*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
+#include "config.h"
+#include <stdio.h>
#include <stdlib.h>
+#ifdef HAVE_UNISTD_H
+# include <unistd.h>
+#endif
#include <errno.h>
#include <vlc/sout.h>
#include "vlc_httpd.h"
+#include "vlc_url.h"
#include "network.h"
+#include "charset.h"
/*****************************************************************************
* Module descriptor
*****************************************************************************/
+
+#define MTU_REDUCE 50
+
#define DST_TEXT N_("Destination")
#define DST_LONGTEXT N_( \
- "Allows you to specify the output URL used for the streaming output." )
+ "This is the output URL that will be used." )
#define SDP_TEXT N_("SDP")
#define SDP_LONGTEXT N_( \
- "Allows you to specify the SDP used for the streaming output. " \
- "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." )
+ "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 " \
+ "access the SDP via HTTP, rtsp://location for RTSP access, and sap:// " \
+ "for the SDP to be announced via SAP." )
#define MUX_TEXT N_("Muxer")
#define MUX_LONGTEXT N_( \
- "Allows you to specify the muxer used for the streaming output." )
+ "This allows you to specify the muxer used for the streaming output. " \
+ "Default is to use no muxer (standard RTP stream)." )
#define NAME_TEXT N_("Session name")
#define NAME_LONGTEXT N_( \
- "Allows you to specify the session name used for the streaming output." )
+ "This is the name of the session that will be announced in the SDP " \
+ "(Session Descriptor)." )
#define DESC_TEXT N_("Session description")
#define DESC_LONGTEXT N_( \
- "Allows you to give a broader description of the stream." )
+ "This allows you to give a broader description of the stream, that will " \
+ "be announced in the SDP (Session Descriptor)." )
#define URL_TEXT N_("Session URL")
#define URL_LONGTEXT N_( \
- "Allows you to specify a URL with additional information on the stream." )
+ "This allows you to give an 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")
#define EMAIL_LONGTEXT N_( \
- "Allows you to specify contact e-mail address for this session." )
-
+ "This allows you to give a contact mail address for the stream, that will " \
+ "be announced in the SDP (Session Descriptor)." )
#define PORT_TEXT N_("Port")
#define PORT_LONGTEXT N_( \
- "Allows you to specify the base port used for the RTP streaming." )
+ "This allows you to specify the base port for the RTP streaming." )
#define PORT_AUDIO_TEXT N_("Audio port")
#define PORT_AUDIO_LONGTEXT N_( \
- "Allows you to specify the default audio port used for the RTP streaming." )
+ "This allows you to specify the default audio port for the RTP streaming." )
#define PORT_VIDEO_TEXT N_("Video port")
#define PORT_VIDEO_LONGTEXT N_( \
- "Allows you to specify the default video port used for the RTP streaming." )
+ "This allows you to specify the default video port for the RTP streaming." )
-#define TTL_TEXT N_("Time To Live")
+#define TTL_TEXT N_("Time-To-Live (TTL)")
#define TTL_LONGTEXT N_( \
- "Allows you to specify the time to live for the output stream." )
+ "This allows you to specify the Time-To-Live for the output stream." )
+
+#define RFC3016_TEXT N_("MP4A LATM")
+#define RFC3016_LONGTEXT N_( \
+ "This allows you to stream MPEG4 LATM audio streams (see RFC3016)." )
static int Open ( vlc_object_t * );
static void Close( vlc_object_t * );
add_integer( SOUT_CFG_PREFIX "port-video", 1232, NULL, PORT_VIDEO_TEXT,
PORT_VIDEO_LONGTEXT, VLC_TRUE );
- add_integer( SOUT_CFG_PREFIX "ttl", 1, NULL, TTL_TEXT,
+ add_integer( SOUT_CFG_PREFIX "ttl", 0, NULL, TTL_TEXT,
TTL_LONGTEXT, VLC_TRUE );
+ add_bool( SOUT_CFG_PREFIX "mp4a-latm", 0, NULL, RFC3016_TEXT,
+ RFC3016_LONGTEXT, VLC_FALSE );
+
set_callbacks( Open, Close );
vlc_module_end();
*****************************************************************************/
static const char *ppsz_sout_options[] = {
"dst", "name", "port", "port-audio", "port-video", "*sdp", "ttl", "mux",
- "description", "url","email", NULL
+ "description", "url","email", "mp4a-latm", NULL
};
static sout_stream_id_t *Add ( sout_stream_t *, es_format_t * );
int i_port_audio;
int i_port_video;
int i_ttl;
+ vlc_bool_t b_latm;
/* when need to use a private one or when using muxer */
int i_payload_type;
char *psz_destination;
int i_port;
int i_cat;
+ int i_bitrate;
/* Packetizer specific fields */
pf_rtp_packetizer_t pf_packetize;
{
sout_stream_t *p_stream = (sout_stream_t*)p_this;
sout_instance_t *p_sout = p_stream->p_sout;
- sout_stream_sys_t *p_sys;
+ sout_stream_sys_t *p_sys = NULL;
+ config_chain_t *p_cfg = NULL;
vlc_value_t val;
+ vlc_bool_t b_rtsp = VLC_FALSE;
- sout_CfgParse( p_stream, SOUT_CFG_PREFIX, ppsz_sout_options, p_stream->p_cfg );
+ config_ChainParse( p_stream, SOUT_CFG_PREFIX,
+ ppsz_sout_options, p_stream->p_cfg );
p_sys = malloc( sizeof( sout_stream_sys_t ) );
p_sys->psz_session_name = strdup( "NONE" );
}
- if( !p_sys->psz_destination || *p_sys->psz_destination == '\0' )
+ for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
{
- sout_cfg_t *p_cfg;
- vlc_bool_t b_ok = VLC_FALSE;
-
- for( p_cfg = p_stream->p_cfg; p_cfg != NULL; p_cfg = p_cfg->p_next )
+ if( !strcmp( p_cfg->psz_name, "sdp" ) )
{
- if( !strcmp( p_cfg->psz_name, "sdp" ) )
+ if( p_cfg->psz_value && !strncasecmp( p_cfg->psz_value, "rtsp", 4 ) )
{
- if( p_cfg->psz_value && !strncasecmp( p_cfg->psz_value, "rtsp", 4 ) )
- {
- b_ok = VLC_TRUE;
- break;
- }
+ b_rtsp = VLC_TRUE;
+ break;
}
}
- if( !b_ok )
- {
- vlc_value_t val2;
- var_Get( p_stream, SOUT_CFG_PREFIX "sdp", &val2 );
- if( !strncasecmp( val2.psz_string, "rtsp", 4 ) )
- b_ok = VLC_TRUE;
- free( val2.psz_string );
- }
+ }
+ if( !b_rtsp )
+ {
+ vlc_value_t val2;
+ var_Get( p_stream, SOUT_CFG_PREFIX "sdp", &val2 );
+ if( !strncasecmp( val2.psz_string, "rtsp", 4 ) )
+ b_rtsp = VLC_TRUE;
+ free( val2.psz_string );
+ }
- if( !b_ok )
+ if( !p_sys->psz_destination || *p_sys->psz_destination == '\0' )
+ {
+ if( !b_rtsp )
{
msg_Err( p_stream, "missing destination and not in rtsp mode" );
free( p_sys );
}
var_Get( p_stream, SOUT_CFG_PREFIX "ttl", &val );
- if( ( val.i_int > 255 ) || ( val.i_int < 0 ) )
+ if( val.i_int == 0 )
+ {
+ /* 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
+ * we ask the core. FIXME: broken when neither sout-rtp-ttl nor
+ * ttl are set. */
+ val.i_int = config_GetInt( p_stream, "ttl" );
+ }
+ if( val.i_int > 255 ) val.i_int = 255;
+ /* must not exceed 999 once formatted */
+
+ if( val.i_int < 0 )
{
msg_Err( p_stream, "illegal TTL %d", val.i_int );
free( p_sys );
}
p_sys->i_ttl = val.i_int;
+
+ var_Get( p_stream, SOUT_CFG_PREFIX "mp4a-latm", &val );
+ p_sys->b_latm = val.b_bool;
+
p_sys->i_payload_type = 96;
p_sys->i_es = 0;
p_sys->es = NULL;
sout_access_out_t *p_grab;
char *psz_rtpmap, url[NI_MAXHOST + 8], access[17], psz_ttl[5], ipv;
- if( !p_sys->psz_destination || *p_sys->psz_destination == '\0' )
+ if( b_rtsp )
+ {
+ msg_Err( p_stream, "muxing is not supported in RTSP mode" );
+ free( p_sys );
+ return VLC_EGENERIC;
+ }
+ else if( !p_sys->psz_destination || *p_sys->psz_destination == '\0' )
{
msg_Err( p_stream, "rtp needs a destination when muxing" );
free( p_sys );
url[sizeof( url ) - 1] = '\0';
/* FIXME: we should check that url is a numerical address, otherwise
* the SDP will be quite broken (regardless of the IP protocol version)
+ * Also it might be IPv6 with no ':' if it is a DNS name.
*/
ipv = ( strchr( p_sys->psz_destination, ':' ) != NULL ) ? '6' : '4';
return VLC_EGENERIC;
}
p_sys->i_mtu = config_GetInt( p_stream, "mtu" ); /* XXX beurk */
- if( p_sys->i_mtu <= 16 )
+ if( p_sys->i_mtu <= 16 + MTU_REDUCE )
{
/* better than nothing */
p_sys->i_mtu = 1500;
}
+ p_sys->i_mtu -= MTU_REDUCE;
/* the access out grabber TODO export it as sout_AccessOutGrabberNew */
p_grab = p_sys->p_grab =
}
/* create the SDP for a muxed stream (only once) */
- /* FIXME http://www.faqs.org/rfcs/rfc2327.html
- All text fields should be UTF-8 encoded. Use global a:charset to announce this.
+ /* FIXME http://www.faqs.org/rfcs/rfc4566.html
o= - should be local username (no spaces allowed)
- o= time should be hashed with some other value to garantue uniqueness
- o= we need IP6 support?
+ o= time should be hashed with some other value to garantee uniqueness
o= don't use the localhost address. use fully qualified domain name or IP4 address
- p= international phone number (pass via vars?)
- c= IP6 support
- a= recvonly (missing)
- a= type:broadcast (missing)
- a= charset: (normally charset should be UTF-8, this can be used to override s= and i=)
+ a= source-filter: we need our source address
a= x-plgroup: (missing)
RTP packets need to get the correct src IP address */
if( net_AddressIsMulticast( (vlc_object_t *)p_stream, p_sys->psz_destination ) )
"i=%s\r\n"
"u=%s\r\n"
"e=%s\r\n"
+ "c=IN IP%c %s%s\r\n"
"t=0 0\r\n" /* permanent stream */ /* when scheduled from vlm, we should set this info correctly */
"a=tool:"PACKAGE_STRING"\r\n"
- "c=IN IP%c %s%s\r\n"
+ "a=recvonly\r\n"
+ "a=type:broadcast\r\n"
"m=video %d RTP/AVP %d\r\n"
"a=rtpmap:%d %s\r\n",
p_sys->i_sdp_id, p_sys->i_sdp_version,
ipv, p_sys->psz_destination, psz_ttl,
p_sys->i_port, p_sys->i_payload_type,
p_sys->i_payload_type, psz_rtpmap );
- fprintf( stderr, "sdp=%s", p_sys->psz_sdp );
+ msg_Dbg( p_stream, "sdp=%s", p_sys->psz_sdp );
/* create the rtp context */
p_sys->ssrc[0] = rand()&0xff;
var_Get( p_stream, SOUT_CFG_PREFIX "sdp", &val );
if( *val.psz_string )
{
- sout_cfg_t *p_cfg;
+ config_chain_t *p_cfg;
SDPHandleUrl( p_stream, val.psz_string );
{
block_Release( p_sys->packet );
}
+ if( p_sys->b_export_sap )
+ {
+ p_sys->p_mux = NULL;
+ SapSetup( p_stream );
+ }
}
while( p_sys->i_rtsp > 0 )
{
httpd_HostDelete( p_sys->p_rtsp_host );
}
-#if 0
- /* why? is this disabled? */
if( p_sys->psz_session_name )
{
free( p_sys->psz_session_name );
free( p_sys->psz_session_email );
p_sys->psz_session_email = NULL;
}
-#endif
if( p_sys->psz_sdp )
{
free( p_sys->psz_sdp );
+ p_sys->psz_sdp = NULL;
+ }
+ if( p_sys->b_export_sdp_file )
+ {
+#ifdef HAVE_UNISTD_H
+ unlink( p_sys->psz_sdp_file );
+#endif
+ free( p_sys->psz_sdp_file );
}
free( p_sys );
}
{
if( p_sys->p_httpd_file )
{
- msg_Err( p_stream, "You can used sdp=http:// only once" );
+ msg_Err( p_stream, "you can use sdp=http:// only once" );
return;
}
{
if( p_sys->p_rtsp_url )
{
- msg_Err( p_stream, "You can used sdp=rtsp:// only once" );
+ msg_Err( p_stream, "you can use sdp=rtsp:// only once" );
return;
}
{
if( p_sys->b_export_sdp_file )
{
- msg_Err( p_stream, "You can used sdp=file:// only once" );
+ msg_Err( p_stream, "you can use sdp=file:// only once" );
return;
}
p_sys->b_export_sdp_file = VLC_TRUE;
{
i_size += strlen( "a=fmtp:* *\r\n" ) + strlen( id->psz_fmtp ) + 10;
}
+ if ( id->i_bitrate)
+ {
+ i_size += strlen( "b=AS: *\r\n") + 10;
+ }
if( b_rtsp )
{
- i_size += strlen( "a=control:*/trackid=*\r\n" ) + strlen( p_sys->psz_rtsp_control ) + 10;
+ i_size += strlen( "a=control:*/trackID=*\r\n" ) + strlen( p_sys->psz_rtsp_control ) + 10;
}
}
+ if( p_sys->p_mux )
+ {
+ i_size += strlen( "m=video %d RTP/AVP %d\r\n" ) +10 +10;
+ }
ipv = ( strchr( psz_destination, ':' ) != NULL ) ? '6' : '4';
if( net_AddressIsMulticast( (vlc_object_t *)p_stream, psz_destination ) )
{
/* Add the ttl if it is a multicast address */
- p += sprintf( p, "/%d\r\n", p_sys->i_ttl );
+ /* FIXME: 1 is not a correct default value in the case of IPv6 */
+ p += sprintf( p, "/%d\r\n", p_sys->i_ttl ?: 1 );
}
else
{
p += sprintf( p, "a=fmtp:%d %s\r\n", id->i_payload_type,
id->psz_fmtp );
}
+ if ( id->i_bitrate)
+ {
+ p += sprintf(p,"b=AS:%d\r\n",id->i_bitrate);
+ }
if( b_rtsp )
{
- p += sprintf( p, "a=control:%s/trackid=%d\r\n", p_sys->psz_rtsp_control, i );
+ p += sprintf( p, "a=control:trackID=%d\r\n", i );
}
}
+ if( p_sys->p_mux )
+ {
+ p += sprintf( p, "m=video %d RTP/AVP %d\r\n",
+ p_sys->i_port, p_sys->i_payload_type );
+ }
return psz_sdp;
}
static int rtp_packetize_ac3 ( sout_stream_t *, sout_stream_id_t *, block_t * );
static int rtp_packetize_split( sout_stream_t *, sout_stream_id_t *, block_t * );
static int rtp_packetize_mp4a ( sout_stream_t *, sout_stream_id_t *, block_t * );
+static int rtp_packetize_mp4a_latm ( sout_stream_t *, sout_stream_id_t *, block_t * );
static int rtp_packetize_h263 ( sout_stream_t *, sout_stream_id_t *, block_t * );
+static int rtp_packetize_h264 ( sout_stream_t *, sout_stream_id_t *, block_t * );
static int rtp_packetize_amr ( sout_stream_t *, sout_stream_id_t *, block_t * );
static void sprintf_hexa( char *s, uint8_t *p_data, int i_data )
s[2*i_data] = '\0';
}
+static const char basis_64[] =
+ "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789+/";
+
+int ap_base64encode_len(int len)
+{
+ return ((len + 2) / 3 * 4) + 1;
+}
+
+int ap_base64encode_binary(char *encoded,
+ const unsigned char *string, int len)
+{
+ int i;
+ char *p;
+
+ p = encoded;
+ for (i = 0; i < len - 2; i += 3) {
+ *p++ = basis_64[(string[i] >> 2) & 0x3F];
+ *p++ = basis_64[((string[i] & 0x3) << 4) |
+ ((int) (string[i + 1] & 0xF0) >> 4)];
+ *p++ = basis_64[((string[i + 1] & 0xF) << 2) |
+ ((int) (string[i + 2] & 0xC0) >> 6)];
+ *p++ = basis_64[string[i + 2] & 0x3F];
+ }
+ if (i < len) {
+ *p++ = basis_64[(string[i] >> 2) & 0x3F];
+ if (i == (len - 1)) {
+ *p++ = basis_64[((string[i] & 0x3) << 4)];
+ *p++ = '=';
+ }
+ else {
+ *p++ = basis_64[((string[i] & 0x3) << 4) |
+ ((int) (string[i + 1] & 0xF0) >> 4)];
+ *p++ = basis_64[((string[i + 1] & 0xF) << 2)];
+ }
+ *p++ = '=';
+ }
+
+ *p++ = '\0';
+ return p - encoded;
+}
+
+int ap_base64encode(char *encoded, const char *string, int len)
+{
+ return ap_base64encode_binary(encoded, (const unsigned char *) string, len);
+}
+
+char *b64_encode(char *buf, int len)
+{
+ int elen;
+ char *out;
+
+ if(len == 0)
+ len = strlen(buf);
+
+ elen = ap_base64encode_len(len);
+ out = (char *) malloc(sizeof(char) * (elen + 1));
+
+ ap_base64encode(out, buf, len);
+
+ return out;
+}
+
static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
{
sout_instance_t *p_sout = p_stream->p_sout;
char url[NI_MAXHOST + 8];
/* first try to create the access out */
- if( p_sys->i_ttl > 0 )
+ if( p_sys->i_ttl )
{
snprintf( access, sizeof( access ), "udp{raw,ttl=%d}",
p_sys->i_ttl );
id->psz_rtpmap = strdup( "H263-1998/90000" );
id->pf_packetize = rtp_packetize_h263;
break;
+ case VLC_FOURCC( 'h', '2', '6', '4' ):
+ id->i_payload_type = p_sys->i_payload_type++;
+ id->i_clock_rate = 90000;
+ id->psz_rtpmap = strdup( "H264/90000" );
+ id->pf_packetize = rtp_packetize_h264;
+ if( p_fmt->i_extra > 0 )
+ {
+ uint8_t *p_buffer = p_fmt->p_extra;
+ int i_buffer = p_fmt->i_extra;
+ char *p_64_sps = NULL;
+ char *p_64_pps = NULL;
+ char hexa[6];
+
+ while( i_buffer > 4 &&
+ p_buffer[0] == 0 && p_buffer[1] == 0 &&
+ p_buffer[2] == 0 && p_buffer[3] == 1 )
+ {
+ const int i_nal_type = p_buffer[4]&0x1f;
+ int i_offset = 1;
+ int i_size = 0;
+ int i_startcode = 0;
+ int i_encoded = 0;
+
+ msg_Dbg( p_stream, "we found a startcode for NAL with TYPE:%d", i_nal_type );
+
+ for( i_offset = 1; i_offset+3 < i_buffer ; i_offset++)
+ {
+ if( p_buffer[i_offset] == 0 && p_buffer[i_offset+1] == 0 && p_buffer[i_offset+2] == 0 && p_buffer[i_offset+3] == 1 )
+ {
+ /* we found another startcode */
+ i_startcode = i_offset;
+ break;
+ }
+ }
+ i_size = i_startcode ? i_startcode : i_buffer;
+ if( i_nal_type == 7 )
+ {
+ p_64_sps = (char *)malloc( ap_base64encode_len( i_size - 4) );
+ i_encoded = ap_base64encode_binary( p_64_sps, &p_buffer[4], i_size - 4 );
+ p_64_sps[i_encoded] = '\0';
+ sprintf_hexa( hexa, &p_buffer[5], 3 );
+ hexa[6] = '\0';
+ }
+ if( i_nal_type == 8 )
+ {
+ p_64_pps = (char *)malloc( ap_base64encode_len( i_size - 4) );
+ i_encoded = ap_base64encode_binary( p_64_pps, &p_buffer[4], i_size - 4 );
+ p_64_pps[i_encoded] = '\0';
+ }
+ i_buffer -= i_size;
+ p_buffer += i_size;
+ }
+ /* FIXME: All this is a bit unsafe */
+ asprintf( &id->psz_fmtp, "packetization-mode=1;profile-level-id=%s;sprop-parameter-sets=%s,%s;", hexa, p_64_sps, p_64_pps );
+ free( p_64_sps );
+ free( p_64_pps );
+ }
+ else
+ id->psz_fmtp = strdup( "packetization-mode=1" );
+if( p_fmt->i_extra > 0 )
+msg_Dbg( p_stream, "WE HAVE %d bytes extra data", p_fmt->i_extra );
+ break;
case VLC_FOURCC( 'm', 'p', '4', 'v' ):
{
}
case VLC_FOURCC( 'm', 'p', '4', 'a' ):
{
- char hexa[2*p_fmt->i_extra +1];
-
id->i_payload_type = p_sys->i_payload_type++;
id->i_clock_rate = p_fmt->audio.i_rate;
- id->psz_rtpmap = malloc( strlen( "mpeg4-generic/" ) + 12 );
- sprintf( id->psz_rtpmap, "mpeg4-generic/%d", p_fmt->audio.i_rate );
- id->pf_packetize = rtp_packetize_mp4a;
- id->psz_fmtp = malloc( 200 + 2 * p_fmt->i_extra );
- sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
- sprintf( id->psz_fmtp,
- "streamtype=5; profile-level-id=15; mode=AAC-hbr; "
- "config=%s; SizeLength=13;IndexLength=3; "
- "IndexDeltaLength=3; Profile=1;", hexa );
+
+ if(!p_sys->b_latm)
+ {
+ char hexa[2*p_fmt->i_extra +1];
+
+ id->psz_rtpmap = malloc( strlen( "mpeg4-generic/" ) + 12 );
+ sprintf( id->psz_rtpmap, "mpeg4-generic/%d", p_fmt->audio.i_rate );
+ id->pf_packetize = rtp_packetize_mp4a;
+ id->psz_fmtp = malloc( 200 + 2 * p_fmt->i_extra );
+ sprintf_hexa( hexa, p_fmt->p_extra, p_fmt->i_extra );
+ sprintf( id->psz_fmtp,
+ "streamtype=5; profile-level-id=15; mode=AAC-hbr; "
+ "config=%s; SizeLength=13;IndexLength=3; "
+ "IndexDeltaLength=3; Profile=1;", hexa );
+ }
+ else
+ {
+ char hexa[13];
+ int i;
+ unsigned char config[6];
+ unsigned int aacsrates[15] = {
+ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
+ 16000, 12000, 11025, 8000, 7350, 0, 0 };
+
+ for( i = 0; i < 15; i++ )
+ if( p_fmt->audio.i_rate == aacsrates[i] )
+ break;
+
+ config[0]=0x40;
+ config[1]=0;
+ config[2]=0x20|i;
+ config[3]=p_fmt->audio.i_channels<<4;
+ config[4]=0x3f;
+ config[5]=0xc0;
+
+ asprintf( &id->psz_rtpmap, "MP4A-LATM/%d/%d",
+ p_fmt->audio.i_rate, p_fmt->audio.i_channels );
+ id->pf_packetize = rtp_packetize_mp4a_latm;
+ sprintf_hexa( hexa, config, 6 );
+ asprintf( &id->psz_fmtp, "profile-level-id=15; "
+ "object=2; cpresent=0; config=%s", hexa );
+ }
break;
}
case VLC_FOURCC( 's', 'a', 'm', 'r' ):
id->ssrc[3] = rand()&0xff;
id->i_sequence = rand()&0xffff;
id->i_timestamp_start = rand()&0xffffffff;
+ id->i_bitrate = p_fmt->i_bitrate/1000; /* Stream bitrate in kbps */
id->i_mtu = config_GetInt( p_stream, "mtu" ); /* XXX beuk */
- if( id->i_mtu <= 16 )
+ if( id->i_mtu <= 16 + MTU_REDUCE )
{
/* better than nothing */
id->i_mtu = 1500;
}
- msg_Dbg( p_stream, "using mtu=%d", id->i_mtu );
+ id->i_mtu -= MTU_REDUCE;
+ msg_Dbg( p_stream, "maximum RTP packet size: %d bytes", id->i_mtu );
if( p_sys->p_rtsp_url )
{
char psz_urlc[strlen( p_sys->psz_rtsp_control ) + 1 + 10];
- sprintf( psz_urlc, "%s/trackid=%d", p_sys->psz_rtsp_path, p_sys->i_es );
- fprintf( stderr, "rtsp: adding %s\n", psz_urlc );
+ sprintf( psz_urlc, "%s/trackID=%d", p_sys->psz_rtsp_path, p_sys->i_es );
+ msg_Dbg( p_stream, "rtsp: adding %s\n", psz_urlc );
id->p_rtsp_url = httpd_UrlNewUnique( p_sys->p_rtsp_host, psz_urlc, NULL, NULL, NULL );
if( id->p_rtsp_url )
p_sys->i_sdp_version++;
- fprintf( stderr, "sdp=%s", p_sys->psz_sdp );
+ msg_Dbg( p_stream, "sdp=%s", p_sys->psz_sdp );
/* Update SDP (sap/file) */
if( p_sys->b_export_sap ) SapSetup( p_stream );
if( id->rtsp_access ) free( id->rtsp_access );
/* Update SDP (sap/file) */
- if( p_sys->b_export_sap ) SapSetup( p_stream );
+ if( p_sys->b_export_sap && !p_sys->p_mux ) SapSetup( p_stream );
if( p_sys->b_export_sdp_file ) FileSetup( p_stream );
free( id );
sout_stream_sys_t *p_sys = p_stream->p_sys;
FILE *f;
- if( ( f = fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
+ if( ( f = utf8_fopen( p_sys->psz_sdp_file, "wt" ) ) == NULL )
{
msg_Err( p_stream, "cannot open file '%s' (%s)",
p_sys->psz_sdp_file, strerror(errno) );
static rtsp_client_t *RtspClientGet( sout_stream_t *p_stream, char *psz_session )
{
int i;
+
+ if( psz_session ) return NULL;
+
for( i = 0; i < p_stream->p_sys->i_rtsp; i++ )
{
if( !strcmp( p_stream->p_sys->rtsp[i]->psz_session, psz_session ) )
{
sout_stream_sys_t *p_sys = p_stream->p_sys;
- fprintf( stderr, "rtsp setup: %s : %d / %s\n", url->psz_host, url->i_port, url->psz_path );
+ msg_Dbg( p_stream, "rtsp setup: %s : %d / %s\n", url->psz_host, url->i_port, url->psz_path );
p_sys->p_rtsp_host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host, url->i_port > 0 ? url->i_port : 554 );
if( p_sys->p_rtsp_host == NULL )
{
sout_stream_t *p_stream = (sout_stream_t*)p_args;
sout_stream_sys_t *p_sys = p_stream->p_sys;
- char *psz_destination = p_sys->psz_destination;
- char *psz_session = NULL;
+ char *psz_destination = p_sys->psz_destination;
+ char *psz_session = NULL;
+ char *psz_cseq = NULL;
+ int i_cseq = 0;
if( answer == NULL || query == NULL )
{
return VLC_SUCCESS;
}
- fprintf( stderr, "RtspCallback query: type=%d\n", query->i_type );
+ //fprintf( stderr, "RtspCallback query: type=%d\n", query->i_type );
answer->i_proto = HTTPD_PROTO_RTSP;
answer->i_version= query->i_version;
answer->i_status = 200;
answer->psz_status = strdup( "OK" );
httpd_MsgAdd( answer, "Content-type", "%s", "application/sdp" );
-
+ httpd_MsgAdd( answer, "Content-Base", "%s/", p_sys->psz_rtsp_control );
answer->p_body = (uint8_t *)psz_sdp;
answer->i_body = strlen( psz_sdp );
break;
default:
return VLC_EGENERIC;
}
- httpd_MsgAdd( answer, "Server", "VLC Server" );
+ httpd_MsgAdd( answer, "Server", PACKAGE_STRING );
httpd_MsgAdd( answer, "Content-Length", "%d", answer->i_body );
- httpd_MsgAdd( answer, "Cseq", "%d", atoi( httpd_MsgGet( query, "Cseq" ) ) );
+ psz_cseq = httpd_MsgGet( query, "Cseq" );
+ if( psz_cseq )
+ i_cseq = atoi( psz_cseq );
+ else
+ i_cseq = 0;
+ httpd_MsgAdd( answer, "Cseq", "%d", i_cseq );
httpd_MsgAdd( answer, "Cache-Control", "%s", "no-cache" );
if( psz_session )
return VLC_SUCCESS;
}
-static int RtspCallbackId( httpd_callback_sys_t *p_args,
+static int RtspCallbackId( httpd_callback_sys_t *p_args,
httpd_client_t *cl,
httpd_message_t *answer, httpd_message_t *query )
{
sout_stream_id_t *id = (sout_stream_id_t*)p_args;
sout_stream_t *p_stream = id->p_stream;
sout_stream_sys_t *p_sys = p_stream->p_sys;
- char *psz_session = NULL;
+ char *psz_session = NULL;
+ char *psz_cseq = NULL;
+ int i_cseq = 0;
+
if( answer == NULL || query == NULL )
{
return VLC_SUCCESS;
}
- fprintf( stderr, "RtspCallback query: type=%d\n", query->i_type );
+ //fprintf( stderr, "RtspCallback query: type=%d\n", query->i_type );
answer->i_proto = HTTPD_PROTO_RTSP;
answer->i_version= query->i_version;
{
char *psz_transport = httpd_MsgGet( query, "Transport" );
- fprintf( stderr, "HTTPD_MSG_SETUP: transport=%s\n", psz_transport );
+ //fprintf( stderr, "HTTPD_MSG_SETUP: transport=%s\n", psz_transport );
if( strstr( psz_transport, "multicast" ) && id->psz_destination )
{
- fprintf( stderr, "HTTPD_MSG_SETUP: multicast\n" );
+ //fprintf( stderr, "HTTPD_MSG_SETUP: multicast\n" );
answer->i_status = 200;
answer->psz_status = strdup( "OK" );
answer->i_body = 0;
answer->p_body = NULL;
psz_session = httpd_MsgGet( query, "Session" );
- if( *psz_session == 0 )
+ if( !psz_session )
{
psz_session = malloc( 100 );
sprintf( psz_session, "%d", rand() );
}
httpd_MsgAdd( answer, "Transport",
"RTP/AVP/UDP;destination=%s;port=%d-%d;ttl=%d",
- id->psz_destination, id->i_port,id->i_port+1, p_sys->i_ttl );
+ id->psz_destination, id->i_port,id->i_port+1,
+ p_sys->i_ttl );
}
else if( strstr( psz_transport, "unicast" ) && strstr( psz_transport, "client_port=" ) )
{
break;
}
- fprintf( stderr, "HTTPD_MSG_SETUP: unicast ip=%s port=%d\n",
- ip, i_port );
+ //fprintf( stderr, "HTTPD_MSG_SETUP: unicast ip=%s port=%d\n", ip, i_port );
psz_session = httpd_MsgGet( query, "Session" );
- if( *psz_session == 0 )
+ if( !psz_session )
{
psz_session = malloc( 100 );
sprintf( psz_session, "%d", rand() );
}
/* first try to create the access out */
- if( p_sys->i_ttl > 0 )
+ if( p_sys->i_ttl )
snprintf( psz_access, sizeof( psz_access ),
"udp{raw,ttl=%d}", p_sys->i_ttl );
else
- strncpy( psz_access, "udp{raw}", sizeof( psz_access ) );
- psz_access[sizeof( psz_access ) - 1] = '\0';
+ strlcpy( psz_access, "udp{raw}", sizeof( psz_access ) );
snprintf( psz_url, sizeof( psz_url ),
( strchr( ip, ':' ) != NULL ) ? "[%s]:%d" : "%s:%d",
}
httpd_MsgAdd( answer, "Server", "VLC Server" );
httpd_MsgAdd( answer, "Content-Length", "%d", answer->i_body );
- httpd_MsgAdd( answer, "Cseq", "%d", atoi( httpd_MsgGet( query, "Cseq" ) ) );
+ psz_cseq = httpd_MsgGet( query, "Cseq" );
+ if( psz_cseq )
+ i_cseq = atoi( psz_cseq );
+ else
+ i_cseq = 0;
+ httpd_MsgAdd( answer, "Cseq", "%d", i_cseq );
httpd_MsgAdd( answer, "Cache-Control", "%s", "no-cache" );
if( psz_session )
return VLC_SUCCESS;
}
+
static int rtp_packetize_ac3( sout_stream_t *p_stream, sout_stream_id_t *id,
block_t *in )
{
return VLC_SUCCESS;
}
+/* rfc3016 */
+static int rtp_packetize_mp4a_latm( sout_stream_t *p_stream, sout_stream_id_t *id,
+ block_t *in )
+{
+ int i_max = id->i_mtu - 14; /* payload max in one packet */
+ int latmhdrsize = in->i_buffer / 0xff + 1;
+ int i_count = ( in->i_buffer + i_max - 1 ) / i_max;
+
+ uint8_t *p_data = in->p_buffer, *p_header = NULL;
+ int i_data = in->i_buffer;
+ int i;
+
+ for( i = 0; i < i_count; i++ )
+ {
+ int i_payload = __MIN( i_max, i_data );
+ block_t *out;
+
+ if( i != 0 )
+ latmhdrsize = 0;
+ out = block_New( p_stream, 12 + latmhdrsize + i_payload );
+
+ /* rtp common header */
+ rtp_packetize_common( id, out, ((i == i_count - 1) ? 1 : 0),
+ (in->i_pts > 0 ? in->i_pts : in->i_dts) );
+
+ if( i == 0 )
+ {
+ int tmp = in->i_buffer;
+
+ p_header=out->p_buffer+12;
+ while( tmp > 0xfe )
+ {
+ *p_header = 0xff;
+ p_header++;
+ tmp -= 0xff;
+ }
+ *p_header = tmp;
+ }
+
+ memcpy( &out->p_buffer[12+latmhdrsize], p_data, i_payload );
+
+ out->i_buffer = 12 + latmhdrsize + i_payload;
+ out->i_dts = in->i_dts + i * in->i_length / i_count;
+ out->i_length = in->i_length / i_count;
+
+ rtp_packetize_send( id, out );
+
+ p_data += i_payload;
+ i_data -= i_payload;
+ }
+
+ return VLC_SUCCESS;
+}
+
static int rtp_packetize_l16( sout_stream_t *p_stream, sout_stream_id_t *id,
block_t *in )
{
return VLC_SUCCESS;
}
+/* rfc3984 */
+static int rtp_packetize_h264( sout_stream_t *p_stream, sout_stream_id_t *id,
+ block_t *in )
+{
+ int i_max = id->i_mtu - 12; /* payload max in one packet */
+ uint8_t *p_data = in->p_buffer;
+ int i_data = in->i_buffer;
+ block_t *out;
+ int i_nal_type;
+ int i_payload;
+
+ while( i_data > 4 )
+ {
+ if( p_data[0] == 0x00 && p_data[1] == 0x00 && p_data[2] == 0x01 && /* startcode */
+ (p_data[3]&0x1f) > 0 && (p_data[3]&0x1f) < 24 ) /* naltype should be between 1 and 23 */
+ {
+ p_data += 3;
+ i_data -= 3;
+ i_nal_type = p_data[0]&0x1f;
+
+ /* Skip global headers */
+ if( i_nal_type == 7 || i_nal_type == 8 )
+ continue;
+
+ if( i_data <= i_max ) /* The whole pack will fit in one rtp payload */
+ {
+ /* single NAL */
+ i_payload = __MIN( i_max, i_data );
+ out = block_New( p_stream, 12 + i_payload );
+
+ /* rtp common header */
+ rtp_packetize_common( id, out, 1,
+ in->i_pts > 0 ? in->i_pts : in->i_dts );
+
+ memcpy( &out->p_buffer[12], p_data, i_payload );
+
+ out->i_buffer+= i_payload;
+ out->i_dts = in->i_dts;
+ out->i_length = in->i_length;
+
+ rtp_packetize_send( id, out );
+ p_data += i_payload;
+ i_data -= i_payload;
+ /*msg_Dbg( p_stream, "nal-out plain %d %02x", i_payload, out->p_buffer[16] );*/
+ }
+ else
+ {
+ /* FU-A */
+ uint8_t nalh; /* The nalheader byte */
+ int start=1, end=0;
+
+ /* skip but remember the nalh byte */
+ nalh = *p_data;
+ p_data++;
+ i_data--;
+
+ i_max = id->i_mtu - 14;
+
+ /*msg_Dbg( p_stream, "nal-out fragmented %02x %d", nalh, i_rest);*/
+
+ while( end == 0 )
+ {
+ i_payload = __MIN( i_max, i_data );
+ out = block_New( p_stream, 14 + i_payload );
+
+ if( i_data == i_payload )
+ end = 1;
+
+ /* rtp common header */
+ rtp_packetize_common( id, out, (end)?1:0,
+ in->i_pts > 0 ? in->i_pts : in->i_dts );
+
+ /* FU indicator */
+ out->p_buffer[12] = (nalh&0x60)|28;
+ /* FU header */
+ out->p_buffer[13] = (start<<7)|(end<<6)|(nalh&0x1f);
+
+ memcpy( &out->p_buffer[14], p_data, i_payload );
+
+ out->i_buffer = 14 + i_payload;
+ /* The dts should "slide" ? See the i_count trick and adapt */
+ out->i_dts = in->i_dts;
+
+ rtp_packetize_send( id, out );
+ /*msg_Dbg( p_stream, "nal-out fragmented: frag %d %d %02x %02x %d", start,end,
+ out->p_buffer[12], out->p_buffer[13], i_payload );*/
+
+ i_data -= i_payload;
+ p_data +- i_payload;
+ start = 0;
+ }
+ }
+ }
+ else
+ {
+ p_data++;
+ i_data--;
+ }
+ }
+ return VLC_SUCCESS;
+}
+
static int rtp_packetize_amr( sout_stream_t *p_stream, sout_stream_id_t *id,
block_t *in )
{
(in->i_pts > 0 ? in->i_pts : in->i_dts) );
/* Payload header */
out->p_buffer[12] = 0xF0; /* CMR */
- out->p_buffer[13] = 0x00; /* ToC */ /* FIXME: frame type */
+ out->p_buffer[13] = p_data[0]&0x7C; /* ToC */ /* FIXME: frame type */
/* FIXME: are we fed multiple frames ? */
- memcpy( &out->p_buffer[14], p_data, i_payload );
+ memcpy( &out->p_buffer[14], p_data+1, i_payload-1 );
- out->i_buffer = 14 + i_payload;
+ out->i_buffer = 14 + i_payload-1;
out->i_dts = in->i_dts + i * in->i_length / i_count;
out->i_length = in->i_length / i_count;