[vlc-devel] [PATCH] rtp sout: implement rtptime parameter

Pierre Ynard linkfanel at yahoo.fr
Sat Dec 5 15:12:35 CET 2009


This adds support for the rtptime parameter in the RTP-Info RTSP header.
It is needed by RealPlayer, otherwise it will start playing the stream
or not depending on the time of the day.


diff --git a/modules/stream_out/rtp.c b/modules/stream_out/rtp.c
index 837270b..f7f80ed 100644
--- a/modules/stream_out/rtp.c
+++ b/modules/stream_out/rtp.c
@@ -301,6 +301,7 @@ struct sout_stream_id_t
 
     sout_stream_t *p_stream;
     /* rtp field */
+    uint32_t    i_timestamp;
     uint16_t    i_sequence;
     uint8_t     i_payload_type;
     uint8_t     ssrc[4];
@@ -909,6 +910,8 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
 
     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)))
@@ -1662,6 +1665,13 @@ uint16_t rtp_get_seq( const sout_stream_id_t *id )
     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 )
@@ -1698,6 +1708,7 @@ void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
     memcpy( out->p_buffer + 8, id->ssrc, 4 );
 
     out->i_buffer = 12;
+    id->i_timestamp = i_timestamp;
     id->i_sequence++;
 }
 
diff --git a/modules/stream_out/rtp.h b/modules/stream_out/rtp.h
index aec9dde..1f9841c 100644
--- a/modules/stream_out/rtp.h
+++ b/modules/stream_out/rtp.h
@@ -39,6 +39,7 @@ char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url );
 int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux );
 void rtp_del_sink( sout_stream_id_t *id, int fd );
 uint16_t rtp_get_seq( const sout_stream_id_t *id );
+uint32_t rtp_get_ts( const sout_stream_id_t *id );
 unsigned rtp_get_num( const sout_stream_id_t *id );
 
 /* RTP packetization */
diff --git a/modules/stream_out/rtsp.c b/modules/stream_out/rtsp.c
index 523a1d4..2c57b3b 100644
--- a/modules/stream_out/rtsp.c
+++ b/modules/stream_out/rtsp.c
@@ -635,7 +635,8 @@ static int RtspHandler( rtsp_stream_t *rtsp, rtsp_stream_id_t *id,
             {
                 /* FIXME: we really need to limit the number of tracks... */
                 char info[ses->trackc * ( strlen( control )
-                              + sizeof("url=/trackID=123;seq=65535, ") ) + 1];
+                              + sizeof("url=/trackID=123;seq=65535;"
+                                       "rtptime=4294967295, ") ) + 1];
                 size_t infolen = 0;
 
                 for( int i = 0; i < ses->trackc; i++ )
@@ -649,10 +650,11 @@ static int RtspHandler( rtsp_stream_t *rtsp, rtsp_stream_id_t *id,
                             rtp_add_sink( tr->id, tr->fd, false );
                         }
                         infolen += sprintf( info + infolen,
-                                            "url=%s/trackID=%u;seq=%u, ",
+                                    "url=%s/trackID=%u;seq=%u;rtptime=%u, ",
                                             control,
                                             rtp_get_num( tr->id ),
-                                            rtp_get_seq( tr->id ) );
+                                            rtp_get_seq( tr->id ),
+                                            rtp_get_ts( tr->id ) );
                     }
                 }
                 if( infolen > 0 )


Regards,

-- 
Pierre Ynard
"Une âme dans un corps, c'est comme un dessin sur une feuille de papier."



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