*
* 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., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
-#include <stdlib.h>
+#include "config.h"
+#ifdef HAVE_UNISTD_H
+# include <unistd.h>
+#endif
#include <errno.h>
#include <vlc/vlc.h>
-#include <vlc/input.h>
-#include <vlc/sout.h>
+#include <vlc_sout.h>
+#include <vlc_block.h>
-#include "vlc_httpd.h"
-#include "network.h"
+#include <vlc_httpd.h>
+#include <vlc_url.h>
+#include <vlc_network.h>
+#include <vlc_charset.h>
+#include <vlc_strings.h>
/*****************************************************************************
* Module descriptor
*****************************************************************************/
+
+#define MTU_REDUCE 50
+
#define DST_TEXT N_("Destination")
#define DST_LONGTEXT N_( \
- "Allows you to specify the output URL used for the streaming output." )
+ "This is the output URL that will be used." )
#define SDP_TEXT N_("SDP")
#define SDP_LONGTEXT N_( \
- "Allows you to specify the SDP used for the streaming output. " \
- "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." )
+ "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 MUX_TEXT N_("Muxer")
#define MUX_LONGTEXT N_( \
- "Allows you to specify the muxer used for the streaming output." )
+ "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_( \
- "Allows you to specify the session name used for the streaming output." )
+ "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_( \
- "Allows you to give a broader description of the stream." )
+ "This allows you to give a broader description of the stream, that will " \
+ "be announced in the SDP (Session Descriptor)." )
#define URL_TEXT N_("Session URL")
#define URL_LONGTEXT N_( \
- "Allows you to specify a URL with additional information on the stream." )
+ "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_( \
- "Allows you to specify contact e-mail address for this session." )
-
+ "This allows you to give a contact mail address for the stream, that will " \
+ "be announced in the SDP (Session Descriptor)." )
#define PORT_TEXT N_("Port")
#define PORT_LONGTEXT N_( \
- "Allows you to specify the base port used for the RTP streaming." )
+ "This allows you to specify the base port for the RTP streaming." )
#define PORT_AUDIO_TEXT N_("Audio port")
#define PORT_AUDIO_LONGTEXT N_( \
- "Allows you to specify the default audio port used for the RTP streaming." )
+ "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_( \
- "Allows you to specify the default video port used for the RTP streaming." )
+ "This allows you to specify the default video port for the RTP streaming." )
-#define TTL_TEXT N_("Time To Live")
+#define TTL_TEXT N_("Hop limit (TTL)")
#define TTL_LONGTEXT N_( \
- "Allows you to specify the time to live for the output stream." )
+ "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \
+ "the multicast packets sent by the stream output (0 = use operating " \
+ "system built-in default).")
+
+#define 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 * );
add_string( SOUT_CFG_PREFIX "email", "", NULL, EMAIL_TEXT,
EMAIL_LONGTEXT, VLC_TRUE );
- add_integer( SOUT_CFG_PREFIX "port-audio", 1234, NULL, PORT_AUDIO_TEXT,
- PORT_LONGTEXT, VLC_TRUE );
- add_integer( SOUT_CFG_PREFIX "port-video", 1236, NULL, PORT_VIDEO_TEXT,
- PORT_LONGTEXT, VLC_TRUE );
- add_integer( SOUT_CFG_PREFIX "port", 1238, NULL, PORT_TEXT,
+ add_integer( SOUT_CFG_PREFIX "port", 1234, NULL, PORT_TEXT,
PORT_LONGTEXT, VLC_TRUE );
+ add_integer( SOUT_CFG_PREFIX "port-audio", 1230, NULL, PORT_AUDIO_TEXT,
+ PORT_AUDIO_LONGTEXT, VLC_TRUE );
+ add_integer( SOUT_CFG_PREFIX "port-video", 1232, NULL, PORT_VIDEO_TEXT,
+ PORT_VIDEO_LONGTEXT, VLC_TRUE );
add_integer( SOUT_CFG_PREFIX "ttl", 0, NULL, TTL_TEXT,
TTL_LONGTEXT, VLC_TRUE );
+ add_bool( SOUT_CFG_PREFIX "mp4a-latm", 0, NULL, RFC3016_TEXT,
+ RFC3016_LONGTEXT, VLC_FALSE );
+
set_callbacks( Open, Close );
vlc_module_end();
*****************************************************************************/
static const char *ppsz_sout_options[] = {
"dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
- "description", "url","email", NULL
+ "description", "url","email", "mp4a-latm", NULL
};
static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
int i_port_audio;
int i_port_video;
int i_ttl;
+ vlc_bool_t b_latm;
/* when need to use a private one or when using muxer */
int i_payload_type;
char *psz_destination;
int i_port;
int i_cat;
+ int i_bitrate;
/* Packetizer specific fields */
pf_rtp_packetizer_t pf_packetize;
httpd_message_t *, httpd_message_t * );
-static rtsp_client_t *RtspClientNew( sout_stream_t *, char *psz_session );
-static rtsp_client_t *RtspClientGet( sout_stream_t *, char *psz_session );
+static rtsp_client_t *RtspClientNew( sout_stream_t *, const char *psz_session );
+static rtsp_client_t *RtspClientGet( sout_stream_t *, const char *psz_session );
static void RtspClientDel( sout_stream_t *, rtsp_client_t * );
/*****************************************************************************
{
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;
+ sout_stream_sys_t *p_sys = NULL;
+ config_chain_t *p_cfg = NULL;
vlc_value_t val;
+ vlc_bool_t b_rtsp = VLC_FALSE;
- sout_CfgParse( p_stream, SOUT_CFG_PREFIX, ppsz_sout_options, p_stream->p_cfg );
+ 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_GetString( p_stream, SOUT_CFG_PREFIX "dst" );
- if( *p_sys->psz_destination == '\0' )
- {
- free( p_sys->psz_destination );
- p_sys->psz_destination = NULL;
- }
-
+ 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_name = strdup( "NONE" );
}
- if( !p_sys->psz_destination || *p_sys->psz_destination == '\0' )
+ for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
{
- sout_cfg_t *p_cfg;
- vlc_bool_t b_ok = VLC_FALSE;
-
- 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" ) )
+ if( p_cfg->psz_value && !strncasecmp( p_cfg->psz_value, "rtsp", 4 ) )
{
- if( p_cfg->psz_value && !strncasecmp( p_cfg->psz_value, "rtsp", 4 ) )
- {
- b_ok = VLC_TRUE;
- break;
- }
+ b_rtsp = VLC_TRUE;
+ break;
}
}
- if( !b_ok )
- {
- vlc_value_t val2;
- var_Get( p_stream, SOUT_CFG_PREFIX "sdp", &val2 );
- if( !strncasecmp( val2.psz_string, "rtsp", 4 ) )
- b_ok = VLC_TRUE;
- free( val2.psz_string );
- }
+ }
+ if( !b_rtsp )
+ {
+ vlc_value_t val2;
+ var_Get( p_stream, SOUT_CFG_PREFIX "sdp", &val2 );
+ if( !strncasecmp( val2.psz_string, "rtsp", 4 ) )
+ b_rtsp = VLC_TRUE;
+ free( val2.psz_string );
+ }
- if( !b_ok )
+ if( !p_sys->psz_destination || *p_sys->psz_destination == '\0' )
+ {
+ if( !b_rtsp )
{
msg_Err( p_stream, "missing destination and not in rtsp mode" );
free( p_sys );
return VLC_EGENERIC;
}
- var_Get( p_stream, SOUT_CFG_PREFIX "ttl", &val );
- p_sys->i_ttl = val.i_int;
+ p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
+ if( p_sys->i_ttl == 0 )
+ {
+ /* Normally, we should let the default hop limit up to the core,
+ * but we have to know it to build our SDP properly, which is why
+ * we ask the core. FIXME: broken when neither sout-rtp-ttl nor
+ * ttl are set. */
+ p_sys->i_ttl = config_GetInt( p_stream, "ttl" );
+ }
+ 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->i_payload_type = 96;
p_sys->i_es = 0;
if( *val.psz_string )
{
sout_access_out_t *p_grab;
+ const char *psz_rtpmap;
+ char url[NI_MAXHOST + 8], access[22], psz_ttl[5], ipv;
- char *psz_rtpmap;
- char access[100];
- char url[p_sys->psz_destination ? strlen( p_sys->psz_destination ) + 1 + 12+1 : 14];
+ if( b_rtsp )
+ {
+ msg_Err( p_stream, "muxing is not supported in RTSP mode" );
+ free( p_sys );
+ return VLC_EGENERIC;
+ }
+ else if( !p_sys->psz_destination || *p_sys->psz_destination == '\0' )
+ {
+ msg_Err( p_stream, "RTP needs a destination when muxing" );
+ free( p_sys );
+ return VLC_EGENERIC;
+ }
/* Check muxer type */
- if( !strncasecmp( val.psz_string, "ps", 2 ) || !strncasecmp( val.psz_string, "mpeg1", 5 ) )
+ if( !strncasecmp( val.psz_string, "ps", 2 )
+ || !strncasecmp( val.psz_string, "mpeg1", 5 ) )
{
psz_rtpmap = "MP2P/90000";
}
else
{
msg_Err( p_stream, "unsupported muxer type with rtp (only ts/ps)" );
+ free( p_sys );
return VLC_EGENERIC;
}
/* create the access out */
if( p_sys->i_ttl > 0 )
{
- sprintf( access, "udp{raw,ttl=%d}", p_sys->i_ttl );
+ sprintf( access, "udp{raw,rtcp,ttl=%d}", p_sys->i_ttl );
}
else
{
- sprintf( access, "udp{raw}" );
+ strcat( access, "udp{raw,rtcp}" );
}
- sprintf( url, "%s:%d", p_sys->psz_destination, p_sys->i_port );
+
+ /* FIXME: we should check that url is a numerical address, otherwise
+ * the SDP will be quite broken (regardless of the IP protocol version)
+ * Also it might be IPv6 with no ':' if it is a DNS name.
+ */
+ ipv = ( strchr( p_sys->psz_destination, ':' ) != NULL ) ? '6' : '4';
+
+ /* IPv6 needs brackets if not already present */
+ if ((ipv == '6') && (p_sys->psz_destination[0] != '['))
+ snprintf (url, sizeof (url), "[%s]:%d",p_sys->psz_destination,
+ p_sys->i_port);
+ else
+ snprintf (url, sizeof (url), "%s:%d",p_sys->psz_destination,
+ p_sys->i_port);
+
if( !( p_sys->p_access = sout_AccessOutNew( p_sout, access, url ) ) )
{
msg_Err( p_stream, "cannot create the access out for %s://%s",
return VLC_EGENERIC;
}
p_sys->i_mtu = config_GetInt( p_stream, "mtu" ); /* XXX beurk */
- if( p_sys->i_mtu <= 16 )
+ if( p_sys->i_mtu <= 16 + MTU_REDUCE )
{
/* better than nothing */
p_sys->i_mtu = 1500;
}
+ p_sys->i_mtu -= MTU_REDUCE;
/* the access out grabber TODO export it as sout_AccessOutGrabberNew */
p_grab = p_sys->p_grab =
p_grab->p_sout = p_sout;
p_grab->psz_access = strdup( "grab" );
p_grab->p_cfg = NULL;
- p_grab->psz_name = strdup( "" );
+ 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;
}
/* create the SDP for a muxed stream (only once) */
- /* FIXME http://www.faqs.org/rfcs/rfc2327.html
- All text fields should be UTF-8 encoded. Use global a:charset to announce this.
+ /* FIXME http://www.faqs.org/rfcs/rfc4566.html
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= time should be hashed with some other value to garantee uniqueness
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
- c= /ttl should only be added in case of multicast
- 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= source-filter: we need our source address
a= x-plgroup: (missing)
RTP packets need to get the correct src IP address */
- p_sys->psz_sdp =
- malloc( 10 + 30 + 10 + strlen( p_sys->psz_session_name ) +
- 10 + strlen( p_sys->psz_session_description ) + 10 + strlen( p_sys->psz_session_url ) +
- 10 + strlen( p_sys->psz_session_email ) + 10 + strlen( p_sys->psz_destination ) +
- 10 + 10 + strlen( PACKAGE_STRING ) +
- 20 + 10 + 20 + 10 + strlen( psz_rtpmap ) );
- sprintf( p_sys->psz_sdp,
- "v=0\r\n"
- "o=- "I64Fd" %d IN IP4 127.0.0.1\r\n"
- "s=%s\r\n"
- "i=%s\r\n"
- "u=%s\r\n"
- "e=%s\r\n"
- "t=0 0\r\n" /* permanent stream */ /* when scheduled from vlm, we should set this info correctly */
- "a=tool:"PACKAGE_STRING"\r\n"
- "c=IN IP4 %s/%d\r\n"
- "m=video %d RTP/AVP %d\r\n"
- "a=rtpmap:%d %s\r\n",
- p_sys->i_sdp_id, p_sys->i_sdp_version,
- p_sys->psz_session_name,
- p_sys->psz_session_description,
- p_sys->psz_session_url,
- p_sys->psz_session_email,
- p_sys->psz_destination, p_sys->i_ttl,
- p_sys->i_port, p_sys->i_payload_type,
- p_sys->i_payload_type, psz_rtpmap );
- fprintf( stderr, "sdp=%s", p_sys->psz_sdp );
+ if( ipv == '4'
+ && net_AddressIsMulticast( VLC_OBJECT(p_stream), p_sys->psz_destination ) )
+ {
+ snprintf( psz_ttl, sizeof( psz_ttl ), "/%d", p_sys->i_ttl );
+ psz_ttl[sizeof( psz_ttl ) - 1] = '\0';
+ }
+ else
+ {
+ psz_ttl[0] = '\0';
+ }
+
+ asprintf( &p_sys->psz_sdp,
+ "v=0\r\n"
+ /* FIXME: source address not known :( */
+ "o=- "I64Fd" %d IN IP%c %s\r\n"
+ "s=%s\r\n"
+ "i=%s\r\n"
+ "u=%s\r\n"
+ "e=%s\r\n"
+ "c=IN IP%c %s%s\r\n"
+ "t=0 0\r\n" /* permanent stream */ /* when scheduled from vlm, we should set this info correctly */
+ "a=tool:"PACKAGE_STRING"\r\n"
+ "a=recvonly\r\n"
+ "a=type:broadcast\r\n"
+ "m=video %d RTP/AVP %d\r\n"
+ "a=rtpmap:%d %s\r\n",
+ p_sys->i_sdp_id, p_sys->i_sdp_version,
+ ipv, ipv == '6' ? "::1" : "127.0.0.1" /* FIXME */,
+ p_sys->psz_session_name,
+ p_sys->psz_session_description,
+ p_sys->psz_session_url,
+ p_sys->psz_session_email,
+ ipv, p_sys->psz_destination, psz_ttl,
+ p_sys->i_port, p_sys->i_payload_type,
+ p_sys->i_payload_type, psz_rtpmap );
+ msg_Dbg( p_stream, "sdp=%s", p_sys->psz_sdp );
/* create the rtp context */
p_sys->ssrc[0] = rand()&0xff;
var_Get( p_stream, SOUT_CFG_PREFIX "sdp", &val );
if( *val.psz_string )
{
- sout_cfg_t *p_cfg;
+ config_chain_t *p_cfg;
SDPHandleUrl( p_stream, val.psz_string );
if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' )
continue;
- if( !strcmp( p_cfg->psz_value, val.psz_string ) ) /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
+ /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */
+ if( !strcmp( p_cfg->psz_value, val.psz_string ) )
continue;
SDPHandleUrl( p_stream, p_cfg->psz_value );
{
block_Release( p_sys->packet );
}
+ if( p_sys->b_export_sap )
+ {
+ p_sys->p_mux = NULL;
+ SapSetup( p_stream );
+ }
}
while( p_sys->i_rtsp > 0 )
- {
RtspClientDel( p_stream, p_sys->rtsp[0] );
- }
vlc_mutex_destroy( &p_sys->lock_sdp );
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 );
- }
+
if( p_sys->p_rtsp_url )
- {
httpd_UrlDelete( p_sys->p_rtsp_url );
- }
+
if( p_sys->p_rtsp_host )
- {
httpd_HostDelete( p_sys->p_rtsp_host );
- }
-#if 0
- /* why? is this disabled? */
+
if( p_sys->psz_session_name )
- {
free( p_sys->psz_session_name );
- p_sys->psz_session_name = NULL;
- }
+
if( p_sys->psz_session_description )
- {
free( p_sys->psz_session_description );
- p_sys->psz_session_description = NULL;
- }
+
if( p_sys->psz_session_url )
- {
free( p_sys->psz_session_url );
- p_sys->psz_session_url = NULL;
- }
+
if( p_sys->psz_session_email )
- {
free( p_sys->psz_session_email );
- p_sys->psz_session_email = NULL;
- }
-#endif
+
if( p_sys->psz_sdp )
- {
free( p_sys->psz_sdp );
+
+ if( p_sys->b_export_sdp_file )
+ {
+#ifdef HAVE_UNISTD_H
+ unlink( p_sys->psz_sdp_file );
+#endif
+ free( p_sys->psz_sdp_file );
}
+ if( p_sys->psz_destination )
+ free( p_sys->psz_destination );
free( p_sys );
}
{
if( p_sys->p_httpd_file )
{
- msg_Err( p_stream, "You can used sdp=http:// only once" );
+ msg_Err( p_stream, "you can use sdp=http:// only once" );
return;
}
{
if( p_sys->p_rtsp_url )
{
- msg_Err( p_stream, "You can used sdp=rtsp:// only once" );
+ msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
return;
}
{
if( p_sys->b_export_sdp_file )
{
- msg_Err( p_stream, "You can used sdp=file:// only once" );
+ msg_Err( p_stream, "you can use sdp=file:// only once" );
return;
}
p_sys->b_export_sdp_file = VLC_TRUE;
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
- c= /ttl should only be added in case of multicast
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( sout_stream_t *p_stream, char *psz_destination, vlc_bool_t b_rtsp )
+static char *SDPGenerate( const sout_stream_t *p_stream,
+ const char *psz_destination, vlc_bool_t b_rtsp )
{
sout_stream_sys_t *p_sys = p_stream->p_sys;
int i_size;
- char *psz_sdp, *p;
+ char *psz_sdp, *p, ipv;
int i;
- i_size = strlen( "v=0\r\n" ) +
- strlen( "o=- * * IN IP4 127.0.0.1\r\n" ) + 10 + 10 +
- strlen( "s=*\r\n" ) + strlen( p_sys->psz_session_name ) +
- strlen( "i=*\r\n" ) + strlen( p_sys->psz_session_description ) +
- strlen( "u=*\r\n" ) + strlen( p_sys->psz_session_url ) +
- strlen( "e=*\r\n" ) + strlen( p_sys->psz_session_email ) +
- strlen( "t=0 0\r\n" ) + /* permanent stream */ /* when scheduled from vlm, we should set this info correctly */
- strlen( "a=tool:"PACKAGE_STRING"\r\n" ) +
- strlen( "c=IN IP4 */*\r\n" ) + 20 + 10 +
- strlen( psz_destination ? psz_destination : "0.0.0.0") ;
+ /* 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" ) + /* permanent stream */ /* when scheduled from vlm, we should set this info correctly */
+ sizeof( "b=RR:0\r\n" ) +
+ sizeof( "a=tool:"PACKAGE_STRING"\r\n" ) +
+ sizeof( "c=IN IP4 */*\r\n" ) + 20 + 10 +
+ strlen( psz_destination ) ;
for( i = 0; i < p_sys->i_es; i++ )
{
sout_stream_id_t *id = p_sys->es[i];
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( b_rtsp )
{
- i_size += strlen( "a=control:*/trackid=*\r\n" ) + strlen( p_sys->psz_rtsp_control ) + 10;
+ i_size += strlen( "a=control:*/trackID=*\r\n" ) + strlen( p_sys->psz_rtsp_control ) + 10;
}
}
+ if( p_sys->p_mux )
+ {
+ i_size += strlen( "m=video %d RTP/AVP %d\r\n" ) +10 +10;
+ }
+
+ ipv = ( strchr( psz_destination, ':' ) != NULL ) ? '6' : '4';
p = psz_sdp = malloc( i_size );
p += sprintf( p, "v=0\r\n" );
- p += sprintf( p, "o=- "I64Fd" %d IN IP4 127.0.0.1\r\n",
- p_sys->i_sdp_id, p_sys->i_sdp_version );
+ p += sprintf( p, "o=- "I64Fd" %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, "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, "c=IN IP4 %s/%d\r\n", psz_destination ? psz_destination : "0.0.0.0",
- p_sys->i_ttl );
+ p += sprintf( p, "c=IN IP%c %s", ipv, psz_destination );
+
+ 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" );
for( i = 0; i < p_sys->i_es; i++ )
{
{
continue;
}
+ if ( id->i_bitrate )
+ {
+ p += sprintf(p,"b=AS:%d\r\n",id->i_bitrate);
+ }
if( id->psz_rtpmap )
{
p += sprintf( p, "a=rtpmap:%d %s\r\n", id->i_payload_type,
}
if( b_rtsp )
{
- p += sprintf( p, "a=control:%s/trackid=%d\r\n", p_sys->psz_rtsp_control, i );
+ p += sprintf( p, "a=control:/trackID=%d\r\n", i );
}
}
+ if( p_sys->p_mux )
+ {
+ p += sprintf( p, "m=video %d RTP/AVP %d\r\n",
+ p_sys->i_port, p_sys->i_payload_type );
+ }
return psz_sdp;
}
static int rtp_packetize_ac3 ( sout_stream_t *, sout_stream_id_t *, block_t * );
static int rtp_packetize_split( sout_stream_t *, sout_stream_id_t *, block_t * );
static int rtp_packetize_mp4a ( sout_stream_t *, sout_stream_id_t *, block_t * );
+static int rtp_packetize_mp4a_latm ( sout_stream_t *, sout_stream_id_t *, block_t * );
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 void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
if( p_sys->psz_destination )
{
- char access[100];
- char url[strlen( p_sys->psz_destination ) + 1 + 12 + 1];
+ char access[22];
+ char url[NI_MAXHOST + 8];
/* first try to create the access out */
- if( p_sys->i_ttl > 0 )
+ if( p_sys->i_ttl )
{
- sprintf( access, "udp{raw,ttl=%d}", p_sys->i_ttl );
+ snprintf( access, sizeof( access ), "udp{raw,rtcp,ttl=%d}",
+ p_sys->i_ttl );
+ access[sizeof( access ) - 1] = '\0';
}
else
- {
- sprintf( access, "udp{raw}" );
- }
- sprintf( url, "%s:%d", p_sys->psz_destination, i_port );
+ strcpy( access, "udp{raw,rtcp}" );
+
+ snprintf( url, sizeof( url ), (( p_sys->psz_destination[0] != '[' ) &&
+ strchr( p_sys->psz_destination, ':' )) ? "[%s]:%d" : "%s:%d",
+ p_sys->psz_destination, i_port );
+ url[sizeof( url ) - 1] = '\0';
+
if( ( p_access = sout_AccessOutNew( p_sout, access, url ) ) == NULL )
{
msg_Err( p_stream, "cannot create the access out for %s://%s",
id->psz_rtpmap = strdup( "H263-1998/90000" );
id->pf_packetize = rtp_packetize_h263;
break;
+ case VLC_FOURCC( 'h', '2', '6', '4' ):
+ id->i_payload_type = p_sys->i_payload_type++;
+ id->i_clock_rate = 90000;
+ id->psz_rtpmap = strdup( "H264/90000" );
+ 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 );
+ if( p_64_sps )
+ free( p_64_sps );
+ if( p_64_pps )
+ free( p_64_pps );
+ }
+ if( !id->psz_fmtp )
+ id->psz_fmtp = strdup( "packetization-mode=1" );
+ break;
case VLC_FOURCC( 'm', 'p', '4', 'v' ):
{
}
case VLC_FOURCC( 'm', 'p', '4', 'a' ):
{
- char hexa[2*p_fmt->i_extra +1];
-
id->i_payload_type = p_sys->i_payload_type++;
id->i_clock_rate = p_fmt->audio.i_rate;
- id->psz_rtpmap = malloc( strlen( "mpeg4-generic/" ) + 12 );
- sprintf( id->psz_rtpmap, "mpeg4-generic/%d", p_fmt->audio.i_rate );
- id->pf_packetize = rtp_packetize_mp4a;
- id->psz_fmtp = malloc( 200 + 2 * p_fmt->i_extra );
- sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
- sprintf( id->psz_fmtp,
- "streamtype=5; profile-level-id=15; mode=AAC-hbr; "
- "config=%s; SizeLength=13;IndexLength=3; "
- "IndexDeltaLength=3; Profile=1;", hexa );
+
+ if(!p_sys->b_latm)
+ {
+ char hexa[2*p_fmt->i_extra +1];
+
+ id->psz_rtpmap = malloc( strlen( "mpeg4-generic/" ) + 12 );
+ sprintf( id->psz_rtpmap, "mpeg4-generic/%d", p_fmt->audio.i_rate );
+ id->pf_packetize = rtp_packetize_mp4a;
+ id->psz_fmtp = malloc( 200 + 2 * p_fmt->i_extra );
+ sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
+ sprintf( id->psz_fmtp,
+ "streamtype=5; profile-level-id=15; mode=AAC-hbr; "
+ "config=%s; SizeLength=13;IndexLength=3; "
+ "IndexDeltaLength=3; Profile=1;", hexa );
+ }
+ 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;
+
+ asprintf( &id->psz_rtpmap, "MP4A-LATM/%d/%d",
+ p_fmt->audio.i_rate, p_fmt->audio.i_channels );
+ id->pf_packetize = rtp_packetize_mp4a_latm;
+ sprintf_hexa( hexa, config, 6 );
+ asprintf( &id->psz_fmtp, "profile-level-id=15; "
+ "object=2; cpresent=0; config=%s", hexa );
+ }
break;
}
case VLC_FOURCC( 's', 'a', 'm', 'r' ):
id->ssrc[3] = rand()&0xff;
id->i_sequence = rand()&0xffff;
id->i_timestamp_start = rand()&0xffffffff;
+ id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
id->i_mtu = config_GetInt( p_stream, "mtu" ); /* XXX beuk */
- if( id->i_mtu <= 16 )
+ if( id->i_mtu <= 16 + MTU_REDUCE )
{
/* better than nothing */
id->i_mtu = 1500;
}
- msg_Dbg( p_stream, "using mtu=%d", id->i_mtu );
+ id->i_mtu -= MTU_REDUCE;
+ msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
if( p_sys->p_rtsp_url )
{
char psz_urlc[strlen( p_sys->psz_rtsp_control ) + 1 + 10];
- sprintf( psz_urlc, "%s/trackid=%d", p_sys->psz_rtsp_path, p_sys->i_es );
- fprintf( stderr, "rtsp: adding %s\n", psz_urlc );
+ sprintf( psz_urlc, "%s/trackID=%d", p_sys->psz_rtsp_path, p_sys->i_es );
+ msg_Dbg( p_stream, "rtsp: adding %s\n", psz_urlc );
id->p_rtsp_url = httpd_UrlNewUnique( p_sys->p_rtsp_host, psz_urlc, NULL, NULL, NULL );
if( id->p_rtsp_url )
{
+ httpd_UrlCatch( id->p_rtsp_url, HTTPD_MSG_DESCRIBE, RtspCallbackId, (void*)id );
httpd_UrlCatch( id->p_rtsp_url, HTTPD_MSG_SETUP, RtspCallbackId, (void*)id );
- //httpd_UrlCatch( id->p_rtsp_url, HTTPD_MSG_PLAY, RtspCallback, (void*)p_stream );
- //httpd_UrlCatch( id->p_rtsp_url, HTTPD_MSG_PAUSE, RtspCallback, (void*)p_stream );
+ httpd_UrlCatch( id->p_rtsp_url, HTTPD_MSG_PLAY, RtspCallbackId, (void*)id );
+ httpd_UrlCatch( id->p_rtsp_url, HTTPD_MSG_PAUSE, RtspCallbackId, (void*)id );
+ httpd_UrlCatch( id->p_rtsp_url, HTTPD_MSG_TEARDOWN, RtspCallbackId, (void*)id );
}
}
p_sys->i_sdp_version++;
- fprintf( stderr, "sdp=%s", p_sys->psz_sdp );
+ msg_Dbg( p_stream, "sdp=%s", p_sys->psz_sdp );
/* Update SDP (sap/file) */
if( p_sys->b_export_sap ) SapSetup( p_stream );
if( id->rtsp_access ) free( id->rtsp_access );
/* Update SDP (sap/file) */
- if( p_sys->b_export_sap ) SapSetup( p_stream );
+ if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
if( p_sys->b_export_sdp_file ) FileSetup( p_stream );
free( id );
unsigned int i_data = p_buffer->i_buffer;
unsigned int i_max = p_sys->i_mtu - 12;
- int i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
+ unsigned i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max;
while( i_data > 0 )
{
p_sys->i_sequence++;
}
- i_size = __MIN( i_data, p_sys->i_mtu - p_sys->packet->i_buffer );
+ i_size = __MIN( i_data,
+ (unsigned)(p_sys->i_mtu - p_sys->packet->i_buffer) );
memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer],
p_data, i_size );
{
sout_stream_sys_t *p_sys = p_stream->p_sys;
sout_instance_t *p_sout = p_stream->p_sout;
- /* FIXME: use sout_AnnounceMethodCreate */
- announce_method_t *p_method = (announce_method_t *)
- malloc(sizeof(announce_method_t));
+ announce_method_t *p_method = sout_SAPMethod();
/* Remove the previous session */
if( p_sys->p_session != NULL)
sout_AnnounceSessionDestroy( p_sys->p_session );
p_sys->p_session = NULL;
}
- p_method->i_type = METHOD_TYPE_SAP;
- p_method->psz_address = NULL; /* FIXME */
- if( p_sys->i_es > 0 && p_sys->psz_sdp && *p_sys->psz_sdp )
+ if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp )
{
- p_sys->p_session = sout_AnnounceRegisterSDP( p_sout, p_sys->psz_sdp,
+ p_sys->p_session = sout_AnnounceRegisterSDP( p_sout, SOUT_CFG_PREFIX,
+ p_sys->psz_sdp,
+ p_sys->psz_destination,
p_method );
}
- free( p_method );
+ sout_MethodRelease( p_method );
return VLC_SUCCESS;
}
sout_stream_sys_t *p_sys = p_stream->p_sys;
FILE *f;
- if( ( f = fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
+ 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) );
{
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 );
+ 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,
/****************************************************************************
* RTSP:
****************************************************************************/
-static rtsp_client_t *RtspClientNew( sout_stream_t *p_stream, char *psz_session )
+static rtsp_client_t *RtspClientNew( sout_stream_t *p_stream, const char *psz_session )
{
rtsp_client_t *rtsp = malloc( sizeof( rtsp_client_t ));
- rtsp->psz_session = psz_session;
+ rtsp->psz_session = strdup( psz_session );
rtsp->i_last = 0;
rtsp->b_playing = VLC_FALSE;
rtsp->i_id = 0;
return rtsp;
}
-static rtsp_client_t *RtspClientGet( sout_stream_t *p_stream, char *psz_session )
+static rtsp_client_t *RtspClientGet( sout_stream_t *p_stream, const char *psz_session )
{
int i;
+
+ if( !psz_session ) return NULL;
+
for( i = 0; i < p_stream->p_sys->i_rtsp; i++ )
{
if( !strcmp( p_stream->p_sys->rtsp[i]->psz_session, psz_session ) )
{
sout_stream_sys_t *p_sys = p_stream->p_sys;
- fprintf( stderr, "rtsp setup: %s : %d / %s\n", url->psz_host, url->i_port, url->psz_path );
+ msg_Dbg( p_stream, "rtsp setup: %s : %d / %s\n", url->psz_host, url->i_port, url->psz_path );
p_sys->p_rtsp_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host, url->i_port > 0 ? url->i_port : 554 );
if( p_sys->p_rtsp_host == NULL )
return VLC_EGENERIC;
}
httpd_UrlCatch( p_sys->p_rtsp_url, HTTPD_MSG_DESCRIBE, RtspCallback, (void*)p_stream );
+ httpd_UrlCatch( p_sys->p_rtsp_url, HTTPD_MSG_SETUP, RtspCallback, (void*)p_stream );
httpd_UrlCatch( p_sys->p_rtsp_url, HTTPD_MSG_PLAY, RtspCallback, (void*)p_stream );
httpd_UrlCatch( p_sys->p_rtsp_url, HTTPD_MSG_PAUSE, RtspCallback, (void*)p_stream );
httpd_UrlCatch( p_sys->p_rtsp_url, HTTPD_MSG_TEARDOWN, RtspCallback, (void*)p_stream );
{
sout_stream_t *p_stream = (sout_stream_t*)p_args;
sout_stream_sys_t *p_sys = p_stream->p_sys;
- char *psz_destination = p_sys->psz_destination;
- char *psz_session = NULL;
+ char *psz_destination = p_sys->psz_destination;
+ const char *psz_session = NULL;
+ const char *psz_cseq = NULL;
if( answer == NULL || query == NULL )
{
return VLC_SUCCESS;
}
- fprintf( stderr, "RtspCallback query: type=%d\n", query->i_type );
+ //fprintf( stderr, "RtspCallback query: type=%d\n", query->i_type );
answer->i_proto = HTTPD_PROTO_RTSP;
answer->i_version= query->i_version;
answer->i_type = HTTPD_MSG_ANSWER;
+ answer->i_body = 0;
+ answer->p_body = NULL;
+ if( httpd_MsgGet( query, "Require" ) != NULL )
+ answer->i_status = 551;
+ else
switch( query->i_type )
{
case HTTPD_MSG_DESCRIBE:
char *psz_sdp = SDPGenerate( p_stream, psz_destination ? psz_destination : "0.0.0.0", VLC_TRUE );
answer->i_status = 200;
- answer->psz_status = strdup( "OK" );
- httpd_MsgAdd( answer, "Content-type", "%s", "application/sdp" );
-
- answer->p_body = psz_sdp;
+ httpd_MsgAdd( answer, "Content-Type", "%s", "application/sdp" );
+ httpd_MsgAdd( answer, "Content-Base", "%s", p_sys->psz_rtsp_control );
+ answer->p_body = (uint8_t *)psz_sdp;
answer->i_body = strlen( psz_sdp );
break;
}
+ case HTTPD_MSG_SETUP:
+ answer->i_status = 459;
+ break;
+
case HTTPD_MSG_PLAY:
{
rtsp_client_t *rtsp;
/* for now only multicast so easy */
answer->i_status = 200;
- answer->psz_status = strdup( "OK" );
- answer->i_body = 0;
- answer->p_body = NULL;
psz_session = httpd_MsgGet( query, "Session" );
rtsp = RtspClientGet( p_stream, psz_session );
}
break;
}
+
case HTTPD_MSG_PAUSE:
- /* FIXME */
- return VLC_EGENERIC;
+ answer->i_status = 405;
+ httpd_MsgAdd( answer, "Allow", "DESCRIBE, PLAY, TEARDOWN" );
+ break;
+
case HTTPD_MSG_TEARDOWN:
{
rtsp_client_t *rtsp;
/* for now only multicast so easy again */
answer->i_status = 200;
- answer->psz_status = strdup( "OK" );
- answer->i_body = 0;
- answer->p_body = NULL;
psz_session = httpd_MsgGet( query, "Session" );
rtsp = RtspClientGet( p_stream, psz_session );
default:
return VLC_EGENERIC;
}
- httpd_MsgAdd( answer, "Server", "VLC Server" );
+
+ httpd_MsgAdd( answer, "Server", PACKAGE_STRING );
httpd_MsgAdd( answer, "Content-Length", "%d", answer->i_body );
- httpd_MsgAdd( answer, "Cseq", "%d", atoi( httpd_MsgGet( query, "Cseq" ) ) );
+ psz_cseq = httpd_MsgGet( query, "Cseq" );
+ httpd_MsgAdd( answer, "Cseq", "%s", psz_cseq ? psz_cseq : "0" );
httpd_MsgAdd( answer, "Cache-Control", "%s", "no-cache" );
if( psz_session )
- {
httpd_MsgAdd( answer, "Session", "%s;timeout=5", psz_session );
- }
return VLC_SUCCESS;
}
-static int RtspCallbackId( httpd_callback_sys_t *p_args,
- httpd_client_t *cl,
- httpd_message_t *answer, httpd_message_t *query )
+
+/** Finds the next transport choice */
+static inline const char *transport_next( const char *str )
+{
+ /* Looks for comma */
+ str = strchr( str, ',' );
+ if( str == NULL )
+ return NULL; /* No more transport options */
+
+ str++; /* skips comma */
+ while( strchr( "\r\n\t ", *str ) )
+ str++;
+
+ return (*str) ? str : NULL;
+}
+
+
+/** Finds the next transport parameter */
+static inline const char *parameter_next( const char *str )
+{
+ while( strchr( ",;", *str ) == NULL )
+ str++;
+
+ return (*str == ';') ? (str + 1) : NULL;
+}
+
+
+static int RtspCallbackId( httpd_callback_sys_t *p_args,
+ httpd_client_t *cl,
+ httpd_message_t *answer, httpd_message_t *query )
{
sout_stream_id_t *id = (sout_stream_id_t*)p_args;
sout_stream_t *p_stream = id->p_stream;
sout_stream_sys_t *p_sys = p_stream->p_sys;
- char *psz_session = NULL;
+ char psz_session_init[21];
+ const char *psz_session;
+ const char *psz_cseq;
if( answer == NULL || query == NULL )
- {
return VLC_SUCCESS;
- }
- fprintf( stderr, "RtspCallback query: type=%d\n", query->i_type );
+ //fprintf( stderr, "RtspCallback query: type=%d\n", query->i_type );
+ /* */
answer->i_proto = HTTPD_PROTO_RTSP;
answer->i_version= query->i_version;
answer->i_type = HTTPD_MSG_ANSWER;
+ answer->i_body = 0;
+ answer->p_body = NULL;
+ /* Create new session ID if needed */
+ psz_session = httpd_MsgGet( query, "Session" );
+ if( psz_session == NULL )
+ {
+ /* FIXME: should be somewhat secure randomness */
+ snprintf( psz_session_init, sizeof(psz_session_init), I64Fd,
+ NTPtime64() + rand() );
+ }
+
+ if( httpd_MsgGet( query, "Require" ) != NULL )
+ answer->i_status = 551;
+ else
switch( query->i_type )
{
case HTTPD_MSG_SETUP:
{
- char *psz_transport = httpd_MsgGet( query, "Transport" );
+ answer->i_status = 461;
- fprintf( stderr, "HTTPD_MSG_SETUP: transport=%s\n", psz_transport );
-
- if( strstr( psz_transport, "multicast" ) && id->psz_destination )
- {
- fprintf( stderr, "HTTPD_MSG_SETUP: multicast\n" );
- answer->i_status = 200;
- answer->psz_status = strdup( "OK" );
- answer->i_body = 0;
- answer->p_body = NULL;
- psz_session = httpd_MsgGet( query, "Session" );
- if( *psz_session == 0 )
- {
- psz_session = malloc( 100 );
- sprintf( psz_session, "%d", rand() );
- }
- httpd_MsgAdd( answer, "Transport",
- "RTP/AVP/UDP;destination=%s;port=%d-%d;ttl=%d",
- id->psz_destination, id->i_port,id->i_port+1, p_sys->i_ttl );
- }
- else if( strstr( psz_transport, "unicast" ) && strstr( psz_transport, "client_port=" ) )
+ for( const char *tpt = httpd_MsgGet( query, "Transport" );
+ tpt != NULL;
+ tpt = transport_next( tpt ) )
{
- int i_port = atoi( strstr( psz_transport, "client_port=" ) + strlen("client_port=") );
- char *ip = httpd_ClientIP( cl );
+ vlc_bool_t b_multicast = VLC_TRUE, b_unsupp = VLC_FALSE;
+ unsigned loport = 5004, hiport = 5005; /* from RFC3551 */
- char psz_access[100];
- char psz_url[100];
-
- sout_access_out_t *p_access;
-
- rtsp_client_t *rtsp = NULL;
+ /* Check transport protocol. */
+ /* Currently, we only support RTP/AVP over UDP */
+ if( strncmp( tpt, "RTP/AVP", 7 ) )
+ continue;
+ tpt += 7;
+ if( strncmp( tpt, "/UDP", 4 ) == 0 )
+ tpt += 4;
+ if( strchr( ";,", *tpt ) == NULL )
+ continue;
- if( ip == NULL )
+ /* Parse transport options */
+ for( const char *opt = parameter_next( tpt );
+ opt != NULL;
+ opt = parameter_next( opt ) )
{
- answer->i_status = 400;
- answer->psz_status = strdup( "Internal server error" );
- answer->i_body = 0;
- answer->p_body = NULL;
- break;
+ if( strncmp( opt, "multicast", 9 ) == 0)
+ b_multicast = VLC_TRUE;
+ else
+ if( strncmp( opt, "unicast", 7 ) == 0 )
+ b_multicast = VLC_FALSE;
+ else
+ if( sscanf( opt, "client_port=%u-%u", &loport, &hiport ) == 2 )
+ ;
+ else
+ if( strncmp( opt, "mode=", 5 ) == 0 )
+ {
+ if( strncasecmp( opt + 5, "\"PLAY\"", 6 ) )
+ {
+ /* Not playing?! */
+ b_unsupp = VLC_TRUE;
+ break;
+ }
+ }
+ else
+ {
+ /*
+ * Every other option is unsupported:
+ *
+ * "source" and "append" are invalid.
+ *
+ * For multicast, "port", "layers", "ttl" are set by the
+ * stream output configuration.
+ *
+ * For unicast, we do not allow "destination" as it
+ * carries a DoS risk, and we decide on "server_port".
+ *
+ * "interleaved" and "ssrc" are not implemented.
+ */
+ b_unsupp = VLC_TRUE;
+ break;
+ }
}
- fprintf( stderr, "HTTPD_MSG_SETUP: unicast ip=%s port=%d\n",
- ip, i_port );
+ if( b_unsupp )
+ continue;
- psz_session = httpd_MsgGet( query, "Session" );
- if( *psz_session == 0 )
+ if( b_multicast )
{
- psz_session = malloc( 100 );
- sprintf( psz_session, "%d", rand() );
+ if( id->psz_destination == NULL )
+ continue;
+
+ answer->i_status = 200;
- rtsp = RtspClientNew( p_stream, psz_session );
+ httpd_MsgAdd( answer, "Transport",
+ "RTP/AVP/UDP;destination=%s;port=%d-%d;ttl=%d",
+ id->psz_destination, id->i_port, id->i_port+1,
+ ( p_sys->i_ttl > 0 ) ? p_sys->i_ttl : 1 );
}
else
{
- rtsp = RtspClientGet( p_stream, psz_session );
- if( rtsp == NULL )
+ char ip[NI_MAXNUMERICHOST], psz_access[22],
+ url[NI_MAXNUMERICHOST + 8];
+ sout_access_out_t *p_access;
+ rtsp_client_t *rtsp = NULL;
+
+ if( ( hiport - loport ) > 1 )
+ continue;
+
+ if( psz_session == NULL )
+ {
+ psz_session = psz_session_init;
+ rtsp = RtspClientNew( p_stream, psz_session );
+ }
+ else
+ {
+ /* FIXME: we probably need to remove an access out,
+ * if there is already one for the same ID */
+ rtsp = RtspClientGet( p_stream, psz_session );
+ if( rtsp == NULL )
+ {
+ answer->i_status = 454;
+ continue;
+ }
+ }
+
+ if( httpd_ClientIP( cl, ip ) == NULL )
{
- /* FIXME right error code */
- answer->i_status = 400;
- answer->psz_status = strdup( "Unknown session id" );
- answer->i_body = 0;
- answer->p_body = NULL;
- free( ip );
+ answer->i_status = 500;
+ continue;
+ }
+
+ if( p_sys->i_ttl )
+ snprintf( psz_access, sizeof( psz_access ),
+ "udp{raw,rtcp,ttl=%d}", p_sys->i_ttl );
+ else
+ strcpy( psz_access, "udp{raw,rtcp}" );
+
+ snprintf( url, sizeof( url ),
+ ( strchr( ip, ':' ) != NULL ) ? "[%s]:%d" : "%s:%d",
+ ip, loport );
+
+ p_access = sout_AccessOutNew( p_stream->p_sout,
+ psz_access, url );
+ if( p_access == NULL )
+ {
+ msg_Err( p_stream,
+ "cannot create access output for %s://%s",
+ psz_access, url );
+ answer->i_status = 500;
break;
}
- }
- /* first try to create the access out */
- if( p_sys->i_ttl > 0 )
- sprintf( psz_access, "udp{raw,ttl=%d}", p_sys->i_ttl );
- else
- sprintf( psz_access, "udp{raw}" );
- sprintf( psz_url, "%s:%d", ip, i_port );
- free( ip );
-
- if( ( p_access = sout_AccessOutNew( p_stream->p_sout, psz_access, psz_url ) ) == NULL )
- {
- msg_Err( p_stream, "cannot create the access out for %s://%s",
- psz_access, psz_url );
- answer->i_status = 400;
- answer->psz_status = strdup( "Server internal error" );
- answer->i_body = 0;
- answer->p_body = NULL;
- break;
- }
+ TAB_APPEND( rtsp->i_id, rtsp->id, id );
+ TAB_APPEND( rtsp->i_access, rtsp->access, p_access );
- TAB_APPEND( rtsp->i_id, rtsp->id, id );
- TAB_APPEND( rtsp->i_access, rtsp->access, p_access );
+ char *src = var_GetNonEmptyString (p_access, "src-addr");
+ int sport = var_GetInteger (p_access, "src-port");
- answer->i_status = 200;
- answer->psz_status = strdup( "OK" );
- answer->i_body = 0;
- answer->p_body = NULL;
+ httpd_ServerIP( cl, ip );
+ fprintf( stderr, "src = %s, ip = %s\n", src, ip );
- httpd_MsgAdd( answer, "Transport",
- "RTP/AVP/UDP;client_port=%d-%d", i_port, i_port + 1 );
- }
- else /* TODO strstr( psz_transport, "interleaved" ) ) */
- {
- answer->i_status = 400;
- answer->psz_status = strdup( "Bad Request" );
- answer->i_body = 0;
- answer->p_body = NULL;
+ if( ( src != NULL ) && strcmp( src, ip ) )
+ {
+ /* Specify source IP if it is different from the RTSP
+ * control connection server address */
+ httpd_MsgAdd( answer, "Transport",
+ "RTP/AVP/UDP;unicast;source=%s;"
+ "client_port=%u-%u;server_port=%u-%u",
+ src, loport, hiport, sport, sport + 1 );
+ }
+ else
+ {
+ httpd_MsgAdd( answer, "Transport",
+ "RTP/AVP/UDP;unicast;"
+ "client_port=%u-%u;server_port=%u-%u",
+ loport, hiport, sport, sport + 1 );
+ }
+
+ answer->i_status = 200;
+ free( src );
+ }
+ break;
}
break;
}
default:
+ answer->i_status = 460;
+ break;
+
return VLC_EGENERIC;
}
- httpd_MsgAdd( answer, "Server", "VLC Server" );
+
+ httpd_MsgAdd( answer, "Server", PACKAGE_STRING );
httpd_MsgAdd( answer, "Content-Length", "%d", answer->i_body );
- httpd_MsgAdd( answer, "Cseq", "%d", atoi( httpd_MsgGet( query, "Cseq" ) ) );
+ psz_cseq = httpd_MsgGet( query, "Cseq" );
+ httpd_MsgAdd( answer, "Cseq", "%s", psz_cseq ? psz_cseq : "0");
httpd_MsgAdd( answer, "Cache-Control", "%s", "no-cache" );
if( psz_session )
- {
httpd_MsgAdd( answer, "Session", "%s"/*;timeout=5*/, psz_session );
- }
return VLC_SUCCESS;
}
return VLC_SUCCESS;
}
+
static int rtp_packetize_ac3( sout_stream_t *p_stream, sout_stream_id_t *id,
block_t *in )
{
return VLC_SUCCESS;
}
+/* rfc3016 */
+static int rtp_packetize_mp4a_latm( sout_stream_t *p_stream, sout_stream_id_t *id,
+ block_t *in )
+{
+ int i_max = id->i_mtu - 14; /* payload max in one packet */
+ int latmhdrsize = in->i_buffer / 0xff + 1;
+ int i_count = ( in->i_buffer + i_max - 1 ) / i_max;
+
+ uint8_t *p_data = in->p_buffer, *p_header = NULL;
+ int i_data = in->i_buffer;
+ int i;
+
+ for( i = 0; i < i_count; i++ )
+ {
+ int i_payload = __MIN( i_max, i_data );
+ block_t *out;
+
+ if( i != 0 )
+ latmhdrsize = 0;
+ out = block_New( p_stream, 12 + latmhdrsize + i_payload );
+
+ /* rtp common header */
+ rtp_packetize_common( id, out, ((i == i_count - 1) ? 1 : 0),
+ (in->i_pts > 0 ? in->i_pts : in->i_dts) );
+
+ if( i == 0 )
+ {
+ int tmp = in->i_buffer;
+
+ p_header=out->p_buffer+12;
+ while( tmp > 0xfe )
+ {
+ *p_header = 0xff;
+ p_header++;
+ tmp -= 0xff;
+ }
+ *p_header = tmp;
+ }
+
+ memcpy( &out->p_buffer[12+latmhdrsize], p_data, i_payload );
+
+ out->i_buffer = 12 + latmhdrsize + i_payload;
+ out->i_dts = in->i_dts + i * in->i_length / i_count;
+ out->i_length = in->i_length / i_count;
+
+ rtp_packetize_send( id, out );
+
+ p_data += i_payload;
+ i_data -= i_payload;
+ }
+
+ return VLC_SUCCESS;
+}
+
static int rtp_packetize_l16( sout_stream_t *p_stream, sout_stream_id_t *id,
block_t *in )
{
return VLC_SUCCESS;
}
+/* rfc3984 */
+static int rtp_packetize_h264_nal( sout_stream_t *p_stream, sout_stream_id_t *id,
+ const uint8_t *p_data, int i_data, int64_t i_pts, int64_t i_dts, vlc_bool_t b_last, int64_t i_length )
+{
+ const int i_max = id->i_mtu - 12; /* payload max in one packet */
+ int i_nal_hdr;
+ int i_nal_type;
+
+ if( i_data < 5 )
+ return VLC_SUCCESS;
+
+ i_nal_hdr = p_data[3];
+ i_nal_type = i_nal_hdr&0x1f;
+ if( i_nal_type == 7 || i_nal_type == 8 )
+ {
+ /* XXX Why do you want to remove them ? It will break streaming with
+ * SPS/PPS change (broadcast) ? */
+ return VLC_SUCCESS;
+ }
+
+ /* Skip start code */
+ p_data += 3;
+ i_data -= 3;
+
+ /* */
+ if( i_data <= i_max )
+ {
+ /* Single NAL unit packet */
+ block_t *out = block_New( p_stream, 12 + i_data );
+ out->i_dts = i_dts;
+ out->i_length = i_length;
+
+ /* */
+ rtp_packetize_common( id, out, b_last, i_pts );
+ out->i_buffer = 12 + i_data;
+
+ memcpy( &out->p_buffer[12], p_data, i_data );
+
+ rtp_packetize_send( id, out );
+ }
+ else
+ {
+ /* FU-A Fragmentation Unit without interleaving */
+ const int i_count = ( i_data-1 + i_max-2 - 1 ) / (i_max-2);
+ int i;
+
+ p_data++;
+ i_data--;
+
+ for( i = 0; i < i_count; i++ )
+ {
+ const int i_payload = __MIN( i_data, i_max-2 );
+ block_t *out = block_New( p_stream, 12 + 2 + i_payload );
+ out->i_dts = i_dts + i * i_length / i_count;
+ out->i_length = i_length / i_count;
+
+ /* */
+ rtp_packetize_common( id, out, (b_last && i_payload == i_data), i_pts );
+ out->i_buffer = 14 + i_payload;
+
+ /* FU indicator */
+ out->p_buffer[12] = 0x00 | (i_nal_hdr & 0x60) | 28;
+ /* FU header */
+ out->p_buffer[13] = ( i == 0 ? 0x80 : 0x00 ) | ( (i == i_count-1) ? 0x40 : 0x00 ) | i_nal_type;
+ memcpy( &out->p_buffer[14], p_data, i_payload );
+
+ rtp_packetize_send( id, out );
+
+ i_data -= i_payload;
+ p_data += i_payload;
+ }
+ }
+ return VLC_SUCCESS;
+}
+
+static int rtp_packetize_h264( sout_stream_t *p_stream, sout_stream_id_t *id,
+ block_t *in )
+{
+ const uint8_t *p_buffer = in->p_buffer;
+ int i_buffer = in->i_buffer;
+
+ while( i_buffer > 4 && ( p_buffer[0] != 0 || p_buffer[1] != 0 || p_buffer[2] != 1 ) )
+ {
+ i_buffer--;
+ p_buffer++;
+ }
+
+ /* Split nal units */
+ while( i_buffer > 4 )
+ {
+ int i_offset;
+ int i_size = i_buffer;
+ int i_skip = i_buffer;
+
+ /* search nal end */
+ for( i_offset = 4; i_offset+2 < i_buffer ; i_offset++)
+ {
+ if( p_buffer[i_offset] == 0 && p_buffer[i_offset+1] == 0 && p_buffer[i_offset+2] == 1 )
+ {
+ /* we found another startcode */
+ 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,
+ (in->i_pts > 0 ? in->i_pts : in->i_dts), in->i_dts,
+ (i_size >= i_buffer), in->i_length * i_size / in->i_buffer );
+
+ i_buffer -= i_skip;
+ p_buffer += i_skip;
+ }
+ return VLC_SUCCESS;
+}
+
static int rtp_packetize_amr( sout_stream_t *p_stream, sout_stream_id_t *id,
block_t *in )
{
(in->i_pts > 0 ? in->i_pts : in->i_dts) );
/* Payload header */
out->p_buffer[12] = 0xF0; /* CMR */
- out->p_buffer[13] = 0x00; /* ToC */ /* FIXME: frame type */
+ out->p_buffer[13] = p_data[0]&0x7C; /* ToC */ /* FIXME: frame type */
/* FIXME: are we fed multiple frames ? */
- memcpy( &out->p_buffer[14], p_data, i_payload );
+ memcpy( &out->p_buffer[14], p_data+1, i_payload-1 );
- out->i_buffer = 14 + i_payload;
+ out->i_buffer = 14 + i_payload-1;
out->i_dts = in->i_dts + i * in->i_length / i_count;
out->i_length = in->i_length / i_count;