[Bug 754575] trick modes in gst-rtsp-server

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=754575

Sebastian Dröge (slomo) <slomo at coaxion.net> changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
 Attachment #339331|none                        |needs-work
             status|                            |

--- Comment #26 from Sebastian Dröge (slomo) <slomo at coaxion.net> ---
Review of attachment 339331:
 --> (https://bugzilla.gnome.org/review?bug=754575&attachment=339331)

::: gst/rtsp-server/rtsp-client.c
@@ +1202,3 @@
+
+  /* as for the speed header, just map it to rate */
+  *rate = *speed;

RTP timestamps are running_time, yes. Meaning, e.g.: if rate=2, applied_rate=1,
then timestamps are as before but you get the data twice as fast as (original)
real-time. If rate=1, applied_rate=2, then timestamps are all divided by 2 but
you get them in real-time.

That's exactly how the distinction between Speed and Scale could be achieved
here. Just that it's not the decision of the client in the end, but of the
server. Unless we extend the seek events by an applied_rate field.

::: gst/rtsp-server/rtsp-client.h
@@ +89,3 @@
  *   be sent for a tunneled connection. The response can be modified. Since
1.4
+ * @adjust_play_mode: called when a PLAY request is received to allow
+ *   implementations to modify scale, speed, or seek flags for the seek. Since
1.12

Document which of the arguments are in, which are out, and which are both
("ref" in C# speak)

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