_id_sys_t *id, unsigned ptime_ms, size_t bytes)
{
/* Samples per second */
size_t spl = (id->rtp_fmt.clock_rate - 1) * ptime_ms / 1000 + 1;
bytes *= id->rtp_fmt.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);
}
uint32_t rtp_compute_ts( unsigned i_clock_rate, int64_t i_pts )
{
/* This is an overflow-proof way of doing:
* return i_pts * (int64_t)i_clock_rate / CLOCK_FREQ;
*
* NOTE: this plays nice with offsets because the (equivalent)
* calculations are linear. */
lldiv_t q = lldiv(i_pts, CLOCK_FREQ);
return q.quot * (int64_t)i_clock_rate
+ q.rem * (int64_t)i_clock_rate / CLOCK_FREQ;
}
/** Add an ES as a new RTP stream */
static sout_stream_id_sys_t *Add( sout_stream_t *p_stream,
const 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;
char *psz_sdp;
sout_stream_id_sys_t *id = malloc( sizeof( *id ) );
if( unlikely(id == NULL) )
return NULL;
id->p_stream = p_stream;
id->i_mtu = var_InheritInteger( 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 );
#ifdef HAVE_SRTP
id->srtp = NULL;
#endif
vlc_mutex_init( &id->lock_sink );
id->sinkc = 0;
id->sinkv = NULL;
id->rtsp_id = NULL;
id->p_fifo = NULL;
id->listen.fd = NULL;
id->b_first_packet = true;
id->i_caching =
(int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
bool format = false;
if (p_sys->p_vod_media != NULL)
{
id->rtp_fmt.ptname = NULL;
uint32_t ssrc;
int val = vod_init_id(p_sys->p_vod_media, p_sys->psz_vod_session,
p_fmt ? p_fmt->i_id : 0, id, &id->rtp_fmt,
&ssrc, &id->i_seq_sent_next);