#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>
#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_TEXT N_("Session descriptipn")
#define DESC_LONGTEXT N_( \
- "This allows you to give a broader description of the stream, that will " \
- "be announced in the SDP (Session Descriptor)." )
+ "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 " \
"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)." )
+ "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." )
"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 DCCP_TEXT N_("DCCP transport")
+#define DCCP_LONGTEXT N_( \
+ "This enables DCCP instead of UDP as a transport for RTP." )
+#define TCP_TEXT N_("TCP transport")
+#define TCP_LONGTEXT N_( \
+ "This enables TCP instead of UDP as a transport for RTP." )
+#define UDP_LITE_TEXT N_("UDP-Lite transport")
+#define UDP_LITE_LONGTEXT N_( \
+ "This enables UDP-Lite instead of UDP as a transport for RTP." )
+
#define RFC3016_TEXT N_("MP4A LATM")
#define RFC3016_LONGTEXT N_( \
"This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
add_string( SOUT_CFG_PREFIX "mux", "", NULL, MUX_TEXT,
MUX_LONGTEXT, VLC_TRUE );
- add_string( SOUT_CFG_PREFIX "name", "NONE", NULL, NAME_TEXT,
+ add_string( SOUT_CFG_PREFIX "name", "", NULL, NAME_TEXT,
NAME_LONGTEXT, VLC_TRUE );
add_string( SOUT_CFG_PREFIX "description", "", NULL, DESC_TEXT,
DESC_LONGTEXT, VLC_TRUE );
URL_LONGTEXT, VLC_TRUE );
add_string( SOUT_CFG_PREFIX "email", "", NULL, EMAIL_TEXT,
EMAIL_LONGTEXT, VLC_TRUE );
+ add_string( SOUT_CFG_PREFIX "phone", "", NULL, PHONE_TEXT,
+ PHONE_LONGTEXT, VLC_TRUE );
add_integer( SOUT_CFG_PREFIX "port", 1234, NULL, PORT_TEXT,
PORT_LONGTEXT, VLC_TRUE );
add_integer( SOUT_CFG_PREFIX "ttl", 0, NULL, TTL_TEXT,
TTL_LONGTEXT, VLC_TRUE );
+ add_bool( SOUT_CFG_PREFIX "rtcp-mux", VLC_FALSE, NULL,
+ RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, VLC_FALSE );
+
+ add_bool( SOUT_CFG_PREFIX "dccp", VLC_FALSE, NULL,
+ DCCP_TEXT, DCCP_LONGTEXT, VLC_FALSE );
+ add_bool( SOUT_CFG_PREFIX "tcp", VLC_FALSE, NULL,
+ TCP_TEXT, TCP_LONGTEXT, VLC_FALSE );
+ add_bool( SOUT_CFG_PREFIX "udplite", VLC_FALSE, NULL,
+ UDP_LITE_TEXT, UDP_LITE_LONGTEXT, VLC_FALSE );
+
add_bool( SOUT_CFG_PREFIX "mp4a-latm", 0, NULL, RFC3016_TEXT,
RFC3016_LONGTEXT, VLC_FALSE );
*****************************************************************************/
static const char *ppsz_sout_options[] = {
"dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
- "description", "url","email", "mp4a-latm", NULL
+ "description", "url", "email", "phone",
+ "rtcp-mux", "dccp", "tcp", "udplite",
+ "mp4a-latm", NULL
};
static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
struct sout_stream_sys_t
{
- /* sdp */
- int64_t i_sdp_id;
- int i_sdp_version;
+ /* SDP */
char *psz_sdp;
vlc_mutex_t lock_sdp;
- char *psz_session_name;
- char *psz_session_description;
- char *psz_session_url;
- char *psz_session_email;
-
- /* */
+ /* SDP to disk */
vlc_bool_t b_export_sdp_file;
char *psz_sdp_file;
- /* sap */
+
+ /* SDP via SAP */
vlc_bool_t 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;
- int i_port;
- int i_port_audio;
- int i_port_video;
- int i_ttl;
+ char *psz_destination;
+ uint8_t proto;
+ uint8_t i_ttl;
+ uint16_t i_port;
+ uint16_t i_port_audio;
+ uint16_t i_port_video;
vlc_bool_t b_latm;
+ vlc_bool_t rtcp_mux;
/* when need to use a private one or when using muxer */
int i_payload_type;
typedef int (*pf_rtp_packetizer_t)( sout_stream_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 */
- uint32_t i_timestamp_start;
uint16_t i_sequence;
uint8_t i_payload_type;
uint8_t ssrc[4];
/* Packets sinks */
vlc_mutex_t lock_sink;
- int fdc;
- int *fdv;
+ int sinkc;
+ rtp_sink_t *sinkv;
rtsp_stream_id_t *rtsp_id;
+ int *listen_fd;
block_fifo_t *p_fifo;
int64_t i_caching;
return VLC_ENOMEM;
p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" );
- p_sys->psz_session_name = var_GetString( p_stream, SOUT_CFG_PREFIX "name" );
- p_sys->psz_session_description = var_GetString( p_stream, SOUT_CFG_PREFIX "description" );
- p_sys->psz_session_url = var_GetString( p_stream, SOUT_CFG_PREFIX "url" );
- p_sys->psz_session_email = var_GetString( p_stream, SOUT_CFG_PREFIX "email" );
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;
p_sys->i_port_video = 0;
}
- if( !p_sys->psz_session_name )
- {
- if( p_sys->psz_destination )
- p_sys->psz_session_name = strdup( p_sys->psz_destination );
- else
- p_sys->psz_session_name = strdup( "NONE" );
- }
-
for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
{
- if( !strcmp( p_cfg->psz_name, "sdp" ) )
+ if( !strcmp( p_cfg->psz_name, "sdp" )
+ && ( p_cfg->psz_value != NULL )
+ && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) )
{
- if( p_cfg->psz_value && !strncasecmp( p_cfg->psz_value, "rtsp", 4 ) )
- {
- b_rtsp = VLC_TRUE;
- break;
- }
+ b_rtsp = VLC_TRUE;
+ break;
}
}
if( !b_rtsp )
psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" );
if( psz != NULL )
{
- if( !strncasecmp( psz, "rtsp", 4 ) )
+ if( !strncasecmp( psz, "rtsp:", 5 ) )
b_rtsp = VLC_TRUE;
free( psz );
}
}
- if( p_sys->psz_destination == NULL )
+ /* Transport protocol */
+ p_sys->proto = IPPROTO_UDP;
+
+ if( var_GetBool( p_stream, SOUT_CFG_PREFIX "dccp" ) )
{
- if( !b_rtsp )
- {
- msg_Err( p_stream, "missing destination and not in RTSP mode" );
- free( p_sys );
- return VLC_EGENERIC;
- }
+ p_sys->proto = IPPROTO_DCCP;
+ p_sys->rtcp_mux = VLC_TRUE; /* Force RTP/RTCP mux */
}
- else if( p_sys->i_port <= 0 )
+#if 0
+ else
+ if( var_GetBool( p_stream, SOUT_CFG_PREFIX "tcp" ) )
{
- msg_Err( p_stream, "invalid port" );
+ p_sys->proto = IPPROTO_TCP;
+ p_sys->rtcp_mux = VLC_TRUE; /* Force RTP/RTCP mux */
+ }
+ else
+#endif
+ if( var_GetBool( p_stream, SOUT_CFG_PREFIX "udplite" ) )
+ p_sys->proto = IPPROTO_UDPLITE;
+
+ 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;
}
* ttl are set. */
p_sys->i_ttl = config_GetInt( p_stream, "ttl" );
}
- if( p_sys->i_ttl > 255 )
- p_sys->i_ttl = 255;
- /* must not exceed 999 once formatted */
p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
p_sys->rtsp = NULL;
p_sys->psz_sdp = NULL;
- p_sys->i_sdp_id = mdate();
- p_sys->i_sdp_version = 1;
- p_sys->psz_sdp = NULL;
-
p_sys->b_export_sap = VLC_FALSE;
p_sys->b_export_sdp_file = VLC_FALSE;
p_sys->p_session = NULL;
if( psz != NULL )
{
sout_stream_id_t *id;
- const char *psz_rtpmap;
- int i_payload_type;
/* Check muxer type */
- if( !strncasecmp( psz, "ps", 2 )
- || !strncasecmp( psz, "mpeg1", 5 ) )
- {
- psz_rtpmap = "MP2P/90000";
- }
- else if( !strncasecmp( psz, "ts", 2 ) )
- {
- psz_rtpmap = "MP2T/90000";
- i_payload_type = 33;
- }
- else
+ 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 );
return VLC_EGENERIC;
}
- id->psz_rtpmap = strdup( psz_rtpmap );
- id->i_payload_type = i_payload_type;
-
p_sys->packet = NULL;
p_stream->pf_add = MuxAdd;
continue;
/* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
- if( !strcmp( p_cfg->psz_value, psz ) )
+ if( !strcmp( p_cfg->psz_value, psz ) )
continue;
SDPHandleUrl( p_stream, p_cfg->psz_value );
if( p_sys->p_httpd_host )
httpd_HostDelete( p_sys->p_httpd_host );
- free( p_sys->psz_session_name );
- free( p_sys->psz_session_description );
- free( p_sys->psz_session_url );
- free( p_sys->psz_session_email );
free( p_sys->psz_sdp );
if( p_sys->b_export_sdp_file )
id->i_port, id->i_port + 1 );
}
}
- else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
+ else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
{
p_sys->b_export_sap = VLC_TRUE;
/*****************************************************************************
* SDPGenerate
*****************************************************************************/
- /* FIXME http://www.faqs.org/rfcs/rfc2327.html
- All text fields should be UTF-8 encoded. Use global a:charset to announce this.
- o= - should be local username (no spaces allowed)
- o= time should be hashed with some other value to garantue uniqueness
- o= we need IP6 support?
- o= don't use the localhost address. use fully qualified domain name or IP4 address
- p= international phone number (pass via vars?)
- c= IP6 support
- a= recvonly (missing)
- a= type:broadcast (missing)
- a= charset: (normally charset should be UTF-8, this can be used to override s= and i=)
- a= x-plgroup: (missing)
- RTP packets need to get the correct src IP address */
/*static*/
char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url )
{
const sout_stream_sys_t *p_sys = p_stream->p_sys;
- size_t i_size;
- const char *psz_destination = p_sys->psz_destination;
- char *psz_sdp, *p, ipv;
+ char *psz_sdp;
+ struct sockaddr_storage dst;
+ socklen_t dstlen;
int i;
/*
* When we have a fixed destination (typically when we do multicast),
* output chain with two different RTSP URLs if you need to handle this
* scenario.
*/
- int inclport = (psz_destination != NULL);
-
- /* FIXME: breaks IP version check on unknown destination */
- if( psz_destination == NULL )
- psz_destination = "0.0.0.0";
-
- i_size = sizeof( "v=0\r\n" ) +
- sizeof( "o=- * * IN IP4 127.0.0.1\r\n" ) + 10 + 10 +
- sizeof( "s=*\r\n" ) + strlen( p_sys->psz_session_name ) +
- sizeof( "i=*\r\n" ) + strlen( p_sys->psz_session_description ) +
- sizeof( "u=*\r\n" ) + strlen( p_sys->psz_session_url ) +
- sizeof( "e=*\r\n" ) + strlen( p_sys->psz_session_email ) +
- sizeof( "t=0 0\r\n" ) +
- sizeof( "b=RR:0\r\n" ) +
- sizeof( "a=tool:"PACKAGE_STRING"\r\n" ) +
- sizeof( "a=recvonly\r\n" ) +
- sizeof( "a=type:broadcast\r\n" ) +
- sizeof( "c=IN IP4 */*\r\n" ) + 20 + 10 +
- strlen( psz_destination ) ;
- for( i = 0; i < p_sys->i_es; i++ )
+ int inclport;
+
+ if( p_sys->psz_destination != NULL )
{
- sout_stream_id_t *id = p_sys->es[i];
+ inclport = 1;
- i_size += strlen( "m=**d*o * RTP/AVP *\r\n" ) + 10 + 10;
- if ( id->i_bitrate )
- {
- i_size += strlen( "b=AS: *\r\n") + 10;
- }
- if( id->psz_rtpmap )
- {
- i_size += strlen( "a=rtpmap:* *\r\n" ) + strlen( id->psz_rtpmap )+10;
- }
- if( id->psz_fmtp )
- {
- i_size += strlen( "a=fmtp:* *\r\n" ) + strlen( id->psz_fmtp ) + 10;
- }
- if( rtsp_url != NULL )
- {
- i_size += strlen( "a=control:*/trackID=*\r\n" ) + strlen( rtsp_url ) + 10;
- }
+ /* 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;
- ipv = ( strchr( psz_destination, ':' ) != NULL ) ? '6' : '4';
-
- p = psz_sdp = malloc( i_size );
- p += sprintf( p, "v=0\r\n" );
- p += sprintf( p, "o=- "I64Fu" %d IN IP%c %s\r\n",
- p_sys->i_sdp_id, p_sys->i_sdp_version,
- ipv, ipv == '6' ? "::1" : "127.0.0.1" );
- if( *p_sys->psz_session_name )
- p += sprintf( p, "s=%s\r\n", p_sys->psz_session_name );
- if( *p_sys->psz_session_description )
- p += sprintf( p, "i=%s\r\n", p_sys->psz_session_description );
- if( *p_sys->psz_session_url )
- p += sprintf( p, "u=%s\r\n", p_sys->psz_session_url );
- if( *p_sys->psz_session_email )
- p += sprintf( p, "e=%s\r\n", p_sys->psz_session_email );
+ /* 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 );
+ }
- p += sprintf( p, "t=0 0\r\n" ); /* permanent stream */
- /* when scheduled from vlm, we should set this info correctly */
- p += sprintf( p, "a=tool:"PACKAGE_STRING"\r\n" );
- p += sprintf( p, "a=recvonly\r\n" );
- p += sprintf( p, "a=type:broadcast\r\n" );
+ psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
+ NULL, 0, (struct sockaddr *)&dst, dstlen );
+ if( psz_sdp == NULL )
+ return NULL;
- p += sprintf( p, "c=IN IP%c %s", ipv, psz_destination );
+ /* TODO: a=source-filter */
+ if( p_sys->rtcp_mux )
+ sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL );
- if( ( ipv == 4 )
- && net_AddressIsMulticast( (vlc_object_t *)p_stream, psz_destination ) )
- {
- /* Add the deprecated TTL field if it is an IPv4 multicast address */
- p += sprintf( p, "/%d", p_sys->i_ttl ?: 1 );
- }
- p += sprintf( p, "\r\n" );
- p += sprintf( p, "b=RR:0\r\n" );
+ 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 */
- if( id->i_cat == AUDIO_ES )
- mime_major = "audio";
- else
- if( id->i_cat == VIDEO_ES )
- mime_major = "video";
- else
- continue;
-
- p += sprintf( p, "m=%s %d RTP/AVP %d\r\n", mime_major,
- inclport * id->i_port, id->i_payload_type );
-
- if ( id->i_bitrate )
+ switch( id->i_cat )
{
- p += sprintf(p,"b=AS:%d\r\n",id->i_bitrate);
+ case VIDEO_ES:
+ mime_major = "video";
+ break;
+ case AUDIO_ES:
+ mime_major = "audio";
+ break;
+ case SPU_ES:
+ mime_major = "text";
+ break;
+ default:
+ continue;
}
- if( id->psz_rtpmap )
+
+ if( rtsp_url == NULL )
{
- p += sprintf( p, "a=rtpmap:%d %s\r\n", id->i_payload_type,
- id->psz_rtpmap );
+ 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;
+ }
}
- if( id->psz_fmtp )
+
+ sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
+ id->i_payload_type, VLC_FALSE, id->i_bitrate,
+ id->psz_rtpmap, id->psz_fmtp);
+
+ if( rtsp_url != NULL )
{
- p += sprintf( p, "a=fmtp:%d %s\r\n", id->i_payload_type,
- id->psz_fmtp );
+ assert( strlen( rtsp_url ) > 0 );
+ vlc_bool_t addslash = ( rtsp_url[strlen( rtsp_url ) - 1] != '/' );
+ sdp_AddAttribute ( &psz_sdp, "control",
+ addslash ? "%s/trackID=%u" : "%strackID=%u",
+ rtsp_url, i );
}
- if( rtsp_url != NULL )
+ else
{
- p += sprintf( p, "a=control:/trackID=%d\r\n", i );
+ if( id->listen_fd != NULL )
+ sdp_AddAttribute( &psz_sdp, "setup", "passive" );
+#if 0
+ if( p_sys->proto == IPPROTO_DCCP )
+ sdp_AddAttribute( &psz_sdp, "dccp-service-code",
+ "SC:RTP%c", toupper( mime_major[0] ) );
+#endif
}
}
static int rtp_packetize_h263 ( sout_stream_t *, sout_stream_id_t *, block_t * );
static int rtp_packetize_h264 ( sout_stream_t *, sout_stream_id_t *, block_t * );
static int rtp_packetize_amr ( sout_stream_t *, sout_stream_id_t *, block_t * );
+static int rtp_packetize_spx ( sout_stream_t *, sout_stream_id_t *, block_t * );
+static int rtp_packetize_t140 ( sout_stream_t *, sout_stream_id_t *, block_t * );
static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
{
* 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;
+ 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;
id->p_stream = p_stream;
- id->i_timestamp_start = rand()&0xffffffff;
id->i_sequence = rand()&0xffff;
id->i_payload_type = p_sys->i_payload_type;
id->ssrc[0] = rand()&0xff;
msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
vlc_mutex_init( p_stream, &id->lock_sink );
- id->fdc = 0;
- id->fdv = NULL;
+ 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");
- id->p_fifo = block_FifoNew( p_stream );
-
- if( vlc_thread_create( id, "RTP send thread", ThreadSend,
- VLC_THREAD_PRIORITY_HIGHEST, VLC_FALSE ) )
- {
- vlc_mutex_destroy( &id->lock_sink );
- vlc_object_destroy( id );
- return NULL;
- }
if( p_sys->psz_destination != NULL )
+ switch( p_sys->proto )
+ {
+ case IPPROTO_TCP:
+ case IPPROTO_DCCP:
+ 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;
+ }
+ 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;
+ }
+ rtp_add_sink( id, fd, p_sys->rtcp_mux );
+ }
+ }
+
+ if( p_fmt == NULL )
{
- 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, IPPROTO_UDP );
+ char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
- if( fd == -1 )
+ if( psz == NULL ) /* Uho! */
+ ;
+ else
+ if( strncmp( psz, "ts", 2 ) == 0 )
+ {
+ id->i_payload_type = 33;
+ id->psz_rtpmap = strdup( "MP2T/90000" );
+ }
+ else
{
- msg_Err( p_stream, "cannot create RTP socket" );
- vlc_thread_join( id );
- vlc_mutex_destroy( &id->lock_sink );
- vlc_object_destroy( id );
- return NULL;
+ id->psz_rtpmap = strdup( "MP2P/90000" );
}
- rtp_add_sink( id, fd );
}
-
- if( p_fmt == NULL )
- ;
else
switch( p_fmt->i_codec )
{
char *p_64_pps = NULL;
char hexa[6+1];
- while( i_buffer > 4 &&
+ while( i_buffer > 4 &&
p_buffer[0] == 0 && p_buffer[1] == 0 &&
p_buffer[2] == 0 && p_buffer[3] == 1 )
{
id->psz_fmtp = strdup( "octet-align=1" );
id->i_clock_rate = p_fmt->audio.i_rate;
id->pf_packetize = rtp_packetize_amr;
- break;
+ break;
case VLC_FOURCC( 's', 'a', 'w', 'b' ):
id->psz_rtpmap = strdup( p_fmt->audio.i_channels == 2 ?
"AMR-WB/16000/2" : "AMR-WB/16000" );
id->psz_fmtp = strdup( "octet-align=1" );
id->i_clock_rate = p_fmt->audio.i_rate;
id->pf_packetize = rtp_packetize_amr;
- break;
+ break;
+ case VLC_FOURCC( 's', 'p', 'x', ' ' ):
+ id->i_payload_type = p_sys->i_payload_type++;
+ if( asprintf( &id->psz_rtpmap, "SPEEX/%d",
+ p_fmt->audio.i_rate ) == -1)
+ id->psz_rtpmap = NULL;
+ id->i_clock_rate = p_fmt->audio.i_rate;
+ id->pf_packetize = rtp_packetize_spx;
+ break;
+ case VLC_FOURCC( 't', '1', '4', '0' ):
+ id->psz_rtpmap = strdup( "t140/1000" );
+ 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 );
- if( id->fdc > 0 )
- rtp_del_sink( id, id->fdv[0] );
- vlc_thread_join( id );
- vlc_mutex_destroy( &id->lock_sink );
- vlc_object_destroy( id );
- return NULL;
+ goto error;
}
+ if( cscov != -1 )
+ cscov += 8 /* UDP */ + 12 /* RTP */;
+ if( id->sinkc > 0 )
+ net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
+
if( id->i_payload_type == p_sys->i_payload_type )
p_sys->i_payload_type++;
p_sys->psz_destination,
p_sys->i_ttl, id->i_port, id->i_port + 1 );
+ id->p_fifo = block_FifoNew( p_stream );
+ if( vlc_thread_create( id, "RTP send thread", ThreadSend,
+ VLC_THREAD_PRIORITY_HIGHEST, VLC_FALSE ) )
+ goto error;
+
/* Update p_sys context */
vlc_mutex_lock( &p_sys->lock_es );
TAB_APPEND( p_sys->i_es, p_sys->es, id );
p_sys->psz_sdp = psz_sdp;
vlc_mutex_unlock( &p_sys->lock_sdp );
- p_sys->i_sdp_version++;
-
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->b_export_sdp_file ) FileSetup( p_stream );
- vlc_object_attach( id, 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_object_kill( id );
+ if( id->p_fifo != NULL )
+ {
+ vlc_object_kill( id );
+ block_FifoWake( id->p_fifo );
+ vlc_thread_join( id );
+ block_FifoRelease( id->p_fifo );
+ }
vlc_mutex_lock( &p_sys->lock_es );
TAB_REMOVE( p_sys->i_es, p_sys->es, id );
if( id->rtsp_id )
RtspDelId( p_sys->rtsp, id->rtsp_id );
- if( id->fdc > 0 )
- rtp_del_sink( id, id->fdv[0] ); /* sink for explicit dst= */
+ if( id->sinkc > 0 )
+ rtp_del_sink( id, id->sinkv[0].rtp_fd ); /* sink for explicit dst= */
+ if( id->listen_fd != NULL )
+ net_ListenClose( id->listen_fd );
- vlc_thread_join( id );
vlc_mutex_destroy( &id->lock_sink );
- block_FifoRelease( id->p_fifo );
/* Update SDP (sap/file) */
if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
{
sout_stream_sys_t *p_sys = p_stream->p_sys;
sout_instance_t *p_sout = p_stream->p_sout;
- announce_method_t *p_method = sout_SAPMethod();
/* Remove the previous session */
if( p_sys->p_session != NULL)
{
sout_AnnounceUnRegister( p_sout, p_sys->p_session);
- sout_AnnounceSessionDestroy( 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, SOUT_CFG_PREFIX,
p_sys->psz_sdp,
p_sys->psz_destination,
p_method );
+ sout_MethodRelease( p_method );
}
- sout_MethodRelease( p_method );
return VLC_SUCCESS;
}
if( ( f = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
{
- msg_Err( p_stream, "cannot open file '%s' (%s)",
- p_sys->psz_sdp_file, strerror(errno) );
+ msg_Err( p_stream, "cannot open file '%s' (%m)",
+ p_sys->psz_sdp_file );
return VLC_EGENERIC;
}
{
sout_stream_id_t *id = (sout_stream_id_t *)p_this;
unsigned i_caching = id->i_caching;
+#ifdef HAVE_TEE
+ int fd[5] = { -1, -1, -1, -1, -1 };
+
+ if( pipe( fd ) )
+ fd[0] = fd[1] = -1;
+ else
+ if( pipe( fd ) )
+ fd[2] = fd[3] = -1;
+ else
+ fd[4] = open( "/dev/null", O_WRONLY );
+#endif
while( !id->b_die )
{
block_t *out = block_FifoGet( id->p_fifo );
+ if( out == NULL )
+ continue; /* Forced wakeup */
+
mtime_t i_date = out->i_dts + i_caching;
+ ssize_t len = out->i_buffer;
+#ifdef HAVE_TEE
+ if( fd[4] != -1 )
+ len = write( fd[1], out->p_buffer, len);
+ if( len == -1 )
+ continue; /* Uho - should not happen */
+#endif
mwait( i_date );
vlc_mutex_lock( &id->lock_sink );
- for( int i = 0; i < id->fdc; i++ )
- send( id->fdv[i], out->p_buffer, out->i_buffer, 0 );
+ unsigned deadc = 0; /* How many dead sockets? */
+ int deadv[id->sinkc]; /* Dead sockets list */
+
+ for( int i = 0; i < id->sinkc; i++ )
+ {
+ SendRTCP( id->sinkv[i].rtcp, out );
+
+#ifdef HAVE_TEE
+ tee( fd[0], fd[3], len, 0 );
+ if( splice( fd[2], NULL, id->sinkv[i].rtp_fd, NULL, len,
+ SPLICE_F_NONBLOCK ) >= 0 )
+ continue;
+ if( errno == EAGAIN )
+ continue;
+
+ /* splice failed */
+ splice( fd[2], NULL, fd[4], NULL, len, 0 );
+#else
+ if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
+ continue;
+#endif
+ /* Retry sending to root out soft-errors */
+ if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
+ continue;
+
+ deadv[deadc++] = id->sinkv[i].rtp_fd;
+ }
vlc_mutex_unlock( &id->lock_sink );
block_Release( out );
+#ifdef HAVE_TEE
+ splice( fd[0], NULL, fd[4], NULL, len, 0 );
+#endif
+
+ for( unsigned i = 0; i < deadc; i++ )
+ {
+ msg_Dbg( id, "removing socket %d", deadv[i] );
+ rtp_del_sink( id, deadv[i] );
+ }
+
+ /* Hopefully we won't overflow the SO_MAXCONN accept queue */
+ while( id->listen_fd != NULL )
+ {
+ int fd = net_Accept( id, id->listen_fd, 0 );
+ if( fd == -1 )
+ break;
+ msg_Dbg( id, "adding socket %d", fd );
+ rtp_add_sink( id, fd, VLC_TRUE );
+ }
}
+
+#ifdef HAVE_TEE
+ for( unsigned i = 0; i < 5; i++ )
+ close( fd[i] );
+#endif
}
static inline void rtp_packetize_send( sout_stream_id_t *id, block_t *out )
block_FifoPut( id->p_fifo, out );
}
-int rtp_add_sink( sout_stream_id_t *id, int fd )
+int rtp_add_sink( sout_stream_id_t *id, int fd, vlc_bool_t rtcp_mux )
{
+ 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, "RTCP failed!" );
+
vlc_mutex_lock( &id->lock_sink );
- TAB_APPEND( id->fdc, id->fdv, fd );
+ INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink );
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 );
- TAB_REMOVE( id->fdc, id->fdv, fd );
+ 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 );
- net_Close( fd );
+
+ CloseRTCP( sink.rtcp );
+ net_Close( sink.rtp_fd );
+}
+
+uint16_t rtp_get_seq( const sout_stream_id_t *id )
+{
+ /* This will return values for the next packet.
+ * Accounting for caching would not be totally trivial. */
+ return id->i_sequence;
+}
+
+/* FIXME: this is pretty bad - if we remove and then insert an ES
+ * the number will get unsynched from inside RTSP */
+unsigned rtp_get_num( const sout_stream_id_t *id )
+{
+ sout_stream_sys_t *p_sys = id->p_stream->p_sys;
+ int i;
+
+ vlc_mutex_lock( &p_sys->lock_es );
+ for( i = 0; i < p_sys->i_es; i++ )
+ {
+ if( id == p_sys->es[i] )
+ break;
+ }
+ vlc_mutex_unlock( &p_sys->lock_es );
+
+ return i;
}
+
/****************************************************************************
* rtp_packetize_*:
****************************************************************************/
i_size = i_offset - ( p_buffer[i_offset-1] == 0 ? 1 : 0);
i_skip = i_offset;
break;
- }
+ }
}
/* TODO add STAP-A to remove a lot of overhead with small slice/sei/... */
rtp_packetize_h264_nal( p_stream, id, p_buffer, i_size,
return VLC_SUCCESS;
}
+static int rtp_packetize_t140( sout_stream_t *p_stream, sout_stream_id_t *id,
+ block_t *in )
+{
+ const size_t i_max = id->i_mtu - 12;
+ const uint8_t *p_data = in->p_buffer;
+ size_t i_data = in->i_buffer;
+
+ for( unsigned i_packet = 0; i_data > 0; i_packet++ )
+ {
+ size_t i_payload = i_data;
+
+ /* Make sure we stop on an UTF-8 character boundary
+ * (assuming the input is valid UTF-8) */
+ if( i_data > i_max )
+ {
+ i_payload = i_max;
+
+ while( ( p_data[i_payload] & 0xC0 ) == 0x80 )
+ {
+ if( i_payload == 0 )
+ return VLC_SUCCESS; /* fishy input! */
+
+ i_payload--;
+ }
+ }
+
+ block_t *out = block_New( p_stream, 12 + i_payload );
+ if( out == NULL )
+ return VLC_SUCCESS;
+
+ rtp_packetize_common( id, out, 0, in->i_pts + i_packet );
+ memcpy( out->p_buffer + 12, p_data, i_payload );
+
+ out->i_buffer = 12 + i_payload;
+ out->i_dts = out->i_pts;
+ out->i_length = 0;
+
+ rtp_packetize_send( id, out );
+
+ p_data += i_payload;
+ i_data -= i_payload;
+ }
+
+ return VLC_SUCCESS;
+}
+
/*****************************************************************************
* Non-RTP mux
*****************************************************************************/
p_grab->pf_write = AccessOutGrabberWrite;
return p_grab;
}
+
+static int rtp_packetize_spx( sout_stream_t *p_stream, sout_stream_id_t *id,
+ block_t *in )
+{
+ uint8_t *p_buffer = in->p_buffer;
+ int i_data_size, i_payload_size, i_payload_padding;
+ i_data_size = i_payload_size = in->i_buffer;
+ i_payload_padding = 0;
+ block_t *p_out;
+
+ if ( in->i_buffer + 12 > id->i_mtu )
+ {
+ msg_Warn( p_stream, "Cannot send packet larger than output MTU" );
+ return VLC_SUCCESS;
+ }
+
+ /*
+ RFC for Speex in RTP says that each packet must end on an octet
+ boundary. So, we check to see if the number of bytes % 4 is zero.
+ If not, we have to add some padding.
+
+ This MAY be overkill since packetization is handled elsewhere and
+ appears to ensure the octet boundary. However, better safe than
+ sorry.
+ */
+ if ( i_payload_size % 4 )
+ {
+ i_payload_padding = 4 - ( i_payload_size % 4 );
+ i_payload_size += i_payload_padding;
+ }
+
+ /*
+ Allocate a new RTP p_output block of the appropriate size.
+ Allow for 12 extra bytes of RTP header.
+ */
+ p_out = block_New( p_stream, 12 + i_payload_size );
+
+ if ( i_payload_padding )
+ {
+ /*
+ The padding is required to be a zero followed by all 1s.
+ */
+ char c_first_pad, c_remaining_pad;
+ c_first_pad = 0x7F;
+ c_remaining_pad = 0xFF;
+
+ /*
+ Allow for 12 bytes before the i_data_size because
+ of the expected RTP header added during
+ rtp_packetize_common.
+ */
+ p_out->p_buffer[12 + i_data_size] = c_first_pad;
+ switch (i_payload_padding)
+ {
+ case 2:
+ p_out->p_buffer[12 + i_data_size + 1] = c_remaining_pad;
+ break;
+ case 3:
+ p_out->p_buffer[12 + i_data_size + 1] = c_remaining_pad;
+ p_out->p_buffer[12 + i_data_size + 2] = c_remaining_pad;
+ break;
+ }
+ }
+
+ /* Add the RTP header to our p_output buffer. */
+ rtp_packetize_common( id, p_out, 0, (in->i_pts > 0 ? in->i_pts : in->i_dts) );
+ /* Copy the Speex payload to the p_output buffer */
+ memcpy( &p_out->p_buffer[12], p_buffer, i_data_size );
+
+ p_out->i_buffer = 12 + i_payload_size;
+ p_out->i_dts = in->i_dts;
+ p_out->i_length = in->i_length;
+
+ /* Queue the buffer for actual transmission. */
+ rtp_packetize_send( id, p_out );
+ return VLC_SUCCESS;
+}
projects / vlc / blobdiff