* 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 <vlc_rand.h>
+#include <srtp.h>
#include "rtp.h"
#include <errno.h>
+#include <assert.h>
+
/*****************************************************************************
* Module descriptor
*****************************************************************************/
#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")
#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 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 );
+ 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 );
+ 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_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 );
+ 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", "key", "salt",
"mp4a-latm", NULL
};
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 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 );
/* */
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;
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_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 )
}
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
p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
- p_sys->i_payload_type = 96;
+ p_sys->payload_bitmap = 0;
p_sys->i_es = 0;
p_sys->es = NULL;
p_sys->rtsp = NULL;
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 )
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;
}
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;
}
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;
}
/*****************************************************************************
* 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;
/* 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->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 );
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] ) );
}
}
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 )
sout_stream_id_t *id;
int i_port, cscov = -1;
char *psz_sdp;
+ int i_port_audio_option = var_GetInteger( p_stream, "port-audio" );
+ int i_port_video_option = var_GetInteger( p_stream, "port-video" );
+
+ 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 )
while( i_port == 0 )
{
- if( p_sys->i_port != p_sys->i_port_audio
- && p_sys->i_port != p_sys->i_port_video )
+ if( p_sys->i_port != i_port_audio_option
+ && p_sys->i_port != i_port_video_option )
{
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;
- 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->psz_enc = NULL;
id->psz_fmtp = NULL;
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;
{
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);
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 )
{
id->psz_enc = "MP2P";
}
+ free( psz );
}
else
switch( p_fmt->i_codec )
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' ):
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;
id->pf_packetize = rtp_packetize_amr;
break;
case VLC_FOURCC( 's', 'p', 'x', ' ' ):
- id->i_payload_type = p_sys->i_payload_type++;
id->psz_enc = "SPEEX";
id->pf_packetize = rtp_packetize_spx;
break;
"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 */
if( cscov != -1 )
cscov += 8 /* UDP */ + 12 /* RTP */;
if( id->sinkc > 0 )
net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
-
- if( id->i_payload_type == p_sys->i_payload_type )
- p_sys->i_payload_type++;
+#endif
if( p_sys->rtsp != NULL )
id->rtsp_id = RtspAddId( p_sys->rtsp, id, p_sys->i_es,
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 */
if( id->p_fifo != NULL )
{
vlc_object_kill( id );
- block_FifoWake( id->p_fifo );
vlc_thread_join( id );
block_FifoRelease( id->p_fifo );
}
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 );
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 );
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 );
/****************************************************************************
* RTP send
****************************************************************************/
-static void ThreadSend( vlc_object_t *p_this )
+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;
- while( !id->b_die )
+ for (;;)
{
block_t *out = block_FifoGet( id->p_fifo );
- if( out == NULL )
- continue; /* Forced wakeup */
+ 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;
+ }
- mtime_t i_date = out->i_dts + i_caching;
- ssize_t len = out->i_buffer;
+ if (out)
+ 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? */
for( int i = 0; i < id->sinkc; i++ )
{
- SendRTCP( id->sinkv[i].rtcp, out );
+ 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;
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 )
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;
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( "" );
projects / vlc / blobdiff