* 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 <fenrir@via.ecp.fr>
*
# include "config.h"
#endif
-#include <vlc/vlc.h>
+#include <vlc_common.h>
+#include <vlc_plugin.h>
#include <vlc_sout.h>
#include <vlc_block.h>
#include <vlc_network.h>
#include <vlc_charset.h>
#include <vlc_strings.h>
+#include <srtp.h>
#include "rtp.h"
#include <errno.h>
+#include <assert.h>
+
/*****************************************************************************
* Module descriptor
*****************************************************************************/
"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. " \
#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",
};
#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 );
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_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 );
/*****************************************************************************
* 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
};
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 );
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
{
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;
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 )
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 )
{
/*****************************************************************************
* 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;
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 )
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;
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 )
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' ):
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;
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 );
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 );
{
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 );
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;
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++ )
{
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 )
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 / I64C(1000000);
+ 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;
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
*****************************************************************************/
}
-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];
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 &&
}
-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;
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