/***************************************************************************** * rtp.c: rtp stream output module ***************************************************************************** * Copyright (C) 2003-2004 the VideoLAN team * Copyright © 2007-2008 Rémi Denis-Courmont * * Authors: Laurent Aimar * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA. *****************************************************************************/ /***************************************************************************** * Preamble *****************************************************************************/ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include #include #include #include #include #include #include #include "rtp.h" #ifdef HAVE_UNISTD_H # include # include # include # 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 /***************************************************************************** * Module descriptor *****************************************************************************/ #define DEST_TEXT N_("Destination") #define DEST_LONGTEXT N_( \ "This is the output URL that will be used." ) #define SDP_TEXT N_("SDP") #define SDP_LONGTEXT N_( \ "This allows you to specify how the SDP (Session Descriptor) for this RTP "\ "session will be made available. You must use an url: http://location to " \ "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \ "for the SDP to be announced via SAP." ) #define SAP_TEXT N_("SAP announcing") #define SAP_LONGTEXT N_("Announce this session with SAP.") #define MUX_TEXT N_("Muxer") #define MUX_LONGTEXT N_( \ "This allows you to specify the muxer used for the streaming output. " \ "Default is to use no muxer (standard RTP stream)." ) #define NAME_TEXT N_("Session name") #define NAME_LONGTEXT N_( \ "This is the name of the session that will be announced in the SDP " \ "(Session Descriptor)." ) #define DESC_TEXT N_("Session description") #define DESC_LONGTEXT N_( \ "This allows you to give a short description with details about the stream, " \ "that will be announced in the SDP (Session Descriptor)." ) #define URL_TEXT N_("Session URL") #define URL_LONGTEXT N_( \ "This allows you to give an URL with more details about the stream " \ "(often the website of the streaming organization), that will " \ "be announced in the SDP (Session Descriptor)." ) #define EMAIL_TEXT N_("Session email") #define EMAIL_LONGTEXT N_( \ "This allows you to give a contact mail address for the stream, that will " \ "be announced in the SDP (Session Descriptor)." ) #define PHONE_TEXT N_("Session phone number") #define PHONE_LONGTEXT N_( \ "This allows you to give a contact telephone number for the stream, that will " \ "be announced in the SDP (Session Descriptor)." ) #define PORT_TEXT N_("Port") #define PORT_LONGTEXT N_( \ "This allows you to specify the base port for the RTP streaming." ) #define PORT_AUDIO_TEXT N_("Audio port") #define PORT_AUDIO_LONGTEXT N_( \ "This allows you to specify the default audio port for the RTP streaming." ) #define PORT_VIDEO_TEXT N_("Video port") #define PORT_VIDEO_LONGTEXT N_( \ "This allows you to specify the default video port for the RTP streaming." ) #define TTL_TEXT N_("Hop limit (TTL)") #define TTL_LONGTEXT N_( \ "This is the hop limit (also known as \"Time-To-Live\" or TTL) of " \ "the multicast packets sent by the stream output (-1 = use operating " \ "system built-in default).") #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing") #define RTCP_MUX_LONGTEXT N_( \ "This sends and receives RTCP packet multiplexed over the same port " \ "as RTP packets." ) #define PROTO_TEXT N_("Transport protocol") #define PROTO_LONGTEXT N_( \ "This selects which transport protocol to use for RTP." ) #define SRTP_KEY_TEXT N_("SRTP key (hexadecimal)") #define SRTP_KEY_LONGTEXT N_( \ "RTP packets will be integrity-protected and ciphered "\ "with this Secure RTP master shared secret key.") #define SRTP_SALT_TEXT N_("SRTP salt (hexadecimal)") #define SRTP_SALT_LONGTEXT N_( \ "Secure RTP requires a (non-secret) master salt value.") static const char *const ppsz_protos[] = { "dccp", "sctp", "tcp", "udp", "udplite", }; static const char *const ppsz_protocols[] = { "DCCP", "SCTP", "TCP", "UDP", "UDP-Lite", }; #define RFC3016_TEXT N_("MP4A LATM") #define RFC3016_LONGTEXT N_( \ "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." ) static int Open ( vlc_object_t * ); static void Close( vlc_object_t * ); #define SOUT_CFG_PREFIX "sout-rtp-" #define MAX_EMPTY_BLOCKS 200 vlc_module_begin(); set_shortname( N_("RTP")); set_description( N_("RTP stream output") ); set_capability( "sout stream", 0 ); add_shortcut( "rtp" ); set_category( CAT_SOUT ); set_subcategory( SUBCAT_SOUT_STREAM ); add_string( SOUT_CFG_PREFIX "dst", "", NULL, DEST_TEXT, DEST_LONGTEXT, true ); add_string( SOUT_CFG_PREFIX "sdp", "", NULL, SDP_TEXT, SDP_LONGTEXT, true ); add_string( SOUT_CFG_PREFIX "mux", "", NULL, MUX_TEXT, MUX_LONGTEXT, true ); add_bool( SOUT_CFG_PREFIX "sap", false, NULL, SAP_TEXT, SAP_LONGTEXT, true ); add_string( SOUT_CFG_PREFIX "name", "", NULL, NAME_TEXT, NAME_LONGTEXT, true ); add_string( SOUT_CFG_PREFIX "description", "", NULL, DESC_TEXT, DESC_LONGTEXT, true ); add_string( SOUT_CFG_PREFIX "url", "", NULL, URL_TEXT, URL_LONGTEXT, true ); add_string( SOUT_CFG_PREFIX "email", "", NULL, EMAIL_TEXT, EMAIL_LONGTEXT, true ); add_string( SOUT_CFG_PREFIX "phone", "", NULL, PHONE_TEXT, PHONE_LONGTEXT, true ); add_string( SOUT_CFG_PREFIX "proto", "udp", NULL, PROTO_TEXT, PROTO_LONGTEXT, false ); change_string_list( ppsz_protos, ppsz_protocols, NULL ); add_integer( SOUT_CFG_PREFIX "port", 50004, NULL, PORT_TEXT, PORT_LONGTEXT, true ); add_integer( SOUT_CFG_PREFIX "port-audio", 50000, NULL, PORT_AUDIO_TEXT, PORT_AUDIO_LONGTEXT, true ); add_integer( SOUT_CFG_PREFIX "port-video", 50002, NULL, PORT_VIDEO_TEXT, PORT_VIDEO_LONGTEXT, true ); add_integer( SOUT_CFG_PREFIX "ttl", -1, NULL, TTL_TEXT, TTL_LONGTEXT, true ); add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL, RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false ); add_string( SOUT_CFG_PREFIX "key", "", NULL, SRTP_KEY_TEXT, SRTP_KEY_LONGTEXT, false ); add_string( SOUT_CFG_PREFIX "salt", "", NULL, SRTP_SALT_TEXT, SRTP_SALT_LONGTEXT, false ); add_bool( SOUT_CFG_PREFIX "mp4a-latm", 0, NULL, RFC3016_TEXT, RFC3016_LONGTEXT, false ); set_callbacks( Open, Close ); vlc_module_end(); /***************************************************************************** * Exported prototypes *****************************************************************************/ static const char *const ppsz_sout_options[] = { "dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux", "sap", "description", "url", "email", "phone", "proto", "rtcp-mux", "key", "salt", "mp4a-latm", NULL }; static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * ); static int Del ( sout_stream_t *, sout_stream_id_t * ); static int Send( sout_stream_t *, sout_stream_id_t *, block_t* ); static sout_stream_id_t *MuxAdd ( sout_stream_t *, es_format_t * ); static int MuxDel ( sout_stream_t *, sout_stream_id_t * ); static int MuxSend( sout_stream_t *, sout_stream_id_t *, block_t* ); static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout ); static void* ThreadSend( vlc_object_t *p_this ); static void SDPHandleUrl( sout_stream_t *, const char * ); static int SapSetup( sout_stream_t *p_stream ); static int FileSetup( sout_stream_t *p_stream ); static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t * ); struct sout_stream_sys_t { /* SDP */ char *psz_sdp; vlc_mutex_t lock_sdp; /* SDP to disk */ bool b_export_sdp_file; 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; /* */ char *psz_destination; uint32_t payload_bitmap; uint16_t i_port; uint16_t i_port_audio; uint16_t i_port_video; uint8_t proto; bool rtcp_mux; int i_ttl:9; bool b_latm; /* in case we do TS/PS over rtp */ sout_mux_t *p_mux; sout_access_out_t *p_grab; block_t *packet; /* */ vlc_mutex_t lock_es; int i_es; sout_stream_id_t **es; }; typedef int (*pf_rtp_packetizer_t)( sout_stream_id_t *, block_t * ); typedef struct rtp_sink_t { int rtp_fd; rtcp_sender_t *rtcp; } rtp_sink_t; struct sout_stream_id_t { VLC_COMMON_MEMBERS sout_stream_t *p_stream; /* rtp field */ uint16_t i_sequence; uint8_t i_payload_type; uint8_t ssrc[4]; /* for sdp */ const char *psz_enc; char *psz_fmtp; int i_clock_rate; int i_port; int i_cat; int i_channels; int i_bitrate; /* Packetizer specific fields */ int i_mtu; srtp_session_t *srtp; pf_rtp_packetizer_t pf_packetize; /* Packets sinks */ vlc_mutex_t lock_sink; int sinkc; rtp_sink_t *sinkv; rtsp_stream_id_t *rtsp_id; int *listen_fd; block_fifo_t *p_fifo; int64_t i_caching; }; /***************************************************************************** * Open: *****************************************************************************/ static int Open( vlc_object_t *p_this ) { sout_stream_t *p_stream = (sout_stream_t*)p_this; sout_instance_t *p_sout = p_stream->p_sout; sout_stream_sys_t *p_sys = NULL; config_chain_t *p_cfg = NULL; char *psz; bool b_rtsp = false; config_ChainParse( p_stream, SOUT_CFG_PREFIX, ppsz_sout_options, p_stream->p_cfg ); p_sys = malloc( sizeof( sout_stream_sys_t ) ); if( p_sys == NULL ) return VLC_ENOMEM; p_sys->psz_destination = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "dst" ); p_sys->i_port = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port" ); p_sys->i_port_audio = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-audio" ); p_sys->i_port_video = var_GetInteger( p_stream, SOUT_CFG_PREFIX "port-video" ); p_sys->rtcp_mux = var_GetBool( p_stream, SOUT_CFG_PREFIX "rtcp-mux" ); p_sys->psz_sdp_file = NULL; if( p_sys->i_port_audio == p_sys->i_port_video ) { msg_Err( p_stream, "audio and video port cannot be the same" ); p_sys->i_port_audio = 0; p_sys->i_port_video = 0; } for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next ) { if( !strcmp( p_cfg->psz_name, "sdp" ) && ( p_cfg->psz_value != NULL ) && !strncasecmp( p_cfg->psz_value, "rtsp:", 5 ) ) { b_rtsp = true; break; } } if( !b_rtsp ) { psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" ); if( psz != NULL ) { if( !strncasecmp( psz, "rtsp:", 5 ) ) b_rtsp = true; free( psz ); } } /* Transport protocol */ p_sys->proto = IPPROTO_UDP; psz = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"proto"); if ((psz == NULL) || !strcasecmp (psz, "udp")) (void)0; /* default */ else if (!strcasecmp (psz, "dccp")) { p_sys->proto = IPPROTO_DCCP; p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */ } #if 0 else if (!strcasecmp (psz, "sctp")) { p_sys->proto = IPPROTO_TCP; p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */ } #endif #if 0 else if (!strcasecmp (psz, "tcp")) { p_sys->proto = IPPROTO_TCP; p_sys->rtcp_mux = true; /* Force RTP/RTCP mux */ } #endif else if (!strcasecmp (psz, "udplite") || !strcasecmp (psz, "udp-lite")) p_sys->proto = IPPROTO_UDPLITE; else msg_Warn (p_this, "unknown or unsupported transport protocol \"%s\"", psz); free (psz); var_Create (p_this, "dccp-service", VLC_VAR_STRING); if( ( p_sys->psz_destination == NULL ) && !b_rtsp ) { msg_Err( p_stream, "missing destination and not in RTSP mode" ); free( p_sys ); return VLC_EGENERIC; } p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" ); if( p_sys->i_ttl == -1 ) { /* Normally, we should let the default hop limit up to the core, * but we have to know it to 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" ); } p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" ); p_sys->payload_bitmap = 0; 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->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_es ); psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" ); if( psz != NULL ) { sout_stream_id_t *id; /* Check muxer type */ if( strncasecmp( psz, "ps", 2 ) && strncasecmp( psz, "mpeg1", 5 ) && strncasecmp( psz, "ts", 2 ) ) { msg_Err( p_stream, "unsupported muxer type for RTP (only TS/PS)" ); free( psz ); vlc_mutex_destroy( &p_sys->lock_sdp ); vlc_mutex_destroy( &p_sys->lock_es ); free( p_sys ); return VLC_EGENERIC; } p_sys->p_grab = GrabberCreate( p_stream ); p_sys->p_mux = sout_MuxNew( p_sout, psz, p_sys->p_grab ); free( psz ); if( p_sys->p_mux == NULL ) { msg_Err( p_stream, "cannot create muxer" ); sout_AccessOutDelete( p_sys->p_grab ); vlc_mutex_destroy( &p_sys->lock_sdp ); vlc_mutex_destroy( &p_sys->lock_es ); free( p_sys ); 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 ); return VLC_EGENERIC; } p_sys->packet = NULL; p_stream->pf_add = MuxAdd; p_stream->pf_del = MuxDel; p_stream->pf_send = MuxSend; } else { p_sys->p_mux = NULL; p_sys->p_grab = NULL; p_stream->pf_add = Add; p_stream->pf_del = Del; p_stream->pf_send = Send; } if( var_GetBool( p_stream, SOUT_CFG_PREFIX"sap" ) ) SDPHandleUrl( p_stream, "sap" ); psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "sdp" ); if( psz != NULL ) { config_chain_t *p_cfg; SDPHandleUrl( p_stream, psz ); for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next ) { if( !strcmp( p_cfg->psz_name, "sdp" ) ) { if( p_cfg->psz_value == NULL || *p_cfg->psz_value == '\0' ) continue; /* needed both :sout-rtp-sdp= and rtp{sdp=} can be used */ if( !strcmp( p_cfg->psz_value, psz ) ) continue; SDPHandleUrl( p_stream, p_cfg->psz_value ); } } free( psz ); } /* update p_sout->i_out_pace_nocontrol */ p_stream->p_sout->i_out_pace_nocontrol++; return VLC_SUCCESS; } /***************************************************************************** * Close: *****************************************************************************/ static void Close( vlc_object_t * p_this ) { sout_stream_t *p_stream = (sout_stream_t*)p_this; sout_stream_sys_t *p_sys = p_stream->p_sys; /* update p_sout->i_out_pace_nocontrol */ p_stream->p_sout->i_out_pace_nocontrol--; if( p_sys->p_mux ) { assert( p_sys->i_es == 1 ); Del( p_stream, p_sys->es[0] ); sout_MuxDelete( p_sys->p_mux ); 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_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->b_export_sdp_file ) { #ifdef HAVE_UNISTD_H unlink( p_sys->psz_sdp_file ); #endif free( p_sys->psz_sdp_file ); } free( p_sys->psz_destination ); free( p_sys ); } /***************************************************************************** * SDPHandleUrl: *****************************************************************************/ static void SDPHandleUrl( sout_stream_t *p_stream, const char *psz_url ) { sout_stream_sys_t *p_sys = p_stream->p_sys; vlc_url_t url; vlc_UrlParse( &url, psz_url, 0 ); if( url.psz_protocol && !strcasecmp( url.psz_protocol, "http" ) ) { if( p_sys->p_httpd_file ) { msg_Err( p_stream, "you can use sdp=http:// only once" ); goto out; } if( HttpSetup( p_stream, &url ) ) { msg_Err( p_stream, "cannot export SDP as HTTP" ); } } else if( url.psz_protocol && !strcasecmp( url.psz_protocol, "rtsp" ) ) { if( p_sys->rtsp != NULL ) { msg_Err( p_stream, "you can use sdp=rtsp:// only once" ); goto out; } /* FIXME test if destination is multicast or no destination at all */ p_sys->rtsp = RtspSetup( p_stream, &url ); if( p_sys->rtsp == NULL ) { msg_Err( p_stream, "cannot export SDP as RTSP" ); } if( p_sys->p_mux != NULL ) { 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 ); } } 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->b_export_sdp_file ) { 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 ); } else { msg_Warn( p_stream, "unknown protocol for SDP (%s)", url.psz_protocol ); } out: vlc_UrlClean( &url ); } /***************************************************************************** * SDPGenerate *****************************************************************************/ /*static*/ char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url ) { const sout_stream_sys_t *p_sys = p_stream->p_sys; char *psz_sdp; struct sockaddr_storage dst; socklen_t dstlen; int i; /* * When we have a fixed destination (typically when we do multicast), * we need to put the actual port numbers in the SDP. * When there is no fixed destination, we only support RTSP unicast * on-demand setup, so we should rather let the clients decide which ports * to use. * When there is both a fixed destination and RTSP unicast, we need to * put port numbers used by the fixed destination, otherwise the SDP would * become totally incorrect for multicast use. It should be noted that * port numbers from SDP with RTSP are only "recommendation" from the * server to the clients (per RFC2326), so only broken clients will fail * to handle this properly. There is no solution but to use two differents * output chain with two different RTSP URLs if you need to handle this * scenario. */ int inclport; if( p_sys->psz_destination != NULL ) { inclport = 1; /* Oh boy, this is really ugly! (+ race condition on lock_es) */ dstlen = sizeof( dst ); if( p_sys->es[0]->listen_fd != NULL ) getsockname( p_sys->es[0]->listen_fd[0], (struct sockaddr *)&dst, &dstlen ); else getpeername( p_sys->es[0]->sinkv[0].rtp_fd, (struct sockaddr *)&dst, &dstlen ); } else { inclport = 0; /* Dummy destination address for RTSP */ memset (&dst, 0, sizeof( struct sockaddr_in ) ); dst.ss_family = AF_INET; #ifdef HAVE_SA_LEN dst.ss_len = #endif dstlen = sizeof( struct sockaddr_in ); } psz_sdp = vlc_sdp_Start( VLC_OBJECT( p_stream ), SOUT_CFG_PREFIX, NULL, 0, (struct sockaddr *)&dst, dstlen ); if( psz_sdp == NULL ) return NULL; /* TODO: a=source-filter */ if( p_sys->rtcp_mux ) sdp_AddAttribute( &psz_sdp, "rtcp-mux", NULL ); if( rtsp_url != NULL ) sdp_AddAttribute ( &psz_sdp, "control", "%s", rtsp_url ); /* FIXME: locking?! */ for( i = 0; i < p_sys->i_es; i++ ) { sout_stream_id_t *id = p_sys->es[i]; const char *mime_major; /* major MIME type */ const char *proto = "RTP/AVP"; /* protocol */ switch( id->i_cat ) { case VIDEO_ES: mime_major = "video"; break; case AUDIO_ES: mime_major = "audio"; break; case SPU_ES: mime_major = "text"; break; default: continue; } 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); 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 ); } else { if( id->listen_fd != NULL ) sdp_AddAttribute( &psz_sdp, "setup", "passive" ); if( p_sys->proto == IPPROTO_DCCP ) sdp_AddAttribute( &psz_sdp, "dccp-service-code", "SC:RTP%c", toupper( mime_major[0] ) ); } } return psz_sdp; } /***************************************************************************** * RTP mux *****************************************************************************/ static void sprintf_hexa( char *s, uint8_t *p_data, int i_data ) { static const char hex[16] = "0123456789abcdef"; int i; for( i = 0; i < i_data; i++ ) { s[2*i+0] = hex[(p_data[i]>>4)&0xf]; s[2*i+1] = hex[(p_data[i] )&0xf]; } s[2*i_data] = '\0'; } /** * Shrink the MTU down to a fixed packetization time (for audio). */ static void rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes) { /* Samples per second */ size_t spl = (id->i_clock_rate - 1) * ptime_ms / 1000 + 1; bytes *= id->i_channels; spl *= bytes; if (spl < rtp_mtu (id)) /* MTU is big enough for ptime */ id->i_mtu = 12 + spl; else /* MTU is too small for ptime, align to a sample boundary */ id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes); } /** 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; 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 ) 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; } while( i_port == 0 ) { if( p_sys->i_port != p_sys->i_port_audio && p_sys->i_port != p_sys->i_port_video ) { i_port = p_sys->i_port; p_sys->i_port += 2; break; } p_sys->i_port += 2; } id->p_stream = p_stream; id->i_sequence = rand()&0xffff; /* 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->ssrc[0] = rand()&0xff; id->ssrc[1] = rand()&0xff; id->ssrc[2] = rand()&0xff; id->ssrc[3] = rand()&0xff; id->psz_enc = NULL; id->psz_fmtp = NULL; id->i_clock_rate = 90000; /* most common case for video */ id->i_channels = 0; id->i_port = i_port; if( p_fmt != NULL ) { id->i_cat = p_fmt->i_cat; if( p_fmt->i_cat == AUDIO_ES ) { id->i_clock_rate = p_fmt->audio.i_rate; id->i_channels = p_fmt->audio.i_channels; } id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */ } else { id->i_cat = VIDEO_ES; id->i_bitrate = 0; } id->i_mtu = 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"); if (key) { id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10, SRTP_PRF_AES_CM, SRTP_RCC_MODE1); if (id->srtp == NULL) { free (key); goto error; } char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt"); errno = srtp_setkeystring (id->srtp, key, salt ? salt : ""); free (salt); free (key); if (errno) { msg_Err (p_stream, "bad SRTP key/salt combination (%m)"); goto error; } id->i_sequence = 0; /* FIXME: awful hack for libvlc_srtp */ } 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"); if( p_sys->psz_destination != NULL ) switch( p_sys->proto ) { case IPPROTO_DCCP: { const char *code; switch (id->i_cat) { case VIDEO_ES: code = "RTPV"; break; case AUDIO_ES: code = "RTPARTPV"; break; case SPU_ES: code = "RTPTRPTV"; break; default: code = "RTPORTPV"; break; } var_SetString (p_stream, "dccp-service", code); } /* fall through */ case IPPROTO_TCP: id->listen_fd = net_Listen( VLC_OBJECT(p_stream), p_sys->psz_destination, i_port, p_sys->proto ); if( id->listen_fd == NULL ) { msg_Err( p_stream, "passive COMEDIA RTP socket failed" ); goto error; } break; default: { int ttl = (p_sys->i_ttl > 0) ? p_sys->i_ttl : -1; int fd = net_ConnectDgram( p_stream, p_sys->psz_destination, i_port, ttl, p_sys->proto ); if( fd == -1 ) { msg_Err( p_stream, "cannot create RTP socket" ); goto error; } rtp_add_sink( id, fd, p_sys->rtcp_mux ); } } if( p_fmt == NULL ) { char *psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" ); if( psz == NULL ) /* Uho! */ ; else if( strncmp( psz, "ts", 2 ) == 0 ) { id->i_payload_type = 33; id->psz_enc = "MP2T"; } else { id->psz_enc = "MP2P"; } free( psz ); } else switch( p_fmt->i_codec ) { case VLC_FOURCC( 'u', 'l', 'a', 'w' ): 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_FOURCC( 'a', 'l', 'a', 'w' ): if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 8000 ) id->i_payload_type = 8; id->psz_enc = "PCMA"; id->pf_packetize = rtp_packetize_split; rtp_set_ptime (id, 20, 1); break; case VLC_FOURCC( 's', '1', '6', 'b' ): 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; rtp_set_ptime (id, 20, 2); break; case VLC_FOURCC( 'u', '8', ' ', ' ' ): id->psz_enc = "L8"; id->pf_packetize = rtp_packetize_split; rtp_set_ptime (id, 20, 1); break; case VLC_FOURCC( 'm', 'p', 'g', 'a' ): case VLC_FOURCC( 'm', 'p', '3', ' ' ): 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_FOURCC( 'm', 'p', 'g', 'v' ): id->i_payload_type = 32; id->psz_enc = "MPV"; id->pf_packetize = rtp_packetize_mpv; break; case VLC_FOURCC( 'G', '7', '2', '6' ): case VLC_FOURCC( 'g', '7', '2', '6' ): 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_FOURCC( 'a', '5', '2', ' ' ): id->psz_enc = "ac3"; id->pf_packetize = rtp_packetize_ac3; break; case VLC_FOURCC( 'H', '2', '6', '3' ): id->psz_enc = "H263-1998"; id->pf_packetize = rtp_packetize_h263; break; case VLC_FOURCC( 'h', '2', '6', '4' ): 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_FOURCC( 'm', 'p', '4', 'v' ): { 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_FOURCC( 'm', 'p', '4', 'a' ): { 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_FOURCC( 's', 'a', 'm', 'r' ): id->psz_enc = "AMR"; id->psz_fmtp = strdup( "octet-align=1" ); id->pf_packetize = rtp_packetize_amr; break; case VLC_FOURCC( 's', 'a', 'w', 'b' ): id->psz_enc = "AMR-WB"; id->psz_fmtp = strdup( "octet-align=1" ); id->pf_packetize = rtp_packetize_amr; break; case VLC_FOURCC( 's', 'p', 'x', ' ' ): id->psz_enc = "SPEEX"; id->pf_packetize = rtp_packetize_spx; break; case VLC_FOURCC( 't', '1', '4', '0' ): id->psz_enc = "t140" ; id->i_clock_rate = 1000; id->pf_packetize = rtp_packetize_t140; break; default: msg_Err( p_stream, "cannot add this stream (unsupported " "codec:%4.4s)", (char*)&p_fmt->i_codec ); goto error; } if (id->i_payload_type >= 96) /* Mark dynamic payload type in use */ p_sys->payload_bitmap |= 1 << (id->i_payload_type - 96); if( cscov != -1 ) cscov += 8 /* UDP */ + 12 /* RTP */; if( id->sinkc > 0 ) net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 ); 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->p_fifo = block_FifoNew(); if( vlc_thread_create( id, "RTP send thread", ThreadSend, VLC_THREAD_PRIORITY_HIGHEST, false ) ) 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->b_export_sdp_file ) FileSetup( p_stream ); return id; error: Del( p_stream, id ); return NULL; } static int Del( sout_stream_t *p_stream, sout_stream_id_t *id ) { sout_stream_sys_t *p_sys = p_stream->p_sys; 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)); free( id->psz_fmtp ); 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->srtp != NULL ) srtp_destroy( id->srtp ); 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 ); vlc_object_detach( id ); vlc_object_release( id ); return VLC_SUCCESS; } static int Send( sout_stream_t *p_stream, sout_stream_id_t *id, block_t *p_buffer ) { block_t *p_next; assert( p_stream->p_sys->p_mux == NULL ); (void)p_stream; while( p_buffer != NULL ) { p_next = p_buffer->p_next; if( id->pf_packetize( id, p_buffer ) ) break; block_Release( p_buffer ); p_buffer = p_next; } return VLC_SUCCESS; } /**************************************************************************** * SAP: ****************************************************************************/ static int SapSetup( sout_stream_t *p_stream ) { sout_stream_sys_t *p_sys = p_stream->p_sys; sout_instance_t *p_sout = p_stream->p_sout; /* Remove the previous session */ if( p_sys->p_session != NULL) { sout_AnnounceUnRegister( p_sout, p_sys->p_session); p_sys->p_session = NULL; } if( ( p_sys->i_es > 0 || p_sys->p_mux ) && p_sys->psz_sdp && *p_sys->psz_sdp ) { announce_method_t *p_method = sout_SAPMethod(); p_sys->p_session = sout_AnnounceRegisterSDP( p_sout, p_sys->psz_sdp, p_sys->psz_destination, p_method ); sout_MethodRelease( p_method ); } return VLC_SUCCESS; } /**************************************************************************** * File: ****************************************************************************/ static int FileSetup( sout_stream_t *p_stream ) { sout_stream_sys_t *p_sys = p_stream->p_sys; FILE *f; if( ( f = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL ) { msg_Err( p_stream, "cannot open file '%s' (%m)", p_sys->psz_sdp_file ); return VLC_EGENERIC; } fputs( p_sys->psz_sdp, f ); fclose( f ); return VLC_SUCCESS; } /**************************************************************************** * HTTP: ****************************************************************************/ static int HttpCallback( httpd_file_sys_t *p_args, httpd_file_t *, uint8_t *p_request, uint8_t **pp_data, int *pi_data ); static int HttpSetup( sout_stream_t *p_stream, const vlc_url_t *url) { sout_stream_sys_t *p_sys = p_stream->p_sys; p_sys->p_httpd_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host, url->i_port > 0 ? url->i_port : 80 ); if( p_sys->p_httpd_host ) { p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host, url->psz_path ? url->psz_path : "/", "application/sdp", NULL, NULL, NULL, HttpCallback, (void*)p_sys ); } if( p_sys->p_httpd_file == NULL ) { return VLC_EGENERIC; } return VLC_SUCCESS; } static int HttpCallback( httpd_file_sys_t *p_args, httpd_file_t *f, uint8_t *p_request, uint8_t **pp_data, int *pi_data ) { VLC_UNUSED(f); VLC_UNUSED(p_request); sout_stream_sys_t *p_sys = (sout_stream_sys_t*)p_args; vlc_mutex_lock( &p_sys->lock_sdp ); if( p_sys->psz_sdp && *p_sys->psz_sdp ) { *pi_data = strlen( p_sys->psz_sdp ); *pp_data = malloc( *pi_data ); memcpy( *pp_data, p_sys->psz_sdp, *pi_data ); } else { *pp_data = NULL; *pi_data = 0; } vlc_mutex_unlock( &p_sys->lock_sdp ); return VLC_SUCCESS; } /**************************************************************************** * RTP send ****************************************************************************/ static void* ThreadSend( vlc_object_t *p_this ) { sout_stream_id_t *id = (sout_stream_id_t *)p_this; unsigned i_caching = id->i_caching; for (;;) { block_t *out = block_FifoGet( id->p_fifo ); block_cleanup_push (out); if( id->srtp ) { /* FIXME: this is awfully inefficient */ size_t len = out->i_buffer; out = block_Realloc( out, 0, len + 10 ); out->i_buffer = len; int canc = vlc_savecancel (); int val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 ); vlc_restorecancel (canc); if( val ) { errno = val; msg_Dbg( id, "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; ssize_t len = out->i_buffer; int canc = vlc_savecancel (); vlc_mutex_lock( &id->lock_sink ); unsigned deadc = 0; /* How many dead sockets? */ int deadv[id->sinkc]; /* Dead sockets list */ for( int i = 0; i < id->sinkc; i++ ) { if( !id->srtp ) /* FIXME: SRTCP support */ SendRTCP( id->sinkv[i].rtcp, out ); if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 ) continue; /* Retry sending to root out soft-errors */ if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 ) continue; deadv[deadc++] = id->sinkv[i].rtp_fd; } vlc_mutex_unlock( &id->lock_sink ); block_Release( out ); for( unsigned i = 0; i < deadc; i++ ) { msg_Dbg( id, "removing socket %d", deadv[i] ); rtp_del_sink( id, deadv[i] ); } /* Hopefully we won't overflow the SO_MAXCONN accept queue */ while( id->listen_fd != NULL ) { int fd = net_Accept( id, id->listen_fd, 0 ); if( fd == -1 ) break; msg_Dbg( id, "adding socket %d", fd ); rtp_add_sink( id, fd, true ); } vlc_restorecancel (canc); } return NULL; } int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux ) { rtp_sink_t sink = { fd, NULL }; sink.rtcp = OpenRTCP( VLC_OBJECT( id->p_stream ), fd, IPPROTO_UDP, rtcp_mux ); if( sink.rtcp == NULL ) msg_Err( id, "RTCP failed!" ); vlc_mutex_lock( &id->lock_sink ); INSERT_ELEM( id->sinkv, id->sinkc, id->sinkc, sink ); vlc_mutex_unlock( &id->lock_sink ); return VLC_SUCCESS; } void rtp_del_sink( sout_stream_id_t *id, int fd ) { rtp_sink_t sink = { fd, NULL }; /* NOTE: must be safe to use if fd is not included */ vlc_mutex_lock( &id->lock_sink ); for( int i = 0; i < id->sinkc; i++ ) { if (id->sinkv[i].rtp_fd == fd) { sink = id->sinkv[i]; REMOVE_ELEM( id->sinkv, id->sinkc, i ); break; } } vlc_mutex_unlock( &id->lock_sink ); CloseRTCP( sink.rtcp ); net_Close( sink.rtp_fd ); } uint16_t rtp_get_seq( const sout_stream_id_t *id ) { /* This will return values for the next packet. * Accounting for caching would not be totally trivial. */ return id->i_sequence; } /* FIXME: this is pretty bad - if we remove and then insert an ES * the number will get unsynched from inside RTSP */ unsigned rtp_get_num( const sout_stream_id_t *id ) { sout_stream_sys_t *p_sys = id->p_stream->p_sys; int i; vlc_mutex_lock( &p_sys->lock_es ); for( i = 0; i < p_sys->i_es; i++ ) { if( id == p_sys->es[i] ) break; } vlc_mutex_unlock( &p_sys->lock_es ); return i; } 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); out->p_buffer[0] = 0x80; out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type; out->p_buffer[2] = ( id->i_sequence >> 8)&0xff; out->p_buffer[3] = ( id->i_sequence )&0xff; out->p_buffer[4] = ( i_timestamp >> 24 )&0xff; out->p_buffer[5] = ( i_timestamp >> 16 )&0xff; out->p_buffer[6] = ( i_timestamp >> 8 )&0xff; out->p_buffer[7] = ( i_timestamp )&0xff; memcpy( out->p_buffer + 8, id->ssrc, 4 ); out->i_buffer = 12; id->i_sequence++; } void rtp_packetize_send( sout_stream_id_t *id, block_t *out ) { block_FifoPut( id->p_fifo, out ); } /** * @return configured max RTP payload size (including payload type-specific * headers, excluding RTP and transport headers) */ size_t rtp_mtu (const sout_stream_id_t *id) { return id->i_mtu - 12; } /***************************************************************************** * Non-RTP mux *****************************************************************************/ /** Add an ES to a non-RTP muxed stream */ static sout_stream_id_t *MuxAdd( sout_stream_t *p_stream, es_format_t *p_fmt ) { sout_input_t *p_input; sout_mux_t *p_mux = p_stream->p_sys->p_mux; assert( p_mux != NULL ); p_input = sout_MuxAddStream( p_mux, p_fmt ); if( p_input == NULL ) { msg_Err( p_stream, "cannot add this stream to the muxer" ); return NULL; } return (sout_stream_id_t *)p_input; } static int MuxSend( sout_stream_t *p_stream, sout_stream_id_t *id, block_t *p_buffer ) { sout_mux_t *p_mux = p_stream->p_sys->p_mux; assert( p_mux != NULL ); sout_MuxSendBuffer( p_mux, (sout_input_t *)id, p_buffer ); return VLC_SUCCESS; } /** Remove an ES from a non-RTP muxed stream */ static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id ) { sout_mux_t *p_mux = p_stream->p_sys->p_mux; assert( p_mux != NULL ); sout_MuxDeleteStream( p_mux, (sout_input_t *)id ); return VLC_SUCCESS; } static ssize_t AccessOutGrabberWriteBuffer( sout_stream_t *p_stream, const block_t *p_buffer ) { sout_stream_sys_t *p_sys = p_stream->p_sys; sout_stream_id_t *id = p_sys->es[0]; int64_t i_dts = p_buffer->i_dts; uint8_t *p_data = p_buffer->p_buffer; size_t i_data = p_buffer->i_buffer; size_t i_max = id->i_mtu - 12; size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max; while( i_data > 0 ) { size_t i_size; /* output complete packet */ if( p_sys->packet && p_sys->packet->i_buffer + i_data > i_max ) { rtp_packetize_send( id, p_sys->packet ); p_sys->packet = NULL; } if( p_sys->packet == NULL ) { /* allocate a new packet */ p_sys->packet = block_New( p_stream, id->i_mtu ); rtp_packetize_common( id, p_sys->packet, 1, i_dts ); p_sys->packet->i_dts = i_dts; p_sys->packet->i_length = p_buffer->i_length / i_packet; i_dts += p_sys->packet->i_length; } i_size = __MIN( i_data, (unsigned)(id->i_mtu - p_sys->packet->i_buffer) ); memcpy( &p_sys->packet->p_buffer[p_sys->packet->i_buffer], p_data, i_size ); p_sys->packet->i_buffer += i_size; p_data += i_size; i_data -= i_size; } return VLC_SUCCESS; } static ssize_t AccessOutGrabberWrite( sout_access_out_t *p_access, block_t *p_buffer ) { sout_stream_t *p_stream = (sout_stream_t*)p_access->p_sys; while( p_buffer ) { block_t *p_next; AccessOutGrabberWriteBuffer( p_stream, p_buffer ); p_next = p_buffer->p_next; block_Release( p_buffer ); p_buffer = p_next; } return VLC_SUCCESS; } static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream ) { sout_access_out_t *p_grab; p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) ); if( p_grab == NULL ) return NULL; p_grab->p_module = NULL; p_grab->psz_access = strdup( "grab" ); p_grab->p_cfg = NULL; p_grab->psz_path = strdup( "" ); p_grab->p_sys = (sout_access_out_sys_t *)p_stream; p_grab->pf_seek = NULL; p_grab->pf_write = AccessOutGrabberWrite; vlc_object_attach( p_grab, p_stream ); return p_grab; }