#include <vlc_charset.h>
#include <vlc_strings.h>
#include <vlc_rand.h>
-#include <srtp.h>
+#ifdef HAVE_SRTP
+# include <srtp.h>
+#endif
#include "rtp.h"
add_bool( SOUT_CFG_PREFIX "rtcp-mux", false, NULL,
RTCP_MUX_TEXT, RTCP_MUX_LONGTEXT, false )
+#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,
+ add_bool( SOUT_CFG_PREFIX "mp4a-latm", false, NULL, RFC3016_TEXT,
RFC3016_LONGTEXT, false )
set_callbacks( Open, Close )
static sout_access_out_t *GrabberCreate( sout_stream_t *p_sout );
static void* ThreadSend( vlc_object_t *p_this );
+static void *rtp_listen_thread( void * );
static void SDPHandleUrl( sout_stream_t *, const char * );
vlc_mutex_t lock_sdp;
/* SDP to disk */
- bool b_export_sdp_file;
char *psz_sdp_file;
/* SDP via SAP */
sout_stream_t *p_stream;
/* rtp field */
+ uint32_t i_timestamp;
uint16_t i_sequence;
uint8_t i_payload_type;
uint8_t ssrc[4];
/* Packetizer specific fields */
int i_mtu;
+#ifdef HAVE_SRTP
srtp_session_t *srtp;
+#endif
pf_rtp_packetizer_t pf_packetize;
/* Packets sinks */
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;
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->psz_sdp_file = NULL;
-
if( p_sys->i_port_audio && p_sys->i_port_video == p_sys->i_port_audio )
{
msg_Err( p_stream, "audio and video RTP port must be distinct" );
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;
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 );
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 );
}
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
{
/* 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,
}
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",
* 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;
- 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)
{
return NULL;
}
- id = vlc_object_create( p_stream, sizeof( sout_stream_id_t ) );
- if( id == NULL )
- 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 != i_port_audio_option
- && p_sys->i_port != i_port_video_option )
+ if (p == 0)
{
- i_port = p_sys->i_port;
+ 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_timestamp = 0; /* It will be filled when the first packet is sent */
+
/* Look for free dymanic payload type */
id->i_payload_type = 96;
while (p_sys->payload_bitmap & (1 << (id->i_payload_type - 96)))
id->i_mtu = 576 - 20 - 8; /* pessimistic */
msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
- id->srtp = NULL;
id->pf_packetize = NULL;
+#ifdef HAVE_SRTP
+ id->srtp = NULL;
+
char *key = var_CreateGetNonEmptyString (p_stream, SOUT_CFG_PREFIX"key");
if (key)
{
}
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");
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:
rtp_set_ptime (id, 20, 1);
break;
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;
id->i_payload_type = 10;
}
id->psz_enc = "L16";
- id->pf_packetize = rtp_packetize_split;
+ 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_CODEC_U8:
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:
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)
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 )
/* 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;
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));
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 );
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)",
block_t *out = block_FifoGet( id->p_fifo );
block_cleanup_push (out);
+#ifdef HAVE_SRTP
if( id->srtp )
{ /* FIXME: this is awfully inefficient */
size_t len = out->i_buffer;
else
out->i_buffer = len;
}
-
if (out)
+#endif
mwait (out->i_dts + i_caching);
vlc_cleanup_pop ();
if (out == NULL)
for( int i = 0; i < id->sinkc; i++ )
{
+#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 )
msg_Dbg( id, "removing socket %d", deadv[i] );
rtp_del_sink( id, deadv[i] );
}
-
- /* 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 );
- }
vlc_restorecancel (canc);
}
return NULL;
}
+
+/* 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 );
+ vlc_restorecancel( canc );
+ }
+
+ assert( 0 );
+}
+
+
int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux )
{
rtp_sink_t sink = { fd, NULL };
return id->i_sequence;
}
+uint32_t rtp_get_ts( const sout_stream_id_t *id )
+{
+ /* ... and this will return the value for the last packet.
+ * Lame, but close enough. */
+ return id->i_timestamp;
+}
+
/* 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 )
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);
+ uint32_t i_timestamp = i_pts * (int64_t)id->i_clock_rate / CLOCK_FREQ;
out->p_buffer[0] = 0x80;
out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type;
memcpy( out->p_buffer + 8, id->ssrc, 4 );
out->i_buffer = 12;
+ id->i_timestamp = i_timestamp;
id->i_sequence++;
}
projects / vlc / blobdiff