Dinamically managing the buffer size in the RTP client side

[Javier Gálvez Guerrero]

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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[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=[hidden email]] 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=[hidden email]] ha entrado en la sala.
(15:30:19) pentarius: hello *
(15:30:34) pentarius: wtay, are you there?
(15:32:33) Judo [n=[hidden email]] 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 <[hidden email]> 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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