[vlc-devel] commit: rtp sout: implement rtptime parameter (Pierre Ynard )
git version control
git at videolan.org
Sat Dec 19 18:48:24 CET 2009
vlc | branch: master | Pierre Ynard <linkfanel at yahoo.fr> | Sat Dec 19 18:44:02 2009 +0100| [2039a06bce2fa27120f596b9c9760f650a79f782] | committer: Pierre Ynard
rtp sout: implement rtptime parameter
This adds support for the rtptime parameter in the RTP-Info RTSP header.
It provides synchronization at start-up between RTP elementary streams,
and is required by some clients to start playing the stream correctly.
> http://git.videolan.org/gitweb.cgi/vlc.git/?a=commit;h=2039a06bce2fa27120f596b9c9760f650a79f782
---
modules/stream_out/rtp.c | 73 ++++++++++++++++++++++++++++++++++++++++++++-
modules/stream_out/rtp.h | 2 +
modules/stream_out/rtsp.c | 9 ++++--
3 files changed, 80 insertions(+), 4 deletions(-)
diff --git a/modules/stream_out/rtp.c b/modules/stream_out/rtp.c
index 0e65085..124d685 100644
--- a/modules/stream_out/rtp.c
+++ b/modules/stream_out/rtp.c
@@ -275,6 +275,12 @@ struct sout_stream_sys_t
/* RTSP */
rtsp_stream_t *rtsp;
+ /* RTSP NPT and timestamp computations */
+ mtime_t i_npt_zero; /* when NPT=0 packet is sent */
+ int64_t i_pts_zero; /* predicts PTS of NPT=0 packet */
+ int64_t i_pts_offset; /* matches actual PTS to prediction */
+ vlc_mutex_t lock_ts;
+
/* */
char *psz_destination;
uint32_t payload_bitmap;
@@ -313,6 +319,8 @@ struct sout_stream_id_t
/* rtp field */
uint16_t i_sequence;
uint8_t i_payload_type;
+ bool b_ts_init;
+ uint32_t i_ts_offset;
uint8_t ssrc[4];
/* for rtsp */
@@ -459,6 +467,14 @@ static int Open( vlc_object_t *p_this )
p_sys->b_latm = var_GetBool( p_stream, SOUT_CFG_PREFIX "mp4a-latm" );
+ /* NPT=0 time will be determined when we packetize the first packet
+ * (of any ES). But we want to be able to report rtptime in RTSP
+ * without waiting. So until then, we use an arbitrary reference
+ * PTS for timestamp computations, and then actual PTS will catch
+ * up using offsets. */
+ p_sys->i_npt_zero = VLC_TS_INVALID;
+ p_sys->i_pts_zero = mdate(); /* arbitrary value, could probably be
+ * random */
p_sys->payload_bitmap = 0;
p_sys->i_es = 0;
p_sys->es = NULL;
@@ -475,6 +491,7 @@ static int Open( vlc_object_t *p_this )
p_stream->p_sys = p_sys;
vlc_mutex_init( &p_sys->lock_sdp );
+ vlc_mutex_init( &p_sys->lock_ts );
vlc_mutex_init( &p_sys->lock_es );
psz = var_GetNonEmptyString( p_stream, SOUT_CFG_PREFIX "mux" );
@@ -606,6 +623,7 @@ static void Close( vlc_object_t * p_this )
RtspUnsetup( p_sys->rtsp );
vlc_mutex_destroy( &p_sys->lock_sdp );
+ vlc_mutex_destroy( &p_sys->lock_ts );
vlc_mutex_destroy( &p_sys->lock_es );
if( p_sys->p_httpd_file )
@@ -870,6 +888,13 @@ rtp_set_ptime (sout_stream_id_t *id, unsigned ptime_ms, size_t bytes)
id->i_mtu = 12 + (((id->i_mtu - 12) / bytes) * bytes);
}
+uint32_t rtp_compute_ts( const sout_stream_id_t *id, int64_t i_pts )
+{
+ /* NOTE: this plays nice with offsets because the calculations are
+ * linear. */
+ return i_pts * (int64_t)id->i_clock_rate / CLOCK_FREQ;
+}
+
/** Add an ES as a new RTP stream */
static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
{
@@ -1307,6 +1332,12 @@ static sout_stream_id_t *Add( sout_stream_t *p_stream, es_format_t *p_fmt )
net_SetCSCov( id->sinkv[0].rtp_fd, cscov, -1 );
#endif
+ vlc_mutex_lock( &p_sys->lock_ts );
+ id->b_ts_init = ( p_sys->i_npt_zero != VLC_TS_INVALID );
+ vlc_mutex_unlock( &p_sys->lock_ts );
+ if( id->b_ts_init )
+ id->i_ts_offset = rtp_compute_ts( id, p_sys->i_pts_offset );
+
if( p_sys->rtsp != NULL )
id->rtsp_id = RtspAddId( p_sys->rtsp, id, p_sys->i_es,
GetDWBE( id->ssrc ),
@@ -1689,6 +1720,26 @@ uint16_t rtp_get_seq( sout_stream_id_t *id )
return seq;
}
+/* Return a timestamp corresponding to packets being sent now, and that
+ * can be passed to rtp_compute_ts() to get rtptime values for each ES. */
+int64_t rtp_get_ts( const sout_stream_t *p_stream )
+{
+ sout_stream_sys_t *p_sys = p_stream->p_sys;
+ mtime_t i_npt_zero;
+ vlc_mutex_lock( &p_sys->lock_ts );
+ i_npt_zero = p_sys->i_npt_zero;
+ vlc_mutex_unlock( &p_sys->lock_ts );
+
+ if( i_npt_zero == VLC_TS_INVALID )
+ return p_sys->i_pts_zero;
+
+ mtime_t now = mdate();
+ if( now < i_npt_zero )
+ return p_sys->i_pts_zero;
+
+ return p_sys->i_pts_zero + (now - i_npt_zero);
+}
+
/* 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 )
@@ -1711,7 +1762,27 @@ unsigned rtp_get_num( const sout_stream_id_t *id )
void rtp_packetize_common( sout_stream_id_t *id, block_t *out,
int b_marker, int64_t i_pts )
{
- uint32_t i_timestamp = i_pts * (int64_t)id->i_clock_rate / CLOCK_FREQ;
+ if( !id->b_ts_init )
+ {
+ sout_stream_sys_t *p_sys = id->p_stream->p_sys;
+ vlc_mutex_lock( &p_sys->lock_ts );
+ if( p_sys->i_npt_zero == VLC_TS_INVALID )
+ {
+ /* This is the first packet of any ES. We initialize the
+ * NPT=0 time reference, and the offset to match the
+ * arbitrary PTS reference. */
+ p_sys->i_npt_zero = i_pts + id->i_caching;
+ p_sys->i_pts_offset = p_sys->i_pts_zero - i_pts;
+ }
+ vlc_mutex_unlock( &p_sys->lock_ts );
+
+ /* And in any case this is the first packet of this ES, so we
+ * initialize the offset for this ES. */
+ id->i_ts_offset = rtp_compute_ts( id, p_sys->i_pts_offset );
+ id->b_ts_init = true;
+ }
+
+ uint32_t i_timestamp = rtp_compute_ts( id, i_pts ) + id->i_ts_offset;
out->p_buffer[0] = 0x80;
out->p_buffer[1] = (b_marker?0x80:0x00)|id->i_payload_type;
diff --git a/modules/stream_out/rtp.h b/modules/stream_out/rtp.h
index 5372146..7bc61cc 100644
--- a/modules/stream_out/rtp.h
+++ b/modules/stream_out/rtp.h
@@ -36,9 +36,11 @@ void RtspDelId( rtsp_stream_t *rtsp, rtsp_stream_id_t * );
char *SDPGenerate( const sout_stream_t *p_stream, const char *rtsp_url );
+uint32_t rtp_compute_ts( const sout_stream_id_t *id, int64_t i_pts );
int rtp_add_sink( sout_stream_id_t *id, int fd, bool rtcp_mux, uint16_t *seq );
void rtp_del_sink( sout_stream_id_t *id, int fd );
uint16_t rtp_get_seq( sout_stream_id_t *id );
+int64_t rtp_get_ts( const sout_stream_t *p_stream );
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 a8e3027..952b1a8 100644
--- a/modules/stream_out/rtsp.c
+++ b/modules/stream_out/rtsp.c
@@ -638,8 +638,10 @@ 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;
+ int64_t ts = rtp_get_ts( rtsp->owner );
for( int i = 0; i < ses->trackc; i++ )
{
@@ -655,10 +657,11 @@ static int RtspHandler( rtsp_stream_t *rtsp, rtsp_stream_id_t *id,
else
seq = rtp_get_seq( tr->id );
infolen += sprintf( info + infolen,
- "url=%s/trackID=%u;seq=%u, ",
+ "url=%s/trackID=%u;seq=%u;rtptime=%u, ",
control,
rtp_get_num( tr->id ),
- seq );
+ seq,
+ rtp_compute_ts( tr->id, ts ) );
}
}
if( infolen > 0 )
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