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vlc/modules/stream_out/rtp.c

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/*****************************************************************************
* 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_fs.h>
#include <vlc_strings.h>
#include <vlc_rand.h>
#ifdef HAVE_SRTP
# include <srtp.h>
#endif
#include "rtp.h"
#ifdef HAVE_UNISTD_H
# include <sys/types.h>
# include <unistd.h>
#endif
#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 <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 (-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.")
#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 )
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", 5004, NULL, PORT_TEXT,
PORT_LONGTEXT, true )
add_integer( SOUT_CFG_PREFIX "port-audio", 0, NULL, PORT_AUDIO_TEXT,
PORT_AUDIO_LONGTEXT, true )
add_integer( SOUT_CFG_PREFIX "port-video", 0, NULL, PORT_VIDEO_TEXT,
PORT_VIDEO_LONGTEXT, true )
add_integer( SOUT_CFG_PREFIX "ttl", -1, NULL, TTL_TEXT,
TTL_LONGTEXT, true )
add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL,
RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000, NULL,
CACHING_TEXT, CACHING_LONGTEXT, true )
#ifdef HAVE_SRTP
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 )
#endif
add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, 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", "caching", "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( 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 * );
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;
uint32_t payload_bitmap;
uint16_t i_port;
uint16_t i_port_audio;
uint16_t i_port_video;
uint8_t proto;
bool rtcp_mux;
int i_ttl:9;
bool b_latm;
/* 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
{
sout_stream_t *p_stream;
/* rtp field */
uint16_t i_sequence;
uint8_t i_payload_type;
bool b_ts_init;
uint32_t i_ts_offset;
uint8_t ssrc[4];
/* for rtsp */
uint16_t i_seq_sent_next;
/* 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;
#ifdef HAVE_SRTP
srtp_session_t *srtp;
#endif
pf_rtp_packetizer_t pf_packetize;
/* 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_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" );
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);
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 == -1 )
{
/* Normally, we should let the default hop limit up to the core,
* but we have to know it to write our RTSP headers properly,
* which is why we ask the core. FIXME: broken when neither
* sout-rtp-ttl nor ttl are set. */
p_sys->i_ttl = var_InheritInteger( p_stream, "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. 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 = mdate(); /* arbitrary value, could probably be
* random */
p_sys->payload_bitmap = 0xFFFFFFFF;
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 )
{
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->psz_destination );
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->psz_destination );
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->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;
}
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 );
sout_MuxDelete( p_sys->p_mux );
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->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_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 )
{
#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" );
else
if( p_sys->p_mux != NULL )
{
sout_stream_id_t *id = p_sys->es[0];
id->rtsp_id = RtspAddId( p_sys->rtsp, id, 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->psz_sdp_file != NULL )
{
msg_Err( p_stream, "you can use sdp=file:// only once" );
goto out;
}
p_sys->psz_sdp_file = make_path( 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) )
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_t *id = p_sys->es[i];
const char *mime_major; /* major MIME type */
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;
}
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( !p_sys->rtcp_mux && (id->i_port & 1) ) /* cf RFC4566 §5.14 */
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( mime_major[0] ) );
}
}
out:
vlc_mutex_unlock( &p_sys->lock_es );
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);
}
uint32_t rtp_compute_ts( const sout_stream_id_t *id, int64_t i_pts )
{
/* NOTE: this plays nice with offsets because the calculations are
* linear. */
return i_pts * (int64_t)id->i_clock_rate / CLOCK_FREQ;
}
/** 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;
char *psz_sdp;
if (0 == p_sys->payload_bitmap)
{
msg_Err (p_stream, "too many RTP elementary streams");
return NULL;
}
/* Choose the port */
uint16_t 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;
else
if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
i_port = p_sys->i_port_video;
/* We do not need the ES lock (p_sys->lock_es) here, because this is the
* only one thread that can *modify* the ES table. The ES lock protects
* the other threads from our modifications (TAB_APPEND, TAB_REMOVE). */
for (int i = 0; i_port && (i < p_sys->i_es); i++)
if (i_port == p_sys->es[i]->i_port)
i_port = 0; /* Port already in use! */
for (uint16_t p = p_sys->i_port; i_port == 0; p += 2)
{
if (p == 0)
{
msg_Err (p_stream, "too many RTP elementary streams");
return NULL;
}
i_port = p;
for (int i = 0; i_port && (i < p_sys->i_es); i++)
if (p == p_sys->es[i]->i_port)
i_port = 0;
}
sout_stream_id_t *id = malloc( sizeof( *id ) );
if( unlikely(id == NULL) )
return NULL;
id->p_stream = p_stream;
/* Look for free dymanic payload type */
id->i_payload_type = 96 + clz32 (p_sys->payload_bitmap);
assert (id->i_payload_type < 128);
vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
id->i_seq_sent_next = id->i_sequence;
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 = 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 );
id->pf_packetize = NULL;
#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->i_caching =
(int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
#ifdef HAVE_SRTP
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;
}
id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
}
#endif
if( p_sys->psz_destination != NULL )
switch( p_sys->proto )
{
case IPPROTO_DCCP:
{
const char *code;
switch (id->i_cat)
{
case VIDEO_ES: code = "RTPV"; break;
case AUDIO_ES: code = "RTPARTPV"; break;
case SPU_ES: code = "RTPTRTPV"; break;
default: code = "RTPORTPV"; break;
}
var_SetString (p_stream, "dccp-service", code);
} /* fall through */
case IPPROTO_TCP:
id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
p_sys->psz_destination, i_port,
p_sys->proto );
if( id->listen.fd == NULL )
{
msg_Err( p_stream, "passive COMEDIA RTP socket failed" );
goto error;
}
if( vlc_clone( &id->listen.thread, rtp_listen_thread, id,
VLC_THREAD_PRIORITY_LOW ) )
{
net_ListenClose( id->listen.fd );
id->listen.fd = NULL;
goto error;
}
break;
default:
{
int ttl = (p_sys->i_ttl >= 0) ? p_sys->i_ttl : -1;
int fd = net_ConnectDgram( p_stream, p_sys->psz_destination,
i_port, ttl, p_sys->proto );
if( fd == -1 )
{
msg_Err( p_stream, "cannot create RTP socket" );
goto error;
}
/* Ignore any unexpected incoming packet (including RTCP-RR
* packets in case of rtcp-mux) */
setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
sizeof (int));
rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL );
}
}
if( p_fmt == NULL )
{
char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
if( psz == NULL ) /* Uho! */
;
else
if( strncmp( psz, "ts", 2 ) == 0 )
{
id->i_payload_type = 33;
id->psz_enc = "MP2T";
}
else
{
id->psz_enc = "MP2P";
}
free( psz );
}
else
switch( p_fmt->i_codec )
{
case VLC_CODEC_MULAW:
if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
id->i_payload_type = 0;
id->psz_enc = "PCMU";
id->pf_packetize = rtp_packetize_split;
rtp_set_ptime (id, 20, 1);
break;
case VLC_CODEC_ALAW:
if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 )
id->i_payload_type = 8;
id->psz_enc = "PCMA";
id->pf_packetize = rtp_packetize_split;
rtp_set_ptime (id, 20, 1);
break;
case VLC_CODEC_S16B:
case VLC_CODEC_S16L:
if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 )
{
id->i_payload_type = 11;
}
else if( p_fmt->audio.i_channels == 2 &&
p_fmt->audio.i_rate == 44100 )
{
id->i_payload_type = 10;
}
id->psz_enc = "L16";
if( p_fmt->i_codec == VLC_CODEC_S16B )
id->pf_packetize = rtp_packetize_split;
else
id->pf_packetize = rtp_packetize_swab;
rtp_set_ptime (id, 20, 2);
break;
case VLC_CODEC_U8:
id->psz_enc = "L8";
id->pf_packetize = rtp_packetize_split;
rtp_set_ptime (id, 20, 1);
break;
case VLC_CODEC_MPGA:
id->i_payload_type = 14;
id->psz_enc = "MPA";
id->i_clock_rate = 90000; /* not 44100 */
id->pf_packetize = rtp_packetize_mpa;
break;
case VLC_CODEC_MPGV:
id->i_payload_type = 32;
id->psz_enc = "MPV";
id->pf_packetize = rtp_packetize_mpv;
break;
case VLC_CODEC_ADPCM_G726:
switch( p_fmt->i_bitrate / 1000 )
{
case 16:
id->psz_enc = "G726-16";
id->pf_packetize = rtp_packetize_g726_16;
break;
case 24:
id->psz_enc = "G726-24";
id->pf_packetize = rtp_packetize_g726_24;
break;
case 32:
id->psz_enc = "G726-32";
id->pf_packetize = rtp_packetize_g726_32;
break;
case 40:
id->psz_enc = "G726-40";
id->pf_packetize = rtp_packetize_g726_40;
break;
default:
msg_Err( p_stream, "cannot add this stream (unsupported "
"G.726 bit rate: %u)", p_fmt->i_bitrate );
goto error;
}
break;
case VLC_CODEC_A52:
id->psz_enc = "ac3";
id->pf_packetize = rtp_packetize_ac3;
break;
case VLC_CODEC_H263:
id->psz_enc = "H263-1998";
id->pf_packetize = rtp_packetize_h263;
break;
case VLC_CODEC_H264:
id->psz_enc = "H264";
id->pf_packetize = rtp_packetize_h264;
id->psz_fmtp = NULL;
if( p_fmt->i_extra > 0 )
{
uint8_t *p_buffer = p_fmt->p_extra;
int i_buffer = p_fmt->i_extra;
char *p_64_sps = NULL;
char *p_64_pps = NULL;
char hexa[6+1];
while( i_buffer > 4 &&
p_buffer[0] == 0 && p_buffer[1] == 0 &&
p_buffer[2] == 0 && p_buffer[3] == 1 )
{
const int i_nal_type = p_buffer[4]&0x1f;
int i_offset;
int i_size = 0;
msg_Dbg( p_stream, "we found a startcode for NAL with TYPE:%d", i_nal_type );
i_size = i_buffer;
for( i_offset = 4; i_offset+3 < i_buffer ; i_offset++)
{
if( !memcmp (p_buffer + i_offset, "\x00\x00\x00\x01", 4 ) )
{
/* we found another startcode */
i_size = i_offset;
break;
}
}
if( i_nal_type == 7 )
{
p_64_sps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
sprintf_hexa( hexa, &p_buffer[5], 3 );
}
else if( i_nal_type == 8 )
{
p_64_pps = vlc_b64_encode_binary( &p_buffer[4], i_size - 4 );
}
i_buffer -= i_size;
p_buffer += i_size;
}
/* */
if( p_64_sps && p_64_pps &&
( asprintf( &id->psz_fmtp,
"packetization-mode=1;profile-level-id=%s;"
"sprop-parameter-sets=%s,%s;", hexa, p_64_sps,
p_64_pps ) == -1 ) )
id->psz_fmtp = NULL;
free( p_64_sps );
free( p_64_pps );
}
if( !id->psz_fmtp )
id->psz_fmtp = strdup( "packetization-mode=1" );
break;
case VLC_CODEC_MP4V:
{
id->psz_enc = "MP4V-ES";
id->pf_packetize = rtp_packetize_split;
if( p_fmt->i_extra > 0 )
{
char hexa[2*p_fmt->i_extra +1];
sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
if( asprintf( &id->psz_fmtp,
"profile-level-id=3; config=%s;", hexa ) == -1 )
id->psz_fmtp = NULL;
}
break;
}
case VLC_CODEC_MP4A:
{
if(!p_sys->b_latm)
{
char hexa[2*p_fmt->i_extra +1];
id->psz_enc = "mpeg4-generic";
id->pf_packetize = rtp_packetize_mp4a;
sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
if( asprintf( &id->psz_fmtp,
"streamtype=5; profile-level-id=15; "
"mode=AAC-hbr; config=%s; SizeLength=13; "
"IndexLength=3; IndexDeltaLength=3; Profile=1;",
hexa ) == -1 )
id->psz_fmtp = NULL;
}
else
{
char hexa[13];
int i;
unsigned char config[6];
unsigned int aacsrates[15] = {
96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
16000, 12000, 11025, 8000, 7350, 0, 0 };
for( i = 0; i < 15; i++ )
if( p_fmt->audio.i_rate == aacsrates[i] )
break;
config[0]=0x40;
config[1]=0;
config[2]=0x20|i;
config[3]=p_fmt->audio.i_channels<<4;
config[4]=0x3f;
config[5]=0xc0;
id->psz_enc = "MP4A-LATM";
id->pf_packetize = rtp_packetize_mp4a_latm;
sprintf_hexa( hexa, config, 6 );
if( asprintf( &id->psz_fmtp, "profile-level-id=15; "
"object=2; cpresent=0; config=%s", hexa ) == -1 )
id->psz_fmtp = NULL;
}
break;
}
case VLC_CODEC_AMR_NB:
id->psz_enc = "AMR";
id->psz_fmtp = strdup( "octet-align=1" );
id->pf_packetize = rtp_packetize_amr;
break;
case VLC_CODEC_AMR_WB:
id->psz_enc = "AMR-WB";
id->psz_fmtp = strdup( "octet-align=1" );
id->pf_packetize = rtp_packetize_amr;
break;
case VLC_CODEC_SPEEX:
id->psz_enc = "SPEEX";
id->pf_packetize = rtp_packetize_spx;
break;
case VLC_CODEC_ITU_T140:
id->psz_enc = "t140" ;
id->i_clock_rate = 1000;
id->pf_packetize = rtp_packetize_t140;
break;
default:
msg_Err( p_stream, "cannot add this stream (unsupported "
"codec: %4.4s)", (char*)&p_fmt->i_codec );
goto error;
}
if (id->i_payload_type >= 96)
/* Mark dynamic payload type in use */
p_sys->payload_bitmap &= ~(1 << (127 - id->i_payload_type));
#if 0 /* No payload formats sets this at the moment */
int cscov = -1;
if( cscov != -1 )
cscov += 8 /* UDP */ + 12 /* RTP */;
if( id->sinkc > 0 )
net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
#endif
vlc_mutex_lock( &p_sys->lock_ts );
id->b_ts_init = ( p_sys->i_npt_zero != VLC_TS_INVALID );
vlc_mutex_unlock( &p_sys->lock_ts );
if( id->b_ts_init )
id->i_ts_offset = rtp_compute_ts( id, p_sys->i_pts_offset );
if( p_sys->rtsp != NULL )
id->rtsp_id = RtspAddId( p_sys->rtsp, id,
GetDWBE( id->ssrc ),
p_sys->psz_destination,
p_sys->i_ttl, id->i_port, id->i_port + 1 );
id->p_fifo = block_FifoNew();
if( unlikely(id->p_fifo == NULL) )
goto error;
if( vlc_clone( &id->thread, ThreadSend, id, VLC_THREAD_PRIORITY_HIGHEST ) )
{
block_FifoRelease( id->p_fifo );
id->p_fifo = NULL;
goto error;
}
/* Update p_sys context */
vlc_mutex_lock( &p_sys->lock_es );
TAB_APPEND( p_sys->i_es, p_sys->es, id );
vlc_mutex_unlock( &p_sys->lock_es );
psz_sdp = SDPGenerate( p_stream, NULL );
vlc_mutex_lock( &p_sys->lock_sdp );
free( p_sys->psz_sdp );
p_sys->psz_sdp = psz_sdp;
vlc_mutex_unlock( &p_sys->lock_sdp );
msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp );
/* Update SDP (sap/file) */
if( p_sys->b_export_sap ) SapSetup( p_stream );
if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
return id;
error:
Del( p_stream, id );
return NULL;
}
static int Del( sout_stream_t *p_stream, sout_stream_id_t *id )
{
sout_stream_sys_t *p_sys = p_stream->p_sys;
vlc_mutex_lock( &p_sys->lock_es );
TAB_REMOVE( p_sys->i_es, p_sys->es, id );
vlc_mutex_unlock( &p_sys->lock_es );
if( likely(id->p_fifo != NULL) )
{
vlc_cancel( id->thread );
vlc_join( id->thread, NULL );
block_FifoRelease( id->p_fifo );
}
/* Release dynamic payload type */
if (id->i_payload_type >= 96)
p_sys->payload_bitmap |= 1 << (127 - id->i_payload_type);
free( id->psz_fmtp );
if( id->rtsp_id )
RtspDelId( p_sys->rtsp, id->rtsp_id );
if( id->listen.fd != NULL )
{
vlc_cancel( id->listen.thread );
vlc_join( id->listen.thread, NULL );
net_ListenClose( id->listen.fd );
}
/* Delete remaining sinks (incoming connections or explicit
* outgoing dst=) */
while( id->sinkc > 0 )
rtp_del_sink( id, id->sinkv[0].rtp_fd );
#ifdef HAVE_SRTP
if( id->srtp != NULL )
srtp_destroy( id->srtp );
#endif
vlc_mutex_destroy( &id->lock_sink );
/* Update SDP (sap/file) */
if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
free( id );
return VLC_SUCCESS;
}
static int Send( sout_stream_t *p_stream, sout_stream_id_t *id,
block_t *p_buffer )
{
block_t *p_next;
assert( p_stream->p_sys->p_mux == NULL );
(void)p_stream;
while( p_buffer != NULL )
{
p_next = p_buffer->p_next;
if( id->pf_packetize( id, p_buffer ) )
break;
block_Release( p_buffer );
p_buffer = p_next;
}
return VLC_SUCCESS;
}
/****************************************************************************
* SAP:
****************************************************************************/
static int SapSetup( sout_stream_t *p_stream )
{
sout_stream_sys_t *p_sys = p_stream->p_sys;
sout_instance_t *p_sout = p_stream->p_sout;
/* Remove the previous session */
if( p_sys->p_session != NULL)
{
sout_AnnounceUnRegister( p_sout, p_sys->p_session);
p_sys->p_session = NULL;
}
if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp )
{
announce_method_t *p_method = sout_SAPMethod();
p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
p_sys->psz_sdp,
p_sys->psz_destination,
p_method );
sout_MethodRelease( p_method );
}
return VLC_SUCCESS;
}
/****************************************************************************
* File:
****************************************************************************/
static int FileSetup( sout_stream_t *p_stream )
{
sout_stream_sys_t *p_sys = p_stream->p_sys;
FILE *f;
if( p_sys->psz_sdp == NULL )
return VLC_EGENERIC; /* too early */
if( ( f = vlc_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
{
msg_Err( p_stream, "cannot open file '%s' (%m)",
p_sys->psz_sdp_file );
return VLC_EGENERIC;
}
fputs( p_sys->psz_sdp, f );
fclose( f );
return VLC_SUCCESS;
}
/****************************************************************************
* HTTP:
****************************************************************************/
static int HttpCallback( httpd_file_sys_t *p_args,
httpd_file_t *, uint8_t *p_request,
uint8_t **pp_data, int *pi_data );
static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url)
{
sout_stream_sys_t *p_sys = p_stream->p_sys;
p_sys->p_httpd_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host,
url->i_port > 0 ? url->i_port : 80 );
if( p_sys->p_httpd_host )
{
p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
url->psz_path ? url->psz_path : "/",
"application/sdp",
NULL, NULL, NULL,
HttpCallback, (void*)p_sys );
}
if( p_sys->p_httpd_file == NULL )
{
return VLC_EGENERIC;
}
return VLC_SUCCESS;
}
static int HttpCallback( httpd_file_sys_t *p_args,
httpd_file_t *f, uint8_t *p_request,
uint8_t **pp_data, int *pi_data )
{
VLC_UNUSED(f); VLC_UNUSED(p_request);
sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args;
vlc_mutex_lock( &p_sys->lock_sdp );
if( p_sys->psz_sdp && *p_sys->psz_sdp )
{
*pi_data = strlen( p_sys->psz_sdp );
*pp_data = malloc( *pi_data );
memcpy( *pp_data, p_sys->psz_sdp, *pi_data );
}
else
{
*pp_data = NULL;
*pi_data = 0;
}
vlc_mutex_unlock( &p_sys->lock_sdp );
return VLC_SUCCESS;
}
/****************************************************************************
* RTP send
****************************************************************************/
static void* ThreadSend( void *data )
{
#ifdef WIN32
# define ECONNREFUSED WSAECONNREFUSED
# define ENOPROTOOPT WSAENOPROTOOPT
# define EHOSTUNREACH WSAEHOSTUNREACH
# define ENETUNREACH WSAENETUNREACH
# define ENETDOWN WSAENETDOWN
# define ENOBUFS WSAENOBUFS
# define EAGAIN WSAEWOULDBLOCK
# define EWOULDBLOCK WSAEWOULDBLOCK
#endif
sout_stream_id_t *id = data;
unsigned i_caching = id->i_caching;
for (;;)
{
block_t *out = block_FifoGet( id->p_fifo );
block_cleanup_push (out);
#ifdef HAVE_SRTP
if( id->srtp )
{ /* FIXME: this is awfully inefficient */
size_t len = out->i_buffer;
out = block_Realloc( out, 0, len + 10 );
out->i_buffer = len;
int canc = vlc_savecancel ();
int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 );
vlc_restorecancel (canc);
if( val )
{
errno = val;
msg_Dbg( id->p_stream, "SRTP sending error: %m" );
block_Release( out );
out = NULL;
}
else
out->i_buffer = len;
}
if (out)
#endif
mwait (out->i_dts + i_caching);
vlc_cleanup_pop ();
if (out == NULL)
continue;
ssize_t len = out->i_buffer;
int canc = vlc_savecancel ();
vlc_mutex_lock( &id->lock_sink );
unsigned deadc = 0; /* How many dead sockets? */
int deadv[id->sinkc]; /* Dead sockets list */
for( int i = 0; i < id->sinkc; i++ )
{
#ifdef HAVE_SRTP
if( !id->srtp ) /* FIXME: SRTCP support */
#endif
SendRTCP( id->sinkv[i].rtcp, out );
if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
continue;
switch( net_errno )
{
/* Soft errors (e.g. ICMP): */
case ECONNREFUSED: /* Port unreachable */
case ENOPROTOOPT:
#ifdef EPROTO
case EPROTO: /* Protocol unreachable */
#endif
case EHOSTUNREACH: /* Host unreachable */
case ENETUNREACH: /* Network unreachable */
case ENETDOWN: /* Entire network down */
send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 );
/* Transient congestion: */
case ENOMEM: /* out of socket buffers */
case ENOBUFS:
case EAGAIN:
#if (EAGAIN != EWOULDBLOCK)
case EWOULDBLOCK:
#endif
continue;
}
deadv[deadc++] = id->sinkv[i].rtp_fd;
}
id->i_seq_sent_next = ntohs(((uint16_t *) out->p_buffer)[1]) + 1;
vlc_mutex_unlock( &id->lock_sink );
block_Release( out );
for( unsigned i = 0; i < deadc; i++ )
{
msg_Dbg( id->p_stream, "removing socket %d", deadv[i] );
rtp_del_sink( id, deadv[i] );
}
vlc_restorecancel (canc);
}
return NULL;
}
/* This thread dequeues incoming connections (DCCP streaming) */
static void *rtp_listen_thread( void *data )
{
sout_stream_id_t *id = data;
assert( id->listen.fd != NULL );
for( ;; )
{
int fd = net_Accept( id->p_stream, id->listen.fd );
if( fd == -1 )
continue;
int canc = vlc_savecancel( );
rtp_add_sink( id, fd, true, NULL );
vlc_restorecancel( canc );
}
assert( 0 );
}
int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux, uint16_t *seq )
{
rtp_sink_t sink = { fd, NULL };
sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP,
rtcp_mux );
if( sink.rtcp == NULL )
msg_Err( id->p_stream, "RTCP failed!" );
vlc_mutex_lock( &id->lock_sink );
INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
if( seq != NULL )
*seq = id->i_seq_sent_next;
vlc_mutex_unlock( &id->lock_sink );
return VLC_SUCCESS;
}
void rtp_del_sink( sout_stream_id_t *id, int fd )
{
rtp_sink_t sink = { fd, NULL };
/* NOTE: must be safe to use if fd is not included */
vlc_mutex_lock( &id->lock_sink );
for( int i = 0; i < id->sinkc; i++ )
{
if (id->sinkv[i].rtp_fd == fd)
{
sink = id->sinkv[i];
REMOVE_ELEM( id->sinkv, id->sinkc, i );
break;
}
}
vlc_mutex_unlock( &id->lock_sink );
CloseRTCP( sink.rtcp );
net_Close( sink.rtp_fd );
}
uint16_t rtp_get_seq( sout_stream_id_t *id )
{
/* This will return values for the next packet. */
uint16_t seq;
vlc_mutex_lock( &id->lock_sink );
seq = id->i_seq_sent_next;
vlc_mutex_unlock( &id->lock_sink );
return seq;
}
/* Return a timestamp corresponding to packets being sent now, and that
* can be passed to rtp_compute_ts() to get rtptime values for each ES. */
int64_t rtp_get_ts( const sout_stream_t *p_stream )
{
sout_stream_sys_t *p_sys = p_stream->p_sys;
mtime_t i_npt_zero;
vlc_mutex_lock( &p_sys->lock_ts );
i_npt_zero = p_sys->i_npt_zero;
vlc_mutex_unlock( &p_sys->lock_ts );
if( i_npt_zero == VLC_TS_INVALID )
return p_sys->i_pts_zero;
mtime_t now = mdate();
if( now < i_npt_zero )
return p_sys->i_pts_zero;
return p_sys->i_pts_zero + (now - i_npt_zero);
}
void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
int b_marker, int64_t i_pts )
{
if( !id->b_ts_init )
{
sout_stream_sys_t *p_sys = id->p_stream->p_sys;
vlc_mutex_lock( &p_sys->lock_ts );
if( p_sys->i_npt_zero == VLC_TS_INVALID )
{
/* This is the first packet of any ES. We initialize the
* NPT=0 time reference, and the offset to match the
* arbitrary PTS reference. */
p_sys->i_npt_zero = i_pts + id->i_caching;
p_sys->i_pts_offset = p_sys->i_pts_zero - i_pts;
}
vlc_mutex_unlock( &p_sys->lock_ts );
/* And in any case this is the first packet of this ES, so we
* initialize the offset for this ES. */
id->i_ts_offset = rtp_compute_ts( id, p_sys->i_pts_offset );
id->b_ts_init = true;
}
uint32_t i_timestamp = rtp_compute_ts( id, i_pts ) + id->i_ts_offset;
out->p_buffer[0] = 0x80;
out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type;
out->p_buffer[2] = ( id->i_sequence >> 8)&0xff;
out->p_buffer[3] = ( id->i_sequence )&0xff;
out->p_buffer[4] = ( i_timestamp >> 24 )&0xff;
out->p_buffer[5] = ( i_timestamp >> 16 )&0xff;
out->p_buffer[6] = ( i_timestamp >> 8 )&0xff;
out->p_buffer[7] = ( i_timestamp )&0xff;
memcpy( out->p_buffer + 8, id->ssrc, 4 );
out->i_buffer = 12;
id->i_sequence++;
}
void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
{
block_FifoPut( id->p_fifo, out );
}
/**
* @return configured max RTP payload size (including payload type-specific
* headers, excluding RTP and transport headers)
*/
size_t rtp_mtu (const sout_stream_id_t *id)
{
return id->i_mtu - 12;
}
/*****************************************************************************
* Non-RTP mux
*****************************************************************************/
/** Add an ES to a non-RTP muxed stream */
static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt )
{
sout_input_t *p_input;
sout_mux_t *p_mux = p_stream->p_sys->p_mux;
assert( p_mux != NULL );
p_input = sout_MuxAddStream( p_mux, p_fmt );
if( p_input == NULL )
{
msg_Err( p_stream, "cannot add this stream to the muxer" );
return NULL;
}
return (sout_stream_id_t *)p_input;
}
static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id,
block_t *p_buffer )
{
sout_mux_t *p_mux = p_stream->p_sys->p_mux;
assert( p_mux != NULL );
sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer );
return VLC_SUCCESS;
}
/** Remove an ES from a non-RTP muxed stream */
static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id )
{
sout_mux_t *p_mux = p_stream->p_sys->p_mux;
assert( p_mux != NULL );
sout_MuxDeleteStream( p_mux, (sout_input_t *)id );
return VLC_SUCCESS;
}
static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream,
const block_t *p_buffer )
{
sout_stream_sys_t *p_sys = p_stream->p_sys;
sout_stream_id_t *id = p_sys->es[0];
int64_t i_dts = p_buffer->i_dts;
uint8_t *p_data = p_buffer->p_buffer;
size_t i_data = p_buffer->i_buffer;
size_t i_max = id->i_mtu - 12;
size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
while( i_data > 0 )
{
size_t i_size;
/* output complete packet */
if( p_sys->packet &&
p_sys->packet->i_buffer + i_data > i_max )
{
rtp_packetize_send( id, p_sys->packet );
p_sys->packet = NULL;
}
if( p_sys->packet == NULL )
{
/* allocate a new packet */
p_sys->packet = block_New( p_stream, id->i_mtu );
rtp_packetize_common( id, p_sys->packet, 1, i_dts );
p_sys->packet->i_dts = i_dts;
p_sys->packet->i_length = p_buffer->i_length / i_packet;
i_dts += p_sys->packet->i_length;
}
i_size = __MIN( i_data,
(unsigned)(id->i_mtu - p_sys->packet->i_buffer) );
memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
p_data, i_size );
p_sys->packet->i_buffer += i_size;
p_data += i_size;
i_data -= i_size;
}
return VLC_SUCCESS;
}
static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access,
block_t *p_buffer )
{
sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys;
while( p_buffer )
{
block_t *p_next;
AccessOutGrabberWriteBuffer( p_stream, p_buffer );
p_next = p_buffer->p_next;
block_Release( p_buffer );
p_buffer = p_next;
}
return VLC_SUCCESS;
}
static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream )
{
sout_access_out_t *p_grab;
p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
if( p_grab == NULL )
return NULL;
p_grab->p_module = NULL;
p_grab->psz_access = strdup( "grab" );
p_grab->p_cfg = NULL;
p_grab->psz_path = strdup( "" );
p_grab->p_sys = (sout_access_out_sys_t *)p_stream;
p_grab->pf_seek = NULL;
p_grab->pf_write = AccessOutGrabberWrite;
vlc_object_attach( p_grab, p_stream );
return p_grab;
}