X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=modules%2Fstream_out%2Frtp.c;h=fb79a6881801efd34feb9e7338aa63b2e4cc85f1;hb=67398e1a639bc5b4c2769b19048ba5390a42c831;hp=b0f6dd0ee0d76d6201076fa8570e16d820c4cfdd;hpb=d1d3dc1d109110bf68cb048c429f6f05a3839200;p=vlc diff --git a/modules/stream_out/rtp.c b/modules/stream_out/rtp.c index b0f6dd0ee0..fb79a68818 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 +39,7 @@ #include #include #include +#include #include "rtp.h" @@ -60,6 +61,8 @@ #include +#include + /***************************************************************************** * Module descriptor *****************************************************************************/ @@ -73,6 +76,8 @@ "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. " \ @@ -125,6 +130,15 @@ #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", }; @@ -144,8 +158,8 @@ static void Close( vlc_object_t * ); #define MAX_EMPTY_BLOCKS 200 vlc_module_begin(); - set_shortname( _("RTP")); - set_description( _("RTP stream output") ); + set_shortname( N_("RTP")); + set_description( N_("RTP stream output") ); set_capability( "sout stream", 0 ); add_shortcut( "rtp" ); set_category( CAT_SOUT ); @@ -158,6 +172,8 @@ vlc_module_begin(); 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 ); @@ -185,6 +201,11 @@ vlc_module_begin(); 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 ); @@ -194,10 +215,10 @@ 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", - "description", "url", "email", "phone", - "proto", "rtcp-mux", + "sap", "description", "url", "email", "phone", + "proto", "rtcp-mux", "key", "salt", "mp4a-latm", NULL }; @@ -213,7 +234,7 @@ static int MuxSend( sout_stream_t *, sout_stream_id_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 *, 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 ); @@ -264,8 +285,7 @@ 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 { @@ -293,8 +313,9 @@ struct sout_stream_id_t int i_bitrate; /* Packetizer specific fields */ + int i_mtu; + srtp_session_t *srtp; pf_rtp_packetizer_t pf_packetize; - int i_mtu; /* Packets sinks */ vlc_mutex_t lock_sink; @@ -434,8 +455,8 @@ static int Open( vlc_object_t *p_this ) p_stream->p_sys = p_sys; - vlc_mutex_init( p_stream, &p_sys->lock_sdp ); - vlc_mutex_init( p_stream, &p_sys->lock_es ); + 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 ) @@ -496,6 +517,9 @@ static int Open( vlc_object_t *p_this ) 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 ) { @@ -583,7 +607,7 @@ 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; @@ -802,6 +826,22 @@ 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); +} /** Add an ES as a new RTP stream */ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) @@ -877,14 +917,38 @@ 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 ); - vlc_mutex_init( p_stream, &id->lock_sink ); + 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; @@ -958,13 +1022,15 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) 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' ): 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' ): if( p_fmt->audio.i_channels == 1 && p_fmt->audio.i_rate == 44100 ) @@ -977,16 +1043,19 @@ 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; + 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_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' ): @@ -1162,7 +1231,7 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt ) p_sys->psz_destination, p_sys->i_ttl, id->i_port, id->i_port + 1 ); - id->p_fifo = block_FifoNew( p_stream ); + id->p_fifo = block_FifoNew(); if( vlc_thread_create( id, "RTP send thread", ThreadSend, VLC_THREAD_PRIORITY_HIGHEST, false ) ) goto error; @@ -1222,6 +1291,8 @@ static int Del( sout_stream_t *p_stream, sout_stream_id_t *id ) 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 ); @@ -1240,11 +1311,12 @@ 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 ); @@ -1361,33 +1433,33 @@ 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; -#ifdef HAVE_TEE - int fd[5] = { -1, -1, -1, -1, -1 }; - if( pipe( fd ) ) - fd[0] = fd[1] = -1; - else - if( pipe( fd ) ) - fd[2] = fd[3] = -1; - else - fd[4] = open( "/dev/null", O_WRONLY ); -#endif - - while( !id->b_die ) + while( vlc_object_alive (id) ) { block_t *out = block_FifoGet( id->p_fifo ); if( out == NULL ) continue; /* Forced wakeup */ + 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 val = srtp_send( id->srtp, out->p_buffer, &len, len + 10 ); + if( val ) + { + errno = val; + msg_Dbg( id, "SRTP sending error: %m" ); + block_Release( out ); + continue; + } + out->i_buffer = len; + } + mtime_t i_date = out->i_dts + i_caching; ssize_t len = out->i_buffer; -#ifdef HAVE_TEE - if( fd[4] != -1 ) - len = write( fd[1], out->p_buffer, len); - if( len == -1 ) - continue; /* Uho - should not happen */ -#endif mwait( i_date ); vlc_mutex_lock( &id->lock_sink ); @@ -1396,22 +1468,11 @@ static void ThreadSend( vlc_object_t *p_this ) for( int i = 0; i < id->sinkc; i++ ) { - SendRTCP( id->sinkv[i].rtcp, out ); - -#ifdef HAVE_TEE - tee( fd[0], fd[3], len, 0 ); - if( splice( fd[2], NULL, id->sinkv[i].rtp_fd, NULL, len, - SPLICE_F_NONBLOCK ) >= 0 ) - continue; - if( errno == EAGAIN ) - continue; + if( !id->srtp ) /* FIXME: SRTCP support */ + SendRTCP( id->sinkv[i].rtcp, out ); - /* splice failed */ - splice( fd[2], NULL, fd[4], NULL, len, 0 ); -#else if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 ) continue; -#endif /* Retry sending to root out soft-errors */ if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 ) continue; @@ -1419,11 +1480,7 @@ static void ThreadSend( vlc_object_t *p_this ) deadv[deadc++] = id->sinkv[i].rtp_fd; } vlc_mutex_unlock( &id->lock_sink ); - block_Release( out ); -#ifdef HAVE_TEE - splice( fd[0], NULL, fd[4], NULL, len, 0 ); -#endif for( unsigned i = 0; i < deadc; i++ ) { @@ -1441,11 +1498,6 @@ static void ThreadSend( vlc_object_t *p_this ) rtp_add_sink( id, fd, true ); } } - -#ifdef HAVE_TEE - for( unsigned i = 0; i < 5; i++ ) - close( fd[i] ); -#endif } int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux ) @@ -1543,16 +1595,6 @@ 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 *****************************************************************************/ @@ -1597,8 +1639,8 @@ 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]; @@ -1606,14 +1648,14 @@ 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 && @@ -1648,8 +1690,8 @@ 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;