#include <vlc_rand.h>
#ifdef HAVE_SRTP
# include <srtp.h>
+# include <gcrypt.h>
+# include <vlc_gcrypt.h>
#endif
#include "rtp.h"
#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 " \
+ "session will be made available. You must use a 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")
"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 " \
+ "This allows you to give a 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")
"negative value or zero disables timeouts. The default is 60 (one " \
"minute)." )
+#define RTSP_USER_TEXT N_("Username")
+#define RTSP_USER_LONGTEXT N_("User name that will be " \
+ "requested to access the stream." )
+#define RTSP_PASS_TEXT N_("Password")
+#define RTSP_PASS_LONGTEXT N_("Password that will be " \
+ "requested to access the stream." )
+
static int Open ( vlc_object_t * );
static void Close( vlc_object_t * );
RTSP_HOST_LONGTEXT, true )
add_integer( "rtsp-timeout", 60, RTSP_TIMEOUT_TEXT,
RTSP_TIMEOUT_LONGTEXT, true )
+ add_string( "sout-rtsp-user", "",
+ RTSP_USER_TEXT, RTSP_USER_LONGTEXT, true )
+ add_password( "sout-rtsp-pwd", "",
+ RTSP_PASS_TEXT, RTSP_PASS_LONGTEXT, true )
vlc_module_end ()
uint16_t i_port_video;
uint8_t proto;
bool rtcp_mux;
- int i_ttl:9;
bool b_latm;
/* VoD */
sout_stream_t *p_stream;
/* rtp field */
uint16_t i_sequence;
+ bool b_first_packet;
bool b_ts_init;
uint32_t i_ts_offset;
uint8_t ssrc[4];
return VLC_EGENERIC;
}
- p_sys->i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
- if( p_sys->i_ttl == -1 )
+ int i_ttl = var_GetInteger( p_stream, SOUT_CFG_PREFIX "ttl" );
+ if( i_ttl != -1 )
{
- /* Normally, we should let the default hop limit up to the core,
- * but we have to know it to write our RTSP headers properly,
- * which is why we ask the core. FIXME: broken when neither
- * sout-rtp-ttl nor ttl are set. */
- p_sys->i_ttl = var_InheritInteger( p_stream, "ttl" );
+ var_Create( p_stream, "ttl", VLC_VAR_INTEGER );
+ var_SetInteger( p_stream, "ttl", i_ttl );
}
p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
goto out;
}
- p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), NULL, &url );
+ if( url.psz_host != NULL && *url.psz_host )
+ {
+ /* msg_Err( p_stream, "\"%s\" RTSP host ignored", url.psz_host );
+ msg_Info( p_stream, "Pass --rtsp-host=%s on the command line "
+ "instead.", url.psz_host ); */
+
+ var_Create( p_stream, "rtsp-host", VLC_VAR_STRING );
+ var_SetString( p_stream, "rtsp-host", url.psz_host );
+ }
+ /* if( url.i_port != 0 )
+ {
+ msg_Err( p_stream, "\"%u\" RTSP port ignored", url.i_port );
+ msg_Info( p_stream, "Pass --rtsp-port=%u on the command line "
+ "instead.", url.i_port );
+ } */
+
+ if( url.i_port <= 0 ) url.i_port = 554;
+ var_Create( p_stream, "rtsp-port", VLC_VAR_INTEGER );
+ var_SetInteger( p_stream, "rtsp-port", url.i_port );
+
+ p_sys->rtsp = RtspSetup( VLC_OBJECT(p_stream), NULL, url.psz_path );
if( p_sys->rtsp == NULL )
msg_Err( p_stream, "cannot export SDP as RTSP" );
}
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] ) );
+ "SC:RTP%c",
+ toupper( (unsigned char)mime_major[0] ) );
}
}
out:
id->p_fifo = NULL;
id->listen.fd = NULL;
+ id->b_first_packet = true;
id->i_caching =
(int64_t)1000 * var_GetInteger( p_stream, SOUT_CFG_PREFIX "caching");
id->i_sequence = id->i_seq_sent_next;
}
/* vod_init_id() may fail either because the ES wasn't found in
- * the VoD media, or because that track wasn't SETUP. In the
+ * the VoD media, or because the RTSP session is gone. In the
* former case, id->rtp_fmt was left untouched. */
format = (id->rtp_fmt.ptname != NULL);
}
}
#ifdef HAVE_SRTP
- char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
+ char *key = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
if (key)
{
+ vlc_gcrypt_init ();
id->srtp = srtp_create (SRTP_ENCR_AES_CM, SRTP_AUTH_HMAC_SHA1, 10,
SRTP_PRF_AES_CM, SRTP_RCC_MODE1);
if (id->srtp == NULL)
goto error;
}
- char *salt = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
+ char *salt = var_GetNonEmptyString (p_stream, SOUT_CFG_PREFIX"salt");
errno = srtp_setkeystring (id->srtp, key, salt ? salt : "");
free (salt);
free (key);
id->i_seq_sent_next = id->i_sequence;
+ int mcast_fd = -1;
if( p_sys->psz_destination != NULL )
{
/* Choose the port */
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 );
+ i_port, -1, p_sys->proto );
if( fd == -1 )
{
msg_Err( p_stream, "cannot create RTP socket" );
setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &(int){ 0 },
sizeof (int));
rtp_add_sink( id, fd, p_sys->rtcp_mux, NULL );
+ /* FIXME: test if this is multicast */
+ mcast_fd = fd;
}
}
}
p_sys->i_pts_offset );
if( p_sys->rtsp != NULL )
- id->rtsp_id = RtspAddId( p_sys->rtsp, id,
- GetDWBE( id->ssrc ),
- id->rtp_fmt.clock_rate,
- p_sys->psz_destination,
- p_sys->i_ttl, id->i_port, id->i_port + 1 );
+ id->rtsp_id = RtspAddId( p_sys->rtsp, id, GetDWBE( id->ssrc ),
+ id->rtp_fmt.clock_rate, mcast_fd );
id->p_fifo = block_FifoNew();
if( unlikely(id->p_fifo == NULL) )
/* Send a Vorbis/Theora Packed Configuration packet (RFC 5215 ยง3.1)
* as the first packet of the stream */
- if (id->i_sequence == id->i_seq_sent_next
- && (!strcmp(id->rtp_fmt.ptname, "vorbis")
- || !strcmp(id->rtp_fmt.ptname, "theora")))
+ if (id->b_first_packet)
+ {
+ id->b_first_packet = false;
+ if (!strcmp(id->rtp_fmt.ptname, "vorbis") ||
+ !strcmp(id->rtp_fmt.ptname, "theora"))
rtp_packetize_xiph_config(id, id->rtp_fmt.fmtp,
p_buffer->i_pts);
+ }
if( id->rtp_fmt.pf_packetize( id, p_buffer ) )
break;
}
if( p_sys->i_es > 0 && 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 );
- }
+ p_sys->psz_destination );
return VLC_SUCCESS;
}
{
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 );
+ p_sys->p_httpd_host = vlc_http_HostNew( VLC_OBJECT(p_stream) );
if( p_sys->p_httpd_host )
{
p_sys->p_httpd_file = httpd_FileNew( p_sys->p_httpd_host,
out->i_buffer = len;
}
if (out)
-#endif
mwait (out->i_dts + i_caching);
vlc_cleanup_pop ();
if (out == NULL)
continue;
+#else
+ mwait (out->i_dts + i_caching);
+ vlc_cleanup_pop ();
+#endif
ssize_t len = out->i_buffer;
int canc = vlc_savecancel ();
#endif
SendRTCP( id->sinkv[i].rtcp, out );
- if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) >= 0 )
- continue;
- switch( net_errno )
+ if( send( id->sinkv[i].rtp_fd, out->p_buffer, len, 0 ) == -1
+ && net_errno != EAGAIN && net_errno != EWOULDBLOCK
+ && net_errno != ENOBUFS && net_errno != ENOMEM )
{
- /* 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 */
+ int type;
+ getsockopt( id->sinkv[i].rtp_fd, SOL_SOCKET, SO_TYPE,
+ &type, &(socklen_t){ sizeof(type) });
+ if( type == SOCK_DGRAM )
+ /* ICMP soft error: ignore and retry */
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;
+ else
+ /* Broken connection */
+ deadv[deadc++] = id->sinkv[i].rtp_fd;
}
-
- 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 );
uint64_t i_ts_init;
/* As per RFC 2326, session identifiers are at least 8 bytes long */
strncpy((char *)&i_ts_init, psz_vod_session, sizeof(uint64_t));
- i_ts_init ^= (uint64_t) p_media;
+ i_ts_init ^= (uintptr_t)p_media;
/* Limit the timestamp to 48 bytes, this is enough and allows us
* to stay away from overflows */
i_ts_init &= 0xFFFFFFFFFFFF;
/* 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.
- * If the stream output is not started, the initial timestamp that will
- * be used with the first packets is returned instead. */
+ * Also return the NPT corresponding to this timestamp. If the stream
+ * output is not started, the initial timestamp that will be used with
+ * the first packets for NPT=0 is returned instead. */
int64_t rtp_get_ts( const sout_stream_t *p_stream, const sout_stream_id_t *id,
- const vod_media_t *p_media, const char *psz_vod_session )
+ const vod_media_t *p_media, const char *psz_vod_session,
+ int64_t *p_npt )
{
+ if (p_npt != NULL)
+ *p_npt = 0;
+
if (id != NULL)
p_stream = id->p_stream;
if( now < i_npt_zero )
return p_sys->i_pts_zero;
- return p_sys->i_pts_zero + (now - i_npt_zero);
+ int64_t npt = now - i_npt_zero;
+ if (p_npt != NULL)
+ *p_npt = npt;
+
+ return p_sys->i_pts_zero + npt;
}
void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
{
sout_access_out_t *p_grab;
- p_grab = vlc_object_create( p_stream->p_sout, sizeof( *p_grab ) );
+ p_grab = vlc_object_create( p_stream, sizeof( *p_grab ) );
if( p_grab == NULL )
return NULL;
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;
}
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