From ce7a4746ad10d451e5e2807be44181df9456d6f0 Mon Sep 17 00:00:00 2001 From: Pierre Ynard Date: Sun, 6 Dec 2009 16:50:43 +0100 Subject: [PATCH] rtp sout: implement rtptime parameter 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. --- modules/stream_out/rtp.c | 11 +++++++++++ modules/stream_out/rtp.h | 1 + modules/stream_out/rtsp.c | 11 ++++++++--- 3 files changed, 20 insertions(+), 3 deletions(-) diff --git a/modules/stream_out/rtp.c b/modules/stream_out/rtp.c index 837270b5f0..f7f80edeb0 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 aec9dde925..1f9841c780 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 523a1d4afd..2e29e15089 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++ ) @@ -648,11 +649,15 @@ 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, ", + "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 ) -- 2.39.2