]> git.sesse.net Git - vlc/commitdiff
Revert "rtp sout: implement rtptime parameter"
authorPierre Ynard <linkfanel@yahoo.fr>
Mon, 7 Dec 2009 19:28:34 +0000 (20:28 +0100)
committerPierre Ynard <linkfanel@yahoo.fr>
Mon, 7 Dec 2009 19:28:34 +0000 (20:28 +0100)
This reverts commit ce7a4746ad10d451e5e2807be44181df9456d6f0.

Signed-off-by: Pierre Ynard <linkfanel@yahoo.fr>
modules/stream_out/rtp.c
modules/stream_out/rtp.h
modules/stream_out/rtsp.c

index edd53fd0dcdb6d0db27cfc1f9c08de15c2d3e950..c720aa5b6787245f960f5a31c562d146713bf997 100644 (file)
@@ -301,7 +301,6 @@ 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];
@@ -910,8 +909,6 @@ 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)))
@@ -1669,13 +1666,6 @@ 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 )
@@ -1712,7 +1702,6 @@ 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++;
 }
 
index 1f9841c78008050a0b95b4a98781eeff8d709cf7..aec9dde9253f84e366015148e8df0792430f8c48 100644 (file)
@@ -39,7 +39,6 @@ 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 */
index 1dca3de1bd41004b85ff6e9d22de403d6755649e..61d81fac656ac4ba68fe47b7daa99ab7369333fd 100644 (file)
@@ -638,8 +638,7 @@ 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;"
-                                       "rtptime=4294967295, ") ) + 1];
+                              + sizeof("url=/trackID=123;seq=65535, ") ) + 1];
                 size_t infolen = 0;
 
                 for( int i = 0; i < ses->trackc; i++ )
@@ -652,15 +651,11 @@ static int RtspHandler( rtsp_stream_t *rtsp, rtsp_stream_id_t *id,
                             tr->playing = true;
                             rtp_add_sink( tr->id, tr->fd, false );
                         }
-                        /* This is racy, as the first packets may have
-                         * already been sent before we fetch this info:
-                         * these extra packets might confuse the client. */
                         infolen += sprintf( info + infolen,
-                                    "url=%s/trackID=%u;seq=%u;rtptime=%u, ",
+                                            "url=%s/trackID=%u;seq=%u, ",
                                             control,
                                             rtp_get_num( tr->id ),
-                                            rtp_get_seq( tr->id ),
-                                            rtp_get_ts( tr->id ) );
+                                            rtp_get_seq( tr->id ) );
                     }
                 }
                 if( infolen > 0 )