diff options
| -rw-r--r-- | ChangeLog | 9 | ||||
| -rw-r--r-- | gst/rtsp/gstrtspsrc.c | 30 | ||||
| -rw-r--r-- | gst/rtsp/gstrtspsrc.h | 1 |
3 files changed, 31 insertions, 9 deletions
@@ -1,3 +1,12 @@ +2007-09-28 Wim Taymans <wim.taymans@gmail.com> + + * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved), + (gst_rtspsrc_play): + * gst/rtsp/gstrtspsrc.h: + In TCP mode, only timestamp the first buffer. TCP is not real time and + it does not make sense to try to skew compensate, also some servers send + the first batch of data in a burst. + 2007-09-27 Tim-Philipp Müller <tim at centricular dot net> * gst/matroska/matroska-demux.c: diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index 7a309303..23f4764b 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -2450,15 +2450,6 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src) /* don't need message anymore */ gst_rtsp_message_unset (&message); - /* set reception timestamp */ - GST_OBJECT_LOCK (src); - if (GST_ELEMENT_CLOCK (src)) { - GstClockTime now = gst_clock_get_time (GST_ELEMENT_CLOCK (src)); - - GST_BUFFER_TIMESTAMP (buf) = now - GST_ELEMENT_CAST (src)->base_time; - } - GST_OBJECT_UNLOCK (src); - GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size, channel); @@ -2473,10 +2464,30 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src) gst_buffer_set_caps (buf, stream->caps); } + if (src->base_time == -1) { + /* Take current running_time. This timestamp will be put on + * the first buffer of each stream because we are a live source and so we + * timestamp with the running_time. When we are dealing with TCP, we also + * only timestamp the first buffer (using the DISCONT flag) because a server + * typically bursts data, for which we don't want to compensate by speeding + * up the media. The other timestamps will be interpollated from this one + * using the RTP timestamps. */ + GST_OBJECT_LOCK (src); + if (GST_ELEMENT_CLOCK (src)) { + GstClockTime now = gst_clock_get_time (GST_ELEMENT_CLOCK (src)); + + src->base_time = now - GST_ELEMENT_CAST (src)->base_time; + } + GST_OBJECT_UNLOCK (src); + } + if (stream->discont && !is_rtcp) { /* mark first RTP buffer as discont */ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT); stream->discont = FALSE; + /* first buffer gets the timestamp, other buffers are not timestamped and + * their presentation time will be interpollated from the rtp timestamps. */ + GST_BUFFER_TIMESTAMP (buf) = src->base_time; } /* chain to the peer pad */ @@ -4157,6 +4168,7 @@ gst_rtspsrc_play (GstRTSPSrc * src) gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src)); } src->running = TRUE; + src->base_time = -1; src->state = GST_RTSP_STATE_PLAYING; gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, FALSE); gst_task_start (src->task); diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h index d956d7e1..29f23ce4 100644 --- a/gst/rtsp/gstrtspsrc.h +++ b/gst/rtsp/gstrtspsrc.h @@ -135,6 +135,7 @@ struct _GstRTSPSrc { gint free_channel; GstEvent *close_segment; GstEvent *start_segment; + GstClockTime base_time; /* UDP mode loop */ gint loop_cmd; |