gst/rtsp/gstrtspsrc.*: In TCP mode, only timestamp the first buffer. TCP is not real time and it does not make sense ...

Original commit message from CVS:
* 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.

2 files changed, 22 insertions, 9 deletions

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;