[gst-devel] Dinamically managing the buffer size in the RTP client side

[Javier Gálvez Guerrero]

(15:14:56) El tema de #gstreamer es: Bad/FFmpeg/Ugly are out! | Current:
Core 0.10.22, Base 0.10.22, Good 0.10.14, Bad 0.10.11, Ugly 0.10.11, FFMpeg
0.10.7, GNonLin 0.10.10.2, Python 0.10.14 |
http://gstreamer.freedesktop.org/wiki | http://build.gstreamer.org
*(15:15:51) dulceangustia: how can I dynamically change the buffer size of a
RTSP/RTP gstreamer client?*
*(15:23:34) cotigao: dulceangustia, there is a latency property  for
rtspsrc/gstrtpbin, that you can try*
(15:24:22) Muelli [n=muelli at e177230009.adsl.alicedsl.de] ha entrado en la
sala.
*(15:24:29) cotigao: dulceangustia, which map to the gstrtpjitterbuffer*
(15:25:11) alex3f ha salido de la sala.
*(15:26:43) dulceangustia: cotigao, so I can dynamically change this
property while receiving RTP data?
(15:28:17) cotigao: dulceangustia, yes *
(15:28:49) thaytan: wtay:
http://bugzilla.gnome.org/show_bug.cgi?id=577843<- I'd hunt this down,
but running out of time before the taxi comes
*(15:29:00) dulceangustia: cotigao, great. thanks a lot! =)*
(15:29:04) thaytan: it's a regression over playbin1 dvd handling though
(15:29:46) _ke ha salido de la sala (quit: Success).
(15:29:50) pentarius [n=pentariu at fokus9206.fokus.fraunhofer.de] ha entrado
en la sala.
(15:30:19) pentarius: hello *
(15:30:34) pentarius: wtay, are you there?
(15:32:33) Judo [n=judo at dhcp-9687ab2f.rescomp.arizona.edu] ha entrado en la
sala.
(15:33:26) pentarius: Or maybe someone else, I have strange effects on the
gstrtpbin. I use it for sending two video streams and one audio stream in
one session. On the receiver side video1 and audio is smoothly. But video
two is very slow. If i start receiving video2 in an extra pipeline - without
the bin - it is shown correctly
(15:33:46) pentarius: any suggestions what I'm doing wrong?
(15:34:00) pentarius: or some explanations?
*(15:36:21) cotigao: dulceangustia, although i am not sure if rtspsrc
propagates the latency value to the rtpmanager again (i.e after configuring
the rtpmanager)
(15:37:42) dulceangustia: cotigao, so you don't know if any change in the
latency property will take any effect once the streaming session has begun
(15:38:54) cotigao: dulceangustia, setting the property directly on
gstrtpbin will do*
(15:38:56) pentarius: anybody an idea where to digg?
*(15:39:46) dulceangustia: cotigao, ok, I count on it. thanks*

El día 31 de marzo de 2009 23:36, Javier Gálvez Guerrero <
javier.galvez.guerrero at gmail.com> escribió:
> Hi,
>
> In order to provide with a seamless video streaming service to the user
when
> performing a handover between two different networks, I would like to
> increase the buffer size in the client side while receiving RTSP/RTP data,
> so the period of time that the client device is not attached to any
network
> can be "hidden" to the user while playing the buffered content. Once the
new
> connection has successfully been established, the streaming session could
be
> continued through this new link and the buffer size configured back to the
> previous value.
>
> So, it is possible to dinamically change the buffer size of the
> corresponding element or it must be configured prior to the streaming
> session? Any suggestion will be welcome.
>
>
> Regards,
> Javi
>
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