(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 | <a href="http://gstreamer.freedesktop.org/wiki">http://gstreamer.freedesktop.org/wiki</a> | <a href="http://build.gstreamer.org">http://build.gstreamer.org</a> (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=<a href="mailto:muelli@e177230009.adsl.alicedsl.de">muelli@e177230009.adsl.alicedsl.de</a>] 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: <a href="http://bugzilla.gnome.org/show_bug.cgi?id=577843">http://bugzilla.gnome.org/show_bug.cgi?id=577843</a> <- 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=<a href="mailto:pentariu@fokus9206.fokus.fraunhofer.de">pentariu@fokus9206.fokus.fraunhofer.de</a>] ha entrado en la sala. (15:30:19) pentarius: hello * (15:30:34) pentarius: wtay, are you there? (15:32:33) Judo [n=<a href="mailto:judo@dhcp-9687ab2f.rescomp.arizona.edu">judo@dhcp-9687ab2f.rescomp.arizona.edu</a>] 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 <<a href="mailto:javier.galvez.guerrero@gmail.com">javier.galvez.guerrero@gmail.com</a>> 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 >