[vlc-devel] commit: Revert "rtp sout: implement rtptime parameter" (Pierre Ynard )
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git at videolan.org
Mon Dec 7 20:28:57 CET 2009
vlc | branch: master | Pierre Ynard <linkfanel at yahoo.fr> | Mon Dec 7 20:28:34 2009 +0100| [5fb806bce8305c9acf910d8a87033dde5d9c003a] | committer: Pierre Ynard
Revert "rtp sout: implement rtptime parameter"
This reverts commit ce7a4746ad10d451e5e2807be44181df9456d6f0.
Signed-off-by: Pierre Ynard <linkfanel at yahoo.fr>
> http://git.videolan.org/gitweb.cgi/vlc.git/?a=commit;h=5fb806bce8305c9acf910d8a87033dde5d9c003a
---
modules/stream_out/rtp.c | 11 -----------
modules/stream_out/rtp.h | 1 -
modules/stream_out/rtsp.c | 11 +++--------
3 files changed, 3 insertions(+), 20 deletions(-)
diff --git a/modules/stream_out/rtp.c b/modules/stream_out/rtp.c
index edd53fd..c720aa5 100644
--- a/modules/stream_out/rtp.c
+++ b/modules/stream_out/rtp.c
@@ -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++;
}
diff --git a/modules/stream_out/rtp.h b/modules/stream_out/rtp.h
index 1f9841c..aec9dde 100644
--- a/modules/stream_out/rtp.h
+++ b/modules/stream_out/rtp.h
@@ -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 */
diff --git a/modules/stream_out/rtsp.c b/modules/stream_out/rtsp.c
index 1dca3de..61d81fa 100644
--- a/modules/stream_out/rtsp.c
+++ b/modules/stream_out/rtsp.c
@@ -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 )
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