/*****************************************************************************
* rtp.c: rtp stream output module
*****************************************************************************
- * Copyright (C) 2003-2004 the VideoLAN team
+ * Copyright (C) 2003-2004, 2010 the VideoLAN team
* Copyright © 2007-2008 Rémi Denis-Courmont
*
* Authors: Laurent Aimar <fenrir@via.ecp.fr>
+ * Pierre Ynard
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
#include <vlc_httpd.h>
#include <vlc_url.h>
#include <vlc_network.h>
-#include <vlc_charset.h>
-#include <vlc_strings.h>
+#include <vlc_fs.h>
#include <vlc_rand.h>
-#include <srtp.h>
+#ifdef HAVE_SRTP
+# include <srtp.h>
+# include <gcrypt.h>
+# include <vlc_gcrypt.h>
+#endif
#include "rtp.h"
#ifdef HAVE_UNISTD_H
# include <sys/types.h>
# include <unistd.h>
-# include <fcntl.h>
-# include <sys/stat.h>
+#endif
+#ifdef HAVE_ARPA_INET_H
+# include <arpa/inet.h>
#endif
#ifdef HAVE_LINUX_DCCP_H
# include <linux/dccp.h>
#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 " \
+ "session will be made available. You must use a url: http://location to " \
"access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
"for the SDP to be announced via SAP." )
#define SAP_TEXT N_("SAP announcing")
"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 " \
+ "This allows you to give a URL with more details about the stream " \
"(often the website of the streaming organization), that will " \
"be announced in the SDP (Session Descriptor)." )
#define EMAIL_TEXT N_("Session email")
"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.")
+ "with this Secure RTP master shared secret key. "\
+ "This must be a 32-character-long hexadecimal string.")
#define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)")
#define SRTP_SALT_LONGTEXT N_( \
- "Secure RTP requires a (non-secret) master salt value.")
+ "Secure RTP requires a (non-secret) master salt value. " \
+ "This must be a 28-character-long hexadecimal string.")
static const char *const ppsz_protos[] = {
"dccp", "sctp", "tcp", "udp", "udplite",
#define RFC3016_LONGTEXT N_( \
"This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
+#define RTSP_HOST_TEXT N_( "RTSP host address" )
+#define RTSP_HOST_LONGTEXT N_( \
+ "This defines the address, port and path the RTSP VOD server will listen " \
+ "on.\nSyntax is address:port/path. The default is to listen on all "\
+ "interfaces (address 0.0.0.0), on port 554, with no path.\nTo listen " \
+ "only on the local interface, use \"localhost\" as address." )
+
+#define RTSP_TIMEOUT_TEXT N_( "RTSP session timeout (s)" )
+#define RTSP_TIMEOUT_LONGTEXT N_( "RTSP sessions will be closed after " \
+ "not receiving any RTSP request for this long. Setting it to a " \
+ "negative value or zero disables timeouts. The default is 60 (one " \
+ "minute)." )
+
+#define RTSP_USER_TEXT N_("Username")
+#define RTSP_USER_LONGTEXT N_("User name that will be " \
+ "requested to access the stream." )
+#define RTSP_PASS_TEXT N_("Password")
+#define RTSP_PASS_LONGTEXT N_("Password that will be " \
+ "requested to access the stream." )
+
static int Open ( vlc_object_t * );
static void Close( vlc_object_t * );
set_shortname( N_("RTP"))
set_description( N_("RTP stream output") )
set_capability( "sout stream", 0 )
- add_shortcut( "rtp" )
+ add_shortcut( "rtp", "vod" )
set_category( CAT_SOUT )
set_subcategory( SUBCAT_SOUT_STREAM )
- add_string( SOUT_CFG_PREFIX "dst", "", NULL, DEST_TEXT,
+ add_string( SOUT_CFG_PREFIX "dst", "", DEST_TEXT,
DEST_LONGTEXT, true )
- add_string( SOUT_CFG_PREFIX "sdp", "", NULL, SDP_TEXT,
+ add_string( SOUT_CFG_PREFIX "sdp", "", SDP_TEXT,
SDP_LONGTEXT, true )
- add_string( SOUT_CFG_PREFIX "mux", "", NULL, MUX_TEXT,
+ add_string( SOUT_CFG_PREFIX "mux", "", MUX_TEXT,
MUX_LONGTEXT, true )
- add_bool( SOUT_CFG_PREFIX "sap", false, NULL, SAP_TEXT, SAP_LONGTEXT,
+ add_bool( SOUT_CFG_PREFIX "sap", false, SAP_TEXT, SAP_LONGTEXT,
true )
- add_string( SOUT_CFG_PREFIX "name", "", NULL, NAME_TEXT,
+ add_string( SOUT_CFG_PREFIX "name", "", NAME_TEXT,
NAME_LONGTEXT, true )
- add_string( SOUT_CFG_PREFIX "description", "", NULL, DESC_TEXT,
+ add_string( SOUT_CFG_PREFIX "description", "", DESC_TEXT,
DESC_LONGTEXT, true )
- add_string( SOUT_CFG_PREFIX "url", "", NULL, URL_TEXT,
+ add_string( SOUT_CFG_PREFIX "url", "", URL_TEXT,
URL_LONGTEXT, true )
- add_string( SOUT_CFG_PREFIX "email", "", NULL, EMAIL_TEXT,
+ add_string( SOUT_CFG_PREFIX "email", "", EMAIL_TEXT,
EMAIL_LONGTEXT, true )
- add_string( SOUT_CFG_PREFIX "phone", "", NULL, PHONE_TEXT,
+ add_string( SOUT_CFG_PREFIX "phone", "", PHONE_TEXT,
PHONE_LONGTEXT, true )
- add_string( SOUT_CFG_PREFIX "proto", "udp", NULL, PROTO_TEXT,
+ add_string( SOUT_CFG_PREFIX "proto", "udp", PROTO_TEXT,
PROTO_LONGTEXT, false )
change_string_list( ppsz_protos, ppsz_protocols, NULL )
- add_integer( SOUT_CFG_PREFIX "port", 5004, NULL, PORT_TEXT,
+ add_integer( SOUT_CFG_PREFIX "port", 5004, PORT_TEXT,
PORT_LONGTEXT, true )
- add_integer( SOUT_CFG_PREFIX "port-audio", 0, NULL, PORT_AUDIO_TEXT,
+ add_integer( SOUT_CFG_PREFIX "port-audio", 0, PORT_AUDIO_TEXT,
PORT_AUDIO_LONGTEXT, true )
- add_integer( SOUT_CFG_PREFIX "port-video", 0, NULL, PORT_VIDEO_TEXT,
+ add_integer( SOUT_CFG_PREFIX "port-video", 0, PORT_VIDEO_TEXT,
PORT_VIDEO_LONGTEXT, true )
- add_integer( SOUT_CFG_PREFIX "ttl", -1, NULL, TTL_TEXT,
+ add_integer( SOUT_CFG_PREFIX "ttl", -1, TTL_TEXT,
TTL_LONGTEXT, true )
- add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL,
+ add_bool( SOUT_CFG_PREFIX "rtcp-mux", false,
RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
+ add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000,
+ CACHING_TEXT, CACHING_LONGTEXT, true )
- add_string( SOUT_CFG_PREFIX "key", "", NULL,
+#ifdef HAVE_SRTP
+ add_string( SOUT_CFG_PREFIX "key", "",
SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false )
- add_string( SOUT_CFG_PREFIX "salt", "", NULL,
+ add_string( SOUT_CFG_PREFIX "salt", "",
SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false )
+#endif
- add_bool( SOUT_CFG_PREFIX "mp4a-latm", 0, NULL, RFC3016_TEXT,
+ add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, RFC3016_TEXT,
RFC3016_LONGTEXT, false )
set_callbacks( Open, Close )
+
+ add_submodule ()
+ set_shortname( N_("RTSP VoD" ) )
+ set_description( N_("RTSP VoD server") )
+ set_category( CAT_SOUT )
+ set_subcategory( SUBCAT_SOUT_VOD )
+ set_capability( "vod server", 10 )
+ set_callbacks( OpenVoD, CloseVoD )
+ add_shortcut( "rtsp" )
+ add_string ( "rtsp-host", NULL, RTSP_HOST_TEXT,
+ RTSP_HOST_LONGTEXT, true )
+ add_integer( "rtsp-timeout", 60, RTSP_TIMEOUT_TEXT,
+ RTSP_TIMEOUT_LONGTEXT, true )
+ add_string( "sout-rtsp-user", "",
+ RTSP_USER_TEXT, RTSP_USER_LONGTEXT, true )
+ add_password( "sout-rtsp-pwd", "",
+ RTSP_PASS_TEXT, RTSP_PASS_LONGTEXT, true )
+
vlc_module_end ()
/*****************************************************************************
static const char *const ppsz_sout_options[] = {
"dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
"sap", "description", "url", "email", "phone",
- "proto", "rtcp-mux", "key", "salt",
+ "proto", "rtcp-mux", "caching", "key", "salt",
"mp4a-latm", NULL
};
block_t* );
static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
-static void* ThreadSend( vlc_object_t *p_this );
+static void* ThreadSend( void * );
+static void *rtp_listen_thread( void * );
static void SDPHandleUrl( sout_stream_t *, const char * );
static int FileSetup( sout_stream_t *p_stream );
static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * );
+static int64_t rtp_init_ts( const vod_media_t *p_media,
+ const char *psz_vod_session );
+
struct sout_stream_sys_t
{
/* SDP */
vlc_mutex_t lock_sdp;
/* SDP to disk */
- bool b_export_sdp_file;
char *psz_sdp_file;
/* SDP via SAP */
/* 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;
+ /* VoD */
+ vod_media_t *p_vod_media;
+ char *psz_vod_session;
+
/* in case we do TS/PS over rtp */
sout_mux_t *p_mux;
sout_access_out_t *p_grab;
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;
struct sout_stream_id_t
{
- VLC_COMMON_MEMBERS
-
sout_stream_t *p_stream;
/* rtp field */
uint16_t i_sequence;
- uint8_t i_payload_type;
+ bool b_first_packet;
+ bool b_ts_init;
+ uint32_t i_ts_offset;
uint8_t ssrc[4];
+ /* for rtsp */
+ uint16_t i_seq_sent_next;
+
/* for sdp */
- const char *psz_enc;
- char *psz_fmtp;
- int i_clock_rate;
+ rtp_format_t rtp_fmt;
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;
- pf_rtp_packetizer_t pf_packetize;
+#endif
/* Packets sinks */
+ vlc_thread_t thread;
vlc_mutex_t lock_sink;
int sinkc;
rtp_sink_t *sinkv;
rtsp_stream_id_t *rtsp_id;
- int *listen_fd;
+ struct {
+ int *fd;
+ vlc_thread_t thread;
+ } listen;
block_fifo_t *p_fifo;
int64_t i_caching;
p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" );
p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" );
- p_sys->psz_sdp_file = NULL;
-
if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
{
msg_Err( p_stream, "audio and video RTP port must be distinct" );
free (psz);
var_Create (p_this, "dccp-service", VLC_VAR_STRING);
- if( ( p_sys->psz_destination == NULL ) && !b_rtsp )
+ p_sys->p_vod_media = NULL;
+ p_sys->psz_vod_session = NULL;
+
+ if (! strcmp(p_stream->psz_name, "vod"))
+ {
+ /* The VLM stops all instances before deleting a media, so this
+ * reference will remain valid during the lifetime of the rtp
+ * stream output. */
+ p_sys->p_vod_media = var_InheritAddress(p_stream, "vod-media");
+
+ if (p_sys->p_vod_media != NULL)
+ {
+ p_sys->psz_vod_session = var_InheritString(p_stream, "vod-session");
+ if (p_sys->psz_vod_session == NULL)
+ {
+ msg_Err(p_stream, "missing VoD session");
+ free(p_sys);
+ return VLC_EGENERIC;
+ }
+
+ const char *mux = vod_get_mux(p_sys->p_vod_media);
+ var_SetString(p_stream, SOUT_CFG_PREFIX "mux", mux);
+ }
+ }
+
+ if( p_sys->psz_destination == NULL && !b_rtsp
+ && p_sys->p_vod_media == NULL )
{
msg_Err( p_stream, "missing destination and not in RTSP mode" );
free( p_sys );
return VLC_EGENERIC;
}
- p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
- if( p_sys->i_ttl == -1 )
+ int i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
+ if( i_ttl != -1 )
{
- /* 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" );
+ var_Create( p_stream, "ttl", VLC_VAR_INTEGER );
+ var_SetInteger( p_stream, "ttl", i_ttl );
}
p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
- p_sys->payload_bitmap = 0;
+ /* NPT=0 time will be determined when we packetize the first packet
+ * (of any ES). But we want to be able to report rtptime in RTSP
+ * without waiting (and already did in the VoD case). So until then,
+ * we use an arbitrary reference PTS for timestamp computations, and
+ * then actual PTS will catch up using offsets. */
+ p_sys->i_npt_zero = VLC_TS_INVALID;
+ p_sys->i_pts_zero = rtp_init_ts(p_sys->p_vod_media,
+ p_sys->psz_vod_session);
p_sys->i_es = 0;
p_sys->es = NULL;
p_sys->rtsp = NULL;
p_sys->psz_sdp = NULL;
p_sys->b_export_sap = false;
- p_sys->b_export_sdp_file = 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 )
msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" );
free( psz );
vlc_mutex_destroy( &p_sys->lock_sdp );
+ vlc_mutex_destroy( &p_sys->lock_ts );
vlc_mutex_destroy( &p_sys->lock_es );
+ free( p_sys->psz_vod_session );
free( p_sys->psz_destination );
free( p_sys );
return VLC_EGENERIC;
msg_Err( p_stream, "cannot create muxer" );
sout_AccessOutDelete( p_sys->p_grab );
vlc_mutex_destroy( &p_sys->lock_sdp );
+ vlc_mutex_destroy( &p_sys->lock_ts );
vlc_mutex_destroy( &p_sys->lock_es );
- free( p_sys->psz_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_vod_session );
free( p_sys->psz_destination );
free( p_sys );
return VLC_EGENERIC;
/* update p_sout->i_out_pace_nocontrol */
p_stream->p_sout->i_out_pace_nocontrol++;
+ if( p_sys->p_mux != NULL )
+ {
+ sout_stream_id_t *id = Add( p_stream, NULL );
+ if( id == NULL )
+ {
+ Close( p_this );
+ return VLC_EGENERIC;
+ }
+ }
+
return VLC_SUCCESS;
}
if( p_sys->p_mux )
{
- assert( p_sys->i_es == 1 );
- Del( p_stream, p_sys->es[0] );
+ assert( p_sys->i_es <= 1 );
sout_MuxDelete( p_sys->p_mux );
+ if ( p_sys->i_es > 0 )
+ Del( p_stream, p_sys->es[0] );
sout_AccessOutDelete( p_sys->p_grab );
+
if( p_sys->packet )
{
block_Release( p_sys->packet );
}
- if( p_sys->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 )
free( p_sys->psz_sdp );
- if( p_sys->b_export_sdp_file )
+ 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_vod_session );
free( p_sys->psz_destination );
free( p_sys );
}
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 )
+ if( url.psz_host != NULL && *url.psz_host )
{
- sout_stream_id_t *id = p_sys->es[0];
- id->rtsp_id = RtspAddId( p_sys->rtsp, id, 0, GetDWBE( id->ssrc ),
- p_sys->psz_destination, p_sys->i_ttl,
- id->i_port, id->i_port + 1 );
+ /* msg_Err( p_stream, "\"%s\" RTSP host ignored", url.psz_host );
+ msg_Info( p_stream, "Pass --rtsp-host=%s on the command line "
+ "instead.", url.psz_host ); */
+
+ var_Create( p_stream, "rtsp-host", VLC_VAR_STRING );
+ var_SetString( p_stream, "rtsp-host", url.psz_host );
}
+ /* if( url.i_port != 0 )
+ {
+ msg_Err( p_stream, "\"%u\" RTSP port ignored", url.i_port );
+ msg_Info( p_stream, "Pass --rtsp-port=%u on the command line "
+ "instead.", url.i_port );
+ } */
+
+ if( url.i_port <= 0 ) url.i_port = 554;
+ var_Create( p_stream, "rtsp-port", VLC_VAR_INTEGER );
+ var_SetInteger( p_stream, "rtsp-port", url.i_port );
+
+ p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), NULL, url.psz_path );
+ if( p_sys->rtsp == NULL )
+ msg_Err( p_stream, "cannot export SDP as RTSP" );
}
else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) ||
( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) )
}
else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) )
{
- if( p_sys->b_export_sdp_file )
+ if( p_sys->psz_sdp_file != NULL )
{
msg_Err( p_stream, "you can use sdp=file:// only once" );
goto out;
}
- p_sys->b_export_sdp_file = true;
- psz_url = &psz_url[5];
- if( psz_url[0] == '/' && psz_url[1] == '/' )
- psz_url += 2;
- p_sys->psz_sdp_file = strdup( psz_url );
+ p_sys->psz_sdp_file = make_path( psz_url );
+ if( p_sys->psz_sdp_file == NULL )
+ goto out;
+ FileSetup( p_stream );
}
else
{
* SDPGenerate
*****************************************************************************/
/*static*/
-char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url )
+char *SDPGenerate( sout_stream_t *p_stream, const char *rtsp_url )
{
- const sout_stream_sys_t *p_sys = p_stream->p_sys;
- char *psz_sdp;
+ sout_stream_sys_t *p_sys = p_stream->p_sys;
+ char *psz_sdp = NULL;
struct sockaddr_storage dst;
socklen_t dstlen;
int i;
*/
int inclport;
+ vlc_mutex_lock( &p_sys->lock_es );
+ if( unlikely(p_sys->i_es == 0 || (rtsp_url != NULL && !p_sys->es[0]->rtsp_id)) )
+ goto out; /* hmm... */
+
if( p_sys->psz_destination != NULL )
{
inclport = 1;
- /* Oh boy, this is really ugly! (+ race condition on lock_es) */
+ /* 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],
+ 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,
{
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 */
- memset (&dst, 0, sizeof( struct sockaddr_in ) );
- dst.ss_family = AF_INET;
+ 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 =
+ dst.ss_len = dstlen;
#endif
- dstlen = sizeof( struct sockaddr_in );
}
psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX,
NULL, 0, (struct sockaddr *)&dst, dstlen );
if( psz_sdp == NULL )
- return NULL;
+ goto out;
/* TODO: a=source-filter */
if( p_sys->rtcp_mux )
if( rtsp_url != NULL )
sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url );
- /* FIXME: locking?! */
+ 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];
+ rtp_format_t *rtp_fmt = &id->rtp_fmt;
const char *mime_major; /* major MIME type */
- const char *proto = "RTP/AVP"; /* protocol */
- switch( id->i_cat )
+ switch( rtp_fmt->cat )
{
case VIDEO_ES:
mime_major = "video";
continue;
}
- if( rtsp_url == NULL )
- {
- switch( p_sys->proto )
- {
- case IPPROTO_UDP:
- break;
- case IPPROTO_TCP:
- proto = "TCP/RTP/AVP";
- break;
- case IPPROTO_DCCP:
- proto = "DCCP/RTP/AVP";
- break;
- case IPPROTO_UDPLITE:
- continue;
- }
- }
-
sdp_AddMedia( &psz_sdp, mime_major, proto, inclport * id->i_port,
- id->i_payload_type, false, id->i_bitrate,
- id->psz_enc, id->i_clock_rate, id->i_channels,
- id->psz_fmtp);
+ rtp_fmt->payload_type, false, rtp_fmt->bitrate,
+ rtp_fmt->ptname, rtp_fmt->clock_rate, rtp_fmt->channels,
+ rtp_fmt->fmtp);
- if( !p_sys->rtcp_mux && (id->i_port & 1) ) /* cf RFC4566 §5.14 */
+ /* cf RFC4566 §5.14 */
+ if( inclport && !p_sys->rtcp_mux && (id->i_port & 1) )
sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 );
if( rtsp_url != NULL )
{
- assert( strlen( rtsp_url ) > 0 );
- bool addslash = ( rtsp_url[strlen( rtsp_url ) - 1] != '/' );
- sdp_AddAttribute ( &psz_sdp, "control",
- addslash ? "%s/trackID=%u" : "%strackID=%u",
- rtsp_url, i );
+ 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 )
+ 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] ) );
+ "SC:RTP%c",
+ toupper( (unsigned char)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).
*/
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;
+ size_t spl = (id->rtp_fmt.clock_rate - 1) * ptime_ms / 1000 + 1;
+ bytes *= id->rtp_fmt.channels;
spl *= bytes;
if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */
id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
}
+uint32_t rtp_compute_ts( unsigned i_clock_rate, int64_t i_pts )
+{
+ /* NOTE: this plays nice with offsets because the calculations are
+ * linear. */
+ return i_pts * (int64_t)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;
- sout_stream_id_t *id;
- int i_port, cscov = -1;
char *psz_sdp;
- int i_port_audio_option = var_GetInteger( p_stream, "port-audio" );
- int i_port_video_option = var_GetInteger( p_stream, "port-video" );
-
- if (0xffffffff == p_sys->payload_bitmap)
- {
- msg_Err (p_stream, "too many RTP elementary streams");
- return NULL;
- }
- id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) );
- if( id == NULL )
+ sout_stream_id_t *id = malloc( sizeof( *id ) );
+ if( unlikely(id == NULL) )
return NULL;
- vlc_object_attach( id, p_stream );
-
- /* Choose the port */
- i_port = 0;
- if( p_fmt == NULL )
- ;
- else
- if( p_fmt->i_cat == AUDIO_ES && p_sys->i_port_audio > 0 )
- {
- i_port = p_sys->i_port_audio;
- p_sys->i_port_audio = 0;
- }
- else
- if( p_fmt->i_cat == VIDEO_ES && p_sys->i_port_video > 0 )
- {
- i_port = p_sys->i_port_video;
- p_sys->i_port_video = 0;
- }
+ id->p_stream = p_stream;
- while( i_port == 0 )
- {
- if( p_sys->i_port != i_port_audio_option
- && p_sys->i_port != i_port_video_option )
- {
- i_port = p_sys->i_port;
- }
- p_sys->i_port += 2;
- }
+ 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->p_stream = p_stream;
+#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;
- /* Look for free dymanic payload type */
- id->i_payload_type = 96;
- while (p_sys->payload_bitmap & (1 << (id->i_payload_type - 96)))
- id->i_payload_type++;
- assert (id->i_payload_type < 128);
+ id->b_first_packet = true;
+ id->i_caching =
+ (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence));
vlc_rand_bytes (id->ssrc, sizeof (id->ssrc));
- 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 )
+ bool format = false;
+
+ if (p_sys->p_vod_media != NULL)
{
- id->i_cat = p_fmt->i_cat;
- if( p_fmt->i_cat == AUDIO_ES )
+ id->rtp_fmt.ptname = NULL;
+ uint32_t ssrc;
+ int val = vod_init_id(p_sys->p_vod_media, p_sys->psz_vod_session,
+ p_fmt ? p_fmt->i_id : 0, id, &id->rtp_fmt,
+ &ssrc, &id->i_seq_sent_next);
+ if (val == VLC_SUCCESS)
{
- id->i_clock_rate = p_fmt->audio.i_rate;
- id->i_channels = p_fmt->audio.i_channels;
+ memcpy(id->ssrc, &ssrc, sizeof(id->ssrc));
+ /* This is ugly, but id->i_seq_sent_next needs to be
+ * initialized inside vod_init_id() to avoid race
+ * conditions. */
+ id->i_sequence = id->i_seq_sent_next;
}
- id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
+ /* vod_init_id() may fail either because the ES wasn't found in
+ * the VoD media, or because the RTSP session is gone. In the
+ * former case, id->rtp_fmt was left untouched. */
+ format = (id->rtp_fmt.ptname != NULL);
}
- else
+
+ if (!format)
{
- id->i_cat = VIDEO_ES;
- id->i_bitrate = 0;
+ id->rtp_fmt.fmtp = NULL; /* don't free() garbage on error */
+ char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
+ if (p_fmt == NULL && psz == NULL)
+ goto error;
+ int val = rtp_get_fmt(VLC_OBJECT(p_stream), p_fmt, psz, &id->rtp_fmt);
+ free( psz );
+ if (val != VLC_SUCCESS)
+ goto error;
}
- id->i_mtu = config_GetInt( p_stream, "mtu" );
- if( id->i_mtu <= 12 + 16 )
- id->i_mtu = 576 - 20 - 8; /* pessimistic */
- msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
-
- id->srtp = NULL;
- id->pf_packetize = NULL;
-
- char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
+#ifdef HAVE_SRTP
+ char *key = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
if (key)
{
+ vlc_gcrypt_init ();
id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
if (id->srtp == NULL)
goto error;
}
- char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
+ char *salt = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
free (salt);
free (key);
}
id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */
}
+#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");
+ id->i_seq_sent_next = id->i_sequence;
+ int mcast_fd = -1;
if( p_sys->psz_destination != 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");
+ goto error;
+ }
+ 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;
+ }
+
+ id->i_port = i_port;
+
+ int type = SOCK_STREAM;
+
switch( p_sys->proto )
{
+#ifdef SOCK_DCCP
case IPPROTO_DCCP:
{
const char *code;
- switch (id->i_cat)
+ switch (id->rtp_fmt.cat)
{
case VIDEO_ES: code = "RTPV"; break;
case AUDIO_ES: code = "RTPARTPV"; break;
default: code = "RTPORTPV"; break;
}
var_SetString (p_stream, "dccp-service", code);
+ type = SOCK_DCCP;
} /* fall through */
+#endif
case IPPROTO_TCP:
- id->listen_fd = net_Listen( VLC_OBJECT(p_stream),
+ id->listen.fd = net_Listen( VLC_OBJECT(p_stream),
p_sys->psz_destination, i_port,
- p_sys->proto );
- if( id->listen_fd == NULL )
+ type, 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 );
+ i_port, -1, 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 );
+ /* 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 );
+ /* FIXME: test if this is multicast */
+ mcast_fd = fd;
}
}
-
- 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
+
+ if( p_fmt != NULL )
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;
+ case VLC_CODEC_U8:
rtp_set_ptime (id, 20, 1);
break;
case VLC_CODEC_S16B:
- 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";
- id->pf_packetize = rtp_packetize_split;
+ case VLC_CODEC_S16L:
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;
- }
- 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:
- {
- char hexa[2*p_fmt->i_extra +1];
-
- id->psz_enc = "MP4V-ES";
- id->pf_packetize = rtp_packetize_split;
- if( p_fmt->i_extra > 0 )
- {
- 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;
+ break;
}
- if (id->i_payload_type >= 96)
- /* Mark dynamic payload type in use */
- p_sys->payload_bitmap |= 1 << (id->i_payload_type - 96);
#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->rtp_fmt.clock_rate,
+ p_sys->i_pts_offset );
+
if( p_sys->rtsp != NULL )
- id->rtsp_id = RtspAddId( p_sys->rtsp, id, p_sys->i_es,
- GetDWBE( id->ssrc ),
- p_sys->psz_destination,
- p_sys->i_ttl, id->i_port, id->i_port + 1 );
+ id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ),
+ id->rtp_fmt.clock_rate, mcast_fd );
id->p_fifo = block_FifoNew();
- if( vlc_thread_create( id, "RTP send thread", ThreadSend,
- VLC_THREAD_PRIORITY_HIGHEST ) )
+ 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 );
/* Update SDP (sap/file) */
if( p_sys->b_export_sap ) SapSetup( p_stream );
- if( p_sys->b_export_sdp_file ) FileSetup( p_stream );
+ if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
return id;
{
sout_stream_sys_t *p_sys = p_stream->p_sys;
- if( id->p_fifo != NULL )
- {
- vlc_object_kill( id );
- 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 );
vlc_mutex_unlock( &p_sys->lock_es );
- /* Release port */
- if( id->i_port == var_GetInteger( p_stream, "port-audio" ) )
- p_sys->i_port_audio = id->i_port;
- if( id->i_port == var_GetInteger( p_stream, "port-video" ) )
- p_sys->i_port_video = id->i_port;
- /* Release dynamic payload type */
- if (id->i_payload_type >= 96)
- p_sys->payload_bitmap &= ~(1 << (id->i_payload_type - 96));
+ if( likely(id->p_fifo != NULL) )
+ {
+ vlc_cancel( id->thread );
+ vlc_join( id->thread, NULL );
+ block_FifoRelease( id->p_fifo );
+ }
- free( id->psz_fmtp );
+ free( id->rtp_fmt.fmtp );
+ if (p_sys->p_vod_media != NULL)
+ vod_detach_id(p_sys->p_vod_media, p_sys->psz_vod_session, id);
if( id->rtsp_id )
RtspDelId( p_sys->rtsp, id->rtsp_id );
- 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 );
+ 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->b_export_sdp_file ) FileSetup( p_stream );
+ if( p_sys->b_export_sap ) SapSetup( p_stream );
+ if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream );
- vlc_object_detach( id );
- vlc_object_release( id );
+ free( id );
return VLC_SUCCESS;
}
while( p_buffer != NULL )
{
p_next = p_buffer->p_next;
- if( id->pf_packetize( id, p_buffer ) )
+
+ /* Send a Vorbis/Theora Packed Configuration packet (RFC 5215 §3.1)
+ * as the first packet of the stream */
+ if (id->b_first_packet)
+ {
+ id->b_first_packet = false;
+ if (!strcmp(id->rtp_fmt.ptname, "vorbis") ||
+ !strcmp(id->rtp_fmt.ptname, "theora"))
+ rtp_packetize_xiph_config(id, id->rtp_fmt.fmtp,
+ p_buffer->i_pts);
+ }
+
+ if( id->rtp_fmt.pf_packetize( id, p_buffer ) )
break;
block_Release( p_buffer );
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();
+ if( p_sys->i_es > 0 && p_sys->psz_sdp && *p_sys->psz_sdp )
p_sys->p_session = sout_AnnounceRegisterSDP( p_sout,
p_sys->psz_sdp,
- p_sys->psz_destination,
- p_method );
- sout_MethodRelease( p_method );
- }
+ p_sys->psz_destination );
return VLC_SUCCESS;
}
sout_stream_sys_t *p_sys = p_stream->p_sys;
FILE *f;
- if( ( f = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
+ 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 );
{
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 );
+ p_sys->p_httpd_host = vlc_http_HostNew( VLC_OBJECT(p_stream) );
if( p_sys->p_httpd_host )
{
p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
/****************************************************************************
* RTP send
****************************************************************************/
-static void* ThreadSend( vlc_object_t *p_this )
+static void* ThreadSend( void *data )
{
#ifdef WIN32
# define ECONNREFUSED WSAECONNREFUSED
# define EAGAIN WSAEWOULDBLOCK
# define EWOULDBLOCK WSAEWOULDBLOCK
#endif
- sout_stream_id_t *id = (sout_stream_id_t *)p_this;
+ 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;
if( val )
{
errno = val;
- msg_Dbg( id, "SRTP sending error: %m" );
+ msg_Dbg( id->p_stream, "SRTP sending error: %m" );
block_Release( out );
out = NULL;
}
else
out->i_buffer = len;
}
-
if (out)
mwait (out->i_dts + i_caching);
vlc_cleanup_pop ();
if (out == NULL)
continue;
+#else
+ mwait (out->i_dts + i_caching);
+ vlc_cleanup_pop ();
+#endif
ssize_t len = out->i_buffer;
int canc = vlc_savecancel ();
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 )
+ if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) == -1
+ && net_errno != EAGAIN && net_errno != EWOULDBLOCK
+ && net_errno != ENOBUFS && net_errno != ENOMEM )
{
- /* 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 */
+ int type;
+ getsockopt( id->sinkv[i].rtp_fd, SOL_SOCKET, SO_TYPE,
+ &type, &(socklen_t){ sizeof(type) });
+ if( type == SOCK_DGRAM )
+ /* ICMP soft error: ignore and retry */
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;
+ else
+ /* Broken connection */
+ deadv[deadc++] = id->sinkv[i].rtp_fd;
}
-
- 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, "removing socket %d", deadv[i] );
+ msg_Dbg( id->p_stream, "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, true );
- }
vlc_restorecancel (canc);
}
return NULL;
}
-int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux )
+
+/* 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, "RTCP failed!" );
+ 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;
}
net_Close( sink.rtp_fd );
}
-uint16_t rtp_get_seq( const sout_stream_id_t *id )
+uint16_t rtp_get_seq( 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;
+ /* 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;
}
-/* 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 )
+/* Return an arbitrary initial timestamp for RTP timestamp computations.
+ * RFC 3550 states that the resulting initial RTP timestamps SHOULD be
+ * random (although we use the same reference for all the ES as a
+ * feature). In the VoD case, this function is called independently
+ * from several parts of the code, so we need to always return the same
+ * value. */
+static int64_t rtp_init_ts( const vod_media_t *p_media,
+ const char *psz_vod_session )
{
- sout_stream_sys_t *p_sys = id->p_stream->p_sys;
- int i;
+ if (p_media == NULL || psz_vod_session == NULL)
+ return mdate();
+
+ uint64_t i_ts_init;
+ /* As per RFC 2326, session identifiers are at least 8 bytes long */
+ strncpy((char *)&i_ts_init, psz_vod_session, sizeof(uint64_t));
+ i_ts_init ^= (uintptr_t)p_media;
+ /* Limit the timestamp to 48 bytes, this is enough and allows us
+ * to stay away from overflows */
+ i_ts_init &= 0xFFFFFFFFFFFF;
+ return i_ts_init;
+}
- 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 a timestamp corresponding to packets being sent now, and that
+ * can be passed to rtp_compute_ts() to get rtptime values for each ES.
+ * Also return the NPT corresponding to this timestamp. If the stream
+ * output is not started, the initial timestamp that will be used with
+ * the first packets for NPT=0 is returned instead. */
+int64_t rtp_get_ts( const sout_stream_t *p_stream, const sout_stream_id_t *id,
+ const vod_media_t *p_media, const char *psz_vod_session,
+ int64_t *p_npt )
+{
+ if (p_npt != NULL)
+ *p_npt = 0;
- return i;
-}
+ if (id != NULL)
+ p_stream = id->p_stream;
+
+ if (p_stream == NULL)
+ return rtp_init_ts(p_media, psz_vod_session);
+
+ 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;
+
+ int64_t npt = now - i_npt_zero;
+ if (p_npt != NULL)
+ *p_npt = npt;
+
+ return p_sys->i_pts_zero + npt;
+}
void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
int b_marker, int64_t i_pts )
{
- uint32_t i_timestamp = i_pts * (int64_t)id->i_clock_rate / INT64_C(1000000);
+ 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->rtp_fmt.clock_rate,
+ p_sys->i_pts_offset );
+ id->b_ts_init = true;
+ }
+
+ uint32_t i_timestamp = rtp_compute_ts( id->rtp_fmt.clock_rate, 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[1] = (b_marker?0x80:0x00)|id->rtp_fmt.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;
{
sout_access_out_t *p_grab;
- p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
+ p_grab = vlc_object_create( p_stream, sizeof( *p_grab ) );
if( p_grab == NULL )
return NULL;
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;
}
projects / vlc / blobdiff