Commit 2039a06b authored by Pierre Ynard's avatar 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.
parent 7a73b990
......@@ -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;
......
......@@ -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 */
......
......@@ -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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