/***************************************************************************** * rtp.c: rtp stream output module ***************************************************************************** * Copyright (C) 2003-2004, 2010 the VideoLAN team * Copyright © 2007-2008 Rémi Denis-Courmont * * Authors: Laurent Aimar * Pierre Ynard * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA. *****************************************************************************/ /***************************************************************************** * Preamble *****************************************************************************/ #ifdef HAVE_CONFIG_H # include "config.h" #endif #define VLC_MODULE_LICENSE VLC_LICENSE_GPL_2_PLUS #include #include #include #include #include #include #include #include #include #include #ifdef HAVE_SRTP # include # include # include #endif #include "rtp.h" #include #include #ifdef HAVE_ARPA_INET_H # include #endif #ifdef HAVE_LINUX_DCCP_H # include #endif #ifndef IPPROTO_DCCP # define IPPROTO_DCCP 33 #endif #ifndef IPPROTO_UDPLITE # define IPPROTO_UDPLITE 136 #endif #include #include #include /***************************************************************************** * Module descriptor *****************************************************************************/ #define DEST_TEXT N_("Destination") #define DEST_LONGTEXT N_( \ "This is the output URL that will be used." ) #define SDP_TEXT N_("SDP") #define SDP_LONGTEXT N_( \ "This allows you to specify how the SDP (Session Descriptor) for this RTP "\ "session will be made available. You must use a url: http://location to " \ "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \ "for the SDP to be announced via SAP." ) #define SAP_TEXT N_("SAP announcing") #define SAP_LONGTEXT N_("Announce this session with SAP.") #define MUX_TEXT N_("Muxer") #define MUX_LONGTEXT N_( \ "This allows you to specify the muxer used for the streaming output. " \ "Default is to use no muxer (standard RTP stream)." ) #define NAME_TEXT N_("Session name") #define NAME_LONGTEXT N_( \ "This is the name of the session that will be announced in the SDP " \ "(Session Descriptor)." ) #define CAT_TEXT N_("Session category") #define CAT_LONGTEXT N_( \ "This allows you to specify a category for the session, " \ "that will be announced if you choose to use SAP." ) #define DESC_TEXT N_("Session description") #define DESC_LONGTEXT N_( \ "This allows you to give a short description with details about the stream, " \ "that will be announced in the SDP (Session Descriptor)." ) #define URL_TEXT N_("Session URL") #define URL_LONGTEXT N_( \ "This allows you to give a URL with more details about the stream " \ "(often the website of the streaming organization), that will " \ "be announced in the SDP (Session Descriptor)." ) #define EMAIL_TEXT N_("Session email") #define EMAIL_LONGTEXT N_( \ "This allows you to give a contact mail address for the stream, that will " \ "be announced in the SDP (Session Descriptor)." ) #define PORT_TEXT N_("Port") #define PORT_LONGTEXT N_( \ "This allows you to specify the base port for the RTP streaming." ) #define PORT_AUDIO_TEXT N_("Audio port") #define PORT_AUDIO_LONGTEXT N_( \ "This allows you to specify the default audio port for the RTP streaming." ) #define PORT_VIDEO_TEXT N_("Video port") #define PORT_VIDEO_LONGTEXT N_( \ "This allows you to specify the default video port for the RTP streaming." ) #define TTL_TEXT N_("Hop limit (TTL)") #define TTL_LONGTEXT N_( \ "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \ "the multicast packets sent by the stream output (-1 = use operating " \ "system built-in default).") #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing") #define RTCP_MUX_LONGTEXT N_( \ "This sends and receives RTCP packet multiplexed over the same port " \ "as RTP packets." ) #define CACHING_TEXT N_("Caching value (ms)") #define CACHING_LONGTEXT N_( \ "Default caching value for outbound RTP streams. This " \ "value should be set in milliseconds." ) #define PROTO_TEXT N_("Transport protocol") #define PROTO_LONGTEXT N_( \ "This selects which transport protocol to use for RTP." ) #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)") #define SRTP_KEY_LONGTEXT N_( \ "RTP packets will be integrity-protected and ciphered "\ "with this Secure RTP master shared secret key. "\ "This must be a 32-character-long hexadecimal string.") #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)") #define SRTP_SALT_LONGTEXT N_( \ "Secure RTP requires a (non-secret) master salt value. " \ "This must be a 28-character-long hexadecimal string.") static const char *const ppsz_protos[] = { "dccp", "sctp", "tcp", "udp", "udplite", }; static const char *const ppsz_protocols[] = { "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite", }; #define RFC3016_TEXT N_("MP4A LATM") #define RFC3016_LONGTEXT N_( \ "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." ) #define RTSP_TIMEOUT_TEXT N_( "RTSP session timeout (s)" ) #define RTSP_TIMEOUT_LONGTEXT N_( "RTSP sessions will be closed after " \ "not receiving any RTSP request for this long. Setting it to a " \ "negative value or zero disables timeouts. The default is 60 (one " \ "minute)." ) #define RTSP_USER_TEXT N_("Username") #define RTSP_USER_LONGTEXT N_("Username that will be " \ "requested to access the stream." ) #define RTSP_PASS_TEXT N_("Password") #define RTSP_PASS_LONGTEXT N_("Password that will be " \ "requested to access the stream." ) static int Open ( vlc_object_t * ); static void Close( vlc_object_t * ); #define SOUT_CFG_PREFIX "sout-rtp-" #define MAX_EMPTY_BLOCKS 200 vlc_module_begin () set_shortname( N_("RTP")) set_description( N_("RTP stream output") ) set_capability( "sout stream", 0 ) add_shortcut( "rtp", "vod" ) set_category( CAT_SOUT ) set_subcategory( SUBCAT_SOUT_STREAM ) add_string( SOUT_CFG_PREFIX "dst", "", DEST_TEXT, DEST_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "sdp", "", SDP_TEXT, SDP_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "mux", "", MUX_TEXT, MUX_LONGTEXT, true ) add_bool( SOUT_CFG_PREFIX "sap", false, SAP_TEXT, SAP_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "name", "", NAME_TEXT, NAME_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "cat", "", CAT_TEXT, CAT_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "description", "", DESC_TEXT, DESC_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "url", "", URL_TEXT, URL_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "email", "", EMAIL_TEXT, EMAIL_LONGTEXT, true ) add_obsolete_string( SOUT_CFG_PREFIX "phone" ) /* since 3.0.0 */ add_string( SOUT_CFG_PREFIX "proto", "udp", PROTO_TEXT, PROTO_LONGTEXT, false ) change_string_list( ppsz_protos, ppsz_protocols ) add_integer( SOUT_CFG_PREFIX "port", 5004, PORT_TEXT, PORT_LONGTEXT, true ) add_integer( SOUT_CFG_PREFIX "port-audio", 0, PORT_AUDIO_TEXT, PORT_AUDIO_LONGTEXT, true ) add_integer( SOUT_CFG_PREFIX "port-video", 0, PORT_VIDEO_TEXT, PORT_VIDEO_LONGTEXT, true ) add_integer( SOUT_CFG_PREFIX "ttl", -1, TTL_TEXT, TTL_LONGTEXT, true ) add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false ) add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000, CACHING_TEXT, CACHING_LONGTEXT, true ) #ifdef HAVE_SRTP add_string( SOUT_CFG_PREFIX "key", "", SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false ) add_string( SOUT_CFG_PREFIX "salt", "", SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false ) #endif add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, RFC3016_TEXT, RFC3016_LONGTEXT, false ) set_callbacks( Open, Close ) add_submodule () set_shortname( N_("RTSP VoD" ) ) set_description( N_("RTSP VoD server") ) set_category( CAT_SOUT ) set_subcategory( SUBCAT_SOUT_VOD ) set_capability( "vod server", 10 ) set_callbacks( OpenVoD, CloseVoD ) add_shortcut( "rtsp" ) add_integer( "rtsp-timeout", 60, RTSP_TIMEOUT_TEXT, RTSP_TIMEOUT_LONGTEXT, true ) add_string( "sout-rtsp-user", "", RTSP_USER_TEXT, RTSP_USER_LONGTEXT, true ) add_password( "sout-rtsp-pwd", "", RTSP_PASS_TEXT, RTSP_PASS_LONGTEXT, true ) vlc_module_end () /***************************************************************************** * Exported prototypes *****************************************************************************/ static const char *const ppsz_sout_options[] = { "dst", "name", "cat", "port", "port-audio", "port-video", "*sdp", "ttl", "mux", "sap", "description", "url", "email", "proto", "rtcp-mux", "caching", #ifdef HAVE_SRTP "key", "salt", #endif "mp4a-latm", NULL }; static sout_stream_id_sys_t *Add( sout_stream_t *, const es_format_t * ); static void Del ( sout_stream_t *, sout_stream_id_sys_t * ); static int Send( sout_stream_t *, sout_stream_id_sys_t *, block_t* ); static sout_stream_id_sys_t *MuxAdd( sout_stream_t *, const es_format_t * ); static void MuxDel ( sout_stream_t *, sout_stream_id_sys_t * ); static int MuxSend( sout_stream_t *, sout_stream_id_sys_t *, block_t* ); static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout ); static void* ThreadSend( void * ); static void *rtp_listen_thread( void * ); static void SDPHandleUrl( sout_stream_t *, const char * ); static int SapSetup( sout_stream_t *p_stream ); static int FileSetup( sout_stream_t *p_stream ); static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * ); static int64_t rtp_init_ts( const vod_media_t *p_media, const char *psz_vod_session ); struct sout_stream_sys_t { /* SDP */ char *psz_sdp; vlc_mutex_t lock_sdp; /* SDP to disk */ char *psz_sdp_file; /* SDP via SAP */ bool b_export_sap; session_descriptor_t *p_session; /* SDP via HTTP */ httpd_host_t *p_httpd_host; httpd_file_t *p_httpd_file; /* RTSP */ rtsp_stream_t *rtsp; /* RTSP NPT and timestamp computations */ vlc_tick_t i_npt_zero; /* when NPT=0 packet is sent */ int64_t i_pts_zero; /* predicts PTS of NPT=0 packet */ int64_t i_pts_offset; /* matches actual PTS to prediction */ vlc_mutex_t lock_ts; /* */ char *psz_destination; uint16_t i_port; uint16_t i_port_audio; uint16_t i_port_video; uint8_t proto; bool rtcp_mux; bool b_latm; /* VoD */ vod_media_t *p_vod_media; char *psz_vod_session; /* in case we do TS/PS over rtp */ sout_mux_t *p_mux; sout_access_out_t *p_grab; block_t *packet; /* */ vlc_mutex_t lock_es; int i_es; sout_stream_id_sys_t **es; }; typedef struct rtp_sink_t { int rtp_fd; rtcp_sender_t *rtcp; } rtp_sink_t; struct sout_stream_id_sys_t { sout_stream_t *p_stream; /* rtp field */ /* For RFC 4175, seqnum is extended to 32-bits */ uint32_t i_sequence; bool b_first_packet; bool b_ts_init; uint32_t i_ts_offset; uint8_t ssrc[4]; /* for rtsp */ uint16_t i_seq_sent_next; /* for sdp */ rtp_format_t rtp_fmt; int i_port; /* Packetizer specific fields */ int i_mtu; #ifdef HAVE_SRTP srtp_session_t *srtp; #endif /* Packets sinks */ vlc_thread_t thread; vlc_mutex_t lock_sink; int sinkc; rtp_sink_t *sinkv; rtsp_stream_id_t *rtsp_id; struct { int *fd; vlc_thread_t thread; } listen; block_fifo_t *p_fifo; int64_t i_caching; }; /***************************************************************************** * Open: *****************************************************************************/ static int Open( vlc_object_t *p_this ) { sout_stream_t *p_stream = (sout_stream_t*)p_this; sout_stream_sys_t *p_sys = NULL; char *psz; bool b_rtsp = false; config_ChainParse( p_stream, SOUT_CFG_PREFIX, ppsz_sout_options, p_stream->p_cfg ); p_sys = malloc( sizeof( sout_stream_sys_t ) ); if( p_sys == NULL ) return VLC_ENOMEM; p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" ); p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" ); p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" ); p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" ); p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" ); if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio ) { msg_Err( p_stream, "audio and video RTP port must be distinct" ); free( p_sys->psz_destination ); free( p_sys ); return VLC_EGENERIC; } for( config_chain_t *p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next ) { if( !strcmp( p_cfg->psz_name, "sdp" ) && ( p_cfg->psz_value != NULL ) && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) ) { b_rtsp = true; break; } } if( !b_rtsp ) { psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" ); if( psz != NULL ) { if( !strncasecmp( psz, "rtsp:", 5 ) ) b_rtsp = true; free( psz ); } } /* Transport protocol */ p_sys->proto = IPPROTO_UDP; psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto"); if ((psz == NULL) || !strcasecmp (psz, "udp")) (void)0; /* default */ else if (!strcasecmp (psz, "dccp")) { p_sys->proto = IPPROTO_DCCP; p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */ } #if 0 else if (!strcasecmp (psz, "sctp")) { p_sys->proto = IPPROTO_TCP; p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */ } #endif #if 0 else if (!strcasecmp (psz, "tcp")) { p_sys->proto = IPPROTO_TCP; p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */ } #endif else if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite")) p_sys->proto = IPPROTO_UDPLITE; else msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"", psz); free (psz); var_Create (p_this, "dccp-service", VLC_VAR_STRING); p_sys->p_vod_media = NULL; p_sys->psz_vod_session = NULL; if (! strcmp(p_stream->psz_name, "vod")) { /* The VLM stops all instances before deleting a media, so this * reference will remain valid during the lifetime of the rtp * stream output. */ p_sys->p_vod_media = var_InheritAddress(p_stream, "vod-media"); if (p_sys->p_vod_media != NULL) { p_sys->psz_vod_session = var_InheritString(p_stream, "vod-session"); if (p_sys->psz_vod_session == NULL) { msg_Err(p_stream, "missing VoD session"); free(p_sys); return VLC_EGENERIC; } const char *mux = vod_get_mux(p_sys->p_vod_media); var_SetString(p_stream, SOUT_CFG_PREFIX "mux", mux); } } if( p_sys->psz_destination == NULL && !b_rtsp && p_sys->p_vod_media == NULL ) { msg_Err( p_stream, "missing destination and not in RTSP mode" ); free( p_sys ); return VLC_EGENERIC; } int i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" ); if( i_ttl != -1 ) { var_Create( p_stream, "ttl", VLC_VAR_INTEGER ); var_SetInteger( p_stream, "ttl", i_ttl ); } p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" ); /* NPT=0 time will be determined when we packetize the first packet * (of any ES). But we want to be able to report rtptime in RTSP * without waiting (and already did in the VoD case). So until then, * we use an arbitrary reference PTS for timestamp computations, and * then actual PTS will catch up using offsets. */ p_sys->i_npt_zero = VLC_TICK_INVALID; p_sys->i_pts_zero = rtp_init_ts(p_sys->p_vod_media, p_sys->psz_vod_session); p_sys->i_es = 0; p_sys->es = NULL; p_sys->rtsp = NULL; p_sys->psz_sdp = NULL; p_sys->b_export_sap = false; p_sys->p_session = NULL; p_sys->psz_sdp_file = NULL; p_sys->p_httpd_host = NULL; p_sys->p_httpd_file = NULL; p_stream->p_sys = p_sys; vlc_mutex_init( &p_sys->lock_sdp ); vlc_mutex_init( &p_sys->lock_ts ); vlc_mutex_init( &p_sys->lock_es ); psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" ); if( psz != NULL ) { /* Check muxer type */ if( strncasecmp( psz, "ps", 2 ) && strncasecmp( psz, "mpeg1", 5 ) && strncasecmp( psz, "ts", 2 ) ) { msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" ); free( psz ); vlc_mutex_destroy( &p_sys->lock_sdp ); vlc_mutex_destroy( &p_sys->lock_ts ); vlc_mutex_destroy( &p_sys->lock_es ); free( p_sys->psz_vod_session ); free( p_sys->psz_destination ); free( p_sys ); return VLC_EGENERIC; } p_sys->p_grab = GrabberCreate( p_stream ); p_sys->p_mux = sout_MuxNew( p_stream->p_sout, psz, p_sys->p_grab ); free( psz ); if( p_sys->p_mux == NULL ) { msg_Err( p_stream, "cannot create muxer" ); sout_AccessOutDelete( p_sys->p_grab ); vlc_mutex_destroy( &p_sys->lock_sdp ); vlc_mutex_destroy( &p_sys->lock_ts ); vlc_mutex_destroy( &p_sys->lock_es ); free( p_sys->psz_vod_session ); free( p_sys->psz_destination ); free( p_sys ); return VLC_EGENERIC; } p_sys->packet = NULL; p_stream->pf_add = MuxAdd; p_stream->pf_del = MuxDel; p_stream->pf_send = MuxSend; } else { p_sys->p_mux = NULL; p_sys->p_grab = NULL; p_stream->pf_add = Add; p_stream->pf_del = Del; p_stream->pf_send = Send; } p_stream->pace_nocontrol = true; if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) ) SDPHandleUrl( p_stream, "sap://" ); psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" ); if( psz != NULL ) { config_chain_t *p_cfg; SDPHandleUrl( p_stream, psz ); for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next ) { if( !strcmp( p_cfg->psz_name, "sdp" ) ) { if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' ) continue; /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */ if( !strcmp( p_cfg->psz_value, psz ) ) continue; SDPHandleUrl( p_stream, p_cfg->psz_value ); } } free( psz ); } if( p_sys->p_mux != NULL ) { sout_stream_id_sys_t *id = Add( p_stream, NULL ); if( id == NULL ) { Close( p_this ); return VLC_EGENERIC; } } return VLC_SUCCESS; } /***************************************************************************** * Close: *****************************************************************************/ static void Close( vlc_object_t * p_this ) { sout_stream_t *p_stream = (sout_stream_t*)p_this; sout_stream_sys_t *p_sys = p_stream->p_sys; if( p_sys->p_mux ) { assert( p_sys->i_es <= 1 ); sout_MuxDelete( p_sys->p_mux ); if ( p_sys->i_es > 0 ) Del( p_stream, p_sys->es[0] ); sout_AccessOutDelete( p_sys->p_grab ); if( p_sys->packet ) { block_Release( p_sys->packet ); } } if( p_sys->rtsp != NULL ) RtspUnsetup( p_sys->rtsp ); vlc_mutex_destroy( &p_sys->lock_sdp ); vlc_mutex_destroy( &p_sys->lock_ts ); vlc_mutex_destroy( &p_sys->lock_es ); if( p_sys->p_httpd_file ) httpd_FileDelete( p_sys->p_httpd_file ); if( p_sys->p_httpd_host ) httpd_HostDelete( p_sys->p_httpd_host ); free( p_sys->psz_sdp ); if( p_sys->psz_sdp_file != NULL ) { unlink( p_sys->psz_sdp_file ); free( p_sys->psz_sdp_file ); } free( p_sys->psz_vod_session ); free( p_sys->psz_destination ); free( p_sys ); } /***************************************************************************** * SDPHandleUrl: *****************************************************************************/ static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url ) { sout_stream_sys_t *p_sys = p_stream->p_sys; vlc_url_t url; vlc_UrlParse( &url, psz_url ); if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) ) { if( p_sys->p_httpd_file ) { msg_Err( p_stream, "you can use sdp=http:// only once" ); goto out; } if( HttpSetup( p_stream, &url ) ) { msg_Err( p_stream, "cannot export SDP as HTTP" ); } } else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) ) { if( p_sys->rtsp != NULL ) { msg_Err( p_stream, "you can use sdp=rtsp:// only once" ); goto out; } if( url.psz_host != NULL && *url.psz_host ) { msg_Warn( p_stream, "\"%s\" RTSP host might be ignored in " "multiple-host configurations, use at your own risks.", url.psz_host ); msg_Info( p_stream, "Consider passing --rtsp-host=IP on the " "command line instead." ); var_Create( p_stream, "rtsp-host", VLC_VAR_STRING ); var_SetString( p_stream, "rtsp-host", url.psz_host ); } if( url.i_port != 0 ) { /* msg_Info( p_stream, "Consider passing --rtsp-port=%u on " "the command line instead.", url.i_port ); */ var_Create( p_stream, "rtsp-port", VLC_VAR_INTEGER ); var_SetInteger( p_stream, "rtsp-port", url.i_port ); } p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), NULL, url.psz_path ); if( p_sys->rtsp == NULL ) msg_Err( p_stream, "cannot export SDP as RTSP" ); } else if( ( url.psz_protocol && !strcasecmp( url.psz_protocol, "sap" ) ) || ( url.psz_host && !strcasecmp( url.psz_host, "sap" ) ) ) { p_sys->b_export_sap = true; SapSetup( p_stream ); } else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "file" ) ) { if( p_sys->psz_sdp_file != NULL ) { msg_Err( p_stream, "you can use sdp=file:// only once" ); goto out; } p_sys->psz_sdp_file = vlc_uri2path( psz_url ); if( p_sys->psz_sdp_file == NULL ) goto out; FileSetup( p_stream ); } else { msg_Warn( p_stream, "unknown protocol for SDP (%s)", url.psz_protocol ); } out: vlc_UrlClean( &url ); } /***************************************************************************** * SDPGenerate *****************************************************************************/ /*static*/ char *SDPGenerate( sout_stream_t *p_stream, const char *rtsp_url ) { sout_stream_sys_t *p_sys = p_stream->p_sys; struct vlc_memstream sdp; struct sockaddr_storage dst; char *psz_sdp = NULL; socklen_t dstlen; int i; /* * When we have a fixed destination (typically when we do multicast), * we need to put the actual port numbers in the SDP. * When there is no fixed destination, we only support RTSP unicast * on-demand setup, so we should rather let the clients decide which ports * to use. * When there is both a fixed destination and RTSP unicast, we need to * put port numbers used by the fixed destination, otherwise the SDP would * become totally incorrect for multicast use. It should be noted that * port numbers from SDP with RTSP are only "recommendation" from the * server to the clients (per RFC2326), so only broken clients will fail * to handle this properly. There is no solution but to use two differents * output chain with two different RTSP URLs if you need to handle this * scenario. */ int inclport; vlc_mutex_lock( &p_sys->lock_es ); if( unlikely(p_sys->i_es == 0 || (rtsp_url != NULL && !p_sys->es[0]->rtsp_id)) ) goto out; /* hmm... */ if( p_sys->psz_destination != NULL ) { inclport = 1; /* Oh boy, this is really ugly! */ dstlen = sizeof( dst ); if( p_sys->es[0]->listen.fd != NULL ) getsockname( p_sys->es[0]->listen.fd[0], (struct sockaddr *)&dst, &dstlen ); else getpeername( p_sys->es[0]->sinkv[0].rtp_fd, (struct sockaddr *)&dst, &dstlen ); } else { inclport = 0; /* Check against URL format rtsp://[]:/ */ bool ipv6 = rtsp_url != NULL && strlen( rtsp_url ) > 7 && rtsp_url[7] == '['; /* Dummy destination address for RTSP */ dstlen = ipv6 ? sizeof( struct sockaddr_in6 ) : sizeof( struct sockaddr_in ); memset (&dst, 0, dstlen); dst.ss_family = ipv6 ? AF_INET6 : AF_INET; #ifdef HAVE_SA_LEN dst.ss_len = dstlen; #endif } if( vlc_sdp_Start( &sdp, VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX, NULL, 0, (struct sockaddr *)&dst, dstlen ) ) goto out; /* TODO: a=source-filter */ if( p_sys->rtcp_mux ) sdp_AddAttribute( &sdp, "rtcp-mux", NULL ); if( rtsp_url != NULL ) sdp_AddAttribute ( &sdp, "control", "%s", rtsp_url ); const char *proto = "RTP/AVP"; /* protocol */ if( rtsp_url == NULL ) { switch( p_sys->proto ) { case IPPROTO_UDP: break; case IPPROTO_TCP: proto = "TCP/RTP/AVP"; break; case IPPROTO_DCCP: proto = "DCCP/RTP/AVP"; break; case IPPROTO_UDPLITE: return psz_sdp; } } for( i = 0; i < p_sys->i_es; i++ ) { sout_stream_id_sys_t *id = p_sys->es[i]; rtp_format_t *rtp_fmt = &id->rtp_fmt; const char *mime_major; /* major MIME type */ switch( rtp_fmt->cat ) { case VIDEO_ES: mime_major = "video"; break; case AUDIO_ES: mime_major = "audio"; break; case SPU_ES: mime_major = "text"; break; default: continue; } sdp_AddMedia( &sdp, mime_major, proto, inclport * id->i_port, rtp_fmt->payload_type, false, rtp_fmt->bitrate, rtp_fmt->ptname, rtp_fmt->clock_rate, rtp_fmt->channels, rtp_fmt->fmtp); /* cf RFC4566 §5.14 */ if( inclport && !p_sys->rtcp_mux && (id->i_port & 1) ) sdp_AddAttribute( &sdp, "rtcp", "%u", id->i_port + 1 ); if( rtsp_url != NULL ) { char *track_url = RtspAppendTrackPath( id->rtsp_id, rtsp_url ); if( track_url != NULL ) { sdp_AddAttribute( &sdp, "control", "%s", track_url ); free( track_url ); } } else { if( id->listen.fd != NULL ) sdp_AddAttribute( &sdp, "setup", "passive" ); if( p_sys->proto == IPPROTO_DCCP ) sdp_AddAttribute( &sdp, "dccp-service-code", "SC:RTP%c", toupper( (unsigned char)mime_major[0] ) ); } } if( vlc_memstream_close( &sdp ) == 0 ) psz_sdp = sdp.ptr; out: vlc_mutex_unlock( &p_sys->lock_es ); return psz_sdp; } /***************************************************************************** * RTP mux *****************************************************************************/ /** * Shrink the MTU down to a fixed packetization time (for audio). */ static void rtp_set_ptime (sout_stream_id_sys_t *id, unsigned ptime_ms, size_t bytes) { /* Samples per second */ size_t spl = (id->rtp_fmt.clock_rate - 1) * ptime_ms / 1000 + 1; bytes *= id->rtp_fmt.channels; spl *= bytes; if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */ id->i_mtu = 12 + spl; else /* MTU is too small for ptime, align to a sample boundary */ id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes); } uint32_t rtp_compute_ts( unsigned i_clock_rate, int64_t i_pts ) { /* This is an overflow-proof way of doing: * return i_pts * (int64_t)i_clock_rate / CLOCK_FREQ; * * NOTE: this plays nice with offsets because the (equivalent) * calculations are linear. */ lldiv_t q = lldiv(i_pts, CLOCK_FREQ); return q.quot * (int64_t)i_clock_rate + q.rem * (int64_t)i_clock_rate / CLOCK_FREQ; } /** Add an ES as a new RTP stream */ static sout_stream_id_sys_t *Add( sout_stream_t *p_stream, const es_format_t *p_fmt ) { /* NOTE: As a special case, if we use a non-RTP * mux (TS/PS), then p_fmt is NULL. */ sout_stream_sys_t *p_sys = p_stream->p_sys; char *psz_sdp; sout_stream_id_sys_t *id = malloc( sizeof( *id ) ); if( unlikely(id == NULL) ) return NULL; id->p_stream = p_stream; id->i_mtu = var_InheritInteger( p_stream, "mtu" ); if( id->i_mtu <= 12 + 16 ) id->i_mtu = 576 - 20 - 8; /* pessimistic */ msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu ); #ifdef HAVE_SRTP id->srtp = NULL; #endif vlc_mutex_init( &id->lock_sink ); id->sinkc = 0; id->sinkv = NULL; id->rtsp_id = NULL; id->p_fifo = NULL; id->listen.fd = NULL; id->b_first_packet = true; id->i_caching = (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching"); vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence)); vlc_rand_bytes (id->ssrc, sizeof (id->ssrc)); bool format = false; if (p_sys->p_vod_media != NULL) { id->rtp_fmt.ptname = NULL; uint32_t ssrc; int val = vod_init_id(p_sys->p_vod_media, p_sys->psz_vod_session, p_fmt ? p_fmt->i_id : 0, id, &id->rtp_fmt, &ssrc, &id->i_seq_sent_next); if (val == VLC_SUCCESS) { 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; } /* 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); } if (!format) { 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; } #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) { free (key); goto error; } char *salt = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt"); int val = srtp_setkeystring (id->srtp, key, salt ? salt : ""); free (salt); free (key); if (val) { msg_Err (p_stream, "bad SRTP key/salt combination (%s)", vlc_strerror_c(val)); goto error; } id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */ } #endif 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->rtp_fmt.cat) { case VIDEO_ES: code = "RTPV"; break; case AUDIO_ES: code = "RTPARTPV"; break; case SPU_ES: code = "RTPTRTPV"; break; default: code = "RTPORTPV"; break; } var_SetString (p_stream, "dccp-service", code); type = SOCK_DCCP; } #endif /* fall through */ case IPPROTO_TCP: id->listen.fd = net_Listen( VLC_OBJECT(p_stream), p_sys->psz_destination, i_port, 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 fd = net_ConnectDgram( p_stream, p_sys->psz_destination, i_port, -1, p_sys->proto ); if( fd == -1 ) { msg_Err( p_stream, "cannot create RTP socket" ); goto error; } /* Ignore any unexpected incoming packet (including RTCP-RR * packets in case of rtcp-mux) */ setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 }, sizeof (int)); rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL ); /* FIXME: test if this is multicast */ mcast_fd = fd; } } } if( p_fmt != NULL ) switch( p_fmt->i_codec ) { case VLC_CODEC_MULAW: case VLC_CODEC_ALAW: case VLC_CODEC_U8: rtp_set_ptime (id, 20, 1); break; case VLC_CODEC_S16B: case VLC_CODEC_S16L: rtp_set_ptime (id, 20, 2); break; case VLC_CODEC_S24B: rtp_set_ptime (id, 20, 3); break; default: break; } #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_TICK_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, GetDWBE( id->ssrc ), id->rtp_fmt.clock_rate, mcast_fd ); id->p_fifo = block_FifoNew(); if( unlikely(id->p_fifo == NULL) ) goto error; if( vlc_clone( &id->thread, ThreadSend, id, VLC_THREAD_PRIORITY_HIGHEST ) ) { block_FifoRelease( id->p_fifo ); id->p_fifo = NULL; goto error; } /* Update p_sys context */ vlc_mutex_lock( &p_sys->lock_es ); TAB_APPEND( p_sys->i_es, p_sys->es, id ); vlc_mutex_unlock( &p_sys->lock_es ); psz_sdp = SDPGenerate( p_stream, NULL ); vlc_mutex_lock( &p_sys->lock_sdp ); free( p_sys->psz_sdp ); p_sys->psz_sdp = psz_sdp; vlc_mutex_unlock( &p_sys->lock_sdp ); msg_Dbg( p_stream, "sdp=\n%s", p_sys->psz_sdp ); /* Update SDP (sap/file) */ if( p_sys->b_export_sap ) SapSetup( p_stream ); if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream ); return id; error: Del( p_stream, id ); return NULL; } static void Del( sout_stream_t *p_stream, sout_stream_id_sys_t *id ) { sout_stream_sys_t *p_sys = p_stream->p_sys; vlc_mutex_lock( &p_sys->lock_es ); TAB_REMOVE( p_sys->i_es, p_sys->es, id ); vlc_mutex_unlock( &p_sys->lock_es ); if( likely(id->p_fifo != NULL) ) { vlc_cancel( id->thread ); vlc_join( id->thread, NULL ); block_FifoRelease( id->p_fifo ); } 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->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 ) SapSetup( p_stream ); if( p_sys->psz_sdp_file != NULL ) FileSetup( p_stream ); free( id ); } static int Send( sout_stream_t *p_stream, sout_stream_id_sys_t *id, block_t *p_buffer ) { assert( p_stream->p_sys->p_mux == NULL ); (void)p_stream; while( p_buffer != NULL ) { block_t *p_next = p_buffer->p_next; p_buffer->p_next = NULL; /* 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; p_buffer = p_next; } return VLC_SUCCESS; } /**************************************************************************** * SAP: ****************************************************************************/ static int SapSetup( sout_stream_t *p_stream ) { sout_stream_sys_t *p_sys = p_stream->p_sys; /* Remove the previous session */ if( p_sys->p_session != NULL) { sout_AnnounceUnRegister( p_stream, p_sys->p_session); p_sys->p_session = NULL; } if( p_sys->i_es > 0 && p_sys->psz_sdp && *p_sys->psz_sdp ) p_sys->p_session = sout_AnnounceRegisterSDP( p_stream, p_sys->psz_sdp, p_sys->psz_destination ); return VLC_SUCCESS; } /**************************************************************************** * File: ****************************************************************************/ static int FileSetup( sout_stream_t *p_stream ) { sout_stream_sys_t *p_sys = p_stream->p_sys; FILE *f; if( p_sys->psz_sdp == NULL ) return VLC_EGENERIC; /* too early */ if( ( f = vlc_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL ) { msg_Err( p_stream, "cannot open file '%s' (%s)", p_sys->psz_sdp_file, vlc_strerror_c(errno) ); return VLC_EGENERIC; } fputs( p_sys->psz_sdp, f ); fclose( f ); return VLC_SUCCESS; } /**************************************************************************** * HTTP: ****************************************************************************/ static int HttpCallback( httpd_file_sys_t *p_args, httpd_file_t *, uint8_t *p_request, uint8_t **pp_data, int *pi_data ); static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url) { sout_stream_sys_t *p_sys = p_stream->p_sys; p_sys->p_httpd_host = 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, url->psz_path ? url->psz_path : "/", "application/sdp", NULL, NULL, HttpCallback, (void*)p_sys ); } if( p_sys->p_httpd_file == NULL ) { return VLC_EGENERIC; } return VLC_SUCCESS; } static int HttpCallback( httpd_file_sys_t *p_args, httpd_file_t *f, uint8_t *p_request, uint8_t **pp_data, int *pi_data ) { VLC_UNUSED(f); VLC_UNUSED(p_request); sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args; vlc_mutex_lock( &p_sys->lock_sdp ); if( p_sys->psz_sdp && *p_sys->psz_sdp ) { *pi_data = strlen( p_sys->psz_sdp ); *pp_data = malloc( *pi_data ); memcpy( *pp_data, p_sys->psz_sdp, *pi_data ); } else { *pp_data = NULL; *pi_data = 0; } vlc_mutex_unlock( &p_sys->lock_sdp ); return VLC_SUCCESS; } /**************************************************************************** * RTP send ****************************************************************************/ static void* ThreadSend( void *data ) { #ifdef _WIN32 # define ENOBUFS WSAENOBUFS # define EAGAIN WSAEWOULDBLOCK # define EWOULDBLOCK WSAEWOULDBLOCK #endif sout_stream_id_sys_t *id = data; unsigned i_caching = id->i_caching; for (;;) { block_t *out = block_FifoGet( id->p_fifo ); block_cleanup_push (out); #ifdef HAVE_SRTP if( id->srtp ) { /* FIXME: this is awfully inefficient */ size_t len = out->i_buffer; out = block_Realloc( out, 0, len + 10 ); out->i_buffer = len; int canc = vlc_savecancel (); int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 ); vlc_restorecancel (canc); if( val ) { msg_Dbg( id->p_stream, "SRTP sending error: %s", vlc_strerror_c(val) ); 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 (); vlc_mutex_lock( &id->lock_sink ); unsigned deadc = 0; /* How many dead sockets? */ int deadv[id->sinkc ? id->sinkc : 1]; /* Dead sockets list */ for( int i = 0; i < id->sinkc; i++ ) { #ifdef HAVE_SRTP if( !id->srtp ) /* FIXME: SRTCP support */ #endif SendRTCP( id->sinkv[i].rtcp, out ); if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) == -1 && net_errno != EAGAIN && net_errno != EWOULDBLOCK && net_errno != ENOBUFS && net_errno != ENOMEM ) { 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 ); else /* Broken connection */ deadv[deadc++] = id->sinkv[i].rtp_fd; } } id->i_seq_sent_next = ntohs(((uint16_t *) out->p_buffer)[1]) + 1; vlc_mutex_unlock( &id->lock_sink ); block_Release( out ); for( unsigned i = 0; i < deadc; i++ ) { msg_Dbg( id->p_stream, "removing socket %d", deadv[i] ); rtp_del_sink( id, deadv[i] ); } vlc_restorecancel (canc); } return NULL; } /* This thread dequeues incoming connections (DCCP streaming) */ static void *rtp_listen_thread( void *data ) { sout_stream_id_sys_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 ); } vlc_assert_unreachable(); } int rtp_add_sink( sout_stream_id_sys_t *id, int fd, bool rtcp_mux, uint16_t *seq ) { rtp_sink_t sink = { fd, NULL }; sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP, rtcp_mux ); if( sink.rtcp == NULL ) msg_Err( id->p_stream, "RTCP failed!" ); vlc_mutex_lock( &id->lock_sink ); TAB_APPEND(id->sinkc, id->sinkv, sink); if( seq != NULL ) *seq = id->i_seq_sent_next; vlc_mutex_unlock( &id->lock_sink ); return VLC_SUCCESS; } void rtp_del_sink( sout_stream_id_sys_t *id, int fd ) { rtp_sink_t sink = { fd, NULL }; /* NOTE: must be safe to use if fd is not included */ vlc_mutex_lock( &id->lock_sink ); for( int i = 0; i < id->sinkc; i++ ) { if (id->sinkv[i].rtp_fd == fd) { sink = id->sinkv[i]; TAB_ERASE(id->sinkc, id->sinkv, i); break; } } vlc_mutex_unlock( &id->lock_sink ); CloseRTCP( sink.rtcp ); net_Close( sink.rtp_fd ); } uint16_t rtp_get_seq( sout_stream_id_sys_t *id ) { /* This will return values for the next packet. */ uint16_t seq; vlc_mutex_lock( &id->lock_sink ); seq = id->i_seq_sent_next; vlc_mutex_unlock( &id->lock_sink ); return seq; } /* Return 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 ) { 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 bits, this is enough and allows us * to stay away from overflows */ i_ts_init &= 0xFFFFFFFFFFFF; return i_ts_init; } /* 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_sys_t *id, const vod_media_t *p_media, const char *psz_vod_session, int64_t *p_npt ) { if (p_npt != NULL) *p_npt = 0; 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; vlc_tick_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_TICK_INVALID ) return p_sys->i_pts_zero; vlc_tick_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_sys_t *id, block_t *out, bool b_m_bit, int64_t i_pts ) { if( !id->b_ts_init ) { sout_stream_sys_t *p_sys = id->p_stream->p_sys; vlc_mutex_lock( &p_sys->lock_ts ); if( p_sys->i_npt_zero == VLC_TICK_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_m_bit?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; out->p_buffer[5] = ( i_timestamp >> 16 )&0xff; out->p_buffer[6] = ( i_timestamp >> 8 )&0xff; out->p_buffer[7] = ( i_timestamp )&0xff; memcpy( out->p_buffer + 8, id->ssrc, 4 ); id->i_sequence++; } uint16_t rtp_get_extended_sequence( sout_stream_id_sys_t *id ) { return id->i_sequence >> 16; } void rtp_packetize_send( sout_stream_id_sys_t *id, block_t *out ) { block_FifoPut( id->p_fifo, out ); } /** * @return configured max RTP payload size (including payload type-specific * headers, excluding RTP and transport headers) */ size_t rtp_mtu (const sout_stream_id_sys_t *id) { return id->i_mtu - 12; } /***************************************************************************** * Non-RTP mux *****************************************************************************/ /** Add an ES to a non-RTP muxed stream */ static sout_stream_id_sys_t *MuxAdd( sout_stream_t *p_stream, const es_format_t *p_fmt ) { sout_input_t *p_input; sout_mux_t *p_mux = p_stream->p_sys->p_mux; assert( p_mux != NULL ); p_input = sout_MuxAddStream( p_mux, p_fmt ); if( p_input == NULL ) { msg_Err( p_stream, "cannot add this stream to the muxer" ); return NULL; } return (sout_stream_id_sys_t *)p_input; } static int MuxSend( sout_stream_t *p_stream, sout_stream_id_sys_t *id, block_t *p_buffer ) { sout_mux_t *p_mux = p_stream->p_sys->p_mux; assert( p_mux != NULL ); return sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer ); } /** Remove an ES from a non-RTP muxed stream */ static void MuxDel( sout_stream_t *p_stream, sout_stream_id_sys_t *id ) { sout_mux_t *p_mux = p_stream->p_sys->p_mux; assert( p_mux != NULL ); sout_MuxDeleteStream( p_mux, (sout_input_t *)id ); } static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream, const block_t *p_buffer ) { sout_stream_sys_t *p_sys = p_stream->p_sys; sout_stream_id_sys_t *id = p_sys->es[0]; int64_t i_dts = p_buffer->i_dts; uint8_t *p_data = p_buffer->p_buffer; size_t i_data = p_buffer->i_buffer; size_t i_max = id->i_mtu - 12; bool b_dis = (p_buffer->i_flags & BLOCK_FLAG_DISCONTINUITY); size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max; while( i_data > 0 ) { size_t i_size; /* output complete packet */ if( p_sys->packet && p_sys->packet->i_buffer + i_data > i_max ) { rtp_packetize_send( id, p_sys->packet ); p_sys->packet = NULL; } if( p_sys->packet == NULL ) { /* allocate a new packet */ p_sys->packet = block_Alloc( id->i_mtu ); /* m-bit is discontinuity for MPEG1/2 PS and TS, RFC2250 2.1 */ rtp_packetize_common( id, p_sys->packet, b_dis, i_dts ); p_sys->packet->i_buffer = 12; p_sys->packet->i_dts = i_dts; p_sys->packet->i_length = p_buffer->i_length / i_packet; i_dts += p_sys->packet->i_length; b_dis = false; } i_size = __MIN( i_data, (unsigned)(id->i_mtu - p_sys->packet->i_buffer) ); memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer], p_data, i_size ); p_sys->packet->i_buffer += i_size; p_data += i_size; i_data -= i_size; } return VLC_SUCCESS; } static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access, block_t *p_buffer ) { sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys; while( p_buffer ) { block_t *p_next; AccessOutGrabberWriteBuffer( p_stream, p_buffer ); p_next = p_buffer->p_next; block_Release( p_buffer ); p_buffer = p_next; } return VLC_SUCCESS; } static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream ) { sout_access_out_t *p_grab; p_grab = vlc_object_create( p_stream, sizeof( *p_grab ) ); if( p_grab == NULL ) return NULL; p_grab->p_module = NULL; p_grab->psz_access = strdup( "grab" ); p_grab->p_cfg = NULL; p_grab->psz_path = strdup( "" ); p_grab->p_sys = (sout_access_out_sys_t *)p_stream; p_grab->pf_seek = NULL; p_grab->pf_write = AccessOutGrabberWrite; return p_grab; } void rtp_get_video_geometry( sout_stream_id_sys_t *id, int *width, int *height ) { int ret = sscanf( id->rtp_fmt.fmtp, "%*s width=%d; height=%d; ", width, height ); assert( ret == 2 ); }