X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=modules%2Fstream_out%2Frtp.c;h=ace40571e4f3c5420943203bb24af57ddcb7f006;hb=4d269a64d89a1bc5ee1d25d6aa0ecc71049a16d8;hp=6765e7923f26ae81bbed2798275078f2f5159fdd;hpb=6da90a1716250d282f16dc6bc9dacec5b9514caf;p=vlc diff --git a/modules/stream_out/rtp.c b/modules/stream_out/rtp.c index 6765e7923f..ace40571e4 100644 --- a/modules/stream_out/rtp.c +++ b/modules/stream_out/rtp.c @@ -2,8 +2,7 @@ * rtp.c: rtp stream output module ***************************************************************************** * Copyright (C) 2003-2004 the VideoLAN team - * Copyright © 2007 Rémi Denis-Courmont - * $Id$ + * Copyright © 2007-2008 Rémi Denis-Courmont * * Authors: Laurent Aimar * @@ -30,7 +29,8 @@ # include "config.h" #endif -#include +#include +#include #include #include @@ -39,6 +39,10 @@ #include #include #include +#include +#ifdef HAVE_SRTP +# include +#endif #include "rtp.h" @@ -48,6 +52,9 @@ # include # include #endif +#ifdef HAVE_ARPA_INET_H +# include +#endif #ifdef HAVE_LINUX_DCCP_H # include #endif @@ -60,6 +67,8 @@ #include +#include + /***************************************************************************** * Module descriptor *****************************************************************************/ @@ -115,7 +124,7 @@ #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 (0 = use operating " \ + "the multicast packets sent by the stream output (-1 = use operating " \ "system built-in default).") #define RTCP_MUX_TEXT N_("RTP/RTCP multiplexing") @@ -123,10 +132,24 @@ "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.") + +#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", }; @@ -145,63 +168,71 @@ static void Close( vlc_object_t * ); #define SOUT_CFG_PREFIX "sout-rtp-" #define MAX_EMPTY_BLOCKS 200 -vlc_module_begin(); - set_shortname( _("RTP")); - set_description( _("RTP stream output") ); - set_capability( "sout stream", 0 ); - add_shortcut( "rtp" ); - set_category( CAT_SOUT ); - set_subcategory( SUBCAT_SOUT_STREAM ); +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 ); - change_unsafe(); + DEST_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "sdp", "", NULL, SDP_TEXT, - SDP_LONGTEXT, true ); + SDP_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "mux", "", NULL, MUX_TEXT, - MUX_LONGTEXT, true ); + MUX_LONGTEXT, true ) add_bool( SOUT_CFG_PREFIX "sap", false, NULL, SAP_TEXT, SAP_LONGTEXT, - true ); + true ) add_string( SOUT_CFG_PREFIX "name", "", NULL, NAME_TEXT, - NAME_LONGTEXT, true ); + NAME_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "description", "", NULL, DESC_TEXT, - DESC_LONGTEXT, true ); + DESC_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "url", "", NULL, URL_TEXT, - URL_LONGTEXT, true ); + URL_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "email", "", NULL, EMAIL_TEXT, - EMAIL_LONGTEXT, true ); + EMAIL_LONGTEXT, true ) add_string( SOUT_CFG_PREFIX "phone", "", NULL, PHONE_TEXT, - PHONE_LONGTEXT, true ); + 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", 0, NULL, TTL_TEXT, - TTL_LONGTEXT, true ); + PROTO_LONGTEXT, false ) + change_string_list( ppsz_protos, ppsz_protocols, NULL ) + add_integer( SOUT_CFG_PREFIX "port", 5004, NULL, PORT_TEXT, + PORT_LONGTEXT, true ) + add_integer( SOUT_CFG_PREFIX "port-audio", 0, NULL, PORT_AUDIO_TEXT, + PORT_AUDIO_LONGTEXT, true ) + add_integer( SOUT_CFG_PREFIX "port-video", 0, 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 ); + RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false ) + add_integer( SOUT_CFG_PREFIX "caching", DEFAULT_PTS_DELAY / 1000, NULL, + CACHING_TEXT, CACHING_LONGTEXT, true ) + +#ifdef HAVE_SRTP + 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 ) +#endif - add_bool( SOUT_CFG_PREFIX "mp4a-latm", 0, NULL, RFC3016_TEXT, - RFC3016_LONGTEXT, false ); + add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, NULL, RFC3016_TEXT, + RFC3016_LONGTEXT, false ) - set_callbacks( Open, Close ); -vlc_module_end(); + set_callbacks( Open, Close ) +vlc_module_end () /***************************************************************************** * Exported prototypes *****************************************************************************/ -static const char *ppsz_sout_options[] = { +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", + "proto", "rtcp-mux", "caching", "key", "salt", "mp4a-latm", NULL }; @@ -215,9 +246,10 @@ 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* ThreadSend( vlc_object_t *p_this ); +static void *rtp_listen_thread( void * ); -static void SDPHandleUrl( sout_stream_t *, char * ); +static void SDPHandleUrl( sout_stream_t *, const char * ); static int SapSetup( sout_stream_t *p_stream ); static int FileSetup( sout_stream_t *p_stream ); @@ -230,7 +262,6 @@ struct sout_stream_sys_t vlc_mutex_t lock_sdp; /* SDP to disk */ - bool b_export_sdp_file; char *psz_sdp_file; /* SDP via SAP */ @@ -244,18 +275,22 @@ struct sout_stream_sys_t /* 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; - uint8_t proto; - uint8_t i_ttl; + uint32_t payload_bitmap; uint16_t i_port; uint16_t i_port_audio; uint16_t i_port_video; - bool b_latm; - bool rtcp_mux; - - /* when need to use a private one or when using muxer */ - int i_payload_type; + 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; @@ -268,8 +303,7 @@ struct sout_stream_sys_t sout_stream_id_t **es; }; -typedef int (*pf_rtp_packetizer_t)( sout_stream_t *, sout_stream_id_t *, - block_t * ); +typedef int (*pf_rtp_packetizer_t)( sout_stream_id_t *, block_t * ); typedef struct rtp_sink_t { @@ -285,8 +319,13 @@ struct sout_stream_id_t /* rtp field */ uint16_t i_sequence; uint8_t i_payload_type; + 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; @@ -297,15 +336,21 @@ struct sout_stream_id_t int i_bitrate; /* Packetizer specific fields */ + int i_mtu; +#ifdef HAVE_SRTP + srtp_session_t *srtp; +#endif pf_rtp_packetizer_t pf_packetize; - int i_mtu; /* Packets sinks */ 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; @@ -335,15 +380,14 @@ static int Open( vlc_object_t *p_this ) 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->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 ) + if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio ) { - msg_Err( p_stream, "audio and video port cannot be the same" ); - p_sys->i_port_audio = 0; - p_sys->i_port_video = 0; + msg_Err( p_stream, "audio and video RTP port must be distinct" ); + free( p_sys->psz_destination ); + free( p_sys ); + return VLC_EGENERIC; } for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next ) @@ -412,7 +456,7 @@ static int Open( vlc_object_t *p_this ) } p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" ); - if( p_sys->i_ttl == 0 ) + 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 @@ -423,15 +467,23 @@ static int Open( vlc_object_t *p_this ) p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" ); - p_sys->i_payload_type = 96; + /* 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. 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 = mdate(); /* arbitrary value, could probably be + * random */ + 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->psz_sdp_file = NULL; p_sys->p_httpd_host = NULL; p_sys->p_httpd_file = NULL; @@ -439,6 +491,7 @@ static int Open( vlc_object_t *p_this ) 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" ); @@ -455,6 +508,7 @@ static int Open( vlc_object_t *p_this ) free( psz ); vlc_mutex_destroy( &p_sys->lock_sdp ); vlc_mutex_destroy( &p_sys->lock_es ); + free( p_sys->psz_destination ); free( p_sys ); return VLC_EGENERIC; } @@ -469,6 +523,7 @@ static int Open( vlc_object_t *p_this ) sout_AccessOutDelete( p_sys->p_grab ); vlc_mutex_destroy( &p_sys->lock_sdp ); vlc_mutex_destroy( &p_sys->lock_es ); + free( p_sys->psz_destination ); free( p_sys ); return VLC_EGENERIC; } @@ -480,6 +535,7 @@ static int Open( vlc_object_t *p_this ) sout_AccessOutDelete( p_sys->p_grab ); vlc_mutex_destroy( &p_sys->lock_sdp ); vlc_mutex_destroy( &p_sys->lock_es ); + free( p_sys->psz_destination ); free( p_sys ); return VLC_EGENERIC; } @@ -547,10 +603,11 @@ static void Close( vlc_object_t * p_this ) if( p_sys->p_mux ) { assert( p_sys->i_es == 1 ); - Del( p_stream, p_sys->es[0] ); sout_MuxDelete( p_sys->p_mux ); + Del( p_stream, p_sys->es[0] ); sout_AccessOutDelete( p_sys->p_grab ); + if( p_sys->packet ) { block_Release( p_sys->packet ); @@ -566,6 +623,7 @@ static void Close( vlc_object_t * p_this ) 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 ) @@ -576,7 +634,7 @@ static void Close( vlc_object_t * p_this ) 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 ); @@ -590,7 +648,7 @@ static void Close( vlc_object_t * p_this ) /***************************************************************************** * SDPHandleUrl: *****************************************************************************/ -static void SDPHandleUrl( sout_stream_t *p_stream, char *psz_url ) +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; @@ -620,14 +678,12 @@ static void SDPHandleUrl( sout_stream_t *p_stream, char *psz_url ) /* 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 ) { sout_stream_id_t *id = p_sys->es[0]; - id->rtsp_id = RtspAddId( p_sys->rtsp, id, 0, GetDWBE( id->ssrc ), + id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ), p_sys->psz_destination, p_sys->i_ttl, id->i_port, id->i_port + 1 ); } @@ -640,16 +696,19 @@ static void SDPHandleUrl( sout_stream_t *p_stream, char *psz_url ) } 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 ); + if( p_sys->psz_sdp_file == NULL ) + goto out; + decode_URI( p_sys->psz_sdp_file ); /* FIXME? */ + FileSetup( p_stream ); } else { @@ -695,8 +754,8 @@ char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url ) /* 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], + 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, @@ -771,6 +830,9 @@ char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url ) id->psz_enc, id->i_clock_rate, id->i_channels, id->psz_fmtp); + if( !p_sys->rtcp_mux && (id->i_port & 1) ) /* cf RFC4566 §5.14 */ + sdp_AddAttribute ( &psz_sdp, "rtcp", "%u", id->i_port + 1 ); + if( rtsp_url != NULL ) { assert( strlen( rtsp_url ) > 0 ); @@ -781,10 +843,10 @@ char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_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", + sdp_AddAttribute( &psz_sdp, "dccp-service-code", "SC:RTP%c", toupper( mime_major[0] ) ); } } @@ -809,6 +871,29 @@ static void sprintf_hexa( char *s, uint8_t *p_data, int i_data ) 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); +} + +uint32_t rtp_compute_ts( const sout_stream_id_t *id, int64_t i_pts ) +{ + /* NOTE: this plays nice with offsets because the calculations are + * linear. */ + return i_pts * (int64_t)id->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 ) @@ -817,51 +902,61 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) * 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; - id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) ); - if( id == NULL ) + if (0xffffffff == p_sys->payload_bitmap) + { + msg_Err (p_stream, "too many RTP elementary streams"); return NULL; - vlc_object_attach( id, p_stream ); + } /* Choose the port */ - i_port = 0; + 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; - 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 ) + /* 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_sys->i_port != p_sys->i_port_audio - && p_sys->i_port != p_sys->i_port_video ) + if (p == 0) { - i_port = p_sys->i_port; - p_sys->i_port += 2; - break; + msg_Err (p_stream, "too many RTP elementary streams"); + return NULL; } - p_sys->i_port += 2; + 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 = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) ); + if( id == NULL ) + return NULL; + vlc_object_attach( id, p_stream ); + id->p_stream = p_stream; - id->i_sequence = rand()&0xffff; - id->i_payload_type = p_sys->i_payload_type; - id->ssrc[0] = rand()&0xff; - id->ssrc[1] = rand()&0xff; - id->ssrc[2] = rand()&0xff; - id->ssrc[3] = rand()&0xff; + /* 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); + + vlc_rand_bytes (&id->i_sequence, sizeof (id->i_sequence)); + vlc_rand_bytes (id->ssrc, sizeof (id->ssrc)); + + id->i_seq_sent_next = id->i_sequence; id->psz_enc = NULL; id->psz_fmtp = NULL; @@ -884,19 +979,46 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) id->i_bitrate = 0; } - id->pf_packetize = NULL; 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->pf_packetize = NULL; + +#ifdef HAVE_SRTP + id->srtp = 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 */ + } +#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->listen.fd = NULL; id->i_caching = (int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching"); @@ -911,25 +1033,32 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) { case VIDEO_ES: code = "RTPV"; break; case AUDIO_ES: code = "RTPARTPV"; break; - case SPU_ES: code = "RTPTRPTV"; break; + case SPU_ES: code = "RTPTRTPV"; 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), + id->listen.fd = net_Listen( VLC_OBJECT(p_stream), p_sys->psz_destination, i_port, p_sys->proto ); - if( id->listen_fd == NULL ) + 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 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 ) @@ -937,7 +1066,11 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) 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 ); } } @@ -957,23 +1090,27 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) { id->psz_enc = "MP2P"; } + free( psz ); } else switch( p_fmt->i_codec ) { - case VLC_FOURCC( 'u', 'l', 'a', 'w' ): + 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_l8; + id->pf_packetize = rtp_packetize_split; + rtp_set_ptime (id, 20, 1); break; - case VLC_FOURCC( 'a', 'l', 'a', 'w' ): + 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_l8; + id->pf_packetize = rtp_packetize_split; + rtp_set_ptime (id, 20, 1); break; - case VLC_FOURCC( 's', '1', '6', 'b' ): + case VLC_CODEC_S16B: + case VLC_CODEC_S16L: if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 ) { id->i_payload_type = 11; @@ -984,32 +1121,62 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) id->i_payload_type = 10; } id->psz_enc = "L16"; - id->pf_packetize = rtp_packetize_l16; + if( p_fmt->i_codec == VLC_CODEC_S16B ) + id->pf_packetize = rtp_packetize_split; + else + id->pf_packetize = rtp_packetize_swab; + rtp_set_ptime (id, 20, 2); break; - case VLC_FOURCC( 'u', '8', ' ', ' ' ): + case VLC_CODEC_U8: id->psz_enc = "L8"; - id->pf_packetize = rtp_packetize_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', ' ' ): + 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_FOURCC( 'm', 'p', 'g', 'v' ): + case VLC_CODEC_MPGV: id->i_payload_type = 32; id->psz_enc = "MPV"; id->pf_packetize = rtp_packetize_mpv; break; - case VLC_FOURCC( 'a', '5', '2', ' ' ): + 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; + default: + msg_Err( p_stream, "cannot add this stream (unsupported " + "G.726 bit rate: %u)", p_fmt->i_bitrate ); + goto error; + } + break; + case VLC_CODEC_A52: id->psz_enc = "ac3"; id->pf_packetize = rtp_packetize_ac3; break; - case VLC_FOURCC( 'H', '2', '6', '3' ): + case VLC_CODEC_H263: id->psz_enc = "H263-1998"; id->pf_packetize = rtp_packetize_h263; break; - case VLC_FOURCC( 'h', '2', '6', '4' ): + case VLC_CODEC_H264: id->psz_enc = "H264"; id->pf_packetize = rtp_packetize_h264; id->psz_fmtp = NULL; @@ -1068,7 +1235,7 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) id->psz_fmtp = strdup( "packetization-mode=1" ); break; - case VLC_FOURCC( 'm', 'p', '4', 'v' ): + case VLC_CODEC_MP4V: { char hexa[2*p_fmt->i_extra +1]; @@ -1083,7 +1250,7 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) } break; } - case VLC_FOURCC( 'm', 'p', '4', 'a' ): + case VLC_CODEC_MP4A: { if(!p_sys->b_latm) { @@ -1128,22 +1295,21 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) } break; } - case VLC_FOURCC( 's', 'a', 'm', 'r' ): + 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_FOURCC( 's', 'a', 'w', 'b' ): + 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_FOURCC( 's', 'p', 'x', ' ' ): - id->i_payload_type = p_sys->i_payload_type++; + case VLC_CODEC_SPEEX: id->psz_enc = "SPEEX"; id->pf_packetize = rtp_packetize_spx; break; - case VLC_FOURCC( 't', '1', '4', '0' ): + case VLC_CODEC_ITU_T140: id->psz_enc = "t140" ; id->i_clock_rate = 1000; id->pf_packetize = rtp_packetize_t140; @@ -1151,27 +1317,36 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) default: msg_Err( p_stream, "cannot add this stream (unsupported " - "codec:%4.4s)", (char*)&p_fmt->i_codec ); + "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 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 - if( id->i_payload_type == p_sys->i_payload_type ) - p_sys->i_payload_type++; + 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, p_sys->i_pts_offset ); if( p_sys->rtsp != NULL ) - id->rtsp_id = RtspAddId( p_sys->rtsp, id, p_sys->i_es, + id->rtsp_id = RtspAddId( p_sys->rtsp, id, 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 ) ) + VLC_THREAD_PRIORITY_HIGHEST ) ) goto error; /* Update p_sys context */ @@ -1190,7 +1365,7 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) /* 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; @@ -1206,7 +1381,6 @@ static int Del( sout_stream_t *p_stream, sout_stream_id_t *id ) if( id->p_fifo != NULL ) { vlc_object_kill( id ); - block_FifoWake( id->p_fifo ); vlc_thread_join( id ); block_FifoRelease( id->p_fifo ); } @@ -1215,11 +1389,9 @@ static int Del( sout_stream_t *p_stream, sout_stream_id_t *id ) 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 ); @@ -1227,14 +1399,22 @@ static int Del( sout_stream_t *p_stream, sout_stream_id_t *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 ); + } +#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->psz_sdp_file != NULL ) FileSetup( p_stream ); vlc_object_detach( id ); vlc_object_release( id ); @@ -1247,11 +1427,12 @@ static int Send( sout_stream_t *p_stream, sout_stream_id_t *id, 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( p_stream, id, p_buffer ) ) + if( id->pf_packetize( id, p_buffer ) ) break; block_Release( p_buffer ); @@ -1296,6 +1477,9 @@ 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 = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL ) { msg_Err( p_stream, "cannot open file '%s' (%m)", @@ -1364,21 +1548,55 @@ static int HttpCallback( httpd_file_sys_t *p_args, /**************************************************************************** * RTP send ****************************************************************************/ -static void ThreadSend( vlc_object_t *p_this ) +static void* ThreadSend( vlc_object_t *p_this ) { +#ifdef WIN32 +# define ECONNREFUSED WSAECONNREFUSED +# define ENOPROTOOPT WSAENOPROTOOPT +# define EHOSTUNREACH WSAEHOSTUNREACH +# define ENETUNREACH WSAENETUNREACH +# define ENETDOWN WSAENETDOWN +# define ENOBUFS WSAENOBUFS +# define EAGAIN WSAEWOULDBLOCK +# define EWOULDBLOCK WSAEWOULDBLOCK +#endif sout_stream_id_t *id = (sout_stream_id_t *)p_this; unsigned i_caching = id->i_caching; - while( !id->b_die ) + for (;;) { block_t *out = block_FifoGet( id->p_fifo ); - if( out == NULL ) - continue; /* Forced wakeup */ - - mtime_t i_date = out->i_dts + i_caching; - ssize_t len = out->i_buffer; + 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 ) + { + errno = val; + msg_Dbg( id, "SRTP sending error: %m" ); + block_Release( out ); + out = NULL; + } + else + out->i_buffer = len; + } + if (out) +#endif + mwait (out->i_dts + i_caching); + vlc_cleanup_pop (); + if (out == NULL) + continue; - mwait( i_date ); + ssize_t len = out->i_buffer; + int canc = vlc_savecancel (); vlc_mutex_lock( &id->lock_sink ); unsigned deadc = 0; /* How many dead sockets? */ @@ -1386,16 +1604,38 @@ static void ThreadSend( vlc_object_t *p_this ) for( int i = 0; i < id->sinkc; i++ ) { - SendRTCP( id->sinkv[i].rtcp, out ); +#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; - /* Retry sending to root out soft-errors */ - if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 ) - continue; + switch( net_errno ) + { + /* 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 */ + 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; + } 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 ); @@ -1404,20 +1644,34 @@ static void ThreadSend( vlc_object_t *p_this ) msg_Dbg( id, "removing socket %d", deadv[i] ); rtp_del_sink( id, deadv[i] ); } + vlc_restorecancel (canc); + } + return NULL; +} - /* 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 ); - } + +/* 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, 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 ) + +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, @@ -1427,6 +1681,8 @@ int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux ) 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; } @@ -1452,36 +1708,62 @@ void rtp_del_sink( sout_stream_id_t *id, int fd ) 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 a timestamp corresponding to packets being sent now, and that + * can be passed to rtp_compute_ts() to get rtptime values for each ES. */ +int64_t rtp_get_ts( const sout_stream_t *p_stream ) { - sout_stream_sys_t *p_sys = id->p_stream->p_sys; - int i; + 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 ); - 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 ); + if( i_npt_zero == VLC_TS_INVALID ) + return p_sys->i_pts_zero; - return i; -} + mtime_t now = mdate(); + if( now < i_npt_zero ) + return p_sys->i_pts_zero; + return p_sys->i_pts_zero + (now - i_npt_zero); +} 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, p_sys->i_pts_offset ); + id->b_ts_init = true; + } + + uint32_t i_timestamp = rtp_compute_ts( id, i_pts ) + id->i_ts_offset; out->p_buffer[0] = 0x80; out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type; @@ -1512,16 +1794,6 @@ size_t rtp_mtu (const sout_stream_id_t *id) return id->i_mtu - 12; } -/** - * @return number of audio samples to include for a given packetization time - * (this really only makes sense for audio formats). - */ -size_t rtp_plen (const sout_stream_id_t * id, unsigned ptime_ms) -{ - return id->i_channels * (((id->i_clock_rate - 1) * ptime_ms / 1000) + 1); -} - - /***************************************************************************** * Non-RTP mux *****************************************************************************/ @@ -1566,8 +1838,8 @@ static int MuxDel( sout_stream_t *p_stream, sout_stream_id_t *id ) } -static int AccessOutGrabberWriteBuffer( sout_stream_t *p_stream, - const block_t *p_buffer ) +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]; @@ -1575,14 +1847,14 @@ static int AccessOutGrabberWriteBuffer( sout_stream_t *p_stream, int64_t i_dts = p_buffer->i_dts; uint8_t *p_data = p_buffer->p_buffer; - unsigned int i_data = p_buffer->i_buffer; - unsigned int i_max = id->i_mtu - 12; + size_t i_data = p_buffer->i_buffer; + size_t i_max = id->i_mtu - 12; - unsigned i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max; + size_t i_packet = ( p_buffer->i_buffer + i_max - 1 ) / i_max; while( i_data > 0 ) { - unsigned int i_size; + size_t i_size; /* output complete packet */ if( p_sys->packet && @@ -1617,8 +1889,8 @@ static int AccessOutGrabberWriteBuffer( sout_stream_t *p_stream, } -static int AccessOutGrabberWrite( sout_access_out_t *p_access, - block_t *p_buffer ) +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; @@ -1646,7 +1918,6 @@ static sout_access_out_t *GrabberCreate( sout_stream_t *p_stream ) return NULL; p_grab->p_module = NULL; - p_grab->p_sout = p_stream->p_sout; p_grab->psz_access = strdup( "grab" ); p_grab->p_cfg = NULL; p_grab->psz_path = strdup( "" );